From dupvinc at yahoo.fr  Tue May  1 03:48:28 2007
From: dupvinc at yahoo.fr (vincent dup)
Date: Tue, 1 May 2007 07:48:28 +0000 (GMT)
Subject: [cisco-voip]  Strange IP Communicator problem
Message-ID: <996109.51065.qm@web26002.mail.ukl.yahoo.com>

Hi,

I have had exactly the same problem with IP communicator using cisco VPn client.
Just upgrade to cisco vpn client version > 4.8 and it will be ok

Regards,
Vinc



      
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From baryonyx5 at gmail.com  Tue May  1 09:41:30 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Tue, 1 May 2007 08:41:30 -0500
Subject: [cisco-voip] CCM 5.1 Attendant Console Presence...
In-Reply-To: <5d093f9a0704300840p77700cdbhcbe0ac954134cc6f@mail.gmail.com>
References: <5d093f9a0704271151r4d65d3d5g7cc98e32867f6644@mail.gmail.com>
	<9568EBEC-4895-4B7F-A828-D0A8EB8807F1@cisco.com>
	<5d093f9a0704300824y798c101as7b33f75957568f75@mail.gmail.com>
	<041C7048-240C-4C66-9C62-1E3ACC5E8CB8@cisco.com>
	<5d093f9a0704300840p77700cdbhcbe0ac954134cc6f@mail.gmail.com>
Message-ID: <a749d0620705010641k2411e35dvf0e06ff548676455@mail.gmail.com>

If you have more than one NIC in the PC you may have to tell AC which IP to
bind to in the advanced settings.  I see this frequently on laptops with a
wired and wireless NIC both active and on PCs with vpn client software.

On 4/30/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> OK, let me check...
>
>
> J
>
> On 4/30/07, Ryan Ratliff <rratliff at cisco.com> wrote:
> >
> > I thought we were talking linestates?
> >
> > Did you look at the TCD trace to see if it was sending linestate
> > updates?
> > In the AC console go to Settings->Advanced.  Do you have the correct
> > IP address listed under the section "Local Host IP Address (for Line
> > State)".  If you don't set one it will show the auto selected IP
> > address and the port TCD is sending to.
> >
> > -Ryan
> >
> > On Apr 30, 2007, at 11:24 AM, Jonathan Charles wrote:
> >
> > There are no firewalls (it just boring InterVLAN routing on a 6513)...
> >
> > Also, user's phone gets caller-ID and calling name, but AC does
> > not... (there is just the one AC on the network).
> >
> >
> >
> > Jonathan
> >
> > On 4/30/07, Ryan Ratliff < rratliff at cisco.com> wrote: This will be the
> > same as any AC line state issue.   Check firewalls,
> > AC console for which NIC it is using, etc.  The LSS updates come as
> > UDP packets from TCD to your PC.   Look in the TCD traces to make
> > sure it is sending them (search for your IP address) and after that
> > it's all up to the network.
> >
> >
> > -Ryan
> >
> > On Apr 27, 2007, at 2:51 PM, Jonathan Charles wrote:
> >
> > So, supposedly attendant console is supposed to show line state of
> > speed dials and address book, but everyone shows as just a question
> > mark...
> >
> > Anyone have any idea how to correct?
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
kris seraphine
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From toddnh65 at gmail.com  Tue May  1 10:10:29 2007
From: toddnh65 at gmail.com (Todd Franklin)
Date: Tue, 1 May 2007 10:10:29 -0400
Subject: [cisco-voip] Copying a Phone
Message-ID: <6101bda50705010710m7eeb2927m965d4cd93362be9f@mail.gmail.com>

Hey everyone.
I have a situation whereby our admission room has 3 Cisco 7960s with 7914s
attached.  All 14 lines on the 7914 are allocated and have text labels.
Anyways, we decided to add a 4th phone.  So I copied one working phone, and
plugged in the new MAC address, gave it an extension when it prompted me to,
and voila, all set.  Except my 7914 had only the "hidden extension" numbers,
not the text.  I had to go into each line in CCM and assign the text.  I am
sure there is a way to "copy it all" rather than the way I went.  Or is
there?  just curious, because I have a similar setup we are adding a phone
to in the next few weeks.

Thanks.
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From jonvoip at gmail.com  Tue May  1 10:43:58 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 1 May 2007 09:43:58 -0500
Subject: [cisco-voip] CCM 5.1 Attendant Console Presence...
In-Reply-To: <a749d0620705010641k2411e35dvf0e06ff548676455@mail.gmail.com>
References: <5d093f9a0704271151r4d65d3d5g7cc98e32867f6644@mail.gmail.com>
	<9568EBEC-4895-4B7F-A828-D0A8EB8807F1@cisco.com>
	<5d093f9a0704300824y798c101as7b33f75957568f75@mail.gmail.com>
	<041C7048-240C-4C66-9C62-1E3ACC5E8CB8@cisco.com>
	<5d093f9a0704300840p77700cdbhcbe0ac954134cc6f@mail.gmail.com>
	<a749d0620705010641k2411e35dvf0e06ff548676455@mail.gmail.com>
Message-ID: <5d093f9a0705010743v481d483fh324eacfaef507394@mail.gmail.com>

Only one NIC in the PC, I also manually entered the IP address of the AC PC
in the AC...



Jonathan

On 5/1/07, Kris Seraphine <baryonyx5 at gmail.com> wrote:
>
> If you have more than one NIC in the PC you may have to tell AC which IP
> to bind to in the advanced settings.  I see this frequently on laptops with
> a wired and wireless NIC both active and on PCs with vpn client software.
>
> On 4/30/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > OK, let me check...
> >
> >
> > J
> >
> > On 4/30/07, Ryan Ratliff < rratliff at cisco.com> wrote:
> > >
> > > I thought we were talking linestates?
> > >
> > > Did you look at the TCD trace to see if it was sending linestate
> > > updates?
> > > In the AC console go to Settings->Advanced.  Do you have the correct
> > > IP address listed under the section "Local Host IP Address (for Line
> > > State)".  If you don't set one it will show the auto selected IP
> > > address and the port TCD is sending to.
> > >
> > > -Ryan
> > >
> > > On Apr 30, 2007, at 11:24 AM, Jonathan Charles wrote:
> > >
> > > There are no firewalls (it just boring InterVLAN routing on a 6513)...
> > >
> > >
> > > Also, user's phone gets caller-ID and calling name, but AC does
> > > not... (there is just the one AC on the network).
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On 4/30/07, Ryan Ratliff < rratliff at cisco.com> wrote: This will be the
> > > same as any AC line state issue.   Check firewalls,
> > > AC console for which NIC it is using, etc.  The LSS updates come as
> > > UDP packets from TCD to your PC.   Look in the TCD traces to make
> > > sure it is sending them (search for your IP address) and after that
> > > it's all up to the network.
> > >
> > >
> > > -Ryan
> > >
> > > On Apr 27, 2007, at 2:51 PM, Jonathan Charles wrote:
> > >
> > > So, supposedly attendant console is supposed to show line state of
> > > speed dials and address book, but everyone shows as just a question
> > > mark...
> > >
> > > Anyone have any idea how to correct?
> > >
> > >
> > >
> > > Jonathan
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> kris seraphine
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From asobihoudai at yahoo.com  Tue May  1 10:58:16 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 1 May 2007 07:58:16 -0700 (PDT)
Subject: [cisco-voip] Copying a Phone
In-Reply-To: <6101bda50705010710m7eeb2927m965d4cd93362be9f@mail.gmail.com>
Message-ID: <904895.92691.qm@web57208.mail.re3.yahoo.com>

If there *is* a way to copy them all at once, I'd sure
like to know about it because I've been configuring
all 7960+7914s manually.

--- Todd Franklin <toddnh65 at gmail.com> wrote:

> Hey everyone.
> I have a situation whereby our admission room has 3
> Cisco 7960s with 7914s
> attached.  All 14 lines on the 7914 are allocated
> and have text labels.
> Anyways, we decided to add a 4th phone.  So I copied
> one working phone, and
> plugged in the new MAC address, gave it an extension
> when it prompted me to,
> and voila, all set.  Except my 7914 had only the
> "hidden extension" numbers,
> not the text.  I had to go into each line in CCM and
> assign the text.  I am
> sure there is a way to "copy it all" rather than the
> way I went.  Or is
> there?  just curious, because I have a similar setup
> we are adding a phone
> to in the next few weeks.
> 
> Thanks.
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From Johnny.Crothers at vanco.co.uk  Tue May  1 11:01:19 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers)
Date: Tue, 1 May 2007 16:01:19 +0100
Subject: [cisco-voip] Copying a Phone
In-Reply-To: <904895.92691.qm@web57208.mail.re3.yahoo.com>
Message-ID: <A56BD242744BF540B15D1672E0E3E875F0CD62@LON-INT-MSGVS01.internal.vancocorp.com>

The only way I can think of is to export the device using BAT which will
include all the line configurations, change the MAC address and
re-import?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: 01 May 2007 15:58
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Copying a Phone

If there *is* a way to copy them all at once, I'd sure
like to know about it because I've been configuring
all 7960+7914s manually.

--- Todd Franklin <toddnh65 at gmail.com> wrote:

> Hey everyone.
> I have a situation whereby our admission room has 3
> Cisco 7960s with 7914s
> attached.  All 14 lines on the 7914 are allocated
> and have text labels.
> Anyways, we decided to add a 4th phone.  So I copied
> one working phone, and
> plugged in the new MAC address, gave it an extension
> when it prompted me to,
> and voila, all set.  Except my 7914 had only the
> "hidden extension" numbers,
> not the text.  I had to go into each line in CCM and
> assign the text.  I am
> sure there is a way to "copy it all" rather than the
> way I went.  Or is
> there?  just curious, because I have a similar setup
> we are adding a phone
> to in the next few weeks.
> 
> Thanks.
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From CBeck at careered.com  Tue May  1 09:38:53 2007
From: CBeck at careered.com (Chris Beck - CEC)
Date: Tue, 1 May 2007 08:38:53 -0500
Subject: [cisco-voip] Wierd phone issue
Message-ID: <70E40EF625463544B814B0819734EF2C04C72723@cecemx002.corporate.careered.com>

I am having a problem at one site with 7940s and 7960s.

 

The phones look normal, but sporadically, when you call them you get
fast busy.  When the user attempts a call, they get fast busy.  If the
user reboots the phone (by removing power), it reboots and works fine.
It has happened to multiple phones, but only at this one site.

 

Phones are running 8.0.4, and are serviced by a cm 4.1.3sr3 cluster.

 

Has anyone seen this one before?

 

All help appreciated, as we have been chasing our tail for a while on
this one.

 

-Chris

 

Chris Beck

Enterprise Voice Architect

Technology Integration Services Team

Information Technology

Career Education Corporation

Greenspoint Office - 847-851-7350

EDC Voice - 847-783-8141

Mobile - 847-542-0118

Fax - 847-874-5741

email - cbeck at careered.com <mailto:cbeck at careered.com> 

 

 

 

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From erik.stillman at shearman.com  Tue May  1 11:35:57 2007
From: erik.stillman at shearman.com (Erik Stillman)
Date: Tue, 1 May 2007 11:35:57 -0400
Subject: [cisco-voip] Strange IPC problem
In-Reply-To: <mailman.12210.1178031711.1489.cisco-voip@puck.nether.net>
Message-ID: <OFF8E7F83B.3463A8BC-ON852572CE.00533DBC-852572CE.0055B083@shearman.com>

Thanks to all the replies about IPC.

Here's some of the info in response:

We're running a Nortel VPN client called Contivity, version V06_01.102
The concentrator is a Nortel appliance running V05_00.303

The computer is one of our standard images;
but I will definitely look into that idea of a software conflict.


(Embedded image moved to file: pic12044.jpg)

Erik


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This communication is not intended to be used, and cannot be used,
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***********************************************************************

This transmittal and/or attachments may be a confidential
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From jonvoip at gmail.com  Tue May  1 12:40:12 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 1 May 2007 11:40:12 -0500
Subject: [cisco-voip] Weird Problem with 7970s...
Message-ID: <5d093f9a0705010940g3cff06a9sc913ee504efbecef@mail.gmail.com>

CallManager 5.1.1b... we have the CFB and CFNA set to go to vmail (check
boxes checked)...

We have noticed that every once in a while calls do not foward to voicemail,
they just ring and ring (calls to other extensions do go to voicemail) and
the only solution is to uncheck the boxes, save, check the boxes again and
save.

The customer does not find this to be an acceptable solution... is this a
bug?

7970s are running 8.2.1



Jonathan
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From erik.stillman at shearman.com  Tue May  1 12:57:51 2007
From: erik.stillman at shearman.com (Erik Stillman)
Date: Tue, 1 May 2007 12:57:51 -0400
Subject: [cisco-voip] Weird phone issue
In-Reply-To: <mailman.12217.1178033768.1489.cisco-voip@puck.nether.net>
Message-ID: <OF3FDFBC29.2C688412-ON852572CE.005CD4F1-852572CE.005D2FE8@shearman.com>

Chris, I would update the firmware on that single phone to the latest code
and test again. (like 8.21 at least)
This can be done by loading the FW into the TFTP server and specifying the
Phone Load Name on the Device config page.

Be very careful to check the Device Defaults page and copy down the current
settings for the model you are updating.
I've seen on many occasions that the defaults will be updated automatically
by the install. I think this is why Cisco offers a zip file of the loads.

Erik



***********************************************************************

This communication is not intended to be used, and cannot be used,
by the recipient or any other person for the purpose of avoiding United
States federal tax penalties that may be imposed on the recipient or
such other person. In addition, if any United States federal tax
advice contained in this communication is used or referred to in
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*************************************************************************


From steffen.bruening at de.logicalis.com  Wed May  2 03:44:32 2007
From: steffen.bruening at de.logicalis.com (Stift)
Date: Wed, 02 May 2007 09:44:32 +0200
Subject: [cisco-voip] Auto Call Pickup discard the zeros
In-Reply-To: <1D39D54FDA99E84D9016063F57CBE60B025519D1@LULUAEXC34.lu.kworld.kpmg.com>
References: <f0plgr$87o$1@sea.gmane.org><1D39D54FDA99E84D9016063F57CBE60B02551636@LULUAEXC34.lu.kworld.kpmg.com>	<f0pm2v$95f$1@sea.gmane.org><1D39D54FDA99E84D9016063F57CBE60B02551764@LULUAEXC34.lu.kworld.kpmg.com>	<f0sp96$r7o$1@sea.gmane.org><1D39D54FDA99E84D9016063F57CBE60B02551969@LULUAEXC34.lu.kworld.kpmg.com>	<f0suef$ggh$1@sea.gmane.org>
	<1D39D54FDA99E84D9016063F57CBE60B025519D1@LULUAEXC34.lu.kworld.kpmg.com>
Message-ID: <f19fh1$td6$1@sea.gmane.org>

Sorry for the delay of my awnser.

Yes there are translations for outgoing calls: discard digits PreDot

Lefevre, Olivier schrieb:
> And for outgoing call?
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
> Sent: vendredi, 27 avril 2007 15:39
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Auto Call Pickup discard the zeros
> 
> 
> The pickup groups are in the same partition as the phones. And there are
> 
> no translation pattern for incoming calls.
> 
> Lefevre, Olivier schrieb:
>> Is the pickup group on a special range, partition???
>> It could be display configure on the pisckup group. Check your
>> translation pattern if the pickup group number is not include in one
> of
>> those.
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
>> Sent: vendredi, 27 avril 2007 14:11
>> To: cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] Auto Call Pickup discard the zeros
>>
>> yes it is the same problem.
>>
>> Lefevre, Olivier schrieb:
>>> Is it the same if you call from foreign country??
>>>
>>> -----Original Message-----
>>> From: cisco-voip-bounces at puck.nether.net
>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
>>> Sent: jeudi, 26 avril 2007 09:58
>>> To: cisco-voip at puck.nether.net
>>> Subject: Re: [cisco-voip] Auto Call Pickup discard the zeros
>>>
>>> CCM4.1.3sr4d
>>>
>>> Lefevre, Olivier schrieb:
>>>> Witch version of CCM are you using?
>>>>
>>>> -----Original Message-----
>>>> From: cisco-voip-bounces at puck.nether.net
>>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
>>>> Sent: jeudi, 26 avril 2007 09:49
>>>> To: cisco-voip at puck.nether.net
>>>> Subject: Re: [cisco-voip] Auto Call Pickup discard the zeros
>>>>
>>>> Hello Olivier,
>>>>
>>>> no I don't have configured any mask on it. All phones and numbers
> are
>>> in
>>>> the same partition.
>>>>
>>>> Another problem is after I picked up the call, I can't see the call
>> in
>>>> the received calls directory.
>>>>
>>>> Lefevre, Olivier schrieb:
>>>>> Have you any mask configure on it? It could came from translation
>>>>> between partition.
>>>>>
>>>>> -----Original Message-----
>>>>> From: cisco-voip-bounces at puck.nether.net
>>>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
>>>>> Sent: mercredi, 25 avril 2007 17:39
>>>>> To: cisco-voip at puck.nether.net
>>>>> Subject: [cisco-voip] Auto Call Pickup discard the zeros
>>>>>
>>>>>
>>>>> I have set Auto Call Pickup to the Value True and when I pickup now
>> a
>>>>> call from a phone all Zeros from the begin of a number with will
>>>>> discard.
>>>>>
>>>>> Example:
>>>>>
>>>>> I call from my cell phone (0173.....) to my colleague (on his phone
>> I
>>>>> see my number with a second zero for the outside dial
> (00173......),
>>>>> then I press pickup and on my phone only see 173......
>>>>>
>>>>> When set Auto Call Pickup back to the Value False, then everything
>>>> works
>>>>>   fine and I see all zeros.
>>>>>
>>>>> Any Ideas or Solutions?
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>> The information in this e-mail is confidential and may be legally
>>>> privileged. It is intended
>>>>> solely for the addressee. Access to this e-mail by anyone else is
>>>> unauthorized. If you have 
>>>>> received this communication in error, please address with the
>> subject
>>>> heading "Received in 
>>>>> error," send back to the sender, then delete the e-mail and destroy
>>>> any copies of it. If you 
>>>>> are not the intended recipient, any disclosure, copying,
>> distribution
>>>> or any action taken or 
>>>>> omitted to be taken in reliance on it, is prohibited and may be
>>>> unlawful. Any opinions or 
>>>>> advice contained in this e-mail are subject to the terms and
>>>> conditions expressed in the 
>>>>> governing KPMG client engagement letter. Opinions, conclusions and
>>>> other information in this 
>>>>> e-mail and any attachments that do not relate to the official
>>> business
>>>> of the firm are 
>>>>> neither given nor endorsed by it. KPMG cannot guarantee that e-mail
>>>> communications are secure 
>>>>> or error-free, as information could be intercepted, corrupted,
>>>> amended, lost, destroyed, 
>>>>> arrive late or incomplete, or contain viruses.
>>>>>  
>>>>> KPMG is the Luxembourg member firm of KPMG international. KPMG
>>>> International is a Swiss 
>>>>> cooperative that serves as a coordinating entity for a network of
>>>> independent firms operating 
>>>>> under the KPMG name. KPMG International provides no services to
>>>> clients. Each member firm of 
>>>>> KPMG International is a legally distinct and separate entity and
>> each
>>>> describes itself as 
>>>>> such. Information about the structure and jurisdiction of your
> local
>>>> KPMG member firm can be 
>>>>> obtained from your KPMG representative.
>>>>>
>>>>> This footnote also confirms that this e-mail message has been swept
>>> by
>>>> AntiVirus software for 
>>>>> the presence of computer viruses. However, the ultimate
>>> responsibility
>>>> for virus checking 
>>>>> lies with the recipient of this e-mail.
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>> The information in this e-mail is confidential and may be legally
>>> privileged. It is intended
>>>> solely for the addressee. Access to this e-mail by anyone else is
>>> unauthorized. If you have 
>>>> received this communication in error, please address with the
> subject
>>> heading "Received in 
>>>> error," send back to the sender, then delete the e-mail and destroy
>>> any copies of it. If you 
>>>> are not the intended recipient, any disclosure, copying,
> distribution
>>> or any action taken or 
>>>> omitted to be taken in reliance on it, is prohibited and may be
>>> unlawful. Any opinions or 
>>>> advice contained in this e-mail are subject to the terms and
>>> conditions expressed in the 
>>>> governing KPMG client engagement letter. Opinions, conclusions and
>>> other information in this 
>>>> e-mail and any attachments that do not relate to the official
>> business
>>> of the firm are 
>>>> neither given nor endorsed by it. KPMG cannot guarantee that e-mail
>>> communications are secure 
>>>> or error-free, as information could be intercepted, corrupted,
>>> amended, lost, destroyed, 
>>>> arrive late or incomplete, or contain viruses.
>>>>  
>>>> KPMG is the Luxembourg member firm of KPMG international. KPMG
>>> International is a Swiss 
>>>> cooperative that serves as a coordinating entity for a network of
>>> independent firms operating 
>>>> under the KPMG name. KPMG International provides no services to
>>> clients. Each member firm of 
>>>> KPMG International is a legally distinct and separate entity and
> each
>>> describes itself as 
>>>> such. Information about the structure and jurisdiction of your local
>>> KPMG member firm can be 
>>>> obtained from your KPMG representative.
>>>>
>>>> This footnote also confirms that this e-mail message has been swept
>> by
>>> AntiVirus software for 
>>>> the presence of computer viruses. However, the ultimate
>> responsibility
>>> for virus checking 
>>>> lies with the recipient of this e-mail.
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>> The information in this e-mail is confidential and may be legally
>> privileged. It is intended
>>> solely for the addressee. Access to this e-mail by anyone else is
>> unauthorized. If you have 
>>> received this communication in error, please address with the subject
>> heading "Received in 
>>> error," send back to the sender, then delete the e-mail and destroy
>> any copies of it. If you 
>>> are not the intended recipient, any disclosure, copying, distribution
>> or any action taken or 
>>> omitted to be taken in reliance on it, is prohibited and may be
>> unlawful. Any opinions or 
>>> advice contained in this e-mail are subject to the terms and
>> conditions expressed in the 
>>> governing KPMG client engagement letter. Opinions, conclusions and
>> other information in this 
>>> e-mail and any attachments that do not relate to the official
> business
>> of the firm are 
>>> neither given nor endorsed by it. KPMG cannot guarantee that e-mail
>> communications are secure 
>>> or error-free, as information could be intercepted, corrupted,
>> amended, lost, destroyed, 
>>> arrive late or incomplete, or contain viruses.
>>>  
>>> KPMG is the Luxembourg member firm of KPMG international. KPMG
>> International is a Swiss 
>>> cooperative that serves as a coordinating entity for a network of
>> independent firms operating 
>>> under the KPMG name. KPMG International provides no services to
>> clients. Each member firm of 
>>> KPMG International is a legally distinct and separate entity and each
>> describes itself as 
>>> such. Information about the structure and jurisdiction of your local
>> KPMG member firm can be 
>>> obtained from your KPMG representative.
>>>
>>> This footnote also confirms that this e-mail message has been swept
> by
>> AntiVirus software for 
>>> the presence of computer viruses. However, the ultimate
> responsibility
>> for virus checking 
>>> lies with the recipient of this e-mail.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> The information in this e-mail is confidential and may be legally
> privileged. It is intended
>> solely for the addressee. Access to this e-mail by anyone else is
> unauthorized. If you have 
>> received this communication in error, please address with the subject
> heading "Received in 
>> error," send back to the sender, then delete the e-mail and destroy
> any copies of it. If you 
>> are not the intended recipient, any disclosure, copying, distribution
> or any action taken or 
>> omitted to be taken in reliance on it, is prohibited and may be
> unlawful. Any opinions or 
>> advice contained in this e-mail are subject to the terms and
> conditions expressed in the 
>> governing KPMG client engagement letter. Opinions, conclusions and
> other information in this 
>> e-mail and any attachments that do not relate to the official business
> of the firm are 
>> neither given nor endorsed by it. KPMG cannot guarantee that e-mail
> communications are secure 
>> or error-free, as information could be intercepted, corrupted,
> amended, lost, destroyed, 
>> arrive late or incomplete, or contain viruses.
>>  
>> KPMG is the Luxembourg member firm of KPMG international. KPMG
> International is a Swiss 
>> cooperative that serves as a coordinating entity for a network of
> independent firms operating 
>> under the KPMG name. KPMG International provides no services to
> clients. Each member firm of 
>> KPMG International is a legally distinct and separate entity and each
> describes itself as 
>> such. Information about the structure and jurisdiction of your local
> KPMG member firm can be 
>> obtained from your KPMG representative.
>>
>> This footnote also confirms that this e-mail message has been swept by
> AntiVirus software for 
>> the presence of computer viruses. However, the ultimate responsibility
> for virus checking 
>> lies with the recipient of this e-mail.
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> The information in this e-mail is confidential and may be legally privileged. It is intended
> solely for the addressee. Access to this e-mail by anyone else is unauthorized. If you have 
> received this communication in error, please address with the subject heading "Received in 
> error," send back to the sender, then delete the e-mail and destroy any copies of it. If you 
> are not the intended recipient, any disclosure, copying, distribution or any action taken or 
> omitted to be taken in reliance on it, is prohibited and may be unlawful. Any opinions or 
> advice contained in this e-mail are subject to the terms and conditions expressed in the 
> governing KPMG client engagement letter. Opinions, conclusions and other information in this 
> e-mail and any attachments that do not relate to the official business of the firm are 
> neither given nor endorsed by it. KPMG cannot guarantee that e-mail communications are secure 
> or error-free, as information could be intercepted, corrupted, amended, lost, destroyed, 
> arrive late or incomplete, or contain viruses.
>  
> KPMG is the Luxembourg member firm of KPMG international. KPMG International is a Swiss 
> cooperative that serves as a coordinating entity for a network of independent firms operating 
> under the KPMG name. KPMG International provides no services to clients. Each member firm of 
> KPMG International is a legally distinct and separate entity and each describes itself as 
> such. Information about the structure and jurisdiction of your local KPMG member firm can be 
> obtained from your KPMG representative.
> 
> This footnote also confirms that this e-mail message has been swept by AntiVirus software for 
> the presence of computer viruses. However, the ultimate responsibility for virus checking 
> lies with the recipient of this e-mail.


From olivier.lefevre at kpmg.lu  Wed May  2 03:54:51 2007
From: olivier.lefevre at kpmg.lu (Lefevre, Olivier)
Date: Wed, 2 May 2007 09:54:51 +0200
Subject: [cisco-voip] Auto Call Pickup discard the zeros
In-Reply-To: <f19fh1$td6$1@sea.gmane.org>
References: <f0plgr$87o$1@sea.gmane.org><1D39D54FDA99E84D9016063F57CBE60B02551636@LULUAEXC34.lu.kworld.kpmg.com>	<f0pm2v$95f$1@sea.gmane.org><1D39D54FDA99E84D9016063F57CBE60B02551764@LULUAEXC34.lu.kworld.kpmg.com>	<f0sp96$r7o$1@sea.gmane.org><1D39D54FDA99E84D9016063F57CBE60B02551969@LULUAEXC34.lu.kworld.kpmg.com>	<f0suef$ggh$1@sea.gmane.org><1D39D54FDA99E84D9016063F57CBE60B025519D1@LULUAEXC34.lu.kworld.kpmg.com>
	<f19fh1$td6$1@sea.gmane.org>
Message-ID: <1D39D54FDA99E84D9016063F57CBE60B02551C70@LULUAEXC34.lu.kworld.kpmg.com>

try to suppress this rules. Notmarly it works after that

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
Sent: mercredi, 02 mai 2007 09:45
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Auto Call Pickup discard the zeros

Sorry for the delay of my awnser.

Yes there are translations for outgoing calls: discard digits PreDot

Lefevre, Olivier schrieb:
> And for outgoing call?
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
> Sent: vendredi, 27 avril 2007 15:39
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Auto Call Pickup discard the zeros
> 
> 
> The pickup groups are in the same partition as the phones. And there
are
> 
> no translation pattern for incoming calls.
> 
> Lefevre, Olivier schrieb:
>> Is the pickup group on a special range, partition???
>> It could be display configure on the pisckup group. Check your
>> translation pattern if the pickup group number is not include in one
> of
>> those.
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
>> Sent: vendredi, 27 avril 2007 14:11
>> To: cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] Auto Call Pickup discard the zeros
>>
>> yes it is the same problem.
>>
>> Lefevre, Olivier schrieb:
>>> Is it the same if you call from foreign country??
>>>
>>> -----Original Message-----
>>> From: cisco-voip-bounces at puck.nether.net
>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
>>> Sent: jeudi, 26 avril 2007 09:58
>>> To: cisco-voip at puck.nether.net
>>> Subject: Re: [cisco-voip] Auto Call Pickup discard the zeros
>>>
>>> CCM4.1.3sr4d
>>>
>>> Lefevre, Olivier schrieb:
>>>> Witch version of CCM are you using?
>>>>
>>>> -----Original Message-----
>>>> From: cisco-voip-bounces at puck.nether.net
>>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
>>>> Sent: jeudi, 26 avril 2007 09:49
>>>> To: cisco-voip at puck.nether.net
>>>> Subject: Re: [cisco-voip] Auto Call Pickup discard the zeros
>>>>
>>>> Hello Olivier,
>>>>
>>>> no I don't have configured any mask on it. All phones and numbers
> are
>>> in
>>>> the same partition.
>>>>
>>>> Another problem is after I picked up the call, I can't see the call
>> in
>>>> the received calls directory.
>>>>
>>>> Lefevre, Olivier schrieb:
>>>>> Have you any mask configure on it? It could came from translation
>>>>> between partition.
>>>>>
>>>>> -----Original Message-----
>>>>> From: cisco-voip-bounces at puck.nether.net
>>>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stift
>>>>> Sent: mercredi, 25 avril 2007 17:39
>>>>> To: cisco-voip at puck.nether.net
>>>>> Subject: [cisco-voip] Auto Call Pickup discard the zeros
>>>>>
>>>>>
>>>>> I have set Auto Call Pickup to the Value True and when I pickup
now
>> a
>>>>> call from a phone all Zeros from the begin of a number with will
>>>>> discard.
>>>>>
>>>>> Example:
>>>>>
>>>>> I call from my cell phone (0173.....) to my colleague (on his
phone
>> I
>>>>> see my number with a second zero for the outside dial
> (00173......),
>>>>> then I press pickup and on my phone only see 173......
>>>>>
>>>>> When set Auto Call Pickup back to the Value False, then everything
>>>> works
>>>>>   fine and I see all zeros.
>>>>>
>>>>> Any Ideas or Solutions?
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip at puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>> The information in this e-mail is confidential and may be legally
>>>> privileged. It is intended
>>>>> solely for the addressee. Access to this e-mail by anyone else is
>>>> unauthorized. If you have 
>>>>> received this communication in error, please address with the
>> subject
>>>> heading "Received in 
>>>>> error," send back to the sender, then delete the e-mail and
destroy
>>>> any copies of it. If you 
>>>>> are not the intended recipient, any disclosure, copying,
>> distribution
>>>> or any action taken or 
>>>>> omitted to be taken in reliance on it, is prohibited and may be
>>>> unlawful. Any opinions or 
>>>>> advice contained in this e-mail are subject to the terms and
>>>> conditions expressed in the 
>>>>> governing KPMG client engagement letter. Opinions, conclusions and
>>>> other information in this 
>>>>> e-mail and any attachments that do not relate to the official
>>> business
>>>> of the firm are 
>>>>> neither given nor endorsed by it. KPMG cannot guarantee that
e-mail
>>>> communications are secure 
>>>>> or error-free, as information could be intercepted, corrupted,
>>>> amended, lost, destroyed, 
>>>>> arrive late or incomplete, or contain viruses.
>>>>>  
>>>>> KPMG is the Luxembourg member firm of KPMG international. KPMG
>>>> International is a Swiss 
>>>>> cooperative that serves as a coordinating entity for a network of
>>>> independent firms operating 
>>>>> under the KPMG name. KPMG International provides no services to
>>>> clients. Each member firm of 
>>>>> KPMG International is a legally distinct and separate entity and
>> each
>>>> describes itself as 
>>>>> such. Information about the structure and jurisdiction of your
> local
>>>> KPMG member firm can be 
>>>>> obtained from your KPMG representative.
>>>>>
>>>>> This footnote also confirms that this e-mail message has been
swept
>>> by
>>>> AntiVirus software for 
>>>>> the presence of computer viruses. However, the ultimate
>>> responsibility
>>>> for virus checking 
>>>>> lies with the recipient of this e-mail.
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>> The information in this e-mail is confidential and may be legally
>>> privileged. It is intended
>>>> solely for the addressee. Access to this e-mail by anyone else is
>>> unauthorized. If you have 
>>>> received this communication in error, please address with the
> subject
>>> heading "Received in 
>>>> error," send back to the sender, then delete the e-mail and destroy
>>> any copies of it. If you 
>>>> are not the intended recipient, any disclosure, copying,
> distribution
>>> or any action taken or 
>>>> omitted to be taken in reliance on it, is prohibited and may be
>>> unlawful. Any opinions or 
>>>> advice contained in this e-mail are subject to the terms and
>>> conditions expressed in the 
>>>> governing KPMG client engagement letter. Opinions, conclusions and
>>> other information in this 
>>>> e-mail and any attachments that do not relate to the official
>> business
>>> of the firm are 
>>>> neither given nor endorsed by it. KPMG cannot guarantee that e-mail
>>> communications are secure 
>>>> or error-free, as information could be intercepted, corrupted,
>>> amended, lost, destroyed, 
>>>> arrive late or incomplete, or contain viruses.
>>>>  
>>>> KPMG is the Luxembourg member firm of KPMG international. KPMG
>>> International is a Swiss 
>>>> cooperative that serves as a coordinating entity for a network of
>>> independent firms operating 
>>>> under the KPMG name. KPMG International provides no services to
>>> clients. Each member firm of 
>>>> KPMG International is a legally distinct and separate entity and
> each
>>> describes itself as 
>>>> such. Information about the structure and jurisdiction of your
local
>>> KPMG member firm can be 
>>>> obtained from your KPMG representative.
>>>>
>>>> This footnote also confirms that this e-mail message has been swept
>> by
>>> AntiVirus software for 
>>>> the presence of computer viruses. However, the ultimate
>> responsibility
>>> for virus checking 
>>>> lies with the recipient of this e-mail.
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>> The information in this e-mail is confidential and may be legally
>> privileged. It is intended
>>> solely for the addressee. Access to this e-mail by anyone else is
>> unauthorized. If you have 
>>> received this communication in error, please address with the
subject
>> heading "Received in 
>>> error," send back to the sender, then delete the e-mail and destroy
>> any copies of it. If you 
>>> are not the intended recipient, any disclosure, copying,
distribution
>> or any action taken or 
>>> omitted to be taken in reliance on it, is prohibited and may be
>> unlawful. Any opinions or 
>>> advice contained in this e-mail are subject to the terms and
>> conditions expressed in the 
>>> governing KPMG client engagement letter. Opinions, conclusions and
>> other information in this 
>>> e-mail and any attachments that do not relate to the official
> business
>> of the firm are 
>>> neither given nor endorsed by it. KPMG cannot guarantee that e-mail
>> communications are secure 
>>> or error-free, as information could be intercepted, corrupted,
>> amended, lost, destroyed, 
>>> arrive late or incomplete, or contain viruses.
>>>  
>>> KPMG is the Luxembourg member firm of KPMG international. KPMG
>> International is a Swiss 
>>> cooperative that serves as a coordinating entity for a network of
>> independent firms operating 
>>> under the KPMG name. KPMG International provides no services to
>> clients. Each member firm of 
>>> KPMG International is a legally distinct and separate entity and
each
>> describes itself as 
>>> such. Information about the structure and jurisdiction of your local
>> KPMG member firm can be 
>>> obtained from your KPMG representative.
>>>
>>> This footnote also confirms that this e-mail message has been swept
> by
>> AntiVirus software for 
>>> the presence of computer viruses. However, the ultimate
> responsibility
>> for virus checking 
>>> lies with the recipient of this e-mail.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> The information in this e-mail is confidential and may be legally
> privileged. It is intended
>> solely for the addressee. Access to this e-mail by anyone else is
> unauthorized. If you have 
>> received this communication in error, please address with the subject
> heading "Received in 
>> error," send back to the sender, then delete the e-mail and destroy
> any copies of it. If you 
>> are not the intended recipient, any disclosure, copying, distribution
> or any action taken or 
>> omitted to be taken in reliance on it, is prohibited and may be
> unlawful. Any opinions or 
>> advice contained in this e-mail are subject to the terms and
> conditions expressed in the 
>> governing KPMG client engagement letter. Opinions, conclusions and
> other information in this 
>> e-mail and any attachments that do not relate to the official
business
> of the firm are 
>> neither given nor endorsed by it. KPMG cannot guarantee that e-mail
> communications are secure 
>> or error-free, as information could be intercepted, corrupted,
> amended, lost, destroyed, 
>> arrive late or incomplete, or contain viruses.
>>  
>> KPMG is the Luxembourg member firm of KPMG international. KPMG
> International is a Swiss 
>> cooperative that serves as a coordinating entity for a network of
> independent firms operating 
>> under the KPMG name. KPMG International provides no services to
> clients. Each member firm of 
>> KPMG International is a legally distinct and separate entity and each
> describes itself as 
>> such. Information about the structure and jurisdiction of your local
> KPMG member firm can be 
>> obtained from your KPMG representative.
>>
>> This footnote also confirms that this e-mail message has been swept
by
> AntiVirus software for 
>> the presence of computer viruses. However, the ultimate
responsibility
> for virus checking 
>> lies with the recipient of this e-mail.
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> The information in this e-mail is confidential and may be legally
privileged. It is intended
> solely for the addressee. Access to this e-mail by anyone else is
unauthorized. If you have 
> received this communication in error, please address with the subject
heading "Received in 
> error," send back to the sender, then delete the e-mail and destroy
any copies of it. If you 
> are not the intended recipient, any disclosure, copying, distribution
or any action taken or 
> omitted to be taken in reliance on it, is prohibited and may be
unlawful. Any opinions or 
> advice contained in this e-mail are subject to the terms and
conditions expressed in the 
> governing KPMG client engagement letter. Opinions, conclusions and
other information in this 
> e-mail and any attachments that do not relate to the official business
of the firm are 
> neither given nor endorsed by it. KPMG cannot guarantee that e-mail
communications are secure 
> or error-free, as information could be intercepted, corrupted,
amended, lost, destroyed, 
> arrive late or incomplete, or contain viruses.
>  
> KPMG is the Luxembourg member firm of KPMG international. KPMG
International is a Swiss 
> cooperative that serves as a coordinating entity for a network of
independent firms operating 
> under the KPMG name. KPMG International provides no services to
clients. Each member firm of 
> KPMG International is a legally distinct and separate entity and each
describes itself as 
> such. Information about the structure and jurisdiction of your local
KPMG member firm can be 
> obtained from your KPMG representative.
> 
> This footnote also confirms that this e-mail message has been swept by
AntiVirus software for 
> the presence of computer viruses. However, the ultimate responsibility
for virus checking 
> lies with the recipient of this e-mail.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
The information in this e-mail is confidential and may be legally privileged. It is intended
solely for the addressee. Access to this e-mail by anyone else is unauthorized. If you have 
received this communication in error, please address with the subject heading "Received in 
error," send back to the sender, then delete the e-mail and destroy any copies of it. If you 
are not the intended recipient, any disclosure, copying, distribution or any action taken or 
omitted to be taken in reliance on it, is prohibited and may be unlawful. Any opinions or 
advice contained in this e-mail are subject to the terms and conditions expressed in the 
governing KPMG client engagement letter. Opinions, conclusions and other information in this 
e-mail and any attachments that do not relate to the official business of the firm are 
neither given nor endorsed by it. KPMG cannot guarantee that e-mail communications are secure 
or error-free, as information could be intercepted, corrupted, amended, lost, destroyed, 
arrive late or incomplete, or contain viruses.
 
KPMG is the Luxembourg member firm of KPMG international. KPMG International is a Swiss 
cooperative that serves as a coordinating entity for a network of independent firms operating 
under the KPMG name. KPMG International provides no services to clients. Each member firm of 
KPMG International is a legally distinct and separate entity and each describes itself as 
such. Information about the structure and jurisdiction of your local KPMG member firm can be 
obtained from your KPMG representative.

This footnote also confirms that this e-mail message has been swept by AntiVirus software for 
the presence of computer viruses. However, the ultimate responsibility for virus checking 
lies with the recipient of this e-mail.



From cj.alan at gmail.com  Wed May  2 04:16:58 2007
From: cj.alan at gmail.com (Alan Su)
Date: Wed, 2 May 2007 16:16:58 +0800
Subject: [cisco-voip] MGCP gateway + CCM4.x + SIP IVR: "busy signal" relay
	problem
Message-ID: <8f9f19f00705020116j77de9240q1b6dca674a538fdb@mail.gmail.com>

Dear All,

Here is my lab environment:

                    MGCP GW ===== PSTN
                      |
    ---------------------------------------------------------
        |                 |                 |
    IP Phone   CCM 4.x   IP-IVR(DN:1234)

(ps. MGCP using T1 PRI to PSTN, CCM using SIP trunk to connect to IVR
server)


When I use IP Phone to dial DN (1234), CCM will route the call to IP-IVR via
SIP trunk, and then user can follow the voice prompt to enter password and
outisde phone number. If the password authentication passed, the outside
number will be forwarded to CCM and then routed to MGCP GW. The normal call
process is working just fine.

Now we want to have another voice prompt when the calling number is busy,
but it can't work as expected. When busy tone comes out, I see no signal
passed back to CCM and IVR server. It will take about 1 - 2 mins and then
IVR display message "486". Is the problem in GW or CCM? Is it possible that
I can make the busy signal passed to CCM/IVR as soon as possible? Thanks a
lot for your comment.

Regards,
Alan
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From aelnagar at ACT-EG.COM  Wed May  2 04:53:47 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Wed, 2 May 2007 11:53:47 +0300
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A029@mail.ACT-EG.COM>

Hello all;
 
A customer wants to upgrade from CM 4.1 to 5.1, what I am worried about is the licsences issue. How will I get the DLUs for callmanager 5.1 we have a lic. with each phone but how we will get the DLUs??
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com
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From akoul at dataways.gr  Wed May  2 05:13:50 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Wed, 02 May 2007 12:13:50 +0300
Subject: [cisco-voip] Phone Registration Rejected after adding BLFs on 7914
Message-ID: <4638564E.2070500@dataways.gr>

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From ealeatherman at gmail.com  Wed May  2 08:16:45 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 2 May 2007 08:16:45 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED88A029@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED88A029@mail.ACT-EG.COM>
Message-ID: <94a1afde0705020516g74d35ceck1ed62f9f820568ac@mail.gmail.com>

hi Ahmed,

When you upgrade, callmanager will generate a temporary license for all the
devices you have configured from 4.1. If I understand correctly, you just
have to send that off to cisco's licensing group and have it converted to a
real 5.2 license with the appropriate number of DLUs. If its anything like
Unity licensing, the turn-around is pretty quick. I'm assuming you need a
SASU agreement for callmanager for them to do it.

We are getting ready to do the same process here this summer with our 4.1system.

Hope that helps.

Ed


On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>  Hello all;
>
> A customer wants to upgrade from CM 4.1 to 5.1, what I am worried about is
> the licsences issue. How will I get the DLUs for callmanager 5.1 we have a
> lic. with each phone but how we will get the DLUs??
>
>  Thanks and Best Regards
> *
> Ahmed A. Elnagar
> *Network Field Engineer
>
> Advanced Computer Technology (ACT)
> 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> Postal Code:12411 Cairo Egypt
>
> *Mob**:* +2010-2833868
> *Website**: *www.act-eg.com
> *E-mail**: *aelnagar at act-eg.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From aelnagar at ACT-EG.COM  Wed May  2 08:19:16 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Wed, 2 May 2007 15:19:16 +0300
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
References: <88C8D019DA547C4BABECB9A806ED22ED88A029@mail.ACT-EG.COM>
	<94a1afde0705020516g74d35ceck1ed62f9f820568ac@mail.gmail.com>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A02D@mail.ACT-EG.COM>

thanks Ed for your reply, yes you are right I contacted Partner help and they told me that during the upgrade a file will be generated that when given to cisco they will give us back the DLUs equivalent to what configured on the system. I think this is an illegal workaround for the lic. issue as by that someone could buy phones without lic. and then setup them with CM 4.1 and then upgrade and he will be having a DLUs for free. I think Cisco has to do something to solve this.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: Ed Leatherman [mailto:ealeatherman at gmail.com]
Sent: Wed 02-May-07 3:16 PM
To: Ahmed Elnagar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1


hi Ahmed,

When you upgrade, callmanager will generate a temporary license for all the devices you have configured from 4.1. If I understand correctly, you just have to send that off to cisco's licensing group and have it converted to a real 5.2 license with the appropriate number of DLUs. If its anything like Unity licensing, the turn-around is pretty quick. I'm assuming you need a SASU agreement for callmanager for them to do it.

We are getting ready to do the same process here this summer with our 4.1 system.

Hope that helps.

Ed



On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote: 

	Hello all;
	 
	A customer wants to upgrade from CM 4.1 to 5.1, what I am worried about is the licsences issue. How will I get the DLUs for callmanager 5.1 we have a lic. with each phone but how we will get the DLUs?? 
	 
	
	Thanks and Best Regards
	
	Ahmed A. Elnagar
	Network Field Engineer 
	 
	Advanced Computer Technology (ACT)
	16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
	Postal Code:12411 Cairo Egypt
	
	Mob : +2010-2833868
	Website: www.act-eg.com <http://www.act-eg.com/> 
	E-mail: aelnagar at act-eg.com

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip 
	
	




-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 
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From jefflin.choi at gmail.com  Wed May  2 08:24:37 2007
From: jefflin.choi at gmail.com (Jefflin Choi)
Date: Wed, 2 May 2007 20:24:37 +0800
Subject: [cisco-voip] conference bridge
Message-ID: <5c72afa80705020524l5e81d0ces6bde0f991e4b8f91@mail.gmail.com>

Hi All,

We have a central office using g711 and a remote branch with g729(Due to 64k
framerelay link).

 We only have dsp's at the central site. In order to make a conference, we
configured it as conference bridge and place it as media resource for the
remote and the central site.

The problem is, when doing adhoc conference with the remote branch, the
quality of the remote phone degrades to an unacceptable situation.  Without
conference, the voice quality is ok.

Testing the conference with 100mbps link, voice quality is better so it
seems like bandwidth is the issue.

Since configuring this for the first time...had few questions hope someone
could enlighten me.
1> what will be the bandwidth required when the remote branch did an adhoc
conference with for example 2 phones. 2 phones at central site that uses
g711. Is it still 24k?

2>I had this configuration where the maximum session is 2. Is this 2
conference with 8 participants? Can the i use this as 5 conference with 3
participants?

3>Can i use a transcoder instead of conference bridge and transcode the g729
of the remote to g711 then the participants will use the callmanager's
software conference bridge to do a conference using all g711?

4>Trying to configure the Media resource group list using the software
conference bridge and fallback to the the enhance ios conference bridge if
in case g729 comes in but it didn't work. Seems like i have to configure
every phone on central and remote to use the enhance ios conference bridge.
Is there a way to allow this?

Thanks in adv.
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From ealeatherman at gmail.com  Wed May  2 08:29:37 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 2 May 2007 08:29:37 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED88A02D@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED88A029@mail.ACT-EG.COM>
	<94a1afde0705020516g74d35ceck1ed62f9f820568ac@mail.gmail.com>
	<88C8D019DA547C4BABECB9A806ED22ED88A02D@mail.ACT-EG.COM>
Message-ID: <94a1afde0705020529p414ddb22y6ed5704cd7eedfd7@mail.gmail.com>

I thought the same thing. I guess the added cost of trying to figure out how
many licenses you OR your company have ever purchased through potentially
multiple vendors, not even counting companies getting bought out or moving
hardware between systems was higher than what they would expect to lose from
people not buying licenses in the first place.

My only annoyance with the system is if I have 300 phones in stock that
arent in use when I upgrade, I need to scan them all in and provision them
in callmanager anyway just to get the DLU's from them (since they were
purchased with old device licenses). OR try to convince cisco to give me
more DLU's for them :)

On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>  thanks Ed for your reply, yes you are right I contacted Partner help and
> they told me that during the upgrade a file will be generated that when
> given to cisco they will give us back the DLUs equivalent to what configured
> on the system. I think this is an illegal workaround for the lic. issue as
> by that someone could buy phones without lic. and then setup them with CM
> 4.1 and then upgrade and he will be having a DLUs for free. I think Cisco
> has to do something to solve this.
>
>  Thanks and Best Regards
> *
> Ahmed A. Elnagar
> *Network Field Engineer
>
> Advanced Computer Technology (ACT)
> 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> Postal Code:12411 Cairo Egypt
>
> *Mob**:* +2010-2833868
> *Website**: *www.act-eg.com
> *E-mail**: *aelnagar at act-eg.com
>
> ------------------------------
> *From:* Ed Leatherman [mailto:ealeatherman at gmail.com]
> *Sent:* Wed 02-May-07 3:16 PM
> *To:* Ahmed Elnagar
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
> hi Ahmed,
>
> When you upgrade, callmanager will generate a temporary license for all
> the devices you have configured from 4.1. If I understand correctly, you
> just have to send that off to cisco's licensing group and have it converted
> to a real 5.2 license with the appropriate number of DLUs. If its anything
> like Unity licensing, the turn-around is pretty quick. I'm assuming you need
> a SASU agreement for callmanager for them to do it.
>
> We are getting ready to do the same process here this summer with our 4.1system.
>
> Hope that helps.
>
> Ed
>
>
> On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
> >
> >  Hello all;
> >
> > A customer wants to upgrade from CM 4.1 to 5.1, what I am worried about
> > is the licsences issue. How will I get the DLUs for callmanager 5.1 we
> > have a lic. with each phone but how we will get the DLUs??
> >
> >  Thanks and Best Regards
> > *
> > Ahmed A. Elnagar
> > *Network Field Engineer
> >
> > Advanced Computer Technology (ACT)
> > 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> > Postal Code:12411 Cairo Egypt
> >
> > *Mob** :* +2010-2833868
> > *Website**: *www.act-eg.com
> > *E-mail**: *aelnagar at act-eg.com
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From jonvoip at gmail.com  Wed May  2 09:18:42 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 2 May 2007 08:18:42 -0500
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <94a1afde0705020516g74d35ceck1ed62f9f820568ac@mail.gmail.com>
References: <88C8D019DA547C4BABECB9A806ED22ED88A029@mail.ACT-EG.COM>
	<94a1afde0705020516g74d35ceck1ed62f9f820568ac@mail.gmail.com>
Message-ID: <5d093f9a0705020618k5193f05co8020f7712f82d768@mail.gmail.com>

Actually, you don't need anything from Cisco, you don't even need to contact
Cisco in any way.

Just go to system, licensing, upload license file on your newly upgraded CCM
5.1 box, you will see a license file in there... click 'view license' and
copy that info to notepad.

Now go to OS platform and go to network and get the MAC address.

Go here:
https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet?FormId=806and
enter the MAC address and paste the info from notepad, fill in the
form.

You should have your real license in about 4 minutes.



Jonathan

On 5/2/07, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> hi Ahmed,
>
> When you upgrade, callmanager will generate a temporary license for all
> the devices you have configured from 4.1. If I understand correctly, you
> just have to send that off to cisco's licensing group and have it converted
> to a real 5.2 license with the appropriate number of DLUs. If its anything
> like Unity licensing, the turn-around is pretty quick. I'm assuming you need
> a SASU agreement for callmanager for them to do it.
>
> We are getting ready to do the same process here this summer with our 4.1system.
>
> Hope that helps.
>
> Ed
>
>
> On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
> >  Hello all;
> >
> > A customer wants to upgrade from CM 4.1 to 5.1, what I am worried about
> > is the licsences issue. How will I get the DLUs for callmanager 5.1 we
> > have a lic. with each phone but how we will get the DLUs??
> >
> >  Thanks and Best Regards
> > *
> > Ahmed A. Elnagar
> > *Network Field Engineer
> >
> > Advanced Computer Technology (ACT)
> > 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> > Postal Code:12411 Cairo Egypt
> >
> > *Mob** :* +2010-2833868
> > * Website**: * www.act-eg.com
> > *E-mail**: * aelnagar at act-eg.com
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From cbomba at s4nets.com  Wed May  2 09:05:15 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Wed, 2 May 2007 09:05:15 -0400 (EDT)
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
Message-ID: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>

Who said you had to buy the phones.  You can just put phones in with bogus MAC addresses and it will generate DLUs for those.  I thought they had to be registered but turns out that they just have to be present in the system.  You can create IP communicators or any other phone to boost the DLUs.  That is a big problem since anybody can do this.

Chris

--original message--

thanks Ed for your reply, yes you are right I contacted Partner help and they told me that during the upgrade a file will be generated that when given to cisco they will give us back the DLUs equivalent to what configured on the system. I think this is an illegal workaround for the lic. issue as by that someone could buy phones without lic. and then setup them with CM 4.1 and then upgrade and he will be having a DLUs for free. I think Cisco has to do something to solve this.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: Ed Leatherman [mailto:ealeatherman at gmail.com]
Sent: Wed 02-May-07 3:16 PM
To: Ahmed Elnagar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1


hi Ahmed,

When you upgrade, callmanager will generate a temporary license for all the devices you have configured from 4.1. If I understand correctly, you just have to send that off to cisco's licensing group and have it converted to a real 5.2 license with the appropriate number of DLUs. If its anything like Unity licensing, the turn-around is pretty quick. I'm assuming you need a SASU agreement for callmanager for them to do it.

We are getting ready to do the same process here this summer with our 4.1 system.

Hope that helps.

Ed



On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote: 

	Hello all;
	 
	A customer wants to upgrade from CM 4.1 to 5.1, what I am worried about is the licsences issue. How will I get the DLUs for callmanager 5.1 we have a lic. with each phone but how we will get the DLUs?? 
	 
	
	Thanks and Best Regards
	
	Ahmed A. Elnagar
	Network Field Engineer 
	 
	Advanced Computer Technology (ACT)
	16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
	Postal Code:12411 Cairo Egypt
	
	Mob : +2010-2833868
	Website: www.act-eg.com <http://www.act-eg.com/> 
	E-mail: aelnagar at act-eg.com

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip 
	
	




-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 
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------------------------------

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


End of cisco-voip Digest, Vol 51, Issue 4
*****************************************


From Scott.Voll at wesd.org  Wed May  2 11:00:23 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 2 May 2007 08:00:23 -0700
Subject: [cisco-voip] conference bridge
In-Reply-To: <5c72afa80705020524l5e81d0ces6bde0f991e4b8f91@mail.gmail.com>
References: <5c72afa80705020524l5e81d0ces6bde0f991e4b8f91@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE6F0@Apollo.wesd.org>

1.         With locations it should setup two 24k streams but if your
using CM I'm not sure how that works because I thought it only did G711
so maybe it's trying to do 160K(80k each) and that's why it seems bad.

2.         Is this the IOS or CM?  Are you talking about Meetme or
Adhoc.  I think your better option over a 64k FR link would be Meetme
with an IOS conference bridge at the central site so only one stream is
going over the FR link.  Maxes are configured in the advanced services
page of CM.

3.         I believe that software conference bridge in CM is G711 only
if I remember right.  I would always use an IOS Conference resource if
possible.

4.         I think the IOS is always the better option. But that's just
my two cents.

 

Scott

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jefflin Choi
Sent: Wednesday, May 02, 2007 5:25 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] conference bridge

 

Hi All,

 

We have a central office using g711 and a remote branch with g729(Due to
64k framerelay link). 

 

We only have dsp's at the central site. In order to make a conference,
we configured it as conference bridge and place it as media resource for
the remote and the central site.

 

The problem is, when doing adhoc conference with the remote branch, the
quality of the remote phone degrades to an unacceptable situation.
Without conference, the voice quality is ok. 

 

Testing the conference with 100mbps link, voice quality is better so it
seems like bandwidth is the issue.

 

Since configuring this for the first time...had few questions hope
someone could enlighten me.

1> what will be the bandwidth required when the remote branch did an
adhoc conference with for example 2 phones. 2 phones at central site
that uses g711. Is it still 24k?

 

2>I had this configuration where the maximum session is 2. Is this 2
conference with 8 participants? Can the i use this as 5 conference with
3 participants?

 

3>Can i use a transcoder instead of conference bridge and transcode the
g729 of the remote to g711 then the participants will use the
callmanager's software conference bridge to do a conference using all
g711? 

 

4>Trying to configure the Media resource group list using the software
conference bridge and fallback to the the enhance ios conference bridge
if in case g729 comes in but it didn't work. Seems like i have to
configure every phone on central and remote to use the enhance ios
conference bridge. Is there a way to allow this? 

 

Thanks in adv.

 

 

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From Roconnell at unislumin.com  Wed May  2 11:10:16 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Wed, 2 May 2007 11:10:16 -0400
Subject: [cisco-voip] Unity Voicemail CallManager applet
In-Reply-To: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
Message-ID: <18082B35CF0EBD4C8D265874508A615C05BCECBB@hermes.unislumin.com>

Hello all,

Does anyone know if that applet plugin for CallManager that allows you
to add mailbox's right from the CCM GUI is supported in CM 5.x? The
following URL has mention on how to set it up "see step 19" 

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/sys_a
d/5_0_4/ccmsys/a06unity.htm#wp1024205

My thoughts are that this is a typo and that we can't use
voicemailbox.asp on Callmanager 5 since it's an appliance?

Please correct me if I'm wrong

Ryan
Attention:

Privileged/Confidential Information may be contained in this message.  Disclosure to any person other than the named recipient is unauthorized.  If you are not the intended recipient, please delete all copies of this information and kindly notify the sender by reply email.  Opinions, conclusions and other information in this message that do not relate to the official business of UNIS LUMIN Inc. shall be understood as neither given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries reserve the right to monitor all e-mail communications through its networks.  Thank you.


From akoul at dataways.gr  Wed May  2 11:35:00 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Wed, 02 May 2007 18:35:00 +0300
Subject: [cisco-voip] Phone Registration Rejected after adding BLFs on
 7914
In-Reply-To: <769361.78989.qm@web57202.mail.re3.yahoo.com>
References: <769361.78989.qm@web57202.mail.re3.yahoo.com>
Message-ID: <4638AFA4.5060508@dataways.gr>

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From asobihoudai at yahoo.com  Wed May  2 11:31:29 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 2 May 2007 08:31:29 -0700 (PDT)
Subject: [cisco-voip] Phone Registration Rejected after adding BLFs on
	7914
In-Reply-To: <4638564E.2070500@dataways.gr>
Message-ID: <769361.78989.qm@web57202.mail.re3.yahoo.com>

Sorry for my density...what are BLFs?

Paul


--- Anthony Kouloglou <akoul at dataways.gr> wrote:


---------------------------------
    Hi,
i have a 7961G running SCCP41.8-2-2SR1S on a ccm
5.0(4)
The phone has also 2 expansion modules 7914 (5.0.2)
The phone has only one line and all the rest are BLF
SD (5 from thephone + 2x14 from modules)

I have used 5 first BLFs from the phone, 14 BLFs from
the first 7914and 2 from the second.

When i try to add more extensions on the rest BLFs of 
the second7914 the phones gets unregistered and
displays
"Registration Rejected..."
Is it some kind of cisco bug?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From rratliff at cisco.com  Wed May  2 12:37:33 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 2 May 2007 12:37:33 -0400
Subject: [cisco-voip] Phone Registration Rejected after adding BLFs on
	7914
In-Reply-To: <4638AFA4.5060508@dataways.gr>
References: <769361.78989.qm@web57202.mail.re3.yahoo.com>
	<4638AFA4.5060508@dataways.gr>
Message-ID: <2E9CD916-3624-4317-8465-47CA0EF472B5@cisco.com>

Sounds like CSCsg12278 - BLF,StationMaxButtonTemplateSize is reached,  
user gets registration rej.

Looks like the limit is determined by the number of CMs in your CM  
group.

Fixed in 5.1(1b) and a 5.0(4)ES but you should be running 5.1.1 anyway.

-Ryan

On May 2, 2007, at 11:35 AM, Anthony Kouloglou wrote:

Busy Lamp Fields,
they monitor a line state of another extension and they are also  
speed dials.
I have noticed that i can use only 21 BLFs!!
When i configure the 22nd, i get registration rejected!
When i erase one of the 22 BLFS (on the device, 1st 7914 or 2nd 7914)  
it gets registered again!!
Here is what i got from debugging:

05/02/2007 16:59:10.975 CCM|Processing StationReg. regCount: 1  
DeviceName=SEP001B0CDB3376, TCPPid = [1.100.7.13678],  
IPAddr=192.168.101.150, Port=50293, Device Controller=[1,60,5904]| 
<CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
0020>
05/02/2007 16:59:10.975 CCM|StationD:    (0005904)  
wait_register_StationRegister - Check MAC address & protocolVer  
(mac=, protocolVer=-2056257527)|<CLID::StandAloneCluster><NID:: 
192.168.101.5><CT::1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::Detailed><MASK::0800>
05/02/2007 16:59:11.035 CCM|ProcessDb - ERROR  
readStationRegistrationProfileByDeviceName -  
readStationTspByDeviceName failed|<CLID::StandAloneCluster><NID:: 
192.168.101.5><CT::1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::0800>
05/02/2007 16:59:11.035 CCM|TspError - Error in TSP. Port IsoEthPort: 
0 Port DSL:0 Name of Device:SEP001B0CDB3376 App ID:Cisco CallManager  
Cluster ID:StandAloneCluster Node ID:ufbccm| 
<CLID::StandAloneCluster><NID:: 
192.168.101.5><CT::AlarmSEP001B0CDB3376><DEV::SEP001B0CDB3376><LVL::All> 
<MASK::ffff>
05/02/2007 16:59:11.035 CCM|StationD:    (0005904) Error,  
registrationProfile_discovery_DbStationRegistrationProfileErr| 
<CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::0800>
05/02/2007 16:59:11.035 CCM|StationD:    (0005904) RegisterReject  
text='Error: DB Config'.|<CLID::StandAloneCluster><NID:: 
192.168.101.5><CT::1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
0020>
05/02/2007 16:59:11.076 CCM|StationInit: (0005904) alarmSeverity=2  
text="24: Name=SEP001B0CDB3376 Load= SCCP41.8-2-2SR1S Last=Phone-Reg- 
Rej" parm1=0(0) parm2=0(0).|<CLID::StandAloneCluster><NID:: 
192.168.101.5><CT::1,100,59,1.1905172><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
0020>
05/02/2007 16:59:11.173 CCM|StationInit:   TCPPid = [1.100.7.13678] 
Socket Broken. DeviceName=SEP001B0CDB3376,IPAddr=192.168.101.150,  
Port=0xc475, Device Controller=[1,60,5904]| 
<CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::0800>
05/02/2007 16:59:11.173 CCM|Closing Station connection  
DeviceName=SEP001B0CDB3376, TCPPid = [1.100.7.13678],  
IPAddr=192.168.101.150, Port=50293, Device Controller=[1,60,5904]| 
<CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
0020>
05/02/2007 16:59:11.174 CCM|DeviceManager:star_DeviceStop  
Name=SEP001B0CDB3376 Key= RegisterDevice=81| 
<CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::Detailed><MASK::ffffff>
05/02/2007 16:59:11.174 CCM|DeviceManager:star_DeviceStop Device =  
SEP001B0CDB3376 not found in the table. RegisterDevice=81| 
<CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::ffffff>
05/02/2007 16:59:13.039 CCM|StationInit:   TCPPid = [1.100.7.13678]  
StationCloseReq received: output|<CLID::StandAloneCluster><NID:: 
192.168.101.5><CT::1,100,59,1.1905171><IP:: 
192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
0800>



Paul Choi wrote:
> Sorry for my density...what are BLFs? Paul --- Anthony Kouloglou  
> <akoul at dataways.gr> wrote: --------------------------------- Hi, i  
> have a 7961G running SCCP41.8-2-2SR1S on a ccm 5.0(4) The phone has  
> also 2 expansion modules 7914 (5.0.2) The phone has only one line  
> and all the rest are BLF SD (5 from thephone + 2x14 from modules) I  
> have used 5 first BLFs from the phone, 14 BLFs from the first  
> 7914and 2 from the second. When i try to add more extensions on the  
> rest BLFs of the second7914 the phones gets unregistered and  
> displays "Registration Rejected..." Is it some kind of cisco bug?
>> _______________________________________________ cisco-voip mailing  
>> list cisco-voip at puck.nether.net https://puck.nether.net/mailman/ 
>> listinfo/cisco-voip
> __________________________________________________ Do You Yahoo!?  
> Tired of spam? Yahoo! Mail has the best spam protection around  
> http://mail.yahoo.com
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jgrace at adcapnet.com  Wed May  2 14:48:26 2007
From: jgrace at adcapnet.com (James Grace)
Date: Wed, 2 May 2007 14:48:26 -0400
Subject: [cisco-voip] setting up t1 tie line E&M
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C643@mail.adcapnetsvc.local>

?
i need to set up a t1 tie line from my voice gateway to a ivr server.  i need to provide clocking on my voice gateway side.  im doing a mgcp e&m 24 channel.  whats the cmds for clocking to make sure im the clocking source
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
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From pmowry at getgds.com  Wed May  2 14:53:29 2007
From: pmowry at getgds.com (Patrick Mowry)
Date: Wed, 2 May 2007 13:53:29 -0500
Subject: [cisco-voip] Upgrade to 4.2 or 4.3?
In-Reply-To: <mailman.5.1178121605.30589.cisco-voip@puck.nether.net>
References: <mailman.5.1178121605.30589.cisco-voip@puck.nether.net>
Message-ID: <4C792A099259634C83F97067671B37FE02EFD322@GDSEXC.gds.pri>

Hello,
  I need to upgrade a 4.0.2 CallManager Cluster to something newer for
continuing support and adding IPCC to the mix.  I was looking at
upgrading to 4.2 and using IPCC 3.5(4), but the customer has asked about
Cisco's stance on the continued support for Win2K.  The concern is
Microsoft's support ends in June of 2010, and they may not be ready for
another CallManager upgrade by then.

  Is there an official stance on what will happen to Win2K based systems
by then?  

  Is anyone running 4.3 with IPCC that can take away my fear of going to
it with all the backend changes?  I view 4.3 as being like 5.0.

Thanks,

-Patrick


From CarlosOrtiz at bayviewfinancial.com  Wed May  2 15:58:18 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 2 May 2007 15:58:18 -0400
Subject: [cisco-voip] CCM BARS Restore
Message-ID: <OF5752E89C.F087E47B-ON852572CF.006BA08A-852572CF.006DB559@bftg.com>

I just did a BARS 4.10 restore of CCM 4.1.3.   The only errors in the log 
are because the 2 subscribers could not be connected to which I expected. 
Everything else shows completed successfully.  When I reboot the server 
the following services are in a hung "Starting" state. 

Cisco Call Mgr
CDR Insert
CTI Manager
Database Layer Monitor
RIS Data Collector
TFTP

Anyone seen this before?
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From bms314 at gmail.com  Wed May  2 16:15:30 2007
From: bms314 at gmail.com (Brian)
Date: Wed, 2 May 2007 15:15:30 -0500
Subject: [cisco-voip] LDAP Directory Integration in Call Manager 5.1
In-Reply-To: <201EE253-4A57-4515-90EF-C8CCC67C83E0@cisco.com>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451E1C@CSEXCH01.charles-stanley.co.uk>
	<201EE253-4A57-4515-90EF-C8CCC67C83E0@cisco.com>
Message-ID: <7aaaf7f0705021315l59ea9462v2c4d74cf3b9a5ddf@mail.gmail.com>

Which traces would you recommend running to verify the sync is doing
something?

Brian

On 4/4/07, Ryan Ratliff <rratliff at cisco.com> wrote:
>
> Make sure you have activated the LDAP Synch service, then try the
> synch again.  Other than that you can look at traces for it to make
> sure it is doing something.  I ended up leaving mine overnight and my
> lab AD has all of 2 end users.
>
> -Ryan
>
> On Apr 4, 2007, at 10:12 AM, Nick Kassel wrote:
>
> Thank you Ryan, I have added the correct format in and CM has accepted
> it and updated the configuration, however I have tried to 'Perform Full
> Sync Now' and it says update successful but how do I know if something
> is happening, I have left it for a while but no users have been
> imported.
>
> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> Sent: 04 April 2007 13:50
> To: Nick Kassel
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] LDAP Directory Integration in Call Manager 5.1
>
> The LDAP manager can just be user at domain, no need to use the full
> dn=... format.  The user search base does need to be something like
> cn=Users,dc=domain,dc=com.  My ldap manager is a domain admin but
> since we don't write anything to AD I'm pretty sure all it needs is
> read permission.
>
> Note that you will need to restart Tomcat on your subscribers after
> enabling the ldap synch. The pub picks up the change automatically
> (ie for CCMUser auth) but the subs don't for some reason.
>
> -Ryan
>
> On Apr 4, 2007, at 8:47 AM, Nick Kassel wrote:
>
> I am trying to test integrating CCM 5.1 with Active Directory in
> vmware, has anyone done this in 5.1, It should be fairly straight
> forward but I am unsure if the LDAP Manager distinguished name and
> LDAP user search base I am entering are in the correct format. Also
> what rights are needed for the AD user for this to take place.
>
>
> If anyone could provide there configuration format I would be very
> grateful.
>
>
>
>
>
>
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> https://puck.nether.net/mailman/listinfo/cisco-voip
> ********************SAVE PAPER - THINK BEFORE YOU
> PRINT**************************************************
>
> The information contained in this e-mail is strictly confidential,
> some or all of which may be legally privileged.
> Access to this e-mail by any person other than the recipient is
> prohibited. If you have received this message in
> error, any use, disclosure, copying, printing, distribution of,
> replying to or any action taken or omitted to be
> taken in reliance on this e-mail, is prohibited. Please advise the
> sender immediately should this e-mail have been
> incorrectly addressed or transmitted, and then delete the email and
> any attachment sent with it from your computer.
>
> You are advised that urgent, time sensitive and confidential
> communications should not be sent by e-mail. You
> accept that any instructions are deemed to have been given at the
> time the recipient(s) accesses them and that
> delivery receipt does not constitute acknowledgement or receipt by
> the intended recipient(s).
>
> You acknowledge that e-mails are not secure and you accept the risk
> of malfunction, viruses, unauthorised
> interference, mis-delivery or delay. Charles Stanley reserves the
> right to monitor and/or record emails sent and
> received via its network for any lawful business purpose in
> accordance with applicable law and regulations.
> ************************************************************************
> *******************************
>
>
> Charles Stanley & Co. Ltd
> Registered Office: 25 Luke Street London EC2A 4AR
>
> Tel: 0207 739 8200 Fax: 0207 739 7798
> Registered in England No. 1903304
>
> Charles Stanley Sutherlands and Charles Stanley Securities are
> divisions of Charles Stanley & Co. Ltd
>
> Authorised and Regulated by the Financial Services Authority, Member
> of the London Stock Exchange, The
> International Capital Market Association and The London International
> Financial Futures & Options Exchange.
>
> This footnote also confirms that this email message has been swept by
> McAfee VirusScan and SurfControl Email
> Filter software.
>
> _______________________________________________
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From jonvoip at gmail.com  Wed May  2 18:41:24 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 2 May 2007 17:41:24 -0500
Subject: [cisco-voip] IPCC 4.0(5) SQL
Message-ID: <5d093f9a0705021541p26cd29balc5ac7bfefca94008@mail.gmail.com>

OK, so I am installing a quad-server IPCC 4.05 cluster (two CRS Engines, two
DB/Recording), I install SQL on the two DB servers and not the CRS Engine,
right? Or does the CRS Engine need SQL to store the IPCC DB?

Also, I want to see if I understand the installation instructions...

I am going to install IPCC on the first CRS node, uncheck everything except
the CRS Engine and install... once I get the popup saying I can install the
other servers, I do so... at what point do I install the second CRS Engine?

I am going to point all of the servers at the same LDAP Profile, is this the
method by which it knows they are all in the same cluster?




Jonathan
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From akoul at dataways.gr  Thu May  3 01:54:52 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 03 May 2007 08:54:52 +0300
Subject: [cisco-voip] Phone Registration Rejected after adding BLFs on
 7914
In-Reply-To: <2E9CD916-3624-4317-8465-47CA0EF472B5@cisco.com>
References: <769361.78989.qm@web57202.mail.re3.yahoo.com>
	<4638AFA4.5060508@dataways.gr>
	<2E9CD916-3624-4317-8465-47CA0EF472B5@cisco.com>
Message-ID: <4639792C.6040504@dataways.gr>

Thanks Ryan,
so there is no way overcoming the problem by editing a parameter or 
something like this?
I have a cluster of 2 CCMs each running 5.0.4.2000-1

Cheers,
Anthony


Ryan Ratliff wrote:
> Sounds like CSCsg12278 - BLF,StationMaxButtonTemplateSize is reached, 
> user gets registration rej.
>
> Looks like the limit is determined by the number of CMs in your CM group.
>
> Fixed in 5.1(1b) and a 5.0(4)ES but you should be running 5.1.1 anyway.
>
> -Ryan
>
> On May 2, 2007, at 11:35 AM, Anthony Kouloglou wrote:
>
> Busy Lamp Fields,
> they monitor a line state of another extension and they are also speed 
> dials.
> I have noticed that i can use only 21 BLFs!!
> When i configure the 22nd, i get registration rejected!
> When i erase one of the 22 BLFS (on the device, 1st 7914 or 2nd 7914) 
> it gets registered again!!
> Here is what i got from debugging:
>
> 05/02/2007 16:59:10.975 CCM|Processing StationReg. regCount: 1 
> DeviceName=SEP001B0CDB3376, TCPPid = [1.100.7.13678], 
> IPAddr=192.168.101.150, Port=50293, Device 
> Controller=[1,60,5904]|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::State 
> Transition><MASK::0020>
> 05/02/2007 16:59:10.975 CCM|StationD:    (0005904) 
> wait_register_StationRegister - Check MAC address & protocolVer (mac=, 
> protocolVer=-2056257527)|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::Detailed><MASK::0800> 
>
> 05/02/2007 16:59:11.035 CCM|ProcessDb - ERROR 
> readStationRegistrationProfileByDeviceName - 
> readStationTspByDeviceName 
> failed|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::0800> 
>
> 05/02/2007 16:59:11.035 CCM|TspError - Error in TSP. Port IsoEthPort:0 
> Port DSL:0 Name of Device:SEP001B0CDB3376 App ID:Cisco CallManager 
> Cluster ID:StandAloneCluster Node 
> ID:ufbccm|<CLID::StandAloneCluster><NID::192.168.101.5><CT::AlarmSEP001B0CDB3376><DEV::SEP001B0CDB3376><LVL::All><MASK::ffff> 
>
> 05/02/2007 16:59:11.035 CCM|StationD:    (0005904) Error, 
> registrationProfile_discovery_DbStationRegistrationProfileErr|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::0800> 
>
> 05/02/2007 16:59:11.035 CCM|StationD:    (0005904) RegisterReject 
> text='Error: DB 
> Config'.|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::State 
> Transition><MASK::0020>
> 05/02/2007 16:59:11.076 CCM|StationInit: (0005904) alarmSeverity=2 
> text="24: Name=SEP001B0CDB3376 Load= SCCP41.8-2-2SR1S 
> Last=Phone-Reg-Rej" parm1=0(0) 
> parm2=0(0).|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905172><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::State 
> Transition><MASK::0020>
> 05/02/2007 16:59:11.173 CCM|StationInit:   TCPPid = 
> [1.100.7.13678]Socket Broken. 
> DeviceName=SEP001B0CDB3376,IPAddr=192.168.101.150, Port=0xc475, Device 
> Controller=[1,60,5904]|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::0800> 
>
> 05/02/2007 16:59:11.173 CCM|Closing Station connection 
> DeviceName=SEP001B0CDB3376, TCPPid = [1.100.7.13678], 
> IPAddr=192.168.101.150, Port=50293, Device 
> Controller=[1,60,5904]|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::State 
> Transition><MASK::0020>
> 05/02/2007 16:59:11.174 CCM|DeviceManager:star_DeviceStop 
> Name=SEP001B0CDB3376 Key= 
> RegisterDevice=81|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::Detailed><MASK::ffffff> 
>
> 05/02/2007 16:59:11.174 CCM|DeviceManager:star_DeviceStop Device = 
> SEP001B0CDB3376 not found in the table. 
> RegisterDevice=81|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::ffffff> 
>
> 05/02/2007 16:59:13.039 CCM|StationInit:   TCPPid = [1.100.7.13678] 
> StationCloseReq received: 
> output|<CLID::StandAloneCluster><NID::192.168.101.5><CT::1,100,59,1.1905171><IP::192.168.101.150><DEV::SEP001B0CDB3376><LVL::State 
> Transition><MASK::0800>
>
>
>
> Paul Choi wrote:
>> Sorry for my density...what are BLFs? Paul --- Anthony Kouloglou 
>> <akoul at dataways.gr> wrote: --------------------------------- Hi, i 
>> have a 7961G running SCCP41.8-2-2SR1S on a ccm 5.0(4) The phone has 
>> also 2 expansion modules 7914 (5.0.2) The phone has only one line and 
>> all the rest are BLF SD (5 from thephone + 2x14 from modules) I have 
>> used 5 first BLFs from the phone, 14 BLFs from the first 7914and 2 
>> from the second. When i try to add more extensions on the rest BLFs 
>> of the second7914 the phones gets unregistered and displays 
>> "Registration Rejected..." Is it some kind of cisco bug?
>>> _______________________________________________ cisco-voip mailing 
>>> list cisco-voip at puck.nether.net 
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> __________________________________________________ Do You Yahoo!? 
>> Tired of spam? Yahoo! Mail has the best spam protection around 
>> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From rratliff at cisco.com  Thu May  3 08:33:14 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 3 May 2007 08:33:14 -0400
Subject: [cisco-voip] Phone Registration Rejected after adding BLFs on
	7914
In-Reply-To: <4639792C.6040504@dataways.gr>
References: <769361.78989.qm@web57202.mail.re3.yahoo.com>
	<4638AFA4.5060508@dataways.gr>
	<2E9CD916-3624-4317-8465-47CA0EF472B5@cisco.com>
	<4639792C.6040504@dataways.gr>
Message-ID: <AB00CA67-C54D-420F-8138-4D0A01769D7E@cisco.com>

No, you can either remove one of the servers from the CM list or stay  
below the limit described until you upgrade.

In case you didn't look the bug up on the web:
Symptom:
When adding 15 BLF(BusyLampField), you will get registration rejected  
on 7970

Conditions:
CallManager 5.0(4), 3 CallManager cluster

Workaround:
Use 14 BLF.

Further Problem Description:
StationMaxButtonTemplateSize is set to 42. The BlfArray is bounded to  
that limit. The problem here is if the phone is assigned to a  
CallManagerGroup that contains 3 CallManager, then 14 Blf entries  
will return 14 * 3 = 42 entries in busylampfield query.
So limitation is like so.
Only two CCM, you can add up to 21 blf. 2 * 21=42
Only one CCM, you can add 35  3*14=42
the max button we can have from our IP phone is 7970 + 2 side car =  
36 button. The first line must be line, then 36 -1 = 35 BLF max.
For single CCM, the limit is no longer 42 as phone button template  
can onyl support up to 36.

-Ryan

On May 3, 2007, at 1:54 AM, Anthony Kouloglou wrote:

Thanks Ryan,
so there is no way overcoming the problem by editing a parameter or  
something like this?
I have a cluster of 2 CCMs each running 5.0.4.2000-1

Cheers,
Anthony


Ryan Ratliff wrote:
> Sounds like CSCsg12278 - BLF,StationMaxButtonTemplateSize is  
> reached, user gets registration rej.
>
> Looks like the limit is determined by the number of CMs in your CM  
> group.
>
> Fixed in 5.1(1b) and a 5.0(4)ES but you should be running 5.1.1  
> anyway.
>
> -Ryan
>
> On May 2, 2007, at 11:35 AM, Anthony Kouloglou wrote:
>
> Busy Lamp Fields,
> they monitor a line state of another extension and they are also  
> speed dials.
> I have noticed that i can use only 21 BLFs!!
> When i configure the 22nd, i get registration rejected!
> When i erase one of the 22 BLFS (on the device, 1st 7914 or 2nd  
> 7914) it gets registered again!!
> Here is what i got from debugging:
>
> 05/02/2007 16:59:10.975 CCM|Processing StationReg. regCount: 1  
> DeviceName=SEP001B0CDB3376, TCPPid = [1.100.7.13678],  
> IPAddr=192.168.101.150, Port=50293, Device Controller=[1,60,5904]| 
> <CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
> 1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
> 0020>
> 05/02/2007 16:59:10.975 CCM|StationD:    (0005904)  
> wait_register_StationRegister - Check MAC address & protocolVer  
> (mac=, protocolVer=-2056257527)|<CLID::StandAloneCluster><NID:: 
> 192.168.101.5><CT::1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::Detailed><MASK::0800>
> 05/02/2007 16:59:11.035 CCM|ProcessDb - ERROR  
> readStationRegistrationProfileByDeviceName -  
> readStationTspByDeviceName failed|<CLID::StandAloneCluster><NID:: 
> 192.168.101.5><CT::1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::0800>
> 05/02/2007 16:59:11.035 CCM|TspError - Error in TSP. Port  
> IsoEthPort:0 Port DSL:0 Name of Device:SEP001B0CDB3376 App ID:Cisco  
> CallManager Cluster ID:StandAloneCluster Node ID:ufbccm| 
> <CLID::StandAloneCluster><NID:: 
> 192.168.101.5><CT::AlarmSEP001B0CDB3376><DEV::SEP001B0CDB3376><LVL::Al 
> l><MASK::ffff>
> 05/02/2007 16:59:11.035 CCM|StationD:    (0005904) Error,  
> registrationProfile_discovery_DbStationRegistrationProfileErr| 
> <CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
> 1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::0800>
> 05/02/2007 16:59:11.035 CCM|StationD:    (0005904) RegisterReject  
> text='Error: DB Config'.|<CLID::StandAloneCluster><NID:: 
> 192.168.101.5><CT::1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
> 0020>
> 05/02/2007 16:59:11.076 CCM|StationInit: (0005904) alarmSeverity=2  
> text="24: Name=SEP001B0CDB3376 Load= SCCP41.8-2-2SR1S Last=Phone- 
> Reg-Rej" parm1=0(0) parm2=0(0).|<CLID::StandAloneCluster><NID:: 
> 192.168.101.5><CT::1,100,59,1.1905172><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
> 0020>
> 05/02/2007 16:59:11.173 CCM|StationInit:   TCPPid = [1.100.7.13678] 
> Socket Broken. DeviceName=SEP001B0CDB3376,IPAddr=192.168.101.150,  
> Port=0xc475, Device Controller=[1,60,5904]| 
> <CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
> 1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::0800>
> 05/02/2007 16:59:11.173 CCM|Closing Station connection  
> DeviceName=SEP001B0CDB3376, TCPPid = [1.100.7.13678],  
> IPAddr=192.168.101.150, Port=50293, Device Controller=[1,60,5904]| 
> <CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
> 1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
> 0020>
> 05/02/2007 16:59:11.174 CCM|DeviceManager:star_DeviceStop  
> Name=SEP001B0CDB3376 Key= RegisterDevice=81| 
> <CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
> 1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::Detailed><MASK::ffffff>
> 05/02/2007 16:59:11.174 CCM|DeviceManager:star_DeviceStop Device =  
> SEP001B0CDB3376 not found in the table. RegisterDevice=81| 
> <CLID::StandAloneCluster><NID::192.168.101.5><CT:: 
> 1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::Error><MASK::ffffff>
> 05/02/2007 16:59:13.039 CCM|StationInit:   TCPPid = [1.100.7.13678]  
> StationCloseReq received: output|<CLID::StandAloneCluster><NID:: 
> 192.168.101.5><CT::1,100,59,1.1905171><IP:: 
> 192.168.101.150><DEV::SEP001B0CDB3376><LVL::State Transition><MASK:: 
> 0800>
>
>
>
> Paul Choi wrote:
>> Sorry for my density...what are BLFs? Paul --- Anthony Kouloglou  
>> <akoul at dataways.gr> wrote: --------------------------------- Hi, i  
>> have a 7961G running SCCP41.8-2-2SR1S on a ccm 5.0(4) The phone  
>> has also 2 expansion modules 7914 (5.0.2) The phone has only one  
>> line and all the rest are BLF SD (5 from thephone + 2x14 from  
>> modules) I have used 5 first BLFs from the phone, 14 BLFs from the  
>> first 7914and 2 from the second. When i try to add more extensions  
>> on the rest BLFs of the second7914 the phones gets unregistered  
>> and displays "Registration Rejected..." Is it some kind of cisco bug?
>>> _______________________________________________ cisco-voip  
>>> mailing list cisco-voip at puck.nether.net https://puck.nether.net/ 
>>> mailman/listinfo/cisco-voip
>> __________________________________________________ Do You Yahoo!?  
>> Tired of spam? Yahoo! Mail has the best spam protection around  
>> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From CarlosOrtiz at bayviewfinancial.com  Thu May  3 08:55:10 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 3 May 2007 08:55:10 -0400
Subject: [cisco-voip] CCM BARS Restore
In-Reply-To: <OF5752E89C.F087E47B-ON852572CF.006BA08A-852572CF.006DB51B@LocalDomain>
Message-ID: <OF56BD7CC8.D9A1911F-ON852572D0.0046CD64-852572D0.0046F839@bftg.com>

Update on this.  I came in this morning and all the services are now 
actually started.  I still can't login to the CCM admin page though.  I am 
checking CCO now for help but if anyone has seen this please comment.

Carlos





Carlos Ortiz/MIA/BAY/BFTG 
05/02/2007 03:58 PM

To
Cisco-Voip List
cc

Subject
CCM BARS Restore





I just did a BARS 4.10 restore of CCM 4.1.3.   The only errors in the log 
are because the 2 subscribers could not be connected to which I expected. 
Everything else shows completed successfully.  When I reboot the server 
the following services are in a hung "Starting" state. 

Cisco Call Mgr
CDR Insert
CTI Manager
Database Layer Monitor
RIS Data Collector
TFTP

Anyone seen this before?
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From Johnny.Crothers at vanco.co.uk  Thu May  3 09:04:04 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers)
Date: Thu, 3 May 2007 14:04:04 +0100
Subject: [cisco-voip] CCM BARS Restore
In-Reply-To: <OF56BD7CC8.D9A1911F-ON852572D0.0046CD64-852572D0.0046F839@bftg.com>
Message-ID: <A56BD242744BF540B15D1672E0E3E875F0D3FB@LON-INT-MSGVS01.internal.vancocorp.com>

Check that the WWW service is started.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: 03 May 2007 13:55
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM BARS Restore

 


Update on this.  I came in this morning and all the services are now
actually started.  I still can't login to the CCM admin page though.  I
am checking CCO now for help but if anyone has seen this please comment.


Carlos 





Carlos Ortiz/MIA/BAY/BFTG 

05/02/2007 03:58 PM 

To

Cisco-Voip List 

cc

 

Subject

CCM BARS Restore

 

 

 



I just did a BARS 4.10 restore of CCM 4.1.3.   The only errors in the
log are because the 2 subscribers could not be connected to which I
expected.  Everything else shows completed successfully.  When I reboot
the server the following services are in a hung "Starting" state.   

Cisco Call Mgr 
CDR Insert 
CTI Manager 
Database Layer Monitor 
RIS Data Collector 
TFTP 

Anyone seen this before? 



**********************************************************************
Any opinions expressed in the email are those of the individual and not necessarily the company. This email and any files transmitted with it are confidential and solely for the use of the intended recipient.  If you are not the intended recipient or the person responsible for delivering it to the intended recipient, be advised that you have received this email in error and that any dissemination, distribution, copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with the content of this email please e-mail to: e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software viruses which could damage your own computer system. While the sender has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening any attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
**********************************************************************

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From CarlosOrtiz at bayviewfinancial.com  Thu May  3 09:43:35 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 3 May 2007 09:43:35 -0400
Subject: [cisco-voip] Fw:  CCM BARS Restore
Message-ID: <OFC6588F5B.5A734429-ON852572D0.004B6405-852572D0.004B66C9@bftg.com>

----- Forwarded by Carlos Ortiz/MIA/BAY/BFTG on 05/03/2007 09:43 AM -----

Carlos Ortiz/MIA/BAY/BFTG
05/03/2007 09:43 AM

To
"Johnny Crothers" <Johnny.Crothers at vanco.co.uk>
cc

Subject
Re: [cisco-voip] CCM BARS Restore





Yes it is along with the IIS admin. 

After rebooting again all services are started except the Call Mgr Service 
which is in a "starting" state.  This was what it looked like this morning 
when I came in but after a few minutes the Call Mgr service status showed 
started.   I rebooted to see if it does the same thing again, but at this 
point it's in a "starting" state.



"Johnny Crothers" <Johnny.Crothers at vanco.co.uk> 
Sent by: cisco-voip-bounces at puck.nether.net
05/03/2007 09:04 AM

To
<CarlosOrtiz at bayviewfinancial.com>, <cisco-voip at puck.nether.net>
cc

Subject
Re: [cisco-voip] CCM BARS Restore






Check that the WWW service is started.
 

From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: 03 May 2007 13:55
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM BARS Restore
 

Update on this.  I came in this morning and all the services are now 
actually started.  I still can't login to the CCM admin page though.  I am 
checking CCO now for help but if anyone has seen this please comment. 

Carlos 




Carlos Ortiz/MIA/BAY/BFTG 
05/02/2007 03:58 PM 


To
Cisco-Voip List 
cc
 
Subject
CCM BARS Restore
 


 
 



I just did a BARS 4.10 restore of CCM 4.1.3.   The only errors in the log 
are because the 2 subscribers could not be connected to which I expected. 
Everything else shows completed successfully.  When I reboot the server 
the following services are in a hung "Starting" state.   

Cisco Call Mgr 
CDR Insert 
CTI Manager 
Database Layer Monitor 
RIS Data Collector 
TFTP 

Anyone seen this before? 


**********************************************************************
Any opinions expressed in the email are those of the individual and not 
necessarily the company. This email and any files transmitted with it are 
confidential and solely for the use of the intended recipient. If you are 
not the intended recipient or the person responsible for delivering it to 
the intended recipient, be advised that you have received this email in 
error and that any dissemination, distribution, copying or use is strictly 
prohibited.

If you have received this email in error, or if you are concerned with the 
content of this email please e-mail to: e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software viruses 
which could damage your own computer system. While the sender has taken 
every reasonable precaution to minimise this risk, we cannot accept 
liability for any damage which you sustain as a result of software 
viruses. You should carry out your own virus checks before opening any 
attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John 
Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
**********************************************************************
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From CarlosOrtiz at bayviewfinancial.com  Thu May  3 10:01:39 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 3 May 2007 10:01:39 -0400
Subject: [cisco-voip] CCM BARS Restore
In-Reply-To: <A56BD242744BF540B15D1672E0E3E875F0D448@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <OFEF284CAD.94C3FC27-ON852572D0.004CDBF3-852572D0.004D0E72@bftg.com>

Yes it's plugged in.  I have copied files to and from the server (patches, 
etc) and I am remote controlling the box right now.

Carlos



"Johnny Crothers" <Johnny.Crothers at vanco.co.uk> 
05/03/2007 09:48 AM

To
<CarlosOrtiz at bayviewfinancial.com>
cc

Subject
RE: [cisco-voip] CCM BARS Restore






Just wandering? but is your NIC plugged in? I know this is probably a dumb 
question, but a lot of services wont start if the NIC with the correct IP 
is actually connected. 

From: CarlosOrtiz at bayviewfinancial.com 
[mailto:CarlosOrtiz at bayviewfinancial.com] 
Sent: 03 May 2007 14:43
To: Johnny Crothers
Subject: Re: [cisco-voip] CCM BARS Restore
 

Yes it is along with the IIS admin.   

After rebooting again all services are started except the Call Mgr Service 
which is in a "starting" state.  This was what it looked like this morning 
when I came in but after a few minutes the Call Mgr service status showed 
started.   I rebooted to see if it does the same thing again, but at this 
point it's in a "starting" state. 


"Johnny Crothers" <Johnny.Crothers at vanco.co.uk> 
Sent by: cisco-voip-bounces at puck.nether.net 
05/03/2007 09:04 AM 


To
<CarlosOrtiz at bayviewfinancial.com>, <cisco-voip at puck.nether.net> 
cc
 
Subject
Re: [cisco-voip] CCM BARS Restore
 


 
 




Check that the WWW service is started. 
  
 


From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: 03 May 2007 13:55
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM BARS Restore 
  

Update on this.  I came in this morning and all the services are now 
actually started.  I still can't login to the CCM admin page though.  I am 
checking CCO now for help but if anyone has seen this please comment. 

Carlos 


Carlos Ortiz/MIA/BAY/BFTG 
05/02/2007 03:58 PM 
 


To
Cisco-Voip List 
cc
 
Subject
CCM BARS Restore

 
 


 
 




I just did a BARS 4.10 restore of CCM 4.1.3.   The only errors in the log 
are because the 2 subscribers could not be connected to which I expected. 
Everything else shows completed successfully.  When I reboot the server 
the following services are in a hung "Starting" state.   

Cisco Call Mgr 
CDR Insert 
CTI Manager 
Database Layer Monitor 
RIS Data Collector 
TFTP 

Anyone seen this before? 


**********************************************************************
Any opinions expressed in the email are those of the individual and not 
necessarily the company. This email and any files transmitted with it are 
confidential and solely for the use of the intended recipient. If you are 
not the intended recipient or the person responsible for delivering it to 
the intended recipient, be advised that you have received this email in 
error and that any dissemination, distribution, copying or use is strictly 
prohibited.

If you have received this email in error, or if you are concerned with the 
content of this email please e-mail to: e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software viruses 
which could damage your own computer system. While the sender has taken 
every reasonable precaution to minimise this risk, we cannot accept 
liability for any damage which you sustain as a result of software 
viruses. You should carry out your own virus checks before opening any 
attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John 
Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
**********************************************************************
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From gmukelabai at CAZ.GOV.ZM  Thu May  3 10:06:56 2007
From: gmukelabai at CAZ.GOV.ZM (Garry Mukelabai)
Date: Thu, 3 May 2007 16:06:56 +0200
Subject: [cisco-voip] CISCO VOIP and Video Conferencing System sought.
Message-ID: <02E078BE53FE0D4FB99E1F050DFA00124962BA@server02.caz.gov.zm>

Hi All,

 

Greetings.

 

I am new to this forum just joined today. 

 

I have been reading with great interest the advancements in CISCO IP
telephony and Video Conferencing system. At the Communications Authority
of Zambia, the telecoms regulatory authority for the country we wish to
deploy such a system.

 

As a qualified CCNA, I have had a glimpse of experience in IP telephony
where I worked as an assistant to a CISCO engineer who installed an IP
telephone system based on the CISCO 3845 Router, 7912 IP Phones and Call
Manager Express at a quasi-government institution somewhere in 2005 (I
was working for a CISCO partner Company at the time). This system
terminated in the local PSTN.

 

Our organisation is made up of the following three sites: 

1.	HQ 50 users 
2.	Site office (20 users) 4 kilometres away connected by a
point-to-point wireless bridge
3.	Regional Office (10 users) 330 kilometres away soon to be
connected by routing through ISP using E1 microwave.

 

Our object is to enable the whole organisation communicate seamlessly on
voice, video and data. Using all features such as meetings via video
conferencing.

 

Initially I would therefore like to get as much understanding abut the
systems as I can and be in touch (as I am) with people in the field that
are knowledgeable on this subject especially considering that the
expected investment is huge.

 

I already have an approved budget for the above and need to start on the
ground-work as this was scheduled for the first quarter of this
financial year (April 2007 - June 2007)

 

Cheers.

 

Garry Mukelabai

Manager Information Systems

 

Communications Authority - Zambia

Plot 3141

Corner Lumumba and Buyantanshi Roads

P O Box 36871

Lusaka, ZAMBIA

 

Tel +260 1 241236/246702

Fax + 260 1 246701

cell +260 97 870519

email  <mailto:gmukelabai at caz.gov.zm> gmukelabai at caz.gov.zm

 <http://www.caz.gov.zm> www.caz.gov.zm

 



 
<http://928426.sigclick.mailinfo.com/sigclick/0B01090C/01044F01/054F0302
318323.jpg>  


No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.467 / Virus Database: 269.6.2/785 - Release Date: 5/2/2007
14:16


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From lelio at uoguelph.ca  Thu May  3 10:22:22 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 3 May 2007 10:22:22 -0400
Subject: [cisco-voip] speed dials on 7911s
Message-ID: <12b701c78d8e$76d98db0$47196883@cfs.uoguelph.ca>

On the 7912s, with the handset onhook, the blue bar is used to scroll through speed dials. If none are configured, the lines say 'not configured' or something like that.

Can anyone confirm if the behaviour of the 7911s is the same?

Thanks, Lelio

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From akoul at dataways.gr  Thu May  3 10:27:39 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 03 May 2007 17:27:39 +0300
Subject: [cisco-voip] Easy sftp server (windows 32bit) for CCM 5.x backup?
Message-ID: <4639F15B.5070507@dataways.gr>

Does cisco propose any sftp server for backup of the CCM 5.x?
Can i use a second CCM (eg subscriber) for that?

Thanks
Anthony

From rratliff at cisco.com  Thu May  3 10:54:39 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 3 May 2007 10:54:39 -0400
Subject: [cisco-voip] Easy sftp server (windows 32bit) for CCM 5.x
	backup?
In-Reply-To: <4639F15B.5070507@dataways.gr>
References: <4639F15B.5070507@dataways.gr>
Message-ID: <9EB0CFF0-A42A-4104-A06A-BD1A99E7A98A@cisco.com>

We use freesshd (freesshd.com) here in the lab.

Your CM is not an SFTP server so you cannot use a subscriber for it.

-Ryan

On May 3, 2007, at 10:27 AM, Anthony Kouloglou wrote:

Does cisco propose any sftp server for backup of the CCM 5.x?
Can i use a second CCM (eg subscriber) for that?

Thanks
Anthony
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From CarlosOrtiz at bayviewfinancial.com  Thu May  3 11:16:50 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 3 May 2007 11:16:50 -0400
Subject: [cisco-voip] Perflib error on Call Mgr
Message-ID: <OF043E7126.DD151121-ON852572D0.00536716-852572D0.0053F065@bftg.com>

Event Type: Error
Event Source: Perflib
Event Category: None
Event ID: 1008
Date: 1/25/2006
Time: 11:42:34 AM
User: N/A
Computer: SA-CCM2
Description:
The Open Procedure for service "DHCPServer" in DLL 
"C:\WINNT\system32\perfctrs.dll"failed.  
Performance data for this service will not be available. 
Status code  returned is data DWORD 0. 
Data:
0000: 05 00 00 00         



This error is a know bug on Call Mgr that is eating up log space. 

http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K31823130


 Has anyone executed the proposed fix?  From reading I believe it just 
requires a registry edit to remove from th logs.  Has anyone done this 
already with no issues?

Carlos
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From CarlosOrtiz at bayviewfinancial.com  Thu May  3 11:18:49 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Thu, 3 May 2007 11:18:49 -0400
Subject: [cisco-voip] Phone URI's
In-Reply-To: <f1b65f880704302013j3e11bba9q8f93d25da3b8e343@mail.gmail.com>
Message-ID: <OF9298F19B.8421A15A-ON852572D0.00541B09-852572D0.00541EED@bftg.com>

That did the trick. 

Thanks.....



"Justin Steinberg" <jsteinberg at gmail.com> 
04/30/2007 11:13 PM

To
"CarlosOrtiz at bayviewfinancial.com" <CarlosOrtiz at bayviewfinancial.com>
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] Phone URI's






Here

<CiscoIPPhoneExecute>
<ExecuteItem Priority="1" URL="Key:Hold"\>
<CiscoIPPhoneExecute\>

Or if that doesn't work try this

<CiscoIPPhoneExecute> 
<ExecuteItem Priority="1" URL="Key:Soft2"\>
<CiscoIPPhoneExecute\>

If needed, replace the numeral '2' above with the softkey position where 
hold is located.  Starting from left, 1, 2, 3 or 4.

Justin

On 4/30/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com > wrote:

I have a person in house who wrote a script that allows you to initiate a 
call from an IP phone given it's IP , waits 6 secs and then calls the next 
number in a list.  This was done to test consecutive DID numbers.  He has 
been searching through the Cisco development materials to see if we can 
now initiate a call place the call on hold and then hang up when we would 
like.   We have been looking at the URI objects but can't find one for the 
Hold softkey.   Has anyone been able to accomplish this function? 

Carlos
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 


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From rratliff at cisco.com  Thu May  3 11:22:34 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 3 May 2007 11:22:34 -0400
Subject: [cisco-voip] Perflib error on Call Mgr
In-Reply-To: <OF043E7126.DD151121-ON852572D0.00536716-852572D0.0053F065@bftg.com>
References: <OF043E7126.DD151121-ON852572D0.00536716-852572D0.0053F065@bftg.com>
Message-ID: <C474ABED-0891-4CF3-AFFB-5F324F7004C8@cisco.com>

You aren't removing anything, just disabling that specific perfmon  
counter.  I've done it many times with various customers.

-Ryan

On May 3, 2007, at 11:16 AM, CarlosOrtiz at bayviewfinancial.com wrote:





Event Type: Error
Event Source: Perflib
Event Category: None
Event ID: 1008
Date: 1/25/2006
Time: 11:42:34 AM
User: N/A
Computer: SA-CCM2
Description:
The Open Procedure for service "DHCPServer" in DLL "C:\WINNT\system32 
\perfctrs.dll"failed.
Performance data for this service will not be available.
Status code  returned is data DWORD 0.
Data:
0000: 05 00 00 00




This error is a know bug on Call Mgr that is eating up log space.

http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K31823130


  Has anyone executed the proposed fix?  From reading I believe it  
just requires a registry edit to remove from th logs.  Has anyone  
done this already with no issues?

Carlos
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From VLoschiavo at data-corporation.com  Thu May  3 13:12:05 2007
From: VLoschiavo at data-corporation.com (Vince Loschiavo)
Date: Thu, 3 May 2007 13:12:05 -0400
Subject: [cisco-voip] Easy sftp server (windows 32bit) for CCM 5.x
	backup?
In-Reply-To: <4639F15B.5070507@dataways.gr>
References: <4639F15B.5070507@dataways.gr>
Message-ID: <CB4BC8C24EF6E149AED9FDB9B9AF57133924E6@MAIL02.hq.data-corporation.com>

I generally recommend: 
http://sshwindows.sourceforge.net/ 


____________________________________________________________
Vincent Loschiavo
Senior Consultant
Datacorp
18503 Pines Blvd
Suite 212
Pembroke Pines, FL, 33029

"Cisco Advanced Unified Communications Partner"
Cell Phone:        786-282-1164
Office Phone:    305-594-6933
____________________________________________________________ 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony
Kouloglou
Sent: Thursday, May 03, 2007 10:28 AM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] Easy sftp server (windows 32bit) for CCM 5.x
backup?

Does cisco propose any sftp server for backup of the CCM 5.x?
Can i use a second CCM (eg subscriber) for that?

Thanks
Anthony
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




From Peterson_Erik at aac.com  Thu May  3 12:36:25 2007
From: Peterson_Erik at aac.com (Peterson_Erik at aac.com)
Date: Thu, 3 May 2007 12:36:25 -0400
Subject: [cisco-voip] CM 4.2(3) SR1 500 Internal Server Error
Message-ID: <OF799C3F1E.C74A30B8-ON852572D0.005AD6A7-852572D0.005B3A3C@aac.com>

Anyone seen a 500 Internal Server Error on a 4.2(3) installation.  I saw 
one post in a NetPro forum that suggested it was a permissions error.  I 
saw this on a newly installed subscriber with OS 4.4a SR5, on a 7825-H2.

From bloemhard_johan at silverfalls.k12.or.us  Thu May  3 13:48:24 2007
From: bloemhard_johan at silverfalls.k12.or.us (Johan Bloemhard)
Date: Thu, 3 May 2007 10:48:24 -0700
Subject: [cisco-voip] Force IP Communicator to use internal microphone
Message-ID: <C763DC29A34B6B418AFBA81F75B2633AAA3561@msexch2.silverfalls.k12.or.us>

Anyone able to force IP Communicator to use the laptop internal
microphone?  I am out of the office and forgot a headhest... I set my
recording source to internal through "Recording Control" then when I
start the call, IP Communicator switches to External and won't let me
select Internal.

 

I know it's not the best practice for echo reasons but just wondered if
it was possible?

 

I have a SigmaTel Audio device on Latitude D820.

 

TIA

 

Johan Bloemhard 
___________________________________ 
Silver Falls School District 4J 
802 Schlador Street 
Silverton, OR 97381 
w: 503.873.6331 

 

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From ealeatherman at gmail.com  Thu May  3 14:19:02 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 3 May 2007 14:19:02 -0400
Subject: [cisco-voip] cisco emergency responder - missing phone(s)
Message-ID: <94a1afde0705031119v3e49256axfe2ed6d6bc50798d@mail.gmail.com>

Curious if anyone has seen this before....

We have CER v1.3 paired up with our callmanager cluster. Probably 100 or so
switches setup on it, 99.9% of the time everything works peachy. There are
however a handful of phones that for some reason do not get discovered on
their respective switches. The phone is listed in CER as unlocated, so I
know it is getting the phone info from callmanager, but the switchport that
we know it is plugged into on the switch in CER shows no phone. however
there are many other phones on the switch that are discovered, i'm not sure
what the difference is. The phone can talk CDP with the switch, I can see
the CDP info on the phone's webpage, with the correct interface etc.

I've tried forcing the locate switch-ports process on that switch, but its
not coming back with any different information, and I see no SNMP errors
about that switch in the app log.

any ideas?

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Thu May  3 14:50:06 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 3 May 2007 14:50:06 -0400
Subject: [cisco-voip] cisco emergency responder - missing phone(s)
In-Reply-To: <310F10740485D9469F0D8417950B252D027398EE@CTG-MSNEXC01.staff.berbee.com>
References: <94a1afde0705031119v3e49256axfe2ed6d6bc50798d@mail.gmail.com>
	<310F10740485D9469F0D8417950B252D027398EE@CTG-MSNEXC01.staff.berbee.com>
Message-ID: <94a1afde0705031150u2bd2e0e2saa13c942172fea07@mail.gmail.com>

I don't believe so, i'll check into that for sure.. wasn't even aware there
was one. Thanks Chris!

On 5/3/07, Clouse, Chris <chris.clouse at berbee.com> wrote:
>
>  Have you applied the MSDE update patch?  I've seen this fix that .1% of
> issues.
>
>
>
> Christopher Clouse, CCNP CCDP CCVP MCP Network+
>
> Berbee
>
> Network Voice Engineer
>
> 7145 Boone Avenue North, Suite 140
>
> Brooklyn Park, MN 55428
>
> *763.592.5887 Direct
>
> *763.535.7428 Fax
>
> *612.419.9751 Mobile
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Ed Leatherman
> *Sent:* Thursday, May 03, 2007 1:19 PM
> *To:* ciscovoip
> *Subject:* [cisco-voip] cisco emergency responder - missing phone(s)
>
>
>
> Curious if anyone has seen this before....
>
> We have CER v1.3 paired up with our callmanager cluster. Probably 100 or
> so switches setup on it, 99.9% of the time everything works peachy. There
> are however a handful of phones that for some reason do not get discovered
> on their respective switches. The phone is listed in CER as unlocated, so I
> know it is getting the phone info from callmanager, but the switchport that
> we know it is plugged into on the switch in CER shows no phone. however
> there are many other phones on the switch that are discovered, i'm not sure
> what the difference is. The phone can talk CDP with the switch, I can see
> the CDP info on the phone's webpage, with the correct interface etc.
>
> I've tried forcing the locate switch-ports process on that switch, but its
> not coming back with any different information, and I see no SNMP errors
> about that switch in the app log.
>
> any ideas?
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From bob at smalltime.com  Thu May  3 14:50:47 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Thu, 03 May 2007 13:50:47 -0500
Subject: [cisco-voip] Force IP Communicator to use internal microphone
In-Reply-To: <C763DC29A34B6B418AFBA81F75B2633AAA3561@msexch2.silverfalls.k12.or.us>
References: <C763DC29A34B6B418AFBA81F75B2633AAA3561@msexch2.silverfalls.k12.or.us>
Message-ID: <463A2F07.9080709@smalltime.com>

Johan Bloemhard wrote:
> Anyone able to force IP Communicator to use the laptop internal 
> microphone?  I am out of the office and forgot a headhest? I set my 
> recording source to internal through ?Recording Control? then when I 
> start the call, IP Communicator switches to External and won?t let me 
> select Internal.

Did you re-run the audio tuning wizard?  Are you _sure_ that your laptop 
has a microphone?  We found out the hard way that Latitude D410's _do 
not_ have a microphone.

From baryonyx5 at gmail.com  Thu May  3 15:23:43 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Thu, 3 May 2007 14:23:43 -0500
Subject: [cisco-voip] LDAP Directory Integration in Call Manager 5.1
In-Reply-To: <7aaaf7f0705021315l59ea9462v2c4d74cf3b9a5ddf@mail.gmail.com>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451E1C@CSEXCH01.charles-stanley.co.uk>
	<201EE253-4A57-4515-90EF-C8CCC67C83E0@cisco.com>
	<7aaaf7f0705021315l59ea9462v2c4d74cf3b9a5ddf@mail.gmail.com>
Message-ID: <a749d0620705031223l26e756f9h78e3b9e9687490bc@mail.gmail.com>

I think it's the Cisco DirSync traces you want.

On 5/2/07, Brian <bms314 at gmail.com> wrote:
>
> Which traces would you recommend running to verify the sync is doing
> something?
>
> Brian
>
> On 4/4/07, Ryan Ratliff < rratliff at cisco.com> wrote:
> >
> > Make sure you have activated the LDAP Synch service, then try the
> > synch again.  Other than that you can look at traces for it to make
> > sure it is doing something.  I ended up leaving mine overnight and my
> > lab AD has all of 2 end users.
> >
> > -Ryan
> >
> > On Apr 4, 2007, at 10:12 AM, Nick Kassel wrote:
> >
> > Thank you Ryan, I have added the correct format in and CM has accepted
> > it and updated the configuration, however I have tried to 'Perform Full
> > Sync Now' and it says update successful but how do I know if something
> > is happening, I have left it for a while but no users have been
> > imported.
> >
> > -----Original Message-----
> > From: Ryan Ratliff [mailto:rratliff at cisco.com]
> > Sent: 04 April 2007 13:50
> > To: Nick Kassel
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] LDAP Directory Integration in Call Manager 5.1
> >
> > The LDAP manager can just be user at domain , no need to use the full
> > dn=... format.  The user search base does need to be something like
> > cn=Users,dc=domain,dc=com.  My ldap manager is a domain admin but
> > since we don't write anything to AD I'm pretty sure all it needs is
> > read permission.
> >
> > Note that you will need to restart Tomcat on your subscribers after
> > enabling the ldap synch. The pub picks up the change automatically
> > (ie for CCMUser auth) but the subs don't for some reason.
> >
> > -Ryan
> >
> > On Apr 4, 2007, at 8:47 AM, Nick Kassel wrote:
> >
> > I am trying to test integrating CCM 5.1 with Active Directory in
> > vmware, has anyone done this in 5.1, It should be fairly straight
> > forward but I am unsure if the LDAP Manager distinguished name and
> > LDAP user search base I am entering are in the correct format. Also
> > what rights are needed for the AD user for this to take place.
> >
> >
> > If anyone could provide there configuration format I would be very
> > grateful.
> >
> >
> >
> >
> >
> >
> > ********************SAVE PAPER - THINK BEFORE YOU
> > PRINT**************************************************
> >
> > The information contained in this e-mail is strictly confidential,
> > some or all of which may be legally privileged.
> > Access to this e-mail by any person other than the recipient is
> > prohibited. If you have received this message in
> > error, any use, disclosure, copying, printing, distribution of,
> > replying to or any action taken or omitted to be
> > taken in reliance on this e-mail, is prohibited. Please advise the
> > sender immediately should this e-mail have been
> > incorrectly addressed or transmitted, and then delete the email and
> > any attachment sent with it from your computer.
> >
> > You are advised that urgent, time sensitive and confidential
> > communications should not be sent by e-mail. You
> > accept that any instructions are deemed to have been given at the
> > time the recipient(s) accesses them and that
> > delivery receipt does not constitute acknowledgement or receipt by
> > the intended recipient(s).
> >
> > You acknowledge that e-mails are not secure and you accept the risk
> > of malfunction, viruses, unauthorised
> > interference, mis-delivery or delay. Charles Stanley reserves the
> > right to monitor and/or record emails sent and
> > received via its network for any lawful business purpose in
> > accordance with applicable law and regulations.
> > ************************************************************************
> >
> >
> > *******************************
> >
> >
> > Charles Stanley & Co. Ltd
> > Registered Office: 25 Luke Street London EC2A 4AR
> >
> > Tel: 0207 739 8200 Fax: 0207 739 7798
> > Registered in England No. 1903304
> >
> > Charles Stanley Sutherlands and Charles Stanley Securities are
> > divisions of Charles Stanley & Co. Ltd
> >
> > Authorised and Regulated by the Financial Services Authority, Member
> > of the London Stock Exchange, The
> > International Capital Market Association and The London International
> > Financial Futures & Options Exchange.
> >
> > This footnote also confirms that this email message has been swept by
> > McAfee VirusScan and SurfControl Email
> > Filter software.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > ********************SAVE PAPER - THINK BEFORE YOU
> > PRINT**************************************************
> >
> > The information contained in this e-mail is strictly confidential,
> > some or all of which may be legally privileged.
> > Access to this e-mail by any person other than the recipient is
> > prohibited. If you have received this message in
> > error, any use, disclosure, copying, printing, distribution of,
> > replying to or any action taken or omitted to be
> > taken in reliance on this e-mail, is prohibited. Please advise the
> > sender immediately should this e-mail have been
> > incorrectly addressed or transmitted, and then delete the email and
> > any attachment sent with it from your computer.
> >
> > You are advised that urgent, time sensitive and confidential
> > communications should not be sent by e-mail. You
> > accept that any instructions are deemed to have been given at the
> > time the recipient(s) accesses them and that
> > delivery receipt does not constitute acknowledgement or receipt by
> > the intended recipient(s).
> >
> > You acknowledge that e-mails are not secure and you accept the risk
> > of malfunction, viruses, unauthorised
> > interference, mis-delivery or delay. Charles Stanley reserves the
> > right to monitor and/or record emails sent and
> > received via its network for any lawful business purpose in
> > accordance with applicable law and regulations.
> > ************************************************************************
> > *******************************
> >
> >
> > Charles Stanley & Co. Ltd
> > Registered Office: 25 Luke Street London EC2A 4AR
> >
> > Tel: 0207 739 8200 Fax: 0207 739 7798
> > Registered in England No. 1903304
> >
> > Charles Stanley Sutherlands and Charles Stanley Securities are
> > divisions of Charles Stanley & Co. Ltd
> >
> > Authorised and Regulated by the Financial Services Authority, Member
> > of the London Stock Exchange, The
> > International Capital Market Association and The London International
> > Financial Futures & Options Exchange.
> >
> > This footnote also confirms that this email message has been swept by
> > McAfee VirusScan and SurfControl Email
> > Filter software.
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
kris seraphine
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From baryonyx5 at gmail.com  Thu May  3 15:27:25 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Thu, 3 May 2007 14:27:25 -0500
Subject: [cisco-voip] speed dials on 7911s
In-Reply-To: <12b701c78d8e$76d98db0$47196883@cfs.uoguelph.ca>
References: <12b701c78d8e$76d98db0$47196883@cfs.uoguelph.ca>
Message-ID: <a749d0620705031227q499dc06cpea60240e3739996d@mail.gmail.com>

On a 7911 running load 8-2-1S pressing the blue bar does bring up the speed
dials list when the handset is on-hook.

On 5/3/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  On the 7912s, with the handset onhook, the blue bar is used to scroll
> through speed dials. If none are configured, the lines say 'not configured'
> or something like that.
>
> Can anyone confirm if the behaviour of the 7911s is the same?
>
> Thanks, Lelio
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
kris seraphine
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From jgrace at adcapnet.com  Thu May  3 16:05:47 2007
From: jgrace at adcapnet.com (James Grace)
Date: Thu, 3 May 2007 16:05:47 -0400
Subject: [cisco-voip] why do i need cmd " card type 0 0
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C648@mail.adcapnetsvc.local>

?
I added a VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 to my 2821 router.  and for some reason i needed to add the "card type 0 0" command to the router before i could see the t1 controllers.  This stump me be cause i never had to do this before.  Can someone explain to me why now i have to add this command
 
 
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
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From baryonyx5 at gmail.com  Thu May  3 17:19:38 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Thu, 3 May 2007 16:19:38 -0500
Subject: [cisco-voip] why do i need cmd " card type 0 0
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C648@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C648@mail.adcapnetsvc.local>
Message-ID: <a749d0620705031419x45c60291ib24bda7898ad1b3c@mail.gmail.com>

i believe that's how you set the card to be in T1 or E1 mode. The command is
card type t1 or e1 0 0 .

On 5/3/07, James Grace <jgrace at adcapnet.com> wrote:
>
> ? I added a VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 to my
> 2821 router.  and for some reason i needed to add the "card type 0 0"
> command to the router before i could see the t1 controllers.  This stump me
> be cause i never had to do this before.  Can someone explain to me why now i
> have to add this command
>
>
>
>  James Grace
>
> Systems Engineer
> Adcap Network Systems, Inc.
> 6865 Shiloh Road East, Suite 250
> Alpharetta, GA 30005
>
> office: 678-456-6986
> direct: 678-456-6761
> jgrace at adcapnet.com
> www.adcapnet.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
kris seraphine
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From baryonyx5 at gmail.com  Thu May  3 17:21:08 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Thu, 3 May 2007 16:21:08 -0500
Subject: [cisco-voip] speed dials on 7911s
In-Reply-To: <144201c78db9$6f424a30$47196883@cfs.uoguelph.ca>
References: <12b701c78d8e$76d98db0$47196883@cfs.uoguelph.ca>
	<a749d0620705031227q499dc06cpea60240e3739996d@mail.gmail.com>
	<144201c78db9$6f424a30$47196883@cfs.uoguelph.ca>
Message-ID: <a749d0620705031421v686b3670mf0200d9a2c51e1cd@mail.gmail.com>

The only difference I'm aware of is that the 7911 shows a big Cisco logo on
the display when idle.  Kind of a built-in idle URL.

On 5/3/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  cool. thanks. i'm guessing these phones are gonna be almost exactly the
> same.
>
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
> ----- Original Message -----
> *From:* Kris Seraphine <baryonyx5 at gmail.com>
> *To:* Lelio Fulgenzi <lelio at uoguelph.ca>
> *Cc:* cisco-voip at puck.nether.net
> *Sent:* Thursday, May 03, 2007 3:27 PM
> *Subject:* Re: [cisco-voip] speed dials on 7911s
>
> On a 7911 running load 8-2-1S pressing the blue bar does bring up the
> speed dials list when the handset is on-hook.
>
> On 5/3/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote:
> >
> >  On the 7912s, with the handset onhook, the blue bar is used to scroll
> > through speed dials. If none are configured, the lines say 'not configured'
> > or something like that.
> >
> > Can anyone confirm if the behaviour of the 7911s is the same?
> >
> > Thanks, Lelio
> >
> >
> > --------------------------------------------------------------------------------
> > Lelio Fulgenzi, B.A.
> > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > ...there's no such thing as a bad timbit...
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> kris seraphine
>
>


-- 
kris seraphine
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From holman at post.harvard.edu  Thu May  3 17:41:52 2007
From: holman at post.harvard.edu (Candace Holman)
Date: Thu, 3 May 2007 17:41:52 -0400
Subject: [cisco-voip] cisco emergency responder - missing phone(s)
Message-ID: <81746c9a0705031441y3069e184s6c3cc8307e311e2b@mail.gmail.com>

Ed,

You've probably already checked, but is there a different phone model or
firmware for the misbehaving devices?  How do the switchport configurations
compare?  Same vlan, same acl?

Candace


> From: "Ed Leatherman" <ealeatherman at gmail.com>
> To: ciscovoip <cisco-voip at puck.nether.net>
> Date: Thu, 3 May 2007 14:19:02 -0400
> Subject: [cisco-voip] cisco emergency responder - missing phone(s)
> Curious if anyone has seen this before....
>
> We have CER v1.3 paired up with our callmanager cluster. Probably 100 or
> so switches setup on it, 99.9% of the time everything works peachy. There
> are however a handful of phones that for some reason do not get discovered
> on their respective switches. The phone is listed in CER as unlocated, so I
> know it is getting the phone info from callmanager, but the switchport that
> we know it is plugged into on the switch in CER shows no phone. however
> there are many other phones on the switch that are discovered, i'm not sure
> what the difference is. The phone can talk CDP with the switch, I can see
> the CDP info on the phone's webpage, with the correct interface etc.
>
> I've tried forcing the locate switch-ports process on that switch, but its
> not coming back with any different information, and I see no SNMP errors
> about that switch in the app log.
>
> any ideas?
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>
>
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From bloemhard_johan at silverfalls.k12.or.us  Thu May  3 17:48:46 2007
From: bloemhard_johan at silverfalls.k12.or.us (Johan Bloemhard)
Date: Thu, 3 May 2007 14:48:46 -0700
Subject: [cisco-voip] Force IP Communicator to use internal microphone
In-Reply-To: <463A2F07.9080709@smalltime.com>
Message-ID: <C763DC29A34B6B418AFBA81F75B2633AAA369D@msexch2.silverfalls.k12.or.us>

Robert,

Yes, I did verify that it has a microphone.  Went into sound recorder
and was able to successfully record my voice.

Yes, I have run the tuning wizard multiple times and messed extensively
with the settings.

Johan Bloemhard 
___________________________________ 
Silver Falls School District 4J 
802 Schlador Street 
Silverton, OR 97381 
w: 503.873.6331 

-----Original Message-----
From: Robert Kulagowski [mailto:bob at smalltime.com] 
Sent: Thursday, May 03, 2007 11:51 AM
To: Johan Bloemhard
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Force IP Communicator to use internal
microphone

Johan Bloemhard wrote:
> Anyone able to force IP Communicator to use the laptop internal 
> microphone?  I am out of the office and forgot a headhest... I set my 
> recording source to internal through "Recording Control" then when I 
> start the call, IP Communicator switches to External and won't let me 
> select Internal.

Did you re-run the audio tuning wizard?  Are you _sure_ that your laptop

has a microphone?  We found out the hard way that Latitude D410's _do 
not_ have a microphone.


From asobihoudai at yahoo.com  Thu May  3 18:26:11 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 3 May 2007 15:26:11 -0700 (PDT)
Subject: [cisco-voip] Dialing from the directory
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C648@mail.adcapnetsvc.local>
Message-ID: <412711.49521.qm@web57203.mail.re3.yahoo.com>

I can't find the solution someone posted about dialing
from the corporate directory using CCM 4.1(3). I have
a customer that is complaining about having to edit
the number before dialing the number. How do I
automatically append the appropriate access digits to
the numbers?

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From lelio at uoguelph.ca  Thu May  3 18:32:14 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 3 May 2007 18:32:14 -0400
Subject: [cisco-voip] speed dials on 7911s
References: <12b701c78d8e$76d98db0$47196883@cfs.uoguelph.ca>
	<a749d0620705031227q499dc06cpea60240e3739996d@mail.gmail.com>
	<144201c78db9$6f424a30$47196883@cfs.uoguelph.ca>
	<a749d0620705031421v686b3670mf0200d9a2c51e1cd@mail.gmail.com>
Message-ID: <000e01c78dd2$e6c22440$c901a8c0@pdp11>

So did the 7912s - you can reload this image. I hope!
  ----- Original Message ----- 
  From: Kris Seraphine 
  To: Lelio Fulgenzi 
  Cc: ciscovoip 
  Sent: Thursday, May 03, 2007 5:21 PM
  Subject: Re: [cisco-voip] speed dials on 7911s


  The only difference I'm aware of is that the 7911 shows a big Cisco logo on the display when idle.  Kind of a built-in idle URL.  


  On 5/3/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
    cool. thanks. i'm guessing these phones are gonna be almost exactly the same.


    --------------------------------------------------------------------------------
    Lelio Fulgenzi, B.A.
    Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
    (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
    ...there's no such thing as a bad timbit...

      ----- Original Message ----- 
      From: Kris Seraphine 
      To: Lelio Fulgenzi 
      Cc: cisco-voip at puck.nether.net 
      Sent: Thursday, May 03, 2007 3:27 PM
      Subject: Re: [cisco-voip] speed dials on 7911s


      On a 7911 running load 8-2-1S pressing the blue bar does bring up the speed dials list when the handset is on-hook.  


      On 5/3/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote: 
        On the 7912s, with the handset onhook, the blue bar is used to scroll through speed dials. If none are configured, the lines say 'not configured' or something like that.

        Can anyone confirm if the behaviour of the 7911s is the same?

        Thanks, Lelio

        --------------------------------------------------------------------------------
        Lelio Fulgenzi, B.A.
        Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
        (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
        ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
        ...there's no such thing as a bad timbit...


        _______________________________________________
        cisco-voip mailing list
        cisco-voip at puck.nether.net
        https://puck.nether.net/mailman/listinfo/cisco-voip





      -- 
      kris seraphine 



  -- 
  kris seraphine 
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From Joe.Pollere at us.didata.com  Thu May  3 20:40:12 2007
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Thu, 3 May 2007 20:40:12 -0400
Subject: [cisco-voip] Unity VM message converation timestamp
Message-ID: <C65033A817B5734392E55CB9DF4461040ABB1906@USNAEXCH.na.didata.local>

 

The client has a site in Austrailia, Unity is in NJ. Time zone for those
subscribers is set for GMT +10:00 (Sydney). When a user retrieves a
message it says message left "tomorrow" at hh:mm. It actually says
tomorrow. The timestamp is correct and in reference to NJ it is actually
tomorrow. I've never seen this behavior before. 

 

Any Ideas why it is saying Tomorrow?

 

Unity 4.1.1 btw

 

Thanks,

 

Joe

 



-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From fwn at feasible.net  Fri May  4 03:37:42 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Fri, 4 May 2007 00:37:42 -0700
Subject: [cisco-voip] Unity VM message converation timestamp
In-Reply-To: <C65033A817B5734392E55CB9DF4461040ABB1906@USNAEXCH.na.didata.local>
Message-ID: <008201c78e1f$211b0de0$0afd000a@e12556>

Seen this yet?
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCsc81385
 
-Fred Nielsen

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Pollere (US)
Sent: Thursday, May 03, 2007 5:40 PM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] Unity VM message converation timestamp



 

The client has a site in Austrailia, Unity is in NJ. Time zone for those
subscribers is set for GMT +10:00 (Sydney). When a user retrieves a message
it says message left "tomorrow" at hh:mm. It actually says tomorrow. The
timestamp is correct and in reference to NJ it is actually tomorrow. I've
never seen this behavior before. 

 

Any Ideas why it is saying Tomorrow?

 

Unity 4.1.1 btw

 

Thanks,

 

Joe

 


  _____  

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee, you
are hereby notified that you have received this communication in error and
that any use or reproduction of this email or its contents is strictly
prohibited and may be unlawful. If you have received this communication in
error, please notify us immediately by replying to this message and deleting
it from your computer. Thank you. 

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From voip at mrga.ch  Fri May  4 03:39:40 2007
From: voip at mrga.ch (voip at mrga.ch)
Date: Fri, 04 May 2007 09:39:40 +0200
Subject: [cisco-voip] CM new Publisher Hardware
Message-ID: <1178264380.463ae33c90753@www.mail2web.ch>

Hello all

we plan to upgrade our server hardware. We now run our publisher on a HP DL380
G3. We plan to upgrade the hardware to a HP DL380 G5.
We are running a CM 3.3(4) with BARS, BAT/TAPS (Extended Services) and MLA.
What software do I have to install (and configure??) on the new server before I
can start the restore of the BARS backup??

Thanks for feedback
Reto




From akoul at dataways.gr  Fri May  4 04:05:23 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Fri, 04 May 2007 11:05:23 +0300
Subject: [cisco-voip] Call Waiting and tone for caller in 5.X
Message-ID: <463AE943.8000608@dataways.gr>

I have many requests for that:
is there a way for a caller to hear a different tone when he/she calls 
an extension with call waiting enabled?

If not, is there a plan by Cisco to cover this issue in the future?

Cheers,
Anthony


From ealeatherman at gmail.com  Fri May  4 08:12:40 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 4 May 2007 08:12:40 -0400
Subject: [cisco-voip] cisco emergency responder - missing phone(s)
In-Reply-To: <81746c9a0705031441y3069e184s6c3cc8307e311e2b@mail.gmail.com>
References: <81746c9a0705031441y3069e184s6c3cc8307e311e2b@mail.gmail.com>
Message-ID: <94a1afde0705040512s538c6ed8m3fa0f718d40f049f@mail.gmail.com>

Same model, firmware, and port configs. going to try the MSDE service pack
this weekend and a general reboot this weekend and see if it helps.

On 5/3/07, Candace Holman <holman at post.harvard.edu> wrote:
>
> Ed,
>
> You've probably already checked, but is there a different phone model or
> firmware for the misbehaving devices?  How do the switchport configurations
> compare?  Same vlan, same acl?
>
> Candace
>
>
> > From: "Ed Leatherman" <ealeatherman at gmail.com>
> > To: ciscovoip <cisco-voip at puck.nether.net>
> > Date: Thu, 3 May 2007 14:19:02 -0400
> > Subject: [cisco-voip] cisco emergency responder - missing phone(s)
> > Curious if anyone has seen this before....
> >
> > We have CER v1.3 paired up with our callmanager cluster. Probably 100 or
> > so switches setup on it, 99.9% of the time everything works peachy.
> > There are however a handful of phones that for some reason do not get
> > discovered on their respective switches. The phone is listed in CER as
> > unlocated, so I know it is getting the phone info from callmanager, but the
> > switchport that we know it is plugged into on the switch in CER shows no
> > phone. however there are many other phones on the switch that are
> > discovered, i'm not sure what the difference is. The phone can talk CDP with
> > the switch, I can see the CDP info on the phone's webpage, with the correct
> > interface etc.
> >
> > I've tried forcing the locate switch-ports process on that switch, but
> > its not coming back with any different information, and I see no SNMP errors
> > about that switch in the app log.
> >
> > any ideas?
> >
> > --
> > Ed Leatherman
> > Senior Voice Engineer
> > West Virginia University
> > Telecommunications and Network Operations
> >
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From jonvoip at gmail.com  Fri May  4 08:45:40 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 4 May 2007 07:45:40 -0500
Subject: [cisco-voip] 7914 with 7951
Message-ID: <5d093f9a0705040545p20241f90ob5f7a80838ae8e3d@mail.gmail.com>

Got a customer who connected 7914s to a few 7941s... apparently there is no
option on the 7941 phone page to add it. There is on the 7961...

Any workaround or solution for this?




Jonathan
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From Nick.Kassel at Charles-Stanley.co.uk  Fri May  4 08:51:34 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri, 4 May 2007 13:51:34 +0100
Subject: [cisco-voip] Unity 4.2 Install Question
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451EA3@CSEXCH01.charles-stanley.co.uk>

If I am installing Unity 4.2 on windows 2003 server and going to
integrate with AD and Exchange 2003 on the domain, do I have to install
the Exchange Deployment tools on the Unity server, I am having an issue
getting the files to run and I was wondering whether I could carry on
with the installation without them.

 

This is a lab system in VMware.

 

 

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From fwn at feasible.net  Fri May  4 09:06:14 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Fri, 4 May 2007 06:06:14 -0700
Subject: [cisco-voip] Unity 4.2 Install Question
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451EA3@CSEXCH01.charles-stanley.co.uk>
Message-ID: <009b01c78e4d$0543e4b0$0afd000a@e12556>

It is definitely necessary to install Exchange System Manager on the Unity
server prior to installing the Unity application.  ESM provides the
necessary MAPI libraries that Unity will leverage to communicate with
Exchange servers.
 
-Fred Nielsen

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Friday, May 04, 2007 5:52 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity 4.2 Install Question



If I am installing Unity 4.2 on windows 2003 server and going to integrate
with AD and Exchange 2003 on the domain, do I have to install the Exchange
Deployment tools on the Unity server, I am having an issue getting the files
to run and I was wondering whether I could carry on with the installation
without them.

 

This is a lab system in VMware.

 

 




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****************************************************************************
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Charles Stanley Sutherlands and Charles Stanley Securities are divisions of
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From rratliff at cisco.com  Fri May  4 09:11:24 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 4 May 2007 09:11:24 -0400
Subject: [cisco-voip] 7914 with 7951
In-Reply-To: <5d093f9a0705040545p20241f90ob5f7a80838ae8e3d@mail.gmail.com>
References: <5d093f9a0705040545p20241f90ob5f7a80838ae8e3d@mail.gmail.com>
Message-ID: <1961E43B-551F-49A0-8078-37FF7D2AA73A@cisco.com>

Looks like neither 4.1, 4.2, or 5.1 have the option to add a 7912 to  
a 7941 or 7940.

I believe the intent is that if you need 14 extra lines you won't be  
using a phone with only 2 to begin with.

-Ryan

On May 4, 2007, at 8:45 AM, Jonathan Charles wrote:

Got a customer who connected 7914s to a few 7941s... apparently there  
is no option on the 7941 phone page to add it. There is on the 7961...

Any workaround or solution for this?




Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jonvoip at gmail.com  Fri May  4 09:18:49 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 4 May 2007 08:18:49 -0500
Subject: [cisco-voip] Updating Lines via Bulk Admin in CCM5.1
Message-ID: <5d093f9a0705040618h7611e1d9q5037bf05eb1e68e3@mail.gmail.com>

Is there a way to do it?




Jonathan
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From Nick.Kassel at Charles-Stanley.co.uk  Fri May  4 09:19:26 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri, 4 May 2007 14:19:26 +0100
Subject: [cisco-voip] Unity 4.2 Install Question
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451EA4@CSEXCH01.charles-stanley.co.uk>

Thanks, unfortunately I can't get it to run on the Unity box, I run the
setup for Exchange then click on the Exchange Deployment Tools and I get
a blank screen. However If I run this from the same location on a
Windows XP workstation it works fine. Any idea why Windows 2003 server
may be blocking it.

 

________________________________

From: Fred Nielsen [mailto:fwn at feasible.net] 
Sent: 04 May 2007 14:06
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity 4.2 Install Question

 

It is definitely necessary to install Exchange System Manager on the
Unity server prior to installing the Unity application.  ESM provides
the necessary MAPI libraries that Unity will leverage to communicate
with Exchange servers.

 

-Fred Nielsen

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Friday, May 04, 2007 5:52 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity 4.2 Install Question

If I am installing Unity 4.2 on windows 2003 server and going to
integrate with AD and Exchange 2003 on the domain, do I have to install
the Exchange Deployment tools on the Unity server, I am having an issue
getting the files to run and I was wondering whether I could carry on
with the installation without them.

 

This is a lab system in VMware.

 

 

 


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************************************************************************
*******************************


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Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
of Charles Stanley & Co. Ltd  

Authorised and Regulated by the Financial Services Authority, Member of
the London Stock Exchange, The 
International Capital Market Association and The London International
Financial Futures & Options Exchange.

This footnote also confirms that this email message has been swept by
McAfee VirusScan and SurfControl Email 
Filter software. 

 

********************SAVE PAPER - THINK BEFORE YOU PRINT**************************************************

The information contained in this e-mail is strictly confidential, some or all of which may be legally privileged. 
Access to this e-mail by any person other than the recipient is prohibited. If you have received this message in
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*******************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions of Charles Stanley & Co. Ltd  

Authorised and Regulated by the Financial Services Authority, Member of the London Stock Exchange, The 
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From jonvoip at gmail.com  Fri May  4 09:24:41 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 4 May 2007 08:24:41 -0500
Subject: [cisco-voip] 7914 with 7951
In-Reply-To: <1961E43B-551F-49A0-8078-37FF7D2AA73A@cisco.com>
References: <5d093f9a0705040545p20241f90ob5f7a80838ae8e3d@mail.gmail.com>
	<1961E43B-551F-49A0-8078-37FF7D2AA73A@cisco.com>
Message-ID: <5d093f9a0705040624v3d8891abia0b721d2752d4559@mail.gmail.com>

While it makes sense, it is annoying... the customer didn't ship any 7961s
to this site.


Jonathan

On 5/4/07, Ryan Ratliff <rratliff at cisco.com> wrote:
>
> Looks like neither 4.1, 4.2, or 5.1 have the option to add a 7912 to
> a 7941 or 7940.
>
> I believe the intent is that if you need 14 extra lines you won't be
> using a phone with only 2 to begin with.
>
> -Ryan
>
> On May 4, 2007, at 8:45 AM, Jonathan Charles wrote:
>
> Got a customer who connected 7914s to a few 7941s... apparently there
> is no option on the 7941 phone page to add it. There is on the 7961...
>
> Any workaround or solution for this?
>
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From fwn at feasible.net  Fri May  4 09:33:14 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Fri, 4 May 2007 06:33:14 -0700
Subject: [cisco-voip] Unity 4.2 Install Question
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451EA4@CSEXCH01.charles-stanley.co.uk>
Message-ID: <00a901c78e50$cb1a3a10$0afd000a@e12556>

I've occasionally seen issues on Win2003 running through the autorun guided
setup screen that you are using, though my issues are somewhat different.
You might want to try shortcutting directly into Exchange setup by running
it from \SETUP\I386.  If this works, take care to select options to only
install management tools and not a full version of Exchange. 
 
-Fred Nielsen

  _____  

From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Friday, May 04, 2007 6:19 AM
To: Fred Nielsen; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity 4.2 Install Question



Thanks, unfortunately I can't get it to run on the Unity box, I run the
setup for Exchange then click on the Exchange Deployment Tools and I get a
blank screen. However If I run this from the same location on a Windows XP
workstation it works fine. Any idea why Windows 2003 server may be blocking
it.

 

  _____  

From: Fred Nielsen [mailto:fwn at feasible.net] 
Sent: 04 May 2007 14:06
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity 4.2 Install Question

 

It is definitely necessary to install Exchange System Manager on the Unity
server prior to installing the Unity application.  ESM provides the
necessary MAPI libraries that Unity will leverage to communicate with
Exchange servers.

 

-Fred Nielsen

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Friday, May 04, 2007 5:52 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity 4.2 Install Question

If I am installing Unity 4.2 on windows 2003 server and going to integrate
with AD and Exchange 2003 on the domain, do I have to install the Exchange
Deployment tools on the Unity server, I am having an issue getting the files
to run and I was wondering whether I could carry on with the installation
without them.

 

This is a lab system in VMware.

 

 

 



********************SAVE PAPER - THINK BEFORE YOU
PRINT**************************************************

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accept that any instructions are deemed to have been given at the time the
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intended recipient(s). 

You acknowledge that e-mails are not secure and you accept the risk of
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monitor and/or record emails sent and
received via its network for any lawful business purpose in accordance with
applicable law and regulations.
****************************************************************************
***************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions of
Charles Stanley & Co. Ltd  

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, The 
International Capital Market Association and The London International
Financial Futures & Options Exchange.

This footnote also confirms that this email message has been swept by McAfee
VirusScan and SurfControl Email 
Filter software. 

 




********************SAVE PAPER - THINK BEFORE YOU
PRINT**************************************************

The information contained in this e-mail is strictly confidential, some or
all of which may be legally privileged. 
Access to this e-mail by any person other than the recipient is prohibited.
If you have received this message in
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You are advised that urgent, time sensitive and confidential communications
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accept that any instructions are deemed to have been given at the time the
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delivery receipt does not constitute acknowledgement or receipt by the
intended recipient(s). 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised 
interference, mis-delivery or delay. Charles Stanley reserves the right to
monitor and/or record emails sent and
received via its network for any lawful business purpose in accordance with
applicable law and regulations.
****************************************************************************
***************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions of
Charles Stanley & Co. Ltd  

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, The 
International Capital Market Association and The London International
Financial Futures & Options Exchange.

This footnote also confirms that this email message has been swept by McAfee
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From fwn at feasible.net  Fri May  4 09:35:05 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Fri, 4 May 2007 06:35:05 -0700
Subject: [cisco-voip] 7914 with 7951
In-Reply-To: <5d093f9a0705040624v3d8891abia0b721d2752d4559@mail.gmail.com>
Message-ID: <00b401c78e51$0d13e010$0afd000a@e12556>

7914 data sheet specifically says 796x/797x phones only.
 
-Fred Nielsen
 
  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Friday, May 04, 2007 6:25 AM
To: Ryan Ratliff
Cc: ciscovoip
Subject: Re: [cisco-voip] 7914 with 7951


While it makes sense, it is annoying... the customer didn't ship any 7961s
to this site.


Jonathan


On 5/4/07, Ryan Ratliff < rratliff at cisco.com <mailto:rratliff at cisco.com> >
wrote: 

Looks like neither 4.1, 4.2, or 5.1 have the option to add a 7912 to 
a 7941 or 7940.

I believe the intent is that if you need 14 extra lines you won't be
using a phone with only 2 to begin with.

-Ryan

On May 4, 2007, at 8:45 AM, Jonathan Charles wrote:

Got a customer who connected 7914s to a few 7941s... apparently there
is no option on the 7941 phone page to add it. There is on the 7961...

Any workaround or solution for this?




Jonathan
_______________________________________________ 
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 



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From fwn at feasible.net  Fri May  4 09:41:56 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Fri, 4 May 2007 06:41:56 -0700
Subject: [cisco-voip] Updating Lines via Bulk Admin in CCM5.1
In-Reply-To: <5d093f9a0705040618h7611e1d9q5037bf05eb1e68e3@mail.gmail.com>
Message-ID: <00d101c78e52$02719890$0afd000a@e12556>

Yup, did you look for it?  You should find it in roughly the same place it
has always been (Phones\Add-Update Lines).  Works really well.
 
-Fred Nielsen


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Friday, May 04, 2007 6:19 AM
To: ciscovoip
Subject: [cisco-voip] Updating Lines via Bulk Admin in CCM5.1


Is there a way to do it?




Jonathan

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From jonvoip at gmail.com  Fri May  4 09:53:00 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 4 May 2007 08:53:00 -0500
Subject: [cisco-voip] Updating Lines via Bulk Admin in CCM5.1
In-Reply-To: <00d101c78e52$02719890$0afd000a@e12556>
References: <5d093f9a0705040618h7611e1d9q5037bf05eb1e68e3@mail.gmail.com>
	<00d101c78e52$02719890$0afd000a@e12556>
Message-ID: <5d093f9a0705040653p38847526pfdfb9b41f21a1a4a@mail.gmail.com>

Working on no sleep.



Jonathan

On 5/4/07, Fred Nielsen <fwn at feasible.net> wrote:
>
>  Yup, did you look for it?  You should find it in roughly the same place
> it has always been (Phones\Add-Update Lines).  Works really well.
>
> -Fred Nielsen
>
>  ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Jonathan Charles
> *Sent:* Friday, May 04, 2007 6:19 AM
> *To:* ciscovoip
> *Subject:* [cisco-voip] Updating Lines via Bulk Admin in CCM5.1
>
> Is there a way to do it?
>
>
>
>
> Jonathan
>
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From erik.stillman at shearman.com  Fri May  4 09:59:36 2007
From: erik.stillman at shearman.com (Erik Stillman)
Date: Fri, 4 May 2007 09:59:36 -0400
Subject: [cisco-voip] Dialing from the directory
In-Reply-To: <mailman.12520.1178264299.1489.cisco-voip@puck.nether.net>
Message-ID: <OF5FF88751.CFFDA1E2-ON852572D1.004C9037-852572D1.004CDFB3@shearman.com>

Hi,

Sounds like there are outside numbers in the corporate directory. Are these
for other offices?

Certainly with AD, anything can be populated in the appropriate fields. By
default, the Corp Directory looks at the Telephone Number field.

Erik

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by the recipient or any other person for the purpose of avoiding United
States federal tax penalties that may be imposed on the recipient or
such other person. In addition, if any United States federal tax
advice contained in this communication is used or referred to in
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or other entity, investment plan, concept, structure or arrangement
(which should be assumed to be the case by a recipient or other
person who is not our client with respect to the subject matter of the
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to support the promotion or marketing of the transactions or matters
addressed by the advice and (ii) the recipient or other person should
seek advice based on the recipient or other person's particular
circumstances from an independent tax advisor. For further information,
please go to http://www.shearman.com/disclaimer/tax_disclosure.html

***********************************************************************

This transmittal and/or attachments may be a confidential
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*************************************************************************


From erik.stillman at shearman.com  Fri May  4 10:17:04 2007
From: erik.stillman at shearman.com (Erik Stillman)
Date: Fri, 4 May 2007 10:17:04 -0400
Subject: [cisco-voip] Perflib error on Call Mgr
In-Reply-To: <mailman.2.1178208005.65090.cisco-voip@puck.nether.net>
Message-ID: <OF2E7DAC7C.BC09DA53-ON852572D1.004DE417-852572D1.004E7940@shearman.com>

I've seen this when the DHCP service is added under  Add/Remove Programs >
Windows Components > Networking Services. Try removing it .... may do the
same thing as the registry entry.

Event Type: Error
Event Source: Perflib
Event Category: None
Event ID: 1008
Date: 1/25/2006
Time: 11:42:34 AM
User: N/A
Computer: SA-CCM2
Description:
The Open Procedure for service "DHCPServer" in DLL "C:\WINNT\system32
\perfctrs.dll"failed.
Performance data for this service will not be available.
Status code  returned is data DWORD 0.
Data:
0000: 05 00 00 00



-Erik

***********************************************************************

This communication is not intended to be used, and cannot be used,
by the recipient or any other person for the purpose of avoiding United
States federal tax penalties that may be imposed on the recipient or
such other person. In addition, if any United States federal tax
advice contained in this communication is used or referred to in
promoting, marketing or recommending any corporation, partnership
or other entity, investment plan, concept, structure or arrangement
(which should be assumed to be the case by a recipient or other
person who is not our client with respect to the subject matter of the
communication), then (i) such tax advice should be construed as written
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addressed by the advice and (ii) the recipient or other person should
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please go to http://www.shearman.com/disclaimer/tax_disclosure.html

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Shearman & Sterling LLP is a limited liability partnership organized in
the United States under the laws of the State of Delaware, which laws
limit the personal liability of partners.

*************************************************************************


From akoul at dataways.gr  Fri May  4 10:23:34 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Fri, 04 May 2007 17:23:34 +0300
Subject: [cisco-voip] Cannot transfer an off net call to another codec
 through an h323 gw
Message-ID: <463B41E6.3040306@dataways.gr>

Hi,
i have problems in transfering an incoming call established with g711 
codec, to another device that uses g729.
So, there is a CCM 5.X with extensions 1XXX and 2XXX for example.

1.VGW (H323)
2. ExtA 1000
   ExtB 1001

3  ExtC 2000

A, B and voicegw are in the same devicepool and use g711.
C is in another device pool that use g729 for calls to the other device 
pool.

GW has 2 different voip dial-peers:
 one with g711 to connect with extensions A, B, and
 one with g729 to connect with C
and one pots dial-peer to call outside to the TELCO numbers

Incoming calls to vgw from the telco, work fine either to extensions A,B 
or C directly (depending on the dial-peer, it uses the appropriate codec).
When A, calls B directly, B can tranfer the call to C
When A, receives a call from the Telco, cannot tranfer it to C: the call 
drops and busy tone is played to both ends.

Does, h323 needs something special for this config?

thanks
anthony




From Roconnell at unislumin.com  Fri May  4 10:29:46 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Fri, 4 May 2007 10:29:46 -0400
Subject: [cisco-voip] Unity Voicemail CallManager applet
In-Reply-To: <18082B35CF0EBD4C8D265874508A615C05BCECBB@hermes.unislumin.com>
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
	<18082B35CF0EBD4C8D265874508A615C05BCECBB@hermes.unislumin.com>
Message-ID: <18082B35CF0EBD4C8D265874508A615C05BCF056@hermes.unislumin.com>


Nobody has any thoughts on this? Nobody has set this up?


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan O'Connell
Sent: Wednesday, May 02, 2007 9:10 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Voicemail CallManager applet

Hello all,

Does anyone know if that applet plugin for CallManager that allows you
to add mailbox's right from the CCM GUI is supported in CM 5.x? The
following URL has mention on how to set it up "see step 19" 

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/sys_a
d/5_0_4/ccmsys/a06unity.htm#wp1024205

My thoughts are that this is a typo and that we can't use
voicemailbox.asp on Callmanager 5 since it's an appliance?

Please correct me if I'm wrong

Ryan
Attention:

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_______________________________________________
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From Uni.Lee at mms.gov  Fri May  4 10:43:42 2007
From: Uni.Lee at mms.gov (Lee, Uni Y)
Date: Fri, 4 May 2007 10:43:42 -0400
Subject: [cisco-voip] PowerChute install on VoIP servers
Message-ID: <129D9C95B82B2B47A19A6DD244AB83690217E6DF@IMSHEXPRI03.service.agency.mms.pri>

We are using APC as UPS for our VoIP servers. Servers are all windows
2000 platform. Did anyone install the APC Powerchute software on VoIP
servers to shutdown when APC goes to battery mode? Did anyone had issue
with it?

Thank in advance.


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From akoul at dataways.gr  Fri May  4 10:53:17 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Fri, 04 May 2007 17:53:17 +0300
Subject: [cisco-voip] Cannot transfer an off net call to another codec
 through an h323 gw
In-Reply-To: <310F10740485D9469F0D8417950B252D02739D0F@CTG-MSNEXC01.staff.berbee.com>
References: <463B41E6.3040306@dataways.gr>
	<310F10740485D9469F0D8417950B252D02739D0F@CTG-MSNEXC01.staff.berbee.com>
Message-ID: <463B48DD.6010207@dataways.gr>

Here it is

all 9xx extensions and voice gateway are in a region that uses g711.

only 964 is on a region that uses g729.


!
resource policy
!
network-clock-participate wic 0
!
!
ip cef
!
!
no ip domain lookup
!
isdn switch-type basic-net3
!
voice-card 0
 no dspfarm
!
!
voice call disc-pi-off
voice rtp send-recv
!
voice service voip
!
!
!
!
!        
!
!
!
!
!
!
!
!
voice translation-rule 1
 rule 1 /^51/ /951/
 rule 2 /^52/ /952/
 rule 3 /^53/ /290/
 rule 4 /^54/ /954/
 rule 5 /^55/ /955/
 rule 6 /^56/ /956/
 rule 7 /^57/ /957/
 rule 8 /^61/ /961/
 rule 9 /^63/ /963/
 rule 10 /^64/ /964/
 rule 15 /.*/ /290/
!
!
voice translation-profile TP01
 translate called 1
!
!
!
!

!
!
!
!
!
interface FastEthernet0/0
 no ip address
 no ip route-cache cef
 no ip route-cache
 duplex full
 speed 100
 no mop enabled
!
interface FastEthernet0/0.1
 encapsulation dot1Q 1 native
 ip address 213.140.132.21 255.255.255.192
 no ip route-cache
!        
interface FastEthernet0/0.100
 encapsulation dot1Q 100
 ip address 192.168.2.254 255.255.255.0
 no ip route-cache
 h323-gateway voip interface
 h323-gateway voip bind srcaddr 192.168.2.254
!
interface FastEthernet0/1
 ip address 192.168.101.1 255.255.255.0
 duplex full
 speed 100
!
interface BRI0/0/0
 no ip address
 isdn switch-type basic-net3
 isdn overlap-receiving
 isdn point-to-point-setup
 isdn incoming-voice voice
 isdn static-tei 0
!
interface BRI0/0/1
 no ip address
 isdn switch-type basic-net3
 isdn overlap-receiving
 isdn point-to-point-setup
 isdn incoming-voice voice
 isdn static-tei 0
!
ip route 0.0.0.0 0.0.0.0 213.140.132.14
!
!
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:
no ip nat create flow-entries
!
kron policy-list testpl
 cli show runnin
!
dialer-list 1 protocol ip permit
snmp-server community public RO
route-map test permit 10
!
!
!
!        
control-plane
!
!
!
voice-port 0/0/0
 translation-profile incoming TP01
 compand-type a-law
 bearer-cap Speech
!
voice-port 0/0/1
 translation-profile incoming TP01
 compand-type a-law
 bearer-cap Speech
!
voice-port 0/1/0
 cptone GR
 connection plar 953
 caller-id enable
!
voice-port 0/1/1
!
!
!
!
!
dial-peer voice 1 pots
 destination-pattern 0T
 direct-inward-dial
 port 0/0/0
!
dial-peer voice 2 pots
 preference 1
 destination-pattern 0T
 direct-inward-dial
 port 0/0/1
!
dial-peer voice 3 pots
 destination-pattern 069T
 port 0/1/0
 prefix 69
!
dial-peer voice 10 voip
 destination-pattern 9..
 session target ipv4:192.168.2.5
 dtmf-relay h245-alphanumeric
 codec g711alaw
 fax rate disable
 fax protocol pass-through g711alaw
 no vad
!
dial-peer voice 11 voip
 destination-pattern 2..
 session target ipv4:192.168.2.5
 dtmf-relay h245-alphanumeric
 codec g711alaw
 fax rate disable
 fax protocol pass-through g711alaw
 no vad
!
dial-peer voice 4 pots
 preference 1
 destination-pattern 069T
 port 0/0/0
 prefix 178969
!
dial-peer voice 5 pots
 preference 1
 destination-pattern 069T
 port 0/0/1
 prefix 178969
!
dial-peer voice 12 voip
 destination-pattern 97.
 session target ipv4:10.0.0.109
 dtmf-relay h245-alphanumeric
!
dial-peer voice 22 voip
 destination-pattern 14..
 session target ipv4:10.200.4.5
 codec g711alaw
!
dial-peer voice 964 voip
 destination-pattern 964
 session target ipv4:192.168.2.5
 dtmf-relay h245-alphanumeric
!
!
no dial-peer outbound status-check pots
!
!
call-manager-fallback
 max-conferences 8 gain -6
 transfer-system full-consult
 ip source-address 192.168.2.254 port 2000
 max-ephones 20
 max-dn 40
 no huntstop
 pickup 999
 alias 1 121 to 952
 alias 2 122 to 955
 alias 3 123 to 956
 alias 4 124 to 954
 alias 5 125 to 964
 alias 6 126 to 957
 alias 7 127 to 951
 alias 8 128 to 961
 alias 9 110 to 953
 alias 10 999 to 956
 alias 11 999 to 964
!
!
line con 0
line aux 0
line vty 0 4
 

Clouse, Chris wrote:
> Can you send me a copy of your config?
>
> Christopher Clouse, CCNP CCDP CCVP MCP Network+
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony
> Kouloglou
> Sent: Friday, May 04, 2007 9:24 AM
> To: cisco-voip at puck-nether.net
> Subject: [cisco-voip] Cannot transfer an off net call to another codec
> through an h323 gw
>
> Hi,
> i have problems in transfering an incoming call established with g711 
> codec, to another device that uses g729.
> So, there is a CCM 5.X with extensions 1XXX and 2XXX for example.
>
> 1.VGW (H323)
> 2. ExtA 1000
>    ExtB 1001
>
> 3  ExtC 2000
>
> A, B and voicegw are in the same devicepool and use g711.
> C is in another device pool that use g729 for calls to the other device 
> pool.
>
> GW has 2 different voip dial-peers:
>  one with g711 to connect with extensions A, B, and
>  one with g729 to connect with C
> and one pots dial-peer to call outside to the TELCO numbers
>
> Incoming calls to vgw from the telco, work fine either to extensions A,B
>
> or C directly (depending on the dial-peer, it uses the appropriate
> codec).
> When A, calls B directly, B can tranfer the call to C
> When A, receives a call from the Telco, cannot tranfer it to C: the call
>
> drops and busy tone is played to both ends.
>
> Does, h323 needs something special for this config?
>
> thanks
> anthony
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>   

From asobihoudai at yahoo.com  Fri May  4 11:14:23 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 4 May 2007 08:14:23 -0700 (PDT)
Subject: [cisco-voip] Dialing from the directory
In-Reply-To: <OF5FF88751.CFFDA1E2-ON852572D1.004C9037-852572D1.004CDFB3@shearman.com>
Message-ID: <913934.35121.qm@web57208.mail.re3.yahoo.com>

Erik,

It's not that they're outside lines. It is just that
the customer requires 10-digit dialing from office to
office and 9 is the access key. So I'll have
900-555-1212 as a number listed in the directory,
users will try to dial that number directly from the
listing without prepending a "91" and of course their
call fails. I'm trying to figure out a way around
this.

Cheers,
Paul


 
____________________________________________________________________________________
Now that's room service!  Choose from over 150,000 hotels
in 45,000 destinations on Yahoo! Travel to find your fit.
http://farechase.yahoo.com/promo-generic-14795097

From asobihoudai at yahoo.com  Fri May  4 11:17:42 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 4 May 2007 08:17:42 -0700 (PDT)
Subject: [cisco-voip] 7914 with 7951
In-Reply-To: <5d093f9a0705040545p20241f90ob5f7a80838ae8e3d@mail.gmail.com>
Message-ID: <239716.77683.qm@web57203.mail.re3.yahoo.com>

Jonathan,

My understanding is that 7941s are not compatible with
7914 expansion modules. You either use a 7961 with one
or two expansion modules. I swore that I saw somewhere
that 7941s were able to be used with 7914s but after
further reading, I didn't see that to be the case.

Paul


--- Jonathan Charles <jonvoip at gmail.com> wrote:

> Got a customer who connected 7914s to a few 7941s...
> apparently there is no
> option on the 7941 phone page to add it. There is on
> the 7961...
> 
> Any workaround or solution for this?
> 
> 
> 
> 
> Jonathan
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From akoul at dataways.gr  Fri May  4 12:01:09 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Fri, 04 May 2007 19:01:09 +0300
Subject: [cisco-voip] Cannot transfer an off net call to another codec
 through an h323 gw
In-Reply-To: <310F10740485D9469F0D8417950B252D0276BF2E@CTG-MSNEXC01.staff.berbee.com>
References: <463B41E6.3040306@dataways.gr>
	<310F10740485D9469F0D8417950B252D02739D0F@CTG-MSNEXC01.staff.berbee.com>
	<463B48DD.6010207@dataways.gr>
	<310F10740485D9469F0D8417950B252D0276BF2E@CTG-MSNEXC01.staff.berbee.com>
Message-ID: <463B58C5.6000403@dataways.gr>

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From bob at smalltime.com  Fri May  4 12:16:21 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Fri, 04 May 2007 11:16:21 -0500
Subject: [cisco-voip] Forwarding a call to a home phone but _with_ Unity VM
 if it's not answered
Message-ID: <463B5C55.3050303@smalltime.com>

So I have a number of users that have phones at home.  We'd like to 
setup their DIDs so that an incoming call gets sent to their home phone. 
  If they don't answer their phone at home, it should end up in Unity.

Right now it's kind of a kluge, and doesn't do what the users want.

DID is setup to CFA to VM.
Their VM account is set to a blank greeting, and after the greeting the 
call gets sent to a Call Handler.  _That_ call handler is owned by the 
user, and has their personal message on it.  The call handler has a Call 
Transfer back to CallManager to a dummy number that's on a CTI port. 
The dummy number is CFA to the user's home number.

The transfer on the call handler is set as supervised, 5 rings.

So, all that, and here's what happens:
Incoming call goes to Unity, Unity sends it to the handler, handler 
makes the call out.  This part is all OK.  The issue is the supervised 
transfer.  The caller doesn't hear any call progress notification, and 
if I turn off the "Wait while I transfer your call", then it may be 15 
seconds of silence before it comes back to Unity and you hear the 
personal greeting.

If I release to switch, the call never comes back to Unity.

I tried setting up an audio source using a wav file of the standard call 
progress "ring", but it doesn't seem like the call is really on hold, so 
the caller still gets silence.

Unity 4.2(1), CM 4.1.3sr2



From aman.chugh at gmail.com  Fri May  4 13:06:26 2007
From: aman.chugh at gmail.com (Aman Chugh)
Date: Fri, 4 May 2007 22:36:26 +0530
Subject: [cisco-voip] FXS port getting stuck
Message-ID: <d4c697ad0705041006s8ccb571n263a26306f2ff299@mail.gmail.com>

Group,

I have a fxs port on a router which seems to be acting up, I have 2851
router with 12.3.11T with a vic4fxs card, The Issue seems to be with only
one port which gets stuck after it gets a call and then remains off hook and
I have to do a shut no shut on the port to bring it to normal , Sometimes it
takes a lot of time with that as well, Am I looking at bad port and would
need to rma the card , any suggestions

Router is an MGCP g/w on cm.
CM 4.1.3

TIA
Aman
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From CarlosOrtiz at bayviewfinancial.com  Fri May  4 13:26:35 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 4 May 2007 13:26:35 -0400
Subject: [cisco-voip] IPCC Express BARS Restore
Message-ID: <OFACFA1B1F.15647294-ON852572D1.005F7A11-852572D1.005FD188@bftg.com>

Can someone confirm that I need to use the "Restoring Cisco CRS" section 
of the install guide to restore to a different server (Fresh OS Install)? 
The procedure calls for you to 

1)start the installation on the IPCC server, Get to the repair option 
menu, choose restore
2) Go to the "BARS Server" (Publisher I assume) and start the restore 
procedure

follow the rest of the procedure to step 22.

Anyone done this before?

Carlos 
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From rratliff at cisco.com  Fri May  4 13:42:36 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 4 May 2007 13:42:36 -0400
Subject: [cisco-voip] CM new Publisher Hardware
In-Reply-To: <1178264380.463ae33c90753@www.mail2web.ch>
References: <1178264380.463ae33c90753@www.mail2web.ch>
Message-ID: <CBE60DD7-5945-49A4-A36B-48D207AB9EC4@cisco.com>

http://www.cisco.com/univercd/cc/td/doc/product/voice/backup/bars40/ 
ag-4012/restore1.htm#wp1031115

-Ryan

On May 4, 2007, at 3:39 AM, voip at mrga.ch wrote:

Hello all

we plan to upgrade our server hardware. We now run our publisher on a  
HP DL380
G3. We plan to upgrade the hardware to a HP DL380 G5.
We are running a CM 3.3(4) with BARS, BAT/TAPS (Extended Services)  
and MLA.
What software do I have to install (and configure??) on the new  
server before I
can start the restore of the BARS backup??

Thanks for feedback
Reto



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From asobihoudai at yahoo.com  Fri May  4 13:57:09 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 4 May 2007 10:57:09 -0700 (PDT)
Subject: [cisco-voip] Dialing from the directory
In-Reply-To: <OF26F56AA8.83F9719D-ON852572D1.00567B47-852572D1.005680BA@shearman.com>
Message-ID: <182381.26302.qm@web57203.mail.re3.yahoo.com>

Erik,

Yes, I've thought about a translation pattern. Someone
posted some sort of solution to this issue a while
back but I can't find it in the archives for some
silly reason. The customer has five digit intraoffice
dialing and ten-digit dialing to other remote offices.
All DNs are, for the most part, ten digits with the
exception to local paging, etc. How would I make
people able to search the directory number and dial
from the directory when it displays a ten-digit DN? I
can't automatically append a "91" to all ten-digit
numbers.


Paul




--- Erik Stillman <erik.stillman at shearman.com> wrote:

> What about translation patterns?
> 
> 
> 
> 
>                                                     
>                       
>   Paul Choi                                         
>                       
>   <asobihoudai at yahoo.com>                           
>                       
>                                                     
>                    To 
>   05/04/2007 11:14 AM                  Erik Stillman
>                       
>                                       
> <erik.stillman at shearman.com>,       
>                                       
> cisco-voip at puck.nether.net          
>                                                     
>                    cc 
>                                                     
>                       
> 
> Erik,
> 
> It's not that they're outside lines. It is just that
> the customer requires 10-digit dialing from office
> to
> office and 9 is the access key. So I'll have
> 900-555-1212 as a number listed in the directory,
> users will try to dial that number directly from the
> listing without prepending a "91" and of course
> their
> call fails. I'm trying to figure out a way around
> this.
> 
> Cheers,
> Paul
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
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From CarlosOrtiz at bayviewfinancial.com  Fri May  4 13:59:10 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 4 May 2007 13:59:10 -0400
Subject: [cisco-voip] Perflib error on Call Mgr
In-Reply-To: <OF2E7DAC7C.BC09DA53-ON852572D1.004DE417-852572D1.004E7940@shearman.com>
Message-ID: <OFA5D35E40.2F7F20F3-ON852572D1.0062A918-852572D1.0062CD2F@bftg.com>

This was a new install so there was no adding or removing.  The only thing 
I worry about is that Cisco didn't recommend that as a fix.  Seems like it 
would be a viable option I suppose but not mentioned in there bug notes.



Erik Stillman <erik.stillman at shearman.com> 
Sent by: cisco-voip-bounces at puck.nether.net
05/04/2007 10:17 AM

To
cisco-voip at puck.nether.net
cc

Subject
Re: [cisco-voip] Perflib error on Call Mgr






I've seen this when the DHCP service is added under  Add/Remove Programs >
Windows Components > Networking Services. Try removing it .... may do the
same thing as the registry entry.

Event Type: Error
Event Source: Perflib
Event Category: None
Event ID: 1008
Date: 1/25/2006
Time: 11:42:34 AM
User: N/A
Computer: SA-CCM2
Description:
The Open Procedure for service "DHCPServer" in DLL "C:\WINNT\system32
\perfctrs.dll"failed.
Performance data for this service will not be available.
Status code  returned is data DWORD 0.
Data:
0000: 05 00 00 00



-Erik

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From asobihoudai at yahoo.com  Fri May  4 13:59:05 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 4 May 2007 10:59:05 -0700 (PDT)
Subject: [cisco-voip] FXS port getting stuck
In-Reply-To: <d4c697ad0705041006s8ccb571n263a26306f2ff299@mail.gmail.com>
Message-ID: <306271.68122.qm@web57215.mail.re3.yahoo.com>

Have you narrowed the issue down to the card itself?
Is the device connected to that port terminating the
connection properly or not? Have you tried different
signaling schemes?

--- Aman Chugh <aman.chugh at gmail.com> wrote:

> Group,
> 
> I have a fxs port on a router which seems to be
> acting up, I have 2851
> router with 12.3.11T with a vic4fxs card, The Issue
> seems to be with only
> one port which gets stuck after it gets a call and
> then remains off hook and
> I have to do a shut no shut on the port to bring it
> to normal , Sometimes it
> takes a lot of time with that as well, Am I looking
> at bad port and would
> need to rma the card , any suggestions
> 
> Router is an MGCP g/w on cm.
> CM 4.1.3
> 
> TIA
> Aman
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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From fwn at feasible.net  Fri May  4 14:06:16 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Fri, 4 May 2007 11:06:16 -0700
Subject: [cisco-voip] Forwarding a call to a home phone but _with_ Unity
	VM if it's not answered
In-Reply-To: <463B5C55.3050303@smalltime.com>
Message-ID: <00ec01c78e76$ef9e90e0$0afd000a@e12556>

This may not be what you are looking for.. But mobility manager (or Unified
Mobility now I think it is called) does what you are needing quite nicely. 

Also, I do seem to remember that Unity does in fact place the call on hold
(from CM's perspective) while doing a transfer (though I'm not 100% clear on
whether this is definitely true for a *supervised* transfer), so you may
want to look into the MoH setup further.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert Kulagowski
Sent: Friday, May 04, 2007 9:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Forwarding a call to a home phone but _with_ Unity VM
if it's not answered

So I have a number of users that have phones at home.  We'd like to setup
their DIDs so that an incoming call gets sent to their home phone. 
  If they don't answer their phone at home, it should end up in Unity.

Right now it's kind of a kluge, and doesn't do what the users want.

DID is setup to CFA to VM.
Their VM account is set to a blank greeting, and after the greeting the call
gets sent to a Call Handler.  _That_ call handler is owned by the user, and
has their personal message on it.  The call handler has a Call Transfer back
to CallManager to a dummy number that's on a CTI port. 
The dummy number is CFA to the user's home number.

The transfer on the call handler is set as supervised, 5 rings.

So, all that, and here's what happens:
Incoming call goes to Unity, Unity sends it to the handler, handler makes
the call out.  This part is all OK.  The issue is the supervised transfer.
The caller doesn't hear any call progress notification, and if I turn off
the "Wait while I transfer your call", then it may be 15 seconds of silence
before it comes back to Unity and you hear the personal greeting.

If I release to switch, the call never comes back to Unity.

I tried setting up an audio source using a wav file of the standard call
progress "ring", but it doesn't seem like the call is really on hold, so the
caller still gets silence.

Unity 4.2(1), CM 4.1.3sr2


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From msmith at laurinburg.org  Fri May  4 15:22:31 2007
From: msmith at laurinburg.org (Marcus Smith)
Date: Fri, 4 May 2007 14:22:31 -0500
Subject: [cisco-voip] vpn
Message-ID: <002f01c78e81$8fc3d5d0$1e00a8c0@laurinburg.org>

I have a question that is not for this forum, but I would like some input.  The question is, if on my pix 515 firewall, I set the vpngroup idle-time to 2400,  why then does my user keep informing me that his Microsoft vpn connection is dropping out after 10 mins.


Thank You,

Marcus Smith
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From erik.stillman at shearman.com  Fri May  4 15:22:13 2007
From: erik.stillman at shearman.com (Erik Stillman)
Date: Fri, 4 May 2007 15:22:13 -0400
Subject: [cisco-voip] Perflib error on Call Mgr
In-Reply-To: <mailman.12597.1178301515.1489.cisco-voip@puck.nether.net>
Message-ID: <OF4DFEC2F6.B765A6CD-ON852572D1.006A50F4-852572D1.006A61B3@shearman.com>

Carlos,

I found this option to be enabled by default. Have you checked your server?




***********************************************************************

This communication is not intended to be used, and cannot be used,
by the recipient or any other person for the purpose of avoiding United
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such other person.  In addition, if any United States federal tax advice
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From CarlosOrtiz at bayviewfinancial.com  Fri May  4 16:54:09 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 4 May 2007 16:54:09 -0400
Subject: [cisco-voip] Perflib error on Call Mgr
In-Reply-To: <OF4DFEC2F6.B765A6CD-ON852572D1.006A50F4-852572D1.006A61B3@shearman.com>
Message-ID: <OFD6C692B2.9A901789-ON852572D1.0072B76E-852572D1.0072D244@bftg.com>

Yeah DHCP is there but it's disabled.....ANyway I am going to try the 
Cisco recommended fix.



Erik Stillman <erik.stillman at shearman.com> 
Sent by: cisco-voip-bounces at puck.nether.net
05/04/2007 03:22 PM

To
cisco-voip at puck.nether.net
cc

Subject
Re: [cisco-voip] Perflib error on Call Mgr






Carlos,

I found this option to be enabled by default. Have you checked your 
server?




***********************************************************************

This communication is not intended to be used, and cannot be used,
by the recipient or any other person for the purpose of avoiding United
States federal tax penalties that may be imposed on the recipient or
such other person.  In addition, if any United States federal tax advice
contained in this communication is used or referred to in promoting,
marketing or recommending any corporation, partnership or other entity,
investment plan, concept, structure or arrangement (which should be
assumed to be the case by a recipient or other person who is not our
client with respect to the subject matter of the communication), then
(i) such tax advice should be construed as written to support the
promotion or marketing of the transactions or matters addressed by the
advice and (ii) the recipient or other person should seek advice based on
the recipient or other person's particular circumstances from an
independent tax advisor.  For further information, please go to
http://www.shearman.com/disclaimer/tax_disclosure.html

***********************************************************************

This transmittal and/or attachments may be a confidential
attorney-client communication or may otherwise be privileged or
confidential. If you are not the intended recipient, you are hereby
notified that you have received this transmittal in error; any review,
dissemination, distribution or copying of this transmittal is strictly
prohibited. If you have received this transmittal and/or attachments in
error, please notify us immediately by reply or by telephone (call us
collect at +1 212-848-8400) and immediately delete this message and
all its attachments.

Shearman & Sterling LLP is a limited liability partnership organized in
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limit the personal liability of partners.

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From erickbe at yahoo.com  Fri May  4 18:36:58 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri, 4 May 2007 15:36:58 -0700 (PDT)
Subject: [cisco-voip] Forwarding a call to a home phone but _with_ Unity
	VM if it's not answered
Message-ID: <661141.293.qm@web37213.mail.mud.yahoo.com>

Unity does place the call on hold, so CCM MoH would be played for both release to switch and supervisor transfer. The release to switch is a real quick transfer (unity puts call on hold, places call on other port, then finishes transfer quick). 

The MoH source used, would come from the voice mail port configuration on the call manager side.

----- Original Message ----
From: Fred Nielsen <fwn at feasible.net>
To: Robert Kulagowski <bob at smalltime.com>; cisco-voip at puck.nether.net
Sent: Friday, May 4, 2007 1:06:16 PM
Subject: Re: [cisco-voip] Forwarding a call to a home phone but _with_ Unity VM if it's not answered

This may not be what you are looking for.. But mobility manager (or Unified
Mobility now I think it is called) does what you are needing quite nicely. 

Also, I do seem to remember that Unity does in fact place the call on hold
(from CM's perspective) while doing a transfer (though I'm not 100% clear on
whether this is definitely true for a *supervised* transfer), so you may
want to look into the MoH setup further.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert Kulagowski
Sent: Friday, May 04, 2007 9:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Forwarding a call to a home phone but _with_ Unity VM
if it's not answered

So I have a number of users that have phones at home.  We'd like to setup
their DIDs so that an incoming call gets sent to their home phone. 
  If they don't answer their phone at home, it should end up in Unity.

Right now it's kind of a kluge, and doesn't do what the users want.

DID is setup to CFA to VM.
Their VM account is set to a blank greeting, and after the greeting the call
gets sent to a Call Handler.  _That_ call handler is owned by the user, and
has their personal message on it.  The call handler has a Call Transfer back
to CallManager to a dummy number that's on a CTI port. 
The dummy number is CFA to the user's home number.

The transfer on the call handler is set as supervised, 5 rings.

So, all that, and here's what happens:
Incoming call goes to Unity, Unity sends it to the handler, handler makes
the call out.  This part is all OK.  The issue is the supervised transfer.
The caller doesn't hear any call progress notification, and if I turn off
the "Wait while I transfer your call", then it may be 15 seconds of silence
before it comes back to Unity and you hear the personal greeting.

If I release to switch, the call never comes back to Unity.

I tried setting up an audio source using a wav file of the standard call
progress "ring", but it doesn't seem like the call is really on hold, so the
caller still gets silence.

Unity 4.2(1), CM 4.1.3sr2


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip




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From netsecuredata at gmail.com  Sat May  5 22:50:48 2007
From: netsecuredata at gmail.com (Jorge Evangelista)
Date: Sat, 5 May 2007 21:50:48 -0500
Subject: [cisco-voip] Reset an FXO port
In-Reply-To: <310F10740485D9469F0D8417950B252D026F3C70@CTG-MSNEXC01.staff.berbee.com>
References: <6101bda50704301051i577d3c3dj6507c27d883e8451@mail.gmail.com>
	<c6f32ce40704301053h64f00419p3952bb7290fcf35d@mail.gmail.com>
	<6101bda50704301117u1bb79397yf9b5c47859479112@mail.gmail.com>
	<310F10740485D9469F0D8417950B252D026F3C70@CTG-MSNEXC01.staff.berbee.com>
Message-ID: <de85c96f0705051950l78769b2fk786f8d3fa9822eed@mail.gmail.com>

Hi,

You can use    clear voice port 0/0/0


On 4/30/07, Clouse, Chris <chris.clouse at berbee.com> wrote:
>
>
>
> Can you post the port configuration?
>
>
>
> -Christopher
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Todd Franklin
> Sent: Monday, April 30, 2007 1:17 PM
> To: Adam; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Reset an FXO port
>
>
>
>
> Yeah I did that.  But still no go.
> Here is the sho voice port 0/0/0 command output:
>
> ho voice port 0/0/0
>
> Foreign Exchange Office 0/0/0 Slot is 0, Sub-unit is 0, Port is 0
>  Type of VoicePort is FXO
>  Operation State is DORMANT
>  Administrative State is UP
>  The Last Interface Down Failure Cause is Administrative Shutdown
>  Description is 555-123-4567
>  Noise Regeneration is enabled
>  Non Linear Processing is enabled
>  Non Linear Mute is disabled
>  Non Linear Threshold is -21 dB
>  Music On Hold Threshold is Set to -38 dBm
>  In Gain is Set to 0 dB
>  Out Attenuation is Set to 3 dB
>  Echo Cancellation is enabled
>  Echo Cancellation NLP mute is disabled
>  Echo Cancellation NLP threshold is -21 dB
>  Echo Cancel Coverage is set to 48 ms
>  Echo Cancel worst case ERL is set to 6 dB
>  Playout-delay Mode is set to adaptive
>  Playout-delay Nominal is set to 60 ms
>  Playout-delay Maximum is set to 250 ms
>  Playout-delay Minimum mode is set to default, value 40 ms
>  Playout-delay Fax is set to 300 ms
>  Connection Mode is plar
>  Connection Number is 2002
>  Initial Time Out is set to 10 s
>  Interdigit Time Out is set to 10 s
>  Call Disconnect Time Out is set to 60 s
>  Ringing Time Out is set to 180 s
>  Wait Release Time Out is set to 30 s
>  Companding Type is u-law
>  Region Tone is set for US
>
>  Analog Info Follows:
>  Currently processing none
>  Maintenance Mode Set to None (not in mtc mode)
>  Number of signaling protocol errors are 0
>  Impedance is set to 600r Ohm
>  Station name None, Station number None
>  Translation profile (Incoming):
>  Translation profile (Outgoing):
>
>  Voice card specific Info Follows:
>  Signal Type is loopStart
>  Battery-Reversal is enabled
>  Number Of Rings is set to 1
>  Supervisory Disconnect is signal
>  Answer Supervision is inactive
>  Hook Status is On Hook
>  Ring Detect Status is inactive
>  Ring Ground Status is inactive
>  Tip Ground Status is inactive
>  Dial Out Type is dtmf
>  Digit Duration Timing is set to 100 ms
>  InterDigit Duration Timing is set to 100 ms
>  Pulse Rate Timing is set to 10 pulses/second
>  InterDigit Pulse Duration Timing is set to 750 ms
>  Percent Break of Pulse is 60 percent
>  GuardOut timer is 2000 ms
>  Minimum ring duration timer is 125 ms
>  Hookflash-in Timing is set to 600 ms
>  Hookflash-out Timing is set to 500 ms
>  Supervisory Disconnnct Timing is set to 350 ms
>
> I think I am missing something basic.  This has happened before and the only
> way to fix it was to reboot the 2801.
>
> Todd
>
>
> On 4/30/07, Adam <cisco at adman.net> wrote:
>
> How about a 'shutdown' followed by 'no shutdown' in the voice-port
> configuration?
>
> -Adam
>
> On 4/30/07, Todd Franklin <toddnh65 at gmail.com> wrote:
> > Apparently my FXO port is hung internally.  I can't easily reboot this box
> > since it has my PRI on it as well.
> > Is there a command to just reset the FXO port?  Right now I have my telco
> > busying it out so it will fall over to the next line (which comes in on
> the
> > PRI).
> >
> > Thanks anyone!
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
"The network is the computer"


From aelnagar at ACT-EG.COM  Sun May  6 02:06:05 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Sun, 6 May 2007 09:06:05 +0300
Subject: [cisco-voip] Call Waiting and tone for caller in 5.X
References: <463AE943.8000608@dataways.gr>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A035@mail.ACT-EG.COM>

yes I had this issue too, but did not find a sol.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Anthony Kouloglou
Sent: Fri 04-May-07 11:05 AM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] Call Waiting and tone for caller in 5.X



I have many requests for that:
is there a way for a caller to hear a different tone when he/she calls
an extension with call waiting enabled?

If not, is there a plan by Cisco to cover this issue in the future?

Cheers,
Anthony

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From jgrace at adcapnet.com  Sun May  6 17:04:31 2007
From: jgrace at adcapnet.com (James Grace)
Date: Sun, 6 May 2007 17:04:31 -0400
Subject: [cisco-voip] trying to set external # mask base on acct codes
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C64C@mail.adcapnetsvc.local>

?
trying to set external # mask base on acct codes.  basically we want to set  external # mask base on the call acct code give.  Have anybody done this before.
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
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From voip at mrga.ch  Mon May  7 08:07:35 2007
From: voip at mrga.ch (Reto Gassmann)
Date: Mon, 07 May 2007 14:07:35 +0200
Subject: [cisco-voip] Device Pack ciscocm.4-2-3-DevPack-20.exe
Message-ID: <1178539655.463f16878fc70@www.mail2web.ch>

Hi all

Last week I downloaded and installed the Device Pack
ciscocm.4-2-3-DevPack-20.exe on our Callmanager 4.2(3). 
I just visited the software download page and found only the Device Pack
ciscocm.4-2-3-DevPack-13.exe.
Am I blind or has Cisco removed the newer Device Pack or what happend to it.

Any ideas???

Thanks Reto




From Matt.Slaga at us.didata.com  Mon May  7 08:11:58 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 7 May 2007 08:11:58 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>

Actually, this is not the case.  It will only generate DLUs for phones
that are registered with the CallManager server.  It will skip any
device that is not registered and not count it in the DLU list.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Christopher M.
Bomba
Sent: Wednesday, May 02, 2007 9:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1

Who said you had to buy the phones.  You can just put phones in with
bogus MAC addresses and it will generate DLUs for those.  I thought they
had to be registered but turns out that they just have to be present in
the system.  You can create IP communicators or any other phone to boost
the DLUs.  That is a big problem since anybody can do this.

Chris

--original message--

thanks Ed for your reply, yes you are right I contacted Partner help and
they told me that during the upgrade a file will be generated that when
given to cisco they will give us back the DLUs equivalent to what
configured on the system. I think this is an illegal workaround for the
lic. issue as by that someone could buy phones without lic. and then
setup them with CM 4.1 and then upgrade and he will be having a DLUs for
free. I think Cisco has to do something to solve this.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: Ed Leatherman [mailto:ealeatherman at gmail.com]
Sent: Wed 02-May-07 3:16 PM
To: Ahmed Elnagar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1


hi Ahmed,

When you upgrade, callmanager will generate a temporary license for all
the devices you have configured from 4.1. If I understand correctly, you
just have to send that off to cisco's licensing group and have it
converted to a real 5.2 license with the appropriate number of DLUs. If
its anything like Unity licensing, the turn-around is pretty quick. I'm
assuming you need a SASU agreement for callmanager for them to do it.

We are getting ready to do the same process here this summer with our
4.1 system.

Hope that helps.

Ed



On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote: 

	Hello all;
	 
	A customer wants to upgrade from CM 4.1 to 5.1, what I am
worried about is the licsences issue. How will I get the DLUs for
callmanager 5.1 we have a lic. with each phone but how we will get the
DLUs?? 
	 
	
	Thanks and Best Regards
	
	Ahmed A. Elnagar
	Network Field Engineer 
	 
	Advanced Computer Technology (ACT)
	16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
	Postal Code:12411 Cairo Egypt
	
	Mob : +2010-2833868
	Website: www.act-eg.com <http://www.act-eg.com/> 
	E-mail: aelnagar at act-eg.com

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip 
	
	




-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 
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End of cisco-voip Digest, Vol 51, Issue 4
*****************************************

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From toddnh65 at gmail.com  Mon May  7 08:14:15 2007
From: toddnh65 at gmail.com (Todd Franklin)
Date: Mon, 7 May 2007 08:14:15 -0400
Subject: [cisco-voip] SmartNet or Not
Message-ID: <6101bda50705070514m30909626q5898df4a38905a81@mail.gmail.com>

Hey everyone!

I just got one of those weird "third party" mailings that it was time to
renew my SmartNet.  Sort of like those ones you get about renewing your
domain for $75 when you can do it anywhere else for $15.

Anyway, it got me to thinking. Do you carry SmartNet on EVERYTHING??
Phones?  All switches?   I am just curious, trying to get an idea as I look
at my smartnet renewal costs!!!!

Todd
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From kdupree at rmwc.edu  Mon May  7 08:18:10 2007
From: kdupree at rmwc.edu (Kathy Dupree)
Date: Mon, 7 May 2007 08:18:10 -0400
Subject: [cisco-voip] MotoQ
Message-ID: <6E497ADB607656479C24E6D7BF6B505A0828259C@exchange.rmwc.edu>

I was asked today by a user if there is any way for the MotoQ to read
the voicemail emails that unity sends out.  I know people with
Blackberries have to buy software but I don't know anything about the
MotoQ.  Does anyone know?  Thanks

 

Katherine O. Dupree

Randolph-Macon Woman's College

Main 243

Telecommunications Support/Infrastructure Coordinator

kdupree at rmwc.edu <mailto:kdupree at rmwc.edu> 

(434) 947 8371

 

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From sdjokic at mds.co.yu  Mon May  7 08:29:26 2007
From: sdjokic at mds.co.yu (Djokic Sinisa)
Date: Mon, 7 May 2007 14:29:26 +0200
Subject: [cisco-voip] ccm4.1.3 and ipccx 4.0 ip address change !!!
Message-ID: <200705071229.l47CTWwC055700@puck.nether.net>

 

hi to all..

can anybody point me to a procedure of changing ip addresses on CCM cluster
and ipcc express..

the design is as follows:

 

we have CCM PUB and CCM SUB - version 4.1.3..

on the CCM PUB we have co-resident installation of IPCC express 4.0..

 

anyway, besides the procedure, has anybody try it and what is experience
like, any issues etc..

 

thanx..

 

regards..

 

Sinisa Djokic 

System Engineer
CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 




  

MDS Informaticki inzenjering
Bul. Arsenija Carnojevica 170
11070 Novi Beograd, Serbia
Tel: +381 11 2015 200, 2015 273
Fax: +381 11 3194 954
www.mds.co.yu
sdjokic at mds.co.yu

 

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From mlorad1 at towson.edu  Mon May  7 08:30:10 2007
From: mlorad1 at towson.edu (Loraditch, Matthew*)
Date: Mon, 7 May 2007 08:30:10 -0400
Subject: [cisco-voip] MotoQ
In-Reply-To: <6E497ADB607656479C24E6D7BF6B505A0828259C@exchange.rmwc.edu>
References: <6E497ADB607656479C24E6D7BF6B505A0828259C@exchange.rmwc.edu>
Message-ID: <A57E8EB3EA7B264FBDF12D930FED574F910504@argon.towson.edu>

While I can't say for sure, Windows Mobile does support standard audio files so I don't see why it wouldn't. You may not be able to play it directly from mobile outlook, but you should be able to open it up in media player.

 

Matthew Loraditch 

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kathy Dupree
Sent: Monday, May 07, 2007 8:18 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MotoQ

 

I was asked today by a user if there is any way for the MotoQ to read the voicemail emails that unity sends out.  I know people with Blackberries have to buy software but I don't know anything about the MotoQ.  Does anyone know?  Thanks

 

Katherine O. Dupree

Randolph-Macon Woman's College

Main 243

Telecommunications Support/Infrastructure Coordinator

kdupree at rmwc.edu

(434) 947 8371

 

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From vloschiavo at data-corporation.com  Mon May  7 08:37:23 2007
From: vloschiavo at data-corporation.com (=?UTF-8?B?VmluY2UgTG9zY2hpYXZv?=)
Date: Mon, 7 May 2007 12:37:23 +0000
Subject: [cisco-voip] MotoQ
In-Reply-To: <A57E8EB3EA7B264FBDF12D930FED574F910504@argon.towson.edu>
References: <6E497ADB607656479C24E6D7BF6B505A0828259C@exchange.rmwc.edu>
	<A57E8EB3EA7B264FBDF12D930FED574F910504@argon.towson.edu>
Message-ID: <1664666681-1178541526-cardhu_blackberry.rim.net-1350441227-@bxe035-cell01.bisx.prod.on.blackberry>

You can change the codec that unity stores the messages in. 

Change to the GSM codec and they should be able to play them. 

Of course, this is a global change. 
Regards,

Vince Loschiavo
Senior Cisco Engineer
DATACORP
7862821164

Sent via BlackBerry from T-Mobile    

-----Original Message-----
From: "Loraditch, Matthew*" <mlorad1 at towson.edu>
Date: Mon, 7 May 2007 08:30:10 
To:<cisco-voip at puck.nether.net>
Subject: Re: [cisco-voip] MotoQ

While I can?t say for sure, Windows Mobile does support standard audio files so I don?t see why it wouldn?t. You may not be able to play it directly from mobile outlook, but you should be able to open it up in media player.
 
?
 
 
Matthew Loraditch 
 
?
 
 
 
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kathy Dupree
 Sent: Monday, May 07, 2007 8:18 AM
 To: cisco-voip at puck.nether.net
 Subject: [cisco-voip] MotoQ
 
?
 
I was asked today by a user if there is any way for the MotoQ to read the voicemail emails that unity sends out.? I know people with Blackberries have to buy software but I don?t know anything about the MotoQ.? Does anyone know?? Thanks
 
?
 
Katherine O. Dupree
 
Randolph-Macon Woman's College
 
Main 243
 
Telecommunications Support/Infrastructure Coordinator
 
kdupree at rmwc.edu
 
(434) 947 8371
 
?
 _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From voip at mrga.ch  Mon May  7 08:42:58 2007
From: voip at mrga.ch (Reto Gassmann)
Date: Mon, 07 May 2007 14:42:58 +0200
Subject: [cisco-voip] ccm4.1.3 and ipccx 4.0 ip address change !!!
In-Reply-To: <200705071229.l47CTWwC055700@puck.nether.net>
References: <200705071229.l47CTWwC055700@puck.nether.net>
Message-ID: <1178541778.463f1ed26af9d@www.mail2web.ch>

Hi

check this link:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094601.shtml

Reto

Zitat von Djokic Sinisa <sdjokic at mds.co.yu>:

>  
> 
> hi to all..
> 
> can anybody point me to a procedure of changing ip addresses on CCM cluster
> and ipcc express..
> 
> the design is as follows:
> 
>  
> 
> we have CCM PUB and CCM SUB - version 4.1.3..
> 
> on the CCM PUB we have co-resident installation of IPCC express 4.0..
> 
>  
> 
> anyway, besides the procedure, has anybody try it and what is experience
> like, any issues etc..
> 
>  
> 
> thanx..
> 
>  
> 
> regards..
> 
>  
> 
> Sinisa Djokic 
> 
> System Engineer
> CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified
> 
>  
> 
> 
> 
> 
>   
> 
> MDS Informaticki inzenjering
> Bul. Arsenija Carnojevica 170
> 11070 Novi Beograd, Serbia
> Tel: +381 11 2015 200, 2015 273
> Fax: +381 11 3194 954
> www.mds.co.yu
> sdjokic at mds.co.yu
> 
>  
> 
> 





From ealeatherman at gmail.com  Mon May  7 08:49:15 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 7 May 2007 08:49:15 -0400
Subject: [cisco-voip] SmartNet or Not
In-Reply-To: <6101bda50705070514m30909626q5898df4a38905a81@mail.gmail.com>
References: <6101bda50705070514m30909626q5898df4a38905a81@mail.gmail.com>
Message-ID: <94a1afde0705070549l2eea7400o9086ea3514ef0c3b@mail.gmail.com>

Hi Todd,

We do not carry smartnet on phones. out of several thousand, we've had maybe
3 go bad - I am thinking about sending them off to a repair outfit to see if
its worth it. We do cover everything else on the voice side with smartnet.

I'm not 100% sure how our network ops folks cover stuff, I know they do not
cover all of their catalyst access layer switches but instead just use the
warranty for a replacement if one goes bad. they have spares of everything
they use so they can replace something immediately while Cisco sends a new
switch. I think they do cover switches that are running enhanced IOS code.

On 5/7/07, Todd Franklin <toddnh65 at gmail.com> wrote:
>
> Hey everyone!
>
> I just got one of those weird "third party" mailings that it was time to
> renew my SmartNet.  Sort of like those ones you get about renewing your
> domain for $75 when you can do it anywhere else for $15.
>
> Anyway, it got me to thinking. Do you carry SmartNet on EVERYTHING??
> Phones?  All switches?   I am just curious, trying to get an idea as I look
> at my smartnet renewal costs!!!!
>
> Todd
>
>
-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Mon May  7 08:53:47 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 7 May 2007 08:53:47 -0400
Subject: [cisco-voip] SmartNet or Not
References: <6101bda50705070514m30909626q5898df4a38905a81@mail.gmail.com>
Message-ID: <003f01c790a6$c0bc3840$47196883@cfs.uoguelph.ca>

We don't smartnet phones or anything with a lifetime warranty. We spare them instead. We have also elected not to smartnet our APs - we'll just replace them when they go bad.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Todd Franklin 
  To: cisco-voip at puck.nether.net 
  Sent: Monday, May 07, 2007 8:14 AM
  Subject: [cisco-voip] SmartNet or Not


  Hey everyone!

  I just got one of those weird "third party" mailings that it was time to renew my SmartNet.  Sort of like those ones you get about renewing your domain for $75 when you can do it anywhere else for $15. 

  Anyway, it got me to thinking. Do you carry SmartNet on EVERYTHING??  Phones?  All switches?   I am just curious, trying to get an idea as I look at my smartnet renewal costs!!!!

  Todd



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From sdjokic at mds.co.yu  Mon May  7 09:05:23 2007
From: sdjokic at mds.co.yu (Djokic Sinisa)
Date: Mon, 7 May 2007 15:05:23 +0200
Subject: [cisco-voip] ccm4.1.3 and ipccx 4.0 ip address change !!!
In-Reply-To: <1178541778.463f1ed26af9d@www.mail2web.ch>
Message-ID: <200705071305.l47D5U4d076005@puck.nether.net>

 

 

thanx guys..

i did find those links..

i also find this one considering IPCC express co-res with CCM..

 

Check page 260, section Scenario 1: CRS Collocated with CCM on
<http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_4_0/eng
lish/administ/crs401ag.pdf>
http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_4_0/engl
ish/administ/crs401ag.pdf for details on changing IP address..

 

so the CCM part is allways the same ( hoping it goes well ) but CRS is
not..which one to use..

and i'm just currious with the sequence of events..

this doc about CRS tells me:

 

1. Stop the CRS Node manager ( IPCC express service ) from the Services
window
2. Modify the IP address using Windows LAN TCP/IP settings
3. Change CCM PUB IP address using CallManager IP address change guides -
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_tech_n
ote09186a0080094601.shtml


4. After whole CCM IP address change process is finished proceed with the
IPCC express change process through the end from procedure 

CRS Collocated with CCM on
<http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_4_0/eng
lish/administ/crs401ag.pdf>
http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_4_0/engl
ish/administ/crs401ag.pdf

 

 

what about subscriber..everything the same as for publisher?..

 

Sinisa Djokic 

System Engineer

CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 

MDS Informaticki inzenjering

Bul. Arsenija Carnojevica 170

11070 Novi Beograd, Serbia

Tel: +381 11 2015 200, 2015 273

Fax: +381 11 3194 954

www.mds.co.yu

sdjokic at mds.co.yu

      

 

-----Original Message-----
From: Reto Gassmann [mailto:voip at mrga.ch] 
Sent: Monday, May 07, 2007 2:43 PM
To: Djokic Sinisa
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm4.1.3 and ipccx 4.0 ip address change !!!

 

Hi

 

check this link:

 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a0080094601.shtml

 

Reto

 

Zitat von Djokic Sinisa <sdjokic at mds.co.yu>:

 

>  

> 

> hi to all..

> 

> can anybody point me to a procedure of changing ip addresses on CCM
cluster

> and ipcc express..

> 

> the design is as follows:

> 

>  

> 

> we have CCM PUB and CCM SUB - version 4.1.3..

> 

> on the CCM PUB we have co-resident installation of IPCC express 4.0..

> 

>  

> 

> anyway, besides the procedure, has anybody try it and what is experience

> like, any issues etc..

> 

>  

> 

> thanx..

> 

>  

> 

> regards..

> 

>  

> 

> Sinisa Djokic 

> 

> System Engineer

> CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

> 

>  

> 

> 

> 

> 

>   

> 

> MDS Informaticki inzenjering

> Bul. Arsenija Carnojevica 170

> 11070 Novi Beograd, Serbia

> Tel: +381 11 2015 200, 2015 273

> Fax: +381 11 3194 954

> www.mds.co.yu

> sdjokic at mds.co.yu

> 

>  

> 

> 

 

 

 

 

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From voip at mrga.ch  Mon May  7 09:58:09 2007
From: voip at mrga.ch (Reto Gassmann)
Date: Mon, 07 May 2007 15:58:09 +0200
Subject: [cisco-voip] Networ Locale Installer 4.2.1.3000
Message-ID: <1178546289.463f30713192f@www.mail2web.ch>

Hi all

I try to update my network locals on a CCM 4.2(3). However, if I want to apply
the file cm-locale-combined_network-4.2.1.3000.exe I get an error message that
the AutoRegister Phone Prompts are not in the package and the installation
fails.

I choose Germany, France, Italy and Switzerland to install.

Any ideas??

Reto




From cbomba at s4nets.com  Mon May  7 10:16:02 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Mon, 7 May 2007 10:16:02 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
	<C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>
Message-ID: <000901c790b2$3e33d070$ba9b7150$@com>

That is good.  I was told both.  It would have been a shame to see people
abusing this.

Chris

-----Original Message-----
From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Monday, May 07, 2007 8:12 AM
To: cbomba at s4nets.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1

Actually, this is not the case.  It will only generate DLUs for phones
that are registered with the CallManager server.  It will skip any
device that is not registered and not count it in the DLU list.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Christopher M.
Bomba
Sent: Wednesday, May 02, 2007 9:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1

Who said you had to buy the phones.  You can just put phones in with
bogus MAC addresses and it will generate DLUs for those.  I thought they
had to be registered but turns out that they just have to be present in
the system.  You can create IP communicators or any other phone to boost
the DLUs.  That is a big problem since anybody can do this.

Chris

--original message--

thanks Ed for your reply, yes you are right I contacted Partner help and
they told me that during the upgrade a file will be generated that when
given to cisco they will give us back the DLUs equivalent to what
configured on the system. I think this is an illegal workaround for the
lic. issue as by that someone could buy phones without lic. and then
setup them with CM 4.1 and then upgrade and he will be having a DLUs for
free. I think Cisco has to do something to solve this.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: Ed Leatherman [mailto:ealeatherman at gmail.com]
Sent: Wed 02-May-07 3:16 PM
To: Ahmed Elnagar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1


hi Ahmed,

When you upgrade, callmanager will generate a temporary license for all
the devices you have configured from 4.1. If I understand correctly, you
just have to send that off to cisco's licensing group and have it
converted to a real 5.2 license with the appropriate number of DLUs. If
its anything like Unity licensing, the turn-around is pretty quick. I'm
assuming you need a SASU agreement for callmanager for them to do it.

We are getting ready to do the same process here this summer with our
4.1 system.

Hope that helps.

Ed



On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote: 

	Hello all;
	 
	A customer wants to upgrade from CM 4.1 to 5.1, what I am
worried about is the licsences issue. How will I get the DLUs for
callmanager 5.1 we have a lic. with each phone but how we will get the
DLUs?? 
	 
	
	Thanks and Best Regards
	
	Ahmed A. Elnagar
	Network Field Engineer 
	 
	Advanced Computer Technology (ACT)
	16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
	Postal Code:12411 Cairo Egypt
	
	Mob : +2010-2833868
	Website: www.act-eg.com <http://www.act-eg.com/> 
	E-mail: aelnagar at act-eg.com

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip 
	
	




-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 
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End of cisco-voip Digest, Vol 51, Issue 4
*****************************************

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From bob at smalltime.com  Mon May  7 10:18:47 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Mon, 07 May 2007 09:18:47 -0500
Subject: [cisco-voip] Networ Locale Installer 4.2.1.3000
In-Reply-To: <1178546289.463f30713192f@www.mail2web.ch>
References: <1178546289.463f30713192f@www.mail2web.ch>
Message-ID: <463F3547.9000800@smalltime.com>

Reto Gassmann wrote:
> Hi all
> 
> I try to update my network locals on a CCM 4.2(3). However, if I want to apply
> the file cm-locale-combined_network-4.2.1.3000.exe I get an error message that
> the AutoRegister Phone Prompts are not in the package and the installation
> fails.
> 
> I choose Germany, France, Italy and Switzerland to install.

I had exactly the same problem, opened a TAC case (605506783). 
Basically, the TAC guy refused to understand that the .exe was finding 
TAPS on the system, automatically adding TAPS, and then failing because 
the TAPS prompts weren't actually in the .exe.

You can even do a custom install, select only one region, then click 
Next.  It automatically adds TAPS prompts!  And TAPS prompts aren't even 
an option in the list of regions that you can select, so you can't 
de-select them.

Eventually, I gave up trying to convince the TAC engineer and had him 
close the case.  Maybe you'll have better luck.


From keith.klevenski at rig.net  Mon May  7 10:26:15 2007
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Mon, 7 May 2007 09:26:15 -0500
Subject: [cisco-voip] Lost Subscriber
Message-ID: <7F0CE0A1A0961E498A41329535342266011AC07C@HOUEXH02.rignet.org>

Hi,

 

We lost both hard drives on a subscriber in a 4.1.3 cluster last night.
The publisher is fine.  What is the best way to add the same subscriber
back to the cluster?  We popped in a couple of spare hard drives and
want to add the same subscriber (name and IP) back to the cluster.  Do I
just add the subscriber to the cluster just like it was brand new?  I
assume I may need to remove the subscriber from the cluster (remove
server and CallManager from ccmadmin) as well before doing this?  Anyone
been through this before and has some quick tips?

 

Thanks! 

 

Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077         

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

http://www.rig.net <http://www.rig.net/> 

 

"Extending the power of your knowledge"

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From ealeatherman at gmail.com  Mon May  7 10:37:07 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 7 May 2007 10:37:07 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <000901c790b2$3e33d070$ba9b7150$@com>
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
	<C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>
	<000901c790b2$3e33d070$ba9b7150$@com>
Message-ID: <94a1afde0705070737j39f244c6ge77f400cba50c595@mail.gmail.com>

We're doing a mock upgrade this week in the lab, I'll see what it does. I've
been told both things also.

We're loading one of our BARS backups onto a test server and then doing the
upgrade process on it.. so should be very obvious if it gives me licenses
for all the unregistered devices on the cluster or just the few that we
endup plugging up in the lab.

On 5/7/07, Christopher M. Bomba <cbomba at s4nets.com> wrote:
>
> That is good.  I was told both.  It would have been a shame to see people
> abusing this.
>
> Chris
>
> -----Original Message-----
> From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com]
> Sent: Monday, May 07, 2007 8:12 AM
> To: cbomba at s4nets.com; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
> Actually, this is not the case.  It will only generate DLUs for phones
> that are registered with the CallManager server.  It will skip any
> device that is not registered and not count it in the DLU list.
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Christopher M.
> Bomba
> Sent: Wednesday, May 02, 2007 9:05 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
> Who said you had to buy the phones.  You can just put phones in with
> bogus MAC addresses and it will generate DLUs for those.  I thought they
> had to be registered but turns out that they just have to be present in
> the system.  You can create IP communicators or any other phone to boost
> the DLUs.  That is a big problem since anybody can do this.
>
> Chris
>
> --original message--
>
> thanks Ed for your reply, yes you are right I contacted Partner help and
> they told me that during the upgrade a file will be generated that when
> given to cisco they will give us back the DLUs equivalent to what
> configured on the system. I think this is an illegal workaround for the
> lic. issue as by that someone could buy phones without lic. and then
> setup them with CM 4.1 and then upgrade and he will be having a DLUs for
> free. I think Cisco has to do something to solve this.
>
> Thanks and Best Regards
>
> Ahmed A. Elnagar
> Network Field Engineer
>
> Advanced Computer Technology (ACT)
> 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> Postal Code:12411 Cairo Egypt
>
> Mob: +2010-2833868
> Website: www.act-eg.com
> E-mail: aelnagar at act-eg.com
>
> ________________________________
>
> From: Ed Leatherman [mailto:ealeatherman at gmail.com]
> Sent: Wed 02-May-07 3:16 PM
> To: Ahmed Elnagar
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
>
> hi Ahmed,
>
> When you upgrade, callmanager will generate a temporary license for all
> the devices you have configured from 4.1. If I understand correctly, you
> just have to send that off to cisco's licensing group and have it
> converted to a real 5.2 license with the appropriate number of DLUs. If
> its anything like Unity licensing, the turn-around is pretty quick. I'm
> assuming you need a SASU agreement for callmanager for them to do it.
>
> We are getting ready to do the same process here this summer with our
> 4.1 system.
>
> Hope that helps.
>
> Ed
>
>
>
> On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>         Hello all;
>
>         A customer wants to upgrade from CM 4.1 to 5.1, what I am
> worried about is the licsences issue. How will I get the DLUs for
> callmanager 5.1 we have a lic. with each phone but how we will get the
> DLUs??
>
>
>         Thanks and Best Regards
>
>         Ahmed A. Elnagar
>         Network Field Engineer
>
>         Advanced Computer Technology (ACT)
>         16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
>         Postal Code:12411 Cairo Egypt
>
>         Mob : +2010-2833868
>         Website: www.act-eg.com <http://www.act-eg.com/>
>         E-mail: aelnagar at act-eg.com
>
>         _______________________________________________
>         cisco-voip mailing list
>         cisco-voip at puck.nether.net
>         https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> -------------- next part --------------
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> 78/attachment.html
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> End of cisco-voip Digest, Vol 51, Issue 4
> *****************************************
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
>
> _______________________________________________
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> cisco-voip at puck.nether.net
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>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From cbomba at s4nets.com  Mon May  7 10:40:21 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Mon, 7 May 2007 10:40:21 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <94a1afde0705070737j39f244c6ge77f400cba50c595@mail.gmail.com>
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>	
	<C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>	
	<000901c790b2$3e33d070$ba9b7150$@com>
	<94a1afde0705070737j39f244c6ge77f400cba50c595@mail.gmail.com>
Message-ID: <002701c790b5$a46792c0$ed36b840$@com>

Yes, please let me know.

 

Thanks,

 

Chris

 

From: Ed Leatherman [mailto:ealeatherman at gmail.com] 
Sent: Monday, May 07, 2007 10:37 AM
To: Christopher M. Bomba
Cc: Matt Slaga (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1

 

We're doing a mock upgrade this week in the lab, I'll see what it does. I've been told both things also.

We're loading one of our BARS backups onto a test server and then doing the upgrade process on it.. so should be very obvious if it gives me licenses for all the unregistered devices on the cluster or just the few that we endup plugging up in the lab. 

On 5/7/07, Christopher M. Bomba <cbomba at s4nets.com> wrote:

That is good.  I was told both.  It would have been a shame to see people
abusing this.

Chris

-----Original Message-----
From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com ]
Sent: Monday, May 07, 2007 8:12 AM
To: cbomba at s4nets.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1

Actually, this is not the case.  It will only generate DLUs for phones
that are registered with the CallManager server.  It will skip any
device that is not registered and not count it in the DLU list. 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Christopher M.
Bomba
Sent: Wednesday, May 02, 2007 9:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1

Who said you had to buy the phones.  You can just put phones in with
bogus MAC addresses and it will generate DLUs for those.  I thought they
had to be registered but turns out that they just have to be present in 
the system.  You can create IP communicators or any other phone to boost
the DLUs.  That is a big problem since anybody can do this.

Chris

--original message--

thanks Ed for your reply, yes you are right I contacted Partner help and 
they told me that during the upgrade a file will be generated that when
given to cisco they will give us back the DLUs equivalent to what
configured on the system. I think this is an illegal workaround for the
lic. issue as by that someone could buy phones without lic. and then
setup them with CM 4.1 and then upgrade and he will be having a DLUs for
free. I think Cisco has to do something to solve this.

Thanks and Best Regards 

Ahmed A. Elnagar
Network Field Engineer

Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: Ed Leatherman [mailto: <mailto:ealeatherman at gmail.com>  ealeatherman at gmail.com]
Sent: Wed 02-May-07 3:16 PM
To: Ahmed Elnagar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1


hi Ahmed,

When you upgrade, callmanager will generate a temporary license for all
the devices you have configured from 4.1. If I understand correctly, you
just have to send that off to cisco's licensing group and have it 
converted to a real 5.2 license with the appropriate number of DLUs. If
its anything like Unity licensing, the turn-around is pretty quick. I'm
assuming you need a SASU agreement for callmanager for them to do it. 

We are getting ready to do the same process here this summer with our
4.1 system.

Hope that helps.

Ed



On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com > wrote:

        Hello all;

        A customer wants to upgrade from CM 4.1 to 5.1, what I am
worried about is the licsences issue. How will I get the DLUs for
callmanager 5.1 we have a lic. with each phone but how we will get the 
DLUs??


        Thanks and Best Regards

        Ahmed A. Elnagar
        Network Field Engineer

        Advanced Computer Technology (ACT)
        16 Fawzy Ramah St.Off Shehab St.Mohandessin , Giza, Egypt
        Postal Code:12411 Cairo Egypt

        Mob : +2010-2833868
        Website: www.act-eg.com <http://www.act-eg.com/ >
        E-mail: aelnagar at act-eg.com

        _______________________________________________
        cisco-voip mailing list
         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
        https://puck.nether.net/mailman/listinfo/cisco-voip






--
Ed Leatherman
Senior Voice Engineer 
West Virginia University
Telecommunications and Network Operations
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------------------------------

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


End of cisco-voip Digest, Vol 51, Issue 4 
*****************************************

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the 
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be 
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.


_______________________________________________ 
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 




-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 

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From bcarter at sentinel.com  Mon May  7 10:48:37 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 7 May 2007 09:48:37 -0500
Subject: [cisco-voip] OT: Cisco Approved Memory
Message-ID: <6A0744639B86DD459433108DBF101D26015D0745@MAIL01.sentinel.com>

Little off topic.  I have a customer who wants to upgrade the DRAM in
Sup720 modules.  I have seen some 3rd party vendors offering "Cisco
Approved" memory?  Is this accurate ?  Does Cisco now have "approved"
memory vendors?  I am looking at a list price difference of $4,800 vs
$500.
 
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From gokhanciscottl at yahoo.com  Mon May  7 10:24:23 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Mon, 7 May 2007 07:24:23 -0700 (PDT)
Subject: [cisco-voip] ccm 4.1.3 voip backbone quality issue
Message-ID: <468184.27456.qm@web39608.mail.mud.yahoo.com>

hi

i have ccm 4.1.3...  
for international calls i routed the calls to ISP voip backbone but the quality is not good
they asked me to give them ccm logs for these calls.
what kind of logs can i support and also how??
thanks a lot

also i asked them QOS config on the routerA which is beetween our network and this isp.
But they told me that they made 512 rate limit in their routerB and there was no need to any qos config in the routerA in my side

(my LAN )ccm 4.1.3  ,ip phones , routerA.>>>.......leased line.......>>>routerB(isp router )

what do you think?
thanks a lot
Gokhan


 
____________________________________________________________________________________
Food fight? Enjoy some healthy debate 
in the Yahoo! Answers Food & Drink Q&A.
http://answers.yahoo.com/dir/?link=list&sid=396545367
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From lelio at uoguelph.ca  Mon May  7 11:26:45 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 7 May 2007 11:26:45 -0400
Subject: [cisco-voip] OT: Cisco Approved Memory
References: <6A0744639B86DD459433108DBF101D26015D0745@MAIL01.sentinel.com>
Message-ID: <010201c790bc$1f313b40$47196883@cfs.uoguelph.ca>


I guess if you don't tell them you have third party memory, how are they going to know? ;)


http://www.cisco.com/en/US/products/prod_warranties_item09186a00800b5594.html


      Products & Services
      Third Party Components - Cisco Policy 
           
           
           
           
      Cisco Guideline:
      Third Party Components Support
      The Cisco guideline for support and warranty services for the use of third-party memory, cables, gigabit interface controllers (GBICs), filters, or other non-Cisco components is as follows:

        a.. When a customer reports a product fault or defect and Cisco believes the fault or defect can be traced to the use of third-party memory products, cables, GBIC's, filters, or other non-Cisco components by a customer or reseller, then, at Cisco's discretion, Cisco may withhold support under warranty or a Cisco support program such as SMARTnetT service. 
        a.. When a product fault or defect occurs in the network, and Cisco concludes that the fault or defect is not attributable to the use of third-party memory, cables, GBICs, filters, or other non-Cisco components installed by a customer or reseller, Cisco will continue to provide support for the affected product under warranty or covered by a Cisco support program. 
      The nature of the defect or error is the key to determining Cisco support obligations.
      Cisco also reserves the right to charge the customer per then current time and material rates for services provided to the customer when Cisco determines, after having provided such services, that the root cause of the defective product was caused by a third party vendor supplied product.




     

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Carter, Bill 
  To: cisco-voip at puck.nether.net 
  Sent: Monday, May 07, 2007 10:48 AM
  Subject: [cisco-voip] OT: Cisco Approved Memory


  Little off topic.  I have a customer who wants to upgrade the DRAM in Sup720 modules.  I have seen some 3rd party vendors offering "Cisco Approved" memory?  Is this accurate ?  Does Cisco now have "approved" memory vendors?  I am looking at a list price difference of $4,800 vs $500.



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From asobihoudai at yahoo.com  Mon May  7 10:29:16 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Mon, 7 May 2007 07:29:16 -0700 (PDT)
Subject: [cisco-voip] FAX relay vs. FAX passthrough
In-Reply-To: <463F3547.9000800@smalltime.com>
Message-ID: <881917.32110.qm@web57207.mail.re3.yahoo.com>

I'm soliciting for opinions on using FAX relay as
opposed to FAX passthrough.

We're using G.729 for FAX on a sizable cluster and
would like to get FAX to work using this codec. I know
that passthrough requires G.711 to work so I'd like to
hear on any opinions for anyone that has successfully
configured relay. 

I'm changing the changing the protocol to Cisco FAX
under voice service voip configuration mode. I'm not
looking forward to configuring several VG248s for FAX
relay either but I'll do whatever it takes.

Paul


 
____________________________________________________________________________________
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Get new email alerts with the free Yahoo! Toolbar.
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From jonvoip at gmail.com  Mon May  7 11:58:48 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 7 May 2007 10:58:48 -0500
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
	<C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0705070858o1b8db828ic6634ddd0bd46800@mail.gmail.com>

This is NOT true.

I just did an upgrade (3.3 to 5.1)... and we built a new server with CCM3.3,
restored the backup (in a lab, no phones present), upgraded to 4.1, ran DMA
and upgraded to 5.1 and restored the DMA data.

The license was for all of the phones configured (zero phones were
registered at the time DMA was run), and the customer had a full upgrade
license for all configured phones.



Jonathan

On 5/7/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
>
> Actually, this is not the case.  It will only generate DLUs for phones
> that are registered with the CallManager server.  It will skip any
> device that is not registered and not count it in the DLU list.
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Christopher M.
> Bomba
> Sent: Wednesday, May 02, 2007 9:05 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
> Who said you had to buy the phones.  You can just put phones in with
> bogus MAC addresses and it will generate DLUs for those.  I thought they
> had to be registered but turns out that they just have to be present in
> the system.  You can create IP communicators or any other phone to boost
> the DLUs.  That is a big problem since anybody can do this.
>
> Chris
>
> --original message--
>
> thanks Ed for your reply, yes you are right I contacted Partner help and
> they told me that during the upgrade a file will be generated that when
> given to cisco they will give us back the DLUs equivalent to what
> configured on the system. I think this is an illegal workaround for the
> lic. issue as by that someone could buy phones without lic. and then
> setup them with CM 4.1 and then upgrade and he will be having a DLUs for
> free. I think Cisco has to do something to solve this.
>
> Thanks and Best Regards
>
> Ahmed A. Elnagar
> Network Field Engineer
>
> Advanced Computer Technology (ACT)
> 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> Postal Code:12411 Cairo Egypt
>
> Mob: +2010-2833868
> Website: www.act-eg.com
> E-mail: aelnagar at act-eg.com
>
> ________________________________
>
> From: Ed Leatherman [mailto:ealeatherman at gmail.com]
> Sent: Wed 02-May-07 3:16 PM
> To: Ahmed Elnagar
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
>
> hi Ahmed,
>
> When you upgrade, callmanager will generate a temporary license for all
> the devices you have configured from 4.1. If I understand correctly, you
> just have to send that off to cisco's licensing group and have it
> converted to a real 5.2 license with the appropriate number of DLUs. If
> its anything like Unity licensing, the turn-around is pretty quick. I'm
> assuming you need a SASU agreement for callmanager for them to do it.
>
> We are getting ready to do the same process here this summer with our
> 4.1 system.
>
> Hope that helps.
>
> Ed
>
>
>
> On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>         Hello all;
>
>         A customer wants to upgrade from CM 4.1 to 5.1, what I am
> worried about is the licsences issue. How will I get the DLUs for
> callmanager 5.1 we have a lic. with each phone but how we will get the
> DLUs??
>
>
>         Thanks and Best Regards
>
>         Ahmed A. Elnagar
>         Network Field Engineer
>
>         Advanced Computer Technology (ACT)
>         16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
>         Postal Code:12411 Cairo Egypt
>
>         Mob : +2010-2833868
>         Website: www.act-eg.com <http://www.act-eg.com/>
>         E-mail: aelnagar at act-eg.com
>
>         _______________________________________________
>         cisco-voip mailing list
>         cisco-voip at puck.nether.net
>         https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> -------------- next part --------------
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> URL:
> https://puck.nether.net/pipermail/cisco-voip/attachments/20070502/c662e8
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>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> End of cisco-voip Digest, Vol 51, Issue 4
> *****************************************
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
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From Scott.Voll at wesd.org  Mon May  7 12:04:21 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 7 May 2007 09:04:21 -0700
Subject: [cisco-voip] SmartNet or Not
In-Reply-To: <6101bda50705070514m30909626q5898df4a38905a81@mail.gmail.com>
References: <6101bda50705070514m30909626q5898df4a38905a81@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE70D@Apollo.wesd.org>

We only smartnet things that if they go bad we would be "up a creek with
out a paddle"  phones --> No, Access Switches --> No, Distribution /
core  switches / routers we do.  We also keep SmartNet on all Software
and VGW's, But we also keep spares on things we don't smartnet.

 

Hope that helps.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Monday, May 07, 2007 5:14 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SmartNet or Not

 

Hey everyone!

I just got one of those weird "third party" mailings that it was time to
renew my SmartNet.  Sort of like those ones you get about renewing your
domain for $75 when you can do it anywhere else for $15. 

Anyway, it got me to thinking. Do you carry SmartNet on EVERYTHING??
Phones?  All switches?   I am just curious, trying to get an idea as I
look at my smartnet renewal costs!!!!

Todd

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From Matt.Slaga at us.didata.com  Mon May  7 12:12:21 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 7 May 2007 12:12:21 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <5d093f9a0705070858o1b8db828ic6634ddd0bd46800@mail.gmail.com>
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
	<C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>
	<5d093f9a0705070858o1b8db828ic6634ddd0bd46800@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0057BE625@USNAEXCH.na.didata.local>

Interesting.  Cisco was hit hard on this at the UC 6.0 Partner Beta
training, and the IPC BU repeated many times that the phones had to be
registered to be counted in the license file.

 

 

 

From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Monday, May 07, 2007 11:59 AM
To: Matt Slaga (US)
Cc: cbomba at s4nets.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1

 

This is NOT true.

I just did an upgrade (3.3 to 5.1)... and we built a new server with
CCM3.3, restored the backup (in a lab, no phones present), upgraded to
4.1, ran DMA and upgraded to 5.1 and restored the DMA data. 

The license was for all of the phones configured (zero phones were
registered at the time DMA was run), and the customer had a full upgrade
license for all configured phones.



Jonathan

On 5/7/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:

Actually, this is not the case.  It will only generate DLUs for phones
that are registered with the CallManager server.  It will skip any
device that is not registered and not count it in the DLU list.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Christopher M.
Bomba
Sent: Wednesday, May 02, 2007 9:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1

Who said you had to buy the phones.  You can just put phones in with
bogus MAC addresses and it will generate DLUs for those.  I thought they
had to be registered but turns out that they just have to be present in 
the system.  You can create IP communicators or any other phone to boost
the DLUs.  That is a big problem since anybody can do this.

Chris

--original message--

thanks Ed for your reply, yes you are right I contacted Partner help and

they told me that during the upgrade a file will be generated that when
given to cisco they will give us back the DLUs equivalent to what
configured on the system. I think this is an illegal workaround for the
lic. issue as by that someone could buy phones without lic. and then
setup them with CM 4.1 and then upgrade and he will be having a DLUs for
free. I think Cisco has to do something to solve this.

Thanks and Best Regards 

Ahmed A. Elnagar
Network Field Engineer

Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: Ed Leatherman [mailto: ealeatherman at gmail.com
<mailto:ealeatherman at gmail.com> ]
Sent: Wed 02-May-07 3:16 PM
To: Ahmed Elnagar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1


hi Ahmed,

When you upgrade, callmanager will generate a temporary license for all
the devices you have configured from 4.1. If I understand correctly, you
just have to send that off to cisco's licensing group and have it 
converted to a real 5.2 license with the appropriate number of DLUs. If
its anything like Unity licensing, the turn-around is pretty quick. I'm
assuming you need a SASU agreement for callmanager for them to do it. 

We are getting ready to do the same process here this summer with our
4.1 system.

Hope that helps.

Ed



On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com > wrote:

        Hello all;

        A customer wants to upgrade from CM 4.1 to 5.1, what I am
worried about is the licsences issue. How will I get the DLUs for
callmanager 5.1 we have a lic. with each phone but how we will get the 
DLUs??


        Thanks and Best Regards

        Ahmed A. Elnagar
        Network Field Engineer

        Advanced Computer Technology (ACT)
        16 Fawzy Ramah St.Off Shehab St.Mohandessin , Giza, Egypt
        Postal Code:12411 Cairo Egypt

        Mob : +2010-2833868
        Website: www.act-eg.com <http://www.act-eg.com/ >
        E-mail: aelnagar at act-eg.com

        _______________________________________________
        cisco-voip mailing list
         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
        https://puck.nether.net/mailman/listinfo/cisco-voip






--
Ed Leatherman
Senior Voice Engineer 
West Virginia University
Telecommunications and Network Operations
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From aelnagar at ACT-EG.COM  Mon May  7 12:23:02 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Mon, 7 May 2007 19:23:02 +0300
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71><C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>
	<5d093f9a0705070858o1b8db828ic6634ddd0bd46800@mail.gmail.com>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A040@mail.ACT-EG.COM>

So from where did u got the DLUs, did you bought DLUs for the system or what??
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
Sent: Mon 07-May-07 6:58 PM
To: Matt Slaga (US)
Cc: cbomba at s4nets.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1


This is NOT true.

I just did an upgrade (3.3 to 5.1)... and we built a new server with CCM3.3, restored the backup (in a lab, no phones present), upgraded to 4.1, ran DMA and upgraded to 5.1 and restored the DMA data. 

The license was for all of the phones configured (zero phones were registered at the time DMA was run), and the customer had a full upgrade license for all configured phones.



Jonathan


On 5/7/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote: 

	Actually, this is not the case.  It will only generate DLUs for phones
	that are registered with the CallManager server.  It will skip any
	device that is not registered and not count it in the DLU list.
	
	
	
	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
	[mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Christopher M.
	Bomba
	Sent: Wednesday, May 02, 2007 9:05 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
	
	Who said you had to buy the phones.  You can just put phones in with
	bogus MAC addresses and it will generate DLUs for those.  I thought they
	had to be registered but turns out that they just have to be present in 
	the system.  You can create IP communicators or any other phone to boost
	the DLUs.  That is a big problem since anybody can do this.
	
	Chris
	
	--original message--
	
	thanks Ed for your reply, yes you are right I contacted Partner help and 
	they told me that during the upgrade a file will be generated that when
	given to cisco they will give us back the DLUs equivalent to what
	configured on the system. I think this is an illegal workaround for the
	lic. issue as by that someone could buy phones without lic. and then
	setup them with CM 4.1 and then upgrade and he will be having a DLUs for
	free. I think Cisco has to do something to solve this.
	
	Thanks and Best Regards 
	
	Ahmed A. Elnagar
	Network Field Engineer
	
	Advanced Computer Technology (ACT)
	16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
	Postal Code:12411 Cairo Egypt
	
	Mob: +2010-2833868
	Website: www.act-eg.com <http://www.act-eg.com/> 
	E-mail: aelnagar at act-eg.com
	
	________________________________
	
	From: Ed Leatherman [mailto: ealeatherman at gmail.com <mailto:ealeatherman at gmail.com> ]
	Sent: Wed 02-May-07 3:16 PM
	To: Ahmed Elnagar
	Cc: cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
	
	
	hi Ahmed,
	
	When you upgrade, callmanager will generate a temporary license for all
	the devices you have configured from 4.1. If I understand correctly, you
	just have to send that off to cisco's licensing group and have it 
	converted to a real 5.2 license with the appropriate number of DLUs. If
	its anything like Unity licensing, the turn-around is pretty quick. I'm
	assuming you need a SASU agreement for callmanager for them to do it. 
	
	We are getting ready to do the same process here this summer with our
	4.1 system.
	
	Hope that helps.
	
	Ed
	
	
	
	On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com > wrote:
	
	        Hello all;
	
	        A customer wants to upgrade from CM 4.1 to 5.1, what I am
	worried about is the licsences issue. How will I get the DLUs for
	callmanager 5.1 we have a lic. with each phone but how we will get the 
	DLUs??
	
	
	        Thanks and Best Regards
	
	        Ahmed A. Elnagar
	        Network Field Engineer
	
	        Advanced Computer Technology (ACT)
	        16 Fawzy Ramah St.Off Shehab St.Mohandessin , Giza, Egypt
	        Postal Code:12411 Cairo Egypt
	
	        Mob : +2010-2833868
	        Website: www.act-eg.com <http://www.act-eg.com/>  <http://www.act-eg.com/ >
	        E-mail: aelnagar at act-eg.com
	
	        _______________________________________________
	        cisco-voip mailing list
	         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
	        https://puck.nether.net/mailman/listinfo/cisco-voip
	
	
	
	
	
	
	--
	Ed Leatherman
	Senior Voice Engineer 
	West Virginia University
	Telecommunications and Network Operations
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	*****************************************
	
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From CarlosOrtiz at bayviewfinancial.com  Mon May  7 12:29:39 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Mon, 7 May 2007 12:29:39 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0057BE625@USNAEXCH.na.didata.local>
Message-ID: <OFE4AEA40C.62F5FCE5-ON852572D4.005A8008-852572D4.005A9AAD@bftg.com>

Could it be that phones that were at least registered before and now 
offline are ok, but phones that are "not found" meaning never registered 
to the system are not?

Carlos



"Matt Slaga \(US\)" <Matt.Slaga at us.didata.com> 
Sent by: cisco-voip-bounces at puck.nether.net
05/07/2007 12:12 PM

To
"Jonathan Charles" <jonvoip at gmail.com>
cc
cbomba at s4nets.com, cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1






Interesting.  Cisco was hit hard on this at the UC 6.0 Partner Beta 
training, and the IPC BU repeated many times that the phones had to be 
registered to be counted in the license file.
 
 
 
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Monday, May 07, 2007 11:59 AM
To: Matt Slaga (US)
Cc: cbomba at s4nets.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
 
This is NOT true.

I just did an upgrade (3.3 to 5.1)... and we built a new server with 
CCM3.3, restored the backup (in a lab, no phones present), upgraded to 
4.1, ran DMA and upgraded to 5.1 and restored the DMA data. 

The license was for all of the phones configured (zero phones were 
registered at the time DMA was run), and the customer had a full upgrade 
license for all configured phones.



Jonathan
On 5/7/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
Actually, this is not the case.  It will only generate DLUs for phones
that are registered with the CallManager server.  It will skip any
device that is not registered and not count it in the DLU list.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Christopher M.
Bomba
Sent: Wednesday, May 02, 2007 9:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1

Who said you had to buy the phones.  You can just put phones in with
bogus MAC addresses and it will generate DLUs for those.  I thought they
had to be registered but turns out that they just have to be present in 
the system.  You can create IP communicators or any other phone to boost
the DLUs.  That is a big problem since anybody can do this.

Chris

--original message--

thanks Ed for your reply, yes you are right I contacted Partner help and 
they told me that during the upgrade a file will be generated that when
given to cisco they will give us back the DLUs equivalent to what
configured on the system. I think this is an illegal workaround for the
lic. issue as by that someone could buy phones without lic. and then
setup them with CM 4.1 and then upgrade and he will be having a DLUs for
free. I think Cisco has to do something to solve this.

Thanks and Best Regards 

Ahmed A. Elnagar
Network Field Engineer

Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: Ed Leatherman [mailto: ealeatherman at gmail.com]
Sent: Wed 02-May-07 3:16 PM
To: Ahmed Elnagar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1


hi Ahmed,

When you upgrade, callmanager will generate a temporary license for all
the devices you have configured from 4.1. If I understand correctly, you
just have to send that off to cisco's licensing group and have it 
converted to a real 5.2 license with the appropriate number of DLUs. If
its anything like Unity licensing, the turn-around is pretty quick. I'm
assuming you need a SASU agreement for callmanager for them to do it. 

We are getting ready to do the same process here this summer with our
4.1 system.

Hope that helps.

Ed



On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com > wrote:

        Hello all;

        A customer wants to upgrade from CM 4.1 to 5.1, what I am
worried about is the licsences issue. How will I get the DLUs for
callmanager 5.1 we have a lic. with each phone but how we will get the 
DLUs??


        Thanks and Best Regards

        Ahmed A. Elnagar
        Network Field Engineer

        Advanced Computer Technology (ACT)
        16 Fawzy Ramah St.Off Shehab St.Mohandessin , Giza, Egypt
        Postal Code:12411 Cairo Egypt

        Mob : +2010-2833868
        Website: www.act-eg.com <http://www.act-eg.com/ >
        E-mail: aelnagar at act-eg.com

        _______________________________________________
        cisco-voip mailing list
         cisco-voip at puck.nether.net
        https://puck..nether.net/mailman/listinfo/cisco-voip






--
Ed Leatherman
Senior Voice Engineer 
West Virginia University
Telecommunications and Network Operations
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https://puck..nether.net/mailman/listinfo/cisco-voip


End of cisco-voip Digest, Vol 51, Issue 4 
*****************************************

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-----------------------------------------
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From aelnagar at ACT-EG.COM  Mon May  7 12:35:14 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Mon, 7 May 2007 19:35:14 +0300
Subject: [cisco-voip] FAX relay vs. FAX passthrough
References: <881917.32110.qm@web57207.mail.re3.yahoo.com>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A041@mail.ACT-EG.COM>

this is a sample dial-peer that I used to relay faxes from PSTN to a FAX server thorugh an H323 gateway
 
dial-peer voice 345 voip
 description Test_Fax
 destination-pattern 345
 session protocol sipv2
 session target ipv4:x.x.x.x
 fax-relay ecm disable
 dtmf-relay rtp-nte
 codec g711ulaw
 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
 no vad
 
here are some of my notes:
 
*Regarding CM 4.x fax relay with T.38 protocol is supported in H323 mode only no T.38 with sip is supported (that is why I made the dial-peer on the gateway as my FAX server only supports T.38 with SIP)
 
*Remember ro disable the ecm function under the dial-peer (the highlighted command) because this function sometimes makes problems with transmission of the fax.
*I didnot try using G.729 codec but it works fine with G.711 codec with me.
 
A very useful command for debugging is debug voice call fax it will be very helpful with you (and I think it is the only debug command available for FAX debugguing)
 
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Paul Choi
Sent: Mon 07-May-07 5:29 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] FAX relay vs. FAX passthrough



I'm soliciting for opinions on using FAX relay as
opposed to FAX passthrough.

We're using G.729 for FAX on a sizable cluster and
would like to get FAX to work using this codec. I know
that passthrough requires G.711 to work so I'd like to
hear on any opinions for anyone that has successfully
configured relay.

I'm changing the changing the protocol to Cisco FAX
under voice service voip configuration mode. I'm not
looking forward to configuring several VG248s for FAX
relay either but I'll do whatever it takes.

Paul



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From akoul at dataways.gr  Mon May  7 12:37:36 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Mon, 07 May 2007 19:37:36 +0300
Subject: [cisco-voip] How to immediate divert a call to a specific Ext but
 different from VMail Pilot?
Message-ID: <463F55D0.9080101@dataways.gr>

How is it possible to do that?
I want to use the immediate divert to send it to another extension

Thanks
Anthony

From jgrace at adcapnet.com  Mon May  7 13:51:24 2007
From: jgrace at adcapnet.com (James Grace)
Date: Mon, 7 May 2007 13:51:24 -0400
Subject: [cisco-voip] Unable to fwd message to another subscriber in unity
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C64F@mail.adcapnetsvc.local>

?
for some reason we are Unable to fwd message to another subscriber in unity.  we are running unity 4.2 unified /exchange 2003.  I'm not sure where to start trouble shooting this problem. 
 
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
<http://www.adcapnet.com/>  
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From baryonyx5 at gmail.com  Mon May  7 14:00:47 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Mon, 7 May 2007 13:00:47 -0500
Subject: [cisco-voip] Cannot transfer an off net call to another codec
	through an h323 gw
In-Reply-To: <463B41E6.3040306@dataways.gr>
References: <463B41E6.3040306@dataways.gr>
Message-ID: <a749d0620705071100w56db2880k6bd274815f72058@mail.gmail.com>

Do you see anything in the CM event logs about an unavailable MTP resource?


Also, enabling slow start on the gateway often solves problems like this.
It's a setting under voice services voip -> h323 I think.

On 5/4/07, Anthony Kouloglou <akoul at dataways.gr> wrote:
>
> Hi,
> i have problems in transfering an incoming call established with g711
> codec, to another device that uses g729.
> So, there is a CCM 5.X with extensions 1XXX and 2XXX for example.
>
> 1.VGW (H323)
> 2. ExtA 1000
>    ExtB 1001
>
> 3  ExtC 2000
>
> A, B and voicegw are in the same devicepool and use g711.
> C is in another device pool that use g729 for calls to the other device
> pool.
>
> GW has 2 different voip dial-peers:
> one with g711 to connect with extensions A, B, and
> one with g729 to connect with C
> and one pots dial-peer to call outside to the TELCO numbers
>
> Incoming calls to vgw from the telco, work fine either to extensions A,B
> or C directly (depending on the dial-peer, it uses the appropriate codec).
> When A, calls B directly, B can tranfer the call to C
> When A, receives a call from the Telco, cannot tranfer it to C: the call
> drops and busy tone is played to both ends.
>
> Does, h323 needs something special for this config?
>
> thanks
> anthony
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
kris seraphine
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From kdupree at rmwc.edu  Mon May  7 14:14:10 2007
From: kdupree at rmwc.edu (Kathy Dupree)
Date: Mon, 7 May 2007 14:14:10 -0400
Subject: [cisco-voip] FW: MotoQ
Message-ID: <6E497ADB607656479C24E6D7BF6B505A082828BF@exchange.rmwc.edu>

 

I was asked today by a user if there is any way for the MotoQ to read
the voicemail emails that unity sends out.  I know people with
Blackberries have to buy software but I don't know anything about the
MotoQ.  Does anyone know?  Thanks

 

Katherine O. Dupree

Randolph-Macon Woman's College

Main 243

Telecommunications Support/Infrastructure Coordinator

kdupree at rmwc.edu

(434) 947 8371

 

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From fwn at feasible.net  Mon May  7 14:15:40 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Mon, 7 May 2007 11:15:40 -0700
Subject: [cisco-voip] Unable to fwd message to another subscriber in
	unity
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C64F@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C64F@mail.adcapnetsvc.local>
Message-ID: <ca09e3750705071115pfc9f5b7i4f41a316584db1c8@mail.gmail.com>

James, can you be a little more specific?  From what interface are you
trying to forward from (Outlook or phone)?  Where to (subscriber on same
server or different server)?  What happens when you try?

-Fred Nielsen

On 5/7/07, James Grace <jgrace at adcapnet.com> wrote:
>
> ?  for some reason we are Unable to fwd message to another subscriber in
> unity.  we are running unity 4.2 unified /exchange 2003.  I'm not sure
> where to start trouble shooting this problem.
>
>
>  James Grace
>
> Systems Engineer
> Adcap Network Systems, Inc.
> 6865 Shiloh Road East, Suite 250
> Alpharetta, GA 30005
>
>  <http://www.adcapnet.com/>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From rratliff at cisco.com  Mon May  7 14:35:43 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 7 May 2007 14:35:43 -0400
Subject: [cisco-voip] How to immediate divert a call to a specific Ext
	but different from VMail Pilot?
In-Reply-To: <463F55D0.9080101@dataways.gr>
References: <463F55D0.9080101@dataways.gr>
Message-ID: <5B7EA966-E807-4059-B3AA-47AA28C55512@cisco.com>

Can't do it.  The iDivert feature redirects to your vm pilot.  You  
can trick it by setting the vm pilot to point to your other extension  
but then the messages button on the phone would also ring that other  
extension.

-Ryan

On May 7, 2007, at 12:37 PM, Anthony Kouloglou wrote:

How is it possible to do that?
I want to use the immediate divert to send it to another extension

Thanks
Anthony
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From GRhodes at rbh.com  Mon May  7 14:47:38 2007
From: GRhodes at rbh.com (Rhodes, Geoff)
Date: Mon, 7 May 2007 14:47:38 -0400
Subject: [cisco-voip] FW: MotoQ
In-Reply-To: <6E497ADB607656479C24E6D7BF6B505A082828BF@exchange.rmwc.edu>
References: <6E497ADB607656479C24E6D7BF6B505A082828BF@exchange.rmwc.edu>
Message-ID: <6350516B6159444E9D646523EB4526F403EB9FC3@apollo.rbh.local>

I believe there's a Windows Mobile version of The Core Pocket Media
Player which works.
http://tcpmp.corecodec.org/about
 
Geoff Rhodes
Director, Information Technology
Robinson, Bradshaw & Hinson, P.A.
704 / 377-8188
grhodes at rbh.com
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kathy Dupree
Sent: Monday, May 07, 2007 2:14 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] FW: MotoQ



 

I was asked today by a user if there is any way for the MotoQ to read
the voicemail emails that unity sends out.  I know people with
Blackberries have to buy software but I don't know anything about the
MotoQ.  Does anyone know?  Thanks

 

Katherine O. Dupree

Randolph-Macon Woman's College

Main 243

Telecommunications Support/Infrastructure Coordinator

kdupree at rmwc.edu

(434) 947 8371

 

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From lelio at uoguelph.ca  Mon May  7 15:34:46 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 7 May 2007 15:34:46 -0400
Subject: [cisco-voip] 7911 - better or worse?
Message-ID: <011b01c790de$c4e240d0$47196883@cfs.uoguelph.ca>

So I'm reviewing the 7911 and I don't like it so far. You can't force a background image and it looks like you need a CCNA just to reset it back to factory defaults. I'm sure someone was dying to put in "Step 4: Make sure you are hopping on one foot"


------------------------------------------------------------------------------
  To perform a factory reset of a phone, follow these steps: 

  Procedure 

  Step 1 Unplug the power cable from the phone and then plug it back in. 

  The phone begins its power up cycle. 

  Step 2 While the phone is powering up, and before the Applications Menu button flashes on and off, press and hold #. 

  Continue to hold # until the message LED on the handset flashes on and off in sequence in red. 

  Step 3 Release # and press 123456789*0#. 

  You can press a key twice in a row, but if you press the keys out of sequence, the factory reset will not take place. 

  After you press these keys, the message LED on the handset flashes faster in red and the phone goes through the factory reset process. 

  Do not power down the phone until it completes the factory reset process and the main screen appears. 


------------------------------------------------------------------------------


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From CarlosOrtiz at bayviewfinancial.com  Mon May  7 15:45:21 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Mon, 7 May 2007 15:45:21 -0400
Subject: [cisco-voip] CSS precedence
Message-ID: <OFEE509754.D17B86BC-ON852572D4.00680C24-852572D4.006C857E@bftg.com>

I am testing a remote location gateway by adding a DID from that location 
to my phone.  I assigned the remote site CSS to the remote line and also 
have a CSS assigned to my phone.  When I take the REMOTE LINE offhook and 
dial, it is exiting my local gateway.   In previous experience I have done 
this and it will exit the remote gateway per CSS precedence rules.  Am I 
missing something here or is this a bug?

Carlos
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From CarlosOrtiz at bayviewfinancial.com  Mon May  7 15:49:33 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Mon, 7 May 2007 15:49:33 -0400
Subject: [cisco-voip] CSS precedence
In-Reply-To: <OFEE509754.D17B86BC-ON852572D4.00680C24-852572D4.006C857E@bftg.com>
Message-ID: <OF5507F31D.C29CBCC4-ON852572D4.006CDA5A-852572D4.006CE818@bftg.com>

Meant to say CallMgr 4.1.3 (sr2)




CarlosOrtiz at bayviewfinancial.com 
Sent by: cisco-voip-bounces at puck.nether.net
05/07/2007 03:45 PM

To
cisco-voip at puck.nether.net
cc

Subject
[cisco-voip] CSS precedence







I am testing a remote location gateway by adding a DID from that location 
to my phone.  I assigned the remote site CSS to the remote line and also 
have a CSS assigned to my phone.  When I take the REMOTE LINE offhook and 
dial, it is exiting my local gateway.   In previous experience I have done 
this and it will exit the remote gateway per CSS precedence rules.  Am I 
missing something here or is this a bug? 

Carlos_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From Matt.Slaga at us.didata.com  Mon May  7 15:57:38 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 7 May 2007 15:57:38 -0400
Subject: [cisco-voip] 7911 - better or worse?
In-Reply-To: <011b01c790de$c4e240d0$47196883@cfs.uoguelph.ca>
References: <011b01c790de$c4e240d0$47196883@cfs.uoguelph.ca>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005800D02@USNAEXCH.na.didata.local>

I've deployed these for several customers (along with the 7905s, 7912s
and 7910s in the old days) who thought they would save a few bucks by
giving the general population cheap phones to save a few bucks.  

 

"Excuse me, but I need to remove your Avaya with the nice big display
and change it out with this cheap Cisco phone".   All the customers that
attempted this regretted it after the fact.  End user acceptance was in
the dumps.  "It's a one-way speakerphone" or "why the heck is the hold
button a big red stop sign?" or "I don't have voicemail, why can't you
get rid of that messages softkey?" or many other wonderful comments.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Monday, May 07, 2007 3:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7911 - better or worse?

 

So I'm reviewing the 7911 and I don't like it so far. You can't force a
background image and it looks like you need a CCNA just to reset it back
to factory defaults. I'm sure someone was dying to put in "Step 4: Make
sure you are hopping on one foot"

 

________________________________

	To perform a factory reset of a phone, follow these steps: 

	Procedure 

	Step 1  Unplug the power cable from the phone and then plug it
back in. 

	The phone begins its power up cycle. 

	Step 2 While the phone is powering up, and before the
Applications Menu button flashes on and off, press and hold #. 

	Continue to hold # until the message LED on the handset flashes
on and off in sequence in red. 

	Step 3 Release # and press 123456789*0#. 

	You can press a key twice in a row, but if you press the keys
out of sequence, the factory reset will not take place. 

	After you press these keys, the message LED on the handset
flashes faster in red and the phone goes through the factory reset
process. 

	Do not power down the phone until it completes the factory reset
process and the main screen appears. 

________________________________

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
....there's no such thing as a bad timbit...



-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
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From joerg.wesely at intact-is.com  Mon May  7 16:01:31 2007
From: joerg.wesely at intact-is.com (=?ISO-8859-15?Q?J=F6rg_Wesely?=)
Date: Mon, 07 May 2007 22:01:31 +0200
Subject: [cisco-voip] CCM 4.2: Configure "monitor"-Buttons on 7910
Message-ID: <f1o0jm$du5$1@sea.gmane.org>

Hi,

on CCME, I can configure "Monitor"-Buttons, that means a 
speeddial-button that also gets lit when the line is in use.

Is it possible to achieve the same on a CCM 4.2 system?

Regards

J?rg


From joerg.wesely at intact-is.com  Mon May  7 16:10:37 2007
From: joerg.wesely at intact-is.com (=?ISO-8859-15?Q?J=F6rg_Wesely?=)
Date: Mon, 07 May 2007 22:10:37 +0200
Subject: [cisco-voip] CCM 4.2: Directory for external numbers
Message-ID: <f1o14o$gvv$1@sea.gmane.org>

Hi,

I would like to have an additional directory for external Numbers, like 
the way it can be done on CCME.
Is it possible to get something like this with CCM 4.2?

Would it also be possible to let the phone display the caller's name 
instead of his number when he is listed in this directory? (the way 
every  mobile phone can do it...)

thanks in advance

J?rg


From Scott.Voll at wesd.org  Mon May  7 16:24:28 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 7 May 2007 13:24:28 -0700
Subject: [cisco-voip] CSS precedence
In-Reply-To: <OF5507F31D.C29CBCC4-ON852572D4.006CDA5A-852572D4.006CE818@bftg.com>
References: <OFEE509754.D17B86BC-ON852572D4.00680C24-852572D4.006C857E@bftg.com>
	<OF5507F31D.C29CBCC4-ON852572D4.006CDA5A-852572D4.006CE818@bftg.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE721@Apollo.wesd.org>

Lines + phone CSS = Device CSS.  Check the phone CSS to not include the
Partitions you want each phone line to be able to go out.

 

Scott

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: Monday, May 07, 2007 12:50 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CSS precedence

 


Meant to say CallMgr 4.1.3 (sr2) 




CarlosOrtiz at bayviewfinancial.com 
Sent by: cisco-voip-bounces at puck.nether.net 

05/07/2007 03:45 PM 

To

cisco-voip at puck.nether.net 

cc

 

Subject

[cisco-voip] CSS precedence

 

 

 





I am testing a remote location gateway by adding a DID from that
location to my phone.  I assigned the remote site CSS to the remote line
and also have a CSS assigned to my phone.  When I take the REMOTE LINE
offhook and dial, it is exiting my local gateway.   In previous
experience I have done this and it will exit the remote gateway per CSS
precedence rules.  Am I missing something here or is this a bug? 

Carlos_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From lelio at uoguelph.ca  Mon May  7 16:27:08 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 7 May 2007 16:27:08 -0400
Subject: [cisco-voip] CSS precedence
References: <OFEE509754.D17B86BC-ON852572D4.00680C24-852572D4.006C857E@bftg.com><OF5507F31D.C29CBCC4-ON852572D4.006CDA5A-852572D4.006CE818@bftg.com>
	<407055A92CECCB499C922A2D35FC19A608DEE721@Apollo.wesd.org>
Message-ID: <018401c790e6$15a7ba20$47196883@cfs.uoguelph.ca>

and forwarding CSS = forwarding CSS. no concatenation at all. that would make sense. ;)

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Voll, Scott 
  To: CarlosOrtiz at bayviewfinancial.com ; cisco-voip at puck.nether.net 
  Sent: Monday, May 07, 2007 4:24 PM
  Subject: Re: [cisco-voip] CSS precedence


  Lines + phone CSS = Device CSS.  Check the phone CSS to not include the Partitions you want each phone line to be able to go out.

   

  Scott

   

   


------------------------------------------------------------------------------

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of CarlosOrtiz at bayviewfinancial.com
  Sent: Monday, May 07, 2007 12:50 PM
  To: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] CSS precedence

   


  Meant to say CallMgr 4.1.3 (sr2) 




        CarlosOrtiz at bayviewfinancial.com 
        Sent by: cisco-voip-bounces at puck.nether.net 

        05/07/2007 03:45 PM 
       To
             cisco-voip at puck.nether.net 
             
              cc
              
             
              Subject
             [cisco-voip] CSS precedence
             

         

               
              
             

       





  I am testing a remote location gateway by adding a DID from that location to my phone.  I assigned the remote site CSS to the remote line and also have a CSS assigned to my phone.  When I take the REMOTE LINE offhook and dial, it is exiting my local gateway.   In previous experience I have done this and it will exit the remote gateway per CSS precedence rules.  Am I missing something here or is this a bug? 

  Carlos_______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From BRiley at jackhenry.com  Mon May  7 16:28:57 2007
From: BRiley at jackhenry.com (Bill Riley III)
Date: Mon, 7 May 2007 15:28:57 -0500
Subject: [cisco-voip] CCM 4.2: Directory for external numbers
In-Reply-To: <f1o14o$gvv$1@sea.gmane.org>
References: <f1o14o$gvv$1@sea.gmane.org>
Message-ID: <2B3745BC9D31514381A6FCC5E72D9373016249DC@SMOMAIL.jhacorp.com>

You can take the same XML file from CCME and create a website on Unity, or another server, then drop that XML file into the website. Then create an IP phone server that points the URL of the XML file.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of J?rg Wesely
Sent: Monday, May 07, 2007 3:11 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM 4.2: Directory for external numbers

Hi,

I would like to have an additional directory for external Numbers, like 
the way it can be done on CCME.
Is it possible to get something like this with CCM 4.2?

Would it also be possible to let the phone display the caller's name 
instead of his number when he is listed in this directory? (the way 
every  mobile phone can do it...)

thanks in advance

J?rg

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Mon May  7 16:31:35 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 7 May 2007 16:31:35 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
Message-ID: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>

Do I need a devpack to run my 7911s? Or can I just download the zip files? What do the devpacks do?

The readme for 8.2(2) SR2 says the following. But the 7911 is running fine with SCCP11.7-2-3S (whatever that is)


  http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.8-2-2SR2-readme.htm

  Table 1: Compatible Cisco Unified Call Manager Release for 7911G (No support for 7906G in these releases)

      Call Manager Version
     Executable on Software Page
     
      3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe
     ciscocm.3-3-5-sr1.exe; ciscocm.3-3-DevPack-64.exe
     
      4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe 
     ciscocm.4-0-2a-sr2b.exe; ciscocm.4-0-DevPack-43.exe
     
      4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe
     ciscocm.4-1-3-sr1.exe; ciscocm.4-1-DevPack-23.exe
     

 Table 2: Compatible Cisco Unified Call Manager Release that supports both 7906G and 7911G

      Call Manager Version
     Executable on Software Page
     
      3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later
     ciscocm.3-3-5-sr2.exe; ciscocm.3-3-DevPack-68.1.exe
     
      4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later
     ciscocm.4-1-3-sr3a.exe; ciscocm.4-1-DevPack-28.2.exe
     
      4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later
     ciscocm.4-2-1-sr1.exe; ciscocm.4-2-1-DevPack-03.3.exe
     



--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From Scott.Voll at wesd.org  Mon May  7 16:45:36 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 7 May 2007 13:45:36 -0700
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
In-Reply-To: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE725@Apollo.wesd.org>

Personally I like the Zip.  I unpackage them, insert, bounce the TFTP,
setup the device defaults and I'm good.  The EXE requires a reboot.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Monday, May 07, 2007 1:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] devpack vs .EXE vs. ZIP

 

Do I need a devpack to run my 7911s? Or can I just download the zip
files? What do the devpacks do?

 

The readme for 8.2(2) SR2 says the following. But the 7911 is running
fine with SCCP11.7-2-3S (whatever that is)

 

 

	
http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.8-2
-2SR2-readme.htm

	 

	Table 1: Compatible Cisco Unified Call Manager Release for 7911G
(No support for 7906G in these releases)

	 

Call Manager Version

Executable on Software Page

3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe

ciscocm.3-3-5-sr1.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A>
; ciscocm.3-3-DevPack-64.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=show
C2A> 

4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe 

ciscocm.4-0-2a-sr2b.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&st
atus=showC2A> ; ciscocm.4-0-DevPack-43.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild
&status=showC2A> 

4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe

ciscocm.4-1-3-sr1.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&stat
us=showC2A> ; ciscocm.4-1-DevPack-23.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild
&status=showC2A> 

Table 2: Compatible Cisco Unified Call Manager Release that supports
both 7906G and 7911G

Call Manager Version

Executable on Software Page

3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later

ciscocm.3-3-5-sr2.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A>
; ciscocm.3-3-DevPack-68.1.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=sh
owC2A> 

4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later

ciscocm.4-1-3-sr3a.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&sta
tus=showC2A> ; ciscocm.4-1-DevPack-28.2.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebui
ld&status=showC2A> 

4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later

ciscocm.4-2-1-sr1.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&stat
us=showC2A> ; ciscocm.4-2-1-DevPack-03.3.exe
<http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco
/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tableb
uild&status=showC2A> 

 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

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From lelio at uoguelph.ca  Mon May  7 16:47:26 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 7 May 2007 16:47:26 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>
	<407055A92CECCB499C922A2D35FC19A608DEE725@Apollo.wesd.org>
Message-ID: <01c401c790e8$ecca45c0$47196883@cfs.uoguelph.ca>

i've never had to bounce the tFTP server after copying over ZIP files. 

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Voll, Scott 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Monday, May 07, 2007 4:45 PM
  Subject: RE: [cisco-voip] devpack vs .EXE vs. ZIP


  Personally I like the Zip.  I unpackage them, insert, bounce the TFTP, setup the device defaults and I'm good.  The EXE requires a reboot.

   

  Scott

   


------------------------------------------------------------------------------

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Monday, May 07, 2007 1:32 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] devpack vs .EXE vs. ZIP

   

  Do I need a devpack to run my 7911s? Or can I just download the zip files? What do the devpacks do?

   

  The readme for 8.2(2) SR2 says the following. But the 7911 is running fine with SCCP11.7-2-3S (whatever that is)

   

   

    http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.8-2-2SR2-readme.htm

     

    Table 1: Compatible Cisco Unified Call Manager Release for 7911G (No support for 7906G in these releases)

     

        Call Manager Version
       Executable on Software Page
       
        3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe
       ciscocm.3-3-5-sr1.exe; ciscocm.3-3-DevPack-64.exe
       
        4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe 
       ciscocm.4-0-2a-sr2b.exe; ciscocm.4-0-DevPack-43.exe
       
        4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe
       ciscocm.4-1-3-sr1.exe; ciscocm.4-1-DevPack-23.exe
       

  Table 2: Compatible Cisco Unified Call Manager Release that supports both 7906G and 7911G

        Call Manager Version
       Executable on Software Page
       
        3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later
       ciscocm.3-3-5-sr2.exe; ciscocm.3-3-DevPack-68.1.exe
       
        4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later
       ciscocm.4-1-3-sr3a.exe; ciscocm.4-1-DevPack-28.2.exe
       
        4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later
       ciscocm.4-2-1-sr1.exe; ciscocm.4-2-1-DevPack-03.3.exe
       

   

   

  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  ...there's no such thing as a bad timbit...
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From dpriebe at theriverbank.com  Mon May  7 17:31:46 2007
From: dpriebe at theriverbank.com (Duane Priebe)
Date: Mon, 7 May 2007 16:31:46 -0500
Subject: [cisco-voip] Phones in a different device pools can't transfer to
	an extension setup as a IPCC pilot point
Message-ID: <43D799CBE4331D44999AB7C9CD67F30202B8A50A@NTEXCHANGE.riverbank.com>


I have an issue where phones that are in different device pools (except
one) are unable to dial an extension setup to be a pilot point for a
IPCC contact center queue.  They just get a pause then a busy.  Can
anyone point me in the right direction as to what I'm missing?  Thanks!

Duane
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From Scott.Voll at wesd.org  Mon May  7 17:35:08 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 7 May 2007 14:35:08 -0700
Subject: [cisco-voip] Phones in a different device pools can't transfer
	toan extension setup as a IPCC pilot point
In-Reply-To: <43D799CBE4331D44999AB7C9CD67F30202B8A50A@NTEXCHANGE.riverbank.com>
References: <43D799CBE4331D44999AB7C9CD67F30202B8A50A@NTEXCHANGE.riverbank.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE728@Apollo.wesd.org>

Sounds like a codec problem.  IPCCx only works with G711.  setup some
Xcoding resources.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Duane Priebe
Sent: Monday, May 07, 2007 2:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Phones in a different device pools can't transfer
toan extension setup as a IPCC pilot point

 

 

I have an issue where phones that are in different device pools (except
one) are unable to dial an extension setup to be a pilot point for a
IPCC contact center queue.  They just get a pause then a busy.  Can
anyone point me in the right direction as to what I'm missing?  Thanks!

Duane 

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From gary at lefman.co.uk  Mon May  7 17:55:18 2007
From: gary at lefman.co.uk (Gary Lefman)
Date: Mon, 7 May 2007 22:55:18 +0100
Subject: [cisco-voip] Network Locale Installer 4.2.1.3000
In-Reply-To: <mailman.12835.1178548633.1489.cisco-voip@puck.nether.net>
Message-ID: <200705072155.WAA04323@cisco.com>

Reto,

Collect the installation log files from here:

	C:\Program Files\Common Files\Cisco\Logs\LocaleInstaller\

Open a case with Cisco TAC and request the log files be sent to the
localisation team for investigation.

Is the server/was the server configured as a CRS?

Regards,

Labmonkey

----------------------------------------------------------------------
Date: Mon, 07 May 2007 15:58:09 +0200
From: Reto Gassmann <voip at mrga.ch>
Subject: [cisco-voip] Networ Locale Installer 4.2.1.3000
To: cisco-voip at puck.nether.net
Message-ID: <1178546289.463f30713192f at www.mail2web.ch>
Content-Type: text/plain; charset=ISO-8859-1

Hi all

I try to update my network locals on a CCM 4.2(3). However, if I want to
apply the file cm-locale-combined_network-4.2.1.3000.exe I get an error
message that the AutoRegister Phone Prompts are not in the package and the
installation fails.

I choose Germany, France, Italy and Switzerland to install.

Any ideas??

Reto

From pat-cv at wcyv.com  Mon May  7 19:42:50 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Mon, 07 May 2007 19:42:50 -0400
Subject: [cisco-voip] Unable to fwd message to another subscriber in
	unity
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C64F@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C64F@mail.adcapnetsvc.local>
Message-ID: <463FB97A.6030507@wcyv.com>

Sounds like Unity doesn't have send-as permissions. Check the security 
tab on one of the users who can't send messages to other users. Your 
UnityMsgStoreSvc account needs to have send-as permissions there. 
Assuming it doesn't, you need to run permissions wizard.

James Grace wrote:
> ?
> for some reason we are Unable to fwd message to another subscriber in unity.  we are running unity 4.2 unified /exchange 2003.  I'm not sure where to start trouble shooting this problem. 
>  
>  
> James Grace 
>  
> Systems Engineer
> Adcap Network Systems, Inc.
> 6865 Shiloh Road East, Suite 250
> Alpharetta, GA 30005
>  
> <http://www.adcapnet.com/>  
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From bo.vestergaard at btinternet.com  Mon May  7 19:36:06 2007
From: bo.vestergaard at btinternet.com (BO VESTERGAARD)
Date: Tue, 8 May 2007 00:36:06 +0100 (BST)
Subject: [cisco-voip] CCME and TCL scripts
Message-ID: <117437.24531.qm@web86009.mail.ird.yahoo.com>

I have installed my first CCME on a 2811 and everything works fine. I can make calls to Cisco telephones and the PSTN. However, I wanted to write a .tcl script and load it as an application. No matter what I try I cannot get it working. The router refuses to load the script.
   
  In order to keep it simple I have created a very short script (doesn't work either). It goes like this.
   
  proc act_Setup { } {
  puts "--Starting the script--\n"
}
  #-----------------------------------------------
#  State Machine
#-----------------------------------------------
  set fsm(CALL_INIT,ev_setup_indication)    "act_Setup    same_state"
  fsm define fsm CALL_INIT

  When I try to load it I get loads of errors on the debug
   
  yourname#call application voice load test
yourname#
May  7 23:37:09.407: //-1//HIFS:/hifs_ifs_cb: hifs ifs file read succeeded. size=280, url=flash:test.tcl
May  7 23:37:09.411: //-1//HIFS:/hifs_free_idata: hifs_free_idata: 0x46637660
May  7 23:37:09.411: //-1//HIFS:/hifs_hold_idata: hifs_hold_idata: 0x46637660
May  7 23:37:09.411: //-1//AFW_:EE45C869F8000:/Tcl_Link: Linking script test
May  7 23:37:09.415: //-1//PACK:/PkgParamRelock:  ElementGet failed
May  7 23:37:09.415: //-1//PACK:/PkgParamRelock:  ElementGet failed
May  7 23:37:09.415: //-1//PACK:EE45C869F8000:/PkgRequire:  can't find package callindss_xwork
May  7 23:37:09.415: //-1//PACK:EE45C869F8000:/PkgRequire:  can't find package mwi_xwork
May  7 23:37:09.415: //-1//PACK:/PkgParamRelock:  ElementGet failed
May  7 23:37:09.415: //-1//PACK:/PkgParamRelock:  ElementGet failed
May  7 23:37:09.415: //-1//PACK:/PkgParamRelock:  ElementGet failed
May  7 23:37:09.415: //-1//PACK:/PkgParamRelock:  ElementGet failed
May  7 23:37:09.419: //-1//PACK:/PkgParamRelock:  ElementGet failed
May  7 23:37:09.419: //-1//PACK:/PkgParamRelock:  ElementGet failed
May  7 23:37:09.419: //-1//PACK:EE45C869F8000:/PkgRequire:  can't find package callindss_xwork
May  7 23:37:09.419: //-1//PACK:EE45C869F8000:/PkgRequire:  can't find package mwi_xwork
May  7 23:37:09.419: //-1//PACK:/PkgParamRelock:  ElementGet failed
May  7 23:37:09.419: //-1//PACK:/PkgParamRelock:  ElementGet failed

  Another strange thing is that I do not have the debug ivr command in the CLI.
   
  I have tried CME 3.3. and 4.0(2) on different versions of IOS but they all do the same. At the moment I am running c2800nm-ipvoice-mz.124-11.T2.bin
   
  Does anyone have any idea what is going here? Would appreciate a bit of help as I have run out of things to try :(
   
   
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From fwn at feasible.net  Mon May  7 22:09:09 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Mon, 7 May 2007 19:09:09 -0700
Subject: [cisco-voip] CSS precedence
In-Reply-To: <018401c790e6$15a7ba20$47196883@cfs.uoguelph.ca>
References: <OFEE509754.D17B86BC-ON852572D4.00680C24-852572D4.006C857E@bftg.com>
	<OF5507F31D.C29CBCC4-ON852572D4.006CDA5A-852572D4.006CE818@bftg.com>
	<407055A92CECCB499C922A2D35FC19A608DEE721@Apollo.wesd.org>
	<018401c790e6$15a7ba20$47196883@cfs.uoguelph.ca>
Message-ID: <ca09e3750705071909i1dc9d822xbd2103d2cf8248a@mail.gmail.com>

concatenated CSS is line + device.  Partitions in the line CSS are ordered
above the device CSS, but this is only considered if you have an exact
match.  In other words, only when your dialed string matches two identical
patterns is the partition order itself really taken into account.

I suspect what is happening for you is that whatever you are dialing is
matching a more specific pattern within the partitions listed in the device
CSS, thereby bypassing a more widely matched pattern that you are hoping to
match on the line CSS.

Let us know if this makes some sense.

-Fred Nielsen

On 5/7/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

>  and forwarding CSS = forwarding CSS. no concatenation at all. that would
> make sense. ;)
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
>
>  ----- Original Message -----
> *From:* Voll, Scott <Scott.Voll at wesd.org>
> *To:* CarlosOrtiz at bayviewfinancial.com ; cisco-voip at puck.nether.net
> *Sent:* Monday, May 07, 2007 4:24 PM
> *Subject:* Re: [cisco-voip] CSS precedence
>
>
>
> Lines + phone CSS = Device CSS.  Check the phone CSS to not include the
> Partitions you want each phone line to be able to go out.
>
>
>
> Scott
>
>
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *
> CarlosOrtiz at bayviewfinancial.com
> *Sent:* Monday, May 07, 2007 12:50 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] CSS precedence
>
>
>
>
> Meant to say CallMgr 4.1.3 (sr2)
>
>
>   *CarlosOrtiz at bayviewfinancial.com*
> Sent by: cisco-voip-bounces at puck.nether.net
>
> 05/07/2007 03:45 PM
>
> To
>
> cisco-voip at puck.nether.net
>
> cc
>
>
>
> Subject
>
> [cisco-voip] CSS precedence
>
>
>
>
>
>
>
>
>
>
>
> I am testing a remote location gateway by adding a DID from that location
> to my phone.  I assigned the remote site CSS to the remote line and also
> have a CSS assigned to my phone.  When I take the *REMOTE LINE* offhook
> and dial, it is exiting my local gateway.   In previous experience I have
> done this and it will exit the remote gateway per CSS precedence rules.  Am
> I missing something here or is this a bug?
>
> Carlos_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From jonvoip at gmail.com  Tue May  8 00:53:31 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 7 May 2007 23:53:31 -0500
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0057BE625@USNAEXCH.na.didata.local>
References: <39528.192.168.1.71.1178111115.webmail@192.168.1.71>
	<C1FE15183DA37645BC0633BC604E44F0057BDFD3@USNAEXCH.na.didata.local>
	<5d093f9a0705070858o1b8db828ic6634ddd0bd46800@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F0057BE625@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0705072153n1ab6be20tcbd12a5ef327cf6f@mail.gmail.com>

That is absolutely idiotic.

What if you are doing a server replacement for a 3.3 to 6.0 server? How
would you do it???




Jonathan

On 5/7/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
>
>  Interesting.  Cisco was hit hard on this at the UC 6.0 Partner Beta
> training, and the IPC BU repeated many times that the phones had to be
> registered to be counted in the license file.
>
>
>
>
>
>
>
> *From:* Jonathan Charles [mailto:jonvoip at gmail.com]
> *Sent:* Monday, May 07, 2007 11:59 AM
> *To:* Matt Slaga (US)
> *Cc:* cbomba at s4nets.com; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
>
>
> This is NOT true.
>
> I just did an upgrade (3.3 to 5.1)... and we built a new server with
> CCM3.3, restored the backup (in a lab, no phones present), upgraded to 4.1,
> ran DMA and upgraded to 5.1 and restored the DMA data.
>
> The license was for all of the phones configured (zero phones were
> registered at the time DMA was run), and the customer had a full upgrade
> license for all configured phones.
>
>
>
> Jonathan
>
> On 5/7/07, *Matt Slaga (US)* <Matt.Slaga at us.didata.com> wrote:
>
> Actually, this is not the case.  It will only generate DLUs for phones
> that are registered with the CallManager server.  It will skip any
> device that is not registered and not count it in the DLU list.
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Christopher M.
> Bomba
> Sent: Wednesday, May 02, 2007 9:05 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
> Who said you had to buy the phones.  You can just put phones in with
> bogus MAC addresses and it will generate DLUs for those.  I thought they
> had to be registered but turns out that they just have to be present in
> the system.  You can create IP communicators or any other phone to boost
> the DLUs.  That is a big problem since anybody can do this.
>
> Chris
>
> --original message--
>
> thanks Ed for your reply, yes you are right I contacted Partner help and
> they told me that during the upgrade a file will be generated that when
> given to cisco they will give us back the DLUs equivalent to what
> configured on the system. I think this is an illegal workaround for the
> lic. issue as by that someone could buy phones without lic. and then
> setup them with CM 4.1 and then upgrade and he will be having a DLUs for
> free. I think Cisco has to do something to solve this.
>
> Thanks and Best Regards
>
> Ahmed A. Elnagar
> Network Field Engineer
>
> Advanced Computer Technology (ACT)
> 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> Postal Code:12411 Cairo Egypt
>
> Mob: +2010-2833868
> Website: www.act-eg.com
> E-mail: aelnagar at act-eg.com
>
> ________________________________
>
> From: Ed Leatherman [mailto: ealeatherman at gmail.com]
> Sent: Wed 02-May-07 3:16 PM
> To: Ahmed Elnagar
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
>
> hi Ahmed,
>
> When you upgrade, callmanager will generate a temporary license for all
> the devices you have configured from 4.1. If I understand correctly, you
> just have to send that off to cisco's licensing group and have it
> converted to a real 5.2 license with the appropriate number of DLUs. If
> its anything like Unity licensing, the turn-around is pretty quick. I'm
> assuming you need a SASU agreement for callmanager for them to do it.
>
> We are getting ready to do the same process here this summer with our
> 4.1 system.
>
> Hope that helps.
>
> Ed
>
>
>
> On 5/2/07, Ahmed Elnagar <aelnagar at act-eg.com > wrote:
>
>         Hello all;
>
>         A customer wants to upgrade from CM 4.1 to 5.1, what I am
> worried about is the licsences issue. How will I get the DLUs for
> callmanager 5.1 we have a lic. with each phone but how we will get the
> DLUs??
>
>
>         Thanks and Best Regards
>
>         Ahmed A. Elnagar
>         Network Field Engineer
>
>         Advanced Computer Technology (ACT)
>         16 Fawzy Ramah St.Off Shehab St.Mohandessin , Giza, Egypt
>         Postal Code:12411 Cairo Egypt
>
>         Mob : +2010-2833868
>         Website: www.act-eg.com <http://www.act-eg.com/ >
>         E-mail: aelnagar at act-eg.com
>
>         _______________________________________________
>         cisco-voip mailing list
>          cisco-voip at puck.nether.net
>         https://puck..nether.net/mailman/listinfo/cisco-voip<https://puck.nether.net/mailman/listinfo/cisco-voip>
>
>
>
>
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> -------------- next part --------------
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>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck..nether.net/mailman/listinfo/cisco-voip<https://puck.nether.net/mailman/listinfo/cisco-voip>
>
>
> End of cisco-voip Digest, Vol 51, Issue 4
> *****************************************
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck..nether.net/mailman/listinfo/cisco-voip<https://puck.nether.net/mailman/listinfo/cisco-voip>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck..nether.net/mailman/listinfo/cisco-voip
> <https://puck.nether.net/mailman/listinfo/cisco-voip>
>
>
>
> ------------------------------
>
> * Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
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From jonvoip at gmail.com  Tue May  8 00:54:55 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 7 May 2007 23:54:55 -0500
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <OFE4AEA40C.62F5FCE5-ON852572D4.005A8008-852572D4.005A9AAD@bftg.com>
References: <C1FE15183DA37645BC0633BC604E44F0057BE625@USNAEXCH.na.didata.local>
	<OFE4AEA40C.62F5FCE5-ON852572D4.005A8008-852572D4.005A9AAD@bftg.com>
Message-ID: <5d093f9a0705072154k5cfcf6cdg17b51b5f757b5625@mail.gmail.com>

You mean 'Unknown'... that is probable....

To be honest, I would suspect they would not carry the licenses over for
phones that are in an unknown state... but unregistered doesn't make
sense... what if a phone is at a remote site and the wan is down while you
are doing the upgrade, do you just get hosed?



Jonathan

On 5/7/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com> wrote:
>
>
> Could it be that phones that were at least registered before and now
> offline are ok, but phones that are "not found" meaning never registered to
> the system are not?
>
> Carlos
>
>
>  *"Matt Slaga \(US\)" <Matt.Slaga at us.didata.com>*
> Sent by: cisco-voip-bounces at puck.nether.net
>
> 05/07/2007 12:12 PM
>   To
> "Jonathan Charles" <jonvoip at gmail.com>  cc
> cbomba at s4nets.com, cisco-voip at puck.nether.net  Subject
> Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
>
>
>
>
>
> Interesting.  Cisco was hit hard on this at the UC 6.0 Partner Beta
> training, and the IPC BU repeated many times that the phones had to be
> registered to be counted in the license file.
>
>
>
> *From:* Jonathan Charles [mailto:jonvoip at gmail.com] *
> Sent:* Monday, May 07, 2007 11:59 AM*
> To:* Matt Slaga (US)*
> Cc:* cbomba at s4nets.com; cisco-voip at puck.nether.net*
> Subject:* Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
> This is NOT true.
>
> I just did an upgrade (3.3 to 5.1)... and we built a new server with
> CCM3.3, restored the backup (in a lab, no phones present), upgraded to 4.1,
> ran DMA and upgraded to 5.1 and restored the DMA data.
>
> The license was for all of the phones configured (zero phones were
> registered at the time DMA was run), and the customer had a full upgrade
> license for all configured phones.
>
>
>
> Jonathan
> On 5/7/07, *Matt Slaga (US)* <*Matt.Slaga at us.didata.com*<Matt.Slaga at us.didata.com>>
> wrote:
> Actually, this is not the case.  It will only generate DLUs for phones
> that are registered with the CallManager server.  It will skip any
> device that is not registered and not count it in the DLU list.
>
>
>
> -----Original Message-----
> From: *cisco-voip-bounces at puck.nether.net*<cisco-voip-bounces at puck.nether.net>
> [mailto:*cisco-voip-bounces at puck.nether.net *<cisco-voip-bounces at puck.nether.net>]
> On Behalf Of Christopher M.
> Bomba
> Sent: Wednesday, May 02, 2007 9:05 AM
> To: *cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net>
> Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
> Who said you had to buy the phones.  You can just put phones in with
> bogus MAC addresses and it will generate DLUs for those.  I thought they
> had to be registered but turns out that they just have to be present in
> the system.  You can create IP communicators or any other phone to boost
> the DLUs.  That is a big problem since anybody can do this.
>
> Chris
>
> --original message--
>
> thanks Ed for your reply, yes you are right I contacted Partner help and
> they told me that during the upgrade a file will be generated that when
> given to cisco they will give us back the DLUs equivalent to what
> configured on the system. I think this is an illegal workaround for the
> lic. issue as by that someone could buy phones without lic. and then
> setup them with CM 4.1 and then upgrade and he will be having a DLUs for
> free. I think Cisco has to do something to solve this.
>
> Thanks and Best Regards
>
> Ahmed A. Elnagar
> Network Field Engineer
>
> Advanced Computer Technology (ACT)
> 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> Postal Code:12411 Cairo Egypt
>
> Mob: +2010-2833868
> Website: *www.act-eg.com* <http://www.act-eg.com/>
> E-mail: *aelnagar at act-eg.com* <aelnagar at act-eg.com>
>
> ________________________________
>
> From: Ed Leatherman [mailto:* ealeatherman at gmail.com*<ealeatherman at gmail.com>
> ]
> Sent: Wed 02-May-07 3:16 PM
> To: Ahmed Elnagar
> Cc: *cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net>
> Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
>
>
> hi Ahmed,
>
> When you upgrade, callmanager will generate a temporary license for all
> the devices you have configured from 4.1. If I understand correctly, you
> just have to send that off to cisco's licensing group and have it
> converted to a real 5.2 license with the appropriate number of DLUs. If
> its anything like Unity licensing, the turn-around is pretty quick. I'm
> assuming you need a SASU agreement for callmanager for them to do it.
>
> We are getting ready to do the same process here this summer with our
> 4.1 system.
>
> Hope that helps.
>
> Ed
>
>
>
> On 5/2/07, Ahmed Elnagar <*aelnagar at act-eg.com * <aelnagar at act-eg.com>>
> wrote:
>
>        Hello all;
>
>        A customer wants to upgrade from CM 4.1 to 5.1, what I am
> worried about is the licsences issue. How will I get the DLUs for
> callmanager 5.1 we have a lic. with each phone but how we will get the
> DLUs??
>
>
>        Thanks and Best Regards
>
>        Ahmed A. Elnagar
>        Network Field Engineer
>
>        Advanced Computer Technology (ACT)
>        16 Fawzy Ramah St.Off Shehab St.Mohandessin , Giza, Egypt
>        Postal Code:12411 Cairo Egypt
>
>        Mob : +2010-2833868
>        Website: *www.act-eg.com* <http://www.act-eg.com/> <*
> http://www.act-eg.com/* <http://www.act-eg.com/> >
>        E-mail: *aelnagar at act-eg.com* <aelnagar at act-eg.com>
>
>        _______________________________________________
>        cisco-voip mailing list
>        * cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net>
>        *https://puck..nether.net/mailman/listinfo/cisco-voip*<https://puck.nether.net/mailman/listinfo/cisco-voip>
>
>
>
>
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> -------------- next part --------------
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> 78/attachment.html
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list*
> **cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net>*
> **https://puck..nether.net/mailman/listinfo/cisco-voip*<https://puck.nether.net/mailman/listinfo/cisco-voip>
>
>
> End of cisco-voip Digest, Vol 51, Issue 4
> *****************************************
>
> _______________________________________________
> cisco-voip mailing list*
> **cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net>*
> **https://puck..nether.net/mailman/listinfo/cisco-voip*<https://puck.nether.net/mailman/listinfo/cisco-voip>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
> _______________________________________________
> cisco-voip mailing list*
> **cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net>*
> **https://puck..nether.net/mailman/listinfo/cisco-voip *<https://puck.nether.net/mailman/listinfo/cisco-voip>
>
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
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From gokhanciscottl at yahoo.com  Tue May  8 02:50:17 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Mon, 7 May 2007 23:50:17 -0700 (PDT)
Subject: [cisco-voip] attendant console broadcast option is missing
Message-ID: <865923.2019.qm@web39613.mail.mud.yahoo.com>

hi

i configure ccm 4.1.3  auto attendant . but there is no BROADCAST option for pilot point.
only 2 options  which are    first available and longest idle.

but according to 4.1.3 ccm services and features document there should be broadcast option also

what am i missing

thanks
Gokhan


 
____________________________________________________________________________________
Don't pick lemons.
See all the new 2007 cars at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html 
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From roda_3tel at yahoo.com  Tue May  8 02:37:38 2007
From: roda_3tel at yahoo.com (Aing Roda)
Date: Mon, 7 May 2007 23:37:38 -0700 (PDT)
Subject: [cisco-voip] PGW 2000 - Calling Number Issue
Message-ID: <379448.8765.qm@web50811.mail.re2.yahoo.com>

Hi,
   
  I am connecting my PGW 2000 softswitch to Huawei switch. When calls are sent from the PGW2000 to Huawei, we can see complete information. However, when calls are sent from Huawei to PGW2000, we can not see ANI. Can anyone help to identify what is the cause of the problem?
   
  Regards,
  Roda

       
---------------------------------
Ahhh...imagining that irresistible "new car" smell?
 Check outnew cars at Yahoo! Autos.
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From steffen.bruening at de.logicalis.com  Tue May  8 03:45:28 2007
From: steffen.bruening at de.logicalis.com (Stift)
Date: Tue, 08 May 2007 09:45:28 +0200
Subject: [cisco-voip] Which has the higher priority for calls? Dial-peer or
	translation-rule/profile?
Message-ID: <f1p9qp$set$1@sea.gmane.org>

Hi,

I have a pots dial-peer with destination-pattern 9T on my Test-CME. And 
now I want to reject some premium numbers an do so I have made a 
translation rule with "reject /^0180/. With test voice translation-rule 
it works, but with my phone I can still dial a 0180 number.

So I ask you: Has the pots dial-peer a higher priority as the 
translation-rule?

Thanks for your help


From aelnagar at ACT-EG.COM  Tue May  8 04:10:00 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 8 May 2007 11:10:00 +0300
Subject: [cisco-voip] CCM 4.2: Configure "monitor"-Buttons on 7910
References: <f1o0jm$du5$1@sea.gmane.org>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A044@mail.ACT-EG.COM>

unfortunatly not supported, I opened cases with cisco and alot of search with no benefit. But alot of third party applications do it see berbee if they have one also check this site http://ipcelerate.com/ 
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of J?rg Wesely
Sent: Mon 07-May-07 11:01 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM 4.2: Configure "monitor"-Buttons on 7910



Hi,

on CCME, I can configure "Monitor"-Buttons, that means a
speeddial-button that also gets lit when the line is in use.

Is it possible to achieve the same on a CCM 4.2 system?

Regards

J?rg

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From tettokalle at hotmail.com  Tue May  8 06:47:30 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Tue, 08 May 2007 12:47:30 +0200
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
In-Reply-To: <01c401c790e8$ecca45c0$47196883@cfs.uoguelph.ca>
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>	<407055A92CECCB499C922A2D35FC19A608DEE725@Apollo.wesd.org>
	<01c401c790e8$ecca45c0$47196883@cfs.uoguelph.ca>
Message-ID: <f1pkg8$1bi$1@sea.gmane.org>

then your tftp caching is disabled ;-)

Lelio Fulgenzi schrieb:
> i've never had to bounce the tFTP server after copying over ZIP files.
>  
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
> 
>     ----- Original Message -----
>     *From:* Voll, Scott <mailto:Scott.Voll at wesd.org>
>     *To:* Lelio Fulgenzi <mailto:lelio at uoguelph.ca> ;
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     *Sent:* Monday, May 07, 2007 4:45 PM
>     *Subject:* RE: [cisco-voip] devpack vs .EXE vs. ZIP
> 
>     Personally I like the Zip.  I unpackage them, insert, bounce the
>     TFTP, setup the device defaults and I?m good.  The EXE requires a
>     reboot.
> 
>      
> 
>     Scott
> 
>      
> 
>     ------------------------------------------------------------------------
> 
>     *From:* cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
>     [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Lelio
>     Fulgenzi
>     *Sent:* Monday, May 07, 2007 1:32 PM
>     *To:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     *Subject:* [cisco-voip] devpack vs .EXE vs. ZIP
> 
>      
> 
>     Do I need a devpack to run my 7911s? Or can I just download the zip
>     files? What do the devpacks do?
> 
>      
> 
>     The readme for 8.2(2) SR2 says the following. But the 7911 is
>     running fine with SCCP11.7-2-3S (whatever that is)
> 
>      
> 
>      
> 
>         http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.8-2-2SR2-readme.htm
> 
>          
> 
>         Table 1: Compatible Cisco Unified Call Manager Release for 7911G
>         (No support for 7906G in these releases)
> 
>          
> 
>     *Call Manager Version*
> 
>     	
> 
>     *Executable on Software Page*
> 
>     3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe
> 
>     	
> 
>     ciscocm.3-3-5-sr1.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A>;
>     ciscocm.3-3-DevPack-64.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A>
> 
>     4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe
> 
>     	
> 
>     ciscocm.4-0-2a-sr2b.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A>;
>     ciscocm.4-0-DevPack-43.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A>
> 
>     4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe
> 
>     	
> 
>     ciscocm.4-1-3-sr1.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A>;
>     ciscocm.4-1-DevPack-23.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A>
> 
>     Table 2: Compatible Cisco Unified Call Manager Release that supports
>     both 7906G and 7911G
> 
>     *Call Manager Version*
> 
>     	
> 
>     *Executable on Software Page*
> 
>     3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later
> 
>     	
> 
>     ciscocm.3-3-5-sr2.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A>;
>     ciscocm.3-3-DevPack-68.1.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A>
> 
>     4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later
> 
>     	
> 
>     ciscocm.4-1-3-sr3a.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A>;
>     ciscocm.4-1-DevPack-28.2.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A>
> 
>     4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later
> 
>     	
> 
>     ciscocm.4-2-1-sr1.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A>;
>     ciscocm.4-2-1-DevPack-03.3.exe
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A>
> 
>      
> 
>      
> 
>     --------------------------------------------------------------------------------
>     Lelio Fulgenzi, B.A.
>     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
>     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>     ...there's no such thing as a bad timbit...
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Tue May  8 08:54:48 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 8 May 2007 08:54:48 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>	<407055A92CECCB499C922A2D35FC19A608DEE725@Apollo.wesd.org><01c401c790e8$ecca45c0$47196883@cfs.uoguelph.ca>
	<f1pkg8$1bi$1@sea.gmane.org>
Message-ID: <005501c79170$0f7837d0$47196883@cfs.uoguelph.ca>

Is this what you are referring to?

  Clusterwide Parameters(Parameters that apply to all servers)   
   
  Build CNF Files*   Build Selective
  Enable Caching of Constant and Bin Files at Startup*   True
  Enable Caching of Configuration Files*   True

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Kalle Tetto 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, May 08, 2007 6:47 AM
  Subject: Re: [cisco-voip] devpack vs .EXE vs. ZIP


  then your tftp caching is disabled ;-)

  Lelio Fulgenzi schrieb:
  > i've never had to bounce the tFTP server after copying over ZIP files.
  >  
  > --------------------------------------------------------------------------------
  > Lelio Fulgenzi, B.A.
  > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  > ...there's no such thing as a bad timbit...
  > 
  >     ----- Original Message -----
  >     *From:* Voll, Scott <mailto:Scott.Voll at wesd.org>
  >     *To:* Lelio Fulgenzi <mailto:lelio at uoguelph.ca> ;
  >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >     *Sent:* Monday, May 07, 2007 4:45 PM
  >     *Subject:* RE: [cisco-voip] devpack vs .EXE vs. ZIP
  > 
  >     Personally I like the Zip.  I unpackage them, insert, bounce the
  >     TFTP, setup the device defaults and I?m good.  The EXE requires a
  >     reboot.
  > 
  >      
  > 
  >     Scott
  > 
  >      
  > 
  >     ------------------------------------------------------------------------
  > 
  >     *From:* cisco-voip-bounces at puck.nether.net
  >     <mailto:cisco-voip-bounces at puck.nether.net>
  >     [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Lelio
  >     Fulgenzi
  >     *Sent:* Monday, May 07, 2007 1:32 PM
  >     *To:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >     *Subject:* [cisco-voip] devpack vs .EXE vs. ZIP
  > 
  >      
  > 
  >     Do I need a devpack to run my 7911s? Or can I just download the zip
  >     files? What do the devpacks do?
  > 
  >      
  > 
  >     The readme for 8.2(2) SR2 says the following. But the 7911 is
  >     running fine with SCCP11.7-2-3S (whatever that is)
  > 
  >      
  > 
  >      
  > 
  >         http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.8-2-2SR2-readme.htm
  > 
  >          
  > 
  >         Table 1: Compatible Cisco Unified Call Manager Release for 7911G
  >         (No support for 7906G in these releases)
  > 
  >          
  > 
  >     *Call Manager Version*
  > 
  >     
  > 
  >     *Executable on Software Page*
  > 
  >     3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe
  > 
  >     
  > 
  >     ciscocm.3-3-5-sr1.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A>;
  >     ciscocm.3-3-DevPack-64.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A>
  > 
  >     4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe
  > 
  >     
  > 
  >     ciscocm.4-0-2a-sr2b.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A>;
  >     ciscocm.4-0-DevPack-43.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A>
  > 
  >     4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe
  > 
  >     
  > 
  >     ciscocm.4-1-3-sr1.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A>;
  >     ciscocm.4-1-DevPack-23.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A>
  > 
  >     Table 2: Compatible Cisco Unified Call Manager Release that supports
  >     both 7906G and 7911G
  > 
  >     *Call Manager Version*
  > 
  >     
  > 
  >     *Executable on Software Page*
  > 
  >     3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later
  > 
  >     
  > 
  >     ciscocm.3-3-5-sr2.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A>;
  >     ciscocm.3-3-DevPack-68.1.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A>
  > 
  >     4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later
  > 
  >     
  > 
  >     ciscocm.4-1-3-sr3a.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A>;
  >     ciscocm.4-1-DevPack-28.2.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A>
  > 
  >     4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later
  > 
  >     
  > 
  >     ciscocm.4-2-1-sr1.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A>;
  >     ciscocm.4-2-1-DevPack-03.3.exe
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A>
  > 
  >      
  > 
  >      
  > 
  >     --------------------------------------------------------------------------------
  >     Lelio Fulgenzi, B.A.
  >     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  >     ...there's no such thing as a bad timbit...
  > 
  > 
  > ------------------------------------------------------------------------
  > 
  > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip

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From ealeatherman at gmail.com  Tue May  8 09:00:28 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 8 May 2007 09:00:28 -0400
Subject: [cisco-voip] cisco emergency responder - missing phone(s)
In-Reply-To: <94a1afde0705040512s538c6ed8m3fa0f718d40f049f@mail.gmail.com>
References: <81746c9a0705031441y3069e184s6c3cc8307e311e2b@mail.gmail.com>
	<94a1afde0705040512s538c6ed8m3fa0f718d40f049f@mail.gmail.com>
Message-ID: <94a1afde0705080600y73acb04elba946aa0f71acc91@mail.gmail.com>

No luck with the MSDE service pack (well, it is installed, just didnt seem
to fix it). Anyone know what OID is used by CER to poll the switch?

On 5/4/07, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> Same model, firmware, and port configs. going to try the MSDE service pack
> this weekend and a general reboot this weekend and see if it helps.
>
> On 5/3/07, Candace Holman <holman at post.harvard.edu> wrote:
>
> > Ed,
> >
> > You've probably already checked, but is there a different phone model or
> > firmware for the misbehaving devices?  How do the switchport configurations
> > compare?  Same vlan, same acl?
> >
> > Candace
> >
> >
> > > From: "Ed Leatherman" <ealeatherman at gmail.com>
> > > To: ciscovoip < cisco-voip at puck.nether.net>
> > > Date: Thu, 3 May 2007 14:19:02 -0400
> > > Subject: [cisco-voip] cisco emergency responder - missing phone(s)
> > > Curious if anyone has seen this before....
> > >
> > > We have CER v1.3 paired up with our callmanager cluster. Probably 100
> > > or so switches setup on it, 99.9% of the time everything works peachy.
> > > There are however a handful of phones that for some reason do not get
> > > discovered on their respective switches. The phone is listed in CER as
> > > unlocated, so I know it is getting the phone info from callmanager, but the
> > > switchport that we know it is plugged into on the switch in CER shows no
> > > phone. however there are many other phones on the switch that are
> > > discovered, i'm not sure what the difference is. The phone can talk CDP with
> > > the switch, I can see the CDP info on the phone's webpage, with the correct
> > > interface etc.
> > >
> > > I've tried forcing the locate switch-ports process on that switch, but
> > > its not coming back with any different information, and I see no SNMP errors
> > > about that switch in the app log.
> > >
> > > any ideas?
> > >
> > > --
> > > Ed Leatherman
> > > Senior Voice Engineer
> > > West Virginia University
> > > Telecommunications and Network Operations
> > >
> > >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From CarlosOrtiz at bayviewfinancial.com  Tue May  8 09:05:13 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 8 May 2007 09:05:13 -0400
Subject: [cisco-voip] CSS precedence
In-Reply-To: <ca09e3750705071909i1dc9d822xbd2103d2cf8248a@mail.gmail.com>
Message-ID: <OFF359368F.4EA29CAD-ON852572D5.00473275-852572D5.0047E3C7@bftg.com>

Yeah,  I am not new to this so it makes perfect sense.  I think the key 
was the "exact match" on the route pattern.  I looked at the route 
patterns and they are slightly different.  On a previous system that I 
administered I always had a consistent route pattern for long distance so 
the CSS of the line would take precedence.  I assumed that this inherited 
config was consistent but that's not the case.  Here was the difference.

9.1[2-9]XX[2-9]XXXXXX - Matching 
9.1[2-9]XXXXXXXXX - Not matching
I assume it matches because of the 2-9 highlighted above versus the X.
Thx......





"Fred Nielsen" <fwn at feasible.net> 
Sent by: cisco-voip-bounces at puck.nether.net
05/07/2007 10:09 PM

To
"Lelio Fulgenzi" <lelio at uoguelph.ca>, CarlosOrtiz at bayviewfinancial.com
cc
"Voll, Scott" <Scott.Voll at wesd.org>, cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] CSS precedence






concatenated CSS is line + device.  Partitions in the line CSS are ordered 
above the device CSS, but this is only considered if you have an exact 
match.  In other words, only when your dialed string matches two identical 
patterns is the partition order itself really taken into account. 
 
I suspect what is happening for you is that whatever you are dialing is 
matching a more specific pattern within the partitions listed in the 
device CSS, thereby bypassing a more widely matched pattern that you are 
hoping to match on the line CSS. 
 
Let us know if this makes some sense.
 
-Fred Nielsen

On 5/7/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
and forwarding CSS = forwarding CSS. no concatenation at all. that would 
make sense. ;)
 
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
 
----- Original Message ----- 
From: Voll, Scott 
To: CarlosOrtiz at bayviewfinancial.com ; cisco-voip at puck.nether.net 
Sent: Monday, May 07, 2007 4:24 PM
Subject: Re: [cisco-voip] CSS precedence

 
Lines + phone CSS = Device CSS.  Check the phone CSS to not include the 
Partitions you want each phone line to be able to go out. 
 
Scott
 
 

From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: Monday, May 07, 2007 12:50 PM
To: cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] CSS precedence
 

Meant to say CallMgr 4.1.3 (sr2) 



CarlosOrtiz at bayviewfinancial.com 
Sent by: cisco-voip-bounces at puck.nether.net 
05/07/2007 03:45 PM 


To
cisco-voip at puck.nether.net 
cc
 
Subject
[cisco-voip] CSS precedence
 


 
 





I am testing a remote location gateway by adding a DID from that location 
to my phone.  I assigned the remote site CSS to the remote line and also 
have a CSS assigned to my phone.  When I take the REMOTE LINE offhook and 
dial, it is exiting my local gateway.   In previous experience I have done 
this and it will exit the remote gateway per CSS precedence rules.  Am I 
missing something here or is this a bug? 

Carlos_______________________________________________ 
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From tettokalle at hotmail.com  Tue May  8 09:06:47 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Tue, 08 May 2007 15:06:47 +0200
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
In-Reply-To: <005501c79170$0f7837d0$47196883@cfs.uoguelph.ca>
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>	<407055A92CECCB499C922A2D35FC19A608DEE725@Apollo.wesd.org><01c401c790e8$ecca45c0$47196883@cfs.uoguelph.ca>	<f1pkg8$1bi$1@sea.gmane.org>
	<005501c79170$0f7837d0$47196883@cfs.uoguelph.ca>
Message-ID: <f1psld$vsv$1@sea.gmane.org>

yes it is. I ran into a similar problem, as I wanted to build my own 
phone backgrounds. I had to restart the tftp service or disable caching.

cheers

Lelio Fulgenzi schrieb:
> Is this what you are referring to?
>  
> 
>     Clusterwide Parameters(Parameters that apply to all servers)  
>      
>     Build CNF Files*   Build Selective
>     Enable Caching of Constant and Bin Files at Startup*   True
>     Enable Caching of Configuration Files*   True
> 
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
> 
>     ----- Original Message -----
>     *From:* Kalle Tetto <mailto:tettokalle at hotmail.com>
>     *To:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     *Sent:* Tuesday, May 08, 2007 6:47 AM
>     *Subject:* Re: [cisco-voip] devpack vs .EXE vs. ZIP
> 
>     then your tftp caching is disabled ;-)
> 
>     Lelio Fulgenzi schrieb:
>      > i've never had to bounce the tFTP server after copying over ZIP
>     files.
>      > 
>      >
>     --------------------------------------------------------------------------------
>      > Lelio Fulgenzi, B.A.
>      > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
>      > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>      >
>     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>      > ...there's no such thing as a bad timbit...
>      >
>      >     ----- Original Message -----
>      >     *From:* Voll, Scott <mailto:Scott.Voll at wesd.org>
>      >     *To:* Lelio Fulgenzi <mailto:lelio at uoguelph.ca> ;
>      >     cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
>      >     *Sent:* Monday, May 07, 2007 4:45 PM
>      >     *Subject:* RE: [cisco-voip] devpack vs .EXE vs. ZIP
>      >
>      >     Personally I like the Zip.  I unpackage them, insert, bounce the
>      >     TFTP, setup the device defaults and I?m good.  The EXE requires a
>      >     reboot.
>      >
>      >     
>      >
>      >     Scott
>      >
>      >     
>      >
>      >    
>     ------------------------------------------------------------------------
>      >
>      >     *From:* cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
>      >     <mailto:cisco-voip-bounces at puck.nether.net>
>      >     [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Lelio
>      >     Fulgenzi
>      >     *Sent:* Monday, May 07, 2007 1:32 PM
>      >     *To:* cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
>      >     *Subject:* [cisco-voip] devpack vs .EXE vs. ZIP
>      >
>      >     
>      >
>      >     Do I need a devpack to run my 7911s? Or can I just download
>     the zip
>      >     files? What do the devpacks do?
>      >
>      >     
>      >
>      >     The readme for 8.2(2) SR2 says the following. But the 7911 is
>      >     running fine with SCCP11.7-2-3S (whatever that is)
>      >
>      >     
>      >
>      >     
>      >
>      >        
>     http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.8-2-2SR2-readme.htm
>      >
>      >         
>      >
>      >         Table 1: Compatible Cisco Unified Call Manager Release
>     for 7911G
>      >         (No support for 7906G in these releases)
>      >
>      >         
>      >
>      >     *Call Manager Version*
>      >
>      >    
>      >
>      >     *Executable on Software Page*
>      >
>      >     3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe
>      >
>      >    
>      >
>      >     ciscocm.3-3-5-sr1.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A>>;
>      >     ciscocm.3-3-DevPack-64.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A>>
>      >
>      >     4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe
>      >
>      >    
>      >
>      >     ciscocm.4-0-2a-sr2b.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A>>;
>      >     ciscocm.4-0-DevPack-43.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A>>
>      >
>      >     4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe
>      >
>      >    
>      >
>      >     ciscocm.4-1-3-sr1.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A>>;
>      >     ciscocm.4-1-DevPack-23.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A>>
>      >
>      >     Table 2: Compatible Cisco Unified Call Manager Release that
>     supports
>      >     both 7906G and 7911G
>      >
>      >     *Call Manager Version*
>      >
>      >    
>      >
>      >     *Executable on Software Page*
>      >
>      >     3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later
>      >
>      >    
>      >
>      >     ciscocm.3-3-5-sr2.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A>>;
>      >     ciscocm.3-3-DevPack-68.1.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A>>
>      >
>      >     4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later
>      >
>      >    
>      >
>      >     ciscocm.4-1-3-sr3a.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A>>;
>      >     ciscocm.4-1-DevPack-28.2.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A>>
>      >
>      >     4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later
>      >
>      >    
>      >
>      >     ciscocm.4-2-1-sr1.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A>>;
>      >     ciscocm.4-2-1-DevPack-03.3.exe
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A>>
>      >
>      >     
>      >
>      >     
>      >
>      >    
>     --------------------------------------------------------------------------------
>      >     Lelio Fulgenzi, B.A.
>      >     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario
>     N1G 2W1
>      >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>      >    
>     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>      >     ...there's no such thing as a bad timbit...
>      >
>      >
>      >
>     ------------------------------------------------------------------------
>      >
>      > _______________________________________________
>      > cisco-voip mailing list
>      > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>      > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Tue May  8 09:09:50 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 8 May 2007 09:09:50 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>	<407055A92CECCB499C922A2D35FC19A608DEE725@Apollo.wesd.org><01c401c790e8$ecca45c0$47196883@cfs.uoguelph.ca>	<f1pkg8$1bi$1@sea.gmane.org><005501c79170$0f7837d0$47196883@cfs.uoguelph.ca>
	<f1psld$vsv$1@sea.gmane.org>
Message-ID: <00a601c79172$29689ed0$47196883@cfs.uoguelph.ca>

Wierd. It looks like I have caching enabled and I didn't have to restart my TFTP server to get the new images.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Kalle Tetto 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, May 08, 2007 9:06 AM
  Subject: Re: [cisco-voip] devpack vs .EXE vs. ZIP


  yes it is. I ran into a similar problem, as I wanted to build my own 
  phone backgrounds. I had to restart the tftp service or disable caching.

  cheers

  Lelio Fulgenzi schrieb:
  > Is this what you are referring to?
  >  
  > 
  >     Clusterwide Parameters(Parameters that apply to all servers)  
  >      
  >     Build CNF Files*   Build Selective
  >     Enable Caching of Constant and Bin Files at Startup*   True
  >     Enable Caching of Configuration Files*   True
  > 
  > --------------------------------------------------------------------------------
  > Lelio Fulgenzi, B.A.
  > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  > ...there's no such thing as a bad timbit...
  > 
  >     ----- Original Message -----
  >     *From:* Kalle Tetto <mailto:tettokalle at hotmail.com>
  >     *To:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >     *Sent:* Tuesday, May 08, 2007 6:47 AM
  >     *Subject:* Re: [cisco-voip] devpack vs .EXE vs. ZIP
  > 
  >     then your tftp caching is disabled ;-)
  > 
  >     Lelio Fulgenzi schrieb:
  >      > i've never had to bounce the tFTP server after copying over ZIP
  >     files.
  >      > 
  >      >
  >     --------------------------------------------------------------------------------
  >      > Lelio Fulgenzi, B.A.
  >      > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  >      > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >      >
  >     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  >      > ...there's no such thing as a bad timbit...
  >      >
  >      >     ----- Original Message -----
  >      >     *From:* Voll, Scott <mailto:Scott.Voll at wesd.org>
  >      >     *To:* Lelio Fulgenzi <mailto:lelio at uoguelph.ca> ;
  >      >     cisco-voip at puck.nether.net
  >     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
  >      >     *Sent:* Monday, May 07, 2007 4:45 PM
  >      >     *Subject:* RE: [cisco-voip] devpack vs .EXE vs. ZIP
  >      >
  >      >     Personally I like the Zip.  I unpackage them, insert, bounce the
  >      >     TFTP, setup the device defaults and I?m good.  The EXE requires a
  >      >     reboot.
  >      >
  >      >     
  >      >
  >      >     Scott
  >      >
  >      >     
  >      >
  >      >    
  >     ------------------------------------------------------------------------
  >      >
  >      >     *From:* cisco-voip-bounces at puck.nether.net
  >     <mailto:cisco-voip-bounces at puck.nether.net>
  >      >     <mailto:cisco-voip-bounces at puck.nether.net>
  >      >     [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Lelio
  >      >     Fulgenzi
  >      >     *Sent:* Monday, May 07, 2007 1:32 PM
  >      >     *To:* cisco-voip at puck.nether.net
  >     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
  >      >     *Subject:* [cisco-voip] devpack vs .EXE vs. ZIP
  >      >
  >      >     
  >      >
  >      >     Do I need a devpack to run my 7911s? Or can I just download
  >     the zip
  >      >     files? What do the devpacks do?
  >      >
  >      >     
  >      >
  >      >     The readme for 8.2(2) SR2 says the following. But the 7911 is
  >      >     running fine with SCCP11.7-2-3S (whatever that is)
  >      >
  >      >     
  >      >
  >      >     
  >      >
  >      >        
  >     http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.8-2-2SR2-readme.htm
  >      >
  >      >         
  >      >
  >      >         Table 1: Compatible Cisco Unified Call Manager Release
  >     for 7911G
  >      >         (No support for 7906G in these releases)
  >      >
  >      >         
  >      >
  >      >     *Call Manager Version*
  >      >
  >      >    
  >      >
  >      >     *Executable on Software Page*
  >      >
  >      >     3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe
  >      >
  >      >    
  >      >
  >      >     ciscocm.3-3-5-sr1.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A>>;
  >      >     ciscocm.3-3-DevPack-64.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A>>
  >      >
  >      >     4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe
  >      >
  >      >    
  >      >
  >      >     ciscocm.4-0-2a-sr2b.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A>>;
  >      >     ciscocm.4-0-DevPack-43.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A>>
  >      >
  >      >     4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe
  >      >
  >      >    
  >      >
  >      >     ciscocm.4-1-3-sr1.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A>>;
  >      >     ciscocm.4-1-DevPack-23.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A>>
  >      >
  >      >     Table 2: Compatible Cisco Unified Call Manager Release that
  >     supports
  >      >     both 7906G and 7911G
  >      >
  >      >     *Call Manager Version*
  >      >
  >      >    
  >      >
  >      >     *Executable on Software Page*
  >      >
  >      >     3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later
  >      >
  >      >    
  >      >
  >      >     ciscocm.3-3-5-sr2.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A>>;
  >      >     ciscocm.3-3-DevPack-68.1.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A>>
  >      >
  >      >     4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later
  >      >
  >      >    
  >      >
  >      >     ciscocm.4-1-3-sr3a.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A>>;
  >      >     ciscocm.4-1-DevPack-28.2.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A>>
  >      >
  >      >     4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later
  >      >
  >      >    
  >      >
  >      >     ciscocm.4-2-1-sr1.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A>>;
  >      >     ciscocm.4-2-1-DevPack-03.3.exe
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A>>
  >      >
  >      >     
  >      >
  >      >     
  >      >
  >      >    
  >     --------------------------------------------------------------------------------
  >      >     Lelio Fulgenzi, B.A.
  >      >     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario
  >     N1G 2W1
  >      >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >      >    
  >     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  >      >     ...there's no such thing as a bad timbit...
  >      >
  >      >
  >      >
  >     ------------------------------------------------------------------------
  >      >
  >      > _______________________________________________
  >      > cisco-voip mailing list
  >      > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >      > https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  >     _______________________________________________
  >     cisco-voip mailing list
  >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >     https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  > ------------------------------------------------------------------------
  > 
  > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip

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From Matt.Slaga at us.didata.com  Tue May  8 09:11:37 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 8 May 2007 09:11:37 -0400
Subject: [cisco-voip] CSS precedence
In-Reply-To: <OFF359368F.4EA29CAD-ON852572D5.00473275-852572D5.0047E3C7@bftg.com>
References: <ca09e3750705071909i1dc9d822xbd2103d2cf8248a@mail.gmail.com>
	<OFF359368F.4EA29CAD-ON852572D5.00473275-852572D5.0047E3C7@bftg.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0058012BD@USNAEXCH.na.didata.local>

Yes, [2-9] is a closer match than X.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: Tuesday, May 08, 2007 9:05 AM
To: Fred Nielsen
Cc: cisco-voip-bounces at puck.nether.net; Voll, Scott;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CSS precedence

 


Yeah,  I am not new to this so it makes perfect sense.  I think the key
was the "exact match" on the route pattern.  I looked at the route
patterns and they are slightly different.  On a previous system that I
administered I always had a consistent route pattern for long distance
so the CSS of the line would take precedence.  I assumed that this
inherited config was consistent but that's not the case.  Here was the
difference. 

9.1[2-9]XX[2-9]XXXXXX - Matching   

9.1[2-9]XXXXXXXXX - Not matching 

I assume it matches because of the 2-9 highlighted above versus the X. 

Thx...... 





"Fred Nielsen" <fwn at feasible.net> 
Sent by: cisco-voip-bounces at puck.nether.net 

05/07/2007 10:09 PM 

To

"Lelio Fulgenzi" <lelio at uoguelph.ca>, CarlosOrtiz at bayviewfinancial.com 

cc

"Voll, Scott" <Scott.Voll at wesd.org>, cisco-voip at puck.nether.net 

Subject

Re: [cisco-voip] CSS precedence

 

		




concatenated CSS is line + device.  Partitions in the line CSS are
ordered above the device CSS, but this is only considered if you have an
exact match.  In other words, only when your dialed string matches two
identical patterns is the partition order itself really taken into
account. 
  
I suspect what is happening for you is that whatever you are dialing is
matching a more specific pattern within the partitions listed in the
device CSS, thereby bypassing a more widely matched pattern that you are
hoping to match on the line CSS. 
  
Let us know if this makes some sense. 
  
-Fred Nielsen

On 5/7/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote: 
and forwarding CSS = forwarding CSS. no concatenation at all. that would
make sense. ;) 
  
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
....there's no such thing as a bad timbit...

----- Original Message ----- 
From: Voll, Scott <mailto:Scott.Voll at wesd.org>  
To: CarlosOrtiz at bayviewfinancial.com ; cisco-voip at puck.nether.net 
Sent: Monday, May 07, 2007 4:24 PM 
Subject: Re: [cisco-voip] CSS precedence 

Lines + phone CSS = Device CSS.  Check the phone CSS to not include the
Partitions you want each phone line to be able to go out. 

  

Scott 

  

  

________________________________

From: cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net>
[mailto:cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
<mailto:CarlosOrtiz at bayviewfinancial.com> 
Sent: Monday, May 07, 2007 12:50 PM
To: cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
Subject: Re: [cisco-voip] CSS precedence 

  


Meant to say CallMgr 4.1.3 (sr2) 

CarlosOrtiz at bayviewfinancial.com
<mailto:CarlosOrtiz at bayviewfinancial.com>  
Sent by: cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net>  

05/07/2007 03:45 PM 

 

To

cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 

cc

  

Subject

[cisco-voip] CSS precedence

  

 

  

 





I am testing a remote location gateway by adding a DID from that
location to my phone.  I assigned the remote site CSS to the remote line
and also have a CSS assigned to my phone.  When I take the REMOTE LINE
offhook and dial, it is exiting my local gateway.   In previous
experience I have done this and it will exit the remote gateway per CSS
precedence rules.  Am I missing something here or is this a bug? 

Carlos_______________________________________________ 
cisco-voip mailing list
cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
https://puck.nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 

________________________________

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



-----------------------------------------
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From fwn at feasible.net  Tue May  8 09:16:17 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Tue, 8 May 2007 06:16:17 -0700
Subject: [cisco-voip] CSS precedence
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0058012BD@USNAEXCH.na.didata.local>
References: <ca09e3750705071909i1dc9d822xbd2103d2cf8248a@mail.gmail.com>
	<OFF359368F.4EA29CAD-ON852572D5.00473275-852572D5.0047E3C7@bftg.com>
	<C1FE15183DA37645BC0633BC604E44F0058012BD@USNAEXCH.na.didata.local>
Message-ID: <ca09e3750705080616k6285dc31scff0b20c678ae3bf@mail.gmail.com>

or another way to say it, more specific, matching fewer possible numbers

-Fred Nielsen


On 5/8/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
>
>  Yes, [2-9] is a closer match than X.
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *
> CarlosOrtiz at bayviewfinancial.com
> *Sent:* Tuesday, May 08, 2007 9:05 AM
> *To:* Fred Nielsen
> *Cc:* cisco-voip-bounces at puck.nether.net; Voll, Scott;
> cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] CSS precedence
>
>
>
>
> Yeah,  I am not new to this so it makes perfect sense.  I think the key
> was the "exact match" on the route pattern.  I looked at the route patterns
> and they are slightly different.  On a previous system that I administered I
> always had a consistent route pattern for long distance so the CSS of the
> line would take precedence.  I assumed that this inherited config was
> consistent but that's not the case.  Here was the difference.
>
> 9.1[2-9]XX[2-9]XXXXXX - Matching
>
> 9.1[2-9]XXXXXXXXX - Not matching
>
> I assume it matches because of the 2-9 highlighted above versus the X.
>
> Thx......
>
>
>
>   *"Fred Nielsen" <fwn at feasible.net>*
> Sent by: cisco-voip-bounces at puck.nether.net
>
> 05/07/2007 10:09 PM
>
> To
>
> "Lelio Fulgenzi" <lelio at uoguelph.ca>, CarlosOrtiz at bayviewfinancial.com
>
> cc
>
> "Voll, Scott" <Scott.Voll at wesd.org>, cisco-voip at puck.nether.net
>
> Subject
>
> Re: [cisco-voip] CSS precedence
>
>
>
>
>
>
> concatenated CSS is line + device.  Partitions in the line CSS are ordered
> above the device CSS, but this is only considered if you have an exact
> match.  In other words, only when your dialed string matches two identical
> patterns is the partition order itself really taken into account.
>
> I suspect what is happening for you is that whatever you are dialing is
> matching a more specific pattern within the partitions listed in the device
> CSS, thereby bypassing a more widely matched pattern that you are hoping to
> match on the line CSS.
>
> Let us know if this makes some sense.
>
> -Fred Nielsen
>
> On 5/7/07, *Lelio Fulgenzi* <lelio at uoguelph.ca> wrote:
> and forwarding CSS = forwarding CSS. no concatenation at all. that would
> make sense. ;)
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ....there's no such thing as a bad timbit...
>
> ----- Original Message -----
> *From:* Voll, Scott <Scott.Voll at wesd.org>
> *To:* CarlosOrtiz at bayviewfinancial.com ; cisco-voip at puck.nether.net
> *Sent:* Monday, May 07, 2007 4:24 PM
> *Subject:* Re: [cisco-voip] CSS precedence
>
> Lines + phone CSS = Device CSS.  Check the phone CSS to not include the
> Partitions you want each phone line to be able to go out.
>
>
>
> Scott
>
>
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *
> CarlosOrtiz at bayviewfinancial.com*
> Sent: *Monday, May 07, 2007 12:50 PM*
> To:* cisco-voip at puck.nether.net *
> Subject:* Re: [cisco-voip] CSS precedence
>
>
>
>
> Meant to say CallMgr 4.1.3 (sr2)
>
> *CarlosOrtiz at bayviewfinancial.com* <CarlosOrtiz at bayviewfinancial.com>
> Sent by: cisco-voip-bounces at puck.nether.net
>
> 05/07/2007 03:45 PM
>
>
>
> To
>
> cisco-voip at puck.nether.net
>
> cc
>
>
>
> Subject
>
> [cisco-voip] CSS precedence
>
>
>
>
>
>
>
>
>
>
>
>
>
> I am testing a remote location gateway by adding a DID from that location
> to my phone.  I assigned the remote site CSS to the remote line and also
> have a CSS assigned to my phone.  When I take the *REMOTE LINE* offhook
> and dial, it is exiting my local gateway.   In previous experience I have
> done this and it will exit the remote gateway per CSS precedence rules.  Am
> I missing something here or is this a bug?
>
> Carlos_______________________________________________
> cisco-voip mailing list*
> *cisco-voip at puck.nether.net*
> *https://puck.nether.net/mailman/listinfo/cisco-voip
>  ------------------------------
>
> _______________________________________________
> cisco-voip mailing list*
> *cisco-voip at puck.nether.net *
> *https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list*
> *cisco-voip at puck.nether.net *
> *https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> ------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
>
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From tettokalle at hotmail.com  Tue May  8 09:19:50 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Tue, 08 May 2007 15:19:50 +0200
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
In-Reply-To: <00a601c79172$29689ed0$47196883@cfs.uoguelph.ca>
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>	<407055A92CECCB499C922A2D35FC19A608DEE725@Apollo.wesd.org><01c401c790e8$ecca45c0$47196883@cfs.uoguelph.ca>	<f1pkg8$1bi$1@sea.gmane.org><005501c79170$0f7837d0$47196883@cfs.uoguelph.ca>	<f1psld$vsv$1@sea.gmane.org>
	<00a601c79172$29689ed0$47196883@cfs.uoguelph.ca>
Message-ID: <f1ptdr$3k2$1@sea.gmane.org>

CCM 4.2(3)

Lelio Fulgenzi schrieb:
> Wierd. It looks like I have caching enabled and I didn't have to restart 
> my TFTP server to get the new images.
>  
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
> 
>     ----- Original Message -----
>     *From:* Kalle Tetto <mailto:tettokalle at hotmail.com>
>     *To:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     *Sent:* Tuesday, May 08, 2007 9:06 AM
>     *Subject:* Re: [cisco-voip] devpack vs .EXE vs. ZIP
> 
>     yes it is. I ran into a similar problem, as I wanted to build my own
>     phone backgrounds. I had to restart the tftp service or disable caching.
> 
>     cheers
> 
>     Lelio Fulgenzi schrieb:
>      > Is this what you are referring to?
>      > 
>      >
>      >     Clusterwide Parameters(Parameters that apply to all servers) 
>      >     
>      >     Build CNF Files*   Build Selective
>      >     Enable Caching of Constant and Bin Files at Startup*   True
>      >     Enable Caching of Configuration Files*   True
>      >
>      >
>     --------------------------------------------------------------------------------
>      > Lelio Fulgenzi, B.A.
>      > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
>      > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>      >
>     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>      > ...there's no such thing as a bad timbit...
>      >
>      >     ----- Original Message -----
>      >     *From:* Kalle Tetto <mailto:tettokalle at hotmail.com>
>      >     *To:* cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
>      >     *Sent:* Tuesday, May 08, 2007 6:47 AM
>      >     *Subject:* Re: [cisco-voip] devpack vs .EXE vs. ZIP
>      >
>      >     then your tftp caching is disabled ;-)
>      >
>      >     Lelio Fulgenzi schrieb:
>      >      > i've never had to bounce the tFTP server after copying
>     over ZIP
>      >     files.
>      >      >
>      >      >
>      >    
>     --------------------------------------------------------------------------------
>      >      > Lelio Fulgenzi, B.A.
>      >      > Senior Analyst (CCS) * University of Guelph * Guelph,
>     Ontario N1G 2W1
>      >      > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>      >      >
>      >    
>     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>      >      > ...there's no such thing as a bad timbit...
>      >      >
>      >      >     ----- Original Message -----
>      >      >     *From:* Voll, Scott <mailto:Scott.Voll at wesd.org>
>      >      >     *To:* Lelio Fulgenzi <mailto:lelio at uoguelph.ca> ;
>      >      >     cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>
>      >     <mailto:cisco-voip at puck.nether.net>
>     <mailto:cisco-voip at puck.nether.net>
>      >      >     *Sent:* Monday, May 07, 2007 4:45 PM
>      >      >     *Subject:* RE: [cisco-voip] devpack vs .EXE vs. ZIP
>      >      >
>      >      >     Personally I like the Zip.  I unpackage them, insert,
>     bounce the
>      >      >     TFTP, setup the device defaults and I?m good.  The EXE
>     requires a
>      >      >     reboot.
>      >      >
>      >      >    
>      >      >
>      >      >     Scott
>      >      >
>      >      >    
>      >      >
>      >      >   
>      >    
>     ------------------------------------------------------------------------
>      >      >
>      >      >     *From:* cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
>      >     <mailto:cisco-voip-bounces at puck.nether.net>
>      >      >     <mailto:cisco-voip-bounces at puck.nether.net>
>      >      >     [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf
>     Of *Lelio
>      >      >     Fulgenzi
>      >      >     *Sent:* Monday, May 07, 2007 1:32 PM
>      >      >     *To:* cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>
>      >     <mailto:cisco-voip at puck.nether.net>
>     <mailto:cisco-voip at puck.nether.net>
>      >      >     *Subject:* [cisco-voip] devpack vs .EXE vs. ZIP
>      >      >
>      >      >    
>      >      >
>      >      >     Do I need a devpack to run my 7911s? Or can I just
>     download
>      >     the zip
>      >      >     files? What do the devpacks do?
>      >      >
>      >      >    
>      >      >
>      >      >     The readme for 8.2(2) SR2 says the following. But the
>     7911 is
>      >      >     running fine with SCCP11.7-2-3S (whatever that is)
>      >      >
>      >      >    
>      >      >
>      >      >    
>      >      >
>      >      >       
>      >    
>     http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.8-2-2SR2-readme.htm
>      >      >
>      >      >        
>      >      >
>      >      >         Table 1: Compatible Cisco Unified Call Manager Release
>      >     for 7911G
>      >      >         (No support for 7906G in these releases)
>      >      >
>      >      >        
>      >      >
>      >      >     *Call Manager Version*
>      >      >
>      >      >   
>      >      >
>      >      >     *Executable on Software Page*
>      >      >
>      >      >     3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe
>      >      >
>      >      >   
>      >      >
>      >      >     ciscocm.3-3-5-sr1.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A>>>;
>      >      >     ciscocm.3-3-DevPack-64.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A>>>
>      >      >
>      >      >     4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe
>      >      >
>      >      >   
>      >      >
>      >      >     ciscocm.4-0-2a-sr2b.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A>>>;
>      >      >     ciscocm.4-0-DevPack-43.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A>>>
>      >      >
>      >      >     4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe
>      >      >
>      >      >   
>      >      >
>      >      >     ciscocm.4-1-3-sr1.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A>>>;
>      >      >     ciscocm.4-1-DevPack-23.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A>>>
>      >      >
>      >      >     Table 2: Compatible Cisco Unified Call Manager Release
>     that
>      >     supports
>      >      >     both 7906G and 7911G
>      >      >
>      >      >     *Call Manager Version*
>      >      >
>      >      >   
>      >      >
>      >      >     *Executable on Software Page*
>      >      >
>      >      >     3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later
>      >      >
>      >      >   
>      >      >
>      >      >     ciscocm.3-3-5-sr2.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A>>>;
>      >      >     ciscocm.3-3-DevPack-68.1.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A>>>
>      >      >
>      >      >     4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or
>     later
>      >      >
>      >      >   
>      >      >
>      >      >     ciscocm.4-1-3-sr3a.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A>>>;
>      >      >     ciscocm.4-1-DevPack-28.2.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A>>>
>      >      >
>      >      >     4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or
>     later
>      >      >
>      >      >   
>      >      >
>      >      >     ciscocm.4-2-1-sr1.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A>>>;
>      >      >     ciscocm.4-2-1-DevPack-03.3.exe
>      >      >   
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A>
>      >    
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A
>     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A>>>
>      >      >
>      >      >    
>      >      >
>      >      >    
>      >      >
>      >      >   
>      >    
>     --------------------------------------------------------------------------------
>      >      >     Lelio Fulgenzi, B.A.
>      >      >     Senior Analyst (CCS) * University of Guelph * Guelph,
>     Ontario
>      >     N1G 2W1
>      >      >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>      >      >   
>      >    
>     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>      >      >     ...there's no such thing as a bad timbit...
>      >      >
>      >      >
>      >      >
>      >    
>     ------------------------------------------------------------------------
>      >      >
>      >      > _______________________________________________
>      >      > cisco-voip mailing list
>      >      > cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
>      >      > https://puck.nether.net/mailman/listinfo/cisco-voip
>      >
>      >     _______________________________________________
>      >     cisco-voip mailing list
>      >     cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
>      >     https://puck.nether.net/mailman/listinfo/cisco-voip
>      >
>      >
>      >
>     ------------------------------------------------------------------------
>      >
>      > _______________________________________________
>      > cisco-voip mailing list
>      > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>      > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From tettokalle at hotmail.com  Tue May  8 09:25:52 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Tue, 08 May 2007 15:25:52 +0200
Subject: [cisco-voip] attendant console broadcast option is missing
In-Reply-To: <865923.2019.qm@web39613.mail.mud.yahoo.com>
References: <865923.2019.qm@web39613.mail.mud.yahoo.com>
Message-ID: <f1ptp6$3k2$2@sea.gmane.org>

the option you are looking for is not on the CCM admin page.

go to c:\program files\cisco\callmanagerattendant\bin and start 
acconfig.bat. Here you'll find your option.

Assure you are on the server the Cisco TCD is running on.

Cheers

gokhan senol schrieb:
> hi
> 
>  
> 
> i configure ccm 4.1.3  auto attendant . but there is no BROADCAST option 
> for pilot point.
> 
> only 2 options  which are    first available and longest idle.
> 
>  
> 
> but according to 4.1.3 ccm services and features document there should 
> be broadcast option also
> 
>  
> 
> what am i missing
> 
>  
> 
> thanks
> 
> Gokhan
> 
>  
> 
> 
> ------------------------------------------------------------------------
> Don't be flakey. Get Yahoo! Mail for Mobile 
> <http://us.rd.yahoo.com/evt=43909/*http://mobile.yahoo.com/mail> and
> always stay connected 
> <http://us.rd.yahoo.com/evt=43909/*http://mobile.yahoo.com/mail> to friends.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Tue May  8 09:28:50 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 8 May 2007 09:28:50 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>	<407055A92CECCB499C922A2D35FC19A608DEE725@Apollo.wesd.org><01c401c790e8$ecca45c0$47196883@cfs.uoguelph.ca>	<f1pkg8$1bi$1@sea.gmane.org><005501c79170$0f7837d0$47196883@cfs.uoguelph.ca>	<f1psld$vsv$1@sea.gmane.org><00a601c79172$29689ed0$47196883@cfs.uoguelph.ca>
	<f1ptdr$3k2$1@sea.gmane.org>
Message-ID: <00e601c79174$d0a86840$47196883@cfs.uoguelph.ca>

4.1.2 sr-blah
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Kalle Tetto 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, May 08, 2007 9:19 AM
  Subject: Re: [cisco-voip] devpack vs .EXE vs. ZIP


  CCM 4.2(3)

  Lelio Fulgenzi schrieb:
  > Wierd. It looks like I have caching enabled and I didn't have to restart 
  > my TFTP server to get the new images.
  >  
  > --------------------------------------------------------------------------------
  > Lelio Fulgenzi, B.A.
  > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  > ...there's no such thing as a bad timbit...
  > 
  >     ----- Original Message -----
  >     *From:* Kalle Tetto <mailto:tettokalle at hotmail.com>
  >     *To:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >     *Sent:* Tuesday, May 08, 2007 9:06 AM
  >     *Subject:* Re: [cisco-voip] devpack vs .EXE vs. ZIP
  > 
  >     yes it is. I ran into a similar problem, as I wanted to build my own
  >     phone backgrounds. I had to restart the tftp service or disable caching.
  > 
  >     cheers
  > 
  >     Lelio Fulgenzi schrieb:
  >      > Is this what you are referring to?
  >      > 
  >      >
  >      >     Clusterwide Parameters(Parameters that apply to all servers) 
  >      >     
  >      >     Build CNF Files*   Build Selective
  >      >     Enable Caching of Constant and Bin Files at Startup*   True
  >      >     Enable Caching of Configuration Files*   True
  >      >
  >      >
  >     --------------------------------------------------------------------------------
  >      > Lelio Fulgenzi, B.A.
  >      > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  >      > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >      >
  >     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  >      > ...there's no such thing as a bad timbit...
  >      >
  >      >     ----- Original Message -----
  >      >     *From:* Kalle Tetto <mailto:tettokalle at hotmail.com>
  >      >     *To:* cisco-voip at puck.nether.net
  >     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
  >      >     *Sent:* Tuesday, May 08, 2007 6:47 AM
  >      >     *Subject:* Re: [cisco-voip] devpack vs .EXE vs. ZIP
  >      >
  >      >     then your tftp caching is disabled ;-)
  >      >
  >      >     Lelio Fulgenzi schrieb:
  >      >      > i've never had to bounce the tFTP server after copying
  >     over ZIP
  >      >     files.
  >      >      >
  >      >      >
  >      >    
  >     --------------------------------------------------------------------------------
  >      >      > Lelio Fulgenzi, B.A.
  >      >      > Senior Analyst (CCS) * University of Guelph * Guelph,
  >     Ontario N1G 2W1
  >      >      > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >      >      >
  >      >    
  >     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  >      >      > ...there's no such thing as a bad timbit...
  >      >      >
  >      >      >     ----- Original Message -----
  >      >      >     *From:* Voll, Scott <mailto:Scott.Voll at wesd.org>
  >      >      >     *To:* Lelio Fulgenzi <mailto:lelio at uoguelph.ca> ;
  >      >      >     cisco-voip at puck.nether.net
  >     <mailto:cisco-voip at puck.nether.net>
  >      >     <mailto:cisco-voip at puck.nether.net>
  >     <mailto:cisco-voip at puck.nether.net>
  >      >      >     *Sent:* Monday, May 07, 2007 4:45 PM
  >      >      >     *Subject:* RE: [cisco-voip] devpack vs .EXE vs. ZIP
  >      >      >
  >      >      >     Personally I like the Zip.  I unpackage them, insert,
  >     bounce the
  >      >      >     TFTP, setup the device defaults and I?m good.  The EXE
  >     requires a
  >      >      >     reboot.
  >      >      >
  >      >      >    
  >      >      >
  >      >      >     Scott
  >      >      >
  >      >      >    
  >      >      >
  >      >      >   
  >      >    
  >     ------------------------------------------------------------------------
  >      >      >
  >      >      >     *From:* cisco-voip-bounces at puck.nether.net
  >     <mailto:cisco-voip-bounces at puck.nether.net>
  >      >     <mailto:cisco-voip-bounces at puck.nether.net>
  >      >      >     <mailto:cisco-voip-bounces at puck.nether.net>
  >      >      >     [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf
  >     Of *Lelio
  >      >      >     Fulgenzi
  >      >      >     *Sent:* Monday, May 07, 2007 1:32 PM
  >      >      >     *To:* cisco-voip at puck.nether.net
  >     <mailto:cisco-voip at puck.nether.net>
  >      >     <mailto:cisco-voip at puck.nether.net>
  >     <mailto:cisco-voip at puck.nether.net>
  >      >      >     *Subject:* [cisco-voip] devpack vs .EXE vs. ZIP
  >      >      >
  >      >      >    
  >      >      >
  >      >      >     Do I need a devpack to run my 7911s? Or can I just
  >     download
  >      >     the zip
  >      >      >     files? What do the devpacks do?
  >      >      >
  >      >      >    
  >      >      >
  >      >      >     The readme for 8.2(2) SR2 says the following. But the
  >     7911 is
  >      >      >     running fine with SCCP11.7-2-3S (whatever that is)
  >      >      >
  >      >      >    
  >      >      >
  >      >      >    
  >      >      >
  >      >      >       
  >      >    
  >     http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.8-2-2SR2-readme.htm
  >      >      >
  >      >      >        
  >      >      >
  >      >      >         Table 1: Compatible Cisco Unified Call Manager Release
  >      >     for 7911G
  >      >      >         (No support for 7906G in these releases)
  >      >      >
  >      >      >        
  >      >      >
  >      >      >     *Call Manager Version*
  >      >      >
  >      >      >   
  >      >      >
  >      >      >     *Executable on Software Page*
  >      >      >
  >      >      >     3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe
  >      >      >
  >      >      >   
  >      >      >
  >      >      >     ciscocm.3-3-5-sr1.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr1.exe&app=Tablebuild&status=showC2A>>>;
  >      >      >     ciscocm.3-3-DevPack-64.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-64.exe&app=Tablebuild&status=showC2A>>>
  >      >      >
  >      >      >     4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe
  >      >      >
  >      >      >   
  >      >      >
  >      >      >     ciscocm.4-0-2a-sr2b.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-2a-sr2b.exe&app=Tablebuild&status=showC2A>>>;
  >      >      >     ciscocm.4-0-DevPack-43.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.0/ciscocm.4-0-DevPack-43.exe&app=Tablebuild&status=showC2A>>>
  >      >      >
  >      >      >     4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe
  >      >      >
  >      >      >   
  >      >      >
  >      >      >     ciscocm.4-1-3-sr1.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr1.exe&app=Tablebuild&status=showC2A>>>;
  >      >      >     ciscocm.4-1-DevPack-23.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-23.exe&app=Tablebuild&status=showC2A>>>
  >      >      >
  >      >      >     Table 2: Compatible Cisco Unified Call Manager Release
  >     that
  >      >     supports
  >      >      >     both 7906G and 7911G
  >      >      >
  >      >      >     *Call Manager Version*
  >      >      >
  >      >      >   
  >      >      >
  >      >      >     *Executable on Software Page*
  >      >      >
  >      >      >     3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later
  >      >      >
  >      >      >   
  >      >      >
  >      >      >     ciscocm.3-3-5-sr2.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-5-sr2.exe&app=Tablebuild&status=showC2A>>>;
  >      >      >     ciscocm.3-3-DevPack-68.1.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/callmgr/3.3/ciscocm.3-3-DevPack-68.1.exe&app=Tablebuild&status=showC2A>>>
  >      >      >
  >      >      >     4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or
  >     later
  >      >      >
  >      >      >   
  >      >      >
  >      >      >     ciscocm.4-1-3-sr3a.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-3-sr3a.exe&app=Tablebuild&status=showC2A>>>;
  >      >      >     ciscocm.4-1-DevPack-28.2.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.1/ciscocm.4-1-DevPack-28.2.exe&app=Tablebuild&status=showC2A>>>
  >      >      >
  >      >      >     4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or
  >     later
  >      >      >
  >      >      >   
  >      >      >
  >      >      >     ciscocm.4-2-1-sr1.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-sr1.exe&app=Tablebuild&status=showC2A>>>;
  >      >      >     ciscocm.4-2-1-DevPack-03.3.exe
  >      >      >   
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A>
  >      >    
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A
  >     <http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/crypto/3DES/voice/callmgr/4.2/ciscocm.4-2-1-DevPack-03.3.exe&app=Tablebuild&status=showC2A>>>
  >      >      >
  >      >      >    
  >      >      >
  >      >      >    
  >      >      >
  >      >      >   
  >      >    
  >     --------------------------------------------------------------------------------
  >      >      >     Lelio Fulgenzi, B.A.
  >      >      >     Senior Analyst (CCS) * University of Guelph * Guelph,
  >     Ontario
  >      >     N1G 2W1
  >      >      >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >      >      >   
  >      >    
  >     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  >      >      >     ...there's no such thing as a bad timbit...
  >      >      >
  >      >      >
  >      >      >
  >      >    
  >     ------------------------------------------------------------------------
  >      >      >
  >      >      > _______________________________________________
  >      >      > cisco-voip mailing list
  >      >      > cisco-voip at puck.nether.net
  >     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
  >      >      > https://puck.nether.net/mailman/listinfo/cisco-voip
  >      >
  >      >     _______________________________________________
  >      >     cisco-voip mailing list
  >      >     cisco-voip at puck.nether.net
  >     <mailto:cisco-voip at puck.nether.net> <mailto:cisco-voip at puck.nether.net>
  >      >     https://puck.nether.net/mailman/listinfo/cisco-voip
  >      >
  >      >
  >      >
  >     ------------------------------------------------------------------------
  >      >
  >      > _______________________________________________
  >      > cisco-voip mailing list
  >      > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >      > https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  >     _______________________________________________
  >     cisco-voip mailing list
  >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >     https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  > ------------------------------------------------------------------------
  > 
  > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip

  _______________________________________________
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From MLinsemier at apcapital.com  Tue May  8 09:31:43 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Tue, 8 May 2007 09:31:43 -0400
Subject: [cisco-voip] Voicemail in SRST Mode
In-Reply-To: <865923.2019.qm@web39613.mail.mud.yahoo.com>
References: <865923.2019.qm@web39613.mail.mud.yahoo.com>
Message-ID: <AB0A35F2486F81468971AA68558BE719063FBB47@smieln01winm01.micoanet.com>

Currently we are running CallManager 4.1(3)sr3 along with Unity 4.2(1).
We currently are using MGCP to control all of our gateways from a
centralized cluster and utilize SRST at remote sites for failover.   I
was wondering if anyone has been successful, while in SRST mode, of
redirecting an inbound call after a set amount of rings to their
voicemail server, and have the subscribers message play rather than the
main Unity greeting.  We are using ISDN PRI at our remote sites. I know
that there were some commands that you could configure to send digits to
the Unity server, but from what I understand those were only supported
in a analog configuration.

 

Below is a excerpt of my call-manager fallback config:

 

call-manager-fallback

  voicemail 91517xxxxxxx

 call-forward busy 91517xxxxxxx

 call-forward noan 9517xxxxxxx timeout 12

 

Any help would be greatly appreciated.

 

Matt

 

 


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From aelnagar at ACT-EG.COM  Tue May  8 09:37:25 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 8 May 2007 16:37:25 +0300
Subject: [cisco-voip] CDR wrong number
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A045@mail.ACT-EG.COM>

I have a similiare problem to that one any ideas??
 
One of our peering partners send their traffic to our gateway with a tech-prefix 
of 1947#. 
Our country code is 94. 

Therefore our POTS dial-peer on gateway is- 
destination-pattern 1947#94T 

Call termination is happening fine. 

BUT, our CDR records the incoming call as 1947#94xxxxxx and billing system identifies it as a USA call.(since it starts from 1) 

We want our CDR to be 94xxxxxx ONLY.(without the tech-prefix) 

Could you please give me an advice to get this thing done? 

 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com
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From ealeatherman at gmail.com  Tue May  8 09:52:37 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 8 May 2007 09:52:37 -0400
Subject: [cisco-voip] Traffic Simulation
Message-ID: <94a1afde0705080652s379ae3a7xe20fc10be4a3af94@mail.gmail.com>

Hello all,

Anyone have a recommendation for VoIP traffic simulation? I have a colleague
here that is setting up a lab for some certification studying, he wants to
be able to put some traffic on it for QoS etc. I suggested IP SLA might do
something like that but he only has 1 router that can run the IOS version
needed. He wants to put some RTP and maybe some SCCP or SIP traffic on the
the network.

Thanks!

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From jsteinberg at gmail.com  Tue May  8 09:55:15 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 8 May 2007 09:55:15 -0400
Subject: [cisco-voip] Voicemail in SRST Mode
In-Reply-To: <AB0A35F2486F81468971AA68558BE719063FBB47@smieln01winm01.micoanet.com>
References: <865923.2019.qm@web39613.mail.mud.yahoo.com>
	<AB0A35F2486F81468971AA68558BE719063FBB47@smieln01winm01.micoanet.com>
Message-ID: <f1b65f880705080655w6c701da9hc34d3d23f7f8b418@mail.gmail.com>

If you have a PRI at both the SRST & Unity site the easiest integration is
using RDNIS.  However, not all Telco providers support end to end RDNIS.
Check to see if yours does by debugging q931 on both your SRST site and your
Unity site.

You will see a debug similiar to below when the SRST router forwards the
call to

Calling Party Number i = 0x0081, '5555553435'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '15555551212'
Plan:Unknown, Type:Unknown
Redirecting Number i = 0x000082, '1001'
Plan:Unknown, Type:Unknown

Where 5555553435 is a PSTN caller who is calling SRST extension 1001 and the
call is forwarded to Unity pilot 5555551212.

You should then perform the same debug on the Unity PRI gateway to see if
the RDNIS comes through, if so then you just need to make sure the RDNIS
number matches the Unity extension and via ccmadmin enable inbound RDNIS on
the MGCP T1 PRI at the Unity site

If your Telco doesn't support RDNIS then you will have to use vm-integration
which is typically used for FXO lines but can also be used for pri.

Justin

On 5/8/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>  Currently we are running CallManager 4.1(3)sr3 along with Unity 4.2(1).
> We currently are using MGCP to control all of our gateways from a
> centralized cluster and utilize SRST at remote sites for failover.   I was
> wondering if anyone has been successful, while in SRST mode, of redirecting
> an inbound call after a set amount of rings to their voicemail server, and
> have the subscribers message play rather than the main Unity greeting.  We
> are using ISDN PRI at our remote sites. I know that there were some commands
> that you could configure to send digits to the Unity server, but from what I
> understand those were only supported in a analog configuration.
>
>
>
> Below is a excerpt of my call-manager fallback config:
>
>
>
> call-manager-fallback
>
>   voicemail 91517xxxxxxx
>
>  call-forward busy 91517xxxxxxx
>
>  call-forward noan 9517xxxxxxx timeout 12
>
>
>
> Any help would be greatly appreciated.
>
>
>
> Matt
>
>
>
>
>
> ------------------------------
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be
> protected by one or more legal privileges. It is intended solely for the use
> of the addressee identified above. If you are not the intended recipient,
> any use, disclosure, copying or distribution of this communication is
> UNAUTHORIZED. Neither this information block, the typed name of the sender,
> nor anything else in this message is intended to constitute an electronic
> signature unless a specific statement to the contrary is included in this
> message. If you have received this communication in error, please
> immediately contact me and delete this communication from your computer.
> Thank you.
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From lelio at uoguelph.ca  Tue May  8 09:56:36 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 8 May 2007 09:56:36 -0400
Subject: [cisco-voip] which ios to use for 3845
Message-ID: <014901c79178$b1d6dce0$47196883@cfs.uoguelph.ca>

my 3845 was just dropped off. 

what IOS are people recommending for this thing? primarily used as a voice gateway (2 T1/PRIs using MGCP) and for SRST. I don't think I will be doing much transcoding, (other than from PSTN to voip) or conferencing.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From MLinsemier at apcapital.com  Tue May  8 10:15:16 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Tue, 8 May 2007 10:15:16 -0400
Subject: [cisco-voip] Voicemail in SRST Mode
In-Reply-To: <f1b65f880705080707m657e0d2cmc6837da4f1a6cfcb@mail.gmail.com>
References: <865923.2019.qm@web39613.mail.mud.yahoo.com>
	<AB0A35F2486F81468971AA68558BE719063FBB47@smieln01winm01.micoanet.com>
	<f1b65f880705080655w6c701da9hc34d3d23f7f8b418@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE719063FBB96@smieln01winm01.micoanet.com>
	<f1b65f880705080707m657e0d2cmc6837da4f1a6cfcb@mail.gmail.com>
Message-ID: <AB0A35F2486F81468971AA68558BE719063FBBBB@smieln01winm01.micoanet.com>

Again thanks for the additional information.  Because I don't want to
solely limit myself to MGCP knowledge, does the same go for H.323 other
then RDNIS has to be enabled at the gateway rather than in CCM.

 

From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Tuesday, May 08, 2007 10:08 AM
To: Linsemier, Matthew
Subject: Re: [cisco-voip] Voicemail in SRST Mode

 

no, DID will not cause any problem.  The extension of the SRST phone is
used as the redirecting number.  Hopefully, your extension is the last
3,4, or 5 digits of your DID.

Easiest way to understand how this works is to do a 'debug isdn q931'
while in SRST mode and place a call from the outside to an IP phone and
watch the outbound fwd leg to unity. 

On 5/8/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:

Justin,

 

Thanks for the response.  I also should have mentioned that we have DID
at all of our sites.  From what you are saying, it sounds like this
could cause problems because there would be no redirecting number?

 

Matt

 

From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Tuesday, May 08, 2007 9:55 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Voicemail in SRST Mode

 

If you have a PRI at both the SRST & Unity site the easiest integration
is using RDNIS.  However, not all Telco providers support end to end
RDNIS.  Check to see if yours does by debugging q931 on both your SRST
site and your Unity site. 

You will see a debug similiar to below when the SRST router forwards the
call to 

Calling Party Number i = 0x0081, '5555553435'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '15555551212' 
Plan:Unknown, Type:Unknown 
Redirecting Number i = 0x000082, '1001'
Plan:Unknown, Type:Unknown 

Where 5555553435 is a PSTN caller who is calling SRST extension 1001 and
the call is forwarded to Unity pilot 5555551212.

You should then perform the same debug on the Unity PRI gateway to see
if the RDNIS comes through, if so then you just need to make sure the
RDNIS number matches the Unity extension and via ccmadmin enable inbound
RDNIS on the MGCP T1 PRI at the Unity site 

If your Telco doesn't support RDNIS then you will have to use
vm-integration which is typically used for FXO lines but can also be
used for pri.

Justin

On 5/8/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:

Currently we are running CallManager 4.1(3)sr3 along with Unity 4.2(1).
We currently are using MGCP to control all of our gateways from a
centralized cluster and utilize SRST at remote sites for failover.   I
was wondering if anyone has been successful, while in SRST mode, of
redirecting an inbound call after a set amount of rings to their
voicemail server, and have the subscribers message play rather than the
main Unity greeting.  We are using ISDN PRI at our remote sites. I know
that there were some commands that you could configure to send digits to
the Unity server, but from what I understand those were only supported
in a analog configuration.

 

Below is a excerpt of my call-manager fallback config:

 

call-manager-fallback

  voicemail 91517xxxxxxx

 call-forward busy 91517xxxxxxx

 call-forward noan 9517xxxxxxx timeout 12

 

Any help would be greatly appreciated.

 

Matt

 

 

________________________________

CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may be
protected by one or more legal privileges. It is intended solely for the
use of the addressee identified above. If you are not the intended
recipient, any use, disclosure, copying or distribution of this
communication is UNAUTHORIZED. Neither this information block, the typed
name of the sender, nor anything else in this message is intended to
constitute an electronic signature unless a specific statement to the
contrary is included in this message. If you have received this
communication in error, please immediately contact me and delete this
communication from your computer. Thank you. 

________________________________


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

 

________________________________

CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may be
protected by one or more legal privileges. It is intended solely for the
use of the addressee identified above. If you are not the intended
recipient, any use, disclosure, copying or distribution of this
communication is UNAUTHORIZED. Neither this information block, the typed
name of the sender, nor anything else in this message is intended to
constitute an electronic signature unless a specific statement to the
contrary is included in this message. If you have received this
communication in error, please immediately contact me and delete this
communication from your computer. Thank you. 

________________________________

 


CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may
be protected by one or more legal privileges. It is intended
solely for the use of the addressee identified above. If you
are not the intended recipient, any use, disclosure, copying
or distribution of this communication is UNAUTHORIZED. Neither
this information block, the typed name of the sender, nor
anything else in this message is intended to constitute an
electronic signature unless a specific statement to the
contrary is included in this message. If you have received this
communication in error, please immediately contact me and delete
this communication from your computer. Thank you.
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From asobihoudai at yahoo.com  Tue May  8 10:19:52 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 8 May 2007 07:19:52 -0700 (PDT)
Subject: [cisco-voip] Which has the higher priority for calls? Dial-peer
	or translation-rule/profile?
In-Reply-To: <f1p9qp$set$1@sea.gmane.org>
Message-ID: <677803.43531.qm@web57212.mail.re3.yahoo.com>

Steffen,

Did you create a voice translation profile and then
apply that to your dial peer?


--- Stift <steffen.bruening at de.logicalis.com> wrote:

> Hi,
> 
> I have a pots dial-peer with destination-pattern 9T
> on my Test-CME. And 
> now I want to reject some premium numbers an do so I
> have made a 
> translation rule with "reject /^0180/. With test
> voice translation-rule 
> it works, but with my phone I can still dial a 0180
> number.
> 
> So I ask you: Has the pots dial-peer a higher
> priority as the 
> translation-rule?
> 
> Thanks for your help
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From voip at mrga.ch  Tue May  8 10:31:02 2007
From: voip at mrga.ch (Reto Gassmann)
Date: Tue, 08 May 2007 16:31:02 +0200
Subject: [cisco-voip] Call forward with hunt groups
Message-ID: <1178634662.464089a63d076@www.mail2web.ch>

Hi all

I try to set up a call forward for a hunt group. I just played a bit around with
settings and found out that I can specify the Hunt Forward Setting on the Hunt
Pilot. The option Forward Hunt No Answer works fine if nobody picks up the
phone after the Maximum Hunt Timer Time. However if nobody is logged in or
everybody is busy the Forward Hunt Busy Desination becomes active.

My question is: Is it possible to forward a Hunt Group to one destination
(answering machine)?

any idea?

Reto




From asobihoudai at yahoo.com  Tue May  8 10:38:39 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 8 May 2007 07:38:39 -0700 (PDT)
Subject: [cisco-voip] Call forward with hunt groups
In-Reply-To: <1178634662.464089a63d076@www.mail2web.ch>
Message-ID: <173888.3788.qm@web57205.mail.re3.yahoo.com>

CCME? CCM? What version? . . .


--- Reto Gassmann <voip at mrga.ch> wrote:

> Hi all
> 
> I try to set up a call forward for a hunt group. I
> just played a bit around with
> settings and found out that I can specify the Hunt
> Forward Setting on the Hunt
> Pilot. The option Forward Hunt No Answer works fine
> if nobody picks up the
> phone after the Maximum Hunt Timer Time. However if
> nobody is logged in or
> everybody is busy the Forward Hunt Busy Desination
> becomes active.
> 
> My question is: Is it possible to forward a Hunt
> Group to one destination
> (answering machine)?
> 
> any idea?
> 
> Reto
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From voip at mrga.ch  Tue May  8 10:42:59 2007
From: voip at mrga.ch (Reto Gassmann)
Date: Tue, 08 May 2007 16:42:59 +0200
Subject: [cisco-voip] Call forward with hunt groups
In-Reply-To: <173888.3788.qm@web57205.mail.re3.yahoo.com>
References: <173888.3788.qm@web57205.mail.re3.yahoo.com>
Message-ID: <1178635379.46408c7324d33@www.mail2web.ch>

Sorry

we have a Callmanager 4.2(3).

Zitat von Paul Choi <asobihoudai at yahoo.com>:

> CCME? CCM? What version? . . .
> 
> 
> --- Reto Gassmann <voip at mrga.ch> wrote:
> 
> > Hi all
> > 
> > I try to set up a call forward for a hunt group. I
> > just played a bit around with
> > settings and found out that I can specify the Hunt
> > Forward Setting on the Hunt
> > Pilot. The option Forward Hunt No Answer works fine
> > if nobody picks up the
> > phone after the Maximum Hunt Timer Time. However if
> > nobody is logged in or
> > everybody is busy the Forward Hunt Busy Desination
> > becomes active.
> > 
> > My question is: Is it possible to forward a Hunt
> > Group to one destination
> > (answering machine)?
> > 
> > any idea?
> > 
> > Reto
> > 
> > 
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around 
> http://mail.yahoo.com 
> 





From bob at smalltime.com  Tue May  8 10:43:22 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Tue, 08 May 2007 09:43:22 -0500
Subject: [cisco-voip] Voicemail in SRST Mode
In-Reply-To: <AB0A35F2486F81468971AA68558BE719063FBB47@smieln01winm01.micoanet.com>
References: <865923.2019.qm@web39613.mail.mud.yahoo.com>
	<AB0A35F2486F81468971AA68558BE719063FBB47@smieln01winm01.micoanet.com>
Message-ID: <46408C8A.5080804@smalltime.com>

You need to get your Cisco SE / Account Manager to get you a copy of the 
  rdnis_save.tcl script, which you'll run on your routers, and then the 
script that comes along with it that you run on IPCC (you can use the 
five-seat version that comes with CM)

In the call-manager-fallback part you specify the destination-pattern 
for busy and noan as a voip dialpeer that has the rdnis_save.tcl service.

I think you'll find that the vm-integration won't let you insert pauses, 
and if you use a PRI it sends all the digits at once, so you can't get 
into the correct Unity user mailbox and you just end up at the main 
greeting.

From birnit at nait.ca  Tue May  8 09:55:45 2007
From: birnit at nait.ca (Tkachuk,Birni)
Date: Tue, 8 May 2007 07:55:45 -0600
Subject: [cisco-voip] IPCC Enterprise
Message-ID: <8BF2352FAC40504B8D83F9403E4CCE3159B235@EXGMETRO.nait.ca>

We are looking for a site that has implemented Cisco IPCC Enterprise.
We are at a crossroads in making a decision between Express and
Enterprise and would like to visit with an organization who has taken
the plunge with Enterprise.  If a visit is not a possibility a
teleconference would be next best. Any help/advise will be greatly
appreciated.

Birni Tkachuk
Voice Services Administrator
NAIT Information Services Division 	
Ph:  780-471-7532
Fax 780-491-3083
	

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From lelio at uoguelph.ca  Tue May  8 10:56:04 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 8 May 2007 10:56:04 -0400
Subject: [cisco-voip] vg224 and DC power
Message-ID: <01d001c79181$008fa0d0$47196883@cfs.uoguelph.ca>

anybody out there using vg224s with DC power?

the documents say it works, but it's not a standard DC input, i.e. terminals, it is a plug of some type. there is no direction on how to get this plug.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From rratliff at cisco.com  Tue May  8 11:18:48 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 8 May 2007 11:18:48 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
In-Reply-To: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>
Message-ID: <8A04A629-4472-40FC-8731-410585F86A0D@cisco.com>

Dev Packs will do database changes as well as copy new load files.

Dev Packs are how we introduce new phone support, etc as well.

The phone load exe files only restart TFTP when it says it is  
restarting services, btw.

-Ryan

On May 7, 2007, at 4:31 PM, Lelio Fulgenzi wrote:

Do I need a devpack to run my 7911s? Or can I just download the zip  
files? What do the devpacks do?

The readme for 8.2(2) SR2 says the following. But the 7911 is running  
fine with SCCP11.7-2-3S (whatever that is)


http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp. 
8-2-2SR2-readme.htm

Table 1: Compatible Cisco Unified Call Manager Release for 7911G (No  
support for 7906G in these releases)

Call Manager Version

Executable on Software Page

3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe

ciscocm.3-3-5-sr1.exe; ciscocm.3-3-DevPack-64.exe

4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe

ciscocm.4-0-2a-sr2b.exe; ciscocm.4-0-DevPack-43.exe

4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe

ciscocm.4-1-3-sr1.exe; ciscocm.4-1-DevPack-23.exe

  Table 2: Compatible Cisco Unified Call Manager Release that  
supports both 7906G and 7911G

Call Manager Version

Executable on Software Page

3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later

ciscocm.3-3-5-sr2.exe; ciscocm.3-3-DevPack-68.1.exe

4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later

ciscocm.4-1-3-sr3a.exe; ciscocm.4-1-DevPack-28.2.exe

4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later

ciscocm.4-2-1-sr1.exe; ciscocm.4-2-1-DevPack-03.3.exe



------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
...there's no such thing as a bad timbit...
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From aelnagar at ACT-EG.COM  Tue May  8 11:37:08 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 8 May 2007 18:37:08 +0300
Subject: [cisco-voip] 7985 Screen saver
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A047@mail.ACT-EG.COM>

Hello all;
 
Regarding the IP Phone 7985 any one has a XML to act as a screen saver?
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com
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From lelio at uoguelph.ca  Tue May  8 11:42:37 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 8 May 2007 11:42:37 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>
	<8A04A629-4472-40FC-8731-410585F86A0D@cisco.com>
Message-ID: <01e701c79187$8152fae0$47196883@cfs.uoguelph.ca>

Fair enough. But do I need it for 4.1(3) sr XX and 7911s?


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Ryan Ratliff 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Tuesday, May 08, 2007 11:18 AM
  Subject: Re: [cisco-voip] devpack vs .EXE vs. ZIP


  Dev Packs will do database changes as well as copy new load files.

  Dev Packs are how we introduce new phone support, etc as well.

  The phone load exe files only restart TFTP when it says it is  
  restarting services, btw.

  -Ryan

  On May 7, 2007, at 4:31 PM, Lelio Fulgenzi wrote:

  Do I need a devpack to run my 7911s? Or can I just download the zip  
  files? What do the devpacks do?

  The readme for 8.2(2) SR2 says the following. But the 7911 is running  
  fine with SCCP11.7-2-3S (whatever that is)


  http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp. 
  8-2-2SR2-readme.htm

  Table 1: Compatible Cisco Unified Call Manager Release for 7911G (No  
  support for 7906G in these releases)

  Call Manager Version

  Executable on Software Page

  3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe

  ciscocm.3-3-5-sr1.exe; ciscocm.3-3-DevPack-64.exe

  4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe

  ciscocm.4-0-2a-sr2b.exe; ciscocm.4-0-DevPack-43.exe

  4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe

  ciscocm.4-1-3-sr1.exe; ciscocm.4-1-DevPack-23.exe

    Table 2: Compatible Cisco Unified Call Manager Release that  
  supports both 7906G and 7911G

  Call Manager Version

  Executable on Software Page

  3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later

  ciscocm.3-3-5-sr2.exe; ciscocm.3-3-DevPack-68.1.exe

  4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later

  ciscocm.4-1-3-sr3a.exe; ciscocm.4-1-DevPack-28.2.exe

  4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later

  ciscocm.4-2-1-sr1.exe; ciscocm.4-2-1-DevPack-03.3.exe



  ------------------------------------------------------------------------ 
  --------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  ...there's no such thing as a bad timbit...
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From d4rkf1ber at gmail.com  Tue May  8 12:24:37 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Tue, 8 May 2007 11:24:37 -0500
Subject: [cisco-voip] Looking for suggestions/options on a 911 problem
Message-ID: <a1b07ea60705080924k5bc86a4j40557594b5789203@mail.gmail.com>

Trying to find the simpliest way to resolve an issue.  Dealing with a
CM setup that has 5 locations, all phones are configured with a device
pool specific to the location they are in.

The phones are all in the same general CSS however and the line on
each phone is in the same partition and CSS as each other.

So the only thing that defines the location of the phone is the Device Pool.

I don't think I have a way to specify what gateway a particular device
pool should use for 911 calls can I?

I think I could do something if each locations phone was in a specific
CSS maybe?

The locations have some older 3745 gateway devices not being used
anymore but they have a live PSTN line connected, the idea was to send
911 calls only out that old PSTN line but with the phones all being in
the same CSS and stuff I just don't see an easy way to accomplish it
based just on device pool.

Am I missing anything?  Or does this assumption seem right, and I will
have to probably put them in specific CSS's based on location as well?

From bcarter at sentinel.com  Tue May  8 12:40:00 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Tue, 8 May 2007 11:40:00 -0500
Subject: [cisco-voip] Looking for suggestions/options on a 911 problem
In-Reply-To: <a1b07ea60705080924k5bc86a4j40557594b5789203@mail.gmail.com>
References: <a1b07ea60705080924k5bc86a4j40557594b5789203@mail.gmail.com>
Message-ID: <6A0744639B86DD459433108DBF101D26015D07F0@MAIL01.sentinel.com>

You are going to have to create separate CSS for each location. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dark Fiber
Sent: Tuesday, May 08, 2007 11:25 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Looking for suggestions/options on a 911 problem

Trying to find the simpliest way to resolve an issue.  Dealing with a CM
setup that has 5 locations, all phones are configured with a device pool
specific to the location they are in.

The phones are all in the same general CSS however and the line on each
phone is in the same partition and CSS as each other.

So the only thing that defines the location of the phone is the Device
Pool.

I don't think I have a way to specify what gateway a particular device
pool should use for 911 calls can I?

I think I could do something if each locations phone was in a specific
CSS maybe?

The locations have some older 3745 gateway devices not being used
anymore but they have a live PSTN line connected, the idea was to send
911 calls only out that old PSTN line but with the phones all being in
the same CSS and stuff I just don't see an easy way to accomplish it
based just on device pool.

Am I missing anything?  Or does this assumption seem right, and I will
have to probably put them in specific CSS's based on location as well?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From rratliff at cisco.com  Tue May  8 12:43:02 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 8 May 2007 12:43:02 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
In-Reply-To: <01e701c79187$8152fae0$47196883@cfs.uoguelph.ca>
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>
	<8A04A629-4472-40FC-8731-410585F86A0D@cisco.com>
	<01e701c79187$8152fae0$47196883@cfs.uoguelph.ca>
Message-ID: <145E4F66-17DF-4097-8F87-8056E528ADB1@cisco.com>

4.1(3)SR4b has support for 7911s.

-Ryan

On May 8, 2007, at 11:42 AM, Lelio Fulgenzi wrote:

Fair enough. But do I need it for 4.1(3) sr XX and 7911s?


------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
...there's no such thing as a bad timbit...
----- Original Message -----
From: Ryan Ratliff
To: Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net
Sent: Tuesday, May 08, 2007 11:18 AM
Subject: Re: [cisco-voip] devpack vs .EXE vs. ZIP

Dev Packs will do database changes as well as copy new load files.

Dev Packs are how we introduce new phone support, etc as well.

The phone load exe files only restart TFTP when it says it is
restarting services, btw.

-Ryan

On May 7, 2007, at 4:31 PM, Lelio Fulgenzi wrote:

Do I need a devpack to run my 7911s? Or can I just download the zip
files? What do the devpacks do?

The readme for 8.2(2) SR2 says the following. But the 7911 is running
fine with SCCP11.7-2-3S (whatever that is)


http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.
8-2-2SR2-readme.htm

Table 1: Compatible Cisco Unified Call Manager Release for 7911G (No
support for 7906G in these releases)

Call Manager Version

Executable on Software Page

3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe

ciscocm.3-3-5-sr1.exe; ciscocm.3-3-DevPack-64.exe

4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe

ciscocm.4-0-2a-sr2b.exe; ciscocm.4-0-DevPack-43.exe

4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe

ciscocm.4-1-3-sr1.exe; ciscocm.4-1-DevPack-23.exe

   Table 2: Compatible Cisco Unified Call Manager Release that
supports both 7906G and 7911G

Call Manager Version

Executable on Software Page

3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later

ciscocm.3-3-5-sr2.exe; ciscocm.3-3-DevPack-68.1.exe

4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later

ciscocm.4-1-3-sr3a.exe; ciscocm.4-1-DevPack-28.2.exe

4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later

ciscocm.4-2-1-sr1.exe; ciscocm.4-2-1-DevPack-03.3.exe



------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
...there's no such thing as a bad timbit...
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From rratliff at cisco.com  Tue May  8 12:44:41 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 8 May 2007 12:44:41 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
In-Reply-To: <145E4F66-17DF-4097-8F87-8056E528ADB1@cisco.com>
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>
	<8A04A629-4472-40FC-8731-410585F86A0D@cisco.com>
	<01e701c79187$8152fae0$47196883@cfs.uoguelph.ca>
	<145E4F66-17DF-4097-8F87-8056E528ADB1@cisco.com>
Message-ID: <F2C3D9B5-DCB8-4591-9CDE-3F44957979C7@cisco.com>

More specifically,
Q. What Cisco CallManager release is required for the Cisco IP Phone  
7911G?
A. Customers are required to use the most recent version of each  
Cisco CallManager train for full support: Version 3.3 (5)SR1, 4.0(2) 
SR2b, or 4.1(3)SR1.

http://www.cisco.com/en/US/partner/products/hw/phones/ps379/ 
products_qanda_item0900aecd8039de5e.shtml

-Ryan

On May 8, 2007, at 12:43 PM, Ryan Ratliff wrote:

4.1(3)SR4b has support for 7911s.

-Ryan

On May 8, 2007, at 11:42 AM, Lelio Fulgenzi wrote:

Fair enough. But do I need it for 4.1(3) sr XX and 7911s?


------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
...there's no such thing as a bad timbit...
----- Original Message -----
From: Ryan Ratliff
To: Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net
Sent: Tuesday, May 08, 2007 11:18 AM
Subject: Re: [cisco-voip] devpack vs .EXE vs. ZIP

Dev Packs will do database changes as well as copy new load files.

Dev Packs are how we introduce new phone support, etc as well.

The phone load exe files only restart TFTP when it says it is
restarting services, btw.

-Ryan

On May 7, 2007, at 4:31 PM, Lelio Fulgenzi wrote:

Do I need a devpack to run my 7911s? Or can I just download the zip
files? What do the devpacks do?

The readme for 8.2(2) SR2 says the following. But the 7911 is running
fine with SCCP11.7-2-3S (whatever that is)


http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.
8-2-2SR2-readme.htm

Table 1: Compatible Cisco Unified Call Manager Release for 7911G (No
support for 7906G in these releases)

Call Manager Version

Executable on Software Page

3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe

ciscocm.3-3-5-sr1.exe; ciscocm.3-3-DevPack-64.exe

4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe

ciscocm.4-0-2a-sr2b.exe; ciscocm.4-0-DevPack-43.exe

4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe

ciscocm.4-1-3-sr1.exe; ciscocm.4-1-DevPack-23.exe

    Table 2: Compatible Cisco Unified Call Manager Release that
supports both 7906G and 7911G

Call Manager Version

Executable on Software Page

3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later

ciscocm.3-3-5-sr2.exe; ciscocm.3-3-DevPack-68.1.exe

4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later

ciscocm.4-1-3-sr3a.exe; ciscocm.4-1-DevPack-28.2.exe

4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later

ciscocm.4-2-1-sr1.exe; ciscocm.4-2-1-DevPack-03.3.exe



------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
...there's no such thing as a bad timbit...
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From CarlosOrtiz at bayviewfinancial.com  Tue May  8 12:43:38 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 8 May 2007 12:43:38 -0400
Subject: [cisco-voip] Looking for suggestions/options on a 911 problem
In-Reply-To: <a1b07ea60705080924k5bc86a4j40557594b5789203@mail.gmail.com>
Message-ID: <OF034911B7.B6F5D77C-ON852572D5.005B0C83-852572D5.005BE2F0@bftg.com>

1)You need  a specific CSS for each location. 
2)Create a 911 partition specific to each location. 
        e.g  Site1_911_PT, Site2_911 , etc.

3) Place the 911 Partition inside the CSS specific to that site
4) Create 911 route pattern for each location within the corresponding 
partition and the Route List for that location

If your route lists contain route groups that pertain to other location 
you will have to create a route list specifically for 911 purposes to keep 
it local to the site.

Carlos






"Dark Fiber" <d4rkf1ber at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
05/08/2007 12:24 PM

To
cisco-voip at puck.nether.net
cc

Subject
[cisco-voip] Looking for suggestions/options on a 911 problem






Trying to find the simpliest way to resolve an issue.  Dealing with a
CM setup that has 5 locations, all phones are configured with a device
pool specific to the location they are in.

The phones are all in the same general CSS however and the line on
each phone is in the same partition and CSS as each other.

So the only thing that defines the location of the phone is the Device 
Pool.

I don't think I have a way to specify what gateway a particular device
pool should use for 911 calls can I?

I think I could do something if each locations phone was in a specific
CSS maybe?

The locations have some older 3745 gateway devices not being used
anymore but they have a live PSTN line connected, the idea was to send
911 calls only out that old PSTN line but with the phones all being in
the same CSS and stuff I just don't see an easy way to accomplish it
based just on device pool.

Am I missing anything?  Or does this assumption seem right, and I will
have to probably put them in specific CSS's based on location as well?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From jason.aarons at us.didata.com  Tue May  8 13:16:49 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 8 May 2007 13:16:49 -0400
Subject: [cisco-voip] IP-IVR  JTAPI setup...
Message-ID: <C1FE15183DA37645BC0633BC604E44F00580189E@USNAEXCH.na.didata.local>

 

In the "JTAPI Provider" section of the IP-IVR configuration, we normally
enter the CallManager machines that will be handling the JTAPI duties.
When you enter one or more entries, you get a pop-up that asks for the
CallManager Administration credentials.  This works fine when I use the
Publisher, but it fails with any Subscriber (we have 4 other
subscribers).  And sure enough, if I try to log into CallManager
Administration on either of the Subscriber machines with the same
"ccmadministrator" account and password, it does not work.

 

 



-----------------------------------------
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This e-mail communication and any attachments may contain
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From lelio at uoguelph.ca  Tue May  8 13:31:18 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 8 May 2007 13:31:18 -0400
Subject: [cisco-voip] devpack vs .EXE vs. ZIP
References: <018b01c790e6$b518fce0$47196883@cfs.uoguelph.ca>
	<8A04A629-4472-40FC-8731-410585F86A0D@cisco.com>
	<01e701c79187$8152fae0$47196883@cfs.uoguelph.ca>
	<145E4F66-17DF-4097-8F87-8056E528ADB1@cisco.com>
Message-ID: <004701c79196$b02364e0$47196883@cfs.uoguelph.ca>

Cool - thanks. We have sr4d, so that's why it works without a dev pack.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Ryan Ratliff 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Tuesday, May 08, 2007 12:43 PM
  Subject: Re: [cisco-voip] devpack vs .EXE vs. ZIP


  4.1(3)SR4b has support for 7911s.

  -Ryan

  On May 8, 2007, at 11:42 AM, Lelio Fulgenzi wrote:

  Fair enough. But do I need it for 4.1(3) sr XX and 7911s?


  ------------------------------------------------------------------------ 
  --------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  ...there's no such thing as a bad timbit...
  ----- Original Message -----
  From: Ryan Ratliff
  To: Lelio Fulgenzi
  Cc: cisco-voip at puck.nether.net
  Sent: Tuesday, May 08, 2007 11:18 AM
  Subject: Re: [cisco-voip] devpack vs .EXE vs. ZIP

  Dev Packs will do database changes as well as copy new load files.

  Dev Packs are how we introduce new phone support, etc as well.

  The phone load exe files only restart TFTP when it says it is
  restarting services, btw.

  -Ryan

  On May 7, 2007, at 4:31 PM, Lelio Fulgenzi wrote:

  Do I need a devpack to run my 7911s? Or can I just download the zip
  files? What do the devpacks do?

  The readme for 8.2(2) SR2 says the following. But the 7911 is running
  fine with SCCP11.7-2-3S (whatever that is)


  http://ftp-sj.cisco.com/cisco/voice/ip-7900ser/cmterm-7911_7906-sccp.
  8-2-2SR2-readme.htm

  Table 1: Compatible Cisco Unified Call Manager Release for 7911G (No
  support for 7906G in these releases)

  Call Manager Version

  Executable on Software Page

  3.3(5)SR1 or later & ciscocm.3-3-DevPack-64.exe

  ciscocm.3-3-5-sr1.exe; ciscocm.3-3-DevPack-64.exe

  4.0(2)SR2b or later & ciscocm.4-0-DevPack-43.exe

  ciscocm.4-0-2a-sr2b.exe; ciscocm.4-0-DevPack-43.exe

  4.1(3)SR1 or later & ciscocm.4-1-DevPack-23.exe

  ciscocm.4-1-3-sr1.exe; ciscocm.4-1-DevPack-23.exe

     Table 2: Compatible Cisco Unified Call Manager Release that
  supports both 7906G and 7911G

  Call Manager Version

  Executable on Software Page

  3.3(5)SR2 or later & ciscocm.3-3-DevPack-68.1.exe or later

  ciscocm.3-3-5-sr2.exe; ciscocm.3-3-DevPack-68.1.exe

  4.1(3)SR3a or later & ciscocm.4-1-DevPack-28.2.exe or later

  ciscocm.4-1-3-sr3a.exe; ciscocm.4-1-DevPack-28.2.exe

  4.2(1)SR1 or later & ciscocm.4-2-1-DevPack-03.3.exe or later

  ciscocm.4-2-1-sr1.exe; ciscocm.4-2-1-DevPack-03.3.exe



  ------------------------------------------------------------------------
  --------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  ...there's no such thing as a bad timbit...
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip

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From bob at smalltime.com  Tue May  8 14:23:27 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Tue, 08 May 2007 13:23:27 -0500
Subject: [cisco-voip] Forwarding a call to a home phone but _with_ Unity
 VM if it's not answered
In-Reply-To: <661141.293.qm@web37213.mail.mud.yahoo.com>
References: <661141.293.qm@web37213.mail.mud.yahoo.com>
Message-ID: <4640C01F.8050707@smalltime.com>

Erick Bergquist wrote:
> Unity does place the call on hold, so CCM MoH would be played for
> both release to switch and supervisor transfer. The release to switch
> is a real quick transfer (unity puts call on hold, places call on
> other port, then finishes transfer quick).
> 
> The MoH source used, would come from the voice mail port
> configuration on the call manager side.

Doesn't seem to be working in my case; the Unity voice ports belong to a 
  Unity device pool; that device pool has a "MOH Test MRGL" and the 
Network Hold and User hold sources are the same.  If I configure 7970's 
with this MRGL, then they hear the MOH when we do tests, but I still 
don't get anything when Unity does the supervised call transfer outbound.

I've opened a TAC case, but if anyone has any ideas...


From rratliff at cisco.com  Tue May  8 16:23:16 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 8 May 2007 16:23:16 -0400
Subject: [cisco-voip] IP-IVR  JTAPI setup...
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00580189E@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00580189E@USNAEXCH.na.didata.local>
Message-ID: <B8B7D887-5EB8-4B87-854A-8A4700AA540D@cisco.com>

Either A) you've never bounced IIS since enabling MLA on the  
subscribers... or B) you haven't installed MLA on your subscribers  
(older CM versions only)... or C) somehow the ccmadministrator  
password has gotten out of synch on your subs (adminutility should be  
able to test this).


-Ryan

On May 8, 2007, at 1:16 PM, Jason Aarons ((US)) wrote:



In the ?JTAPI Provider? section of the IP-IVR configuration, we  
normally enter the CallManager machines that will be handling the  
JTAPI duties.  When you enter one or more entries, you get a pop-up  
that asks for the CallManager Administration credentials.  This works  
fine when I use the Publisher, but it fails with any Subscriber (we  
have 4 other subscribers).  And sure enough, if I try to log into  
CallManager Administration on either of the Subscriber machines with  
the same ?ccmadministrator? account and password, it does not work.







Disclaimer: This e-mail communication and any attachments may contain  
confidential and privileged information and is for use by the  
designated addressee(s) named above only. If you are not the intended  
addressee, you are hereby notified that you have received this  
communication in error and that any use or reproduction of this email  
or its contents is strictly prohibited and may be unlawful. If you  
have received this communication in error, please notify us  
immediately by replying to this message and deleting it from your  
computer. Thank you.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From CBeck at careered.com  Tue May  8 21:39:38 2007
From: CBeck at careered.com (Chris Beck - CEC)
Date: Tue, 8 May 2007 20:39:38 -0500
Subject: [cisco-voip] IP-IVR  JTAPI setup...
References: <C1FE15183DA37645BC0633BC604E44F00580189E@USNAEXCH.na.didata.local>
Message-ID: <70E40EF625463544B814B0819734EF2C186E1A@cecemx002.corporate.careered.com>

I am taking a bit of a guess here, but it sounds like the dcdirectory replication is broken between callmanagers.  
 
depending on the size of the ldap directory store, there is a script file called reconfig_cluster on the server that will rebuild replication.  Make sure there are no special characters in the dcdirectory password specified during rebuild.
 
I found a good document on CCO about how to fix this.  I will see if I can find it agian.
 
-Chris

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
Sent: Tue 5/8/2007 12:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IP-IVR JTAPI setup...



 

In the "JTAPI Provider" section of the IP-IVR configuration, we normally enter the CallManager machines that will be handling the JTAPI duties.  When you enter one or more entries, you get a pop-up that asks for the CallManager Administration credentials.  This works fine when I use the Publisher, but it fails with any Subscriber (we have 4 other subscribers).  And sure enough, if I try to log into CallManager Administration on either of the Subscriber machines with the same "ccmadministrator" account and password, it does not work.

 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 


From jason.aarons at us.didata.com  Tue May  8 22:13:32 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 8 May 2007 22:13:32 -0400
Subject: [cisco-voip] IP-IVR  JTAPI setup...
In-Reply-To: <70E40EF625463544B814B0819734EF2C186E1A@cecemx002.corporate.careered.com>
References: <C1FE15183DA37645BC0633BC604E44F00580189E@USNAEXCH.na.didata.local>
	<70E40EF625463544B814B0819734EF2C186E1A@cecemx002.corporate.careered.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0058385EF@USNAEXCH.na.didata.local>

AD integrated, we can login to subs CCMAdmin using AD accounts that were
setup in CCMAdmin\MLA, just can't use ccmadministrator on subs -jason

-----Original Message-----
From: Chris Beck - CEC [mailto:CBeck at careered.com] 
Sent: Tuesday, May 08, 2007 9:40 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IP-IVR JTAPI setup...

I am taking a bit of a guess here, but it sounds like the dcdirectory
replication is broken between callmanagers.  
 
depending on the size of the ldap directory store, there is a script
file called reconfig_cluster on the server that will rebuild
replication.  Make sure there are no special characters in the
dcdirectory password specified during rebuild.
 
I found a good document on CCO about how to fix this.  I will see if I
can find it agian.
 
-Chris

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
Sent: Tue 5/8/2007 12:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IP-IVR JTAPI setup...



 

In the "JTAPI Provider" section of the IP-IVR configuration, we normally
enter the CallManager machines that will be handling the JTAPI duties.
When you enter one or more entries, you get a pop-up that asks for the
CallManager Administration credentials.  This works fine when I use the
Publisher, but it fails with any Subscriber (we have 4 other
subscribers).  And sure enough, if I try to log into CallManager
Administration on either of the Subscriber machines with the same
"ccmadministrator" account and password, it does not work.

 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 
-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.


From kds850 at gmail.com  Wed May  9 00:08:39 2007
From: kds850 at gmail.com (kevin k)
Date: Wed, 9 May 2007 00:08:39 -0400
Subject: [cisco-voip] building scalable dial plan
Message-ID: <40d7ab0a0705082108h391784d1o87a57258060e6b33@mail.gmail.com>

new to voip. reading and reading.
situation.
central ccm cluster.

i have over 200 sites.
5 sites - voip
rest centrex + key system

the debate is what to do with dial plan.

users at each site are used and fond of using 4 digit dn internally.
but looking ahead 4 digit dn will not suffice.

can you experts out there, suggest how to give 4 digit dn internally and 10
digit outside of the site, inter-site.????

or any thought?

i appreciate any comment.
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From jonvoip at gmail.com  Wed May  9 01:13:16 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 9 May 2007 00:13:16 -0500
Subject: [cisco-voip] Call Forward All on second line
Message-ID: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>

Got a receptionist, the main number comes in on her second line.

When she goes to lunch, she wants to CFA it to another woman.

She can't (That key is not active here) via the phone.

She CAN do it thru the web page (ccmuser)

Is there a service param that I can change so she can do it via the phone
(she wants to be able to do it quickly, so that when she needs to step away
she can just forward the phone)...




Jonathan
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From bob at smalltime.com  Wed May  9 01:33:10 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 09 May 2007 00:33:10 -0500
Subject: [cisco-voip] Call Forward All on second line
In-Reply-To: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>
References: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>
Message-ID: <46415D16.8000602@smalltime.com>

Jonathan Charles wrote:
> Got a receptionist, the main number comes in on her second line.
> 
> When she goes to lunch, she wants to CFA it to another woman.
> 
> She can't (That key is not active here) via the phone.
> 
> She CAN do it thru the web page (ccmuser)

I think I'll let Lelio handle this one.

From aelnagar at ACT-EG.COM  Wed May  9 03:54:30 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Wed, 9 May 2007 10:54:30 +0300
Subject: [cisco-voip] IPTD exam
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A04A@mail.ACT-EG.COM>

Hello all
 
I know that is out of our scope but I am doing IPTD exam and I have problem with that question.
 
 

One possible cluster arrangement for mycompany.com is to use two MCS-7830 servers in a 1:1 redundancy scheme with 50-50 load balancing. The cluster will support 3000 device units and 1500 phones. You have calculated that one of the servers will be supporting 312 non-IP phone weight values. There will be 8 phones with a BHCA of 15 registered to this sever along with 12 Phones with a BACH of 8. What is the maximum numbers of phones with a BHCA of 6 or fewer that this one server can support in this arrangement?

 

any ideas how to solve?

 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com
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From Johnny.Crothers at vanco.co.uk  Wed May  9 06:15:57 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers)
Date: Wed, 9 May 2007 11:15:57 +0100
Subject: [cisco-voip] (no subject)
Message-ID: <A56BD242744BF540B15D1672E0E3E875F9EDC5@LON-INT-MSGVS01.internal.vancocorp.com>

Hi everyone,

 

Little off topic, but can someone point me in the direction of the URL
for CCIE Voice Lab kit lists?

 

Thanks!!

 



**********************************************************************
Any opinions expressed in the email are those of the individual and not necessarily the company. This email and any files transmitted with it are confidential and solely for the use of the intended recipient.  If you are not the intended recipient or the person responsible for delivering it to the intended recipient, be advised that you have received this email in error and that any dissemination, distribution, copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with the content of this email please e-mail to: e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software viruses which could damage your own computer system. While the sender has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening any attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
**********************************************************************

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From sil at infiltrated.net  Wed May  9 07:00:03 2007
From: sil at infiltrated.net (J. Oquendo)
Date: Wed, 09 May 2007 07:00:03 -0400
Subject: [cisco-voip] (no subject)
In-Reply-To: <A56BD242744BF540B15D1672E0E3E875F9EDC5@LON-INT-MSGVS01.internal.vancocorp.com>
References: <A56BD242744BF540B15D1672E0E3E875F9EDC5@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <4641A9B3.3030600@infiltrated.net>

Johnny Crothers wrote:
>
> Hi everyone,
>
>  
>
> Little off topic, but can someone point me in the direction of the URL 
> for CCIE Voice Lab kit lists?
>
>  
>
> Thanks!!
>

http://www.ciscokits.com/cube/index.php?act=viewProd&productId=65&ccSID-b4c410adddf67168ce2ac0e2807326f8=0c4cb61ec9012e1e72c462807520c452
http://www.cisco.com/web/learning/le3/ccie/voice/lab_equipment.html
http://www.cciesecrets.net/books_voice.htm

IMO, you could scour eBay for MC3810's to replace any 36xx or 26xx 
series w/FXS||O cards. They have them built in.

www.infiltrated.net/wiredHome/ (my security lab with 3810's... used and 
tested in real world VoIP production once upon a time)

-- 
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
echo infiltrated.net|sed 's/^/sil@/g' 

"Wise men talk because they have something to say;
fools, because they have to say something." -- Plato


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From jsteinberg at gmail.com  Wed May  9 08:49:11 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 9 May 2007 08:49:11 -0400
Subject: [cisco-voip] Call Forward All on second line
In-Reply-To: <46415D16.8000602@smalltime.com>
References: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>
	<46415D16.8000602@smalltime.com>
Message-ID: <f1b65f880705090549v308311ddx9dfaf1da06c4df7e@mail.gmail.com>

I know this subject has been discussed countless times and Lelio has the bug
id as his homepage.....

But, just recently someone posted the myvoicesource.org website on here and
I was checking it out the other day and they indeed have a IP phone service
that advertises to handle the "i want to forward my non-primary line from my
phone scenario".

Has anyone tried this?  I plan to lab it but haven't gotten around to it.


http://www.myvoipsource.org/index.php?option=com_content&task=blogcategory&id=75&Itemid=85&limit=9&limitstart=9


I'm only mentioning this part because it took me a few minutes to figure it
out, in order to download the script, you must click 'login' on the left and
then register for a free account.

Justin



On 5/9/07, Robert Kulagowski <bob at smalltime.com> wrote:
>
> Jonathan Charles wrote:
> > Got a receptionist, the main number comes in on her second line.
> >
> > When she goes to lunch, she wants to CFA it to another woman.
> >
> > She can't (That key is not active here) via the phone.
> >
> > She CAN do it thru the web page (ccmuser)
>
> I think I'll let Lelio handle this one.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From jsteinberg at gmail.com  Wed May  9 08:49:50 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 9 May 2007 08:49:50 -0400
Subject: [cisco-voip] Call Forward All on second line
In-Reply-To: <f1b65f880705090549v308311ddx9dfaf1da06c4df7e@mail.gmail.com>
References: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>
	<46415D16.8000602@smalltime.com>
	<f1b65f880705090549v308311ddx9dfaf1da06c4df7e@mail.gmail.com>
Message-ID: <f1b65f880705090549j24bc6653o82e6d3570b13f3c3@mail.gmail.com>

I forgot to mention that all the scripts on that site appear to be
opensource...

:)

On 5/9/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> I know this subject has been discussed countless times and Lelio has the
> bug id as his homepage.....
>
> But, just recently someone posted the myvoicesource.org website on here
> and I was checking it out the other day and they indeed have a IP phone
> service that advertises to handle the "i want to forward my non-primary line
> from my phone scenario".
>
> Has anyone tried this?  I plan to lab it but haven't gotten around to it.
>
>
>
> http://www.myvoipsource.org/index.php?option=com_content&task=blogcategory&id=75&Itemid=85&limit=9&limitstart=9
>
>
> I'm only mentioning this part because it took me a few minutes to figure
> it out, in order to download the script, you must click 'login' on the left
> and then register for a free account.
>
> Justin
>
>
>
> On 5/9/07, Robert Kulagowski <bob at smalltime.com> wrote:
> >
> > Jonathan Charles wrote:
> > > Got a receptionist, the main number comes in on her second line.
> > >
> > > When she goes to lunch, she wants to CFA it to another woman.
> > >
> > > She can't (That key is not active here) via the phone.
> > >
> > > She CAN do it thru the web page (ccmuser)
> >
> > I think I'll let Lelio handle this one.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
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From CarlosOrtiz at bayviewfinancial.com  Wed May  9 08:56:47 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 9 May 2007 08:56:47 -0400
Subject: [cisco-voip] Call Forward All on second line
In-Reply-To: <f1b65f880705090549j24bc6653o82e6d3570b13f3c3@mail.gmail.com>
Message-ID: <OFBC537AF3.9DB60DF5-ON852572D6.00470829-852572D6.00471D99@bftg.com>

Lelio,

Warm up that lab and let us know if you get it to work...... ;)





"Justin Steinberg" <jsteinberg at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
05/09/2007 08:49 AM

To
"Robert Kulagowski" <bob at smalltime.com>
cc
ciscovoip <cisco-voip at puck.nether.net>
Subject
Re: [cisco-voip] Call Forward All on second line






I forgot to mention that all the scripts on that site appear to be 
opensource...

:)

On 5/9/07, Justin Steinberg < jsteinberg at gmail.com> wrote:
I know this subject has been discussed countless times and Lelio has the 
bug id as his homepage..... 

But, just recently someone posted the myvoicesource.org website on here 
and I was checking it out the other day and they indeed have a IP phone 
service that advertises to handle the "i want to forward my non-primary 
line from my phone scenario". 

Has anyone tried this?  I plan to lab it but haven't gotten around to it.


http://www.myvoipsource.org/index.php?option=com_content&task=blogcategory&id=75&Itemid=85&limit=9&limitstart=9



I'm only mentioning this part because it took me a few minutes to figure 
it out, in order to download the script, you must click 'login' on the 
left and then register for a free account. 

Justin




On 5/9/07, Robert Kulagowski < bob at smalltime.com> wrote:
Jonathan Charles wrote:
> Got a receptionist, the main number comes in on her second line.
>
> When she goes to lunch, she wants to CFA it to another woman.
>
> She can't (That key is not active here) via the phone. 
>
> She CAN do it thru the web page (ccmuser)

I think I'll let Lelio handle this one.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From lelio at uoguelph.ca  Wed May  9 09:04:45 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 9 May 2007 09:04:45 -0400
Subject: [cisco-voip] Call Forward All on second line
References: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>
	<46415D16.8000602@smalltime.com>
Message-ID: <002101c7923a$9da58f60$47196883@cfs.uoguelph.ca>

can't be done with a native callmanager install. you need to install third party apps/services. the phone will not behave as with the primary line forwarded though....no feedback regarding forwarding state.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Robert Kulagowski 
  To: ciscovoip 
  Sent: Wednesday, May 09, 2007 1:33 AM
  Subject: Re: [cisco-voip] Call Forward All on second line


  Jonathan Charles wrote:
  > Got a receptionist, the main number comes in on her second line.
  > 
  > When she goes to lunch, she wants to CFA it to another woman.
  > 
  > She can't (That key is not active here) via the phone.
  > 
  > She CAN do it thru the web page (ccmuser)

  I think I'll let Lelio handle this one.
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From tettokalle at hotmail.com  Wed May  9 09:55:09 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Wed, 09 May 2007 15:55:09 +0200
Subject: [cisco-voip] Weird shared line problem, calls get dropped
Message-ID: <f1sjs5$2be$1@sea.gmane.org>

Hi there,

I have a strange problem, running a support line on several phones, all 
using EM. The first line is the DDI and the second line is a shared line 
for the department. If now someone is having a call on the shared line, 
and a second call is coming in, the shared line rings on all phoness, 
someone picks up and we have two calls properly set up.

If now the fisrt person ends his call and hangs up the other call on the 
shared line is most of the time also dropped. Has anyone had this 
problem before? I can't explain, why this should happen, Is there a 
thing like a conference going on, for the CCM?

I can't see anything in the trace, seems for the CCM a normal operation, 
I will raise the debug level and see, if I see any hint.

Many thanks,
Cheers


From rratliff at cisco.com  Wed May  9 10:12:02 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 9 May 2007 10:12:02 -0400
Subject: [cisco-voip] IP-IVR  JTAPI setup...
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0058385EF@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00580189E@USNAEXCH.na.didata.local><70E40EF625463544B814B0819734EF2C186E1A@cecemx002.corporate.careered.com>
	<C1FE15183DA37645BC0633BC604E44F0058385EF@USNAEXCH.na.didata.local>
Message-ID: <001e01c79244$03f863b0$74c16640@amer.cisco.com>

What version of CM? 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (US)
Sent: Tuesday, May 08, 2007 10:14 PM
To: Chris Beck - CEC; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IP-IVR JTAPI setup...

AD integrated, we can login to subs CCMAdmin using AD accounts that were
setup in CCMAdmin\MLA, just can't use ccmadministrator on subs -jason

-----Original Message-----
From: Chris Beck - CEC [mailto:CBeck at careered.com]
Sent: Tuesday, May 08, 2007 9:40 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IP-IVR JTAPI setup...

I am taking a bit of a guess here, but it sounds like the dcdirectory
replication is broken between callmanagers.  
 
depending on the size of the ldap directory store, there is a script file
called reconfig_cluster on the server that will rebuild replication.  Make
sure there are no special characters in the dcdirectory password specified
during rebuild.
 
I found a good document on CCO about how to fix this.  I will see if I can
find it agian.
 
-Chris

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
Sent: Tue 5/8/2007 12:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IP-IVR JTAPI setup...



 

In the "JTAPI Provider" section of the IP-IVR configuration, we normally
enter the CallManager machines that will be handling the JTAPI duties.
When you enter one or more entries, you get a pop-up that asks for the
CallManager Administration credentials.  This works fine when I use the
Publisher, but it fails with any Subscriber (we have 4 other subscribers).
And sure enough, if I try to log into CallManager Administration on either
of the Subscriber machines with the same "ccmadministrator" account and
password, it does not work.

 

 

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From Roconnell at unislumin.com  Wed May  9 10:12:21 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Wed, 9 May 2007 10:12:21 -0400
Subject: [cisco-voip] building scalable dial plan
In-Reply-To: <40d7ab0a0705082108h391784d1o87a57258060e6b33@mail.gmail.com>
References: <40d7ab0a0705082108h391784d1o87a57258060e6b33@mail.gmail.com>
Message-ID: <18082B35CF0EBD4C8D265874508A615C05BCF639@hermes.unislumin.com>

Hi Kevin,

 

I had a very informative discussion few months ago with Dan in this
group. I will just repost our conversation, about scalable dialplans.
Read from the bottom up!!

 

Ryan,

    

     -Yes each location has there own customized translation pattern to
expand a 4 digit call to a full 10 digit extension.

     - Funnny you should ask.  No I don't like to send the @ route to
the GateKeeper.  I like to be more specific.  I have been thinking of
adding that to the design but I would prefer to perform more testing.  I
would love to finally be able to bring up my missed call log, see a
missed call from an E164 number, hit the dial key and have it route the
call correctly.  The typical user doesn't like to see the prepended 91
in their call logs.  I agree. 

     -Wow you have put a lot of thought into your dialplan, good for
you.  I only configure zones for DID numbers.  The sites operate just as
you described.  I put non DID extensions into a seperate partition.
Then give them a fake E164 number beggining with a 1.  111.901.4567.
The remainder of the number can mirror the DID(s) for the site.  You
want to plan ahead to make sure you will never have overlap in Unity.
CallManager ois finished here.

     -Centralized Unity just works.  No translations, no masks.  The CSS
for Unity includes all SITE-DID partitons as well as SITE-NonDID
partitions.  There is no overlap so Unity is capable of sending MWI
On/Off as well as transferring to all extensions.  This also serves as a
back door transfer to non DID extensions. 

     -CAC is used here.  The LOCATIONS in CallManager are enforced and
you have additional bandwidth policing at the GateKeeper.

 

Everything just works!

 

  
 

On 3/12/07, Ryan O'Connell <Roconnell at unislumin.com> wrote: 

Dan,

 

I like it, very slick. Thank you for putting such detail in your
response as well. I hope you don't mind me asking a few more questions
I'm intrigued now. Besides I'm still early in the design phase and you
may sway me to alter my plans. 

 

-        In your design you have all your internal DN's set to be E.164
numbers. Are you using some sort of Translation patterns at each of the
sites to maintain 4 digit dial within a given site?

-        I don't have a lot of experience with Gatekeepers so let me ask
you this when using a Gatekeeper controlled trunk is it just one route
into CallManager like a default route? Using your example: User dials
91.6025028867, the RG-1 strips predot and sends it to Gatekeeper. I
assume there is a default route command that sends all 10 digit numbers
to CCM. Once it gets there since the Trunks Inbound CSS has all
"SITE_EXTENTIONS" partitions so therefore the call completes. 

-        What about non-DID's at remote sites do you have them in
another partition or something that the Gatekeeper can't see? Example at
your remote site you don't use DID's so do you still assign them a E.164
number? What happens if that E.164 number is an actual number for
MacDonalds at that remote location but it happens to be the same number
that you assigned to a non-DID at the remote location?

-        What about voicemail, I'm not sure if you're using centralized
voicemail with Unity but that is what I'm using. How are you handling
Trap connections and MWI for remote locations? Do those outbound Unity
ports have the same CSS that you applied to your Gatekeeper controlled
trunk? 

-        With your design are you still using CAC? 

 

Aren't Dialplans fun!!

 

Cheers Ryan

 

________________________________

From: Dan Cavanaugh [mailto:dcavanaugh at gmail.com]
Sent: Mon 3/12/2007 6:33 PM
To: Ryan O'Connell
Cc: Justin Steinberg; cisco-voip at puck.nether.net 


Subject: Re: [cisco-voip] MWI with overlapping numbers




 

The GateKeeper helps you perform a basic translation.  The way I have it
configured is each site on the network uses (2) route lists, SITE-RL and
SITE-GK-RL.  All local route patterns, local 7 digit, 911, etc. use the
SITE-RL which points to the local VG.  You are done there.  Any
91.[2-9]xx[2-9]xxxxxx dial pattern is routed to the SITE-GK-RL.  Keep in
mind that ALL your extensions at all sites are the full E164 number,
6025028867.  A user dials 91.6025028867, not knowing if this number is
on your network or a local McDonalds.  The SITE-GK-RL consists of (3)
route groups and does the following:

(RG-1)  Strip the preDot and send it to the GateKeeper.  The GateKeeper
will maintain ranges of extensions available on the CallManager cluster
or multiple clusters(60250288..).  The GateKeeper controlled trunks to
CallManager will have a CSS that has access to all SITE-EXTENSIONS
partitions.  If the 10 digit number matches a defined range on the
GateKeeper, the call is placed ON-Net.  If this request fails, --> RG-2.


(RG-2)  Send it to GateKeeper.  The GateKeeper will maintain a list of
91xxx....... routes.  The xxx will be the local area code for each
gateway registered to GateKeeper.  If the GateKeeper has a gateway
registered for 91602......., the call will be routed to a gateway on the
network and the call will be placed as a local call.  If there is not a
registered gateway with the 91602 zone defined, -->RG-3. 

(RG-3)  Finally send the call to the local VG as a long distance call.

 

You said you weren't doing any intercluster trunks.  The only trunk
needed would be to the GateKeeper.  It would take place of your CSS and
PT manipulations.  This is very modular and scalable in design.  It also
allows you to just plug in another site(or company) overnight without
further thought about overlapping dialplans.  Users don't need to be
notified about dialing rules for ON-Net vs. OFF-Net calls, they just
simply call each other the same way they always have.  What it does not
accomplish is the silly abbreviated dialing between sites.  Everyone
would love to dial across the nation using only 4 digits.  This is a
pipe dream and you will spend many nights undoing what you thought would
scale for your company.  

 



 

On 3/11/07, Ryan O'Connell <Roconnell at unislumin.com > wrote: 

Hi Justin

 

I would like to hear more about your comments you made about problems
with CTI and AC line status. How will this be a problem? 

 

Also Dan, why introduce a Gatekeeper how will this help on a centralized
design, I'm not doing any intercluster trunks? 

 

Ryan

 

 

 

________________________________

From: Dan Cavanaugh [mailto: dcavanaugh at gmail.com] 
Sent: Friday, March 09, 2007 10:26 AM 
To: Justin Steinberg
Cc: Ryan O'Connell; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MWI with overlapping numbers

 

Total agreement with Justin.  What you have done at this point is
introduced layers of translation and complexity that don't need to be
there.  A nice design is exactly what Justin suggests below, with the
addition of a GateKeeper.  The Gatekeeper will add the filter you need
to determine if an extension is on your cluster. 

On 3/6/07, Justin Steinberg < jsteinberg at gmail.com
<mailto:jsteinberg at gmail.com> > wrote: 

You will notice other problems with this approach such as:

CDR issues since CDR is not partition aware 
CTI issues 
AC issues with line state updates

I think it would make life easier to use E164 numbers as the DN and use
translation patterns within each site to translate four digits to E164. 

Just my thoughts, and I do realize making this conversion would be a
real pain. :) 

Justin

On 3/6/07, Ryan O'Connell < Roconnell at unislumin.com
<mailto:Roconnell at unislumin.com> > wrote:

Fixed it

 

The fix was to change the MWI CSS to search the translation Table
instead of each sites local partition. And to change the MWI
notification in Unity back to "X" 

 

Ryan

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ryan O'Connell 
Sent: Tue 3/6/2007 3:07 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MWI with overlapping numbers 

 

Hey all,

 

I'm stuck on this one. Here is the scenario.

 

IPT customer, with lots of phones. They are using a Centralized model
where CallManger and Unity are at the head office providing services for
head office and all branch locations. This is a big customer and DN
overlapp is a real issue so they decided they wanted to go 4 digit
dialing within a given site, and 10 digit dialing between sites. 

 

No problem

 

*	I setup each location with a different <sitename>Oncluster
Partition and added 4 digit DN's to these partitions. 
*	phones can dial each other using 4 digits within a site 
*	when they dial another site the must dial 11 digits. example to
dial Edmonton they dial 917808885180 
*	A translation pattern is used to strip the pattern and search
the correct sites partition then strip the 91780888 off the number
leaving 5180 
*	For centralized voicemail, I used different "VoiceMailProfiles"
to prefix the numbers going into Unity so when someone from Edmonton
dials into Unity 780888 is prefixed to the 5180 putting them into the
correct mailbox. 
*	To light MWI lamps, I had to turn Multitenant Mode on so that VM
Ports can see the translations created and all works fine 

HERE is the problem

*	When someone has the same last 4 digit dn's at different sites,
MWI only works for the site who's partition comes first in the MWI CSS  

Example Calgary phone with DN 4035555180 and an Edmonton Phone
7808885180

*	Unity mailbox are as follows Calgary user is 4035555180 and the
Edmonton User is 7808885180 
*	both message notification feilds under each subscriber have been
changed from the default X to 5180 because the actual extension is only
4 digits not 10 (I THINK HERE IS MY PROBLEM BUT DON'T KNOW HOW TO FIX
IT) 
*	I leave a message for the Edmonton user 
*	MWI outbound ports use the Translation pattern to translate
7808885180 to 5180 in the edmOnCluster partition. And tell the phone to
dial the lamp code 
*	The MWI-ON CSS has partitions in the order of cgyOncluster -
then edmOncluster. It doesn't know better it just matches 5180 to the
cgyOncluster and the lamp turns on. 

I know you may need a decoder ring to understand this email, but any
help is appreciated

 

Ryno

________________________________

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of kevin k
Sent: Tuesday, May 08, 2007 10:09 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] building scalable dial plan

 

new to voip. reading and reading.

situation.

central ccm cluster.

 

i have over 200 sites.

5 sites - voip

rest centrex + key system

 

the debate is what to do with dial plan.

 

users at each site are used and fond of using 4 digit dn internally.

but looking ahead 4 digit dn will not suffice.

 

can you experts out there, suggest how to give 4 digit dn internally and
10 digit outside of the site, inter-site.????
 

or any thought?

 

i appreciate any comment.

 


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From jonvoip at gmail.com  Wed May  9 10:23:38 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 9 May 2007 09:23:38 -0500
Subject: [cisco-voip] Call Forward All on second line
In-Reply-To: <002101c7923a$9da58f60$47196883@cfs.uoguelph.ca>
References: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>
	<46415D16.8000602@smalltime.com>
	<002101c7923a$9da58f60$47196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0705090723t3d4904a8sccbd60786108432a@mail.gmail.com>

So, is this a 'no, it can't be done?'

If so, I have a simple solution... move the line they need to CFA to the
primary, make their personal line, line 2 (they don't need to forward
that...)



Jonathan

On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  can't be done with a native callmanager install. you need to install
> third party apps/services. the phone will not behave as with the primary
> line forwarded though....no feedback regarding forwarding state.
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
> ----- Original Message -----
> *From:* Robert Kulagowski <bob at smalltime.com>
> *To:* ciscovoip <cisco-voip at puck.nether.net>
> *Sent:* Wednesday, May 09, 2007 1:33 AM
> *Subject:* Re: [cisco-voip] Call Forward All on second line
>
> Jonathan Charles wrote:
> > Got a receptionist, the main number comes in on her second line.
> >
> > When she goes to lunch, she wants to CFA it to another woman.
> >
> > She can't (That key is not active here) via the phone.
> >
> > She CAN do it thru the web page (ccmuser)
>
> I think I'll let Lelio handle this one.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From lelio at uoguelph.ca  Wed May  9 10:25:14 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 9 May 2007 10:25:14 -0400
Subject: [cisco-voip] Call Forward All on second line
References: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>
	<46415D16.8000602@smalltime.com>
	<002101c7923a$9da58f60$47196883@cfs.uoguelph.ca>
	<5d093f9a0705090723t3d4904a8sccbd60786108432a@mail.gmail.com>
Message-ID: <00dc01c79245$dc564af0$47196883@cfs.uoguelph.ca>

Basically, yes, no it can not be done. We've done the same line swaps to appease the masses in some situations.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Jonathan Charles 
  To: Lelio Fulgenzi 
  Cc: ciscovoip 
  Sent: Wednesday, May 09, 2007 10:23 AM
  Subject: Re: [cisco-voip] Call Forward All on second line


  So, is this a 'no, it can't be done?'

  If so, I have a simple solution... move the line they need to CFA to the primary, make their personal line, line 2 (they don't need to forward that...)



  Jonathan


  On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
    can't be done with a native callmanager install. you need to install third party apps/services. the phone will not behave as with the primary line forwarded though....no feedback regarding forwarding state.

    --------------------------------------------------------------------------------
    Lelio Fulgenzi, B.A.
    Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
    (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
    ...there's no such thing as a bad timbit...

      ----- Original Message ----- 
      From: Robert Kulagowski 
      To: ciscovoip 
      Sent: Wednesday, May 09, 2007 1:33 AM
      Subject: Re: [cisco-voip] Call Forward All on second line


      Jonathan Charles wrote:
      > Got a receptionist, the main number comes in on her second line.
      > 
      > When she goes to lunch, she wants to CFA it to another woman.
      > 
      > She can't (That key is not active here) via the phone.
      > 
      > She CAN do it thru the web page (ccmuser)

      I think I'll let Lelio handle this one.
      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip


    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip



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From jonvoip at gmail.com  Wed May  9 10:26:59 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 9 May 2007 09:26:59 -0500
Subject: [cisco-voip] Call Forward All on second line
In-Reply-To: <00dc01c79245$dc564af0$47196883@cfs.uoguelph.ca>
References: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>
	<46415D16.8000602@smalltime.com>
	<002101c7923a$9da58f60$47196883@cfs.uoguelph.ca>
	<5d093f9a0705090723t3d4904a8sccbd60786108432a@mail.gmail.com>
	<00dc01c79245$dc564af0$47196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0705090726y5357bc3fnabc030500eccc1b3@mail.gmail.com>

that blows and stuff...

Sounds like feature request time (cuz those ALWAYS result in the new feature
being in the next release!)



Jonathan

On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  Basically, yes, no it can not be done. We've done the same line swaps to
> appease the masses in some situations.
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
> ----- Original Message -----
> *From:* Jonathan Charles <jonvoip at gmail.com>
> *To:* Lelio Fulgenzi <lelio at uoguelph.ca>
> *Cc:* ciscovoip <cisco-voip at puck.nether.net>
> *Sent:* Wednesday, May 09, 2007 10:23 AM
> *Subject:* Re: [cisco-voip] Call Forward All on second line
>
> So, is this a 'no, it can't be done?'
>
> If so, I have a simple solution... move the line they need to CFA to the
> primary, make their personal line, line 2 (they don't need to forward
> that...)
>
>
>
> Jonathan
>
> On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> >
> >  can't be done with a native callmanager install. you need to install
> > third party apps/services. the phone will not behave as with the primary
> > line forwarded though....no feedback regarding forwarding state.
> >
> > --------------------------------------------------------------------------------
> > Lelio Fulgenzi, B.A.
> > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > ...there's no such thing as a bad timbit...
> >
> > ----- Original Message -----
> > *From:* Robert Kulagowski <bob at smalltime.com>
> > *To:* ciscovoip <cisco-voip at puck.nether.net>
> > *Sent:* Wednesday, May 09, 2007 1:33 AM
> > *Subject:* Re: [cisco-voip] Call Forward All on second line
> >
> > Jonathan Charles wrote:
> > > Got a receptionist, the main number comes in on her second line.
> > >
> > > When she goes to lunch, she wants to CFA it to another woman.
> > >
> > > She can't (That key is not active here) via the phone.
> > >
> > > She CAN do it thru the web page (ccmuser)
> >
> > I think I'll let Lelio handle this one.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
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From c3voip at nc.rr.com  Wed May  9 10:43:52 2007
From: c3voip at nc.rr.com (c3voip)
Date: Wed, 9 May 2007 10:43:52 -0400
Subject: [cisco-voip] Call Forward All on second line
In-Reply-To: <5d093f9a0705090726y5357bc3fnabc030500eccc1b3@mail.gmail.com>
References: <5d093f9a0705082213y2ba77e61t30ac546f11a57624@mail.gmail.com>
	<46415D16.8000602@smalltime.com>
	<002101c7923a$9da58f60$47196883@cfs.uoguelph.ca>
	<5d093f9a0705090723t3d4904a8sccbd60786108432a@mail.gmail.com>
	<00dc01c79245$dc564af0$47196883@cfs.uoguelph.ca>
	<5d093f9a0705090726y5357bc3fnabc030500eccc1b3@mail.gmail.com>
Message-ID: <001801c79248$7683fad0$b610800a@RCCNT.RTI.ORG>

I have a few users that need this feature, and I have implemented the
IPPhone callForwarder IP Service from myvoipsource.org.  I have had a very
positive response to the usefulness of it.

 

-C

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Wednesday, May 09, 2007 10:27 AM
To: Lelio Fulgenzi
Cc: ciscovoip
Subject: Re: [cisco-voip] Call Forward All on second line

 

that blows and stuff...

Sounds like feature request time (cuz those ALWAYS result in the new feature
being in the next release!)



Jonathan

On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

Basically, yes, no it can not be done. We've done the same line swaps to
appease the masses in some situations.

 

----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

----- Original Message ----- 

From: Jonathan <mailto:jonvoip at gmail.com>  Charles 

To: Lelio <mailto:lelio at uoguelph.ca>  Fulgenzi 

Cc: ciscovoip <mailto:cisco-voip at puck.nether.net>  

Sent: Wednesday, May 09, 2007 10:23 AM

Subject: Re: [cisco-voip] Call Forward All on second line

 

So, is this a 'no, it can't be done?'

If so, I have a simple solution... move the line they need to CFA to the
primary, make their personal line, line 2 (they don't need to forward
that...)



Jonathan

On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote: 

can't be done with a native callmanager install. you need to install third
party apps/services. the phone will not behave as with the primary line
forwarded though....no feedback regarding forwarding state.



----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

----- Original Message ----- 

From: Robert <mailto:bob at smalltime.com>  Kulagowski 

To: ciscovoip <mailto:cisco-voip at puck.nether.net>  

Sent: Wednesday, May 09, 2007 1:33 AM

Subject: Re: [cisco-voip] Call Forward All on second line

 

Jonathan Charles wrote:
> Got a receptionist, the main number comes in on her second line.
> 
> When she goes to lunch, she wants to CFA it to another woman.
> 
> She can't (That key is not active here) via the phone.
> 
> She CAN do it thru the web page (ccmuser)

I think I'll let Lelio handle this one.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip

 

 

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From Scott.Voll at wesd.org  Wed May  9 12:31:37 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 9 May 2007 09:31:37 -0700
Subject: [cisco-voip] attendant console broadcast option is missing
In-Reply-To: <f1ptp6$3k2$2@sea.gmane.org>
References: <865923.2019.qm@web39613.mail.mud.yahoo.com>
	<f1ptp6$3k2$2@sea.gmane.org>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE733@Apollo.wesd.org>

It's under Line group configuration.

Distribution Algorithm

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kalle Tetto
Sent: Tuesday, May 08, 2007 6:26 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] attendant console broadcast option is missing

the option you are looking for is not on the CCM admin page.

go to c:\program files\cisco\callmanagerattendant\bin and start 
acconfig.bat. Here you'll find your option.

Assure you are on the server the Cisco TCD is running on.

Cheers

gokhan senol schrieb:
> hi
> 
>  
> 
> i configure ccm 4.1.3  auto attendant . but there is no BROADCAST
option 
> for pilot point.
> 
> only 2 options  which are    first available and longest idle.
> 
>  
> 
> but according to 4.1.3 ccm services and features document there should

> be broadcast option also
> 
>  
> 
> what am i missing
> 
>  
> 
> thanks
> 
> Gokhan
> 
>  
> 
> 
>
------------------------------------------------------------------------
> Don't be flakey. Get Yahoo! Mail for Mobile 
> <http://us.rd.yahoo.com/evt=43909/*http://mobile.yahoo.com/mail> and
> always stay connected 
> <http://us.rd.yahoo.com/evt=43909/*http://mobile.yahoo.com/mail> to
friends.
> 
> 
>
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> https://puck.nether.net/mailman/listinfo/cisco-voip

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From Scott.Voll at wesd.org  Wed May  9 12:39:28 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 9 May 2007 09:39:28 -0700
Subject: [cisco-voip] which ios to use for 3845
In-Reply-To: <014901c79178$b1d6dce0$47196883@cfs.uoguelph.ca>
References: <014901c79178$b1d6dce0$47196883@cfs.uoguelph.ca>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE734@Apollo.wesd.org>

Personally I like to start with the newest mainline code.  But I don't
know if the 3845 has a mainline code yet.

 

I guess the newest T train.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, May 08, 2007 6:57 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] which ios to use for 3845

 

my 3845 was just dropped off. 

 

what IOS are people recommending for this thing? primarily used as a
voice gateway (2 T1/PRIs using MGCP) and for SRST. I don't think I will
be doing much transcoding, (other than from PSTN to voip) or
conferencing.

 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

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From jason.aarons at us.didata.com  Wed May  9 13:36:40 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 9 May 2007 13:36:40 -0400
Subject: [cisco-voip] IP-IVR  JTAPI setup...
In-Reply-To: <001e01c79244$03f863b0$74c16640@amer.cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F00580189E@USNAEXCH.na.didata.local><70E40EF625463544B814B0819734EF2C186E1A@cecemx002.corporate.careered.com>
	<C1FE15183DA37645BC0633BC604E44F0058385EF@USNAEXCH.na.didata.local>
	<001e01c79244$03f863b0$74c16640@amer.cisco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005838D30@USNAEXCH.na.didata.local>

4.1(3)SR4d, no ES -jason

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Wednesday, May 09, 2007 10:12 AM
To: Jason Aarons (US); 'Chris Beck - CEC'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IP-IVR JTAPI setup...

What version of CM? 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Tuesday, May 08, 2007 10:14 PM
To: Chris Beck - CEC; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IP-IVR JTAPI setup...

AD integrated, we can login to subs CCMAdmin using AD accounts that were
setup in CCMAdmin\MLA, just can't use ccmadministrator on subs -jason

-----Original Message-----
From: Chris Beck - CEC [mailto:CBeck at careered.com]
Sent: Tuesday, May 08, 2007 9:40 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IP-IVR JTAPI setup...

I am taking a bit of a guess here, but it sounds like the dcdirectory
replication is broken between callmanagers.  
 
depending on the size of the ldap directory store, there is a script
file
called reconfig_cluster on the server that will rebuild replication.
Make
sure there are no special characters in the dcdirectory password
specified
during rebuild.
 
I found a good document on CCO about how to fix this.  I will see if I
can
find it agian.
 
-Chris

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
Sent: Tue 5/8/2007 12:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IP-IVR JTAPI setup...



 

In the "JTAPI Provider" section of the IP-IVR configuration, we normally
enter the CallManager machines that will be handling the JTAPI duties.
When you enter one or more entries, you get a pop-up that asks for the
CallManager Administration credentials.  This works fine when I use the
Publisher, but it fails with any Subscriber (we have 4 other
subscribers).
And sure enough, if I try to log into CallManager Administration on
either
of the Subscriber machines with the same "ccmadministrator" account and
password, it does not work.

 

 

________________________________

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From moody at arizona.edu  Wed May  9 14:04:01 2007
From: moody at arizona.edu (Walt Moody)
Date: Wed, 09 May 2007 11:04:01 -0700
Subject: [cisco-voip] Area Code vs NXX Conflict
Message-ID: <46420D11.4050604@arizona.edu>

Here's the setup:

The University of Arizona - in Tucson - uses seven digit dialing for
both on Campus and off Campus local calls.  We do not use 9+ or any
other PBX "outside access" codes.  NANP long distance is 1+ 10 digits,
IDDD is 011+.

Except:  When dialing from our Tucson Campus to our Phoenix Campus, we
dial 602827XXXX.  We do not use a "1" or any other prefix digit to when
we dial a "free" or on-net call.

Still with me?   OK, try this: We want to set up our dial plans so that
we can use 10 digits to call from the Tucson Campus to anywhere in the
Phoenix rate center, which consists of Area Codes 480, 602, and 623.
That's easy enough to do, except (there's that word again) 623 is both
an Area Code in Phoenix and an NXX in Tucson.

I need to find a way to tell the difference between, for example, the
dialed digits (623) 555-1212 and 623-5551 and route the 10 digit "623"
numbers to our PRIs in Phoenix and the 7 digit "623" numbers to the
local Tucson PRIs.

Prefix digits are not an option -- there are several PBXs in addition
to the CallManagers that are part of the University's voice services,
and no real good way to change them from their current dial plans.

Ideas, anyone?

Thanks,

-walt

From lelio at uoguelph.ca  Wed May  9 14:13:03 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 9 May 2007 14:13:03 -0400
Subject: [cisco-voip] Area Code vs NXX Conflict
References: <46420D11.4050604@arizona.edu>
Message-ID: <008101c79265$af6236b0$47196883@cfs.uoguelph.ca>

If you can live with interdigit timeout, then you should be able to program things so that you can dial either number and get through. Users will have to wait 10 seconds (or whatever your parameter is configured for) before 7 digit calls are placed -or- if I'm not mistaken, they can press # to complete the dialing process (you might need an extra route pattern with the # at the end though as well).

Other than that, if you are looking to be able to dial a 7 digit and 10 digit call with the same beginning three digits and not have interdigit timeout at the end, I do not think you will be able to do this.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Walt Moody 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, May 09, 2007 2:04 PM
  Subject: [cisco-voip] Area Code vs NXX Conflict


  Here's the setup:

  The University of Arizona - in Tucson - uses seven digit dialing for
  both on Campus and off Campus local calls.  We do not use 9+ or any
  other PBX "outside access" codes.  NANP long distance is 1+ 10 digits,
  IDDD is 011+.

  Except:  When dialing from our Tucson Campus to our Phoenix Campus, we
  dial 602827XXXX.  We do not use a "1" or any other prefix digit to when
  we dial a "free" or on-net call.

  Still with me?   OK, try this: We want to set up our dial plans so that
  we can use 10 digits to call from the Tucson Campus to anywhere in the
  Phoenix rate center, which consists of Area Codes 480, 602, and 623.
  That's easy enough to do, except (there's that word again) 623 is both
  an Area Code in Phoenix and an NXX in Tucson.

  I need to find a way to tell the difference between, for example, the
  dialed digits (623) 555-1212 and 623-5551 and route the 10 digit "623"
  numbers to our PRIs in Phoenix and the 7 digit "623" numbers to the
  local Tucson PRIs.

  Prefix digits are not an option -- there are several PBXs in addition
  to the CallManagers that are part of the University's voice services,
  and no real good way to change them from their current dial plans.

  Ideas, anyone?

  Thanks,

  -walt
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From bprentiss at gmail.com  Wed May  9 14:23:17 2007
From: bprentiss at gmail.com (Brian Prentiss)
Date: Wed, 9 May 2007 11:23:17 -0700
Subject: [cisco-voip] Area Code vs NXX Conflict
In-Reply-To: <008101c79265$af6236b0$47196883@cfs.uoguelph.ca>
References: <46420D11.4050604@arizona.edu>
	<008101c79265$af6236b0$47196883@cfs.uoguelph.ca>
Message-ID: <7e82b2130705091123j9172c3bq3636182ff1ac516a@mail.gmail.com>

The thing is with the first option here, is that potentially you would have
a similiar situation with the other area codes at some point, as the
population growth forces more NXX assignments in the 520 area code.  Also,
as people wait for that interdigit timeout, they might be thinking "did it
go through?" and that might be a bit of a challenge support wise.

Are you planning on doing the same with the 928 area code that is in the
other part of the state?

Regards,
Brian


On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  If you can live with interdigit timeout, then you should be able to
> program things so that you can dial either number and get through. Users
> will have to wait 10 seconds (or whatever your parameter is configured for)
> before 7 digit calls are placed -or- if I'm not mistaken, they can press #
> to complete the dialing process (you might need an extra route pattern with
> the # at the end though as well).
>
> Other than that, if you are looking to be able to dial a 7 digit and 10
> digit call with the same beginning three digits and not have interdigit
> timeout at the end, I do not think you will be able to do this.
>
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
> ----- Original Message -----
> *From:* Walt Moody <moody at arizona.edu>
> *To:* cisco-voip at puck.nether.net
> *Sent:* Wednesday, May 09, 2007 2:04 PM
> *Subject:* [cisco-voip] Area Code vs NXX Conflict
>
>
> Here's the setup:
>
> The University of Arizona - in Tucson - uses seven digit dialing for
> both on Campus and off Campus local calls.  We do not use 9+ or any
> other PBX "outside access" codes.  NANP long distance is 1+ 10 digits,
> IDDD is 011+.
>
> Except:  When dialing from our Tucson Campus to our Phoenix Campus, we
> dial 602827XXXX.  We do not use a "1" or any other prefix digit to when
> we dial a "free" or on-net call.
>
> Still with me?   OK, try this: We want to set up our dial plans so that
> we can use 10 digits to call from the Tucson Campus to anywhere in the
> Phoenix rate center, which consists of Area Codes 480, 602, and 623.
> That's easy enough to do, except (there's that word again) 623 is both
> an Area Code in Phoenix and an NXX in Tucson.
>
> I need to find a way to tell the difference between, for example, the
> dialed digits (623) 555-1212 and 623-5551 and route the 10 digit "623"
> numbers to our PRIs in Phoenix and the 7 digit "623" numbers to the
> local Tucson PRIs.
>
> Prefix digits are not an option -- there are several PBXs in addition
> to the CallManagers that are part of the University's voice services,
> and no real good way to change them from their current dial plans.
>
> Ideas, anyone?
>
> Thanks,
>
> -walt
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From GRhodes at rbh.com  Wed May  9 15:44:58 2007
From: GRhodes at rbh.com (Rhodes, Geoff)
Date: Wed, 9 May 2007 15:44:58 -0400
Subject: [cisco-voip] Echo on IP Phones through PRI
In-Reply-To: <418387.45400.qm@web37209.mail.mud.yahoo.com>
References: <418387.45400.qm@web37209.mail.mud.yahoo.com>
Message-ID: <6350516B6159444E9D646523EB4526F403EB9FFD@apollo.rbh.local>

Please excuse the newbie question, but how do I determine the DSP
version my MGCP gateway is running?  It's a CMM blade in our 6509
switch.  The IOS on it looks to be 12.4(8a).  I'd like to document what
release we are running for future needs.

We get complaints ever once in a while about echo problems.  Most of the
time we blow them off unless someone is constantly complaining.

Here are our settings in CCMAdmin for the CMM Gateway:
Echo Cancellation Enable = Enable
Echo Cancel Coverage = 32
Input Gain = 3
Output Attenuation = -3

Thanks for your help. 

Geoff Rhodes
Director, Information Technology
Robinson, Bradshaw & Hinson, P.A.
704 / 377-8188
grhodes at rbh.com
 


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
Sent: Wednesday, April 18, 2007 9:12 AM
To: Matthew Melbourne; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Echo on IP Phones through PRI

DSP Fimware 4.4.708 has issues which can cause echo, etc. Many people on
the cisco NetPro forums have ran into echo issues with this DSP
firmware.  I don't have the exact bug id's handy, but you can try
upgrading to a current 12.4 mainline release if your routers DRAM/flash
can take it. 12.4(13a) has 4.4.23 DSP ware. So far, I have not ran into
any issues yet with DSP firmware 4.4.21 or higher in 12.4 mainline but
YMMV. 

If you can't update the IOS, and I don't believe there is a newer 12.3T
release with a newer firmware embedded in it, you can get a standalone
DSP firmware image file to put on the router flash that is a newer
version from TAC. 

As far as finding out what gateway a call is using, probably the
quickest way is to go to the phones web page while the call is connected
and look at the stream page for the remote IP address. Other then that,
you would need to check out CDR and/or QRT records if they use QRT
softkey, etc and go into each gateway. 

----- Original Message ----
From: Matthew Melbourne <matt at melbourne.org.uk>
To: cisco-voip at puck.nether.net
Sent: Wednesday, April 18, 2007 7:50:22 AM
Subject: [cisco-voip] Echo on IP Phones through PRI

We are getting sporadic issues from users on IP phones who are hearing
echo on external calls. From this document is would suggest that echo on
IP Phones is caused by hybrid echo and not any local acoustic echo.

http://www.cisco.com/warp/public/788/voice-qos/echo_ipphone_gateway.html

The MGCP-controlled voice gateways are all running 12.3(14)T7 with DSP
firmware 4.4.708. In CallManager (4.1(3)SR2), echo cancellation is on
for each port, with a "default" echo cancellation coverage of 64ms.
Input gain and output attenuation are set to 0dB.

I have seen suggestions to include the "echo-cancel erl worst-case 3"
command on the voice ports. Can this help in situations of occasional
echo?

Are there any other parameters changes which could help? Could it be a
code/DSP firmware issue? I haven't found a specific echo-related bug
relating to this particular IOS and its embedded DSP firmware.

On occasions, the start of the call is fine but echo comes and goes
during the call.

Is there a quick way of finding out which voice gateway a call is
terminated on (we have 8, and all could be used in a round-robin fashion
for outbound calls).

Cheers,

Matt

--
Matthew Melbourne
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From jason.aarons at us.didata.com  Wed May  9 15:51:58 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 9 May 2007 15:51:58 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
Message-ID: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>

I'm trying to find a reference about server hostname changes for
production CallManager 4.1, I recall it's not advised but can't find a
article on cisco.com stating this -jason

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 



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From danie.dutoit at us.didata.com  Wed May  9 15:55:21 2007
From: danie.dutoit at us.didata.com (Danie Du Toit (US))
Date: Wed, 9 May 2007 15:55:21 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>
Message-ID: <3D2A9A1D2B2F0E49B70E6E882E89890B02C5845A@usbossvex01.na.didata.local>


Unable to Change the Server Name for Cisco CallManager


 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/prod_troubleshooting_guide_chapter09186a00803fa956.html#wp1037996


________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Wed 5/9/2007 3:51 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?



I'm trying to find a reference about server hostname changes for production CallManager 4.1, I recall it's not advised but can't find a article on cisco.com stating this -jason

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact the Dimension Data NOC at 800-974-6584.

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 



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From rratliff at cisco.com  Wed May  9 15:58:13 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 9 May 2007 15:58:13 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>
Message-ID: <002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>

No way, no how.

The only time you can change the hostname of a CM server is before  
you install CallManager on it.

-Ryan

On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:

I?m trying to find a reference about server hostname changes for  
production CallManager 4.1, I recall it?s not advised but can?t find  
a article on cisco.com stating this -jason





Jason Aarons

Consultant

http://www.dimensiondata.com/na

904-338-3245 cell



For urgent issues notify your Project Manager, for 24x7 support  
contact the Dimension Data NOC at 800-974-6584.





Disclaimer: This e-mail communication and any attachments may contain  
confidential and privileged information and is for use by the  
designated addressee(s) named above only. If you are not the intended  
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computer. Thank you.

_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Matt.Slaga at us.didata.com  Wed May  9 16:05:24 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 9 May 2007 16:05:24 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
In-Reply-To: <002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>
	<002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00583915E@USNAEXCH.na.didata.local>

If the customer really pushes you, create a DNS C-name record.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, May 09, 2007 3:58 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?

No way, no how.

The only time you can change the hostname of a CM server is before  
you install CallManager on it.

-Ryan

On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:

I'm trying to find a reference about server hostname changes for  
production CallManager 4.1, I recall it's not advised but can't find  
a article on cisco.com stating this -jason





Jason Aarons

Consultant

http://www.dimensiondata.com/na

904-338-3245 cell



For urgent issues notify your Project Manager, for 24x7 support  
contact the Dimension Data NOC at 800-974-6584.





Disclaimer: This e-mail communication and any attachments may contain  
confidential and privileged information and is for use by the  
designated addressee(s) named above only. If you are not the intended  
addressee, you are hereby notified that you have received this  
communication in error and that any use or reproduction of this email  
or its contents is strictly prohibited and may be unlawful. If you  
have received this communication in error, please notify us  
immediately by replying to this message and deleting it from your  
computer. Thank you.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip
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From pmowry at getgds.com  Wed May  9 16:18:10 2007
From: pmowry at getgds.com (Patrick Mowry)
Date: Wed, 9 May 2007 15:18:10 -0500
Subject: [cisco-voip] Area Code vs NXX Conflict
In-Reply-To: <mailman.13196.1178740589.1489.cisco-voip@puck.nether.net>
References: <mailman.13196.1178740589.1489.cisco-voip@puck.nether.net>
Message-ID: <4C792A099259634C83F97067671B37FE02EFD79B@GDSEXC.gds.pri>

"We do not use 9+ or any other PBX "outside access" codes.  NANP long
distance is 1+ 10 digits, IDDD is 011+."  

  With this in mind, wouldn't it be natural for your users to dial a 1
in front of phoenix calls?  So instead of educating your users to dial
just the 10 digits for Phoenix calls, would it not be both easier and
have less support calls if you let the users dial the 1 and then throw
it away when you don't need it? Either that or require 10 digits for all
"free" calls.

You are only saving them from dialing 1 digit and I think it would be
easier to let them place calls like they would naturally dial at home.

----------------------------------------------------------------------

Message: 1
Date: Wed, 9 May 2007 11:23:17 -0700
From: "Brian Prentiss" <bprentiss at gmail.com>
Subject: Re: [cisco-voip] Area Code vs NXX Conflict
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Cc: cisco-voip at puck.nether.net
Message-ID:
	<7e82b2130705091123j9172c3bq3636182ff1ac516a at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

The thing is with the first option here, is that potentially you would
have
a similiar situation with the other area codes at some point, as the
population growth forces more NXX assignments in the 520 area code.
Also,
as people wait for that interdigit timeout, they might be thinking "did
it
go through?" and that might be a bit of a challenge support wise.

Are you planning on doing the same with the 928 area code that is in the
other part of the state?

Regards,
Brian



From moody at arizona.edu  Wed May  9 16:18:40 2007
From: moody at arizona.edu (Walt Moody)
Date: Wed, 09 May 2007 13:18:40 -0700
Subject: [cisco-voip] Area Code vs NXX Conflict
In-Reply-To: <7e82b2130705091123j9172c3bq3636182ff1ac516a@mail.gmail.com>
References: <46420D11.4050604@arizona.edu>	
	<008101c79265$af6236b0$47196883@cfs.uoguelph.ca>
	<7e82b2130705091123j9172c3bq3636182ff1ac516a@mail.gmail.com>
Message-ID: <46422CA0.9000805@arizona.edu>

Brian,

I'm aware of the potential for future conflicts.  Right now, the only
conflict is 623.  If at some far in the future date we have PRIs in
928, and 928 is in conflict, we'll have to deal with it.

Lelio,

Interdigit time outs could be a problem, but the more pressing problem 
is routes.  How do you tell 623[2-9]XXXXXX to go to the PSTN trunks in
Phoenix, and 623[2-9]XXX to go to Tucson's PSTN trunks?  There doesn't
seem to be a way to build a conflict table with different routing for
seven digits versus ten.

-walt

Brian Prentiss wrote:
> The thing is with the first option here, is that potentially you would 
> have a similiar situation with the other area codes at some point, as 
> the population growth forces more NXX assignments in the 520 area code.  
> Also, as people wait for that interdigit timeout, they might be thinking 
> "did it go through?" and that might be a bit of a challenge support wise.
>  
> Are you planning on doing the same with the 928 area code that is in the 
> other part of the state?
>  
> Regards,
> Brian
> 
>  
> On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca <mailto:lelio at uoguelph.ca>> 
> wrote:
> 
>     If you can live with interdigit timeout, then you should be able to
>     program things so that you can dial either number and get through.
>     Users will have to wait 10 seconds (or whatever your parameter is
>     configured for) before 7 digit calls are placed -or- if I'm not
>     mistaken, they can press # to complete the dialing process (you
>     might need an extra route pattern with the # at the end though as
>     well).
>      
>     Other than that, if you are looking to be able to dial a 7 digit and
>     10 digit call with the same beginning three digits and not have
>     interdigit timeout at the end, I do not think you will be able to do
>     this.
>      
>      
>     --------------------------------------------------------------------------------
>     Lelio Fulgenzi, B.A.
>     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
>     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>     ...there's no such thing as a bad timbit...
> 
>         ----- Original Message -----
>         From: Walt Moody <mailto:moody at arizona.edu>
>         To: cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>         Sent: Wednesday, May 09, 2007 2:04 PM
>         Subject: [cisco-voip] Area Code vs NXX Conflict
> 
>          
>         Here's the setup:
> 
>         The University of Arizona - in Tucson - uses seven digit dialing for
>         both on Campus and off Campus local calls.  We do not use 9+ or any
>         other PBX "outside access" codes.  NANP long distance is 1+ 10
>         digits,
>         IDDD is 011+.
> 
>         Except:  When dialing from our Tucson Campus to our Phoenix
>         Campus, we
>         dial 602827XXXX.  We do not use a "1" or any other prefix digit
>         to when
>         we dial a "free" or on-net call.
> 
>         Still with me?   OK, try this: We want to set up our dial plans
>         so that
>         we can use 10 digits to call from the Tucson Campus to anywhere
>         in the
>         Phoenix rate center, which consists of Area Codes 480, 602, and
>         623.
>         That's easy enough to do, except (there's that word again) 623
>         is both
>         an Area Code in Phoenix and an NXX in Tucson.
> 
>         I need to find a way to tell the difference between, for
>         example, the
>         dialed digits (623) 555-1212 and 623-5551 and route the 10 digit
>         "623"
>         numbers to our PRIs in Phoenix and the 7 digit "623" numbers to the
>         local Tucson PRIs.
> 
>         Prefix digits are not an option -- there are several PBXs in
>         addition
>         to the CallManagers that are part of the University's voice
>         services,
>         and no real good way to change them from their current dial plans.
> 
>         Ideas, anyone?
> 
>         Thanks,
> 
>         -walt
>         _______________________________________________
>         cisco-voip mailing list
>         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>         https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 

From lelio at uoguelph.ca  Wed May  9 16:23:49 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 9 May 2007 16:23:49 -0400
Subject: [cisco-voip] Area Code vs NXX Conflict
References: <46420D11.4050604@arizona.edu>	<008101c79265$af6236b0$47196883@cfs.uoguelph.ca><7e82b2130705091123j9172c3bq3636182ff1ac516a@mail.gmail.com>
	<46422CA0.9000805@arizona.edu>
Message-ID: <014301c79277$f44d5220$47196883@cfs.uoguelph.ca>

Routing is not a problem, the system will always route to the most precise. if you put in two route patterns one with 7 digits and one with 10, the calls will go where you want them to go.
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Walt Moody 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, May 09, 2007 4:18 PM
  Subject: Re: [cisco-voip] Area Code vs NXX Conflict


  Brian,

  I'm aware of the potential for future conflicts.  Right now, the only
  conflict is 623.  If at some far in the future date we have PRIs in
  928, and 928 is in conflict, we'll have to deal with it.

  Lelio,

  Interdigit time outs could be a problem, but the more pressing problem 
  is routes.  How do you tell 623[2-9]XXXXXX to go to the PSTN trunks in
  Phoenix, and 623[2-9]XXX to go to Tucson's PSTN trunks?  There doesn't
  seem to be a way to build a conflict table with different routing for
  seven digits versus ten.

  -walt

  Brian Prentiss wrote:
  > The thing is with the first option here, is that potentially you would 
  > have a similiar situation with the other area codes at some point, as 
  > the population growth forces more NXX assignments in the 520 area code.  
  > Also, as people wait for that interdigit timeout, they might be thinking 
  > "did it go through?" and that might be a bit of a challenge support wise.
  >  
  > Are you planning on doing the same with the 928 area code that is in the 
  > other part of the state?
  >  
  > Regards,
  > Brian
  > 
  >  
  > On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca <mailto:lelio at uoguelph.ca>> 
  > wrote:
  > 
  >     If you can live with interdigit timeout, then you should be able to
  >     program things so that you can dial either number and get through.
  >     Users will have to wait 10 seconds (or whatever your parameter is
  >     configured for) before 7 digit calls are placed -or- if I'm not
  >     mistaken, they can press # to complete the dialing process (you
  >     might need an extra route pattern with the # at the end though as
  >     well).
  >      
  >     Other than that, if you are looking to be able to dial a 7 digit and
  >     10 digit call with the same beginning three digits and not have
  >     interdigit timeout at the end, I do not think you will be able to do
  >     this.
  >      
  >      
  >     --------------------------------------------------------------------------------
  >     Lelio Fulgenzi, B.A.
  >     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >     ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  >     ...there's no such thing as a bad timbit...
  > 
  >         ----- Original Message -----
  >         From: Walt Moody <mailto:moody at arizona.edu>
  >         To: cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >         Sent: Wednesday, May 09, 2007 2:04 PM
  >         Subject: [cisco-voip] Area Code vs NXX Conflict
  > 
  >          
  >         Here's the setup:
  > 
  >         The University of Arizona - in Tucson - uses seven digit dialing for
  >         both on Campus and off Campus local calls.  We do not use 9+ or any
  >         other PBX "outside access" codes.  NANP long distance is 1+ 10
  >         digits,
  >         IDDD is 011+.
  > 
  >         Except:  When dialing from our Tucson Campus to our Phoenix
  >         Campus, we
  >         dial 602827XXXX.  We do not use a "1" or any other prefix digit
  >         to when
  >         we dial a "free" or on-net call.
  > 
  >         Still with me?   OK, try this: We want to set up our dial plans
  >         so that
  >         we can use 10 digits to call from the Tucson Campus to anywhere
  >         in the
  >         Phoenix rate center, which consists of Area Codes 480, 602, and
  >         623.
  >         That's easy enough to do, except (there's that word again) 623
  >         is both
  >         an Area Code in Phoenix and an NXX in Tucson.
  > 
  >         I need to find a way to tell the difference between, for
  >         example, the
  >         dialed digits (623) 555-1212 and 623-5551 and route the 10 digit
  >         "623"
  >         numbers to our PRIs in Phoenix and the 7 digit "623" numbers to the
  >         local Tucson PRIs.
  > 
  >         Prefix digits are not an option -- there are several PBXs in
  >         addition
  >         to the CallManagers that are part of the University's voice
  >         services,
  >         and no real good way to change them from their current dial plans.
  > 
  >         Ideas, anyone?
  > 
  >         Thanks,
  > 
  >         -walt
  >         _______________________________________________
  >         cisco-voip mailing list
  >         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >         https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  >     _______________________________________________
  >     cisco-voip mailing list
  >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >     https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Scott.Voll at wesd.org  Wed May  9 16:35:46 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 9 May 2007 13:35:46 -0700
Subject: [cisco-voip] Area Code vs NXX Conflict
In-Reply-To: <46422CA0.9000805@arizona.edu>
References: <46420D11.4050604@arizona.edu>	<008101c79265$af6236b0$47196883@cfs.uoguelph.ca><7e82b2130705091123j9172c3bq3636182ff1ac516a@mail.gmail.com>
	<46422CA0.9000805@arizona.edu>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE738@Apollo.wesd.org>

It's a lot of data entry...... but put one in for every prefix.

6233[2-3]xxxxx
6234[5,7]xxxxx

Etc.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Walt Moody
Sent: Wednesday, May 09, 2007 1:19 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Area Code vs NXX Conflict

Brian,

I'm aware of the potential for future conflicts.  Right now, the only
conflict is 623.  If at some far in the future date we have PRIs in
928, and 928 is in conflict, we'll have to deal with it.

Lelio,

Interdigit time outs could be a problem, but the more pressing problem 
is routes.  How do you tell 623[2-9]XXXXXX to go to the PSTN trunks in
Phoenix, and 623[2-9]XXX to go to Tucson's PSTN trunks?  There doesn't
seem to be a way to build a conflict table with different routing for
seven digits versus ten.

-walt

Brian Prentiss wrote:
> The thing is with the first option here, is that potentially you would

> have a similiar situation with the other area codes at some point, as 
> the population growth forces more NXX assignments in the 520 area
code.  
> Also, as people wait for that interdigit timeout, they might be
thinking 
> "did it go through?" and that might be a bit of a challenge support
wise.
>  
> Are you planning on doing the same with the 928 area code that is in
the 
> other part of the state?
>  
> Regards,
> Brian
> 
>  
> On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca
<mailto:lelio at uoguelph.ca>> 
> wrote:
> 
>     If you can live with interdigit timeout, then you should be able
to
>     program things so that you can dial either number and get through.
>     Users will have to wait 10 seconds (or whatever your parameter is
>     configured for) before 7 digit calls are placed -or- if I'm not
>     mistaken, they can press # to complete the dialing process (you
>     might need an extra route pattern with the # at the end though as
>     well).
>      
>     Other than that, if you are looking to be able to dial a 7 digit
and
>     10 digit call with the same beginning three digits and not have
>     interdigit timeout at the end, I do not think you will be able to
do
>     this.
>      
>      
>
------------------------------------------------------------------------
--------
>     Lelio Fulgenzi, B.A.
>     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G
2W1
>     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>     ...there's no such thing as a bad timbit...
> 
>         ----- Original Message -----
>         From: Walt Moody <mailto:moody at arizona.edu>
>         To: cisco-voip at puck.nether.net
<mailto:cisco-voip at puck.nether.net>
>         Sent: Wednesday, May 09, 2007 2:04 PM
>         Subject: [cisco-voip] Area Code vs NXX Conflict
> 
>          
>         Here's the setup:
> 
>         The University of Arizona - in Tucson - uses seven digit
dialing for
>         both on Campus and off Campus local calls.  We do not use 9+
or any
>         other PBX "outside access" codes.  NANP long distance is 1+ 10
>         digits,
>         IDDD is 011+.
> 
>         Except:  When dialing from our Tucson Campus to our Phoenix
>         Campus, we
>         dial 602827XXXX.  We do not use a "1" or any other prefix
digit
>         to when
>         we dial a "free" or on-net call.
> 
>         Still with me?   OK, try this: We want to set up our dial
plans
>         so that
>         we can use 10 digits to call from the Tucson Campus to
anywhere
>         in the
>         Phoenix rate center, which consists of Area Codes 480, 602,
and
>         623.
>         That's easy enough to do, except (there's that word again) 623
>         is both
>         an Area Code in Phoenix and an NXX in Tucson.
> 
>         I need to find a way to tell the difference between, for
>         example, the
>         dialed digits (623) 555-1212 and 623-5551 and route the 10
digit
>         "623"
>         numbers to our PRIs in Phoenix and the 7 digit "623" numbers
to the
>         local Tucson PRIs.
> 
>         Prefix digits are not an option -- there are several PBXs in
>         addition
>         to the CallManagers that are part of the University's voice
>         services,
>         and no real good way to change them from their current dial
plans.
> 
>         Ideas, anyone?
> 
>         Thanks,
> 
>         -walt
>         _______________________________________________
>         cisco-voip mailing list
>         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>         https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jrobbins at mercurypay.com  Wed May  9 16:39:45 2007
From: jrobbins at mercurypay.com (Jerri Robbins)
Date: Wed, 9 May 2007 14:39:45 -0600
Subject: [cisco-voip] Message waiting light on - no messages
Message-ID: <9015A93106739144995D6FD24A0F7A560868408C@hydrogen.mercurypay.com>

End users started having problems with their message waiting indicators
this a.m.  We restarted our unity server, but the lights are still on
with no messages.  Any suggestions?

 

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From garethsc at gmail.com  Wed May  9 16:46:21 2007
From: garethsc at gmail.com (Gareth Cafirma)
Date: Wed, 9 May 2007 16:46:21 -0400
Subject: [cisco-voip] Automatic CFAll?
Message-ID: <8dc3ea820705091346u73d211a5o1390b5d470678733@mail.gmail.com>

Hello Guys,

Is it possible to do a "automatic" call forward all to an auto attendant?
The client has a 5.1 CCM, unity connection 1.2. Here is the scenario,
operator leaves her desk and forward all calls to the nightbell, after 6 PM,
they want to remove the forward to nightbell and forward it to auto
attendant instead.
Can this be done? Any suggestions will be appreciated.

Gareth
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From lelio at uoguelph.ca  Wed May  9 16:49:10 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 9 May 2007 16:49:10 -0400
Subject: [cisco-voip] can't register new gateway
Message-ID: <016e01c7927b$7fe2c830$47196883@cfs.uoguelph.ca>

so i can't seem to ge this new 3845 gateway to register to my publisher. i can ping both ways (pub <-> gw) so communications is there. i must be missing some configuration command, but can seem to get it. i made sure to:
  a.. enable routing
  b.. make the description of the gateway equal the domain on the router
  c.. add the ccm-manager config statement
is there an mgcp command that i'm missing? config and show ccm-manager follows

grrrrrr

------------------------------------------------------------------------------


  !
  version 12.4
  service timestamps debug datetime localtime
  service timestamps log datetime localtime
  no service password-encryption
  !
  hostname vgw-kemptville
  !
  boot-start-marker
  boot-end-marker
  !
  card type t1 0 0
  card type t1 0 1
  logging buffered 4096 informational
  no logging console
  !
  aaa new-model
  !
  !
  aaa authentication login default group tacacs+ enable
  aaa authentication login virt-users group tacacs+ enable
  aaa authentication login console group tacacs+ enable
  !
  aaa session-id common
  clock timezone EST -5
  clock summer-time EDT recurring
  no network-clock-participate wic 0
  no network-clock-participate wic 1
  ip cef
  !
  !
  no ip dhcp use vrf connected
  !
  ip dhcp pool dhcp_vlan826
     network 10.104.126.0 255.255.255.128
     option 150 ip 10.104.21.121
     default-router 10.104.126.1
  !
  !
  ip domain name net.uoguelph.ca
  voice-card 0
   no dspfarm
  !
  !
  controller T1 0/0/0
   framing esf
   linecode b8zs
  !
  controller T1 0/1/0
   framing esf
   linecode b8zs
  !
  !
  !
  interface GigabitEthernet0/0
   description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE
   no ip address
   duplex auto
   speed auto
   media-type rj45
   no keepalive
  !
  interface GigabitEthernet0/0.129
   encapsulation dot1Q 129 native
   ip address 192.168.129.66 255.255.255.0
  !
  interface GigabitEthernet0/0.725
   encapsulation dot1Q 725
   ip address 10.104.125.99 255.255.255.128
  !
  interface GigabitEthernet0/0.826
   encapsulation dot1Q 826
   ip address 10.104.126.1 255.255.255.128
  !
  ip default-gateway 192.168.129.1
  ip route 0.0.0.0 0.0.0.0 192.168.129.1
  ip route 10.0.0.0 255.0.0.0 10.104.125.1
  !
  !
  !
  ccm-manager fallback-mgcp
  ccm-manager redundant-host 10.104.21.109
  ccm-manager mgcp
  ccm-manager music-on-hold
  ccm-manager config server 10.104.21.109
  ccm-manager config
  !
  mgcp

------------------------------------------------------------------------------

  vgw-kemptville#  show ccm-manager
  MGCP Domain Name: vgw-kemptville.net.uoguelph.ca
  Priority        Status                   Host
  ============================================================
  Primary         None
  First Backup    Down                     10.104.21.109
  Second Backup   None

  Current active Call Manager:    None
  Backhaul/Redundant link port:   2428
  Failover Interval:              30 seconds
  Keepalive Interval:             15 seconds
  Last keepalive sent:            16:33:45 EDT May 9 2007 (elapsed time: 00:14:33)
  Last MGCP traffic time:         16:34:18 EDT May 9 2007 (elapsed time: 00:14:01)
  Last failover time:             16:34:18 EDT May 9 2007 from (0.0.0.0)
  Last switchback time:           None
  Switchback mode:                Graceful
  MGCP Fallback mode:             Enabled/OFF
  Last MGCP Fallback start time:  None
  Last MGCP Fallback end time:    None
  MGCP Download Tones:            Disabled

  Backhaul/Redundant link is down
  Configuration Auto-Download Information
  =======================================
   No configurations downloaded
  Current state: Downloading XML file
  Configuration Download statistics:
          Download Attempted             : 1
            Download Successful          : 1
            Download Failed              : 0
          Configuration Attempted        : 0
            Configuration Successful     : 0
            Configuration Failed(Parsing): 0
            Configuration Failed(config) : 0
  Last config download command:
  Configuration Error History:
  FAX mode: cisco
  vgw-kemptville#

------------------------------------------------------------------------------

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From Scott.Voll at wesd.org  Wed May  9 16:49:27 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 9 May 2007 13:49:27 -0700
Subject: [cisco-voip] Message waiting light on - no messages
In-Reply-To: <9015A93106739144995D6FD24A0F7A560868408C@hydrogen.mercurypay.com>
References: <9015A93106739144995D6FD24A0F7A560868408C@hydrogen.mercurypay.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE739@Apollo.wesd.org>

Do a resync on Unity....... Depending on # of subscribers this could
take a while.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jerri Robbins
Sent: Wednesday, May 09, 2007 1:40 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Message waiting light on - no messages

 

End users started having problems with their message waiting indicators
this a.m.  We restarted our unity server, but the lights are still on
with no messages.  Any suggestions?

 

-------------- next part --------------
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From Scott.Voll at wesd.org  Wed May  9 16:54:39 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 9 May 2007 13:54:39 -0700
Subject: [cisco-voip] can't register new gateway
In-Reply-To: <016e01c7927b$7fe2c830$47196883@cfs.uoguelph.ca>
References: <016e01c7927b$7fe2c830$47196883@cfs.uoguelph.ca>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE73A@Apollo.wesd.org>

So you have a DNS entry for vgw-kemptville.uoguelph.ca = the ip address?

 

And the MGCP gateway is vgw-kemptville.uoguelph.ca?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, May 09, 2007 1:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] can't register new gateway

 

so i can't seem to ge this new 3845 gateway to register to my publisher.
i can ping both ways (pub <-> gw) so communications is there. i must be
missing some configuration command, but can seem to get it. i made sure
to:

*	enable routing
*	make the description of the gateway equal the domain on the
router
*	add the ccm-manager config statement

is there an mgcp command that i'm missing? config and show ccm-manager
follows

 

grrrrrr

	
________________________________


	
	!
	version 12.4
	service timestamps debug datetime localtime
	service timestamps log datetime localtime
	no service password-encryption
	!
	hostname vgw-kemptville
	!
	boot-start-marker
	boot-end-marker
	!
	card type t1 0 0
	card type t1 0 1
	logging buffered 4096 informational
	no logging console
	!
	aaa new-model
	!
	!
	aaa authentication login default group tacacs+ enable
	aaa authentication login virt-users group tacacs+ enable
	aaa authentication login console group tacacs+ enable
	!
	aaa session-id common
	clock timezone EST -5
	clock summer-time EDT recurring
	no network-clock-participate wic 0
	no network-clock-participate wic 1
	ip cef
	!
	!
	no ip dhcp use vrf connected
	!
	ip dhcp pool dhcp_vlan826
	   network 10.104.126.0 255.255.255.128
	   option 150 ip 10.104.21.121
	   default-router 10.104.126.1
	!
	!
	ip domain name net.uoguelph.ca
	voice-card 0
	 no dspfarm
	!
	!
	controller T1 0/0/0
	 framing esf
	 linecode b8zs
	!
	controller T1 0/1/0
	 framing esf
	 linecode b8zs
	!
	!
	!
	interface GigabitEthernet0/0
	 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE
	 no ip address
	 duplex auto
	 speed auto
	 media-type rj45
	 no keepalive
	!
	interface GigabitEthernet0/0.129
	 encapsulation dot1Q 129 native
	 ip address 192.168.129.66 255.255.255.0
	!
	interface GigabitEthernet0/0.725
	 encapsulation dot1Q 725
	 ip address 10.104.125.99 255.255.255.128
	!
	interface GigabitEthernet0/0.826
	 encapsulation dot1Q 826
	 ip address 10.104.126.1 255.255.255.128
	!
	ip default-gateway 192.168.129.1
	ip route 0.0.0.0 0.0.0.0 192.168.129.1
	ip route 10.0.0.0 255.0.0.0 10.104.125.1
	!
	!
	!
	ccm-manager fallback-mgcp
	ccm-manager redundant-host 10.104.21.109
	ccm-manager mgcp
	ccm-manager music-on-hold
	ccm-manager config server 10.104.21.109
	ccm-manager config
	!
	mgcp

	
________________________________


	vgw-kemptville#  show ccm-manager
	MGCP Domain Name: vgw-kemptville.net.uoguelph.ca
	Priority        Status                   Host
	============================================================
	Primary         None
	First Backup    Down                     10.104.21.109
	Second Backup   None

	 

	Current active Call Manager:    None
	Backhaul/Redundant link port:   2428
	Failover Interval:              30 seconds
	Keepalive Interval:             15 seconds
	Last keepalive sent:            16:33:45 EDT May 9 2007 (elapsed
time: 00:14:33)
	Last MGCP traffic time:         16:34:18 EDT May 9 2007 (elapsed
time: 00:14:01)
	Last failover time:             16:34:18 EDT May 9 2007 from
(0.0.0.0)
	Last switchback time:           None
	Switchback mode:                Graceful
	MGCP Fallback mode:             Enabled/OFF
	Last MGCP Fallback start time:  None
	Last MGCP Fallback end time:    None
	MGCP Download Tones:            Disabled

	 

	Backhaul/Redundant link is down
	Configuration Auto-Download Information
	=======================================
	 No configurations downloaded
	Current state: Downloading XML file
	Configuration Download statistics:
	        Download Attempted             : 1
	          Download Successful          : 1
	          Download Failed              : 0
	        Configuration Attempted        : 0
	          Configuration Successful     : 0
	          Configuration Failed(Parsing): 0
	          Configuration Failed(config) : 0
	Last config download command:
	Configuration Error History:
	FAX mode: cisco
	vgw-kemptville#

	
________________________________


------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

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From jsteinberg at gmail.com  Wed May  9 17:04:04 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 9 May 2007 17:04:04 -0400
Subject: [cisco-voip] can't register new gateway
In-Reply-To: <016e01c7927b$7fe2c830$47196883@cfs.uoguelph.ca>
References: <016e01c7927b$7fe2c830$47196883@cfs.uoguelph.ca>
Message-ID: <f1b65f880705091404i5ac6d666v82d28f2bdf579964@mail.gmail.com>

I don't see any of the MGCP commands in the config but I can't tell whether
they are missing or just cut off in your email.

On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  so i can't seem to ge this new 3845 gateway to register to my publisher.
> i can ping both ways (pub <-> gw) so communications is there. i must be
> missing some configuration command, but can seem to get it. i made sure to:
>
>    - enable routing
>    - make the description of the gateway equal the domain on the router
>    - add the ccm-manager config statement
>
> is there an mgcp command that i'm missing? config and show ccm-manager
> follows
>
> grrrrrr
>
>  ------------------------------
>
> !
> version 12.4
> service timestamps debug datetime localtime
> service timestamps log datetime localtime
> no service password-encryption
> !
> hostname vgw-kemptville
> !
> boot-start-marker
> boot-end-marker
> !
> card type t1 0 0
> card type t1 0 1
> logging buffered 4096 informational
> no logging console
> !
> aaa new-model
> !
> !
> aaa authentication login default group tacacs+ enable
> aaa authentication login virt-users group tacacs+ enable
> aaa authentication login console group tacacs+ enable
> !
> aaa session-id common
> clock timezone EST -5
> clock summer-time EDT recurring
> no network-clock-participate wic 0
> no network-clock-participate wic 1
> ip cef
> !
> !
> no ip dhcp use vrf connected
> !
> ip dhcp pool dhcp_vlan826
>    network 10.104.126.0 255.255.255.128
>    option 150 ip 10.104.21.121
>    default-router 10.104.126.1
> !
> !
> ip domain name net.uoguelph.ca
> voice-card 0
>  no dspfarm
> !
> !
> controller T1 0/0/0
>  framing esf
>  linecode b8zs
> !
> controller T1 0/1/0
>  framing esf
>  linecode b8zs
> !
> !
> !
> interface GigabitEthernet0/0
>  description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE
>  no ip address
>  duplex auto
>  speed auto
>  media-type rj45
>  no keepalive
> !
> interface GigabitEthernet0/0.129
>  encapsulation dot1Q 129 native
>  ip address 192.168.129.66 255.255.255.0
> !
> interface GigabitEthernet0/0.725
>  encapsulation dot1Q 725
>  ip address 10.104.125.99 255.255.255.128
> !
> interface GigabitEthernet0/0.826
>  encapsulation dot1Q 826
>  ip address 10.104.126.1 255.255.255.128
> !
> ip default-gateway 192.168.129.1
> ip route 0.0.0.0 0.0.0.0 192.168.129.1
> ip route 10.0.0.0 255.0.0.0 10.104.125.1
> !
> !
> !
> ccm-manager fallback-mgcp
> ccm-manager redundant-host 10.104.21.109
> ccm-manager mgcp
> ccm-manager music-on-hold
> ccm-manager config server 10.104.21.109
> ccm-manager config
> !
> mgcp
>  ------------------------------
>  vgw-kemptville#  show ccm-manager
> MGCP Domain Name: vgw-kemptville.net.uoguelph.ca
> Priority        Status                   Host
> ============================================================
> Primary         None
> First Backup    Down                     10.104.21.109
> Second Backup   None
>
> Current active Call Manager:    None
> Backhaul/Redundant link port:   2428
> Failover Interval:              30 seconds
> Keepalive Interval:             15 seconds
> Last keepalive sent:            16:33:45 EDT May 9 2007 (elapsed time:
> 00:14:33)
> Last MGCP traffic time:         16:34:18 EDT May 9 2007 (elapsed time:
> 00:14:01)
> Last failover time:             16:34:18 EDT May 9 2007 from (0.0.0.0)
> Last switchback time:           None
> Switchback mode:                Graceful
> MGCP Fallback mode:             Enabled/OFF
> Last MGCP Fallback start time:  None
> Last MGCP Fallback end time:    None
> MGCP Download Tones:            Disabled
>
> Backhaul/Redundant link is down
> Configuration Auto-Download Information
> =======================================
>  No configurations downloaded
> Current state: Downloading XML file
> Configuration Download statistics:
>         Download Attempted             : 1
>           Download Successful          : 1
>           Download Failed              : 0
>         Configuration Attempted        : 0
>           Configuration Successful     : 0
>           Configuration Failed(Parsing): 0
>           Configuration Failed(config) : 0
> Last config download command:
> Configuration Error History:
> FAX mode: cisco
> vgw-kemptville#
> ------------------------------
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From Scott.Voll at wesd.org  Wed May  9 17:09:31 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 9 May 2007 14:09:31 -0700
Subject: [cisco-voip] can't register new gateway
In-Reply-To: <f1b65f880705091404i5ac6d666v82d28f2bdf579964@mail.gmail.com>
References: <016e01c7927b$7fe2c830$47196883@cfs.uoguelph.ca>
	<f1b65f880705091404i5ac6d666v82d28f2bdf579964@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE73D@Apollo.wesd.org>

Good call...... where is your mgcp call-agent XXXXX statement?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Wednesday, May 09, 2007 2:04 PM
To: Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] can't register new gateway

 

I don't see any of the MGCP commands in the config but I can't tell
whether they are missing or just cut off in your email.

On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

so i can't seem to ge this new 3845 gateway to register to my publisher.
i can ping both ways (pub <-> gw) so communications is there. i must be
missing some configuration command, but can seem to get it. i made sure
to:

*	enable routing
*	make the description of the gateway equal the domain on the
router
*	add the ccm-manager config statement

is there an mgcp command that i'm missing? config and show ccm-manager
follows

 

grrrrrr

	
________________________________


	
	!
	version 12.4
	service timestamps debug datetime localtime
	service timestamps log datetime localtime
	no service password-encryption
	!
	hostname vgw-kemptville
	!
	boot-start-marker
	boot-end-marker
	!
	card type t1 0 0
	card type t1 0 1
	logging buffered 4096 informational
	no logging console
	!
	aaa new-model
	!
	!
	aaa authentication login default group tacacs+ enable
	aaa authentication login virt-users group tacacs+ enable
	aaa authentication login console group tacacs+ enable
	!
	aaa session-id common
	clock timezone EST -5
	clock summer-time EDT recurring
	no network-clock-participate wic 0
	no network-clock-participate wic 1
	ip cef
	!
	!
	no ip dhcp use vrf connected
	!
	ip dhcp pool dhcp_vlan826
	   network 10.104.126.0 255.255.255.128
	   option 150 ip 10.104.21.121
	   default-router 10.104.126.1
	!
	!
	ip domain name net.uoguelph.ca
	voice-card 0
	 no dspfarm
	!
	!
	controller T1 0/0/0
	 framing esf
	 linecode b8zs
	!
	controller T1 0/1/0
	 framing esf
	 linecode b8zs
	!
	!
	!
	interface GigabitEthernet0/0
	 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE
	 no ip address
	 duplex auto
	 speed auto
	 media-type rj45
	 no keepalive
	!
	interface GigabitEthernet0/0.129
	 encapsulation dot1Q 129 native
	 ip address 192.168.129.66 255.255.255.0
	!
	interface GigabitEthernet0/0.725
	 encapsulation dot1Q 725
	 ip address 10.104.125.99 255.255.255.128
	!
	interface GigabitEthernet0/0.826
	 encapsulation dot1Q 826
	 ip address 10.104.126.1 255.255.255.128
	!
	ip default-gateway 192.168.129.1
	ip route 0.0.0.0 0.0.0.0 192.168.129.1
	ip route 10.0.0.0 255.0.0.0 10.104.125.1
	!
	!
	!
	ccm-manager fallback-mgcp
	ccm-manager redundant-host 10.104.21.109
	ccm-manager mgcp
	ccm-manager music-on-hold
	ccm-manager config server 10.104.21.109
	ccm-manager config
	!
	mgcp

	
________________________________


	vgw-kemptville#  show ccm-manager
	MGCP Domain Name: vgw-kemptville.net.uoguelph.ca
	Priority        Status                   Host
	============================================================
	Primary         None
	First Backup    Down                     10.104.21.109
	Second Backup   None

	 

	Current active Call Manager:    None
	Backhaul/Redundant link port:   2428
	Failover Interval:              30 seconds
	Keepalive Interval:             15 seconds
	Last keepalive sent:            16:33:45 EDT May 9 2007 (elapsed
time: 00:14:33)
	Last MGCP traffic time:         16:34:18 EDT May 9 2007 (elapsed
time: 00:14:01)
	Last failover time:             16:34:18 EDT May 9 2007 from
(0.0.0.0)
	Last switchback time:           None
	Switchback mode:                Graceful
	MGCP Fallback mode:             Enabled/OFF
	Last MGCP Fallback start time:  None
	Last MGCP Fallback end time:    None
	MGCP Download Tones:            Disabled

	 

	Backhaul/Redundant link is down
	Configuration Auto-Download Information
	=======================================
	 No configurations downloaded
	Current state: Downloading XML file
	Configuration Download statistics:
	        Download Attempted             : 1
	          Download Successful          : 1
	          Download Failed              : 0
	        Configuration Attempted        : 0
	          Configuration Successful     : 0
	          Configuration Failed(Parsing): 0
	          Configuration Failed(config) : 0
	Last config download command:
	Configuration Error History:
	FAX mode: cisco
	vgw-kemptville#

	
________________________________


------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From axe7961 at hotmail.com  Wed May  9 17:34:39 2007
From: axe7961 at hotmail.com (Axel Wilzinski)
Date: Wed, 9 May 2007 17:34:39 -0400
Subject: [cisco-voip] Conference Bridge on IOS Router
Message-ID: <BAY103-W720D38C19CF741F4477F6BA3B0@phx.gbl>

Has anyone had any luck getting local conferencing working on an IOS gateway?  We are using MGCP with a remote Call Manager (4.2.3).  The local gateway has a PVDM2 with 32 DSP's.  Currently, conference calls traverse the WAN even for a local conference call.
 
Any help would be appreciated.
 
Thanks!
_________________________________________________________________
Create the ultimate e-mail address book. Import your contacts to Windows Live Hotmail.
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From pmowry at getgds.com  Wed May  9 17:35:06 2007
From: pmowry at getgds.com (Patrick Mowry)
Date: Wed, 9 May 2007 16:35:06 -0500
Subject: [cisco-voip] can't register new gateway
In-Reply-To: <mailman.13208.1178744647.1489.cisco-voip@puck.nether.net>
References: <mailman.13208.1178744647.1489.cisco-voip@puck.nether.net>
Message-ID: <4C792A099259634C83F97067671B37FE02EFD7B5@GDSEXC.gds.pri>

>From your config:
 no network-clock-participate wic 0
 no network-clock-participate wic 1

I think this may be keeping the controller configs from ccm-manager
download from taking effect.  Take the no out from in front and force
another ccm-manager download and see what happens.  Also you can do a
debug on the download to see it CallManager is responding with a config
file and verify the router is sending the device name the same as you
have configured in CallManager.  I do not believe DNS resolution is
required.

-Patrick
________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, May 09, 2007 1:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] can't register new gateway

 

so i can't seem to ge this new 3845 gateway to register to my publisher.
i can ping both ways (pub <-> gw) so communications is there. i must be
missing some configuration command, but can seem to get it. i made sure
to:

*	enable routing
*	make the description of the gateway equal the domain on the
router
*	add the ccm-manager config statement

is there an mgcp command that i'm missing? config and show ccm-manager
follows

 

grrrrrr


From jason.aarons at us.didata.com  Wed May  9 17:42:41 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 9 May 2007 17:42:41 -0400
Subject: [cisco-voip] Area Code vs NXX Conflict
In-Reply-To: <46422CA0.9000805@arizona.edu>
References: <46420D11.4050604@arizona.edu>	<008101c79265$af6236b0$47196883@cfs.uoguelph.ca><7e82b2130705091123j9172c3bq3636182ff1ac516a@mail.gmail.com>
	<46422CA0.9000805@arizona.edu>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00586BDF8@USNAEXCH.na.didata.local>

You would have to get a list of NPA-XXXs for that CO and expand the
route pattern. I've done this for TEHO (Tail End Hop Off) routing,
Atlanta had hundreds of NPA-XXXs...headache to keep up with..

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Walt Moody
Sent: Wednesday, May 09, 2007 4:19 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Area Code vs NXX Conflict

Brian,

I'm aware of the potential for future conflicts.  Right now, the only
conflict is 623.  If at some far in the future date we have PRIs in
928, and 928 is in conflict, we'll have to deal with it.

Lelio,

Interdigit time outs could be a problem, but the more pressing problem 
is routes.  How do you tell 623[2-9]XXXXXX to go to the PSTN trunks in
Phoenix, and 623[2-9]XXX to go to Tucson's PSTN trunks?  There doesn't
seem to be a way to build a conflict table with different routing for
seven digits versus ten.

-walt

Brian Prentiss wrote:
> The thing is with the first option here, is that potentially you would

> have a similiar situation with the other area codes at some point, as 
> the population growth forces more NXX assignments in the 520 area
code.  
> Also, as people wait for that interdigit timeout, they might be
thinking 
> "did it go through?" and that might be a bit of a challenge support
wise.
>  
> Are you planning on doing the same with the 928 area code that is in
the 
> other part of the state?
>  
> Regards,
> Brian
> 
>  
> On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca
<mailto:lelio at uoguelph.ca>> 
> wrote:
> 
>     If you can live with interdigit timeout, then you should be able
to
>     program things so that you can dial either number and get through.
>     Users will have to wait 10 seconds (or whatever your parameter is
>     configured for) before 7 digit calls are placed -or- if I'm not
>     mistaken, they can press # to complete the dialing process (you
>     might need an extra route pattern with the # at the end though as
>     well).
>      
>     Other than that, if you are looking to be able to dial a 7 digit
and
>     10 digit call with the same beginning three digits and not have
>     interdigit timeout at the end, I do not think you will be able to
do
>     this.
>      
>      
>
------------------------------------------------------------------------
--------
>     Lelio Fulgenzi, B.A.
>     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G
2W1
>     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>     ...there's no such thing as a bad timbit...
> 
>         ----- Original Message -----
>         From: Walt Moody <mailto:moody at arizona.edu>
>         To: cisco-voip at puck.nether.net
<mailto:cisco-voip at puck.nether.net>
>         Sent: Wednesday, May 09, 2007 2:04 PM
>         Subject: [cisco-voip] Area Code vs NXX Conflict
> 
>          
>         Here's the setup:
> 
>         The University of Arizona - in Tucson - uses seven digit
dialing for
>         both on Campus and off Campus local calls.  We do not use 9+
or any
>         other PBX "outside access" codes.  NANP long distance is 1+ 10
>         digits,
>         IDDD is 011+.
> 
>         Except:  When dialing from our Tucson Campus to our Phoenix
>         Campus, we
>         dial 602827XXXX.  We do not use a "1" or any other prefix
digit
>         to when
>         we dial a "free" or on-net call.
> 
>         Still with me?   OK, try this: We want to set up our dial
plans
>         so that
>         we can use 10 digits to call from the Tucson Campus to
anywhere
>         in the
>         Phoenix rate center, which consists of Area Codes 480, 602,
and
>         623.
>         That's easy enough to do, except (there's that word again) 623
>         is both
>         an Area Code in Phoenix and an NXX in Tucson.
> 
>         I need to find a way to tell the difference between, for
>         example, the
>         dialed digits (623) 555-1212 and 623-5551 and route the 10
digit
>         "623"
>         numbers to our PRIs in Phoenix and the 7 digit "623" numbers
to the
>         local Tucson PRIs.
> 
>         Prefix digits are not an option -- there are several PBXs in
>         addition
>         to the CallManagers that are part of the University's voice
>         services,
>         and no real good way to change them from their current dial
plans.
> 
>         Ideas, anyone?
> 
>         Thanks,
> 
>         -walt
>         _______________________________________________
>         cisco-voip mailing list
>         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>         https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.


From Scott.Voll at wesd.org  Wed May  9 18:42:08 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 9 May 2007 15:42:08 -0700
Subject: [cisco-voip] VGW + throughput
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE742@Apollo.wesd.org>

What should I expect for Throughput on a ISR VGW.        

 

2801 = ?mb

2811 = ?mb

2821 =

2851 =

3825 =

 

Plus they will be doing a couple pots lines and ~8 FXS/FXO ports, and
QoS . No T1's, ATM, Frame, etc.  just Ethernet IP.

 

Thanks

 

Scott

 

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From jason.aarons at us.didata.com  Wed May  9 19:44:32 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 9 May 2007 19:44:32 -0400
Subject: [cisco-voip] IOS strip digits from 8 to 4
Message-ID: <C1FE15183DA37645BC0633BC604E44F00586BF11@USNAEXCH.na.didata.local>

Telco is delivering 8 digits in Malaysia, I only need 4 digits
significant.  I don't want to ask them to change from 8 digits to 4.

 

This is all h323 dial-peers, some DIDs go to CallManager, some DIDs go
to CVP, etc. I can't use MGCP or do a translation-pattern in CCM.

 

How best should I reduce the 8 digits received to 4 via IOS Gateway?

 

DID Range 8808-87XX

 

num-exp 880887.. ....

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 



-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
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From lelio at uoguelph.ca  Wed May  9 19:55:39 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 9 May 2007 19:55:39 -0400
Subject: [cisco-voip] Area Code vs NXX Conflict
References: <46420D11.4050604@arizona.edu>	<008101c79265$af6236b0$47196883@cfs.uoguelph.ca><7e82b2130705091123j9172c3bq3636182ff1ac516a@mail.gmail.com><46422CA0.9000805@arizona.edu>
	<C1FE15183DA37645BC0633BC604E44F00586BDF8@USNAEXCH.na.didata.local>
Message-ID: <002a01c79295$8d9d7f00$47196883@cfs.uoguelph.ca>

I use this for this sort of stuff....

http://www.localcallingguide.com/

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Jason Aarons (US) 
  To: Walt Moody ; cisco-voip at puck.nether.net 
  Sent: Wednesday, May 09, 2007 5:42 PM
  Subject: Re: [cisco-voip] Area Code vs NXX Conflict


  You would have to get a list of NPA-XXXs for that CO and expand the
  route pattern. I've done this for TEHO (Tail End Hop Off) routing,
  Atlanta had hundreds of NPA-XXXs...headache to keep up with..

  -----Original Message-----
  From: cisco-voip-bounces at puck.nether.net
  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Walt Moody
  Sent: Wednesday, May 09, 2007 4:19 PM
  To: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Area Code vs NXX Conflict

  Brian,

  I'm aware of the potential for future conflicts.  Right now, the only
  conflict is 623.  If at some far in the future date we have PRIs in
  928, and 928 is in conflict, we'll have to deal with it.

  Lelio,

  Interdigit time outs could be a problem, but the more pressing problem 
  is routes.  How do you tell 623[2-9]XXXXXX to go to the PSTN trunks in
  Phoenix, and 623[2-9]XXX to go to Tucson's PSTN trunks?  There doesn't
  seem to be a way to build a conflict table with different routing for
  seven digits versus ten.

  -walt

  Brian Prentiss wrote:
  > The thing is with the first option here, is that potentially you would

  > have a similiar situation with the other area codes at some point, as 
  > the population growth forces more NXX assignments in the 520 area
  code.  
  > Also, as people wait for that interdigit timeout, they might be
  thinking 
  > "did it go through?" and that might be a bit of a challenge support
  wise.
  >  
  > Are you planning on doing the same with the 928 area code that is in
  the 
  > other part of the state?
  >  
  > Regards,
  > Brian
  > 
  >  
  > On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca
  <mailto:lelio at uoguelph.ca>> 
  > wrote:
  > 
  >     If you can live with interdigit timeout, then you should be able
  to
  >     program things so that you can dial either number and get through.
  >     Users will have to wait 10 seconds (or whatever your parameter is
  >     configured for) before 7 digit calls are placed -or- if I'm not
  >     mistaken, they can press # to complete the dialing process (you
  >     might need an extra route pattern with the # at the end though as
  >     well).
  >      
  >     Other than that, if you are looking to be able to dial a 7 digit
  and
  >     10 digit call with the same beginning three digits and not have
  >     interdigit timeout at the end, I do not think you will be able to
  do
  >     this.
  >      
  >      
  >
  ------------------------------------------------------------------------
  --------
  >     Lelio Fulgenzi, B.A.
  >     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G
  2W1
  >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  >     ...there's no such thing as a bad timbit...
  > 
  >         ----- Original Message -----
  >         From: Walt Moody <mailto:moody at arizona.edu>
  >         To: cisco-voip at puck.nether.net
  <mailto:cisco-voip at puck.nether.net>
  >         Sent: Wednesday, May 09, 2007 2:04 PM
  >         Subject: [cisco-voip] Area Code vs NXX Conflict
  > 
  >          
  >         Here's the setup:
  > 
  >         The University of Arizona - in Tucson - uses seven digit
  dialing for
  >         both on Campus and off Campus local calls.  We do not use 9+
  or any
  >         other PBX "outside access" codes.  NANP long distance is 1+ 10
  >         digits,
  >         IDDD is 011+.
  > 
  >         Except:  When dialing from our Tucson Campus to our Phoenix
  >         Campus, we
  >         dial 602827XXXX.  We do not use a "1" or any other prefix
  digit
  >         to when
  >         we dial a "free" or on-net call.
  > 
  >         Still with me?   OK, try this: We want to set up our dial
  plans
  >         so that
  >         we can use 10 digits to call from the Tucson Campus to
  anywhere
  >         in the
  >         Phoenix rate center, which consists of Area Codes 480, 602,
  and
  >         623.
  >         That's easy enough to do, except (there's that word again) 623
  >         is both
  >         an Area Code in Phoenix and an NXX in Tucson.
  > 
  >         I need to find a way to tell the difference between, for
  >         example, the
  >         dialed digits (623) 555-1212 and 623-5551 and route the 10
  digit
  >         "623"
  >         numbers to our PRIs in Phoenix and the 7 digit "623" numbers
  to the
  >         local Tucson PRIs.
  > 
  >         Prefix digits are not an option -- there are several PBXs in
  >         addition
  >         to the CallManagers that are part of the University's voice
  >         services,
  >         and no real good way to change them from their current dial
  plans.
  > 
  >         Ideas, anyone?
  > 
  >         Thanks,
  > 
  >         -walt
  >         _______________________________________________
  >         cisco-voip mailing list
  >         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >         https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  >     _______________________________________________
  >     cisco-voip mailing list
  >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >     https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
  -----------------------------------------
  Disclaimer:

  This e-mail communication and any attachments may contain
  confidential and privileged information and is for use by the
  designated addressee(s) named above only.  If you are not the
  intended addressee, you are hereby notified that you have received
  this communication in error and that any use or reproduction of
  this email or its contents is strictly prohibited and may be
  unlawful.  If you have received this communication in error, please
  notify us immediately by replying to this message and deleting it
  from your computer. Thank you.

  _______________________________________________
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  https://puck.nether.net/mailman/listinfo/cisco-voip
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From cbomba at s4nets.com  Wed May  9 19:50:33 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Wed, 9 May 2007 19:50:33 -0400 (EDT)
Subject: [cisco-voip] cisco-voip Digest, Vol 51, Issue 58
Message-ID: <54676.192.168.1.35.1178754633.webmail@192.168.1.35>

voice-card 0
 dspfarm
 dsp services dspfarm


sccp local Loopback0
sccp ccm 10.x.x.x identifier 2 <---SUBSCRIBER
sccp ccm 10.x.x.x identifier 1 <---PUBLISHER
sccp
!
sccp ccm group 1
 bind interface Loopback0
 associate ccm 2 priority 2
 associate ccm 1 priority 1
 associate profile 5 register CFB_NAME   <---NAME it CFB_Something
 switchback method graceful
 switchback interval 120
!
dspfarm profile 5 conference
 codec g711ulaw
 codec g729r8
 codec g729ar8
 codec g729abr8
 maximum sessions 16
 associate application SCCP


-----Original Message-----

Message: 2
Date: Wed, 9 May 2007 17:34:39 -0400
From: Axel Wilzinski <axe7961 at hotmail.com>
Subject: [cisco-voip] Conference Bridge on IOS Router
To: <cisco-voip at puck.nether.net>
Message-ID: <BAY103-W720D38C19CF741F4477F6BA3B0 at phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"

Has anyone had any luck getting local conferencing working on an IOS gateway?  We are using MGCP with a remote Call Manager (4.2.3).  The local gateway has a PVDM2 with 32 DSP's.  Currently, conference calls traverse the WAN even for a local conference call.
 
Any help would be appreciated.
 
Thanks!


From cbomba at s4nets.com  Wed May  9 20:04:08 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Wed, 9 May 2007 20:04:08 -0400 (EDT)
Subject: [cisco-voip] cisco-voip Digest, Vol 51, Issue 59
Message-ID: <47416.192.168.1.71.1178755448.webmail@192.168.1.71>

Try this:

voice translation-rule 1
 rule 1 /^.*\(....\)/ /\1/
!
!
voice translation-profile 8-to-4
 translate called 1

dial-peer voice 5 voip
 translation-profile outgoing 8-to-4
 preference 1
 destination-pattern 880887..
 progress_ind setup enable 3
 voice-class codec 1
 session target ipv4:X.X.X.X
 dtmf-relay h245-alphanumeric
 no vad


Chris Bomba

-----Original Message-----
Date: Wed, 9 May 2007 19:44:32 -0400
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
Subject: [cisco-voip] IOS strip digits from 8 to 4
To: <cisco-voip at puck.nether.net>
Message-ID:
	<C1FE15183DA37645BC0633BC604E44F00586BF11 at USNAEXCH.na.didata.local>
Content-Type: text/plain; charset="us-ascii"

Telco is delivering 8 digits in Malaysia, I only need 4 digits
significant.  I don't want to ask them to change from 8 digits to 4.

 

This is all h323 dial-peers, some DIDs go to CallManager, some DIDs go
to CVP, etc. I can't use MGCP or do a translation-pattern in CCM.

 

How best should I reduce the 8 digits received to 4 via IOS Gateway?

 

DID Range 8808-87XX

 

num-exp 880887.. ....

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell


From cbomba at s4nets.com  Wed May  9 19:56:22 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Wed, 9 May 2007 19:56:22 -0400 (EDT)
Subject: [cisco-voip] cisco-voip Digest, Vol 51, Issue 58
Message-ID: <39436.192.168.1.70.1178754982.webmail@192.168.1.70>

Right, I don't see the call agent information and in CallManager make sure you are matching the device name up with the MGCP Domain Name in a "show ccm-manager".  


Try this:

ccm-manager fallback-mgcp
ccm-manager redundant-host 10.104.21.109
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.104.21.109
ccm-manager config
!
mgcp
mgcp call-agent 10.104.21.110 2427 service-type mgcp version 0.1
mgcp bind control source-interface GigabitEthernet0/0.826 or 725
mgcp bind media source-interface GigabitEthernet0/0.826 or 725


I am not sure which sub-interface is the voice vlan and I am assuming 10.104.21.109 is the Publisher so you might have chosen 110 as the Sub. 



Chris Bomba


-----Original Message-----
Date: Wed, 9 May 2007 14:09:31 -0700
From: "Voll, Scott" <Scott.Voll at wesd.org>
Subject: Re: [cisco-voip] can't register new gateway
To: "Justin Steinberg" <jsteinberg at gmail.com>,	"Lelio Fulgenzi"
	<lelio at uoguelph.ca>
Cc: cisco-voip at puck.nether.net
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE73D at Apollo.wesd.org>
Content-Type: text/plain; charset="us-ascii"

Good call...... where is your mgcp call-agent XXXXX statement?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Wednesday, May 09, 2007 2:04 PM
To: Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] can't register new gateway

 

I don't see any of the MGCP commands in the config but I can't tell
whether they are missing or just cut off in your email.

On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

so i can't seem to ge this new 3845 gateway to register to my publisher.
i can ping both ways (pub <-> gw) so communications is there. i must be
missing some configuration command, but can seem to get it. i made sure
to:

*	enable routing
*	make the description of the gateway equal the domain on the
router
*	add the ccm-manager config statement

is there an mgcp command that i'm missing? config and show ccm-manager
follows

 

grrrrrr

	
________________________________


	
	!
	version 12.4
	service timestamps debug datetime localtime
	service timestamps log datetime localtime
	no service password-encryption
	!
	hostname vgw-kemptville
	!
	boot-start-marker
	boot-end-marker
	!
	card type t1 0 0
	card type t1 0 1
	logging buffered 4096 informational
	no logging console
	!
	aaa new-model
	!
	!
	aaa authentication login default group tacacs+ enable
	aaa authentication login virt-users group tacacs+ enable
	aaa authentication login console group tacacs+ enable
	!
	aaa session-id common
	clock timezone EST -5
	clock summer-time EDT recurring
	no network-clock-participate wic 0
	no network-clock-participate wic 1
	ip cef
	!
	!
	no ip dhcp use vrf connected
	!
	ip dhcp pool dhcp_vlan826
	   network 10.104.126.0 255.255.255.128
	   option 150 ip 10.104.21.121
	   default-router 10.104.126.1
	!
	!
	ip domain name net.uoguelph.ca
	voice-card 0
	 no dspfarm
	!
	!
	controller T1 0/0/0
	 framing esf
	 linecode b8zs
	!
	controller T1 0/1/0
	 framing esf
	 linecode b8zs
	!
	!
	!
	interface GigabitEthernet0/0
	 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE
	 no ip address
	 duplex auto
	 speed auto
	 media-type rj45
	 no keepalive
	!
	interface GigabitEthernet0/0.129
	 encapsulation dot1Q 129 native
	 ip address 192.168.129.66 255.255.255.0
	!
	interface GigabitEthernet0/0.725
	 encapsulation dot1Q 725
	 ip address 10.104.125.99 255.255.255.128
	!
	interface GigabitEthernet0/0.826
	 encapsulation dot1Q 826
	 ip address 10.104.126.1 255.255.255.128
	!
	ip default-gateway 192.168.129.1
	ip route 0.0.0.0 0.0.0.0 192.168.129.1
	ip route 10.0.0.0 255.0.0.0 10.104.125.1
	!
	!
	!
	ccm-manager fallback-mgcp
	ccm-manager redundant-host 10.104.21.109
	ccm-manager mgcp
	ccm-manager music-on-hold
	ccm-manager config server 10.104.21.109
	ccm-manager config
	!
	mgcp

	
________________________________


	vgw-kemptville#  show ccm-manager
	MGCP Domain Name: vgw-kemptville.net.uoguelph.ca
	Priority        Status                   Host
	============================================================
	Primary         None
	First Backup    Down                     10.104.21.109
	Second Backup   None

	 

	Current active Call Manager:    None
	Backhaul/Redundant link port:   2428
	Failover Interval:              30 seconds
	Keepalive Interval:             15 seconds
	Last keepalive sent:            16:33:45 EDT May 9 2007 (elapsed
time: 00:14:33)
	Last MGCP traffic time:         16:34:18 EDT May 9 2007 (elapsed
time: 00:14:01)
	Last failover time:             16:34:18 EDT May 9 2007 from
(0.0.0.0)
	Last switchback time:           None
	Switchback mode:                Graceful
	MGCP Fallback mode:             Enabled/OFF
	Last MGCP Fallback start time:  None
	Last MGCP Fallback end time:    None
	MGCP Download Tones:            Disabled

	 

	Backhaul/Redundant link is down
	Configuration Auto-Download Information
	=======================================
	 No configurations downloaded
	Current state: Downloading XML file
	Configuration Download statistics:
	        Download Attempted             : 1
	          Download Successful          : 1
	          Download Failed              : 0
	        Configuration Attempted        : 0
	          Configuration Successful     : 0
	          Configuration Failed(Parsing): 0
	          Configuration Failed(config) : 0
	Last config download command:
	Configuration Error History:
	FAX mode: cisco
	vgw-kemptville#



_


From lelio at uoguelph.ca  Wed May  9 20:53:01 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 9 May 2007 20:53:01 -0400
Subject: [cisco-voip] cisco-voip Digest, Vol 51, Issue 58
References: <39436.192.168.1.70.1178754982.webmail@192.168.1.70>
Message-ID: <007801c7929d$8fa2cb40$47196883@cfs.uoguelph.ca>

Thanks for all the suggestions. Unfortunately, none of them have worked. Some comments:
  a.. I do not have any dns entries, however, I did not have any before during my testing so I'm not sure that's it.
  b.. As far as the MGCP commands are concerned, I'm pretty sure I only had the first MGCP command as the first, and the rest were downloaded via the callmanager download.
  c.. As far as the device pools are concerned, again, I'm pretty sure they were not the same during the testing and all worked ok.
  d.. I tried the network-clock-participate commands, still no go.
I had a previous router/gateway with the same IP address configured, so that might be causing problems. Tomorrow I will delete it and all it's dependancies and try again.

If that doesn't work, it's TAC time. 

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Christopher M. Bomba 
  To: cisco-voip at puck.nether.net 
  Cc: lelio at uoguelph.ca 
  Sent: Wednesday, May 09, 2007 7:56 PM
  Subject: RE: cisco-voip Digest, Vol 51, Issue 58


  Right, I don't see the call agent information and in CallManager make sure you are matching the device name up with the MGCP Domain Name in a "show ccm-manager".  


  Try this:

  ccm-manager fallback-mgcp
  ccm-manager redundant-host 10.104.21.109
  ccm-manager mgcp
  ccm-manager music-on-hold
  ccm-manager config server 10.104.21.109
  ccm-manager config
  !
  mgcp
  mgcp call-agent 10.104.21.110 2427 service-type mgcp version 0.1
  mgcp bind control source-interface GigabitEthernet0/0.826 or 725
  mgcp bind media source-interface GigabitEthernet0/0.826 or 725


  I am not sure which sub-interface is the voice vlan and I am assuming 10.104.21.109 is the Publisher so you might have chosen 110 as the Sub. 



  Chris Bomba


  -----Original Message-----
  Date: Wed, 9 May 2007 14:09:31 -0700
  From: "Voll, Scott" <Scott.Voll at wesd.org>
  Subject: Re: [cisco-voip] can't register new gateway
  To: "Justin Steinberg" <jsteinberg at gmail.com>, "Lelio Fulgenzi"
  <lelio at uoguelph.ca>
  Cc: cisco-voip at puck.nether.net
  Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE73D at Apollo.wesd.org>
  Content-Type: text/plain; charset="us-ascii"

  Good call...... where is your mgcp call-agent XXXXX statement?

   

  Scott

   

  ________________________________

  From: cisco-voip-bounces at puck.nether.net
  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
  Steinberg
  Sent: Wednesday, May 09, 2007 2:04 PM
  To: Lelio Fulgenzi
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] can't register new gateway

   

  I don't see any of the MGCP commands in the config but I can't tell
  whether they are missing or just cut off in your email.

  On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

  so i can't seem to ge this new 3845 gateway to register to my publisher.
  i can ping both ways (pub <-> gw) so communications is there. i must be
  missing some configuration command, but can seem to get it. i made sure
  to:

  * enable routing
  * make the description of the gateway equal the domain on the
  router
  * add the ccm-manager config statement

  is there an mgcp command that i'm missing? config and show ccm-manager
  follows

   

  grrrrrr


  ________________________________



  !
  version 12.4
  service timestamps debug datetime localtime
  service timestamps log datetime localtime
  no service password-encryption
  !
  hostname vgw-kemptville
  !
  boot-start-marker
  boot-end-marker
  !
  card type t1 0 0
  card type t1 0 1
  logging buffered 4096 informational
  no logging console
  !
  aaa new-model
  !
  !
  aaa authentication login default group tacacs+ enable
  aaa authentication login virt-users group tacacs+ enable
  aaa authentication login console group tacacs+ enable
  !
  aaa session-id common
  clock timezone EST -5
  clock summer-time EDT recurring
  no network-clock-participate wic 0
  no network-clock-participate wic 1
  ip cef
  !
  !
  no ip dhcp use vrf connected
  !
  ip dhcp pool dhcp_vlan826
     network 10.104.126.0 255.255.255.128
     option 150 ip 10.104.21.121
     default-router 10.104.126.1
  !
  !
  ip domain name net.uoguelph.ca
  voice-card 0
  no dspfarm
  !
  !
  controller T1 0/0/0
  framing esf
  linecode b8zs
  !
  controller T1 0/1/0
  framing esf
  linecode b8zs
  !
  !
  !
  interface GigabitEthernet0/0
  description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE
  no ip address
  duplex auto
  speed auto
  media-type rj45
  no keepalive
  !
  interface GigabitEthernet0/0.129
  encapsulation dot1Q 129 native
  ip address 192.168.129.66 255.255.255.0
  !
  interface GigabitEthernet0/0.725
  encapsulation dot1Q 725
  ip address 10.104.125.99 255.255.255.128
  !
  interface GigabitEthernet0/0.826
  encapsulation dot1Q 826
  ip address 10.104.126.1 255.255.255.128
  !
  ip default-gateway 192.168.129.1
  ip route 0.0.0.0 0.0.0.0 192.168.129.1
  ip route 10.0.0.0 255.0.0.0 10.104.125.1
  !
  !
  !
  ccm-manager fallback-mgcp
  ccm-manager redundant-host 10.104.21.109
  ccm-manager mgcp
  ccm-manager music-on-hold
  ccm-manager config server 10.104.21.109
  ccm-manager config
  !
  mgcp


  ________________________________


  vgw-kemptville#  show ccm-manager
  MGCP Domain Name: vgw-kemptville.net.uoguelph.ca
  Priority        Status                   Host
  ============================================================
  Primary         None
  First Backup    Down                     10.104.21.109
  Second Backup   None



  Current active Call Manager:    None
  Backhaul/Redundant link port:   2428
  Failover Interval:              30 seconds
  Keepalive Interval:             15 seconds
  Last keepalive sent:            16:33:45 EDT May 9 2007 (elapsed
  time: 00:14:33)
  Last MGCP traffic time:         16:34:18 EDT May 9 2007 (elapsed
  time: 00:14:01)
  Last failover time:             16:34:18 EDT May 9 2007 from
  (0.0.0.0)
  Last switchback time:           None
  Switchback mode:                Graceful
  MGCP Fallback mode:             Enabled/OFF
  Last MGCP Fallback start time:  None
  Last MGCP Fallback end time:    None
  MGCP Download Tones:            Disabled



  Backhaul/Redundant link is down
  Configuration Auto-Download Information
  =======================================
  No configurations downloaded
  Current state: Downloading XML file
  Configuration Download statistics:
          Download Attempted             : 1
            Download Successful          : 1
            Download Failed              : 0
          Configuration Attempted        : 0
            Configuration Successful     : 0
            Configuration Failed(Parsing): 0
            Configuration Failed(config) : 0
  Last config download command:
  Configuration Error History:
  FAX mode: cisco
  vgw-kemptville#



  _

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From jsteinberg at gmail.com  Wed May  9 22:01:34 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 9 May 2007 22:01:34 -0400
Subject: [cisco-voip] can't register new gateway
In-Reply-To: <4C792A099259634C83F97067671B37FE02EFD7B5@GDSEXC.gds.pri>
References: <mailman.13208.1178744647.1489.cisco-voip@puck.nether.net>
	<4C792A099259634C83F97067671B37FE02EFD7B5@GDSEXC.gds.pri>
Message-ID: <f1b65f880705091901v3dcbfd3es7cab76f04c05c392@mail.gmail.com>

Is your gateway name configured as

vgw-kemptville.net.uoguelph.ca

I see you mentioned that you put the domain name in the description
field, but the domain name should be concatenated with the hostname
and the entire string is used as the device name for the mgcp gw
configuration in ccmadmin.


On 5/9/07, Patrick Mowry <pmowry at getgds.com> wrote:
> >From your config:
>  no network-clock-participate wic 0
>  no network-clock-participate wic 1
>
> I think this may be keeping the controller configs from ccm-manager
> download from taking effect.  Take the no out from in front and force
> another ccm-manager download and see what happens.  Also you can do a
> debug on the download to see it CallManager is responding with a config
> file and verify the router is sending the device name the same as you
> have configured in CallManager.  I do not believe DNS resolution is
> required.
>
> -Patrick
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
> Sent: Wednesday, May 09, 2007 1:49 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] can't register new gateway
>
>
>
> so i can't seem to ge this new 3845 gateway to register to my publisher.
> i can ping both ways (pub <-> gw) so communications is there. i must be
> missing some configuration command, but can seem to get it. i made sure
> to:
>
> *       enable routing
> *       make the description of the gateway equal the domain on the
> router
> *       add the ccm-manager config statement
>
> is there an mgcp command that i'm missing? config and show ccm-manager
> follows
>
>
>
> grrrrrr
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From mikelydick at yahoo.com  Wed May  9 21:14:16 2007
From: mikelydick at yahoo.com (Mike Lydick)
Date: Wed, 9 May 2007 18:14:16 -0700 (PDT)
Subject: [cisco-voip] IOS strip digits from 8 to 4
Message-ID: <863704.24403.qm@web90405.mail.mud.yahoo.com>


Just a guess but how about:

Translation-rule 1
rule 1 ..../..../ /..../


----- Original Message ----
From: "cisco-voip-request at puck.nether.net" <cisco-voip-request at puck.nether.net>
To: cisco-voip at puck.nether.net
Sent: Wednesday, May 9, 2007 7:56:06 PM
Subject: cisco-voip Digest, Vol 51, Issue 59

Send cisco-voip mailing list submissions to
    cisco-voip at puck.nether.net

To subscribe or unsubscribe via the World Wide Web, visit
    https://puck.nether.net/mailman/listinfo/cisco-voip
or, via email, send a message with subject or body 'help' to
    cisco-voip-request at puck.nether.net

You can reach the person managing the list at
    cisco-voip-owner at puck.nether.net

When replying, please edit your Subject line so it is more specific
than "Re: Contents of cisco-voip digest..."


Today's Topics:

   1. Re: Area Code vs NXX Conflict (Jason Aarons (US))
   2. VGW + throughput (Voll, Scott)
   3. IOS strip digits from 8 to 4 (Jason Aarons (US))
   4. Re: Area Code vs NXX Conflict (Lelio Fulgenzi)


----------------------------------------------------------------------

Message: 1
Date: Wed, 9 May 2007 17:42:41 -0400
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
Subject: Re: [cisco-voip] Area Code vs NXX Conflict
To: "Walt Moody" <moody at arizona.edu>, <cisco-voip at puck.nether.net>
Message-ID:
    <C1FE15183DA37645BC0633BC604E44F00586BDF8 at USNAEXCH.na.didata.local>
Content-Type: text/plain;    charset="us-ascii"

You would have to get a list of NPA-XXXs for that CO and expand the
route pattern. I've done this for TEHO (Tail End Hop Off) routing,
Atlanta had hundreds of NPA-XXXs...headache to keep up with..

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Walt Moody
Sent: Wednesday, May 09, 2007 4:19 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Area Code vs NXX Conflict

Brian,

I'm aware of the potential for future conflicts.  Right now, the only
conflict is 623.  If at some far in the future date we have PRIs in
928, and 928 is in conflict, we'll have to deal with it.

Lelio,

Interdigit time outs could be a problem, but the more pressing problem 
is routes.  How do you tell 623[2-9]XXXXXX to go to the PSTN trunks in
Phoenix, and 623[2-9]XXX to go to Tucson's PSTN trunks?  There doesn't
seem to be a way to build a conflict table with different routing for
seven digits versus ten.

-walt

Brian Prentiss wrote:
> The thing is with the first option here, is that potentially you would

> have a similiar situation with the other area codes at some point, as 
> the population growth forces more NXX assignments in the 520 area
code.  
> Also, as people wait for that interdigit timeout, they might be
thinking 
> "did it go through?" and that might be a bit of a challenge support
wise.
>  
> Are you planning on doing the same with the 928 area code that is in
the 
> other part of the state?
>  
> Regards,
> Brian
> 
>  
> On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca
<mailto:lelio at uoguelph.ca>> 
> wrote:
> 
>     If you can live with interdigit timeout, then you should be able
to
>     program things so that you can dial either number and get through.
>     Users will have to wait 10 seconds (or whatever your parameter is
>     configured for) before 7 digit calls are placed -or- if I'm not
>     mistaken, they can press # to complete the dialing process (you
>     might need an extra route pattern with the # at the end though as
>     well).
>      
>     Other than that, if you are looking to be able to dial a 7 digit
and
>     10 digit call with the same beginning three digits and not have
>     interdigit timeout at the end, I do not think you will be able to
do
>     this.
>      
>      
>
------------------------------------------------------------------------
--------
>     Lelio Fulgenzi, B.A.
>     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G
2W1
>     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>     ...there's no such thing as a bad timbit...
> 
>         ----- Original Message -----
>         From: Walt Moody <mailto:moody at arizona.edu>
>         To: cisco-voip at puck.nether.net
<mailto:cisco-voip at puck.nether.net>
>         Sent: Wednesday, May 09, 2007 2:04 PM
>         Subject: [cisco-voip] Area Code vs NXX Conflict
> 
>          
>         Here's the setup:
> 
>         The University of Arizona - in Tucson - uses seven digit
dialing for
>         both on Campus and off Campus local calls.  We do not use 9+
or any
>         other PBX "outside access" codes.  NANP long distance is 1+ 10
>         digits,
>         IDDD is 011+.
> 
>         Except:  When dialing from our Tucson Campus to our Phoenix
>         Campus, we
>         dial 602827XXXX.  We do not use a "1" or any other prefix
digit
>         to when
>         we dial a "free" or on-net call.
> 
>         Still with me?   OK, try this: We want to set up our dial
plans
>         so that
>         we can use 10 digits to call from the Tucson Campus to
anywhere
>         in the
>         Phoenix rate center, which consists of Area Codes 480, 602,
and
>         623.
>         That's easy enough to do, except (there's that word again) 623
>         is both
>         an Area Code in Phoenix and an NXX in Tucson.
> 
>         I need to find a way to tell the difference between, for
>         example, the
>         dialed digits (623) 555-1212 and 623-5551 and route the 10
digit
>         "623"
>         numbers to our PRIs in Phoenix and the 7 digit "623" numbers
to the
>         local Tucson PRIs.
> 
>         Prefix digits are not an option -- there are several PBXs in
>         addition
>         to the CallManagers that are part of the University's voice
>         services,
>         and no real good way to change them from their current dial
plans.
> 
>         Ideas, anyone?
> 
>         Thanks,
> 
>         -walt
>         _______________________________________________
>         cisco-voip mailing list
>         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>         https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 2
Date: Wed, 9 May 2007 15:42:08 -0700
From: "Voll, Scott" <Scott.Voll at wesd.org>
Subject: [cisco-voip] VGW + throughput
To: <cisco-voip at puck.nether.net>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE742 at Apollo.wesd.org>
Content-Type: text/plain; charset="us-ascii"

What should I expect for Throughput on a ISR VGW.        

 

2801 = ?mb

2811 = ?mb

2821 =

2851 =

3825 =

 

Plus they will be doing a couple pots lines and ~8 FXS/FXO ports, and
QoS . No T1's, ATM, Frame, etc.  just Ethernet IP.

 

Thanks

 

Scott

 

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Message: 3
Date: Wed, 9 May 2007 19:44:32 -0400
From: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>
Subject: [cisco-voip] IOS strip digits from 8 to 4
To: <cisco-voip at puck.nether.net>
Message-ID:
    <C1FE15183DA37645BC0633BC604E44F00586BF11 at USNAEXCH.na.didata.local>
Content-Type: text/plain; charset="us-ascii"

Telco is delivering 8 digits in Malaysia, I only need 4 digits
significant.  I don't want to ask them to change from 8 digits to 4.

 

This is all h323 dial-peers, some DIDs go to CallManager, some DIDs go
to CVP, etc. I can't use MGCP or do a translation-pattern in CCM.

 

How best should I reduce the 8 digits received to 4 via IOS Gateway?

 

DID Range 8808-87XX

 

num-exp 880887.. ....

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 



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Message: 4
Date: Wed, 9 May 2007 19:55:39 -0400
From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Subject: Re: [cisco-voip] Area Code vs NXX Conflict
To: "Jason Aarons \(US\)" <jason.aarons at us.didata.com>,    "Walt Moody"
    <moody at arizona.edu>, <cisco-voip at puck.nether.net>
Message-ID: <002a01c79295$8d9d7f00$47196883 at cfs.uoguelph.ca>
Content-Type: text/plain; charset="iso-8859-1"

I use this for this sort of stuff....

http://www.localcallingguide.com/

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Jason Aarons (US) 
  To: Walt Moody ; cisco-voip at puck.nether.net 
  Sent: Wednesday, May 09, 2007 5:42 PM
  Subject: Re: [cisco-voip] Area Code vs NXX Conflict


  You would have to get a list of NPA-XXXs for that CO and expand the
  route pattern. I've done this for TEHO (Tail End Hop Off) routing,
  Atlanta had hundreds of NPA-XXXs...headache to keep up with..

  -----Original Message-----
  From: cisco-voip-bounces at puck.nether.net
  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Walt Moody
  Sent: Wednesday, May 09, 2007 4:19 PM
  To: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Area Code vs NXX Conflict

  Brian,

  I'm aware of the potential for future conflicts.  Right now, the only
  conflict is 623.  If at some far in the future date we have PRIs in
  928, and 928 is in conflict, we'll have to deal with it.

  Lelio,

  Interdigit time outs could be a problem, but the more pressing problem 
  is routes.  How do you tell 623[2-9]XXXXXX to go to the PSTN trunks in
  Phoenix, and 623[2-9]XXX to go to Tucson's PSTN trunks?  There doesn't
  seem to be a way to build a conflict table with different routing for
  seven digits versus ten.

  -walt

  Brian Prentiss wrote:
  > The thing is with the first option here, is that potentially you would

  > have a similiar situation with the other area codes at some point, as 
  > the population growth forces more NXX assignments in the 520 area
  code.  
  > Also, as people wait for that interdigit timeout, they might be
  thinking 
  > "did it go through?" and that might be a bit of a challenge support
  wise.
  >  
  > Are you planning on doing the same with the 928 area code that is in
  the 
  > other part of the state?
  >  
  > Regards,
  > Brian
  > 
  >  
  > On 5/9/07, Lelio Fulgenzi <lelio at uoguelph.ca
  <mailto:lelio at uoguelph.ca>> 
  > wrote:
  > 
  >     If you can live with interdigit timeout, then you should be able
  to
  >     program things so that you can dial either number and get through.
  >     Users will have to wait 10 seconds (or whatever your parameter is
  >     configured for) before 7 digit calls are placed -or- if I'm not
  >     mistaken, they can press # to complete the dialing process (you
  >     might need an extra route pattern with the # at the end though as
  >     well).
  >      
  >     Other than that, if you are looking to be able to dial a 7 digit
  and
  >     10 digit call with the same beginning three digits and not have
  >     interdigit timeout at the end, I do not think you will be able to
  do
  >     this.
  >      
  >      
  >
  ------------------------------------------------------------------------
  --------
  >     Lelio Fulgenzi, B.A.
  >     Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G
  2W1
  >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  >     ...there's no such thing as a bad timbit...
  > 
  >         ----- Original Message -----
  >         From: Walt Moody <mailto:moody at arizona.edu>
  >         To: cisco-voip at puck.nether.net
  <mailto:cisco-voip at puck.nether.net>
  >         Sent: Wednesday, May 09, 2007 2:04 PM
  >         Subject: [cisco-voip] Area Code vs NXX Conflict
  > 
  >          
  >         Here's the setup:
  > 
  >         The University of Arizona - in Tucson - uses seven digit
  dialing for
  >         both on Campus and off Campus local calls.  We do not use 9+
  or any
  >         other PBX "outside access" codes.  NANP long distance is 1+ 10
  >         digits,
  >         IDDD is 011+.
  > 
  >         Except:  When dialing from our Tucson Campus to our Phoenix
  >         Campus, we
  >         dial 602827XXXX.  We do not use a "1" or any other prefix
  digit
  >         to when
  >         we dial a "free" or on-net call.
  > 
  >         Still with me?   OK, try this: We want to set up our dial
  plans
  >         so that
  >         we can use 10 digits to call from the Tucson Campus to
  anywhere
  >         in the
  >         Phoenix rate center, which consists of Area Codes 480, 602,
  and
  >         623.
  >         That's easy enough to do, except (there's that word again) 623
  >         is both
  >         an Area Code in Phoenix and an NXX in Tucson.
  > 
  >         I need to find a way to tell the difference between, for
  >         example, the
  >         dialed digits (623) 555-1212 and 623-5551 and route the 10
  digit
  >         "623"
  >         numbers to our PRIs in Phoenix and the 7 digit "623" numbers
  to the
  >         local Tucson PRIs.
  > 
  >         Prefix digits are not an option -- there are several PBXs in
  >         addition
  >         to the CallManagers that are part of the University's voice
  >         services,
  >         and no real good way to change them from their current dial
  plans.
  > 
  >         Ideas, anyone?
  > 
  >         Thanks,
  > 
  >         -walt
  >         _______________________________________________
  >         cisco-voip mailing list
  >         cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >         https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  >     _______________________________________________
  >     cisco-voip mailing list
  >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
  >     https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  _______________________________________________
  cisco-voip mailing list
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  https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Wed May  9 22:23:47 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 9 May 2007 22:23:47 -0400
Subject: [cisco-voip] can't register new gateway
References: <mailman.13208.1178744647.1489.cisco-voip@puck.nether.net>
	<4C792A099259634C83F97067671B37FE02EFD7B5@GDSEXC.gds.pri>
	<f1b65f880705091901v3dcbfd3es7cab76f04c05c392@mail.gmail.com>
Message-ID: <001401c792aa$3d8fac30$c901a8c0@pdp11>

Thanks Justin. I was just going to reply to the list about that. I tried 
that and it registered. I must have misread my old config, I will have to 
review it, I don't think it had an 'ip domain name' config statement.

It took a while with the nested pre-requisites, but I finally got things 
back to the way they were. I will try again in the morning to see what steps 
I have to take to get things working again.

----- Original Message ----- 
From: "Justin Steinberg" <jsteinberg at gmail.com>
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Cc: <cisco-voip at puck.nether.net>
Sent: Wednesday, May 09, 2007 10:01 PM
Subject: Re: [cisco-voip] can't register new gateway


> Is your gateway name configured as
>
> vgw-kemptville.net.uoguelph.ca
>
> I see you mentioned that you put the domain name in the description
> field, but the domain name should be concatenated with the hostname
> and the entire string is used as the device name for the mgcp gw
> configuration in ccmadmin.
>
>
> On 5/9/07, Patrick Mowry <pmowry at getgds.com> wrote:
>> >From your config:
>>  no network-clock-participate wic 0
>>  no network-clock-participate wic 1
>>
>> I think this may be keeping the controller configs from ccm-manager
>> download from taking effect.  Take the no out from in front and force
>> another ccm-manager download and see what happens.  Also you can do a
>> debug on the download to see it CallManager is responding with a config
>> file and verify the router is sending the device name the same as you
>> have configured in CallManager.  I do not believe DNS resolution is
>> required.
>>
>> -Patrick
>> ________________________________
>>
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
>> Sent: Wednesday, May 09, 2007 1:49 PM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] can't register new gateway
>>
>>
>>
>> so i can't seem to ge this new 3845 gateway to register to my publisher.
>> i can ping both ways (pub <-> gw) so communications is there. i must be
>> missing some configuration command, but can seem to get it. i made sure
>> to:
>>
>> *       enable routing
>> *       make the description of the gateway equal the domain on the
>> router
>> *       add the ccm-manager config statement
>>
>> is there an mgcp command that i'm missing? config and show ccm-manager
>> follows
>>
>>
>>
>> grrrrrr
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> 


From nws at fredf.org  Wed May  9 22:36:18 2007
From: nws at fredf.org (Kent Roberts)
Date: Wed, 9 May 2007 20:36:18 -0600
Subject: [cisco-voip] Conference Bridge on IOS Router
In-Reply-To: <BAY103-W720D38C19CF741F4477F6BA3B0@phx.gbl>
References: <BAY103-W720D38C19CF741F4477F6BA3B0@phx.gbl>
Message-ID: <3e6f56310705091936j57c14580rff64b5006c7a616f@mail.gmail.com>

24 of those dsp's are gonna be tied up on a t1 card.  Thats if you have
one.  Need a little more info tho to really help.  The problem we ran into
is the t1's ate 24  dsp channels, and since the transcoder config is usually
first in the ios, if there any configured theres no room left for the
conferencing....   If you don't have a T1 card in there, then you got plenty
of resources.  What kind of issue are you having?   Do you have the SCCP
config setup?   well frist I should ask what kind of IOS device is this, an
what has the PVDMS in it.  IS it onboard like an ISR router or off board in
an nm-hdv slot.   There are several configs to address it, just depends on
what equipment you have.

On 5/9/07, Axel Wilzinski <axe7961 at hotmail.com> wrote:
>
> Has anyone had any luck getting local conferencing working on an IOS
> gateway?  We are using MGCP with a remote Call Manager (4.2.3).  The local
> gateway has a PVDM2 with 32 DSP's.  Currently, conference calls traverse the
> WAN even for a local conference call.
>
> Any help would be appreciated.
>
> Thanks!
>
> ------------------------------
> Add some color. Personalize your inbox with your favorite colors. Try it!<http://www.windowslive-hotmail.com/learnmore/personalize.html?locale=en-us&ocid=RMT_TAGLM_HMWL_reten_addcolor_0507>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From aelnagar at ACT-EG.COM  Thu May 10 02:24:32 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Thu, 10 May 2007 09:24:32 +0300
Subject: [cisco-voip] can't register new gateway
References: <mailman.13208.1178744647.1489.cisco-voip@puck.nether.net><4C792A099259634C83F97067671B37FE02EFD7B5@GDSEXC.gds.pri><f1b65f880705091901v3dcbfd3es7cab76f04c05c392@mail.gmail.com>
	<001401c792aa$3d8fac30$c901a8c0@pdp11>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A04D@mail.ACT-EG.COM>

to know the exact name that the gateway register with, from a near other cisco device issue the command show cdp ne and you will find the exact name that the gateway register with.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
Sent: Thu 10-May-07 5:23 AM
To: Justin Steinberg
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] can't register new gateway



Thanks Justin. I was just going to reply to the list about that. I tried
that and it registered. I must have misread my old config, I will have to
review it, I don't think it had an 'ip domain name' config statement.

It took a while with the nested pre-requisites, but I finally got things
back to the way they were. I will try again in the morning to see what steps
I have to take to get things working again.

----- Original Message -----
From: "Justin Steinberg" <jsteinberg at gmail.com>
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Cc: <cisco-voip at puck.nether.net>
Sent: Wednesday, May 09, 2007 10:01 PM
Subject: Re: [cisco-voip] can't register new gateway


> Is your gateway name configured as
>
> vgw-kemptville.net.uoguelph.ca
>
> I see you mentioned that you put the domain name in the description
> field, but the domain name should be concatenated with the hostname
> and the entire string is used as the device name for the mgcp gw
> configuration in ccmadmin.
>
>
> On 5/9/07, Patrick Mowry <pmowry at getgds.com> wrote:
>> >From your config:
>>  no network-clock-participate wic 0
>>  no network-clock-participate wic 1
>>
>> I think this may be keeping the controller configs from ccm-manager
>> download from taking effect.  Take the no out from in front and force
>> another ccm-manager download and see what happens.  Also you can do a
>> debug on the download to see it CallManager is responding with a config
>> file and verify the router is sending the device name the same as you
>> have configured in CallManager.  I do not believe DNS resolution is
>> required.
>>
>> -Patrick
>> ________________________________
>>
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
>> Sent: Wednesday, May 09, 2007 1:49 PM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] can't register new gateway
>>
>>
>>
>> so i can't seem to ge this new 3845 gateway to register to my publisher.
>> i can ping both ways (pub <-> gw) so communications is there. i must be
>> missing some configuration command, but can seem to get it. i made sure
>> to:
>>
>> *       enable routing
>> *       make the description of the gateway equal the domain on the
>> router
>> *       add the ccm-manager config statement
>>
>> is there an mgcp command that i'm missing? config and show ccm-manager
>> follows
>>
>>
>>
>> grrrrrr
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From aelnagar at ACT-EG.COM  Thu May 10 02:31:27 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Thu, 10 May 2007 09:31:27 +0300
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local><002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00583915E@USNAEXCH.na.didata.local>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A04E@mail.ACT-EG.COM>

dont even try it you will break your CM replication and will go through alot of very strange problems.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Matt Slaga (US)
Sent: Wed 09-May-07 11:05 PM
To: Ryan Ratliff; Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?



If the customer really pushes you, create a DNS C-name record.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, May 09, 2007 3:58 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?

No way, no how.

The only time you can change the hostname of a CM server is before 
you install CallManager on it.

-Ryan

On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:

I'm trying to find a reference about server hostname changes for 
production CallManager 4.1, I recall it's not advised but can't find 
a article on cisco.com stating this -jason





Jason Aarons

Consultant

http://www.dimensiondata.com/na

904-338-3245 cell



For urgent issues notify your Project Manager, for 24x7 support 
contact the Dimension Data NOC at 800-974-6584.





Disclaimer: This e-mail communication and any attachments may contain 
confidential and privileged information and is for use by the 
designated addressee(s) named above only. If you are not the intended 
addressee, you are hereby notified that you have received this 
communication in error and that any use or reproduction of this email 
or its contents is strictly prohibited and may be unlawful. If you 
have received this communication in error, please notify us 
immediately by replying to this message and deleting it from your 
computer. Thank you.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
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-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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From aelnagar at ACT-EG.COM  Thu May 10 02:34:06 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Thu, 10 May 2007 09:34:06 +0300
Subject: [cisco-voip] Echo on IP Phones through PRI
References: <418387.45400.qm@web37209.mail.mud.yahoo.com>
	<6350516B6159444E9D646523EB4526F403EB9FFD@apollo.rbh.local>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A04F@mail.ACT-EG.COM>

try to increase your output attenuation ( you set it to -3 which is a gain of 3) play with this vlaue but with positive values not negative and it will do it for you, the echo problem is mainly affected by this value as this controls the output power off your gateway, the higher this level the higher is the reflected signal (echo) from teleco. I am assuming that the IP Phone side is the one hearing the echo.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Rhodes, Geoff
Sent: Wed 09-May-07 10:44 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Echo on IP Phones through PRI



Please excuse the newbie question, but how do I determine the DSP
version my MGCP gateway is running?  It's a CMM blade in our 6509
switch.  The IOS on it looks to be 12.4(8a).  I'd like to document what
release we are running for future needs.

We get complaints ever once in a while about echo problems.  Most of the
time we blow them off unless someone is constantly complaining.

Here are our settings in CCMAdmin for the CMM Gateway:
Echo Cancellation Enable = Enable
Echo Cancel Coverage = 32
Input Gain = 3
Output Attenuation = -3

Thanks for your help.

Geoff Rhodes
Director, Information Technology
Robinson, Bradshaw & Hinson, P.A.
704 / 377-8188
grhodes at rbh.com



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
Sent: Wednesday, April 18, 2007 9:12 AM
To: Matthew Melbourne; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Echo on IP Phones through PRI

DSP Fimware 4.4.708 has issues which can cause echo, etc. Many people on
the cisco NetPro forums have ran into echo issues with this DSP
firmware.  I don't have the exact bug id's handy, but you can try
upgrading to a current 12.4 mainline release if your routers DRAM/flash
can take it. 12.4(13a) has 4.4.23 DSP ware. So far, I have not ran into
any issues yet with DSP firmware 4.4.21 or higher in 12.4 mainline but
YMMV.

If you can't update the IOS, and I don't believe there is a newer 12.3T
release with a newer firmware embedded in it, you can get a standalone
DSP firmware image file to put on the router flash that is a newer
version from TAC.

As far as finding out what gateway a call is using, probably the
quickest way is to go to the phones web page while the call is connected
and look at the stream page for the remote IP address. Other then that,
you would need to check out CDR and/or QRT records if they use QRT
softkey, etc and go into each gateway.

----- Original Message ----
From: Matthew Melbourne <matt at melbourne.org.uk>
To: cisco-voip at puck.nether.net
Sent: Wednesday, April 18, 2007 7:50:22 AM
Subject: [cisco-voip] Echo on IP Phones through PRI

We are getting sporadic issues from users on IP phones who are hearing
echo on external calls. From this document is would suggest that echo on
IP Phones is caused by hybrid echo and not any local acoustic echo.

http://www.cisco.com/warp/public/788/voice-qos/echo_ipphone_gateway.html

The MGCP-controlled voice gateways are all running 12.3(14)T7 with DSP
firmware 4.4.708. In CallManager (4.1(3)SR2), echo cancellation is on
for each port, with a "default" echo cancellation coverage of 64ms.
Input gain and output attenuation are set to 0dB.

I have seen suggestions to include the "echo-cancel erl worst-case 3"
command on the voice ports. Can this help in situations of occasional
echo?

Are there any other parameters changes which could help? Could it be a
code/DSP firmware issue? I haven't found a specific echo-related bug
relating to this particular IOS and its embedded DSP firmware.

On occasions, the start of the call is fine but echo comes and goes
during the call.

Is there a quick way of finding out which voice gateway a call is
terminated on (we have 8, and all could be used in a round-robin fashion
for outbound calls).

Cheers,

Matt

--
Matthew Melbourne
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around
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_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From akoul at dataways.gr  Thu May 10 02:39:59 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 10 May 2007 09:39:59 +0300
Subject: [cisco-voip] Conference Bridge on IOS Router
In-Reply-To: <BAY103-W720D38C19CF741F4477F6BA3B0@phx.gbl>
References: <BAY103-W720D38C19CF741F4477F6BA3B0@phx.gbl>
Message-ID: <4642BE3F.7080209@dataways.gr>

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From gokhanciscottl at yahoo.com  Thu May 10 04:37:54 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Thu, 10 May 2007 01:37:54 -0700 (PDT)
Subject: [cisco-voip] callerID is missing on plar connection
Message-ID: <750049.85198.qm@web39615.mail.mud.yahoo.com>

hi 
i know here is ccm group but i need help about an issue  related to  ccme 


2851 ccme 3.3 
manager has direct analogue line. For this i configured plar connection on fxo port for his second line. 
he can receive calls from this analogue line and he can also call outside from this line. 
config is below 
problem is CallerID is missing for incoming calls .on phone display unknown number. 
i directly connected analgue phone to the line and checked if the telco sends callrID . it is ok. on my cell phone i saw the callerID.  
but when i connected it to the fxo port and made plar connection callerID is missing.


voice-port 0/2/3 
description plar to ephone63 button2
timeouts wait-release 3 
connection plar opx 99 
caller-id enable 

!
dial-peer voice 800 pots
 incoming called-number .
 port 0/2/3
!
dial-peer voice 801 pots
 destination-pattern 801
 port 0/2/3
!

ephone-dn 2 
number 99 
pickup-group 14 
label direct line 
hold-alert 30 originator 
trunk 801 timeout 10 

ephone 63 
ephone-template 1 
mac-address 0019.2F3A.2807 
type 7940 
button 1:12 2:2 

thanks a lot 


any idea? 
thanks 
Gokhan

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
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From matt at melbourne.org.uk  Thu May 10 06:34:44 2007
From: matt at melbourne.org.uk (Matthew Melbourne)
Date: Thu, 10 May 2007 11:34:44 +0100
Subject: [cisco-voip] Echo on IP Phones through PRI
In-Reply-To: <8a1f34b50705091546s70b357c6p702138bb016f45ed@mail.gmail.com>
References: <b718d1880704180550r52c715c5x80e83be222546ba2@mail.gmail.com>
	<8a1f34b50705091546s70b357c6p702138bb016f45ed@mail.gmail.com>
Message-ID: <b718d1880705100334r797c5d9dp68e6674e21dc02b4@mail.gmail.com>

Thanks Steve. We have an upgrade planned for our voice gateways in an
forthcoming maintenance window, where we'll try 12.4(latest) mainline.
Are you aware of any bugs which document this issue?

Cheers,

Matt

On 09/05/07, Steve G <smgustafson at gmail.com> wrote:
> 12.3(14) will deffinietly give you echo issues.  it is the DSP firmware.
> either get new DSP firmware from TAC, or upgrade IOS.  I am using 12.4(8)b
> with no issues.
>
> Steve
>
>
> On 4/18/07, Matthew Melbourne <matt at melbourne.org.uk> wrote:
> >
> > We are getting sporadic issues from users on IP phones who are hearing
> > echo on external calls. From this document is would suggest that echo
> > on IP Phones is caused by hybrid echo and not any local acoustic echo.
> >
> >
> http://www.cisco.com/warp/public/788/voice-qos/echo_ipphone_gateway.html
> >
> > The MGCP-controlled voice gateways are all running 12.3(14)T7 with DSP
> > firmware 4.4.708. In CallManager (4.1(3)SR2), echo cancellation is on
> > for each port, with a "default" echo cancellation coverage of 64ms.
> > Input gain and output attenuation are set to 0dB.
> >
> > I have seen suggestions to include the "echo-cancel erl worst-case 3"
> > command on the voice ports. Can this help in situations of occasional
> > echo?
> >
> > Are there any other parameters changes which could help? Could it be a
> > code/DSP firmware issue? I haven't found a specific echo-related bug
> > relating to this particular IOS and its embedded DSP firmware.
> >
> > On occasions, the start of the call is fine but echo comes and goes
> > during the call.
> >
> > Is there a quick way of finding out which voice gateway a call is
> > terminated on (we have 8, and all could be used in a round-robin
> > fashion for outbound calls).
> >
> > Cheers,
> >
> > Matt
> >
> > --
> > Matthew Melbourne
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>


-- 
Matthew Melbourne

From ealeatherman at gmail.com  Thu May 10 08:26:10 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 10 May 2007 08:26:10 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED88A04E@mail.ACT-EG.COM>
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>
	<002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00583915E@USNAEXCH.na.didata.local>
	<88C8D019DA547C4BABECB9A806ED22ED88A04E@mail.ACT-EG.COM>
Message-ID: <94a1afde0705100526ycf3693co47630081d6c970d9@mail.gmail.com>

Related question, and I think the answer is also probably no but want to
verify...

When upgrading from ccm 4.1 to 5, would it be possible to change the host
name of the primary node in the 5.0 cluster after loading the DMA data but
prior to adding any subscriber nodes?

Trying to come up with a way to upgrade our system to 5 but leave 2 nodes in
the old cluster active to operate the call center users and IPCCX - or at
least determine if its feasible. so basically to split the cluster up.

On 5/10/07, Ahmed Elnagar <aelnagar at act-eg.com> wrote:
>
>  dont even try it you will break your CM replication and will go through
> alot of very strange problems.
>
>  Thanks and Best Regards
> *
> Ahmed A. Elnagar
> *Network Field Engineer
>
> Advanced Computer Technology (ACT)
> 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> Postal Code:12411 Cairo Egypt
>
> *Mob**:* +2010-2833868
> *Website**: *www.act-eg.com
> *E-mail**: *aelnagar at act-eg.com
>
> ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net on behalf of Matt Slaga (US)
> *Sent:* Wed 09-May-07 11:05 PM
> *To:* Ryan Ratliff; Jason Aarons (US)
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?
>
>  If the customer really pushes you, create a DNS C-name record.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at puck.nether.net>]
> On Behalf Of Ryan Ratliff
> Sent: Wednesday, May 09, 2007 3:58 PM
> To: Jason Aarons (US)
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?
>
> No way, no how.
>
> The only time you can change the hostname of a CM server is before
> you install CallManager on it.
>
> -Ryan
>
> On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:
>
> I'm trying to find a reference about server hostname changes for
> production CallManager 4.1, I recall it's not advised but can't find
> a article on cisco.com stating this -jason
>
>
>
>
>
> Jason Aarons
>
> Consultant
>
> http://www.dimensiondata.com/na
>
> 904-338-3245 cell
>
>
>
> For urgent issues notify your Project Manager, for 24x7 support
> contact the Dimension Data NOC at 800-974-6584.
>
>
>
>
>
> Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the intended
> addressee, you are hereby notified that you have received this
> communication in error and that any use or reproduction of this email
> or its contents is strictly prohibited and may be unlawful. If you
> have received this communication in error, please notify us
> immediately by replying to this message and deleting it from your
> computer. Thank you.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Thu May 10 08:29:26 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 10 May 2007 08:29:26 -0400
Subject: [cisco-voip] Automatic CFAll?
In-Reply-To: <8dc3ea820705091346u73d211a5o1390b5d470678733@mail.gmail.com>
References: <8dc3ea820705091346u73d211a5o1390b5d470678733@mail.gmail.com>
Message-ID: <94a1afde0705100529k5ef0b773saef63496f5c16b50@mail.gmail.com>

Could use time of day based routing to send calls to the auto attendant
after 6PM and then back to the operator's phone at whatever time in the
morning. The operator would still have to remove the forwarding on their set
in the morning when they get in, so it wouldnt go to the nighbell.

This doesn't really change the call forwarding settings but i think it would
have the effect you want.

On 5/9/07, Gareth Cafirma <garethsc at gmail.com> wrote:
>
> Hello Guys,
>
> Is it possible to do a "automatic" call forward all to an auto attendant?
> The client has a 5.1 CCM, unity connection 1.2. Here is the scenario,
> operator leaves her desk and forward all calls to the nightbell, after 6 PM,
> they want to remove the forward to nightbell and forward it to auto
> attendant instead.
> Can this be done? Any suggestions will be appreciated.
>
> Gareth
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From aelnagar at ACT-EG.COM  Thu May 10 08:34:27 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Thu, 10 May 2007 15:34:27 +0300
Subject: [cisco-voip] callerID is missing on plar connection
References: <750049.85198.qm@web39615.mail.mud.yahoo.com>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A059@mail.ACT-EG.COM>

FXO has problems when in MGCP mode with callmanager, it has to be in h323 mode to display the caller ID. I dont know whtere it is the same for CME or not
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of gokhan senol
Sent: Thu 10-May-07 11:37 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] callerID is missing on plar connection



hi 

i know here is ccm group but i need help about an issue  related to  ccme 

 


2851 ccme 3.3 
manager has direct analogue line. For this i configured plar connection on fxo port for his second line. 
he can receive calls from this analogue line and he can also call outside from this line. 
config is below 
problem is CallerID is missing for incoming calls .on phone display unknown number. 

i directly connected analgue phone to the line and checked if the telco sends callrID . it is ok. on my cell phone i saw the callerID.  

but when i connected it to the fxo port and made plar connection callerID is missing.

 


voice-port 0/2/3 

description plar to ephone63 button2
timeouts wait-release 3 
connection plar opx 99 
caller-id enable 

 

!
dial-peer voice 800 pots
 incoming called-number .
 port 0/2/3
!
dial-peer voice 801 pots
 destination-pattern 801
 port 0/2/3
!

 

ephone-dn 2 
number 99 
pickup-group 14 
label direct line 
hold-alert 30 originator 
trunk 801 timeout 10 

 

ephone 63 
ephone-template 1 
mac-address 0019.2F3A.2807 
type 7940 
button 1:12 2:2 

 

thanks a lot 

 


any idea? 
thanks 
Gokhan 

 


 


________________________________

Ahhh...imagining that irresistible "new car" smell?
Check out new cars at Yahoo! Autos. <http://us.rd.yahoo.com/evt=48245/*http://autos.yahoo.com/new_cars.html;_ylc=X3oDMTE1YW1jcXJ2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDbmV3LWNhcnM->  
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From Matt.Slaga at us.didata.com  Thu May 10 08:36:39 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 10 May 2007 08:36:39 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
In-Reply-To: <3e6f56310705091939h7aa83074w4073bd8db67cbc75@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>
	<002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00583915E@USNAEXCH.na.didata.local>
	<3e6f56310705091939h7aa83074w4073bd8db67cbc75@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00586C107@USNAEXCH.na.didata.local>

Perhaps I didn't clarify enough here.  Create a C Name record for the
new name, leave the original name as is.

 

From: Kent Roberts [mailto:kent at fredf.org] 
Sent: Wednesday, May 09, 2007 10:39 PM
To: Matt Slaga (US)
Cc: Ryan Ratliff; Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?

 

Name changes are not TAC supported, and it really, really breaks things!

 



 

On 5/9/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote: 

If the customer really pushes you, create a DNS C-name record.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, May 09, 2007 3:58 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?

No way, no how.

The only time you can change the hostname of a CM server is before
you install CallManager on it.

-Ryan

On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:

I'm trying to find a reference about server hostname changes for 
production CallManager 4.1, I recall it's not advised but can't find
a article on cisco.com stating this -jason





Jason Aarons

Consultant

http://www.dimensiondata.com/na

904-338-3245 cell



For urgent issues notify your Project Manager, for 24x7 support
contact the Dimension Data NOC at 800-974-6584.





Disclaimer: This e-mail communication and any attachments may contain 
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the intended
addressee, you are hereby notified that you have received this
communication in error and that any use or reproduction of this email 
or its contents is strictly prohibited and may be unlawful. If you
have received this communication in error, please notify us
immediately by replying to this message and deleting it from your
computer. Thank you. 

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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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https://puck.nether.net/mailman/listinfo/cisco-voip
-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the 
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

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https://puck.nether.net/mailman/listinfo/cisco-voip 

 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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From Nicolas.Maton at colt.net  Thu May 10 08:30:57 2007
From: Nicolas.Maton at colt.net (Maton, Nicolas)
Date: Thu, 10 May 2007 13:30:57 +0100
Subject: [cisco-voip] unsubscribe
In-Reply-To: <94a1afde0705100529k5ef0b773saef63496f5c16b50@mail.gmail.com>
Message-ID: <44F679B1D2FB1748B2775781D1BCC6AA821FC7@ULONCTMVMAI005.EU.COLT>

 
 
 

Nicolas Maton
Engineer IP/DATA
COLT
Zweefvliegtuigstraat
1130 Haren
Belgium

Email: nicolas.maton at colt.net
www.colt.net <http://www.colt.net/>  
Data | Voice | Managed Services 

	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
	Sent: donderdag 10 mei 2007 14:29
	To: Gareth Cafirma
	Cc: cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Automatic CFAll?
	
	
	Could use time of day based routing to send calls to the auto
attendant after 6PM and then back to the operator's phone at whatever
time in the morning. The operator would still have to remove the
forwarding on their set in the morning when they get in, so it wouldnt
go to the nighbell. 
	
	This doesn't really change the call forwarding settings but i
think it would have the effect you want.
	
	
	On 5/9/07, Gareth Cafirma < garethsc at gmail.com
<mailto:garethsc at gmail.com> > wrote: 

		Hello Guys,
		 
		Is it possible to do a "automatic" call forward all to
an auto attendant? The client has a 5.1 CCM, unity connection 1.2. Here
is the scenario, operator leaves her desk and forward all calls to the
nightbell, after 6 PM, they want to remove the forward to nightbell and
forward it to auto attendant instead. 
		Can this be done? Any suggestions will be appreciated.
		
		 
		Gareth

		_______________________________________________
		cisco-voip mailing list
		cisco-voip at puck.nether.net
		https://puck.nether.net/mailman/listinfo/cisco-voip
		
		




	-- 
	Ed Leatherman
	Senior Voice Engineer
	West Virginia University
	Telecommunications and Network Operations 



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From Matt.Slaga at us.didata.com  Thu May 10 09:05:57 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 10 May 2007 09:05:57 -0400
Subject: [cisco-voip] callerID is missing on plar connection
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED88A059@mail.ACT-EG.COM>
References: <750049.85198.qm@web39615.mail.mud.yahoo.com>
	<88C8D019DA547C4BABECB9A806ED22ED88A059@mail.ACT-EG.COM>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00586C19C@USNAEXCH.na.didata.local>

You might need to alter the "caller-id attenuation", and possibly
(though I doubt this one) change your signaling (loop start/ground
start).

 

By default, the global command "caller-id enable" is active, try
entering it again and see if it makes a difference.

 

Also try "caller-id altering ring 1" (before doing this, hook up your
analog phone again and see how many rings before the telco sends
caller-id information)

 

And "caller-id alerting dsp-pre-alloc"

 

Also do a "show voice port x/x" and look for the "Caller ID Info
Follows".

 

Use the "Show call history" to check information received by the caller
id receiver.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Thursday, May 10, 2007 8:34 AM
To: gokhan senol; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] callerID is missing on plar connection

 

FXO has problems when in MGCP mode with callmanager, it has to be in
h323 mode to display the caller ID. I dont know whtere it is the same
for CME or not

 

Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer

 

Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of gokhan senol
Sent: Thu 10-May-07 11:37 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] callerID is missing on plar connection

hi 

i know here is ccm group but i need help about an issue  related to
ccme 

 


2851 ccme 3.3 
manager has direct analogue line. For this i configured plar connection
on fxo port for his second line. 
he can receive calls from this analogue line and he can also call
outside from this line. 
config is below 
problem is CallerID is missing for incoming calls .on phone display
unknown number. 

i directly connected analgue phone to the line and checked if the telco
sends callrID . it is ok. on my cell phone i saw the callerID.  

but when i connected it to the fxo port and made plar connection
callerID is missing.

 


voice-port 0/2/3 

description plar to ephone63 button2
timeouts wait-release 3 
connection plar opx 99 
caller-id enable 

 

!
dial-peer voice 800 pots
 incoming called-number .
 port 0/2/3
!
dial-peer voice 801 pots
 destination-pattern 801
 port 0/2/3
!

 

ephone-dn 2 
number 99 
pickup-group 14 
label direct line 
hold-alert 30 originator 
trunk 801 timeout 10 

 

ephone 63 
ephone-template 1 
mac-address 0019.2F3A.2807 
type 7940 
button 1:12 2:2 

 

thanks a lot 

 


any idea? 
thanks 
Gokhan 

 


 

 

________________________________

Ahhh...imagining that irresistible "new car" smell?
Check out new cars at Yahoo! Autos.
<http://us.rd.yahoo.com/evt=48245/*http:/autos.yahoo.com/new_cars.html;_
ylc=X3oDMTE1YW1jcXJ2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDbmV3LWNhcnM-
>  




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From Nick.Kassel at Charles-Stanley.co.uk  Thu May 10 09:25:00 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 10 May 2007 14:25:00 +0100
Subject: [cisco-voip] MCS 7835 H2 Servers RAID configuration
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451EC9@CSEXCH01.charles-stanley.co.uk>

I have just installed 3 new MCS 7835 H2 servers with 5.1.1b, which have
the serial attached scsi drives. What I would like to know is how do I
check the Raid configuration on these as on the Publisher the 2 hard
drive leds do not come on at all and on the 1st subscriber only 1 drive
has the led working, the 3rd subscriber both lights are working fine.

 

What's the best way to monitor these servers or can I only do this
through RTMT.

 

Regards

 

Nick

 

 

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From garethsc at gmail.com  Thu May 10 09:55:26 2007
From: garethsc at gmail.com (Gareth Cafirma)
Date: Thu, 10 May 2007 09:55:26 -0400
Subject: [cisco-voip] Automatic CFAll?
In-Reply-To: <94a1afde0705100529k5ef0b773saef63496f5c16b50@mail.gmail.com>
References: <8dc3ea820705091346u73d211a5o1390b5d470678733@mail.gmail.com>
	<94a1afde0705100529k5ef0b773saef63496f5c16b50@mail.gmail.com>
Message-ID: <8dc3ea820705100655n4af97c5fk71c67fc79d73b58@mail.gmail.com>

Thanks Ed,

I have read the documents about time of day and time schedule, but from what
I understand it restricts calls and forwarding option if time schedule is in
place, but where to the forwarding to AA takes place?

G


On 5/10/07, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> Could use time of day based routing to send calls to the auto attendant
> after 6PM and then back to the operator's phone at whatever time in the
> morning. The operator would still have to remove the forwarding on their set
> in the morning when they get in, so it wouldnt go to the nighbell.
>
> This doesn't really change the call forwarding settings but i think it
> would have the effect you want.
>
>  On 5/9/07, Gareth Cafirma < garethsc at gmail.com> wrote:
>
> >  Hello Guys,
> >
> > Is it possible to do a "automatic" call forward all to an auto
> > attendant? The client has a 5.1 CCM, unity connection 1.2. Here is the
> > scenario, operator leaves her desk and forward all calls to the nightbell,
> > after 6 PM, they want to remove the forward to nightbell and forward it to
> > auto attendant instead.
> > Can this be done? Any suggestions will be appreciated.
> >
> > Gareth
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
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From tettokalle at hotmail.com  Thu May 10 10:02:25 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Thu, 10 May 2007 16:02:25 +0200
Subject: [cisco-voip] HP Teaming Interface
Message-ID: <f1v8lq$6kl$1@sea.gmane.org>

Hi

yesterday the Teaming Interface has suddenly died on one of my MCS-7825 
Server, has anyone of you had this before? I'm now running only with 1 
NIC, but I dont want, that the server dies again.

Cheers


From lelio at uoguelph.ca  Thu May 10 10:16:13 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 10 May 2007 10:16:13 -0400
Subject: [cisco-voip] codec complexity flex command
Message-ID: <010501c7930d$c42b52e0$47196883@cfs.uoguelph.ca>

I tried entering the "codec complexity flex" command under my voice-card and although it took it, when I do a show run, it only comes up with the following:
  voice-card 0
   no dspfarm
Is the flex code the default? What show command can I use to verify that flex is being used?

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From pmarques74 at gmail.com  Thu May 10 10:31:03 2007
From: pmarques74 at gmail.com (pedro marques)
Date: Thu, 10 May 2007 15:31:03 +0100
Subject: [cisco-voip] Calls from the PSTN getting 'No circuit/channel
	available'
Message-ID: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>

Hi,

I have a CCM running 4.2 connecting through MGCP to a 2821 Gateway.
After some debugging and traces enabled on the CCM I noticed that the calls
that are getting disconnected from the PSTN all fall in the first 5 channels
of the E1 PRI.

These are the relevant commands on the 2821:

controller E1 0/0/0
 pri-group timeslots 1-16 service mgcp
!
interface Serial0/0/0:15
 no ip address
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 isdn bchan-number-order ascending
 no cdp enable
!
ccm-manager mgcp
ccm-manager config server <CCM ip address>
!
mgcp
mgcp call-agent <CCM ip address> service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp package-capability rtp-package
mgcp default-package line-package
mgcp sdp simple
!
mgcp profile default

These are two snippets from a 'debug isdn q931':

*May 10 11:32:02.582: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref =
0x007F
        Sending Complete
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18384
                Preferred, Channel 4
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0083, 'calling party number'
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0x81, 'called party number'
                Plan:ISDN, Type:Unknown
*May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref
= 0x807F
        Channel ID i = 0xA98384
                Exclusive, Channel 4
*May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref
= 0x807F
        Cause i = 0x80A2 - No circuit/channel available
*May 10 11:32:02.790: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref =
0x007F
*May 10 11:32:02.802: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x807F


*May 10 11:33:35.062: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref =
0x0057
        Sending Complete
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18383
                Preferred, Channel 3
        Progress Ind i = 0x8283 - Origination address is non-ISDN
        Calling Party Number i = 0x0083, 'calling party number'
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0x81, 'called party number'
                Plan:ISDN, Type:Unknown
*May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref
= 0x8057
        Channel ID i = 0xA98383
                Exclusive, Channel 3
*May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref
= 0x8057
        Cause i = 0x80A2 - No circuit/channel available

In attachment I include a file (copy of ccm00000065.txt) that has the
relevant information taken from a CCM Trace log and run through Voice Log
Translator.

With all this information I went to check the MGCP Gateway configuration in
CCMAdmin. Some of the configuration elements included on the E1 endpoint
configuration page and that are somewhat confusing me (mind you that this is
an inherited configuration) are as follows:

Channel Selection Order: BOTTOM_UP (since the configuration on the gateway's
interface Serial0/0/0:15 is for bchan-number-order ascending, shouldn't
these parameters match by using the TOP_DOWN option?)

Channel IE Type: Use Number When 1B (This is an option that I don't fully
understand. Any help on this would be appreciated. But to what I understand,
and since the E1 PRI configuration is only used for half a primary (15
channels) shouldn't this parameter be set to 'Timeslot Number'?)

TIA


-- 
Pedro Miguel Marques
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05/10/2007 12:26:07.800	Send	10.105.0.20	Q.931	DISCONNECT	0x800D	S0/SU0/DS1-0/1
Message Header
08 02 80 0D 45
 -- DISCONNECT, pd = 8, callref = 0x800D

Information Element(s)
 08 02 80 A2
 -- Cause i = 0x0800A2 - No channel available 

Entire Message:
0010: 08 02 80 0D 45 08 02 80 A2
--------------------------------------------------------------------

05/10/2007 12:26:27.863	Send	10.105.0.20	Q.931	DISCONNECT	0x8036	S0/SU0/DS1-0/3
Message Header
08 02 80 36 45
 -- DISCONNECT, pd = 8, callref = 0x8036

Information Element(s)
 08 02 80 A2
 -- Cause i = 0x0800A2 - No channel available 

Entire Message:
0010: 08 02 80 36 45 08 02 80 A2
--------------------------------------------------------------------

05/10/2007 12:27:48.615	Send	10.105.0.20	Q.931	DISCONNECT	0x807F	S0/SU0/DS1-0/4
Message Header
08 02 80 7F 45
 -- DISCONNECT, pd = 8, callref = 0x807F

Information Element(s)
 08 02 80 A2
 -- Cause i = 0x0800A2 - No channel available 

Entire Message:
0010: 08 02 80 7F 45 08 02 80 A2
--------------------------------------------------------------------

05/10/2007 12:28:04.100	Send	10.105.0.20	Q.931	DISCONNECT	0x800B	S0/SU0/DS1-0/5
Message Header
08 02 80 0B 45
 -- DISCONNECT, pd = 8, callref = 0x800B

Information Element(s)
 08 02 80 A2
 -- Cause i = 0x0800A2 - No channel available 

Entire Message:
0010: 08 02 80 0B 45 08 02 80 A2
--------------------------------------------------------------------


05/10/2007 12:28:27.179	Send	10.105.0.20	Q.931	DISCONNECT	0x8001	S0/SU0/DS1-0/2
Message Header
08 02 80 01 45
 -- DISCONNECT, pd = 8, callref = 0x8001

Information Element(s)
 08 02 80 A2
 -- Cause i = 0x0800A2 - No channel available 

Entire Message:
0010: 08 02 80 01 45 08 02 80 A2
--------------------------------------------------------------------

05/10/2007 12:28:29.522	Send	10.105.0.20	Q.931	DISCONNECT	0x8069	S0/SU0/DS1-0/2
Message Header
08 02 80 69 45
 -- DISCONNECT, pd = 8, callref = 0x8069

Information Element(s)
 08 02 80 A2
 -- Cause i = 0x0800A2 - No channel available 

Entire Message:
0010: 08 02 80 69 45 08 02 80 A2
--------------------------------------------------------------------

05/10/2007 12:28:37.304	Send	10.105.0.20	Q.931	DISCONNECT	0x8075	S0/SU0/DS1-0/2
Message Header
08 02 80 75 45
 -- DISCONNECT, pd = 8, callref = 0x8075

Information Element(s)
 08 02 80 A2
 -- Cause i = 0x0800A2 - No channel available 

Entire Message:
0010: 08 02 80 75 45 08 02 80 A2
--------------------------------------------------------------------



05/10/2007 12:29:21.102	Send	10.105.0.20	Q.931	DISCONNECT	0x8057	S0/SU0/DS1-0/3
Message Header
08 02 80 57 45
 -- DISCONNECT, pd = 8, callref = 0x8057

Information Element(s)
 08 02 80 A2
 -- Cause i = 0x0800A2 - No channel available 

Entire Message:
0010: 08 02 80 57 45 08 02 80 A2
--------------------------------------------------------------------

From jonvoip at gmail.com  Thu May 10 10:42:40 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 10 May 2007 09:42:40 -0500
Subject: [cisco-voip] IOS strip digits from 8 to 4
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00586BF11@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00586BF11@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0705100742h6c636544vcc98f4d3da7b0f3f@mail.gmail.com>

On the inbound peer

incoming called-number ....



Jonathan

On 5/9/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>  Telco is delivering 8 digits in Malaysia, I only need 4 digits
> significant.  I don't want to ask them to change from 8 digits to 4.
>
>
>
> This is all h323 dial-peers, some DIDs go to CallManager, some DIDs go to
> CVP, etc. I can't use MGCP or do a translation-pattern in CCM.
>
>
>
> How best should I reduce the 8 digits received to 4 via IOS Gateway?
>
>
>
> DID Range 8808-87XX
>
>
>
> num-exp 880887.. ?.
>
>
>
>
>
> Jason Aarons
>
> Consultant
>
> http://www.dimensiondata.com/na
>
> 904-338-3245 cell
>
>
>
> For urgent issues notify your Project Manager, for 24x7 support contact
> the Dimension Data NOC at 800-974-6584.
>
>
>
> ------------------------------
>
> * Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
> addressee(s) named above only. If you are not the intended addressee, you
> are hereby notified that you have received this communication in error and
> that any use or reproduction of this email or its contents is strictly
> prohibited and may be unlawful. If you have received this communication in
> error, please notify us immediately by replying to this message and deleting
> it from your computer. Thank you. *
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From jonvoip at gmail.com  Thu May 10 10:45:04 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 10 May 2007 09:45:04 -0500
Subject: [cisco-voip] MCS 7835 H2 Servers RAID configuration
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451EC9@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451EC9@CSEXCH01.charles-stanley.co.uk>
Message-ID: <5d093f9a0705100745x4673ec3es13c22ff4437715cc@mail.gmail.com>

Well, you could do a reboot and see what the raid status is...

Also, IIRC, the LEDs are disabled on the 7825s..

(big sticker over the drvies when they ship says so...)


Jonathan

On 5/10/07, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote:
>
>  I have just installed 3 new MCS 7835 H2 servers with 5.1.1b, which have
> the serial attached scsi drives. What I would like to know is how do I check
> the Raid configuration on these as on the Publisher the 2 hard drive leds do
> not come on at all and on the 1st subscriber only 1 drive has the led
> working, the 3rd subscriber both lights are working fine.
>
>
>
> What's the best way to monitor these servers or can I only do this through
> RTMT.
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>
> Regards
>
>
>
> Nick
>
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From jonvoip at gmail.com  Thu May 10 10:47:08 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 10 May 2007 09:47:08 -0500
Subject: [cisco-voip] Calls from the PSTN getting 'No circuit/channel
	available'
In-Reply-To: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>
References: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>
Message-ID: <5d093f9a0705100747m2c309f45l342f93c10fc14769@mail.gmail.com>

This message means that the digits being sent did not match any
route-pattern, route point, translation pattern or DN.

You stripped out the digits under called-party number, so I can't tell you
what to search for in Route-Plan report...

Just make sure the digits being sent, match something on CCM.



Jonathan



On 5/10/07, pedro marques <pmarques74 at gmail.com> wrote:
>
> Hi,
>
> I have a CCM running 4.2 connecting through MGCP to a 2821 Gateway.
> After some debugging and traces enabled on the CCM I noticed that the
> calls that are getting disconnected from the PSTN all fall in the first 5
> channels of the E1 PRI.
>
> These are the relevant commands on the 2821:
>
> controller E1 0/0/0
>  pri-group timeslots 1-16 service mgcp
> !
> interface Serial0/0/0:15
>  no ip address
>  isdn switch-type primary-net5
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  isdn bchan-number-order ascending
>  no cdp enable
> !
> ccm-manager mgcp
> ccm-manager config server <CCM ip address>
> !
> mgcp
> mgcp call-agent <CCM ip address> service-type mgcp version 0.1
> mgcp dtmf-relay voip codec all mode out-of-band
> mgcp package-capability rtp-package
> mgcp default-package line-package
> mgcp sdp simple
> !
> mgcp profile default
>
> These are two snippets from a 'debug isdn q931':
>
> *May 10 11:32:02.582: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref =
> 0x007F
>         Sending Complete
>         Bearer Capability i = 0x9090A3
>                 Standard = CCITT
>                 Transfer Capability = 3.1kHz Audio
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA18384
>                 Preferred, Channel 4
>         Progress Ind i = 0x8183 - Origination address is non-ISDN
>         Calling Party Number i = 0x0083, 'calling party number'
>                 Plan:Unknown, Type:Unknown
>         Called Party Number i = 0x81, 'called party number'
>                 Plan:ISDN, Type:Unknown
> *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
> callref = 0x807F
>         Channel ID i = 0xA98384
>                 Exclusive, Channel 4
> *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8
> callref = 0x807F
>         Cause i = 0x80A2 - No circuit/channel available
> *May 10 11:32:02.790: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref
> = 0x007F
> *May 10 11:32:02.802: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8
> callref = 0x807F
>
>
> *May 10 11:33:35.062: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref =
> 0x0057
>         Sending Complete
>         Bearer Capability i = 0x9090A3
>                 Standard = CCITT
>                 Transfer Capability = 3.1kHz Audio
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA18383
>                 Preferred, Channel 3
>         Progress Ind i = 0x8283 - Origination address is non-ISDN
>         Calling Party Number i = 0x0083, 'calling party number'
>                 Plan:Unknown, Type:Unknown
>         Called Party Number i = 0x81, 'called party number'
>                 Plan:ISDN, Type:Unknown
> *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
> callref = 0x8057
>         Channel ID i = 0xA98383
>                 Exclusive, Channel 3
> *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8
> callref = 0x8057
>         Cause i = 0x80A2 - No circuit/channel available
>
> In attachment I include a file (copy of ccm00000065.txt) that has the
> relevant information taken from a CCM Trace log and run through Voice Log
> Translator.
>
> With all this information I went to check the MGCP Gateway configuration
> in CCMAdmin. Some of the configuration elements included on the E1 endpoint
> configuration page and that are somewhat confusing me (mind you that this is
> an inherited configuration) are as follows:
>
> Channel Selection Order: BOTTOM_UP (since the configuration on the
> gateway's interface Serial0/0/0:15 is for bchan-number-order ascending,
> shouldn't these parameters match by using the TOP_DOWN option?)
>
> Channel IE Type: Use Number When 1B (This is an option that I don't fully
> understand. Any help on this would be appreciated. But to what I understand,
> and since the E1 PRI configuration is only used for half a primary (15
> channels) shouldn't this parameter be set to 'Timeslot Number'?)
>
> TIA
>
>
> --
> Pedro Miguel Marques
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From pmarques74 at gmail.com  Thu May 10 11:02:37 2007
From: pmarques74 at gmail.com (pedro marques)
Date: Thu, 10 May 2007 16:02:37 +0100
Subject: [cisco-voip] Calls from the PSTN getting 'No circuit/channel
	available'
In-Reply-To: <5d093f9a0705100747m2c309f45l342f93c10fc14769@mail.gmail.com>
References: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>
	<5d093f9a0705100747m2c309f45l342f93c10fc14769@mail.gmail.com>
Message-ID: <d2216e4e0705100802r450ad5f8l4d7886be8d95bbff@mail.gmail.com>

Hi Jonathan,

I include the same debugs with the calling and called party numbers:

*May 10 11:32:02.582: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref =
0x007F
        Sending Complete
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18384
                Preferred, Channel 4
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0083, '244560496'
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0x81, '3300'
                Plan:ISDN, Type:Unknown
*May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref
= 0x807F
        Channel ID i = 0xA98384
                Exclusive, Channel 4
*May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref
= 0x807F
        Cause i = 0x80A2 - No circuit/channel available
*May 10 11:32:02.790: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref =
0x007F
*May 10 11:32:02.802: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x807F


*May 10 11:33:35.062: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref =
0x0057
        Sending Complete
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18383
                Preferred, Channel 3
        Progress Ind i = 0x8283 - Origination address is non-ISDN
        Calling Party Number i = 0x0083, '244502150'
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0x81, '3300'
                Plan:ISDN, Type:Unknown
*May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref
= 0x8057
        Channel ID i = 0xA98383
                Exclusive, Channel 3
*May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref
= 0x8057
        Cause i = 0x80A2 - No circuit/channel available

The '3300' number is associated with two translation patterns. One points to
a Hunt Pilot (5930) that is associated to a partition (WorkHours) which is
further associated to a Time Schedule containing several Time Periods. The
other one points directly to a DN (5250) associated to a different partition
(NonWorkHours) further associated with a Time Schedule that contains several
Time Periods.

On 10/05/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> This message means that the digits being sent did not match any
> route-pattern, route point, translation pattern or DN.
>
> You stripped out the digits under called-party number, so I can't tell you
> what to search for in Route-Plan report...
>
> Just make sure the digits being sent, match something on CCM.
>
>
>
> Jonathan
>
>
>
> On 5/10/07, pedro marques < pmarques74 at gmail.com> wrote:
>
> > Hi,
> >
> > I have a CCM running 4.2 connecting through MGCP to a 2821 Gateway.
> > After some debugging and traces enabled on the CCM I noticed that the
> > calls that are getting disconnected from the PSTN all fall in the first 5
> > channels of the E1 PRI.
> >
> > These are the relevant commands on the 2821:
> >
> > controller E1 0/0/0
> >  pri-group timeslots 1-16 service mgcp
> > !
> > interface Serial0/0/0:15
> >  no ip address
> >  isdn switch-type primary-net5
> >  isdn incoming-voice voice
> >  isdn bind-l3 ccm-manager
> >  isdn bchan-number-order ascending
> >  no cdp enable
> > !
> > ccm-manager mgcp
> > ccm-manager config server <CCM ip address>
> > !
> > mgcp
> > mgcp call-agent <CCM ip address> service-type mgcp version 0.1
> > mgcp dtmf-relay voip codec all mode out-of-band
> > mgcp package-capability rtp-package
> > mgcp default-package line-package
> > mgcp sdp simple
> > !
> > mgcp profile default
> >
> > These are two snippets from a 'debug isdn q931':
> >
> > *May 10 11:32:02.582: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref
> > = 0x007F
> >         Sending Complete
> >         Bearer Capability i = 0x9090A3
> >                 Standard = CCITT
> >                 Transfer Capability = 3.1kHz Audio
> >                 Transfer Mode = Circuit
> >                 Transfer Rate = 64 kbit/s
> >         Channel ID i = 0xA18384
> >                 Preferred, Channel 4
> >         Progress Ind i = 0x8183 - Origination address is non-ISDN
> >         Calling Party Number i = 0x0083, 'calling party number'
> >                 Plan:Unknown, Type:Unknown
> >         Called Party Number i = 0x81, 'called party number'
> >                 Plan:ISDN, Type:Unknown
> > *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
> > callref = 0x807F
> >         Channel ID i = 0xA98384
> >                 Exclusive, Channel 4
> > *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8
> > callref = 0x807F
> >         Cause i = 0x80A2 - No circuit/channel available
> > *May 10 11:32:02.790: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8
> > callref = 0x007F
> > *May 10 11:32:02.802: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8
> > callref = 0x807F
> >
> >
> > *May 10 11:33:35.062: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref
> > = 0x0057
> >         Sending Complete
> >         Bearer Capability i = 0x9090A3
> >                 Standard = CCITT
> >                 Transfer Capability = 3.1kHz Audio
> >                 Transfer Mode = Circuit
> >                 Transfer Rate = 64 kbit/s
> >         Channel ID i = 0xA18383
> >                 Preferred, Channel 3
> >         Progress Ind i = 0x8283 - Origination address is non-ISDN
> >         Calling Party Number i = 0x0083, 'calling party number'
> >                 Plan:Unknown, Type:Unknown
> >         Called Party Number i = 0x81, 'called party number'
> >                 Plan:ISDN, Type:Unknown
> > *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
> > callref = 0x8057
> >         Channel ID i = 0xA98383
> >                 Exclusive, Channel 3
> > *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8
> > callref = 0x8057
> >         Cause i = 0x80A2 - No circuit/channel available
> >
> > In attachment I include a file (copy of ccm00000065.txt) that has the
> > relevant information taken from a CCM Trace log and run through Voice Log
> > Translator.
> >
> > With all this information I went to check the MGCP Gateway configuration
> > in CCMAdmin. Some of the configuration elements included on the E1 endpoint
> > configuration page and that are somewhat confusing me (mind you that this is
> > an inherited configuration) are as follows:
> >
> > Channel Selection Order: BOTTOM_UP (since the configuration on the
> > gateway's interface Serial0/0/0:15 is for bchan-number-order ascending,
> > shouldn't these parameters match by using the TOP_DOWN option?)
> >
> > Channel IE Type: Use Number When 1B (This is an option that I don't
> > fully understand. Any help on this would be appreciated. But to what I
> > understand, and since the E1 PRI configuration is only used for half a
> > primary (15 channels) shouldn't this parameter be set to 'Timeslot
> > Number'?)
> >
> > TIA
> >
> >
> > --
> > Pedro Miguel Marques
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>


-- 
Pedro Miguel Marques
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From pmarques74 at gmail.com  Thu May 10 11:07:51 2007
From: pmarques74 at gmail.com (pedro marques)
Date: Thu, 10 May 2007 16:07:51 +0100
Subject: [cisco-voip] Calls from the PSTN getting 'No circuit/channel
	available'
In-Reply-To: <d2216e4e0705100802r450ad5f8l4d7886be8d95bbff@mail.gmail.com>
References: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>
	<5d093f9a0705100747m2c309f45l342f93c10fc14769@mail.gmail.com>
	<d2216e4e0705100802r450ad5f8l4d7886be8d95bbff@mail.gmail.com>
Message-ID: <d2216e4e0705100807s15d5e2c8u6a1083e42c716310@mail.gmail.com>

??What IOS version?

Chris,

The gateway is running 12.4(3f).


Pedro Miguel Marques
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From chris.clouse at berbee.com  Thu May 10 11:09:17 2007
From: chris.clouse at berbee.com (Clouse, Chris)
Date: Thu, 10 May 2007 10:09:17 -0500
Subject: [cisco-voip] Calls from the PSTN getting 'No circuit/channel
	available'
In-Reply-To: <d2216e4e0705100807s15d5e2c8u6a1083e42c716310@mail.gmail.com>
References: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>
	<5d093f9a0705100747m2c309f45l342f93c10fc14769@mail.gmail.com>
	<d2216e4e0705100802r450ad5f8l4d7886be8d95bbff@mail.gmail.com>
	<d2216e4e0705100807s15d5e2c8u6a1083e42c716310@mail.gmail.com>
Message-ID: <310F10740485D9469F0D8417950B252D027AFB46@CTG-MSNEXC01.staff.berbee.com>

I've seen bugs with E1s in the 12.4(3x) code.  Switch to 12.4(7d) and
see if that helps.

 

Christopher Clouse, CCNP CCDP CCVP MCP Network+

Berbee

Network Voice Engineer

7145 Boone Avenue North, Suite 140

Brooklyn Park, MN 55428

*763.592.5887 Direct

*763.535.7428 Fax

*612.419.9751 Mobile

 

From: pedro marques [mailto:pmarques74 at gmail.com] 
Sent: Thursday, May 10, 2007 10:08 AM
To: Clouse, Chris
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Calls from the PSTN getting 'No
circuit/channel available'

 

>>What IOS version?

Chris,

The gateway is running 12.4(3f).


Pedro Miguel Marques 

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From pmarques74 at gmail.com  Thu May 10 11:14:23 2007
From: pmarques74 at gmail.com (pedro marques)
Date: Thu, 10 May 2007 16:14:23 +0100
Subject: [cisco-voip] codec complexity flex command
In-Reply-To: <010501c7930d$c42b52e0$47196883@cfs.uoguelph.ca>
References: <010501c7930d$c42b52e0$47196883@cfs.uoguelph.ca>
Message-ID: <d2216e4e0705100814m76c71df2q5c3ed3397dddcc9c@mail.gmail.com>

Lelio,

If your hardware is using the C5510 chipset I believe the default mode would
be flex mode. Anyhow the command 'show voice dsp voice' will show what
chipset(s) is being used on the hardware.

On 10/05/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  I tried entering the "codec complexity flex" command under my voice-card
> and although it took it, when I do a show run, it only comes up with the
> following:
>
> voice-card 0
>  no dspfarm
>
> Is the flex code the default? What show command can I use to verify that
> flex is being used?
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Pedro Miguel Marques
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From matt at saskin.net  Thu May 10 11:33:59 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 10 May 2007 11:33:59 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
In-Reply-To: <94a1afde0705100526ycf3693co47630081d6c970d9@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>	<002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>	<C1FE15183DA37645BC0633BC604E44F00583915E@USNAEXCH.na.didata.local>	<88C8D019DA547C4BABECB9A806ED22ED88A04E@mail.ACT-EG.COM>
	<94a1afde0705100526ycf3693co47630081d6c970d9@mail.gmail.com>
Message-ID: <46433B67.3090308@saskin.net>

How do you intend to create a publisher in the "old" cluster once you 
split them?

In theory the subscribers will keep chugging along with some end-user 
loss of features...well, and the lack of ability to make configuration 
changes.

-matt

Ed Leatherman wrote:
> Related question, and I think the answer is also probably no but want 
> to verify...
>
> When upgrading from ccm 4.1 to 5, would it be possible to change the 
> host name of the primary node in the 5.0 cluster after loading the DMA 
> data but prior to adding any subscriber nodes?
>
> Trying to come up with a way to upgrade our system to 5 but leave 2 
> nodes in the old cluster active to operate the call center users and 
> IPCCX - or at least determine if its feasible. so basically to split 
> the cluster up.
>
> On 5/10/07, *Ahmed Elnagar* <aelnagar at act-eg.com 
> <mailto:aelnagar at act-eg.com>> wrote:
>
>     dont even try it you will break your CM replication and will go
>     through alot of very strange problems.
>      
>     Thanks and Best Regards
>     *
>     Ahmed A. Elnagar
>     *Network Field Engineer
>      
>     Advanced Computer Technology (ACT)
>     16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
>     Postal Code:12411 Cairo Egypt
>
>     *Mob** :* +2010-2833868
>     *Website**: * www.act-eg.com <http://www.act-eg.com>
>     *E-mail**: * aelnagar at act-eg.com <mailto:aelnagar at act-eg.com>
>
>     ------------------------------------------------------------------------
>     *From:* cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net> on behalf of Matt
>     Slaga (US)
>     *Sent:* Wed 09-May-07 11:05 PM
>     *To:* Ryan Ratliff; Jason Aarons (US)
>
>     *Cc:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     *Subject:* Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?
>
>     If the customer really pushes you, create a DNS C-name record.
>
>     -----Original Message-----
>     From: cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
>     [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
>     Sent: Wednesday, May 09, 2007 3:58 PM
>     To: Jason Aarons (US)
>     Cc: cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     Subject: Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?
>
>     No way, no how.
>
>     The only time you can change the hostname of a CM server is before 
>     you install CallManager on it.
>
>     -Ryan
>
>     On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:
>
>     I'm trying to find a reference about server hostname changes for 
>     production CallManager 4.1, I recall it's not advised but can't find 
>     a article on cisco.com <http://cisco.com> stating this -jason
>
>
>
>
>
>     Jason Aarons
>
>     Consultant
>
>     http://www.dimensiondata.com/na <http://www.dimensiondata.com/na>
>
>     904-338-3245 cell
>
>
>
>     For urgent issues notify your Project Manager, for 24x7 support 
>     contact the Dimension Data NOC at 800-974-6584.
>
>
>
>
>
>     Disclaimer: This e-mail communication and any attachments may
>     contain 
>     confidential and privileged information and is for use by the 
>     designated addressee(s) named above only. If you are not the intended 
>     addressee, you are hereby notified that you have received this 
>     communication in error and that any use or reproduction of this
>     email 
>     or its contents is strictly prohibited and may be unlawful. If you 
>     have received this communication in error, please notify us 
>     immediately by replying to this message and deleting it from your 
>     computer. Thank you.
>
>     _______________________________________________
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>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>     -----------------------------------------
>     Disclaimer:
>
>     This e-mail communication and any attachments may contain
>     confidential and privileged information and is for use by the
>     designated addressee(s) named above only.  If you are not the
>     intended addressee, you are hereby notified that you have received
>     this communication in error and that any use or reproduction of
>     this email or its contents is strictly prohibited and may be
>     unlawful.  If you have received this communication in error, please
>     notify us immediately by replying to this message and deleting it
>     from your computer. Thank you.
>
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> -- 
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   



From matt at saskin.net  Thu May 10 11:36:36 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 10 May 2007 11:36:36 -0400
Subject: [cisco-voip] CDR wrong number
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22ED88A045@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22ED88A045@mail.ACT-EG.COM>
Message-ID: <46433C04.2010400@saskin.net>

I *think* that if you were use a translation pattern on the inbound 
calls (i.e.; place it in a partition that comes first in the CSS that 
the gateway calls come inbound into) and strip the digits at the 
translation pattern level then you will end up with the post-translation 
digits in the CDR's - this is assuming you're running MGCP gateways.

If h323, you can take care of stripping on the inbound dial-peer before 
you ever hit CM.

-matt

Ahmed Elnagar wrote:
> *I have a similiare problem to that one any ideas??*
>  
> One of our peering partners send their traffic to our gateway with a 
> tech-prefix
> of 1947#.
> Our country code is 94.
>
> Therefore our POTS dial-peer on gateway is-
> destination-pattern 1947#94T
>
> Call termination is happening fine.
>
> BUT, our CDR records the incoming call as 1947#94xxxxxx and billing 
> system identifies it as a USA call.(since it starts from 1)
>
> We want our CDR to be 94xxxxxx ONLY.(without the tech-prefix)
>
> Could you please give me an advice to get this thing done?
>  
> Thanks and Best Regards
> *
> Ahmed A. Elnagar
> *Network Field Engineer
>  
> Advanced Computer Technology (ACT)
> 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> Postal Code:12411 Cairo Egypt
>
> *Mob**:* +2010-2833868
> *Website**: *www.act-eg.com
> *E-mail**: *aelnagar at act-eg.com <mailto:aelnagar at act-eg.com>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   



From jeremie.ceintrey at gmail.com  Thu May 10 11:39:07 2007
From: jeremie.ceintrey at gmail.com (Jeremie Ceintrey)
Date: Thu, 10 May 2007 17:39:07 +0200
Subject: [cisco-voip] Thomson st 2030 on Callmanager
Message-ID: <dc3a89dd0705100839j610038c4xbf690b5c38605478@mail.gmail.com>

Hi,

I'm trying to configure a ST 2030 Thomson phone on a Cisco Callmanager 5 but
it seems that it doesn't work. the phone receive an error message just after
it receives 200 OK sip message from the Callmanager (error 500 Internal
Server error)

has anyone ever configured this kind of sip phone on a Callmanager 5?

Thanks.

J?r?mie.
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From fwn at feasible.net  Thu May 10 11:44:01 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Thu, 10 May 2007 08:44:01 -0700
Subject: [cisco-voip] MCS 7835 H2 Servers RAID configuration
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451EC9@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451EC9@CSEXCH01.charles-stanley.co.uk>
Message-ID: <ca09e3750705100844v2a1cad67vab7e5261dd7c33f5@mail.gmail.com>

"show hardware" from the CLI will provide a basic indication of RAID status

On 5/10/07, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote:
>
>  I have just installed 3 new MCS 7835 H2 servers with 5.1.1b, which have
> the serial attached scsi drives. What I would like to know is how do I check
> the Raid configuration on these as on the Publisher the 2 hard drive leds do
> not come on at all and on the 1st subscriber only 1 drive has the led
> working, the 3rd subscriber both lights are working fine.
>
>
>
> What's the best way to monitor these servers or can I only do this through
> RTMT.
>
>
>
> Regards
>
>
>
> Nick
>
>
>
>
>
>
>
>
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From ealeatherman at gmail.com  Thu May 10 11:44:55 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 10 May 2007 11:44:55 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
In-Reply-To: <46433B67.3090308@saskin.net>
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>
	<002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00583915E@USNAEXCH.na.didata.local>
	<88C8D019DA547C4BABECB9A806ED22ED88A04E@mail.ACT-EG.COM>
	<94a1afde0705100526ycf3693co47630081d6c970d9@mail.gmail.com>
	<46433B67.3090308@saskin.net>
Message-ID: <94a1afde0705100844y6f7557adm732eb3b3e4939b33@mail.gmail.com>

My thought was to do an install on a new server for the 5.1 publisher node
w/new IP Address and restore the DMA data file onto that, thereby not even
touching the 4.1 publisher (other than running DMA). Would also leave 1
subscriber node from the 4.1 cluster alone, and upgrade the others and point
them to the new 5.1 pub. Just not sure what would happen having two clusters
with publishers having the same host name (maybe no problem?)

On 5/10/07, Matthew Saskin <matt at saskin.net> wrote:
>
> How do you intend to create a publisher in the "old" cluster once you
> split them?
>
> In theory the subscribers will keep chugging along with some end-user
> loss of features...well, and the lack of ability to make configuration
> changes.
>
> -matt
>
> Ed Leatherman wrote:
> > Related question, and I think the answer is also probably no but want
> > to verify...
> >
> > When upgrading from ccm 4.1 to 5, would it be possible to change the
> > host name of the primary node in the 5.0 cluster after loading the DMA
> > data but prior to adding any subscriber nodes?
> >
> > Trying to come up with a way to upgrade our system to 5 but leave 2
> > nodes in the old cluster active to operate the call center users and
> > IPCCX - or at least determine if its feasible. so basically to split
> > the cluster up.
> >
> > On 5/10/07, *Ahmed Elnagar* <aelnagar at act-eg.com
> > <mailto:aelnagar at act-eg.com>> wrote:
> >
> >     dont even try it you will break your CM replication and will go
> >     through alot of very strange problems.
> >
> >     Thanks and Best Regards
> >     *
> >     Ahmed A. Elnagar
> >     *Network Field Engineer
> >
> >     Advanced Computer Technology (ACT)
> >     16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> >     Postal Code:12411 Cairo Egypt
> >
> >     *Mob** :* +2010-2833868
> >     *Website**: * www.act-eg.com <http://www.act-eg.com>
> >     *E-mail**: * aelnagar at act-eg.com <mailto:aelnagar at act-eg.com>
> >
> >
> ------------------------------------------------------------------------
> >     *From:* cisco-voip-bounces at puck.nether.net
> >     <mailto:cisco-voip-bounces at puck.nether.net> on behalf of Matt
> >     Slaga (US)
> >     *Sent:* Wed 09-May-07 11:05 PM
> >     *To:* Ryan Ratliff; Jason Aarons (US)
> >
> >     *Cc:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     *Subject:* Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?
> >
> >     If the customer really pushes you, create a DNS C-name record.
> >
> >     -----Original Message-----
> >     From: cisco-voip-bounces at puck.nether.net
> >     <mailto:cisco-voip-bounces at puck.nether.net>
> >     [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan
> Ratliff
> >     Sent: Wednesday, May 09, 2007 3:58 PM
> >     To: Jason Aarons (US)
> >     Cc: cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     Subject: Re: [cisco-voip] Changing 4.1(3) hostname article on CCO?
> >
> >     No way, no how.
> >
> >     The only time you can change the hostname of a CM server is before
> >     you install CallManager on it.
> >
> >     -Ryan
> >
> >     On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:
> >
> >     I'm trying to find a reference about server hostname changes for
> >     production CallManager 4.1, I recall it's not advised but can't find
> >     a article on cisco.com <http://cisco.com> stating this -jason
> >
> >
> >
> >
> >
> >     Jason Aarons
> >
> >     Consultant
> >
> >     http://www.dimensiondata.com/na <http://www.dimensiondata.com/na>
> >
> >     904-338-3245 cell
> >
> >
> >
> >     For urgent issues notify your Project Manager, for 24x7 support
> >     contact the Dimension Data NOC at 800-974-6584.
> >
> >
> >
> >
> >
> >     Disclaimer: This e-mail communication and any attachments may
> >     contain
> >     confidential and privileged information and is for use by the
> >     designated addressee(s) named above only. If you are not the
> intended
> >     addressee, you are hereby notified that you have received this
> >     communication in error and that any use or reproduction of this
> >     email
> >     or its contents is strictly prohibited and may be unlawful. If you
> >     have received this communication in error, please notify us
> >     immediately by replying to this message and deleting it from your
> >     computer. Thank you.
> >
> >     _______________________________________________
> >     cisco-voip mailing list
> >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >     _______________________________________________
> >     cisco-voip mailing list
> >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >     -----------------------------------------
> >     Disclaimer:
> >
> >     This e-mail communication and any attachments may contain
> >     confidential and privileged information and is for use by the
> >     designated addressee(s) named above only.  If you are not the
> >     intended addressee, you are hereby notified that you have received
> >     this communication in error and that any use or reproduction of
> >     this email or its contents is strictly prohibited and may be
> >     unlawful.  If you have received this communication in error, please
> >     notify us immediately by replying to this message and deleting it
> >     from your computer. Thank you.
> >
> >     _______________________________________________
> >     cisco-voip mailing list
> >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >     _______________________________________________
> >     cisco-voip mailing list
> >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >
> > --
> > Ed Leatherman
> > Senior Voice Engineer
> > West Virginia University
> > Telecommunications and Network Operations
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Thu May 10 12:33:55 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 10 May 2007 12:33:55 -0400
Subject: [cisco-voip] MLPP and 911 calls
Message-ID: <01c201c79321$00cd3200$47196883@cfs.uoguelph.ca>

I'm looking to configure MLPP for 911 calls on our network. I've read through the documentation, but am a little confused as to when indication/preemption need to be enabled. I'm pretty sure that I don't need preemption enabled since I won't be preempting calls to IP phones, but not sure if I need indication enabled if I am going to be placing calls that will be preempted on a gateway.

Here's what I was thinking I have to do:
  a.. leave enterprise parameter MLPP domain as 0
  b.. leave enterprise parameter MLPP Indication Status = MLPP Indication turned off.
  c.. leave enterprise parameter MLPP Preemption Setting  = No preemption allowed.
  d.. configure 911 route pattern as executive override
  e.. configure trunks with MLPP Preemption Forceful
Will the call about to be pre-empted hear anything?

Any comments would be helpful.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From axe7961 at hotmail.com  Thu May 10 12:42:41 2007
From: axe7961 at hotmail.com (Axel Wilzinski)
Date: Thu, 10 May 2007 12:42:41 -0400
Subject: [cisco-voip] Conference Bridge on IOS Router
Message-ID: <BAY103-W2D24B5997102BCDCA7D66BA3A0@phx.gbl>

Yes, we have a T1 PRI card.  In my original message I said we have 32 DSP's, but I was wrong, we have 64.We have a 2851 with a with a PVDM2 upgraded to 64 DSP's.   In addition to the VWIC PRI card, we have a high density 8FXS/DID module (EVM-HD-8FXS/DID) with a 6 port FXO expansion module (EM-HDA-6FXO).  The IOS version is 12.4(11)T1.  (All MGCP)When our VoIP consultant originally installed our gateway, he spent several hours trying to get the conference bridge to work and somehow determined that our configuration is not supported.  He said we had enough DSP's but they could not be used for both the bridge and the PRI's and FXS and FXO ports.  I didn't know enough about how all this works at the time to question him further.  Now that several months have passed and I have had some time to get familiar with Call Manager and other gateways that we installed without the consultant, I'm interested in researching this again.I will look at the Cisco link that was sent to me in this thread and read the documentation closer.ThanksDate: Wed, 9 May 2007 20:36:18 -0600From: nws at fredf.orgTo: axe7961 at hotmail.comSubject: Re: [cisco-voip] Conference Bridge on IOS RouterCC: cisco-voip at puck.nether.net24 of those dsp's are gonna be tied up on a t1 card.  Thats if you have one.  Need a little more info tho to really help.  The problem we ran into is the t1's ate 24  dsp channels, and since the transcoder config is usually first in the ios, if there any configured theres no room left for the conferencing....   If you don't have a T1 card in there, then you got plenty of resources.  What kind of issue are you having?   Do you have the SCCP config setup?   well frist I should ask what kind of IOS device is this, an what has the PVDMS in it.  IS it onboard like an ISR router or off board in an nm-hdv slot.   There are several configs to address it, just depends on what equipment you have.

On 5/9/07, Axel Wilzinski <axe7961 at hotmail.com> wrote:

Has anyone had any luck getting local conferencing working on an IOS gateway?  We are using MGCP with a remote Call Manager (4.2.3).  The local gateway has a PVDM2 with 32 DSP's.  Currently, conference calls traverse the WAN even for a local conference call.
 Any help would be appreciated. Thanks!

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From axe7961 at hotmail.com  Thu May 10 12:52:02 2007
From: axe7961 at hotmail.com (Axel Wilzinski)
Date: Thu, 10 May 2007 12:52:02 -0400
Subject: [cisco-voip] Echo on IP Phones through PRI
Message-ID: <BAY103-W12FE4D6B2850172B084912BA3A0@phx.gbl>

We had a similar problem that was related to slip seconds on the T1.  It was interesting because even though the slip seconds were consistent, the echo was not.  The root of the problem was caused by not having a "network-clock-select" statement in the config.  This problem drove us crazy for a couple of weeks because we upgraded IOS and DSP firmware per Cisco's instructions to no avail.Do a show controllers t1 0/0/0 (or whatever port the PRI is on) and make sure it's clean.  You should see a lot of zero's.> Date: Thu, 10 May 2007 11:34:44 +0100> From: matt at melbourne.org.uk> To: smgustafson at gmail.com> CC: cisco-voip at puck.nether.net> Subject: Re: [cisco-voip] Echo on IP Phones through PRI> > Thanks Steve. We have an upgrade planned for our voice gateways in an> forthcoming maintenance window, where we'll try 12.4(latest) mainline.> Are you aware of any bugs which document this issue?> > Cheers,> > Matt> > On 09/05/07, Steve G <smgustafson at gmail.com> wrote:> > 12.3(14) will deffinietly give you echo issues.  it is the DSP firmware.> > either get new DSP firmware from TAC, or upgrade IOS.  I am using 12.4(8)b> > with no issues.> >> > Steve> >> >> > On 4/18/07, Matthew Melbourne <matt at melbourne.org.uk> wrote:> > >> > > We are getting sporadic issues from users on IP phones who are hearing> > > echo on external calls. From this document is would suggest that echo> > > on IP Phones is caused by hybrid echo and not any local acoustic echo.> > >> > >> > http://www.cisco.com/warp/public/788/voice-qos/echo_ipphone_gateway.html> > >> > > The MGCP-controlled voice gateways are all running 12.3(14)T7 with DSP> > > firmware 4.4.708. In CallManager (4.1(3)SR2), echo cancellation is on> > > for each port, with a "default" echo cancellation coverage of 64ms.> > > Input gain and output attenuation are set to 0dB.> > >> > > I have seen suggestions to include the "echo-cancel erl worst-case 3"> > > command on the voice ports. Can this help in situations of occasional> > > echo?> > >> > > Are there any other parameters changes which could help? Could it be a> > > code/DSP firmware issue? I haven't found a specific echo-related bug> > > relating to this particular IOS and its embedded DSP firmware.> > >> > > On occasions, the start of the call is fine but echo comes and goes> > > during the call.> > >> > > Is there a quick way of finding out which voice gateway a call is> > > terminated on (we have 8, and all could be used in a round-robin> > > fashion for outbound calls).> > >> > > Cheers,> > >> > > Matt> > >> > > --> > > Matthew Melbourne> > > _______________________________________________> > > cisco-voip mailing list> > > cisco-voip at puck.nether.net> > > https://puck.nether.net/mailman/listinfo/cisco-voip> > >> >> >> > > -- > Matthew Melbourne> _______________________________________________> cisco-voip mailing list> cisco-voip at puck.nether.net> https://puck.nether.net/mailman/listinfo/cisco-voip
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From smgustafson at gmail.com  Thu May 10 12:57:27 2007
From: smgustafson at gmail.com (Steve G)
Date: Thu, 10 May 2007 09:57:27 -0700
Subject: [cisco-voip] Echo on IP Phones through PRI
In-Reply-To: <b718d1880705100334r797c5d9dp68e6674e21dc02b4@mail.gmail.com>
References: <b718d1880704180550r52c715c5x80e83be222546ba2@mail.gmail.com>
	<8a1f34b50705091546s70b357c6p702138bb016f45ed@mail.gmail.com>
	<b718d1880705100334r797c5d9dp68e6674e21dc02b4@mail.gmail.com>
Message-ID: <8a1f34b50705100957u4b85ddc1i1e58dd86e0b41349@mail.gmail.com>

CSCsd54344


On 5/10/07, Matthew Melbourne <matt at melbourne.org.uk> wrote:
>
> Thanks Steve. We have an upgrade planned for our voice gateways in an
> forthcoming maintenance window, where we'll try 12.4(latest) mainline.
> Are you aware of any bugs which document this issue?
>
> Cheers,
>
> Matt
>
> On 09/05/07, Steve G <smgustafson at gmail.com> wrote:
> > 12.3(14) will deffinietly give you echo issues.  it is the DSP firmware.
> > either get new DSP firmware from TAC, or upgrade IOS.  I am using 12.4
> (8)b
> > with no issues.
> >
> > Steve
> >
> >
> > On 4/18/07, Matthew Melbourne <matt at melbourne.org.uk> wrote:
> > >
> > > We are getting sporadic issues from users on IP phones who are hearing
> > > echo on external calls. From this document is would suggest that echo
> > > on IP Phones is caused by hybrid echo and not any local acoustic echo.
> > >
> > >
> > http://www.cisco.com/warp/public/788/voice-qos/echo_ipphone_gateway.html
> > >
> > > The MGCP-controlled voice gateways are all running 12.3(14)T7 with DSP
> > > firmware 4.4.708. In CallManager (4.1(3)SR2), echo cancellation is on
> > > for each port, with a "default" echo cancellation coverage of 64ms.
> > > Input gain and output attenuation are set to 0dB.
> > >
> > > I have seen suggestions to include the "echo-cancel erl worst-case 3"
> > > command on the voice ports. Can this help in situations of occasional
> > > echo?
> > >
> > > Are there any other parameters changes which could help? Could it be a
> > > code/DSP firmware issue? I haven't found a specific echo-related bug
> > > relating to this particular IOS and its embedded DSP firmware.
> > >
> > > On occasions, the start of the call is fine but echo comes and goes
> > > during the call.
> > >
> > > Is there a quick way of finding out which voice gateway a call is
> > > terminated on (we have 8, and all could be used in a round-robin
> > > fashion for outbound calls).
> > >
> > > Cheers,
> > >
> > > Matt
> > >
> > > --
> > > Matthew Melbourne
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
>
>
> --
> Matthew Melbourne
>
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From matt at saskin.net  Thu May 10 13:01:51 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 10 May 2007 13:01:51 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
In-Reply-To: <94a1afde0705100844y6f7557adm732eb3b3e4939b33@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>	
	<002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>	
	<C1FE15183DA37645BC0633BC604E44F00583915E@USNAEXCH.na.didata.local>	
	<88C8D019DA547C4BABECB9A806ED22ED88A04E@mail.ACT-EG.COM>	
	<94a1afde0705100526ycf3693co47630081d6c970d9@mail.gmail.com>	
	<46433B67.3090308@saskin.net>
	<94a1afde0705100844y6f7557adm732eb3b3e4939b33@mail.gmail.com>
Message-ID: <46434FFF.4060501@saskin.net>

Ahh, I misunderstood and thought you were planning on upgrading the pub 
from the existing cluster.  I can't even guess how the name change idea 
would work out.  Do you have a spare (off-network) server to test with?  
VM perhaps?

As for the multiple servers with the same hostname, that shouldn't cause 
issues provided you're relying on local name resolution instead of DNS, 
but this is just a semi-educated guess :)

-matt

Ed Leatherman wrote:
> My thought was to do an install on a new server for the 5.1 publisher 
> node w/new IP Address and restore the DMA data file onto that, thereby 
> not even touching the 4.1 publisher (other than running DMA). Would 
> also leave 1 subscriber node from the 4.1 cluster alone, and upgrade 
> the others and point them to the new 5.1 pub. Just not sure what would 
> happen having two clusters with publishers having the same host name 
> (maybe no problem?)
>
> On 5/10/07, *Matthew Saskin* <matt at saskin.net 
> <mailto:matt at saskin.net>> wrote:
>
>     How do you intend to create a publisher in the "old" cluster once you
>     split them?
>
>     In theory the subscribers will keep chugging along with some end-user
>     loss of features...well, and the lack of ability to make
>     configuration
>     changes.
>
>     -matt
>
>     Ed Leatherman wrote:
>     > Related question, and I think the answer is also probably no but
>     want
>     > to verify...
>     >
>     > When upgrading from ccm 4.1 to 5, would it be possible to change
>     the
>     > host name of the primary node in the 5.0 cluster after loading
>     the DMA
>     > data but prior to adding any subscriber nodes?
>     >
>     > Trying to come up with a way to upgrade our system to 5 but leave 2
>     > nodes in the old cluster active to operate the call center users and
>     > IPCCX - or at least determine if its feasible. so basically to split
>     > the cluster up.
>     >
>     > On 5/10/07, *Ahmed Elnagar* < aelnagar at act-eg.com
>     <mailto:aelnagar at act-eg.com>
>     > <mailto:aelnagar at act-eg.com <mailto:aelnagar at act-eg.com>>> wrote:
>     >
>     >     dont even try it you will break your CM replication and will go
>     >     through alot of very strange problems.
>     >
>     >     Thanks and Best Regards
>     >     *
>     >     Ahmed A. Elnagar
>     >     *Network Field Engineer
>     >
>     >     Advanced Computer Technology (ACT)
>     >     16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
>     >     Postal Code:12411 Cairo Egypt
>     >
>     >     *Mob** :* +2010-2833868
>     >     *Website**: * www.act-eg.com <http://www.act-eg.com>
>     <http://www.act-eg.com>
>     >     *E-mail**: * aelnagar at act-eg.com
>     <mailto:aelnagar at act-eg.com> <mailto:aelnagar at act-eg.com
>     <mailto:aelnagar at act-eg.com>>
>     >
>     >    
>     ------------------------------------------------------------------------
>     >     *From:* cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
>     >     <mailto: cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>> on behalf of Matt
>     >     Slaga (US)
>     >     *Sent:* Wed 09-May-07 11:05 PM
>     >     *To:* Ryan Ratliff; Jason Aarons (US)
>     >
>     >     *Cc:* cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>
>     <mailto:cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>>
>     >     *Subject:* Re: [cisco-voip] Changing 4.1(3) hostname article
>     on CCO?
>     >
>     >     If the customer really pushes you, create a DNS C-name record.
>     >
>     >     -----Original Message-----
>     >     From: cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>
>     >     <mailto:cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>>
>     >     [mailto: cisco-voip-bounces at puck.nether.net
>     <mailto:cisco-voip-bounces at puck.nether.net>] On Behalf Of Ryan Ratliff
>     >     Sent: Wednesday, May 09, 2007 3:58 PM
>     >     To: Jason Aarons (US)
>     >     Cc: cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>
>     <mailto:cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>>
>     >     Subject: Re: [cisco-voip] Changing 4.1(3) hostname article
>     on CCO?
>     >
>     >     No way, no how.
>     >
>     >     The only time you can change the hostname of a CM server is
>     before
>     >     you install CallManager on it.
>     >
>     >     -Ryan
>     >
>     >     On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:
>     >
>     >     I'm trying to find a reference about server hostname changes for
>     >     production CallManager 4.1, I recall it's not advised but
>     can't find
>     >     a article on cisco.com <http://cisco.com> <http://cisco.com>
>     stating this -jason
>     >
>     >
>     >
>     >
>     >
>     >     Jason Aarons
>     >
>     >     Consultant
>     >
>     >     http://www.dimensiondata.com/na
>     <http://www.dimensiondata.com/na>
>     >
>     >     904-338-3245 cell
>     >
>     >
>     >
>     >     For urgent issues notify your Project Manager, for 24x7 support
>     >     contact the Dimension Data NOC at 800-974-6584.
>     >
>     >
>     >
>     >
>     >
>     >     Disclaimer: This e-mail communication and any attachments may
>     >     contain
>     >     confidential and privileged information and is for use by the
>     >     designated addressee(s) named above only. If you are not the
>     intended
>     >     addressee, you are hereby notified that you have received this
>     >     communication in error and that any use or reproduction of this
>     >     email
>     >     or its contents is strictly prohibited and may be unlawful.
>     If you
>     >     have received this communication in error, please notify us
>     >     immediately by replying to this message and deleting it from
>     your
>     >     computer. Thank you.
>     >
>     >     _______________________________________________
>     >     cisco-voip mailing list
>     >     cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>
>     <mailto:cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>>
>     >     https://puck.nether.net/mailman/listinfo/cisco-voip
>     >
>     >     _______________________________________________
>     >     cisco-voip mailing list
>     >     cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net> <mailto:
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>>
>     >     https://puck.nether.net/mailman/listinfo/cisco-voip
>     >     -----------------------------------------
>     >     Disclaimer:
>     >
>     >     This e-mail communication and any attachments may contain
>     >     confidential and privileged information and is for use by the
>     >     designated addressee(s) named above only.  If you are not the
>     >     intended addressee, you are hereby notified that you have
>     received
>     >     this communication in error and that any use or reproduction of
>     >     this email or its contents is strictly prohibited and may be
>     >     unlawful.  If you have received this communication in error,
>     please
>     >     notify us immediately by replying to this message and
>     deleting it
>     >     from your computer. Thank you.
>     >
>     >     _______________________________________________
>     >     cisco-voip mailing list
>     >     cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>
>     <mailto:cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>>
>     >     https://puck.nether.net/mailman/listinfo/cisco-voip
>     >
>     >
>     >     _______________________________________________
>     >     cisco-voip mailing list
>     >     cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>
>     <mailto:cisco-voip at puck.nether.net
>     <mailto:cisco-voip at puck.nether.net>>
>     >     https://puck.nether.net/mailman/listinfo/cisco-voip
>     >
>     >
>     >
>     >
>     > --
>     > Ed Leatherman
>     > Senior Voice Engineer
>     > West Virginia University
>     > Telecommunications and Network Operations
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > cisco-voip mailing list
>     > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     > https://puck.nether.net/mailman/listinfo/cisco-voip
>     >
>
>
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>     <https://puck.nether.net/mailman/listinfo/cisco-voip>
>
>
>
>
> -- 
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations 



From Scott.Voll at wesd.org  Thu May 10 13:04:29 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu, 10 May 2007 10:04:29 -0700
Subject: [cisco-voip] Echo on IP Phones through PRI
In-Reply-To: <8a1f34b50705100957u4b85ddc1i1e58dd86e0b41349@mail.gmail.com>
References: <b718d1880704180550r52c715c5x80e83be222546ba2@mail.gmail.com><8a1f34b50705091546s70b357c6p702138bb016f45ed@mail.gmail.com><b718d1880705100334r797c5d9dp68e6674e21dc02b4@mail.gmail.com>
	<8a1f34b50705100957u4b85ddc1i1e58dd86e0b41349@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE74C@Apollo.wesd.org>

I'm running the CMM with 12.4.7a that has DSPware 4.4.15 does the newest
12.4 have the 4.4.18?

 

Scott

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Steve G
Sent: Thursday, May 10, 2007 9:57 AM
To: Matthew Melbourne
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Echo on IP Phones through PRI

 

CSCsd54344

 

On 5/10/07, Matthew Melbourne <matt at melbourne.org.uk> wrote: 

Thanks Steve. We have an upgrade planned for our voice gateways in an
forthcoming maintenance window, where we'll try 12.4(latest) mainline.
Are you aware of any bugs which document this issue?

Cheers,

Matt

On 09/05/07, Steve G <smgustafson at gmail.com> wrote:
> 12.3(14) will deffinietly give you echo issues.  it is the DSP
firmware.
> either get new DSP firmware from TAC, or upgrade IOS.  I am using
12.4(8)b
> with no issues.
>
> Steve
>
>
> On 4/18/07, Matthew Melbourne <matt at melbourne.org.uk> wrote:
> >
> > We are getting sporadic issues from users on IP phones who are
hearing
> > echo on external calls. From this document is would suggest that
echo 
> > on IP Phones is caused by hybrid echo and not any local acoustic
echo.
> >
> >
>
http://www.cisco.com/warp/public/788/voice-qos/echo_ipphone_gateway.html

> >
> > The MGCP-controlled voice gateways are all running 12.3(14)T7 with
DSP
> > firmware 4.4.708. In CallManager (4.1(3)SR2), echo cancellation is
on
> > for each port, with a "default" echo cancellation coverage of 64ms. 
> > Input gain and output attenuation are set to 0dB.
> >
> > I have seen suggestions to include the "echo-cancel erl worst-case
3"
> > command on the voice ports. Can this help in situations of
occasional 
> > echo?
> >
> > Are there any other parameters changes which could help? Could it be
a
> > code/DSP firmware issue? I haven't found a specific echo-related bug
> > relating to this particular IOS and its embedded DSP firmware. 
> >
> > On occasions, the start of the call is fine but echo comes and goes
> > during the call.
> >
> > Is there a quick way of finding out which voice gateway a call is
> > terminated on (we have 8, and all could be used in a round-robin 
> > fashion for outbound calls).
> >
> > Cheers,
> >
> > Matt
> >
> > --
> > Matthew Melbourne
> > _______________________________________________ 
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip 
> >
>
>


--
Matthew Melbourne

 

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From jsteinberg at gmail.com  Thu May 10 14:26:30 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 10 May 2007 14:26:30 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
In-Reply-To: <46434FFF.4060501@saskin.net>
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>
	<002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00583915E@USNAEXCH.na.didata.local>
	<88C8D019DA547C4BABECB9A806ED22ED88A04E@mail.ACT-EG.COM>
	<94a1afde0705100526ycf3693co47630081d6c970d9@mail.gmail.com>
	<46433B67.3090308@saskin.net>
	<94a1afde0705100844y6f7557adm732eb3b3e4939b33@mail.gmail.com>
	<46434FFF.4060501@saskin.net>
Message-ID: <f1b65f880705101126ue2e5f80y7e1cf61404e2fae1@mail.gmail.com>

I haven't tested this but I think it might work.

The link below indicates that changing the hostname as part of the DMA
upgrade process from CM 4.x to CM 5.x is supported.  If you don't want
to read the link, here is the important sentence.

"This feature allows you to import your 4.x data to a 5.1(1) server
without having to preserve the IP Address or Host Name. The IP Address
and/or Host name of the 5.1(1) server can differ from the 4.x
servername."

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_upgrade_guides09186a0080792e84.html#wp46919

You would then use TFTP settings to control which callmanager phones
registered with.

I'd open a TAC case to have them confirm this but this seems like a sane idea.

Justin

On 5/10/07, Matthew Saskin <matt at saskin.net> wrote:
> Ahh, I misunderstood and thought you were planning on upgrading the pub
> from the existing cluster.  I can't even guess how the name change idea
> would work out.  Do you have a spare (off-network) server to test with?
> VM perhaps?
>
> As for the multiple servers with the same hostname, that shouldn't cause
> issues provided you're relying on local name resolution instead of DNS,
> but this is just a semi-educated guess :)
>
> -matt
>
> Ed Leatherman wrote:
> > My thought was to do an install on a new server for the 5.1 publisher
> > node w/new IP Address and restore the DMA data file onto that, thereby
> > not even touching the 4.1 publisher (other than running DMA). Would
> > also leave 1 subscriber node from the 4.1 cluster alone, and upgrade
> > the others and point them to the new 5.1 pub. Just not sure what would
> > happen having two clusters with publishers having the same host name
> > (maybe no problem?)
> >
> > On 5/10/07, *Matthew Saskin* <matt at saskin.net
> > <mailto:matt at saskin.net>> wrote:
> >
> >     How do you intend to create a publisher in the "old" cluster once you
> >     split them?
> >
> >     In theory the subscribers will keep chugging along with some end-user
> >     loss of features...well, and the lack of ability to make
> >     configuration
> >     changes.
> >
> >     -matt
> >
> >     Ed Leatherman wrote:
> >     > Related question, and I think the answer is also probably no but
> >     want
> >     > to verify...
> >     >
> >     > When upgrading from ccm 4.1 to 5, would it be possible to change
> >     the
> >     > host name of the primary node in the 5.0 cluster after loading
> >     the DMA
> >     > data but prior to adding any subscriber nodes?
> >     >
> >     > Trying to come up with a way to upgrade our system to 5 but leave 2
> >     > nodes in the old cluster active to operate the call center users and
> >     > IPCCX - or at least determine if its feasible. so basically to split
> >     > the cluster up.
> >     >
> >     > On 5/10/07, *Ahmed Elnagar* < aelnagar at act-eg.com
> >     <mailto:aelnagar at act-eg.com>
> >     > <mailto:aelnagar at act-eg.com <mailto:aelnagar at act-eg.com>>> wrote:
> >     >
> >     >     dont even try it you will break your CM replication and will go
> >     >     through alot of very strange problems.
> >     >
> >     >     Thanks and Best Regards
> >     >     *
> >     >     Ahmed A. Elnagar
> >     >     *Network Field Engineer
> >     >
> >     >     Advanced Computer Technology (ACT)
> >     >     16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> >     >     Postal Code:12411 Cairo Egypt
> >     >
> >     >     *Mob** :* +2010-2833868
> >     >     *Website**: * www.act-eg.com <http://www.act-eg.com>
> >     <http://www.act-eg.com>
> >     >     *E-mail**: * aelnagar at act-eg.com
> >     <mailto:aelnagar at act-eg.com> <mailto:aelnagar at act-eg.com
> >     <mailto:aelnagar at act-eg.com>>
> >     >
> >     >
> >     ------------------------------------------------------------------------
> >     >     *From:* cisco-voip-bounces at puck.nether.net
> >     <mailto:cisco-voip-bounces at puck.nether.net>
> >     >     <mailto: cisco-voip-bounces at puck.nether.net
> >     <mailto:cisco-voip-bounces at puck.nether.net>> on behalf of Matt
> >     >     Slaga (US)
> >     >     *Sent:* Wed 09-May-07 11:05 PM
> >     >     *To:* Ryan Ratliff; Jason Aarons (US)
> >     >
> >     >     *Cc:* cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>
> >     <mailto:cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>>
> >     >     *Subject:* Re: [cisco-voip] Changing 4.1(3) hostname article
> >     on CCO?
> >     >
> >     >     If the customer really pushes you, create a DNS C-name record.
> >     >
> >     >     -----Original Message-----
> >     >     From: cisco-voip-bounces at puck.nether.net
> >     <mailto:cisco-voip-bounces at puck.nether.net>
> >     >     <mailto:cisco-voip-bounces at puck.nether.net
> >     <mailto:cisco-voip-bounces at puck.nether.net>>
> >     >     [mailto: cisco-voip-bounces at puck.nether.net
> >     <mailto:cisco-voip-bounces at puck.nether.net>] On Behalf Of Ryan Ratliff
> >     >     Sent: Wednesday, May 09, 2007 3:58 PM
> >     >     To: Jason Aarons (US)
> >     >     Cc: cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>
> >     <mailto:cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>>
> >     >     Subject: Re: [cisco-voip] Changing 4.1(3) hostname article
> >     on CCO?
> >     >
> >     >     No way, no how.
> >     >
> >     >     The only time you can change the hostname of a CM server is
> >     before
> >     >     you install CallManager on it.
> >     >
> >     >     -Ryan
> >     >
> >     >     On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:
> >     >
> >     >     I'm trying to find a reference about server hostname changes for
> >     >     production CallManager 4.1, I recall it's not advised but
> >     can't find
> >     >     a article on cisco.com <http://cisco.com> <http://cisco.com>
> >     stating this -jason
> >     >
> >     >
> >     >
> >     >
> >     >
> >     >     Jason Aarons
> >     >
> >     >     Consultant
> >     >
> >     >     http://www.dimensiondata.com/na
> >     <http://www.dimensiondata.com/na>
> >     >
> >     >     904-338-3245 cell
> >     >
> >     >
> >     >
> >     >     For urgent issues notify your Project Manager, for 24x7 support
> >     >     contact the Dimension Data NOC at 800-974-6584.
> >     >
> >     >
> >     >
> >     >
> >     >
> >     >     Disclaimer: This e-mail communication and any attachments may
> >     >     contain
> >     >     confidential and privileged information and is for use by the
> >     >     designated addressee(s) named above only. If you are not the
> >     intended
> >     >     addressee, you are hereby notified that you have received this
> >     >     communication in error and that any use or reproduction of this
> >     >     email
> >     >     or its contents is strictly prohibited and may be unlawful.
> >     If you
> >     >     have received this communication in error, please notify us
> >     >     immediately by replying to this message and deleting it from
> >     your
> >     >     computer. Thank you.
> >     >
> >     >     _______________________________________________
> >     >     cisco-voip mailing list
> >     >     cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>
> >     <mailto:cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>>
> >     >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >     >
> >     >     _______________________________________________
> >     >     cisco-voip mailing list
> >     >     cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net> <mailto:
> >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>>
> >     >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >     >     -----------------------------------------
> >     >     Disclaimer:
> >     >
> >     >     This e-mail communication and any attachments may contain
> >     >     confidential and privileged information and is for use by the
> >     >     designated addressee(s) named above only.  If you are not the
> >     >     intended addressee, you are hereby notified that you have
> >     received
> >     >     this communication in error and that any use or reproduction of
> >     >     this email or its contents is strictly prohibited and may be
> >     >     unlawful.  If you have received this communication in error,
> >     please
> >     >     notify us immediately by replying to this message and
> >     deleting it
> >     >     from your computer. Thank you.
> >     >
> >     >     _______________________________________________
> >     >     cisco-voip mailing list
> >     >     cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>
> >     <mailto:cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>>
> >     >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >     >
> >     >
> >     >     _______________________________________________
> >     >     cisco-voip mailing list
> >     >     cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>
> >     <mailto:cisco-voip at puck.nether.net
> >     <mailto:cisco-voip at puck.nether.net>>
> >     >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >     >
> >     >
> >     >
> >     >
> >     > --
> >     > Ed Leatherman
> >     > Senior Voice Engineer
> >     > West Virginia University
> >     > Telecommunications and Network Operations
> >     >
> >     ------------------------------------------------------------------------
> >     >
> >     > _______________________________________________
> >     > cisco-voip mailing list
> >     > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     > https://puck.nether.net/mailman/listinfo/cisco-voip
> >     >
> >
> >
> >     _______________________________________________
> >     cisco-voip mailing list
> >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     https://puck.nether.net/mailman/listinfo/cisco-voip
> >     <https://puck.nether.net/mailman/listinfo/cisco-voip>
> >
> >
> >
> >
> > --
> > Ed Leatherman
> > Senior Voice Engineer
> > West Virginia University
> > Telecommunications and Network Operations
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From ealeatherman at gmail.com  Thu May 10 15:34:40 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 10 May 2007 15:34:40 -0400
Subject: [cisco-voip] Changing 4.1(3) hostname article on CCO?
In-Reply-To: <f1b65f880705101126ue2e5f80y7e1cf61404e2fae1@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F0058390F0@USNAEXCH.na.didata.local>
	<002BA443-BCB8-40FF-8E15-ABC0D1ED931E@cisco.com>
	<C1FE15183DA37645BC0633BC604E44F00583915E@USNAEXCH.na.didata.local>
	<88C8D019DA547C4BABECB9A806ED22ED88A04E@mail.ACT-EG.COM>
	<94a1afde0705100526ycf3693co47630081d6c970d9@mail.gmail.com>
	<46433B67.3090308@saskin.net>
	<94a1afde0705100844y6f7557adm732eb3b3e4939b33@mail.gmail.com>
	<46434FFF.4060501@saskin.net>
	<f1b65f880705101126ue2e5f80y7e1cf61404e2fae1@mail.gmail.com>
Message-ID: <94a1afde0705101234j4de53322j3f2899e4a01fb5b4@mail.gmail.com>

Awesome thanks Justin, I never saw that. We'll probably do a test
run-through in the lab just to be on the safe side.

On 5/10/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
>
> I haven't tested this but I think it might work.
>
> The link below indicates that changing the hostname as part of the DMA
> upgrade process from CM 4.x to CM 5.x is supported.  If you don't want
> to read the link, here is the important sentence.
>
> "This feature allows you to import your 4.x data to a 5.1(1) server
> without having to preserve the IP Address or Host Name. The IP Address
> and/or Host name of the 5.1(1) server can differ from the 4.x
> servername."
>
>
> http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_upgrade_guides09186a0080792e84.html#wp46919
>
> You would then use TFTP settings to control which callmanager phones
> registered with.
>
> I'd open a TAC case to have them confirm this but this seems like a sane
> idea.
>
> Justin
>
> On 5/10/07, Matthew Saskin <matt at saskin.net> wrote:
> > Ahh, I misunderstood and thought you were planning on upgrading the pub
> > from the existing cluster.  I can't even guess how the name change idea
> > would work out.  Do you have a spare (off-network) server to test with?
> > VM perhaps?
> >
> > As for the multiple servers with the same hostname, that shouldn't cause
> > issues provided you're relying on local name resolution instead of DNS,
> > but this is just a semi-educated guess :)
> >
> > -matt
> >
> > Ed Leatherman wrote:
> > > My thought was to do an install on a new server for the 5.1 publisher
> > > node w/new IP Address and restore the DMA data file onto that, thereby
> > > not even touching the 4.1 publisher (other than running DMA). Would
> > > also leave 1 subscriber node from the 4.1 cluster alone, and upgrade
> > > the others and point them to the new 5.1 pub. Just not sure what would
> > > happen having two clusters with publishers having the same host name
> > > (maybe no problem?)
> > >
> > > On 5/10/07, *Matthew Saskin* <matt at saskin.net
> > > <mailto:matt at saskin.net>> wrote:
> > >
> > >     How do you intend to create a publisher in the "old" cluster once
> you
> > >     split them?
> > >
> > >     In theory the subscribers will keep chugging along with some
> end-user
> > >     loss of features...well, and the lack of ability to make
> > >     configuration
> > >     changes.
> > >
> > >     -matt
> > >
> > >     Ed Leatherman wrote:
> > >     > Related question, and I think the answer is also probably no but
> > >     want
> > >     > to verify...
> > >     >
> > >     > When upgrading from ccm 4.1 to 5, would it be possible to change
> > >     the
> > >     > host name of the primary node in the 5.0 cluster after loading
> > >     the DMA
> > >     > data but prior to adding any subscriber nodes?
> > >     >
> > >     > Trying to come up with a way to upgrade our system to 5 but
> leave 2
> > >     > nodes in the old cluster active to operate the call center users
> and
> > >     > IPCCX - or at least determine if its feasible. so basically to
> split
> > >     > the cluster up.
> > >     >
> > >     > On 5/10/07, *Ahmed Elnagar* < aelnagar at act-eg.com
> > >     <mailto:aelnagar at act-eg.com>
> > >     > <mailto:aelnagar at act-eg.com <mailto:aelnagar at act-eg.com>>>
> wrote:
> > >     >
> > >     >     dont even try it you will break your CM replication and will
> go
> > >     >     through alot of very strange problems.
> > >     >
> > >     >     Thanks and Best Regards
> > >     >     *
> > >     >     Ahmed A. Elnagar
> > >     >     *Network Field Engineer
> > >     >
> > >     >     Advanced Computer Technology (ACT)
> > >     >     16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> > >     >     Postal Code:12411 Cairo Egypt
> > >     >
> > >     >     *Mob** :* +2010-2833868
> > >     >     *Website**: * www.act-eg.com <http://www.act-eg.com>
> > >     <http://www.act-eg.com>
> > >     >     *E-mail**: * aelnagar at act-eg.com
> > >     <mailto:aelnagar at act-eg.com> <mailto:aelnagar at act-eg.com
> > >     <mailto:aelnagar at act-eg.com>>
> > >     >
> > >     >
> > >
> ------------------------------------------------------------------------
> > >     >     *From:* cisco-voip-bounces at puck.nether.net
> > >     <mailto:cisco-voip-bounces at puck.nether.net>
> > >     >     <mailto: cisco-voip-bounces at puck.nether.net
> > >     <mailto:cisco-voip-bounces at puck.nether.net>> on behalf of Matt
> > >     >     Slaga (US)
> > >     >     *Sent:* Wed 09-May-07 11:05 PM
> > >     >     *To:* Ryan Ratliff; Jason Aarons (US)
> > >     >
> > >     >     *Cc:* cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>
> > >     <mailto:cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>>
> > >     >     *Subject:* Re: [cisco-voip] Changing 4.1(3) hostname article
> > >     on CCO?
> > >     >
> > >     >     If the customer really pushes you, create a DNS C-name
> record.
> > >     >
> > >     >     -----Original Message-----
> > >     >     From: cisco-voip-bounces at puck.nether.net
> > >     <mailto:cisco-voip-bounces at puck.nether.net>
> > >     >     <mailto:cisco-voip-bounces at puck.nether.net
> > >     <mailto:cisco-voip-bounces at puck.nether.net>>
> > >     >     [mailto: cisco-voip-bounces at puck.nether.net
> > >     <mailto:cisco-voip-bounces at puck.nether.net>] On Behalf Of Ryan
> Ratliff
> > >     >     Sent: Wednesday, May 09, 2007 3:58 PM
> > >     >     To: Jason Aarons (US)
> > >     >     Cc: cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>
> > >     <mailto:cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>>
> > >     >     Subject: Re: [cisco-voip] Changing 4.1(3) hostname article
> > >     on CCO?
> > >     >
> > >     >     No way, no how.
> > >     >
> > >     >     The only time you can change the hostname of a CM server is
> > >     before
> > >     >     you install CallManager on it.
> > >     >
> > >     >     -Ryan
> > >     >
> > >     >     On May 9, 2007, at 3:51 PM, Jason Aarons ((US)) wrote:
> > >     >
> > >     >     I'm trying to find a reference about server hostname changes
> for
> > >     >     production CallManager 4.1, I recall it's not advised but
> > >     can't find
> > >     >     a article on cisco.com <http://cisco.com> <http://cisco.com>
> > >     stating this -jason
> > >     >
> > >     >
> > >     >
> > >     >
> > >     >
> > >     >     Jason Aarons
> > >     >
> > >     >     Consultant
> > >     >
> > >     >     http://www.dimensiondata.com/na
> > >     <http://www.dimensiondata.com/na>
> > >     >
> > >     >     904-338-3245 cell
> > >     >
> > >     >
> > >     >
> > >     >     For urgent issues notify your Project Manager, for 24x7
> support
> > >     >     contact the Dimension Data NOC at 800-974-6584.
> > >     >
> > >     >
> > >     >
> > >     >
> > >     >
> > >     >     Disclaimer: This e-mail communication and any attachments
> may
> > >     >     contain
> > >     >     confidential and privileged information and is for use by
> the
> > >     >     designated addressee(s) named above only. If you are not the
> > >     intended
> > >     >     addressee, you are hereby notified that you have received
> this
> > >     >     communication in error and that any use or reproduction of
> this
> > >     >     email
> > >     >     or its contents is strictly prohibited and may be unlawful.
> > >     If you
> > >     >     have received this communication in error, please notify us
> > >     >     immediately by replying to this message and deleting it from
> > >     your
> > >     >     computer. Thank you.
> > >     >
> > >     >     _______________________________________________
> > >     >     cisco-voip mailing list
> > >     >     cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>
> > >     <mailto:cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>>
> > >     >     https://puck.nether.net/mailman/listinfo/cisco-voip
> > >     >
> > >     >     _______________________________________________
> > >     >     cisco-voip mailing list
> > >     >     cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net> <mailto:
> > >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>>
> > >     >     https://puck.nether.net/mailman/listinfo/cisco-voip
> > >     >     -----------------------------------------
> > >     >     Disclaimer:
> > >     >
> > >     >     This e-mail communication and any attachments may contain
> > >     >     confidential and privileged information and is for use by
> the
> > >     >     designated addressee(s) named above only.  If you are not
> the
> > >     >     intended addressee, you are hereby notified that you have
> > >     received
> > >     >     this communication in error and that any use or reproduction
> of
> > >     >     this email or its contents is strictly prohibited and may be
> > >     >     unlawful.  If you have received this communication in error,
> > >     please
> > >     >     notify us immediately by replying to this message and
> > >     deleting it
> > >     >     from your computer. Thank you.
> > >     >
> > >     >     _______________________________________________
> > >     >     cisco-voip mailing list
> > >     >     cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>
> > >     <mailto:cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>>
> > >     >     https://puck.nether.net/mailman/listinfo/cisco-voip
> > >     >
> > >     >
> > >     >     _______________________________________________
> > >     >     cisco-voip mailing list
> > >     >     cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>
> > >     <mailto:cisco-voip at puck.nether.net
> > >     <mailto:cisco-voip at puck.nether.net>>
> > >     >     https://puck.nether.net/mailman/listinfo/cisco-voip
> > >     >
> > >     >
> > >     >
> > >     >
> > >     > --
> > >     > Ed Leatherman
> > >     > Senior Voice Engineer
> > >     > West Virginia University
> > >     > Telecommunications and Network Operations
> > >     >
> > >
> ------------------------------------------------------------------------
> > >     >
> > >     > _______________________________________________
> > >     > cisco-voip mailing list
> > >     > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> > >     > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >     >
> > >
> > >
> > >     _______________________________________________
> > >     cisco-voip mailing list
> > >     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> > >     https://puck.nether.net/mailman/listinfo/cisco-voip
> > >     <https://puck.nether.net/mailman/listinfo/cisco-voip>
> > >
> > >
> > >
> > >
> > > --
> > > Ed Leatherman
> > > Senior Voice Engineer
> > > West Virginia University
> > > Telecommunications and Network Operations
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Thu May 10 15:46:53 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 10 May 2007 15:46:53 -0400
Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
In-Reply-To: <5d093f9a0705072154k5cfcf6cdg17b51b5f757b5625@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F0057BE625@USNAEXCH.na.didata.local>
	<OFE4AEA40C.62F5FCE5-ON852572D4.005A8008-852572D4.005A9AAD@bftg.com>
	<5d093f9a0705072154k5cfcf6cdg17b51b5f757b5625@mail.gmail.com>
Message-ID: <94a1afde0705101246ge46d329ye9b550b377ec0cb4@mail.gmail.com>

We did a test upgrade yesterday:

Restored a BARS backup of our production pub to our test server, ran DMA (so
had about 3300 unregistered devices in the database at the time DMA was
run), then installed CCM 5 and loaded that DMA onto it. CallManager reports
approx 14000 DLU's. At 4 DLU's per 7960 thats in the right ballpark.

I'll see about doing a new DMA and add some "unknown" devices.

On another note... lesson learned: make sure you specify how you want your
DLU's if you order new ones. I just ordered 300 7960 phones + 1200 DLU's and
the DLU's arrived in packets of 10... yes I now have 15 pounds of license
units. I guess I didn't specify to the vendor that I wanted the license
packs in quantities that made sense.

On 5/8/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> You mean 'Unknown'... that is probable....
>
> To be honest, I would suspect they would not carry the licenses over for
> phones that are in an unknown state... but unregistered doesn't make
> sense... what if a phone is at a remote site and the wan is down while you
> are doing the upgrade, do you just get hosed?
>
>
>
> Jonathan
>
> On 5/7/07, CarlosOrtiz at bayviewfinancial.com <CarlosOrtiz at bayviewfinancial.com>
> wrote:
> >
> >
> > Could it be that phones that were at least registered before and now
> > offline are ok, but phones that are "not found" meaning never registered to
> > the system are not?
> >
> > Carlos
> >
> >
> >  *"Matt Slaga \(US\)" <Matt.Slaga at us.didata.com>*
> > Sent by: cisco-voip-bounces at puck.nether.net
> >
> > 05/07/2007 12:12 PM
> >   To
> > "Jonathan Charles" <jonvoip at gmail.com>  cc
> > cbomba at s4nets.com, cisco-voip at puck.nether.net  Subject
> > Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
> >
> >
> >
> >
> >
> >
> > Interesting.  Cisco was hit hard on this at the UC 6.0 Partner Beta
> > training, and the IPC BU repeated many times that the phones had to be
> > registered to be counted in the license file.
> >
> >
> >
> > *From:* Jonathan Charles [mailto:jonvoip at gmail.com] *
> > Sent:* Monday, May 07, 2007 11:59 AM*
> > To:* Matt Slaga (US)*
> > Cc:* cbomba at s4nets.com; cisco-voip at puck.nether.net*
> > Subject:* Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
> >
> > This is NOT true.
> >
> > I just did an upgrade (3.3 to 5.1)... and we built a new server with
> > CCM3.3, restored the backup (in a lab, no phones present), upgraded to
> > 4.1, ran DMA and upgraded to 5.1 and restored the DMA data.
> >
> > The license was for all of the phones configured (zero phones were
> > registered at the time DMA was run), and the customer had a full upgrade
> > license for all configured phones.
> >
> >
> >
> > Jonathan
> > On 5/7/07, *Matt Slaga (US)* <*Matt.Slaga at us.didata.com*<Matt.Slaga at us.didata.com>> wrote:
> > Actually, this is not the case.  It will only generate DLUs for phones
> > that are registered with the CallManager server.  It will skip any
> > device that is not registered and not count it in the DLU list.
> >
> >
> >
> > -----Original Message-----
> > From: *cisco-voip-bounces at puck.nether.net *<cisco-voip-bounces at puck.nether.net>
> > [mailto:*cisco-voip-bounces at puck.nether.net *<cisco-voip-bounces at puck.nether.net>]
> > On Behalf Of Christopher M.
> > Bomba
> > Sent: Wednesday, May 02, 2007 9:05 AM
> > To: *cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net>
> > Subject: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
> >
> > Who said you had to buy the phones.  You can just put phones in with
> > bogus MAC addresses and it will generate DLUs for those.  I thought they
> > had to be registered but turns out that they just have to be present in
> > the system.  You can create IP communicators or any other phone to boost
> > the DLUs.  That is a big problem since anybody can do this.
> >
> > Chris
> >
> > --original message--
> >
> > thanks Ed for your reply, yes you are right I contacted Partner help and
> >
> > they told me that during the upgrade a file will be generated that when
> > given to cisco they will give us back the DLUs equivalent to what
> > configured on the system. I think this is an illegal workaround for the
> > lic. issue as by that someone could buy phones without lic. and then
> > setup them with CM 4.1 and then upgrade and he will be having a DLUs for
> > free. I think Cisco has to do something to solve this.
> >
> > Thanks and Best Regards
> >
> > Ahmed A. Elnagar
> > Network Field Engineer
> >
> > Advanced Computer Technology (ACT)
> > 16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt
> > Postal Code:12411 Cairo Egypt
> >
> > Mob: +2010-2833868
> > Website: *www.act-eg.com* <http://www.act-eg.com/>
> > E-mail: *aelnagar at act-eg.com* <aelnagar at act-eg.com>
> >
> > ________________________________
> >
> > From: Ed Leatherman [mailto:* ealeatherman at gmail.com*<ealeatherman at gmail.com>
> > ]
> > Sent: Wed 02-May-07 3:16 PM
> > To: Ahmed Elnagar
> > Cc: *cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net>
> > Subject: Re: [cisco-voip] Callmanager upgrade 4.1 >>> 5.1
> >
> >
> > hi Ahmed,
> >
> > When you upgrade, callmanager will generate a temporary license for all
> > the devices you have configured from 4.1. If I understand correctly, you
> > just have to send that off to cisco's licensing group and have it
> > converted to a real 5.2 license with the appropriate number of DLUs. If
> > its anything like Unity licensing, the turn-around is pretty quick. I'm
> > assuming you need a SASU agreement for callmanager for them to do it.
> >
> > We are getting ready to do the same process here this summer with our
> > 4.1 system.
> >
> > Hope that helps.
> >
> > Ed
> >
> >
> >
> > On 5/2/07, Ahmed Elnagar <*aelnagar at act-eg.com * <aelnagar at act-eg.com>>
> > wrote:
> >
> >        Hello all;
> >
> >        A customer wants to upgrade from CM 4.1 to 5.1, what I am
> > worried about is the licsences issue. How will I get the DLUs for
> > callmanager 5.1 we have a lic. with each phone but how we will get the
> > DLUs??
> >
> >
> >        Thanks and Best Regards
> >
> >        Ahmed A. Elnagar
> >        Network Field Engineer
> >
> >        Advanced Computer Technology (ACT)
> >        16 Fawzy Ramah St.Off Shehab St.Mohandessin , Giza, Egypt
> >        Postal Code:12411 Cairo Egypt
> >
> >        Mob : +2010-2833868
> >        Website: *www.act-eg.com* <http://www.act-eg.com/> <*
> > http://www.act-eg.com/* <http://www.act-eg.com/> >
> >        E-mail: *aelnagar at act-eg.com* <aelnagar at act-eg.com>
> >
> >        _______________________________________________
> >        cisco-voip mailing list
> >        * cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net>
> >        *https://puck..nether.net/mailman/listinfo/cisco-voip *<https://puck.nether.net/mailman/listinfo/cisco-voip>
> >
> >
> >
> >
> >
> >
> > --
> > Ed Leatherman
> > Senior Voice Engineer
> > West Virginia University
> > Telecommunications and Network Operations
> > -------------- next part --------------
> > An HTML attachment was scrubbed...
> > URL:*
> > **https://puck.nether.net/pipermail/cisco-voip/attachments/20070502/c662e8
> > *<https://puck.nether.net/pipermail/cisco-voip/attachments/20070502/c662e8>
> > 78/attachment.html
> >
> > ------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list*
> > **cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net> *
> > **https://puck..nether.net/mailman/listinfo/cisco-voip *<https://puck.nether.net/mailman/listinfo/cisco-voip>
> >
> >
> > End of cisco-voip Digest, Vol 51, Issue 4
> > *****************************************
> >
> > _______________________________________________
> > cisco-voip mailing list*
> > **cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net> *
> > **https://puck..nether.net/mailman/listinfo/cisco-voip *<https://puck.nether.net/mailman/listinfo/cisco-voip>
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain
> > confidential and privileged information and is for use by the
> > designated addressee(s) named above only.  If you are not the
> > intended addressee, you are hereby notified that you have received
> > this communication in error and that any use or reproduction of
> > this email or its contents is strictly prohibited and may be
> > unlawful.  If you have received this communication in error, please
> > notify us immediately by replying to this message and deleting it
> > from your computer. Thank you.
> >
> > _______________________________________________
> > cisco-voip mailing list*
> > **cisco-voip at puck.nether.net* <cisco-voip at puck.nether.net> *
> > **https://puck..nether.net/mailman/listinfo/cisco-voip *<https://puck.nether.net/mailman/listinfo/cisco-voip>
> >
> >
> > ------------------------------
> >
> > *Disclaimer: This e-mail communication and any attachments may contain
> > confidential and privileged information and is for use by the designated
> > addressee(s) named above only. If you are not the intended addressee, you
> > are hereby notified that you have received this communication in error and
> > that any use or reproduction of this email or its contents is strictly
> > prohibited and may be unlawful. If you have received this communication in
> > error, please notify us immediately by replying to this message and deleting
> > it from your computer. Thank you. *
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Thu May 10 16:54:15 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 10 May 2007 16:54:15 -0400
Subject: [cisco-voip] which ios to use for 3845
References: <014901c79178$b1d6dce0$47196883@cfs.uoguelph.ca>
	<6A0744639B86DD459433108DBF101D26015D07F3@MAIL01.sentinel.com>
Message-ID: <034001c79345$5f0b48b0$47196883@cfs.uoguelph.ca>

Well, considering we've got 7911s we need SRST for, looks like it's 12.4(9)T or better for me!

Talk about bleeding edge...at least it looks like 12.4(11)T2 is LD and not ED.

Thanks for all your suggestions.



--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Carter, Bill 
  To: Lelio Fulgenzi 
  Sent: Tuesday, May 08, 2007 12:45 PM
  Subject: RE: [cisco-voip] which ios to use for 3845


  12.4(6)T7 or 12.4(7)b

  http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/system/gbtst4x/4_4/rng44nip.htm#wp1261437

  Cisco "Golden Bridge" documents
  http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/system/gbtst4x/4_4/index.htm





------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Tuesday, May 08, 2007 8:57 AM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] which ios to use for 3845


  my 3845 was just dropped off. 

  what IOS are people recommending for this thing? primarily used as a voice gateway (2 T1/PRIs using MGCP) and for SRST. I don't think I will be doing much transcoding, (other than from PSTN to voip) or conferencing.


  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  ...there's no such thing as a bad timbit...
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From jason.aarons at us.didata.com  Thu May 10 21:13:19 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 10 May 2007 21:13:19 -0400
Subject: [cisco-voip] Difference between these two
Message-ID: <C1FE15183DA37645BC0633BC604E44F00589E156@USNAEXCH.na.didata.local>

Can anyone clarify the differences between the two sets of commands
below?

 

Newer config 12.4.11.T2

voice service voip

   h323

       call preserve

 

 

Older running-config 12.4.4XC2

voice class h323 1

   h225 timeout tcp establish 0

      call preserve

 

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
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From jefflin.choi at gmail.com  Thu May 10 23:17:44 2007
From: jefflin.choi at gmail.com (Jefflin Choi)
Date: Fri, 11 May 2007 11:17:44 +0800
Subject: [cisco-voip] Cisco Desktop Supervisor or Cisco Agent Desktop
	through VPN
Message-ID: <5c72afa80705102017s7811973bmdd26e168a28fe352@mail.gmail.com>

Hi All,

>From link:
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a00801cf916.shtml
*"Note: *Officially, Cisco does not support the installation of Cisco
Desktop Supervisor or Cisco Agent Desktop through any kind of firewalls."

Is this same information as compared with vpn dialup client setup?

We had a remote location connecting via vpn. Installation went through
unlike on the link. Call also goes through with a ring however no audio is
heard. Using ip communicator, it works fine without any problems.

Is the port being used by the cisco agent desktop same as the ip
communicator? Or is it not a supported scenario?

Thanks,
Jeff
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From olivier.lefevre at kpmg.lu  Fri May 11 05:45:36 2007
From: olivier.lefevre at kpmg.lu (Lefevre, Olivier)
Date: Fri, 11 May 2007 11:45:36 +0200
Subject: [cisco-voip] =?iso-8859-1?q?RE=A0=3A__Difference_between_these_tw?=
	=?iso-8859-1?q?o?=
References: <C1FE15183DA37645BC0633BC604E44F00589E156@USNAEXCH.na.didata.local>
Message-ID: <1D39D54FDA99E84D9016063F57CBE60B028339DC@LULUAEXC34.lu.kworld.kpmg.com>

Hi,
 
The difference is in the newer is the service apply on all h323 dial pear. and the one is only apply on the dial pear where the class is apply.
 
Olivier

________________________________

De: cisco-voip-bounces at puck.nether.net de la part de Jason Aarons (US)
Date: ven. 11/05/2007 3:13
?: cisco-voip at puck.nether.net
Objet : [cisco-voip] Difference between these two



Can anyone clarify the differences between the two sets of commands below?

 

Newer config 12.4.11.T2

voice service voip

   h323

       call preserve

 

 

Older running-config 12.4.4XC2

voice class h323 1

   h225 timeout tcp establish 0

      call preserve

 

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact the Dimension Data NOC at 800-974-6584.

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 

The information in this e-mail is confidential and may be legally privileged. It is intended
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the presence of computer viruses. However, the ultimate responsibility for virus checking 
lies with the recipient of this e-mail.



From paukstis at gmail.com  Fri May 11 08:06:08 2007
From: paukstis at gmail.com (Arunas)
Date: Fri, 11 May 2007 15:06:08 +0300
Subject: [cisco-voip] IPCC Express service activation
Message-ID: <84602f830705110506v5541d2c8se2925c236bbdf954@mail.gmail.com>

Hi All,

We have IPCC express 4.0.5a running on separate server and CRS Editor
service in shutdown state. Could anybody give me an advice to get this
service up and running.  I can't find any relevant information on cisco.com.

Many thanks.
Arunas
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From paukstis at gmail.com  Fri May 11 09:21:54 2007
From: paukstis at gmail.com (Arunas)
Date: Fri, 11 May 2007 16:21:54 +0300
Subject: [cisco-voip] IPCC Express service activation
In-Reply-To: <BD89624EE83BD349BD31A08B453DE24401650E66@uogexb201.cfs.uoguelph.ca>
References: <84602f830705110506v5541d2c8se2925c236bbdf954@mail.gmail.com>
	<BD89624EE83BD349BD31A08B453DE24401650E66@uogexb201.cfs.uoguelph.ca>
Message-ID: <84602f830705110621r757dc114w2a0c8befa5c5802@mail.gmail.com>

CRS Editor service not starting when I've started CRS Editor. Problem is
that if CRS editor service in shutdown state I can't debug my script. When I
got this issue (can't debug script) I've started to look at services an
found this problem.
Any ideas ?

Thanks,
Arunas

On 5/11/07, Rasim Duric <rduric at uoguelph.ca> wrote:
>
>  Starting/Closing of the CRS editor will start/shutdown CRS Editor
> service.
>
>
>
> Rasim Duric
> Network Analyst (CCS)
> University of Guelph
> Guelph, N1G 2W1, ON
> 519-824-4120x53146
> rduric at uoguelph.ca
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Arunas
> *Sent:* Friday, May 11, 2007 8:06 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] IPCC Express service activation
>
>
>
> Hi All,
>
> We have IPCC express 4.0.5a running on separate server and CRS Editor
> service in shutdown state. Could anybody give me an advice to get this
> service up and running.  I can't find any relevant information on
> cisco.com.
>
> Many thanks.
> Arunas
>
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From lelio at uoguelph.ca  Fri May 11 09:52:59 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 11 May 2007 09:52:59 -0400
Subject: [cisco-voip] which ios to use for 3845
Message-ID: <004201c793d3$af522fd0$47196883@cfs.uoguelph.ca>

You really gotta like it when documentation conflicts.

And from srst 3.3 "supported firmware, platforms, etc", we get a table that says 7911g is supported.

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2169/prod_installation_guide09186a00805fdf87.html

>From srst 4.1 datasheet we get a table that says 7911g is not supported until 4.0

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2169/products_data_sheet0900aecd805e1e79.html

"round and round she goes, where she stops, nobody knows."


      Cisco Unified IP Phone Support
     
      Cisco Unified IP Phone or Cisco Endpoint
     File Name of Compatible Cisco Unified CallManager
      Firmware for SCCP 
     File Name of Compatible
      Cisco Unified CallManager 
      Firmware for SIP 
     
      Cisco ATA-186 

      Cisco ATA-188
     ATA030100SCCP040211A.zup
     -
     
      Cisco IP Communicator 2.0*
     NA
     NA
     
      Cisco Unified IP Phone 7902G
     CP7902040000SCCP040701A.sbin
     -
     
      Cisco Unified IP Phone 7905G
     CP7902040000SCCP040701A.sbin
     -
     
      Cisco Unified IP Phone 7910
     P00403020214.bin
     -
     
      Cisco IP Phone 7911G*
     cmterm-7911-sccp.7-2-1.exe
     -
     


  Table 3. Cisco Unified IP Phone Support Using SCCP

        Phone
       Cisco Unified SRST 2.1
       Cisco Unified SRST 3.3
       Cisco Unified SRST 3.4
       Cisco Unified SRST 4.0
       Cisco Unified SRST 4.1
       
        Cisco Unified IP Phone 7970G and 7971G-GE models
       -
       X
       X
       X
       X
       
        Cisco Unified IP Phone 7960G and 7940G models
       X
       X
       X
       X
       X
       
        Cisco Unified IP Phone 7961G, 7941G, 7961G-GE, and 7941G-GE models
       -
       X
       X
       X
       X
       
        Cisco Unified IP Conference Station 7935
       X
       X
       X
       X
       X
       
        Cisco Unified IP Conference Station 7936
       -
       X
       X
       X
       X
       
        Cisco Unified IP Phone 7912G
       -
       X
       X
       X
       X
       
        Cisco Unified IP Phone 7911G
       -
       -
       -
       X
       X
       


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: cisco-voip at puck.nether.net 
  Sent: Thursday, May 10, 2007 4:54 PM
  Subject: Re: [cisco-voip] which ios to use for 3845


  Well, considering we've got 7911s we need SRST for, looks like it's 12.4(9)T or better for me!

  Talk about bleeding edge...at least it looks like 12.4(11)T2 is LD and not ED.

  Thanks for all your suggestions.



  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  ...there's no such thing as a bad timbit...

    ----- Original Message ----- 
    From: Carter, Bill 
    To: Lelio Fulgenzi 
    Sent: Tuesday, May 08, 2007 12:45 PM
    Subject: RE: [cisco-voip] which ios to use for 3845


    12.4(6)T7 or 12.4(7)b

    http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/system/gbtst4x/4_4/rng44nip.htm#wp1261437

    Cisco "Golden Bridge" documents
    http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/system/gbtst4x/4_4/index.htm





----------------------------------------------------------------------------
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
    Sent: Tuesday, May 08, 2007 8:57 AM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] which ios to use for 3845


    my 3845 was just dropped off. 

    what IOS are people recommending for this thing? primarily used as a voice gateway (2 T1/PRIs using MGCP) and for SRST. I don't think I will be doing much transcoding, (other than from PSTN to voip) or conferencing.


    --------------------------------------------------------------------------------
    Lelio Fulgenzi, B.A.
    Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
    (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
    ...there's no such thing as a bad timbit...
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From Scott.Voll at wesd.org  Fri May 11 10:49:48 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 11 May 2007 07:49:48 -0700
Subject: [cisco-voip] IPCC Express service activation
In-Reply-To: <84602f830705110621r757dc114w2a0c8befa5c5802@mail.gmail.com>
References: <84602f830705110506v5541d2c8se2925c236bbdf954@mail.gmail.com><BD89624EE83BD349BD31A08B453DE24401650E66@uogexb201.cfs.uoguelph.ca>
	<84602f830705110621r757dc114w2a0c8befa5c5802@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE751@Apollo.wesd.org>

Why don't you run it on your local PC?  Is it supported on the server?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Arunas
Sent: Friday, May 11, 2007 6:22 AM
To: Rasim Duric; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC Express service activation

 

CRS Editor service not starting when I've started CRS Editor. Problem is
that if CRS editor service in shutdown state I can't debug my script.
When I got this issue (can't debug script) I've started to look at
services an found this problem.
Any ideas ?

Thanks,
Arunas

On 5/11/07, Rasim Duric <rduric at uoguelph.ca > wrote:

Starting/Closing of the CRS editor will start/shutdown CRS Editor
service.

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Arunas
Sent: Friday, May 11, 2007 8:06 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Express service activation

 

Hi All,

We have IPCC express 4.0.5a running on separate server and CRS Editor
service in shutdown state. Could anybody give me an advice to get this
service up and running.  I can't find any relevant information on
cisco.com.

Many thanks.
Arunas

 

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From jgrace at adcapnet.com  Fri May 11 11:06:01 2007
From: jgrace at adcapnet.com (James Grace)
Date: Fri, 11 May 2007 11:06:01 -0400
Subject: [cisco-voip] IPCCX trying to display Customer name in enterprise
	data
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C657@mail.adcapnetsvc.local>

?
I need to modify the enterprise data with a new field to display a name base on incoming calls to the ipccx.  
ive don't this before in ver 3.5 but its not working in 4.0.5.  so far ive done the following steps 
Goto enterprise data 

Add a field  called display and give it a index number 

Then goto the layout (default) and add the field to the layout

Go down to the workflow 

And make sure the show enterprise and allow agent to edit enterprise

On the script 

After the select resource set 

I Change it to select and add the set enterprise call info step # This is where im lost #

I created a variable (string) with the name I want  "NAME"

add the connect step and also added a goto queue under the fail 

 

does anyone have a sample script for this 

 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
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From rduric at uoguelph.ca  Fri May 11 11:41:37 2007
From: rduric at uoguelph.ca (Rasim Duric)
Date: Fri, 11 May 2007 11:41:37 -0400
Subject: [cisco-voip] IPCCX trying to display Customer name in
	enterprisedata
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C657@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C657@mail.adcapnetsvc.local>
Message-ID: <BD89624EE83BD349BD31A08B453DE24401650E97@uogexb201.cfs.uoguelph.ca>

I had the same problem during the migration from 3.5 to 4.0.5 and this
is the document that could help:

http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_not
e09186a00806b136d.shtml

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Friday, May 11, 2007 11:06 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCCX trying to display Customer name in
enterprisedata

 

?

I need to modify the enterprise data with a new field to display a name
base on incoming calls to the ipccx.  

ive don't this before in ver 3.5 but its not working in 4.0.5.  so far
ive done the following steps 

Goto enterprise data 

Add a field  called display and give it a index number 

Then goto the layout (default) and add the field to the layout

Go down to the workflow 

And make sure the show enterprise and allow agent to edit enterprise

On the script 

After the select resource set 

I Change it to select and add the set enterprise call info step # This
is where im lost #

I created a variable (string) with the name I want  "NAME"

add the connect step and also added a goto queue under the fail 

 

does anyone have a sample script for this 

 

James Grace 

 

Systems Engineer

Adcap Network Systems, Inc.

6865 Shiloh Road East, Suite 250

Alpharetta, GA 30005

 

office: 678-456-6986

direct: 678-456-6761

jgrace at adcapnet.com

www.adcapnet.com <http://www.adcapnet.com/> 

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From lelio at uoguelph.ca  Fri May 11 12:05:05 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 11 May 2007 12:05:05 -0400
Subject: [cisco-voip] gateway registration flapping
Message-ID: <010201c793e6$24216d00$47196883@cfs.uoguelph.ca>

Not sure why, but it seems like the gateway registration to callmanager is flapping. No event viewer log entries correspond to this. Any idea why this might be happening?


  vgw-kemptville# show ccm-manager
  MGCP Domain Name: vgw-kemptville
  Priority        Status                   Host
  ============================================================
  Primary         Registered               10.104.21.121
  First Backup    Down                     10.104.21.109
  Second Backup   None

  Current active Call Manager:    10.104.21.121
  Backhaul/Redundant link port:   2428
  Failover Interval:              30 seconds
  Keepalive Interval:             15 seconds
  Last keepalive sent:            12:01:28 EDT May 11 2007 (elapsed time: 00:00:12)
  Last MGCP traffic time:         12:01:28 EDT May 11 2007 (elapsed time: 00:00:12)
  Last failover time:             11:58:52 EDT May 11 2007 from (10.104.21.121)
  Last switchback time:           11:58:58 EDT May 11 2007 from (10.104.21.109)
  Switchback mode:                Graceful
  MGCP Fallback mode:             Not Selected
  Last MGCP Fallback start time:  None
  Last MGCP Fallback end time:    None
  MGCP Download Tones:            Disabled

  Backhaul Link info:
      Link Protocol:      TCP

  vgw-kemptville# show ccm-manager
  MGCP Domain Name: vgw-kemptville
  Priority        Status                   Host
  ============================================================
  Primary         Registered               10.104.21.121
  First Backup    Backup Ready             10.104.21.109
  Second Backup   None


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From jason.aarons at us.didata.com  Fri May 11 12:28:45 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 11 May 2007 12:28:45 -0400
Subject: [cisco-voip] Anyone used the WIC-1AM for managing remote router ?
Message-ID: <C1FE15183DA37645BC0633BC604E44F00589E824@USNAEXCH.na.didata.local>

Can anyone attest to using the WIC-1AM for managing a router?

 

From: Nick Monfreda (US) 
Sent: Friday, May 11, 2007 12:10 PM
To: 
Subject: RE: ISR Question

 

Will this part work?

 

WIC-1AM

 

 

________________________________

From: Jason Aarons (US) 
Sent: Friday, May 11, 2007 12:00 PM



 

Yes but it's expensive.  I recall it's like a WIC-AM (analog modem). For
single device use the USR5686E or like.

http://www.cisco.com/en/US/tech/tk801/tk36/technologies_tech_note09186a0
0800941c9.shtml

 

 

 

For large data center we used the 2511, it's EoL, replaced with a
AS2511-RJ, I think.

 

Also the Digi PortServer TS 16 is really cheap. Single modem can control
up to 16 ports.

 

Question:

I am having a brain freeze. 

 

Does Cisco have a module that can plug into an ISR 2800 that has a modem
port that someone can dial into that will allow remote management of the
router if the WAN goes down? 

 

 




-----------------------------------------
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This e-mail communication and any attachments may contain
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From jim at tgasolutions.com  Fri May 11 13:24:06 2007
From: jim at tgasolutions.com (Jim McBurnett)
Date: Fri, 11 May 2007 13:24:06 -0400
Subject: [cisco-voip] Anyone used the WIC-1AM for managing remote router
	?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00589E824@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00589E824@USNAEXCH.na.didata.local>
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC31B589@exchange.TGASolutions.local>

I have a few customers doing this...

It works good...

 

jim

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Friday, May 11, 2007 12:29 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Anyone used the WIC-1AM for managing remote router
?

 

Can anyone attest to using the WIC-1AM for managing a router?

 

From: Nick Monfreda (US) 
Sent: Friday, May 11, 2007 12:10 PM
To: 
Subject: RE: ISR Question

 

Will this part work?

 

WIC-1AM

 

 

________________________________

From: Jason Aarons (US) 
Sent: Friday, May 11, 2007 12:00 PM

 

Yes but it's expensive.  I recall it's like a WIC-AM (analog modem). For
single device use the USR5686E or like.

http://www.cisco.com/en/US/tech/tk801/tk36/technologies_tech_note09186a0
0800941c9.shtml

 

 

 

For large data center we used the 2511, it's EoL, replaced with a
AS2511-RJ, I think.

 

Also the Digi PortServer TS 16 is really cheap. Single modem can control
up to 16 ports.

 

Question:

I am having a brain freeze. 

 

Does Cisco have a module that can plug into an ISR 2800 that has a modem
port that someone can dial into that will allow remote management of the
router if the WAN goes down? 

 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 

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From bcarter at sentinel.com  Fri May 11 13:52:54 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Fri, 11 May 2007 12:52:54 -0500
Subject: [cisco-voip] which ios to use for 3845
In-Reply-To: <004201c793d3$af522fd0$47196883@cfs.uoguelph.ca>
References: <004201c793d3$af522fd0$47196883@cfs.uoguelph.ca>
Message-ID: <6A0744639B86DD459433108DBF101D26016AE697@MAIL01.sentinel.com>

I have SRST version 3.3 gateways with 7911g's.  They worked in our
testing.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, May 11, 2007 8:53 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] which ios to use for 3845


You really gotta like it when documentation conflicts.
 
And from srst 3.3 "supported firmware, platforms, etc", we get a table
that says 7911g is supported.
 
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2169/prod_inst
allation_guide09186a00805fdf87.html
 
>From srst 4.1 datasheet we get a table that says 7911g is not supported
until 4.0
 
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2169/products_
data_sheet0900aecd805e1e79.html
 
"round and round she goes, where she stops, nobody knows."
 
 
Cisco Unified IP Phone Support

Cisco Unified IP Phone or Cisco Endpoint

File Name of Compatible Cisco Unified CallManager
Firmware for SCCP 

File Name of Compatible
Cisco Unified CallManager 
Firmware for SIP 

Cisco ATA-186 

Cisco ATA-188

ATA030100SCCP040211A.zup

-

Cisco IP Communicator 2.0*

NA

NA

Cisco Unified IP Phone 7902G

CP7902040000SCCP040701A.sbin

-

Cisco Unified IP Phone 7905G

CP7902040000SCCP040701A.sbin

-

Cisco Unified IP Phone 7910

P00403020214.bin

-

Cisco IP Phone 7911G*

cmterm-7911-sccp.7-2-1.exe

-

 

	Table 3. Cisco Unified IP Phone Support Using SCCP

	
Phone

Cisco Unified SRST 2.1

Cisco Unified SRST 3.3

Cisco Unified SRST 3.4

Cisco Unified SRST 4.0

Cisco Unified SRST 4.1

Cisco Unified IP Phone 7970G and 7971G-GE models

-

X

X

X

X

Cisco Unified IP Phone 7960G and 7940G models

X

X

X

X

X

Cisco Unified IP Phone 7961G, 7941G, 7961G-GE, and 7941G-GE models

-

X

X

X

X

Cisco Unified IP Conference Station 7935

X

X

X

X

X

Cisco Unified IP Conference Station 7936

-

X

X

X

X

Cisco Unified IP Phone 7912G

-

X

X

X

X

Cisco Unified IP Phone 7911G

-

-

-

X

X

 
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...


	----- Original Message ----- 
	From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  
	To: cisco-voip at puck.nether.net 
	Sent: Thursday, May 10, 2007 4:54 PM
	Subject: Re: [cisco-voip] which ios to use for 3845

	Well, considering we've got 7911s we need SRST for, looks like
it's 12.4(9)T or better for me!
	 
	Talk about bleeding edge...at least it looks like 12.4(11)T2 is
LD and not ED.
	 
	Thanks for all your suggestions.
	 
	 
	 
	
------------------------------------------------------------------------
--------
	Lelio Fulgenzi, B.A.
	Senior Analyst (CCS) * University of Guelph * Guelph, Ontario
N1G 2W1
	(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
	
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
	...there's no such thing as a bad timbit...
	

		----- Original Message ----- 
		From: Carter, Bill <mailto:bcarter at sentinel.com>  
		To: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  
		Sent: Tuesday, May 08, 2007 12:45 PM
		Subject: RE: [cisco-voip] which ios to use for 3845

		12.4(6)T7 or 12.4(7)b
		 
	
http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/sys
tem/gbtst4x/4_4/rng44nip.htm#wp1261437
		 
		Cisco "Golden Bridge" documents
	
http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/sys
tem/gbtst4x/4_4/index.htm
	
<http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/sy
stem/gbtst4x/4_4/rng44nip.htm>  
		 

________________________________

		From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
		Sent: Tuesday, May 08, 2007 8:57 AM
		To: cisco-voip at puck.nether.net
		Subject: [cisco-voip] which ios to use for 3845
		
		
		my 3845 was just dropped off. 
		 
		what IOS are people recommending for this thing?
primarily used as a voice gateway (2 T1/PRIs using MGCP) and for SRST. I
don't think I will be doing much transcoding, (other than from PSTN to
voip) or conferencing.
		 
		 
	
------------------------------------------------------------------------
--------
		Lelio Fulgenzi, B.A.
		Senior Analyst (CCS) * University of Guelph * Guelph,
Ontario N1G 2W1
		(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
	
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
		...there's no such thing as a bad timbit...
		

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From lelio at uoguelph.ca  Fri May 11 13:55:03 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 11 May 2007 13:55:03 -0400
Subject: [cisco-voip] which ios to use for 3845
References: <004201c793d3$af522fd0$47196883@cfs.uoguelph.ca>
	<6A0744639B86DD459433108DBF101D26016AE697@MAIL01.sentinel.com>
Message-ID: <006401c793f5$81b805a0$47196883@cfs.uoguelph.ca>

too bad the call-manager-fall back doesn't support 7911 for max-lines. or the 7936 for that matter.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Carter, Bill 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Friday, May 11, 2007 1:52 PM
  Subject: RE: [cisco-voip] which ios to use for 3845


  I have SRST version 3.3 gateways with 7911g's.  They worked in our testing.



------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Friday, May 11, 2007 8:53 AM
  To: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] which ios to use for 3845


  You really gotta like it when documentation conflicts.

  And from srst 3.3 "supported firmware, platforms, etc", we get a table that says 7911g is supported.

  http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2169/prod_installation_guide09186a00805fdf87.html

  From srst 4.1 datasheet we get a table that says 7911g is not supported until 4.0

  http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2169/products_data_sheet0900aecd805e1e79.html

  "round and round she goes, where she stops, nobody knows."


        Cisco Unified IP Phone Support
       
        Cisco Unified IP Phone or Cisco Endpoint
       File Name of Compatible Cisco Unified CallManager
        Firmware for SCCP 
       File Name of Compatible
        Cisco Unified CallManager 
        Firmware for SIP 
       
        Cisco ATA-186 

        Cisco ATA-188
       ATA030100SCCP040211A.zup
       -
       
        Cisco IP Communicator 2.0*
       NA
       NA
       
        Cisco Unified IP Phone 7902G
       CP7902040000SCCP040701A.sbin
       -
       
        Cisco Unified IP Phone 7905G
       CP7902040000SCCP040701A.sbin
       -
       
        Cisco Unified IP Phone 7910
       P00403020214.bin
       -
       
        Cisco IP Phone 7911G*
       cmterm-7911-sccp.7-2-1.exe
       -
       


    Table 3. Cisco Unified IP Phone Support Using SCCP

          Phone
         Cisco Unified SRST 2.1
         Cisco Unified SRST 3.3
         Cisco Unified SRST 3.4
         Cisco Unified SRST 4.0
         Cisco Unified SRST 4.1
         
          Cisco Unified IP Phone 7970G and 7971G-GE models
         -
         X
         X
         X
         X
         
          Cisco Unified IP Phone 7960G and 7940G models
         X
         X
         X
         X
         X
         
          Cisco Unified IP Phone 7961G, 7941G, 7961G-GE, and 7941G-GE models
         -
         X
         X
         X
         X
         
          Cisco Unified IP Conference Station 7935
         X
         X
         X
         X
         X
         
          Cisco Unified IP Conference Station 7936
         -
         X
         X
         X
         X
         
          Cisco Unified IP Phone 7912G
         -
         X
         X
         X
         X
         
          Cisco Unified IP Phone 7911G
         -
         -
         -
         X
         X
         


  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  ...there's no such thing as a bad timbit...

    ----- Original Message ----- 
    From: Lelio Fulgenzi 
    To: cisco-voip at puck.nether.net 
    Sent: Thursday, May 10, 2007 4:54 PM
    Subject: Re: [cisco-voip] which ios to use for 3845


    Well, considering we've got 7911s we need SRST for, looks like it's 12.4(9)T or better for me!

    Talk about bleeding edge...at least it looks like 12.4(11)T2 is LD and not ED.

    Thanks for all your suggestions.



    --------------------------------------------------------------------------------
    Lelio Fulgenzi, B.A.
    Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
    (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
    ...there's no such thing as a bad timbit...

      ----- Original Message ----- 
      From: Carter, Bill 
      To: Lelio Fulgenzi 
      Sent: Tuesday, May 08, 2007 12:45 PM
      Subject: RE: [cisco-voip] which ios to use for 3845


      12.4(6)T7 or 12.4(7)b

      http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/system/gbtst4x/4_4/rng44nip.htm#wp1261437

      Cisco "Golden Bridge" documents
      http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/system/gbtst4x/4_4/index.htm





--------------------------------------------------------------------------
      From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
      Sent: Tuesday, May 08, 2007 8:57 AM
      To: cisco-voip at puck.nether.net
      Subject: [cisco-voip] which ios to use for 3845


      my 3845 was just dropped off. 

      what IOS are people recommending for this thing? primarily used as a voice gateway (2 T1/PRIs using MGCP) and for SRST. I don't think I will be doing much transcoding, (other than from PSTN to voip) or conferencing.


      --------------------------------------------------------------------------------
      Lelio Fulgenzi, B.A.
      Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
      (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
      ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
      ...there's no such thing as a bad timbit...
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From jgrace at adcapnet.com  Fri May 11 16:11:12 2007
From: jgrace at adcapnet.com (James Grace)
Date: Fri, 11 May 2007 16:11:12 -0400
Subject: [cisco-voip] unable to here prompts on ipccx
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C658@mail.adcapnetsvc.local>

?
for some reason when i call one of my triggers for ipccx, from an internal ip phone, Im not able to hear the prompts.  now the call is going thru the scripts from top to bottom. and all the options are working there.  but just no audio.  Now this may have something to do with the fact that the ipccx server is at a colo.  Could i be having a codec issue 
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
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From matt at saskin.net  Fri May 11 16:21:14 2007
From: matt at saskin.net (Matthew Saskin)
Date: Fri, 11 May 2007 16:21:14 -0400
Subject: [cisco-voip] unable to here prompts on ipccx
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C658@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C658@mail.adcapnetsvc.local>
Message-ID: <4644D03A.3060008@saskin.net>

codec issue or one way traffic I would think.

1 - can you ping the ipccx box, can the ipccx box ping you?
2 - any firewalls/vpn/etc. in between?  If so, are you allowing RTP 
traffic through in both directions?
3 - what codec do you have configured for calls between the location 
that your IPCCx route points/ports live in and the location your phone 
lives in.  If necessary, are there transcoding resources available to 
both?  Are there enough transcoding resources?

-matt

James Grace wrote:
> ?
> for some reason when i call one of my triggers for ipccx, from an 
> internal ip phone, Im not able to hear the prompts.  now the call is 
> going thru the scripts from top to bottom. and all the options are 
> working there.  but just no audio.  Now this may have something to do 
> with the fact that the ipccx server is at a colo.  Could i be having a 
> codec issue
>  
> James Grace
>  
> Systems Engineer
> Adcap Network Systems, Inc.
> 6865 Shiloh Road East, Suite 250
> Alpharetta, GA 30005
>  
> office: 678-456-6986
> direct: 678-456-6761
> jgrace at adcapnet.com <mailto:jgrace at adcapnet.com>
> www.adcapnet.com <http://www.adcapnet.com/>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   



From jgrace at adcapnet.com  Fri May 11 17:08:24 2007
From: jgrace at adcapnet.com (James Grace)
Date: Fri, 11 May 2007 17:08:24 -0400
Subject: [cisco-voip] my 7941 phone are making my swich ports bounce
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C65A@mail.adcapnetsvc.local>

?
for some reason we have a group of 7941 that we plug in to the 3560 poe sw.  when they power up it causes the sw port to go up and down. on the phone it shows Ethernet disconnect and them registering and back to disconnect.  on the sw its just going up and down.  Now we have a few phones that are up and fine.  but the bulk of them have this problem.  Have anybody else seen this. its almost like the auto duplex is not working for the phones
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
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From aizhar at ccbootcamp.com  Fri May 11 18:46:44 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Fri, 11 May 2007 15:46:44 -0700
Subject: [cisco-voip] my 7941 phone are making my swich ports bounce
Message-ID: <E76ACA8C13EF1847BC69538B522C36634D3516@nlimail.nlidomain>


Have you tested it locally, might be cabling issues ...

Does it work eventually after all this up down thing?

--------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take Cisco Learning credits!
www.ccbootcamp.com (Cisco Training and Advanced Technology Rental Racks)
www.routerie.com (Routing and Switching Forums)
www.securityie.com (Security Forums)
www.voiceie.com (Voice Forums)



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Friday, May 11, 2007 14:08
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] my 7941 phone are making my swich ports bounce [bcc][faked-from][html-rem]

?
for some reason we have a group of 7941 that we plug in to the 3560 poe sw.  when they power up it causes the sw port to go up and down. on the phone it shows Ethernet disconnect and them registering and back to disconnect.  on the sw its just going up and down.  Now we have a few phones that are up and fine.  but the bulk of them have this problem.  Have anybody else seen this. its almost like the auto duplex is not working for the phones
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.467 / Virus Database: 269.6.8/797 - Release Date: 5/10/2007 17:10
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.467 / Virus Database: 269.6.8/797 - Release Date: 5/10/2007 17:10
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.467 / Virus Database: 269.6.8/797 - Release Date: 5/10/2007 17:10
 


From aizhar at ccbootcamp.com  Fri May 11 18:57:25 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Fri, 11 May 2007 15:57:25 -0700
Subject: [cisco-voip] gateway registration flapping
In-Reply-To: <010201c793e6$24216d00$47196883@cfs.uoguelph.ca>
Message-ID: <E76ACA8C13EF1847BC69538B522C36634D3519@nlimail.nlidomain>


Looks like a network issue, between the backup ccm and the gw.

Has it always worked or is this a new install?

--------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take Cisco Learning credits!
www.ccbootcamp.com (Cisco Training and Advanced Technology Rental Racks)
www.routerie.com (Routing and Switching Forums)
www.securityie.com (Security Forums)
www.voiceie.com (Voice Forums)



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, May 11, 2007 09:05
To: 
Subject: [cisco-voip] gateway registration flapping [bcc][faked-from][html-rem]

Not sure why, but it seems like the gateway registration to callmanager is flapping. No event viewer log entries correspond to this. Any idea why this might be happening?


  vgw-kemptville# show ccm-manager
  MGCP Domain Name: vgw-kemptville
  Priority        Status                   Host
  ============================================================
  Primary         Registered               10.104.21.121
  First Backup    Down                     10.104.21.109
  Second Backup   None

  Current active Call Manager:    10.104.21.121
  Backhaul/Redundant link port:   2428
  Failover Interval:              30 seconds
  Keepalive Interval:             15 seconds
  Last keepalive sent:            12:01:28 EDT May 11 2007 (elapsed time: 00:00:12)
  Last MGCP traffic time:         12:01:28 EDT May 11 2007 (elapsed time: 00:00:12)
  Last failover time:             11:58:52 EDT May 11 2007 from (10.104.21.121)
  Last switchback time:           11:58:58 EDT May 11 2007 from (10.104.21.109)
  Switchback mode:                Graceful
  MGCP Fallback mode:             Not Selected
  Last MGCP Fallback start time:  None
  Last MGCP Fallback end time:    None
  MGCP Download Tones:            Disabled

  Backhaul Link info:
      Link Protocol:      TCP

  vgw-kemptville# show ccm-manager
  MGCP Domain Name: vgw-kemptville
  Priority        Status                   Host
  ============================================================
  Primary         Registered               10.104.21.121
  First Backup    Backup Ready             10.104.21.109
  Second Backup   None


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.467 / Virus Database: 269.6.8/797 - Release Date: 5/10/2007 17:10
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.467 / Virus Database: 269.6.8/797 - Release Date: 5/10/2007 17:10
 


From jonvoip at gmail.com  Sat May 12 05:38:05 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 12 May 2007 04:38:05 -0500
Subject: [cisco-voip] New IPCC cluster, error when adding JTAPI Triggers...
Message-ID: <5d093f9a0705120238t52032b53td9d040503a25fc55@mail.gmail.com>

IPCC 4.0(5a), w/ CCM 4.1(3)sr4d, four server HA cluster...


When I try to add a JTAPI Trigger, I get:

"Could not update the user on the CallManager." Right next to the DN I
added...

I am not sure the cluster is in sync, how do I check, and how do I get it in
sync if it isn't?




Jonathan
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From jason.aarons at us.didata.com  Sat May 12 11:45:22 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Sat, 12 May 2007 11:45:22 -0400
Subject: [cisco-voip] Annoying CDP message with 7941G-GE in 4.1(3)SR4d
Message-ID: <C1FE15183DA37645BC0633BC604E44F00589EEE4@USNAEXCH.na.didata.local>

This is driving me crazy, any way to disable these pop ups when updating
a 7941G-GE ? Since the default is CDP enabled maybe show the popup
message when they set CDP to disabled?

 

Pop-Up

Cisco Discovery Protocol (CDP): Switch Port: Warning! CDP should be
disabled on the Network port if this phone is connected to a non-Cisco
switch. For futher details, consult hte Cisco  CallManager
Administration Guide. Ok.

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
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From tettokalle at hotmail.com  Sat May 12 13:32:57 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Sat, 12 May 2007 19:32:57 +0200
Subject: [cisco-voip] Calls from the PSTN getting 'No circuit/channel
	available'
In-Reply-To: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>
References: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>
Message-ID: <f24tok$sld$1@sea.gmane.org>

Hi, I had this often before, try to play around with bchan order. change 
from bottom up to top down. I also had, that I simply could not use any 
of 4 channels out of 30. So contact also your telco and ask for the 
channels and if they have any problem.

I have also seen, that someone has bought 16 channels and the telco only 
had 12 delivered.

So I think it is a isdn layer 3 channel selection problem, of a channel 
that simply not exists.

Cheers

pedro marques schrieb:
> Hi,
> 
> I have a CCM running 4.2 connecting through MGCP to a 2821 Gateway.
> After some debugging and traces enabled on the CCM I noticed that the 
> calls that are getting disconnected from the PSTN all fall in the first 
> 5 channels of the E1 PRI.
> 
> These are the relevant commands on the 2821:
> 
> controller E1 0/0/0
>  pri-group timeslots 1-16 service mgcp
> !
> interface Serial0/0/0:15
>  no ip address
>  isdn switch-type primary-net5
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  isdn bchan-number-order ascending
>  no cdp enable
> !
> ccm-manager mgcp
> ccm-manager config server <CCM ip address>
> !
> mgcp
> mgcp call-agent <CCM ip address> service-type mgcp version 0.1
> mgcp dtmf-relay voip codec all mode out-of-band
> mgcp package-capability rtp-package
> mgcp default-package line-package
> mgcp sdp simple
> !
> mgcp profile default
> 
> These are two snippets from a 'debug isdn q931':
> 
> *May 10 11:32:02.582: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref 
> = 0x007F
>         Sending Complete
>         Bearer Capability i = 0x9090A3
>                 Standard = CCITT
>                 Transfer Capability = 3.1kHz Audio
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA18384
>                 Preferred, Channel 4
>         Progress Ind i = 0x8183 - Origination address is non-ISDN 
>         Calling Party Number i = 0x0083, 'calling party number'
>                 Plan:Unknown, Type:Unknown
>         Called Party Number i = 0x81, 'called party number'
>                 Plan:ISDN, Type:Unknown
> *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  
> callref = 0x807F
>         Channel ID i = 0xA98384
>                 Exclusive, Channel 4
> *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  
> callref = 0x807F
>         Cause i = 0x80A2 - No circuit/channel available
> *May 10 11:32:02.790: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  
> callref = 0x007F
> *May 10 11:32:02.802: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8  
> callref = 0x807F
> 
> 
> *May 10 11:33:35.062: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref 
> = 0x0057
>         Sending Complete
>         Bearer Capability i = 0x9090A3
>                 Standard = CCITT
>                 Transfer Capability = 3.1kHz Audio
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA18383
>                 Preferred, Channel 3
>         Progress Ind i = 0x8283 - Origination address is non-ISDN 
>         Calling Party Number i = 0x0083, 'calling party number' 
>                 Plan:Unknown, Type:Unknown
>         Called Party Number i = 0x81, 'called party number'
>                 Plan:ISDN, Type:Unknown
> *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  
> callref = 0x8057
>         Channel ID i = 0xA98383
>                 Exclusive, Channel 3
> *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  
> callref = 0x8057
>         Cause i = 0x80A2 - No circuit/channel available
> 
> In attachment I include a file (copy of ccm00000065.txt) that has the 
> relevant information taken from a CCM Trace log and run through Voice 
> Log Translator.
> 
> With all this information I went to check the MGCP Gateway configuration 
> in CCMAdmin. Some of the configuration elements included on the E1 
> endpoint configuration page and that are somewhat confusing me (mind you 
> that this is an inherited configuration) are as follows:
> 
> Channel Selection Order: BOTTOM_UP (since the configuration on the 
> gateway's interface Serial0/0/0:15 is for bchan-number-order ascending, 
> shouldn't these parameters match by using the TOP_DOWN option?)
> 
> Channel IE Type: Use Number When 1B (This is an option that I don't 
> fully understand. Any help on this would be appreciated. But to what I 
> understand, and since the E1 PRI configuration is only used for half a 
> primary (15 channels) shouldn't this parameter be set to 'Timeslot Number'?)
> 
> TIA
> 
> 
> -- 
> Pedro Miguel Marques
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From jason.aarons at us.didata.com  Sat May 12 14:00:14 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Sat, 12 May 2007 14:00:14 -0400
Subject: [cisco-voip] Calls from the PSTN getting 'No
	circuit/channelavailable'
In-Reply-To: <f24tok$sld$1@sea.gmane.org>
References: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>
	<f24tok$sld$1@sea.gmane.org>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00589EEEC@USNAEXCH.na.didata.local>

Fractional PRI isn't supported on MGCP. Workaround set all channels in
your pri-group timeslots, then busyout the out of service b-channels.

Example
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-dms100
 isdn incoming-voice voice
 isdn service b_channel 11 state 2
 isdn service b_channel 12 state 2
 isdn service b_channel 13 state 2
 isdn service b_channel 14 state 2
 isdn service b_channel 15 state 2
 isdn service b_channel 16 state 2
 isdn service b_channel 17 state 2
 isdn service b_channel 18 state 2
 isdn service b_channel 19 state 2
 isdn service b_channel 20 state 2
 isdn service b_channel 21 state 2
 isdn service b_channel 22 state 2
 isdn bind-l3 ccm-manager
 no cdp enable

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kalle Tetto
Sent: Saturday, May 12, 2007 1:33 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Calls from the PSTN getting 'No
circuit/channelavailable'

Hi, I had this often before, try to play around with bchan order. change

from bottom up to top down. I also had, that I simply could not use any 
of 4 channels out of 30. So contact also your telco and ask for the 
channels and if they have any problem.

I have also seen, that someone has bought 16 channels and the telco only

had 12 delivered.

So I think it is a isdn layer 3 channel selection problem, of a channel 
that simply not exists.

Cheers

pedro marques schrieb:
> Hi,
> 
> I have a CCM running 4.2 connecting through MGCP to a 2821 Gateway.
> After some debugging and traces enabled on the CCM I noticed that the 
> calls that are getting disconnected from the PSTN all fall in the
first 
> 5 channels of the E1 PRI.
> 
> These are the relevant commands on the 2821:
> 
> controller E1 0/0/0
>  pri-group timeslots 1-16 service mgcp
> !
> interface Serial0/0/0:15
>  no ip address
>  isdn switch-type primary-net5
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  isdn bchan-number-order ascending
>  no cdp enable
> !
> ccm-manager mgcp
> ccm-manager config server <CCM ip address>
> !
> mgcp
> mgcp call-agent <CCM ip address> service-type mgcp version 0.1
> mgcp dtmf-relay voip codec all mode out-of-band
> mgcp package-capability rtp-package
> mgcp default-package line-package
> mgcp sdp simple
> !
> mgcp profile default
> 
> These are two snippets from a 'debug isdn q931':
> 
> *May 10 11:32:02.582: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8
callref 
> = 0x007F
>         Sending Complete
>         Bearer Capability i = 0x9090A3
>                 Standard = CCITT
>                 Transfer Capability = 3.1kHz Audio
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA18384
>                 Preferred, Channel 4
>         Progress Ind i = 0x8183 - Origination address is non-ISDN 
>         Calling Party Number i = 0x0083, 'calling party number'
>                 Plan:Unknown, Type:Unknown
>         Called Party Number i = 0x81, 'called party number'
>                 Plan:ISDN, Type:Unknown
> *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  
> callref = 0x807F
>         Channel ID i = 0xA98384
>                 Exclusive, Channel 4
> *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  
> callref = 0x807F
>         Cause i = 0x80A2 - No circuit/channel available
> *May 10 11:32:02.790: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  
> callref = 0x007F
> *May 10 11:32:02.802: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8

> callref = 0x807F
> 
> 
> *May 10 11:33:35.062: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8
callref 
> = 0x0057
>         Sending Complete
>         Bearer Capability i = 0x9090A3
>                 Standard = CCITT
>                 Transfer Capability = 3.1kHz Audio
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA18383
>                 Preferred, Channel 3
>         Progress Ind i = 0x8283 - Origination address is non-ISDN 
>         Calling Party Number i = 0x0083, 'calling party number' 
>                 Plan:Unknown, Type:Unknown
>         Called Party Number i = 0x81, 'called party number'
>                 Plan:ISDN, Type:Unknown
> *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  
> callref = 0x8057
>         Channel ID i = 0xA98383
>                 Exclusive, Channel 3
> *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  
> callref = 0x8057
>         Cause i = 0x80A2 - No circuit/channel available
> 
> In attachment I include a file (copy of ccm00000065.txt) that has the 
> relevant information taken from a CCM Trace log and run through Voice 
> Log Translator.
> 
> With all this information I went to check the MGCP Gateway
configuration 
> in CCMAdmin. Some of the configuration elements included on the E1 
> endpoint configuration page and that are somewhat confusing me (mind
you 
> that this is an inherited configuration) are as follows:
> 
> Channel Selection Order: BOTTOM_UP (since the configuration on the 
> gateway's interface Serial0/0/0:15 is for bchan-number-order
ascending, 
> shouldn't these parameters match by using the TOP_DOWN option?)
> 
> Channel IE Type: Use Number When 1B (This is an option that I don't 
> fully understand. Any help on this would be appreciated. But to what I

> understand, and since the E1 PRI configuration is only used for half a

> primary (15 channels) shouldn't this parameter be set to 'Timeslot
Number'?)
> 
> TIA
> 
> 
> -- 
> Pedro Miguel Marques
> 
> 
>
------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From jonvoip at gmail.com  Sat May 12 14:48:24 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 12 May 2007 13:48:24 -0500
Subject: [cisco-voip] Calls from the PSTN getting 'No circuit/channel
	available'
In-Reply-To: <d2216e4e0705100802r450ad5f8l4d7886be8d95bbff@mail.gmail.com>
References: <d2216e4e0705100731i3b2277c3xcaa431627178e5b@mail.gmail.com>
	<5d093f9a0705100747m2c309f45l342f93c10fc14769@mail.gmail.com>
	<d2216e4e0705100802r450ad5f8l4d7886be8d95bbff@mail.gmail.com>
Message-ID: <5d093f9a0705121148t2d39bdqf38fe4afecfa7708@mail.gmail.com>

Make sure the CSS of the gateway includes the partition of the trans pattern


Jonathan

On 5/10/07, pedro marques <pmarques74 at gmail.com> wrote:
>
> Hi Jonathan,
>
> I include the same debugs with the calling and called party numbers:
>
> *May 10 11:32:02.582: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref =
> 0x007F
>         Sending Complete
>         Bearer Capability i = 0x9090A3
>                 Standard = CCITT
>                 Transfer Capability = 3.1kHz Audio
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA18384
>                 Preferred, Channel 4
>         Progress Ind i = 0x8183 - Origination address is non-ISDN
>         Calling Party Number i = 0x0083, '244560496'
>                 Plan:Unknown, Type:Unknown
>         Called Party Number i = 0x81, '3300'
>                 Plan:ISDN, Type:Unknown
> *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
> callref = 0x807F
>         Channel ID i = 0xA98384
>                 Exclusive, Channel 4
> *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8
> callref = 0x807F
>         Cause i = 0x80A2 - No circuit/channel available
> *May 10 11:32:02.790: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8  callref
> = 0x007F
> *May 10 11:32:02.802: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8
> callref = 0x807F
>
>
> *May 10 11:33:35.062: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref =
> 0x0057
>         Sending Complete
>         Bearer Capability i = 0x9090A3
>                 Standard = CCITT
>                 Transfer Capability = 3.1kHz Audio
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA18383
>                 Preferred, Channel 3
>         Progress Ind i = 0x8283 - Origination address is non-ISDN
>         Calling Party Number i = 0x0083, '244502150'
>                 Plan:Unknown, Type:Unknown
>         Called Party Number i = 0x81, '3300'
>                 Plan:ISDN, Type:Unknown
> *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
> callref = 0x8057
>         Channel ID i = 0xA98383
>                 Exclusive, Channel 3
> *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8
> callref = 0x8057
>         Cause i = 0x80A2 - No circuit/channel available
>
> The '3300' number is associated with two translation patterns. One points
> to a Hunt Pilot (5930) that is associated to a partition (WorkHours) which
> is further associated to a Time Schedule containing several Time Periods.
> The other one points directly to a DN (5250) associated to a different
> partition (NonWorkHours) further associated with a Time Schedule that
> contains several Time Periods.
>
> On 10/05/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > This message means that the digits being sent did not match any
> > route-pattern, route point, translation pattern or DN.
> >
> > You stripped out the digits under called-party number, so I can't tell
> > you what to search for in Route-Plan report...
> >
> > Just make sure the digits being sent, match something on CCM.
> >
> >
> >
> > Jonathan
> >
> >
> >
> > On 5/10/07, pedro marques < pmarques74 at gmail.com> wrote:
> >
> > > Hi,
> > >
> > > I have a CCM running 4.2 connecting through MGCP to a 2821 Gateway.
> > > After some debugging and traces enabled on the CCM I noticed that the
> > > calls that are getting disconnected from the PSTN all fall in the first 5
> > > channels of the E1 PRI.
> > >
> > > These are the relevant commands on the 2821:
> > >
> > > controller E1 0/0/0
> > >  pri-group timeslots 1-16 service mgcp
> > > !
> > > interface Serial0/0/0:15
> > >  no ip address
> > >  isdn switch-type primary-net5
> > >  isdn incoming-voice voice
> > >  isdn bind-l3 ccm-manager
> > >  isdn bchan-number-order ascending
> > >  no cdp enable
> > > !
> > > ccm-manager mgcp
> > > ccm-manager config server <CCM ip address>
> > > !
> > > mgcp
> > > mgcp call-agent <CCM ip address> service-type mgcp version 0.1
> > > mgcp dtmf-relay voip codec all mode out-of-band
> > > mgcp package-capability rtp-package
> > > mgcp default-package line-package
> > > mgcp sdp simple
> > > !
> > > mgcp profile default
> > >
> > > These are two snippets from a 'debug isdn q931':
> > >
> > > *May 10 11:32:02.582: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8
> > > callref = 0x007F
> > >         Sending Complete
> > >         Bearer Capability i = 0x9090A3
> > >                 Standard = CCITT
> > >                 Transfer Capability = 3.1kHz Audio
> > >                 Transfer Mode = Circuit
> > >                 Transfer Rate = 64 kbit/s
> > >         Channel ID i = 0xA18384
> > >                 Preferred, Channel 4
> > >         Progress Ind i = 0x8183 - Origination address is non-ISDN
> > >         Calling Party Number i = 0x0083, 'calling party number'
> > >                 Plan:Unknown, Type:Unknown
> > >         Called Party Number i = 0x81, 'called party number'
> > >                 Plan:ISDN, Type:Unknown
> > > *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
> > > callref = 0x807F
> > >         Channel ID i = 0xA98384
> > >                 Exclusive, Channel 4
> > > *May 10 11:32:02.602: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8
> > > callref = 0x807F
> > >         Cause i = 0x80A2 - No circuit/channel available
> > > *May 10 11:32:02.790: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8
> > > callref = 0x007F
> > > *May 10 11:32:02.802: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8
> > > callref = 0x807F
> > >
> > >
> > > *May 10 11:33:35.062: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8
> > > callref = 0x0057
> > >         Sending Complete
> > >         Bearer Capability i = 0x9090A3
> > >                 Standard = CCITT
> > >                 Transfer Capability = 3.1kHz Audio
> > >                 Transfer Mode = Circuit
> > >                 Transfer Rate = 64 kbit/s
> > >         Channel ID i = 0xA18383
> > >                 Preferred, Channel 3
> > >         Progress Ind i = 0x8283 - Origination address is non-ISDN
> > >         Calling Party Number i = 0x0083, 'calling party number'
> > >                 Plan:Unknown, Type:Unknown
> > >         Called Party Number i = 0x81, 'called party number'
> > >                 Plan:ISDN, Type:Unknown
> > > *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8
> > > callref = 0x8057
> > >         Channel ID i = 0xA98383
> > >                 Exclusive, Channel 3
> > > *May 10 11:33:35.086: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8
> > > callref = 0x8057
> > >         Cause i = 0x80A2 - No circuit/channel available
> > >
> > > In attachment I include a file (copy of ccm00000065.txt) that has the
> > > relevant information taken from a CCM Trace log and run through Voice Log
> > > Translator.
> > >
> > > With all this information I went to check the MGCP Gateway
> > > configuration in CCMAdmin. Some of the configuration elements included on
> > > the E1 endpoint configuration page and that are somewhat confusing me (mind
> > > you that this is an inherited configuration) are as follows:
> > >
> > > Channel Selection Order: BOTTOM_UP (since the configuration on the
> > > gateway's interface Serial0/0/0:15 is for bchan-number-order
> > > ascending, shouldn't these parameters match by using the TOP_DOWN option?)
> > >
> > > Channel IE Type: Use Number When 1B (This is an option that I don't
> > > fully understand. Any help on this would be appreciated. But to what I
> > > understand, and since the E1 PRI configuration is only used for half a
> > > primary (15 channels) shouldn't this parameter be set to 'Timeslot
> > > Number'?)
> > >
> > > TIA
> > >
> > >
> > > --
> > > Pedro Miguel Marques
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >
> >
>
>
> --
> Pedro Miguel Marques
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From jonvoip at gmail.com  Sun May 13 12:53:25 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 13 May 2007 11:53:25 -0500
Subject: [cisco-voip] CCM 3.3 to 4.1 upgrade, update lines...
Message-ID: <5d093f9a0705130953m5e638b40t6b53e35c944d346d@mail.gmail.com>

So, standard 3.3 setup, each DN in a  line 1 and line 2 partition...

Upgraded to 4.1(3) and was wondering if there is a batch file that will
delete the second line, change the max calls and busy trigger on line 1 to 4
and 2, respectively and change the CFNA and CFB settings on line 1 to go to
vmail.

This is happening enough that such a batch file would be immensely valuable.






Jonathan
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From aizhar at ccbootcamp.com  Sun May 13 14:40:39 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Sun, 13 May 2007 11:40:39 -0700
Subject: [cisco-voip] CCM 3.3 to 4.1 upgrade, update lines...
References: <5d093f9a0705130953m5e638b40t6b53e35c944d346d@mail.gmail.com>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FA7@nlimail.nlidomain>

You can do most of it with BAT, if not all.
 
Haven't seen any batch for that sort of things yet ...
 
Regards,
 
Avner Izhar, CCIE #15999 (Voice)
Network Learning Inc.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
Sent: Sun 5/13/2007 9:53 AM
To: ciscovoip
Subject: [cisco-voip] CCM 3.3 to 4.1 upgrade, update lines...



So, standard 3.3 setup, each DN in a  line 1 and line 2 partition...

Upgraded to 4.1(3) and was wondering if there is a batch file that will
delete the second line, change the max calls and busy trigger on line 1 to 4
and 2, respectively and change the CFNA and CFB settings on line 1 to go to
vmail.

This is happening enough that such a batch file would be immensely valuable.






Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From fwn at feasible.net  Sun May 13 15:08:23 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Sun, 13 May 2007 12:08:23 -0700
Subject: [cisco-voip] CCM 3.3 to 4.1 upgrade, update lines...
In-Reply-To: <5d093f9a0705130953m5e638b40t6b53e35c944d346d@mail.gmail.com>
References: <5d093f9a0705130953m5e638b40t6b53e35c944d346d@mail.gmail.com>
Message-ID: <ca09e3750705131208m2b6359a7kcc8084e80ea9403@mail.gmail.com>

BAT will do most, and the AXL API allows access to everything that BAT can't
do.  AXL calls can be batch scripted to do just about *anything* you can do
via the GUI.

On 5/13/07, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> So, standard 3.3 setup, each DN in a  line 1 and line 2 partition...
>
> Upgraded to 4.1(3) and was wondering if there is a batch file that will
> delete the second line, change the max calls and busy trigger on line 1 to 4
> and 2, respectively and change the CFNA and CFB settings on line 1 to go to
> vmail.
>
> This is happening enough that such a batch file would be immensely
> valuable.
>
>
>
>
>
>
> Jonathan
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From joerg.wesely at intact-is.com  Sun May 13 15:32:12 2007
From: joerg.wesely at intact-is.com (=?ISO-8859-15?Q?J=F6rg_Wesely?=)
Date: Sun, 13 May 2007 21:32:12 +0200
Subject: [cisco-voip] How many DSPs for one E1
Message-ID: <f27p4r$6eu$1@sea.gmane.org>

Hi,

I've got a 2851 here that terminates one E1-line (ISDN PRI) for voice 
that has some DSPs inserted (see show inventory below).

Can anyone tell me how many DSPs I need for this E1 and how many of them 
can I use in a DSPfarm for transcoding and conference briding? I've 
found some documentation about how many XCODE-sessions these DSPs can 
handle, but nothing about the ones used for the ISDN termination.

Thanks in advance


J?rg




Show inventory gives this output:
---------------
NAME: "2851 chassis", DESCR: "2851 chassis"
PID: CISCO2851         , VID: V03 , SN: FTX1107A4QC

NAME: "Four port FXS DID voice interface daughtercard on Slot 0 SubSlot 
0", DESCR: "Four port FXS DID voice interface daughtercard"
PID: VIC-4FXS/DID=     , VID: 3.1, SN: FOC103324K3

NAME: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 
SubSlot 1", DESCR: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1"
PID: VWIC2-2MFT-T1/E1  , VID: V01 , SN: FOC10273FY4

NAME: "PVDMII DSP SIMM with three DSPs on Slot 0 SubSlot 4", DESCR: 
"PVDMII DSP SIMM with three DSPs"
PID: PVDM2-48          , VID: V01 , SN: FOC11050MBR

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 5", DESCR: 
"PVDMII DSP SIMM with four DSPs"
PID: PVDM2-64          , VID: V01 , SN: FOC10482Q30

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 6", DESCR: 
"PVDMII DSP SIMM with four DSPs"
PID: PVDM2-64          , VID: V01 , SN: FOC10482Q38

NAME: "NM-SE on Slot 1", DESCR: "NM-SE"
PID: NM-CUE            , VID: V03, SN: FOC11045YTG

NAME: "40GB IDE Disc Daughter Card on Slot 1 SubSlot 0", DESCR: "40GB 
IDE Disc Daughter Card"
PID:                   , VID:    , SN: FOC11053WB3
------------


From embeleco at gmail.com  Sun May 13 21:15:24 2007
From: embeleco at gmail.com (embeleco)
Date: Sun, 13 May 2007 21:15:24 -0400
Subject: [cisco-voip] DialPeer redundancy voip/pots with RSVP on the network
Message-ID: <589f5b010705131815w7671327awe7003c12fe3c01c4@mail.gmail.com>

Hi.

Currently I have an RSVP enable network that is running VoIP. On the VoIP
router I have 2 dialpeers one pointing to the VoIP network and the other to
the POTS (T1/PRI) with the same destination-pattern and using a preference
of 2 on pots dialpeer. On the VoIP dialpeer I have the req-qos controlled
load and the acc-qos in order to make the call using RSVP reservation for
QoS.

My question: If for some reason the VoIP network can't be use because of
RSVP lack of bandwidth end-to-end.. does the VoIP router scan the next
dial-peer (i.e. the POTS dialpeer)?? Or it will stop on the voip dialpeer.

Thanks in advanced,
-JJ
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From aizhar at ccbootcamp.com  Sun May 13 21:42:45 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Sun, 13 May 2007 18:42:45 -0700
Subject: [cisco-voip] How many DSPs for one E1
References: <f27p4r$6eu$1@sea.gmane.org>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FA9@nlimail.nlidomain>

Hi,
 
You have 2 E1 ports and 4 FXS, this are served by 176 DSPs.
 
In general you can dedicate 108 DSPs to XCODE tasks, the exact amount is a product of what codecs
 are you going to transcode from, since transcoding to g.729 requires less then to g.723.
 
It is advisable to run your data on cisco's dsp calculator at:
http://www.cisco.com/cgi-bin/Support/DSP/cisco_prodsel.pl
 
Regards,
Avner Izhar, CCIE #15999 (Voice)
Network Learning Inc.
 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of J?rg Wesely
Sent: Sun 5/13/2007 12:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How many DSPs for one E1



Hi,

I've got a 2851 here that terminates one E1-line (ISDN PRI) for voice
that has some DSPs inserted (see show inventory below).

Can anyone tell me how many DSPs I need for this E1 and how many of them
can I use in a DSPfarm for transcoding and conference briding? I've
found some documentation about how many XCODE-sessions these DSPs can
handle, but nothing about the ones used for the ISDN termination.

Thanks in advance


J?rg




Show inventory gives this output:
---------------
NAME: "2851 chassis", DESCR: "2851 chassis"
PID: CISCO2851         , VID: V03 , SN: FTX1107A4QC

NAME: "Four port FXS DID voice interface daughtercard on Slot 0 SubSlot
0", DESCR: "Four port FXS DID voice interface daughtercard"
PID: VIC-4FXS/DID=     , VID: 3.1, SN: FOC103324K3

NAME: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0
SubSlot 1", DESCR: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1"
PID: VWIC2-2MFT-T1/E1  , VID: V01 , SN: FOC10273FY4

NAME: "PVDMII DSP SIMM with three DSPs on Slot 0 SubSlot 4", DESCR:
"PVDMII DSP SIMM with three DSPs"
PID: PVDM2-48          , VID: V01 , SN: FOC11050MBR

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 5", DESCR:
"PVDMII DSP SIMM with four DSPs"
PID: PVDM2-64          , VID: V01 , SN: FOC10482Q30

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 6", DESCR:
"PVDMII DSP SIMM with four DSPs"
PID: PVDM2-64          , VID: V01 , SN: FOC10482Q38

NAME: "NM-SE on Slot 1", DESCR: "NM-SE"
PID: NM-CUE            , VID: V03, SN: FOC11045YTG

NAME: "40GB IDE Disc Daughter Card on Slot 1 SubSlot 0", DESCR: "40GB
IDE Disc Daughter Card"
PID:                   , VID:    , SN: FOC11053WB3
------------

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From voip at mrga.ch  Mon May 14 02:52:57 2007
From: voip at mrga.ch (Reto Gassmann)
Date: Mon, 14 May 2007 08:52:57 +0200
Subject: [cisco-voip] BARS Scheduler
Message-ID: <1179125577.464807498b220@www.mail2web.ch>

Hi all

I installed a new Version of BARS (4.0.12) this weekend and configured it to
backup the Callmanger and the two IVR Servers. I ran a backup and it produced
almost 3 GB of Data.
Since the IVRs don't change data very often (and most of the Data (2.5G) comes
from the IVRs), is it possible to set up different schedules? One for the CM
that runs every night and one for the IVRs that runs every week.

Has anone allready doen something like that??

Thanks for feedback
Reto 



From Ruben.Montes at eu.didata.com  Mon May 14 04:39:14 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Mon, 14 May 2007 10:39:14 +0200
Subject: [cisco-voip] Changing DNS domain in CCM environment
References: <f27p4r$6eu$1@sea.gmane.org>
Message-ID: <FA233791F398B848B978464507739E8F3E4256@EUBEBRUSVEX1.eu.didata.local>

Hello,

I have a CCM 5.1 cluster working with name resolution (DNS) instead of IP addresses to resolve the CCM names. Now, the IT department wants me to chenge the DNS domain where the CCMs and the phones are located (the old domain is going to dissapear).

Is there any problem in doing such action? Do you know which is the porcedure to do this?

Thanks in advance,

Ruben

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From c719562 at gmail.com  Mon May 14 07:39:02 2007
From: c719562 at gmail.com (Serg)
Date: Mon, 14 May 2007 21:39:02 +1000
Subject: [cisco-voip] Cisco CallManager with H.323 Gateway - Sample IOS
	Configuration
Message-ID: <f3e392860705140439y61f7bc3buf51f9d12f52a5fce@mail.gmail.com>

Hi All,
          I'm having some trouble configuring a Cisco 2610 fitted with an
NM-2V and 2 x VIC-2FXO-ME (Australia).

I'm currently using IOS 12.1.27b ENTERPRISE PLUS 3DES on the router.

I have tried to follow tried to follow the "Cisco IOS H.323 Gateway
Configuration for Use with Cisco CallManager"
(http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094636.shtml<http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094636.shtml>)
which has been written for a  T1 ISDN PRI trunk rather than a FXO or POTS
port.

I was hoping the configuration would be similar but I'm not having any luck.

Your help would be much appreciated.

Sergio

PS: here is what I have tried so far


version 12.1
no service single-slot-reload-enable
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname CBRVGR01
!
boot system flash c2600-jk2s-mz.121-27b.bin
enable password xxxxxxxxx
!
!
!
!
!
ip subnet-zero
ip domain-name mycompany.com
ip name-server 192.168.168.76
!
isdn voice-call-failure 0
!
!
!
!
voice-port 1/0/0
 echo-cancel coverage 32
 cptone AU
!
voice-port 1/0/1
!
voice-port 1/1/0
!
voice-port 1/1/1
!
voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8
!
!
dial-peer voice 7400 voip
 preference 1
 destination-pattern 74..
 dtmf-relay h245-alphanumeric
 no vad
 session target ipv4:192.168.168.105
!
dial-peer voice 74 pots
 destination-pattern 0T
 direct-inward-dial
 port 1/0/0
!
!
!
interface Ethernet0/0
 ip address 192.168.100.240 255.255.255.0
!
interface BRI0/0
 no ip address
 shutdown
!
ip classless
ip route 0.0.0.0 0.0.0.0 192.168.100.1
no ip http server
!
!
!
banner motd ^C
       ***********************************************
       *  MyCompany's VoiP PSTN Gateway - CBRVGR01  *
       ***********************************************
^C
!
line con 0
 exec-timeout 0 0
 password xxxxxxxxx
line aux 0
line vty 0 4
 exec-timeout 0 0
 password xxxxxxxxx
 login
 transport input telnet
!
end
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From patrick.diener at gmail.com  Mon May 14 07:56:29 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Mon, 14 May 2007 13:56:29 +0200
Subject: [cisco-voip] Cisco CallManager with H.323 Gateway - Sample IOS
	Configuration
In-Reply-To: <f3e392860705140439y61f7bc3buf51f9d12f52a5fce@mail.gmail.com>
References: <f3e392860705140439y61f7bc3buf51f9d12f52a5fce@mail.gmail.com>
Message-ID: <c0deb5b80705140456nf354afepde42b542b361a130@mail.gmail.com>

Hi Sergio

your config. does not look that bad... did you test the config
already? are either in or outbound calls working?
afaik do FXO Ports not support direct inward dial, so you have to dial
the extension twice or use "connection plar" on the voice port to
direct inbound calls directly to an attendant.

to debug the call routing on the ios router use "debug voice dialpeer"
or "debug dialpeer"

to understand call routing on ios platforms go to:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml

Regards
Patrick

On 5/14/07, Serg <c719562 at gmail.com> wrote:
> Hi All,
>           I'm having some trouble configuring a Cisco 2610 fitted with an
> NM-2V and 2 x VIC-2FXO-ME (Australia).
>
> I'm currently using IOS 12.1.27b ENTERPRISE PLUS 3DES on the router.
>
> I have tried to follow tried to follow the "Cisco IOS H.323 Gateway
> Configuration for Use with Cisco CallManager"
> (http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094636.shtml)
> which has been written for a  T1 ISDN PRI trunk rather than a FXO or POTS
> port.
>
> I was hoping the configuration would be similar but I'm not having any luck.
>
> Your help would be much appreciated.
>
> Sergio
>
> PS: here is what I have tried so far
>
>
> version 12.1
> no service single-slot-reload-enable
> service timestamps debug uptime
> service timestamps log uptime
> no service password-encryption
> !
> hostname CBRVGR01
> !
> boot system flash c2600-jk2s-mz.121-27b.bin
> enable password xxxxxxxxx
> !
> !
> !
> !
> !
> ip subnet-zero
> ip domain-name mycompany.com
> ip name-server 192.168.168.76
> !
> isdn voice-call-failure 0
> !
> !
> !
> !
> voice-port 1/0/0
>  echo-cancel coverage 32
>  cptone AU
> !
> voice-port 1/0/1
> !
> voice-port 1/1/0
> !
> voice-port 1/1/1
> !
> voice class codec 1
>  codec preference 1 g711alaw
>   codec preference 2 g711ulaw
>  codec preference 3 g729r8
> !
> !
> dial-peer voice 7400 voip
>  preference 1
>  destination-pattern 74..
>  dtmf-relay h245-alphanumeric
>  no vad
>  session target ipv4: 192.168.168.105
> !
> dial-peer voice 74 pots
>  destination-pattern 0T
>  direct-inward-dial
>  port 1/0/0
> !
> !
> !
> interface Ethernet0/0
>  ip address 192.168.100.240 255.255.255.0
> !
> interface BRI0/0
>  no ip address
>  shutdown
> !
> ip classless
> ip route 0.0.0.0 0.0.0.0 192.168.100.1
> no ip http server
> !
> !
> !
> banner motd ^C
>        ***********************************************
>        *  MyCompany's VoiP PSTN Gateway - CBRVGR01  *
>        ***********************************************
> ^C
> !
> line con 0
>  exec-timeout 0 0
>  password xxxxxxxxx
> line aux 0
> line vty 0 4
>  exec-timeout 0 0
>  password xxxxxxxxx
>  login
>  transport input telnet
> !
> end
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From aelnagar at ACT-EG.COM  Mon May 14 08:44:20 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Mon, 14 May 2007 15:44:20 +0300
Subject: [cisco-voip] Changing DNS domain in CCM environment
References: <f27p4r$6eu$1@sea.gmane.org>
	<FA233791F398B848B978464507739E8F3E4256@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A068@mail.ACT-EG.COM>

very simple I think, just change the DHCP option DNS to the new DNS server IP address.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ruben Montes (Europe)
Sent: Mon 14-May-07 11:39 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Changing DNS domain in CCM environment



Hello,

I have a CCM 5.1 cluster working with name resolution (DNS) instead of IP addresses to resolve the CCM names. Now, the IT department wants me to chenge the DNS domain where the CCMs and the phones are located (the old domain is going to dissapear).

Is there any problem in doing such action? Do you know which is the porcedure to do this?

Thanks in advance,

Ruben

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From MillerS at DicksteinShapiro.COM  Mon May 14 09:55:09 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 14 May 2007 09:55:09 -0400
Subject: [cisco-voip] Changing User Extensions and User Names in the CRS
 Administration Pages
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538975@DCEX2.DSMO.COM>

For some reason, I am having a hard time finding the interface that
allows name changes and extension changes for resources.  Can anyone
point me in the right direction?
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
==============================================================================
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From lelio at uoguelph.ca  Mon May 14 10:37:57 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 14 May 2007 10:37:57 -0400
Subject: [cisco-voip] preemption and srst/dial-peers
Message-ID: <012601c79635$76b9ce30$47196883@cfs.uoguelph.ca>

I'd like to be able to preempt other calls while in srst mode, basically let 911 go out at all costs. I know that I can get analog ports, but I'd rather be able to just punch through a PRI channel.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From aizhar at ccbootcamp.com  Mon May 14 10:53:35 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Mon, 14 May 2007 07:53:35 -0700
Subject: [cisco-voip] Changing User Extensions and User Names in the
	CRSAdministration Pages [bcc][faked-from][html-rem]
References: <B9B5B75DF9145F44A21111DD92CC490006538975@DCEX2.DSMO.COM>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FAC@nlimail.nlidomain>

Guess your talking ipcc here, if so it is coming from the call manager.
 
Users who are marked with the ICD checkbox will appear on ipcc.
 
Hope that helps.
 
Regards,
Avner Izhar, CCIE #15999(Voice)
Network Learning Inc.
www.ccbootcamp.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Miller, Steve
Sent: Mon 5/14/2007 6:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Changing User Extensions and User Names in the CRSAdministration Pages [bcc][faked-from][html-rem]



For some reason, I am having a hard time finding the interface that
allows name changes and extension changes for resources.  Can anyone
point me in the right direction?


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com> 



--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message.

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com <http://www.dicksteinshapiro.com/> 
==============================================================================

_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip


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From ealeatherman at gmail.com  Mon May 14 11:08:04 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 14 May 2007 11:08:04 -0400
Subject: [cisco-voip] Changing User Extensions and User Names in the CRS
	Administration Pages
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538975@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490006538975@DCEX2.DSMO.COM>
Message-ID: <94a1afde0705140808iac04778tf92b85d59757556a@mail.gmail.com>

I believe these changes need to be made on callmanager administration user
directory rather than through CRS

On 5/14/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
>
>  For some reason, I am having a hard time finding the interface that
> allows name changes and extension changes for resources.  Can anyone point
> me in the right direction?
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370 Fax (202)-330-5607
> millers at dicksteinshapiro.com
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message.
>
> To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
> ==============================================================================
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From bcarter at sentinel.com  Mon May 14 11:18:12 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 14 May 2007 10:18:12 -0500
Subject: [cisco-voip] VG224 Analog Voice Cable
Message-ID: <6A0744639B86DD459433108DBF101D26016AE70A@MAIL01.sentinel.com>

My cable vendor is asking for specifications for the Analog Voice Cable
required for the VG224.  All I can find is RJ21X/CA21A with a 50-pin
amphenol connector.
 
What about Male/Female ?
 
 
 
 
Bill Carter
Sr. Business Communications Analyst
Sentinel Technologies
(O) 217.391.5015
 
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From axe7961 at hotmail.com  Mon May 14 11:20:02 2007
From: axe7961 at hotmail.com (Axel Wilzinski)
Date: Mon, 14 May 2007 11:20:02 -0400
Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
Message-ID: <BAY103-W33D04D8806988E1B74F77BA3E0@phx.gbl>

A little background:  We installed 4.2.3 about 8 months ago as part of a pilot program and have a several sites for a total of about 80-90 phones.  We will be adding our main site with approximately 400 users within the next 12 months.  We have one Call Manager Cluster that is sized for all of this along with a Unity server.Should we upgrade to the latest 5.X version and consider using SIP?  Is the upgrade a painful process and is there any long term benefit?  I am a little concerned about staying on the Windows platform and the proprietary SCCP protocol.  At the same time, I'm concerned about the reliability of 5.X and the learning curve.Any insight would be appreciated.Thanks!
_________________________________________________________________
Change is good. See what?s different about Windows Live Hotmail. 
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From lelio at uoguelph.ca  Mon May 14 11:21:44 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 14 May 2007 11:21:44 -0400
Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
References: <BAY103-W33D04D8806988E1B74F77BA3E0@phx.gbl>
Message-ID: <019e01c7963b$94dac2b0$47196883@cfs.uoguelph.ca>

I think many people are staying with SCCP when going to 5.x.x.

The issue you have here is that you can not upgrade from 4.2 to 5.x, you can only upgrade to 6.x from 4.2. Right Scott? ;)


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Axel Wilzinski 
  To: cisco-voip at puck.nether.net 
  Sent: Monday, May 14, 2007 11:20 AM
  Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?


  A little background:  We installed 4.2.3 about 8 months ago as part of a pilot program and have a several sites for a total of about 80-90 phones.  We will be adding our main site with approximately 400 users within the next 12 months.  We have one Call Manager Cluster that is sized for all of this along with a Unity server.

  Should we upgrade to the latest 5.X version and consider using SIP?  Is the upgrade a painful process and is there any long term benefit?  I am a little concerned about staying on the Windows platform and the proprietary SCCP protocol.  At the same time, I'm concerned about the reliability of 5.X and the learning curve.

  Any insight would be appreciated.

  Thanks!



------------------------------------------------------------------------------
  Change is good. See what?s different about Windows Live Hotmail. Check it out! 


------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From bms314 at gmail.com  Mon May 14 11:22:56 2007
From: bms314 at gmail.com (Brian)
Date: Mon, 14 May 2007 10:22:56 -0500
Subject: [cisco-voip] Using CAR to track incoming call information
Message-ID: <7aaaf7f0705140822l7dbda011yf227c3c5c9f1f06@mail.gmail.com>

I've got a newbie question on using CAR on CCM 4.1(3).  We'd like to be able
to create a report to find out how many calls come into our main number.
We'd also like to know how many of these calls are transferred and where
these calls are transferred to.

Does CAR support this kind of reporting?  Or do I need some third party
package to be able to parse CDR's?  Any suggestions would be greatly
appreciated.

Thanks,
Brian
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From rratliff at cisco.com  Mon May 14 11:29:14 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 14 May 2007 11:29:14 -0400
Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
In-Reply-To: <BAY103-W33D04D8806988E1B74F77BA3E0@phx.gbl>
References: <BAY103-W33D04D8806988E1B74F77BA3E0@phx.gbl>
Message-ID: <119165DF-ADBB-411E-8B7C-34DD02B5EE38@cisco.com>

For starters, as Lelio mentioned you cannot upgrade from 4.2 to 5.x.

Going forward if you are staying with Cisco phones, and using them  
with Cisco CallManager, what specifically concerns you about using  
the Cisco-proprietary SCCP protocol?
CallManager has been using SCCP to talk to phones for years now.  The  
SIP client interface is brand new in 5.x.  I'm not saying one is  
better than the other or that you will have problems running all of  
your phones on SIP loads.   I'm curious why you would prefer Cisco's  
SIP phone vs Cisco's SCCP phone.

-Ryan

On May 14, 2007, at 11:20 AM, Axel Wilzinski wrote:

A little background:  We installed 4.2.3 about 8 months ago as part  
of a pilot program and have a several sites for a total of about  
80-90 phones.  We will be adding our main site with approximately 400  
users within the next 12 months.  We have one Call Manager Cluster  
that is sized for all of this along with a Unity server.

Should we upgrade to the latest 5.X version and consider using SIP?   
Is the upgrade a painful process and is there any long term benefit?   
I am a little concerned about staying on the Windows platform and the  
proprietary SCCP protocol.  At the same time, I'm concerned about the  
reliability of 5.X and the learning curve.

Any insight would be appreciated.

Thanks!

Change is good. See what?s different about Windows Live Hotmail.  
Check it out!
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Scott.Voll at wesd.org  Mon May 14 11:29:58 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 14 May 2007 08:29:58 -0700
Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
In-Reply-To: <019e01c7963b$94dac2b0$47196883@cfs.uoguelph.ca>
References: <BAY103-W33D04D8806988E1B74F77BA3E0@phx.gbl>
	<019e01c7963b$94dac2b0$47196883@cfs.uoguelph.ca>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE76A@Apollo.wesd.org>

>From all the stuff I've seen you can only upgrade to 5.1 from 4.1.  if
your running 4.2 or 4.3 you will need to wait for 6.x.

 

I also believe Lelio is right on the SCCP.  SIP only has about 16
features so you loose functionality with SIP.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Monday, May 14, 2007 8:22 AM
To: Axel Wilzinski; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?

 

I think many people are staying with SCCP when going to 5.x.x.

 

The issue you have here is that you can not upgrade from 4.2 to 5.x, you
can only upgrade to 6.x from 4.2. Right Scott? ;)

 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

	----- Original Message ----- 

	From: Axel Wilzinski <mailto:axe7961 at hotmail.com>  

	To: cisco-voip at puck.nether.net 

	Sent: Monday, May 14, 2007 11:20 AM

	Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?

	 

	A little background:  We installed 4.2.3 about 8 months ago as
part of a pilot program and have a several sites for a total of about
80-90 phones.  We will be adding our main site with approximately 400
users within the next 12 months.  We have one Call Manager Cluster that
is sized for all of this along with a Unity server.
	
	Should we upgrade to the latest 5.X version and consider using
SIP?  Is the upgrade a painful process and is there any long term
benefit?  I am a little concerned about staying on the Windows platform
and the proprietary SCCP protocol.  At the same time, I'm concerned
about the reliability of 5.X and the learning curve.
	
	Any insight would be appreciated.
	
	Thanks!

	 

	
________________________________


	Change is good. See what's different about Windows Live Hotmail.
Check it out!
<http://www.windowslive-hotmail.com/learnmore/default.html?locale=en-us&
ocid=RMT_TAGLM_HMWL_reten_changegood_0507>  

	
________________________________


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

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From bob at smalltime.com  Mon May 14 11:34:40 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Mon, 14 May 2007 10:34:40 -0500
Subject: [cisco-voip] Migrating from G.729 Unity back to G.711
Message-ID: <46488190.9020000@smalltime.com>

Enough people are complaining about the recording quality in Unity 
(which is usually actually an issue with cell phones leaving messages 
for Unity subscribers) that we're contemplating going back to G.711 in 
Unity.

If we set the recording format on Unity back to G.711, I'm assuming that 
if bandwidth is sufficient (and we set the codec between regions to 
G.711), then that's pretty much it?  It's not going to require a 
reinstall or anything like that, will it?

Any other caveats?

From ealeatherman at gmail.com  Mon May 14 12:02:10 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 14 May 2007 12:02:10 -0400
Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
In-Reply-To: <BAY103-W33D04D8806988E1B74F77BA3E0@phx.gbl>
References: <BAY103-W33D04D8806988E1B74F77BA3E0@phx.gbl>
Message-ID: <94a1afde0705140902y29e50885y662af033ed81e703@mail.gmail.com>

I can tell you one thing I am looking forward to with the new platform OS -
installing an upgrade on a separate partition and  being able to reboot back
to the old version if something doesn't work.

On 5/14/07, Axel Wilzinski <axe7961 at hotmail.com> wrote:
>
> A little background:  We installed 4.2.3 about 8 months ago as part of a
> pilot program and have a several sites for a total of about 80-90 phones.
> We will be adding our main site with approximately 400 users within the next
> 12 months.  We have one Call Manager Cluster that is sized for all of this
> along with a Unity server.
>
> Should we upgrade to the latest 5.X version and consider using SIP?  Is
> the upgrade a painful process and is there any long term benefit?  I am a
> little concerned about staying on the Windows platform and the proprietary
> SCCP protocol.  At the same time, I'm concerned about the reliability of
> 5.X and the learning curve.
>
> Any insight would be appreciated.
>
> Thanks!
>
> ------------------------------
> Change is good. See what's different about Windows Live Hotmail. Check it
> out!<http://www.windowslive-hotmail.com/learnmore/default.html?locale=en-us&ocid=RMT_TAGLM_HMWL_reten_changegood_0507%0A%0A>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From axe7961 at hotmail.com  Mon May 14 13:50:26 2007
From: axe7961 at hotmail.com (Axel Wilzinski)
Date: Mon, 14 May 2007 13:50:26 -0400
Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
Message-ID: <BAY103-W14E769257AA81816A49758BA3E0@phx.gbl>







Well that makes the decision easy! :)As far as SCCP vs. SIP, I guess I'm just looking at industry trends more than anything.  We have been happy with SCCP for the 8 months that we have been using it.  I just want to make sure we are making the right choice going forward.  From what I can tell, the best choice right now is SCCP, which is fine.  I was also looking at the possibility of supporting 3rd party SIP devices such as conference phones, etc.My bigger concern is on the OS platform as well as learning a whole new interface.  It looks like we would have to wait until a 6.x version to come out anyway, and even then I would want to wait for all the bugs to settle...Thanks again!








> CC: cisco-voip at puck.nether.net> From: rratliff at cisco.com> Subject: Re: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?> Date: Mon, 14 May 2007 11:29:14 -0400> To: axe7961 at hotmail.com> > For starters, as Lelio mentioned you cannot upgrade from 4.2 to 5.x.> > Going forward if you are staying with Cisco phones, and using them  > with Cisco CallManager, what specifically concerns you about using  > the Cisco-proprietary SCCP protocol?> CallManager has been using SCCP to talk to phones for years now.  The  > SIP client interface is brand new in 5.x.  I'm not saying one is  > better than the other or that you will have problems running all of  > your phones on SIP loads.   I'm curious why you would prefer Cisco's  > SIP phone vs Cisco's SCCP phone.> > -Ryan> > On May 14, 2007, at 11:20 AM, Axel Wilzinski wrote:> > A little background:  We installed 4.2.3 about 8 months ago as part  > of a pilot program and have a several sites for a total of about  > 80-90 phones.  We will be adding our main site with approximately 400  > users within the next 12 months.  We have one Call Manager Cluster  > that is sized for all of this along with a Unity server.> > Should we upgrade to the latest 5.X version and consider using SIP?   > Is the upgrade a painful process and is there any long term benefit?   > I am a little concerned about staying on the Windows platform and the  > proprietary SCCP protocol.  At the same time, I'm concerned about the  > reliability of 5.X and the learning curve.> > Any insight would be appreciated.> > Thanks!> > Change is good. See what?s different about Windows Live Hotmail.  > Check it out!> _______________________________________________> cisco-voip mailing list> cisco-voip at puck.nether.net> https://puck.nether.net/mailman/listinfo/cisco-voip
_________________________________________________________________
Download Messenger. Start an i?m conversation. Support a cause. Join now.
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From Scott.Voll at wesd.org  Mon May 14 13:58:16 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 14 May 2007 10:58:16 -0700
Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
In-Reply-To: <BAY103-W14E769257AA81816A49758BA3E0@phx.gbl>
References: <BAY103-W14E769257AA81816A49758BA3E0@phx.gbl>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE772@Apollo.wesd.org>

Really the interface hasn't changed that much from what I can see.  They
make it a little easier to find things from one page.  But most all the
things are in the same place.

 

Scott

 

PS. I'm with Ed and the upgrade process on 5.x.  I'm looking forward to
upgrading this summer.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Axel Wilzinski
Sent: Monday, May 14, 2007 10:50 AM
To: rratliff at cisco.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?

 

Well that makes the decision easy! :)

As far as SCCP vs. SIP, I guess I'm just looking at industry trends more
than anything.  We have been happy with SCCP for the 8 months that we
have been using it.  I just want to make sure we are making the right
choice going forward.  From what I can tell, the best choice right now
is SCCP, which is fine.  I was also looking at the possibility of
supporting 3rd party SIP devices such as conference phones, etc.

My bigger concern is on the OS platform as well as learning a whole new
interface.  It looks like we would have to wait until a 6.x version to
come out anyway, and even then I would want to wait for all the bugs to
settle...

Thanks again!





________________________________

> CC: cisco-voip at puck.nether.net
> From: rratliff at cisco.com
> Subject: Re: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
> Date: Mon, 14 May 2007 11:29:14 -0400
> To: axe7961 at hotmail.com
> 
> For starters, as Lelio mentioned you cannot upgrade from 4.2 to 5.x.
> 
> Going forward if you are staying with Cisco phones, and using them 
> with Cisco CallManager, what specifically concerns you about using 
> the Cisco-proprietary SCCP protocol?
> CallManager has been using SCCP to talk to phones for years now. The 
> SIP client interface is brand new in 5.x. I'm not saying one is 
> better than the other or that you will have problems running all of 
> your phones on SIP loads. I'm curious why you would prefer Cisco's 
> SIP phone vs Cisco's SCCP phone.
> 
> -Ryan
> 
> On May 14, 2007, at 11:20 AM, Axel Wilzinski wrote:
> 
> A little background: We installed 4.2.3 about 8 months ago as part 
> of a pilot program and have a several sites for a total of about 
> 80-90 phones. We will be adding our main site with approximately 400 
> users within the next 12 months. We have one Call Manager Cluster 
> that is sized for all of this along with a Unity server.
> 
> Should we upgrade to the latest 5.X version and consider using SIP? 
> Is the upgrade a painful process and is there any long term benefit? 
> I am a little concerned about staying on the Windows platform and the 
> proprietary SCCP protocol. At the same time, I'm concerned about the 
> reliability of 5.X and the learning curve.
> 
> Any insight would be appreciated.
> 
> Thanks!
> 
> Change is good. See what's different about Windows Live Hotmail. 
> Check it out!
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

 

________________________________

Download Messenger. Start an i'm conversation. Support a cause. Join
Now! <http://im.live.com/messenger/im/home/?source=TAGWL_MAY07> 

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From csvoip at googlemail.com  Mon May 14 14:24:45 2007
From: csvoip at googlemail.com (Nick)
Date: Mon, 14 May 2007 19:24:45 +0100
Subject: [cisco-voip] Presence Status between 2 x 5.1 clusters
Message-ID: <475fda330705141124i4153a0e5o59249bceb2f9a74b@mail.gmail.com>

I have 2 5.1 clusters joined by an inter cluster trunk both with Presence
enabled for all phones in Corporate directory and call lists, is it
possible for the Presence subscription to work across the ICT so I can se
the status of users on the other cluster?

I tried setting up a SIP trunk instead of the H323 trunk but I was unable to
add this to the Route group or do I need both together?
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From erickbe at yahoo.com  Mon May 14 14:42:11 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon, 14 May 2007 11:42:11 -0700 (PDT)
Subject: [cisco-voip] Unity 4.2(1) accounts vanishing on their own
Message-ID: <97462.20885.qm@web37210.mail.mud.yahoo.com>

Anyone ever see unity accounts in an unified setup just go away on their own?  

The exchange mailbox and AD account are there still but the Unity account vanishes out of the blue. We're seeing this on 2 seperate 4.2(1) servers and have been working with TAC. No errors in the event logs indicating a problem, etc.

Regards, Erick




      ____________________________________________________________________________________
Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center.
http://autos.yahoo.com/green_center/ 

From lelio at uoguelph.ca  Mon May 14 15:04:51 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 14 May 2007 15:04:51 -0400
Subject: [cisco-voip] 11D->10D discard : how does it affect 10 digit calls?
Message-ID: <00cc01c7965a$c0306a90$47196883@cfs.uoguelph.ca>

I'd like to have one route pattern only which uses a route filter to capture eligible tollbypass calls. 

If my route filter has an "discard digits: 11D->10D" will 10 digit calls proceed unaffected?

I'm going to test this out, but just thought I'd ask the list to see if others are doing something similar.

It will cut the number of route patterns I need in half.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From JWilliams at ci.carson-city.nv.us  Mon May 14 14:42:40 2007
From: JWilliams at ci.carson-city.nv.us (James Williams)
Date: Mon, 14 May 2007 11:42:40 -0700
Subject: [cisco-voip] CCM 4.2 and Off-Hook
Message-ID: <46484B66.D6AB.003D.0@ci.carson-city.nv.us>

I'm looking for a way to discover the status of all phones in a CCM4.2 cluster (primarily on/off hook, although information on who they're talking to, etc. would be great if available). Is this information just kept in memory on the server (i.e. inaccessible from another program), in the main database somewhere, or is there another location this information is stored? Obviously, the quicker this information updates the better, but the delay can be as great as 15-20 seconds after the status change without much of a problem. Does anyone know where I might be able to get this information?

Thanks in advance,

-James



From WOCAMPO at broward.org  Mon May 14 15:34:31 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Mon, 14 May 2007 15:34:31 -0400
Subject: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE772@Apollo.wesd.org>
References: <BAY103-W14E769257AA81816A49758BA3E0@phx.gbl>
	<407055A92CECCB499C922A2D35FC19A608DEE772@Apollo.wesd.org>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA2F3@UVEXVS01.bc.broward.cty>

I was in a training last week  for call manager 5.1.  I spoke with one
of the Cisco engineers who said Call Manager version 6.0 has been the
most tested one. 

I noticed also that SIP still has some issues that need to be fixed,
especially when they are in SRST mode.

I will also stick with SCCP .

 

 

 

Walter Ocampo

System Network Analyst

Office of Information Technology

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, May 14, 2007 1:58 PM
To: Axel Wilzinski; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?

 

Really the interface hasn't changed that much from what I can see.  They
make it a little easier to find things from one page.  But most all the
things are in the same place.

 

Scott

 

PS. I'm with Ed and the upgrade process on 5.x.  I'm looking forward to
upgrading this summer.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Axel Wilzinski
Sent: Monday, May 14, 2007 10:50 AM
To: rratliff at cisco.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?

 

Well that makes the decision easy! :)

As far as SCCP vs. SIP, I guess I'm just looking at industry trends more
than anything.  We have been happy with SCCP for the 8 months that we
have been using it.  I just want to make sure we are making the right
choice going forward.  From what I can tell, the best choice right now
is SCCP, which is fine.  I was also looking at the possibility of
supporting 3rd party SIP devices such as conference phones, etc.

My bigger concern is on the OS platform as well as learning a whole new
interface.  It looks like we would have to wait until a 6.x version to
come out anyway, and even then I would want to wait for all the bugs to
settle...

Thanks again!

 

________________________________

> CC: cisco-voip at puck.nether.net
> From: rratliff at cisco.com
> Subject: Re: [cisco-voip] Would you upgrade from 4.2.3 to 5.xx?
> Date: Mon, 14 May 2007 11:29:14 -0400
> To: axe7961 at hotmail.com
> 
> For starters, as Lelio mentioned you cannot upgrade from 4.2 to 5.x.
> 
> Going forward if you are staying with Cisco phones, and using them 
> with Cisco CallManager, what specifically concerns you about using 
> the Cisco-proprietary SCCP protocol?
> CallManager has been using SCCP to talk to phones for years now. The 
> SIP client interface is brand new in 5.x. I'm not saying one is 
> better than the other or that you will have problems running all of 
> your phones on SIP loads. I'm curious why you would prefer Cisco's 
> SIP phone vs Cisco's SCCP phone.
> 
> -Ryan
> 
> On May 14, 2007, at 11:20 AM, Axel Wilzinski wrote:
> 
> A little background: We installed 4.2.3 about 8 months ago as part 
> of a pilot program and have a several sites for a total of about 
> 80-90 phones. We will be adding our main site with approximately 400 
> users within the next 12 months. We have one Call Manager Cluster 
> that is sized for all of this along with a Unity server.
> 
> Should we upgrade to the latest 5.X version and consider using SIP? 
> Is the upgrade a painful process and is there any long term benefit? 
> I am a little concerned about staying on the Windows platform and the 
> proprietary SCCP protocol. At the same time, I'm concerned about the 
> reliability of 5.X and the learning curve.
> 
> Any insight would be appreciated.
> 
> Thanks!
> 
> Change is good. See what's different about Windows Live Hotmail. 
> Check it out!
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

 

________________________________

Download Messenger. Start an i'm conversation. Support a cause. Join
Now! <http://im.live.com/messenger/im/home/?source=TAGWL_MAY07> 

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From ealeatherman at gmail.com  Mon May 14 16:13:45 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 14 May 2007 16:13:45 -0400
Subject: [cisco-voip] Unity 4.2(1) accounts vanishing on their own
In-Reply-To: <97462.20885.qm@web37210.mail.mud.yahoo.com>
References: <97462.20885.qm@web37210.mail.mud.yahoo.com>
Message-ID: <94a1afde0705141313n65e48c71lf4633c5227100372@mail.gmail.com>

Only time I've seen it was when the exchange mailbox or AD account were
deleted.

On 5/14/07, Erick Bergquist <erickbe at yahoo.com> wrote:
>
> Anyone ever see unity accounts in an unified setup just go away on their
> own?
>
> The exchange mailbox and AD account are there still but the Unity account
> vanishes out of the blue. We're seeing this on 2 seperate 4.2(1) servers
> and have been working with TAC. No errors in the event logs indicating a
> problem, etc.
>
> Regards, Erick
>
>
>
>
>
>       ____________________________________________________________________________________
> Park yourself in front of a world of choices in alternative vehicles.
> Visit the Yahoo! Auto Green Center.
> http://autos.yahoo.com/green_center/
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From MillerS at DicksteinShapiro.COM  Mon May 14 16:15:07 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 14 May 2007 16:15:07 -0400
Subject: [cisco-voip] Changing User Extensions and User Names in the
 CRSAdministration Pages [bcc][faked-from][html-rem]
References: <B9B5B75DF9145F44A21111DD92CC490006538975@DCEX2.DSMO.COM>
	<E76ACA8C13EF1847BC69538B522C3663560FAC@nlimail.nlidomain>
Message-ID: <B9B5B75DF9145F44A21111DD92CC49000653899D@DCEX2.DSMO.COM>

What is the ICD checkbox?  In the Global Directory?
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Monday, May 14, 2007 10:54 AM
To: Miller, Steve; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]


Guess your talking ipcc here, if so it is coming from the call manager.
 
Users who are marked with the ICD checkbox will appear on ipcc.
 
Hope that helps.
 
Regards,
Avner Izhar, CCIE #15999(Voice)
Network Learning Inc.
www.ccbootcamp.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Miller, Steve
Sent: Mon 5/14/2007 6:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]



For some reason, I am having a hard time finding the interface that
allows name changes and extension changes for resources.  Can anyone
point me in the right direction?


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com> 



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have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message.

To reply to our email administrator directly, send an email to
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http://www.DicksteinShapiro.com <http://www.dicksteinshapiro.com/> 
========================================================================
======

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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



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This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

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From ealeatherman at gmail.com  Mon May 14 16:16:47 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 14 May 2007 16:16:47 -0400
Subject: [cisco-voip] CCM 4.2 and Off-Hook
In-Reply-To: <46484B66.D6AB.003D.0@ci.carson-city.nv.us>
References: <46484B66.D6AB.003D.0@ci.carson-city.nv.us>
Message-ID: <94a1afde0705141316l3987f354p80075dd5553db31b@mail.gmail.com>

How would you need this information, and do you need it for all phones at
once or just arbitrary endpoints? Attendant Console can show you line state
of a desired IP Phone on the system, although it doesn't show any call
details. The attendant just has to locate them in the directory.

On 5/14/07, James Williams <JWilliams at ci.carson-city.nv.us> wrote:
>
> I'm looking for a way to discover the status of all phones in a CCM4.2cluster (primarily on/off hook, although information on who they're talking
> to, etc. would be great if available). Is this information just kept in
> memory on the server (i.e. inaccessible from another program), in the main
> database somewhere, or is there another location this information is stored?
> Obviously, the quicker this information updates the better, but the delay
> can be as great as 15-20 seconds after the status change without much of a
> problem. Does anyone know where I might be able to get this information?
>
> Thanks in advance,
>
> -James
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Mon May 14 16:22:30 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 14 May 2007 16:22:30 -0400
Subject: [cisco-voip] Using CAR to track incoming call information
In-Reply-To: <7aaaf7f0705140822l7dbda011yf227c3c5c9f1f06@mail.gmail.com>
References: <7aaaf7f0705140822l7dbda011yf227c3c5c9f1f06@mail.gmail.com>
Message-ID: <94a1afde0705141322m441df7c2ra710db130e73d820@mail.gmail.com>

I don't think CAR can give you want you are looking for, it seems to be
primarily setup for outbound call billing. You could do a SQL query on your
database to get a number of inbound calls. I'm not sure how you would find
which calls were transfered though from CDR info, would be interesting to
see how if it's possible.



On 5/14/07, Brian <bms314 at gmail.com> wrote:
>
> I've got a newbie question on using CAR on CCM 4.1(3).  We'd like to be
> able to create a report to find out how many calls come into our main
> number.  We'd also like to know how many of these calls are transferred and
> where these calls are transferred to.
>
> Does CAR support this kind of reporting?  Or do I need some third party
> package to be able to parse CDR's?  Any suggestions would be greatly
> appreciated.
>
> Thanks,
> Brian
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From JWilliams at ci.carson-city.nv.us  Mon May 14 16:22:37 2007
From: JWilliams at ci.carson-city.nv.us (James Williams)
Date: Mon, 14 May 2007 13:22:37 -0700
Subject: [cisco-voip] CCM 4.2 and Off-Hook
In-Reply-To: <94a1afde0705141316l3987f354p80075dd5553db31b@mail.gmail.com>
References: <46484B66.D6AB.003D.0@ci.carson-city.nv.us>
	<94a1afde0705141316l3987f354p80075dd5553db31b@mail.gmail.com>
Message-ID: <464862CC.D6AB.003D.0@ci.carson-city.nv.us>

> How would you need this information, and do you need it for all phones at
> once or just arbitrary endpoints? Attendant Console can show you line state
> of a desired IP Phone on the system, although it doesn't show any call
> details. The attendant just has to locate them in the directory.

The format the information is in is more or less irrelevant, as it's going to be going to another program, where I can have it parse whatever CCM will give it.

I realize Attendant Console provides this information, but I need to get it another way so I can integrate that information with some other data. If I can use the UDP transmissions that AC puts out, that would work, but I'd need to know how the two communicate so I can "spoof" the fact that I'm not really an AC client. Does that make sense?

-James




From Brian.Henry at apptis.com  Mon May 14 16:25:29 2007
From: Brian.Henry at apptis.com (Brian Henry)
Date: Mon, 14 May 2007 16:25:29 -0400
Subject: [cisco-voip] MGCP and FXS Ports tied to RightFax
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6C1@CH-EXCH-02.intelltech.apptis.inc>

I have a CCM 4.3(1) with a 2821 running 12.4(11)T MGCP controlled.  There are 3 PRI's coming into the system and 12 FXS ports tied to Caparis RightFax Server 9.X. 
Calls come into CCM just fine and all call routing works even to the RightFax server through RL-->RG configuration that I have route patterns pointing to from within CCM.
 
Call Flow goes PSTN-->2821-MGCP(PRI)-->CCM(4.3(1))-->2821-MGCP(FXS Ports)-->Caparis-RightFax.
 
The faxes are getting to the server, however it is placing them in a default account because there is no DTMF being passed to it according to running the card in debug mode.  So no ANI nor DNIS.
 
I was thinking of going to H323 to make this work if needed but would like to stick with MGCP if possible.
 
Should I switch the "MGCP" command from dtmf out of band to other parameters
or
Will I have to enable Fax-Relay and not passthrough to get this to see the DTMF digits ANI/DNIS?
 
Thanks
 
Brian
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From d4rkf1ber at gmail.com  Mon May 14 16:30:53 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Mon, 14 May 2007 15:30:53 -0500
Subject: [cisco-voip] Transmitting ALI data from Emergency Responder?
Message-ID: <a1b07ea60705141330x10c30818sfca9a756e8bc5042@mail.gmail.com>

I am curious to find out what others have done in regards to
leveraging emergency responder and more specifically working through
the process for transmitting your ALI info.

Personally I have not had a lot of luck in determing the process from
my service provider.

Like many who have CallManager we have PRI's coming into a main
office, we have satellite offices in many different cities with
different PSAP's and such.  The users have phone numbers tied to a DID
range tied to the PRI's and well 911 is an issue.

We plan to leverage ER to identify phones in those offices and tie
them to an ELIN associated with the address of that office.  Nothing
too detailed or granular.

The primary issue at this point is getting info to the service
provider I guess.  I don't think we are suppose to deal with
individual PSAP's.  My understanding is we are suppose to send info to
the service provider of our PRI's, who is suppose to send the info or
pass it on to PSAPs I guess.

Just looking to hear what others have done regarding this process, the
trials and tribulations if you will of figuring out this aspect of the
whole thing, it does seem to be the most important component of
rolling out ER.

later

From Scott.Voll at wesd.org  Mon May 14 18:00:38 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 14 May 2007 15:00:38 -0700
Subject: [cisco-voip] Changing User Extensions and User Names in the
	CRSAdministration Pages [bcc][faked-from][html-rem]
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC49000653899D@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490006538975@DCEX2.DSMO.COM><E76ACA8C13EF1847BC69538B522C3663560FAC@nlimail.nlidomain>
	<B9B5B75DF9145F44A21111DD92CC49000653899D@DCEX2.DSMO.COM>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE777@Apollo.wesd.org>

Yes.  Go into ccmadmin --> users --> search for user -->  assocate phone
--> you will see set as primary and ICD.  Check this box. And click
update.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, May 14, 2007 1:15 PM
To: Avner Izhar; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

 

What is the ICD checkbox?  In the Global Directory?

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

 

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Monday, May 14, 2007 10:54 AM
To: Miller, Steve; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

Guess your talking ipcc here, if so it is coming from the call manager.

 

Users who are marked with the ICD checkbox will appear on ipcc.

 

Hope that helps.

 

Regards,

Avner Izhar, CCIE #15999(Voice)

Network Learning Inc.

www.ccbootcamp.com

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Miller, Steve
Sent: Mon 5/14/2007 6:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

For some reason, I am having a hard time finding the interface that
allows name changes and extension changes for resources.  Can anyone
point me in the right direction?


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com> 



--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
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the sender immediately by reply e-mail message and permanently delete
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========================================================================
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
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any review, use, dissemination, forwarding, printing, copying, or other
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From ANDY.WEBSTER at illinois.gov  Mon May 14 18:13:23 2007
From: ANDY.WEBSTER at illinois.gov (Webster, Andy)
Date: Mon, 14 May 2007 17:13:23 -0500
Subject: [cisco-voip] FW: 4.1 to 5.0 on MCS7835 hardware
References: <B42B3AE11A6CB7439C48FF2B866A7CFE01580074@IL084EX112.illinois.gov>
Message-ID: <B42B3AE11A6CB7439C48FF2B866A7CFE16FD74@IL084EX112.illinois.gov>



 
Hi,

            I have two servers MCS7835-1266-TD.  They're on the list http://www.cisco.com/en/US/customer/products/hw/voiceapp/ps378/prod_brochure0900aecd8062a4f9.html as being unable to upgrade to 5.0 or 6.0.

            Does that mean there is something in the install that won't work?  Or will I just not have support from cisco once I upgrade the servers?  Long term I probably need a hardware upgrade, but will this upgrade work short term?  I have a very low number of phones in use on these servers.

 

Thanks,

Andy

 


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From Scott.Voll at wesd.org  Mon May 14 19:05:42 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 14 May 2007 16:05:42 -0700
Subject: [cisco-voip] FW: 4.1 to 5.0 on MCS7835 hardware
In-Reply-To: <B42B3AE11A6CB7439C48FF2B866A7CFE16FD74@IL084EX112.illinois.gov>
References: <B42B3AE11A6CB7439C48FF2B866A7CFE01580074@IL084EX112.illinois.gov>
	<B42B3AE11A6CB7439C48FF2B866A7CFE16FD74@IL084EX112.illinois.gov>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE779@Apollo.wesd.org>

You have to have a minimum of 2 gig's of ram......... you have to have a
minimum of 72 gig hard drives............. You need a CPU that is twice
the horsepower of a 1266.  I would say NO.  it just flat out will not
work. Or be supported for that matter.

 

If you wanted to spend the money on the upgrade I would put it into new
HP servers.  (next time I upgrade hw it will not be MCS.... They will be
HP)

 

Just my 2 cents.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Webster, Andy
Sent: Monday, May 14, 2007 3:13 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] FW: 4.1 to 5.0 on MCS7835 hardware

 

 


Hi,

            I have two servers MCS7835-1266-TD.  They're on the list
http://www.cisco.com/en/US/customer/products/hw/voiceapp/ps378/prod_broc
hure0900aecd8062a4f9.html as being unable to upgrade to 5.0 or 6.0.

            Does that mean there is something in the install that won't
work?  Or will I just not have support from cisco once I upgrade the
servers?  Long term I probably need a hardware upgrade, but will this
upgrade work short term?  I have a very low number of phones in use on
these servers.



Thanks,

Andy





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From fwn at feasible.net  Mon May 14 19:44:04 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Mon, 14 May 2007 16:44:04 -0700
Subject: [cisco-voip] FW: 4.1 to 5.0 on MCS7835 hardware
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE779@Apollo.wesd.org>
References: <B42B3AE11A6CB7439C48FF2B866A7CFE01580074@IL084EX112.illinois.gov>
	<B42B3AE11A6CB7439C48FF2B866A7CFE16FD74@IL084EX112.illinois.gov>
	<407055A92CECCB499C922A2D35FC19A608DEE779@Apollo.wesd.org>
Message-ID: <ca09e3750705141644o1c97c2d4o505f051caeea3b88@mail.gmail.com>

Agree.  The only way to install on a system that does not meet the minimum
spec's mentioned by Scott is to either install on VMware or hack the
installer itself.

-Fred Nielsen

On 5/14/07, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>  You have to have a minimum of 2 gig's of ram??? you have to have a
> minimum of 72 gig hard drives????. You need a CPU that is twice the
> horsepower of a 1266.  I would say NO.  it just flat out will not work. Or
> be supported for that matter.
>
>
>
> If you wanted to spend the money on the upgrade I would put it into new HP
> servers.  (next time I upgrade hw it will not be MCS?. They will be HP)
>
>
>
> Just my 2 cents.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Webster, Andy
> *Sent:* Monday, May 14, 2007 3:13 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] FW: 4.1 to 5.0 on MCS7835 hardware
>
>
>
>
>
>
> Hi,
>
>             I have two servers MCS7835-1266-TD.  They're on the list
> http://www.cisco.com/en/US/customer/products/hw/voiceapp/ps378/prod_brochure0900aecd8062a4f9.htmlas being unable to upgrade to
> 5.0 or 6.0.
>
>             Does that mean there is something in the install that won't
> work?  Or will I just not have support from cisco once I upgrade the
> servers?  Long term I probably need a hardware upgrade, but will this
> upgrade work short term?  I have a very low number of phones in use on these
> servers.
>
>
>
> Thanks,
>
> Andy
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From lelio at uoguelph.ca  Mon May 14 22:00:40 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 14 May 2007 22:00:40 -0400
Subject: [cisco-voip] ccm-mananger config - getting along without it
Message-ID: <00d101c79694$d7363a50$c901a8c0@pdp11>

It looks like the "ccm-manager config" statement is overwriting my trunk group assignment under the controller T1 interface, soooooooo, it looks like I have to give it up for the manual approach.

Can anyone comment on what I lose and how I get it back with going with the manual configs? I know that I can issue the ccm-manager config statement and then just remove it and things work, but as I go forward, what should I look out for?

When I look at the xml file on the tftp server, there is quite a bit of information there that I don't see in the config anywhere.

Comments? Pointers to documentation?

Thanks, Lelio

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

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From holman at post.harvard.edu  Mon May 14 22:23:57 2007
From: holman at post.harvard.edu (Candace Holman)
Date: Mon, 14 May 2007 22:23:57 -0400
Subject: [cisco-voip] Transmitting ALI data from Emergency Responder?
Message-ID: <81746c9a0705141923ude3f4f9obf5bec4a328bacdc@mail.gmail.com>

It's a bit off topic for this list but may be the only place where hands-on
VoIP/E911 projects might be discussed, so I'd like to hear what others say
too.

There are whole companies devoted to this task and you are right, it is a
major task and can be a hassle.  It's going to depend on the state-supported
E911 infrastructure you deal with, so look up each state you need to do this
for.  A "selective router" in the e911 infrastructure routes calls to the
proper PSAP and each PSAP has a PS/ALI database.  For example, in
Massachusetts there are 2 selective routers and 278 PSAPs.  The MA PSAPs
have a deal with Verizon to be the only PS/ALI uploader so we had to do our
data uploads through Verizon.  We had to use Kermit over a modem, but I
believe they are updating this process.  You will need to press your service
provider for information, but you have to know the right questions to ask.

We only had to do two PSAPs. Since each state's E911 is run independently,
you're going to have to find out what needs to be done for your case/s.
Within CER you generate the upload files, and may need to tweak them for the
PS/ALI formatting (hopefully not too much). Hint - generate one of each NENA
format and you'll be ready when they tell you you sent it in the wrong
format.

At some point you will need to deal with each PSAP to do testing, so get
prepared and know which ones you'll need to contact.  As a professional
courtesy, you don't want to dial 911 for testing without calling ahead
first.

Here's a Vz link that I think someone on this list posted recently.  It
shows what you should be looking for from your provider, and lets you start
building a list of questions to ask:
http://www22.verizon.com/wholesale/local/E911/1,21070,,00.html

Also check out the NENA website and when you think you need a break from it
all lookup the Next Gen 911 (ECRIT) project being defined in the IETF and
googlable in combination with Henning Schulzrinne's name.  I would be
surprised if Cisco were not all over this at some point in the next year or
two.

Candace

From: "Dark Fiber" <d4rkf1ber at gmail.com>
To: cisco-voip at puck.nether.net
Date: Mon, 14 May 2007 15:30:53 -0500

I am curious to find out what others have done in regards to
leveraging emergency responder and more specifically working through
the process for transmitting your ALI info.

Personally I have not had a lot of luck in determing the process from
my service provider.

Like many who have CallManager we have PRI's coming into a main
office, we have satellite offices in many different cities with
different PSAP's and such.  The users have phone numbers tied to a DID
range tied to the PRI's and well 911 is an issue.

We plan to leverage ER to identify phones in those offices and tie
them to an ELIN associated with the address of that office.  Nothing
too detailed or granular.

The primary issue at this point is getting info to the service
provider I guess.  I don't think we are suppose to deal with
individual PSAP's.  My understanding is we are suppose to send info to
the service provider of our PRI's, who is suppose to send the info or
pass it on to PSAPs I guess.

Just looking to hear what others have done regarding this process, the
trials and tribulations if you will of figuring out this aspect of the
whole thing, it does seem to be the most important component of
rolling out ER.

later
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From MillerS at DicksteinShapiro.COM  Mon May 14 22:36:11 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 14 May 2007 22:36:11 -0400
Subject: [cisco-voip] Changing User Extensions and User Names in the
 CRSAdministration Pages [bcc][faked-from][html-rem]
References: <407055A92CECCB499C922A2D35FC19A608DEE777@Apollo.wesd.org>
Message-ID: <B9B5B75DF9145F44A21111DD92CC4900065389A4@DCEX2.DSMO.COM>

OK.  That's good.  Thank you!    But, how do you change the name of the
"resource" on the CRS server which is associated with that ICD
extension?
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, May 14, 2007 6:01 PM
To: Miller, Steve; Avner Izhar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]



Yes.  Go into ccmadmin --> users --> search for user -->  assocate phone
--> you will see set as primary and ICD.  Check this box. And click
update.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, May 14, 2007 1:15 PM
To: Avner Izhar; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

 

What is the ICD checkbox?  In the Global Directory?

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

 

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Monday, May 14, 2007 10:54 AM
To: Miller, Steve; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

Guess your talking ipcc here, if so it is coming from the call manager.

 

Users who are marked with the ICD checkbox will appear on ipcc.

 

Hope that helps.

 

Regards,

Avner Izhar, CCIE #15999(Voice)

Network Learning Inc.

www.ccbootcamp.com

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Miller, Steve
Sent: Mon 5/14/2007 6:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

For some reason, I am having a hard time finding the interface that
allows name changes and extension changes for resources.  Can anyone
point me in the right direction?


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com> 



--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
prohibited. Dickstein Shapiro reserves the right to monitor any
communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message.

To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com <http://www.dicksteinshapiro.com/> 
========================================================================
======

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
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have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message. 
 
To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com
 
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
========================================================================
======

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

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==============================================================================
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From aizhar at ccbootcamp.com  Mon May 14 23:04:29 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Mon, 14 May 2007 20:04:29 -0700
Subject: [cisco-voip] Changing User Extensions and User Names in
	theCRSAdministration Pages
References: <B9B5B75DF9145F44A21111DD92CC4900065389A4@DCEX2.DSMO.COM>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FB4@nlimail.nlidomain>

What you need to do is to create a new user and assign that user to ICD with the extension that was used by the old user. You can't rename a user so it will have to be a delete and create procedure.
 
You should also assign the user too the correct skill or resource group in the IPCC end.
 
Detailed information for this can be found at:
http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_4_0/english/administ/crs401ag.pdf
 
Chapter four would have what you are looking for..
 
Regards,
Avner Izhar, CCIE #15999(Voice)
Network Learning Inc.
www.ccbootcamp.com


________________________________

From: Miller, Steve [mailto:MillerS at DicksteinShapiro.COM]
Sent: Mon 5/14/2007 7:36 PM
To: Voll, Scott; Avner Izhar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in theCRSAdministration Pages [bcc][faked-from][html-rem] [mx][html-rem]



OK.  That's good.  Thank you!    But, how do you change the name of the
"resource" on the CRS server which is associated with that ICD
extension?


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com> 



________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org]
Sent: Monday, May 14, 2007 6:01 PM
To: Miller, Steve; Avner Izhar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]



Yes.  Go into ccmadmin --> users --> search for user -->  assocate phone
--> you will see set as primary and ICD.  Check this box. And click
update.



Scott



________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, May 14, 2007 1:15 PM
To: Avner Izhar; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]



What is the ICD checkbox?  In the Global Directory?



Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com> 





________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com]
Sent: Monday, May 14, 2007 10:54 AM
To: Miller, Steve; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

Guess your talking ipcc here, if so it is coming from the call manager.



Users who are marked with the ICD checkbox will appear on ipcc.



Hope that helps.



Regards,

Avner Izhar, CCIE #15999(Voice)

Network Learning Inc.

www.ccbootcamp.com



________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Miller, Steve
Sent: Mon 5/14/2007 6:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

For some reason, I am having a hard time finding the interface that
allows name changes and extension changes for resources.  Can anyone
point me in the right direction?


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>



--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
prohibited. Dickstein Shapiro reserves the right to monitor any
communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message.

To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com <http://www.dicksteinshapiro.com/>  <http://www.dicksteinshapiro.com/>
========================================================================
======

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
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intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
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communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message.

To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com <http://www.dicksteinshapiro.com/> 
========================================================================
======

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message.

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com <http://www.dicksteinshapiro.com/> 
==============================================================================


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From MillerS at DicksteinShapiro.COM  Mon May 14 23:22:17 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 14 May 2007 23:22:17 -0400
Subject: [cisco-voip] Changing User Extensions and User Names in
 theCRSAdministration Pages
References: <B9B5B75DF9145F44A21111DD92CC4900065389A4@DCEX2.DSMO.COM>
	<E76ACA8C13EF1847BC69538B522C3663560FB4@nlimail.nlidomain>
Message-ID: <B9B5B75DF9145F44A21111DD92CC4900065389A5@DCEX2.DSMO.COM>

Thank you!
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Monday, May 14, 2007 11:04 PM
To: Miller, Steve; Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in
theCRSAdministration Pages


What you need to do is to create a new user and assign that user to ICD
with the extension that was used by the old user. You can't rename a
user so it will have to be a delete and create procedure.
 
You should also assign the user too the correct skill or resource group
in the IPCC end.
 
Detailed information for this can be found at:
http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/apps_4_0/
english/administ/crs401ag.pdf
 
Chapter four would have what you are looking for..
 
Regards,
Avner Izhar, CCIE #15999(Voice)
Network Learning Inc.
www.ccbootcamp.com


________________________________

From: Miller, Steve [mailto:MillerS at DicksteinShapiro.COM]
Sent: Mon 5/14/2007 7:36 PM
To: Voll, Scott; Avner Izhar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in
theCRSAdministration Pages [bcc][faked-from][html-rem] [mx][html-rem]



OK.  That's good.  Thank you!    But, how do you change the name of the
"resource" on the CRS server which is associated with that ICD
extension?


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com> 



________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org]
Sent: Monday, May 14, 2007 6:01 PM
To: Miller, Steve; Avner Izhar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]



Yes.  Go into ccmadmin --> users --> search for user -->  assocate phone
--> you will see set as primary and ICD.  Check this box. And click
update.



Scott



________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, May 14, 2007 1:15 PM
To: Avner Izhar; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]



What is the ICD checkbox?  In the Global Directory?



Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com> 





________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com]
Sent: Monday, May 14, 2007 10:54 AM
To: Miller, Steve; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

Guess your talking ipcc here, if so it is coming from the call manager.



Users who are marked with the ICD checkbox will appear on ipcc.



Hope that helps.



Regards,

Avner Izhar, CCIE #15999(Voice)

Network Learning Inc.

www.ccbootcamp.com



________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Miller, Steve
Sent: Mon 5/14/2007 6:55 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Changing User Extensions and User Names in the
CRSAdministration Pages [bcc][faked-from][html-rem]

For some reason, I am having a hard time finding the interface that
allows name changes and extension changes for resources.  Can anyone
point me in the right direction?


Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>



--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
prohibited. Dickstein Shapiro reserves the right to monitor any
communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message.

To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com <http://www.dicksteinshapiro.com/>
<http://www.dicksteinshapiro.com/>
========================================================================
======

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

--------------------------------------------------------
This e-mail message and any attached files are confidential and are
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person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
prohibited. Dickstein Shapiro reserves the right to monitor any
communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message.

To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com <http://www.dicksteinshapiro.com/> 
========================================================================
======

--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
prohibited. Dickstein Shapiro reserves the right to monitor any
communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message.

To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com <http://www.dicksteinshapiro.com/> 
========================================================================
======



--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
==============================================================================
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From fwn at feasible.net  Mon May 14 23:44:51 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Mon, 14 May 2007 20:44:51 -0700
Subject: [cisco-voip] ccm-mananger config - getting along without it
In-Reply-To: <00d101c79694$d7363a50$c901a8c0@pdp11>
References: <00d101c79694$d7363a50$c901a8c0@pdp11>
Message-ID: <ca09e3750705142044g64b29f49jffd5a837ab562003@mail.gmail.com>

I realize this is not all the information you are looking for... but I never
use that command - even during initial configuration.  I do not like the
some of the changes it tends to apply to router configs, I prefer to be more
explicit and know what I am configuring.  The only caveat I do not like is
that to reset an ISDN interface (truly, meaning drop the D-channel and
reestablish) you have to either "no mgcp" the router or shut/unshut the
interface.  When you have the ccm-manager config command in you can do this
via Reset from CM.

On the flip side, there are a number of customers who benefit from this
approach (i.e. telecom admins who want nothing to do with IOS)... but being
an IOS guy I still dislike it.

-Fred Nielsen

On 5/14/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

>  It looks like the "ccm-manager config" statement is overwriting my trunk
> group assignment under the controller T1 interface, soooooooo, it looks like
> I have to give it up for the manual approach.
>
> Can anyone comment on what I lose and how I get it back with going with
> the manual configs? I know that I can issue the ccm-manager config statement
> and then just remove it and things work, but as I go forward, what should I
> look out for?
>
> When I look at the xml file on the tftp server, there is quite a bit of
> information there that I don't see in the config anywhere.
>
> Comments? Pointers to documentation?
>
> Thanks, Lelio
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From kds20850 at yahoo.com  Tue May 15 00:46:50 2007
From: kds20850 at yahoo.com (kevin k)
Date: Mon, 14 May 2007 21:46:50 -0700 (PDT)
Subject: [cisco-voip] voice monitoring tool
Message-ID: <459516.38525.qm@web57507.mail.re1.yahoo.com>

i want to monitor voice network.
what tool do you recommend?

thanks in advance.


       
____________________________________________________________________________________Luggage? GPS? Comic books? 
Check out fitting gifts for grads at Yahoo! Search
http://search.yahoo.com/search?fr=oni_on_mail&p=graduation+gifts&cs=bz
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From lemon at lemon.za.net  Tue May 15 04:00:58 2007
From: lemon at lemon.za.net (Lemon)
Date: Tue, 15 May 2007 10:00:58 +0200
Subject: [cisco-voip] Extension Mobility
Message-ID: <94e031d30705150100i722a95feg80e628e60946abd1@mail.gmail.com>

Hi, we have the following setup:

CCM 4.2 (1) SR1b with PUB and SUB that's been running for about 6 months now
with no problems. Now we are deploying more phones, 7911G and 7941G, and
even though we can subscribe these phones to the Login services, we we try
to login on the phones we get "host not found". This only applies to these
new 7911G and 7941G's. We deployed 9785's with no problems.

To get around this problem, we stop and start the TFTP services and that
seems to fix it for now.

Anyone come across this?

Thnx

C
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From jeremie.ceintrey at gmail.com  Tue May 15 04:37:21 2007
From: jeremie.ceintrey at gmail.com (Jeremie Ceintrey)
Date: Tue, 15 May 2007 10:37:21 +0200
Subject: [cisco-voip] ST 2030 SIP PHONE on Callmanager 5 and call pickup
	feature
Message-ID: <dc3a89dd0705150137n613d82cdt766d40cbe98b74ac@mail.gmail.com>

Hi,

I would like to know if anybody has success to set call pickup on st 2030
phone registered with SIP protocol on a Cisco Callmanager 5?

Thanks.
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From Johnny.Crothers at vanco.co.uk  Tue May 15 05:12:48 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers)
Date: Tue, 15 May 2007 10:12:48 +0100
Subject: [cisco-voip]  BAR's upgrade 4.0.11 to 4.0.12
Message-ID: <A56BD242744BF540B15D1672E0E3E87501038DE9@LON-INT-MSGVS01.internal.vancocorp.com>

Hey everyone.

 

Just recently upgraded BAR's due to a icompatability error I was
receiving with CAR. Although now I'm receiving this error:

 

[02:00:09 AM] Failed to copy URL Services page. Couldn't find IIS Root
Directory for CCMCIP in x.x.x.x

 

I've tried to restart the IIS service to no avail.

 

Anyone got an idea's?

 



**********************************************************************
Any opinions expressed in the email are those of the individual and not necessarily the company. This email and any files transmitted with it are confidential and solely for the use of the intended recipient.  If you are not the intended recipient or the person responsible for delivering it to the intended recipient, be advised that you have received this email in error and that any dissemination, distribution, copying or use is strictly prohibited.

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The contents of an attachment to this e-mail may contain software viruses which could damage your own computer system. While the sender has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening any attachments to this e-mail. 

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From hemant.taan at gmail.com  Tue May 15 06:05:34 2007
From: hemant.taan at gmail.com (hemant kumar)
Date: Tue, 15 May 2007 15:35:34 +0530
Subject: [cisco-voip] voice monitoring tool
In-Reply-To: <459516.38525.qm@web57507.mail.re1.yahoo.com>
References: <459516.38525.qm@web57507.mail.re1.yahoo.com>
Message-ID: <e8429ac20705150305o543559d0n16c4b844f87dea1e@mail.gmail.com>

monitor means real time monitoring ? or recording of voice calls?


On 5/15/07, kevin k <kds20850 at yahoo.com> wrote:
>
> i want to monitor voice network.
> what tool do you recommend?
>
> thanks in advance.
>
>
> ------------------------------
> Building a website is a piece of cake.
> Yahoo! Small Business gives you all the tools to get online.<http://us.rd.yahoo.com/evt=48251/*http://smallbusiness.yahoo.com/webhosting/?p=PASSPORTPLUS>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From kds20850 at yahoo.com  Tue May 15 06:59:48 2007
From: kds20850 at yahoo.com (kevin k)
Date: Tue, 15 May 2007 18:59:48 +0800 (CST)
Subject: [cisco-voip] voice monitoring tool
Message-ID: <621167.91277.qm@web57508.mail.re1.yahoo.com>

real time monitoring and baselining.
something for enterprise.

excluding RTMT.


----- Original Message ----
From: hemant kumar <hemant.taan at gmail.com>
To: kevin k <kds20850 at yahoo.com>
Cc: cisco-voip at puck.nether.net
Sent: Tuesday, May 15, 2007 6:05:34 AM
Subject: Re: [cisco-voip] voice monitoring tool

monitor means real time monitoring ? or recording of voice calls?



On 5/15/07, kevin k <kds20850 at yahoo.com > wrote:
i want to monitor voice network.
what tool do you recommend?
 
thanks in advance.
 



Building a website is a piece of cake. 
Yahoo! Small Business gives you all the tools to get online.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


       
____________________________________________________________________________________Pinpoint customers who are looking for what you sell. 
http://searchmarketing.yahoo.com/
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From ealeatherman at gmail.com  Tue May 15 08:07:27 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 15 May 2007 08:07:27 -0400
Subject: [cisco-voip] Transmitting ALI data from Emergency Responder?
In-Reply-To: <a1b07ea60705141330x10c30818sfca9a756e8bc5042@mail.gmail.com>
References: <a1b07ea60705141330x10c30818sfca9a756e8bc5042@mail.gmail.com>
Message-ID: <94a1afde0705150507w1f623dfcoa79d7b97077ea91f@mail.gmail.com>

We have PS-ALI service from verizon, which let's us update the records we
use for ELINs as needed. There is a setup fee and a monthly fee depending on
how many database records you need, and at least in our area there are a
minimum number of records you have to buy which would make it more expensive
for smaller companies. We also used to have to update these via Kermit, but
they recently joined modern times and moved to a web-based system which is
much easier to use.

This is a process that I would like to have better understanding of myself.
Between our local Verizon people and the PSAPs I dealt with, I felt that
neither had a complete grasp of what we needed when we started down this
path. We basically had to learn as we went.

On 5/14/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
>
> I am curious to find out what others have done in regards to
> leveraging emergency responder and more specifically working through
> the process for transmitting your ALI info.
>
> Personally I have not had a lot of luck in determing the process from
> my service provider.
>
> Like many who have CallManager we have PRI's coming into a main
> office, we have satellite offices in many different cities with
> different PSAP's and such.  The users have phone numbers tied to a DID
> range tied to the PRI's and well 911 is an issue.
>
> We plan to leverage ER to identify phones in those offices and tie
> them to an ELIN associated with the address of that office.  Nothing
> too detailed or granular.
>
> The primary issue at this point is getting info to the service
> provider I guess.  I don't think we are suppose to deal with
> individual PSAP's.  My understanding is we are suppose to send info to
> the service provider of our PRI's, who is suppose to send the info or
> pass it on to PSAPs I guess.
>
> Just looking to hear what others have done regarding this process, the
> trials and tribulations if you will of figuring out this aspect of the
> whole thing, it does seem to be the most important component of
> rolling out ER.
>
> later
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Tue May 15 08:11:27 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 15 May 2007 08:11:27 -0400
Subject: [cisco-voip] Transmitting ALI data from Emergency Responder?
In-Reply-To: <81746c9a0705141923ude3f4f9obf5bec4a328bacdc@mail.gmail.com>
References: <81746c9a0705141923ude3f4f9obf5bec4a328bacdc@mail.gmail.com>
Message-ID: <94a1afde0705150511m26065f91o2879f65290822c79@mail.gmail.com>

Anyone have info on a reputable company that handles PS-ALI service? If
Verizon is the only ALI uploader in our state, are we stuck with them?

I have a regional campus that needs maybe 15 ELINs, and Verizon will only
sell a minimum of 80 records, which for the size and budget of the campus is
something they are resisting paying for. We can't add them to our own PS-ALI
contract as they are on a different 911 system.

On 5/14/07, Candace Holman <holman at post.harvard.edu> wrote:
>
> It's a bit off topic for this list but may be the only place where
> hands-on VoIP/E911 projects might be discussed, so I'd like to hear what
> others say too.
>
> There are whole companies devoted to this task and you are right, it is a
> major task and can be a hassle.  It's going to depend on the state-supported
> E911 infrastructure you deal with, so look up each state you need to do this
> for.  A "selective router" in the e911 infrastructure routes calls to the
> proper PSAP and each PSAP has a PS/ALI database.  For example, in
> Massachusetts there are 2 selective routers and 278 PSAPs.  The MA PSAPs
> have a deal with Verizon to be the only PS/ALI uploader so we had to do our
> data uploads through Verizon.  We had to use Kermit over a modem, but I
> believe they are updating this process.  You will need to press your service
> provider for information, but you have to know the right questions to ask.
>
> We only had to do two PSAPs. Since each state's E911 is run independently,
> you're going to have to find out what needs to be done for your case/s.
> Within CER you generate the upload files, and may need to tweak them for the
> PS/ALI formatting (hopefully not too much). Hint - generate one of each NENA
> format and you'll be ready when they tell you you sent it in the wrong
> format.
>
> At some point you will need to deal with each PSAP to do testing, so get
> prepared and know which ones you'll need to contact.  As a professional
> courtesy, you don't want to dial 911 for testing without calling ahead
> first.
>
> Here's a Vz link that I think someone on this list posted recently.  It
> shows what you should be looking for from your provider, and lets you start
> building a list of questions to ask:
> http://www22.verizon.com/wholesale/local/E911/1,21070,,00.html
>
> Also check out the NENA website and when you think you need a break from
> it all lookup the Next Gen 911 (ECRIT) project being defined in the IETF and
> googlable in combination with Henning Schulzrinne's name.  I would be
> surprised if Cisco were not all over this at some point in the next year or
> two.
>
> Candace
>
> From: "Dark Fiber" <d4rkf1ber at gmail.com>
> To:  cisco-voip at puck.nether.net
> Date: Mon, 14 May 2007 15:30:53 -0500
>
> I am curious to find out what others have done in regards to
> leveraging emergency responder and more specifically working through
> the process for transmitting your ALI info.
>
> Personally I have not had a lot of luck in determing the process from
> my service provider.
>
> Like many who have CallManager we have PRI's coming into a main
> office, we have satellite offices in many different cities with
> different PSAP's and such.  The users have phone numbers tied to a DID
> range tied to the PRI's and well 911 is an issue.
>
> We plan to leverage ER to identify phones in those offices and tie
> them to an ELIN associated with the address of that office.  Nothing
> too detailed or granular.
>
> The primary issue at this point is getting info to the service
> provider I guess.  I don't think we are suppose to deal with
> individual PSAP's.  My understanding is we are suppose to send info to
> the service provider of our PRI's, who is suppose to send the info or
> pass it on to PSAPs I guess.
>
> Just looking to hear what others have done regarding this process, the
> trials and tribulations if you will of figuring out this aspect of the
> whole thing, it does seem to be the most important component of
> rolling out ER.
>
> later
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Tue May 15 08:15:12 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 15 May 2007 08:15:12 -0400
Subject: [cisco-voip] voice monitoring tool
In-Reply-To: <621167.91277.qm@web57508.mail.re1.yahoo.com>
References: <621167.91277.qm@web57508.mail.re1.yahoo.com>
Message-ID: <94a1afde0705150515l7e206682of0e018a3e8e281b6@mail.gmail.com>

I've heard good things about Qovia, I think Cisco actually resells some of
their network probes.




On 5/15/07, kevin k <kds20850 at yahoo.com> wrote:
>
> real time monitoring and baselining.
> something for enterprise.
>
> excluding RTMT.
>
> ----- Original Message ----
> From: hemant kumar <hemant.taan at gmail.com>
> To: kevin k <kds20850 at yahoo.com>
> Cc: cisco-voip at puck.nether.net
> Sent: Tuesday, May 15, 2007 6:05:34 AM
> Subject: Re: [cisco-voip] voice monitoring tool
>
> monitor means real time monitoring ? or recording of voice calls?
>
>
> On 5/15/07, kevin k <kds20850 at yahoo.com > wrote:
> >
> >  i want to monitor voice network.
> > what tool do you recommend?
> >
> > thanks in advance.
> >
> >
> > ------------------------------
> > Building a website is a piece of cake.
> > Yahoo! Small Business gives you all the tools to get online.<http://us.rd.yahoo.com/evt=48251/*http://smallbusiness.yahoo.com/webhosting/?p=PASSPORTPLUS>
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> ------------------------------
> Pinpoint customers
> <http://us.rd.yahoo.com/evt=48250/*http://searchmarketing.yahoo.com/arp/sponsoredsearch_v9.php?o=US2226&cmp=Yahoo&ctv=AprNI&s=Y&s2=EM&b=50>who
> are looking for what you sell.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From Brian.Henry at apptis.com  Tue May 15 08:16:43 2007
From: Brian.Henry at apptis.com (Brian Henry)
Date: Tue, 15 May 2007 08:16:43 -0400
Subject: [cisco-voip] voice monitoring tool
References: <621167.91277.qm@web57508.mail.re1.yahoo.com>
	<94a1afde0705150515l7e206682of0e018a3e8e281b6@mail.gmail.com>
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6C4@CH-EXCH-02.intelltech.apptis.inc>

yes Cisco doe rebrand some of their appliances and they can be helpful tools when trying to running down voice quality issues.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ed Leatherman
Sent: Tue 5/15/2007 8:15 AM
To: kevin k
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] voice monitoring tool


I've heard good things about Qovia, I think Cisco actually resells some of their network probes.





On 5/15/07, kevin k < kds20850 at yahoo.com <mailto:kds20850 at yahoo.com> > wrote: 

	real time monitoring and baselining.
	something for enterprise.
	 
	excluding RTMT.
	
	
	
	----- Original Message ----
	From: hemant kumar < hemant.taan at gmail.com <mailto:hemant.taan at gmail.com> >
	To: kevin k <kds20850 at yahoo.com>
	Cc: cisco-voip at puck.nether.net
	Sent: Tuesday, May 15, 2007 6:05:34 AM
	Subject: Re: [cisco-voip] voice monitoring tool
	
	monitor means real time monitoring ? or recording of voice calls?
	
	
	
	On 5/15/07, kevin k <kds20850 at yahoo.com > wrote: 

		i want to monitor voice network.
		what tool do you recommend?
		 
		thanks in advance.
		
		 
		
		
________________________________

		Building a website is a piece of cake. 
		Yahoo! Small Business gives you all the tools to get online. <http://us.rd.yahoo.com/evt=48251/*http://smallbusiness.yahoo.com/webhosting/?p=PASSPORTPLUS> 

		_______________________________________________
		cisco-voip mailing list
		cisco-voip at puck.nether.net
		https://puck.nether.net/mailman/listinfo/cisco-voip 
		
		




________________________________

	Pinpoint customers <http://us.rd.yahoo.com/evt=48250/*http://searchmarketing.yahoo.com/arp/sponsoredsearch_v9.php?o=US2226&cmp=Yahoo&ctv=AprNI&s=Y&s2=EM&b=50> who are looking for what you sell. 

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	
	




-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 
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From Johnny.Crothers at vanco.co.uk  Tue May 15 08:45:10 2007
From: Johnny.Crothers at vanco.co.uk (Johnny Crothers)
Date: Tue, 15 May 2007 13:45:10 +0100
Subject: [cisco-voip] voice monitoring tool
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6C4@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <A56BD242744BF540B15D1672E0E3E87501038F8A@LON-INT-MSGVS01.internal.vancocorp.com>

I'm using Prognosis at the moment, looks like quite a good one... suits
large scale deployments, does everything that I have needed so far...

 

Johnny Crothers

VoIP Third Line Engineer

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Henry
Sent: 15 May 2007 13:17
To: Ed Leatherman; kevin k
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] voice monitoring tool

 

yes Cisco doe rebrand some of their appliances and they can be helpful
tools when trying to running down voice quality issues.

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ed Leatherman
Sent: Tue 5/15/2007 8:15 AM
To: kevin k
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] voice monitoring tool

I've heard good things about Qovia, I think Cisco actually resells some
of their network probes.





On 5/15/07, kevin k < kds20850 at yahoo.com <mailto:kds20850 at yahoo.com> >
wrote: 

real time monitoring and baselining.

something for enterprise.

 

excluding RTMT.

----- Original Message ----
From: hemant kumar < hemant.taan at gmail.com
<mailto:hemant.taan at gmail.com> >
To: kevin k <kds20850 at yahoo.com>
Cc: cisco-voip at puck.nether.net
Sent: Tuesday, May 15, 2007 6:05:34 AM
Subject: Re: [cisco-voip] voice monitoring tool

monitor means real time monitoring ? or recording of voice calls?



On 5/15/07, kevin k <kds20850 at yahoo.com > wrote: 

i want to monitor voice network.

what tool do you recommend?

 

thanks in advance.

 

 

________________________________

Building a website is a piece of cake. 
Yahoo! Small Business gives you all the tools to get online.
<http://us.rd.yahoo.com/evt=48251/*http:/smallbusiness.yahoo.com/webhost
ing/?p=PASSPORTPLUS> 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

 

 

 

________________________________

Pinpoint customers
<http://us.rd.yahoo.com/evt=48250/*http:/searchmarketing.yahoo.com/arp/s
ponsoredsearch_v9.php?o=US2226&cmp=Yahoo&ctv=AprNI&s=Y&s2=EM&b=50> who
are looking for what you sell. 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 



**********************************************************************
Any opinions expressed in the email are those of the individual and not necessarily the company. This email and any files transmitted with it are confidential and solely for the use of the intended recipient.  If you are not the intended recipient or the person responsible for delivering it to the intended recipient, be advised that you have received this email in error and that any dissemination, distribution, copying or use is strictly prohibited.

If you have received this email in error, or if you are concerned with the content of this email please e-mail to: e-security.support at vanco.info

The contents of an attachment to this e-mail may contain software viruses which could damage your own computer system. While the sender has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening any attachments to this e-mail. 

Vanco UK Ltd Registered in England No: 2296733 Registered Office: John Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX

Please consider the environment before printing this e-mail
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From lelio at uoguelph.ca  Tue May 15 09:09:21 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 15 May 2007 09:09:21 -0400
Subject: [cisco-voip] Transmitting ALI data from Emergency Responder?
References: <81746c9a0705141923ude3f4f9obf5bec4a328bacdc@mail.gmail.com>
	<94a1afde0705150511m26065f91o2879f65290822c79@mail.gmail.com>
Message-ID: <008d01c796f2$410495a0$47196883@cfs.uoguelph.ca>

Something you can consider as an alternative is asking your PSTN provider to update the ELIN records for you. For example, if you don't need dynamic ELINs and callback, and you have a short number of them, you can add LDNs on your PRI/megalink service and have the records updated for those numbers only.

Of course, all the stars have to align, your provider has to offer this service and you have to be using PRIs, etc, but it might be an option for you. 



--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Ed Leatherman 
  To: Candace Holman 
  Cc: cisco-voip at puck.nether.net 
  Sent: Tuesday, May 15, 2007 8:11 AM
  Subject: Re: [cisco-voip] Transmitting ALI data from Emergency Responder?


  Anyone have info on a reputable company that handles PS-ALI service? If Verizon is the only ALI uploader in our state, are we stuck with them?

  I have a regional campus that needs maybe 15 ELINs, and Verizon will only sell a minimum of 80 records, which for the size and budget of the campus is something they are resisting paying for. We can't add them to our own PS-ALI contract as they are on a different 911 system. 


  On 5/14/07, Candace Holman <holman at post.harvard.edu> wrote:
    It's a bit off topic for this list but may be the only place where hands-on VoIP/E911 projects might be discussed, so I'd like to hear what others say too.

    There are whole companies devoted to this task and you are right, it is a major task and can be a hassle.  It's going to depend on the state-supported E911 infrastructure you deal with, so look up each state you need to do this for.  A "selective router" in the e911 infrastructure routes calls to the proper PSAP and each PSAP has a PS/ALI database.  For example, in Massachusetts there are 2 selective routers and 278 PSAPs.  The MA PSAPs have a deal with Verizon to be the only PS/ALI uploader so we had to do our data uploads through Verizon.  We had to use Kermit over a modem, but I believe they are updating this process.  You will need to press your service provider for information, but you have to know the right questions to ask. 

    We only had to do two PSAPs. Since each state's E911 is run independently, you're going to have to find out what needs to be done for your case/s.  Within CER you generate the upload files, and may need to tweak them for the PS/ALI formatting (hopefully not too much). Hint - generate one of each NENA format and you'll be ready when they tell you you sent it in the wrong format. 

    At some point you will need to deal with each PSAP to do testing, so get prepared and know which ones you'll need to contact.  As a professional courtesy, you don't want to dial 911 for testing without calling ahead first. 

    Here's a Vz link that I think someone on this list posted recently.  It shows what you should be looking for from your provider, and lets you start building a list of questions to ask:
    http://www22.verizon.com/wholesale/local/E911/1,21070,,00.html 

    Also check out the NENA website and when you think you need a break from it all lookup the Next Gen 911 (ECRIT) project being defined in the IETF and googlable in combination with Henning Schulzrinne's name.  I would be surprised if Cisco were not all over this at some point in the next year or two. 

    Candace
      From: "Dark Fiber" <d4rkf1ber at gmail.com>
      To:  cisco-voip at puck.nether.net
      Date: Mon, 14 May 2007 15:30:53 -0500 
       
      I am curious to find out what others have done in regards to
      leveraging emergency responder and more specifically working through
      the process for transmitting your ALI info.

      Personally I have not had a lot of luck in determing the process from 
      my service provider.

      Like many who have CallManager we have PRI's coming into a main
      office, we have satellite offices in many different cities with
      different PSAP's and such.  The users have phone numbers tied to a DID 
      range tied to the PRI's and well 911 is an issue.

      We plan to leverage ER to identify phones in those offices and tie
      them to an ELIN associated with the address of that office.  Nothing
      too detailed or granular. 

      The primary issue at this point is getting info to the service
      provider I guess.  I don't think we are suppose to deal with
      individual PSAP's.  My understanding is we are suppose to send info to
      the service provider of our PRI's, who is suppose to send the info or 
      pass it on to PSAPs I guess.

      Just looking to hear what others have done regarding this process, the
      trials and tribulations if you will of figuring out this aspect of the
      whole thing, it does seem to be the most important component of 
      rolling out ER.

      later

       

    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip





  -- 
  Ed Leatherman
  Senior Voice Engineer
  West Virginia University
  Telecommunications and Network Operations 


------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Nick.Kassel at Charles-Stanley.co.uk  Tue May 15 09:42:02 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 15 May 2007 14:42:02 +0100
Subject: [cisco-voip] Call Park Issue on 5.1
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451EDE@CSEXCH01.charles-stanley.co.uk>

I have an issue with Call park on 5.1(1)b, if I answer a call then put
the call on Park the call park display number does not display until
about 5 seconds later, the incoming number is displayed in the mean
time, this is a global issue and is happening across all sites. I have
checked the service parameters but can not see anything for this, has
anyone else come across this yet?

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From rratliff at cisco.com  Tue May 15 09:49:59 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 15 May 2007 09:49:59 -0400
Subject: [cisco-voip] CCM 4.2 and Off-Hook
In-Reply-To: <464862CC.D6AB.003D.0@ci.carson-city.nv.us>
References: <46484B66.D6AB.003D.0@ci.carson-city.nv.us><94a1afde0705141316l3987f354p80075dd5553db31b@mail.gmail.com>
	<464862CC.D6AB.003D.0@ci.carson-city.nv.us>
Message-ID: <00c201c796f7$ee318620$b241530a@amer.cisco.com>

Look at the Cisco Lines perfmon counter.  I'm not sure if it shows linestate
as much as the number of calls active.  It is organised by partition/dn, not
device. 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Williams
Sent: Monday, May 14, 2007 4:23 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CCM 4.2 and Off-Hook

> How would you need this information, and do you need it for all phones 
> at once or just arbitrary endpoints? Attendant Console can show you 
> line state of a desired IP Phone on the system, although it doesn't 
> show any call details. The attendant just has to locate them in the
directory.

The format the information is in is more or less irrelevant, as it's going
to be going to another program, where I can have it parse whatever CCM will
give it.

I realize Attendant Console provides this information, but I need to get it
another way so I can integrate that information with some other data. If I
can use the UDP transmissions that AC puts out, that would work, but I'd
need to know how the two communicate so I can "spoof" the fact that I'm not
really an AC client. Does that make sense?

-James



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From rratliff at cisco.com  Tue May 15 09:52:08 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 15 May 2007 09:52:08 -0400
Subject: [cisco-voip] ccm-mananger config - getting along without it
In-Reply-To: <ca09e3750705142044g64b29f49jffd5a837ab562003@mail.gmail.com>
References: <00d101c79694$d7363a50$c901a8c0@pdp11>
	<ca09e3750705142044g64b29f49jffd5a837ab562003@mail.gmail.com>
Message-ID: <010b01c796f8$3d8cc540$b241530a@amer.cisco.com>

The main thing you lose is the ability to reset the gateway via the CCMAdmin
web page.  This means that after any config changes on the CM side you have
to do a 'no mgcp' 'mgcp'.
 
-Ryan 
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fred Nielsen
Sent: Monday, May 14, 2007 11:45 PM
To: Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm-mananger config - getting along without it


I realize this is not all the information you are looking for... but I never
use that command - even during initial configuration.  I do not like the
some of the changes it tends to apply to router configs, I prefer to be more
explicit and know what I am configuring.  The only caveat I do not like is
that to reset an ISDN interface (truly, meaning drop the D-channel and
reestablish) you have to either "no mgcp" the router or shut/unshut the
interface.  When you have the ccm-manager config command in you can do this
via Reset from CM. 
 
On the flip side, there are a number of customers who benefit from this
approach (i.e. telecom admins who want nothing to do with IOS)... but being
an IOS guy I still dislike it.
 
-Fred Nielsen

On 5/14/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

It looks like the "ccm-manager config" statement is overwriting my trunk
group assignment under the controller T1 interface, soooooooo, it looks like
I have to give it up for the manual approach. 
 
Can anyone comment on what I lose and how I get it back with going with the
manual configs? I know that I can issue the ccm-manager config statement and
then just remove it and things work, but as I go forward, what should I look
out for? 
 
When I look at the xml file on the tftp server, there is quite a bit of
information there that I don't see in the config anywhere.
 
Comments? Pointers to documentation?
 
Thanks, Lelio
 
----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
 
 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 




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From ealeatherman at gmail.com  Tue May 15 10:04:58 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 15 May 2007 10:04:58 -0400
Subject: [cisco-voip] Transmitting ALI data from Emergency Responder?
In-Reply-To: <008d01c796f2$410495a0$47196883@cfs.uoguelph.ca>
References: <81746c9a0705141923ude3f4f9obf5bec4a328bacdc@mail.gmail.com>
	<94a1afde0705150511m26065f91o2879f65290822c79@mail.gmail.com>
	<008d01c796f2$410495a0$47196883@cfs.uoguelph.ca>
Message-ID: <94a1afde0705150704y6a6fb6e9x8f97d48dcac8342e@mail.gmail.com>

We tried that option, at least in our area verizon wouldn't do it, they said
we had to get the PS-ALI service.

On 5/15/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  Something you can consider as an alternative is asking your PSTN provider
> to update the ELIN records for you. For example, if you don't need dynamic
> ELINs and callback, and you have a short number of them, you can add LDNs on
> your PRI/megalink service and have the records updated for those numbers
> only.
>
> Of course, all the stars have to align, your provider has to offer this
> service and you have to be using PRIs, etc, but it might be an option for
> you.
>
>
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
> ----- Original Message -----
> *From:* Ed Leatherman <ealeatherman at gmail.com>
> *To:* Candace Holman <holman at post.harvard.edu>
> *Cc:* cisco-voip at puck.nether.net
> *Sent:* Tuesday, May 15, 2007 8:11 AM
> *Subject:* Re: [cisco-voip] Transmitting ALI data from Emergency
> Responder?
>
> Anyone have info on a reputable company that handles PS-ALI service? If
> Verizon is the only ALI uploader in our state, are we stuck with them?
>
> I have a regional campus that needs maybe 15 ELINs, and Verizon will only
> sell a minimum of 80 records, which for the size and budget of the campus is
> something they are resisting paying for. We can't add them to our own PS-ALI
> contract as they are on a different 911 system.
>
> On 5/14/07, Candace Holman <holman at post.harvard.edu> wrote:
> >
> > It's a bit off topic for this list but may be the only place where
> > hands-on VoIP/E911 projects might be discussed, so I'd like to hear what
> > others say too.
> >
> > There are whole companies devoted to this task and you are right, it is
> > a major task and can be a hassle.  It's going to depend on the
> > state-supported E911 infrastructure you deal with, so look up each state you
> > need to do this for.  A "selective router" in the e911 infrastructure routes
> > calls to the proper PSAP and each PSAP has a PS/ALI database.  For example,
> > in Massachusetts there are 2 selective routers and 278 PSAPs.  The MA PSAPs
> > have a deal with Verizon to be the only PS/ALI uploader so we had to do our
> > data uploads through Verizon.  We had to use Kermit over a modem, but I
> > believe they are updating this process.  You will need to press your service
> > provider for information, but you have to know the right questions to ask.
> >
> > We only had to do two PSAPs. Since each state's E911 is run
> > independently, you're going to have to find out what needs to be done for
> > your case/s.  Within CER you generate the upload files, and may need to
> > tweak them for the PS/ALI formatting (hopefully not too much). Hint -
> > generate one of each NENA format and you'll be ready when they tell you you
> > sent it in the wrong format.
> >
> > At some point you will need to deal with each PSAP to do testing, so get
> > prepared and know which ones you'll need to contact.  As a professional
> > courtesy, you don't want to dial 911 for testing without calling ahead
> > first.
> >
> > Here's a Vz link that I think someone on this list posted recently.  It
> > shows what you should be looking for from your provider, and lets you start
> > building a list of questions to ask:
> > http://www22.verizon.com/wholesale/local/E911/1,21070,,00.html
> >
> > Also check out the NENA website and when you think you need a break from
> > it all lookup the Next Gen 911 (ECRIT) project being defined in the IETF and
> > googlable in combination with Henning Schulzrinne's name.  I would be
> > surprised if Cisco were not all over this at some point in the next year or
> > two.
> >
> > Candace
> >
> > From: "Dark Fiber" <d4rkf1ber at gmail.com>
> > To:  cisco-voip at puck.nether.net
> > Date: Mon, 14 May 2007 15:30:53 -0500
> >
> > I am curious to find out what others have done in regards to
> > leveraging emergency responder and more specifically working through
> > the process for transmitting your ALI info.
> >
> > Personally I have not had a lot of luck in determing the process from
> > my service provider.
> >
> > Like many who have CallManager we have PRI's coming into a main
> > office, we have satellite offices in many different cities with
> > different PSAP's and such.  The users have phone numbers tied to a DID
> > range tied to the PRI's and well 911 is an issue.
> >
> > We plan to leverage ER to identify phones in those offices and tie
> > them to an ELIN associated with the address of that office.  Nothing
> > too detailed or granular.
> >
> > The primary issue at this point is getting info to the service
> > provider I guess.  I don't think we are suppose to deal with
> > individual PSAP's.  My understanding is we are suppose to send info to
> > the service provider of our PRI's, who is suppose to send the info or
> > pass it on to PSAPs I guess.
> >
> > Just looking to hear what others have done regarding this process, the
> > trials and tribulations if you will of figuring out this aspect of the
> > whole thing, it does seem to be the most important component of
> > rolling out ER.
> >
> > later
> >
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Tue May 15 10:19:04 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 15 May 2007 10:19:04 -0400
Subject: [cisco-voip] ccm-mananger config - getting along without it
References: <00d101c79694$d7363a50$c901a8c0@pdp11>
	<ca09e3750705142044g64b29f49jffd5a837ab562003@mail.gmail.com>
	<010b01c796f8$3d8cc540$b241530a@amer.cisco.com>
Message-ID: <00de01c796fb$fdd8f3c0$47196883@cfs.uoguelph.ca>

Oh - so I still have configuration control from CallManager? All I have to do is the  "no mgcp/mgcp" and it downloads the new configs?

That would be great.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Ryan Ratliff 
  To: 'Fred Nielsen' ; 'Lelio Fulgenzi' 
  Cc: cisco-voip at puck.nether.net 
  Sent: Tuesday, May 15, 2007 9:52 AM
  Subject: RE: [cisco-voip] ccm-mananger config - getting along without it


  The main thing you lose is the ability to reset the gateway via the CCMAdmin web page.  This means that after any config changes on the CM side you have to do a 'no mgcp' 'mgcp'.

  -Ryan 




------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fred Nielsen
  Sent: Monday, May 14, 2007 11:45 PM
  To: Lelio Fulgenzi
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] ccm-mananger config - getting along without it


  I realize this is not all the information you are looking for... but I never use that command - even during initial configuration.  I do not like the some of the changes it tends to apply to router configs, I prefer to be more explicit and know what I am configuring.  The only caveat I do not like is that to reset an ISDN interface (truly, meaning drop the D-channel and reestablish) you have to either "no mgcp" the router or shut/unshut the interface.  When you have the ccm-manager config command in you can do this via Reset from CM. 

  On the flip side, there are a number of customers who benefit from this approach (i.e. telecom admins who want nothing to do with IOS)... but being an IOS guy I still dislike it.

  -Fred Nielsen

  On 5/14/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
    It looks like the "ccm-manager config" statement is overwriting my trunk group assignment under the controller T1 interface, soooooooo, it looks like I have to give it up for the manual approach. 

    Can anyone comment on what I lose and how I get it back with going with the manual configs? I know that I can issue the ccm-manager config statement and then just remove it and things work, but as I go forward, what should I look out for? 

    When I look at the xml file on the tftp server, there is quite a bit of information there that I don't see in the config anywhere.

    Comments? Pointers to documentation?

    Thanks, Lelio

    --------------------------------------------------------------------------------
    Lelio Fulgenzi, B.A.
    Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
    (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
    ...there's no such thing as a bad timbit...



    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip 



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From rratliff at cisco.com  Tue May 15 10:32:15 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 15 May 2007 10:32:15 -0400
Subject: [cisco-voip] ccm-mananger config - getting along without it
In-Reply-To: <00de01c796fb$fdd8f3c0$47196883@cfs.uoguelph.ca>
References: <00d101c79694$d7363a50$c901a8c0@pdp11>
	<ca09e3750705142044g64b29f49jffd5a837ab562003@mail.gmail.com>
	<010b01c796f8$3d8cc540$b241530a@amer.cisco.com>
	<00de01c796fb$fdd8f3c0$47196883@cfs.uoguelph.ca>
Message-ID: <013201c796fd$d5c6b410$b241530a@amer.cisco.com>

No it won't download any configs at all.  The assumption is that anything
you change in CCM that needs an IOS command on the router you will do by
hand.
 

-Ryan 

 

  _____  

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Tuesday, May 15, 2007 10:19 AM
To: Ryan Ratliff; 'Fred Nielsen'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm-mananger config - getting along without it


Oh - so I still have configuration control from CallManager? All I have to
do is the  "no mgcp/mgcp" and it downloads the new configs?
 
That would be great.
 
----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...


----- Original Message ----- 
From: Ryan Ratliff <mailto:rratliff at cisco.com>  
To: 'Fred Nielsen' <mailto:fwn at feasible.net>  ; 'Lelio Fulgenzi'
<mailto:lelio at uoguelph.ca>  
Cc: cisco-voip at puck.nether.net 
Sent: Tuesday, May 15, 2007 9:52 AM
Subject: RE: [cisco-voip] ccm-mananger config - getting along without it

The main thing you lose is the ability to reset the gateway via the CCMAdmin
web page.  This means that after any config changes on the CM side you have
to do a 'no mgcp' 'mgcp'.
 
-Ryan 
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fred Nielsen
Sent: Monday, May 14, 2007 11:45 PM
To: Lelio Fulgenzi
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm-mananger config - getting along without it


I realize this is not all the information you are looking for... but I never
use that command - even during initial configuration.  I do not like the
some of the changes it tends to apply to router configs, I prefer to be more
explicit and know what I am configuring.  The only caveat I do not like is
that to reset an ISDN interface (truly, meaning drop the D-channel and
reestablish) you have to either "no mgcp" the router or shut/unshut the
interface.  When you have the ccm-manager config command in you can do this
via Reset from CM. 
 
On the flip side, there are a number of customers who benefit from this
approach (i.e. telecom admins who want nothing to do with IOS)... but being
an IOS guy I still dislike it.
 
-Fred Nielsen

On 5/14/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

It looks like the "ccm-manager config" statement is overwriting my trunk
group assignment under the controller T1 interface, soooooooo, it looks like
I have to give it up for the manual approach. 
 
Can anyone comment on what I lose and how I get it back with going with the
manual configs? I know that I can issue the ccm-manager config statement and
then just remove it and things work, but as I go forward, what should I look
out for? 
 
When I look at the xml file on the tftp server, there is quite a bit of
information there that I don't see in the config anywhere.
 
Comments? Pointers to documentation?
 
Thanks, Lelio
 
----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
 
 

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 




-------------- next part --------------
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From jsteinberg at gmail.com  Tue May 15 10:34:01 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 15 May 2007 10:34:01 -0400
Subject: [cisco-voip] Call Park Issue on 5.1
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451EDE@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451EDE@CSEXCH01.charles-stanley.co.uk>
Message-ID: <f1b65f880705150734x2cf8ebe1jfecbce6f2cb2f5e7@mail.gmail.com>

I believe this is the default behavior on (some?) newer phone loads.
However, you can use ccm service parameter Caller ID Display Priority
Enabled to change this.  Change this parameter to 'false' and this will
resolve your issue.

On 5/15/07, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote:
>
>  I have an issue with Call park on 5.1(1)b, if I answer a call then put
> the call on Park the call park display number does not display until about 5
> seconds later, the incoming number is displayed in the mean time, this is a
> global issue and is happening across all sites. I have checked the service
> parameters but can not see anything for this, has anyone else come across
> this yet?
>
>
>
>
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From lelio at uoguelph.ca  Tue May 15 11:02:14 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 15 May 2007 11:02:14 -0400
Subject: [cisco-voip] ccm-mananger config - getting along without it
References: <00d101c79694$d7363a50$c901a8c0@pdp11>
	<ca09e3750705142044g64b29f49jffd5a837ab562003@mail.gmail.com>
	<010b01c796f8$3d8cc540$b241530a@amer.cisco.com>
	<00de01c796fb$fdd8f3c0$47196883@cfs.uoguelph.ca>
	<013201c796fd$d5c6b410$b241530a@amer.cisco.com>
Message-ID: <013501c79702$057e23b0$47196883@cfs.uoguelph.ca>

Oh. How do I know what commands require equivelant IOS statements? There's quite a bit on the gateway config page and I don't see anything remotely close to IOS statements. 
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Ryan Ratliff 
  To: 'Lelio Fulgenzi' ; 'Fred Nielsen' 
  Cc: cisco-voip at puck.nether.net 
  Sent: Tuesday, May 15, 2007 10:32 AM
  Subject: RE: [cisco-voip] ccm-mananger config - getting along without it


  No it won't download any configs at all.  The assumption is that anything you change in CCM that needs an IOS command on the router you will do by hand.

  -Ryan 





------------------------------------------------------------------------------
  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
  Sent: Tuesday, May 15, 2007 10:19 AM
  To: Ryan Ratliff; 'Fred Nielsen'
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] ccm-mananger config - getting along without it


  Oh - so I still have configuration control from CallManager? All I have to do is the  "no mgcp/mgcp" and it downloads the new configs?

  That would be great.

  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  ...there's no such thing as a bad timbit...

    ----- Original Message ----- 
    From: Ryan Ratliff 
    To: 'Fred Nielsen' ; 'Lelio Fulgenzi' 
    Cc: cisco-voip at puck.nether.net 
    Sent: Tuesday, May 15, 2007 9:52 AM
    Subject: RE: [cisco-voip] ccm-mananger config - getting along without it


    The main thing you lose is the ability to reset the gateway via the CCMAdmin web page.  This means that after any config changes on the CM side you have to do a 'no mgcp' 'mgcp'.

    -Ryan 




----------------------------------------------------------------------------
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fred Nielsen
    Sent: Monday, May 14, 2007 11:45 PM
    To: Lelio Fulgenzi
    Cc: cisco-voip at puck.nether.net
    Subject: Re: [cisco-voip] ccm-mananger config - getting along without it


    I realize this is not all the information you are looking for... but I never use that command - even during initial configuration.  I do not like the some of the changes it tends to apply to router configs, I prefer to be more explicit and know what I am configuring.  The only caveat I do not like is that to reset an ISDN interface (truly, meaning drop the D-channel and reestablish) you have to either "no mgcp" the router or shut/unshut the interface.  When you have the ccm-manager config command in you can do this via Reset from CM. 

    On the flip side, there are a number of customers who benefit from this approach (i.e. telecom admins who want nothing to do with IOS)... but being an IOS guy I still dislike it.

    -Fred Nielsen

    On 5/14/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
      It looks like the "ccm-manager config" statement is overwriting my trunk group assignment under the controller T1 interface, soooooooo, it looks like I have to give it up for the manual approach. 

      Can anyone comment on what I lose and how I get it back with going with the manual configs? I know that I can issue the ccm-manager config statement and then just remove it and things work, but as I go forward, what should I look out for? 

      When I look at the xml file on the tftp server, there is quite a bit of information there that I don't see in the config anywhere.

      Comments? Pointers to documentation?

      Thanks, Lelio

      --------------------------------------------------------------------------------
      Lelio Fulgenzi, B.A.
      Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
      (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
      ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
      ...there's no such thing as a bad timbit...



      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip 



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From jonvoip at gmail.com  Tue May 15 11:36:39 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 15 May 2007 10:36:39 -0500
Subject: [cisco-voip] BAR's upgrade 4.0.11 to 4.0.12
In-Reply-To: <A56BD242744BF540B15D1672E0E3E87501038DE9@LON-INT-MSGVS01.internal.vancocorp.com>
References: <A56BD242744BF540B15D1672E0E3E87501038DE9@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <5d093f9a0705150836x46b34f3eice92a49ebb8b0b3c@mail.gmail.com>

You are hitting bug CSCsi32637 ..

Release Notes

B>Symptom:
bars backup with 4.0.12

Conditions:

bars 4.0.12

Workaround:

downgrade to 4.0.11

Further Problem Description:

[Warning] Failed to copy URL Services page. Couldn't find IIS Root
Directory for CCMCIP



However, I was under the belief that this was cosmetic in nature...



Jonathan

On 5/15/07, Johnny Crothers <Johnny.Crothers at vanco.co.uk> wrote:
>
>
>
>
> Hey everyone.
>
>
>
> Just recently upgraded BAR's due to a icompatability error I was receiving with CAR. Although now I'm receiving this error:
>
>
>
> [02:00:09 AM] Failed to copy URL Services page. Couldn't find IIS Root Directory for CCMCIP in x.x.x.x
>
>
>
> I've tried to restart the IIS service to no avail.
>
>
>
> Anyone got an idea's?
>
>
>
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From JWilliams at ci.carson-city.nv.us  Tue May 15 11:39:58 2007
From: JWilliams at ci.carson-city.nv.us (James Williams)
Date: Tue, 15 May 2007 08:39:58 -0700
Subject: [cisco-voip] CCM 4.2 and Off-Hook
In-Reply-To: <00c201c796f7$ee318620$b241530a@amer.cisco.com>
References: <46484B66.D6AB.003D.0@ci.carson-city.nv.us><94a1afde0705141316l3987f354p80075dd5553db31b@mail.gmail.com>
	<464862CC.D6AB.003D.0@ci.carson-city.nv.us>
	<00c201c796f7$ee318620$b241530a@amer.cisco.com>
Message-ID: <464971FE.D6AB.003D.0@ci.carson-city.nv.us>

> Look at the Cisco Lines perfmon counter.  I'm not sure if it shows linestate
> as much as the number of calls active.  It is organised by partition/dn, not
> device. 

I'm looking at it now, but it only appears to be tracking calls to voicemail ports (as does Cisco Phones, which I thought might work, as well). Am I looking at the wrong counters?

-James




From ErasmuE4 at telkom.co.za  Tue May 15 11:47:22 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Tue, 15 May 2007 17:47:22 +0200
Subject: [cisco-voip] ipcc express agents not showing in sup desktop app
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01546BE8@CNTRRA20-XCS01.telkom.co.za>


Hi

 

I am busy implementing an IPCC express ver 4.x contact centre. The
script is very simple and have a single team, 12 resources, no skills
based routing and one trigger number going to a singe CSQ, one SUP.
Everything is working fine and integrated with call manager 4.2.
However, I have a strange problem with the supervisor desktop not
showing the agents in his team on his PC. I checked he is assigned to
the team with all his agents and so on. He is also designated the
supervisor role in the user management. When I use my own laptop
instead of his PC I found that when I log in as the sup that I can see
myself as sup and see agents (only logged in two for a test) but the
agents disappear from the sup desktop again after a short while and
occasionally re- appear (this is on my laptop. This shows that there
is already a difference between my laptop and his PC and I therefore
suspect that it is not the IPCC CRS server config etc / or CCM config.
I tried disabling the windows firewall on his machine, my laptop and
also on the agent PCs but it did not make a difference. While this is
happening the CRS is running fine in terms of the script and also the
services in the control centre.

 

Any advice will be appreciated - is it an XP problem?, Firewall
problem? Should we try to re-install the agent and sup desktops after
we disabled the firewalls on the PCs? \

Maybe someone has seen a similar problem. The customer tells me that
the agent and sup PCs are all XP with SP2

 

 

Erik Erasmus

 



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From Scott.Voll at wesd.org  Tue May 15 11:52:14 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 15 May 2007 08:52:14 -0700
Subject: [cisco-voip] ipcc express agents not showing in sup desktop app
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F01546BE8@CNTRRA20-XCS01.telkom.co.za>
References: <472E195E692EE74BAE4E7B2880A78C1F01546BE8@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE782@Apollo.wesd.org>

We have seen this in the past.  In 3.5 it was closing the chat window,
but in 4.x you can't close the chat window.  MS Windows firewall can
cause the issue also.  Is there any other Firewalls between?  Are the
agents PC connected to there phones?  (I don't know if that would help).


 

You are not abnormal, but I'm not sure of the exact problem.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, May 15, 2007 8:47 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ipcc express agents not showing in sup desktop app

 

Hi

 

I am busy implementing an IPCC express ver 4.x contact centre. The
script is very simple and have a single team, 12 resources, no skills
based routing and one trigger number going to a singe CSQ, one SUP.
Everything is working fine and integrated with call manager 4.2.
However, I have a strange problem with the supervisor desktop not
showing the agents in his team on his PC. I checked he is assigned to
the team with all his agents and so on. He is also designated the
supervisor role in the user management. When I use my own laptop instead
of his PC I found that when I log in as the sup that I can see myself as
sup and see agents (only logged in two for a test) but the agents
disappear from the sup desktop again after a short while and
occasionally re- appear (this is on my laptop. This shows that there is
already a difference between my laptop and his PC and I therefore
suspect that it is not the IPCC CRS server config etc / or CCM config. I
tried disabling the windows firewall on his machine, my laptop and also
on the agent PCs but it did not make a difference. While this is
happening the CRS is running fine in terms of the script and also the
services in the control centre.

 

Any advice will be appreciated - is it an XP problem?, Firewall problem?
Should we try to re-install the agent and sup desktops after we disabled
the firewalls on the PCs? \

Maybe someone has seen a similar problem. The customer tells me that the
agent and sup PCs are all XP with SP2

 

 

Erik Erasmus

 

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	
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From JWilliams at ci.carson-city.nv.us  Tue May 15 12:01:21 2007
From: JWilliams at ci.carson-city.nv.us (James Williams)
Date: Tue, 15 May 2007 09:01:21 -0700
Subject: [cisco-voip] CCM 4.2 and Off-Hook
In-Reply-To: <464971FE.D6AB.003D.0@ci.carson-city.nv.us>
References: <46484B66.D6AB.003D.0@ci.carson-city.nv.us><94a1afde0705141316l3987f354p80075dd5553db31b@mail.gmail.com>
	<464862CC.D6AB.003D.0@ci.carson-city.nv.us>
	<00c201c796f7$ee318620$b241530a@amer.cisco.com>
	<464971FE.D6AB.003D.0@ci.carson-city.nv.us>
Message-ID: <46497701.D6AB.003D.0@ci.carson-city.nv.us>

> I'm looking at it now, but it only appears to be tracking calls to voicemail 
> ports (as does Cisco Phones, which I thought might work, as well). Am I 
> looking at the wrong counters?

Scratch that; wasn't looking at the server the phones are all registered to.

Thanks for the pointer in the right direction.

-James




From ErasmuE4 at telkom.co.za  Tue May 15 12:08:44 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Tue, 15 May 2007 18:08:44 +0200
Subject: [cisco-voip] ipcc express agents not showing in sup desktop app
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE782@Apollo.wesd.org>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01546BF1@CNTRRA20-XCS01.telkom.co.za>


Scott - I left the site at about 17 h00 our time and will look into it
again tomorrow. For starters I am going to re-install the desktop for
the sup while the firewall is disabled. 

 

Yes the PCs are directly attached to the relevant phones and the
RMuser controls the mac addresses of all agents and the sup.

 

One thing I forgot to mention is that the agents are configured for
extension mobility but this should not be a problem as far as I know -
they agents I tested were logged in to ext mobility as well as to the
agent desktop. The agent desktop will not allow logging if they are
not logged in to ext mobility anyway because I used their logged in
number on the profile as ICD extension.

 

 

 

Erik

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, May 15, 2007 5:52 PM
To: Erik Erasmus (E); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ipcc express agents not showing in sup
desktop app

 

We have seen this in the past.  In 3.5 it was closing the chat window,
but in 4.x you can't close the chat window.  MS Windows firewall can
cause the issue also.  Is there any other Firewalls between?  Are the
agents PC connected to there phones?  (I don't know if that would
help). 

 

You are not abnormal, but I'm not sure of the exact problem.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, May 15, 2007 8:47 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ipcc express agents not showing in sup desktop
app

 

Hi

 

I am busy implementing an IPCC express ver 4.x contact centre. The
script is very simple and have a single team, 12 resources, no skills
based routing and one trigger number going to a singe CSQ, one SUP.
Everything is working fine and integrated with call manager 4.2.
However, I have a strange problem with the supervisor desktop not
showing the agents in his team on his PC. I checked he is assigned to
the team with all his agents and so on. He is also designated the
supervisor role in the user management. When I use my own laptop
instead of his PC I found that when I log in as the sup that I can see
myself as sup and see agents (only logged in two for a test) but the
agents disappear from the sup desktop again after a short while and
occasionally re- appear (this is on my laptop. This shows that there
is already a difference between my laptop and his PC and I therefore
suspect that it is not the IPCC CRS server config etc / or CCM config.
I tried disabling the windows firewall on his machine, my laptop and
also on the agent PCs but it did not make a difference. While this is
happening the CRS is running fine in terms of the script and also the
services in the control centre.

 

Any advice will be appreciated - is it an XP problem?, Firewall
problem? Should we try to re-install the agent and sup desktops after
we disabled the firewalls on the PCs? \

Maybe someone has seen a similar problem. The customer tells me that
the agent and sup PCs are all XP with SP2

 

 

Erik Erasmus

 

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	


~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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e-mail legal notice available at 
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~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From holman at post.harvard.edu  Tue May 15 12:17:28 2007
From: holman at post.harvard.edu (Candace Holman)
Date: Tue, 15 May 2007 12:17:28 -0400
Subject: [cisco-voip] Transmitting ALI data from Emergency Responder?
In-Reply-To: <94a1afde0705150704y6a6fb6e9x8f97d48dcac8342e@mail.gmail.com>
References: <81746c9a0705141923ude3f4f9obf5bec4a328bacdc@mail.gmail.com>
	<94a1afde0705150511m26065f91o2879f65290822c79@mail.gmail.com>
	<008d01c796f2$410495a0$47196883@cfs.uoguelph.ca>
	<94a1afde0705150704y6a6fb6e9x8f97d48dcac8342e@mail.gmail.com>
Message-ID: <81746c9a0705150917t507310bct744f9f8238aba2f0@mail.gmail.com>

There are a couple of USA-specific psali providers here:
http://nena.org/companyid/cid.asp

Candace

On 5/15/07, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> We tried that option, at least in our area verizon wouldn't do it, they
> said we had to get the PS-ALI service.
>
> On 5/15/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote:
> >
> >  Something you can consider as an alternative is asking your PSTN
> > provider to update the ELIN records for you. For example, if you don't need
> > dynamic ELINs and callback, and you have a short number of them, you can add
> > LDNs on your PRI/megalink service and have the records updated for those
> > numbers only.
> >
> > Of course, all the stars have to align, your provider has to offer this
> > service and you have to be using PRIs, etc, but it might be an option for
> > you.
> >
> >
> >
> >
> > --------------------------------------------------------------------------------
> > Lelio Fulgenzi, B.A.
> > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > ...there's no such thing as a bad timbit...
> >
> >
> >  ----- Original Message -----
> > *From:* Ed Leatherman <ealeatherman at gmail.com>
> > *To:* Candace Holman <holman at post.harvard.edu>
> > *Cc:* cisco-voip at puck.nether.net
> > *Sent:* Tuesday, May 15, 2007 8:11 AM
> > *Subject:* Re: [cisco-voip] Transmitting ALI data from Emergency
> > Responder?
> >
> >
> > Anyone have info on a reputable company that handles PS-ALI service? If
> > Verizon is the only ALI uploader in our state, are we stuck with them?
> >
> > I have a regional campus that needs maybe 15 ELINs, and Verizon will
> > only sell a minimum of 80 records, which for the size and budget of the
> > campus is something they are resisting paying for. We can't add them to our
> > own PS-ALI contract as they are on a different 911 system.
> >
> > On 5/14/07, Candace Holman <holman at post.harvard.edu> wrote:
> > >
> > > It's a bit off topic for this list but may be the only place where
> > > hands-on VoIP/E911 projects might be discussed, so I'd like to hear what
> > > others say too.
> > >
> > > There are whole companies devoted to this task and you are right, it
> > > is a major task and can be a hassle.  It's going to depend on the
> > > state-supported E911 infrastructure you deal with, so look up each state you
> > > need to do this for.  A "selective router" in the e911 infrastructure routes
> > > calls to the proper PSAP and each PSAP has a PS/ALI database.  For example,
> > > in Massachusetts there are 2 selective routers and 278 PSAPs.  The MA PSAPs
> > > have a deal with Verizon to be the only PS/ALI uploader so we had to do our
> > > data uploads through Verizon.  We had to use Kermit over a modem, but I
> > > believe they are updating this process.  You will need to press your service
> > > provider for information, but you have to know the right questions to ask.
> > >
> > > We only had to do two PSAPs. Since each state's E911 is run
> > > independently, you're going to have to find out what needs to be done for
> > > your case/s.  Within CER you generate the upload files, and may need to
> > > tweak them for the PS/ALI formatting (hopefully not too much). Hint -
> > > generate one of each NENA format and you'll be ready when they tell you you
> > > sent it in the wrong format.
> > >
> > > At some point you will need to deal with each PSAP to do testing, so
> > > get prepared and know which ones you'll need to contact.  As a professional
> > > courtesy, you don't want to dial 911 for testing without calling ahead
> > > first.
> > >
> > > Here's a Vz link that I think someone on this list posted recently.
> > > It shows what you should be looking for from your provider, and lets
> > > you start building a list of questions to ask:
> > > http://www22.verizon.com/wholesale/local/E911/1,21070,,00.html
> > >
> > > Also check out the NENA website and when you think you need a break
> > > from it all lookup the Next Gen 911 (ECRIT) project being defined in the
> > > IETF and googlable in combination with Henning Schulzrinne's name.  I would
> > > be surprised if Cisco were not all over this at some point in the next year
> > > or two.
> > >
> > > Candace
> > >
> > > From: "Dark Fiber" <d4rkf1ber at gmail.com>
> > > To:  cisco-voip at puck.nether.net
> > > Date: Mon, 14 May 2007 15:30:53 -0500
> > >
> > > I am curious to find out what others have done in regards to
> > > leveraging emergency responder and more specifically working through
> > > the process for transmitting your ALI info.
> > >
> > > Personally I have not had a lot of luck in determing the process from
> > > my service provider.
> > >
> > > Like many who have CallManager we have PRI's coming into a main
> > > office, we have satellite offices in many different cities with
> > > different PSAP's and such.  The users have phone numbers tied to a DID
> > >
> > > range tied to the PRI's and well 911 is an issue.
> > >
> > > We plan to leverage ER to identify phones in those offices and tie
> > > them to an ELIN associated with the address of that office.  Nothing
> > > too detailed or granular.
> > >
> > > The primary issue at this point is getting info to the service
> > > provider I guess.  I don't think we are suppose to deal with
> > > individual PSAP's.  My understanding is we are suppose to send info to
> > > the service provider of our PRI's, who is suppose to send the info or
> > > pass it on to PSAPs I guess.
> > >
> > > Just looking to hear what others have done regarding this process, the
> > > trials and tribulations if you will of figuring out this aspect of the
> > > whole thing, it does seem to be the most important component of
> > > rolling out ER.
> > >
> > > later
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> >
> >
> > --
> > Ed Leatherman
> > Senior Voice Engineer
> > West Virginia University
> > Telecommunications and Network Operations
> >
> > ------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>
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From aelnagar at ACT-EG.COM  Tue May 15 12:54:45 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 15 May 2007 19:54:45 +0300
Subject: [cisco-voip] Location Based MLPP
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A06B@mail.ACT-EG.COM>

Hello all;
 
Any one configured MLPP or has a link for an easy how to configure this feature, appricate any feedback  
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com
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From ErasmuE4 at telkom.co.za  Tue May 15 14:21:30 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Tue, 15 May 2007 20:21:30 +0200
Subject: [cisco-voip] ntp for call manager - unity connection etc. Question
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01546BF2@CNTRRA20-XCS01.telkom.co.za>


Hi

 

I am planning to use ntp to sync time on a two server call manager
cluster - one publisher and one subscriber. Plus one Unity Connection
server. Was thinking of using the main voice gateway as the NTP master
and the point each server to it. Cisco documentation is conflicting.
One doc says one must point the publisher to the NTP master and then
point the sub to the pub  and other docs on cisco.com say the servers
can not point to another call manager. Using CCM 4.2.

 

 

 

Anyone has any advice - with the conflicting info it might be better
to point all servers directly to the main gateway - acting as the
master ??

 

 

Erik Erasmus



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From cpoper at hamilton.ca  Tue May 15 15:01:41 2007
From: cpoper at hamilton.ca (Poper, Cathy)
Date: Tue, 15 May 2007 15:01:41 -0400
Subject: [cisco-voip] ntp for call manager - unity connection etc.
	Question
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F01546BF2@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <46262932FF67F74CA6D537C31A6969870D4893BE@endeavour.InfoNetwork>

Hi Erik - I followed Cisco docs and modified the ntp.conf file on the
publisher only, to point to our main network gateway.  The 3 subscribers
all specify the publisher's IP address as their source in the ntp.conf
file...they've been keeping time well so far....
 
Cathy
~~~~~~~~~~~~~~~~~~~~ 

	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
	Sent: Tuesday, May 15, 2007 2:22 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] ntp for call manager - unity connection
etc. Question
	
	

	Hi

	 

	I am planning to use ntp to sync time on a two server call
manager cluster - one publisher and one subscriber. Plus one Unity
Connection server. Was thinking of using the main voice gateway as the
NTP master and the point each server to it. Cisco documentation is
conflicting. One doc says one must point the publisher to the NTP master
and then point the sub to the pub  and other docs on cisco.com say the
servers can not point to another call manager. Using CCM 4.2.

	 

	 

	 

	Anyone has any advice - with the conflicting info it might be
better to point all servers directly to the main gateway - acting as the
master ??

	 

	 

	Erik Erasmus

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	

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From ErasmuE4 at telkom.co.za  Tue May 15 15:06:21 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Tue, 15 May 2007 21:06:21 +0200
Subject: [cisco-voip] ntp for call manager - unity connection etc.
	Question
In-Reply-To: <46262932FF67F74CA6D537C31A6969870D4893BE@endeavour.InfoNetwork>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01546BF4@CNTRRA20-XCS01.telkom.co.za>


Thanks 

 

Will do same

 

________________________________

From: Poper, Cathy [mailto:cpoper at hamilton.ca] 
Sent: Tuesday, May 15, 2007 9:02 PM
To: Erik Erasmus (E); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ntp for call manager - unity connection etc.
Question

 

Hi Erik - I followed Cisco docs and modified the ntp.conf file on the
publisher only, to point to our main network gateway.  The 3
subscribers all specify the publisher's IP address as their source in
the ntp.conf file...they've been keeping time well so far....

 

Cathy
~~~~~~~~~~~~~~~~~~~~ 

	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
	Sent: Tuesday, May 15, 2007 2:22 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] ntp for call manager - unity connection
etc. Question

	Hi

	 

	I am planning to use ntp to sync time on a two server call
manager cluster - one publisher and one subscriber. Plus one Unity
Connection server. Was thinking of using the main voice gateway as the
NTP master and the point each server to it. Cisco documentation is
conflicting. One doc says one must point the publisher to the NTP
master and then point the sub to the pub  and other docs on cisco.com
say the servers can not point to another call manager. Using CCM 4.2.

	 

	 

	 

	Anyone has any advice - with the conflicting info it might be
better to point all servers directly to the main gateway - acting as
the master ??

	 

	 

	Erik Erasmus

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

	 



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From baryonyx5 at gmail.com  Tue May 15 15:06:51 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Tue, 15 May 2007 14:06:51 -0500
Subject: [cisco-voip] Call Park Issue on 5.1
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451EDE@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451EDE@CSEXCH01.charles-stanley.co.uk>
Message-ID: <a749d0620705151206u631f6ebeq43f965fdd099a5c9@mail.gmail.com>

Is this only an issue when you park the call immediately after answering
it?  I noticed the same thing during testing because I'd answer the call and
park it right away.  If I'd wait 5 seconds or so first, then when parking
the call the park number displayed.  I wouldn't expect many calls to be
parked this quickly in production so I didn't worry about it too much.

On 5/15/07, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote:
>
>  I have an issue with Call park on 5.1(1)b, if I answer a call then put
> the call on Park the call park display number does not display until about 5
> seconds later, the incoming number is displayed in the mean time, this is a
> global issue and is happening across all sites. I have checked the service
> parameters but can not see anything for this, has anyone else come across
> this yet?
>
>
>
>
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>
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>
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-- 
kris seraphine
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From jonvoip at gmail.com  Tue May 15 15:09:06 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 15 May 2007 14:09:06 -0500
Subject: [cisco-voip] Meetingplace Express unable to relay email thru
	Exchange off net
Message-ID: <5d093f9a0705151209q4ae980ccma73f5f408be9c292@mail.gmail.com>

Installed a new MPX 1.2, when you schedule a meeting, emails get sent
to participants for on net users only (local to the domain), however,
off net (to say, gmail, etc) do not work.

We have tested and the Exchange servers will relay mail.

The SMTP configuration is pretty basic, just an IP for the mail server
with no authentication...

Do we need to authenticate to send mail off net?



Jonathan

From lelio at uoguelph.ca  Tue May 15 15:11:04 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 15 May 2007 15:11:04 -0400
Subject: [cisco-voip] ntp for call manager - unity connection
	etc.Question
References: <46262932FF67F74CA6D537C31A6969870D4893BE@endeavour.InfoNetwork>
Message-ID: <025901c79724$c8f7ac40$47196883@cfs.uoguelph.ca>

MessageWe've set up all servers to sync to our routers. That way, if the publisher is down for any length of time, the time will still remain correct on the subscribers.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Poper, Cathy 
  To: Erik Erasmus (E) ; cisco-voip at puck.nether.net 
  Sent: Tuesday, May 15, 2007 3:01 PM
  Subject: Re: [cisco-voip] ntp for call manager - unity connection etc.Question


  Hi Erik - I followed Cisco docs and modified the ntp.conf file on the publisher only, to point to our main network gateway.  The 3 subscribers all specify the publisher's IP address as their source in the ntp.conf file...they've been keeping time well so far....

  Cathy
  ~~~~~~~~~~~~~~~~~~~~ 
    -----Original Message-----
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus (E)
    Sent: Tuesday, May 15, 2007 2:22 PM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] ntp for call manager - unity connection etc. Question


    Hi

     

    I am planning to use ntp to sync time on a two server call manager cluster - one publisher and one subscriber. Plus one Unity Connection server. Was thinking of using the main voice gateway as the NTP master and the point each server to it. Cisco documentation is conflicting. One doc says one must point the publisher to the NTP master and then point the sub to the pub  and other docs on cisco.com say the servers can not point to another call manager. Using CCM 4.2.

     

     

     

    Anyone has any advice - with the conflicting info it might be better to point all servers directly to the main gateway - acting as the master ??

     

     

    Erik Erasmus

          ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
          This e-mail and its contents are subject to the Telkom SA Limited
          e-mail legal notice available at 
          http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
          ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
         



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Tue May 15 19:03:51 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 15 May 2007 19:03:51 -0400
Subject: [cisco-voip] handling inbound calls during SRST ...
Message-ID: <02ec01c79745$4ddbda60$47196883@cfs.uoguelph.ca>

I think I've come up with a plan that suits my needs regarding inbound dialing during SRST and wouldn't mind hearing what others have to say....

Basically, I'm going to create two dial-peers for both the T1 ports which translate all incoming calls so that it adds a 9 in front of it. Then I will use num-exp to modify these to the appropriate extension.

My reasoning behind this is this:
  a.. by prefixing all incoming calls I am not limited to the number of translation rules (15)
  b.. by using the translation profile incoming on the dialpeer I don't have to worry about affecting numbers dialed by phones onsite
  c.. by using num-exp config statements with 9<blah>, onsite phones don't see any change in dialing behaviour
    a.. if i had used just the number without the 9, users would see matches to non-existant extensions
  d.. by using num-exp I have no limits (not that I've seen)

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From jonvoip at gmail.com  Tue May 15 20:28:01 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 15 May 2007 19:28:01 -0500
Subject: [cisco-voip] Meetingplace Express unable to relay email
	thruExchange off net
In-Reply-To: <5d093f9a0705151713p4fe03785o591c82112ae5c192@mail.gmail.com>
References: <5d093f9a0705151209q4ae980ccma73f5f408be9c292@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F005922FB2@USNAEXCH.na.didata.local>
	<5d093f9a0705151713p4fe03785o591c82112ae5c192@mail.gmail.com>
Message-ID: <5d093f9a0705151728kf1d1da1qe0013ce3aed84246@mail.gmail.com>

I SSHd to the MPX box and then telnetted to the mail server on 25 and
sent mail without a prob...

There is something definitely wrong with Meetingplace Express...

Just out of curiosity, does it matter if they are listed as a guest?



Jonathan

On 5/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> I SSHd to the MPX box and then telnetted to the mail server on 25 and
> sent mail without a prob...
>
> There is something definitely wrong with Meetingplace...
>
>
>
> Jonathan
>
> On 5/15/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> > Can you telnet to smtp server and send the same email as if you were
> > MeetingPlace?
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > Charles
> > Sent: Tuesday, May 15, 2007 3:09 PM
> > To: ciscovoip
> > Subject: [cisco-voip] Meetingplace Express unable to relay email
> > thruExchange off net
> >
> > Installed a new MPX 1.2, when you schedule a meeting, emails get sent
> > to participants for on net users only (local to the domain), however,
> > off net (to say, gmail, etc) do not work.
> >
> > We have tested and the Exchange servers will relay mail.
> >
> > The SMTP configuration is pretty basic, just an IP for the mail server
> > with no authentication...
> >
> > Do we need to authenticate to send mail off net?
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain
> > confidential and privileged information and is for use by the
> > designated addressee(s) named above only.  If you are not the
> > intended addressee, you are hereby notified that you have received
> > this communication in error and that any use or reproduction of
> > this email or its contents is strictly prohibited and may be
> > unlawful.  If you have received this communication in error, please
> > notify us immediately by replying to this message and deleting it
> > from your computer. Thank you.
> >
>

From jonvoip at gmail.com  Tue May 15 20:38:36 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 15 May 2007 19:38:36 -0500
Subject: [cisco-voip] Meetingplace Express unable to relay email
	thruExchange off net
In-Reply-To: <5d093f9a0705151728kf1d1da1qe0013ce3aed84246@mail.gmail.com>
References: <5d093f9a0705151209q4ae980ccma73f5f408be9c292@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F005922FB2@USNAEXCH.na.didata.local>
	<5d093f9a0705151713p4fe03785o591c82112ae5c192@mail.gmail.com>
	<5d093f9a0705151728kf1d1da1qe0013ce3aed84246@mail.gmail.com>
Message-ID: <5d093f9a0705151738t7b449cddx666ca50b313716bb@mail.gmail.com>

OK, I tried something crazy... but it worked (so maybe not so crazy after all!)

I configured a new user with an offnet email address and invited him.
The offnet notication was received.

So, it looks like meetingplace express only sends email notifications
to users configured on the system.

How do we change this?



Jonathan

On 5/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> I SSHd to the MPX box and then telnetted to the mail server on 25 and
> sent mail without a prob...
>
> There is something definitely wrong with Meetingplace Express...
>
> Just out of curiosity, does it matter if they are listed as a guest?
>
>
>
> Jonathan
>
> On 5/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > I SSHd to the MPX box and then telnetted to the mail server on 25 and
> > sent mail without a prob...
> >
> > There is something definitely wrong with Meetingplace...
> >
> >
> >
> > Jonathan
> >
> > On 5/15/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> > > Can you telnet to smtp server and send the same email as if you were
> > > MeetingPlace?
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > > Charles
> > > Sent: Tuesday, May 15, 2007 3:09 PM
> > > To: ciscovoip
> > > Subject: [cisco-voip] Meetingplace Express unable to relay email
> > > thruExchange off net
> > >
> > > Installed a new MPX 1.2, when you schedule a meeting, emails get sent
> > > to participants for on net users only (local to the domain), however,
> > > off net (to say, gmail, etc) do not work.
> > >
> > > We have tested and the Exchange servers will relay mail.
> > >
> > > The SMTP configuration is pretty basic, just an IP for the mail server
> > > with no authentication...
> > >
> > > Do we need to authenticate to send mail off net?
> > >
> > >
> > >
> > > Jonathan
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > > -----------------------------------------
> > > Disclaimer:
> > >
> > > This e-mail communication and any attachments may contain
> > > confidential and privileged information and is for use by the
> > > designated addressee(s) named above only.  If you are not the
> > > intended addressee, you are hereby notified that you have received
> > > this communication in error and that any use or reproduction of
> > > this email or its contents is strictly prohibited and may be
> > > unlawful.  If you have received this communication in error, please
> > > notify us immediately by replying to this message and deleting it
> > > from your computer. Thank you.
> > >
> >
>

From akoul at dataways.gr  Wed May 16 02:44:53 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Wed, 16 May 2007 09:44:53 +0300
Subject: [cisco-voip] Does CSCsb42763 for pickup in Broadcast groups work in
	CCM 5.x?
Message-ID: <464AA865.6040201@dataways.gr>

Hi,
does the engineering fix CSCsb42763 work in CCM 5.0(4a)?
I want to be able to pickup a ringing extension in the same pickup group 
with me but which belongs to a broadcast group.

Thanks
Anthony

From Nick.Kassel at Charles-Stanley.co.uk  Wed May 16 06:05:57 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed, 16 May 2007 11:05:57 +0100
Subject: [cisco-voip] Call Park Issue on 5.1
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451EE6@CSEXCH01.charles-stanley.co.uk>

Thanks Justin, I have changed the parameter to False and now it is
working fine.

 

________________________________

From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: 15 May 2007 15:34
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call Park Issue on 5.1

 

I believe this is the default behavior on (some?) newer phone loads.
However, you can use ccm service parameter Caller ID Display Priority
Enabled to change this.  Change this parameter to 'false' and this will
resolve your issue.

On 5/15/07, Nick Kassel < Nick.Kassel at charles-stanley.co.uk
<mailto:Nick.Kassel at charles-stanley.co.uk> > wrote:

I have an issue with Call park on 5.1(1)b, if I answer a call then put
the call on Park the call park display number does not display until
about 5 seconds later, the incoming number is displayed in the mean
time, this is a global issue and is happening across all sites. I have
checked the service parameters but can not see anything for this, has
anyone else come across this yet?

 


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************************************************************************
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Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
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Authorised and Regulated by the Financial Services Authority, Member of
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Access to this e-mail by any person other than the recipient is prohibited. If you have received this message in
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You are advised that urgent, time sensitive and confidential communications should not be sent by e-mail. You 
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You acknowledge that e-mails are not secure and you accept the risk of malfunction, viruses, unauthorised 
interference, mis-delivery or delay. Charles Stanley reserves the right to monitor and/or record emails sent and
received via its network for any lawful business purpose in accordance with applicable law and regulations.
*******************************************************************************************************


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Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions of Charles Stanley & Co. Ltd  

Authorised and Regulated by the Financial Services Authority, Member of the London Stock Exchange, The 
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From MillerS at DicksteinShapiro.COM  Wed May 16 07:42:28 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Wed, 16 May 2007 07:42:28 -0400
Subject: [cisco-voip] "verifying load" Message on Phone?
Message-ID: <B9B5B75DF9145F44A21111DD92CC4900065389DF@DCEX2.DSMO.COM>

We have a phone stuck on "verifying load".  This is a new phone which we
programmed and are attempting to register at an SRST site.  We have not
run into this problem before.   I am trying to instruct someone "on the
ground" in that location as we are 3000 miles away.  Would a factory
reset help?
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

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From Nick.Kassel at Charles-Stanley.co.uk  Wed May 16 09:43:38 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed, 16 May 2007 14:43:38 +0100
Subject: [cisco-voip] 5.1 Phone Upgrade Issue
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451EEA@CSEXCH01.charles-stanley.co.uk>

I have a very strange issue with one of our remote sites, I have
upgraded some phones to SCCP70.8-2-1S and now they are unable to dial
out through there local H323 gateway, the number will ring twice and
then the call cuts out and it says Busy on the phone, any phones that
have not been upgraded can still dial out fine. 

 

Now if I change the order of the Gateways in the Route group so that
they dial out of the back up gateway at our main site all phones can
dial out fine, I have also upgraded a number of other remote sites with
the same firmware and they are also fine.

 

Can anyone advise what might cause this issue, as I can't understand why
this would happen especially just at one site.

 

  

********************SAVE PAPER - THINK BEFORE YOU PRINT**************************************************

The information contained in this e-mail is strictly confidential, some or all of which may be legally privileged. 
Access to this e-mail by any person other than the recipient is prohibited. If you have received this message in
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taken in reliance on this e-mail, is prohibited. Please advise the sender immediately should this e-mail have been 
incorrectly addressed or transmitted, and then delete the email and any attachment sent with it from your computer.

You are advised that urgent, time sensitive and confidential communications should not be sent by e-mail. You 
accept that any instructions are deemed to have been given at the time the recipient(s) accesses them and that 
delivery receipt does not constitute acknowledgement or receipt by the intended recipient(s). 

You acknowledge that e-mails are not secure and you accept the risk of malfunction, viruses, unauthorised 
interference, mis-delivery or delay. Charles Stanley reserves the right to monitor and/or record emails sent and
received via its network for any lawful business purpose in accordance with applicable law and regulations.
*******************************************************************************************************


Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions of Charles Stanley & Co. Ltd  

Authorised and Regulated by the Financial Services Authority, Member of the London Stock Exchange, The 
International Capital Market Association and The London International Financial Futures & Options Exchange.

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From baryonyx5 at gmail.com  Wed May 16 09:48:28 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Wed, 16 May 2007 08:48:28 -0500
Subject: [cisco-voip] "verifying load" Message on Phone?
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC4900065389DF@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC4900065389DF@DCEX2.DSMO.COM>
Message-ID: <a749d0620705160648i4b45d1b6w1fb58cdb16618f00@mail.gmail.com>

Have you tried power cycling the phone?  Does the phone have connectivity
back to CallManager?  A new phone won't be able to register with your SRST
device unless it has registered with CallManager successfully at least
once.

On 5/16/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
>
>  We have a phone stuck on "verifying load".  This is a new phone which we
> programmed and are attempting to register at an SRST site.  We have not run
> into this problem before.   I am trying to instruct someone "on the ground"
> in that location as we are 3000 miles away.  Would a factory reset help?
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370 Fax (202)-330-5607
> millers at dicksteinshapiro.com
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message.
>
> To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLPhttp://www.DicksteinShapiro.com
> ==============================================================================
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
kris seraphine
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From MillerS at DicksteinShapiro.COM  Wed May 16 09:57:51 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Wed, 16 May 2007 09:57:51 -0400
Subject: [cisco-voip] "verifying load" Message on Phone?
References: <B9B5B75DF9145F44A21111DD92CC4900065389DF@DCEX2.DSMO.COM>
	<a749d0620705160648i4b45d1b6w1fb58cdb16618f00@mail.gmail.com>
Message-ID: <B9B5B75DF9145F44A21111DD92CC4900065389F6@DCEX2.DSMO.COM>

Thanks.  The site is not in SRST mode,  but it is a remote site that
uses SRST for emerg. backup.  Apparently, the phone just took a long
time to load its software.  Thanks all.
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

________________________________

From: Kris Seraphine [mailto:baryonyx5 at gmail.com] 
Sent: Wednesday, May 16, 2007 9:48 AM
To: Miller, Steve
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] "verifying load" Message on Phone?


Have you tried power cycling the phone?  Does the phone have
connectivity back to CallManager?  A new phone won't be able to register
with your SRST device unless it has registered with CallManager
successfully at least once.  


On 5/16/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote: 

	We have a phone stuck on "verifying load".  This is a new phone
which we programmed and are attempting to register at an SRST site.  We
have not run into this problem before.   I am trying to instruct someone
"on the ground" in that location as we are 3000 miles away.  Would a
factory reset help?
	 

	Steve Miller
	Telecom Engineer
	Dickstein Shapiro LLP
	1825 Eye Street NW | Washington, DC 20006
	Tel (202) 420-3370 Fax (202)-330-5607
	millers at dicksteinshapiro.com
<mailto:millers at dicksteinshapiro.com> 

	 

	

	--------------------------------------------------------
	This e-mail message and any attached files are confidential and
are intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
prohibited. Dickstein Shapiro reserves the right to monitor any
communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message. 
	
	
	To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com
	
	
	Dickstein Shapiro LLP
	http://www.DicksteinShapiro.com
	
========================================================================
======

	


	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	
	




-- 
kris seraphine 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

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From jgrace at adcapnet.com  Wed May 16 11:02:43 2007
From: jgrace at adcapnet.com (James Grace)
Date: Wed, 16 May 2007 11:02:43 -0400
Subject: [cisco-voip] unable to fwd Voice messages to other subscribers
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C65C@mail.adcapnetsvc.local>

?
for some reason ip phone users are not able to fwd voice messages to other subscriber on the unity system.  when we go thru the steps we get a " unable to fwd message because the server is unavailable"   now the funny thing is that I can fwd message remotely with my softphone just fine over the vpn. and the receiver of that message can play it with no problem. have anyone seen this before
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
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material from all computers."

This original email was sent to the internet for delivery at 16 May 2007 11:03:17 -0400
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From jason.aarons at us.didata.com  Wed May 16 11:09:41 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 16 May 2007 11:09:41 -0400
Subject: [cisco-voip] BAR's upgrade 4.0.11 to 4.0.12
In-Reply-To: <A56BD242744BF540B15D1672E0E3E87501038DE9@LON-INT-MSGVS01.internal.vancocorp.com>
References: <A56BD242744BF540B15D1672E0E3E87501038DE9@LON-INT-MSGVS01.internal.vancocorp.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005957289@USNAEXCH.na.didata.local>

I've probably done 8 installs with BARS 4.0.12, all gave the same error.
I assume it's a caveat...and a BARS restore won't restore any CCMCIP
customized files..

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Johnny Crothers
Sent: Tuesday, May 15, 2007 5:13 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] BAR's upgrade 4.0.11 to 4.0.12

 

Hey everyone.

 

Just recently upgraded BAR's due to a icompatability error I was
receiving with CAR. Although now I'm receiving this error:

 

[02:00:09 AM] Failed to copy URL Services page. Couldn't find IIS Root
Directory for CCMCIP in x.x.x.x

 

I've tried to restart the IIS service to no avail.

 

Anyone got an idea's?

 



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Any opinions expressed in the email are those of the individual and not
necessarily the company. This email and any files transmitted with it
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you are not the intended recipient or the person responsible for
delivering it to the intended recipient, be advised that you have
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If you have received this email in error, or if you are concerned with
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has taken every reasonable precaution to minimise this risk, we cannot
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Disclaimer:

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From zohaibshabir at gmail.com  Wed May 16 11:10:52 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Wed, 16 May 2007 20:10:52 +0500
Subject: [cisco-voip] Busy greeting in Call Manager 4.2
Message-ID: <1ff12970705160810i2147923s59df83d81c762071@mail.gmail.com>

Dear All,
Is it all possible that i can configure Busy External and Busy Internal to
some local sound source without the help of Cisco Unity.

-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and PreSales)
DWP Group, TECH Division
Ph:+92-302-8232689
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From jonvoip at gmail.com  Wed May 16 11:18:38 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 16 May 2007 10:18:38 -0500
Subject: [cisco-voip] BAR's upgrade 4.0.11 to 4.0.12
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005957289@USNAEXCH.na.didata.local>
References: <A56BD242744BF540B15D1672E0E3E87501038DE9@LON-INT-MSGVS01.internal.vancocorp.com>
	<C1FE15183DA37645BC0633BC604E44F005957289@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0705160818l555ae05hddfe2eb7c82156ba@mail.gmail.com>

I have seen info that this error is just cosmetic, and others where it
says it has something to do with DNS name of the services URL (but
that was for an older version of bars)...


Jonathan

On 5/16/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> I've probably done 8 installs with BARS 4.0.12, all gave the same error. I
> assume it's a caveat?and a BARS restore won't restore any CCMCIP customized
> files..
>
>
>
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Johnny Crothers
>  Sent: Tuesday, May 15, 2007 5:13 AM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] BAR's upgrade 4.0.11 to 4.0.12
>
>
>
>
> Hey everyone.
>
>
>
> Just recently upgraded BAR's due to a icompatability error I was receiving
> with CAR. Although now I'm receiving this error:
>
>
>
> [02:00:09 AM] Failed to copy URL Services page. Couldn't find IIS Root
> Directory for CCMCIP in x.x.x.x
>
>
>
> I've tried to restart the IIS service to no avail.
>
>
>
> Anyone got an idea's?
>
>
>
>
>
> **********************************************************************
>  Any opinions expressed in the email are those of the individual and not
> necessarily the company. This email and any files transmitted with it are
> confidential and solely for the use of the intended recipient. If you are
> not the intended recipient or the person responsible for delivering it to
> the intended recipient, be advised that you have received this email in
> error and that any dissemination, distribution, copying or use is strictly
> prohibited.
>
>  If you have received this email in error, or if you are concerned with the
> content of this email please e-mail to: e-security.support at vanco.info
>
>  The contents of an attachment to this e-mail may contain software viruses
> which could damage your own computer system. While the sender has taken
> every reasonable precaution to minimise this risk, we cannot accept
> liability for any damage which you sustain as a result of software viruses.
> You should carry out your own virus checks before opening any attachments to
> this e-mail.
>
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> Busch House, 277 London Road, Isleworth, Middlesex TW7 5AX
>
>  Please consider the environment before printing this e-mail
> **********************************************************************
>
>
> ________________________________
>
>
>
>
>  Disclaimer: This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the designated
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


From Nick.Kassel at Charles-Stanley.co.uk  Wed May 16 11:25:26 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed, 16 May 2007 16:25:26 +0100
Subject: [cisco-voip] 5.1 Phone Upgrade Issue
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451EEB@CSEXCH01.charles-stanley.co.uk>

Can anyone help with this, the phones are 7970 and the previous firmware
that worked fine is SCCP70.8-0-3S 

 

Why would I be able to dial out with one firmware and not another this
doesn't make sense.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: 16 May 2007 14:44
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 5.1 Phone Upgrade Issue
Importance: High

 

I have a very strange issue with one of our remote sites, I have
upgraded some phones to SCCP70.8-2-1S and now they are unable to dial
out through there local H323 gateway, the number will ring twice and
then the call cuts out and it says Busy on the phone, any phones that
have not been upgraded can still dial out fine. 

 

Now if I change the order of the Gateways in the Route group so that
they dial out of the back up gateway at our main site all phones can
dial out fine, I have also upgraded a number of other remote sites with
the same firmware and they are also fine.

 

Can anyone advise what might cause this issue, as I can't understand why
this would happen especially just at one site.

 

  

 


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Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions of Charles Stanley & Co. Ltd  

Authorised and Regulated by the Financial Services Authority, Member of the London Stock Exchange, The 
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From jonvoip at gmail.com  Wed May 16 12:14:23 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 16 May 2007 11:14:23 -0500
Subject: [cisco-voip] Meetingplace Express unable to relay email
	thruExchange off net
In-Reply-To: <5d093f9a0705151738t7b449cddx666ca50b313716bb@mail.gmail.com>
References: <5d093f9a0705151209q4ae980ccma73f5f408be9c292@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F005922FB2@USNAEXCH.na.didata.local>
	<5d093f9a0705151713p4fe03785o591c82112ae5c192@mail.gmail.com>
	<5d093f9a0705151728kf1d1da1qe0013ce3aed84246@mail.gmail.com>
	<5d093f9a0705151738t7b449cddx666ca50b313716bb@mail.gmail.com>
Message-ID: <5d093f9a0705160914i7d5a46fk65996948b8b3c5ee@mail.gmail.com>

OK, this is resolved.

When you invite people by email, they are invited using the Guest
account, which was disabled, so no emails were sent.



Jonathan

On 5/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> OK, I tried something crazy... but it worked (so maybe not so crazy after all!)
>
> I configured a new user with an offnet email address and invited him.
> The offnet notication was received.
>
> So, it looks like meetingplace express only sends email notifications
> to users configured on the system.
>
> How do we change this?
>
>
>
> Jonathan
>
> On 5/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > I SSHd to the MPX box and then telnetted to the mail server on 25 and
> > sent mail without a prob...
> >
> > There is something definitely wrong with Meetingplace Express...
> >
> > Just out of curiosity, does it matter if they are listed as a guest?
> >
> >
> >
> > Jonathan
> >
> > On 5/15/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > I SSHd to the MPX box and then telnetted to the mail server on 25 and
> > > sent mail without a prob...
> > >
> > > There is something definitely wrong with Meetingplace...
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On 5/15/07, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> > > > Can you telnet to smtp server and send the same email as if you were
> > > > MeetingPlace?
> > > >
> > > > -----Original Message-----
> > > > From: cisco-voip-bounces at puck.nether.net
> > > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > > > Charles
> > > > Sent: Tuesday, May 15, 2007 3:09 PM
> > > > To: ciscovoip
> > > > Subject: [cisco-voip] Meetingplace Express unable to relay email
> > > > thruExchange off net
> > > >
> > > > Installed a new MPX 1.2, when you schedule a meeting, emails get sent
> > > > to participants for on net users only (local to the domain), however,
> > > > off net (to say, gmail, etc) do not work.
> > > >
> > > > We have tested and the Exchange servers will relay mail.
> > > >
> > > > The SMTP configuration is pretty basic, just an IP for the mail server
> > > > with no authentication...
> > > >
> > > > Do we need to authenticate to send mail off net?
> > > >
> > > >
> > > >
> > > > Jonathan
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > > -----------------------------------------
> > > > Disclaimer:
> > > >
> > > > This e-mail communication and any attachments may contain
> > > > confidential and privileged information and is for use by the
> > > > designated addressee(s) named above only.  If you are not the
> > > > intended addressee, you are hereby notified that you have received
> > > > this communication in error and that any use or reproduction of
> > > > this email or its contents is strictly prohibited and may be
> > > > unlawful.  If you have received this communication in error, please
> > > > notify us immediately by replying to this message and deleting it
> > > > from your computer. Thank you.
> > > >
> > >
> >
>

From cbomba at s4nets.com  Wed May 16 13:12:37 2007
From: cbomba at s4nets.com (Christopher M. Bomba)
Date: Wed, 16 May 2007 13:12:37 -0400 (EDT)
Subject: [cisco-voip] Unity Bridge not getting message from Octel
Message-ID: <60255.192.168.1.70.1179335557.webmail@192.168.1.70>

I have a new Unity install with Unity Bridge connecting to 3 different Octel nodes.  I have everything configured and I can send messages from a Unity mailbox to an Octel mailbox on all three nodes.  The problem I am having is when an Octel subscriber tries to send or forward messages to a Unity mailbox.  I see the call come into the Bridge but it never seems to get any further.  Like there isn't any DTMF being sent between them.  

Does anyone know of anything that the Bridge needs to see from the Octel nodes in order for this to work?  Is there any configuration changes that need to be made in the Octel nodes other than assinging the Bridge a serial number and a pilot number to reach it?

Thanks,

Chris


From jonvoip at gmail.com  Wed May 16 13:41:21 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 16 May 2007 12:41:21 -0500
Subject: [cisco-voip] Recurring issue with IPCC Express 4.0(X)
Message-ID: <5d093f9a0705161041q60ffa7bue19d9bf459199ee7@mail.gmail.com>

I have been running into an issue (so many times, I have lost track),
that after a new install of IPCC Express, I go to create the JTAPI
provider and it says Update Successful, restart CRS Engine for changes
to take effect... and when I restart the engine, I go back there and
the provider isn't configured.

I generally have to do it five or six times before it finally takes it
and creates the user on CCM.

What am I missing, why do I have to do it over and over again?



Jonathan

From pat-cv at wcyv.com  Wed May 16 14:15:21 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Wed, 16 May 2007 14:15:21 -0400
Subject: [cisco-voip] unable to fwd Voice messages to other subscribers
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C65C@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C65C@mail.adcapnetsvc.local>
Message-ID: <464B4A39.2080006@wcyv.com>

Sounds like your send-as permissions are mis-configured. If you've 
applied the exchange DST patches, Exchange requires you to have full 
mailbox access AND send-as permissions to do a send-as (which is what 
Unity tries to do when you forward a message). Prior to that, you could 
have either/or and it would work. Verify whether or not your 
UnityMsgStoreSvc account has send-as permissions for the AD account of 
the user sending the message. If it doesn't, re-run latest permissions 
wizard for your version of Unity:

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_field_notice09186a0080704437.shtml

-Pat

James Grace wrote:
> ?
> for some reason ip phone users are not able to fwd voice messages to other subscriber on the unity system.  when we go thru the steps we get a " unable to fwd message because the server is unavailable"   now the funny thing is that I can fwd message remotely with my softphone just fine over the vpn. and the receiver of that message can play it with no problem. have anyone seen this before
>  
> James Grace 
>  
> Systems Engineer
> Adcap Network Systems, Inc.
> 6865 Shiloh Road East, Suite 250
> Alpharetta, GA 30005
>  
> office: 678-456-6986
> direct: 678-456-6761
> jgrace at adcapnet.com
> www.adcapnet.com <http://www.adcapnet.com/> 
> Disclaimer For ADCAP Network Systems
> 
> "The information transmitted is intended only for the person or entity to which it is addressed and may
> contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
> or taking of any action in reliance upon this information by persons or entities other than the intended
> recipient is prohibited. If you received this message in error, please contact the sender and delete the
> material from all computers."
> 
> This original email was sent to the internet for delivery at 16 May 2007 11:03:17 -0400
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
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> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From pat-cv at wcyv.com  Wed May 16 14:18:25 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Wed, 16 May 2007 14:18:25 -0400
Subject: [cisco-voip] Unity Bridge not getting message from Octel
In-Reply-To: <60255.192.168.1.70.1179335557.webmail@192.168.1.70>
References: <60255.192.168.1.70.1179335557.webmail@192.168.1.70>
Message-ID: <464B4AF1.2000005@wcyv.com>

Both servers do a handshake when they first connect, using DTMF tones. 
If that doesn't work, if one of the servers isn't hearing the tones from 
the other, it'll just sit there like that. You should be able to use the 
BANANA tool to view information about your calls and why they failed. 
Take a look at the Bridge troubleshooting guide, it's quite thorough, 
explains what to check, what the error codes mean, etc:

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_feature_guide_chapter09186a00806c15eb.html#wp1074305

-Pat

Christopher M. Bomba wrote:
> I have a new Unity install with Unity Bridge connecting to 3 different Octel nodes.  I have everything configured and I can send messages from a Unity mailbox to an Octel mailbox on all three nodes.  The problem I am having is when an Octel subscriber tries to send or forward messages to a Unity mailbox.  I see the call come into the Bridge but it never seems to get any further.  Like there isn't any DTMF being sent between them.  
> 
> Does anyone know of anything that the Bridge needs to see from the Octel nodes in order for this to work?  Is there any configuration changes that need to be made in the Octel nodes other than assinging the Bridge a serial number and a pilot number to reach it?
> 
> Thanks,
> 
> Chris
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From cpoper at hamilton.ca  Wed May 16 14:50:03 2007
From: cpoper at hamilton.ca (Poper, Cathy)
Date: Wed, 16 May 2007 14:50:03 -0400
Subject: [cisco-voip] IPCC Express Custom Reports
Message-ID: <46262932FF67F74CA6D537C31A6969870D4893D6@endeavour.InfoNetwork>

Does anyone in the group have any experience and/or success in
developing custom reports for IPCC express in-house?  If so,  I'd like
to know what your experiences were, and any hints you can provide..

Thx

Cathy Poper
Network Support Analyst
City of Hamilton-Information Technology Services
55 York Blvd, 6th flr
Hamilton. Ont. Canada
L8L 3K7
Tel: 905-546-2424 x5481
Fax: 905-546-2333
Email: cpoper at hamilton.ca
~~~~~~~~~~~~~~~~~~~~

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From VLoschiavo at data-corporation.com  Wed May 16 15:04:59 2007
From: VLoschiavo at data-corporation.com (Vince Loschiavo)
Date: Wed, 16 May 2007 15:04:59 -0400
Subject: [cisco-voip] How to send unconfigured DIDs to an auto-attendant?
In-Reply-To: <46262932FF67F74CA6D537C31A6969870D4893D6@endeavour.InfoNetwork>
References: <46262932FF67F74CA6D537C31A6969870D4893D6@endeavour.InfoNetwork>
Message-ID: <CB4BC8C24EF6E149AED9FDB9B9AF57133D8BAC@MAIL02.hq.data-corporation.com>

CCM 5.X, Unity 4.2 Unified Messaging, MGCP Gateways,
 
How can I send unconfigured DID's to an auto-attendant.
 
The customer has a block of numbers (xxx) XXX-1000 through 2000.  But
only 200 of the DID's are in use.  Right now if someone dials a number
that is not in use, the caller gets a re-order tone (fast busy).  How
can I forward those (in bulk) to the Auto-Attendant without configuring
CTI Route points for each DID?
 
I suspect that a CTI Route Point with a number of "XXXX" should do the
trick, but I haven't tested.  
 
Has anyone done this before?
 
____________________________________________________________
Vincent Loschiavo
Senior Consultant
Datacorp
18503 Pines Blvd
Suite 212
Pembroke Pines, FL, 33029

"Cisco Advanced Unified Communications Partner"
Cell Phone:        786-282-1164
Office Phone:    305-594-6933
____________________________________________________________ 
 
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From erickbe at yahoo.com  Wed May 16 14:10:10 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 16 May 2007 11:10:10 -0700 (PDT)
Subject: [cisco-voip] FAX not working to one number
Message-ID: <47236.54911.qm@web37201.mail.mud.yahoo.com>

I have a strange FAX issue, where a FAX machine connected to a VG224 can send/receive FAX to any number but is not able to send one outside number a FAX. It can receive FAX's from them fine, just not send. The FAX machines phone or a phone plugged into the FAX phone line can call the # fine so there is not a CSS issue, etc.

The FAX machine is plugged into a VG224 with IOS 12.4(11)T , the VG224 is configured as SCCP (stcapp) and we have tried setting the port capabilty to modem passthrough and not with no change.

The PSTN Gateway is a Cisco 2821 with a PRI, it is MGCP and running IOS 12.4(9)T and has FAX relay disabled. 

We have tried 7 different FAX machines of different models and neither are able to call this particular outside FAX machine when going over IPT. When the FAX machines are plugged into a traditional analog circuit there is not a problem and the FAX works fine.

Any ideas?



 
____________________________________________________________________________________
Sucker-punch spam with award-winning protection. 
Try the free Yahoo! Mail Beta.
http://advision.webevents.yahoo.com/mailbeta/features_spam.html

From jonvoip at gmail.com  Wed May 16 15:15:50 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 16 May 2007 14:15:50 -0500
Subject: [cisco-voip] FAX not working to one number
In-Reply-To: <47236.54911.qm@web37201.mail.mud.yahoo.com>
References: <47236.54911.qm@web37201.mail.mud.yahoo.com>
Message-ID: <5d093f9a0705161215r168c21aam36dd2441b43dffaa@mail.gmail.com>

Why do you think it is a problem on your end?



Jonathan

On 5/16/07, Erick Bergquist <erickbe at yahoo.com> wrote:
> I have a strange FAX issue, where a FAX machine connected to a VG224 can send/receive FAX to any number but is not able to send one outside number a FAX. It can receive FAX's from them fine, just not send. The FAX machines phone or a phone plugged into the FAX phone line can call the # fine so there is not a CSS issue, etc.
>
> The FAX machine is plugged into a VG224 with IOS 12.4(11)T , the VG224 is configured as SCCP (stcapp) and we have tried setting the port capabilty to modem passthrough and not with no change.
>
> The PSTN Gateway is a Cisco 2821 with a PRI, it is MGCP and running IOS 12.4(9)T and has FAX relay disabled.
>
> We have tried 7 different FAX machines of different models and neither are able to call this particular outside FAX machine when going over IPT. When the FAX machines are plugged into a traditional analog circuit there is not a problem and the FAX works fine.
>
> Any ideas?
>
>
>
>
> ____________________________________________________________________________________
> Sucker-punch spam with award-winning protection.
> Try the free Yahoo! Mail Beta.
> http://advision.webevents.yahoo.com/mailbeta/features_spam.html
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From birnit at nait.ca  Wed May 16 15:18:59 2007
From: birnit at nait.ca (Tkachuk,Birni)
Date: Wed, 16 May 2007 13:18:59 -0600
Subject: [cisco-voip] Supressing Incoming CLID on Shared Phones
Message-ID: <8BF2352FAC40504B8D83F9403E4CCE3159B637@EXGMETRO.nait.ca>

Is there any way to supress the Incoming Calling Line ID on shared
lines?  I am anticipating staff concerns about privacy.  

Birni Tkachuk
Voice Services Administrator
NAIT Information Services Division 	
Ph:  780-471-7532
Fax 780-491-3083
	

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From toddnh65 at gmail.com  Wed May 16 15:24:09 2007
From: toddnh65 at gmail.com (Todd Franklin)
Date: Wed, 16 May 2007 15:24:09 -0400
Subject: [cisco-voip] Second Button on 7940
Message-ID: <6101bda50705161224n13769ce7qa715308cae682be5@mail.gmail.com>

My manager has a 7940 and he's complaining about his loss of buttons that he
had on his previous phone.
So I mention this to a friend, and he says "you can make it so when he
presses that 2nd button, he gets a list of extensions to call, and he can
just arrow up and down thru them".  How do I do that???  I guess the first
button is kind of reserved, it's your extension!!!  But how do I do that
with the 2nd button?  Maybe I could just have it that when he presses it, he
gets the names of the six department heads and their extensions.  (the
buttons I am talking about are the 2 to the right of the LCD screen)

Thanks anyone.
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From lelio at uoguelph.ca  Wed May 16 15:31:53 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 16 May 2007 15:31:53 -0400
Subject: [cisco-voip] Second Button on 7940
References: <6101bda50705161224n13769ce7qa715308cae682be5@mail.gmail.com>
Message-ID: <003c01c797f0$db7aecc0$47196883@cfs.uoguelph.ca>

The second button can be defined as a service URL. So if you create a service that outputs XML and the appropriate softkeys, you can do this relatively easily.

That being said, I wouldn't know where to start to build this service, just that if it's there, it's easy to point to.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Todd Franklin 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, May 16, 2007 3:24 PM
  Subject: [cisco-voip] Second Button on 7940


  My manager has a 7940 and he's complaining about his loss of buttons that he had on his previous phone.
  So I mention this to a friend, and he says "you can make it so when he presses that 2nd button, he gets a list of extensions to call, and he can just arrow up and down thru them".  How do I do that???  I guess the first button is kind of reserved, it's your extension!!!  But how do I do that with the 2nd button?  Maybe I could just have it that when he presses it, he gets the names of the six department heads and their extensions.  (the buttons I am talking about are the 2 to the right of the LCD screen) 

  Thanks anyone.





------------------------------------------------------------------------------


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  cisco-voip mailing list
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From lelio at uoguelph.ca  Wed May 16 15:35:46 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 16 May 2007 15:35:46 -0400
Subject: [cisco-voip] Supressing Incoming CLID on Shared Phones
References: <8BF2352FAC40504B8D83F9403E4CCE3159B637@EXGMETRO.nait.ca>
Message-ID: <004b01c797f1$66566900$47196883@cfs.uoguelph.ca>

Supressing Incoming CLID on Shared PhonesYou could create a translation for that extension that masks the calling number.

To ensure that the order of operations is maintained, you would probably want to put all shared lines in a seperate partition and the translation for that shared line in the main partition.

For example

regular_ext_pt contains all regular extensions and the translations to shared lines.
sharedline_pt contains all the shared lines

regular_css contains regular_ext_pt and is assigned to all phones and inbound trunks
sharedline_css contains sharedline_pt and is assigned to the translation so that it can reach shared lines.



--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Tkachuk,Birni 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, May 16, 2007 3:18 PM
  Subject: [cisco-voip] Supressing Incoming CLID on Shared Phones


  Is there any way to supress the Incoming Calling Line ID on shared lines?  I am anticipating staff concerns about privacy.  

  Birni Tkachuk 
  Voice Services Administrator 
  NAIT Information Services Division      
  Ph:  780-471-7532 
  Fax 780-491-3083 
          



------------------------------------------------------------------------------


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From PRuttman at foley.com  Wed May 16 17:16:25 2007
From: PRuttman at foley.com (Ruttman, Pete G.)
Date: Wed, 16 May 2007 16:16:25 -0500
Subject: [cisco-voip] voice monitoring tool
In-Reply-To: <94a1afde0705150515l7e206682of0e018a3e8e281b6@mail.gmail.com>
Message-ID: <5DB17A1560B7EE4186747DF43A6D4C6C6EC251@WIMKEEX07.foleylaw.com>

We were doing an onsite eval of Qovia.  Right in the middle of it Cisco
bought out some part of Qovia and the rest of the company was let go.  I
think Cisco is still reselling the lower end models but I don't know
where it is really all going now.   

 

Pete

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Tuesday, May 15, 2007 7:15 AM
To: kevin k
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] voice monitoring tool

 

I've heard good things about Qovia, I think Cisco actually resells some
of their network probes.





On 5/15/07, kevin k < kds20850 at yahoo.com <mailto:kds20850 at yahoo.com> >
wrote:

real time monitoring and baselining.

something for enterprise.

 

excluding RTMT.

----- Original Message ----
From: hemant kumar < hemant.taan at gmail.com
<mailto:hemant.taan at gmail.com> >
To: kevin k <kds20850 at yahoo.com>
Cc: cisco-voip at puck.nether.net
Sent: Tuesday, May 15, 2007 6:05:34 AM
Subject: Re: [cisco-voip] voice monitoring tool

monitor means real time monitoring ? or recording of voice calls?



On 5/15/07, kevin k <kds20850 at yahoo.com > wrote: 

i want to monitor voice network.

what tool do you recommend?

 

thanks in advance.

 

 

________________________________

Building a website is a piece of cake. 
Yahoo! Small Business gives you all the tools to get online.
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ing/?p=PASSPORTPLUS> 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 

 

 

 

________________________________

Pinpoint customers
<http://us.rd.yahoo.com/evt=48250/*http:/searchmarketing.yahoo.com/arp/s
ponsoredsearch_v9.php?o=US2226&cmp=Yahoo&ctv=AprNI&s=Y&s2=EM&b=50> who
are looking for what you sell. 


_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations 


The preceding email message may be confidential or protected by the attorney-client privilege. It is not intended for transmission to, or receipt by, any unauthorized persons.  If you have received this message in error, please (i) do not read it, (ii) reply to the sender that you received the message in error, and (iii) erase or destroy the message.  Legal advice contained in the preceding message is solely for the benefit of the Foley & Lardner LLP client(s) represented by the Firm in the particular matter that is the subject of this message, and may not be relied upon by any other party.      

  
Internal Revenue Service regulations require that certain types of written advice include a disclaimer. To the extent the preceding message contains advice relating to a Federal tax issue, unless expressly stated otherwise the advice is not intended or written to be used, and it cannot be used by the recipient or any other taxpayer, for the purpose of avoiding Federal tax penalties, and was not written to support the promotion or marketing of any transaction or matter discussed herein.

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From jonvoip at gmail.com  Wed May 16 17:55:06 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 16 May 2007 16:55:06 -0500
Subject: [cisco-voip] Recurring issue with IPCC Express 4.0(X)
In-Reply-To: <5d093f9a0705161041q60ffa7bue19d9bf459199ee7@mail.gmail.com>
References: <5d093f9a0705161041q60ffa7bue19d9bf459199ee7@mail.gmail.com>
Message-ID: <5d093f9a0705161455i240a3052v8488083ff8f3dfe4@mail.gmail.com>

OK, I installed IPCC about six times today (doing cluster testing) and
I figured out the trick...

DO NOT USE THE DEFAULT PASSWORD

If you change it, it works the first time.



Jonathan

On 5/16/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> I have been running into an issue (so many times, I have lost track),
> that after a new install of IPCC Express, I go to create the JTAPI
> provider and it says Update Successful, restart CRS Engine for changes
> to take effect... and when I restart the engine, I go back there and
> the provider isn't configured.
>
> I generally have to do it five or six times before it finally takes it
> and creates the user on CCM.
>
> What am I missing, why do I have to do it over and over again?
>
>
>
> Jonathan
>

From jason.aarons at us.didata.com  Wed May 16 18:04:00 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 16 May 2007 18:04:00 -0400
Subject: [cisco-voip] Having to net stop CSA versus services.msc ?
Message-ID: <C1FE15183DA37645BC0633BC604E44F005957C34@USNAEXCH.na.didata.local>

Has a customer who couldn't stop CSA via the GUI in services.msc in 2003
Server (UCCE ICM/PG/AQ)it just showed stopping but would never stop.
Instead we found net stop csa from dos command line worked. Anyone seen
this before? Known bug? -jason

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 




-----------------------------------------
Disclaimer:

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From ealeatherman at gmail.com  Wed May 16 19:17:36 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 16 May 2007 19:17:36 -0400
Subject: [cisco-voip] voice monitoring tool
In-Reply-To: <5DB17A1560B7EE4186747DF43A6D4C6C6EC251@WIMKEEX07.foleylaw.com>
References: <94a1afde0705150515l7e206682of0e018a3e8e281b6@mail.gmail.com>
	<5DB17A1560B7EE4186747DF43A6D4C6C6EC251@WIMKEEX07.foleylaw.com>
Message-ID: <94a1afde0705161617j3330cebcw67e5c593caa5430b@mail.gmail.com>

Guess that's why i never heard back from the sales guy after they offered us
a demo!

On 5/16/07, Ruttman, Pete G. <PRuttman at foley.com> wrote:
>
>  We were doing an onsite eval of Qovia.  Right in the middle of it Cisco
> bought out some part of Qovia and the rest of the company was let go.  I
> think Cisco is still reselling the lower end models but I don't know where
> it is really all going now.
>
>
>
> Pete
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Ed Leatherman
> *Sent:* Tuesday, May 15, 2007 7:15 AM
> *To:* kevin k
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] voice monitoring tool
>
>
>
> I've heard good things about Qovia, I think Cisco actually resells some of
> their network probes.
>
>
>
>  On 5/15/07, *kevin k* < kds20850 at yahoo.com> wrote:
>
> real time monitoring and baselining.
>
> something for enterprise.
>
>
>
> excluding RTMT.
>
> ----- Original Message ----
> From: hemant kumar < hemant.taan at gmail.com>
> To: kevin k <kds20850 at yahoo.com>
> Cc: cisco-voip at puck.nether.net
> Sent: Tuesday, May 15, 2007 6:05:34 AM
> Subject: Re: [cisco-voip] voice monitoring tool
>
> monitor means real time monitoring ? or recording of voice calls?
>
>  On 5/15/07, *kevin k* <kds20850 at yahoo.com > wrote:
>
> i want to monitor voice network.
>
> what tool do you recommend?
>
>
>
> thanks in advance.
>
>
>
>
>  ------------------------------
>
> Building a website is a piece of cake.
> Yahoo! Small Business gives you all the tools to get online.<http://us.rd.yahoo.com/evt=48251/*http:/smallbusiness.yahoo.com/webhosting/?p=PASSPORTPLUS>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
>  ------------------------------
>
> Pinpoint customers
> <http://us.rd.yahoo.com/evt=48250/*http:/searchmarketing.yahoo.com/arp/sponsoredsearch_v9.php?o=US2226&cmp=Yahoo&ctv=AprNI&s=Y&s2=EM&b=50>who
> are looking for what you sell.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>
> The preceding email message may be confidential or protected by the
> attorney-client privilege. It is not intended for transmission to, or
> receipt by, any unauthorized persons. If you have received this message in
> error, please (i) do not read it, (ii) reply to the sender that you received
> the message in error, and (iii) erase or destroy the message. Legal advice
> contained in the preceding message is solely for the benefit of the Foley &
> Lardner LLP client(s) represented by the Firm in the particular matter that
> is the subject of this message, and may not be relied upon by any other
> party.
>
>
> Internal Revenue Service regulations require that certain types of written
> advice include a disclaimer. To the extent the preceding message contains
> advice relating to a Federal tax issue, unless expressly stated otherwise
> the advice is not intended or written to be used, and it cannot be used by
> the recipient or any other taxpayer, for the purpose of avoiding Federal tax
> penalties, and was not written to support the promotion or marketing of any
> transaction or matter discussed herein.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From asobihoudai at yahoo.com  Wed May 16 18:25:45 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 16 May 2007 15:25:45 -0700 (PDT)
Subject: [cisco-voip] Changing password on CCM 4.1(3) with IPCCx connected
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005957C34@USNAEXCH.na.didata.local>
Message-ID: <204817.30817.qm@web57210.mail.re3.yahoo.com>

I'm trying to figure out how to change the CCM cluster
password but discovered that CRS doesn't like that and
proceeds to refuse to talk to CCM properly. Any hints?

Paul


       
____________________________________________________________________________________Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. 
http://answers.yahoo.com/dir/?link=list&sid=396545433

From erickbe at yahoo.com  Wed May 16 21:05:10 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 16 May 2007 18:05:10 -0700 (PDT)
Subject: [cisco-voip] Having to net stop CSA versus services.msc ?
Message-ID: <482470.19128.qm@web37202.mail.mud.yahoo.com>

I have seen it get stuck stopping from the services.msc once or twice. Not as much as www getting stuck though.

----- Original Message ----
From: Jason Aarons (US) <jason.aarons at us.didata.com>
To: cisco-voip at puck.nether.net
Sent: Wednesday, May 16, 2007 5:04:00 PM
Subject: [cisco-voip] Having to net stop CSA versus services.msc ?







 

 


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Has a customer who couldn?t stop CSA via the GUI in

services.msc in 2003 Server (UCCE ICM/PG/AQ)it just showed stopping but would

never stop.  Instead we found net stop csa from dos command line worked.

Anyone seen this before? Known bug? -jason
 



  
 



  
 





 

  

  Jason

  Aarons
 

  Consultant
 

  

  

  http://www.dimensiondata.com/na
 

  904-338-3245

  cell
 

  

 





  
 



For

urgent issues notify your Project Manager, for 24x7 support contact the

Dimension Data NOC at 800-974-6584.
 



 
 














Disclaimer:



This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only.  If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful.  If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.


_______________________________________________
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From aizhar at ccbootcamp.com  Wed May 16 23:25:53 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Wed, 16 May 2007 20:25:53 -0700
Subject: [cisco-voip] Changing password on CCM 4.1(3) with IPCCx
	connected
References: <204817.30817.qm@web57210.mail.re3.yahoo.com>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FC5@nlimail.nlidomain>

What passwords are you trying to change? administrator? ccmdministrator, Directory Manager?
 
IPCC only cares about administrator/ccmadministrator and DC's Directory Manager.
 
Please provide more details and what exactly ipcc doesn't like about it so I could offer something.
 
-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take Cisco Learning credits!
http://www.ccbootcamp.com <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012


________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Paul Choi
Sent: Wed 5/16/2007 3:25 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Changing password on CCM 4.1(3) with IPCCx connected



I'm trying to figure out how to change the CCM cluster
password but discovered that CRS doesn't like that and
proceeds to refuse to talk to CCM properly. Any hints?

Paul


      
____________________________________________________________________________________Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out.
http://answers.yahoo.com/dir/?link=list&sid=396545433
_______________________________________________
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From mfa at gte.net  Wed May 16 23:38:28 2007
From: mfa at gte.net (Mike Armstrong)
Date: Wed, 16 May 2007 23:38:28 -0400
Subject: [cisco-voip] Second Button on 7940
References: <mailman.14108.1179352166.1489.cisco-voip@puck.nether.net>
Message-ID: <053701c79834$d58c6de0$1402a8c0@MFADesktop>

Or, if he wants real buttons, tell him about 7960s and 7914s (and their 
costs).  You might also point out that the higher up the management chain, 
the fewer buttons one has.  That said, I'd also love to know how to do what 
Lelio suggested.

Mike Armstrong

> Date: Wed, 16 May 2007 15:31:53 -0400
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: Re: [cisco-voip] Second Button on 7940
> To: "Todd Franklin" <toddnh65 at gmail.com>, <cisco-voip at puck.nether.net>
> Message-ID: <003c01c797f0$db7aecc0$47196883 at cfs.uoguelph.ca>
> Content-Type: text/plain; charset="iso-8859-1"
>
> The second button can be defined as a service URL. So if you create a 
> service that outputs XML and the appropriate softkeys, you can do this 
> relatively easily.
>
> That being said, I wouldn't know where to start to build this service, 
> just that if it's there, it's easy to point to.
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> ...there's no such thing as a bad timbit...
>
>  ----- Original Message ----- 
>  From: Todd Franklin
>  To: cisco-voip at puck.nether.net
>  Sent: Wednesday, May 16, 2007 3:24 PM
>  Subject: [cisco-voip] Second Button on 7940
>
>
>  My manager has a 7940 and he's complaining about his loss of buttons that 
> he had on his previous phone.
>  So I mention this to a friend, and he says "you can make it so when he 
> presses that 2nd button, he gets a list of extensions to call, and he can 
> just arrow up and down thru them".  How do I do that???  I guess the first 
> button is kind of reserved, it's your extension!!!  But how do I do that 
> with the 2nd button?  Maybe I could just have it that when he presses it, 
> he gets the names of the six department heads and their extensions.  (the 
> buttons I am talking about are the 2 to the right of the LCD screen)
>
>  Thanks anyone.


From asobihoudai at yahoo.com  Thu May 17 00:05:25 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 16 May 2007 21:05:25 -0700 (PDT)
Subject: [cisco-voip] Changing password on CCM 4.1(3) with IPCCx
	connected
In-Reply-To: <E76ACA8C13EF1847BC69538B522C3663560FC5@nlimail.nlidomain>
Message-ID: <106009.56071.qm@web57211.mail.re3.yahoo.com>

I'm trying to change the Administrator password.

Paul

--- Avner Izhar <aizhar at ccbootcamp.com> wrote:

> What passwords are you trying to change?
> administrator? ccmdministrator, Directory Manager?
>  
> IPCC only cares about administrator/ccmadministrator
> and DC's Directory Manager.
>  
> Please provide more details and what exactly ipcc
> doesn't like about it so I could offer something.
>  
> -------------------------------------------
> Regards,
> Avner Izhar
> Technical Instructor
> CCIE #15999 (Voice)
> Voice: (702) 968-5100
> Fax: (702) 446-8012
> E-mail: aizhar at ccbootcamp.com
> Network Learning Inc - A Cisco Sponsored
> Organization(SO) YES! We take Cisco Learning
> credits!
> http://www.ccbootcamp.com
>
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com>
>  (Cisco Training and Rental Racks)
> http://www.ccbootcamp.com/groupstudy.html
>
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com/groupstudy.html>
>  (groupstudy member discounts!)
> Toll Free: 877-NLI-CCIE (877-654-2243)
> Outside USA: 702-968-5100
> FAX: 702-446-8012
> 
> 
> ________________________________
> 
> From: cisco-voip-bounces at puck.nether.net on behalf
> of Paul Choi
> Sent: Wed 5/16/2007 3:25 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Changing password on CCM
> 4.1(3) with IPCCx connected
> 
> 
> 
> I'm trying to figure out how to change the CCM
> cluster
> password but discovered that CRS doesn't like that
> and
> proceeds to refuse to talk to CCM properly. Any
> hints?
> 
> Paul
> 
> 
>       
>
____________________________________________________________________________________Be
> a better Heartthrob. Get better relationship answers
> from someone who knows. Yahoo! Answers - Check it
> out.
>
http://answers.yahoo.com/dir/?link=list&sid=396545433
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 



       
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From aizhar at ccbootcamp.com  Thu May 17 00:16:31 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Wed, 16 May 2007 21:16:31 -0700
Subject: [cisco-voip] Changing password on CCM 4.1(3) with IPCCx
	connected
References: <106009.56071.qm@web57211.mail.re3.yahoo.com>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FC8@nlimail.nlidomain>

And what do you get from ipcc that doesn't allow it?
 
ipcc version 4.0x will use this user name to create ports and users, is that what you are experiencing?
 
-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take Cisco Learning credits!
http://www.ccbootcamp.com <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012


________________________________

From: Paul Choi [mailto:asobihoudai at yahoo.com]
Sent: Wed 5/16/2007 9:05 PM
To: Avner Izhar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Changing password on CCM 4.1(3) with IPCCx connected



I'm trying to change the Administrator password.

Paul

--- Avner Izhar <aizhar at ccbootcamp.com> wrote:

> What passwords are you trying to change?
> administrator? ccmdministrator, Directory Manager?
> 
> IPCC only cares about administrator/ccmadministrator
> and DC's Directory Manager.
> 
> Please provide more details and what exactly ipcc
> doesn't like about it so I could offer something.
> 
> -------------------------------------------
> Regards,
> Avner Izhar
> Technical Instructor
> CCIE #15999 (Voice)
> Voice: (702) 968-5100
> Fax: (702) 446-8012
> E-mail: aizhar at ccbootcamp.com
> Network Learning Inc - A Cisco Sponsored
> Organization(SO) YES! We take Cisco Learning
> credits!
> http://www.ccbootcamp.com <http://www.ccbootcamp.com/> 
>
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com>
>  (Cisco Training and Rental Racks)
> http://www.ccbootcamp.com/groupstudy.html
>
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com/groupstudy.html>
>  (groupstudy member discounts!)
> Toll Free: 877-NLI-CCIE (877-654-2243)
> Outside USA: 702-968-5100
> FAX: 702-446-8012
>
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net on behalf
> of Paul Choi
> Sent: Wed 5/16/2007 3:25 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Changing password on CCM
> 4.1(3) with IPCCx connected
>
>
>
> I'm trying to figure out how to change the CCM
> cluster
> password but discovered that CRS doesn't like that
> and
> proceeds to refuse to talk to CCM properly. Any
> hints?
>
> Paul
>
>
>      
>
____________________________________________________________________________________Be
> a better Heartthrob. Get better relationship answers
> from someone who knows. Yahoo! Answers - Check it
> out.
>
http://answers.yahoo.com/dir/?link=list&sid=396545433
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From sdjokic at mds.co.yu  Thu May 17 04:01:10 2007
From: sdjokic at mds.co.yu (Djokic Sinisa)
Date: Thu, 17 May 2007 10:01:10 +0200
Subject: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!
Message-ID: <200705170801.l4H818EM017488@puck.nether.net>

hi to all..

 

has anybody experienced the following problem:

 

ccm4.1(3) - we have gatekeeper configured on ccm and intercluster trunk whis
is controlled via that gatekeeper ( GNU )..

there is also a route pattern for certain video calls to go from CCM side
over that intercluster trunk to TANDBERG video terminal ( which is registred
to the same GNU gatekeeper )..

the phone and the intercluster trunk are in the same device pool ( g711 )..

when user  ( with IP phone and VTA ) makes a call to that video number and
succesfull video call is established..

no media resource group list are used at all..

 

ccm4.2(1) - the same setup as before is copied to the new CCM 4.2 in lab..

everything is the same ( including the same gatekeeper ) but it doesn't
work..

there are two things..

if we put that device pool ( the one we're using ) uses media resource group
list a call is established but without video..

if we don't put that device pool ( the one we're using ) uses media resource
group list a call is droped as soon it is established..

 

 

 

does anybody have an idea about this strange behaviour..

is there something we should know about CCM4.2 or is it u bug..

 

regards..

 

Sinisa Djokic 

System Engineer
CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 




  

MDS Informaticki inzenjering
Bul. Arsenija Carnojevica 170
11070 Novi Beograd, Serbia
Tel: +381 11 2015 200, 2015 273
Fax: +381 11 3194 954
www.mds.co.yu
sdjokic at mds.co.yu

 

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From matt at saskin.net  Thu May 17 06:05:20 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 17 May 2007 06:05:20 -0400
Subject: [cisco-voip] Second Button on 7940
In-Reply-To: <053701c79834$d58c6de0$1402a8c0@MFADesktop>
References: <mailman.14108.1179352166.1489.cisco-voip@puck.nether.net>
	<053701c79834$d58c6de0$1402a8c0@MFADesktop>
Message-ID: <464C28E0.7060103@saskin.net>

It's pretty straight-forward to do what Lelio suggested as a phone 
service.  I've got a boatload of 7941's deployed currently, maybe it's 
worth my time to take an hour or two and cobble together a service like 
this.  If I've got some free time to get it done I'll put the source 
code online somewhere.

-matt

Mike Armstrong wrote:
> Or, if he wants real buttons, tell him about 7960s and 7914s (and their 
> costs).  You might also point out that the higher up the management chain, 
> the fewer buttons one has.  That said, I'd also love to know how to do what 
> Lelio suggested.
>
> Mike Armstrong
>
>   
>> Date: Wed, 16 May 2007 15:31:53 -0400
>> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
>> Subject: Re: [cisco-voip] Second Button on 7940
>> To: "Todd Franklin" <toddnh65 at gmail.com>, <cisco-voip at puck.nether.net>
>> Message-ID: <003c01c797f0$db7aecc0$47196883 at cfs.uoguelph.ca>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> The second button can be defined as a service URL. So if you create a 
>> service that outputs XML and the appropriate softkeys, you can do this 
>> relatively easily.
>>
>> That being said, I wouldn't know where to start to build this service, 
>> just that if it's there, it's easy to point to.
>>
>> --------------------------------------------------------------------------------
>> Lelio Fulgenzi, B.A.
>> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
>> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>> ...there's no such thing as a bad timbit...
>>
>>  ----- Original Message ----- 
>>  From: Todd Franklin
>>  To: cisco-voip at puck.nether.net
>>  Sent: Wednesday, May 16, 2007 3:24 PM
>>  Subject: [cisco-voip] Second Button on 7940
>>
>>
>>  My manager has a 7940 and he's complaining about his loss of buttons that 
>> he had on his previous phone.
>>  So I mention this to a friend, and he says "you can make it so when he 
>> presses that 2nd button, he gets a list of extensions to call, and he can 
>> just arrow up and down thru them".  How do I do that???  I guess the first 
>> button is kind of reserved, it's your extension!!!  But how do I do that 
>> with the 2nd button?  Maybe I could just have it that when he presses it, 
>> he gets the names of the six department heads and their extensions.  (the 
>> buttons I am talking about are the 2 to the right of the LCD screen)
>>
>>  Thanks anyone.
>>     
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From akoul at dataways.gr  Thu May 17 07:34:59 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 17 May 2007 14:34:59 +0300
Subject: [cisco-voip] Regarding Pickup and CSCsb42763 Engineer Fix
Message-ID: <464C3DE3.4070807@dataways.gr>

Is there a cisco documented place where i can find more for this 
feature: pickup a line ringing a BROADCAST Line Group when my Extension 
is NOT on this Line Group but i am on the same Call Pickup Group with 
the phones ringing.

I have tried CSCsb42763 on a 5.0(4a) and it turned to a mess.
I couldn't turn the feature off!!
I disabled it from the Hunt Pilot because it didn't work, but then i 
wasn't able to pickup ONE ringing extension from the same Pickup group!!
I kept getting "No call Availlable for Pickup".

So where can i find more about this?

Thanks a lot, Anthony

From Nick.Kassel at Charles-Stanley.co.uk  Thu May 17 08:09:45 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 17 May 2007 13:09:45 +0100
Subject: [cisco-voip] IPCC Express for 5.1
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451EF4@CSEXCH01.charles-stanley.co.uk>

I have been asked to investigate getting IPCC up and running for our
helpdesk, we are running 5.1(1)b, now I know that IPCC 4.5SR2 is the
version for 5.X, but I see that there is also a Data Sheet for IPCC
Express 5.0
http://www.cisco.com/en/US/customer/products/sw/custcosw/ps1846/products
_data_sheet0900aecd805e1d54.html 

 

That's all I can find on CCO for that version and it does not state what
version of Call Manager it's for, does anyone have any further
information on this version IE is it Windows / Linux etc or is this
going to be the version for CM 6.0

 

As soon as it's available we will want to move to 6.0 so I don't want to
install a version of IPCC now and find it will need to be upgraded when
we eventually go to 6.0. 

 

Any info would be greatly appreciated.

 

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From lelio at uoguelph.ca  Thu May 17 09:40:34 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 17 May 2007 09:40:34 -0400
Subject: [cisco-voip] SRST, MGCP -> h.323 and trunk groups
Message-ID: <007f01c79888$f20cd550$47196883@cfs.uoguelph.ca>

Has anyone out there got trunk groups on T1s working during SRST mode?

For some reason, after reboot, the trunk groups no longer work when SRST is triggered. I have removed the ccm-manager config statements so the actual trunk-group commands are there on the T1 controllers, but things don't work until I actually remove the trunk group (in global config) and add the commands back in.




--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...
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From tennille.spence at gsa.gov  Thu May 17 09:35:39 2007
From: tennille.spence at gsa.gov (tennille.spence at gsa.gov)
Date: Thu, 17 May 2007 08:35:39 -0500
Subject: [cisco-voip] voice monitoring tool
In-Reply-To: <94a1afde0705161617j3330cebcw67e5c593caa5430b@mail.gmail.com>
Message-ID: <OF5F70B0FD.C92752CC-ON862572DE.004A124F-862572DE.004AAD1F@gsa.gov>


We had the qovia system and while applying security patches the application
broke.  When I attempted to contact qovia to purchase support I found their
website down and number disconnected. I had heard that cisco had bought
them out entirely but when I asked our cisco rep he responded that they had
not bought them but did purchase certain assets in a fire sale.
http://www.gazette.net/stories/031507/busiflo200830_32322.shtml .I assume
these assets are being incorporated into the cisco voice monitoring tools


                                                                           
                                                                           
             "Ed Leatherman"                                               
             <ealeatherman at gma                                          To 
             il.com>                   "Ruttman, Pete G."                  
             Sent by:                  <PRuttman at foley.com>                
             cisco-voip-bounce                                          cc 
             s at puck.nether.net         cisco-voip at puck.nether.net          
                                                                   Subject 
                                       Re: [cisco-voip] voice monitoring   
             05/16/2007 06:17          tool                                
             PM                                                            
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           




Guess that's why i never heard back from the sales guy after they offered
us a demo!

On 5/16/07, Ruttman, Pete G. <PRuttman at foley.com> wrote:
  We were doing an onsite eval of Qovia.  Right in the middle of it Cisco
  bought out some part of Qovia and the rest of the company was let go.  I
  think Cisco is still reselling the lower end models but I don't know
  where it is really all going now.





  Pete








  From: cisco-voip-bounces at puck.nether.net [mailto:
  cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
  Sent: Tuesday, May 15, 2007 7:15 AM
  To: kevin k
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] voice monitoring tool





  I've heard good things about Qovia, I think Cisco actually resells some
  of their network probes.





  On 5/15/07, kevin k < kds20850 at yahoo.com> wrote:


  real time monitoring and baselining.


  something for enterprise.





  excluding RTMT.


  ----- Original Message ----
  From: hemant kumar < hemant.taan at gmail.com>
  To: kevin k <kds20850 at yahoo.com>
  Cc: cisco-voip at puck.nether.net
  Sent: Tuesday, May 15, 2007 6:05:34 AM
  Subject: Re: [cisco-voip] voice monitoring tool

  monitor means real time monitoring ? or recording of voice calls?



  On 5/15/07, kevin k < kds20850 at yahoo.com > wrote:


  i want to monitor voice network.


  what tool do you recommend?





  thanks in advance.











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  Yahoo! Small Business gives you all the tools to get online.



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  https://puck.nether.net/mailman/listinfo/cisco-voip














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  _______________________________________________
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  https://puck.nether.net/mailman/listinfo/cisco-voip





  --
  Ed Leatherman
  Senior Voice Engineer
  West Virginia University
  Telecommunications and Network Operations

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West Virginia University
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From balbler at nts.at  Thu May 17 10:32:43 2007
From: balbler at nts.at (Bernhard Albler)
Date: Thu, 17 May 2007 16:32:43 +0200
Subject: [cisco-voip] IPCC Express for 5.1
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451EF4@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451EF4@CSEXCH01.charles-stanley.co.uk>
Message-ID: <22202927000062df@nts.at>

Hi Nick,

IPCC-X 4.5 definitely works with CM 5.1, I have customers running this combo. IPCC-X 5.0 is a new version that is part of the Ohio Launch (CM6 and so on).
It offers quite a few new features over 4.5, including integrated outbound, Reskilling from Sup Desktop(I believe it if I see it!), Web Collab option etc. etc. etc.

It's still running on Windows, I am not sure if it is 2k3 only or W2k also (4.1 being the first version for W2k3)
It is supposed to be compatible with CM 5.1 and CM6

Regards
bernhard

------------------------------------------
DI (FH) Bernhard Albler, NTS AG 
Tel.: +43 (316) 405 455 -0, email:?balbler at nts.at?


From aizhar at ccbootcamp.com  Thu May 17 10:38:47 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Thu, 17 May 2007 07:38:47 -0700
Subject: [cisco-voip] SRST, MGCP -> h.323 and trunk groups
References: <007f01c79888$f20cd550$47196883@cfs.uoguelph.ca>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FCD@nlimail.nlidomain>

Should work,
 
Do you have your 'ccm-manager fallback mgcp' and h.323 dial peers set?
 
Try posting the output of 'show ccm-manager' when you are in srst mode for further review.
 
-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take Cisco Learning credits!
http://www.ccbootcamp.com <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012


________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
Sent: Thu 5/17/2007 6:40 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SRST, MGCP -> h.323 and trunk groups



Has anyone out there got trunk groups on T1s working during SRST mode?

For some reason, after reboot, the trunk groups no longer work when SRST is triggered. I have removed the ccm-manager config statements so the actual trunk-group commands are there on the T1 controllers, but things don't work until I actually remove the trunk group (in global config) and add the commands back in.




--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
...there's no such thing as a bad timbit...

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From Nick.Kassel at Charles-Stanley.co.uk  Thu May 17 10:51:12 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 17 May 2007 15:51:12 +0100
Subject: [cisco-voip] IPCC Express for 5.1
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451EF7@CSEXCH01.charles-stanley.co.uk>

Excellent many thanks Bernard that helps me a lot.

-----Original Message-----
From: Bernhard Albler [mailto:balbler at nts.at] 
Sent: 17 May 2007 15:33
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: AW: [cisco-voip] IPCC Express for 5.1

Hi Nick,

IPCC-X 4.5 definitely works with CM 5.1, I have customers running this combo. IPCC-X 5.0 is a new version that is part of the Ohio Launch (CM6 and so on).
It offers quite a few new features over 4.5, including integrated outbound, Reskilling from Sup Desktop(I believe it if I see it!), Web Collab option etc. etc. etc.

It's still running on Windows, I am not sure if it is 2k3 only or W2k also (4.1 being the first version for W2k3)
It is supposed to be compatible with CM 5.1 and CM6

Regards
bernhard

------------------------------------------
DI (FH) Bernhard Albler, NTS AG 
Tel.: +43 (316) 405 455 -0, email:?balbler at nts.at?

********************SAVE PAPER - THINK BEFORE YOU PRINT**************************************************

The information contained in this e-mail is strictly confidential, some or all of which may be legally privileged. 
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delivery receipt does not constitute acknowledgement or receipt by the intended recipient(s). 

You acknowledge that e-mails are not secure and you accept the risk of malfunction, viruses, unauthorised 
interference, mis-delivery or delay. Charles Stanley reserves the right to monitor and/or record emails sent and
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*******************************************************************************************************


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Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
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Charles Stanley Sutherlands and Charles Stanley Securities are divisions of Charles Stanley & Co. Ltd  

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This footnote also confirms that this email message has been swept by McAfee VirusScan and SurfControl Email 
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From rratliff at cisco.com  Thu May 17 12:14:31 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 17 May 2007 12:14:31 -0400
Subject: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!
In-Reply-To: <200705170801.l4H818EM017488@puck.nether.net>
References: <200705170801.l4H818EM017488@puck.nether.net>
Message-ID: <263BB3CF-6383-412E-9D80-240C0C6B5F67@cisco.com>

When a call is dropped as soon as it goes to a connected state this  
usually indicates a codec mismatch, or some other media establishment  
error.

The fact that the call connects with no video when you use the MRGL  
tells me that either an MTP or transcoder is being invoked for that  
call.  Since we have no MTP or Transcoders that support video, you  
get audio only.

You can verify this by looking at the Perfmon counters in CCM, by  
hitting the IP phone's web page and looking at the active RTP stream  
(when the audio only call is up) to see the IP address it is sending  
RTP to, or by examining the CCM trace.

-Ryan

On May 17, 2007, at 4:01 AM, Djokic Sinisa wrote:

hi to all..



has anybody experienced the following problem:



ccm4.1(3) ? we have gatekeeper configured on ccm and intercluster  
trunk whis is controlled via that gatekeeper ( GNU )..

there is also a route pattern for certain video calls to go from CCM  
side over that intercluster trunk to TANDBERG video terminal ( which  
is registred to the same GNU gatekeeper )..

the phone and the intercluster trunk are in the same device pool  
( g711 )..

when user  ( with IP phone and VTA ) makes a call to that video  
number and succesfull video call is established..

no media resource group list are used at all..



ccm4.2(1) - the same setup as before is copied to the new CCM 4.2 in  
lab..

everything is the same ( including the same gatekeeper ) but it  
doesn?t work..

there are two things..

if we put that device pool ( the one we?re using ) uses media  
resource group list a call is established but without video..

if we don?t put that device pool ( the one we?re using ) uses media  
resource group list a call is droped as soon it is established..







does anybody have an idea about this strange behaviour..

is there something we should know about CCM4.2 or is it u bug..



regards..



Sinisa Djokic

System Engineer
CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified



<image001.gif>



MDS Informaticki inzenjering
Bul. Arsenija Carnojevica 170
11070 Novi Beograd, Serbia
Tel: +381 11 2015 200, 2015 273
Fax: +381 11 3194 954
www.mds.co.yu
sdjokic at mds.co.yu



<image001.gif>
_______________________________________________
cisco-voip mailing list
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From rratliff at cisco.com  Thu May 17 12:15:40 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 17 May 2007 12:15:40 -0400
Subject: [cisco-voip] Having to net stop CSA versus services.msc ?
In-Reply-To: <482470.19128.qm@web37202.mail.mud.yahoo.com>
References: <482470.19128.qm@web37202.mail.mud.yahoo.com>
Message-ID: <5DBC4090-33D8-4E2D-8B1B-21322BC9C324@cisco.com>

When this happens with any recent CSA version it means there is an  
active terminal service session where the CSA popup window requesting  
confirmation that you want to shut down the service is waiting for  
user input.

-Ryan

On May 16, 2007, at 9:05 PM, Erick Bergquist wrote:

I have seen it get stuck stopping from the services.msc once or  
twice. Not as much as www getting stuck though.

----- Original Message ----
From: Jason Aarons (US) <jason.aarons at us.didata.com>
To: cisco-voip at puck.nether.net
Sent: Wednesday, May 16, 2007 5:04:00 PM
Subject: [cisco-voip] Having to net stop CSA versus services.msc ?

Has a customer who couldn?t stop CSA via the GUI in services.msc in  
2003 Server (UCCE ICM/PG/AQ)it just showed stopping but would never  
stop.  Instead we found net stop csa from dos command line worked.  
Anyone seen this before? Known bug? -jason



Jason Aarons

Consultant

http://www.dimensiondata.com/na

904-338-3245 cell


For urgent issues notify your Project Manager, for 24x7 support  
contact the Dimension Data NOC at 800-974-6584.




Disclaimer: This e-mail communication and any attachments may contain  
confidential and privileged information and is for use by the  
designated addressee(s) named above only. If you are not the intended  
addressee, you are hereby notified that you have received this  
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or its contents is strictly prohibited and may be unlawful. If you  
have received this communication in error, please notify us  
immediately by replying to this message and deleting it from your  
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From bcarter at sentinel.com  Thu May 17 17:14:18 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 17 May 2007 16:14:18 -0500
Subject: [cisco-voip] voice monitoring tool
In-Reply-To: <OF5F70B0FD.C92752CC-ON862572DE.004A124F-862572DE.004AAD1F@gsa.gov>
References: <94a1afde0705161617j3330cebcw67e5c593caa5430b@mail.gmail.com>
	<OF5F70B0FD.C92752CC-ON862572DE.004A124F-862572DE.004AAD1F@gsa.gov>
Message-ID: <6A0744639B86DD459433108DBF101D26016AE8FD@MAIL01.sentinel.com>

You may have noticed Cisco came out with several VoIP monitoring tools
about the same time Qovia closed their doors. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
tennille.spence at gsa.gov
Sent: Thursday, May 17, 2007 8:36 AM
To: ealeatherman at gmail.com
Cc: cisco-voip-bounces at puck.nether.net; Ruttman, Pete G.;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] voice monitoring tool


We had the qovia system and while applying security patches the
application broke.  When I attempted to contact qovia to purchase
support I found their website down and number disconnected. I had heard
that cisco had bought them out entirely but when I asked our cisco rep
he responded that they had not bought them but did purchase certain
assets in a fire sale.
http://www.gazette.net/stories/031507/busiflo200830_32322.shtml .I
assume these assets are being incorporated into the cisco voice
monitoring tools


 

 

             "Ed Leatherman"

             <ealeatherman at gma
To 
             il.com>                   "Ruttman, Pete G."

             Sent by:                  <PRuttman at foley.com>

             cisco-voip-bounce
cc 
             s at puck.nether.net         cisco-voip at puck.nether.net

 
Subject 
                                       Re: [cisco-voip] voice monitoring

             05/16/2007 06:17          tool

             PM

 

 

 

 

 





Guess that's why i never heard back from the sales guy after they
offered us a demo!

On 5/16/07, Ruttman, Pete G. <PRuttman at foley.com> wrote:
  We were doing an onsite eval of Qovia.  Right in the middle of it
Cisco
  bought out some part of Qovia and the rest of the company was let go.
I
  think Cisco is still reselling the lower end models but I don't know
  where it is really all going now.





  Pete








  From: cisco-voip-bounces at puck.nether.net [mailto:
  cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
  Sent: Tuesday, May 15, 2007 7:15 AM
  To: kevin k
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] voice monitoring tool





  I've heard good things about Qovia, I think Cisco actually resells
some
  of their network probes.





  On 5/15/07, kevin k < kds20850 at yahoo.com> wrote:


  real time monitoring and baselining.


  something for enterprise.





  excluding RTMT.


  ----- Original Message ----
  From: hemant kumar < hemant.taan at gmail.com>
  To: kevin k <kds20850 at yahoo.com>
  Cc: cisco-voip at puck.nether.net
  Sent: Tuesday, May 15, 2007 6:05:34 AM
  Subject: Re: [cisco-voip] voice monitoring tool

  monitor means real time monitoring ? or recording of voice calls?



  On 5/15/07, kevin k < kds20850 at yahoo.com > wrote:


  i want to monitor voice network.


  what tool do you recommend?





  thanks in advance.











  Building a website is a piece of cake.
  Yahoo! Small Business gives you all the tools to get online.



  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip














  Pinpoint customers who are looking for what you sell.



  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip





  --
  Ed Leatherman
  Senior Voice Engineer
  West Virginia University
  Telecommunications and Network Operations

  The preceding email message may be confidential or protected by the
  attorney-client privilege. It is not intended for transmission to, or
  receipt by, any unauthorized persons. If you have received this
message
  in error, please (i) do not read it, (ii) reply to the sender that you
  received the message in error, and (iii) erase or destroy the message.
  Legal advice contained in the preceding message is solely for the
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  of the Foley & Lardner LLP client(s) represented by the Firm in the
  particular matter that is the subject of this message, and may not be
  relied upon by any other party.


  Internal Revenue Service regulations require that certain types of
  written advice include a disclaimer. To the extent the preceding
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  contains advice relating to a Federal tax issue, unless expressly
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  otherwise the advice is not intended or written to be used, and it
cannot
  be used by the recipient or any other taxpayer, for the purpose of
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  _______________________________________________
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--
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
_______________________________________________
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From spackett at gmail.com  Thu May 17 17:15:19 2007
From: spackett at gmail.com (Stu Packett)
Date: Thu, 17 May 2007 14:15:19 -0700
Subject: [cisco-voip] Regular 911 Testing
Message-ID: <feb44fca0705171415x49bf9dabvea9837f081cf4192@mail.gmail.com>

Do any of you do regular 911 testing?  For us, we do a monthly 911 call, but
I wanted to know if there was a more effective way of ensuring that 911
calls will go out.  What methods are you using?  Thanks in advance.
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From bcarter at sentinel.com  Thu May 17 17:15:57 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 17 May 2007 16:15:57 -0500
Subject: [cisco-voip] How to send unconfigured DIDs to an auto-attendant?
In-Reply-To: <CB4BC8C24EF6E149AED9FDB9B9AF57133D8BAC@MAIL02.hq.data-corporation.com>
References: <46262932FF67F74CA6D537C31A6969870D4893D6@endeavour.InfoNetwork>
	<CB4BC8C24EF6E149AED9FDB9B9AF57133D8BAC@MAIL02.hq.data-corporation.com>
Message-ID: <6A0744639B86DD459433108DBF101D26016AE8FE@MAIL01.sentinel.com>

yes, XXXX is the way to do it.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Vince Loschiavo
Sent: Wednesday, May 16, 2007 2:05 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How to send unconfigured DIDs to an
auto-attendant?


CCM 5.X, Unity 4.2 Unified Messaging, MGCP Gateways,
 
How can I send unconfigured DID's to an auto-attendant.
 
The customer has a block of numbers (xxx) XXX-1000 through 2000.  But
only 200 of the DID's are in use.  Right now if someone dials a number
that is not in use, the caller gets a re-order tone (fast busy).  How
can I forward those (in bulk) to the Auto-Attendant without configuring
CTI Route points for each DID?
 
I suspect that a CTI Route Point with a number of "XXXX" should do the
trick, but I haven't tested.  
 
Has anyone done this before?
 
____________________________________________________________
Vincent Loschiavo
Senior Consultant
Datacorp
18503 Pines Blvd
Suite 212
Pembroke Pines, FL, 33029

"Cisco Advanced Unified Communications Partner"
Cell Phone:        786-282-1164
Office Phone:    305-594-6933
____________________________________________________________ 
 
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From bcarter at sentinel.com  Thu May 17 17:17:54 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 17 May 2007 16:17:54 -0500
Subject: [cisco-voip] Regular 911 Testing
In-Reply-To: <feb44fca0705171415x49bf9dabvea9837f081cf4192@mail.gmail.com>
References: <feb44fca0705171415x49bf9dabvea9837f081cf4192@mail.gmail.com>
Message-ID: <6A0744639B86DD459433108DBF101D26016AE8FF@MAIL01.sentinel.com>

You can change the route patterns to instead of the call going to 911 it
goes to your cell phone.  This way you can see caller ID etc, but don't
upset the 911 center.
 
Cisco Operation Manager will simulate 911 calls to verify functionality.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, May 17, 2007 4:15 PM
To: ciscovoip
Subject: [cisco-voip] Regular 911 Testing


Do any of you do regular 911 testing?  For us, we do a monthly 911 call,
but I wanted to know if there was a more effective way of ensuring that
911 calls will go out.  What methods are you using?  Thanks in advance.

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From matt at saskin.net  Thu May 17 17:44:07 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 17 May 2007 17:44:07 -0400
Subject: [cisco-voip] Regular 911 Testing
In-Reply-To: <6A0744639B86DD459433108DBF101D26016AE8FF@MAIL01.sentinel.com>
References: <feb44fca0705171415x49bf9dabvea9837f081cf4192@mail.gmail.com>
	<6A0744639B86DD459433108DBF101D26016AE8FF@MAIL01.sentinel.com>
Message-ID: <464CCCA7.7010601@saskin.net>

Caller ID doesn't mean anything if the ALI data is wrong and the psap 
gets the wrong location to go with it :)

While I don't schedule monthly tests, I do test for every new site 
deployment as well as any time the carrier makes a change to our 
circuits at the site.

-matt

Carter, Bill wrote:
> You can change the route patterns to instead of the call going to 911 it 
> goes to your cell phone.  This way you can see caller ID etc, but don't 
> upset the 911 center.
>  
> Cisco Operation Manager will simulate 911 calls to verify functionality.
> 
> ------------------------------------------------------------------------
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Stu Packett
> *Sent:* Thursday, May 17, 2007 4:15 PM
> *To:* ciscovoip
> *Subject:* [cisco-voip] Regular 911 Testing
> 
> Do any of you do regular 911 testing?  For us, we do a monthly 911 call, 
> but I wanted to know if there was a more effective way of ensuring that 
> 911 calls will go out.  What methods are you using?  Thanks in advance.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From ealeatherman at gmail.com  Thu May 17 18:42:51 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 17 May 2007 18:42:51 -0400
Subject: [cisco-voip] Regular 911 Testing
In-Reply-To: <feb44fca0705171415x49bf9dabvea9837f081cf4192@mail.gmail.com>
References: <feb44fca0705171415x49bf9dabvea9837f081cf4192@mail.gmail.com>
Message-ID: <94a1afde0705171542n60e44dd6rcd90471cd048ae35@mail.gmail.com>

We do a test call usually as part of monthly maintenance but only if we've
patched something in the call path. for instance if all I do is patch
voicemail and antivirus one month, I won't bother the PSAP people with a 911
call.

They are very cooperative when I do call, they even help test callback
through CER when I ask them to.


On 5/17/07, Stu Packett <spackett at gmail.com> wrote:
>
> Do any of you do regular 911 testing?  For us, we do a monthly 911 call,
> but I wanted to know if there was a more effective way of ensuring that 911
> calls will go out.  What methods are you using?  Thanks in advance.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From sdjokic at mds.co.yu  Fri May 18 01:50:10 2007
From: sdjokic at mds.co.yu (Djokic Sinisa)
Date: Fri, 18 May 2007 07:50:10 +0200
Subject: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!
In-Reply-To: <263BB3CF-6383-412E-9D80-240C0C6B5F67@cisco.com>
Message-ID: <200705180550.l4I5oFmI041028@puck.nether.net>

 

thanx ryan..

 

i agree with you..that's what i'm thinking also..

the problem is there shouldn't be any codec mismatch etc..

i have two separate systems - ccm4.1 and ccm4.2, which are configured
identically..

both using the same gatekeeper..

one of them works - ccm4.1..

another doesn't - ccm4.2..

all the device pools use g711 codec..

so, that's the reason i was asking have anybody experienced something like
that ( maybe it's a bug ) or something..

 

thanx..

 

regards..

 

Sinisa Djokic 

System Engineer

CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 

MDS Informaticki inzenjering

Bul. Arsenija Carnojevica 170

11070 Novi Beograd, Serbia

Tel: +381 11 2015 200, 2015 273

Fax: +381 11 3194 954

www.mds.co.yu

sdjokic at mds.co.yu

      

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Thursday, May 17, 2007 6:15 PM
To: Djokic Sinisa
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!

 

When a call is dropped as soon as it goes to a connected state this  

usually indicates a codec mismatch, or some other media establishment  

error.

 

The fact that the call connects with no video when you use the MRGL  

tells me that either an MTP or transcoder is being invoked for that  

call.  Since we have no MTP or Transcoders that support video, you  

get audio only.

 

You can verify this by looking at the Perfmon counters in CCM, by  

hitting the IP phone's web page and looking at the active RTP stream  

(when the audio only call is up) to see the IP address it is sending  

RTP to, or by examining the CCM trace.

 

-Ryan

 

On May 17, 2007, at 4:01 AM, Djokic Sinisa wrote:

 

hi to all..

 

 

 

has anybody experienced the following problem:

 

 

 

ccm4.1(3) - we have gatekeeper configured on ccm and intercluster  

trunk whis is controlled via that gatekeeper ( GNU )..

 

there is also a route pattern for certain video calls to go from CCM  

side over that intercluster trunk to TANDBERG video terminal ( which  

is registred to the same GNU gatekeeper )..

 

the phone and the intercluster trunk are in the same device pool  

( g711 )..

 

when user  ( with IP phone and VTA ) makes a call to that video  

number and succesfull video call is established..

 

no media resource group list are used at all..

 

 

 

ccm4.2(1) - the same setup as before is copied to the new CCM 4.2 in  

lab..

 

everything is the same ( including the same gatekeeper ) but it  

doesn't work..

 

there are two things..

 

if we put that device pool ( the one we're using ) uses media  

resource group list a call is established but without video..

 

if we don't put that device pool ( the one we're using ) uses media  

resource group list a call is droped as soon it is established..

 

 

 

 

 

 

 

does anybody have an idea about this strange behaviour..

 

is there something we should know about CCM4.2 or is it u bug..

 

 

 

regards..

 

 

 

Sinisa Djokic

 

System Engineer

CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 

 

 

<image001.gif>

 

 

 

MDS Informaticki inzenjering

Bul. Arsenija Carnojevica 170

11070 Novi Beograd, Serbia

Tel: +381 11 2015 200, 2015 273

Fax: +381 11 3194 954

www.mds.co.yu

sdjokic at mds.co.yu

 

 

 

<image001.gif>

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cisco-voip mailing list

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From list at monach.com  Fri May 18 04:12:55 2007
From: list at monach.com (Rob)
Date: Fri, 18 May 2007 16:12:55 +0800
Subject: [cisco-voip] SIP extensions and JTAPI
Message-ID: <006c01c79924$58119c50$0834d4f0$@com>

Hi

 

Can anyone tell me whether I can monitor a SIP extension through JTAPI?

 

Basically we currently have Cisco softphones connected to CCM 3.3. We have
an application that monitors the softphones through JTAPI. We're now looking
to swap the Cisco softphone for a SIP softphone. My understanding is that we
will require to register the softphones with a SIP Proxy and configure a SIP
trunk in CCM (we will upgrade to v4). My question is, can we monitor the SIP
extension from JTAPI in the same manner that we could using the Cisco
softphone? That is, make a call, drop a call, receive call progress
messages, etc.

 

Thanks for your help,

 

Rob

 

 

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From mmilanovic at ctiusa.com  Fri May 18 08:36:28 2007
From: mmilanovic at ctiusa.com (Mladen Milanovic)
Date: Fri, 18 May 2007 08:36:28 -0400
Subject: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!
In-Reply-To: <200705180550.l4I5oFmI041028@puck.nether.net>
References: <263BB3CF-6383-412E-9D80-240C0C6B5F67@cisco.com>
	<200705180550.l4I5oFmI041028@puck.nether.net>
Message-ID: <4761C60AD1E74A4D8606CAF48582265A6EFFB2@EXCHANGE.ctiusa.com>

Sinisa,
what do you have in the CCM traces. There should be the reason why calls
are failing. Also, do you have MTP check box "checked" on the H.323 GK
controled trunk
 
Do you have any debugging on GNU. If not, put Cisco IOS router and make
another testing with debugging turned on.
 
If call fails because of codec mismatch you should see that in the h.323
logs. H.245 negotiation will failand you will have description. Also,
you may put a ethereal and capture the packeks.
 
Also, double check do you have fast start enable on the CCM 4.2. I think
that this is a default for 4.2(1). Mabye GNU has a trouble in Media
Negotiation process with the fast start. I am not sure but I think that
fast start is enabled on the CCM 4.2 by default. Try to disable it and
make testing and capture the logs.
 
 
Mladen

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Djokic Sinisa
Sent: Friday, May 18, 2007 1:50 AM
To: 'Ryan Ratliff'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm4.2 video over intercluster trunk issues
!!!



 

thanx ryan..

 

i agree with you..that's what i'm thinking also..

the problem is there shouldn't be any codec mismatch etc..

i have two separate systems - ccm4.1 and ccm4.2, which are configured
identically..

both using the same gatekeeper..

one of them works - ccm4.1..

another doesn't - ccm4.2..

all the device pools use g711 codec..

so, that's the reason i was asking have anybody experienced something
like that ( maybe it's a bug ) or something..

 

thanx..

 

regards..

 

Sinisa Djokic 

System Engineer

CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 

MDS Informaticki inzenjering

Bul. Arsenija Carnojevica 170

11070 Novi Beograd, Serbia

Tel: +381 11 2015 200, 2015 273

Fax: +381 11 3194 954

www.mds.co.yu

sdjokic at mds.co.yu

      

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Thursday, May 17, 2007 6:15 PM
To: Djokic Sinisa
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm4.2 video over intercluster trunk issues
!!!

 

When a call is dropped as soon as it goes to a connected state this  

usually indicates a codec mismatch, or some other media establishment  

error.

 

The fact that the call connects with no video when you use the MRGL  

tells me that either an MTP or transcoder is being invoked for that  

call.  Since we have no MTP or Transcoders that support video, you  

get audio only.

 

You can verify this by looking at the Perfmon counters in CCM, by  

hitting the IP phone's web page and looking at the active RTP stream  

(when the audio only call is up) to see the IP address it is sending  

RTP to, or by examining the CCM trace.

 

-Ryan

 

On May 17, 2007, at 4:01 AM, Djokic Sinisa wrote:

 

hi to all..

 

 

 

has anybody experienced the following problem:

 

 

 

ccm4.1(3) - we have gatekeeper configured on ccm and intercluster  

trunk whis is controlled via that gatekeeper ( GNU )..

 

there is also a route pattern for certain video calls to go from CCM  

side over that intercluster trunk to TANDBERG video terminal ( which  

is registred to the same GNU gatekeeper )..

 

the phone and the intercluster trunk are in the same device pool  

( g711 )..

 

when user  ( with IP phone and VTA ) makes a call to that video  

number and succesfull video call is established..

 

no media resource group list are used at all..

 

 

 

ccm4.2(1) - the same setup as before is copied to the new CCM 4.2 in  

lab..

 

everything is the same ( including the same gatekeeper ) but it  

doesn't work..

 

there are two things..

 

if we put that device pool ( the one we're using ) uses media  

resource group list a call is established but without video..

 

if we don't put that device pool ( the one we're using ) uses media  

resource group list a call is droped as soon it is established..

 

 

 

 

 

 

 

does anybody have an idea about this strange behaviour..

 

is there something we should know about CCM4.2 or is it u bug..

 

 

 

regards..

 

 

 

Sinisa Djokic

 

System Engineer

CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 

 

 

<image001.gif>

 

 

 

MDS Informaticki inzenjering

Bul. Arsenija Carnojevica 170

11070 Novi Beograd, Serbia

Tel: +381 11 2015 200, 2015 273

Fax: +381 11 3194 954

www.mds.co.yu

sdjokic at mds.co.yu

 

 

 

<image001.gif>

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

-------------- next part --------------
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From rratliff at cisco.com  Fri May 18 09:08:33 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 18 May 2007 09:08:33 -0400
Subject: [cisco-voip] SIP extensions and JTAPI
In-Reply-To: <006c01c79924$58119c50$0834d4f0$@com>
References: <006c01c79924$58119c50$0834d4f0$@com>
Message-ID: <B097D9D5-7ECF-47DB-81BB-4139C604E12A@cisco.com>

If the SIP endpoint were registering with CM (5.x) then my guess is  
you could do it.  I seriously doubt Jtapi can control an extension  
that CM only knows about via SIP trunk.

-Ryan

On May 18, 2007, at 4:12 AM, Rob wrote:

Hi



Can anyone tell me whether I can monitor a SIP extension through JTAPI?



Basically we currently have Cisco softphones connected to CCM 3.3. We  
have an application that monitors the softphones through JTAPI. We?re  
now looking to swap the Cisco softphone for a SIP softphone. My  
understanding is that we will require to register the softphones with  
a SIP Proxy and configure a SIP trunk in CCM (we will upgrade to v4).  
My question is, can we monitor the SIP extension from JTAPI in the  
same manner that we could using the Cisco softphone? That is, make a  
call, drop a call, receive call progress messages, etc.



Thanks for your help,



Rob





_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From kdupree at rmwc.edu  Fri May 18 09:20:38 2007
From: kdupree at rmwc.edu (Kathy Dupree)
Date: Fri, 18 May 2007 09:20:38 -0400
Subject: [cisco-voip] importing groups in to Unity
Message-ID: <6E497ADB607656479C24E6D7BF6B505A0840CED3@exchange.rmwc.edu>

Can someone tell me where to find instructions on importing a group into
Unity?  I don't how to import from exchange one user at a time but I
will have groups of users to import for the coming school year.  Thanks

 

Katherine O. Dupree

Randolph-Macon Woman's College

Main 243

Telecommunications Support/Infrastructure Coordinator

kdupree at rmwc.edu <mailto:kdupree at rmwc.edu> 

(434) 947 8371

 

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From sdjokic at mds.co.yu  Fri May 18 09:24:05 2007
From: sdjokic at mds.co.yu (Djokic Sinisa)
Date: Fri, 18 May 2007 15:24:05 +0200
Subject: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!
In-Reply-To: <4761C60AD1E74A4D8606CAF48582265A6EFFB2@EXCHANGE.ctiusa.com>
Message-ID: <200705181324.l4IDO8rU063161@puck.nether.net>

o.k..

 

thanx mladen..

 

well, fast start is not enabled.. as in CCM 4.1..

 

MTP check box is not checked as well.. as in CCM 4.1..

 

two configs on both ccms are exactly the same..

 

to tell you the truth, i was just trying to replicate the situation a.s.a.p.
( i haven't got time to test it with IOS etc. ) since this one is working
just fine..

 

so, i didn't involved in deeper troubleshooting for now..it looks like i'll
have to..:-)..

 

as i said..if it's not a bug, it must been some media communication
differencies in two ccm versions ( ili je kolega u labu nesto uradio sto jos
ne mogu da dokucim - :-) )..

 

anyway i'll doublecheck everything again and  proceed with further testing
and let you know as soon as find time to do it..

 

thanx..

 

regards..

 

 

Sinisa Djokic 

System Engineer
CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 




  

MDS Informaticki inzenjering
Bul. Arsenija Carnojevica 170
11070 Novi Beograd, Serbia
Tel: +381 11 2015 200, 2015 273
Fax: +381 11 3194 954
www.mds.co.yu
sdjokic at mds.co.yu

 

  _____  

From: Mladen Milanovic [mailto:mmilanovic at ctiusa.com] 
Sent: Friday, May 18, 2007 2:36 PM
To: Djokic Sinisa; Ryan Ratliff
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!

 

Sinisa,

what do you have in the CCM traces. There should be the reason why calls are
failing. Also, do you have MTP check box "checked" on the H.323 GK controled
trunk

 

Do you have any debugging on GNU. If not, put Cisco IOS router and make
another testing with debugging turned on.

 

If call fails because of codec mismatch you should see that in the h.323
logs. H.245 negotiation will failand you will have description. Also, you
may put a ethereal and capture the packeks.

 

Also, double check do you have fast start enable on the CCM 4.2. I think
that this is a default for 4.2(1). Mabye GNU has a trouble in Media
Negotiation process with the fast start. I am not sure but I think that fast
start is enabled on the CCM 4.2 by default. Try to disable it and make
testing and capture the logs.

 

 

Mladen

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Djokic Sinisa
Sent: Friday, May 18, 2007 1:50 AM
To: 'Ryan Ratliff'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!

 

thanx ryan..

 

i agree with you..that's what i'm thinking also..

the problem is there shouldn't be any codec mismatch etc..

i have two separate systems - ccm4.1 and ccm4.2, which are configured
identically..

both using the same gatekeeper..

one of them works - ccm4.1..

another doesn't - ccm4.2..

all the device pools use g711 codec..

so, that's the reason i was asking have anybody experienced something like
that ( maybe it's a bug ) or something..

 

thanx..

 

regards..

 

Sinisa Djokic 

System Engineer

CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 

MDS Informaticki inzenjering

Bul. Arsenija Carnojevica 170

11070 Novi Beograd, Serbia

Tel: +381 11 2015 200, 2015 273

Fax: +381 11 3194 954

www.mds.co.yu

sdjokic at mds.co.yu

      

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Thursday, May 17, 2007 6:15 PM
To: Djokic Sinisa
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!

 

When a call is dropped as soon as it goes to a connected state this  

usually indicates a codec mismatch, or some other media establishment  

error.

 

The fact that the call connects with no video when you use the MRGL  

tells me that either an MTP or transcoder is being invoked for that  

call.  Since we have no MTP or Transcoders that support video, you  

get audio only.

 

You can verify this by looking at the Perfmon counters in CCM, by  

hitting the IP phone's web page and looking at the active RTP stream  

(when the audio only call is up) to see the IP address it is sending  

RTP to, or by examining the CCM trace.

 

-Ryan

 

On May 17, 2007, at 4:01 AM, Djokic Sinisa wrote:

 

hi to all..

 

 

 

has anybody experienced the following problem:

 

 

 

ccm4.1(3) - we have gatekeeper configured on ccm and intercluster  

trunk whis is controlled via that gatekeeper ( GNU )..

 

there is also a route pattern for certain video calls to go from CCM  

side over that intercluster trunk to TANDBERG video terminal ( which  

is registred to the same GNU gatekeeper )..

 

the phone and the intercluster trunk are in the same device pool  

( g711 )..

 

when user  ( with IP phone and VTA ) makes a call to that video  

number and succesfull video call is established..

 

no media resource group list are used at all..

 

 

 

ccm4.2(1) - the same setup as before is copied to the new CCM 4.2 in  

lab..

 

everything is the same ( including the same gatekeeper ) but it  

doesn't work..

 

there are two things..

 

if we put that device pool ( the one we're using ) uses media  

resource group list a call is established but without video..

 

if we don't put that device pool ( the one we're using ) uses media  

resource group list a call is droped as soon it is established..

 

 

 

 

 

 

 

does anybody have an idea about this strange behaviour..

 

is there something we should know about CCM4.2 or is it u bug..

 

 

 

regards..

 

 

 

Sinisa Djokic

 

System Engineer

CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified

 

 

 

<image001.gif>

 

 

 

MDS Informaticki inzenjering

Bul. Arsenija Carnojevica 170

11070 Novi Beograd, Serbia

Tel: +381 11 2015 200, 2015 273

Fax: +381 11 3194 954

www.mds.co.yu

sdjokic at mds.co.yu

 

 

 

<image001.gif>

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

https://puck.nether.net/mailman/listinfo/cisco-voip

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From kdupree at rmwc.edu  Fri May 18 09:24:32 2007
From: kdupree at rmwc.edu (Kathy Dupree)
Date: Fri, 18 May 2007 09:24:32 -0400
Subject: [cisco-voip] importing to Unity
Message-ID: <6E497ADB607656479C24E6D7BF6B505A0840CED8@exchange.rmwc.edu>

Sorry, I meant to say I do know how to import one user at a time.

 

Katherine O. Dupree

Randolph-Macon Woman's College

Main 243

Telecommunications Support/Infrastructure Coordinator

kdupree at rmwc.edu <mailto:kdupree at rmwc.edu> 

(434) 947 8371

 

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From matt at saskin.net  Fri May 18 09:25:49 2007
From: matt at saskin.net (Matthew Saskin)
Date: Fri, 18 May 2007 09:25:49 -0400
Subject: [cisco-voip] SIP extensions and JTAPI
In-Reply-To: <B097D9D5-7ECF-47DB-81BB-4139C604E12A@cisco.com>
References: <006c01c79924$58119c50$0834d4f0$@com>
	<B097D9D5-7ECF-47DB-81BB-4139C604E12A@cisco.com>
Message-ID: <464DA95D.7090801@saskin.net>

I would concur.  With an extension hanging off a SIP trunk, callmanager 
just knows about a route pattern to send calls across that trunk - it's 
up to the device serving as your sip proxy to handle extending the call 
to the SIP endpoint, etc., no?

-matt

Ryan Ratliff wrote:
> If the SIP endpoint were registering with CM (5.x) then my guess is  
> you could do it.  I seriously doubt Jtapi can control an extension  
> that CM only knows about via SIP trunk.
> 
> -Ryan
> 
> On May 18, 2007, at 4:12 AM, Rob wrote:
> 
> Hi
> 
> 
> 
> Can anyone tell me whether I can monitor a SIP extension through JTAPI?
> 
> 
> 
> Basically we currently have Cisco softphones connected to CCM 3.3. We  
> have an application that monitors the softphones through JTAPI. We?re  
> now looking to swap the Cisco softphone for a SIP softphone. My  
> understanding is that we will require to register the softphones with  
> a SIP Proxy and configure a SIP trunk in CCM (we will upgrade to v4).  
> My question is, can we monitor the SIP extension from JTAPI in the  
> same manner that we could using the Cisco softphone? That is, make a  
> call, drop a call, receive call progress messages, etc.
> 
> 
> 
> Thanks for your help,
> 
> 
> 
> Rob
> 
> 
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From rratliff at cisco.com  Fri May 18 09:35:50 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 18 May 2007 09:35:50 -0400
Subject: [cisco-voip] importing groups in to Unity
In-Reply-To: <6E497ADB607656479C24E6D7BF6B505A0840CED3@exchange.rmwc.edu>
References: <6E497ADB607656479C24E6D7BF6B505A0840CED3@exchange.rmwc.edu>
Message-ID: <DC8C44DE-FE10-4534-88F9-A5BDC9B32E63@cisco.com>

Check out the bulk subscriber utilities on ciscounitytools.com.   I'm  
pretty sure there's a bulk import utility either already installed on  
Unity or available at that site.

-Ryan

On May 18, 2007, at 9:20 AM, Kathy Dupree wrote:

Can someone tell me where to find instructions on importing a group  
into Unity?  I don?t how to import from exchange one user at a time  
but I will have groups of users to import for the coming school  
year.  Thanks



Katherine O. Dupree

Randolph-Macon Woman's College

Main 243

Telecommunications Support/Infrastructure Coordinator

kdupree at rmwc.edu

(434) 947 8371



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jsteinberg at gmail.com  Fri May 18 09:36:52 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri, 18 May 2007 09:36:52 -0400
Subject: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!
In-Reply-To: <200705181324.l4IDO8rU063161@puck.nether.net>
References: <4761C60AD1E74A4D8606CAF48582265A6EFFB2@EXCHANGE.ctiusa.com>
	<200705181324.l4IDO8rU063161@puck.nether.net>
Message-ID: <f1b65f880705180636i5ded68d1t5e28a50e6dba1ae@mail.gmail.com>

Just a thought here....Can you check on your CCM 4.1 server which is working
properly and see whether the CCM service parameter 'BRQ enabled' is set to
true?  By default, this parameter is false.  Possibly, this has been changed
to true on 4.1 but not on 4.2

Justin

On 5/18/07, Djokic Sinisa <sdjokic at mds.co.yu> wrote:
>
>  o.k..
>
>
>
> thanx mladen..
>
>
>
> well, fast start is not enabled.. as in CCM 4.1..
>
>
>
> MTP check box is not checked as well.. as in CCM 4.1..
>
>
>
> two configs on both ccms are exactly the same..
>
>
>
> to tell you the truth, i was just trying to replicate the situation
> a.s.a.p. ( i haven't got time to test it with IOS etc. ) since this one is
> working just fine..
>
>
>
> so, i didn't involved in deeper troubleshooting for now..it looks like
> i'll have to..J..
>
>
>
> as i said..if it's not a bug, it must been some media communication
> differencies in two ccm versions ( ili je kolega u labu nesto uradio sto jos
> ne mogu da dokucim - J )..
>
>
>
> anyway i'll doublecheck everything again and  proceed with further testing
> and let you know as soon as find time to do it..
>
>
>
> thanx..
>
>
>
> regards..
>
>
>
>
>
> *Sinisa Djokic*
>
> System Engineer
> CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified
>
>
>
>
>
> MDS Informaticki inzenjering
> Bul. Arsenija Carnojevica 170
> 11070 Novi Beograd, Serbia
> Tel: +381 11 2015 200, 2015 273
> Fax: +381 11 3194 954
> www.mds.co.yu
> sdjokic at mds.co.yu
>
>
>  ------------------------------
>
> *From:* Mladen Milanovic [mailto:mmilanovic at ctiusa.com]
> *Sent:* Friday, May 18, 2007 2:36 PM
> *To:* Djokic Sinisa; Ryan Ratliff
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] ccm4.2 video over intercluster trunk issues
> !!!
>
>
>
> Sinisa,
>
> what do you have in the CCM traces. There should be the reason why calls
> are failing. Also, do you have MTP check box "checked" on the H.323 GK
> controled trunk
>
>
>
> Do you have any debugging on GNU. If not, put Cisco IOS router and make
> another testing with debugging turned on.
>
>
>
> If call fails because of codec mismatch you should see that in the h.323logs.
> H.245 negotiation will failand you will have description. Also, you may
> put a ethereal and capture the packeks.
>
>
>
> Also, double check do you have fast start enable on the CCM 4.2. I think
> that this is a default for 4.2(1). Mabye GNU has a trouble in Media
> Negotiation process with the fast start. I am not sure but I think that fast
> start is enabled on the CCM 4.2 by default. Try to disable it and make
> testing and capture the logs.
>
>
>
>
>
> Mladen
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Djokic Sinisa
> *Sent:* Friday, May 18, 2007 1:50 AM
> *To:* 'Ryan Ratliff'
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] ccm4.2 video over intercluster trunk issues
> !!!
>
>
>
> thanx ryan..
>
>
>
> i agree with you..that's what i'm thinking also..
>
> the problem is there shouldn't be any codec mismatch etc..
>
> i have two separate systems ? ccm4.1 and ccm4.2, which are configured
> identically..
>
> both using the same gatekeeper..
>
> one of them works ? ccm4.1..
>
> another doesn't ? ccm4.2..
>
> all the device pools use g711 codec..
>
> so, that's the reason i was asking have anybody experienced something like
> that ( maybe it's a bug ) or something..
>
>
>
> thanx..
>
>
>
> regards..
>
>
>
> Sinisa Djokic
>
> System Engineer
>
> CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified
>
>
>
> MDS Informaticki inzenjering
>
> Bul. Arsenija Carnojevica 170
>
> 11070 Novi Beograd, Serbia
>
> Tel: +381 11 2015 200, 2015 273
>
> Fax: +381 11 3194 954
>
> www.mds.co.yu
>
> sdjokic at mds.co.yu
>
>
>
> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> Sent: Thursday, May 17, 2007 6:15 PM
> To: Djokic Sinisa
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] ccm4.2 video over intercluster trunk issues !!!
>
>
>
> When a call is dropped as soon as it goes to a connected state this
>
> usually indicates a codec mismatch, or some other media establishment
>
> error.
>
>
>
> The fact that the call connects with no video when you use the MRGL
>
> tells me that either an MTP or transcoder is being invoked for that
>
> call.  Since we have no MTP or Transcoders that support video, you
>
> get audio only.
>
>
>
> You can verify this by looking at the Perfmon counters in CCM, by
>
> hitting the IP phone's web page and looking at the active RTP stream
>
> (when the audio only call is up) to see the IP address it is sending
>
> RTP to, or by examining the CCM trace.
>
>
>
> -Ryan
>
>
>
> On May 17, 2007, at 4:01 AM, Djokic Sinisa wrote:
>
>
>
> hi to all..
>
>
>
>
>
>
>
> has anybody experienced the following problem:
>
>
>
>
>
>
>
> ccm4.1(3) ? we have gatekeeper configured on ccm and intercluster
>
> trunk whis is controlled via that gatekeeper ( GNU )..
>
>
>
> there is also a route pattern for certain video calls to go from CCM
>
> side over that intercluster trunk to TANDBERG video terminal ( which
>
> is registred to the same GNU gatekeeper )..
>
>
>
> the phone and the intercluster trunk are in the same device pool
>
> ( g711 )..
>
>
>
> when user  ( with IP phone and VTA ) makes a call to that video
>
> number and succesfull video call is established..
>
>
>
> no media resource group list are used at all..
>
>
>
>
>
>
>
> ccm4.2(1) - the same setup as before is copied to the new CCM 4.2 in
>
> lab..
>
>
>
> everything is the same ( including the same gatekeeper ) but it
>
> doesn't work..
>
>
>
> there are two things..
>
>
>
> if we put that device pool ( the one we're using ) uses media
>
> resource group list a call is established but without video..
>
>
>
> if we don't put that device pool ( the one we're using ) uses media
>
> resource group list a call is droped as soon it is established..
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> does anybody have an idea about this strange behaviour..
>
>
>
> is there something we should know about CCM4.2 or is it u bug..
>
>
>
>
>
>
>
> regards..
>
>
>
>
>
>
>
> Sinisa Djokic
>
>
>
> System Engineer
>
> CCNP,CCDA,CQS-CWLDS,CS-CIPTDS,CS-CIPCCES Cisco Certified
>
>
>
>
>
>
>
> <image001.gif>
>
>
>
>
>
>
>
> MDS Informaticki inzenjering
>
> Bul. Arsenija Carnojevica 170
>
> 11070 Novi Beograd, Serbia
>
> Tel: +381 11 2015 200, 2015 273
>
> Fax: +381 11 3194 954
>
> www.mds.co.yu
>
> sdjokic at mds.co.yu
>
>
>
>
>
>
>
> <image001.gif>
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net
>
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From Matt.Slaga at us.didata.com  Fri May 18 09:48:18 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 18 May 2007 09:48:18 -0400
Subject: [cisco-voip] IP Communicator 2.1 supports Vista
In-Reply-To: <f1b65f880705180636i5ded68d1t5e28a50e6dba1ae@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F03385EA@USNAEXCH.na.didata.local>

Cisco is finally releasing support for Vista in IPComm 2.1.  This is
great news!

 

Table 1 Minimum and Recommended Requirements for Cisco IP Communicator  

 	 
Item 

Description 

 	 
Internal Hardware 

Microsoft Windows Vista: 

* A Microsoft Vista Premium Ready PC. For details about the minimum
hardware requirements for Windows Vista (in addition to the requirements
in this table), search for Premium Ready PC on the Microsoft website or
see this URL: http://support.microsoft.com/kb/919183 

Note  Cisco IP Communicator supports x86-based processors running a
32-bit OS; 64-bit OSs are not supported. 

*Disk space:100 MB free disk space 

*Memory: 1 GB RAM (see the software row in this table for the supported
Vista OSs) 

*A non-ISA full-duplex sound card (integrated or PCI-based) or USB sound
device 

*A 10/100 Mbps Ethernet network interface card 

*SVGA video card 

*800 x 600 x16-bit screen resolution (1024 x 768 x 16-bit or better
recommended) 

 	 
	

 




-----------------------------------------
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designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
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From Brian.Henry at apptis.com  Fri May 18 10:20:42 2007
From: Brian.Henry at apptis.com (Brian Henry)
Date: Fri, 18 May 2007 10:20:42 -0400
Subject: [cisco-voip] importing groups in to Unity
References: <6E497ADB607656479C24E6D7BF6B505A0840CED3@exchange.rmwc.edu>
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6D0@CH-EXCH-02.intelltech.apptis.inc>

You can use Unity's Bulk Import Wizard, so if they are already in AD or if you have a .csv file made it is really easy.  Go to the Unity box and open Cisco Unity Tools, that is where you will find the tool.  It has a good help about how to use.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Kathy Dupree
Sent: Fri 5/18/2007 9:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] importing groups in to Unity



Can someone tell me where to find instructions on importing a group into Unity?  I don't how to import from exchange one user at a time but I will have groups of users to import for the coming school year.  Thanks

 

Katherine O. Dupree

Randolph-Macon Woman's College

Main 243

Telecommunications Support/Infrastructure Coordinator

kdupree at rmwc.edu <mailto:kdupree at rmwc.edu> 

(434) 947 8371

 

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From pat-cv at wcyv.com  Fri May 18 09:34:49 2007
From: pat-cv at wcyv.com (Pat Hayes)
Date: Fri, 18 May 2007 09:34:49 -0400
Subject: [cisco-voip] importing groups in to Unity
In-Reply-To: <6E497ADB607656479C24E6D7BF6B505A0840CED3@exchange.rmwc.edu>
References: <6E497ADB607656479C24E6D7BF6B505A0840CED3@exchange.rmwc.edu>
Message-ID: <464DAB79.6050607@wcyv.com>

The Bulk Import Tool. It lets you import users by CSV or AD Container.

Kathy Dupree wrote:
> Can someone tell me where to find instructions on importing a group into
> Unity?  I don't how to import from exchange one user at a time but I
> will have groups of users to import for the coming school year.  Thanks

From LSOCARRAS at broward.org  Fri May 18 10:56:33 2007
From: LSOCARRAS at broward.org (Socarras, Luis)
Date: Fri, 18 May 2007 10:56:33 -0400
Subject: [cisco-voip] IP Communicator 2.1 supports Vista
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F03385EA@USNAEXCH.na.didata.local>
References: <f1b65f880705180636i5ded68d1t5e28a50e6dba1ae@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F03385EA@USNAEXCH.na.didata.local>
Message-ID: <5BFD4B266FCBED44938AE3BB35ACD0AA01F5CCDF@UVEXVS01.bc.broward.cty>

And an external mic!!

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Friday, May 18, 2007 9:48 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IP Communicator 2.1 supports Vista

 

Cisco is finally releasing support for Vista in IPComm 2.1.  This is
great news!

 

Table 1 Minimum and Recommended Requirements for Cisco IP Communicator  

 

Item 

Description 

 

Internal Hardware 

Microsoft Windows Vista: 

* A Microsoft Vista Premium Ready PC. For details about the minimum
hardware requirements for Windows Vista (in addition to the requirements
in this table), search for Premium Ready PC on the Microsoft website or
see this URL: http://support.microsoft.com/kb/919183 

Note  Cisco IP Communicator supports x86-based processors running a
32-bit OS; 64-bit OSs are not supported. 

*Disk space:100 MB free disk space 

*Memory: 1 GB RAM (see the software row in this table for the supported
Vista OSs) 

*A non-ISA full-duplex sound card (integrated or PCI-based) or USB sound
device 

*A 10/100 Mbps Ethernet network interface card 

*SVGA video card 

*800 x 600 x16-bit screen resolution (1024 x 768 x 16-bit or better
recommended) 

 

		

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 

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From MillerS at DicksteinShapiro.COM  Fri May 18 12:30:07 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Fri, 18 May 2007 12:30:07 -0400
Subject: [cisco-voip] Release Notes for win-OS-Upgrade-K9.2000-4-4a.exe and
 win-OS-Upgrade-K9.2000-4-4a-sr7.exe ???
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538A57@DCEX2.DSMO.COM>

I must be having a brain cramp, because I'm searching the Cisco site,
but I can't find the release notes that go with this OS upgrade.  Aren't
they supposed to accompany the software?  I need to review the details,
because this patch is part of our upgrade tonight.  Please assist if you
can.  Thank you!
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
==============================================================================
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From Peterson_Erik at aac.com  Fri May 18 12:33:56 2007
From: Peterson_Erik at aac.com (Peterson_Erik at aac.com)
Date: Fri, 18 May 2007 12:33:56 -0400
Subject: [cisco-voip] IP Communicator 2.1 supports Vista
Message-ID: <OF08A0B1F2.8466A525-ON852572DF.005AF72D-852572DF.005B0018@aac.com>

But no support for VTA yet.

From jason.aarons at us.didata.com  Fri May 18 13:27:55 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 18 May 2007 13:27:55 -0400
Subject: [cisco-voip] H323 Enhanced Call Preservation and Media Inactivity
	Detection
Message-ID: <C1FE15183DA37645BC0633BC604E44F0059BDA89@USNAEXCH.na.didata.local>

Anyone seen any success or problems with these new 12.4T features?

 

This should keep H323 PSTN calls up when switching into SRST mode;

 

Voice service voip

   H323

    Call preserve

 

Which means if the H225/245 timers expire it won't disconnect a call
leg. So switching from SRST with H323 Gateway hopefully won't disconnect
a call.

 

Since we setup call preserv I had to add media inactivity detection, if
the DSP don't see RTP packets for 50000ms (5000 times 10) then it will
disconnect the call;

Router(config)# ip rtcp report interval 5000

Router(config)# gateway

Router(config-gateway)# timer media-inactive 10

Router(config-gateway)# exit

 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From rleetun at co.boulder.co.us  Fri May 18 13:50:07 2007
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Fri, 18 May 2007 11:50:07 -0600
Subject: [cisco-voip] CallManager 4.1.3 and MPE 2.0
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0059BDA89@USNAEXCH.na.didata.local>
Message-ID: <963CE94787005149A52C17F33A028E1207E61FCE@EVS1B.boco.co.boulder.co.us>

Hi,
 
Is anyone running CallManager 4.1.3 with MPE 2.0?  If so, how is MPE
performing?
 
Thanks in advance.
 
Rob
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From MillerS at DicksteinShapiro.COM  Fri May 18 13:50:08 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Fri, 18 May 2007 13:50:08 -0400
Subject: [cisco-voip] Locating NIC Cards in in the Call Manager
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538A5D@DCEX2.DSMO.COM>

We have teaming NICs in one of our offices and I can't find the devices
in Call Manager. Can you tell me how to view them?  Thank you!
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
==============================================================================
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From bob at smalltime.com  Fri May 18 14:24:26 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Fri, 18 May 2007 13:24:26 -0500
Subject: [cisco-voip] H323 Enhanced Call Preservation and Media
 Inactivity Detection
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0059BDA89@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0059BDA89@USNAEXCH.na.didata.local>
Message-ID: <464DEF5A.1080505@smalltime.com>

Jason Aarons (US) wrote:
> Anyone seen any success or problems with these new 12.4T features?

Been using it.  Wish we didn't have to.  Works well.

From tim.reimers at asheville.k12.nc.us  Fri May 18 14:19:45 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Fri, 18 May 2007 14:19:45 -0400
Subject: [cisco-voip] Blocking calls from unknown caller id? or routing to a
	messagebox?
Message-ID: <050944EE48FF384F9C663F7048C47DA702E1A938@webmail.asheville.k12.nc.us>

Hi -

 

I've been asked to see that our system does not accept calls that don't
deliver caller ID, or deliver all such calls to a mailbox that tells
people they must dial #82 on their phone and then call.

 

Is this possible?

 

System:

 

CCM 4.12

H.323 ISDN PRI gateway with 2 full PRIs.

 

 

Just ask if you need more details about our config from that..

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From jason.aarons at us.didata.com  Fri May 18 15:00:07 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 18 May 2007 15:00:07 -0400
Subject: [cisco-voip] H323 Enhanced Call Preservation and Media
	Inactivity Detection
In-Reply-To: <464DEF5A.1080505@smalltime.com>
References: <C1FE15183DA37645BC0633BC604E44F0059BDA89@USNAEXCH.na.didata.local>
	<464DEF5A.1080505@smalltime.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0059BDC9D@USNAEXCH.na.didata.local>

I shutdown (start/shutdown, maybe I should have just pulled the NIC?) my
subscriber with my phone showed Active Registered, my CP-7941G-GE
running 8.2(2)SR1S dropped the call, phone shows CM Down, features
disabled, then registered with publisher. Test was not the desired
results!

voice service voip 
 allow-connections h323 to h323
 h323
  call preserve

IOS is 12.4(11)T2

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Kulagowski
Sent: Friday, May 18, 2007 2:24 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H323 Enhanced Call Preservation and Media
Inactivity Detection

Jason Aarons (US) wrote:
> Anyone seen any success or problems with these new 12.4T features?

Been using it.  Wish we didn't have to.  Works well.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From bob at smalltime.com  Fri May 18 15:23:39 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Fri, 18 May 2007 14:23:39 -0500
Subject: [cisco-voip] H323 Enhanced Call Preservation and
 Media	Inactivity Detection
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0059BDC9D@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0059BDA89@USNAEXCH.na.didata.local>	<464DEF5A.1080505@smalltime.com>
	<C1FE15183DA37645BC0633BC604E44F0059BDC9D@USNAEXCH.na.didata.local>
Message-ID: <464DFD3B.1030901@smalltime.com>

Jason Aarons (US) wrote:
> I shutdown (start/shutdown, maybe I should have just pulled the NIC?) my
> subscriber with my phone showed Active Registered, my CP-7941G-GE
> running 8.2(2)SR1S dropped the call, phone shows CM Down, features
> disabled, then registered with publisher. Test was not the desired
> results!
> 
> voice service voip 
>  allow-connections h323 to h323
>  h323
>   call preserve
> 
> IOS is 12.4(11)T2

Well, because we're single threaded over the WAN, our situation is that 
we lose connectivity to _all_ CMs, not just one.

Did you try pulling your "WAN" connection instead of shutting down a CM?

We use:

voice service voip
  fax protocol pass-through g711ulaw
  h323
   call preserve limit-media-detection
  modem passthrough nse codec g711ulaw
!
!
voice class codec 1
  codec preference 1 g711ulaw
  codec preference 2 g729r8
!
!
!
voice class h323 1
  h225 timeout tcp establish 5
   call preserve

works on 2851 / 12.4.9(T)

From jrobbins at mercurypay.com  Fri May 18 15:32:26 2007
From: jrobbins at mercurypay.com (Jerri Robbins)
Date: Fri, 18 May 2007 13:32:26 -0600
Subject: [cisco-voip] install problems
Message-ID: <9015A93106739144995D6FD24A0F7A56028FDE5F@hydrogen.mercurypay.com>

When I try to run the .exe file to install the agent desktop, I have another installer pop up and start looking for the directory to pull from.  And it is not a directory that is accessible.  Any suggestions on what to try?

From Brian.Henry at apptis.com  Fri May 18 16:03:44 2007
From: Brian.Henry at apptis.com (Brian Henry)
Date: Fri, 18 May 2007 16:03:44 -0400
Subject: [cisco-voip] H323 Enhanced Call Preservation and
	Media	Inactivity Detection
References: <C1FE15183DA37645BC0633BC604E44F0059BDA89@USNAEXCH.na.didata.local>	<464DEF5A.1080505@smalltime.com>
	<C1FE15183DA37645BC0633BC604E44F0059BDC9D@USNAEXCH.na.didata.local>
	<464DFD3B.1030901@smalltime.com>
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6D1@CH-EXCH-02.intelltech.apptis.inc>

Remember to also change you service parameter in CCM to allow the call preservation as well.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Robert Kulagowski
Sent: Fri 5/18/2007 3:23 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H323 Enhanced Call Preservation and Media Inactivity Detection



Jason Aarons (US) wrote:
> I shutdown (start/shutdown, maybe I should have just pulled the NIC?) my
> subscriber with my phone showed Active Registered, my CP-7941G-GE
> running 8.2(2)SR1S dropped the call, phone shows CM Down, features
> disabled, then registered with publisher. Test was not the desired
> results!
>
> voice service voip
>  allow-connections h323 to h323
>  h323
>   call preserve
>
> IOS is 12.4(11)T2

Well, because we're single threaded over the WAN, our situation is that
we lose connectivity to _all_ CMs, not just one.

Did you try pulling your "WAN" connection instead of shutting down a CM?

We use:

voice service voip
  fax protocol pass-through g711ulaw
  h323
   call preserve limit-media-detection
  modem passthrough nse codec g711ulaw
!
!
voice class codec 1
  codec preference 1 g711ulaw
  codec preference 2 g729r8
!
!
!
voice class h323 1
  h225 timeout tcp establish 5
   call preserve

works on 2851 / 12.4.9(T)
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From asobihoudai at yahoo.com  Fri May 18 16:43:18 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 18 May 2007 13:43:18 -0700 (PDT)
Subject: [cisco-voip] Unity + DTMF relay issue
Message-ID: <872986.95746.qm@web57205.mail.re3.yahoo.com>

I have three offices all sharing one voice gateway.
When they try to check their voicemail remotely, as
soon as they hit pound they hear "sorry I do not
recognize that as a valid entry."

Has anyone encountered this issue before? I'm about to
open a TAC case but if I don't have to I'd be most
relieved.

Dial peers at the shared gateway have dtmf relay
commands on the dial-peers so I don't understand what
the problem could be. I'm also ASSUMING that this is a
DTMF relay issue and not a problem inherent to Unity
which is working fine for all other offices housed
upon it.


 
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From jason.aarons at us.didata.com  Fri May 18 18:23:12 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 18 May 2007 18:23:12 -0400
Subject: [cisco-voip] H323 Enhanced Call Preservation
	andMedia	Inactivity Detection
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6D1@CH-EXCH-02.intelltech.apptis.inc>
References: <C1FE15183DA37645BC0633BC604E44F0059BDA89@USNAEXCH.na.didata.local>	<464DEF5A.1080505@smalltime.com><C1FE15183DA37645BC0633BC604E44F0059BDC9D@USNAEXCH.na.didata.local><464DFD3B.1030901@smalltime.com>
	<2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6D1@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0059BDFA7@USNAEXCH.na.didata.local>

I'm not sure what service parameter you are referring to  -jason

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Henry
Sent: Friday, May 18, 2007 4:04 PM
To: Robert Kulagowski; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H323 Enhanced Call Preservation andMedia
Inactivity Detection

 

Remember to also change you service parameter in CCM to allow the call
preservation as well.

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Robert Kulagowski
Sent: Fri 5/18/2007 3:23 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H323 Enhanced Call Preservation and Media
Inactivity Detection

Jason Aarons (US) wrote:
> I shutdown (start/shutdown, maybe I should have just pulled the NIC?)
my
> subscriber with my phone showed Active Registered, my CP-7941G-GE
> running 8.2(2)SR1S dropped the call, phone shows CM Down, features
> disabled, then registered with publisher. Test was not the desired
> results!
>
> voice service voip
>  allow-connections h323 to h323
>  h323
>   call preserve
>
> IOS is 12.4(11)T2

Well, because we're single threaded over the WAN, our situation is that
we lose connectivity to _all_ CMs, not just one.

Did you try pulling your "WAN" connection instead of shutting down a CM?

We use:

voice service voip
  fax protocol pass-through g711ulaw
  h323
   call preserve limit-media-detection
  modem passthrough nse codec g711ulaw
!
!
voice class codec 1
  codec preference 1 g711ulaw
  codec preference 2 g729r8
!
!
!
voice class h323 1
  h225 timeout tcp establish 5
   call preserve

works on 2851 / 12.4.9(T)
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From lelio at uoguelph.ca  Fri May 18 21:47:49 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 18 May 2007 21:47:49 -0400
Subject: [cisco-voip] vg224 - MGCP or SCCP?
Message-ID: <003001c799b7$b52d7ab0$c901a8c0@pdp11>

Just wondering what others have used out there for their VG224.

I'm not so concerned with features or call survivability right now, but I'm more concerned with having faxes work while in SRST mode. I'm guessing modem/fax passthrough is what I really want.

On a side note, a number of people asked if you need to buy a licence for a VG224. People have been saying no. But according to this documentation, if you use the VG224 is SCCP mode, then you are required to buy a licence.

------------------------------------------------------------------------------
  http://www.cisco.com/en/US/customer/products/ps6441/products_feature_guide09186a0080483a76.html#wp1043332

  Each FXS port that has SCCP enabled counts as a single IP phone for licensing purposes. For example, if you register all 24 ports on a Cisco VG 224: 
  -For Cisco Unified CME, you need to purchase at least an FL-CCME-SMALL license. 

  -For Cisco Unified CallManager, the 24 ports count towards the 2500 limit if you purchase LIC-CCM-4.X-2500. 


------------------------------------------------------------------------------

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From erickbe at yahoo.com  Fri May 18 23:42:54 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri, 18 May 2007 20:42:54 -0700 (PDT)
Subject: [cisco-voip] vg224 - MGCP or SCCP?
Message-ID: <412462.61068.qm@web37202.mail.mud.yahoo.com>

Well,

I am currently dealing with a weird FAX issue with a VG224 in SCCP (stcapp) mode.  SCCP (stcapp) only supports passthrough, not relay.  It looks like I'm going to need to tinker with making one of the ports H323 or MGCP to see if that works better and fixes this FAX problem I'm dealing with. I am working with TAC on it and they are suggesting moving away from SCCP so we can try T.38 FAX Relay which is in the newer IOS versions to see if it fixes this weird FAX problem. I'm neutral on the suggestion until I see what happens and am waiting for some details back from TAC still before I go tinkering with a live system and not mess up stuff that is working fine (other FAX's, etc).

The FAX problem is, a FAX machine on the VG224 can FAX everyone fine except for one number. It can receive FAXs from that number though. A FAX machine attached to an ATA however can dial the number fine and have a good FAX, so theres something a little different with the passthrough on the VG224 in SCCP mode versus the ATA. The PSTN Gateway is a Cisco 2821 with 2 PRIs , MGCP controlled with fax relay disabled at the moment.

FAX's and VoIP are a pain still it seems. It either works or it is an adventure ....



----- Original Message ----
From: Lelio Fulgenzi <lelio at uoguelph.ca>
To: cisco-voip at puck.nether.net
Sent: Friday, May 18, 2007 8:47:49 PM
Subject: [cisco-voip] vg224 - MGCP or SCCP?



 
 


Just wondering what others have used out there for 
their VG224.

 

I'm not so concerned with features or call 
survivability right now, but I'm more concerned with having faxes work while in 
SRST mode. I'm guessing modem/fax passthrough is what I really 
want.

 

On a side note, a number of people asked if you 
need to buy a licence for a VG224. People have been saying no. But according to 
this documentation, if you use the VG224 is SCCP mode, then you are required to 
buy a licence.


  
  
  http://www.cisco.com/en/US/customer/products/ps6441/products_feature_guide09186a0080483a76.html#wp1043332

   

  Each FXS port that has SCCP enabled counts as a 
  single IP phone for licensing purposes. For example, if you register all 24 
  ports on a Cisco VG 224: 
  ?For Cisco Unified CME, you need to purchase at least an FL-CCME-SMALL 
  license. 

  ?For Cisco Unified CallManager, the 24 ports count towards the 2500 
  limit if you purchase LIC-CCM-4.X-2500. 

  
  

  

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https://puck.nether.net/mailman/listinfo/cisco-voip







 
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From MillerS at DicksteinShapiro.COM  Sat May 19 03:01:59 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Sat, 19 May 2007 03:01:59 -0400
Subject: [cisco-voip] "CM down, features disabled" Message
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538A9D@DCEX2.DSMO.COM>

I have see this happen on a number of different occasions when there was
basically no system disruption.  Has anyone seen this message on their
phone in situations where CM was clearly NOT down?
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

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From asobihoudai at yahoo.com  Sat May 19 04:33:05 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Sat, 19 May 2007 01:33:05 -0700 (PDT)
Subject: [cisco-voip] "CM down, features disabled" Message
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538A9D@DCEX2.DSMO.COM>
Message-ID: <366179.42704.qm@web57210.mail.re3.yahoo.com>

--- "Miller, Steve" <MillerS at DicksteinShapiro.COM>
wrote:

> I have see this happen on a number of different
> occasions when there was
> basically no system disruption.  Has anyone seen
> this message on their
> phone in situations where CM was clearly NOT down?
>  

Yes I have seen this only a few weeks ago with a
customer's phone. Unfortunately I can't remember what
the hell I did to resolve it. :(

Paul


       
____________________________________________________________________________________
Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games.
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From aelnagar at ACT-EG.COM  Sat May 19 08:49:52 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Sat, 19 May 2007 15:49:52 +0300
Subject: [cisco-voip] "CM down, features disabled" Message
References: <366179.42704.qm@web57210.mail.re3.yahoo.com>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A071@mail.ACT-EG.COM>

sometimes I see this message with IPcom when connecting using VPN and the connection is not good, but after the call the phone doesnot restart
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Paul Choi
Sent: Sat 19-May-07 11:33 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] "CM down, features disabled" Message



--- "Miller, Steve" <MillerS at DicksteinShapiro.COM>
wrote:

> I have see this happen on a number of different
> occasions when there was
> basically no system disruption.  Has anyone seen
> this message on their
> phone in situations where CM was clearly NOT down?
> 

Yes I have seen this only a few weeks ago with a
customer's phone. Unfortunately I can't remember what
the hell I did to resolve it. :(

Paul


      
____________________________________________________________________________________
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From aelnagar at ACT-EG.COM  Sat May 19 09:21:16 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Sat, 19 May 2007 16:21:16 +0300
Subject: [cisco-voip] FAX not working to one number
References: <47236.54911.qm@web37201.mail.mud.yahoo.com>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A074@mail.ACT-EG.COM>

try to make sure that ECM feature is disabled. this may cause fax failure
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Erick Bergquist
Sent: Wed 16-May-07 9:10 PM
To: ciscovoip
Subject: [cisco-voip] FAX not working to one number



I have a strange FAX issue, where a FAX machine connected to a VG224 can send/receive FAX to any number but is not able to send one outside number a FAX. It can receive FAX's from them fine, just not send. The FAX machines phone or a phone plugged into the FAX phone line can call the # fine so there is not a CSS issue, etc.

The FAX machine is plugged into a VG224 with IOS 12.4(11)T , the VG224 is configured as SCCP (stcapp) and we have tried setting the port capabilty to modem passthrough and not with no change.

The PSTN Gateway is a Cisco 2821 with a PRI, it is MGCP and running IOS 12.4(9)T and has FAX relay disabled.

We have tried 7 different FAX machines of different models and neither are able to call this particular outside FAX machine when going over IPT. When the FAX machines are plugged into a traditional analog circuit there is not a problem and the FAX works fine.

Any ideas?




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From lelio at uoguelph.ca  Sat May 19 08:48:52 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat, 19 May 2007 08:48:52 -0400
Subject: [cisco-voip] vg224 - MGCP or SCCP?
References: <412462.61068.qm@web37202.mail.mud.yahoo.com>
Message-ID: <003201c79a14$117c9830$c901a8c0@pdp11>

Thanks Eric. From the CallManager config page for the VG224, I see MGCP or SCCP. How can you move one port to MGCP/H323? From what I see, it's an all or nothing. I'm likely wrong on this, but I don't see how you would get that to work.

  ----- Original Message ----- 
  From: Erick Bergquist 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Friday, May 18, 2007 11:42 PM
  Subject: Re: [cisco-voip] vg224 - MGCP or SCCP?


  Well,

  I am currently dealing with a weird FAX issue with a VG224 in SCCP (stcapp) mode.  SCCP (stcapp) only supports passthrough, not relay.  It looks like I'm going to need to tinker with making one of the ports H323 or MGCP to see if that works better and fixes this FAX problem I'm dealing with. I am working with TAC on it and they are suggesting moving away from SCCP so we can try T.38 FAX Relay which is in the newer IOS versions to see if it fixes this weird FAX problem. I'm neutral on the suggestion until I see what happens and am waiting for some details back from TAC still before I go tinkering with a live system and not mess up stuff that is working fine (other FAX's, etc).

  The FAX problem is, a FAX machine on the VG224 can FAX everyone fine except for one number. It can receive FAXs from that number though. A FAX machine attached to an ATA however can dial the number fine and have a good FAX, so theres something a little different with the passthrough on the VG224 in SCCP mode versus the ATA. The PSTN Gateway is a Cisco 2821 with 2 PRIs , MGCP controlled with fax relay disabled at the moment.

  FAX's and VoIP are a pain still it seems. It either works or it is an adventure ....




  ----- Original Message ----
  From: Lelio Fulgenzi <lelio at uoguelph.ca>
  To: cisco-voip at puck.nether.net
  Sent: Friday, May 18, 2007 8:47:49 PM
  Subject: [cisco-voip] vg224 - MGCP or SCCP?


  Just wondering what others have used out there for their VG224.

  I'm not so concerned with features or call survivability right now, but I'm more concerned with having faxes work while in SRST mode. I'm guessing modem/fax passthrough is what I really want.

  On a side note, a number of people asked if you need to buy a licence for a VG224. People have been saying no. But according to this documentation, if you use the VG224 is SCCP mode, then you are required to buy a licence.
    http://www.cisco.com/en/US/customer/products/ps6441/products_feature_guide09186a0080483a76.html#wp1043332

    Each FXS port that has SCCP enabled counts as a single IP phone for licensing purposes. For example, if you register all 24 ports on a Cisco VG 224: 
    ?For Cisco Unified CME, you need to purchase at least an FL-CCME-SMALL license. 

    ?For Cisco Unified CallManager, the 24 ports count towards the 2500 limit if you purchase LIC-CCM-4.X-2500. 



  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip





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From erickbe at yahoo.com  Sat May 19 09:15:49 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sat, 19 May 2007 06:15:49 -0700 (PDT)
Subject: [cisco-voip] InsertCDR.exe spiking CPU in 4.1(3)sr4d
Message-ID: <38709.969.qm@web37212.mail.mud.yahoo.com>

Anyone seen this before? Bug toolkit shows one possible match that saids resolved in 4.3. 

All the CDR folders were empty and CDR enabled set to true on all servers in cluster.



 
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with the Yahoo! Search movie showtime shortcut.
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From lelio at uoguelph.ca  Sat May 19 10:39:10 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat, 19 May 2007 10:39:10 -0400
Subject: [cisco-voip] FAX not working to one number
References: <47236.54911.qm@web37201.mail.mud.yahoo.com>
	<88C8D019DA547C4BABECB9A806ED22ED88A074@mail.ACT-EG.COM>
Message-ID: <001e01c79a23$84d0fc90$c901a8c0@pdp11>

[cisco-voip] FAX not working to one numberwhere is this disabled?

  ----- Original Message ----- 
  From: Ahmed Elnagar 
  To: Erick Bergquist ; ciscovoip 
  Sent: Saturday, May 19, 2007 9:21 AM
  Subject: Re: [cisco-voip] FAX not working to one number


  try to make sure that ECM feature is disabled. this may cause fax failure

  Thanks and Best Regards

  Ahmed A. Elnagar
  Network Field Engineer
   
  Advanced Computer Technology (ACT)
  16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
  Postal Code:12411 Cairo Egypt

  Mob: +2010-2833868
  Website: www.act-eg.com
  E-mail: aelnagar at act-eg.com


------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net on behalf of Erick Bergquist
  Sent: Wed 16-May-07 9:10 PM
  To: ciscovoip
  Subject: [cisco-voip] FAX not working to one number


  I have a strange FAX issue, where a FAX machine connected to a VG224 can send/receive FAX to any number but is not able to send one outside number a FAX. It can receive FAX's from them fine, just not send. The FAX machines phone or a phone plugged into the FAX phone line can call the # fine so there is not a CSS issue, etc.

  The FAX machine is plugged into a VG224 with IOS 12.4(11)T , the VG224 is configured as SCCP (stcapp) and we have tried setting the port capabilty to modem passthrough and not with no change.

  The PSTN Gateway is a Cisco 2821 with a PRI, it is MGCP and running IOS 12.4(9)T and has FAX relay disabled.

  We have tried 7 different FAX machines of different models and neither are able to call this particular outside FAX machine when going over IPT. When the FAX machines are plugged into a traditional analog circuit there is not a problem and the FAX works fine.

  Any ideas?




  ____________________________________________________________________________________
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  _______________________________________________
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  https://puck.nether.net/mailman/listinfo/cisco-voip




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From aelnagar at ACT-EG.COM  Sat May 19 10:53:39 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Sat, 19 May 2007 17:53:39 +0300
Subject: [cisco-voip] FAX not working to one number
References: <47236.54911.qm@web37201.mail.mud.yahoo.com>
	<88C8D019DA547C4BABECB9A806ED22ED88A074@mail.ACT-EG.COM>
	<001e01c79a23$84d0fc90$c901a8c0@pdp11>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A076@mail.ACT-EG.COM>

I was using fax realy with a sip connection to a fax server, the configuration that disables the ecm was on the gateway under the voip dial-peer that is pointing to the fax server with this command fax-relay ecm disable I dont know will that be applicable in ur case or not, try to check the the only number that is failing to send faxs to and see what is the type of the fax machine there if it supports ecm then most commonly that is the problem, try search on cisco for ecm for better understanding of this feature, but I was having a case like u exactly I was able to send to all fax numbers excpet one and I discovered that the this one that fails is  a multifuction machine that supports ECM and this doesnot work well with our case.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Sat 19-May-07 5:39 PM
To: Ahmed Elnagar; Erick Bergquist; ciscovoip
Subject: Re: [cisco-voip] FAX not working to one number


where is this disabled?
 

	----- Original Message ----- 
	From: Ahmed Elnagar <mailto:aelnagar at ACT-EG.COM>  
	To: Erick Bergquist <mailto:erickbe at yahoo.com>  ; ciscovoip <mailto:cisco-voip at puck.nether.net>  
	Sent: Saturday, May 19, 2007 9:21 AM
	Subject: Re: [cisco-voip] FAX not working to one number

	try to make sure that ECM feature is disabled. this may cause fax failure
	 
	
	Thanks and Best Regards
	
	Ahmed A. Elnagar
	Network Field Engineer
	 
	Advanced Computer Technology (ACT)
	16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
	Postal Code:12411 Cairo Egypt
	
	Mob: +2010-2833868
	Website: www.act-eg.com <http://www.act-eg.com/> 
	E-mail: aelnagar at act-eg.com

________________________________

	From: cisco-voip-bounces at puck.nether.net on behalf of Erick Bergquist
	Sent: Wed 16-May-07 9:10 PM
	To: ciscovoip
	Subject: [cisco-voip] FAX not working to one number
	
	

	I have a strange FAX issue, where a FAX machine connected to a VG224 can send/receive FAX to any number but is not able to send one outside number a FAX. It can receive FAX's from them fine, just not send. The FAX machines phone or a phone plugged into the FAX phone line can call the # fine so there is not a CSS issue, etc.
	
	The FAX machine is plugged into a VG224 with IOS 12.4(11)T , the VG224 is configured as SCCP (stcapp) and we have tried setting the port capabilty to modem passthrough and not with no change.
	
	The PSTN Gateway is a Cisco 2821 with a PRI, it is MGCP and running IOS 12.4(9)T and has FAX relay disabled.
	
	We have tried 7 different FAX machines of different models and neither are able to call this particular outside FAX machine when going over IPT. When the FAX machines are plugged into a traditional analog circuit there is not a problem and the FAX works fine.
	
	Any ideas?
	
	
	
	
	____________________________________________________________________________________
	Sucker-punch spam with award-winning protection.
	Try the free Yahoo! Mail Beta.
	http://advision.webevents.yahoo.com/mailbeta/features_spam.html
	_______________________________________________
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	https://puck.nether.net/mailman/listinfo/cisco-voip
	

	
________________________________


	

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From erickbe at yahoo.com  Sat May 19 11:30:33 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sat, 19 May 2007 08:30:33 -0700 (PDT)
Subject: [cisco-voip] InsertCDR.exe spiking CPU in 4.1(3)sr4d
Message-ID: <20070519153033.887.qmail@web37203.mail.mud.yahoo.com>

 The bug id I found is CSCsh27547 which seems to match part of the
behavior, because lsass and dllhost also had high cpu usage at the time. But there were no files in the folders to process. 

Anyone know if this bug is going to fixed in a 4.1(3) ES or 4.2? 


----- Original Message ----
From: Erick Bergquist <erickbe at yahoo.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Saturday, May 19, 2007 8:15:49 AM
Subject: [cisco-voip] InsertCDR.exe spiking CPU in 4.1(3)sr4d

Anyone seen this before? Bug toolkit shows one possible match that saids resolved in 4.3. 

All the CDR folders were empty and CDR enabled set to true on all servers in cluster.



 
____________________________________________________________________________________
8:00? 8:25? 8:40? Find a flick in no time 
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip





       
____________________________________________________________________________________Got a little couch potato? 
Check out fun summer activities for kids.
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From MillerS at DicksteinShapiro.COM  Sat May 19 13:15:31 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Sat, 19 May 2007 13:15:31 -0400
Subject: [cisco-voip] Re-enabled NIC on the Call Manager Has Prevented
 Access to the System - Please Help!
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538AAF@DCEX2.DSMO.COM>

We have NICs in a Call Manager Subscriber in NY and one had been
disabled for a long time.  I was told by an expert engineer (who
actually was the architect or our system) that all of the settings for
the cards were correct and that there was probably a cable unplugged.
When I got to the site, there was a cable unplugged so I replaced the
cable and then re-enabled the card that corresponded to that port. At
that point, we rec'd errors in the event log that the system could not
communicate with the NTP server and now we cannot connect to the Call
Manager via RDP or by web interface programming.  All calls are being
processed and there seems to be no current user issue, but I need to
know if I should simply disable the connection again and troubleshoot or
if there is something in the teaming NIC card setup that needs to be
changed.  I have never messed with this before and I want to fix it, but
I don't want to interfere with the phone service.  I'm sure that this is
a simple issue, but I don't want a misstep.  Please let me know if you
are familiar with this.  Thank you!
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

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From MillerS at DicksteinShapiro.COM  Sat May 19 13:46:15 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Sat, 19 May 2007 13:46:15 -0400
Subject: [cisco-voip] Re-enabled NIC on the Call Manager Has Prevented
 Access to the System - Please Help!
References: <B9B5B75DF9145F44A21111DD92CC490006538AAF@DCEX2.DSMO.COM>
	<310F10740485D9469F0D8417950B252D0281F897@CTG-MSNEXC01.staff.berbee.com>
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538AB3@DCEX2.DSMO.COM>

When I look at the team properties, it says "Network Fault Tolerance
Only with Preference Order".  What else do I need to check?  Would
simply disabling the 2nd NIC again temporarily fix the problem?  There
has been no adjustment of the cards (programming) at all at this point.
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

________________________________

From: Clouse, Chris [mailto:chris.clouse at berbee.com] 
Sent: Saturday, May 19, 2007 1:40 PM
To: Miller, Steve
Subject: RE: [cisco-voip] Re-enabled NIC on the Call Manager Has
Prevented Access to the System - Please Help!



The NIC's should not be teamed.  The only method that is supported is
Fault Tolerance Only.  Enable this mode, connect both cables, and try
again.

 

Christopher Clouse, CCNP CCDP CCVP MCP Network+

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Saturday, May 19, 2007 12:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Re-enabled NIC on the Call Manager Has Prevented
Access to the System - Please Help!

 

We have NICs in a Call Manager Subscriber in NY and one had been
disabled for a long time.  I was told by an expert engineer (who
actually was the architect or our system) that all of the settings for
the cards were correct and that there was probably a cable unplugged.
When I got to the site, there was a cable unplugged so I replaced the
cable and then re-enabled the card that corresponded to that port. At
that point, we rec'd errors in the event log that the system could not
communicate with the NTP server and now we cannot connect to the Call
Manager via RDP or by web interface programming.  All calls are being
processed and there seems to be no current user issue, but I need to
know if I should simply disable the connection again and troubleshoot or
if there is something in the teaming NIC card setup that needs to be
changed.  I have never messed with this before and I want to fix it, but
I don't want to interfere with the phone service.  I'm sure that this is
a simple issue, but I don't want a misstep.  Please let me know if you
are familiar with this.  Thank you!

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
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have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
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To reply to our email administrator directly, send an email to
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http://www.DicksteinShapiro.com
========================================================================
======

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

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From jason.aarons at us.didata.com  Sat May 19 15:36:54 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Sat, 19 May 2007 15:36:54 -0400
Subject: [cisco-voip] Re-enabled NIC on the Call Manager Has Prevented
	Access to the System - Please Help!
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538AB3@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490006538AAF@DCEX2.DSMO.COM><310F10740485D9469F0D8417950B252D0281F897@CTG-MSNEXC01.staff.berbee.com>
	<B9B5B75DF9145F44A21111DD92CC490006538AB3@DCEX2.DSMO.COM>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0059BE0C7@USNAEXCH.na.didata.local>

Using the Cisco Media Convergence Server Network Teaming Driver with
Operating System Versions 2000.4.3 and 2000.4.3a

http://www.cisco.com/en/US/products/hw/voiceapp/ps378/prod_installation_
guide09186a00806a476f.html

 

The above document "strongly suggests NFT" in Step6, the default is
Auto. 

 

I don't see anything that states Auto is unsupported by Cisco.  I would
think being able to use two NICS @  1000/Full would be better than one @
1000/Full -jason

 

In Auto mode connected to a Catalyst 3500/4500/6500 with default
settings Auto will select Transmit Load Balancing with Fault Tolerance
(TLB)  based upon the HP Whitepaper below. If you go into Control Panel
> Networks > Teaming Interface > Properties> HP Network Configuration
Utility > Select HP Network Team > Properties > Information it will
shows the actual "Current Mode" that Auto selected. I went into a
OS2000-4-4a-SR6 and found they were Transmit Load Balancing with Fault
Tolerance (TLB) and have been in production for several years with no
issues relating to the NIC teaming. IPCC Express application isn't
supported with NIC teaming and does have problems.

 

ftp://ftp.compaq.com/pub/products/servers/networking/TeamingWP.pdf
<javascript:openExternal('ftp://ftp.compaq.com/pub/products/servers/netw
orking/TeamingWP.pdf')> 

 

 

________________________________

From: Clouse, Chris [mailto:chris.clouse at berbee.com] 
Sent: Saturday, May 19, 2007 1:40 PM
To: Miller, Steve
Subject: RE: [cisco-voip] Re-enabled NIC on the Call Manager Has
Prevented Access to the System - Please Help!

The NIC's should not be teamed.  The only method that is supported is
Fault Tolerance Only.  Enable this mode, connect both cables, and try
again.

 

Christopher Clouse, CCNP CCDP CCVP MCP Network+

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Saturday, May 19, 2007 12:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Re-enabled NIC on the Call Manager Has Prevented
Access to the System - Please Help!

 

We have NICs in a Call Manager Subscriber in NY and one had been
disabled for a long time.  I was told by an expert engineer (who
actually was the architect or our system) that all of the settings for
the cards were correct and that there was probably a cable unplugged.
When I got to the site, there was a cable unplugged so I replaced the
cable and then re-enabled the card that corresponded to that port. At
that point, we rec'd errors in the event log that the system could not
communicate with the NTP server and now we cannot connect to the Call
Manager via RDP or by web interface programming.  All calls are being
processed and there seems to be no current user issue, but I need to
know if I should simply disable the connection again and troubleshoot or
if there is something in the teaming NIC card setup that needs to be
changed.  I have never messed with this before and I want to fix it, but
I don't want to interfere with the phone service.  I'm sure that this is
a simple issue, but I don't want a misstep.  Please let me know if you
are familiar with this.  Thank you!

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com <mailto:millers at dicksteinshapiro.com>  

 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
prohibited. Dickstein Shapiro reserves the right to monitor any
communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message. 
 
To reply to our email administrator directly, send an email to
postmaster at dicksteinshapiro.com
 
Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
========================================================================
======
--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
prohibited. Dickstein Shapiro reserves the right to monitor any
communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message. 
 
To reply to our email administrator directly, send an email to
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Dickstein Shapiro LLP
http://www.DicksteinShapiro.com
========================================================================
======



-----------------------------------------
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From jason.aarons at us.didata.com  Sat May 19 15:38:15 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Sat, 19 May 2007 15:38:15 -0400
Subject: [cisco-voip] InsertCDR.exe spiking CPU in 4.1(3)sr4d
In-Reply-To: <20070519153033.887.qmail@web37203.mail.mud.yahoo.com>
References: <20070519153033.887.qmail@web37203.mail.mud.yahoo.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0059BE0C8@USNAEXCH.na.didata.local>

Verify the CDR Insert service is only running on the publisher -jason

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
Sent: Saturday, May 19, 2007 11:31 AM
To: Erick Bergquist; ciscovoip
Subject: Re: [cisco-voip] InsertCDR.exe spiking CPU in 4.1(3)sr4d

 The bug id I found is CSCsh27547 which seems to match part of the
behavior, because lsass and dllhost also had high cpu usage at the time.
But there were no files in the folders to process. 

Anyone know if this bug is going to fixed in a 4.1(3) ES or 4.2? 


----- Original Message ----
From: Erick Bergquist <erickbe at yahoo.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Saturday, May 19, 2007 8:15:49 AM
Subject: [cisco-voip] InsertCDR.exe spiking CPU in 4.1(3)sr4d

Anyone seen this before? Bug toolkit shows one possible match that saids
resolved in 4.3. 

All the CDR folders were empty and CDR enabled set to true on all
servers in cluster.



 
________________________________________________________________________
____________
8:00? 8:25? 8:40? Find a flick in no time 
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip





       
________________________________________________________________________
____________Got a little couch potato? 
Check out fun summer activities for kids.
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ds&cs=bz 
_______________________________________________
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-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From jason.aarons at us.didata.com  Sat May 19 19:08:54 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Sat, 19 May 2007 19:08:54 -0400
Subject: [cisco-voip] 7941 DTMF fails after switch into SRST 4.0
In-Reply-To: <464DFD3B.1030901@smalltime.com>
References: <C1FE15183DA37645BC0633BC604E44F0059BDA89@USNAEXCH.na.didata.local>	<464DEF5A.1080505@smalltime.com><C1FE15183DA37645BC0633BC604E44F0059BDC9D@USNAEXCH.na.didata.local>
	<464DFD3B.1030901@smalltime.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F0059BE0F8@USNAEXCH.na.didata.local>

I think issuing the "Start, shutdown" causes a graceful shutdown via
SCCP messages to the phones which is why I didn't get call preservation
in the testing.

I retested pulling both CallManager's Ethernet cable and the call stayed
up. DTMF works during the call, once it goes into SRST I lose DTMF
capability. The primary application is CVP (Cisco Voice Portal) I'm not
sure why all three dtmf-relay formats are there but I don't want to
remove them in case they are needed for CVP. There is no Unity.

If I start the call while under SRST I can enter DTMF tones (I'm calling
a move theatre on the PSTN that allows you to press digits to select
show times for my testing) but once CCM is down the DTMF fails.

dial-peer voice 28800 voip
 description To PrimarySubscriber
 preference 1
 destination-pattern 8800
 session target ipv4:10.20.10.4
 dtmf-relay rtp-nte h245-signal h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 2 pots
 description Outbound calls to TM E1 PSTN
 destination-pattern 9T
 incoming called-number .
 direct-inward-dial
 port 0/0/0:15

SRST 4.0(2) in 12.4(11)T2 with CP-7941G-GE with 8.2(2)SR1S

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Kulagowski
Sent: Friday, May 18, 2007 3:24 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H323 Enhanced Call Preservation and Media
Inactivity Detection

Jason Aarons (US) wrote:
> I shutdown (start/shutdown, maybe I should have just pulled the NIC?)
my
> subscriber with my phone showed Active Registered, my CP-7941G-GE
> running 8.2(2)SR1S dropped the call, phone shows CM Down, features
> disabled, then registered with publisher. Test was not the desired
> results!
> 
> voice service voip 
>  allow-connections h323 to h323
>  h323
>   call preserve
> 
> IOS is 12.4(11)T2

Well, because we're single threaded over the WAN, our situation is that 
we lose connectivity to _all_ CMs, not just one.

Did you try pulling your "WAN" connection instead of shutting down a CM?

We use:

voice service voip
  fax protocol pass-through g711ulaw
  h323
   call preserve limit-media-detection
  modem passthrough nse codec g711ulaw
!
!
voice class codec 1
  codec preference 1 g711ulaw
  codec preference 2 g729r8
!
!
!
voice class h323 1
  h225 timeout tcp establish 5
   call preserve

works on 2851 / 12.4.9(T)
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

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Disclaimer:

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this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From fwn at feasible.net  Sun May 20 10:40:21 2007
From: fwn at feasible.net (Fred Nielsen)
Date: Sun, 20 May 2007 07:40:21 -0700
Subject: [cisco-voip] 5.1 Phone Upgrade Issue
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451EEB@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451EEB@CSEXCH01.charles-stanley.co.uk>
Message-ID: <ca09e3750705200740p67b4c1f0n29b8989c7deea254@mail.gmail.com>

Have you thought about codecs?  Is the new 7970 firmware attempting to use
G.722 but the gateway isn't supporting it perhaps?  Since your calls work
out your remote gateway, I'm assuming that you are using G.729 across the
WAN... try looking into the within region default codec.

On 5/16/07, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote:
>
>  Can anyone help with this, the phones are 7970 and the previous firmware
> that worked fine is SCCP70.8-0-3S
>
>
>
> Why would I be able to dial out with one firmware and not another this
> doesn't make sense.
>
>
>
>
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Nick Kassel
> *Sent:* 16 May 2007 14:44
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] 5.1 Phone Upgrade Issue
> *Importance:* High
>
>
>
> I have a very strange issue with one of our remote sites, I have upgraded
> some phones to SCCP70.8-2-1S and now they are unable to dial out through
> there local H323 gateway, the number will ring twice and then the call cuts
> out and it says Busy on the phone, any phones that have not been upgraded
> can still dial out fine.
>
>
>
> Now if I change the order of the Gateways in the Route group so that they
> dial out of the back up gateway at our main site all phones can dial out
> fine, I have also upgraded a number of other remote sites with the same
> firmware and they are also fine.
>
>
>
> Can anyone advise what might cause this issue, as I can't understand why
> this would happen especially just at one site.
>
>
>
>
>
>
>
>
>
> ********************SAVE PAPER - THINK BEFORE YOU PRINT**************************************************
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From SCASPER at mtb.com  Sun May 20 13:45:35 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Sun, 20 May 2007 13:45:35 -0400
Subject: [cisco-voip] which ios to use for 3845
Message-ID: <s6505126.033@comrcm01intgwia.mandtbank.com>

I am running 12.4.12a  on a 3845 that currently support about 100 7911
telephones. Tested SRST Thursday night - it took about 6 minutes for the
phones to register, no problems.

>>> "Lelio Fulgenzi" <lelio at uoguelph.ca> 05/10/07 4:54 PM >>>
Well, considering we've got 7911s we need SRST for, looks like it's
12.4(9)T or better for me!

Talk about bleeding edge...at least it looks like 12.4(11)T2 is LD and
not ED.

Thanks for all your suggestions.



--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^

...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Carter, Bill 
  To: Lelio Fulgenzi 
  Sent: Tuesday, May 08, 2007 12:45 PM
  Subject: RE: [cisco-voip] which ios to use for 3845


  12.4(6)T7 or 12.4(7)b

 
http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/system/gbtst4x/4_4/rng44nip.htm#wp1261437


  Cisco "Golden Bridge" documents
 
http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/system/gbtst4x/4_4/index.htm






------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio
Fulgenzi
  Sent: Tuesday, May 08, 2007 8:57 AM
  To: cisco-voip at puck.nether.net 
  Subject: [cisco-voip] which ios to use for 3845


  my 3845 was just dropped off. 

  what IOS are people recommending for this thing? primarily used as a
voice gateway (2 T1/PRIs using MGCP) and for SRST. I don't think I will
be doing much transcoding, (other than from PSTN to voip) or
conferencing.


 
--------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G
2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  ...there's no such thing as a bad timbit...

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From jonvoip at gmail.com  Sun May 20 18:10:27 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 20 May 2007 17:10:27 -0500
Subject: [cisco-voip] install problems
In-Reply-To: <9015A93106739144995D6FD24A0F7A56028FDE5F@hydrogen.mercurypay.com>
References: <9015A93106739144995D6FD24A0F7A56028FDE5F@hydrogen.mercurypay.com>
Message-ID: <5d093f9a0705201510y10a31de5iee4c5a2f2ce7d961@mail.gmail.com>

Login to the share from that desktop... then run setup.



Jonathan

On 5/18/07, Jerri Robbins <jrobbins at mercurypay.com> wrote:
> When I try to run the .exe file to install the agent desktop, I have another installer pop up and start looking for the directory to pull from.  And it is not a directory that is accessible.  Any suggestions on what to try?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From voip at mrga.ch  Mon May 21 02:35:41 2007
From: voip at mrga.ch (Reto Gassmann)
Date: Mon, 21 May 2007 08:35:41 +0200
Subject: [cisco-voip] Upgrade Order CCM 3.3(4) -> 4.2(3)
Message-ID: <1179729341.46513dbda056a@www.mail2web.ch>

Hello

I am planing the upgrade from our Callmanager 3.3(4) to 4.2(3). Is there a
specific order I have to follow when I install the upgrades? 

The planed path for the publisher upgrade looks like this:
Update Callmanager to 4.2(3)
Update JTAPI 
Update Extended Services to 3.5
Configure Extended Services
Update Extended Services to 3.5 SR2
Update BAT to 5.2(3a)
Update TAPS
Install Locals 4.2.1.3000
Install CCM 4.2(3) SR2

The planed path for the subscribers upgrade looks like this:
Update Callmanager to 4.2(3)
Install Locals 4.2.1.3000
Install CCM 4.2(3) SR2

Thanks for inputs
Reto


From Nick.Kassel at Charles-Stanley.co.uk  Mon May 21 04:55:03 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 21 May 2007 09:55:03 +0100
Subject: [cisco-voip] 5.1 Phone Upgrade Issue
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451F00@CSEXCH01.charles-stanley.co.uk>

We have plenty of bandwidth so use G711 locally and across the WAN,
however when I downgraded the phone to 8-0-4SR3AS it worked fine out of
the local gateway. This branch is using a 2651XM gateway with 12.4 11T,
however all our other sites use 1760 gateways with 12.4 11T and they are
fine.

 

 

________________________________

From: fredless at gmail.com [mailto:fredless at gmail.com] On Behalf Of Fred
Nielsen
Sent: 20 May 2007 15:40
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 5.1 Phone Upgrade Issue

 

Have you thought about codecs?  Is the new 7970 firmware attempting to
use G.722 but the gateway isn't supporting it perhaps?  Since your calls
work out your remote gateway, I'm assuming that you are using G.729
across the WAN... try looking into the within region default codec. 

On 5/16/07, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote: 

Can anyone help with this, the phones are 7970 and the previous firmware
that worked fine is SCCP70.8-0-3S 

 

Why would I be able to dial out with one firmware and not another this
doesn't make sense.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: 16 May 2007 14:44
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 5.1 Phone Upgrade Issue
Importance: High

 

I have a very strange issue with one of our remote sites, I have
upgraded some phones to SCCP70.8-2-1S and now they are unable to dial
out through there local H323 gateway, the number will ring twice and
then the call cuts out and it says Busy on the phone, any phones that
have not been upgraded can still dial out fine. 

 

Now if I change the order of the Gateways in the Route group so that
they dial out of the back up gateway at our main site all phones can
dial out fine, I have also upgraded a number of other remote sites with
the same firmware and they are also fine. 

 

Can anyone advise what might cause this issue, as I can't understand why
this would happen especially just at one site.

 

  

 


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Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
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Authorised and Regulated by the Financial Services Authority, Member of
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Registered Office: 25 Luke Street London EC2A 4AR

Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
of Charles Stanley & Co. Ltd   

Authorised and Regulated by the Financial Services Authority, Member of
the London Stock Exchange, The 
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Financial Futures & Options Exchange. 

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Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions of Charles Stanley & Co. Ltd  

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From Hans-Peter.Walter at tds.de  Mon May 21 05:01:21 2007
From: Hans-Peter.Walter at tds.de (Hans-Peter Walter)
Date: Mon, 21 May 2007 11:01:21 +0200
Subject: [cisco-voip] Antwort: Re: H323 Enhanced Call
	Preservation	andMedia	Inactivity Detection
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0059BDFA7@USNAEXCH.na.didata.local>
Message-ID: <OFA7DB68B8.17860DC1-ONC12572E2.00309551-C12572E2.0031579F@de.tds-global.com>

I had the same problem, and after adding...

voice service voip 
 h323 
 no h225 timeout keepalive 

...I can switch between SRST and CCM without interruptions of calls.
There was no need to change any parameters in CCM for my environment.

HTH,
Hans-Peter






"Jason Aarons (US)" <jason.aarons at us.didata.com> 
Gesendet von: cisco-voip-bounces at puck.nether.net
19.05.2007 00:23

An
"Brian Henry" <Brian.Henry at apptis.com>, "Robert Kulagowski" 
<bob at smalltime.com>, <cisco-voip at puck.nether.net>
Kopie

Thema
Re: [cisco-voip] H323 Enhanced Call Preservation        andMedia 
Inactivity Detection






I?m not sure what service parameter you are referring to  -jason
 
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Henry
Sent: Friday, May 18, 2007 4:04 PM
To: Robert Kulagowski; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H323 Enhanced Call Preservation andMedia 
Inactivity Detection
 
Remember to also change you service parameter in CCM to allow the call 
preservation as well.
 

From: cisco-voip-bounces at puck.nether.net on behalf of Robert Kulagowski
Sent: Fri 5/18/2007 3:23 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H323 Enhanced Call Preservation and Media 
Inactivity Detection
Jason Aarons (US) wrote:
> I shutdown (start/shutdown, maybe I should have just pulled the NIC?) my
> subscriber with my phone showed Active Registered, my CP-7941G-GE
> running 8.2(2)SR1S dropped the call, phone shows CM Down, features
> disabled, then registered with publisher. Test was not the desired
> results!
>
> voice service voip
>  allow-connections h323 to h323
>  h323
>   call preserve
>
> IOS is 12.4(11)T2

Well, because we're single threaded over the WAN, our situation is that
we lose connectivity to _all_ CMs, not just one.

Did you try pulling your "WAN" connection instead of shutting down a CM?

We use:

voice service voip
  fax protocol pass-through g711ulaw
  h323
   call preserve limit-media-detection
  modem passthrough nse codec g711ulaw
!
!
voice class codec 1
  codec preference 1 g711ulaw
  codec preference 2 g729r8
!
!
!
voice class h323 1
  h225 timeout tcp establish 5
   call preserve

works on 2851 / 12.4.9(T)
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From rratliff at cisco.com  Mon May 21 09:06:08 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 21 May 2007 09:06:08 -0400
Subject: [cisco-voip] Re-enabled NIC on the Call Manager Has Prevented
	Access to the System - Please Help!
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0059BE0C7@USNAEXCH.na.didata.local>
References: <B9B5B75DF9145F44A21111DD92CC490006538AAF@DCEX2.DSMO.COM><310F10740485D9469F0D8417950B252D0281F897@CTG-MSNEXC01.staff.berbee.com>
	<B9B5B75DF9145F44A21111DD92CC490006538AB3@DCEX2.DSMO.COM>
	<C1FE15183DA37645BC0633BC604E44F0059BE0C7@USNAEXCH.na.didata.local>
Message-ID: <CB07FEB5-E7B5-4409-9ED4-048060F800AC@cisco.com>

My recommendation has always been to use NFT only, never load balancing.

-Ryan

On May 19, 2007, at 3:36 PM, Jason Aarons ((US)) wrote:

Using the Cisco Media Convergence Server Network Teaming Driver with  
Operating System Versions 2000.4.3 and 2000.4.3a

http://www.cisco.com/en/US/products/hw/voiceapp/ps378/ 
prod_installation_guide09186a00806a476f.html



The above document ?strongly suggests NFT? in Step6, the default is  
Auto.



I don?t see anything that states Auto is unsupported by Cisco.  I  
would think being able to use two NICS @  1000/Full would be better  
than one @ 1000/Full -jason



In Auto mode connected to a Catalyst 3500/4500/6500 with default  
settings Auto will select Transmit Load Balancing with Fault  
Tolerance (TLB)  based upon the HP Whitepaper below. If you go into  
Control Panel > Networks > Teaming Interface > Properties> HP Network  
Configuration Utility > Select HP Network Team > Properties >  
Information it will shows the actual ?Current Mode? that Auto  
selected. I went into a OS2000-4-4a-SR6 and found they were Transmit  
Load Balancing with Fault Tolerance (TLB) and have been in production  
for several years with no issues relating to the NIC teaming. IPCC  
Express application isn?t supported with NIC teaming and does have  
problems.



ftp://ftp.compaq.com/pub/products/servers/networking/TeamingWP.pdf





From: Clouse, Chris [mailto:chris.clouse at berbee.com]
Sent: Saturday, May 19, 2007 1:40 PM
To: Miller, Steve
Subject: RE: [cisco-voip] Re-enabled NIC on the Call Manager Has  
Prevented Access to the System - Please Help!

The NIC?s should not be teamed.  The only method that is supported is  
Fault Tolerance Only.  Enable this mode, connect both cables, and try  
again.



Christopher Clouse, CCNP CCDP CCVP MCP Network+



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Saturday, May 19, 2007 12:16 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Re-enabled NIC on the Call Manager Has  
Prevented Access to the System - Please Help!



We have NICs in a Call Manager Subscriber in NY and one had been  
disabled for a long time.  I was told by an expert engineer (who  
actually was the architect or our system) that all of the settings  
for the cards were correct and that there was probably a cable  
unplugged.  When I got to the site, there was a cable unplugged so I  
replaced the cable and then re-enabled the card that corresponded to  
that port. At that point, we rec'd errors in the event log that the  
system could not communicate with the NTP server and now we cannot  
connect to the Call Manager via RDP or by web interface programming.   
All calls are being processed and there seems to be no current user  
issue, but I need to know if I should simply disable the connection  
again and troubleshoot or if there is something in the teaming NIC  
card setup that needs to be changed.  I have never messed with this  
before and I want to fix it, but I don't want to interfere with the  
phone service.  I'm sure that this is a simple issue, but I don't  
want a misstep.  Please let me know if you are familiar with this.   
Thank you!



Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202)-330-5607
millers at dicksteinshapiro.com



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on its network.  If you have received this confidential communication  
in error, please notify the sender immediately by reply e-mail  
message and permanently delete the original message.  To reply to our  
email administrator directly, send an email to  
postmaster at dicksteinshapiro.com Dickstein Shapiro LLPhttp:// 
www.DicksteinShapiro.com================================================ 
==============================------------------------------------------ 
--------------This e-mail message and any attached files are  
confidential and are intended solely for the use of the addressee(s)  
named above. This communication may contain material protected by  
attorney-client, work product, or other privileges. If you are not  
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confidential communication to the intended recipient, you have  
received this communication in error, and any review, use,  
dissemination, forwarding, printing, copying, or other distribution  
of this e-mail message and any attached files is strictly prohibited.  
Dickstein Shapiro reserves the right to monitor any communication  
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received this confidential communication in error, please notify the  
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==============================


Disclaimer: This e-mail communication and any attachments may contain  
confidential and privileged information and is for use by the  
designated addressee(s) named above only. If you are not the intended  
addressee, you are hereby notified that you have received this  
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From jonvoip at gmail.com  Mon May 21 11:20:36 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 21 May 2007 10:20:36 -0500
Subject: [cisco-voip] CallManager service start timer...
Message-ID: <5d093f9a0705210820t38752406r688fea4a7c78deed@mail.gmail.com>

I have a customer with a huge number of route patterns (3000+), and
every time they reboot the server, the CCM service fails to start
(didn't respond in a timely fashion)...

I have fixed this before, by changing the timeout for the service to
start... but I can't remember what the parameter to change is...

Any ideas?



Jonathan

From rratliff at cisco.com  Mon May 21 11:38:23 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 21 May 2007 11:38:23 -0400
Subject: [cisco-voip] CallManager service start timer...
In-Reply-To: <5d093f9a0705210820t38752406r688fea4a7c78deed@mail.gmail.com>
References: <5d093f9a0705210820t38752406r688fea4a7c78deed@mail.gmail.com>
Message-ID: <1812249E-204B-4ACF-BD9D-6BD97F547489@cisco.com>

Look at the service parameters for CallManager.  There is an entire  
section of various initialization timers (Clusterwide Parameters  
System - General).  The one you want is most likely Route Plan  
Initialization Timer.

-Ryan

On May 21, 2007, at 11:20 AM, Jonathan Charles wrote:

I have a customer with a huge number of route patterns (3000+), and
every time they reboot the server, the CCM service fails to start
(didn't respond in a timely fashion)...

I have fixed this before, by changing the timeout for the service to
start... but I can't remember what the parameter to change is...

Any ideas?



Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From arlen.nascimento at gmail.com  Mon May 21 11:40:43 2007
From: arlen.nascimento at gmail.com (Arlen Nascimento)
Date: Mon, 21 May 2007 11:40:43 -0400
Subject: [cisco-voip] Sip trunk
Message-ID: <51461ca60705210840h5ac89d72m53b09e8df9cc7635@mail.gmail.com>

Good day,

did anyboy here configure a sip trunk between ccme and asterisk?
could you place the code here?
and how do i assign a dial pattern to a specific extension?

Regards

-- 
Arlen Nascimento
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From jonvoip at gmail.com  Mon May 21 11:51:21 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 21 May 2007 10:51:21 -0500
Subject: [cisco-voip] CallManager service start timer...
In-Reply-To: <003c01c79bbe$b2f6d1c0$c901a8c0@pdp11>
References: <5d093f9a0705210820t38752406r688fea4a7c78deed@mail.gmail.com>
	<003c01c79bbe$b2f6d1c0$c901a8c0@pdp11>
Message-ID: <5d093f9a0705210851l382e352dg68cd86442ecb090f@mail.gmail.com>

60 sites, toll bypass like a mofo and a crapload of 10-digit local dialing...


Jonathan

On 5/21/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> 3000?
>
> yowza!
>
> what for?
>
>
> ----- Original Message -----
> From: "Jonathan Charles" <jonvoip at gmail.com>
> To: "ciscovoip" <cisco-voip at puck.nether.net>
> Sent: Monday, May 21, 2007 11:20 AM
> Subject: [cisco-voip] CallManager service start timer...
>
>
> >I have a customer with a huge number of route patterns (3000+), and
> > every time they reboot the server, the CCM service fails to start
> > (didn't respond in a timely fashion)...
> >
> > I have fixed this before, by changing the timeout for the service to
> > start... but I can't remember what the parameter to change is...
> >
> > Any ideas?
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From jonvoip at gmail.com  Mon May 21 11:53:39 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 21 May 2007 10:53:39 -0500
Subject: [cisco-voip] CallManager service start timer...
In-Reply-To: <1812249E-204B-4ACF-BD9D-6BD97F547489@cisco.com>
References: <5d093f9a0705210820t38752406r688fea4a7c78deed@mail.gmail.com>
	<1812249E-204B-4ACF-BD9D-6BD97F547489@cisco.com>
Message-ID: <5d093f9a0705210853w1582bf99y210e18d61882b0b8@mail.gmail.com>

That was it... thnx!


Jonathan

On 5/21/07, Ryan Ratliff <rratliff at cisco.com> wrote:
> Look at the service parameters for CallManager.  There is an entire
> section of various initialization timers (Clusterwide Parameters
> System - General).  The one you want is most likely Route Plan
> Initialization Timer.
>
> -Ryan
>
> On May 21, 2007, at 11:20 AM, Jonathan Charles wrote:
>
> I have a customer with a huge number of route patterns (3000+), and
> every time they reboot the server, the CCM service fails to start
> (didn't respond in a timely fashion)...
>
> I have fixed this before, by changing the timeout for the service to
> start... but I can't remember what the parameter to change is...
>
> Any ideas?
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From matt at saskin.net  Mon May 21 11:53:56 2007
From: matt at saskin.net (Matthew Saskin)
Date: Mon, 21 May 2007 11:53:56 -0400
Subject: [cisco-voip] Sip trunk
In-Reply-To: <51461ca60705210840h5ac89d72m53b09e8df9cc7635@mail.gmail.com>
References: <51461ca60705210840h5ac89d72m53b09e8df9cc7635@mail.gmail.com>
Message-ID: <4651C094.1010606@saskin.net>

I've done SIP trunks between CCM (not express) and asterisk.  You 
looking for the asterisk extensions.conf configuration or the CCM Side 
of things?

-matt

Arlen Nascimento wrote:
> Good day,
> 
> did anyboy here configure a sip trunk between ccme and asterisk?
> could you place the code here?
> and how do i assign a dial pattern to a specific extension?
> 
> Regards
> 
> -- 
> Arlen Nascimento
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From asobihoudai at yahoo.com  Mon May 21 10:58:52 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Mon, 21 May 2007 07:58:52 -0700 (PDT)
Subject: [cisco-voip] LEC not supporting *
In-Reply-To: <CB07FEB5-E7B5-4409-9ED4-048060F800AC@cisco.com>
Message-ID: <909705.39629.qm@web57204.mail.re3.yahoo.com>

Hi everyone,

I just wanted to bring your attention to this oddball
problem I had.

Users dialing into Unity remotely from a certain site
weren't able to login using *. After checking and
double checking the configuration to make sure we had
DTMF relay configured correctly, a colleague suggested
for me to try a different number for sign-in and it
worked. He told me that certain carriers do NOT
support the * key. We're going to propose to the
customer that they use 1 instead of *.

Also, waaay off topic but am I the only one getting a
LOT of 419 emails? 


       
____________________________________________________________________________________Give spam the boot. Take control with tough spam protection in the all-new Yahoo! Mail Beta.
http://advision.webevents.yahoo.com/mailbeta/newmail_html.html 

From steve.siltman at assurant.com  Mon May 21 11:34:10 2007
From: steve.siltman at assurant.com (steve.siltman at assurant.com)
Date: Mon, 21 May 2007 10:34:10 -0500
Subject: [cisco-voip] NetBackup Client
In-Reply-To: <mailman.14585.1179737715.1489.cisco-voip@puck.nether.net>
Message-ID: <OFCC47EABE.31C3B5DB-ON852572E2.005543F0-862572E2.005585AC@assurant.com>

Is it supported to use the Veritas NetBackup client on CallManager 4.1.3?

or is it better to copy the BARS backup file to a network drive?

Thanks,

steve.siltman at assurant.com
**************************************************************************************
This e-mail message and all attachments transmitted with it may contain legally privileged and/or confidential information intended solely for the use of the addressee(s). If the reader of this message is not the intended recipient, you are hereby notified that any reading, dissemination, distribution, copying, forwarding or other use of this message or its attachments is strictly prohibited. If you have received this message in error, please notify the sender immediately and delete this message and all copies and backups thereof.

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From Scott.Voll at wesd.org  Mon May 21 12:06:21 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 21 May 2007 09:06:21 -0700
Subject: [cisco-voip] NetBackup Client
In-Reply-To: <OFCC47EABE.31C3B5DB-ON852572E2.005543F0-862572E2.005585AC@assurant.com>
References: <mailman.14585.1179737715.1489.cisco-voip@puck.nether.net>
	<OFCC47EABE.31C3B5DB-ON852572E2.005543F0-862572E2.005585AC@assurant.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7A5@Apollo.wesd.org>

I don't believe that is supported.

 

May I suggest you bars backup to network drive and then backup to tape
from there.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
steve.siltman at assurant.com
Sent: Monday, May 21, 2007 8:34 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] NetBackup Client

 


Is it supported to use the Veritas NetBackup client on CallManager
4.1.3? 

or is it better to copy the BARS backup file to a network drive? 

Thanks,

steve.siltman at assurant.com 

________________________________

This e-mail message and all attachments transmitted with it may contain
legally privileged and/or confidential information intended solely for
the use of the addressee(s). If the reader of this message is not the
intended recipient, you are hereby notified that any reading,
dissemination, distribution, copying, forwarding or other use of this
message or its attachments is strictly prohibited. If you have received
this message in error, please notify the sender immediately and delete
this message and all copies and backups thereof.

Thank you.

________________________________

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From Scott.Voll at wesd.org  Mon May 21 13:54:02 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 21 May 2007 10:54:02 -0700
Subject: [cisco-voip] CM 5 or 6
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7AD@Apollo.wesd.org>

I'm staining my mind, but I thought I remembered a preso on CM 5 about
location based call being able to be extended stars rather then just
star type topology.  Is this correct?  

 

Thanks

 

Scott

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From August.SARKANY at nextiraone.at  Mon May 21 13:46:47 2007
From: August.SARKANY at nextiraone.at (SARKANY August)
Date: Mon, 21 May 2007 19:46:47 +0200
Subject: [cisco-voip] CA e-Health and e-Health for voice
Message-ID: <114398F6873B1646BA549F43A5BA82D19D8CC0@atmai001.at.nextiraone-eu.ned>

Hello,
 
any experiences with ca e-helath or e-health for voice
and their impact on callmanager.
 
many thanks in advance
 
August

From mmilanovic at ctiusa.com  Mon May 21 14:30:09 2007
From: mmilanovic at ctiusa.com (Mladen Milanovic)
Date: Mon, 21 May 2007 14:30:09 -0400
Subject: [cisco-voip] Sip trunk
In-Reply-To: <4651C094.1010606@saskin.net>
References: <51461ca60705210840h5ac89d72m53b09e8df9cc7635@mail.gmail.com>
	<4651C094.1010606@saskin.net>
Message-ID: <4761C60AD1E74A4D8606CAF48582265A6F0274@EXCHANGE.ctiusa.com>

Do you look for the configuration on the ccme side ?

M 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matthew Saskin
Sent: Monday, May 21, 2007 11:54 AM
To: Arlen Nascimento
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Sip trunk

I've done SIP trunks between CCM (not express) and asterisk.  You
looking for the asterisk extensions.conf configuration or the CCM Side
of things?

-matt

Arlen Nascimento wrote:
> Good day,
> 
> did anyboy here configure a sip trunk between ccme and asterisk?
> could you place the code here?
> and how do i assign a dial pattern to a specific extension?
> 
> Regards
> 
> --
> Arlen Nascimento
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jonvoip at gmail.com  Mon May 21 14:42:01 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 21 May 2007 13:42:01 -0500
Subject: [cisco-voip] "CM down, features disabled" Message
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538A9D@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490006538A9D@DCEX2.DSMO.COM>
Message-ID: <5d093f9a0705211142p2d0971d4sf252096e60f3279a@mail.gmail.com>

Define "down"....

Seriously, I have seen a network hiccup cause this and the only people
who got this message were those who had an active call up (the other
phones reset so fast no one noticed).



Jonathan

On 5/19/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
>
>
> I have see this happen on a number of different occasions when there was
> basically no system disruption.  Has anyone seen this message on their phone
> in situations where CM was clearly NOT down?
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370 Fax (202)-330-5607
> millers at dicksteinshapiro.com
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are intended
> solely for the use of the addressee(s) named above. This communication may
> contain material protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person responsible for
> delivering this confidential communication to the intended recipient, you
> have received this communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or other distribution of this
> e-mail message and any attached files is strictly prohibited. Dickstein
> Shapiro reserves the right to monitor any communication that is created,
> received, or sent on its network. If you have received this confidential
> communication in error, please notify the sender immediately by reply e-mail
> message and permanently delete the original message.
>
> To reply to our email administrator directly, send an email to
> postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
> ==============================================================================
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From MillerS at DicksteinShapiro.COM  Mon May 21 14:45:12 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Mon, 21 May 2007 14:45:12 -0400
Subject: [cisco-voip] "CM down, features disabled" Message
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538AE8@DCEX2.DSMO.COM>

Interesting.  Thanks.

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 
Fax (202)-330-5607
millers at dicksteinshapiro.com 

----- Original Message -----
From: Jonathan Charles <jonvoip at gmail.com>
To: Miller, Steve
Cc: cisco-voip at puck.nether.net <cisco-voip at puck.nether.net>
Sent: Mon May 21 14:42:01 2007
Subject: Re: [cisco-voip] "CM down, features disabled" Message

Define "down"....

Seriously, I have seen a network hiccup cause this and the only people
who got this message were those who had an active call up (the other
phones reset so fast no one noticed).



Jonathan

On 5/19/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
>
>
> I have see this happen on a number of different occasions when there was
> basically no system disruption.  Has anyone seen this message on their phone
> in situations where CM was clearly NOT down?
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370 Fax (202)-330-5607
> millers at dicksteinshapiro.com
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are intended
> solely for the use of the addressee(s) named above. This communication may
> contain material protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person responsible for
> delivering this confidential communication to the intended recipient, you
> have received this communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or other distribution of this
> e-mail message and any attached files is strictly prohibited. Dickstein
> Shapiro reserves the right to monitor any communication that is created,
> received, or sent on its network. If you have received this confidential
> communication in error, please notify the sender immediately by reply e-mail
> message and permanently delete the original message.
>
> To reply to our email administrator directly, send an email to
> postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
> ==============================================================================
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

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From Scott.Voll at wesd.org  Mon May 21 14:53:22 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 21 May 2007 11:53:22 -0700
Subject: [cisco-voip] "CM down, features disabled" Message
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538AE8@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490006538AE8@DCEX2.DSMO.COM>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7B0@Apollo.wesd.org>

I just had this problem...... I had a switch that the Gig port was going
up/up .... Up/down.... Up/ Up........

 

Had to RMA the switch and all is working now.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Monday, May 21, 2007 11:45 AM
To: jonvoip at gmail.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] "CM down, features disabled" Message

 

Interesting.  Thanks.

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370
Fax (202)-330-5607
millers at dicksteinshapiro.com

----- Original Message -----
From: Jonathan Charles <jonvoip at gmail.com>
To: Miller, Steve
Cc: cisco-voip at puck.nether.net <cisco-voip at puck.nether.net>
Sent: Mon May 21 14:42:01 2007
Subject: Re: [cisco-voip] "CM down, features disabled" Message

Define "down"....

Seriously, I have seen a network hiccup cause this and the only people
who got this message were those who had an active call up (the other
phones reset so fast no one noticed).



Jonathan

On 5/19/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
>
>
> I have see this happen on a number of different occasions when there
was
> basically no system disruption.  Has anyone seen this message on their
phone
> in situations where CM was clearly NOT down?
>
>
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370 Fax (202)-330-5607
> millers at dicksteinshapiro.com
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are
intended
> solely for the use of the addressee(s) named above. This communication
may
> contain material protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person
responsible for
> delivering this confidential communication to the intended recipient,
you
> have received this communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or other distribution of
this
> e-mail message and any attached files is strictly prohibited.
Dickstein
> Shapiro reserves the right to monitor any communication that is
created,
> received, or sent on its network. If you have received this
confidential
> communication in error, please notify the sender immediately by reply
e-mail
> message and permanently delete the original message.
>
> To reply to our email administrator directly, send an email to
> postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
>
========================================================================
======
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
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From David_Zilk at adp.com  Mon May 21 14:54:59 2007
From: David_Zilk at adp.com (Zilk, David)
Date: Mon, 21 May 2007 13:54:59 -0500
Subject: [cisco-voip] CNAME via H.323
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3B0241B51B@EXCHANGEDS01.ds.ad.adp.com>

I have come up with a method to correctly send Caller Name information
received via ISDN Facility IE when using H.323 to communicate with Call
Manager, at least in some telco configurations.  

This is accomplished by installing a TCL script on the voice gateway
router.  In general, the TCL script intercepts the call, and before
sending it on to the dial-peer, copies the CNAME info into the field
supported by Call Manager in the H.323 communication.

 

I have tested it in the situation where the Facility IE message is sent
as a separate IE, after the initial call setup, and am interested in
seeing how this script works in other telco environments.  I will make
it available to anyone who wants it, if they are willing to send me some
debugs from the router it is installed on.

 

Please email me directly at david_zilk at adp.com, and I will respond back
with the script and instructions for implementing it.

 

David Zilk

Network Engineer

ADP Dealer Service

 

 



This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From asobihoudai at yahoo.com  Mon May 21 15:42:43 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Mon, 21 May 2007 12:42:43 -0700 (PDT)
Subject: [cisco-voip] Dialing from the IP phone's corporate directory
Message-ID: <75119.67686.qm@web57210.mail.re3.yahoo.com>

My co-worker discovered that the corporate directory
pulls its DNs from Active Directory on the Unity
server. We will most likely append 91 to all DNs under
active directory so users can dial directly using
that.

I hope this information is useful to someone.

Paul


 
____________________________________________________________________________________
It's here! Your new message!  
Get new email alerts with the free Yahoo! Toolbar.
http://tools.search.yahoo.com/toolbar/features/mail/

From LuisYrigoyen at bayviewfinancial.com  Mon May 21 15:52:51 2007
From: LuisYrigoyen at bayviewfinancial.com (LuisYrigoyen at bayviewfinancial.com)
Date: Mon, 21 May 2007 15:52:51 -0400
Subject: [cisco-voip] IPCC HTTP requests
Message-ID: <OFDED41DAC.6797C306-ON852572E2.006CD3CB-852572E2.006D353F@bftg.com>

Hi, 

We have IPCC express enhanced and I'm trying to send an HTTP request to 
website (ASP page). 
I'm trying to send a parameter to an ASP webpage to make it run. All I 
want is to deliver it. 

Has anybody used the Send HTTP Response step to send an HTTP request to an 
external web page?

I was reading the cisco guide and it says to use the Create URL Document + 
the Send HTTP Response step but it doesn't work. 

My Create URL Document: 
URL= http://x.x.x.x/webpage.asp 
Method = Post 
Parameters = name = f, values = "something" 
document = doc 

My Send HTTP Response step: 
Document = doc 


Any help will be greatly appreciated. 

thank you 



Luis Yrigoyen

Network Operations
Bayview Financial, L.P.
4425 Ponce de Leon Blvd.
Coral Gables, FL 33146

T 305.817.5083
F 305.646.4744
E luisyrigoyen at bayviewfinancial.com
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From jonvoip at gmail.com  Mon May 21 16:25:44 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 21 May 2007 15:25:44 -0500
Subject: [cisco-voip] Change passwords on Cisco Emergency Responder...
Message-ID: <5d093f9a0705211325y1b1e5da3r93aad6364055d23c@mail.gmail.com>

Cust is firing someone, need to change passwords on CER... just a
single server, not finding any instructions on Cisco's web page...
kind of an emergency...



Jonathan

From jonvoip at gmail.com  Mon May 21 16:30:18 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 21 May 2007 15:30:18 -0500
Subject: [cisco-voip] Dialing from the IP phone's corporate directory
In-Reply-To: <75119.67686.qm@web57210.mail.re3.yahoo.com>
References: <75119.67686.qm@web57210.mail.re3.yahoo.com>
Message-ID: <5d093f9a0705211330m56bebb72s1a8d3836d35120f5@mail.gmail.com>

This seems highly unlikely. However, it could be done...

Usually, what we when we integrate CCM with AD/LDAP is we connect to
the customer's AD infrastructure, not Unity (Unity can either be
pulling from AD or be its own standalone domain if it is just vmail
only).

It would be very strange if you integrated CCM with Unity's AD infrastructure...



Jonathan

On 5/21/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> My co-worker discovered that the corporate directory
> pulls its DNs from Active Directory on the Unity
> server. We will most likely append 91 to all DNs under
> active directory so users can dial directly using
> that.
>
> I hope this information is useful to someone.
>
> Paul
>
>
>
> ____________________________________________________________________________________
> It's here! Your new message!
> Get new email alerts with the free Yahoo! Toolbar.
> http://tools.search.yahoo.com/toolbar/features/mail/
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Mon May 21 16:48:50 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 21 May 2007 15:48:50 -0500
Subject: [cisco-voip] Change passwords on Cisco Emergency Responder...
In-Reply-To: <5d093f9a0705211325y1b1e5da3r93aad6364055d23c@mail.gmail.com>
References: <5d093f9a0705211325y1b1e5da3r93aad6364055d23c@mail.gmail.com>
Message-ID: <5d093f9a0705211348u544451acn691cb7562fea1cca@mail.gmail.com>

OK, I found out I just need to remove the user from Windows on CER...
however, how do I make sure I don't break the link between the CER and
CCM?



Jonathan

On 5/21/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> Cust is firing someone, need to change passwords on CER... just a
> single server, not finding any instructions on Cisco's web page...
> kind of an emergency...
>
>
>
> Jonathan
>

From Brian.Henry at apptis.com  Mon May 21 18:43:03 2007
From: Brian.Henry at apptis.com (Brian Henry)
Date: Mon, 21 May 2007 18:43:03 -0400
Subject: [cisco-voip] H323 Gateway and 0X3A Bearer Capability Not Available
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6D4@CH-EXCH-02.intelltech.apptis.inc>

I have a CCM 4.3(1) with 2821 VGW 12.4(11)T1 running in H323 mode.
 
Testing:
 
Using LAN Line
I can call into VGW and receive ring tone on call setup/proceed.
 
Using 7961 8.2(2)sr1
I can call out but certain numbers I call do not connect and get, and also do not get ring indicator
call_progress: DEV_CONNECTED VOICE ERROR: Bearer capability not available(0x3A): b channel 2 call id 0x50D
 
IP Communicator 2.1(1)
Can call anywhere and receive ring indicator as well as get connected to calls that I cannot through regular hard phones.
 
I have added the progress_ind setup/connect/progress commands under the pots dial-peers to correct the ring back issue but to no avail.
 
I do not have Display IE checked and I am not running fast-start for inbound nor outbound.
 
DSP's are in flex mode and the firmware is 9.2.2 I do believe.
 
 
Very weird stuff when I can call everywhere with IPC but not a hard phone and the funny thing is that sometimes I can call it.
 
Brian
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From asobihoudai at yahoo.com  Mon May 21 18:35:07 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Mon, 21 May 2007 15:35:07 -0700 (PDT)
Subject: [cisco-voip] Dialing from the IP phone's corporate directory
In-Reply-To: <5d093f9a0705211330m56bebb72s1a8d3836d35120f5@mail.gmail.com>
Message-ID: <165086.72265.qm@web57209.mail.re3.yahoo.com>

Jonathan, 

You're right.

We checked on a different server and it was NOT the
case that we had all of our users in the active
directory.

Back to the drawing board... :(

Paul

--- Jonathan Charles <jonvoip at gmail.com> wrote:

> This seems highly unlikely. However, it could be
> done...
> 
> Usually, what we when we integrate CCM with AD/LDAP
> is we connect to
> the customer's AD infrastructure, not Unity (Unity
> can either be
> pulling from AD or be its own standalone domain if
> it is just vmail
> only).
> 
> It would be very strange if you integrated CCM with
> Unity's AD infrastructure...
> 
> 
> 
> Jonathan
> 
> On 5/21/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> > My co-worker discovered that the corporate
> directory
> > pulls its DNs from Active Directory on the Unity
> > server. We will most likely append 91 to all DNs
> under
> > active directory so users can dial directly using
> > that.
> >
> > I hope this information is useful to someone.
> >
> > Paul
> >
> >
> >
> >
>
____________________________________________________________________________________
> > It's here! Your new message!
> > Get new email alerts with the free Yahoo! Toolbar.
> >
> http://tools.search.yahoo.com/toolbar/features/mail/
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> 



 
____________________________________________________________________________________
Now that's room service!  Choose from over 150,000 hotels
in 45,000 destinations on Yahoo! Travel to find your fit.
http://farechase.yahoo.com/promo-generic-14795097

From chris.clouse at berbee.com  Mon May 21 19:58:31 2007
From: chris.clouse at berbee.com (Clouse, Chris)
Date: Mon, 21 May 2007 18:58:31 -0500
Subject: [cisco-voip] Dialing from the IP phone's corporate directory
In-Reply-To: <165086.72265.qm@web57209.mail.re3.yahoo.com>
References: <5d093f9a0705211330m56bebb72s1a8d3836d35120f5@mail.gmail.com>
	<165086.72265.qm@web57209.mail.re3.yahoo.com>
Message-ID: <310F10740485D9469F0D8417950B252D0285FED9@CTG-MSNEXC01.staff.berbee.com>

You can have the directory application search multiple directories
without being in the domain.  Have you checked the application you were
using for multiple directory entries?



Christopher Clouse, CCNP CCDP CCVP MCP Network+


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 5:35 PM
To: Jonathan Charles
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Dialing from the IP phone's corporate
directory

Jonathan, 

You're right.

We checked on a different server and it was NOT the
case that we had all of our users in the active
directory.

Back to the drawing board... :(

Paul

--- Jonathan Charles <jonvoip at gmail.com> wrote:

> This seems highly unlikely. However, it could be
> done...
> 
> Usually, what we when we integrate CCM with AD/LDAP
> is we connect to
> the customer's AD infrastructure, not Unity (Unity
> can either be
> pulling from AD or be its own standalone domain if
> it is just vmail
> only).
> 
> It would be very strange if you integrated CCM with
> Unity's AD infrastructure...
> 
> 
> 
> Jonathan
> 
> On 5/21/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> > My co-worker discovered that the corporate
> directory
> > pulls its DNs from Active Directory on the Unity
> > server. We will most likely append 91 to all DNs
> under
> > active directory so users can dial directly using
> > that.
> >
> > I hope this information is useful to someone.
> >
> > Paul
> >
> >
> >
> >
>
________________________________________________________________________
____________
> > It's here! Your new message!
> > Get new email alerts with the free Yahoo! Toolbar.
> >
> http://tools.search.yahoo.com/toolbar/features/mail/
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> 



 
________________________________________________________________________
____________
Now that's room service!  Choose from over 150,000 hotels
in 45,000 destinations on Yahoo! Travel to find your fit.
http://farechase.yahoo.com/promo-generic-14795097
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jason.aarons at us.didata.com  Mon May 21 19:58:27 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 21 May 2007 19:58:27 -0400
Subject: [cisco-voip] (no subject)
Message-ID: <C1FE15183DA37645BC0633BC604E44F005A0398E@USNAEXCH.na.didata.local>

If a call comes in for 8900 matching dial-peer 28900 I want to translate
the called number to 8912...

 

What am I doing wrong ?

 

voice translation-rule 1

 rule 1 /987654/ //

 rule 2 /8900/ /8912/

!

!

voice translation-profile 8900-8912

 translate called 1

!

dial-peer voice 28900 voip

translation-profile incoming 8900-8912

 

 

VGW1#test translation-rule 1 8900

Translation Rule 1 is not defined

VGW1#

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From aizhar at ccbootcamp.com  Mon May 21 20:12:37 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Mon, 21 May 2007 17:12:37 -0700
Subject: [cisco-voip] (no subject) [bcc][faked-from][html-rem]
References: <C1FE15183DA37645BC0633BC604E44F005A0398E@USNAEXCH.na.didata.local>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FE2@nlimail.nlidomain>

Try doing 'test voice translation-rule 1', also might want to match /^8900/ so it only match the beginning of the string.
 
-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take Cisco Learning credits!
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http://www.ccbootcamp.com/groupstudy.html <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012


________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jason Aarons (US)
Sent: Mon 5/21/2007 4:58 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] (no subject) [bcc][faked-from][html-rem]



If a call comes in for 8900 matching dial-peer 28900 I want to translate
the called number to 8912...



What am I doing wrong ?



voice translation-rule 1

 rule 1 /987654/ //

 rule 2 /8900/ /8912/

!

!

voice translation-profile 8900-8912

 translate called 1

!

dial-peer voice 28900 voip

translation-profile incoming 8900-8912





VGW1#test translation-rule 1 8900

Translation Rule 1 is not defined

VGW1#





Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na>

904-338-3245 cell



For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.






-----------------------------------------
=
0ADisclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

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From asobihoudai at yahoo.com  Mon May 21 20:19:11 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Mon, 21 May 2007 17:19:11 -0700 (PDT)
Subject: [cisco-voip] (no subject)
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005A0398E@USNAEXCH.na.didata.local>
Message-ID: <952205.6175.qm@web57209.mail.re3.yahoo.com>

You've setup voice translation rules...

you're not testing them with 'voice translation-rule.'

Cheers,
Paul

--- "Jason Aarons (US)" <jason.aarons at us.didata.com>
wrote:

> If a call comes in for 8900 matching dial-peer 28900
> I want to translate
> the called number to 8912...
> 
>  
> 
> What am I doing wrong ?
> 
>  
> 
> voice translation-rule 1
> 
>  rule 1 /987654/ //
> 
>  rule 2 /8900/ /8912/
> 
> !
> 
> !
> 
> voice translation-profile 8900-8912
> 
>  translate called 1
> 
> !
> 
> dial-peer voice 28900 voip
> 
> translation-profile incoming 8900-8912
> 
>  
> 
>  
> 
> VGW1#test translation-rule 1 8900
> 
> Translation Rule 1 is not defined
> 
> VGW1#
> 



       
____________________________________________________________________________________Choose the right car based on your needs.  Check out Yahoo! Autos new Car Finder tool.
http://autos.yahoo.com/carfinder/

From lelio at uoguelph.ca  Mon May 21 20:22:01 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 21 May 2007 20:22:01 -0400
Subject: [cisco-voip] (no subject)
References: <C1FE15183DA37645BC0633BC604E44F005A0398E@USNAEXCH.na.didata.local>
Message-ID: <001201c79c07$37e001c0$47196883@cfs.uoguelph.ca>

Jason, I'm doing something similar. I'm saying "anything that comes in, insert a 9 in the front". The key was the concept of "slices" for me. Also, you might notice that it's "test voice translation-rule" not test translation-rule. 

   Here's my config:

  !
  voice translation-rule 613
   rule 1 /\(.+\)/ /9\1/
  !
  voice translation-profile kc_did_profile
   translate called 613
  !
  dial-peer voice 1 pots
   translation-profile incoming kc_did_profile
   incoming called-number .
   direct-inward-dial
   port 0/0/0:23
  !

  vgw-kemptville#test voice translation-rule 613 ?
    WORD  Number to translate

  vgw-kemptville#test voice translation-rule 613 5195551212
  Matched with rule 1
  Original number: 5195551212     Translated number: 95195551212
  Original number type: none      Translated number type: none
  Original number plan: none      Translated number plan: none


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...

  ----- Original Message ----- 
  From: Jason Aarons (US) 
  To: cisco-voip at puck.nether.net 
  Sent: Monday, May 21, 2007 7:58 PM
  Subject: [cisco-voip] (no subject)


  If a call comes in for 8900 matching dial-peer 28900 I want to translate the called number to 8912.

   

  What am I doing wrong ?

   

  voice translation-rule 1

   rule 1 /987654/ //

   rule 2 /8900/ /8912/

  !

  !

  voice translation-profile 8900-8912

   translate called 1

  !

  dial-peer voice 28900 voip

  translation-profile incoming 8900-8912

   

   

  VGW1#test translation-rule 1 8900

  Translation Rule 1 is not defined

  VGW1#

   

   

        Jason Aarons

        Consultant
       http://www.dimensiondata.com/na

        904-338-3245 cell
       

   

  For urgent issues notify your Project Manager, for 24x7 support contact the Dimension Data NOC at 800-974-6584.

   



------------------------------------------------------------------------------


  Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 



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From d4rkf1ber at gmail.com  Mon May 21 21:21:45 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Mon, 21 May 2007 20:21:45 -0500
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
Message-ID: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com>

Anyone aware of any wireless SCCP phone devices as an alternative to
say a 7920?  One time I saw some lapel type phones demonstrated at a
Cisco event but have never seen anything official from Cisco regarding
them or similiar items.

Just curious, thanks in advanced.

From Cisco.voip at VERIZON.NET  Mon May 21 22:28:08 2007
From: Cisco.voip at VERIZON.NET (Cisco.voip at VERIZON.NET)
Date: Mon, 21 May 2007 22:28:08 -0400
Subject: [cisco-voip] Monitor channel usage mgcp
References: <mailman.14705.1179792893.1489.cisco-voip@puck.nether.net>
Message-ID: <000301c79c18$d670bfd0$0201a8c0@corporation2>

All, I need an application to monitor usage of channels on several pri mgcp 
cisco ios gateways
to provide usage details to mgmt.

What app's do i need to look at.

TIA


From asobihoudai at yahoo.com  Mon May 21 23:00:00 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Mon, 21 May 2007 20:00:00 -0700 (PDT)
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
In-Reply-To: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com>
Message-ID: <411933.7897.qm@web57210.mail.re3.yahoo.com>

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as
> an alternative to
> say a 7920?  One time I saw some lapel type phones
> demonstrated at a
> Cisco event but have never seen anything official
> from Cisco regarding
> them or similiar items.
> 
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



 
____________________________________________________________________________________
8:00? 8:25? 8:40? Find a flick in no time 
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news

From Scott.Voll at wesd.org  Mon May 21 23:01:12 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 21 May 2007 20:01:12 -0700
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
In-Reply-To: <411933.7897.qm@web57210.mail.re3.yahoo.com>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com>
	<411933.7897.qm@web57210.mail.re3.yahoo.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org>

Nokia?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 8:00 PM
To: Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as
> an alternative to
> say a 7920?  One time I saw some lapel type phones
> demonstrated at a
> Cisco event but have never seen anything official
> from Cisco regarding
> them or similiar items.
> 
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



 
________________________________________________________________________
____________
8:00? 8:25? 8:40? Find a flick in no time 
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jim at tgasolutions.com  Mon May 21 23:09:27 2007
From: jim at tgasolutions.com (Jim McBurnett)
Date: Mon, 21 May 2007 23:09:27 -0400
Subject: [cisco-voip] BAT File Issue.
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC31B8E7@exchange.TGASolutions.local>

Hey folks,
I was recently informed that these 2 fields are not supported via BAT in
CCM 5.1
ASCII Display Internal Caller ID
ASCII Line text Label

Now this is really only important when a 5.x cluster is connected to a
4.x cluster.
Then you do not get caller ID on inter cluster calls...

Is there anything coming soon to resolve this? Is it fixed in 6.x?

Thanks,
Jim

From cisco at adman.net  Mon May 21 23:10:52 2007
From: cisco at adman.net (Adam)
Date: Mon, 21 May 2007 22:10:52 -0500
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
In-Reply-To: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com>
Message-ID: <c6f32ce40705212010x2e75e902h83d1b4bdfa4a1673@mail.gmail.com>

I think the devices you are referring to may be the Vocera
communicators (http://www.vocera.com/). They not exactly straight SCCP
phones as they require the Vocera server to function.

-Adam

On 5/21/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
> Anyone aware of any wireless SCCP phone devices as an alternative to
> say a 7920?  One time I saw some lapel type phones demonstrated at a
> Cisco event but have never seen anything official from Cisco regarding
> them or similiar items.
>
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From lelio at uoguelph.ca  Mon May 21 22:31:37 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 21 May 2007 22:31:37 -0400
Subject: [cisco-voip] Monitor channel usage mgcp
References: <mailman.14705.1179792893.1489.cisco-voip@puck.nether.net>
	<000301c79c18$d670bfd0$0201a8c0@corporation2>
Message-ID: <002501c79c19$533e3650$c901a8c0@pdp11>

have you tried the gateway utilization reports? they're pretty useful.

you could look at MRTG and take 1-5 minute snapshots for trends.


----- Original Message ----- 
From: <Cisco.voip at VERIZON.NET>
To: <cisco-voip at puck.nether.net>
Sent: Monday, May 21, 2007 10:28 PM
Subject: [cisco-voip] Monitor channel usage mgcp


> All, I need an application to monitor usage of channels on several pri 
> mgcp
> cisco ios gateways
> to provide usage details to mgmt.
>
> What app's do i need to look at.
>
> TIA
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From Cisco.voip at VERIZON.NET  Mon May 21 23:32:37 2007
From: Cisco.voip at VERIZON.NET (Cisco.voip at VERIZON.NET)
Date: Mon, 21 May 2007 23:32:37 -0400
Subject: [cisco-voip] Monitor channel usage mgcp
References: <mailman.14705.1179792893.1489.cisco-voip@puck.nether.net>
	<000301c79c18$d670bfd0$0201a8c0@corporation2>
	<002501c79c19$533e3650$c901a8c0@pdp11>
Message-ID: <000d01c79c21$d87d1b80$0201a8c0@corporation2>

where are the gw utiliztion reports? I am using CCM 4.2
----- Original Message ----- 
From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
To: <Cisco.voip at VERIZON.NET>; <cisco-voip at puck.nether.net>
Sent: Monday, May 21, 2007 10:31 PM
Subject: Re: [cisco-voip] Monitor channel usage mgcp


> have you tried the gateway utilization reports? they're pretty useful.
> 
> you could look at MRTG and take 1-5 minute snapshots for trends.
> 
> 
> ----- Original Message ----- 
> From: <Cisco.voip at VERIZON.NET>
> To: <cisco-voip at puck.nether.net>
> Sent: Monday, May 21, 2007 10:28 PM
> Subject: [cisco-voip] Monitor channel usage mgcp
> 
> 
>> All, I need an application to monitor usage of channels on several pri 
>> mgcp
>> cisco ios gateways
>> to provide usage details to mgmt.
>>
>> What app's do i need to look at.
>>
>> TIA
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> 
>

From Scott.Voll at wesd.org  Tue May 22 00:31:54 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 21 May 2007 21:31:54 -0700
Subject: [cisco-voip] How do is see if both pub and sub are sync'd?
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7C4@Apollo.wesd.org>

Running CM 4.1.3es100

 

How do I tell if both CM's are sync'd?

 

Scott

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From lelio at uoguelph.ca  Mon May 21 23:36:47 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 21 May 2007 23:36:47 -0400
Subject: [cisco-voip] Monitor channel usage mgcp
References: <mailman.14705.1179792893.1489.cisco-voip@puck.nether.net>
	<000301c79c18$d670bfd0$0201a8c0@corporation2>
	<002501c79c19$533e3650$c901a8c0@pdp11>
	<000d01c79c21$d87d1b80$0201a8c0@corporation2>
Message-ID: <001101c79c22$6d5226b0$c901a8c0@pdp11>

go to serviceability. i believe it's in car.

----- Original Message ----- 
From: <Cisco.voip at VERIZON.NET>
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>; <cisco-voip at puck.nether.net>
Sent: Monday, May 21, 2007 11:32 PM
Subject: Re: [cisco-voip] Monitor channel usage mgcp


> where are the gw utiliztion reports? I am using CCM 4.2
> ----- Original Message ----- 
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> To: <Cisco.voip at VERIZON.NET>; <cisco-voip at puck.nether.net>
> Sent: Monday, May 21, 2007 10:31 PM
> Subject: Re: [cisco-voip] Monitor channel usage mgcp
> 
> 
>> have you tried the gateway utilization reports? they're pretty useful.
>> 
>> you could look at MRTG and take 1-5 minute snapshots for trends.
>> 
>> 
>> ----- Original Message ----- 
>> From: <Cisco.voip at VERIZON.NET>
>> To: <cisco-voip at puck.nether.net>
>> Sent: Monday, May 21, 2007 10:28 PM
>> Subject: [cisco-voip] Monitor channel usage mgcp
>> 
>> 
>>> All, I need an application to monitor usage of channels on several pri 
>>> mgcp
>>> cisco ios gateways
>>> to provide usage details to mgmt.
>>>
>>> What app's do i need to look at.
>>>
>>> TIA
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>> 
>>
>

From aizhar at ccbootcamp.com  Tue May 22 01:09:49 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Mon, 21 May 2007 22:09:49 -0700
Subject: [cisco-voip] How do is see if both pub and sub are sync'd?
	[bcc][faked-from][html-rem]
References: <407055A92CECCB499C922A2D35FC19A608DEE7C4@Apollo.wesd.org>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FE4@nlimail.nlidomain>

Using SQL enterprise manager, see if you have any red X signs on the replication agent.
 
Also create a phone via ccmadmin and verify with the same sql enterprise manager that it is on the subscriber's copy of the database.
 
Detailed instructions can be found at:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00801d11a6.shtml#veri
 
 
-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take Cisco Learning credits!
http://www.ccbootcamp.com <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012


________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Voll, Scott
Sent: Mon 5/21/2007 9:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How do is see if both pub and sub are sync'd? [bcc][faked-from][html-rem]



Running CM 4.1.3es100



How do I tell if both CM's are sync'd?



Scott

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From Scott.Voll at wesd.org  Tue May 22 01:21:44 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 21 May 2007 22:21:44 -0700
Subject: [cisco-voip] How do is see if both pub and sub are sync'd?
	[bcc][faked-from][html-rem]
In-Reply-To: <E76ACA8C13EF1847BC69538B522C3663560FE4@nlimail.nlidomain>
References: <407055A92CECCB499C922A2D35FC19A608DEE7C4@Apollo.wesd.org>
	<E76ACA8C13EF1847BC69538B522C3663560FE4@nlimail.nlidomain>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7C7@Apollo.wesd.org>

THANKS

 

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Monday, May 21, 2007 10:10 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] How do is see if both pub and sub are sync'd?
[bcc][faked-from][html-rem]

 

Using SQL enterprise manager, see if you have any red X signs on the
replication agent.

 

Also create a phone via ccmadmin and verify with the same sql enterprise
manager that it is on the subscriber's copy of the database.

 

Detailed instructions can be found at:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a00801d11a6.shtml#veri

 

 

-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take
Cisco Learning credits!
http://www.ccbootcamp.com
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Voll, Scott
Sent: Mon 5/21/2007 9:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How do is see if both pub and sub are sync'd?
[bcc][faked-from][html-rem]

Running CM 4.1.3es100



How do I tell if both CM's are sync'd?



Scott

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From smarwah at enetsolutions.net  Tue May 22 01:56:40 2007
From: smarwah at enetsolutions.net (Sumit Marwah)
Date: Tue, 22 May 2007 00:56:40 -0500
Subject: [cisco-voip] How do is see if both pub and sub are
	sync'd?[bcc][faked-from][html-rem]
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE7C7@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A608DEE7C4@Apollo.wesd.org><E76ACA8C13EF1847BC69538B522C3663560FE4@nlimail.nlidomain>
	<407055A92CECCB499C922A2D35FC19A608DEE7C7@Apollo.wesd.org>
Message-ID: <FF7F71539DC4E544BA5737C7AA6CEE17368FD9@ems05.ems-1.net>

Scott,

 

You can try using DBhelper tool to check the synchronization of the sql
database and it tries to fix the problem as well.

 

Regards,

Sumit Marwah 
Sr. System Engineer
CCIE Voice # 17400

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, May 22, 2007 12:22 AM
To: Avner Izhar
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] How do is see if both pub and sub are
sync'd?[bcc][faked-from][html-rem]

 

THANKS

 

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Monday, May 21, 2007 10:10 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] How do is see if both pub and sub are sync'd?
[bcc][faked-from][html-rem]

 

Using SQL enterprise manager, see if you have any red X signs on the
replication agent.

 

Also create a phone via ccmadmin and verify with the same sql enterprise
manager that it is on the subscriber's copy of the database.

 

Detailed instructions can be found at:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a00801d11a6.shtml#veri

 

 

-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take
Cisco Learning credits!
http://www.ccbootcamp.com
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Voll, Scott
Sent: Mon 5/21/2007 9:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How do is see if both pub and sub are sync'd?
[bcc][faked-from][html-rem]

Running CM 4.1.3es100



How do I tell if both CM's are sync'd?



Scott

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From tele at plexia.com  Tue May 22 05:34:39 2007
From: tele at plexia.com (tele)
Date: Tue, 22 May 2007 11:34:39 +0200
Subject: [cisco-voip] CME in only SIP scenario
Message-ID: <1179826479.6586.108.camel@mda>

Hi,

I've a particular configuration of a CME 4.1 in a only SIP scenario, CME
register as UA to a SIP proxy.
CME is able to recive a range of number like

01044992XX

and it's able to deliver call to internal phone.
I've interoperability issue with some gateway in our network so i need
to restrict call transfer/forwarding only to CME and not throught the
network.
To be able to achieve this i need the Loopback Call routing function:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00807e9b10.html

But using the loopback call routing i've voice problem, no audio but if
i do a debug voice rtp packet i can see rtp packet going to correct
ports and IP. but i can hear nothing.

Ok attached here there are two configuration, the first one:

cme_with_dialpeer.conf i'm using dial-peer and i've no problem with
voice but i've problem with call transfer/forward cause re-invite for
on-hold goes through the network and i prefer CME handle this.

cme-loopback-dn.cfg i'm using loopback-dn in the ephone-dn, call
signalling is ok but i can't hear voice.

I don't know how to starts troubleshooting this issue, maybe any of you
have an idea.

thank you very much

:tele





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From Ruben.Montes at eu.didata.com  Tue May 22 05:38:01 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Tue, 22 May 2007 11:38:01 +0200
Subject: [cisco-voip] Calling to phones after WAN outage
References: <feb44fca0705171415x49bf9dabvea9837f081cf4192@mail.gmail.com><6A0744639B86DD459433108DBF101D26016AE8FF@MAIL01.sentinel.com>
	<464CCCA7.7010601@saskin.net>
Message-ID: <FA233791F398B848B978464507739E8F86C43A@EUBEBRUSVEX1.eu.didata.local>

Hello,

I have a centralized Hub and Spoke design with a CCM cluster located in the Hub, and IP phones and H323 gateways in the spokes.

To address bandwidth insufficiency, we're going to use AAR. But what happens when the WAN that connects to the sites go down? 

In the remote sites I have SRST with translattion profiles that prepend the digits that are needed to call through the PSTN to reach the central phones.
What should I do in the reverse direction in order to permit the phones in the cenral site to reach the ones that appear as unregistered now using the extensi?n number?

Regards,

Ruben

From jsteinberg at gmail.com  Tue May 22 07:43:53 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 22 May 2007 07:43:53 -0400
Subject: [cisco-voip] Calling to phones after WAN outage
In-Reply-To: <FA233791F398B848B978464507739E8F86C43A@EUBEBRUSVEX1.eu.didata.local>
References: <feb44fca0705171415x49bf9dabvea9837f081cf4192@mail.gmail.com>
	<6A0744639B86DD459433108DBF101D26016AE8FF@MAIL01.sentinel.com>
	<464CCCA7.7010601@saskin.net>
	<FA233791F398B848B978464507739E8F86C43A@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <f1b65f880705220443t11a53573t4d8dda6eba720362@mail.gmail.com>

AAR doesn't inheritly support that scenario where your WAN link is
down - it only reroutes for lack of bandwidth.

You would have to use the 'forward on unregistered' feature that is
available with CCM 4.2.

You would then put in the full E164 number of the phone and if that
phone is unregistered then the call is extended over the PSTN to reach
the phone when it is in SRST mode.  This depends on the SRST site
having its own PSTN circuit / gateway that terminates those E164
numbers.

Justin



On 5/22/07, Ruben Montes (Europe) <Ruben.Montes at eu.didata.com> wrote:
> Hello,
>
> I have a centralized Hub and Spoke design with a CCM cluster located in the Hub, and IP phones and H323 gateways in the spokes.
>
> To address bandwidth insufficiency, we're going to use AAR. But what happens when the WAN that connects to the sites go down?
>
> In the remote sites I have SRST with translattion profiles that prepend the digits that are needed to call through the PSTN to reach the central phones.
> What should I do in the reverse direction in order to permit the phones in the cenral site to reach the ones that appear as unregistered now using the extensi?n number?
>
> Regards,
>
> Ruben
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From Ruben.Montes at eu.didata.com  Tue May 22 07:59:50 2007
From: Ruben.Montes at eu.didata.com (Ruben Montes (Europe))
Date: Tue, 22 May 2007 13:59:50 +0200
Subject: [cisco-voip] Calling to phones after WAN outage
References: <feb44fca0705171415x49bf9dabvea9837f081cf4192@mail.gmail.com>
	<6A0744639B86DD459433108DBF101D26016AE8FF@MAIL01.sentinel.com>
	<464CCCA7.7010601@saskin.net>
	<FA233791F398B848B978464507739E8F86C43A@EUBEBRUSVEX1.eu.didata.local>
	<f1b65f880705220443t11a53573t4d8dda6eba720362@mail.gmail.com>
Message-ID: <FA233791F398B848B978464507739E8F86C519@EUBEBRUSVEX1.eu.didata.local>

Hi,

We're using CCm 5.1... Is there any possibility to do duch a thing in this scenario?

Thanks is advance,

Ruben

-----Mensaje original-----
De: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Enviado el: martes, 22 de mayo de 2007 13:44
Para: Ruben Montes (Europe)
CC: ciscovoip; Jordi Riba (Europe)
Asunto: Re: [cisco-voip] Calling to phones after WAN outage

AAR doesn't inheritly support that scenario where your WAN link is
down - it only reroutes for lack of bandwidth.

You would have to use the 'forward on unregistered' feature that is
available with CCM 4.2.

You would then put in the full E164 number of the phone and if that
phone is unregistered then the call is extended over the PSTN to reach
the phone when it is in SRST mode.  This depends on the SRST site
having its own PSTN circuit / gateway that terminates those E164
numbers.

Justin



On 5/22/07, Ruben Montes (Europe) <Ruben.Montes at eu.didata.com> wrote:
> Hello,
>
> I have a centralized Hub and Spoke design with a CCM cluster located in the Hub, and IP phones and H323 gateways in the spokes.
>
> To address bandwidth insufficiency, we're going to use AAR. But what happens when the WAN that connects to the sites go down?
>
> In the remote sites I have SRST with translattion profiles that prepend the digits that are needed to call through the PSTN to reach the central phones.
> What should I do in the reverse direction in order to permit the phones in the cenral site to reach the ones that appear as unregistered now using the extensi?n number?
>
> Regards,
>
> Ruben
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From rratliff at cisco.com  Tue May 22 08:32:19 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 22 May 2007 08:32:19 -0400
Subject: [cisco-voip] Monitor channel usage mgcp
In-Reply-To: <001101c79c22$6d5226b0$c901a8c0@pdp11>
References: <mailman.14705.1179792893.1489.cisco-voip@puck.nether.net>
	<000301c79c18$d670bfd0$0201a8c0@corporation2>
	<002501c79c19$533e3650$c901a8c0@pdp11>
	<000d01c79c21$d87d1b80$0201a8c0@corporation2>
	<001101c79c22$6d5226b0$c901a8c0@pdp11>
Message-ID: <0B5933CF-3938-458E-BF6F-9D59FA4257B6@cisco.com>

You can also use Perfmon (and thus RTMT).  Go looking at the Cisco  
counters.

-Ryan

On May 21, 2007, at 11:36 PM, Lelio Fulgenzi wrote:

go to serviceability. i believe it's in car.

----- Original Message -----
From: <Cisco.voip at VERIZON.NET>
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>; <cisco-voip at puck.nether.net>
Sent: Monday, May 21, 2007 11:32 PM
Subject: Re: [cisco-voip] Monitor channel usage mgcp


> where are the gw utiliztion reports? I am using CCM 4.2
> ----- Original Message -----
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> To: <Cisco.voip at VERIZON.NET>; <cisco-voip at puck.nether.net>
> Sent: Monday, May 21, 2007 10:31 PM
> Subject: Re: [cisco-voip] Monitor channel usage mgcp
>
>
>> have you tried the gateway utilization reports? they're pretty  
>> useful.
>>
>> you could look at MRTG and take 1-5 minute snapshots for trends.
>>
>>
>> ----- Original Message -----
>> From: <Cisco.voip at VERIZON.NET>
>> To: <cisco-voip at puck.nether.net>
>> Sent: Monday, May 21, 2007 10:28 PM
>> Subject: [cisco-voip] Monitor channel usage mgcp
>>
>>
>>> All, I need an application to monitor usage of channels on  
>>> several pri
>>> mgcp
>>> cisco ios gateways
>>> to provide usage details to mgmt.
>>>
>>> What app's do i need to look at.
>>>
>>> TIA
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From CarlosOrtiz at bayviewfinancial.com  Tue May 22 08:53:27 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 22 May 2007 08:53:27 -0400
Subject: [cisco-voip] RTMT Error Media List Exhausted
Message-ID: <OFECF657D9.77680AB1-ON852572E3.00467344-852572E3.0046D07B@bftg.com>

Last week we started seeing this error message fairly often from the Real 
Time Monitoring Tool.  Apparently one or more of our Media Resource Group 
Lists is reaching its limits but the message does not indicate which one. 
What's the best way to trace down the exact MRGL or item causing this 
message?

Real-Time Monitoring At 15:54:46 on 05/21/2007 on cluster 
MIACALLMGR-Cluster.
Number of MediaListExhausted events exceed 10 within 60 minutes.


Carlos
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From smarwah at enetsolutions.net  Tue May 22 09:06:14 2007
From: smarwah at enetsolutions.net (Sumit Marwah)
Date: Tue, 22 May 2007 08:06:14 -0500
Subject: [cisco-voip] RTMT Error Media List Exhausted
Message-ID: <FF7F71539DC4E544BA5737C7AA6CEE1740BFD1@ems05.ems-1.net>

Carlos,

You can check the Event Viewer of the Call Manager the media resources are registered to. There should be an error message in the application log around the same time the RTMT alert was generated that should tell you which media resource unregistered.

Regards,
Sumit

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net <cisco-voip-bounces at puck.nether.net>
To: cisco-voip at puck.nether.net <cisco-voip at puck.nether.net>
Sent: Tue May 22 07:53:27 2007
Subject: [cisco-voip] RTMT Error Media List Exhausted


Last week we started seeing this error message fairly often from the Real Time Monitoring Tool.  Apparently one or more of our Media Resource Group Lists is reaching its limits but the message does not indicate which one.  What's the best way to trace down the exact MRGL or item causing this message? 

Real-Time Monitoring At 15:54:46 on 05/21/2007 on cluster MIACALLMGR-Cluster.
Number of MediaListExhausted events exceed 10 within 60 minutes. 


Carlos

From d4rkf1ber at gmail.com  Tue May 22 09:37:33 2007
From: d4rkf1ber at gmail.com (Dark Fiber)
Date: Tue, 22 May 2007 08:37:33 -0500
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
In-Reply-To: <c6f32ce40705212010x2e75e902h83d1b4bdfa4a1673@mail.gmail.com>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com>
	<c6f32ce40705212010x2e75e902h83d1b4bdfa4a1673@mail.gmail.com>
Message-ID: <a1b07ea60705220637j6cc26b09h1f30e1e16d2b7e34@mail.gmail.com>

The vocera product is what I saw.  Disappointed to read that it
requires another server.

The 7921G is interesting but still leaves me in the same boat.  Couple
of users who are away from their desk alot need a wireless phone, but
complain about having to carry the 7920.  In other words having to
hold it in their hand or put it down and forget it.  I told them about
the carrying case but they are both women and don't always have a belt
or something to clip it on to.

The vocera product would be the perfect size and fit, just not so sure
about the extra cost and headache of needing a server for a few
handsets.

Scott you mentioned Nokia care to elaborate?  Does Nokia have a SCCP wifi phone?



On 5/21/07, Adam <cisco at adman.net> wrote:
> I think the devices you are referring to may be the Vocera
> communicators (http://www.vocera.com/). They not exactly straight SCCP
> phones as they require the Vocera server to function.
>
> -Adam
>
> On 5/21/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
> > Anyone aware of any wireless SCCP phone devices as an alternative to
> > say a 7920?  One time I saw some lapel type phones demonstrated at a
> > Cisco event but have never seen anything official from Cisco regarding
> > them or similiar items.
> >
> > Just curious, thanks in advanced.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From rleetun at co.boulder.co.us  Tue May 22 09:51:01 2007
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Tue, 22 May 2007 07:51:01 -0600
Subject: [cisco-voip] Sip trunks and multicast
In-Reply-To: <001e01c79a23$84d0fc90$c901a8c0@pdp11>
Message-ID: <963CE94787005149A52C17F33A028E1207E61FDA@EVS1B.boco.co.boulder.co.us>

Good morning,
 
What needs to be configured on a SIP trunk to allow for multicast to
work?  We can make connection to the server, but we do not get one way
or better yet two way audio.
 
Thanks for the assistance.
 
Rob
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From gsobhi at nileonebrokers.com  Tue May 22 09:49:11 2007
From: gsobhi at nileonebrokers.com (George Sobhi)
Date: Tue, 22 May 2007 16:49:11 +0300
Subject: [cisco-voip] MOH file
Message-ID: <NILESERVER16bdVJWrG000007db@nileserver1.niledomain.com>

Dear all,

I have a wma file how can I convert it and tftp it to 2811 cisco router with
12.4(4)XC / CME 4.0(1)

 

thanks

 

George Sobhi

Technical Support Engineer

NILEone BROKERS Co.

4, Syria St, Mohandeseen, Giza-Egypt.

Phone: 202-3363252

Ext.:      121

Fax:      202-3363260

 

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From Scott.Voll at wesd.org  Tue May 22 10:50:00 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 22 May 2007 07:50:00 -0700
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
In-Reply-To: <a1b07ea60705220637j6cc26b09h1f30e1e16d2b7e34@mail.gmail.com>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com><c6f32ce40705212010x2e75e902h83d1b4bdfa4a1673@mail.gmail.com>
	<a1b07ea60705220637j6cc26b09h1f30e1e16d2b7e34@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7CC@Apollo.wesd.org>

http://polydistortion.net/monkey/archives/2006/11/16/003924.html

after doing some surfing it looks like CME.  But as it is SIP I can't
image it not working with cm 5.x.  

I only know anything about this as it was on the list about a year ago
in the UK.

So If you want to ask any more questions......... ask the guy who made
the web page.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dark Fiber
Sent: Tuesday, May 22, 2007 6:38 AM
To: Adam
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

The vocera product is what I saw.  Disappointed to read that it
requires another server.

The 7921G is interesting but still leaves me in the same boat.  Couple
of users who are away from their desk alot need a wireless phone, but
complain about having to carry the 7920.  In other words having to
hold it in their hand or put it down and forget it.  I told them about
the carrying case but they are both women and don't always have a belt
or something to clip it on to.

The vocera product would be the perfect size and fit, just not so sure
about the extra cost and headache of needing a server for a few
handsets.

Scott you mentioned Nokia care to elaborate?  Does Nokia have a SCCP
wifi phone?



On 5/21/07, Adam <cisco at adman.net> wrote:
> I think the devices you are referring to may be the Vocera
> communicators (http://www.vocera.com/). They not exactly straight SCCP
> phones as they require the Vocera server to function.
>
> -Adam
>
> On 5/21/07, Dark Fiber <d4rkf1ber at gmail.com> wrote:
> > Anyone aware of any wireless SCCP phone devices as an alternative to
> > say a 7920?  One time I saw some lapel type phones demonstrated at a
> > Cisco event but have never seen anything official from Cisco
regarding
> > them or similiar items.
> >
> > Just curious, thanks in advanced.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From CarlosOrtiz at bayviewfinancial.com  Tue May 22 10:53:39 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 22 May 2007 10:53:39 -0400
Subject: [cisco-voip] MOH file
In-Reply-To: <NILESERVER16bdVJWrG000007db@nileserver1.niledomain.com>
Message-ID: <OF76FE44DD.8A18923C-ON852572E3.00518550-852572E3.0051D113@bftg.com>

Not sure if Call Mgr can convert wma, but it will definitely convert .wav 
and mp3 formats of the same file.  Then you can just upload the g711 file 
to the flash of the 2811 like you would an IOS file.

Carlos



"George Sobhi" <gsobhi at nileonebrokers.com> 
Sent by: cisco-voip-bounces at puck.nether.net
05/22/2007 09:49 AM

To
<cisco-voip at puck.nether.net>
cc

Subject
[cisco-voip] MOH file






Dear all,
I have a wma file how can I convert it and tftp it to 2811 cisco router 
with 12.4(4)XC / CME 4.0(1)
 
thanks
 
George Sobhi
Technical Support Engineer
NILEone BROKERS Co.
4, Syria St, Mohandeseen, Giza-Egypt.
Phone: 202-3363252
Ext.:      121
Fax:      202-3363260
 _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From CarlosOrtiz at bayviewfinancial.com  Tue May 22 11:02:00 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 22 May 2007 11:02:00 -0400
Subject: [cisco-voip] RTMT Error Media List Exhausted
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE7CB@Apollo.wesd.org>
Message-ID: <OF6D1E87DE.31D5B23A-ON852572E3.00522E49-852572E3.00529492@bftg.com>

Well in our case it looks like the Music on Hold resources are being 
exhausted.  I thought I checked the app log already but upon closer 
examiniation I did find that in the logs.   Are you getting the same 
message from RTMT? If so, then you probably can find a corresponding 
message in the App log although its still pretty general.    It does not 
specify which device was trying to access a specific resource or anything 
like that.

Carlos





"Voll, Scott" <Scott.Voll at wesd.org> 
05/22/2007 10:42 AM

To
<CarlosOrtiz at bayviewfinancial.com>
cc

Subject
RE: [cisco-voip] RTMT Error Media List Exhausted






Let me know what you find.  I always get the messages to and just haven?t 
worried about it.
 
Scott
 

From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: Tuesday, May 22, 2007 5:53 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] RTMT Error Media List Exhausted
 

Last week we started seeing this error message fairly often from the Real 
Time Monitoring Tool.  Apparently one or more of our Media Resource Group 
Lists is reaching its limits but the message does not indicate which one. 
What's the best way to trace down the exact MRGL or item causing this 
message? 

Real-Time Monitoring At 15:54:46 on 05/21/2007 on cluster 
MIACALLMGR-Cluster.
Number of MediaListExhausted events exceed 10 within 60 minutes. 


Carlos

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From asobihoudai at yahoo.com  Tue May 22 11:11:24 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 22 May 2007 08:11:24 -0700 (PDT)
Subject: [cisco-voip] Sip trunks and multicast
In-Reply-To: <963CE94787005149A52C17F33A028E1207E61FDA@EVS1B.boco.co.boulder.co.us>
Message-ID: <93227.53149.qm@web57214.mail.re3.yahoo.com>

If you're not even having one-way audio, then I
suspect there might be some routing issues.

Also, what is this? CCME? CCM? What version? That
information is helpful for anybody to help you.

Paul

--- "Leetun, Rob" <rleetun at co.boulder.co.us> wrote:

> Good morning,
>  
> What needs to be configured on a SIP trunk to allow
> for multicast to
> work?  We can make connection to the server, but we
> do not get one way
> or better yet two way audio.
>  
> Thanks for the assistance.
>  
> Rob
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.
http://answers.yahoo.com/dir/?link=list&sid=396545469

From asobihoudai at yahoo.com  Tue May 22 11:12:27 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Tue, 22 May 2007 08:12:27 -0700 (PDT)
Subject: [cisco-voip] MOH file
In-Reply-To: <OF76FE44DD.8A18923C-ON852572E3.00518550-852572E3.0051D113@bftg.com>
Message-ID: <22534.50219.qm@web57203.mail.re3.yahoo.com>

I'd save that .wma file as a .wav and then convert.

Paul
--- CarlosOrtiz at bayviewfinancial.com wrote:

> Not sure if Call Mgr can convert wma, but it will
> definitely convert .wav 
> and mp3 formats of the same file.  Then you can just
> upload the g711 file 
> to the flash of the 2811 like you would an IOS file.
> 
> Carlos
> 
> 
> 
> "George Sobhi" <gsobhi at nileonebrokers.com> 
> Sent by: cisco-voip-bounces at puck.nether.net
> 05/22/2007 09:49 AM
> 
> To
> <cisco-voip at puck.nether.net>
> cc
> 
> Subject
> [cisco-voip] MOH file
> 
> 
> 
> 
> 
> 
> Dear all,
> I have a wma file how can I convert it and tftp it
> to 2811 cisco router 
> with 12.4(4)XC / CME 4.0(1)
>  
> thanks
>  
> George Sobhi
> Technical Support Engineer
> NILEone BROKERS Co.
> 4, Syria St, Mohandeseen, Giza-Egypt.
> Phone: 202-3363252
> Ext.:      121
> Fax:      202-3363260



       
____________________________________________________________________________________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  

From WOCAMPO at broward.org  Tue May 22 13:28:12 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Tue, 22 May 2007 13:28:12 -0400
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com><411933.7897.qm@web57210.mail.re3.yahoo.com>
	<407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty>

Nokia has developed the Wireless phone with Dual mode.
One mode that  can connect to CCM using only the WLAN radio and the intellisync SCCP client and the other mode for the Wireless telco. 
Regards,

Walter Ocampo
System Network Analyst
Office of Information Technology
540 S.E 3rd. Ave, Suite 300
Fort Lauderdale, FL 33301
(954) 357 8041?Cell (954) 632 8228
wocampo at broward.org
?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, May 21, 2007 11:01 PM
To: Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 8:00 PM
To: Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as
> an alternative to
> say a 7920?  One time I saw some lapel type phones
> demonstrated at a
> Cisco event but have never seen anything official
> from Cisco regarding
> them or similiar items.
> 
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



 
________________________________________________________________________
____________
8:00? 8:25? 8:40? Find a flick in no time 
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From WOCAMPO at broward.org  Tue May 22 13:40:48 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Tue, 22 May 2007 13:40:48 -0400
Subject: [cisco-voip] RTMT Error Media List Exhausted
In-Reply-To: <OF6D1E87DE.31D5B23A-ON852572E3.00522E49-852572E3.00529492@bftg.com>
References: <407055A92CECCB499C922A2D35FC19A608DEE7CB@Apollo.wesd.org>
	<OF6D1E87DE.31D5B23A-ON852572E3.00522E49-852572E3.00529492@bftg.com>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA330@UVEXVS01.bc.broward.cty>

I am getting the same messages. Since is in safe range, I went to Cisco Documentation and they suggest to change the settings. In case you want to change the setting do the following:

Highlight  the alert

Go to alert TAB

Alert Properties and Change  the threshold setting from 0 to 1. 

Clear the alert.  

 

 

 

 

 

Walter Ocampo

System Network Analyst

Office of Information Technology

540 S.E 3rd. Ave, Suite 300

Fort Lauderdale, FL 33301

(954) 357 8041 Cell (954) 632 8228

wocampo at broward.org <mailto:wocampo at broward.org> 

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of CarlosOrtiz at bayviewfinancial.com
Sent: Tuesday, May 22, 2007 11:02 AM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] RTMT Error Media List Exhausted

 


Well in our case it looks like the Music on Hold resources are being exhausted.  I thought I checked the app log already but upon closer examiniation I did find that in the logs.   Are you getting the same message from RTMT? If so, then you probably can find a corresponding message in the App log although its still pretty general.    It does not specify which device was trying to access a specific resource or anything like that. 

Carlos 





"Voll, Scott" <Scott.Voll at wesd.org> 

05/22/2007 10:42 AM 

To

<CarlosOrtiz at bayviewfinancial.com> 

cc

	
Subject

RE: [cisco-voip] RTMT Error Media List Exhausted

 

		




Let me know what you find.  I always get the messages to and just haven?t worried about it. 
  
Scott 
  

 

________________________________


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of CarlosOrtiz at bayviewfinancial.com
Sent: Tuesday, May 22, 2007 5:53 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] RTMT Error Media List Exhausted 
  

Last week we started seeing this error message fairly often from the Real Time Monitoring Tool.  Apparently one or more of our Media Resource Group Lists is reaching its limits but the message does not indicate which one.  What's the best way to trace down the exact MRGL or item causing this message? 

Real-Time Monitoring At 15:54:46 on 05/21/2007 on cluster MIACALLMGR-Cluster.
Number of MediaListExhausted events exceed 10 within 60 minutes. 


Carlos 

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From tim.reimers at asheville.k12.nc.us  Tue May 22 14:05:18 2007
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue, 22 May 2007 14:05:18 -0400
Subject: [cisco-voip] routing calls based on Caller ID?
Message-ID: <050944EE48FF384F9C663F7048C47DA702E1A950@webmail.asheville.k12.nc.us>

Hi everyone-

 

Is it possible to route calls based on the Caller ID of the incoming
call??

 

CCM 4.13

H.323 ISDN PRI gateway

 

 

 

Thanks, Tim

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From jason.aarons at us.didata.com  Tue May 22 14:08:44 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 22 May 2007 14:08:44 -0400
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
In-Reply-To: <F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com><411933.7897.qm@web57210.mail.re3.yahoo.com><407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org>
	<F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005A35AF8@USNAEXCH.na.didata.local>

I wasn't impressed using the Nokia model 60 back to CallManager. For example there is not a WiFi signal indicator on phone menu, you have to dig into menus to find stuff. It wasn't a good PDA overall. I would rate the Palm OS in the middle, Nokia at bottom and Windows Mobile 6 Pro on HTC/6700 at the top.

I'd stick with a Windows Mobile device with WiFi and use the IP Blue product line.

http://www.ipblue.com/download.asp?product=vtgo_ppc

Everyone has their own opinion...


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ocampo, Walter
Sent: Tuesday, May 22, 2007 1:28 PM
To: Voll, Scott; Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia has developed the Wireless phone with Dual mode.
One mode that  can connect to CCM using only the WLAN radio and the intellisync SCCP client and the other mode for the Wireless telco. 
Regards,

Walter Ocampo
System Network Analyst
Office of Information Technology
540 S.E 3rd. Ave, Suite 300
Fort Lauderdale, FL 33301
(954) 357 8041?Cell (954) 632 8228
wocampo at broward.org
?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, May 21, 2007 11:01 PM
To: Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 8:00 PM
To: Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as
> an alternative to
> say a 7920?  One time I saw some lapel type phones
> demonstrated at a
> Cisco event but have never seen anything official
> from Cisco regarding
> them or similiar items.
> 
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



 
________________________________________________________________________
____________
8:00? 8:25? 8:40? Find a flick in no time 
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From rratliff at cisco.com  Tue May 22 14:12:49 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 22 May 2007 14:12:49 -0400
Subject: [cisco-voip] routing calls based on Caller ID?
In-Reply-To: <050944EE48FF384F9C663F7048C47DA702E1A950@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA702E1A950@webmail.asheville.k12.nc.us>
Message-ID: <8ED458AE-D730-44B2-91E9-A00AC615CADF@cisco.com>

Have to use TCL on the router or something like IPCC in CM.  CM  
itself cannot based on ANI.

-Ryan

On May 22, 2007, at 2:05 PM, Tim Reimers wrote:

Hi everyone-



Is it possible to route calls based on the Caller ID of the incoming  
call??



CCM 4.13

H.323 ISDN PRI gateway







Thanks, Tim

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From rratliff at cisco.com  Tue May 22 14:13:36 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 22 May 2007 14:13:36 -0400
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005A35AF8@USNAEXCH.na.didata.local>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com><411933.7897.qm@web57210.mail.re3.yahoo.com><407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org>
	<F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty>
	<C1FE15183DA37645BC0633BC604E44F005A35AF8@USNAEXCH.na.didata.local>
Message-ID: <3241DBC0-9726-45EC-8F48-4B7F0DE73968@cisco.com>

For what it's worth the E61 is the replacement for the E60.  I  
haven't seen either so I couldn't even tell you what it looks like.

-Ryan

On May 22, 2007, at 2:08 PM, Jason Aarons ((US)) wrote:

I wasn't impressed using the Nokia model 60 back to CallManager. For  
example there is not a WiFi signal indicator on phone menu, you have  
to dig into menus to find stuff. It wasn't a good PDA overall. I  
would rate the Palm OS in the middle, Nokia at bottom and Windows  
Mobile 6 Pro on HTC/6700 at the top.

I'd stick with a Windows Mobile device with WiFi and use the IP Blue  
product line.

http://www.ipblue.com/download.asp?product=vtgo_ppc

Everyone has their own opinion...


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Ocampo, Walter
Sent: Tuesday, May 22, 2007 1:28 PM
To: Voll, Scott; Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia has developed the Wireless phone with Dual mode.
One mode that  can connect to CCM using only the WLAN radio and the  
intellisync SCCP client and the other mode for the Wireless telco.
Regards,

Walter Ocampo
System Network Analyst
Office of Information Technology
540 S.E 3rd. Ave, Suite 300
Fort Lauderdale, FL 33301
(954) 357 8041 Cell (954) 632 8228
wocampo at broward.org


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, May 21, 2007 11:01 PM
To: Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 8:00 PM
To: Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as
> an alternative to
> say a 7920?  One time I saw some lapel type phones
> demonstrated at a
> Cisco event but have never seen anything official
> from Cisco regarding
> them or similiar items.
>
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>




________________________________________________________________________
____________
8:00? 8:25? 8:40? Find a flick in no time
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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Disclaimer:

This e-mail communication and any attachments may contain
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From jason.aarons at us.didata.com  Tue May 22 14:27:09 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 22 May 2007 14:27:09 -0400
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
In-Reply-To: <3241DBC0-9726-45EC-8F48-4B7F0DE73968@cisco.com>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com><411933.7897.qm@web57210.mail.re3.yahoo.com><407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org>
	<F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty>
	<C1FE15183DA37645BC0633BC604E44F005A35AF8@USNAEXCH.na.didata.local>
	<3241DBC0-9726-45EC-8F48-4B7F0DE73968@cisco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005A35B74@USNAEXCH.na.didata.local>

I wouldn't call it a replacement, it has the larger screen and keyboard,
but the OS is the same. I never think the N60 will reach the success of
say a iPhone or Windows Mobile or Palm and will fade away...good thing
Sony exited that market!

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, May 22, 2007 2:14 PM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

For what it's worth the E61 is the replacement for the E60.  I  
haven't seen either so I couldn't even tell you what it looks like.

-Ryan

On May 22, 2007, at 2:08 PM, Jason Aarons ((US)) wrote:

I wasn't impressed using the Nokia model 60 back to CallManager. For  
example there is not a WiFi signal indicator on phone menu, you have  
to dig into menus to find stuff. It wasn't a good PDA overall. I  
would rate the Palm OS in the middle, Nokia at bottom and Windows  
Mobile 6 Pro on HTC/6700 at the top.

I'd stick with a Windows Mobile device with WiFi and use the IP Blue  
product line.

http://www.ipblue.com/download.asp?product=vtgo_ppc

Everyone has their own opinion...


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Ocampo, Walter
Sent: Tuesday, May 22, 2007 1:28 PM
To: Voll, Scott; Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia has developed the Wireless phone with Dual mode.
One mode that  can connect to CCM using only the WLAN radio and the  
intellisync SCCP client and the other mode for the Wireless telco.
Regards,

Walter Ocampo
System Network Analyst
Office of Information Technology
540 S.E 3rd. Ave, Suite 300
Fort Lauderdale, FL 33301
(954) 357 8041 Cell (954) 632 8228
wocampo at broward.org


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, May 21, 2007 11:01 PM
To: Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 8:00 PM
To: Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as
> an alternative to
> say a 7920?  One time I saw some lapel type phones
> demonstrated at a
> Cisco event but have never seen anything official
> from Cisco regarding
> them or similiar items.
>
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>




________________________________________________________________________
____________
8:00? 8:25? 8:40? Find a flick in no time
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
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This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

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Disclaimer:

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From jason.aarons at us.didata.com  Tue May 22 14:31:22 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 22 May 2007 14:31:22 -0400
Subject: [cisco-voip] CFwdALL and get That Key is Not Active (7961 > 4.1(3)
	with load 8.2(2)SR1S)
Message-ID: <C1FE15183DA37645BC0633BC604E44F005A35B95@USNAEXCH.na.didata.local>

I press Line 5 on a 7961 and CFwdALL and get That Key is Not Active
here.

 

Line 5 has a good Calling Search Space, maybe there is a enterprise
parameter that was added in 4.x? 

 

This used to work in 3.x (forward on per line). CFwdALL works for line 1
but not lines 2-6 on the 7961....

 

 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
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From smarwah at enetsolutions.net  Tue May 22 14:34:49 2007
From: smarwah at enetsolutions.net (Sumit Marwah)
Date: Tue, 22 May 2007 13:34:49 -0500
Subject: [cisco-voip] CFwdALL and get That Key is Not Active (7961 >
	4.1(3)with load 8.2(2)SR1S)
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005A35B95@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F005A35B95@USNAEXCH.na.didata.local>
Message-ID: <FF7F71539DC4E544BA5737C7AA6CEE1736903C@ems05.ems-1.net>

This can only be done from the ccmuser page. By default the CFwdALL
button is only available for the primary line, i.e. line 1. 

 

Maybe there is a system parameter that can be changed, not sure on that
one.

 

Regards,

 

Sumit Marwah 
Sr. System Engineer
CCIE Voice # 17400

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Tuesday, May 22, 2007 1:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CFwdALL and get That Key is Not Active (7961 >
4.1(3)with load 8.2(2)SR1S)

 

I press Line 5 on a 7961 and CFwdALL and get That Key is Not Active
here.

 

Line 5 has a good Calling Search Space, maybe there is a enterprise
parameter that was added in 4.x? 

 

This used to work in 3.x (forward on per line). CFwdALL works for line 1
but not lines 2-6 on the 7961....

 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
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From tettokalle at hotmail.com  Tue May 22 14:54:26 2007
From: tettokalle at hotmail.com (Kalle Tetto)
Date: Tue, 22 May 2007 20:54:26 +0200
Subject: [cisco-voip] H323 Gateway and 0X3A Bearer Capability Not
	Available
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6D4@CH-EXCH-02.intelltech.apptis.inc>
References: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6D4@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <f2ve96$e2u$1@sea.gmane.org>

try to use another codec on the pots dial-peer, g711ulaw or g711alaw.

cheers

Brian Henry schrieb:
> I have a CCM 4.3(1) with 2821 VGW 12.4(11)T1 running in H323 mode.
>  
> Testing:
>  
> Using LAN Line
> I can call into VGW and receive ring tone on call setup/proceed.
>  
> Using 7961 8.2(2)sr1
> I can call out but certain numbers I call do not connect and get, and 
> also do not get ring indicator
> call_progress: DEV_CONNECTED VOICE ERROR: Bearer capability not 
> available(0x3A): b channel 2 call id 0x50D
>  
> IP Communicator 2.1(1)
> Can call anywhere and receive ring indicator as well as get connected to 
> calls that I cannot through regular hard phones.
>  
> I have added the progress_ind setup/connect/progress commands under the 
> pots dial-peers to correct the ring back issue but to no avail.
>  
> I do not have Display IE checked and I am not running fast-start for 
> inbound nor outbound.
>  
> DSP's are in flex mode and the firmware is 9.2.2 I do believe.
>  
>  
> Very weird stuff when I can call everywhere with IPC but not a hard 
> phone and the funny thing is that sometimes I can call it.
>  
> Brian
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From Brian.Henry at apptis.com  Tue May 22 15:11:30 2007
From: Brian.Henry at apptis.com (Brian Henry)
Date: Tue, 22 May 2007 15:11:30 -0400
Subject: [cisco-voip] H323 Gateway and 0X3A Bearer Capability
	NotAvailable
References: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6D4@CH-EXCH-02.intelltech.apptis.inc>
	<f2ve96$e2u$1@sea.gmane.org>
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6D8@CH-EXCH-02.intelltech.apptis.inc>

turns out that after I left the customer had been playing with VT Advantage and thus after getting this information made sure that under each of the voice-ports(PRI) that the statement "bearer-cap speech" was set.
 
Thanks for the input though
 
B

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Kalle Tetto
Sent: Tue 5/22/2007 2:54 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H323 Gateway and 0X3A Bearer Capability NotAvailable



try to use another codec on the pots dial-peer, g711ulaw or g711alaw.

cheers

Brian Henry schrieb:
> I have a CCM 4.3(1) with 2821 VGW 12.4(11)T1 running in H323 mode.
> 
> Testing:
> 
> Using LAN Line
> I can call into VGW and receive ring tone on call setup/proceed.
> 
> Using 7961 8.2(2)sr1
> I can call out but certain numbers I call do not connect and get, and
> also do not get ring indicator
> call_progress: DEV_CONNECTED VOICE ERROR: Bearer capability not
> available(0x3A): b channel 2 call id 0x50D
> 
> IP Communicator 2.1(1)
> Can call anywhere and receive ring indicator as well as get connected to
> calls that I cannot through regular hard phones.
> 
> I have added the progress_ind setup/connect/progress commands under the
> pots dial-peers to correct the ring back issue but to no avail.
> 
> I do not have Display IE checked and I am not running fast-start for
> inbound nor outbound.
> 
> DSP's are in flex mode and the firmware is 9.2.2 I do believe.
> 
> 
> Very weird stuff when I can call everywhere with IPC but not a hard
> phone and the funny thing is that sometimes I can call it.
> 
> Brian
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip


-------------- next part --------------
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From rleetun at co.boulder.co.us  Tue May 22 15:25:51 2007
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Tue, 22 May 2007 13:25:51 -0600
Subject: [cisco-voip] Spam: Spam: Re: H323 Gateway and 0X3A Bearer
	Capability NotAvailable
In-Reply-To: <f2ve96$e2u$1@sea.gmane.org>
Message-ID: <963CE94787005149A52C17F33A028E1207E61FE2@EVS1B.boco.co.boulder.co.us>

Try setting up your voice port as this: 

 voice-port....
 no vad
 bearer-cap Speech

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kalle Tetto
Sent: Tuesday, May 22, 2007 12:54 PM
To: cisco-voip at puck.nether.net
Subject: Spam: Spam: Re: [cisco-voip] H323 Gateway and 0X3A Bearer
Capability NotAvailable

try to use another codec on the pots dial-peer, g711ulaw or g711alaw.

cheers

Brian Henry schrieb:
> I have a CCM 4.3(1) with 2821 VGW 12.4(11)T1 running in H323 mode.
>  
> Testing:
>  
> Using LAN Line
> I can call into VGW and receive ring tone on call setup/proceed.
>  
> Using 7961 8.2(2)sr1
> I can call out but certain numbers I call do not connect and get, and 
> also do not get ring indicator
> call_progress: DEV_CONNECTED VOICE ERROR: Bearer capability not
> available(0x3A): b channel 2 call id 0x50D
>  
> IP Communicator 2.1(1)
> Can call anywhere and receive ring indicator as well as get connected 
> to calls that I cannot through regular hard phones.
>  
> I have added the progress_ind setup/connect/progress commands under 
> the pots dial-peers to correct the ring back issue but to no avail.
>  
> I do not have Display IE checked and I am not running fast-start for 
> inbound nor outbound.
>  
> DSP's are in flex mode and the firmware is 9.2.2 I do believe.
>  
>  
> Very weird stuff when I can call everywhere with IPC but not a hard 
> phone and the funny thing is that sometimes I can call it.
>  
> Brian
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From rratliff at cisco.com  Tue May 22 16:18:51 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 22 May 2007 16:18:51 -0400
Subject: [cisco-voip] New Cisco Phones
In-Reply-To: <BAY126-DAV7FFE13722D63AF06EAE03BD360@phx.gbl>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com>
	<411933.7897.qm@web57210.mail.re3.yahoo.com>
	<407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org>
	<F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty>
	<C1FE15183DA37645BC0633BC604E44F005A35AF8@USNAEXCH.na.didata.local>
	<3241DBC0-9726-45EC-8F48-4B7F0DE73968@cisco.com>
	<BAY126-DAV7FFE13722D63AF06EAE03BD360@phx.gbl>
Message-ID: <CAFEF0F3-0D5F-4762-99F0-EDE8B6E4CF66@cisco.com>

7931 - http://www.cisco.com/en/US/products/hw/phones/ps379/ 
products_data_sheet0900aecd80534259.html

-Ryan

On May 22, 2007, at 4:12 PM, Dennis Hartmann wrote:


	Does anyone have any information on these new Cisco phones:

7931
7937
7942
7945
7962
7965
7975
3951
3952

    I found a Q & A document on the SIP 3951:
http://www.cisco.com/web/CN/products/products_netsol/voices/products/ 
3951/pd
f/unified_sip_qa.pdf

Sincerely,

Dennis J. Hartmann
White Pine Communications
CCIE# 15651
CCVP/CCSI/CCIP/CCNP/MCSE
dh8 at pobox.com

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Tuesday, May 22, 2007 2:14 PM
To: Jason Aarons ((US))
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

For what it's worth the E61 is the replacement for the E60.  I  
haven't seen
either so I couldn't even tell you what it looks like.

-Ryan

On May 22, 2007, at 2:08 PM, Jason Aarons ((US)) wrote:

I wasn't impressed using the Nokia model 60 back to CallManager. For  
example
there is not a WiFi signal indicator on phone menu, you have to dig into
menus to find stuff. It wasn't a good PDA overall. I would rate the  
Palm OS
in the middle, Nokia at bottom and Windows Mobile 6 Pro on HTC/6700  
at the
top.

I'd stick with a Windows Mobile device with WiFi and use the IP Blue  
product
line.

http://www.ipblue.com/download.asp?product=vtgo_ppc

Everyone has their own opinion...


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Ocampo, Walter
Sent: Tuesday, May 22, 2007 1:28 PM
To: Voll, Scott; Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia has developed the Wireless phone with Dual mode.
One mode that  can connect to CCM using only the WLAN radio and the
intellisync SCCP client and the other mode for the Wireless telco.
Regards,

Walter Ocampo
System Network Analyst
Office of Information Technology
540 S.E 3rd. Ave, Suite 300
Fort Lauderdale, FL 33301
(954) 357 8041 Cell (954) 632 8228
wocampo at broward.org


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, May 21, 2007 11:01 PM
To: Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 8:00 PM
To: Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as an alternative to
> say a 7920?  One time I saw some lapel type phones demonstrated at a
> Cisco event but have never seen anything official from Cisco regarding
> them or similiar items.
>
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>




________________________________________________________________________
____________
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http://tools.search.yahoo.com/shortcuts/#news
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain  
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above only.  If you are not the intended addressee, you are hereby  
notified
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From mlorad1 at towson.edu  Tue May 22 19:47:47 2007
From: mlorad1 at towson.edu (Loraditch, Matthew*)
Date: Tue, 22 May 2007 19:47:47 -0400
Subject: [cisco-voip] New Cisco Phones
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com><411933.7897.qm@web57210.mail.re3.yahoo.com><407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org><F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty><C1FE15183DA37645BC0633BC604E44F005A35AF8@USNAEXCH.na.didata.local><3241DBC0-9726-45EC-8F48-4B7F0DE73968@cisco.com><BAY126-DAV7FFE13722D63AF06EAE03BD360@phx.gbl>
	<CAFEF0F3-0D5F-4762-99F0-EDE8B6E4CF66@cisco.com>
Message-ID: <A57E8EB3EA7B264FBDF12D930FED574F801751@argon.towson.edu>

Where did this list come from?  I know the 7931 has been out for a little while now, maybe a few months, and I recall reading the info about the 3951, but I haven't heard hide nor hair about the others.........
 
Matthew Loraditch 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ryan Ratliff
Sent: Tue 5/22/2007 4:18 PM
To: dh8 at pobox.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] New Cisco Phones



7931 - http://www.cisco.com/en/US/products/hw/phones/ps379/
products_data_sheet0900aecd80534259.html

-Ryan

On May 22, 2007, at 4:12 PM, Dennis Hartmann wrote:


        Does anyone have any information on these new Cisco phones:

7931
7937
7942
7945
7962
7965
7975
3951
3952

    I found a Q & A document on the SIP 3951:
http://www.cisco.com/web/CN/products/products_netsol/voices/products/
3951/pd
f/unified_sip_qa.pdf

Sincerely,

Dennis J. Hartmann
White Pine Communications
CCIE# 15651
CCVP/CCSI/CCIP/CCNP/MCSE
dh8 at pobox.com

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Tuesday, May 22, 2007 2:14 PM
To: Jason Aarons ((US))
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

For what it's worth the E61 is the replacement for the E60.  I 
haven't seen
either so I couldn't even tell you what it looks like.

-Ryan

On May 22, 2007, at 2:08 PM, Jason Aarons ((US)) wrote:

I wasn't impressed using the Nokia model 60 back to CallManager. For 
example
there is not a WiFi signal indicator on phone menu, you have to dig into
menus to find stuff. It wasn't a good PDA overall. I would rate the 
Palm OS
in the middle, Nokia at bottom and Windows Mobile 6 Pro on HTC/6700 
at the
top.

I'd stick with a Windows Mobile device with WiFi and use the IP Blue 
product
line.

http://www.ipblue.com/download.asp?product=vtgo_ppc

Everyone has their own opinion...


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Ocampo, Walter
Sent: Tuesday, May 22, 2007 1:28 PM
To: Voll, Scott; Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia has developed the Wireless phone with Dual mode.
One mode that  can connect to CCM using only the WLAN radio and the
intellisync SCCP client and the other mode for the Wireless telco.
Regards,

Walter Ocampo
System Network Analyst
Office of Information Technology
540 S.E 3rd. Ave, Suite 300
Fort Lauderdale, FL 33301
(954) 357 8041 Cell (954) 632 8228
wocampo at broward.org


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, May 21, 2007 11:01 PM
To: Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 8:00 PM
To: Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as an alternative to
> say a 7920?  One time I saw some lapel type phones demonstrated at a
> Cisco event but have never seen anything official from Cisco regarding
> them or similiar items.
>
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>




________________________________________________________________________
____________
8:00? 8:25? 8:40? Find a flick in no time with the Yahoo! Search movie
showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain 
confidential and
privileged information and is for use by the designated addressee(s) 
named
above only.  If you are not the intended addressee, you are hereby 
notified
that you have received this communication in error and that any use or
reproduction of this email or its contents is strictly prohibited and 
may be
unlawful.  If you have received this communication in error, please 
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From SCASPER at mtb.com  Tue May 22 20:40:48 2007
From: SCASPER at mtb.com (STEVEN CASPER)
Date: Tue, 22 May 2007 20:40:48 -0400
Subject: [cisco-voip] New Cisco Phones
Message-ID: <s6535557.094@comrcm01intgwia.mandtbank.com>

Looks like the 7931 is now supported on CCM. Is it supported on 4.1.3? 
If so, any info on the required Dev pack?

Oh,  by the way love the retro paper label!

Steve

>>> "Ryan Ratliff" <rratliff at cisco.com> 05/22/07 4:18 PM >>>
7931 - http://www.cisco.com/en/US/products/hw/phones/ps379/ 
products_data_sheet0900aecd80534259.html

-Ryan

On May 22, 2007, at 4:12 PM, Dennis Hartmann wrote:


	Does anyone have any information on these new Cisco phones:

7931
7937
7942
7945
7962
7965
7975
3951
3952

    I found a Q & A document on the SIP 3951:
http://www.cisco.com/web/CN/products/products_netsol/voices/products/ 
3951/pd
f/unified_sip_qa.pdf

Sincerely,

Dennis J. Hartmann
White Pine Communications
CCIE# 15651
CCVP/CCSI/CCIP/CCNP/MCSE
dh8 at pobox.com 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Tuesday, May 22, 2007 2:14 PM
To: Jason Aarons ((US))
Cc: cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920
Alternatives?

For what it's worth the E61 is the replacement for the E60.  I  
haven't seen
either so I couldn't even tell you what it looks like.

-Ryan

On May 22, 2007, at 2:08 PM, Jason Aarons ((US)) wrote:

I wasn't impressed using the Nokia model 60 back to CallManager. For  
example
there is not a WiFi signal indicator on phone menu, you have to dig
into
menus to find stuff. It wasn't a good PDA overall. I would rate the  
Palm OS
in the middle, Nokia at bottom and Windows Mobile 6 Pro on HTC/6700  
at the
top.

I'd stick with a Windows Mobile device with WiFi and use the IP Blue  
product
line.

http://www.ipblue.com/download.asp?product=vtgo_ppc 

Everyone has their own opinion...


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Ocampo, Walter
Sent: Tuesday, May 22, 2007 1:28 PM
To: Voll, Scott; Paul Choi; Dark Fiber; cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920
Alternatives?

Nokia has developed the Wireless phone with Dual mode.
One mode that  can connect to CCM using only the WLAN radio and the
intellisync SCCP client and the other mode for the Wireless telco.
Regards,

Walter Ocampo
System Network Analyst
Office of Information Technology
540 S.E 3rd. Ave, Suite 300
Fort Lauderdale, FL 33301
(954) 357 8041 Cell (954) 632 8228
wocampo at broward.org 


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, May 21, 2007 11:01 PM
To: Paul Choi; Dark Fiber; cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920
Alternatives?

Nokia?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 8:00 PM
To: Dark Fiber; cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920
Alternatives?

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as an alternative to
> say a 7920?  One time I saw some lapel type phones demonstrated at a
> Cisco event but have never seen anything official from Cisco
regarding
> them or similiar items.
>
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net 
> https://puck.nether.net/mailman/listinfo/cisco-voip 
>




________________________________________________________________________
____________
8:00? 8:25? 8:40? Find a flick in no time with the Yahoo! Search movie
showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 
_______________________________________________
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cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 
_______________________________________________
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This e-mail communication and any attachments may contain  
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privileged information and is for use by the designated addressee(s)  
named
above only.  If you are not the intended addressee, you are hereby  
notified
that you have received this communication in error and that any use or
reproduction of this email or its contents is strictly prohibited and 

may be
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notify
us immediately by replying to this message and deleting it from your
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From lelio at uoguelph.ca  Tue May 22 20:46:14 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 22 May 2007 20:46:14 -0400
Subject: [cisco-voip] unity express wizard - what do we lose without it
	and/or how do we run it again?
Message-ID: <006801c79cd3$c4337e50$c901a8c0@pdp11>


we cancelled out of the unity express wizard and i'm just wondering what we lose without it?

everything is working fine after editing the components, but i was reading this document

http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml#srst

and it says it would have created the dialpeers if we ran the wizard. i'm just wondering what stuff we lost by not using the wizard.


anyway to get to run the wizard again? it's ok to lose the current config. it's a brandnew install.
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From dhartma5 at optonline.net  Tue May 22 21:50:14 2007
From: dhartma5 at optonline.net (Dennis Hartmann)
Date: Tue, 22 May 2007 21:50:14 -0400
Subject: [cisco-voip] New Cisco Phones
In-Reply-To: <BAY126-DAV7FFE13722D63AF06EAE03BD360@phx.gbl>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com>
	<411933.7897.qm@web57210.mail.re3.yahoo.com>
	<407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org>
	<F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty>
	<C1FE15183DA37645BC0633BC604E44F005A35AF8@USNAEXCH.na.didata.local>
	<3241DBC0-9726-45EC-8F48-4B7F0DE73968@cisco.com>
	<BAY126-DAV7FFE13722D63AF06EAE03BD360@phx.gbl>
Message-ID: <01d101c79cdc$b4e2a940$0e7910ac@e1505>


	Does anyone have any information on these new Cisco phones:

7931
7937
7942
7945
7962
7965
7975
3951
3952

   I found a Q & A document on the SIP 3951:
http://www.cisco.com/web/CN/products/products_netsol/voices/products/3951/pd
f/unified_sip_qa.pdf

Sincerely,

Dennis J. Hartmann
White Pine Communications
CCIE# 15651
CCVP/CCSI/CCIP/CCNP/MCSE
dh8 at pobox.com

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Tuesday, May 22, 2007 2:14 PM
To: Jason Aarons ((US))
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

For what it's worth the E61 is the replacement for the E60.  I haven't seen
either so I couldn't even tell you what it looks like.

-Ryan

On May 22, 2007, at 2:08 PM, Jason Aarons ((US)) wrote:

I wasn't impressed using the Nokia model 60 back to CallManager. For example
there is not a WiFi signal indicator on phone menu, you have to dig into
menus to find stuff. It wasn't a good PDA overall. I would rate the Palm OS
in the middle, Nokia at bottom and Windows Mobile 6 Pro on HTC/6700 at the
top.

I'd stick with a Windows Mobile device with WiFi and use the IP Blue product
line.

http://www.ipblue.com/download.asp?product=vtgo_ppc

Everyone has their own opinion...


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Ocampo, Walter
Sent: Tuesday, May 22, 2007 1:28 PM
To: Voll, Scott; Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia has developed the Wireless phone with Dual mode.
One mode that  can connect to CCM using only the WLAN radio and the
intellisync SCCP client and the other mode for the Wireless telco.
Regards,

Walter Ocampo
System Network Analyst
Office of Information Technology
540 S.E 3rd. Ave, Suite 300
Fort Lauderdale, FL 33301
(954) 357 8041 Cell (954) 632 8228
wocampo at broward.org


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, May 21, 2007 11:01 PM
To: Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Nokia?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Monday, May 21, 2007 8:00 PM
To: Dark Fiber; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?

Cisco 7921?

--- Dark Fiber <d4rkf1ber at gmail.com> wrote:

> Anyone aware of any wireless SCCP phone devices as an alternative to 
> say a 7920?  One time I saw some lapel type phones demonstrated at a 
> Cisco event but have never seen anything official from Cisco regarding 
> them or similiar items.
>
> Just curious, thanks in advanced.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>




________________________________________________________________________
____________
8:00? 8:25? 8:40? Find a flick in no time with the Yahoo! Search movie
showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
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-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain confidential and
privileged information and is for use by the designated addressee(s) named
above only.  If you are not the intended addressee, you are hereby notified
that you have received this communication in error and that any use or
reproduction of this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please notify
us immediately by replying to this message and deleting it from your
computer. Thank you.
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From baryonyx5 at gmail.com  Tue May 22 21:56:40 2007
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Tue, 22 May 2007 20:56:40 -0500
Subject: [cisco-voip] New Cisco Phones
In-Reply-To: <A57E8EB3EA7B264FBDF12D930FED574F801751@argon.towson.edu>
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com>
	<411933.7897.qm@web57210.mail.re3.yahoo.com>
	<407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org>
	<F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty>
	<C1FE15183DA37645BC0633BC604E44F005A35AF8@USNAEXCH.na.didata.local>
	<3241DBC0-9726-45EC-8F48-4B7F0DE73968@cisco.com>
	<BAY126-DAV7FFE13722D63AF06EAE03BD360@phx.gbl>
	<CAFEF0F3-0D5F-4762-99F0-EDE8B6E4CF66@cisco.com>
	<A57E8EB3EA7B264FBDF12D930FED574F801751@argon.towson.edu>
Message-ID: <a749d0620705221856l5d093669h1fda49754fb71e4c@mail.gmail.com>

The 79x5's are listed on the licensing page in CCM 5.1.

On 5/22/07, Loraditch, Matthew* <mlorad1 at towson.edu> wrote:
>
>  Where did this list come from?  I know the 7931 has been out for a little
> while now, maybe a few months, and I recall reading the info about the 3951,
> but I haven't heard hide nor hair about the others.........
>
>  Matthew Loraditch
>
> ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net on behalf of Ryan Ratliff
> *Sent:* Tue 5/22/2007 4:18 PM
> *To:* dh8 at pobox.com
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] New Cisco Phones
>
>  7931 - http://www.cisco.com/en/US/products/hw/phones/ps379/
> products_data_sheet0900aecd80534259.html
>
> -Ryan
>
> On May 22, 2007, at 4:12 PM, Dennis Hartmann wrote:
>
>
>         Does anyone have any information on these new Cisco phones:
>
> 7931
> 7937
> 7942
> 7945
> 7962
> 7965
> 7975
> 3951
> 3952
>
>     I found a Q & A document on the SIP 3951:
> http://www.cisco.com/web/CN/products/products_netsol/voices/products/
> 3951/pd
> f/unified_sip_qa.pdf
>
> Sincerely,
>
> Dennis J. Hartmann
> White Pine Communications
> CCIE# 15651
> CCVP/CCSI/CCIP/CCNP/MCSE
> dh8 at pobox.com
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at puck.nether.net>]
> On Behalf Of Ryan Ratliff
> Sent: Tuesday, May 22, 2007 2:14 PM
> To: Jason Aarons ((US))
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
>
> For what it's worth the E61 is the replacement for the E60.  I
> haven't seen
> either so I couldn't even tell you what it looks like.
>
> -Ryan
>
> On May 22, 2007, at 2:08 PM, Jason Aarons ((US)) wrote:
>
> I wasn't impressed using the Nokia model 60 back to CallManager. For
> example
> there is not a WiFi signal indicator on phone menu, you have to dig into
> menus to find stuff. It wasn't a good PDA overall. I would rate the
> Palm OS
> in the middle, Nokia at bottom and Windows Mobile 6 Pro on HTC/6700
> at the
> top.
>
> I'd stick with a Windows Mobile device with WiFi and use the IP Blue
> product
> line.
>
> http://www.ipblue.com/download.asp?product=vtgo_ppc
>
> Everyone has their own opinion...
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- <cisco-voip->
> bounces at puck.nether.net] On Behalf Of Ocampo, Walter
> Sent: Tuesday, May 22, 2007 1:28 PM
> To: Voll, Scott; Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
>
> Nokia has developed the Wireless phone with Dual mode.
> One mode that  can connect to CCM using only the WLAN radio and the
> intellisync SCCP client and the other mode for the Wireless telco.
> Regards,
>
> Walter Ocampo
> System Network Analyst
> Office of Information Technology
> 540 S.E 3rd. Ave, Suite 300
> Fort Lauderdale, FL 33301
> (954) 357 8041 Cell (954) 632 8228
> wocampo at broward.org
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- <cisco-voip->
> bounces at puck.nether.net] On Behalf Of Voll, Scott
> Sent: Monday, May 21, 2007 11:01 PM
> To: Paul Choi; Dark Fiber; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
>
> Nokia?
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net<cisco-voip-bounces at puck.nether.net>]
> On Behalf Of Paul Choi
> Sent: Monday, May 21, 2007 8:00 PM
> To: Dark Fiber; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives?
>
> Cisco 7921?
>
> --- Dark Fiber <d4rkf1ber at gmail.com> wrote:
>
> > Anyone aware of any wireless SCCP phone devices as an alternative to
> > say a 7920?  One time I saw some lapel type phones demonstrated at a
> > Cisco event but have never seen anything official from Cisco regarding
> > them or similiar items.
> >
> > Just curious, thanks in advanced.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
>
> ________________________________________________________________________
> ____________
> 8:00? 8:25? 8:40? Find a flick in no time with the Yahoo! Search movie
> showtime shortcut.
> http://tools.search.yahoo.com/shortcuts/#news
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and
> privileged information and is for use by the designated addressee(s)
> named
> above only.  If you are not the intended addressee, you are hereby
> notified
> that you have received this communication in error and that any use or
> reproduction of this email or its contents is strictly prohibited and
> may be
> unlawful.  If you have received this communication in error, please
> notify
> us immediately by replying to this message and deleting it from your
> computer. Thank you.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
kris seraphine
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From n68851 at hotmail.com  Wed May 23 00:38:49 2007
From: n68851 at hotmail.com (Bryan Nowak)
Date: Tue, 22 May 2007 23:38:49 -0500
Subject: [cisco-voip] IPCC Name to User Step
References: <a1b07ea60705211821i13de626s77021cc8aae0f943@mail.gmail.com><411933.7897.qm@web57210.mail.re3.yahoo.com><407055A92CECCB499C922A2D35FC19A608DEE7C2@Apollo.wesd.org><F4DD4283A3EF194B84D774C60FB7F786024DA32F@UVEXVS01.bc.broward.cty><C1FE15183DA37645BC0633BC604E44F005A35AF8@USNAEXCH.na.didata.local><3241DBC0-9726-45EC-8F48-4B7F0DE73968@cisco.com><BAY126-DAV7FFE13722D63AF06EAE03BD360@phx.gbl><CAFEF0F3-0D5F-4762-99F0-EDE8B6E4CF66@cisco.com><A57E8EB3EA7B264FBDF12D930FED574F801751@argon.towson.edu>
	<a749d0620705221856l5d093669h1fda49754fb71e4c@mail.gmail.com>
Message-ID: <BAY141-DAV7EE745CAD725035B649AA93350@phx.gbl>

Hi,

We've been trying to utilize the Name to User step in IPCC for multiple scripts, but can't get it to work at all.  No matter what name is entered, it is not found.  We're integrated with AD and the desktop agents are not experiencing any problems.  I'm assuming that the Name to User step would be pulling from the same directory. Is that not the case?  If my assumption is correct, does anyone know what the problem might be?  I'm not "trained" with IPCC and I haven't been able to locate any solid information in the CRS scripting guides or other docs on CCO.  Any ideas?

Thanks,
Bryan
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From gsobhi at nileonebrokers.com  Wed May 23 02:38:36 2007
From: gsobhi at nileonebrokers.com (George Sobhi)
Date: Wed, 23 May 2007 09:38:36 +0300
Subject: [cisco-voip] MOH file
In-Reply-To: <22534.50219.qm@web57203.mail.re3.yahoo.com>
Message-ID: <NILESERVER1NzW3l1GM000007f8@nileserver1.niledomain.com>

How can I convert it?

 

 

-----Original Message-----
From: Paul Choi [mailto:asobihoudai at yahoo.com] 
Sent: Tuesday, May 22, 2007 6:12 PM
To: CarlosOrtiz at bayviewfinancial.com; George Sobhi
Cc: cisco-voip-bounces at puck.nether.net; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MOH file

 

I'd save that .wma file as a .wav and then convert.

 

Paul

--- CarlosOrtiz at bayviewfinancial.com wrote:

 

> Not sure if Call Mgr can convert wma, but it will

> definitely convert .wav 

> and mp3 formats of the same file.  Then you can just

> upload the g711 file 

> to the flash of the 2811 like you would an IOS file.

> 

> Carlos

> 

> 

> 

> "George Sobhi" <gsobhi at nileonebrokers.com> 

> Sent by: cisco-voip-bounces at puck.nether.net

> 05/22/2007 09:49 AM

> 

> To

> <cisco-voip at puck.nether.net>

> cc

> 

> Subject

> [cisco-voip] MOH file

> 

> 

> 

> 

> 

> 

> Dear all,

> I have a wma file how can I convert it and tftp it

> to 2811 cisco router 

> with 12.4(4)XC / CME 4.0(1)

>  

> thanks

>  

> George Sobhi

> Technical Support Engineer

> NILEone BROKERS Co.

> 4, Syria St, Mohandeseen, Giza-Egypt.

> Phone: 202-3363252

> Ext.:      121

> Fax:      202-3363260

 

 

 

       

____________________________________________________________________________
________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's
updated for today's economy) at Yahoo! Games.

http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  

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From newlook110 at yahoo.com  Wed May 23 06:21:15 2007
From: newlook110 at yahoo.com (new look)
Date: Wed, 23 May 2007 03:21:15 -0700 (PDT)
Subject: [cisco-voip] Changing the CCM Password
Message-ID: <993231.47541.qm@web61320.mail.yahoo.com>

All,
How can I change the CCM log in Password. I mean change the Web authentication password.
Thanks


       
____________________________________________________________________________________Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.
http://answers.yahoo.com/dir/?link=list&sid=396545469
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From PEKOU at gr.ibm.com  Wed May 23 07:28:40 2007
From: PEKOU at gr.ibm.com (Evi Pekou)
Date: Wed, 23 May 2007 14:28:40 +0300
Subject: [cisco-voip] SIP trunk calling name restriction
Message-ID: <OF50D13231.FF2DAC0C-ONC22572E4.003E7D04-C22572E4.003F16B3@gr.ibm.com>

Hi all,

I have a CCM 4.1(3) configured with a SIP trunk to a service provider's 
Asterisk. 
The provider has requested for outgoing calls  not to send calling name 
identification at all.
When I configure the "calling name presentation" at SIP trunk 
configuration settings as restricted,
the calling name "Anonymous" is sent from CCM to Asterisk instead of 
user's name. 
Is this parameter configurable? Can I change the "Anonymous" string to 
something else or even better leave the calling name blank?

Thanks,
Evi
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From keli at carocomp.ro  Wed May 23 07:29:26 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Wed, 23 May 2007 14:29:26 +0300
Subject: [cisco-voip] Maximum outgoing call duration
Message-ID: <46542596.7000909@carocomp.ro>

Hi!

I have a client with CallManager 4.1 installed, who wants to limit 
outgoing calls (preferably only on some trunks) to a certain length 
(Like no call can last longer than 7 minutes, for example.)

Is that possible? If so, how? (even third-party solutions would be 
interesting, but only if there's no other way)

thanks,
  Zoltan

From newlook110 at yahoo.com  Wed May 23 06:35:59 2007
From: newlook110 at yahoo.com (new look)
Date: Wed, 23 May 2007 03:35:59 -0700 (PDT)
Subject: [cisco-voip] E1 Problem
Message-ID: <82151.27854.qm@web61311.mail.yahoo.com>

Dear All,
I'm appling a new Cisco IPTelephony system my voice geatway is Cisco router and I'm using E1 Interface 
the problem is the destance between my company and the Telecom exchange is about 5Km so the max bandwidth I have is 512K
So I need to know can I use this channels only?
what is the router configuration ?
By the way I'm using paradyne two units 7996 (DSL/G703)
Thanks


       
____________________________________________________________________________________Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. 
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From keli at carocomp.ro  Wed May 23 07:46:57 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Wed, 23 May 2007 14:46:57 +0300
Subject: [cisco-voip] Messages softkey
Message-ID: <465429B1.3080403@carocomp.ro>

Hi!

I hqave some 7911G (and 7906G) Cisco IP phones, right now on a CCME.

is there a way to disable the "Messages" softkey in either CallManager, 
or CCME? Right now I have a CCME, that does not have any voicemail 
setup, but the Msgs button insists to stay on-screen.

The ephone-template -> softkeys configuration options don't even include 
the Msgs softkey, so I can't enable/disable it, neither can I move it 
around. It seems it gets placed automatically in the third position, or 
less if only one other softkey is defined in the template.

The telephony-service configuration does not include voicemail. IOS is 
C2800NM-SPSERVICESK9-M, Version 12.4(11)T, obviously, with CCME 4.0(2)

As far as I recall, I found it similarly close to impossible to remove 
it from CallManager as well.

Thanks,
  Zoltan

From gokhanciscottl at yahoo.com  Wed May 23 07:11:46 2007
From: gokhanciscottl at yahoo.com (gokhan senol)
Date: Wed, 23 May 2007 04:11:46 -0700 (PDT)
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives
Message-ID: <26048.56727.qm@web39610.mail.mud.yahoo.com>

what about  ' kirk '  products
it is a concentrator and siemens dect   phones register CCM  via this concentrator


       
____________________________________________________________________________________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  
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From ErasmuE4 at telkom.co.za  Wed May 23 08:13:03 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 23 May 2007 14:13:03 +0200
Subject: [cisco-voip] Changing the CCM Password
In-Reply-To: <993231.47541.qm@web61320.mail.yahoo.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01547030@CNTRRA20-XCS01.telkom.co.za>


Look at the multilevel administration docs cisco have 

 

Explain how to set up administration with different access levels and
different passwords etc.

 

 

 

Erik erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of new look
Sent: Wednesday, May 23, 2007 12:21 PM
To: cisco voip
Subject: [cisco-voip] Changing the CCM Password

 

All,

How can I change the CCM log in Password. I mean change the Web
authentication password.

Thanks

 

________________________________

Yahoo! oneSearch: Finally, mobile search that gives answers
<http://us.rd.yahoo.com/evt=48252/*http:/mobile.yahoo.com/mobileweb/on
esearch?refer=1ONXIC> , not web links. 



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From lelio at uoguelph.ca  Wed May 23 08:16:45 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 23 May 2007 08:16:45 -0400
Subject: [cisco-voip] Maximum outgoing call duration
References: <46542596.7000909@carocomp.ro>
Message-ID: <003401c79d34$3ae588d0$47196883@cfs.uoguelph.ca>

Scott Voll can tell you ALL about this......

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: Kelemen Zoltan 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, May 23, 2007 7:29 AM
  Subject: [cisco-voip] Maximum outgoing call duration


  Hi!

  I have a client with CallManager 4.1 installed, who wants to limit 
  outgoing calls (preferably only on some trunks) to a certain length 
  (Like no call can last longer than 7 minutes, for example.)

  Is that possible? If so, how? (even third-party solutions would be 
  interesting, but only if there's no other way)

  thanks,
    Zoltan
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From ErasmuE4 at telkom.co.za  Wed May 23 08:34:21 2007
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 23 May 2007 14:34:21 +0200
Subject: [cisco-voip] attendant console issues - call stuck in call manager Q
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F01547039@CNTRRA20-XCS01.telkom.co.za>


Hi

 

============== just some info and maybe someone else has seen the same
==============================

 

Just implemented Call manager cluster for a customer  with the
following software 

 

 

OS2000-4-3 sr8 

 

CCM 4-2-3 sr1

 

 IPCC Express 4-1

 

 

Everything went fine apart from the attendant consoles. On Day one
when calls started coming into the attendants - when they are all busy
calls go to the Q because I enabled the Q in the accconfig.bat
utility. But then when the attendants are free again the calls are not
de-queued to them. Worked with tac and they gave me an engineering
special version of CCM to load. Upgraded the cluster last night.
Looked better this morning but at 11 am the customer reported that the
same thing happened again so we restarted the cti manager service on
the active call manager. This seems to help but it is not ideal and
eventually we will have to find out why this error is occurring.

 

Will post to the group if we find a way to resolve.

 

So - please be aware that depending on the software version you intend
to install - that you might encounter similar problems 

 

 

Erik Erasmus

 



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From keli at carocomp.ro  Wed May 23 08:36:42 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Wed, 23 May 2007 15:36:42 +0300
Subject: [cisco-voip] Maximum outgoing call duration
In-Reply-To: <003401c79d34$3ae588d0$47196883@cfs.uoguelph.ca>
References: <46542596.7000909@carocomp.ro>
	<003401c79d34$3ae588d0$47196883@cfs.uoguelph.ca>
Message-ID: <4654355A.2030903@carocomp.ro>

A bit more detail? :-) Forgive me, I'm new to the list.

In the mean time, I did find the "Maximum Call Duration Timer" in 
CallManager Service Parameters, however, I'd still be interested if this 
would be possible to apply only to outgoing calls. As far as I 
understand this, this timer limits ALL calls in the system, including 
on-net calls.
Since the client asked for something like 7minutes on external calls, 
limiting all internal (and incoming) calls to 7 minutes as well seems a 
bit harsh.

Zoltan

Lelio Fulgenzi wrote:
> Scott Voll can tell you ALL about this......
>  
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo 
> buffalo."      WJR
>
>     ----- Original Message -----
>     *From:* Kelemen Zoltan <mailto:keli at carocomp.ro>
>     *To:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     *Sent:* Wednesday, May 23, 2007 7:29 AM
>     *Subject:* [cisco-voip] Maximum outgoing call duration
>
>     Hi!
>
>     I have a client with CallManager 4.1 installed, who wants to limit
>     outgoing calls (preferably only on some trunks) to a certain length
>     (Like no call can last longer than 7 minutes, for example.)
>
>     Is that possible? If so, how? (even third-party solutions would be
>     interesting, but only if there's no other way)
>
>     thanks,
>       Zoltan
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>


From Matt.Slaga at us.didata.com  Wed May 23 08:39:42 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 23 May 2007 08:39:42 -0400
Subject: [cisco-voip] MOH file
In-Reply-To: <NILESERVER1NzW3l1GM000007f8@nileserver1.niledomain.com>
References: <22534.50219.qm@web57203.mail.re3.yahoo.com>
	<NILESERVER1NzW3l1GM000007f8@nileserver1.niledomain.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005A6B9D6@USNAEXCH.na.didata.local>

Get a copy of CoolEdit.  It will allow you to convert from almost any
format to almost any format.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of George Sobhi
Sent: Wednesday, May 23, 2007 2:39 AM
To: 'Paul Choi'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MOH file

 

How can I convert it?

 

 

-----Original Message-----
From: Paul Choi [mailto:asobihoudai at yahoo.com] 
Sent: Tuesday, May 22, 2007 6:12 PM
To: CarlosOrtiz at bayviewfinancial.com; George Sobhi
Cc: cisco-voip-bounces at puck.nether.net; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MOH file

 

I'd save that .wma file as a .wav and then convert.

 

Paul

--- CarlosOrtiz at bayviewfinancial.com wrote:

 

> Not sure if Call Mgr can convert wma, but it will

> definitely convert .wav 

> and mp3 formats of the same file.  Then you can just

> upload the g711 file 

> to the flash of the 2811 like you would an IOS file.

> 

> Carlos

> 

> 

> 

> "George Sobhi" <gsobhi at nileonebrokers.com> 

> Sent by: cisco-voip-bounces at puck.nether.net

> 05/22/2007 09:49 AM

> 

> To

> <cisco-voip at puck.nether.net>

> cc

> 

> Subject

> [cisco-voip] MOH file

> 

> 

> 

> 

> 

> 

> Dear all,

> I have a wma file how can I convert it and tftp it

> to 2811 cisco router 

> with 12.4(4)XC / CME 4.0(1)

>  

> thanks

>  

> George Sobhi

> Technical Support Engineer

> NILEone BROKERS Co.

> 4, Syria St, Mohandeseen, Giza-Egypt.

> Phone: 202-3363252

> Ext.:      121

> Fax:      202-3363260

 

 

 

       

________________________________________________________________________
____________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now
(it's updated for today's economy) at Yahoo! Games.

http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  




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From Matt.Slaga at us.didata.com  Wed May 23 08:41:36 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 23 May 2007 08:41:36 -0400
Subject: [cisco-voip] Messages softkey
In-Reply-To: <465429B1.3080403@carocomp.ro>
References: <465429B1.3080403@carocomp.ro>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005A6B9E4@USNAEXCH.na.didata.local>

The messages button is hard coded in to the 7905, 7906, 7911 and 7912.
Other than reprogramming the firmware, there is no way to remove the
softkeys.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kelemen Zoltan
Sent: Wednesday, May 23, 2007 7:47 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Messages softkey

Hi!

I hqave some 7911G (and 7906G) Cisco IP phones, right now on a CCME.

is there a way to disable the "Messages" softkey in either CallManager, 
or CCME? Right now I have a CCME, that does not have any voicemail 
setup, but the Msgs button insists to stay on-screen.

The ephone-template -> softkeys configuration options don't even include

the Msgs softkey, so I can't enable/disable it, neither can I move it 
around. It seems it gets placed automatically in the third position, or 
less if only one other softkey is defined in the template.

The telephony-service configuration does not include voicemail. IOS is 
C2800NM-SPSERVICESK9-M, Version 12.4(11)T, obviously, with CCME 4.0(2)

As far as I recall, I found it similarly close to impossible to remove 
it from CallManager as well.

Thanks,
  Zoltan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
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designated addressee(s) named above only.  If you are not the
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notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From Matt.Slaga at us.didata.com  Wed May 23 08:44:54 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 23 May 2007 08:44:54 -0400
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives
In-Reply-To: <26048.56727.qm@web39610.mail.mud.yahoo.com>
References: <26048.56727.qm@web39610.mail.mud.yahoo.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005A6B9F3@USNAEXCH.na.didata.local>

Spectrum has a SCCP wireless phone that was available prior to the 7920.
They also have a PTT (push to talk) version out now as well.

 

http://www.spectralink.com/products/index.jsp

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of gokhan senol
Sent: Wednesday, May 23, 2007 7:12 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives

 

what about  ' kirk '  products

it is a concentrator and siemens dect   phones register CCM  via this
concentrator

 

 

 

________________________________

Finding fabulous fares is fun.
Let Yahoo! FareChase search your favorite travel sites
<http://farechase.yahoo.com/promo-generic-14795097;_ylc=X3oDMTFtNW45amVp
BF9TAzk3NDA3NTg5BF9zAzI3MTk0ODEEcG9zAzEEc2VjA21haWx0YWdsaW5lBHNsawNxMS0w
Nw--%0d%0a>  to find flight and hotel bargains.




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From Matt.Slaga at us.didata.com  Wed May 23 08:46:00 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 23 May 2007 08:46:00 -0400
Subject: [cisco-voip] Maximum outgoing call duration
In-Reply-To: <4654355A.2030903@carocomp.ro>
References: <46542596.7000909@carocomp.ro><003401c79d34$3ae588d0$47196883@cfs.uoguelph.ca>
	<4654355A.2030903@carocomp.ro>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005A6B9F8@USNAEXCH.na.didata.local>

This is probably something that a PBX firewall could handle.  Check
www.securelogix.com/Pbx-Firewall.html.




-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kelemen Zoltan
Sent: Wednesday, May 23, 2007 8:37 AM
To: Cisco VoIP
Subject: Re: [cisco-voip] Maximum outgoing call duration

A bit more detail? :-) Forgive me, I'm new to the list.

In the mean time, I did find the "Maximum Call Duration Timer" in 
CallManager Service Parameters, however, I'd still be interested if this

would be possible to apply only to outgoing calls. As far as I 
understand this, this timer limits ALL calls in the system, including 
on-net calls.
Since the client asked for something like 7minutes on external calls, 
limiting all internal (and incoming) calls to 7 minutes as well seems a 
bit harsh.

Zoltan

Lelio Fulgenzi wrote:
> Scott Voll can tell you ALL about this......
>  
>
------------------------------------------------------------------------
--------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo 
> buffalo."      WJR
>
>     ----- Original Message -----
>     *From:* Kelemen Zoltan <mailto:keli at carocomp.ro>
>     *To:* cisco-voip at puck.nether.net
<mailto:cisco-voip at puck.nether.net>
>     *Sent:* Wednesday, May 23, 2007 7:29 AM
>     *Subject:* [cisco-voip] Maximum outgoing call duration
>
>     Hi!
>
>     I have a client with CallManager 4.1 installed, who wants to limit
>     outgoing calls (preferably only on some trunks) to a certain
length
>     (Like no call can last longer than 7 minutes, for example.)
>
>     Is that possible? If so, how? (even third-party solutions would be
>     interesting, but only if there's no other way)
>
>     thanks,
>       Zoltan
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From keli at carocomp.ro  Wed May 23 08:51:43 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Wed, 23 May 2007 15:51:43 +0300
Subject: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005A6B9F3@USNAEXCH.na.didata.local>
References: <26048.56727.qm@web39610.mail.mud.yahoo.com>
	<C1FE15183DA37645BC0633BC604E44F005A6B9F3@USNAEXCH.na.didata.local>
Message-ID: <465438DF.9060704@carocomp.ro>

Kirk was bought by Spectralink (who later got bought by Polycom IIRC), 
so you're probably talking about the same solution.

Look for Kirk Wireless Server 600/3 and compatible handsets. You can use 
almost any kind of standard-supporting DECT handset (usually labeled 
"cordeless phones") with a KIRK WS, however, their own models are quite 
good (and very solid) as well.

Zoltan

Matt Slaga (US) wrote:
>
> Spectrum has a SCCP wireless phone that was available prior to the 
> 7920.  They also have a PTT (push to talk) version out now as well.
>
>  
>
> http://www.spectralink.com/products/index.jsp
>
>  
>
>  
>
>  
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *gokhan senol
> *Sent:* Wednesday, May 23, 2007 7:12 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] SCCP / CallManager Wireless 7920 Alternatives
>
>  
>
> what about  ' kirk '  products
>
> it is a concentrator and siemens dect   phones register CCM  via this 
> concentrator
>
>  
>
>  
>
>  
>
> ------------------------------------------------------------------------
>
> Finding fabulous fares is fun.
> Let Yahoo! FareChase search your favorite travel sites 
> <http://farechase.yahoo.com/promo-generic-14795097;_ylc=X3oDMTFtNW45amVpBF9TAzk3NDA3NTg5BF9zAzI3MTk0ODEEcG9zAzEEc2VjA21haWx0YWdsaW5lBHNsawNxMS0wNw--%0d%0a> 
> to find flight and hotel bargains.
>
> ------------------------------------------------------------------------
>
> *Disclaimer: This e-mail communication and any attachments may contain 
> confidential and privileged information and is for use by the 
> designated addressee(s) named above only. If you are not the intended 
> addressee, you are hereby notified that you have received this 
> communication in error and that any use or reproduction of this email 
> or its contents is strictly prohibited and may be unlawful. If you 
> have received this communication in error, please notify us 
> immediately by replying to this message and deleting it from your 
> computer. Thank you. *
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   



From CarlosOrtiz at bayviewfinancial.com  Wed May 23 09:46:07 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Wed, 23 May 2007 09:46:07 -0400
Subject: [cisco-voip] MOH file
In-Reply-To: <NILESERVER1NzW3l1GM000007f8@nileserver1.niledomain.com>
Message-ID: <OF288AD0B2.4DA4D155-ON852572E4.004B8F96-852572E4.004BA237@bftg.com>

Google..... convert wma to .wav   --   find one that's free or free 
trial......



"George Sobhi" <gsobhi at nileonebrokers.com> 
Sent by: cisco-voip-bounces at puck.nether.net
05/23/2007 02:38 AM

To
"'Paul Choi'" <asobihoudai at yahoo.com>
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] MOH file






How can I convert it?
 
 
-----Original Message-----
From: Paul Choi [mailto:asobihoudai at yahoo.com] 
Sent: Tuesday, May 22, 2007 6:12 PM
To: CarlosOrtiz at bayviewfinancial.com; George Sobhi
Cc: cisco-voip-bounces at puck.nether.net; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MOH file
 
I'd save that .wma file as a .wav and then convert.
 
Paul
--- CarlosOrtiz at bayviewfinancial.com wrote:
 
> Not sure if Call Mgr can convert wma, but it will
> definitely convert .wav 
> and mp3 formats of the same file.  Then you can just
> upload the g711 file 
> to the flash of the 2811 like you would an IOS file.
> 
> Carlos
> 
> 
> 
> "George Sobhi" <gsobhi at nileonebrokers.com> 
> Sent by: cisco-voip-bounces at puck.nether.net
> 05/22/2007 09:49 AM
> 
> To
> <cisco-voip at puck.nether.net>
> cc
> 
> Subject
> [cisco-voip] MOH file
> 
> 
> 
> 
> 
> 
> Dear all,
> I have a wma file how can I convert it and tftp it
> to 2811 cisco router 
> with 12.4(4)XC / CME 4.0(1)
> 
> thanks
> 
> George Sobhi
> Technical Support Engineer
> NILEone BROKERS Co.
> 4, Syria St, Mohandeseen, Giza-Egypt.
> Phone: 202-3363252
> Ext.:      121
> Fax:      202-3363260
 
 
 
 
____________________________________________________________________________________Boardwalk 
for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for 
today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From tele at plexia.com  Wed May 23 10:37:56 2007
From: tele at plexia.com (tele)
Date: Wed, 23 May 2007 16:37:56 +0200
Subject: [cisco-voip] CME in only SIP scenario
In-Reply-To: <1179826479.6586.108.camel@mda>
References: <1179826479.6586.108.camel@mda>
Message-ID: <1179931076.6586.191.camel@mda>

Ok so i've made a little step.
What i can seen using debug voip rtp is that the stream is only on "rx",
for example:

May 23 14:08:42.806: RTP(963): fs rx s=192.168.5.2(41000),
d=192.168.5.223(16400), pt=8, ts=2D780, ssrc=BCA
May 23 14:08:42.810: RTP(65010): fs rx s=82.215.172.192(17532),
d=82.215.133.140(17464), pt=8, ts=4EFF9440, ssrc=1959A317

I can image what happens with the rtp stream:

82.215.172.192     82.215.133.140       192.168.5.223      192.168.5.2
UA1 --------------> [CME extip -------- intip CME]-------- UAINT
                  rx                            rx
                  ->                            <-

the problem is that there is no "tx" inside the "loopback call routing".
should be like this:


82.215.172.192     82.215.133.140       192.168.5.223      192.168.5.2
UA1 --------------> [CME extip -------- intip CME]-------- UAINT
                  rx           tx     tx          rx
                  ->           ->     <-          <-


could be a NAT problem cause i'm using two interface one public and one
private? 


thank you

:tele

On Tue, 2007-05-22 at 11:34 +0200, tele wrote:
> Hi,
> 
> I've a particular configuration of a CME 4.1 in a only SIP scenario, CME
> register as UA to a SIP proxy.
> CME is able to recive a range of number like
> 
> 01044992XX
> 
> and it's able to deliver call to internal phone.
> I've interoperability issue with some gateway in our network so i need
> to restrict call transfer/forwarding only to CME and not throught the
> network.
> To be able to achieve this i need the Loopback Call routing function:
> http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00807e9b10.html
> 
> But using the loopback call routing i've voice problem, no audio but if
> i do a debug voice rtp packet i can see rtp packet going to correct
> ports and IP. but i can hear nothing.
> 
> Ok attached here there are two configuration, the first one:
> 
> cme_with_dialpeer.conf i'm using dial-peer and i've no problem with
> voice but i've problem with call transfer/forward cause re-invite for
> on-hold goes through the network and i prefer CME handle this.
> 
> cme-loopback-dn.cfg i'm using loopback-dn in the ephone-dn, call
> signalling is ok but i can't hear voice.
> 
> I don't know how to starts troubleshooting this issue, maybe any of you
> have an idea.
> 
> thank you very much
> 
> :tele
> 
> 
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
cme-test#
May 23 14:08:19.447: voip_rtp_update_callinfo: RTCP enabled
May 23 14:08:19.447: voip_rtp_update_local_ssrc srtp: 0 CallID: 2408 dstCallID: 2407 new ssrc: 0x1CBC05DF
May 23 14:08:19.447: voip_rtcp_get_cname: cname=0.0.0 at 192.168.5.223
May 23 14:08:19.447: voip_rtp_update_callinfo: RTP address/port changes 0:0->C0A80502:A028
May 23 14:08:19.447: voip_rtp_update_callinfo: first_pak=TRUE, new_ssrc=0, type=3
May 23 14:08:19.451:  voip_rtp_update_xmit_info
May 23 14:08:19.451:  voip_rtp_update_xmit_info, dstvdbptr: 859FBC20, dstCallID 2407, gccb: 843F5B10, xmitFunc 804E6CEC,context 0
May 23 14:08:19.451:  voip_rtp_update_xmit_info Context is NULL, exit
May 23 14:08:19.451: voip_rtp_set_non_rtp_call: Non-RTP call end
May 23 14:08:19.451: voip_rtp_exchange_context_info
May 23 14:08:19.455:  voip_rtp_update_xmit_info
May 23 14:08:19.455:  voip_rtp_update_xmit_info, dstvdbptr: 8529B4B4, dstCallID 2407, gccb: 843F5B10, xmitFunc 804E6CEC,context 83CC847C
May 23 14:08:19.455:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 8529B4B4, xmit_info->dstCallID 2407, xmit_info->xmitFunc 804E6CEC, xmit_info->context 83CC847C
May 23 14:08:19.455:  voip xmit info count: 1
May 23 14:08:19.455: voip_rtp_exchange_context_info
May 23 14:08:19.455:  voip_rtp_update_xmit_info
May 23 14:08:19.455:  voip_rtp_update_xmit_info, dstvdbptr: 859FBC20, dstCallID 2407, gccb: 843F5B10, xmitFunc 804E6CEC,context 83CC847C
May 23 14:08:19.455:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 859FBC20, xmit_info->dstCallID 2407, xmit_info->xmitFunc 804E6CEC, xmit_info->context 83CC847C
May 23 14:08:19.459:  voip xmit info count: 1
May 23 14:08:19.459: voip_rtp_exchange_context_info
May 23 14:08:19.459:  voip_rtp_update_xmit_info
May 23 14:08:19.459:  voip_rtp_update_xmit_info, dstvdbptr: 859FBC20, dstCallID 2407, gccb: 843F5B10, xmitFunc 804E6CEC,context 83CC847C
May 23 14:08:19.459:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 859FBC20, xmit_info->dstCallID 2407, xmit_info->xmitFunc 804E6CEC, xmit_info->context 83CC847C
May 23 14:08:19.459:  voip xmit info count: 1
May 23 14:08:19.467: voip_rtcp_start_session:
May 23 14:08:19.467: voip_rtcp_start_session: start session
May 23 14:08:19.488: voip_rtp_create_session idb tableid is 0,session table id is 0
May 23 14:08:19.488: voip_rtp_create_session: callID=2405, dstCallID=2406 laddr=82.215.133.140, lport=17464,raddr=82.215.172.192, rport=17532, type=3, sig_tos=3, ip_tos=5
May 23 14:08:19.492: voip_rtp_update_local_ssrc srtp: 0 CallID: 2405 dstCallID: 2406 new ssrc: 0x68B858C
May 23 14:08:19.492: voip_rtcp_get_cname: cname=0.0.0 at 82.215.133.140
May 23 14:08:19.492:  voip_rtp_update_xmit_info
May 23 14:08:19.492:  voip_rtp_update_xmit_info, dstvdbptr: 859FBF14, dstCallID 2406, gccb: 83CBADB0, xmitFunc 804E6CEC,context 0
May 23 14:08:19.492:  voip_rtp_update_xmit_info Context is NULL, exit
May 23 14:08:19.492: voip_rtp_update_cisco_payloads:
May 23 14:08:19.496: voip_rtp_update_cisco_payloads: The updated cisco reserved payload types are:
         cisco_codec_fax_pt = 96, cisco_codec_fax_ack_pt = 97,
         cisco_rtp_dtmf_relay_pt = 121, cisco_fax_relay_pt = 122,
         cisco_cas_pt = 123, cisco_clear_channel_pt = 125,
         cisco_rtp_type_annex_B = -2053128428
May 23 14:08:19.496: voip_rtp_set_non_rtp_call: Non-RTP call end
May 23 14:08:19.500: voip_rtp_exchange_context_info
May 23 14:08:19.500:  voip_rtp_update_xmit_info
May 23 14:08:19.500:  voip_rtp_update_xmit_info, dstvdbptr: 8529B4B4, dstCallID 2406, gccb: 83CBADB0, xmitFunc 804E6CEC,context 844C598C
May 23 14:08:19.500:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 8529B4B4, xmit_info->dstCallID 2406, xmit_info->xmitFunc 804E6CEC, xmit_info->context 844C598C
May 23 14:08:19.500:  voip xmit info count: 1
May 23 14:08:19.500: voip_rtp_exchange_context_info
May 23 14:08:19.500:  voip_rtp_update_xmit_info
May 23 14:08:19.500:  voip_rtp_update_xmit_info, dstvdbptr: 859FBF14, dstCallID 2406, gccb: 83CBADB0, xmitFunc 804E6CEC,context 844C598C
May 23 14:08:19.504:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 859FBF14, xmit_info->dstCallID 2406, xmit_info->xmitFunc 804E6CEC, xmit_info->context 844C598C
May 23 14:08:19.504:  voip xmit info count: 1
May 23 14:08:19.504: voip_rtp_exchange_context_info
May 23 14:08:19.504:  voip_rtp_update_xmit_info
May 23 14:08:19.504:  voip_rtp_update_xmit_info, dstvdbptr: 859FBF14, dstCallID 2406, gccb: 83CBADB0, xmitFunc 804E6CEC,context 844C598C
May 23 14:08:19.504:  voip_rtp_update_xmit_info Xmit Info node current values xmit_info->dstvdbptr: 859FBF14, xmit_info->dstCallID 2406, xmit_info->xmitFunc 804E6CEC, xmit_info->context 844C598C
May 23 14:08:19.504:  voip xmit info count: 1
May 23 14:08:19.508: voip_rtcp_start_session:
May 23 14:08:19.512: voip_rtcp_start_session: start session
May 23 14:08:19.536: RTP(0): ps rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=50, ssrc=BCA
May 23 14:08:19.552: RTP(1): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F0, ssrc=BCA
May 23 14:08:19.572: RTP(2): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=190, ssrc=BCA
May 23 14:08:19.592: RTP(3): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=230, ssrc=BCA
May 23 14:08:19.612: RTP(4): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D0, ssrc=BCA
May 23 14:08:19.632: RTP(5): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=370, ssrc=BCA
May 23 14:08:19.652: RTP(6): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=410, ssrc=BCA
May 23 14:08:19.672: RTP(7): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4B0, ssrc=BCA
May 23 14:08:19.692: RTP(8): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=550, ssrc=BCA
May 23 14:08:19.712: RTP(9): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5F0, ssrc=BCA
May 23 14:08:19.732: RTP(10): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=690, ssrc=BCA
May 23 14:08:19.752: RTP(11): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=730, ssrc=BCA
May 23 14:08:19.772: RTP(12): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7D0, ssrc=BCA
May 23 14:08:19.792: RTP(13): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=870, ssrc=BCA
May 23 14:08:19.812: RTP(14): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=910, ssrc=BCA
May 23 14:08:19.820: RTP(63865): ps rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCC8A0, ssrc=1959A317
May 23 14:08:19.832: RTP(15): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9B0, ssrc=BCA
May 23 14:08:19.840: RTP(63866): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCC940, ssrc=1959A317
May 23 14:08:19.852: RTP(16): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A50, ssrc=BCA
May 23 14:08:19.860: RTP(63867): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCC9E0, ssrc=1959A317
May 23 14:08:19.872: RTP(17): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AF0, ssrc=BCA
May 23 14:08:19.876: RTP(63868): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCCA80, ssrc=1959A317
May 23 14:08:19.892: RTP(18): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B90, ssrc=BCA
May 23 14:08:19.896: RTP(63869): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCCB20, ssrc=1959A317
May 23 14:08:19.912: RTP(19): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C30, ssrc=BCA
May 23 14:08:19.916: RTP(63870): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCCBC0, ssrc=1959A317
May 23 14:08:19.932: RTP(20): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CD0, ssrc=BCA
May 23 14:08:19.940: RTP(63871): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCCC60, ssrc=1959A317
May 23 14:08:19.952: RTP(21): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D70, ssrc=BCA
May 23 14:08:19.960: RTP(63872): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCCD00, ssrc=1959A317
May 23 14:08:19.972: RTP(22): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E10, ssrc=BCA
May 23 14:08:19.976: RTP(63873): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCCDA0, ssrc=1959A317
May 23 14:08:19.992: RTP(23): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EB0, ssrc=BCA
May 23 14:08:19.996: RTP(63874): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCCE40, ssrc=1959A317
May 23 14:08:20.012: RTP(24): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F50, ssrc=BCA
May 23 14:08:20.016: RTP(63875): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCCEE0, ssrc=1959A317
May 23 14:08:20.032: RTP(25): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FF0, ssrc=BCA
May 23 14:08:20.036: RTP(63876): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCCF80, ssrc=1959A317
May 23 14:08:20.052: RTP(26): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1090, ssrc=BCA
May 23 14:08:20.060: RTP(63877): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD020, ssrc=1959A317
May 23 14:08:20.072: RTP(27): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1130, ssrc=BCA
May 23 14:08:20.076: RTP(63878): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD0C0, ssrc=1959A317
May 23 14:08:20.093: RTP(28): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11D0, ssrc=BCA
May 23 14:08:20.097: RTP(63879): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD160, ssrc=1959A317
May 23 14:08:20.113: RTP(29): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1270, ssrc=BCA
May 23 14:08:20.117: RTP(63880): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD200, ssrc=1959A317
May 23 14:08:20.129: RTP(30): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1310, ssrc=BCA
May 23 14:08:20.137: RTP(63881): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD2A0, ssrc=1959A317
May 23 14:08:20.153: RTP(31): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=13B0, ssrc=BCA
May 23 14:08:20.157: RTP(63882): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD340, ssrc=1959A317
May 23 14:08:20.169: RTP(32): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1450, ssrc=BCA
May 23 14:08:20
cme-test#.181: RTP(63883): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD3E0, ssrc=1959A317
May 23 14:08:20.189: RTP(33): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14F0, ssrc=BCA
May 23 14:08:20.197: RTP(63884): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD480, ssrc=1959A317
May 23 14:08:20.209: RTP(34): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1590, ssrc=BCA
May 23 14:08:20.217: RTP(63885): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD520, ssrc=1959A317
May 23 14:08:20.229: RTP(35): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1630, ssrc=BCA
May 23 14:08:20.241: RTP(63886): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD5C0, ssrc=1959A317
May 23 14:08:20.249: RTP(36): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16D0, ssrc=BCA
May 23 14:08:20.257: RTP(63887): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD660, ssrc=1959A317
May 23 14:08:20.269: RTP(37): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1770, ssrc=BCA
May 23 14:08:20.281: RTP(63888): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD700, ssrc=1959A317
May 23 14:08:20.289: RTP(38): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1810, ssrc=BCA
May 23 14:08:20.297: RTP(63889): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD7A0, ssrc=1959A317
May 23 14:08:20.309: RTP(39): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=18B0, ssrc=BCA
May 23 14:08:20.317: RTP(63890): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD840, ssrc=1959A317
May 23 14:08:20.329: RTP(40): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1950, ssrc=BCA
May 23 14:08:20.337: RTP(63891): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD8E0, ssrc=1959A317
May 23 14:08:20.349: RTP(41): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=19F0, ssrc=BCA
May 23 14:08:20.357: RTP(63892): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCD980, ssrc=1959A317
May 23 14:08:20.369: RTP(42): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A90, ssrc=BCA
May 23 14:08:20.377: RTP(63893): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDA20, ssrc=1959A317
May 23 14:08:20.389: RTP(43): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1B30, ssrc=BCA
May 23 14:08:20.393: RTP(63894): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDAC0, ssrc=1959A317
May 23 14:08:20.409: RTP(44): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1BD0, ssrc=BCA
May 23 14:08:20.417: RTP(63895): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDB60, ssrc=1959A317
May 23 14:08:20.429: RTP(45): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1C70, ssrc=BCA
May 23 14:08:20.437: RTP(63896): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDC00, ssrc=1959A317
May 23 14:08:20.449: RTP(46): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1D10, ssrc=BCA
May 23 14:08:20.469: RTP(47): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1DB0, ssrc=BCA
May 23 14:08:20.473: RTP(63897): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDCA0, ssrc=1959A317
May 23 14:08:20.477: RTP(63898): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDD40, ssrc=1959A317
May 23 14:08:20.489: RTP(48): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E50, ssrc=BCA
May 23 14:08:20.501: RTP(63899): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDDE0, ssrc=1959A317
May 23 14:08:20.509: RTP(49): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EF0, ssrc=BCA
May 23 14:08:20.513: RTP(63900): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDE80, ssrc=1959A317
May 23 14:08:20.529: RTP(50): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F90, ssrc=BCA
May 23 14:08:20.537: RTP(63901): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDF20, ssrc=1959A317
May 23 14:08:20.549: RTP(51): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2030, ssrc=BCA
May 23 14:08:20.557: RTP(63902): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCDFC0, ssrc=1959A317
May 23 14:08:20.569: RTP(52): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=20D0, ssrc=BCA
May 23 14:08:20.577: RTP(63903): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE060, ssrc=1959A317
May 23 14:08:20.589: RTP(53): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2170, ssrc=BCA
May 23 14:08:20.597: RTP(63904): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE100, ssrc=1959A317
May 23 14:08:20.609: RTP(54): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2210, ssrc=BCA
May 23 14:08:20.613: RTP(63905): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE1A0, ssrc=1959A317
May 23 14:08:20.629: RTP(55): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22B0, ssrc=BCA
May 23 14:08:20.637: RTP(63906): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE240, ssrc=1959A317
May 23 14:08:20.649: RTP(56): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2350, ssrc=BCA
May 23 14:08:20.657: RTP(63907): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE2E0, ssrc=1959A317
May 23 14:08:20.669: RTP(57): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=23F0, ssrc=BCA
May 23 14:08:20.677: RTP(63908): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE380, ssrc=1959A317
May 23 14:08:20.690: RTP(58): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2490, ssrc=BCA
May 23 14:08:20.698: RTP(63909): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE420, ssrc=1959A317
May 23 14:08:20.710: RTP(59): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2530, ssrc=BCA
May 23 14:08:20.722: RTP(63910): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE4C0, ssrc=1959A317
May 23 14:08:20.730: RTP(60): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25D0, ssrc=BCA
May 23 14:08:20.734: RTP(63911): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE560, ssrc=1959A317
May 23 14:08:20.750: RTP(61): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2670, ssrc=BCA
May 23 14:08:20.758: RTP(63912): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE600, ssrc=1959A317
May 23 14:08:20.770: RTP(62): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2710, ssrc=BCA
May 23 14:08:20.778: RTP(63913): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE6A0, ssrc=1959A317
May 23 14:08:20.790: RTP(63): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=27B0, ssrc=BCA
May 23 14:08:20.798: RTP(63914): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE740, ssrc=1959A317
May 23 14:08:20.810: RTP(64): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2850, ssrc=BCA
May 23 14:08:20.818: RTP(63915): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE7E0, ssrc=1959A317
May 23 14:08:20.830: RTP(65): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28F0, ssrc=BCA
May 23 14:08:20.834: RTP(63916): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE880, ssrc=1959A317
May 23 14:08:20.850: RTP(66): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2990, ssrc=BCA
May 23 14:08:20.858: RTP(63917): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE920, ssrc=1959A317
May 23 14:08:20.870: RTP(67): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2A30, ssrc=BCA
May 23 14:08:20.878: RTP(63918): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCE9C0, ssrc=1959A317
May 23 14:08:20.890: RTP(68): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2AD0, ssrc=BCA
May 23 14:08:20.898: RTP(63919): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCEA60, ssrc=1959A317
May 23 14:08:20.902: voip_rtcp_stats_req: Calculated remaining disc time is 0 msec...Media Inactivity Criteria is 0
May 23 14:08:20.910: RTP(69): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2B70, ssrc=BCA
May 23 14:08:20.918: RTP(63920): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCEB00, ssrc=1959A317
May 23 14:08:20.930: RTP(70): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C10, ssrc=BCA
May 23 14:08:20.934: RTP(63921): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCEBA0, ssrc=1959A317
May 23 14:08:20.950: RTP(71): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2CB0, ssrc=BCA
May 23 14:08:20.954: RTP(63922): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCEC40, ssrc=1959A317
May 23 14:08:20.970: RTP(72): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D50, ssrc=BCA
May 23 14:08:20.982: RTP(63923): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCECE0, ssrc=1959A317
May 23 14:08:20.990: RTP(73): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2DF0, ssrc=BCA
May 23 14:08:20.998: RTP(63924): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCED80, ssrc=1959A317
May 23 14:08:21.010: RTP(74): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2E90, ssrc=BCA
May 23 14:08:21.018: RTP(63925): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCEE20, ssrc=1959A317
May 23 14:08:21.030: RTP(75): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2F30, ssrc=BCA
May 23 14:08:21.034: RTP(63926): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCEEC0, ssrc=1959A317
May 23 14:08:21.050: RTP(76): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2FD0, ssrc=BCA
May 23 14:08:21.054: RTP(63927): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCEF60, ssrc=1959A317
May 23 14:08:21.070: RTP(77): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3070, ssrc=BCA
May 23 14:08:21.078: RTP(63928): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF000, ssrc=1959A317
May 23 14:08:21.090: RTP(78): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3110, ssrc=BCA
May 23 14:08:21.098: RTP(63929): fs rx
cme-test#
cme-test# s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF0A0, ssrc=1959A317
May 23 14:08:21.110: RTP(79): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=31B0, ssrc=BCA
May 23 14:08:21.130: RTP(80): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3250, ssrc=BCA
May 23 14:08:21.142: RTP(63930): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF140, ssrc=1959A317
May 23 14:08:21.142: RTP(63931): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF1E0, ssrc=1959A317
May 23 14:08:21.150: RTP(81): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=32F0, ssrc=BCA
May 23 14:08:21.154: RTP(63932): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF280, ssrc=1959A317
May 23 14:08:21.170: RTP(82): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3390, ssrc=BCA
May 23 14:08:21.186: RTP(63933): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF320, ssrc=1959A317
May 23 14:08:21.190: RTP(83): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3430, ssrc=BCA
May 23 14:08:21.202: RTP(63934): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF3C0, ssrc=1959A317
May 23 14:08:21.210: RTP(84): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=34D0, ssrc=BCA
May 23 14:08:21.218: RTP(63935): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF460, ssrc=1959A317
May 23 14:08:21.230: RTP(85): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3570, ssrc=BCA
May 23 14:08:21.238: RTP(63936): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF500, ssrc=1959A317
May 23 14:08:21.250: RTP(86): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3610, ssrc=BCA
May 23 14:08:21.258: RTP(63937): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF5A0, ssrc=1959A317
May 23 14:08:21.270: RTP(87): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=36B0, ssrc=BCA
May 23 14:08:21.274: RTP(63938): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF640, ssrc=1959A317
May 23 14:08:21.290: RTP(88): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3750, ssrc=BCA
May 23 14:08:21.299: RTP(63939): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF6E0, ssrc=1959A317
May 23 14:08:21.311: RTP(89): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=37F0, ssrc=BCA
May 23 14:08:21.319: RTP(63940): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF780, ssrc=1959A317
May 23 14:08:21.331: RTP(90): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3890, ssrc=BCA
May 23 14:08:21.339: RTP(63941): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF820, ssrc=1959A317
May 23 14:08:21.351: RTP(91): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3930, ssrc=BCA
May 23 14:08:21.355: RTP(63942): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF8C0, ssrc=1959A317
May 23 14:08:21.371: RTP(92): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=39D0, ssrc=BCA
May 23 14:08:21.375: RTP(63943): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCF960, ssrc=1959A317
May 23 14:08:21.391: RTP(93): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3A70, ssrc=BCA
May 23 14:08:21.395: RTP(63944): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFA00, ssrc=1959A317
May 23 14:08:21.411: RTP(94): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3B10, ssrc=BCA
May 23 14:08:21.419: RTP(63945): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFAA0, ssrc=1959A317
May 23 14:08:21.431: RTP(95): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3BB0, ssrc=BCA
May 23 14:08:21.439: RTP(63946): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFB40, ssrc=1959A317
May 23 14:08:21.451: RTP(96): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3C50, ssrc=BCA
May 23 14:08:21.459: RTP(63947): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFBE0, ssrc=1959A317
May 23 14:08:21.471: RTP(97): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3CF0, ssrc=BCA
May 23 14:08:21.475: RTP(63948): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFC80, ssrc=1959A317
May 23 14:08:21.491: RTP(98): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3D90, ssrc=BCA
May 23 14:08:21.495: RTP(63949): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFD20, ssrc=1959A317
May 23 14:08:21.511: RTP(99): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3E30, ssrc=BCA
May 23 14:08:21.515: RTP(63950): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFDC0, ssrc=1959A317
May 23 14:08:21.531: RTP(100): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3ED0, ssrc=BCA
May 23 14:08:21.539: RTP(63951): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFE60, ssrc=1959A317
May 23 14:08:21.551: RTP(101): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=3F70, ssrc=BCA
May 23 14:08:21.559: RTP(63952): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFF00, ssrc=1959A317
May 23 14:08:21.571: RTP(102): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4010, ssrc=BCA
May 23 14:08:21.575: RTP(63953): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFCFFA0, ssrc=1959A317
May 23 14:08:21.591: RTP(103): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=40B0, ssrc=BCA
May 23 14:08:21.595: RTP(63954): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0040, ssrc=1959A317
May 23 14:08:21.611: RTP(104): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4150, ssrc=BCA
May 23 14:08:21.615: RTP(63955): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD00E0, ssrc=1959A317
May 23 14:08:21.631: RTP(105): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=41F0, ssrc=BCA
May 23 14:08:21.639: RTP(63956): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0180, ssrc=1959A317
May 23 14:08:21.651: RTP(106): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4290, ssrc=BCA
May 23 14:08:21.659: RTP(63957): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0220, ssrc=1959A317
May 23 14:08:21.671: RTP(107): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4330, ssrc=BCA
May 23 14:08:21.683: RTP(63958): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD02C0, ssrc=1959A317
May 23 14:08:21.691: RTP(108): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=43D0, ssrc=BCA
May 23 14:08:21.699: RTP(63959): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0360, ssrc=1959A317
May 23 14:08:21.711: RTP(109): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4470, ssrc=BCA
May 23 14:08:21.715: RTP(63960): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0400, ssrc=1959A317
May 23 14:08:21.731: RTP(110): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4510, ssrc=BCA
May 23 14:08:21.735: RTP(63961): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD04A0, ssrc=1959A317
May 23 14:08:21.751: RTP(111): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=45B0, ssrc=BCA
May 23 14:08:21.759: RTP(63962): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0540, ssrc=1959A317
May 23 14:08:21.771: RTP(112): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4650, ssrc=BCA
May 23 14:08:21.779: RTP(63963): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD05E0, ssrc=1959A317
May 23 14:08:21.791: RTP(113): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=46F0, ssrc=BCA
May 23 14:08:21.799: RTP(63964): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0680, ssrc=1959A317
May 23 14:08:21.811: RTP(114): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4790, ssrc=BCA
May 23 14:08:21.815: RTP(63965): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0720, ssrc=1959A317
May 23 14:08:21.831: RTP(115): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4830, ssrc=BCA
May 23 14:08:21.835: RTP(63966): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD07C0, ssrc=1959A317
May 23 14:08:21.851: RTP(116): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=48D0, ssrc=BCA
May 23 14:08:21.855: RTP(63967): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0860, ssrc=1959A317
May 23 14:08:21.871: RTP(117): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4970, ssrc=BCA
May 23 14:08:21.879: RTP(63968): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0900, ssrc=1959A317
May 23 14:08:21.891: RTP(118): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4A10, ssrc=BCA
May 23 14:08:21.900: RTP(63969): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD09A0, ssrc=1959A317
May 23 14:08:21.912: RTP(119): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4AB0, ssrc=BCA
May 23 14:08:21.920: RTP(63970): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0A40, ssrc=1959A317
May 23 14:08:21.932: RTP(120): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4B50, ssrc=BCA
May 23 14:08:21.936: RTP(63971): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0AE0, ssrc=1959A317
May 23 14:08:21.952: RTP(121): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4BF0, ssrc=BCA
May 23 14:08:21.956: RTP(63972): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0B80, ssrc=1959A317
May 23 14:08:21.972: RTP(122): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4C90, ssrc=BCA
May 23 14:08:21.980: RTP(63973): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0C20, ssrc=1959A317
May 23 14:08:21.992: RTP(123): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4D30, ssrc=BCA
May 23 14:08:22.633: RTP(63974): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0CC0, ssrc=1959A317
May 23 14:08:22.633: RTP(63975): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0D60, ssrc=1959A317
May 23 14:08:22.633: RTP(63976): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0E00, ssrc=1959A317
May 23 14:08:22
cme-test#
cme-test#
cme-test#
cme-test#.637: RTP(63977): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0EA0, ssrc=1959A317
May 23 14:08:22.637: RTP(63978): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0F40, ssrc=1959A317
May 23 14:08:22.637: RTP(63979): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD0FE0, ssrc=1959A317
May 23 14:08:22.641: RTP(63980): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1080, ssrc=1959A317
May 23 14:08:22.641: RTP(63981): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1120, ssrc=1959A317
May 23 14:08:22.641: RTP(63982): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD11C0, ssrc=1959A317
May 23 14:08:22.641: RTP(63983): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1260, ssrc=1959A317
May 23 14:08:22.645: RTP(63984): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1300, ssrc=1959A317
May 23 14:08:22.645: RTP(63985): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD13A0, ssrc=1959A317
May 23 14:08:22.645: RTP(63986): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1440, ssrc=1959A317
May 23 14:08:22.649: RTP(63987): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD14E0, ssrc=1959A317
May 23 14:08:22.649: RTP(63988): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1580, ssrc=1959A317
May 23 14:08:22.649: RTP(63989): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1620, ssrc=1959A317
May 23 14:08:22.649: RTP(63990): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD16C0, ssrc=1959A317
May 23 14:08:22.653: RTP(63991): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1760, ssrc=1959A317
May 23 14:08:22.653: RTP(63992): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1800, ssrc=1959A317
May 23 14:08:22.653: RTP(63993): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD18A0, ssrc=1959A317
May 23 14:08:22.657: RTP(63994): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1940, ssrc=1959A317
May 23 14:08:22.657: RTP(63995): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD19E0, ssrc=1959A317
May 23 14:08:22.657: RTP(63996): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1A80, ssrc=1959A317
May 23 14:08:22.657: RTP(63997): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1B20, ssrc=1959A317
May 23 14:08:22.661: RTP(63998): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1BC0, ssrc=1959A317
May 23 14:08:22.661: RTP(63999): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1C60, ssrc=1959A317
May 23 14:08:22.661: RTP(64000): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1D00, ssrc=1959A317
May 23 14:08:22.661: RTP(64001): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1DA0, ssrc=1959A317
May 23 14:08:22.665: RTP(64002): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1E40, ssrc=1959A317
May 23 14:08:22.665: RTP(64003): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1EE0, ssrc=1959A317
May 23 14:08:22.665: RTP(64004): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD1F80, ssrc=1959A317
May 23 14:08:22.669: RTP(64005): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2020, ssrc=1959A317
May 23 14:08:22.669: RTP(64006): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD20C0, ssrc=1959A317
May 23 14:08:22.669: RTP(64007): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2160, ssrc=1959A317
May 23 14:08:22.669: RTP(124): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4DD0, ssrc=BCA
May 23 14:08:22.673: RTP(125): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4E70, ssrc=BCA
May 23 14:08:22.673: RTP(126): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4F10, ssrc=BCA
May 23 14:08:22.673: RTP(127): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=4FB0, ssrc=BCA
May 23 14:08:22.677: RTP(128): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5050, ssrc=BCA
May 23 14:08:22.677: RTP(129): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=50F0, ssrc=BCA
May 23 14:08:22.677: RTP(130): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5190, ssrc=BCA
May 23 14:08:22.677: RTP(131): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5230, ssrc=BCA
May 23 14:08:22.681: RTP(132): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=52D0, ssrc=BCA
May 23 14:08:22.681: RTP(133): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5370, ssrc=BCA
May 23 14:08:22.681: RTP(134): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5410, ssrc=BCA
May 23 14:08:22.681: RTP(135): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=54B0, ssrc=BCA
May 23 14:08:22.685: RTP(136): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5550, ssrc=BCA
May 23 14:08:22.685: RTP(137): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=55F0, ssrc=BCA
May 23 14:08:22.685: RTP(138): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5690, ssrc=BCA
May 23 14:08:22.689: RTP(139): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5730, ssrc=BCA
May 23 14:08:22.689: RTP(140): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=57D0, ssrc=BCA
May 23 14:08:22.689: RTP(141): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5870, ssrc=BCA
May 23 14:08:22.689: RTP(142): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5910, ssrc=BCA
May 23 14:08:22.693: RTP(143): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=59B0, ssrc=BCA
May 23 14:08:22.693: RTP(144): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5A50, ssrc=BCA
May 23 14:08:22.693: RTP(145): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5AF0, ssrc=BCA
May 23 14:08:22.693: RTP(146): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5B90, ssrc=BCA
May 23 14:08:22.697: RTP(147): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5C30, ssrc=BCA
May 23 14:08:22.697: RTP(148): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5CD0, ssrc=BCA
May 23 14:08:22.697: RTP(149): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5D70, ssrc=BCA
May 23 14:08:22.701: RTP(150): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5E10, ssrc=BCA
May 23 14:08:22.701: RTP(151): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5EB0, ssrc=BCA
May 23 14:08:22.701: RTP(152): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5F50, ssrc=BCA
May 23 14:08:22.701: RTP(153): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=5FF0, ssrc=BCA
May 23 14:08:22.705: RTP(154): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6090, ssrc=BCA
May 23 14:08:22.705: RTP(155): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6130, ssrc=BCA
May 23 14:08:22.705: RTP(156): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=61D0, ssrc=BCA
May 23 14:08:22.705: RTP(157): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6270, ssrc=BCA
May 23 14:08:22.709: RTP(158): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6310, ssrc=BCA
May 23 14:08:22.709: RTP(64008): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2200, ssrc=1959A317
May 23 14:08:22.709: RTP(64009): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD22A0, ssrc=1959A317
May 23 14:08:22.713: RTP(159): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=63B0, ssrc=BCA
May 23 14:08:23.274: RTP(64010): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2340, ssrc=1959A317
May 23 14:08:23.278: RTP(64011): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD23E0, ssrc=1959A317
May 23 14:08:23.278: RTP(64012): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2480, ssrc=1959A317
May 23 14:08:23.278: RTP(64013): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2520, ssrc=1959A317
May 23 14:08:23.282: RTP(64014): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD25C0, ssrc=1959A317
May 23 14:08:23.282: RTP(64015): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2660, ssrc=1959A317
May 23 14:08:23.282: RTP(64016): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2700, ssrc=1959A317
May 23 14:08:23.282: RTP(64017): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD27A0, ssrc=1959A317
May 23 14:08:23.286: RTP(64018): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2840, ssrc=1959A317
May 23 14:08:23.286: RTP(64019): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD28E0, ssrc=1959A317
May 23 14:08:23.286: RTP(64020): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2980, ssrc=1959A317
May 23 14:08:23.290: RTP(64021): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2A20, ssrc=1959A317
May 23 14:08:23.290: RTP(64022): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2AC0, ssrc=1959A317
May 23 14:08:23.290: RTP(64023): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2B60, ssrc=1959A317
May 23 14:08:23.294: RTP(64024): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2C00, ssrc=1959A317
May 23 14:08:23.294: RTP(64025): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2CA0, ssrc=1959A317
May 23 14:08:23.294: RTP(64026): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2D40, ssrc=1959A317
May 23 14:08:23.294: RTP(64027): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2DE0, ssrc=1959A317
May 23 14:08:23.298: RTP(64028): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2E80, ssrc=1959A317
May 23 14:08:23.298: RTP(64029): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2F20, ssrc=1959A317
May 23 14:08:23.298: RTP(64030): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD2FC0, ssrc=1959A317
May 23 14:08:23.298: RTP(64031): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3060, ssrc=1959A317
May 23 14:08:23.302: RTP(64032): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3100, ssrc=1959A317
May 23 14:08:23.302: RTP(64033): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD31A0, ssrc=1959A317
May 23 14:08:23.302: RTP(64034): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3240, ssrc=1959A317
May 23 14:08:23.306: RTP(64035): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD32E0, ssrc=1959A317
May 23 14:08:23.306: RTP(64036): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3380, ssrc=1959A317
May 23 14:08:23.306: RTP(64037): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3420, ssrc=1959A317
May 23 14:08:23.306: RTP(64038): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD34C0, ssrc=1959A317
May 23 14:08:23.310: RTP(64039): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3560, ssrc=1959A317
May 23 14:08:23.310: RTP(160): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6450, ssrc=BCA
May 23 14:08:23.310: RTP(161): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=64F0, ssrc=BCA
May 23 14:08:23.314: RTP(162): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6590, ssrc=BCA
May 23 14:08:23.314: RTP(163): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6630, ssrc=BCA
May 23 14:08:23.314: RTP(164): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=66D0, ssrc=BCA
May 23 14:08:23.314: RTP(165): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6770, ssrc=BCA
May 23 14:08:23.318: RTP(166): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6810, ssrc=BCA
May 23 14:08:23.318: RTP(167): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=68B0, ssrc=BCA
May 23 14:08:23.318: RTP(168): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6950, ssrc=BCA
May 23 14:08:23.318: RTP(169): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=69F0, ssrc=BCA
May 23 14:08:23.322: RTP(170): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6A90, ssrc=BCA
May 23 14:08:23.322: RTP(171): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6B30, ssrc=BCA
May 23 14:08:23.322: RTP(172): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6BD0, ssrc=BCA
May 23 14:08:23.322: RTP(173): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6C70, ssrc=BCA
May 23 14:08:23.326: RTP(174): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6D10, ssrc=BCA
May 23 14:08:23.326: RTP(175): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6DB0, ssrc=BCA
May 23 14:08:23.326: RTP(176): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6E50, ssrc=BCA
May 23 14:08:23.326: RTP(177): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6EF0, ssrc=BCA
May 23 14:08:23.330: RTP(178): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=6F90, ssrc=BCA
May 23 14:08:23.330: RTP(179): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7030, ssrc=BCA
May 23 14:08:23.330: RTP(180): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=70D0, ssrc=BCA
May 23 14:08:23.330: RTP(181): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7170, ssrc=BCA
May 23 14:08:23.334: RTP(182): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7210, ssrc=BCA
May 23 14:08:23.334: RTP(183): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=72B0, ssrc=BCA
May 23 14:08:23.334: RTP(184): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7350, ssrc=BCA
May 23 14:08:23.338: RTP(185): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=73F0, ssrc=BCA
May 23 14:08:23.338: RTP(186): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7490, ssrc=BCA
May 23 14:08:23.338: RTP(187): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7530, ssrc=BCA
May 23 14:08:23.338: RTP(188): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=75D0, ssrc=BCA
May 23 14:08:23.342: RTP(189): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7670, ssrc=BCA
May 23 14:08:23.342: RTP(190): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7710, ssrc=BCA
May 23 14:08:23.342: RTP(64040): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3600, ssrc=1959A317
May 23 14:08:23.350: RTP(64041): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD36A0, ssrc=1959A317
May 23 14:08:23.350: RTP(191): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=77B0, ssrc=BCA
May 23 14:08:23.358: RTP(64042): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3740, ssrc=1959A317
May 23 14:08:23.370: RTP(192): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7850, ssrc=BCA
May 23 14:08:23.378: RTP(64043): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD37E0, ssrc=1959A317
May 23 14:08:23.390: RTP(193): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=78F0, ssrc=BCA
May 23 14:08:23.406: RTP(64044): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3880, ssrc=1959A317
May 23 14:08:23.410: RTP(194): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7990, ssrc=BCA
May 23 14:08:23.418: RTP(64045): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3920, ssrc=1959A317
May 23 14:08:23.430: RTP(195): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7A30, ssrc=BCA
May 23 14:08:23.434: RTP(64046): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD39C0, ssrc=1959A317
May 23 14:08:23.450: RTP(196): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7AD0, ssrc=BCA
May 23 14:08:23.458: RTP(64047): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3A60, ssrc=1959A317
May 23 14:08:23.470: RTP(197): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7B70, ssrc=BCA
May 23 14:08:23.478: RTP(64048): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3B00, ssrc=1959A317
May 23 14:08:23.490: RTP(198): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7C10, ssrc=BCA
May 23 14:08:23.498: RTP(64049): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3BA0, ssrc=1959A317
May 23 14:08:23.510: RTP(199): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7CB0, ssrc=BCA
May 23 14:08:23.518: RTP(64050): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3C40, ssrc=1959A317
May 23 14:08:23.530: RTP(200): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7D50, ssrc=BCA
May 23 14:08:23.534: RTP(64051): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3CE0, ssrc=1959A317
May 23 14:08:23.550: RTP(201): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7DF0, ssrc=BCA
May 23 14:08:23.558: RTP(64052): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3D80, ssrc=1959A317
May 23 14:08:23.570: RTP(202): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7E90, ssrc=BCA
May 23 14:08:23.578: RTP(64053): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3E20, ssrc=1959A317
May 23 14:08:23.590: RTP(203): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7F30, ssrc=BCA
May 23 14:08:23.598: RTP(64054): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3EC0, ssrc=1959A317
May 23 14:08:23.610: RTP(204): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=7FD0, ssrc=BCA
May 23 14:08:23.618: RTP(64055): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD3F60, ssrc=1959A317
May 23 14:08:23.618: RTP(205): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=13, ts=8070, ssrc=BCA
May 23 14:08:23.634: RTP(64056): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4000, ssrc=1959A317
May 23 14:08:23.654: RTP(64057): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD40A0, ssrc=1959A317
May 23 14:08:23.662: voip_rtcp_stats_req: Calculated remaining disc time is 0 msec...Media Inactivity Criteria is 0
May 23 14:08:23.662: voip_rtcp_stats_req: Calculated remaining disc time is 0 msec...Media Inactivity Criteria is 0
May 23 14:08:23.678: RTP(64058): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4140, ssrc=1959A317
May 23 14:08:23.799: RTP(64059): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD41E0, ssrc=1959A317
May 23 14:08:23.799: RTP(64060): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4280, ssrc=1959A317
May 23 14:08:23.799: RTP(64061): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4320, ssrc=1959A317
May 23 14:08:23.803: RTP(64062): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD43C0, ssrc=1959A317
May 23 14:08:23.803: RTP(64063): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4460, ssrc=1959A317
May 23 14:08:23.803: RTP(64064): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4500, ssrc=1959A317
May 23 14:08:24.612: RTP(64065): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD45A0, ssrc=1959A317
May 23 14:08:24.616: RTP(64066): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4640, ssrc=1959A317
May 23 14:08:24.616: RTP(64067): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD46E0, ssrc=1959A317
May 23 14:08:24.616: RTP(64068): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4780, ssrc=1959A317
May 23 14:08:24.620: RTP(64069): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4820, ssrc=1959A317
May 23 14:08:24.620: RTP(64070): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD48C0, ssrc=1959A317
May 23 14:08:24.620: RTP(64071): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4960, ssrc=1959A317
May 23 14:08:24.624: RTP(64072): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4A00, ssrc=1959A317
May 23 14:08:24.624: RTP(64073): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4AA0, ssrc=1959A317
May 23 14:08:24.624: RTP(64074): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4B40, ssrc=1959A317
May 23 14:08:24.624: RTP(64075): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4BE0, ssrc=1959A317
May 23 14:08:24.628: RTP(64076): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4C80, ssrc=1959A317
May 23 14:08:24.628: RTP(64077): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4D20, ssrc=1959A317
May 23 14:08:24.628: RTP(64078): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4DC0, ssrc=1959A317
May 23 14:08:24.628: RTP(64079): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4E60, ssrc=1959A317
May 23 14:08:24.632: RTP(64080): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4F00, ssrc=1959A317
May 23 14:08:24.632: RTP(64081): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD4FA0, ssrc=1959A317
May 23 14:08:24.632: RTP(64082): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5040, ssrc=1959A317
May 23 14:08:24.636: RTP(64083): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD50E0, ssrc=1959A317
May 23 14:08:24.636: RTP(64084): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5180, ssrc=1959A317
May 23 14:08:24.636: RTP(64085): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5220, ssrc=1959A317
May 23 14:08:24.636: RTP(64086): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD52C0, ssrc=1959A317
May 23 14:08:24.640: RTP(64087): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5360, ssrc=1959A317
May 23 14:08:24.640: RTP(64088): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5400, ssrc=1959A317
May 23 14:08:24.640: RTP(64089): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD54A0, ssrc=1959A317
May 23 14:08:24.644: RTP(64090): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5540, ssrc=1959A317
May 23 14:08:24.644: RTP(64091): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD55E0, ssrc=1959A317
May 23 14:08:24.644: RTP(64092): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5680, ssrc=1959A317
May 23 14:08:24.644: RTP(64093): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5720, ssrc=1959A317
May 23 14:08:24.648: RTP(64094): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD57C0, ssrc=1959A317
May 23 14:08:24.648: RTP(64095): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5860, ssrc=1959A317
May 23 14:08:24.648: RTP(64096): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5900, ssrc=1959A317
May 23 14:08:24.652: RTP(64097): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD59A0, ssrc=1959A317
May 23 14:08:24.652: RTP(64098): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5A40, ssrc=1959A317
May 23 14:08:24.652: RTP(64099): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5AE0, ssrc=1959A317
May 23 14:08:24.652: RTP(64100): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5B80, ssrc=1959A317
May 23 14:08:24.656: RTP(64101): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5C20, ssrc=1959A317
May 23 14:08:24.656: RTP(64102): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5CC0, ssrc=1959A317
May 23 14:08:24.656: RTP(64103): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5D60, ssrc=1959A317
May 23 14:08:24.660: RTP(64104): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5E00, ssrc=1959A317
May 23 14:08:24.660: RTP(64105): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5EA0, ssrc=1959A317
May 23 14:08:24.660: RTP(64106): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5F40, ssrc=1959A317
May 23 14:08:24.660: RTP(64107): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD5FE0, ssrc=1959A317
May 23 14:08:24.664: RTP(206): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=98D0, ssrc=BCA
May 23 14:08:24.664: RTP(207): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9970, ssrc=BCA
May 23 14:08:24.664: RTP(208): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9A10, ssrc=BCA
May 23 14:08:24.668: RTP(209): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9AB0, ssrc=BCA
May 23 14:08:24.668: RTP(210): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9B50, ssrc=BCA
May 23 14:08:24.668: RTP(211): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9BF0, ssrc=BCA
May 23 14:08:24.668: RTP(212): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9C90, ssrc=BCA
May 23 14:08:24.672: RTP(213): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9D30, ssrc=BCA
May 23 14:08:24.672: RTP(214): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9DD0, ssrc=BCA
May 23 14:08:24.672: RTP(215): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9E70, ssrc=BCA
May 23 14:08:24.672: RTP(216): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9F10, ssrc=BCA
May 23 14:08:24.676: RTP(217): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=9FB0, ssrc=BCA
May 23 14:08:24.676: RTP(218): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A050, ssrc=BCA
May 23 14:08:24.676: RTP(219): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A0F0, ssrc=BCA
May 23 14:08:24.680: RTP(64108): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6080, ssrc=1959A317
May 23 14:08:24.692: RTP(220): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A190, ssrc=BCA
May 23 14:08:24.700: RTP(64109): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6120, ssrc=1959A317
May 23 14:08:24.712: RTP(221): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A230, ssrc=BCA
May 23 14:08:24.716: RTP(64110): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD61C0, ssrc=1959A317
May 23 14:08:24.732: RTP(222): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A2D0, ssrc=BCA
May 23 14:08:24.736: RTP(64111): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6260, ssrc=1959A317
May 23 14:08:24.752: RTP(223): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A370, ssrc=BCA
May 23 14:08:24.756: RTP(64112): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6300, ssrc=1959A317
May 23 14:08:24.772: RTP(224): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A410, ssrc=BCA
May 23 14:08:24.776: RTP(64113): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD63A0, ssrc=1959A317
May 23 14:08:24.792: RTP(225): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A4B0, ssrc=BCA
May 23 14:08:24.800: RTP(64114): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6440, ssrc=1959A317
May 23 14:08:24.812: RTP(226): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A550, ssrc=BCA
May 23 14:08:24.820: RTP(64115): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD64E0, ssrc=1959A317
May 23 14:08:24.832: RTP(227): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A5F0, ssrc=BCA
May 23 14:08:24.836: RTP(64116): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6580, ssrc=1959A317
May 23 14:08:24.852: RTP(228): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A690, ssrc=BCA
May 23 14:08:24.856: RTP(64117): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6620, ssrc=1959A317
May 23 14:08:24.872: RTP(229): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A730, ssrc=BCA
May 23 14:08:24.876: RTP(64118): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD66C0, ssrc=1959A317
May 23 14:08:24.892: RTP(230): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A7D0, ssrc=BCA
May 23 14:08:24.896: RTP(64119): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6760, ssrc=1959A317
May 23 14:08:24.908: RTP(231): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A870, ssrc=BCA
May 23 14:08:24.920: RTP(64120): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6800, ssrc=1959A317
May 23 14:08:24.928: RTP(232): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A910, ssrc=BCA
May 23 14:08:24.941: RTP(64121): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD68A0, ssrc=1959A317
May 23 14:08:24.953: RTP(233): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=A9B0, ssrc=BCA
May 23 14:08:24.957: RTP(64122): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6940, ssrc=1959A317
May 23 14:08:25.293: RTP(234): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AA50, ssrc=BCA
May 23 14:08:25.293: RTP(235): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AAF0, ssrc=BCA
May 23 14:08:25.293: RTP(236): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AB90, ssrc=BCA
May 23 14:08:25.293: RTP(237): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AC30, ssrc=BCA
May 23 14:08:25.297: RTP(238): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=ACD0, ssrc=BCA
May 23 14:08:25.297: RTP(239): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AD70, ssrc=BCA
May 23 14:08:25.297: RTP(240): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AE10, ssrc=BCA
May 23 14:08:25.297: RTP(241): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AEB0, ssrc=BCA
May 23 14:08:25.301: RTP(242): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AF50, ssrc=BCA
May 23 14:08:25.301: RTP(243): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=AFF0, ssrc=BCA
May 23 14:08:25.301: RTP(244): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B090, ssrc=BCA
May 23 14:08:25.305: RTP(245): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B130, ssrc=BCA
May 23 14:08:25.305: RTP(246): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B1D0, ssrc=BCA
May 23 14:08:25.305: RTP(247): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B270, ssrc=BCA
May 23 14:08:25.305: RTP(248): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B310, ssrc=BCA
May 23 14:08:25.309: RTP(249): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B3B0, ssrc=BCA
May 23 14:08:25.309: RTP(250): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B450, ssrc=BCA
May 23 14:08:25.309: RTP(64123): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD69E0, ssrc=1959A317
May 23 14:08:25.313: RTP(64124): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6A80, ssrc=1959A317
May 23 14:08:25.313: RTP(64125): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6B20, ssrc=1959A317
May 23 14:08:25.313: RTP(64126): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6BC0, ssrc=1959A317
May 23 14:08:25.313: RTP(64127): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6C60, ssrc=1959A317
May 23 14:08:25.317: RTP(64128): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6D00, ssrc=1959A317
May 23 14:08:25.317: RTP(64129): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6DA0, ssrc=1959A317
May 23 14:08:25.317: RTP(64130): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6E40, ssrc=1959A317
May 23 14:08:25.317: RTP(64131): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6EE0, ssrc=1959A317
May 23 14:08:25.321: RTP(64132): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD6F80, ssrc=1959A317
May 23 14:08:25.321: RTP(64133): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7020, ssrc=1959A317
May 23 14:08:25.321: RTP(64134): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD70C0, ssrc=1959A317
May 23 14:08:25.325: RTP(64135): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7160, ssrc=1959A317
May 23 14:08:25.325: RTP(64136): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7200, ssrc=1959A317
May 23 14:08:25.325: RTP(64137): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD72A0, ssrc=1959A317
May 23 14:08:25.325: RTP(64138): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7340, ssrc=1959A317
May 23 14:08:25.329: RTP(64139): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD73E0, ssrc=1959A317
May 23 14:08:25.329: RTP(64140): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7480, ssrc=1959A317
May 23 14:08:25.329: RTP(251): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B4F0, ssrc=BCA
May 23 14:08:25.333: RTP(252): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B590, ssrc=BCA
May 23 14:08:26.066: RTP(64141): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7520, ssrc=1959A317
May 23 14:08:26.066: RTP(64142): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD75C0, ssrc=1959A317
May 23 14:08:26.066: RTP(64143): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7660, ssrc=1959A317
May 23 14:08:26.070: RTP(64144): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7700, ssrc=1959A317
May 23 14:08:26.070: RTP(64145): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD77A0, ssrc=1959A317
May 23 14:08:26.070: RTP(64146): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7840, ssrc=1959A317
May 23 14:08:26.074: RTP(64147): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD78E0, ssrc=1959A317
May 23 14:08:26.074: RTP(64148): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7980, ssrc=1959A317
May 23 14:08:26.074: RTP(64149): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7A20, ssrc=1959A317
May 23 14:08:26.074: RTP(64150): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7AC0, ssrc=1959A317
May 23 14:08:26.078: RTP(64151): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7B60, ssrc=1959A317
May 23 14:08:26.078: RTP(64152): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7C00, ssrc=1959A317
May 23 14:08:26.078: RTP(64153): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7CA0, ssrc=1959A317
May 23 14:08:26.082: RTP(64154): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7D40, ssrc=1959A317
May 23 14:08:26.082: RTP(64155): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7DE0, ssrc=1959A317
May 23 14:08:26.082: RTP(64156): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7E80, ssrc=1959A317
May 23 14:08:26.082: RTP(64157): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7F20, ssrc=1959A317
May 23 14:08:26.086: RTP(64158): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD7FC0, ssrc=1959A317
May 23 14:08:26.086: RTP(64159): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8060, ssrc=1959A317
May 23 14:08:26.086: RTP(64160): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8100, ssrc=1959A317
May 23 14:08:26.090: RTP(64161): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD81A0, ssrc=1959A317
May 23 14:08:26.090: RTP(64162): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8240, ssrc=1959A317
May 23 14:08:26.090: RTP(64163): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD82E0, ssrc=1959A317
May 23 14:08:26.090: RTP(64164): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8380, ssrc=1959A317
May 23 14:08:26.094: RTP(64165): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8420, ssrc=1959A317
May 23 14:08:26.094: RTP(64166): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD84C0, ssrc=1959A317
May 23 14:08:26.094: RTP(64167): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8560, ssrc=1959A317
May 23 14:08:26.098: RTP(64168): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8600, ssrc=1959A317
May 23 14:08:26.098: RTP(64169): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD86A0, ssrc=1959A317
May 23 14:08:26.098: RTP(64170): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8740, ssrc=1959A317
May 23 14:08:26.098: RTP(64171): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD87E0, ssrc=1959A317
May 23 14:08:26.102: RTP(64172): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8880, ssrc=1959A317
May 23 14:08:26.102: RTP(64173): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8920, ssrc=1959A317
May 23 14:08:26.102: RTP(64174): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD89C0, ssrc=1959A317
May 23 14:08:26.106: RTP(64175): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8A60, ssrc=1959A317
May 23 14:08:26.106: RTP(64176): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8B00, ssrc=1959A317
May 23 14:08:26.106: RTP(64177): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8BA0, ssrc=1959A317
May 23 14:08:26.106: RTP(64178): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8C40, ssrc=1959A317
May 23 14:08:26.110: RTP(64179): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8CE0, ssrc=1959A317
May 23 14:08:26.110: RTP(253): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B630, ssrc=BCA
May 23 14:08:26.110: RTP(254): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B6D0, ssrc=BCA
May 23 14:08:26.114: RTP(255): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B770, ssrc=BCA
May 23 14:08:26.114: RTP(256): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B810, ssrc=BCA
May 23 14:08:26.114: RTP(257): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B8B0, ssrc=BCA
May 23 14:08:26.114: RTP(258): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B950, ssrc=BCA
May 23 14:08:26.118: RTP(259): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=B9F0, ssrc=BCA
May 23 14:08:26.118: RTP(260): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=BA90, ssrc=BCA
May 23 14:08:26.118: RTP(261): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=BB30, ssrc=BCA
May 23 14:08:26.118: RTP(262): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=BBD0, ssrc=BCA
May 23 14:08:26.122: RTP(263): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=BC70, ssrc=BCA
May 23 14:08:26.122: RTP(264): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=BD10, ssrc=BCA
May 23 14:08:26.122: RTP(265): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=13, ts=BDB0, ssrc=BCA
May 23 14:08:26.122: RTP(266): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C030, ssrc=BCA
May 23 14:08:26.126: RTP(267): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C0D0, ssrc=BCA
May 23 14:08:26.126: RTP(268): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C170, ssrc=BCA
May 23 14:08:26.126: RTP(269): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C210, ssrc=BCA
May 23 14:08:26.130: RTP(270): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C2B0, ssrc=BCA
May 23 14:08:26.130: RTP(271): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C350, ssrc=BCA
May 23 14:08:26.130: RTP(272): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C3F0, ssrc=BCA
May 23 14:08:26.130: RTP(273): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C490, ssrc=BCA
May 23 14:08:26.134: RTP(274): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C530, ssrc=BCA
May 23 14:08:26.134: RTP(275): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C5D0, ssrc=BCA
May 23 14:08:26.134: RTP(276): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C670, ssrc=BCA
May 23 14:08:26.134: RTP(277): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C710, ssrc=BCA
May 23 14:08:26.138: RTP(278): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C7B0, ssrc=BCA
May 23 14:08:26.138: RTP(279): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C850, ssrc=BCA
May 23 14:08:26.138: RTP(280): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C8F0, ssrc=BCA
May 23 14:08:26.142: RTP(281): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=C990, ssrc=BCA
May 23 14:08:26.142: RTP(282): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CA30, ssrc=BCA
May 23 14:08:26.142: RTP(283): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CAD0, ssrc=BCA
May 23 14:08:26.142: RTP(284): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CB70, ssrc=BCA
May 23 14:08:26.147: RTP(285): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CC10, ssrc=BCA
May 23 14:08:26.147: RTP(286): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CCB0, ssrc=BCA
May 23 14:08:26.147: RTP(287): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CD50, ssrc=BCA
May 23 14:08:26.147: RTP(288): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CDF0, ssrc=BCA
May 23 14:08:26.151: RTP(289): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CE90, ssrc=BCA
May 23 14:08:26.151: RTP(64180): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8D80, ssrc=1959A317
May 23 14:08:26.151: RTP(64181): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8E20, ssrc=1959A317
May 23 14:08:26.151: RTP(290): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CF30, ssrc=BCA
May 23 14:08:26.155: voip_rtcp_stats_req: Calculated remaining disc time is 0 msec...Media Inactivity Criteria is 0
May 23 14:08:26.159: RTP(64182): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8EC0, ssrc=1959A317
May 23 14:08:26.171: RTP(291): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=CFD0, ssrc=BCA
May 23 14:08:26.179: RTP(64183): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD8F60, ssrc=1959A317
May 23 14:08:26.191: RTP(292): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D070, ssrc=BCA
May 23 14:08:26.199: RTP(64184): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9000, ssrc=1959A317
May 23 14:08:26.211: RTP(293): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D110, ssrc=BCA
May 23 14:08:26.219: RTP(64185): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD90A0, ssrc=1959A317
May 23 14:08:26.231: RTP(294): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D1B0, ssrc=BCA
May 23 14:08:26.235: RTP(64186): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9140, ssrc=1959A317
May 23 14:08:26.251: RTP(295): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D250, ssrc=BCA
May 23 14:08:26.255: RTP(64187): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD91E0, ssrc=1959A317
May 23 14:08:26.271: RTP(296): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D2F0, ssrc=BCA
May 23 14:08:26.279: RTP(64188): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9280, ssrc=1959A317
May 23 14:08:26.291: RTP(297): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D390, ssrc=BCA
May 23 14:08:26.299: RTP(64189): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9320, ssrc=1959A317
May 23 14:08:26.311: RTP(298): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D430, ssrc=BCA
May 23 14:08:26.319: RTP(64190): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD93C0, ssrc=1959A317
May 23 14:08:26.331: RTP(299): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D4D0, ssrc=BCA
May 23 14:08:26.335: RTP(64191): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9460, ssrc=1959A317
May 23 14:08:26.351: RTP(300): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D570, ssrc=BCA
May 23 14:08:26.355: RTP(64192): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9500, ssrc=1959A317
May 23 14:08:26.371: RTP(301): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D610, ssrc=BCA
May 23 14:08:26.375: RTP(64193): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD95A0, ssrc=1959A317
May 23 14:08:26.391: RTP(302): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D6B0, ssrc=BCA
May 23 14:08:26.399: RTP(64194): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9640, ssrc=1959A317
May 23 14:08:26.411: RTP(303): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D750, ssrc=BCA
May 23 14:08:26.419: RTP(64195): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD96E0, ssrc=1959A317
May 23 14:08:26.431: RTP(304): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D7F0, ssrc=BCA
May 23 14:08:26.435: RTP(64196): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9780, ssrc=1959A317
May 23 14:08:26.451: RTP(305): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D890, ssrc=BCA
May 23 14:08:26.735: RTP(64197): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9820, ssrc=1959A317
May 23 14:08:26.735: RTP(64198): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD98C0, ssrc=1959A317
May 23 14:08:26.735: RTP(64199): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9960, ssrc=1959A317
May 23 14:08:26.739: RTP(64200): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9A00, ssrc=1959A317
May 23 14:08:26.739: RTP(64201): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9AA0, ssrc=1959A317
May 23 14:08:26.739: RTP(64202): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9B40, ssrc=1959A317
May 23 14:08:26.743: RTP(64203): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9BE0, ssrc=1959A317
May 23 14:08:26.743: RTP(64204): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9C80, ssrc=1959A317
May 23 14:08:26.743: RTP(64205): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9D20, ssrc=1959A317
May 23 14:08:26.743: RTP(64206): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9DC0, ssrc=1959A317
May 23 14:08:26.747: RTP(64207): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9E60, ssrc=1959A317
May 23 14:08:26.747: RTP(64208): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9F00, ssrc=1959A317
May 23 14:08:26.747: RTP(64209): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFD9FA0, ssrc=1959A317
May 23 14:08:26.751: RTP(64210): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA040, ssrc=1959A317
May 23 14:08:26.751: RTP(64211): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA0E0, ssrc=1959A317
May 23 14:08:26.751: RTP(306): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D930, ssrc=BCA
May 23 14:08:26.751: RTP(307): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=D9D0, ssrc=BCA
May 23 14:08:26.756: RTP(308): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=DA70, ssrc=BCA
May 23 14:08:26.756: RTP(309): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=DB10, ssrc=BCA
May 23 14:08:26.756: RTP(310): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=DBB0, ssrc=BCA
May 23 14:08:26.760: RTP(311): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=DC50, ssrc=BCA
May 23 14:08:26.760: RTP(312): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=DCF0, ssrc=BCA
May 23 14:08:26.760: RTP(313): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=DD90, ssrc=BCA
May 23 14:08:26.760: RTP(314): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=DE30, ssrc=BCA
May 23 14:08:26.764: RTP(315): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=DED0, ssrc=BCA
May 23 14:08:26.764: RTP(316): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=DF70, ssrc=BCA
May 23 14:08:26.764: RTP(317): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E010, ssrc=BCA
May 23 14:08:26.764: RTP(318): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E0B0, ssrc=BCA
May 23 14:08:26.768: RTP(319): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E150, ssrc=BCA
May 23 14:08:26.768: RTP(320): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E1F0, ssrc=BCA
May 23 14:08:26.768: RTP(64212): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA180, ssrc=1959A317
May 23 14:08:27.541: RTP(321): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E290, ssrc=BCA
May 23 14:08:27.541: RTP(322): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E330, ssrc=BCA
May 23 14:08:27.541: RTP(323): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E3D0, ssrc=BCA
May 23 14:08:27.541: RTP(324): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E470, ssrc=BCA
May 23 14:08:27.545: RTP(325): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E510, ssrc=BCA
May 23 14:08:27.545: RTP(326): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E5B0, ssrc=BCA
May 23 14:08:27.545: RTP(327): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E650, ssrc=BCA
May 23 14:08:27.549: RTP(328): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E6F0, ssrc=BCA
May 23 14:08:27.549: RTP(329): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E790, ssrc=BCA
May 23 14:08:27.549: RTP(330): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E830, ssrc=BCA
May 23 14:08:27.549: RTP(331): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E8D0, ssrc=BCA
May 23 14:08:27.553: RTP(332): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=E970, ssrc=BCA
May 23 14:08:27.553: RTP(333): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EA10, ssrc=BCA
May 23 14:08:27.553: RTP(334): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EAB0, ssrc=BCA
May 23 14:08:27.553: RTP(335): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EB50, ssrc=BCA
May 23 14:08:27.557: RTP(336): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EBF0, ssrc=BCA
May 23 14:08:27.557: RTP(337): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EC90, ssrc=BCA
May 23 14:08:27.557: RTP(338): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=ED30, ssrc=BCA
May 23 14:08:27.557: RTP(339): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EDD0, ssrc=BCA
May 23 14:08:27.561: RTP(340): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EE70, ssrc=BCA
May 23 14:08:27.561: RTP(341): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EF10, ssrc=BCA
May 23 14:08:27.561: RTP(342): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=EFB0, ssrc=BCA
May 23 14:08:27.565: RTP(343): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F050, ssrc=BCA
May 23 14:08:27.565: RTP(344): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F0F0, ssrc=BCA
May 23 14:08:27.565: RTP(345): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F190, ssrc=BCA
May 23 14:08:27.565: RTP(346): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F230, ssrc=BCA
May 23 14:08:27.569: RTP(347): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F2D0, ssrc=BCA
May 23 14:08:27.569: RTP(348): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F370, ssrc=BCA
May 23 14:08:27.569: RTP(349): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F410, ssrc=BCA
May 23 14:08:27.569: RTP(350): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F4B0, ssrc=BCA
May 23 14:08:27.573: RTP(351): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F550, ssrc=BCA
May 23 14:08:27.573: RTP(352): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F5F0, ssrc=BCA
May 23 14:08:27.573: RTP(353): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F690, ssrc=BCA
May 23 14:08:27.577: RTP(354): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F730, ssrc=BCA
May 23 14:08:27.577: RTP(355): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F7D0, ssrc=BCA
May 23 14:08:27.577: RTP(356): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F870, ssrc=BCA
May 23 14:08:27.577: RTP(357): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F910, ssrc=BCA
May 23 14:08:27.581: RTP(358): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=F9B0, ssrc=BCA
May 23 14:08:27.581: RTP(359): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FA50, ssrc=BCA
May 23 14:08:27.581: RTP(360): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FAF0, ssrc=BCA
May 23 14:08:27.581: RTP(361): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FB90, ssrc=BCA
May 23 14:08:27.585: RTP(64213): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA220, ssrc=1959A317
May 23 14:08:27.585: RTP(64214): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA2C0, ssrc=1959A317
May 23 14:08:27.585: RTP(64215): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA360, ssrc=1959A317
May 23 14:08:27.589: RTP(64216): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA400, ssrc=1959A317
May 23 14:08:27.589: RTP(64217): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA4A0, ssrc=1959A317
May 23 14:08:27.589: RTP(64218): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA540, ssrc=1959A317
May 23 14:08:27.589: RTP(64219): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA5E0, ssrc=1959A317
May 23 14:08:27.593: RTP(64220): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA680, ssrc=1959A317
May 23 14:08:27.593: RTP(64221): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA720, ssrc=1959A317
May 23 14:08:27.593: RTP(64222): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA7C0, ssrc=1959A317
May 23 14:08:27.597: RTP(64223): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA860, ssrc=1959A317
May 23 14:08:27.597: RTP(64224): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA900, ssrc=1959A317
May 23 14:08:27.597: RTP(64225): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDA9A0, ssrc=1959A317
May 23 14:08:27.597: RTP(64226): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDAA40, ssrc=1959A317
May 23 14:08:27.601: RTP(64227): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDAAE0, ssrc=1959A317
May 23 14:08:27.601: RTP(64228): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDAB80, ssrc=1959A317
May 23 14:08:27.601: RTP(64229): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDAC20, ssrc=1959A317
May 23 14:08:27.605: RTP(64230): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDACC0, ssrc=1959A317
May 23 14:08:27.605: RTP(64231): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDAD60, ssrc=1959A317
May 23 14:08:27.605: RTP(64232): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDAE00, ssrc=1959A317
May 23 14:08:27.605: RTP(64233): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDAEA0, ssrc=1959A317
May 23 14:08:27.609: RTP(64234): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDAF40, ssrc=1959A317
May 23 14:08:27.609: RTP(64235): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDAFE0, ssrc=1959A317
May 23 14:08:27.609: RTP(64236): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB080, ssrc=1959A317
May 23 14:08:27.613: RTP(64237): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB120, ssrc=1959A317
May 23 14:08:27.613: RTP(64238): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB1C0, ssrc=1959A317
May 23 14:08:27.613: RTP(64239): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB260, ssrc=1959A317
May 23 14:08:27.613: RTP(64240): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB300, ssrc=1959A317
May 23 14:08:27.617: RTP(64241): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB3A0, ssrc=1959A317
May 23 14:08:27.617: RTP(64242): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB440, ssrc=1959A317
May 23 14:08:27.617: RTP(64243): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB4E0, ssrc=1959A317
May 23 14:08:27.617: RTP(64244): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB580, ssrc=1959A317
May 23 14:08:27.621: RTP(64245): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB620, ssrc=1959A317
May 23 14:08:27.621: RTP(64246): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB6C0, ssrc=1959A317
May 23 14:08:27.621: RTP(64247): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB760, ssrc=1959A317
May 23 14:08:27.621: RTP(64248): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB800, ssrc=1959A317
May 23 14:08:27.625: RTP(64249): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB8A0, ssrc=1959A317
May 23 14:08:27.625: RTP(64250): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB940, ssrc=1959A317
May 23 14:08:27.625: RTP(64251): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDB9E0, ssrc=1959A317
May 23 14:08:27.629: RTP(64252): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDBA80, ssrc=1959A317
May 23 14:08:27.629: RTP(64253): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDBB20, ssrc=1959A317
May 23 14:08:27.629: RTP(64254): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDBBC0, ssrc=1959A317
May 23 14:08:27.629: RTP(64255): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDBC60, ssrc=1959A317
May 23 14:08:27.633: RTP(362): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FC30, ssrc=BCA
May 23 14:08:27.633: RTP(363): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FCD0, ssrc=BCA
May 23 14:08:27.633: RTP(364): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FD70, ssrc=BCA
May 23 14:08:27.641: RTP(64256): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDBD00, ssrc=1959A317
May 23 14:08:27.649: RTP(365): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FE10, ssrc=BCA
May 23 14:08:27.657: RTP(64257): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDBDA0, ssrc=1959A317
May 23 14:08:27.669: RTP(366): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FEB0, ssrc=BCA
May 23 14:08:27.677: RTP(64258): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDBE40, ssrc=1959A317
May 23 14:08:27.689: RTP(367): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FF50, ssrc=BCA
May 23 14:08:27.697: RTP(64259): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDBEE0, ssrc=1959A317
May 23 14:08:27.709: RTP(368): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=FFF0, ssrc=BCA
May 23 14:08:27.717: RTP(64260): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDBF80, ssrc=1959A317
May 23 14:08:27.729: RTP(369): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=10090, ssrc=BCA
May 23 14:08:27.733: RTP(64261): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDC020, ssrc=1959A317
May 23 14:08:27.749: RTP(370): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=10130, ssrc=BCA
May 23 14:08:27.757: RTP(64262): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDC0C0, ssrc=1959A317
May 23 14:08:27.769: RTP(371): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=101D0, ssrc=BCA
May 23 14:08:27.777: RTP(64263): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDC160, ssrc=1959A317
May 23 14:08:27.789: RTP(372): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=10270, ssrc=BCA
May 23 14:08:27.797: RTP(64264): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDC200, ssrc=1959A317
May 23 14:08:27.809: RTP(373): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=10310, ssrc=BCA
May 23 14:08:28.330: RTP(396): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11170, ssrc=BCA
May 23 14:08:28.330: RTP(397): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11210, ssrc=BCA
May 23 14:08:28.330: RTP(398): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=112B0, ssrc=BCA
May 23 14:08:28.334: RTP(399): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11350, ssrc=BCA
May 23 14:08:28.334: RTP(64290): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD240, ssrc=1959A317
May 23 14:08:29.131: RTP(64291): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD2E0, ssrc=1959A317
May 23 14:08:29.131: RTP(64292): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD380, ssrc=1959A317
May 23 14:08:29.135: RTP(64293): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD420, ssrc=1959A317
May 23 14:08:29.135: RTP(64294): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD4C0, ssrc=1959A317
May 23 14:08:29.135: RTP(64295): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD560, ssrc=1959A317
May 23 14:08:29.139: RTP(64296): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD600, ssrc=1959A317
May 23 14:08:29.139: RTP(64297): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD6A0, ssrc=1959A317
May 23 14:08:29.139: RTP(64298): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD740, ssrc=1959A317
May 23 14:08:29.139: RTP(64299): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD7E0, ssrc=1959A317
May 23 14:08:29.143: RTP(64300): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD880, ssrc=1959A317
May 23 14:08:29.143: RTP(64301): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD920, ssrc=1959A317
May 23 14:08:29.143: RTP(64302): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDD9C0, ssrc=1959A317
May 23 14:08:29.147: RTP(64303): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDDA60, ssrc=1959A317
May 23 14:08:29.147: RTP(64304): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDDB00, ssrc=1959A317
May 23 14:08:29.147: RTP(64305): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDDBA0, ssrc=1959A317
May 23 14:08:29.147: RTP(64306): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDDC40, ssrc=1959A317
May 23 14:08:29.151: RTP(64307): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDDCE0, ssrc=1959A317
May 23 14:08:29.151: RTP(64308): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDDD80, ssrc=1959A317
May 23 14:08:29.151: RTP(64309): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDDE20, ssrc=1959A317
May 23 14:08:29.151: RTP(64310): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDDEC0, ssrc=1959A317
May 23 14:08:29.155: RTP(64311): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDDF60, ssrc=1959A317
May 23 14:08:29.155: RTP(64312): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE000, ssrc=1959A317
May 23 14:08:29.155: RTP(64313): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE0A0, ssrc=1959A317
May 23 14:08:29.159: RTP(64314): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE140, ssrc=1959A317
May 23 14:08:29.159: RTP(64315): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE1E0, ssrc=1959A317
May 23 14:08:29.159: RTP(64316): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE280, ssrc=1959A317
May 23 14:08:29.159: RTP(64317): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE320, ssrc=1959A317
May 23 14:08:29.163: RTP(64318): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE3C0, ssrc=1959A317
May 23 14:08:29.163: RTP(64319): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE460, ssrc=1959A317
May 23 14:08:29.163: RTP(64320): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE500, ssrc=1959A317
May 23 14:08:29.167: RTP(64321): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE5A0, ssrc=1959A317
May 23 14:08:29.167: RTP(64322): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE640, ssrc=1959A317
May 23 14:08:29.167: RTP(64323): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE6E0, ssrc=1959A317
May 23 14:08:29.167: RTP(64324): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE780, ssrc=1959A317
May 23 14:08:29.171: RTP(64325): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE820, ssrc=1959A317
May 23 14:08:29.171: RTP(64326): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE8C0, ssrc=1959A317
May 23 14:08:29.171: RTP(64327): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDE960, ssrc=1959A317
May 23 14:08:29.175: RTP(64328): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDEA00, ssrc=1959A317
May 23 14:08:29.175: RTP(64329): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDEAA0, ssrc=1959A317
May 23 14:08:29.175: RTP(64330): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDEB40, ssrc=1959A317
May 23 14:08:29.175: RTP(64331): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDEBE0, ssrc=1959A317
May 23 14:08:29.179: RTP(64332): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDEC80, ssrc=1959A317
May 23 14:08:29.179: RTP(64333): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDED20, ssrc=1959A317
May 23 14:08:29.179: RTP(400): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=113F0, ssrc=BCA
May 23 14:08:29.179: RTP(401): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11490, ssrc=BCA
May 23 14:08:29.184: RTP(402): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11530, ssrc=BCA
May 23 14:08:29.184: RTP(403): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=115D0, ssrc=BCA
May 23 14:08:29.184: RTP(404): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11670, ssrc=BCA
May 23 14:08:29.188: RTP(405): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11710, ssrc=BCA
May 23 14:08:29.188: RTP(406): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=117B0, ssrc=BCA
May 23 14:08:29.188: RTP(407): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11850, ssrc=BCA
May 23 14:08:29.188: RTP(408): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=118F0, ssrc=BCA
May 23 14:08:29.192: RTP(409): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11990, ssrc=BCA
May 23 14:08:29.192: RTP(410): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11A30, ssrc=BCA
May 23 14:08:29.192: RTP(411): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11AD0, ssrc=BCA
May 23 14:08:29.192: RTP(412): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11B70, ssrc=BCA
May 23 14:08:29.196: RTP(413): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11C10, ssrc=BCA
May 23 14:08:29.196: RTP(414): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11CB0, ssrc=BCA
May 23 14:08:29.196: RTP(415): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11D50, ssrc=BCA
May 23 14:08:29.196: RTP(416): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11DF0, ssrc=BCA
May 23 14:08:29.200: RTP(417): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11E90, ssrc=BCA
May 23 14:08:29.200: RTP(418): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11F30, ssrc=BCA
May 23 14:08:29.200: RTP(419): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=11FD0, ssrc=BCA
May 23 14:08:29.200: RTP(420): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12070, ssrc=BCA
May 23 14:08:29.204: RTP(421): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12110, ssrc=BCA
May 23 14:08:29.204: RTP(422): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=121B0, ssrc=BCA
May 23 14:08:29.204: RTP(423): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12250, ssrc=BCA
May 23 14:08:29.208: RTP(424): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=122F0, ssrc=BCA
May 23 14:08:29.208: RTP(425): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12390, ssrc=BCA
May 23 14:08:29.208: RTP(426): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12430, ssrc=BCA
May 23 14:08:29.208: RTP(427): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=124D0, ssrc=BCA
May 23 14:08:29.212: RTP(428): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12570, ssrc=BCA
May 23 14:08:29.212: RTP(429): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12610, ssrc=BCA
May 23 14:08:29.212: RTP(430): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=126B0, ssrc=BCA
May 23 14:08:29.212: RTP(431): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12750, ssrc=BCA
May 23 14:08:29.216: RTP(432): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=127F0, ssrc=BCA
May 23 14:08:29.216: RTP(433): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12890, ssrc=BCA
May 23 14:08:29.216: RTP(434): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12930, ssrc=BCA
May 23 14:08:29.216: RTP(435): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=129D0, ssrc=BCA
May 23 14:08:29.220: RTP(436): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12A70, ssrc=BCA
May 23 14:08:29.220: RTP(437): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12B10, ssrc=BCA
May 23 14:08:29.220: RTP(438): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12BB0, ssrc=BCA
May 23 14:08:29.220: RTP(439): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12C50, ssrc=BCA
May 23 14:08:29.224: RTP(440): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12CF0, ssrc=BCA
May 23 14:08:29.224: RTP(441): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12D90, ssrc=BCA
May 23 14:08:29.224: RTP(442): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12E30, ssrc=BCA
May 23 14:08:29.224: RTP(443): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=12ED0, ssrc=BCA
May 23 14:08:29.228: RTP(64334): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDEDC0, ssrc=1959A317
May 23 14:08:29.228: RTP(64335): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFDEE60, ssrc=1959A317
May 23 14:08:30.814: RTP(483): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=147D0, ssrc=BCA
May 23 14:08:30.814: RTP(484): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14870, ssrc=BCA
May 23 14:08:30.818: RTP(485): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14910, ssrc=BCA
May 23 14:08:30.818: RTP(486): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=149B0, ssrc=BCA
May 23 14:08:30.818: RTP(487): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14A50, ssrc=BCA
May 23 14:08:30.818: RTP(488): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14AF0, ssrc=BCA
May 23 14:08:30.822: RTP(489): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14B90, ssrc=BCA
May 23 14:08:30.822: RTP(490): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14C30, ssrc=BCA
May 23 14:08:30.822: RTP(491): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14CD0, ssrc=BCA
May 23 14:08:30.822: RTP(492): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14D70, ssrc=BCA
May 23 14:08:30.826: RTP(493): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14E10, ssrc=BCA
May 23 14:08:30.826: RTP(494): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14EB0, ssrc=BCA
May 23 14:08:30.826: RTP(495): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14F50, ssrc=BCA
May 23 14:08:30.826: RTP(496): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=14FF0, ssrc=BCA
May 23 14:08:30.830: RTP(497): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15090, ssrc=BCA
May 23 14:08:30.830: RTP(498): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15130, ssrc=BCA
May 23 14:08:30.830: RTP(499): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=151D0, ssrc=BCA
May 23 14:08:30.834: RTP(500): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15270, ssrc=BCA
May 23 14:08:30.834: RTP(501): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15310, ssrc=BCA
May 23 14:08:30.834: RTP(502): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=153B0, ssrc=BCA
May 23 14:08:30.834: RTP(503): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15450, ssrc=BCA
May 23 14:08:30.838: RTP(504): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=154F0, ssrc=BCA
May 23 14:08:30.838: RTP(505): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15590, ssrc=BCA
May 23 14:08:30.838: RTP(506): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15630, ssrc=BCA
May 23 14:08:30.838: RTP(507): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=156D0, ssrc=BCA
May 23 14:08:30.842: RTP(508): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15770, ssrc=BCA
May 23 14:08:30.842: RTP(509): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15810, ssrc=BCA
May 23 14:08:30.842: RTP(510): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=158B0, ssrc=BCA
May 23 14:08:30.846: RTP(511): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15950, ssrc=BCA
May 23 14:08:30.846: RTP(512): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=159F0, ssrc=BCA
May 23 14:08:30.846: RTP(513): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15A90, ssrc=BCA
May 23 14:08:30.846: RTP(514): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15B30, ssrc=BCA
May 23 14:08:30.850: RTP(515): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15BD0, ssrc=BCA
May 23 14:08:30.850: RTP(516): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15C70, ssrc=BCA
May 23 14:08:30.850: RTP(517): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15D10, ssrc=BCA
May 23 14:08:30.850: RTP(518): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15DB0, ssrc=BCA
May 23 14:08:30.854: RTP(519): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15E50, ssrc=BCA
May 23 14:08:30.854: RTP(520): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15EF0, ssrc=BCA
May 23 14:08:30.854: RTP(521): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=15F90, ssrc=BCA
May 23 14:08:30.854: RTP(522): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16030, ssrc=BCA
May 23 14:08:30.858: RTP(523): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=160D0, ssrc=BCA
May 23 14:08:30.858: RTP(524): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16170, ssrc=BCA
May 23 14:08:30.858: RTP(525): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16210, ssrc=BCA
May 23 14:08:30.862: RTP(64415): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2060, ssrc=1959A317
May 23 14:08:30.862: RTP(64416): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2100, ssrc=1959A317
May 23 14:08:30.862: RTP(64417): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE21A0, ssrc=1959A317
May 23 14:08:31.523: RTP(526): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=162B0, ssrc=BCA
May 23 14:08:31.523: RTP(527): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16350, ssrc=BCA
May 23 14:08:31.527: RTP(528): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=163F0, ssrc=BCA
May 23 14:08:31.527: RTP(529): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16490, ssrc=BCA
May 23 14:08:31.527: RTP(530): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16530, ssrc=BCA
May 23 14:08:31.527: RTP(531): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=165D0, ssrc=BCA
May 23 14:08:31.531: RTP(532): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16670, ssrc=BCA
May 23 14:08:31.531: RTP(533): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16710, ssrc=BCA
May 23 14:08:31.531: RTP(534): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=167B0, ssrc=BCA
May 23 14:08:31.531: RTP(535): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16850, ssrc=BCA
May 23 14:08:31.535: RTP(536): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=168F0, ssrc=BCA
May 23 14:08:31.535: RTP(537): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16990, ssrc=BCA
May 23 14:08:31.535: RTP(538): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16A30, ssrc=BCA
May 23 14:08:31.535: RTP(539): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16AD0, ssrc=BCA
May 23 14:08:31.539: RTP(540): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16B70, ssrc=BCA
May 23 14:08:31.539: RTP(541): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16C10, ssrc=BCA
May 23 14:08:31.539: RTP(542): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16CB0, ssrc=BCA
May 23 14:08:31.543: RTP(543): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16D50, ssrc=BCA
May 23 14:08:31.543: RTP(544): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16DF0, ssrc=BCA
May 23 14:08:31.543: RTP(545): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16E90, ssrc=BCA
May 23 14:08:31.543: RTP(546): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16F30, ssrc=BCA
May 23 14:08:31.547: RTP(547): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=16FD0, ssrc=BCA
May 23 14:08:31.547: RTP(548): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=17070, ssrc=BCA
May 23 14:08:31.547: RTP(549): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=17110, ssrc=BCA
May 23 14:08:31.547: RTP(550): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=171B0, ssrc=BCA
May 23 14:08:31.551: RTP(551): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=17250, ssrc=BCA
May 23 14:08:31.551: RTP(552): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=172F0, ssrc=BCA
May 23 14:08:31.551: RTP(553): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=17390, ssrc=BCA
May 23 14:08:31.551: RTP(554): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=17430, ssrc=BCA
May 23 14:08:31.555: RTP(555): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=174D0, ssrc=BCA
May 23 14:08:31.555: RTP(556): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=17570, ssrc=BCA
May 23 14:08:31.555: RTP(557): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=17610, ssrc=BCA
May 23 14:08:31.555: RTP(558): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=176B0, ssrc=BCA
May 23 14:08:31.559: RTP(559): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=17750, ssrc=BCA
May 23 14:08:31.559: RTP(560): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=177F0, ssrc=BCA
May 23 14:08:31.559: RTP(64418): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2240, ssrc=1959A317
May 23 14:08:31.559: RTP(64419): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE22E0, ssrc=1959A317
May 23 14:08:31.563: RTP(64420): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2380, ssrc=1959A317
May 23 14:08:31.563: RTP(64421): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2420, ssrc=1959A317
May 23 14:08:31.563: RTP(64422): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE24C0, ssrc=1959A317
May 23 14:08:31.563: RTP(64423): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2560, ssrc=1959A317
May 23 14:08:31.567: RTP(64424): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2600, ssrc=1959A317
May 23 14:08:31.567: RTP(64425): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE26A0, ssrc=1959A317
May 23 14:08:31.567: RTP(64426): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2740, ssrc=1959A317
May 23 14:08:31.567: RTP(64427): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE27E0, ssrc=1959A317
May 23 14:08:31.571: RTP(64428): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2880, ssrc=1959A317
May 23 14:08:31.571: RTP(64429): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2920, ssrc=1959A317
May 23 14:08:31.571: RTP(64430): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE29C0, ssrc=1959A317
May 23 14:08:31.571: RTP(64431): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2A60, ssrc=1959A317
May 23 14:08:31.575: RTP(64432): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2B00, ssrc=1959A317
May 23 14:08:31.575: RTP(64433): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2BA0, ssrc=1959A317
May 23 14:08:31.575: RTP(64434): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2C40, ssrc=1959A317
May 23 14:08:31.579: RTP(64435): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2CE0, ssrc=1959A317
May 23 14:08:31.579: RTP(64436): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2D80, ssrc=1959A317
May 23 14:08:31.579: RTP(64437): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2E20, ssrc=1959A317
May 23 14:08:31.579: RTP(64438): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2EC0, ssrc=1959A317
May 23 14:08:31.583: RTP(64439): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE2F60, ssrc=1959A317
May 23 14:08:32.733: RTP(64510): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE5BC0, ssrc=1959A317
May 23 14:08:32.733: RTP(599): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=19BE0, ssrc=BCA
May 23 14:08:32.733: RTP(600): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=19C80, ssrc=BCA
May 23 14:08:33.418: RTP(64511): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE5C60, ssrc=1959A317
May 23 14:08:33.418: RTP(64512): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE5D00, ssrc=1959A317
May 23 14:08:33.418: RTP(64513): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE5DA0, ssrc=1959A317
May 23 14:08:33.422: RTP(64514): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE5E40, ssrc=1959A317
May 23 14:08:33.422: RTP(64515): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE5EE0, ssrc=1959A317
May 23 14:08:33.422: RTP(64516): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE5F80, ssrc=1959A317
May 23 14:08:33.422: RTP(64517): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6020, ssrc=1959A317
May 23 14:08:33.426: RTP(64518): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE60C0, ssrc=1959A317
May 23 14:08:33.426: RTP(64519): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6160, ssrc=1959A317
May 23 14:08:33.426: RTP(64520): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6200, ssrc=1959A317
May 23 14:08:33.431: RTP(64521): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE62A0, ssrc=1959A317
May 23 14:08:33.431: RTP(64522): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6340, ssrc=1959A317
May 23 14:08:33.431: RTP(64523): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE63E0, ssrc=1959A317
May 23 14:08:33.431: RTP(64524): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6480, ssrc=1959A317
May 23 14:08:33.435: RTP(64525): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6520, ssrc=1959A317
May 23 14:08:33.435: RTP(64526): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE65C0, ssrc=1959A317
May 23 14:08:33.435: RTP(64527): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6660, ssrc=1959A317
May 23 14:08:33.439: RTP(64528): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6700, ssrc=1959A317
May 23 14:08:33.439: RTP(64529): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE67A0, ssrc=1959A317
May 23 14:08:33.439: RTP(64530): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6840, ssrc=1959A317
May 23 14:08:33.439: RTP(64531): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE68E0, ssrc=1959A317
May 23 14:08:33.443: RTP(64532): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6980, ssrc=1959A317
May 23 14:08:33.443: RTP(64533): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6A20, ssrc=1959A317
May 23 14:08:33.443: RTP(64534): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6AC0, ssrc=1959A317
May 23 14:08:33.447: RTP(64535): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6B60, ssrc=1959A317
May 23 14:08:33.447: RTP(64536): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6C00, ssrc=1959A317
May 23 14:08:33.447: RTP(64537): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6CA0, ssrc=1959A317
May 23 14:08:33.447: RTP(64538): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6D40, ssrc=1959A317
May 23 14:08:33.451: RTP(64539): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6DE0, ssrc=1959A317
May 23 14:08:33.451: RTP(64540): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6E80, ssrc=1959A317
May 23 14:08:33.451: RTP(64541): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6F20, ssrc=1959A317
May 23 14:08:33.451: RTP(64542): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE6FC0, ssrc=1959A317
May 23 14:08:33.455: RTP(64543): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7060, ssrc=1959A317
May 23 14:08:33.455: RTP(64544): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7100, ssrc=1959A317
May 23 14:08:33.455: RTP(64545): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE71A0, ssrc=1959A317
May 23 14:08:33.459: RTP(64546): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7240, ssrc=1959A317
May 23 14:08:33.459: RTP(64547): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE72E0, ssrc=1959A317
May 23 14:08:33.459: RTP(601): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=19D20, ssrc=BCA
May 23 14:08:33.459: RTP(602): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=19DC0, ssrc=BCA
May 23 14:08:33.463: RTP(603): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=19E60, ssrc=BCA
May 23 14:08:33.463: RTP(604): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=19F00, ssrc=BCA
May 23 14:08:33.463: RTP(605): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=19FA0, ssrc=BCA
May 23 14:08:33.467: RTP(606): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A040, ssrc=BCA
May 23 14:08:33.467: RTP(607): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A0E0, ssrc=BCA
May 23 14:08:33.467: RTP(608): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A180, ssrc=BCA
May 23 14:08:33.467: RTP(609): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A220, ssrc=BCA
May 23 14:08:33.471: RTP(610): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A2C0, ssrc=BCA
May 23 14:08:33.471: RTP(611): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A360, ssrc=BCA
May 23 14:08:33.471: RTP(612): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A400, ssrc=BCA
May 23 14:08:33.471: RTP(613): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A4A0, ssrc=BCA
May 23 14:08:33.475: RTP(614): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A540, ssrc=BCA
May 23 14:08:33.475: RTP(615): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A5E0, ssrc=BCA
May 23 14:08:33.475: RTP(616): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A680, ssrc=BCA
May 23 14:08:33.475: RTP(617): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A720, ssrc=BCA
May 23 14:08:33.479: RTP(618): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A7C0, ssrc=BCA
May 23 14:08:33.479: RTP(619): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A860, ssrc=BCA
May 23 14:08:33.479: RTP(620): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A900, ssrc=BCA
May 23 14:08:33.479: RTP(621): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1A9A0, ssrc=BCA
May 23 14:08:33.483: RTP(622): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1AA40, ssrc=BCA
May 23 14:08:33.483: RTP(623): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1AAE0, ssrc=BCA
May 23 14:08:33.483: RTP(624): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1AB80, ssrc=BCA
May 23 14:08:33.483: RTP(625): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1AC20, ssrc=BCA
May 23 14:08:33.487: RTP(626): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1ACC0, ssrc=BCA
May 23 14:08:33.487: RTP(627): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1AD60, ssrc=BCA
May 23 14:08:33.487: RTP(628): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1AE00, ssrc=BCA
May 23 14:08:33.487: RTP(629): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1AEA0, ssrc=BCA
May 23 14:08:33.491: RTP(630): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1AF40, ssrc=BCA
May 23 14:08:33.491: RTP(631): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1AFE0, ssrc=BCA
May 23 14:08:33.491: RTP(632): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1B080, ssrc=BCA
May 23 14:08:33.491: RTP(633): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1B120, ssrc=BCA
May 23 14:08:33.495: RTP(634): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1B1C0, ssrc=BCA
May 23 14:08:33.495: RTP(635): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1B260, ssrc=BCA
May 23 14:08:33.495: RTP(636): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1B300, ssrc=BCA
May 23 14:08:33.495: RTP(637): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1B3A0, ssrc=BCA
May 23 14:08:33.499: RTP(638): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1B440, ssrc=BCA
May 23 14:08:33.499: RTP(64548): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7380, ssrc=1959A317
May 23 14:08:33.499: RTP(64549): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7420, ssrc=1959A317
May 23 14:08:33.499: RTP(639): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1B4E0, ssrc=BCA
May 23 14:08:34.176: RTP(64550): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE74C0, ssrc=1959A317
May 23 14:08:34.180: RTP(64551): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7560, ssrc=1959A317
May 23 14:08:34.180: RTP(64552): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7600, ssrc=1959A317
May 23 14:08:34.180: RTP(64553): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE76A0, ssrc=1959A317
May 23 14:08:34.180: RTP(64554): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7740, ssrc=1959A317
May 23 14:08:34.184: RTP(64555): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE77E0, ssrc=1959A317
May 23 14:08:34.184: RTP(64556): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7880, ssrc=1959A317
May 23 14:08:34.184: RTP(64557): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7920, ssrc=1959A317
May 23 14:08:34.188: RTP(64558): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE79C0, ssrc=1959A317
May 23 14:08:34.188: RTP(64559): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFE7A60, ssrc=1959A317
May 23 14:08:35.314: RTP(714): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E3C0, ssrc=BCA
May 23 14:08:35.314: RTP(715): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E460, ssrc=BCA
May 23 14:08:35.314: RTP(716): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E500, ssrc=BCA
May 23 14:08:35.318: RTP(717): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E5A0, ssrc=BCA
May 23 14:08:35.318: RTP(718): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E640, ssrc=BCA
May 23 14:08:35.318: RTP(719): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E6E0, ssrc=BCA
May 23 14:08:35.318: RTP(720): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E780, ssrc=BCA
May 23 14:08:35.322: RTP(721): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E820, ssrc=BCA
May 23 14:08:35.322: RTP(722): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E8C0, ssrc=BCA
May 23 14:08:35.322: RTP(723): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1E960, ssrc=BCA
May 23 14:08:35.322: RTP(724): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EA00, ssrc=BCA
May 23 14:08:35.326: RTP(725): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EAA0, ssrc=BCA
May 23 14:08:35.326: RTP(726): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EB40, ssrc=BCA
May 23 14:08:35.326: RTP(727): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EBE0, ssrc=BCA
May 23 14:08:35.326: RTP(728): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EC80, ssrc=BCA
May 23 14:08:35.330: RTP(729): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1ED20, ssrc=BCA
May 23 14:08:35.330: RTP(730): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EDC0, ssrc=BCA
May 23 14:08:35.330: RTP(64639): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEAC60, ssrc=1959A317
May 23 14:08:35.334: RTP(64640): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEAD00, ssrc=1959A317
May 23 14:08:36.015: RTP(64641): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEADA0, ssrc=1959A317
May 23 14:08:36.019: RTP(64642): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEAE40, ssrc=1959A317
May 23 14:08:36.019: RTP(64643): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEAEE0, ssrc=1959A317
May 23 14:08:36.019: RTP(64644): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEAF80, ssrc=1959A317
May 23 14:08:36.019: RTP(64645): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB020, ssrc=1959A317
May 23 14:08:36.023: RTP(64646): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB0C0, ssrc=1959A317
May 23 14:08:36.023: RTP(64647): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB160, ssrc=1959A317
May 23 14:08:36.023: RTP(64648): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB200, ssrc=1959A317
May 23 14:08:36.027: RTP(64649): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB2A0, ssrc=1959A317
May 23 14:08:36.027: RTP(64650): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB340, ssrc=1959A317
May 23 14:08:36.027: RTP(64651): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB3E0, ssrc=1959A317
May 23 14:08:36.027: RTP(64652): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB480, ssrc=1959A317
May 23 14:08:36.031: RTP(64653): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB520, ssrc=1959A317
May 23 14:08:36.031: RTP(64654): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB5C0, ssrc=1959A317
May 23 14:08:36.031: RTP(64655): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB660, ssrc=1959A317
May 23 14:08:36.035: RTP(64656): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB700, ssrc=1959A317
May 23 14:08:36.035: RTP(64657): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB7A0, ssrc=1959A317
May 23 14:08:36.035: RTP(64658): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB840, ssrc=1959A317
May 23 14:08:36.035: RTP(64659): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB8E0, ssrc=1959A317
May 23 14:08:36.039: RTP(64660): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEB980, ssrc=1959A317
May 23 14:08:36.039: RTP(64661): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBA20, ssrc=1959A317
May 23 14:08:36.039: RTP(64662): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBAC0, ssrc=1959A317
May 23 14:08:36.039: RTP(64663): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBB60, ssrc=1959A317
May 23 14:08:36.043: RTP(64664): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBC00, ssrc=1959A317
May 23 14:08:36.043: RTP(64665): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBCA0, ssrc=1959A317
May 23 14:08:36.043: RTP(64666): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBD40, ssrc=1959A317
May 23 14:08:36.047: RTP(64667): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBDE0, ssrc=1959A317
May 23 14:08:36.047: RTP(64668): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBE80, ssrc=1959A317
May 23 14:08:36.047: RTP(64669): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBF20, ssrc=1959A317
May 23 14:08:36.047: RTP(64670): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEBFC0, ssrc=1959A317
May 23 14:08:36.051: RTP(64671): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEC060, ssrc=1959A317
May 23 14:08:36.051: RTP(64672): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEC100, ssrc=1959A317
May 23 14:08:36.051: RTP(64673): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEC1A0, ssrc=1959A317
May 23 14:08:36.055: RTP(64674): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEC240, ssrc=1959A317
May 23 14:08:36.055: RTP(64675): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEC2E0, ssrc=1959A317
May 23 14:08:36.055: RTP(64676): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEC380, ssrc=1959A317
May 23 14:08:36.055: RTP(64677): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEC420, ssrc=1959A317
May 23 14:08:36.059: RTP(731): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EE60, ssrc=BCA
May 23 14:08:36.059: RTP(732): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EF00, ssrc=BCA
May 23 14:08:36.059: RTP(733): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1EFA0, ssrc=BCA
May 23 14:08:36.059: RTP(734): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F040, ssrc=BCA
May 23 14:08:36.063: RTP(735): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F0E0, ssrc=BCA
May 23 14:08:36.063: RTP(736): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F180, ssrc=BCA
May 23 14:08:36.063: RTP(737): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F220, ssrc=BCA
May 23 14:08:36.067: RTP(738): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F2C0, ssrc=BCA
May 23 14:08:36.067: RTP(739): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F360, ssrc=BCA
May 23 14:08:36.067: RTP(740): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F400, ssrc=BCA
May 23 14:08:36.067: RTP(741): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F4A0, ssrc=BCA
May 23 14:08:36.071: RTP(742): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F540, ssrc=BCA
May 23 14:08:36.071: RTP(743): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F5E0, ssrc=BCA
May 23 14:08:36.071: RTP(744): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F680, ssrc=BCA
May 23 14:08:36.071: RTP(745): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F720, ssrc=BCA
May 23 14:08:36.075: RTP(746): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F7C0, ssrc=BCA
May 23 14:08:36.075: RTP(747): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F860, ssrc=BCA
May 23 14:08:36.075: RTP(748): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F900, ssrc=BCA
May 23 14:08:36.075: RTP(749): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1F9A0, ssrc=BCA
May 23 14:08:36.079: RTP(750): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FA40, ssrc=BCA
May 23 14:08:36.079: RTP(751): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FAE0, ssrc=BCA
May 23 14:08:36.079: RTP(752): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FB80, ssrc=BCA
May 23 14:08:36.079: RTP(753): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FC20, ssrc=BCA
May 23 14:08:36.083: RTP(754): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FCC0, ssrc=BCA
May 23 14:08:36.083: RTP(755): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FD60, ssrc=BCA
May 23 14:08:36.083: RTP(756): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FE00, ssrc=BCA
May 23 14:08:36.083: RTP(757): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FEA0, ssrc=BCA
May 23 14:08:36.087: RTP(758): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FF40, ssrc=BCA
May 23 14:08:36.087: RTP(759): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=1FFE0, ssrc=BCA
May 23 14:08:36.087: RTP(760): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=20080, ssrc=BCA
May 23 14:08:36.087: RTP(761): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=20120, ssrc=BCA
May 23 14:08:36.091: RTP(762): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=201C0, ssrc=BCA
May 23 14:08:36.091: RTP(763): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=20260, ssrc=BCA
May 23 14:08:36.091: RTP(764): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=20300, ssrc=BCA
May 23 14:08:36.091: RTP(765): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=203A0, ssrc=BCA
May 23 14:08:36.095: RTP(766): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=20440, ssrc=BCA
May 23 14:08:36.095: RTP(767): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=204E0, ssrc=BCA
May 23 14:08:36.095: RTP(768): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=20580, ssrc=BCA
May 23 14:08:36.596: RTP(64704): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFED500, ssrc=1959A317
May 23 14:08:37.277: RTP(794): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=215C0, ssrc=BCA
May 23 14:08:37.281: RTP(795): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21660, ssrc=BCA
May 23 14:08:37.281: RTP(796): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21700, ssrc=BCA
May 23 14:08:37.281: RTP(797): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=217A0, ssrc=BCA
May 23 14:08:37.285: RTP(798): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21840, ssrc=BCA
May 23 14:08:37.285: RTP(799): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=218E0, ssrc=BCA
May 23 14:08:37.285: RTP(800): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21980, ssrc=BCA
May 23 14:08:37.285: RTP(801): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21A20, ssrc=BCA
May 23 14:08:37.289: RTP(802): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21AC0, ssrc=BCA
May 23 14:08:37.289: RTP(803): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21B60, ssrc=BCA
May 23 14:08:37.289: RTP(804): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21C00, ssrc=BCA
May 23 14:08:37.289: RTP(805): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21CA0, ssrc=BCA
May 23 14:08:37.293: RTP(806): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21D40, ssrc=BCA
May 23 14:08:37.293: RTP(807): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21DE0, ssrc=BCA
May 23 14:08:37.293: RTP(808): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21E80, ssrc=BCA
May 23 14:08:37.297: RTP(809): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21F20, ssrc=BCA
May 23 14:08:37.297: RTP(810): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=21FC0, ssrc=BCA
May 23 14:08:37.297: RTP(811): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22060, ssrc=BCA
May 23 14:08:37.297: RTP(812): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22100, ssrc=BCA
May 23 14:08:37.301: RTP(813): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=221A0, ssrc=BCA
May 23 14:08:37.301: RTP(814): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22240, ssrc=BCA
May 23 14:08:37.301: RTP(815): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=222E0, ssrc=BCA
May 23 14:08:37.301: RTP(816): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22380, ssrc=BCA
May 23 14:08:37.305: RTP(817): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22420, ssrc=BCA
May 23 14:08:37.305: RTP(818): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=224C0, ssrc=BCA
May 23 14:08:37.305: RTP(819): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22560, ssrc=BCA
May 23 14:08:37.309: RTP(820): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22600, ssrc=BCA
May 23 14:08:37.309: RTP(821): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=226A0, ssrc=BCA
May 23 14:08:37.309: RTP(822): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22740, ssrc=BCA
May 23 14:08:37.309: RTP(823): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=227E0, ssrc=BCA
May 23 14:08:37.313: RTP(824): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22880, ssrc=BCA
May 23 14:08:37.313: RTP(825): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22920, ssrc=BCA
May 23 14:08:37.313: RTP(826): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=229C0, ssrc=BCA
May 23 14:08:37.313: RTP(827): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22A60, ssrc=BCA
May 23 14:08:37.317: RTP(828): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22B00, ssrc=BCA
May 23 14:08:37.317: RTP(829): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22BA0, ssrc=BCA
May 23 14:08:37.317: RTP(64705): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFED5A0, ssrc=1959A317
May 23 14:08:37.321: RTP(64706): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFED640, ssrc=1959A317
May 23 14:08:37.321: RTP(64707): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFED6E0, ssrc=1959A317
May 23 14:08:37.321: RTP(64708): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFED780, ssrc=1959A317
May 23 14:08:37.321: RTP(64709): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFED820, ssrc=1959A317
May 23 14:08:37.325: RTP(64710): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFED8C0, ssrc=1959A317
May 23 14:08:37.325: RTP(64711): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFED960, ssrc=1959A317
May 23 14:08:37.325: RTP(64712): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDA00, ssrc=1959A317
May 23 14:08:37.329: RTP(64713): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDAA0, ssrc=1959A317
May 23 14:08:37.329: RTP(64714): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDB40, ssrc=1959A317
May 23 14:08:37.329: RTP(64715): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDBE0, ssrc=1959A317
May 23 14:08:37.329: RTP(64716): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDC80, ssrc=1959A317
May 23 14:08:37.333: RTP(64717): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDD20, ssrc=1959A317
May 23 14:08:37.333: RTP(64718): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDDC0, ssrc=1959A317
May 23 14:08:37.333: RTP(64719): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDE60, ssrc=1959A317
May 23 14:08:37.333: RTP(64720): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDF00, ssrc=1959A317
May 23 14:08:37.337: RTP(64721): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEDFA0, ssrc=1959A317
May 23 14:08:37.337: RTP(64722): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE040, ssrc=1959A317
May 23 14:08:37.337: RTP(64723): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE0E0, ssrc=1959A317
May 23 14:08:37.341: RTP(64724): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE180, ssrc=1959A317
May 23 14:08:37.341: RTP(64725): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE220, ssrc=1959A317
May 23 14:08:37.341: RTP(64726): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE2C0, ssrc=1959A317
May 23 14:08:37.341: RTP(64727): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE360, ssrc=1959A317
May 23 14:08:37.345: RTP(64728): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE400, ssrc=1959A317
May 23 14:08:37.345: RTP(64729): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE4A0, ssrc=1959A317
May 23 14:08:37.345: RTP(64730): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE540, ssrc=1959A317
May 23 14:08:37.349: RTP(64731): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE5E0, ssrc=1959A317
May 23 14:08:37.349: RTP(64732): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE680, ssrc=1959A317
May 23 14:08:37.349: RTP(64733): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE720, ssrc=1959A317
May 23 14:08:37.349: RTP(64734): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE7C0, ssrc=1959A317
May 23 14:08:37.353: RTP(64735): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE860, ssrc=1959A317
May 23 14:08:37.353: RTP(64736): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE900, ssrc=1959A317
May 23 14:08:37.353: RTP(64737): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEE9A0, ssrc=1959A317
May 23 14:08:37.357: RTP(64738): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEEA40, ssrc=1959A317
May 23 14:08:37.357: RTP(64739): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEEAE0, ssrc=1959A317
May 23 14:08:37.357: RTP(64740): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEEB80, ssrc=1959A317
May 23 14:08:37.357: RTP(64741): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEEC20, ssrc=1959A317
May 23 14:08:37.361: RTP(64742): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEECC0, ssrc=1959A317
May 23 14:08:37.361: RTP(830): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22C40, ssrc=BCA
May 23 14:08:37.361: RTP(831): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22CE0, ssrc=BCA
May 23 14:08:37.365: RTP(832): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22D80, ssrc=BCA
May 23 14:08:37.377: RTP(64743): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEED60, ssrc=1959A317
May 23 14:08:37.381: RTP(833): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22E20, ssrc=BCA
May 23 14:08:37.397: RTP(64744): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEEE00, ssrc=1959A317
May 23 14:08:37.401: RTP(834): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22EC0, ssrc=BCA
May 23 14:08:37.421: RTP(64745): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEEEA0, ssrc=1959A317
May 23 14:08:37.421: RTP(835): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=22F60, ssrc=BCA
May 23 14:08:37.441: RTP(64746): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEEF40, ssrc=1959A317
May 23 14:08:37.441: RTP(836): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=23000, ssrc=BCA
May 23 14:08:37.453: RTP(64747): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFEEFE0, ssrc=1959A317
May 23 14:08:38.619: RTP(846): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=252B0, ssrc=BCA
May 23 14:08:38.619: RTP(847): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25350, ssrc=BCA
May 23 14:08:38.619: RTP(848): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=253F0, ssrc=BCA
May 23 14:08:38.623: RTP(849): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25490, ssrc=BCA
May 23 14:08:38.623: RTP(64805): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1420, ssrc=1959A317
May 23 14:08:39.284: RTP(850): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25530, ssrc=BCA
May 23 14:08:39.284: RTP(851): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=255D0, ssrc=BCA
May 23 14:08:39.284: RTP(852): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25670, ssrc=BCA
May 23 14:08:39.284: RTP(853): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25710, ssrc=BCA
May 23 14:08:39.288: RTP(854): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=257B0, ssrc=BCA
May 23 14:08:39.288: RTP(855): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25850, ssrc=BCA
May 23 14:08:39.288: RTP(856): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=258F0, ssrc=BCA
May 23 14:08:39.292: RTP(857): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25990, ssrc=BCA
May 23 14:08:39.292: RTP(858): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25A30, ssrc=BCA
May 23 14:08:39.292: RTP(859): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25AD0, ssrc=BCA
May 23 14:08:39.292: RTP(860): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25B70, ssrc=BCA
May 23 14:08:39.296: RTP(861): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25C10, ssrc=BCA
May 23 14:08:39.296: RTP(862): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25CB0, ssrc=BCA
May 23 14:08:39.296: RTP(863): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25D50, ssrc=BCA
May 23 14:08:39.296: RTP(864): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25DF0, ssrc=BCA
May 23 14:08:39.300: RTP(865): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25E90, ssrc=BCA
May 23 14:08:39.300: RTP(866): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25F30, ssrc=BCA
May 23 14:08:39.300: RTP(867): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=25FD0, ssrc=BCA
May 23 14:08:39.304: RTP(868): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26070, ssrc=BCA
May 23 14:08:39.304: RTP(869): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26110, ssrc=BCA
May 23 14:08:39.304: RTP(870): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=261B0, ssrc=BCA
May 23 14:08:39.304: RTP(871): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26250, ssrc=BCA
May 23 14:08:39.308: RTP(872): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=262F0, ssrc=BCA
May 23 14:08:39.308: RTP(873): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26390, ssrc=BCA
May 23 14:08:39.308: RTP(874): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26430, ssrc=BCA
May 23 14:08:39.308: RTP(875): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=264D0, ssrc=BCA
May 23 14:08:39.312: RTP(876): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26570, ssrc=BCA
May 23 14:08:39.312: RTP(877): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26610, ssrc=BCA
May 23 14:08:39.312: RTP(878): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=13, ts=266B0, ssrc=BCA
May 23 14:08:39.312: RTP(64806): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF14C0, ssrc=1959A317
May 23 14:08:39.316: RTP(64807): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1560, ssrc=1959A317
May 23 14:08:39.316: RTP(64808): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1600, ssrc=1959A317
May 23 14:08:39.316: RTP(64809): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF16A0, ssrc=1959A317
May 23 14:08:39.320: RTP(64810): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1740, ssrc=1959A317
May 23 14:08:39.320: RTP(64811): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF17E0, ssrc=1959A317
May 23 14:08:39.320: RTP(64812): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1880, ssrc=1959A317
May 23 14:08:39.320: RTP(64813): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1920, ssrc=1959A317
May 23 14:08:39.324: RTP(64814): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF19C0, ssrc=1959A317
May 23 14:08:39.324: RTP(64815): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1A60, ssrc=1959A317
May 23 14:08:39.324: RTP(64816): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1B00, ssrc=1959A317
May 23 14:08:39.328: RTP(64817): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1BA0, ssrc=1959A317
May 23 14:08:39.328: RTP(64818): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1C40, ssrc=1959A317
May 23 14:08:39.328: RTP(64819): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1CE0, ssrc=1959A317
May 23 14:08:39.328: RTP(64820): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1D80, ssrc=1959A317
May 23 14:08:39.332: RTP(64821): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1E20, ssrc=1959A317
May 23 14:08:39.332: RTP(64822): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1EC0, ssrc=1959A317
May 23 14:08:39.332: RTP(64823): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF1F60, ssrc=1959A317
May 23 14:08:39.332: RTP(64824): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2000, ssrc=1959A317
May 23 14:08:39.336: RTP(64825): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF20A0, ssrc=1959A317
May 23 14:08:39.336: RTP(64826): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2140, ssrc=1959A317
May 23 14:08:39.336: RTP(64827): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF21E0, ssrc=1959A317
May 23 14:08:39.340: RTP(64828): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2280, ssrc=1959A317
May 23 14:08:39.340: RTP(64829): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2320, ssrc=1959A317
May 23 14:08:39.340: RTP(64830): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF23C0, ssrc=1959A317
May 23 14:08:39.340: RTP(64831): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2460, ssrc=1959A317
May 23 14:08:39.344: RTP(64832): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2500, ssrc=1959A317
May 23 14:08:39.344: RTP(64833): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF25A0, ssrc=1959A317
May 23 14:08:39.344: RTP(64834): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2640, ssrc=1959A317
May 23 14:08:39.348: RTP(64835): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF26E0, ssrc=1959A317
May 23 14:08:39.348: RTP(64836): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2780, ssrc=1959A317
May 23 14:08:39.348: RTP(64837): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2820, ssrc=1959A317
May 23 14:08:39.348: RTP(64838): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF28C0, ssrc=1959A317
May 23 14:08:39.352: RTP(64839): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2960, ssrc=1959A317
May 23 14:08:39.352: RTP(64840): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2A00, ssrc=1959A317
May 23 14:08:39.360: RTP(64841): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2AA0, ssrc=1959A317
May 23 14:08:39.384: RTP(64842): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2B40, ssrc=1959A317
May 23 14:08:39.384: RTP(64843): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2BE0, ssrc=1959A317
May 23 14:08:39.400: RTP(879): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26D40, ssrc=BCA
May 23 14:08:39.404: RTP(64844): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2C80, ssrc=1959A317
May 23 14:08:39.420: RTP(880): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26DE0, ssrc=BCA
May 23 14:08:39.428: RTP(64845): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2D20, ssrc=1959A317
May 23 14:08:39.440: RTP(881): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26E80, ssrc=BCA
May 23 14:08:39.444: RTP(64846): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2DC0, ssrc=1959A317
May 23 14:08:39.460: RTP(882): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26F20, ssrc=BCA
May 23 14:08:39.472: RTP(64847): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2E60, ssrc=1959A317
May 23 14:08:39.480: RTP(883): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=26FC0, ssrc=BCA
May 23 14:08:39.488: RTP(64848): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2F00, ssrc=1959A317
May 23 14:08:39.501: RTP(884): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=27060, ssrc=BCA
May 23 14:08:39.521: RTP(885): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=27100, ssrc=BCA
May 23 14:08:39.525: RTP(64849): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF2FA0, ssrc=1959A317
May 23 14:08:39.541: RTP(886): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=271A0, ssrc=BCA
May 23 14:08:39.541: RTP(64850): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3040, ssrc=1959A317
May 23 14:08:39.561: RTP(887): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=27240, ssrc=BCA
May 23 14:08:39.569: RTP(64851): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF30E0, ssrc=1959A317
May 23 14:08:39.573: RTP(64852): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3180, ssrc=1959A317
May 23 14:08:39.573: RTP(888): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=13, ts=272E0, ssrc=BCA
May 23 14:08:39.909: RTP(64868): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3B80, ssrc=1959A317
May 23 14:08:39.909: RTP(64869): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3C20, ssrc=1959A317
May 23 14:08:40.582: RTP(64870): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3CC0, ssrc=1959A317
May 23 14:08:40.582: RTP(64871): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3D60, ssrc=1959A317
May 23 14:08:40.586: RTP(64872): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3E00, ssrc=1959A317
May 23 14:08:40.586: RTP(64873): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3EA0, ssrc=1959A317
May 23 14:08:40.586: RTP(64874): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3F40, ssrc=1959A317
May 23 14:08:40.586: RTP(64875): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF3FE0, ssrc=1959A317
May 23 14:08:40.590: RTP(64876): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4080, ssrc=1959A317
May 23 14:08:40.590: RTP(64877): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4120, ssrc=1959A317
May 23 14:08:40.590: RTP(64878): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF41C0, ssrc=1959A317
May 23 14:08:40.594: RTP(64879): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4260, ssrc=1959A317
May 23 14:08:40.594: RTP(64880): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4300, ssrc=1959A317
May 23 14:08:40.594: RTP(64881): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF43A0, ssrc=1959A317
May 23 14:08:40.594: RTP(64882): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4440, ssrc=1959A317
May 23 14:08:40.598: RTP(64883): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF44E0, ssrc=1959A317
May 23 14:08:40.598: RTP(64884): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4580, ssrc=1959A317
May 23 14:08:40.598: RTP(64885): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4620, ssrc=1959A317
May 23 14:08:40.602: RTP(64886): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF46C0, ssrc=1959A317
May 23 14:08:40.602: RTP(64887): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4760, ssrc=1959A317
May 23 14:08:40.602: RTP(64888): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4800, ssrc=1959A317
May 23 14:08:40.602: RTP(64889): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF48A0, ssrc=1959A317
May 23 14:08:40.606: RTP(64890): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4940, ssrc=1959A317
May 23 14:08:40.606: RTP(64891): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF49E0, ssrc=1959A317
May 23 14:08:40.606: RTP(64892): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4A80, ssrc=1959A317
May 23 14:08:40.610: RTP(64893): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4B20, ssrc=1959A317
May 23 14:08:40.610: RTP(64894): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4BC0, ssrc=1959A317
May 23 14:08:40.610: RTP(64895): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4C60, ssrc=1959A317
May 23 14:08:40.610: RTP(64896): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4D00, ssrc=1959A317
May 23 14:08:40.614: RTP(64897): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4DA0, ssrc=1959A317
May 23 14:08:40.614: RTP(64898): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4E40, ssrc=1959A317
May 23 14:08:40.614: RTP(64899): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4EE0, ssrc=1959A317
May 23 14:08:40.618: RTP(64900): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF4F80, ssrc=1959A317
May 23 14:08:40.618: RTP(64901): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5020, ssrc=1959A317
May 23 14:08:40.618: RTP(64902): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF50C0, ssrc=1959A317
May 23 14:08:40.618: RTP(64903): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5160, ssrc=1959A317
May 23 14:08:40.622: RTP(64904): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5200, ssrc=1959A317
May 23 14:08:40.622: RTP(889): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28550, ssrc=BCA
May 23 14:08:40.622: RTP(890): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=285F0, ssrc=BCA
May 23 14:08:40.626: RTP(891): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28690, ssrc=BCA
May 23 14:08:40.626: RTP(892): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28730, ssrc=BCA
May 23 14:08:40.626: RTP(893): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=287D0, ssrc=BCA
May 23 14:08:40.626: RTP(894): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28870, ssrc=BCA
May 23 14:08:40.630: RTP(895): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28910, ssrc=BCA
May 23 14:08:40.630: RTP(896): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=289B0, ssrc=BCA
May 23 14:08:40.630: RTP(897): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28A50, ssrc=BCA
May 23 14:08:40.630: RTP(898): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28AF0, ssrc=BCA
May 23 14:08:40.634: RTP(899): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28B90, ssrc=BCA
May 23 14:08:40.634: RTP(900): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28C30, ssrc=BCA
May 23 14:08:40.634: RTP(901): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28CD0, ssrc=BCA
May 23 14:08:40.634: RTP(902): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28D70, ssrc=BCA
May 23 14:08:40.638: RTP(903): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28E10, ssrc=BCA
May 23 14:08:40.638: RTP(904): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28EB0, ssrc=BCA
May 23 14:08:40.638: RTP(905): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28F50, ssrc=BCA
May 23 14:08:40.642: RTP(906): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=28FF0, ssrc=BCA
May 23 14:08:40.642: RTP(907): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=29090, ssrc=BCA
May 23 14:08:40.642: RTP(908): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=29130, ssrc=BCA
May 23 14:08:40.642: RTP(909): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=291D0, ssrc=BCA
May 23 14:08:40.646: RTP(910): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=29270, ssrc=BCA
May 23 14:08:40.646: RTP(911): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=29310, ssrc=BCA
May 23 14:08:40.646: RTP(912): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=293B0, ssrc=BCA
May 23 14:08:40.646: RTP(64905): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF52A0, ssrc=1959A317
May 23 14:08:40.650: RTP(913): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=29450, ssrc=BCA
May 23 14:08:40.662: RTP(64906): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5340, ssrc=1959A317
May 23 14:08:40.670: RTP(914): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=294F0, ssrc=BCA
May 23 14:08:40.678: RTP(64907): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF53E0, ssrc=1959A317
May 23 14:08:40.690: RTP(915): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=29590, ssrc=BCA
May 23 14:08:40.711: RTP(916): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=29630, ssrc=BCA
May 23 14:08:40.731: RTP(917): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=296D0, ssrc=BCA
May 23 14:08:40.751: RTP(918): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=29770, ssrc=BCA
May 23 14:08:40.759: RTP(64908): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5480, ssrc=1959A317
May 23 14:08:40.771: RTP(919): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=29810, ssrc=BCA
May 23 14:08:40.771: RTP(920): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=13, ts=29860, ssrc=BCA
May 23 14:08:40.791: RTP(64909): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5520, ssrc=1959A317
May 23 14:08:40.799: RTP(64910): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF55C0, ssrc=1959A317
May 23 14:08:40.827: RTP(64911): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5660, ssrc=1959A317
May 23 14:08:40.839: RTP(64912): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5700, ssrc=1959A317
May 23 14:08:40.863: RTP(64913): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF57A0, ssrc=1959A317
May 23 14:08:40.883: RTP(64914): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5840, ssrc=1959A317
May 23 14:08:40.903: RTP(64915): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF58E0, ssrc=1959A317
May 23 14:08:40.923: RTP(64916): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5980, ssrc=1959A317
May 23 14:08:40.923: voip_rtcp_stats_req: Calculated remaining disc time is 0 msec...Media Inactivity Criteria is 0
May 23 14:08:41.175: RTP(64917): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5A20, ssrc=1959A317
May 23 14:08:41.179: RTP(64918): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5AC0, ssrc=1959A317
May 23 14:08:41.179: RTP(64919): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5B60, ssrc=1959A317
May 23 14:08:41.179: RTP(64920): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5C00, ssrc=1959A317
May 23 14:08:41.183: RTP(64921): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5CA0, ssrc=1959A317
May 23 14:08:41.183: RTP(64922): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5D40, ssrc=1959A317
May 23 14:08:41.183: RTP(64923): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5DE0, ssrc=1959A317
May 23 14:08:41.183: RTP(64924): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5E80, ssrc=1959A317
May 23 14:08:41.187: RTP(64925): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF5F20, ssrc=1959A317
May 23 14:08:42.053: RTP(929): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2B160, ssrc=BCA
May 23 14:08:42.057: RTP(930): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=13, ts=2B200, ssrc=BCA
May 23 14:08:42.057: RTP(931): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2BCF0, ssrc=BCA
May 23 14:08:42.057: RTP(932): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2BD90, ssrc=BCA
May 23 14:08:42.057: RTP(933): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2BE30, ssrc=BCA
May 23 14:08:42.061: RTP(934): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2BED0, ssrc=BCA
May 23 14:08:42.061: RTP(935): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2BF70, ssrc=BCA
May 23 14:08:42.061: RTP(936): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C010, ssrc=BCA
May 23 14:08:42.738: RTP(64973): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF7D20, ssrc=1959A317
May 23 14:08:42.738: RTP(64974): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF7DC0, ssrc=1959A317
May 23 14:08:42.742: RTP(64975): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF7E60, ssrc=1959A317
May 23 14:08:42.742: RTP(64976): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF7F00, ssrc=1959A317
May 23 14:08:42.742: RTP(64977): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF7FA0, ssrc=1959A317
May 23 14:08:42.746: RTP(64978): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8040, ssrc=1959A317
May 23 14:08:42.746: RTP(64979): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF80E0, ssrc=1959A317
May 23 14:08:42.746: RTP(64980): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8180, ssrc=1959A317
May 23 14:08:42.746: RTP(64981): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8220, ssrc=1959A317
May 23 14:08:42.750: RTP(64982): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF82C0, ssrc=1959A317
May 23 14:08:42.750: RTP(64983): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8360, ssrc=1959A317
May 23 14:08:42.750: RTP(64984): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8400, ssrc=1959A317
May 23 14:08:42.754: RTP(64985): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF84A0, ssrc=1959A317
May 23 14:08:42.754: RTP(64986): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8540, ssrc=1959A317
May 23 14:08:42.754: RTP(64987): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF85E0, ssrc=1959A317
May 23 14:08:42.754: RTP(64988): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8680, ssrc=1959A317
May 23 14:08:42.758: RTP(64989): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8720, ssrc=1959A317
May 23 14:08:42.758: RTP(64990): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF87C0, ssrc=1959A317
May 23 14:08:42.758: RTP(64991): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8860, ssrc=1959A317
May 23 14:08:42.762: RTP(64992): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8900, ssrc=1959A317
May 23 14:08:42.762: RTP(64993): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF89A0, ssrc=1959A317
May 23 14:08:42.762: RTP(64994): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8A40, ssrc=1959A317
May 23 14:08:42.762: RTP(64995): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8AE0, ssrc=1959A317
May 23 14:08:42.766: RTP(64996): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8B80, ssrc=1959A317
May 23 14:08:42.766: RTP(64997): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8C20, ssrc=1959A317
May 23 14:08:42.766: RTP(64998): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8CC0, ssrc=1959A317
May 23 14:08:42.770: RTP(64999): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8D60, ssrc=1959A317
May 23 14:08:42.770: RTP(65000): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8E00, ssrc=1959A317
May 23 14:08:42.770: RTP(65001): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8EA0, ssrc=1959A317
May 23 14:08:42.770: RTP(65002): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8F40, ssrc=1959A317
May 23 14:08:42.774: RTP(65003): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF8FE0, ssrc=1959A317
May 23 14:08:42.774: RTP(65004): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9080, ssrc=1959A317
May 23 14:08:42.774: RTP(65005): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9120, ssrc=1959A317
May 23 14:08:42.778: RTP(65006): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF91C0, ssrc=1959A317
May 23 14:08:42.778: RTP(65007): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9260, ssrc=1959A317
May 23 14:08:42.778: RTP(65008): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9300, ssrc=1959A317
May 23 14:08:42.778: RTP(65009): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF93A0, ssrc=1959A317
May 23 14:08:42.782: RTP(937): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C0B0, ssrc=BCA
May 23 14:08:42.782: RTP(938): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C150, ssrc=BCA
May 23 14:08:42.782: RTP(939): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C1F0, ssrc=BCA
May 23 14:08:42.782: RTP(940): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C290, ssrc=BCA
May 23 14:08:42.786: RTP(941): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C330, ssrc=BCA
May 23 14:08:42.786: RTP(942): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C3D0, ssrc=BCA
May 23 14:08:42.786: RTP(943): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C470, ssrc=BCA
May 23 14:08:42.790: RTP(944): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2C510, ssrc=BCA
May 23 14:08:42.790: RTP(945): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=13, ts=2C5B0, ssrc=BCA
May 23 14:08:42.790: RTP(946): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2CCE0, ssrc=BCA
May 23 14:08:42.790: RTP(947): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2CD80, ssrc=BCA
May 23 14:08:42.794: RTP(948): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2CE20, ssrc=BCA
May 23 14:08:42.794: RTP(949): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2CEC0, ssrc=BCA
May 23 14:08:42.794: RTP(950): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2CF60, ssrc=BCA
May 23 14:08:42.794: RTP(951): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D000, ssrc=BCA
May 23 14:08:42.798: RTP(952): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D0A0, ssrc=BCA
May 23 14:08:42.798: RTP(953): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D140, ssrc=BCA
May 23 14:08:42.798: RTP(954): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D1E0, ssrc=BCA
May 23 14:08:42.798: RTP(955): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D280, ssrc=BCA
May 23 14:08:42.802: RTP(956): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D320, ssrc=BCA
May 23 14:08:42.802: RTP(957): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D3C0, ssrc=BCA
May 23 14:08:42.802: RTP(958): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D460, ssrc=BCA
May 23 14:08:42.802: RTP(959): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D500, ssrc=BCA
May 23 14:08:42.806: RTP(960): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D5A0, ssrc=BCA
May 23 14:08:42.806: RTP(961): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D640, ssrc=BCA
May 23 14:08:42.806: RTP(962): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D6E0, ssrc=BCA
May 23 14:08:42.806: RTP(963): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D780, ssrc=BCA
May 23 14:08:42.810: RTP(65010): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9440, ssrc=1959A317
May 23 14:08:42.810: RTP(65011): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF94E0, ssrc=1959A317
May 23 14:08:42.822: RTP(964): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D820, ssrc=BCA
May 23 14:08:42.838: RTP(65012): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9580, ssrc=1959A317
May 23 14:08:42.842: RTP(965): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D8C0, ssrc=BCA
May 23 14:08:42.854: RTP(65013): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9620, ssrc=1959A317
May 23 14:08:42.862: RTP(966): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2D960, ssrc=BCA
May 23 14:08:42.878: RTP(65014): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF96C0, ssrc=1959A317
May 23 14:08:42.882: RTP(967): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2DA00, ssrc=BCA
May 23 14:08:42.898: RTP(65015): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9760, ssrc=1959A317
May 23 14:08:42.898: RTP(968): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2DAA0, ssrc=BCA
May 23 14:08:42.918: RTP(65016): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9800, ssrc=1959A317
May 23 14:08:42.918: RTP(969): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2DB40, ssrc=BCA
May 23 14:08:42.930: RTP(65017): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF98A0, ssrc=1959A317
May 23 14:08:42.938: RTP(970): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2DBE0, ssrc=BCA
May 23 14:08:42.954: RTP(65018): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9940, ssrc=1959A317
May 23 14:08:42.958: RTP(971): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2DC80, ssrc=BCA
May 23 14:08:42.970: RTP(65019): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF99E0, ssrc=1959A317
May 23 14:08:43.283: RTP(983): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2E400, ssrc=BCA
May 23 14:08:43.287: RTP(984): fs rx s=192.168.5.2(41000), d=192.168.5.223(16400), pt=8, ts=2E4A0, ssrc=BCA
May 23 14:08:43.287: RTP(65025): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9DA0, ssrc=1959A317
May 23 14:08:43.287: RTP(65026): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9E40, ssrc=1959A317
May 23 14:08:43.287: RTP(65027): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9EE0, ssrc=1959A317
May 23 14:08:43.291: RTP(65028): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFF9F80, ssrc=1959A317
May 23 14:08:43.291: RTP(65029): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA020, ssrc=1959A317
May 23 14:08:43.291: RTP(65030): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA0C0, ssrc=1959A317
May 23 14:08:43.295: RTP(65031): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA160, ssrc=1959A317
May 23 14:08:43.295: RTP(65032): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA200, ssrc=1959A317
May 23 14:08:43.295: RTP(65033): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA2A0, ssrc=1959A317
May 23 14:08:43.295: RTP(65034): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA340, ssrc=1959A317
May 23 14:08:43.299: RTP(65035): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA3E0, ssrc=1959A317
May 23 14:08:43.299: RTP(65036): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA480, ssrc=1959A317
May 23 14:08:43.299: RTP(65037): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA520, ssrc=1959A317
May 23 14:08:43.303: RTP(65038): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA5C0, ssrc=1959A317
May 23 14:08:43.303: RTP(65039): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA660, ssrc=1959A317
May 23 14:08:43.319: voip_rtp_update_callinfo: first_pak=TRUE, new_ssrc=0, type=3
May 23 14:08:43.319:  voip_rtp_update_xmit_info
May 23 14:08:43.319:  voip_rtp_update_xmit_info, dstvdbptr: 859FBC20, dstCallID 2407, gccb: 843F5B10, xmitFunc 0,context 0
May 23 14:08:43.323:  voip_rtp_update_xmit_info xmit_func is NULL
May 23 14:08:43.323: voip_rtp_stop_disc_timer
May 23 14:08:43.331: voip_rtp_stop_disc_timer
May 23 14:08:43.331: voip_rtcp_remove_ccb
May 23 14:08:43.339: RTP(65040): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA700, ssrc=1959A317
May 23 14:08:43.343: voip_rtcp_close_session
May 23 14:08:43.343: voip_rtcp_stop_session:
May 23 14:08:43.343: voip_rtcp_stop_session: stop session
May 23 14:08:43.343: Cleaning up sess=846F43F8, sp=16400, dp=41000
May 23 14:08:43.363: voip_rtp_update_callinfo: first_pak=TRUE, new_ssrc=0, type=3
May 23 14:08:43.367:  voip_rtp_update_xmit_info
May 23 14:08:43.367:  voip_rtp_update_xmit_info, dstvdbptr: 859FBF14, dstCallID 2406, gccb: 83CBADB0, xmitFunc 0,context 0
May 23 14:08:43.367: RTP(65041): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA7A0, ssrc=1959A317
May 23 14:08:43.367:  voip_rtp_update_xmit_info xmit_func is NULL
May 23 14:08:43.367: voip_rtp_stop_disc_timer
May 23 14:08:43.375: RTP(65042): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA840, ssrc=1959A317
May 23 14:08:43.387: RTP(65043): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA8E0, ssrc=1959A317
May 23 14:08:44.092: RTP(65044): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFA980, ssrc=1959A317
May 23 14:08:44.092: RTP(65045): fs rx s=82.215.172.192(17532), d=82.215.133.140(17464), pt=8, ts=4EFFAA20, ssrc=1959A317
May 23 14:08:44.357: voip_rtp_stop_disc_timer
May 23 14:08:44.357: voip_rtcp_remove_ccb
May 23 14:08:44.365: voip_rtcp_close_session
May 23 14:08:44.365: voip_rtcp_stop_session:
May 23 14:08:44.365: voip_rtcp_stop_session: stop session
May 23 14:08:44.369: Cleaning up sess=83CC827C, sp=17464, dp=17532
cme-test#

From toddnh65 at gmail.com  Wed May 23 10:50:38 2007
From: toddnh65 at gmail.com (Todd Franklin)
Date: Wed, 23 May 2007 10:50:38 -0400
Subject: [cisco-voip] Any Dell POE Switches
Message-ID: <6101bda50705230750t9fd92ado573393ef79a6eb3a@mail.gmail.com>

Anyone using Dell POE switches in their Cisco IP Phone setup?
Dell just quoted me an eye-popping number on their POE switches.
I may try one and just set up a few phones that are non-critical and see how
it goes (*if* it goes, lol)
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From smgustafson at gmail.com  Wed May 23 11:13:59 2007
From: smgustafson at gmail.com (Steve G)
Date: Wed, 23 May 2007 08:13:59 -0700
Subject: [cisco-voip] iDivert on second line
Message-ID: <8a1f34b50705230813k9c66d03sb68d9f220dc74887@mail.gmail.com>

Has anyone been able to get iDivert to work when you are talking on line1
and a new call hunts to your line2?  Is it possible?
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From cisco at adman.net  Wed May 23 11:20:30 2007
From: cisco at adman.net (Adam)
Date: Wed, 23 May 2007 10:20:30 -0500
Subject: [cisco-voip] Any Dell POE Switches
In-Reply-To: <6101bda50705230750t9fd92ado573393ef79a6eb3a@mail.gmail.com>
References: <6101bda50705230750t9fd92ado573393ef79a6eb3a@mail.gmail.com>
Message-ID: <c6f32ce40705230820y58eea086y7f7abba8b50d1515@mail.gmail.com>

Yes I have a customer who has phones running on a Dell switch. The two
things to keep in mind are that the Dell will only provide 802.3af
POE, so if you have 7940s, 7960s or any of the other original phones
it will not work as they require the Cisco pre-standard power. The
second consideration is that without CDP you will have no way to to
automatically configure the separate data and voice VLANs on the phone
if you want to segment that traffic. You can manually configure the
phone to use the VLANs correctly, but it is no longer an hands off
process. If you can live with those two caveats then the price is
significantly lower than the Cisco product... personally this is not a
configuration that I would recommend to a customer, but sometimes they
already have other switches in place.

-Adam

On 5/23/07, Todd Franklin <toddnh65 at gmail.com> wrote:
> Anyone using Dell POE switches in their Cisco IP Phone setup?
> Dell just quoted me an eye-popping number on their POE switches.
> I may try one and just set up a few phones that are non-critical and see how
> it goes (*if* it goes, lol)
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From Scott.Voll at wesd.org  Wed May 23 11:23:33 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 23 May 2007 08:23:33 -0700
Subject: [cisco-voip] Maximum outgoing call duration
In-Reply-To: <003401c79d34$3ae588d0$47196883@cfs.uoguelph.ca>
References: <46542596.7000909@carocomp.ro>
	<003401c79d34$3ae588d0$47196883@cfs.uoguelph.ca>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7D8@Apollo.wesd.org>

I was on a tac call for 12 hours when I found out about the max call
duration parameter and had to call tac back........

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, May 23, 2007 5:17 AM
To: Kelemen Zoltan; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Maximum outgoing call duration

 

Scott Voll can tell you ALL about this......

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."
WJR

	----- Original Message ----- 

	From: Kelemen Zoltan <mailto:keli at carocomp.ro>  

	To: cisco-voip at puck.nether.net 

	Sent: Wednesday, May 23, 2007 7:29 AM

	Subject: [cisco-voip] Maximum outgoing call duration

	 

	Hi!
	
	I have a client with CallManager 4.1 installed, who wants to
limit 
	outgoing calls (preferably only on some trunks) to a certain
length 
	(Like no call can last longer than 7 minutes, for example.)
	
	Is that possible? If so, how? (even third-party solutions would
be 
	interesting, but only if there's no other way)
	
	thanks,
	  Zoltan
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

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From Scott.Voll at wesd.org  Wed May 23 12:11:31 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 23 May 2007 09:11:31 -0700
Subject: [cisco-voip] FW: Cisco Security Response: Cisco CallManager Input
	ValidationVulnerability
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7E4@Apollo.wesd.org>

Hey--

Ryan, Wes, or Mike....... do you have a time line on CM 4.1.3sr5?  and
can I upgrade from ES100 to SR5?

Thanks

Scott

-----Original Message-----
From: cust-security-announce-bounces at cisco.com
[mailto:cust-security-announce-bounces at cisco.com] On Behalf Of Cisco
Systems Product Security Incident Response Team
Sent: Wednesday, May 23, 2007 8:47 AM
To: cust-security-announce at cisco.com
Subject: Cisco Security Response: Cisco CallManager Input
ValidationVulnerability

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Cisco Security Response: Cisco CallManager Input Validation
Vulnerability

http://www.cisco.com/warp/public/707/cisco-sr-20070523-ccm.shtml

Revision 1.0

For Public Release 2007 May 23 1600 UTC (GMT)

-
-----------------------------------------------------------------------

Cisco Response
==============

This is Cisco PSIRT's response to the statements made by Marc Ruef and
Stefan Friedi from scip AG in their message "Cisco CallManager 4.1 Input
Validation Vulnerability," posted on 2007 May 23 at 1600 UTC (GMT).

The original emails were posted to BugTraq and Full-Disclosure.

In their postings, Marc Ruef and Stefan Friedi illustrate how to bypass
the web application firewall used in Cisco CallManager. This means of
bypass can be used to display graphics, scripts, or other information
downloaded from an external web site. This technique may also be used to
conduct cross-site scripting attacks. Cisco confirms that the example
the authors Ruef and Friedi provided bypasses the web application
firewall and that there may be other methods for bypassing the web
application firewall.

Cisco has made improvements to the input validation mechanisms in
CallManager that may mitigate the risks associated with this security
vulnerability. These improvements have been incorporated into 4.2(3)sr2.
Future releases, 3.3(5)sr3, 4.1(3)sr5 and 4.3(1)sr1, will also include
the improvements made to address this bug. This issue is being tracked
by the following Cisco Bug ID:

  * CSCsi12374 - Improvements in User Input Validation

Service releases of CallManager software are available at the following
link:

http://www.cisco.com/public/sw-center/sw-voice.shtml

Additional Information
======================

Cisco CallManager is the software-based call-processing component of the
Cisco IP telephony solution that extends enterprise telephony features
and functions to packet telephony network devices, such as IP phones,
media processing devices, voice-over-IP (VoIP) gateways, and multimedia
applications. The vulnerability described in this response exists in
the web application firewall used in CallManager. This feature is
designed to prevent users from entering malicious code into the input
fields used in CallManager forms. The vulnerability exists because the
web application firewall fails to properly sanitize some potentially
malicious tags.

To exploit these issues an attacker must convince an authenticated
user to follow a specially crafted, malicious URL. A successful attack
may result in the execution of arbitrary script code in the user's web
browser.

For additional information on cross-site scripting (XSS) attacks and the
methods used to exploit such vulnerabilities, please refer to the Cisco
Applied Intelligence Response "Understanding Cross-Site Scripting (XSS)
Threat Vectors," which is available at the following link:

http://www.cisco.com/en/US/products/ps6120/tsd_products_security_respons
e09186a008073f7b3.html

The Cisco PSIRT is not aware of any malicious use of the vulnerability
described in this document.

This issue was reported to Cisco by Marc Ruef and Stefan Friedi from
scip AG. We would like to thank Marc Ruef and Stefan Friedi for bringing
this issue to our attention and for working with us toward coordinated
disclosure of the issue. We greatly appreciate the opportunity to
work with researchers on security vulnerabilities, and welcome the
opportunity to review and assist in product reports.

THIS DOCUMENT IS PROVIDED ON AN "AS IS" BASIS AND DOES NOT IMPLY
ANY KIND OF GUARANTEE OR WARRANTY, INCLUDING THE WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR USE. YOUR USE OF THE
INFORMATION ON THE DOCUMENT OR MATERIALS LINKED FROM THE DOCUMENT IS
AT YOUR OWN RISK. CISCO RESERVES THE RIGHT TO CHANGE OR UPDATE THIS
DOCUMENT AT ANY TIME.

Revision History
================

+-------------------------------------------------------------+
| Revision 1.0   | 2007-May-23   | Initial public release.    |
+-------------------------------------------------------------+

Cisco Security Procedures
=========================

Complete information on reporting security vulnerabilities in Cisco
products, obtaining assistance with security incidents, and registering
to receive security information from Cisco, is available on Cisco's
worldwide website at
http://www.cisco.com/en/US/products/products_security_vulnerability_poli
cy.html.
This includes instructions for press inquiries regarding Cisco security
notices. All Cisco security advisories are available at
http://www.cisco.com/go/psirt.

-
-----------------------------------------------------------------------
All contents are Copyright (C) 2006-2007 Cisco Systems, Inc. All rights
reserved.
-
-----------------------------------------------------------------------

Updated: May 23, 2007                                Document ID: 82462

-
-----------------------------------------------------------------------
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zWQKjcIn7Qw9+OpSHfB7MMc=
=mA2X
-----END PGP SIGNATURE-----
_______________________________________________
cust-security-announce mailing list
cust-security-announce at cisco.com
To unsubscribe, send the command "unsubscribe" in the subject of your
message to cust-security-announce-leave at cisco.com

From jgrace at adcapnet.com  Wed May 23 12:21:13 2007
From: jgrace at adcapnet.com (James Grace)
Date: Wed, 23 May 2007 12:21:13 -0400
Subject: [cisco-voip] how do you set mgcp to 5ess
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C663@mail.adcapnetsvc.local>

?
how do you set mgcp to 5ess  trying to turn up a pri with switch type 5ess  .  for some resome when we reboot the route it reverts back to primary-NI
 
and on the callmanager theres no 5ess
have anybody seen this
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
recipient is prohibited. If you received this message in error, please contact the sender and delete the
material from all computers."

This original email was sent to the internet for delivery at 23 May 2007 12:22:06 -0400
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From jsteinberg at gmail.com  Wed May 23 12:27:18 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 23 May 2007 12:27:18 -0400
Subject: [cisco-voip] FW: Cisco Security Response: Cisco CallManager
	Input ValidationVulnerability
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE7E4@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A608DEE7E4@Apollo.wesd.org>
Message-ID: <f1b65f880705230927u3a91f044pfb8a15b586193146@mail.gmail.com>

I saw ccm 4.1.3sr5, 4.2.3sr2a, and 5.1.2 posted on CCO today.

-Justin

On 5/23/07, Voll, Scott <Scott.Voll at wesd.org> wrote:
> Hey--
>
> Ryan, Wes, or Mike....... do you have a time line on CM 4.1.3sr5?  and
> can I upgrade from ES100 to SR5?
>
> Thanks
>
> Scott
>
> -----Original Message-----
> From: cust-security-announce-bounces at cisco.com
> [mailto:cust-security-announce-bounces at cisco.com] On Behalf Of Cisco
> Systems Product Security Incident Response Team
> Sent: Wednesday, May 23, 2007 8:47 AM
> To: cust-security-announce at cisco.com
> Subject: Cisco Security Response: Cisco CallManager Input
> ValidationVulnerability
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Cisco Security Response: Cisco CallManager Input Validation
> Vulnerability
>
> http://www.cisco.com/warp/public/707/cisco-sr-20070523-ccm.shtml
>
> Revision 1.0
>
> For Public Release 2007 May 23 1600 UTC (GMT)
>
> -
> -----------------------------------------------------------------------
>
> Cisco Response
> ==============
>
> This is Cisco PSIRT's response to the statements made by Marc Ruef and
> Stefan Friedi from scip AG in their message "Cisco CallManager 4.1 Input
> Validation Vulnerability," posted on 2007 May 23 at 1600 UTC (GMT).
>
> The original emails were posted to BugTraq and Full-Disclosure.
>
> In their postings, Marc Ruef and Stefan Friedi illustrate how to bypass
> the web application firewall used in Cisco CallManager. This means of
> bypass can be used to display graphics, scripts, or other information
> downloaded from an external web site. This technique may also be used to
> conduct cross-site scripting attacks. Cisco confirms that the example
> the authors Ruef and Friedi provided bypasses the web application
> firewall and that there may be other methods for bypassing the web
> application firewall.
>
> Cisco has made improvements to the input validation mechanisms in
> CallManager that may mitigate the risks associated with this security
> vulnerability. These improvements have been incorporated into 4.2(3)sr2.
> Future releases, 3.3(5)sr3, 4.1(3)sr5 and 4.3(1)sr1, will also include
> the improvements made to address this bug. This issue is being tracked
> by the following Cisco Bug ID:
>
>   * CSCsi12374 - Improvements in User Input Validation
>
> Service releases of CallManager software are available at the following
> link:
>
> http://www.cisco.com/public/sw-center/sw-voice.shtml
>
> Additional Information
> ======================
>
> Cisco CallManager is the software-based call-processing component of the
> Cisco IP telephony solution that extends enterprise telephony features
> and functions to packet telephony network devices, such as IP phones,
> media processing devices, voice-over-IP (VoIP) gateways, and multimedia
> applications. The vulnerability described in this response exists in
> the web application firewall used in CallManager. This feature is
> designed to prevent users from entering malicious code into the input
> fields used in CallManager forms. The vulnerability exists because the
> web application firewall fails to properly sanitize some potentially
> malicious tags.
>
> To exploit these issues an attacker must convince an authenticated
> user to follow a specially crafted, malicious URL. A successful attack
> may result in the execution of arbitrary script code in the user's web
> browser.
>
> For additional information on cross-site scripting (XSS) attacks and the
> methods used to exploit such vulnerabilities, please refer to the Cisco
> Applied Intelligence Response "Understanding Cross-Site Scripting (XSS)
> Threat Vectors," which is available at the following link:
>
> http://www.cisco.com/en/US/products/ps6120/tsd_products_security_respons
> e09186a008073f7b3.html
>
> The Cisco PSIRT is not aware of any malicious use of the vulnerability
> described in this document.
>
> This issue was reported to Cisco by Marc Ruef and Stefan Friedi from
> scip AG. We would like to thank Marc Ruef and Stefan Friedi for bringing
> this issue to our attention and for working with us toward coordinated
> disclosure of the issue. We greatly appreciate the opportunity to
> work with researchers on security vulnerabilities, and welcome the
> opportunity to review and assist in product reports.
>
> THIS DOCUMENT IS PROVIDED ON AN "AS IS" BASIS AND DOES NOT IMPLY
> ANY KIND OF GUARANTEE OR WARRANTY, INCLUDING THE WARRANTIES OF
> MERCHANTABILITY OR FITNESS FOR A PARTICULAR USE. YOUR USE OF THE
> INFORMATION ON THE DOCUMENT OR MATERIALS LINKED FROM THE DOCUMENT IS
> AT YOUR OWN RISK. CISCO RESERVES THE RIGHT TO CHANGE OR UPDATE THIS
> DOCUMENT AT ANY TIME.
>
> Revision History
> ================
>
> +-------------------------------------------------------------+
> | Revision 1.0   | 2007-May-23   | Initial public release.    |
> +-------------------------------------------------------------+
>
> Cisco Security Procedures
> =========================
>
> Complete information on reporting security vulnerabilities in Cisco
> products, obtaining assistance with security incidents, and registering
> to receive security information from Cisco, is available on Cisco's
> worldwide website at
> http://www.cisco.com/en/US/products/products_security_vulnerability_poli
> cy.html.
> This includes instructions for press inquiries regarding Cisco security
> notices. All Cisco security advisories are available at
> http://www.cisco.com/go/psirt.
>
> -
> -----------------------------------------------------------------------
> All contents are Copyright (C) 2006-2007 Cisco Systems, Inc. All rights
> reserved.
> -
> -----------------------------------------------------------------------
>
> Updated: May 23, 2007                                Document ID: 82462
>
> -
> -----------------------------------------------------------------------
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.6 (GNU/Linux)
>
> iD8DBQFGVGFn8NUAbBmDaxQRAtqYAJ0QYVxfjNxM+fNHZ//KYPOdfmsxaQCfZZRC
> zWQKjcIn7Qw9+OpSHfB7MMc=
> =mA2X
> -----END PGP SIGNATURE-----
> _______________________________________________
> cust-security-announce mailing list
> cust-security-announce at cisco.com
> To unsubscribe, send the command "unsubscribe" in the subject of your
> message to cust-security-announce-leave at cisco.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From rratliff at cisco.com  Wed May 23 12:30:09 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 23 May 2007 12:30:09 -0400
Subject: [cisco-voip] FW: Cisco Security Response: Cisco CallManager
	Input ValidationVulnerability
In-Reply-To: <f1b65f880705230927u3a91f044pfb8a15b586193146@mail.gmail.com>
References: <407055A92CECCB499C922A2D35FC19A608DEE7E4@Apollo.wesd.org>
	<f1b65f880705230927u3a91f044pfb8a15b586193146@mail.gmail.com>
Message-ID: <1B41FAF8-4925-4592-8C4F-B79D9A3F7FE5@cisco.com>

Base ES for SR5 is ES106.  This means anything lower than 106 can  
have SR5 installed over it.

-Ryan

On May 23, 2007, at 12:27 PM, Justin Steinberg wrote:

I saw ccm 4.1.3sr5, 4.2.3sr2a, and 5.1.2 posted on CCO today.

-Justin

On 5/23/07, Voll, Scott <Scott.Voll at wesd.org> wrote:
> Hey--
>
> Ryan, Wes, or Mike....... do you have a time line on CM 4.1.3sr5?  and
> can I upgrade from ES100 to SR5?
>
> Thanks
>
> Scott
>
> -----Original Message-----
> From: cust-security-announce-bounces at cisco.com
> [mailto:cust-security-announce-bounces at cisco.com] On Behalf Of Cisco
> Systems Product Security Incident Response Team
> Sent: Wednesday, May 23, 2007 8:47 AM
> To: cust-security-announce at cisco.com
> Subject: Cisco Security Response: Cisco CallManager Input
> ValidationVulnerability
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Cisco Security Response: Cisco CallManager Input Validation
> Vulnerability
>
> http://www.cisco.com/warp/public/707/cisco-sr-20070523-ccm.shtml
>
> Revision 1.0
>
> For Public Release 2007 May 23 1600 UTC (GMT)
>
> -
> ---------------------------------------------------------------------- 
> -
>
> Cisco Response
> ==============
>
> This is Cisco PSIRT's response to the statements made by Marc Ruef and
> Stefan Friedi from scip AG in their message "Cisco CallManager 4.1  
> Input
> Validation Vulnerability," posted on 2007 May 23 at 1600 UTC (GMT).
>
> The original emails were posted to BugTraq and Full-Disclosure.
>
> In their postings, Marc Ruef and Stefan Friedi illustrate how to  
> bypass
> the web application firewall used in Cisco CallManager. This means of
> bypass can be used to display graphics, scripts, or other information
> downloaded from an external web site. This technique may also be  
> used to
> conduct cross-site scripting attacks. Cisco confirms that the example
> the authors Ruef and Friedi provided bypasses the web application
> firewall and that there may be other methods for bypassing the web
> application firewall.
>
> Cisco has made improvements to the input validation mechanisms in
> CallManager that may mitigate the risks associated with this security
> vulnerability. These improvements have been incorporated into 4.2(3) 
> sr2.
> Future releases, 3.3(5)sr3, 4.1(3)sr5 and 4.3(1)sr1, will also include
> the improvements made to address this bug. This issue is being tracked
> by the following Cisco Bug ID:
>
>   * CSCsi12374 - Improvements in User Input Validation
>
> Service releases of CallManager software are available at the  
> following
> link:
>
> http://www.cisco.com/public/sw-center/sw-voice.shtml
>
> Additional Information
> ======================
>
> Cisco CallManager is the software-based call-processing component  
> of the
> Cisco IP telephony solution that extends enterprise telephony features
> and functions to packet telephony network devices, such as IP phones,
> media processing devices, voice-over-IP (VoIP) gateways, and  
> multimedia
> applications. The vulnerability described in this response exists in
> the web application firewall used in CallManager. This feature is
> designed to prevent users from entering malicious code into the input
> fields used in CallManager forms. The vulnerability exists because the
> web application firewall fails to properly sanitize some potentially
> malicious tags.
>
> To exploit these issues an attacker must convince an authenticated
> user to follow a specially crafted, malicious URL. A successful attack
> may result in the execution of arbitrary script code in the user's web
> browser.
>
> For additional information on cross-site scripting (XSS) attacks  
> and the
> methods used to exploit such vulnerabilities, please refer to the  
> Cisco
> Applied Intelligence Response "Understanding Cross-Site Scripting  
> (XSS)
> Threat Vectors," which is available at the following link:
>
> http://www.cisco.com/en/US/products/ps6120/ 
> tsd_products_security_respons
> e09186a008073f7b3.html
>
> The Cisco PSIRT is not aware of any malicious use of the vulnerability
> described in this document.
>
> This issue was reported to Cisco by Marc Ruef and Stefan Friedi from
> scip AG. We would like to thank Marc Ruef and Stefan Friedi for  
> bringing
> this issue to our attention and for working with us toward coordinated
> disclosure of the issue. We greatly appreciate the opportunity to
> work with researchers on security vulnerabilities, and welcome the
> opportunity to review and assist in product reports.
>
> THIS DOCUMENT IS PROVIDED ON AN "AS IS" BASIS AND DOES NOT IMPLY
> ANY KIND OF GUARANTEE OR WARRANTY, INCLUDING THE WARRANTIES OF
> MERCHANTABILITY OR FITNESS FOR A PARTICULAR USE. YOUR USE OF THE
> INFORMATION ON THE DOCUMENT OR MATERIALS LINKED FROM THE DOCUMENT IS
> AT YOUR OWN RISK. CISCO RESERVES THE RIGHT TO CHANGE OR UPDATE THIS
> DOCUMENT AT ANY TIME.
>
> Revision History
> ================
>
> +-------------------------------------------------------------+
> | Revision 1.0   | 2007-May-23   | Initial public release.    |
> +-------------------------------------------------------------+
>
> Cisco Security Procedures
> =========================
>
> Complete information on reporting security vulnerabilities in Cisco
> products, obtaining assistance with security incidents, and  
> registering
> to receive security information from Cisco, is available on Cisco's
> worldwide website at
> http://www.cisco.com/en/US/products/ 
> products_security_vulnerability_poli
> cy.html.
> This includes instructions for press inquiries regarding Cisco  
> security
> notices. All Cisco security advisories are available at
> http://www.cisco.com/go/psirt.
>
> -
> ---------------------------------------------------------------------- 
> -
> All contents are Copyright (C) 2006-2007 Cisco Systems, Inc. All  
> rights
> reserved.
> -
> ---------------------------------------------------------------------- 
> -
>
> Updated: May 23, 2007                                Document ID:  
> 82462
>
> -
> ---------------------------------------------------------------------- 
> -
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.6 (GNU/Linux)
>
> iD8DBQFGVGFn8NUAbBmDaxQRAtqYAJ0QYVxfjNxM+fNHZ//KYPOdfmsxaQCfZZRC
> zWQKjcIn7Qw9+OpSHfB7MMc=
> =mA2X
> -----END PGP SIGNATURE-----
> _______________________________________________
> cust-security-announce mailing list
> cust-security-announce at cisco.com
> To unsubscribe, send the command "unsubscribe" in the subject of your
> message to cust-security-announce-leave at cisco.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From email at constel.com.br  Wed May 23 13:12:28 2007
From: email at constel.com.br (Constel)
Date: Wed, 23 May 2007 14:12:28 -0300
Subject: [cisco-voip] IP Communicator over VPN
Message-ID: <01b001c79d5d$8ae6cf50$0104640a@microzilio>

Hi

I'm getting some troubles with IP Communicator over VPN. The users complain
they can connect on the VPN Server and register their IP Communicator, but
are not able to make calls. When the New Call button is pressed nothing
happens. It seems it could not pass skinny signaling over VPN.

That problem has began after I changed the VPN server. It was a PIX, now
it's an ASA5500.
I could not see any difference between the PIX and ASA configuration.
CallManager version 4.1(3)sr3c
IP Communicator 2.0.2 and 2.1.1
VPN Client 4.6 and 5.0

Has someone seen that before?
thanks in advance
--
Marcelo Zilio


From matthias.schaerer at anyweb.ch  Wed May 23 13:20:59 2007
From: matthias.schaerer at anyweb.ch (Matthias Schaerer)
Date: Wed, 23 May 2007 19:20:59 +0200
Subject: [cisco-voip] unregistered CIPC on CME 4.1
Message-ID: <OF5B388B35.FF6AA4BD-ONC12572E4.005EE492-C12572E4.005F4E07@anyweb.ch>

Hi,

I have an unregistered IP communicator (2.1) on CME 4.1. When calling from 
an ISDN line a busy/disconnect message is sent back. I can handle this 
situation with call-forward busy, no problem. But now I want to forward 
the calls differently during night service time, but the call-forward busy 
treatment always takes precedence over the night service.

Does anyone have an idea how to solve this?

Thanks for any hint.
Mat
-----------------------------------------------------------------------------------
Matthias Schaerer
Network Engineer, Instructor
Mob: +41 79 419 4557

http://www.anyweb.ch - All you want to know about taking care of networks 
is here.
http://www.anyweb-osys.ch - All you want to know about networking training 
is here.
(Don't be afraid to ask.)

In case of trouble with resending emails to me 
mailto:matthias.schaerer at anyweb.ch
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From newlook110 at yahoo.com  Wed May 23 12:35:07 2007
From: newlook110 at yahoo.com (new look)
Date: Wed, 23 May 2007 09:35:07 -0700 (PDT)
Subject: [cisco-voip] Nokia E61i With CCM4.2(3)
Message-ID: <884740.90387.qm@web61321.mail.yahoo.com>

Dear All,
I have CCM 4.2(3) and Nokia E61i have Call connect installed on it 
what the setps I need to register E61i at the CCM
Thanks


 
____________________________________________________________________________________
Food fight? Enjoy some healthy debate 
in the Yahoo! Answers Food & Drink Q&A.
http://answers.yahoo.com/dir/?link=list&sid=396545367
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From gmcallister at heart.net  Wed May 23 16:10:24 2007
From: gmcallister at heart.net (Greg McAllister)
Date: Wed, 23 May 2007 15:10:24 -0500
Subject: [cisco-voip] CUCM 5.1 fixed-source MOH
Message-ID: <1D890ECE05ECD64AAFA8F2EDA180D30201216B9B@exchange1.Heart.net>

Has anyone successfully implemented the USB adapter for fixed-source MOH
on Callmanager 5.x? We are struggling getting ours working, and there's
not much experience within Cisco on it. We have the required adapter,
the configs are apparently correct, but the device will not start
according to the traces. 

 

Thanks,

Greg McAllister

 

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From Scott.Voll at wesd.org  Wed May 23 16:44:06 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 23 May 2007 13:44:06 -0700
Subject: [cisco-voip] CM 6.0
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7EE@Apollo.wesd.org>

I know that some people where asking about upgrade paths for 5.0, 5.1,
etc.

 

I just seen the upgrade path and if you want to go to any 5.x or 6.0
don't upgrade to 4.3.

 

Stay with 4.1 if you want to go to either 5.x or 6.0

 

If you're not concerned with 5.x you can upgrade to 4.2 and still get to
6.0

 

Scott

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From keli at carocomp.ro  Wed May 23 16:59:09 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Wed, 23 May 2007 23:59:09 +0300
Subject: [cisco-voip] CM 6.0
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE7EE@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A608DEE7EE@Apollo.wesd.org>
Message-ID: <20070523235909.lud9ynhu884kwcwk@mail.carocomp.ro>

Also note, that Cisco Unified Communications Manager (6.0) Business  
Edition, is an upgrade dead-end. According to Cisco there will be no  
upgrade path from that server neither to other versions(?), nor to  
larger installations.

I find quite amusing the new upgrade-policy @ Cisco. Last year, they  
told us at the local Cisco Expo (here in Romania), that 6.0 will be  
platform independent. Now 6.0 is out, it's Linux only (not that I have  
any problem with that :) ) and they devised a nifty scheme, how to get  
more money out of the customers.

Basically, all major upgrades are to be payed for (so far so good, if  
it would be justified) But I think whatever would have been like 5.2  
(or 5.1(2)), what do you know, it's called 6.0 now, and they will have  
for sure one major version upgrade per year.

And now they say 7.0 is going to be platform independent. And they  
would probably deny they ever told 6.0 is going to be ;-)

Zoltan

Quoting "Voll, Scott" <Scott.Voll at wesd.org>:

> I know that some people where asking about upgrade paths for 5.0, 5.1,
> etc.
>
>
>
> I just seen the upgrade path and if you want to go to any 5.x or 6.0
> don't upgrade to 4.3.
>
>
>
> Stay with 4.1 if you want to go to either 5.x or 6.0
>
>
>
> If you're not concerned with 5.x you can upgrade to 4.2 and still get to
> 6.0
>
>
>
> Scott
>
>



----------------------------------------------------------------
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From rratliff at cisco.com  Wed May 23 17:16:34 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 23 May 2007 17:16:34 -0400
Subject: [cisco-voip] CM 6.0
In-Reply-To: <20070523235909.lud9ynhu884kwcwk@mail.carocomp.ro>
References: <407055A92CECCB499C922A2D35FC19A608DEE7EE@Apollo.wesd.org>
	<20070523235909.lud9ynhu884kwcwk@mail.carocomp.ro>
Message-ID: <FC55C47C-90F4-45F4-8FFB-CE6E02721CA1@cisco.com>

As soon as 5.0 came out the plan was announced to be one minor  
version in the fall, major version in the spring.  5.1 was released  
last fall (and 5.1.2 is out now) and 6.0 will be coming this spring/ 
summer.   Look for 6.1 this fall, and 7.0 should follow next spring.

I have no visibility into the marketing/upgrade path licensing stuff  
so can't comment on that part.   There was a change from the  
originally announced plan for the platform-independent version as  
well as which version would be have feature-parity with 4.2 and this  
has caused much confusion.

-Ryan

On May 23, 2007, at 4:59 PM, keli at carocomp.ro wrote:

Also note, that Cisco Unified Communications Manager (6.0) Business
Edition, is an upgrade dead-end. According to Cisco there will be no
upgrade path from that server neither to other versions(?), nor to
larger installations.

I find quite amusing the new upgrade-policy @ Cisco. Last year, they
told us at the local Cisco Expo (here in Romania), that 6.0 will be
platform independent. Now 6.0 is out, it's Linux only (not that I have
any problem with that :) ) and they devised a nifty scheme, how to get
more money out of the customers.

Basically, all major upgrades are to be payed for (so far so good, if
it would be justified) But I think whatever would have been like 5.2
(or 5.1(2)), what do you know, it's called 6.0 now, and they will have
for sure one major version upgrade per year.

And now they say 7.0 is going to be platform independent. And they
would probably deny they ever told 6.0 is going to be ;-)

Zoltan

Quoting "Voll, Scott" <Scott.Voll at wesd.org>:

> I know that some people where asking about upgrade paths for 5.0, 5.1,
> etc.
>
>
>
> I just seen the upgrade path and if you want to go to any 5.x or 6.0
> don't upgrade to 4.3.
>
>
>
> Stay with 4.1 if you want to go to either 5.x or 6.0
>
>
>
> If you're not concerned with 5.x you can upgrade to 4.2 and still  
> get to
> 6.0
>
>
>
> Scott
>
>



----------------------------------------------------------------
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_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From bob at smalltime.com  Wed May 23 18:08:01 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 23 May 2007 17:08:01 -0500
Subject: [cisco-voip] FW: Cisco Security Response: Cisco CallManager
 Input ValidationVulnerability
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE7E4@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A608DEE7E4@Apollo.wesd.org>
Message-ID: <4654BB41.7060007@smalltime.com>

> Ryan, Wes, or Mike....... do you have a time line on CM 4.1.3sr5?  and
> can I upgrade from ES100 to SR5?

I just got an email from the automated notification system on CCO that 
shows that SR5 is already out.

From ahmetcan.keten at gmail.com  Wed May 23 18:08:18 2007
From: ahmetcan.keten at gmail.com (Ahmet Can KETEN)
Date: Thu, 24 May 2007 01:08:18 +0300
Subject: [cisco-voip] Problem with QSIG connection between Alcatel 4400 and
	2811 VGW
Message-ID: <85e03a210705231508u16762c0fg4f99d0b100e08e2d@mail.gmail.com>

Hello My Friends,

I have a problem about the PRI E1 QSIG connection between my 2811 VGW and
Alcatel 4400 PBX. When the analog phone behind the PBX tries to call to the
ip phone that is located at the remote side where the CCM is there, the
analog phone still gets the continuous dial tone although caller dials the
extension number. During the dialing time there is no dial-tone, there is
silence in a normal way but after dialing the extension, caller starts to
get dial-tone again. I realize that for each extension that caller dials
without on-hook the phone, one b channel is assigned to that call but still
there is dial-tone, there is no ring back tone. After the caller dials the
extension, he or she still gets dial-tone and without on-hook the phone, the
caller dials another extension over the same dial-tone, another bchan is
assigned to that call but still there is no ring back tone and still there
is dial tone. After the specific amount of time, the caller gets the
fast-busy tone in a normal way. When I used sh isdn active command during
the dialing time, I see the call. What I should do? Could you help me?

Thanks,
Canket
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From bob at smalltime.com  Wed May 23 19:07:45 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 23 May 2007 18:07:45 -0500
Subject: [cisco-voip] Interop Las Vegas
Message-ID: <4654C941.3030303@smalltime.com>

Anyone else attending?


From Matt.Slaga at us.didata.com  Wed May 23 21:40:19 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 23 May 2007 21:40:19 -0400
Subject: [cisco-voip] CM 6.0
In-Reply-To: <20070523235909.lud9ynhu884kwcwk@mail.carocomp.ro>
References: <407055A92CECCB499C922A2D35FC19A608DEE7EE@Apollo.wesd.org>
	<20070523235909.lud9ynhu884kwcwk@mail.carocomp.ro>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005AA268D@USNAEXCH.na.didata.local>

If you remember just a year and a half ago, they claimed 5.1 would be
platform independent as well.  I've heard 6.0, 6.1 and now 7.0 would
have this.  I think they will push it out possibly even farther because
I really don't think Cisco wants to utilize a Microsoft OS if they don't
have too.  If/when they come out with one, be ready to pay extra for the
Microsoft version since someone is going to have to pay Mr. Gates for
the OS and SQL and you can bet it won't be Microsoft or Cisco.

My 2 cents.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
keli at carocomp.ro
Sent: Wednesday, May 23, 2007 4:59 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 6.0

Also note, that Cisco Unified Communications Manager (6.0) Business  
Edition, is an upgrade dead-end. According to Cisco there will be no  
upgrade path from that server neither to other versions(?), nor to  
larger installations.

I find quite amusing the new upgrade-policy @ Cisco. Last year, they  
told us at the local Cisco Expo (here in Romania), that 6.0 will be  
platform independent. Now 6.0 is out, it's Linux only (not that I have  
any problem with that :) ) and they devised a nifty scheme, how to get  
more money out of the customers.

Basically, all major upgrades are to be payed for (so far so good, if  
it would be justified) But I think whatever would have been like 5.2  
(or 5.1(2)), what do you know, it's called 6.0 now, and they will have  
for sure one major version upgrade per year.

And now they say 7.0 is going to be platform independent. And they  
would probably deny they ever told 6.0 is going to be ;-)

Zoltan

Quoting "Voll, Scott" <Scott.Voll at wesd.org>:

> I know that some people where asking about upgrade paths for 5.0, 5.1,
> etc.
>
>
>
> I just seen the upgrade path and if you want to go to any 5.x or 6.0
> don't upgrade to 4.3.
>
>
>
> Stay with 4.1 if you want to go to either 5.x or 6.0
>
>
>
> If you're not concerned with 5.x you can upgrade to 4.2 and still get
to
> 6.0
>
>
>
> Scott
>
>



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_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
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designated addressee(s) named above only.  If you are not the
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From jason.aarons at us.didata.com  Thu May 24 01:12:25 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 24 May 2007 01:12:25 -0400
Subject: [cisco-voip] CM 6.0
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005AA268D@USNAEXCH.na.didata.local>
References: <407055A92CECCB499C922A2D35FC19A608DEE7EE@Apollo.wesd.org><20070523235909.lud9ynhu884kwcwk@mail.carocomp.ro>
	<C1FE15183DA37645BC0633BC604E44F005AA268D@USNAEXCH.na.didata.local>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005AA2705@USNAEXCH.na.didata.local>

I wonder if we will be on RHE 4 through 2012 unless someone is going to
put money into Matthew Szuliz's pocket for RHE
development/upgrades.....doesn't everyone need their bread buttered?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Wednesday, May 23, 2007 9:40 PM
To: keli at carocomp.ro; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 6.0

If you remember just a year and a half ago, they claimed 5.1 would be
platform independent as well.  I've heard 6.0, 6.1 and now 7.0 would
have this.  I think they will push it out possibly even farther because
I really don't think Cisco wants to utilize a Microsoft OS if they don't
have too.  If/when they come out with one, be ready to pay extra for the
Microsoft version since someone is going to have to pay Mr. Gates for
the OS and SQL and you can bet it won't be Microsoft or Cisco.

My 2 cents.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
keli at carocomp.ro
Sent: Wednesday, May 23, 2007 4:59 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 6.0

Also note, that Cisco Unified Communications Manager (6.0) Business  
Edition, is an upgrade dead-end. According to Cisco there will be no  
upgrade path from that server neither to other versions(?), nor to  
larger installations.

I find quite amusing the new upgrade-policy @ Cisco. Last year, they  
told us at the local Cisco Expo (here in Romania), that 6.0 will be  
platform independent. Now 6.0 is out, it's Linux only (not that I have  
any problem with that :) ) and they devised a nifty scheme, how to get  
more money out of the customers.

Basically, all major upgrades are to be payed for (so far so good, if  
it would be justified) But I think whatever would have been like 5.2  
(or 5.1(2)), what do you know, it's called 6.0 now, and they will have  
for sure one major version upgrade per year.

And now they say 7.0 is going to be platform independent. And they  
would probably deny they ever told 6.0 is going to be ;-)

Zoltan

Quoting "Voll, Scott" <Scott.Voll at wesd.org>:

> I know that some people where asking about upgrade paths for 5.0, 5.1,
> etc.
>
>
>
> I just seen the upgrade path and if you want to go to any 5.x or 6.0
> don't upgrade to 4.3.
>
>
>
> Stay with 4.1 if you want to go to either 5.x or 6.0
>
>
>
> If you're not concerned with 5.x you can upgrade to 4.2 and still get
to
> 6.0
>
>
>
> Scott
>
>



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https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
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unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From ahmetcan.keten at gmail.com  Thu May 24 03:39:22 2007
From: ahmetcan.keten at gmail.com (Ahmet Can KETEN)
Date: Thu, 24 May 2007 10:39:22 +0300
Subject: [cisco-voip] Problem with QSIG connection between Alcatel 4400
	and 2811 VGW
In-Reply-To: <85e03a210705231508u16762c0fg4f99d0b100e08e2d@mail.gmail.com>
References: <85e03a210705231508u16762c0fg4f99d0b100e08e2d@mail.gmail.com>
Message-ID: <85e03a210705240039h694f5b6av6c0bf24e76d773ec@mail.gmail.com>

By the way my debug voip ccapi inout outptut is:


HT_izmit#
*May 24 07:05:00.029: //2004/D3EED2398086/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x46F79238, Tag=0x0, Call Id=2004,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
*May 24 07:05:00.029: //2004/D3EED2398086/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*May 24 07:05:00.029: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*May 24 07:05:00.029: :cc_free_feature_vsa freeing 465247B8
*May 24 07:05:00.029: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*May 24 07:05:00.029:  vsacount in free is 1
*May 24 07:05:00.069: //2005/DA8365E68087/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x46F79238, Tag=0x0, Call Id=2005,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
*May 24 07:05:00.069: //2005/DA8365E68087/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*May 24 07:05:00.069: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*May 24 07:05:00.069: :cc_free_feature_vsa freeing 46524458
*May 24 07:05:00.069: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*May 24 07:05:00.069:  vsacount in free is 0
*May 24 07:05:05.933: //-1/EF5326508088/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=5122
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=604242
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=-1
   cisco-rdnplan=-1
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0

*May 24 07:05:05.937: //-1/EF5326508088/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x46F79238, Call Info(
   Calling Number=5122,(Calling Name=)(TON=Unknown, NPI=Unknown,
Screening=User, Passed, Presentation=Allowed),
   Called Number=604242(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine,
FinalDestinationFlag=FALSE,
   Incoming Dial-peer=101, Progress Indication=NULL(0), Calling IE
Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID
Transparent=FALSE), Call Id=-1
*May 24 07:05:05.937: //-1/EF5326508088/CCAPI/ccCheckClipClir:
   In: Calling Number=5122(TON=Unknown, NPI=Unknown, Screening=User, Passed,
Presentation=Allowed)
*May 24 07:05:05.937: //-1/EF5326508088/CCAPI/ccCheckClipClir:
   Out: Calling Number=5122(TON=Unknown, NPI=Unknown, Screening=User,
Passed, Presentation=Allowed)
*May 24 07:05:05.937: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*May 24 07:05:05.937: :cc_get_feature_vsa malloc success
*May 24 07:05:05.937: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*May 24 07:05:05.937:  cc_get_feature_vsa count is 1
*May 24 07:05:05.937: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

*May 24 07:05:05.937: :FEATURE_VSA attributes are:
feature_name:0,fearture_time:1179796576,feature_id:2006
*May 24 07:05:05.937:
//2006/EF5326508088/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=5122(TON=Unknown, NPI=Unknown, Screening=User,
Passed, Presentation=Allowed),
   Called Number=604242(TON=Unknown, NPI=Unknown))
*May 24 07:05:05.937: //2006/EF5326508088/CCAPI/cc_process_call_setup_ind:
   Event=0x46534528
*May 24 07:05:05.937: //2006/EF5326508088/CCAPI/ccCallSetContext:
   Context=0x45CCF6DC
*May 24 07:05:05.937: //2006/EF5326508088/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 2006 with tag 101 to app
"_ManagedAppProcess_Default"
*May 24 07:05:05.937: //2006/EF5326508088/CCAPI/ccCallSetupAck:
   Call Id=2006
*May 24 07:05:05.937: //2006/EF5326508088/CCAPI/cc_api_set_transfer_info:
   Transfer Number=, Transfer Reason=0x0
*May 24 07:05:05.937: //2006/EF5326508088/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=TRUE, Tone=Dial Tone,
   Tone Direction=Network, Params=0x0, Call Id=2006
*May 24 07:05:05.937: //2006/EF5326508088/CCAPI/ccSetDigitTimeouts:
   Initial Digit Timeout=-1000(ms), Inter Digit Timeout=-1000(ms)
*May 24 07:05:05.941: //2006/EF5326508088/CCAPI/ccSetDigitTimeouts:
   Call Entry(Inter Digit Timeout=10000(ms), Initial Digit
Timeout=10000(ms))
*May 24 07:05:05.941: //2006/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0x7D6, digit_event=0x1, enable=TRUE, consume=FALSE)
*May 24 07:05:05.941: //2006/EF5326508088/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=2006
*May 24 07:05:05.941:
//2006/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x46F79238, callID=0x7D6, disp=0, digit_event=0x1, enable=TRUE,
consume=FALSE)
*May 24 07:05:05.941:
//2006/EF5326508088/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0x0, Interface=0x46F79238, Call Id=2006
*May 24 07:05:05.941:
//2006/EF5326508088/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=10000(ms), Inter Digit
Timeout=10000(ms))
*May 24 07:05:14.805: //2006/EF5326508088/CCAPI/cc_api_call_disconnected:
   Cause Value=16, Interface=0x46F79238, Call Id=2006
*May 24 07:05:14.805: //2006/EF5326508088/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=FALSE, Cause Value=16, Retry Count=0)
*May 24 07:05:14.809: //2006/EF5326508088/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0,
Disconnect Cause=16)
*May 24 07:05:14.809: //2006/EF5326508088/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
*May 24 07:05:14.809: //2006/EF5326508088/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
*May 24 07:05:14.849: //2006/EF5326508088/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x46F79238, Tag=0x0, Call Id=2006,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
*May 24 07:05:14.849: //2006/EF5326508088/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*May 24 07:05:14.849: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*May 24 07:05:14.849: :cc_free_feature_vsa freeing 46524458
*May 24 07:05:14.849: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

*May 24 07:05:14.849:  vsacount in free is 0u all
All possible debugging has been turned off
HT_izmit#
Can you see anything wrong? Thanks from now.


2007/5/24, Ahmet Can KETEN <ahmetcan.keten at gmail.com>:
>
> Hello My Friends,
>
> I have a problem about the PRI E1 QSIG connection between my 2811 VGW and
> Alcatel 4400 PBX. When the analog phone behind the PBX tries to call to the
> ip phone that is located at the remote side where the CCM is there, the
> analog phone still gets the continuous dial tone although caller dials the
> extension number. During the dialing time there is no dial-tone, there is
> silence in a normal way but after dialing the extension, caller starts to
> get dial-tone again. I realize that for each extension that caller dials
> without on-hook the phone, one b channel is assigned to that call but still
> there is dial-tone, there is no ring back tone. After the caller dials the
> extension, he or she still gets dial-tone and without on-hook the phone, the
> caller dials another extension over the same dial-tone, another bchan is
> assigned to that call but still there is no ring back tone and still there
> is dial tone. After the specific amount of time, the caller gets the
> fast-busy tone in a normal way. When I used sh isdn active command during
> the dialing time, I see the call. What I should do? Could you help me?
>
> Thanks,
> Canket
>
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From akoul at dataways.gr  Thu May 24 04:54:51 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 24 May 2007 11:54:51 +0300
Subject: [cisco-voip] Any Dell POE Switches
In-Reply-To: <c6f32ce40705230820y58eea086y7f7abba8b50d1515@mail.gmail.com>
References: <6101bda50705230750t9fd92ado573393ef79a6eb3a@mail.gmail.com>
	<c6f32ce40705230820y58eea086y7f7abba8b50d1515@mail.gmail.com>
Message-ID: <465552DB.9040303@dataways.gr>

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From bridge_stone at gmx.net  Thu May 24 06:13:59 2007
From: bridge_stone at gmx.net (Konne)
Date: Thu, 24 May 2007 12:13:59 +0200
Subject: [cisco-voip] Problem with 7920 - ringing but no sound
Message-ID: <46556567.3010904@gmx.net>

hi @ all,

i have a little problem with my Cisco VoIP. I' ve a WLAN Infrastructure 
and i would like to implement VoIP now.
my problem is that the phones are communicating well, only that i can 
hear nothing.

have you any suggests

thx


my network:


Router2811 with DSP ----- TRUNK dot1q ------ Catalyst3750------------ ACS
                                                                         
                  |
                                                                         
                  |
                                                                         
            AP1240


                                                          ephone1 
7920                     ephone2 7920


my config Router2811:

router2811#sh run
Building configuration...

Current configuration : 2961 bytes
!
! Last configuration change at 11:52:04 MESZ Thu May 24 2007 by router2811
! NVRAM config last updated at 11:52:06 MESZ Thu May 24 2007 by router2811
!
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname router2811
!
boot-start-marker
boot-end-marker
!
logging buffered 51200 warnings
!
clock timezone MEZ 1
clock summer-time MESZ recurring last Sun Mar 2:00 last Sun Oct 3:00
no aaa new-model
ip subnet-zero
!
!
ip cef
!
!
ip domain name +++++
ip name-server 192.168.138.10
ip name-server 192.168.138.11
no ftp-server write-enable
!
voice-card 0
 no dspfarm
 dsp services dspfarm
!
username +++++++ privilege 15 secret 5 ++++++++
!
!
!
!
interface FastEthernet0/0
 no ip address
 duplex auto
 speed auto
!
interface FastEthernet0/0.1
 description Management
 encapsulation dot1Q 1 native
 ip address 192.168.250.20 255.255.255.0
!
interface FastEthernet0/0.138
 description Server
 encapsulation dot1Q 138
 ip address 192.168.138.2 255.255.255.0
 no cdp enable
!
interface FastEthernet0/0.184
 description Admin
 encapsulation dot1Q 184
 ip address 192.168.184.2 255.255.255.0
 no cdp enable
!
interface FastEthernet0/0.880
 description VoIP
 encapsulation dot1Q 880
 ip address 10.95.180.5 255.255.255.0
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
ip classless
!
!
no ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
access-list 10 permit 192.168.184.0 0.0.0.255
access-list 10 permit 192.168.185.0 0.0.0.7
access-list 10 permit 192.168.250.0 0.0.0.255
!
!
!
control-plane
!
telephony-service
 max-ephones 10
 max-dn 10
 ip source-address 10.95.180.5 port 2000
 time-format 24
 date-format dd-mm-yy
 user-locale DE
 network-locale IT
 create cnf-files version-stamp 7960 May 23 2007 12:24:26
 max-conferences 8
 transfer-system full-blind
 transfer-pattern ...
!
!
ephone-dn  1
 number 750
!
!
ephone-dn  2
 number 751
!
!
ephone  1
 mac-address 0019.AAmac
 button  1:1
!
!
!
ephone  2
 mac-address 0019.AAmac2
 button  1:2
!
!
line con 0
 login local
line aux 0
line vty 0 4
 access-class 10 in
 exec-timeout 30 0
 privilege level 15
 login local
 transport input telnet ssh
line vty 5 15
 access-class 10 in
 exec-timeout 30 0
 privilege level 15
 login local
 transport input telnet ssh
!
scheduler allocate 20000 1000
ntp server 62.101.81.203 source FastEthernet0/0.138
ntp server 62.152.126.5 source FastEthernet0/0.138
!
end


From matt at saskin.net  Thu May 24 06:38:12 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 24 May 2007 06:38:12 -0400
Subject: [cisco-voip] Problem with 7920 - ringing but no sound
In-Reply-To: <46556567.3010904@gmx.net>
References: <46556567.3010904@gmx.net>
Message-ID: <46556B14.1080505@saskin.net>

Are you getting no audio at all or only one way audio?

-matt

Konne wrote:
> hi @ all,
>
> i have a little problem with my Cisco VoIP. I' ve a WLAN Infrastructure 
> and i would like to implement VoIP now.
> my problem is that the phones are communicating well, only that i can 
> hear nothing.
>
> have you any suggests
>
> thx
>
>
> my network:
>
>
> Router2811 with DSP ----- TRUNK dot1q ------ Catalyst3750------------ ACS
>                                                                          
>                   |
>                                                                          
>                   |
>                                                                          
>             AP1240
>
>
>                                                           ephone1 
> 7920                     ephone2 7920
>
>
> my config Router2811:
>
> router2811#sh run
> Building configuration...
>
> Current configuration : 2961 bytes
> !
> ! Last configuration change at 11:52:04 MESZ Thu May 24 2007 by router2811
> ! NVRAM config last updated at 11:52:06 MESZ Thu May 24 2007 by router2811
> !
> version 12.3
> service timestamps debug datetime msec
> service timestamps log datetime msec
> service password-encryption
> !
> hostname router2811
> !
> boot-start-marker
> boot-end-marker
> !
> logging buffered 51200 warnings
> !
> clock timezone MEZ 1
> clock summer-time MESZ recurring last Sun Mar 2:00 last Sun Oct 3:00
> no aaa new-model
> ip subnet-zero
> !
> !
> ip cef
> !
> !
> ip domain name +++++
> ip name-server 192.168.138.10
> ip name-server 192.168.138.11
> no ftp-server write-enable
> !
> voice-card 0
>  no dspfarm
>  dsp services dspfarm
> !
> username +++++++ privilege 15 secret 5 ++++++++
> !
> !
> !
> !
> interface FastEthernet0/0
>  no ip address
>  duplex auto
>  speed auto
> !
> interface FastEthernet0/0.1
>  description Management
>  encapsulation dot1Q 1 native
>  ip address 192.168.250.20 255.255.255.0
> !
> interface FastEthernet0/0.138
>  description Server
>  encapsulation dot1Q 138
>  ip address 192.168.138.2 255.255.255.0
>  no cdp enable
> !
> interface FastEthernet0/0.184
>  description Admin
>  encapsulation dot1Q 184
>  ip address 192.168.184.2 255.255.255.0
>  no cdp enable
> !
> interface FastEthernet0/0.880
>  description VoIP
>  encapsulation dot1Q 880
>  ip address 10.95.180.5 255.255.255.0
> !
> interface FastEthernet0/1
>  no ip address
>  shutdown
>  duplex auto
>  speed auto
> !
> ip classless
> !
> !
> no ip http server
> ip http authentication local
> ip http secure-server
> ip http timeout-policy idle 60 life 86400 requests 10000
> !
> access-list 10 permit 192.168.184.0 0.0.0.255
> access-list 10 permit 192.168.185.0 0.0.0.7
> access-list 10 permit 192.168.250.0 0.0.0.255
> !
> !
> !
> control-plane
> !
> telephony-service
>  max-ephones 10
>  max-dn 10
>  ip source-address 10.95.180.5 port 2000
>  time-format 24
>  date-format dd-mm-yy
>  user-locale DE
>  network-locale IT
>  create cnf-files version-stamp 7960 May 23 2007 12:24:26
>  max-conferences 8
>  transfer-system full-blind
>  transfer-pattern ...
> !
> !
> ephone-dn  1
>  number 750
> !
> !
> ephone-dn  2
>  number 751
> !
> !
> ephone  1
>  mac-address 0019.AAmac
>  button  1:1
> !
> !
> !
> ephone  2
>  mac-address 0019.AAmac2
>  button  1:2
> !
> !
> line con 0
>  login local
> line aux 0
> line vty 0 4
>  access-class 10 in
>  exec-timeout 30 0
>  privilege level 15
>  login local
>  transport input telnet ssh
> line vty 5 15
>  access-class 10 in
>  exec-timeout 30 0
>  privilege level 15
>  login local
>  transport input telnet ssh
> !
> scheduler allocate 20000 1000
> ntp server 62.101.81.203 source FastEthernet0/0.138
> ntp server 62.152.126.5 source FastEthernet0/0.138
> !
> end
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From bridge_stone at gmx.net  Thu May 24 07:24:35 2007
From: bridge_stone at gmx.net (Konne)
Date: Thu, 24 May 2007 13:24:35 +0200
Subject: [cisco-voip] Problem with 7920 - ringing but no sound
In-Reply-To: <46556B14.1080505@saskin.net>
References: <46556567.3010904@gmx.net> <46556B14.1080505@saskin.net>
Message-ID: <465575F3.1070503@gmx.net>

hi,

no audio at all, only a noise...

bye

Matthew Saskin wrote:

> Are you getting no audio at all or only one way audio?
>
> -matt
>
> Konne wrote:
>
>> hi @ all,
>>
>> i have a little problem with my Cisco VoIP. I' ve a WLAN 
>> Infrastructure and i would like to implement VoIP now.
>> my problem is that the phones are communicating well, only that i can 
>> hear nothing.
>>
>> have you any suggests
>>
>> thx
>>
>>
>> my network:
>>
>>
>> Router2811 with DSP ----- TRUNK dot1q ------ Catalyst3750------------ 
>> ACS
>>                                                                          
>>                   |
>>                                                                          
>>                   |
>>                                                                          
>>             AP1240
>>
>>
>>                                                           ephone1 
>> 7920                     ephone2 7920
>>
>>
>> my config Router2811:
>>
>> router2811#sh run
>> Building configuration...
>>
>> Current configuration : 2961 bytes
>> !
>> ! Last configuration change at 11:52:04 MESZ Thu May 24 2007 by 
>> router2811
>> ! NVRAM config last updated at 11:52:06 MESZ Thu May 24 2007 by 
>> router2811
>> !
>> version 12.3
>> service timestamps debug datetime msec
>> service timestamps log datetime msec
>> service password-encryption
>> !
>> hostname router2811
>> !
>> boot-start-marker
>> boot-end-marker
>> !
>> logging buffered 51200 warnings
>> !
>> clock timezone MEZ 1
>> clock summer-time MESZ recurring last Sun Mar 2:00 last Sun Oct 3:00
>> no aaa new-model
>> ip subnet-zero
>> !
>> !
>> ip cef
>> !
>> !
>> ip domain name +++++
>> ip name-server 192.168.138.10
>> ip name-server 192.168.138.11
>> no ftp-server write-enable
>> !
>> voice-card 0
>>  no dspfarm
>>  dsp services dspfarm
>> !
>> username +++++++ privilege 15 secret 5 ++++++++
>> !
>> !
>> !
>> !
>> interface FastEthernet0/0
>>  no ip address
>>  duplex auto
>>  speed auto
>> !
>> interface FastEthernet0/0.1
>>  description Management
>>  encapsulation dot1Q 1 native
>>  ip address 192.168.250.20 255.255.255.0
>> !
>> interface FastEthernet0/0.138
>>  description Server
>>  encapsulation dot1Q 138
>>  ip address 192.168.138.2 255.255.255.0
>>  no cdp enable
>> !
>> interface FastEthernet0/0.184
>>  description Admin
>>  encapsulation dot1Q 184
>>  ip address 192.168.184.2 255.255.255.0
>>  no cdp enable
>> !
>> interface FastEthernet0/0.880
>>  description VoIP
>>  encapsulation dot1Q 880
>>  ip address 10.95.180.5 255.255.255.0
>> !
>> interface FastEthernet0/1
>>  no ip address
>>  shutdown
>>  duplex auto
>>  speed auto
>> !
>> ip classless
>> !
>> !
>> no ip http server
>> ip http authentication local
>> ip http secure-server
>> ip http timeout-policy idle 60 life 86400 requests 10000
>> !
>> access-list 10 permit 192.168.184.0 0.0.0.255
>> access-list 10 permit 192.168.185.0 0.0.0.7
>> access-list 10 permit 192.168.250.0 0.0.0.255
>> !
>> !
>> !
>> control-plane
>> !
>> telephony-service
>>  max-ephones 10
>>  max-dn 10
>>  ip source-address 10.95.180.5 port 2000
>>  time-format 24
>>  date-format dd-mm-yy
>>  user-locale DE
>>  network-locale IT
>>  create cnf-files version-stamp 7960 May 23 2007 12:24:26
>>  max-conferences 8
>>  transfer-system full-blind
>>  transfer-pattern ...
>> !
>> !
>> ephone-dn  1
>>  number 750
>> !
>> !
>> ephone-dn  2
>>  number 751
>> !
>> !
>> ephone  1
>>  mac-address 0019.AAmac
>>  button  1:1
>> !
>> !
>> !
>> ephone  2
>>  mac-address 0019.AAmac2
>>  button  1:2
>> !
>> !
>> line con 0
>>  login local
>> line aux 0
>> line vty 0 4
>>  access-class 10 in
>>  exec-timeout 30 0
>>  privilege level 15
>>  login local
>>  transport input telnet ssh
>> line vty 5 15
>>  access-class 10 in
>>  exec-timeout 30 0
>>  privilege level 15
>>  login local
>>  transport input telnet ssh
>> !
>> scheduler allocate 20000 1000
>> ntp server 62.101.81.203 source FastEthernet0/0.138
>> ntp server 62.152.126.5 source FastEthernet0/0.138
>> !
>> end
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>   
>
>


From MillerS at DicksteinShapiro.COM  Thu May 24 07:44:29 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Thu, 24 May 2007 07:44:29 -0400
Subject: [cisco-voip] Turning "ON" Standard Greeting
References: <C1FE15183DA37645BC0633BC604E44F005AA268D@USNAEXCH.na.didata.local>
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538B6B@DCEX2.DSMO.COM>

 I know this seems absurd, but I am having trouble turning on my
Standard Greeting in Unity.  Currently, my "Closed" greeting plays and
there is no option that I can see to turn on the Standard greeting even
immediately after I re-record it.  Please advise.  Thank you!

Steve

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Wednesday, May 23, 2007 9:40 PM
To: keli at carocomp.ro; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 6.0

If you remember just a year and a half ago, they claimed 5.1 would be
platform independent as well.  I've heard 6.0, 6.1 and now 7.0 would
have this.  I think they will push it out possibly even farther because
I really don't think Cisco wants to utilize a Microsoft OS if they don't
have too.  If/when they come out with one, be ready to pay extra for the
Microsoft version since someone is going to have to pay Mr. Gates for
the OS and SQL and you can bet it won't be Microsoft or Cisco.

My 2 cents.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
keli at carocomp.ro
Sent: Wednesday, May 23, 2007 4:59 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 6.0

Also note, that Cisco Unified Communications Manager (6.0) Business  
Edition, is an upgrade dead-end. According to Cisco there will be no  
upgrade path from that server neither to other versions(?), nor to  
larger installations.

I find quite amusing the new upgrade-policy @ Cisco. Last year, they  
told us at the local Cisco Expo (here in Romania), that 6.0 will be  
platform independent. Now 6.0 is out, it's Linux only (not that I have  
any problem with that :) ) and they devised a nifty scheme, how to get  
more money out of the customers.

Basically, all major upgrades are to be payed for (so far so good, if  
it would be justified) But I think whatever would have been like 5.2  
(or 5.1(2)), what do you know, it's called 6.0 now, and they will have  
for sure one major version upgrade per year.

And now they say 7.0 is going to be platform independent. And they  
would probably deny they ever told 6.0 is going to be ;-)

Zoltan

Quoting "Voll, Scott" <Scott.Voll at wesd.org>:

> I know that some people where asking about upgrade paths for 5.0, 5.1,
> etc.
>
>
>
> I just seen the upgrade path and if you want to go to any 5.x or 6.0
> don't upgrade to 4.3.
>
>
>
> Stay with 4.1 if you want to go to either 5.x or 6.0
>
>
>
> If you're not concerned with 5.x you can upgrade to 4.2 and still get
to
> 6.0
>
>
>
> Scott
>
>



----------------------------------------------------------------
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_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
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This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

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==============================================================================


From Brian.Henry at apptis.com  Thu May 24 07:50:45 2007
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu, 24 May 2007 07:50:45 -0400
Subject: [cisco-voip] Turning "ON" Standard Greeting
References: <C1FE15183DA37645BC0633BC604E44F005AA268D@USNAEXCH.na.didata.local>
	<B9B5B75DF9145F44A21111DD92CC490006538B6B@DCEX2.DSMO.COM>
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6DC@CH-EXCH-02.intelltech.apptis.inc>

Disable your closed greeting, but remember to check your schedule to make sure it says ALL-HOURS ALL-DAYS if not then what schedule are you on?
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Miller, Steve
Sent: Thu 5/24/2007 7:44 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Turning "ON" Standard Greeting



 I know this seems absurd, but I am having trouble turning on my
Standard Greeting in Unity.  Currently, my "Closed" greeting plays and
there is no option that I can see to turn on the Standard greeting even
immediately after I re-record it.  Please advise.  Thank you!

Steve

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Wednesday, May 23, 2007 9:40 PM
To: keli at carocomp.ro; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 6.0

If you remember just a year and a half ago, they claimed 5.1 would be
platform independent as well.  I've heard 6.0, 6.1 and now 7.0 would
have this.  I think they will push it out possibly even farther because
I really don't think Cisco wants to utilize a Microsoft OS if they don't
have too.  If/when they come out with one, be ready to pay extra for the
Microsoft version since someone is going to have to pay Mr. Gates for
the OS and SQL and you can bet it won't be Microsoft or Cisco.

My 2 cents.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
keli at carocomp.ro
Sent: Wednesday, May 23, 2007 4:59 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 6.0

Also note, that Cisco Unified Communications Manager (6.0) Business 
Edition, is an upgrade dead-end. According to Cisco there will be no 
upgrade path from that server neither to other versions(?), nor to 
larger installations.

I find quite amusing the new upgrade-policy @ Cisco. Last year, they 
told us at the local Cisco Expo (here in Romania), that 6.0 will be 
platform independent. Now 6.0 is out, it's Linux only (not that I have 
any problem with that :) ) and they devised a nifty scheme, how to get 
more money out of the customers.

Basically, all major upgrades are to be payed for (so far so good, if 
it would be justified) But I think whatever would have been like 5.2 
(or 5.1(2)), what do you know, it's called 6.0 now, and they will have 
for sure one major version upgrade per year.

And now they say 7.0 is going to be platform independent. And they 
would probably deny they ever told 6.0 is going to be ;-)

Zoltan

Quoting "Voll, Scott" <Scott.Voll at wesd.org>:

> I know that some people where asking about upgrade paths for 5.0, 5.1,
> etc.
>
>
>
> I just seen the upgrade path and if you want to go to any 5.x or 6.0
> don't upgrade to 4.3.
>
>
>
> Stay with 4.1 if you want to go to either 5.x or 6.0
>
>
>
> If you're not concerned with 5.x you can upgrade to 4.2 and still get
to
> 6.0
>
>
>
> Scott
>
>



----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.
_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip


--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message.

To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com

Dickstein Shapiro LLP
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==============================================================================

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From akoul at dataways.gr  Thu May 24 09:01:22 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 24 May 2007 16:01:22 +0300
Subject: [cisco-voip] ciscotsp for CCM 5.04: any How To?
Message-ID: <46558CA2.5080301@dataways.gr>

Hi,
1. i have tried to configure it on a ccm 5.0(4) for the first time so i 
can dial contacts from Outlook.
2. I have downloaded the plugin from the admin page of the ccm (version 
5.1.018).
3. I installed it my PC.
4. I configure it: ie user, password and ip of ccm
the user has a phone associated with a phone.

But, in my outlook i cannot see the cisco tsp or the sepxxxx in my dial 
lines.

I looked around but i could not find any guide about this.

Any ideas? Has anyone dealt with this in the past?

From akoul at dataways.gr  Thu May 24 09:26:19 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 24 May 2007 16:26:19 +0300
Subject: [cisco-voip] ciscotsp for CCM 5.04: any How To?
In-Reply-To: <C27AFBCF.21E9%mark.turpin@calence.com>
References: <C27AFBCF.21E9%mark.turpin@calence.com>
Message-ID: <4655927B.4000801@dataways.gr>

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From Matt.Slaga at us.didata.com  Thu May 24 09:39:11 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 24 May 2007 09:39:11 -0400
Subject: [cisco-voip] ciscotsp for CCM 5.04: any How To?
In-Reply-To: <4655927B.4000801@dataways.gr>
References: <C27AFBCF.21E9%mark.turpin@calence.com>
	<4655927B.4000801@dataways.gr>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005AA298C@USNAEXCH.na.didata.local>

There is generic documentation on configuring this on CCO, but it is
also written that this "is not supported by TAC".  The last version they
released instructions was 3.x.  Link below.

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_ad
ministration_guide_chapter09186a00800cdecc.html

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony
Kouloglou
Sent: Thursday, May 24, 2007 9:26 AM
To: Mark Turpin
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] ciscotsp for CCM 5.04: any How To?

 

Yes,
i am running XP pro sp2.
Don't i have to do anything with ccm?
Where can i find a guide for this... :-( 

Mark Turpin wrote: 

Have you rebooted?
 
-Mark
 
On 5/24/07 8:01 AM, "Anthony Kouloglou" <akoul at dataways.gr>
<mailto:akoul at dataways.gr>  wrote:
 
  

	Hi,
	1. i have tried to configure it on a ccm 5.0(4) for the first
time so i
	can dial contacts from Outlook.
	2. I have downloaded the plugin from the admin page of the ccm
(version
	5.1.018).
	3. I installed it my PC.
	4. I configure it: ie user, password and ip of ccm
	the user has a phone associated with a phone.
	 
	But, in my outlook i cannot see the cisco tsp or the sepxxxx in
my dial
	lines.
	 
	I looked around but i could not find any guide about this.
	 
	Any ideas? Has anyone dealt with this in the past?
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	    

 
  



-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From akoul at dataways.gr  Thu May 24 09:41:24 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 24 May 2007 16:41:24 +0300
Subject: [cisco-voip] ciscotsp for CCM 5.04: any How To?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005AA298C@USNAEXCH.na.didata.local>
References: <C27AFBCF.21E9%mark.turpin@calence.com>
	<4655927B.4000801@dataways.gr>
	<C1FE15183DA37645BC0633BC604E44F005AA298C@USNAEXCH.na.didata.local>
Message-ID: <46559604.1060703@dataways.gr>

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From lelio at uoguelph.ca  Thu May 24 10:10:31 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 24 May 2007 10:10:31 -0400
Subject: [cisco-voip] you know you are a CCMadmin weenie when...
Message-ID: <012401c79e0d$49e953a0$47196883@cfs.uoguelph.ca>

you know you are a CCMadmin weenie when...you use CCMadmin instead of your corporate directory to look up phonenumbers.....

*sigh*

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR
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From rleetun at co.boulder.co.us  Thu May 24 11:30:47 2007
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Thu, 24 May 2007 09:30:47 -0600
Subject: [cisco-voip] Open Source MCU
In-Reply-To: <012401c79e0d$49e953a0$47196883@cfs.uoguelph.ca>
Message-ID: <963CE94787005149A52C17F33A028E1207E62003@EVS1B.boco.co.boulder.co.us>

Hi,
 
What is good open source MCU?  Has anyone implemented an open source
MCU, and if so, how did it go?
 
I am plan on using it in a CCM 4.1.3 environment along with a 2821
gatekeeper.
 
Thanks.
 
Rob
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From mfa at gte.net  Thu May 24 11:39:24 2007
From: mfa at gte.net (Mike Armstrong)
Date: Thu, 24 May 2007 11:39:24 -0400
Subject: [cisco-voip]  you know you are a CCMadmin weenie when...
References: <mailman.15097.1180015834.1489.cisco-voip@puck.nether.net>
Message-ID: <016801c79e19$b4fb5650$1402a8c0@MFADesktop>

OTOH, the CCMadmin numbers are correct...

> Date: Thu, 24 May 2007 10:10:31 -0400
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: [cisco-voip] you know you are a CCMadmin weenie when...
> To: <cisco-voip at puck.nether.net>
> Message-ID: <012401c79e0d$49e953a0$47196883 at cfs.uoguelph.ca>
> Content-Type: text/plain; charset="iso-8859-1"
>
> you know you are a CCMadmin weenie when...you use CCMadmin instead of your 
> corporate directory to look up phonenumbers.....
>
> *sigh*


From sil at infiltrated.net  Thu May 24 11:49:46 2007
From: sil at infiltrated.net (J. Oquendo)
Date: Thu, 24 May 2007 11:49:46 -0400
Subject: [cisco-voip] Open Source MCU
In-Reply-To: <963CE94787005149A52C17F33A028E1207E62003@EVS1B.boco.co.boulder.co.us>
References: <963CE94787005149A52C17F33A028E1207E62003@EVS1B.boco.co.boulder.co.us>
Message-ID: <4655B41A.7070608@infiltrated.net>

Leetun, Rob wrote:
> Hi,
>  
> What is good open source MCU?  Has anyone implemented an open source 
> MCU, and if so, how did it go?
>  
> I am plan on using it in a CCM 4.1.3 environment along with a 2821 
> gatekeeper.
>  
> Thanks.
>  
> Rob
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   
Don't know about production systems though...

http://www.openh323.org/
http://www.voxgratia.org/modules/news/


-- 
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
echo infiltrated.net|sed 's/^/sil@/g' 

"Wise men talk because they have something to say;
fools, because they have to say something." -- Plato


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From rafiq at dreamthought.com  Thu May 24 12:20:31 2007
From: rafiq at dreamthought.com (Raf)
Date: Thu, 24 May 2007 17:20:31 +0100 (BST)
Subject: [cisco-voip] Obtaining AXL CCM 5.1 Schema
Message-ID: <20070524172003.J42631@joshua.dreamthought.com>

Hello,

I'm trying to get a hold of the CCM 5.1 AXL schema.  The Cisco site 
requires me to register, etc, and I just want the documentation.

Is there anyway of obtaining this without the hassle of registration?

Apologies for being slightly off-topic.
Thanks in advance.

Raf




From charlieb at cot.net  Thu May 24 12:25:05 2007
From: charlieb at cot.net (Charles J. Boening)
Date: Thu, 24 May 2007 09:25:05 -0700
Subject: [cisco-voip] ciscotsp for CCM 5.04: any How To?
In-Reply-To: <46558CA2.5080301@dataways.gr>
References: <46558CA2.5080301@dataways.gr>
Message-ID: <20DF280A12FBD1478CDCDEA366ACE077FB719B@apollo.calore.local>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Make sure you install the Wave driver.  There are instructions in the
ciscotsp.txt in c:\program files\cisco.  This fixed the problem on my
system.  About the only difference is that I'm running TSP version
5.1(1).

For Windows XP:
- From Control Panel execute the "Add Hardware" utility.  Click the 
"Next" button. Select "Yes, I have already connected the hardware"
Radio button. Click Next. Select "Add a new Hardware device" from the
list.
Click the "Next" button.  Select "Install the hardware that I manually 
select from a list" radio button.  Click the "Next" button. Select
"Sound, video and game controller" when prompted for hardware type. 
Click the "Next" button.  Click the "Have Disk" button when prompted
to "Select a Device Driver".  Click the "Browse" button on the
"Install from Disk" window.  Browse to
C:\Program Files\Cisco\Wave Drivers and select the file "OEMSETUP".
Click "Open" to install the Cisco Wave Driver and select OK.
Highlight the "Cisco TAPI Wave Driver" in Select a Device Driver window
and select 'Next'. Select 'Next' in "Start Hardware Installation"
window.
If Prompted for "Digital signature Not Found" click on "Continue Anyway"

button.
When prompted for "Install from disk 1" for file avaudio32.dll, choose
Browse button and select path "C:\Program Files\Cisco\Wave Drivers" 
and click Open to install the avaudio32.dll. 
When prompted to reboot your machine, do so.



Good luck.


Charlie






> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Anthony Kouloglou
> Sent: Thursday, May 24, 2007 6:01 AM
> To: cisco-voip at puck-nether.net
> Subject: [cisco-voip] ciscotsp for CCM 5.04: any How To?
> 
> Hi,
> 1. i have tried to configure it on a ccm 5.0(4) for the first time so
i
> can dial contacts from Outlook.
> 2. I have downloaded the plugin from the admin page of the ccm
(version
> 5.1.018).
> 3. I installed it my PC.
> 4. I configure it: ie user, password and ip of ccm
> the user has a phone associated with a phone.
> 
> But, in my outlook i cannot see the cisco tsp or the sepxxxx in my
dial
> lines.
> 
> I looked around but i could not find any guide about this.
> 
> Any ideas? Has anyone dealt with this in the past?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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5JClWRDpeQ1rCuw+2gJDc4M=
=qWI8
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From fkauffma at cisco.com  Thu May 24 12:30:04 2007
From: fkauffma at cisco.com (Frank Kauffman IV (fkauffma))
Date: Thu, 24 May 2007 12:30:04 -0400
Subject: [cisco-voip] extension mobility login question
Message-ID: <53E2851455813A40B20FB58AB3A98D0801CB1FAC@xmb-rtp-20d.amer.cisco.com>

If a user is logged into the phone, and the phone is upgraded and has
registered back with Callmanager, will the user automatically be logged
back in to the phone?  Or is it necessary for user to log back in
manually.
 
Best Regards,
 
 	
Frank Kauffman
Customer Support Engineer
Cisco Remote Operations Services

fkauffma at cisco.com
Phone :919.574.0285
Mobile :919.749.4647



7100-8 Kit Creek Road
Research Triangle Park, NC. 27709
United States
www.cisco.com <http://www.cisco.com/> 

 	
 	
This e-mail may contain confidential and privileged material for the
sole use of the intended recipient. Any review, use, distribution or
disclosure by others is strictly prohibited. If you are not the intended
recipient (or authorized to receive for the recipient), please contact
the sender by reply e-mail and delete all copies of this message.	
 


 
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From keli at carocomp.ro  Thu May 24 13:13:17 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Thu, 24 May 2007 20:13:17 +0300
Subject: [cisco-voip] Open Source MCU
In-Reply-To: <4655B41A.7070608@infiltrated.net>
References: <963CE94787005149A52C17F33A028E1207E62003@EVS1B.boco.co.boulder.co.us>
	<4655B41A.7070608@infiltrated.net>
Message-ID: <20070524201317.r97euvankskcg0oo@mail.carocomp.ro>

OpenH323-based GnuGK (or OpenGK) is quite good and stable. However, I  
don't know much about their MCU implementation. I know for sure, that  
the project was stalled for quite a while before it got a fresh boot  
about a year ago (or maybe less)

As far as I know, one of the project's sponsors was Stonevoice, a  
European Cisco partner company, who now has a software MCU in their  
product list. Maybe it's worth checking out their site  
http://www.stonevoice.com/
[They have some other nice & reasonably priced side-products as well  
for CME environments]

regards
Zoltan

Quoting "J. Oquendo" <sil at infiltrated.net>:

> Leetun, Rob wrote:
>> Hi,
>> What is good open source MCU?  Has anyone implemented an open   
>> source MCU, and if so, how did it go?
>> I am plan on using it in a CCM 4.1.3 environment along with a 2821   
>> gatekeeper.
>> Thanks.
>> Rob
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
> Don't know about production systems though...
>
> http://www.openh323.org/
> http://www.voxgratia.org/modules/news/
>
>
> -- 
> ====================================================
> J. Oquendo
> http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
> echo infiltrated.net|sed 's/^/sil@/g' "Wise men talk because they have
> something to say;
> fools, because they have to say something." -- Plato



----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.


From jonvoip at gmail.com  Thu May 24 14:13:10 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 24 May 2007 13:13:10 -0500
Subject: [cisco-voip] Unallocated/Unassigned Number for 911 call...
Message-ID: <5d093f9a0705241113t11f38317y1a48bd0c69ada7a2@mail.gmail.com>

So, made a 911 test call:

VOICEGW#
*May 24 18:13:39.451: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8
callref = 0x004D
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98394
                Exclusive, Channel 20
        Calling Party Number i = 0x0181, '[removed]'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x80, '911'
                Plan:Unknown, Type:Unknown
*May 24 18:13:39.527: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x804D
        Cause i = 0x8081 - Unallocated/unassigned number


Any ideas?



Jonathan

From Derek.Hover at apptis.com  Thu May 24 14:30:33 2007
From: Derek.Hover at apptis.com (Derek Hover)
Date: Thu, 24 May 2007 14:30:33 -0400
Subject: [cisco-voip] CM 4.1(3)SR5
In-Reply-To: <5d093f9a0705241113t11f38317y1a48bd0c69ada7a2@mail.gmail.com>
References: <5d093f9a0705241113t11f38317y1a48bd0c69ada7a2@mail.gmail.com>
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC031226C4@CH-EXCH-02.intelltech.apptis.inc>


Now that it's out, anyone taking a stick to it yet?

Derek

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Thursday, May 24, 2007 2:13 PM
To: ciscovoip
Subject: [cisco-voip] Unallocated/Unassigned Number for 911 call...

So, made a 911 test call:

VOICEGW#
*May 24 18:13:39.451: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8
callref = 0x004D
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98394
                Exclusive, Channel 20
        Calling Party Number i = 0x0181, '[removed]'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x80, '911'
                Plan:Unknown, Type:Unknown
*May 24 18:13:39.527: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x804D
        Cause i = 0x8081 - Unallocated/unassigned number


Any ideas?



Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From Brian.Henry at apptis.com  Thu May 24 14:51:11 2007
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu, 24 May 2007 14:51:11 -0400
Subject: [cisco-voip] CM 4.1(3)SR5
References: <5d093f9a0705241113t11f38317y1a48bd0c69ada7a2@mail.gmail.com>
	<2F09B2162CCDEA4EA05C089B24FBC2BC031226C4@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6DE@CH-EXCH-02.intelltech.apptis.inc>

What is configured on your route pattern for "911"?
Are you using CER? (Cisco Emergency Responder)?
What number should be passed to the PSAP when dialing 911 if you are not using CER? most likely the main number to the business.
Does your Route Pattern or Route List Restrict CLID?
If not
Then make sure you check you calling Parties Mask, but change the CLID to the 10-Digit Main Number, make sure that you allow on your RouteList(on the Route Groups for the RouteList).
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Derek Hover
Sent: Thu 5/24/2007 2:30 PM
To: ciscovoip
Subject: Re: [cisco-voip] CM 4.1(3)SR5




Now that it's out, anyone taking a stick to it yet?

Derek

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Thursday, May 24, 2007 2:13 PM
To: ciscovoip
Subject: [cisco-voip] Unallocated/Unassigned Number for 911 call...

So, made a 911 test call:

VOICEGW#
*May 24 18:13:39.451: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8
callref = 0x004D
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98394
                Exclusive, Channel 20
        Calling Party Number i = 0x0181, '[removed]'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x80, '911'
                Plan:Unknown, Type:Unknown
*May 24 18:13:39.527: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x804D
        Cause i = 0x8081 - Unallocated/unassigned number


Any ideas?



Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From Matt.Slaga at us.didata.com  Thu May 24 15:38:33 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 24 May 2007 15:38:33 -0400
Subject: [cisco-voip] CM 4.1(3)SR5
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6DE@CH-EXCH-02.intelltech.apptis.inc>
References: <5d093f9a0705241113t11f38317y1a48bd0c69ada7a2@mail.gmail.com><2F09B2162CCDEA4EA05C089B24FBC2BC031226C4@CH-EXCH-02.intelltech.apptis.inc>
	<2F09B2162CCDEA4EA05C089B24FBC2BC07AEA6DE@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005AA32B2@USNAEXCH.na.didata.local>

I've seen this almost exact problem  once before.  It was caused by the
carrier forgetting to assigning a PSAP location to the circuit.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Henry
Sent: Thursday, May 24, 2007 2:51 PM
To: Derek Hover; ciscovoip
Subject: Re: [cisco-voip] CM 4.1(3)SR5

 

What is configured on your route pattern for "911"?

Are you using CER? (Cisco Emergency Responder)?

What number should be passed to the PSAP when dialing 911 if you are not
using CER? most likely the main number to the business.

Does your Route Pattern or Route List Restrict CLID?

If not

Then make sure you check you calling Parties Mask, but change the CLID
to the 10-Digit Main Number, make sure that you allow on your
RouteList(on the Route Groups for the RouteList).

 

Brian

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Derek Hover
Sent: Thu 5/24/2007 2:30 PM
To: ciscovoip
Subject: Re: [cisco-voip] CM 4.1(3)SR5

 

Now that it's out, anyone taking a stick to it yet?

Derek

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Thursday, May 24, 2007 2:13 PM
To: ciscovoip
Subject: [cisco-voip] Unallocated/Unassigned Number for 911 call...

So, made a 911 test call:

VOICEGW#
*May 24 18:13:39.451: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8
callref = 0x004D
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98394
                Exclusive, Channel 20
        Calling Party Number i = 0x0181, '[removed]'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x80, '911'
                Plan:Unknown, Type:Unknown
*May 24 18:13:39.527: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x804D
        Cause i = 0x8081 - Unallocated/unassigned number


Any ideas?



Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




-----------------------------------------
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This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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From balbler at nts.at  Thu May 24 16:01:13 2007
From: balbler at nts.at (Bernhard Albler)
Date: Thu, 24 May 2007 22:01:13 +0200
Subject: [cisco-voip] extension mobility login question
In-Reply-To: <53E2851455813A40B20FB58AB3A98D0801CB1FAC@xmb-rtp-20d.amer.cisco.com>
References: <53E2851455813A40B20FB58AB3A98D0801CB1FAC@xmb-rtp-20d.amer.cisco.com>
Message-ID: <4759687500005273@nts.at>

Hi Frank,

from a backend standpoint, a login request actually copies the device profile of the user over the device. Since the change is done in the DB, it will survive a phone upgrade (as well as a failover from server etc.)

regards
bernhard

------------------------------------------
DI (FH) Bernhard Albler, NTS AG
Tel.: +43 (1) 867 67 76 -0, email: balbler at nts.at

?

From jrobbins at mercurypay.com  Thu May 24 17:11:33 2007
From: jrobbins at mercurypay.com (Jerri Robbins)
Date: Thu, 24 May 2007 15:11:33 -0600
Subject: [cisco-voip] Unity question
Message-ID: <9015A93106739144995D6FD24A0F7A5608B5DB1F@hydrogen.mercurypay.com>

I need to set up some DNs to transfer to a CSQ and show which number was
dialed so the Support staff who answers the calls can answer
appropriately - i.e. "Mr. Smith's office" depending on which number was
called.  Is it possible to set up a call handler in Unity to do this?

 

Jerri

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From voipjockey at gmail.com  Thu May 24 17:25:30 2007
From: voipjockey at gmail.com (Jimmy Pale)
Date: Thu, 24 May 2007 16:25:30 -0500
Subject: [cisco-voip] Unity Language Install?
Message-ID: <001201c79e4a$1061c580$31255080$@com>

I am trying install an additional language on Unity. Using the CDs
downloaded from CCO I only get the option to install English though I have
downloaded the CD for the other language I need to install as well. English
United States is the only option during the install with the downloaded CDs.
I have not received the DVD media yet. Can anyone help me with what I am
missing here?

 

Thanks!

 

 

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From jsteinberg at gmail.com  Thu May 24 18:09:04 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 24 May 2007 18:09:04 -0400
Subject: [cisco-voip] Unity question
In-Reply-To: <9015A93106739144995D6FD24A0F7A5608B5DB1F@hydrogen.mercurypay.com>
References: <9015A93106739144995D6FD24A0F7A5608B5DB1F@hydrogen.mercurypay.com>
Message-ID: <f1b65f880705241509p3369d321s7f45034f45260b8a@mail.gmail.com>

Setup a CTI Route point for each DN and set the internal caller ID for
this route point to "Mr. Smith's office". Set the call forward all to
your CSQ and they should see "Forward for Mr. Smith's office" on their
phone.

On 5/24/07, Jerri Robbins <jrobbins at mercurypay.com> wrote:
>
>
>
>
> I need to set up some DNs to transfer to a CSQ and show which number was
> dialed so the Support staff who answers the calls can answer appropriately ?
> i.e. "Mr. Smith's office" depending on which number was called.  Is it
> possible to set up a call handler in Unity to do this?
>
>
>
> Jerri
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From nick.jon.griffin at gmail.com  Thu May 24 18:39:35 2007
From: nick.jon.griffin at gmail.com (Nick Griffin)
Date: Thu, 24 May 2007 17:39:35 -0500
Subject: [cisco-voip] Hunt Groups and DND
Message-ID: <a2b352c40705241539r28305d68h98ec90615ddd4eee@mail.gmail.com>

I have a 5.1 install coming up in the near future. On the customers existing
system they currently use sequential hunt groups in combination with DND.
Their goal is to route customer calls to reps ASAP, the do not disturb
feature keeps the customers wait short by not routing the call to the hunt
group member while that member is in DND mode. Does anyone know of a similar
solution that could be accomplish on CCM5.1? I'm thinking Extension Mobility
with 1 button login, but that seems to take to long with the reset. They
essentially need to be able to "log out" of a hunt group. Looking for any
suggestions.

Regards,

Nick Griffin
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From jsteinberg at gmail.com  Thu May 24 18:49:14 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 24 May 2007 18:49:14 -0400
Subject: [cisco-voip] Hunt Groups and DND
In-Reply-To: <a2b352c40705241539r28305d68h98ec90615ddd4eee@mail.gmail.com>
References: <a2b352c40705241539r28305d68h98ec90615ddd4eee@mail.gmail.com>
Message-ID: <f1b65f880705241549o7176707dnf11f039b3daf82e1@mail.gmail.com>

This exact feature is in 4.2 or 6.0.

You can hack it in 4.1 or 5.1 by using extension mobility and have
their user device profile add a second line that is the hunt group
member.

Others may have different approaches...



On 5/24/07, Nick Griffin <nick.jon.griffin at gmail.com> wrote:
> I have a 5.1 install coming up in the near future. On the customers existing
> system they currently use sequential hunt groups in combination with DND.
> Their goal is to route customer calls to reps ASAP, the do not disturb
> feature keeps the customers wait short by not routing the call to the hunt
> group member while that member is in DND mode. Does anyone know of a similar
> solution that could be accomplish on CCM5.1? I'm thinking Extension Mobility
> with 1 button login, but that seems to take to long with the reset. They
> essentially need to be able to "log out" of a hunt group. Looking for any
> suggestions.
>
>  Regards,
>
> Nick Griffin
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From asobihoudai at yahoo.com  Thu May 24 18:53:44 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 24 May 2007 15:53:44 -0700 (PDT)
Subject: [cisco-voip] Unallocated/Unassigned Number for 911 call...
In-Reply-To: <5d093f9a0705241113t11f38317y1a48bd0c69ada7a2@mail.gmail.com>
Message-ID: <163598.29333.qm@web57214.mail.re3.yahoo.com>

Do you only get this result with 911? What about other
numbers?


--- Jonathan Charles <jonvoip at gmail.com> wrote:

> So, made a 911 test call:
> 
> VOICEGW#
> *May 24 18:13:39.451: ISDN Se0/0/0:23 Q931: TX ->
> SETUP pd = 8
> callref = 0x004D
>         Bearer Capability i = 0x8090A2
>                 Standard = CCITT
>                 Transfer Capability = Speech
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA98394
>                 Exclusive, Channel 20
>         Calling Party Number i = 0x0181, '[removed]'
>                 Plan:ISDN, Type:Unknown
>         Called Party Number i = 0x80, '911'
>                 Plan:Unknown, Type:Unknown
> *May 24 18:13:39.527: ISDN Se0/0/0:23 Q931: RX <-
> RELEASE_COMP pd = 8
> callref = 0x804D
>         Cause i = 0x8081 - Unallocated/unassigned
> number
> 
> 
> Any ideas?
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  

From erickbe at yahoo.com  Thu May 24 21:23:47 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu, 24 May 2007 18:23:47 -0700 (PDT)
Subject: [cisco-voip] FAX not working to one number *FIXED*
Message-ID: <275605.73612.qm@web37205.mail.mud.yahoo.com>

FYI, 

We got the one number working.  We had to force the VG224 to do fax and modem passthrough globally and by setting stcapp to enable passthrough on each port. We had to do this even though SCCP (Stcapp) only supports passthrough.

Commands added to VG224 config,

 


voice service voip 
 fax protocol 
pass-through g711ulaw
 modem passthrough nse codec g711ulaw


and then this, for each voice port...
 


stcapp register capability 2/0 modem-passthrough




----- Original Message ----
From: Lelio Fulgenzi <lelio at uoguelph.ca>
To: Ahmed Elnagar <aelnagar at ACT-EG.COM>; Erick Bergquist <erickbe at yahoo.com>; ciscovoip <cisco-voip at puck.nether.net>
Sent: Saturday, May 19, 2007 9:39:10 AM
Subject: Re: [cisco-voip] FAX not working to one number

[cisco-voip] FAX not working to one number

 
 


where is this disabled?

 


  ----- Original Message ----- 

  From: 
  Ahmed 
  Elnagar 

  To: Erick Bergquist ; ciscovoip 

  Sent: Saturday, May 19, 2007 9:21 
AM

  Subject: Re: [cisco-voip] FAX not working 
  to one number

  


  
  try to make sure that ECM 
  feature is disabled. this may cause fax failure

   


  
  
  Thanks and Best 
  Regards

Ahmed A. Elnagar
Network 
  Field Engineer

    

  Advanced 
  Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab 
  St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: 
  +2010-2833868
Website: www.act-eg.com
E-mail: 
  aelnagar at act-eg.com



  

  
  From: cisco-voip-bounces at puck.nether.net 
  on behalf of Erick Bergquist
Sent: Wed 16-May-07 9:10 
  PM
To: ciscovoip
Subject: [cisco-voip] FAX not working to 
  one number



  
  I have a strange FAX issue, where a FAX machine connected to a 
  VG224 can send/receive FAX to any number but is not able to send one outside 
  number a FAX. It can receive FAX's from them fine, just not send. The FAX 
  machines phone or a phone plugged into the FAX phone line can call the # fine 
  so there is not a CSS issue, etc.

The FAX machine is plugged into a 
  VG224 with IOS 12.4(11)T , the VG224 is configured as SCCP (stcapp) and we 
  have tried setting the port capabilty to modem passthrough and not with no 
  change.

The PSTN Gateway is a Cisco 2821 with a PRI, it is MGCP and 
  running IOS 12.4(9)T and has FAX relay disabled.

We have tried 7 
  different FAX machines of different models and neither are able to call this 
  particular outside FAX machine when going over IPT. When the FAX machines are 
  plugged into a traditional analog circuit there is not a problem and the FAX 
  works fine.

Any 
  ideas?




____________________________________________________________________________________
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  list
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From bcarter at sentinel.com  Thu May 24 21:27:59 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 24 May 2007 20:27:59 -0500
Subject: [cisco-voip] Unallocated/Unassigned Number for 911 call...
In-Reply-To: <163598.29333.qm@web57214.mail.re3.yahoo.com>
References: <5d093f9a0705241113t11f38317y1a48bd0c69ada7a2@mail.gmail.com>
	<163598.29333.qm@web57214.mail.re3.yahoo.com>
Message-ID: <6A0744639B86DD459433108DBF101D26016AEC92@MAIL01.sentinel.com>

Just an update for everyone.  The customer signed a waiver with the
Telco to remove 911 service on this circuit.  Previously 911 service
utilized separate analog circuits.

So the Telco switch was saying 911 was an unallocated number because the
switch wasn't accepting 911 calls on this circuit. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Thursday, May 24, 2007 5:54 PM
To: Jonathan Charles; ciscovoip
Subject: Re: [cisco-voip] Unallocated/Unassigned Number for 911 call...

Do you only get this result with 911? What about other numbers?


--- Jonathan Charles <jonvoip at gmail.com> wrote:

> So, made a 911 test call:
> 
> VOICEGW#
> *May 24 18:13:39.451: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref

> = 0x004D
>         Bearer Capability i = 0x8090A2
>                 Standard = CCITT
>                 Transfer Capability = Speech
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA98394
>                 Exclusive, Channel 20
>         Calling Party Number i = 0x0181, '[removed]'
>                 Plan:ISDN, Type:Unknown
>         Called Party Number i = 0x80, '911'
>                 Plan:Unknown, Type:Unknown *May 24 18:13:39.527: ISDN 
> Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x804D
>         Cause i = 0x8081 - Unallocated/unassigned number
> 
> 
> Any ideas?
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
________________________________________________________________________
____________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now
(it's updated for today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow
_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip

From bcarter at sentinel.com  Thu May 24 21:31:19 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 24 May 2007 20:31:19 -0500
Subject: [cisco-voip] how do you set mgcp to 5ess
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C663@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C663@mail.adcapnetsvc.local>
Message-ID: <6A0744639B86DD459433108DBF101D26016AEC93@MAIL01.sentinel.com>

You should be able to use Primary-NI when the switch type is 5ess for
the MGCP config.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Wednesday, May 23, 2007 11:21 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how do you set mgcp to 5ess


? 
how do you set mgcp to 5ess  trying to turn up a pri with switch type
5ess  .  for some resome when we reboot the route it reverts back to
primary-NI
 
and on the callmanager theres no 5ess
have anybody seen this
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity
to which it is addressed and may
contain confidential and/or privileged material. Any review,
retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or
entities other than the intended
recipient is prohibited. If you received this message in error, please
contact the sender and delete the
material from all computers."

This original email was sent to the internet for delivery at 23 May 2007
12:22:06 -0400
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From lelio at uoguelph.ca  Thu May 24 22:05:16 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 24 May 2007 22:05:16 -0400
Subject: [cisco-voip] FAX not working to one number *FIXED*
References: <275605.73612.qm@web37205.mail.mud.yahoo.com>
Message-ID: <001a01c79e71$23a97670$c901a8c0@pdp11>

[cisco-voip] FAX not working to one numberwe're about to deploy a vg224 with sCCP.

think i should add this to my config from the start?

  ----- Original Message ----- 
  From: Erick Bergquist 
  To: Lelio Fulgenzi ; Ahmed Elnagar ; ciscovoip 
  Sent: Thursday, May 24, 2007 9:23 PM
  Subject: Re: [cisco-voip] FAX not working to one number *FIXED*


  FYI, 

  We got the one number working.  We had to force the VG224 to do fax and modem passthrough globally and by setting stcapp to enable passthrough on each port. We had to do this even though SCCP (Stcapp) only supports passthrough.

  Commands added to VG224 config,


  voice service voip 
   fax protocol pass-through g711ulaw
   modem passthrough nse codec g711ulaw

  and then this, for each voice port...
   

  stcapp register capability 2/0 modem-passthrough




  ----- Original Message ----
  From: Lelio Fulgenzi <lelio at uoguelph.ca>
  To: Ahmed Elnagar <aelnagar at ACT-EG.COM>; Erick Bergquist <erickbe at yahoo.com>; ciscovoip <cisco-voip at puck.nether.net>
  Sent: Saturday, May 19, 2007 9:39:10 AM
  Subject: Re: [cisco-voip] FAX not working to one number


  where is this disabled?

    ----- Original Message ----- 
    From: Ahmed Elnagar 
    To: Erick Bergquist ; ciscovoip 
    Sent: Saturday, May 19, 2007 9:21 AM
    Subject: Re: [cisco-voip] FAX not working to one number


    try to make sure that ECM feature is disabled. this may cause fax failure

    Thanks and Best Regards

    Ahmed A. Elnagar
    Network Field Engineer

    Advanced Computer Technology (ACT)
    16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
    Postal Code:12411 Cairo Egypt

    Mob: +2010-2833868
    Website: www.act-eg.com
    E-mail: aelnagar at act-eg.com


----------------------------------------------------------------------------
    From: cisco-voip-bounces at puck.nether.net on behalf of Erick Bergquist
    Sent: Wed 16-May-07 9:10 PM
    To: ciscovoip
    Subject: [cisco-voip] FAX not working to one number


    I have a strange FAX issue, where a FAX machine connected to a VG224 can send/receive FAX to any number but is not able to send one outside number a FAX. It can receive FAX's from them fine, just not send. The FAX machines phone or a phone plugged into the FAX phone line can call the # fine so there is not a CSS issue, etc.

    The FAX machine is plugged into a VG224 with IOS 12.4(11)T , the VG224 is configured as SCCP (stcapp) and we have tried setting the port capabilty to modem passthrough and not with no change.

    The PSTN Gateway is a Cisco 2821 with a PRI, it is MGCP and running IOS 12.4(9)T and has FAX relay disabled.

    We have tried 7 different FAX machines of different models and neither are able to call this particular outside FAX machine when going over IPT. When the FAX machines are plugged into a traditional analog circuit there is not a problem and the FAX works fine.

    Any ideas?




    ____________________________________________________________________________________
    Sucker-punch spam with award-winning protection.
    Try the free Yahoo! Mail Beta.
    http://advision.webevents.yahoo.com/mailbeta/features_spam.html
    _______________________________________________
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    https://puck.nether.net/mailman/listinfo/cisco-voip




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From voipjockey at gmail.com  Thu May 24 23:06:28 2007
From: voipjockey at gmail.com (Jimmy Pale)
Date: Thu, 24 May 2007 22:06:28 -0500
Subject: [cisco-voip] how do you set mgcp to 5ess
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C663@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C663@mail.adcapnetsvc.local>
Message-ID: <002f01c79e79$b21aca00$16505e00$@com>

5E8 Custom

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Wednesday, May 23, 2007 11:21 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how do you set mgcp to 5ess

 

?

how do you set mgcp to 5ess  trying to turn up a pri with switch type 5ess
.  for some resome when we reboot the route it reverts back to primary-NI

 

and on the callmanager theres no 5ess

have anybody seen this

 

James Grace 

 

Systems Engineer

Adcap Network Systems, Inc.

6865 Shiloh Road East, Suite 250

Alpharetta, GA 30005

 

office: 678-456-6986

direct: 678-456-6761

jgrace at adcapnet.com

www.adcapnet.com <http://www.adcapnet.com/> 

Disclaimer For ADCAP Network Systems
 
"The information transmitted is intended only for the person or entity to
which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission,
dissemination or other use of,
or taking of any action in reliance upon this information by persons or
entities other than the intended
recipient is prohibited. If you received this message in error, please
contact the sender and delete the
material from all computers."
 
This original email was sent to the internet for delivery at 23 May 2007
12:22:06 -0400
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From erickbe at yahoo.com  Thu May 24 22:29:56 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu, 24 May 2007 19:29:56 -0700 (PDT)
Subject: [cisco-voip] FAX not working to one number *FIXED*
Message-ID: <290043.780.qm@web37215.mail.mud.yahoo.com>

Well, since SCCP only supports passthrough, I don't see it causing any problems if the commands are there. I guess the commands force it to skip over figuring out what to do dynamically, but if it can only do passthrough with SCCP then there is a breakdown somewhere. Oh well, I am happy to get past another weird FAX problem with one number and not causing any other FAX problems. 

Now, TAC was suggesting going to Fax Relay now with MGCP or H323 and moving away from passthrough. So there must have been improvements, as before recent IOS versions passthrough was way to go. The new IOS versions (recent 12.4T) has a fax-relay command to make SG3 fax go to G3 fax also, but this is when doing Fax relay. 



----- Original Message ----
From: Lelio Fulgenzi <lelio at uoguelph.ca>
To: Erick Bergquist <erickbe at yahoo.com>; Ahmed Elnagar <aelnagar at ACT-EG.COM>; ciscovoip <cisco-voip at puck.nether.net>
Sent: Thursday, May 24, 2007 9:05:16 PM
Subject: Re: [cisco-voip] FAX not working to one number *FIXED*

[cisco-voip] FAX not working to one number

 
DIV {
MARGIN:0px;}



we're about to deploy a vg224 with 
sCCP.

 

think i should add this to my config from the 
start?

 


  ----- Original Message ----- 

  From: 
  Erick Bergquist 
  

  To: Lelio Fulgenzi ; Ahmed Elnagar 
  ; ciscovoip 

  Sent: Thursday, May 24, 2007 9:23 
PM

  Subject: Re: [cisco-voip] FAX not working 
  to one number *FIXED*

  


  FYI, 
  

We got the one number working.  We had to force the VG224 to do 
  fax and modem passthrough globally and by setting stcapp to enable passthrough 
  on each port. We had to do this even though SCCP (Stcapp) only supports 
  passthrough.

Commands added to VG224 config,

   

  
  voice service voip 
 fax protocol 
  pass-through g711ulaw
 modem passthrough nse codec g711ulaw

  
and then this, for each voice port...
 


  stcapp register capability 2/0 
  modem-passthrough





  
  ----- 
  Original Message ----
From: Lelio Fulgenzi <lelio at uoguelph.ca>
To: 
  Ahmed Elnagar <aelnagar at ACT-EG.COM>; Erick Bergquist 
  <erickbe at yahoo.com>; ciscovoip 
  <cisco-voip at puck.nether.net>
Sent: Saturday, May 19, 2007 9:39:10 
  AM
Subject: Re: [cisco-voip] FAX not working to one number


  

  where is this disabled?

   

  
    ----- 
    Original Message ----- 

    From: 
    Ahmed Elnagar 

    To: 
    Erick Bergquist ; ciscovoip 

    Sent: 
    Saturday, May 19, 2007 9:21 AM

    Subject: 
    Re: [cisco-voip] FAX not working to one number

    


    
    try to make sure that ECM 
    feature is disabled. this may cause fax failure

     


    
    
    Thanks 
    and Best Regards

Ahmed A. Elnagar
Network 
    Field Engineer

     

    Advanced 
    Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab 
    St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo 
    Egypt

Mob: 
    +2010-2833868
Website: 
    www.act-eg.com
E-mail: 
    aelnagar at act-eg.com



    

    
    From: cisco-voip-bounces at puck.nether.net on behalf of Erick 
    Bergquist
Sent: Wed 16-May-07 9:10 PM
To: 
    ciscovoip
Subject: [cisco-voip] FAX not working to one 
    number



    
    I have a strange FAX issue, where a FAX machine connected to 
    a VG224 can send/receive FAX to any number but is not able to send one 
    outside number a FAX. It can receive FAX's from them fine, just not send. 
    The FAX machines phone or a phone plugged into the FAX phone line can call 
    the # fine so there is not a CSS issue, etc.

The FAX machine is 
    plugged into a VG224 with IOS 12.4(11)T , the VG224 is configured as SCCP 
    (stcapp) and we have tried setting the port capabilty to modem passthrough 
    and not with no change.

The PSTN Gateway is a Cisco 2821 with a PRI, 
    it is MGCP and running IOS 12.4(9)T and has FAX relay disabled.

We 
    have tried 7 different FAX machines of different models and neither are able 
    to call this particular outside FAX machine when going over IPT. When the 
    FAX machines are plugged into a traditional analog circuit there is not a 
    problem and the FAX works fine.

Any 
    ideas?




____________________________________________________________________________________
Sucker-punch 
    spam with award-winning protection.
Try the free Yahoo! Mail Beta.
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    list
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From lelio at uoguelph.ca  Thu May 24 22:48:48 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 24 May 2007 22:48:48 -0400
Subject: [cisco-voip] FAX not working to one number *FIXED*
References: <290043.780.qm@web37215.mail.mud.yahoo.com>
Message-ID: <001101c79e77$3883d850$c901a8c0@pdp11>

[cisco-voip] FAX not working to one numberIs it possible to have an VG224 with ports using different protocols?

  ----- Original Message ----- 
  From: Erick Bergquist 
  To: Lelio Fulgenzi ; Ahmed Elnagar ; ciscovoip 
  Sent: Thursday, May 24, 2007 10:29 PM
  Subject: Re: [cisco-voip] FAX not working to one number *FIXED*


  Well, since SCCP only supports passthrough, I don't see it causing any problems if the commands are there. I guess the commands force it to skip over figuring out what to do dynamically, but if it can only do passthrough with SCCP then there is a breakdown somewhere. Oh well, I am happy to get past another weird FAX problem with one number and not causing any other FAX problems. 

  Now, TAC was suggesting going to Fax Relay now with MGCP or H323 and moving away from passthrough. So there must have been improvements, as before recent IOS versions passthrough was way to go. The new IOS versions (recent 12.4T) has a fax-relay command to make SG3 fax go to G3 fax also, but this is when doing Fax relay. 





  ----- Original Message ----
  From: Lelio Fulgenzi <lelio at uoguelph.ca>
  To: Erick Bergquist <erickbe at yahoo.com>; Ahmed Elnagar <aelnagar at ACT-EG.COM>; ciscovoip <cisco-voip at puck.nether.net>
  Sent: Thursday, May 24, 2007 9:05:16 PM
  Subject: Re: [cisco-voip] FAX not working to one number *FIXED*


  we're about to deploy a vg224 with sCCP.

  think i should add this to my config from the start?

    ----- Original Message ----- 
    From: Erick Bergquist 
    To: Lelio Fulgenzi ; Ahmed Elnagar ; ciscovoip 
    Sent: Thursday, May 24, 2007 9:23 PM
    Subject: Re: [cisco-voip] FAX not working to one number *FIXED*


    FYI, 

    We got the one number working.  We had to force the VG224 to do fax and modem passthrough globally and by setting stcapp to enable passthrough on each port. We had to do this even though SCCP (Stcapp) only supports passthrough.

    Commands added to VG224 config,


    voice service voip 
     fax protocol pass-through g711ulaw
     modem passthrough nse codec g711ulaw

    and then this, for each voice port...
     

    stcapp register capability 2/0 modem-passthrough




    ----- Original Message ----
    From: Lelio Fulgenzi <lelio at uoguelph.ca>
    To: Ahmed Elnagar <aelnagar at ACT-EG.COM>; Erick Bergquist <erickbe at yahoo.com>; ciscovoip <cisco-voip at puck.nether.net>
    Sent: Saturday, May 19, 2007 9:39:10 AM
    Subject: Re: [cisco-voip] FAX not working to one number


    where is this disabled?

      ----- Original Message ----- 
      From: Ahmed Elnagar 
      To: Erick Bergquist ; ciscovoip 
      Sent: Saturday, May 19, 2007 9:21 AM
      Subject: Re: [cisco-voip] FAX not working to one number


      try to make sure that ECM feature is disabled. this may cause fax failure

      Thanks and Best Regards

      Ahmed A. Elnagar
      Network Field Engineer

      Advanced Computer Technology (ACT)
      16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
      Postal Code:12411 Cairo Egypt

      Mob: +2010-2833868
      Website: www.act-eg.com
      E-mail: aelnagar at act-eg.com


--------------------------------------------------------------------------
      From: cisco-voip-bounces at puck.nether.net on behalf of Erick Bergquist
      Sent: Wed 16-May-07 9:10 PM
      To: ciscovoip
      Subject: [cisco-voip] FAX not working to one number


      I have a strange FAX issue, where a FAX machine connected to a VG224 can send/receive FAX to any number but is not able to send one outside number a FAX. It can receive FAX's from them fine, just not send. The FAX machines phone or a phone plugged into the FAX phone line can call the # fine so there is not a CSS issue, etc.

      The FAX machine is plugged into a VG224 with IOS 12.4(11)T , the VG224 is configured as SCCP (stcapp) and we have tried setting the port capabilty to modem passthrough and not with no change.

      The PSTN Gateway is a Cisco 2821 with a PRI, it is MGCP and running IOS 12.4(9)T and has FAX relay disabled.

      We have tried 7 different FAX machines of different models and neither are able to call this particular outside FAX machine when going over IPT. When the FAX machines are plugged into a traditional analog circuit there is not a problem and the FAX works fine.

      Any ideas?




      ____________________________________________________________________________________
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      _______________________________________________
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      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip





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From daniel.abelski at gmail.com  Fri May 25 01:16:01 2007
From: daniel.abelski at gmail.com (Daniel A. Abelski)
Date: Fri, 25 May 2007 11:16:01 +0600
Subject: [cisco-voip] =?koi8-r?b?79TXxdQ6ICBPcGVuIFNvdXJjZSBNQ1U=?=
In-Reply-To: <20070524201317.r97euvankskcg0oo@mail.carocomp.ro>
References: <963CE94787005149A52C17F33A028E1207E62003@EVS1B.boco.co.boulder.co.us>
	<4655B41A.7070608@infiltrated.net>
	<20070524201317.r97euvankskcg0oo@mail.carocomp.ro>
Message-ID: <cd0147ab0705242216lbc4bcaehbb2cce7eb7d33c6d@mail.gmail.com>

I was using openmcu. It was ok. But only h.261, g711 and no
transcoding, So if this is right for you - use it.

2007/5/24, keli at carocomp.ro <keli at carocomp.ro>:
> OpenH323-based GnuGK (or OpenGK) is quite good and stable. However, I
> don't know much about their MCU implementation. I know for sure, that
> the project was stalled for quite a while before it got a fresh boot
> about a year ago (or maybe less)
>
> As far as I know, one of the project's sponsors was Stonevoice, a
> European Cisco partner company, who now has a software MCU in their
> product list. Maybe it's worth checking out their site
> http://www.stonevoice.com/
> [They have some other nice & reasonably priced side-products as well
> for CME environments]
>
> regards
> Zoltan
>
> Quoting "J. Oquendo" <sil at infiltrated.net>:
>
> > Leetun, Rob wrote:
> >> Hi,
> >> What is good open source MCU?  Has anyone implemented an open
> >> source MCU, and if so, how did it go?
> >> I am plan on using it in a CCM 4.1.3 environment along with a 2821
> >> gatekeeper.
> >> Thanks.
> >> Rob
> >> ------------------------------------------------------------------------
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> > Don't know about production systems though...
> >
> > http://www.openh323.org/
> > http://www.voxgratia.org/modules/news/
> >
> >
> > --
> > ====================================================
> > J. Oquendo
> > http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
> > echo infiltrated.net|sed 's/^/sil@/g' "Wise men talk because they have
> > something to say;
> > fools, because they have to say something." -- Plato
>
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


-- 
? ?????????,
????????? ?????? ?????????????

From bridge_stone at gmx.net  Fri May 25 04:41:34 2007
From: bridge_stone at gmx.net (Konne)
Date: Fri, 25 May 2007 10:41:34 +0200
Subject: [cisco-voip] Problem with 7920 - ringing but no sound
In-Reply-To: <465575F3.1070503@gmx.net>
References: <46556567.3010904@gmx.net> <46556B14.1080505@saskin.net>
	<465575F3.1070503@gmx.net>
Message-ID: <4656A13E.803@gmx.net>

Konne wrote:

> hi,
>
> no audio at all, only a noise...
>
> bye
>
> Matthew Saskin wrote:
>
>> Are you getting no audio at all or only one way audio?
>>
>> -matt
>>
>> Konne wrote:
>>
>>> hi @ all,
>>>
>>> i have a little problem with my Cisco VoIP. I' ve a WLAN 
>>> Infrastructure and i would like to implement VoIP now.
>>> my problem is that the phones are communicating well, only that i 
>>> can hear nothing.
>>>
>>> have you any suggests
>>>
>>> thx
>>>
>>>
>>> my network:
>>>
>>>
>>> Router2811 with DSP ----- TRUNK dot1q ------ 
>>> Catalyst3750------------ ACS
>>>                                                                          
>>>                   |
>>>                                                                          
>>>                   |
>>>                                                                          
>>>             AP1240
>>>
>>>
>>>                                                           ephone1 
>>> 7920                     ephone2 7920
>>>
>>>
>>> my config Router2811:
>>>
>>> router2811#sh run
>>> Building configuration...
>>>
>>> Current configuration : 2961 bytes
>>> !
>>> ! Last configuration change at 11:52:04 MESZ Thu May 24 2007 by 
>>> router2811
>>> ! NVRAM config last updated at 11:52:06 MESZ Thu May 24 2007 by 
>>> router2811
>>> !
>>> version 12.3
>>> service timestamps debug datetime msec
>>> service timestamps log datetime msec
>>> service password-encryption
>>> !
>>> hostname router2811
>>> !
>>> boot-start-marker
>>> boot-end-marker
>>> !
>>> logging buffered 51200 warnings
>>> !
>>> clock timezone MEZ 1
>>> clock summer-time MESZ recurring last Sun Mar 2:00 last Sun Oct 3:00
>>> no aaa new-model
>>> ip subnet-zero
>>> !
>>> !
>>> ip cef
>>> !
>>> !
>>> ip domain name +++++
>>> ip name-server 192.168.138.10
>>> ip name-server 192.168.138.11
>>> no ftp-server write-enable
>>> !
>>> voice-card 0
>>>  no dspfarm
>>>  dsp services dspfarm
>>> !
>>> username +++++++ privilege 15 secret 5 ++++++++
>>> !
>>> !
>>> !
>>> !
>>> interface FastEthernet0/0
>>>  no ip address
>>>  duplex auto
>>>  speed auto
>>> !
>>> interface FastEthernet0/0.1
>>>  description Management
>>>  encapsulation dot1Q 1 native
>>>  ip address 192.168.250.20 255.255.255.0
>>> !
>>> interface FastEthernet0/0.138
>>>  description Server
>>>  encapsulation dot1Q 138
>>>  ip address 192.168.138.2 255.255.255.0
>>>  no cdp enable
>>> !
>>> interface FastEthernet0/0.184
>>>  description Admin
>>>  encapsulation dot1Q 184
>>>  ip address 192.168.184.2 255.255.255.0
>>>  no cdp enable
>>> !
>>> interface FastEthernet0/0.880
>>>  description VoIP
>>>  encapsulation dot1Q 880
>>>  ip address 10.95.180.5 255.255.255.0
>>> !
>>> interface FastEthernet0/1
>>>  no ip address
>>>  shutdown
>>>  duplex auto
>>>  speed auto
>>> !
>>> ip classless
>>> !
>>> !
>>> no ip http server
>>> ip http authentication local
>>> ip http secure-server
>>> ip http timeout-policy idle 60 life 86400 requests 10000
>>> !
>>> access-list 10 permit 192.168.184.0 0.0.0.255
>>> access-list 10 permit 192.168.185.0 0.0.0.7
>>> access-list 10 permit 192.168.250.0 0.0.0.255
>>> !
>>> !
>>> !
>>> control-plane
>>> !
>>> telephony-service
>>>  max-ephones 10
>>>  max-dn 10
>>>  ip source-address 10.95.180.5 port 2000
>>>  time-format 24
>>>  date-format dd-mm-yy
>>>  user-locale DE
>>>  network-locale IT
>>>  create cnf-files version-stamp 7960 May 23 2007 12:24:26
>>>  max-conferences 8
>>>  transfer-system full-blind
>>>  transfer-pattern ...
>>> !
>>> !
>>> ephone-dn  1
>>>  number 750
>>> !
>>> !
>>> ephone-dn  2
>>>  number 751
>>> !
>>> !
>>> ephone  1
>>>  mac-address 0019.AAmac
>>>  button  1:1
>>> !
>>> !
>>> !
>>> ephone  2
>>>  mac-address 0019.AAmac2
>>>  button  1:2
>>> !
>>> !
>>> line con 0
>>>  login local
>>> line aux 0
>>> line vty 0 4
>>>  access-class 10 in
>>>  exec-timeout 30 0
>>>  privilege level 15
>>>  login local
>>>  transport input telnet ssh
>>> line vty 5 15
>>>  access-class 10 in
>>>  exec-timeout 30 0
>>>  privilege level 15
>>>  login local
>>>  transport input telnet ssh
>>> !
>>> scheduler allocate 20000 1000
>>> ntp server 62.101.81.203 source FastEthernet0/0.138
>>> ntp server 62.152.126.5 source FastEthernet0/0.138
>>> !
>>> end
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>   
>>
>>
>>
>
>


From jeremie.ceintrey at gmail.com  Fri May 25 05:21:51 2007
From: jeremie.ceintrey at gmail.com (Jeremie Ceintrey)
Date: Fri, 25 May 2007 11:21:51 +0200
Subject: [cisco-voip] extension mobility login question
In-Reply-To: <53E2851455813A40B20FB58AB3A98D0801CB1FAC@xmb-rtp-20d.amer.cisco.com>
References: <53E2851455813A40B20FB58AB3A98D0801CB1FAC@xmb-rtp-20d.amer.cisco.com>
Message-ID: <dc3a89dd0705250221y60154563w11e8abce913ff6ab@mail.gmail.com>

Hi Frank,

I've never tried it, but I'm pretty sure that it works like the reset. When
you reset a phone on witch a user is logged in, or when you disconnect a
phone, the user is not logged out...
If test it, please tell me how it works, but i think, to answer your
question, that the user is not logged out.

Best regards.
J?r?mie, France.

2007/5/24, Frank Kauffman IV (fkauffma) <fkauffma at cisco.com>:
>
>  If a user is logged into the phone, and the phone is upgraded and has
> registered back with Callmanager, will the user automatically be logged back
> in to the phone?  Or is it necessary for user to log back in manually.
>
> *Best Regards,*
>
>
> *Frank Kauffman*
> *Customer Support Engineer*
> *Cisco Remote Operations Services
> *
> fkauffma at cisco.com
> Phone :*919.574.0285*
> Mobile :*919.749.4647*
>
> **
> 7100-8 Kit Creek Road
> Research Triangle Park, NC. 27709
> United States
> www.cisco.com
>      This e-mail may contain confidential and privileged material for the
> sole use of the intended recipient. Any review, use, distribution or
> disclosure by others is strictly prohibited. If you are not the intended
> recipient (or authorized to receive for the recipient), please contact the
> sender by reply e-mail and delete all copies of this message.
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From tele at plexia.com  Fri May 25 06:20:10 2007
From: tele at plexia.com (tele)
Date: Fri, 25 May 2007 12:20:10 +0200
Subject: [cisco-voip] CME 4.1 SIP Phone MOH
Message-ID: <1180088410.6586.307.camel@mda>

Dear all,

i've a CME 4.1 and SIP phones and when i put for example an on-net call
on-hold MOH is not working.
I don't know if MOH works only with "ephone" and not with "voice
register-dn" it's not clear, Cisco says: "Configuring MOH for SIP phones
is the same as configuring MOH for SCCP phones."
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00807dab90.html#wp1010333
So the configuration is really simple

I starts troubleshooting MOH for SIP phones, the codec is correct, and
the stream goes in sendonly mode but i can't hear MOH,
debug ccsip all give some informations.
The same MOH works fine in the auto attendant.

Anybody have a working MOH with CME and only SIP phones? could you
please provide to me the debug ccsip all output of the Re-invite for the
on-hold? 
i want to compare my debug with a working MOH with SIP phones.

kind regards,

:tele






From egoppel at gmail.com  Fri May 25 08:49:40 2007
From: egoppel at gmail.com (Erik Goppel)
Date: Fri, 25 May 2007 14:49:40 +0200
Subject: [cisco-voip] Problem with 7920 - ringing but no sound
In-Reply-To: <4656A13E.803@gmx.net>
References: <46556567.3010904@gmx.net> <46556B14.1080505@saskin.net>
	<465575F3.1070503@gmx.net> <4656A13E.803@gmx.net>
Message-ID: <47fb51250705250549tc05008eo8e5a85d2c6c8e864@mail.gmail.com>

Konne,

Is that you have ringing on the phone, but you don`t hear the ringtone? or
is it no audio in your call?
Can you hear the ringtone, if you change it on the phone?
It could be, that the ringtones are not served on you CME, you will have to
include them in your tftp-server configuration, which is missing by the way
in your configuration.
If you are running cme, and the raw files are not uploaded to the phone, but
the are accesible in your (default) ringlist, it cannot play the audio,
excepte for the default raw files that come preloaded in the phone.

So i would advise you to read the CME installation and configuration
document, on how to configure this correctly.

good luck,

erik


2007/5/25, Konne <bridge_stone at gmx.net>:
>
> Konne wrote:
>
> > hi,
> >
> > no audio at all, only a noise...
> >
> > bye
> >
> > Matthew Saskin wrote:
> >
> >> Are you getting no audio at all or only one way audio?
> >>
> >> -matt
> >>
> >> Konne wrote:
> >>
> >>> hi @ all,
> >>>
> >>> i have a little problem with my Cisco VoIP. I' ve a WLAN
> >>> Infrastructure and i would like to implement VoIP now.
> >>> my problem is that the phones are communicating well, only that i
> >>> can hear nothing.
> >>>
> >>> have you any suggests
> >>>
> >>> thx
> >>>
> >>>
> >>> my network:
> >>>
> >>>
> >>> Router2811 with DSP ----- TRUNK dot1q ------
> >>> Catalyst3750------------ ACS
> >>>
> >>>                   |
> >>>
> >>>                   |
> >>>
> >>>             AP1240
> >>>
> >>>
> >>>                                                           ephone1
> >>> 7920                     ephone2 7920
> >>>
> >>>
> >>> my config Router2811:
> >>>
> >>> router2811#sh run
> >>> Building configuration...
> >>>
> >>> Current configuration : 2961 bytes
> >>> !
> >>> ! Last configuration change at 11:52:04 MESZ Thu May 24 2007 by
> >>> router2811
> >>> ! NVRAM config last updated at 11:52:06 MESZ Thu May 24 2007 by
> >>> router2811
> >>> !
> >>> version 12.3
> >>> service timestamps debug datetime msec
> >>> service timestamps log datetime msec
> >>> service password-encryption
> >>> !
> >>> hostname router2811
> >>> !
> >>> boot-start-marker
> >>> boot-end-marker
> >>> !
> >>> logging buffered 51200 warnings
> >>> !
> >>> clock timezone MEZ 1
> >>> clock summer-time MESZ recurring last Sun Mar 2:00 last Sun Oct 3:00
> >>> no aaa new-model
> >>> ip subnet-zero
> >>> !
> >>> !
> >>> ip cef
> >>> !
> >>> !
> >>> ip domain name +++++
> >>> ip name-server 192.168.138.10
> >>> ip name-server 192.168.138.11
> >>> no ftp-server write-enable
> >>> !
> >>> voice-card 0
> >>>  no dspfarm
> >>>  dsp services dspfarm
> >>> !
> >>> username +++++++ privilege 15 secret 5 ++++++++
> >>> !
> >>> !
> >>> !
> >>> !
> >>> interface FastEthernet0/0
> >>>  no ip address
> >>>  duplex auto
> >>>  speed auto
> >>> !
> >>> interface FastEthernet0/0.1
> >>>  description Management
> >>>  encapsulation dot1Q 1 native
> >>>  ip address 192.168.250.20 255.255.255.0
> >>> !
> >>> interface FastEthernet0/0.138
> >>>  description Server
> >>>  encapsulation dot1Q 138
> >>>  ip address 192.168.138.2 255.255.255.0
> >>>  no cdp enable
> >>> !
> >>> interface FastEthernet0/0.184
> >>>  description Admin
> >>>  encapsulation dot1Q 184
> >>>  ip address 192.168.184.2 255.255.255.0
> >>>  no cdp enable
> >>> !
> >>> interface FastEthernet0/0.880
> >>>  description VoIP
> >>>  encapsulation dot1Q 880
> >>>  ip address 10.95.180.5 255.255.255.0
> >>> !
> >>> interface FastEthernet0/1
> >>>  no ip address
> >>>  shutdown
> >>>  duplex auto
> >>>  speed auto
> >>> !
> >>> ip classless
> >>> !
> >>> !
> >>> no ip http server
> >>> ip http authentication local
> >>> ip http secure-server
> >>> ip http timeout-policy idle 60 life 86400 requests 10000
> >>> !
> >>> access-list 10 permit 192.168.184.0 0.0.0.255
> >>> access-list 10 permit 192.168.185.0 0.0.0.7
> >>> access-list 10 permit 192.168.250.0 0.0.0.255
> >>> !
> >>> !
> >>> !
> >>> control-plane
> >>> !
> >>> telephony-service
> >>>  max-ephones 10
> >>>  max-dn 10
> >>>  ip source-address 10.95.180.5 port 2000
> >>>  time-format 24
> >>>  date-format dd-mm-yy
> >>>  user-locale DE
> >>>  network-locale IT
> >>>  create cnf-files version-stamp 7960 May 23 2007 12:24:26
> >>>  max-conferences 8
> >>>  transfer-system full-blind
> >>>  transfer-pattern ...
> >>> !
> >>> !
> >>> ephone-dn  1
> >>>  number 750
> >>> !
> >>> !
> >>> ephone-dn  2
> >>>  number 751
> >>> !
> >>> !
> >>> ephone  1
> >>>  mac-address 0019.AAmac
> >>>  button  1:1
> >>> !
> >>> !
> >>> !
> >>> ephone  2
> >>>  mac-address 0019.AAmac2
> >>>  button  1:2
> >>> !
> >>> !
> >>> line con 0
> >>>  login local
> >>> line aux 0
> >>> line vty 0 4
> >>>  access-class 10 in
> >>>  exec-timeout 30 0
> >>>  privilege level 15
> >>>  login local
> >>>  transport input telnet ssh
> >>> line vty 5 15
> >>>  access-class 10 in
> >>>  exec-timeout 30 0
> >>>  privilege level 15
> >>>  login local
> >>>  transport input telnet ssh
> >>> !
> >>> scheduler allocate 20000 1000
> >>> ntp server 62.101.81.203 source FastEthernet0/0.138
> >>> ntp server 62.152.126.5 source FastEthernet0/0.138
> >>> !
> >>> end
> >>>
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip at puck.nether.net
> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>
> >>
> >>
> >>
> >
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From bridge_stone at gmx.net  Fri May 25 09:13:02 2007
From: bridge_stone at gmx.net (Konne)
Date: Fri, 25 May 2007 15:13:02 +0200
Subject: [cisco-voip] Problem with 7920 - ringing but no sound
In-Reply-To: <47fb51250705250549tc05008eo8e5a85d2c6c8e864@mail.gmail.com>
References: <46556567.3010904@gmx.net> <46556B14.1080505@saskin.net>	
	<465575F3.1070503@gmx.net> <4656A13E.803@gmx.net>
	<47fb51250705250549tc05008eo8e5a85d2c6c8e864@mail.gmail.com>
Message-ID: <4656E0DE.5060907@gmx.net>

hi,

now i've found the problem i have configured the wlancontroller to do 
"Peer to Peer Blocking Mode". now all is ok.

thx

Erik Goppel wrote:

> Konne,
>  
> Is that you have ringing on the phone, but you don`t hear the 
> ringtone? or is it no audio in your call?
> Can you hear the ringtone, if you change it on the phone?
> It could be, that the ringtones are not served on you CME, you will 
> have to include them in your tftp-server configuration, which is 
> missing by the way in your configuration.
> If you are running cme, and the raw files are not uploaded to the 
> phone, but the are accesible in your (default) ringlist, it cannot 
> play the audio, excepte for the default raw files that come preloaded 
> in the phone.
>  
> So i would advise you to read the CME installation and configuration 
> document, on how to configure this correctly.
>  
> good luck,
>  
> erik
>
>  
> 2007/5/25, Konne <bridge_stone at gmx.net <mailto:bridge_stone at gmx.net>>:
>
>     Konne wrote:
>
>     > hi,
>     >
>     > no audio at all, only a noise...
>     >
>     > bye
>     >
>     > Matthew Saskin wrote:
>     >
>     >> Are you getting no audio at all or only one way audio?
>     >>
>     >> -matt
>     >>
>     >> Konne wrote:
>     >>
>     >>> hi @ all,
>     >>>
>     >>> i have a little problem with my Cisco VoIP. I' ve a WLAN
>     >>> Infrastructure and i would like to implement VoIP now.
>     >>> my problem is that the phones are communicating well, only that i
>     >>> can hear nothing.
>     >>>
>     >>> have you any suggests
>     >>>
>     >>> thx
>     >>>
>     >>>
>     >>> my network:
>     >>>
>     >>>
>     >>> Router2811 with DSP ----- TRUNK dot1q ------
>     >>> Catalyst3750------------ ACS
>     >>>
>     >>>                   |
>     >>>
>     >>>                   |
>     >>>
>     >>>             AP1240
>     >>>
>     >>>
>     >>>                                                           ephone1
>     >>> 7920                     ephone2 7920
>     >>>
>     >>>
>     >>> my config Router2811:
>     >>>
>     >>> router2811#sh run
>     >>> Building configuration...
>     >>>
>     >>> Current configuration : 2961 bytes
>     >>> !
>     >>> ! Last configuration change at 11:52:04 MESZ Thu May 24 2007 by
>     >>> router2811
>     >>> ! NVRAM config last updated at 11:52:06 MESZ Thu May 24 2007 by
>     >>> router2811
>     >>> !
>     >>> version 12.3
>     >>> service timestamps debug datetime msec
>     >>> service timestamps log datetime msec
>     >>> service password-encryption
>     >>> !
>     >>> hostname router2811
>     >>> !
>     >>> boot-start-marker
>     >>> boot-end-marker
>     >>> !
>     >>> logging buffered 51200 warnings
>     >>> !
>     >>> clock timezone MEZ 1
>     >>> clock summer-time MESZ recurring last Sun Mar 2:00 last Sun
>     Oct 3:00
>     >>> no aaa new-model
>     >>> ip subnet-zero
>     >>> !
>     >>> !
>     >>> ip cef
>     >>> !
>     >>> !
>     >>> ip domain name +++++
>     >>> ip name-server 192.168.138.10 <http://192.168.138.10>
>     >>> ip name-server 192.168.138.11 <http://192.168.138.11>
>     >>> no ftp-server write-enable
>     >>> !
>     >>> voice-card 0
>     >>>  no dspfarm
>     >>>  dsp services dspfarm
>     >>> !
>     >>> username +++++++ privilege 15 secret 5 ++++++++
>     >>> !
>     >>> !
>     >>> !
>     >>> !
>     >>> interface FastEthernet0/0
>     >>>  no ip address
>     >>>  duplex auto
>     >>>  speed auto
>     >>> !
>     >>> interface FastEthernet0/0.1
>     >>>  description Management
>     >>>  encapsulation dot1Q 1 native
>     >>>  ip address 192.168.250.20 <http://192.168.250.20>
>     255.255.255.0 <http://255.255.255.0>
>     >>> !
>     >>> interface FastEthernet0/0.138
>     >>>  description Server
>     >>>  encapsulation dot1Q 138
>     >>>  ip address 192.168.138.2 <http://192.168.138.2> 255.255.255.0
>     <http://255.255.255.0>
>     >>>  no cdp enable
>     >>> !
>     >>> interface FastEthernet0/0.184
>     >>>  description Admin
>     >>>  encapsulation dot1Q 184
>     >>>  ip address 192.168.184.2 <http://192.168.184.2> 255.255.255.0
>     <http://255.255.255.0>
>     >>>  no cdp enable
>     >>> !
>     >>> interface FastEthernet0/0.880
>     >>>  description VoIP
>     >>>  encapsulation dot1Q 880
>     >>>  ip address 10.95.180.5 <http://10.95.180.5> 255.255.255.0
>     <http://255.255.255.0>
>     >>> !
>     >>> interface FastEthernet0/1
>     >>>  no ip address
>     >>>  shutdown
>     >>>  duplex auto
>     >>>  speed auto
>     >>> !
>     >>> ip classless
>     >>> !
>     >>> !
>     >>> no ip http server
>     >>> ip http authentication local
>     >>> ip http secure-server
>     >>> ip http timeout-policy idle 60 life 86400 requests 10000
>     >>> !
>     >>> access-list 10 permit 192.168.184.0 <http://192.168.184.0>
>     0.0.0.255 <http://0.0.0.255>
>     >>> access-list 10 permit 192.168.185.0 <http://192.168.185.0>
>     0.0.0.7 <http://0.0.0.7>
>     >>> access-list 10 permit 192.168.250.0 <http://192.168.250.0>
>     0.0.0.255 <http://0.0.0.255>
>     >>> !
>     >>> !
>     >>> !
>     >>> control-plane
>     >>> !
>     >>> telephony-service
>     >>>  max-ephones 10
>     >>>  max-dn 10
>     >>>  ip source-address 10.95.180.5 <http://10.95.180.5> port 2000
>     >>>  time-format 24
>     >>>  date-format dd-mm-yy
>     >>>  user-locale DE
>     >>>  network-locale IT
>     >>>  create cnf-files version-stamp 7960 May 23 2007 12:24:26
>     >>>  max-conferences 8
>     >>>  transfer-system full-blind
>     >>>  transfer-pattern ...
>     >>> !
>     >>> !
>     >>> ephone-dn  1
>     >>>  number 750
>     >>> !
>     >>> !
>     >>> ephone-dn  2
>     >>>  number 751
>     >>> !
>     >>> !
>     >>> ephone  1
>     >>>  mac-address 0019.AAmac
>     >>>  button  1:1
>     >>> !
>     >>> !
>     >>> !
>     >>> ephone  2
>     >>>  mac-address 0019.AAmac2
>     >>>  button  1:2
>     >>> !
>     >>> !
>     >>> line con 0
>     >>>  login local
>     >>> line aux 0
>     >>> line vty 0 4
>     >>>  access-class 10 in
>     >>>  exec-timeout 30 0
>     >>>  privilege level 15
>     >>>  login local
>     >>>  transport input telnet ssh
>     >>> line vty 5 15
>     >>>  access-class 10 in
>     >>>  exec-timeout 30 0
>     >>>  privilege level 15
>     >>>  login local
>     >>>  transport input telnet ssh
>     >>> !
>     >>> scheduler allocate 20000 1000
>     >>> ntp server 62.101.81.203 <http://62.101.81.203> source
>     FastEthernet0/0.138
>     >>> ntp server 62.152.126.5 <http://62.152.126.5> source
>     FastEthernet0/0.138
>     >>> !
>     >>> end
>     >>>
>     >>> _______________________________________________
>     >>> cisco-voip mailing list
>     >>> cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     >>> https://puck.nether.net/mailman/listinfo/cisco-voip
>     >>>
>     >>
>     >>
>     >>
>     >
>     >
>
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


From CarlosOrtiz at bayviewfinancial.com  Fri May 25 09:16:57 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 25 May 2007 09:16:57 -0400
Subject: [cisco-voip] IPCC Supervisor Desktop
Message-ID: <OF694EF606.B190F6CB-ON852572E6.0047F531-852572E6.0048F6AC@bftg.com>

We have a problem with the Supervisor Desktop and the Team Skills Stats 
report.   The Column labeled "Oldest in Queue" is supposed to show the 
amount of calls waiting in queue and the wait time of the caller to be 
served next (longest current wait time).   This is only relevant when 
there are no agents ready to take the next call.  In the screenshot below 
you will see that I have 2 agents ready to take calls but the "Oldest in 
Queue" shows 1 caller but no time is incrementing.  This just remains 
there all day inaccurately.  Has anyone seen this or been able to 
determine a cause?



Carlos
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From akoul at dataways.gr  Fri May 25 09:21:57 2007
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Fri, 25 May 2007 16:21:57 +0300
Subject: [cisco-voip] 7970 running SCCP70.8-2-2SR2S resets on CCM
	5.0.4.2000-1
Message-ID: <4656E2F5.6070209@dataways.gr>

Hi, i have one CCM 5.0.4.2000-1  and there is an 7970 which sometimes 
resets several times a day!!
CCM is on a gigabit port of a WS-C2950T-24 and the phones on the same 
switch fast ethernet.
The phone also sometimes gets registered on the SRST which is on the 
same site. it gets unregistered at the same moment and gets registered 
back to the CCM.
It gets power seperately from a power cube (2950 has no POE)
Even if it gets registered on the SRST and back there is no reason for a 
full reset.
Because it has one 7914, it is very noticeable since all lights start 
blinking!

I took a trace from CCM service but nothing came out.
What are most common reasons for reset of a phone?
An other 7970 with 7914 also does the same. The customer has not noticed 
if other phones do the same thing but those 7970 do it for sure.
Could 7914 be the problem?
7914 has S00105000200  image.

Thanks
Anthony

From ealeatherman at gmail.com  Fri May 25 09:34:06 2007
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 25 May 2007 09:34:06 -0400
Subject: [cisco-voip] IPCC Supervisor Desktop
In-Reply-To: <OF694EF606.B190F6CB-ON852572E6.0047F531-852572E6.0048F6AC@bftg.com>
References: <OF694EF606.B190F6CB-ON852572E6.0047F531-852572E6.0048F6AC@bftg.com>
Message-ID: <94a1afde0705250634t289e1cf4y6d633baa37efeef4@mail.gmail.com>

We get this from time to time also Carlos... I'm not sure what causes it.
we're running 4.0.4 with the latest SR. Reseting the CRS engine gets rid of
it at least for us, and I havent seen where it impacts any historical
reports.. mainly its just confusing for the supervisors and is highly
annoying.

On 5/25/07, CarlosOrtiz at bayviewfinancial.com <
CarlosOrtiz at bayviewfinancial.com> wrote:
>
>
> We have a problem with the Supervisor Desktop and the Team Skills Stats
> report.   The Column labeled "Oldest in Queue" is supposed to show the
> amount of calls waiting in queue and the wait time of the caller to be
> served next (longest current wait time).   This is only relevant when there
> are no agents ready to take the next call.  In the screenshot below you will
> see that I have 2 agents ready to take calls but the "Oldest in Queue" shows
> 1 caller but no time is incrementing.  This just remains there all day
> inaccurately.  Has anyone seen this or been able to determine a cause?
>
>
>
> Carlos
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From jgrace at adcapnet.com  Fri May 25 09:44:07 2007
From: jgrace at adcapnet.com (James Grace)
Date: Fri, 25 May 2007 09:44:07 -0400
Subject: [cisco-voip] how to test a t1 cas  what are the cmds
Message-ID: <ECEA9223F645834FB6A65B7CE377B9CD11C667@mail.adcapnetsvc.local>

?
how to test a t1 cas  what are the cmds
Im trying to test some calls going over the t1 cas wink/start connection.  i not sure what cmds i can use
 
James Grace 
 
Systems Engineer
Adcap Network Systems, Inc.
6865 Shiloh Road East, Suite 250
Alpharetta, GA 30005
 
office: 678-456-6986
direct: 678-456-6761
jgrace at adcapnet.com
www.adcapnet.com <http://www.adcapnet.com/> 
Disclaimer For ADCAP Network Systems

"The information transmitted is intended only for the person or entity to which it is addressed and may
contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of,
or taking of any action in reliance upon this information by persons or entities other than the intended
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This original email was sent to the internet for delivery at 25 May 2007 09:44:18 -0400
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From holman at post.harvard.edu  Fri May 25 10:39:00 2007
From: holman at post.harvard.edu (Candace Holman)
Date: Fri, 25 May 2007 10:39:00 -0400
Subject: [cisco-voip] 7970 running SCCP70.8-2-2SR2S resets on CCM
	5.0.4.2000-1
Message-ID: <81746c9a0705250739l57122ef2ne615b519a7e6c6d3@mail.gmail.com>

We saw this happening a number of times and found it was coincident with a
DHCP lease expiration period.  Running Ethereal traces on the phones
confirmed that it reset after a DHCP interaction, and hard coding the IP
address was the workaround.  You could try that while waiting for a solution
or better explanation.

You could also try moving the phone or switch port to another network to see
the behavior is specific to an underprovisioned network.


> Hi, i have one CCM 5.0.4.2000-1  and there is an 7970 which sometimes
> resets several times a day!!
> CCM is on a gigabit port of a WS-C2950T-24 and the phones on the same
> switch fast ethernet.
> The phone also sometimes gets registered on the SRST which is on the
> same site. it gets unregistered at the same moment and gets registered
> back to the CCM.
> It gets power seperately from a power cube (2950 has no POE)
> Even if it gets registered on the SRST and back there is no reason for a
> full reset.
> Because it has one 7914, it is very noticeable since all lights start
> blinking!
>
> I took a trace from CCM service but nothing came out.
> What are most common reasons for reset of a phone?
> An other 7970 with 7914 also does the same. The customer has not noticed
> if other phones do the same thing but those 7970 do it for sure.
> Could 7914 be the problem?
> 7914 has S00105000200  image.
>
> Thanks
> Anthony
>
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From Nick.Kassel at Charles-Stanley.co.uk  Fri May 25 10:38:57 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri, 25 May 2007 15:38:57 +0100
Subject: [cisco-voip] Upgrade to 5.1(2)
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451F11@CSEXCH01.charles-stanley.co.uk>

Has anyone upgraded to 5.1(2) yet, just wondered if anyone had any
issues with it.

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From CarlosOrtiz at bayviewfinancial.com  Fri May 25 10:52:30 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Fri, 25 May 2007 10:52:30 -0400
Subject: [cisco-voip] IPCC Supervisor Desktop
In-Reply-To: <9015A93106739144995D6FD24A0F7A5608B5DFCC@hydrogen.mercurypay.com>
Message-ID: <OFF46B20E8.F533FDC1-ON852572E6.00510D33-852572E6.0051B677@bftg.com>

I can actually create a "stuck call" by conferencing 2 agents OR doing an 
agent-> non agent> agent transfer that is seen in the Real Time reports 
but I am not able to create one that is seen in the Supervisor Desktop. 
The users are executing some sequence that causes it to occur that I 
haven't discovered. Can you answer these questions for me?

1) Were your phantom calls actually seen in the Supervisor Desktop?
2) How long have you been running 4.0.5?
3)  Any other issues present themselves or was it a smooth transition from 
whatever you upgraded from?
4) Did you work a TAC case with this issue and was this their solution?





"Jerri Robbins" <jrobbins at mercurypay.com> 
05/25/2007 10:10 AM

To
<CarlosOrtiz at bayviewfinancial.com>
cc

Subject
RE: [cisco-voip] IPCC Supervisor Desktop






We used to get this problem quite often.  It is a phantom call stuck in 
the queue.  The only way to get rid of it is to restart the CSQ service. 
We upgraded to 4.0.5 and I haven?t seen this at all since then.  It is a 
known bug that you can find in the bug tool on the Cisco site - 
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCsd79734 
 
The upgrade seems to have fixed this error.
 
Jerri
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: Friday, May 25, 2007 7:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC Supervisor Desktop
 

We have a problem with the Supervisor Desktop and the Team Skills Stats 
report.   The Column labeled "Oldest in Queue" is supposed to show the 
amount of calls waiting in queue and the wait time of the caller to be 
served next (longest current wait time).   This is only relevant when 
there are no agents ready to take the next call.  In the screenshot below 
you will see that I have 2 agents ready to take calls but the "Oldest in 
Queue" shows 1 caller but no time is incrementing.  This just remains 
there all day inaccurately.  Has anyone seen this or been able to 
determine a cause? 



Carlos

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From Scott.Voll at wesd.org  Fri May 25 11:14:23 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 25 May 2007 08:14:23 -0700
Subject: [cisco-voip] Unity Language Install?
In-Reply-To: <001201c79e4a$1061c580$31255080$@com>
References: <001201c79e4a$1061c580$31255080$@com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7FA@Apollo.wesd.org>

Do you have the license for more then one language?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jimmy Pale
Sent: Thursday, May 24, 2007 2:26 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Language Install?

 

I am trying install an additional language on Unity. Using the CDs
downloaded from CCO I only get the option to install English though I
have downloaded the CD for the other language I need to install as well.
English United States is the only option during the install with the
downloaded CDs. I have not received the DVD media yet. Can anyone help
me with what I am missing here?

 

Thanks!

 

 

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From Scott.Voll at wesd.org  Fri May 25 11:16:02 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 25 May 2007 08:16:02 -0700
Subject: [cisco-voip] Hunt Groups and DND
In-Reply-To: <f1b65f880705241549o7176707dnf11f039b3daf82e1@mail.gmail.com>
References: <a2b352c40705241539r28305d68h98ec90615ddd4eee@mail.gmail.com>
	<f1b65f880705241549o7176707dnf11f039b3daf82e1@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE7FB@Apollo.wesd.org>

IPCCx.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Thursday, May 24, 2007 3:49 PM
To: Nick Griffin
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Hunt Groups and DND

This exact feature is in 4.2 or 6.0.

You can hack it in 4.1 or 5.1 by using extension mobility and have
their user device profile add a second line that is the hunt group
member.

Others may have different approaches...



On 5/24/07, Nick Griffin <nick.jon.griffin at gmail.com> wrote:
> I have a 5.1 install coming up in the near future. On the customers
existing
> system they currently use sequential hunt groups in combination with
DND.
> Their goal is to route customer calls to reps ASAP, the do not disturb
> feature keeps the customers wait short by not routing the call to the
hunt
> group member while that member is in DND mode. Does anyone know of a
similar
> solution that could be accomplish on CCM5.1? I'm thinking Extension
Mobility
> with 1 button login, but that seems to take to long with the reset.
They
> essentially need to be able to "log out" of a hunt group. Looking for
any
> suggestions.
>
>  Regards,
>
> Nick Griffin
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From lelio at uoguelph.ca  Fri May 25 11:43:02 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 11:43:02 -0400
Subject: [cisco-voip] using "reject" rule to reject outbound called numbers
Message-ID: <016001c79ee3$60f65670$47196883@cfs.uoguelph.ca>

I'm trying to use the reject rule to block outbound called numbers but it's not working. The debugs actually say it's skipping the block rule.

Anyone know what's going on?



------------------------------------------------------------------------------
  config:

  !
  voice translation-profile 999
   translate called 1112
  !
  voice translation-rule 1112
   rule 1 reject /914165551212/
  !
  dial-peer voice 99901 pots
   translation-profile outgoing 999 ***tried incoming as well, it failed***
   destination-pattern 914165551212
   port 0/1/0:23
   forward-digits 11
  !




------------------------------------------------------------------------------
  debug:

    *May 25 11:33:08: //-1/xxxxxxxxxxxx/RXRULE/regxrule_get_profile_from_trunkgroup_internal: Voice port 0x673C1D8C does not belong to a
    ny trunk group
    *May 25 11:33:08: //-1/12C459A180E7/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x662CD204; count=1
    *May 25 11:33:08: //-1/12C459A180E7/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x662CD204; count=0
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_stack_push_RegXruleNumInfo: stack=0x662CD204; count=1
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal: number=61001 type=unknown plan=unknown numbertype=ca
    lling
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_match_internal: Error: ruleset for calling number not found
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal: No match: number=61001 type=unknown plan=unknown
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal: number=914165551212 type=unknown plan=unknown number
    type=called
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_match: Skipping a call block rule; number=914165551212 rule precedence=1
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_match_internal: No match found
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal: No match: number=914165551212 type=unknown plan=unkn
    own
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirec
    t-target
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-target number not found
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal: number= type=unknown plan=unknown numbertype=redirec
    t-called
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_get_RegXrule: Invalid translation ruleset tag=0
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_match_internal: Error: ruleset for redirect-called number not found
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal: No match: number= type=unknown plan=unknown
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_dp_translate: calling_number=61001 calling_octet=0x0
            called_number=914165551212 called_octet=0x80
            redirect_number= redirect_type=0 redirect_plan=0        redirect_PI=0 redirect_SI=0
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_vp_translate: No profile found in voice port or trunk group for outgoing directi
    on
    *May 25 11:33:10: //-1/12C459A180E7/RXRULE/regxrule_vp_translate: calling_number=61001 calling_octet=0x0
            called_number=914165551212 called_octet=0x80
            redirect_number= redirect_type=0 redirect_plan=0
    *May 25 11:33:11: %ISDN-6-CONNECT: Interface Serial0/1/0:22 is now connected to 14165551212 N/A
    *May 25 11:33:13: //-1/12C459A180E7/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x662CD204; count=2
    *May 25 11:33:13: %ISDN-6-CONNECT: Interface Serial0/1/0:22 is now connected to 14165551212 N/A
    *May 25 11:33:13: %ISDN-6-CONNECT: Interface Serial0/1/0:22 is now connected to 14165551212 N/A
    *May 25 11:33:13: %ISDN-6-DISCONNECT: Interface Serial0/1/0:22  disconnected from 14165551212 , call lasted 1 seconds
    *May 25 11:33:13: %ISDN-6-CONNECT: Interface Serial0/1/0:22 is now connected to 14165551212 N/A
    *May 25 11:33:13: %ISDN-6-DISCONNECT: Interface Serial0/1/0:22  disconnected from unknown , call lasted 1 seconds

------------------------------------------------------------------------------


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR
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From asobihoudai at yahoo.com  Fri May 25 11:59:26 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 25 May 2007 08:59:26 -0700 (PDT)
Subject: [cisco-voip] how to test a t1 cas  what are the cmds
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C667@mail.adcapnetsvc.local>
Message-ID: <416815.32540.qm@web57215.mail.re3.yahoo.com>

If you're just trying to test phone calls over your
CAS T1, just like any other connection you can try to
use 'csim start <phonenumber>' to see if you can call
yourself.

Cisco recommends you don't use that command willynilly
since it has been known to crash the box so it is
probably best if you are physically located near it.

Paul

--- James Grace <jgrace at adcapnet.com> wrote:

> ?
> how to test a t1 cas  what are the cmds
> Im trying to test some calls going over the t1 cas
> wink/start connection.  i not sure what cmds i can
> use
>  
> James Grace 
>  
> Systems Engineer
> Adcap Network Systems, Inc.
> 6865 Shiloh Road East, Suite 250
> Alpharetta, GA 30005
>  
> office: 678-456-6986
> direct: 678-456-6761
> jgrace at adcapnet.com
> www.adcapnet.com <http://www.adcapnet.com/> 
> Disclaimer For ADCAP Network Systems
> 
> "The information transmitted is intended only for
> the person or entity to which it is addressed and
> may
> contain confidential and/or privileged material. Any
> review, retransmission, dissemination or other use
> of,
> or taking of any action in reliance upon this
> information by persons or entities other than the
> intended
> recipient is prohibited. If you received this
> message in error, please contact the sender and
> delete the
> material from all computers."
> 
> This original email was sent to the internet for
> delivery at 25 May 2007 09:44:18 -0400
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



 
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From asobihoudai at yahoo.com  Fri May 25 12:05:21 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 25 May 2007 09:05:21 -0700 (PDT)
Subject: [cisco-voip] 7970 running SCCP70.8-2-2SR2S resets on CCM
	5.0.4.2000-1
In-Reply-To: <81746c9a0705250739l57122ef2ne615b519a7e6c6d3@mail.gmail.com>
Message-ID: <579986.67052.qm@web57213.mail.re3.yahoo.com>

Candace,

Interesting. I'll have to keep your fix in mind the
next time I run into this issue since I just upgraded
the customer's phone's firmware and the issue was
resolved.

Paul


--- Candace Holman <holman at post.harvard.edu> wrote:

> We saw this happening a number of times and found it
> was coincident with a
> DHCP lease expiration period.  Running Ethereal
> traces on the phones
> confirmed that it reset after a DHCP interaction,
> and hard coding the IP
> address was the workaround.  You could try that
> while waiting for a solution
> or better explanation.
> 
> You could also try moving the phone or switch port
> to another network to see
> the behavior is specific to an underprovisioned
> network.
> 
> 
> > Hi, i have one CCM 5.0.4.2000-1  and there is an
> 7970 which sometimes
> > resets several times a day!!
> > CCM is on a gigabit port of a WS-C2950T-24 and the
> phones on the same
> > switch fast ethernet.
> > The phone also sometimes gets registered on the
> SRST which is on the
> > same site. it gets unregistered at the same moment
> and gets registered
> > back to the CCM.
> > It gets power seperately from a power cube (2950
> has no POE)
> > Even if it gets registered on the SRST and back
> there is no reason for a
> > full reset.
> > Because it has one 7914, it is very noticeable
> since all lights start
> > blinking!
> >
> > I took a trace from CCM service but nothing came
> out.
> > What are most common reasons for reset of a phone?
> > An other 7970 with 7914 also does the same. The
> customer has not noticed
> > if other phones do the same thing but those 7970
> do it for sure.
> > Could 7914 be the problem?
> > 7914 has S00105000200  image.
> >
> > Thanks
> > Anthony
> >
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
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From asobihoudai at yahoo.com  Fri May 25 12:07:26 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 25 May 2007 09:07:26 -0700 (PDT)
Subject: [cisco-voip] using "reject" rule to reject outbound called
	numbers
In-Reply-To: <016001c79ee3$60f65670$47196883@cfs.uoguelph.ca>
Message-ID: <88464.29286.qm@web57203.mail.re3.yahoo.com>

What does test voice translation rule tell you? Does
it say your translation rule works according to plan?

Paul

--- Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

> I'm trying to use the reject rule to block outbound
> called numbers but it's not working. The debugs
> actually say it's skipping the block rule.
> 
> Anyone know what's going on?
> 
> 
> 
>
------------------------------------------------------------------------------
>   config:
> 
>   !
>   voice translation-profile 999
>    translate called 1112
>   !
>   voice translation-rule 1112
>    rule 1 reject /914165551212/
>   !
>   dial-peer voice 99901 pots
>    translation-profile outgoing 999 ***tried
> incoming as well, it failed***
>    destination-pattern 914165551212
>    port 0/1/0:23
>    forward-digits 11
>   !
> 
> 
> 
> 
>
------------------------------------------------------------------------------
>   debug:
> 
>     *May 25 11:33:08:
>
//-1/xxxxxxxxxxxx/RXRULE/regxrule_get_profile_from_trunkgroup_internal:
> Voice port 0x673C1D8C does not belong to a
>     ny trunk group
>     *May 25 11:33:08:
>
//-1/12C459A180E7/RXRULE/regxrule_stack_pop_RegXruleNumInfo:
> stack=0x662CD204; count=1
>     *May 25 11:33:08:
>
//-1/12C459A180E7/RXRULE/regxrule_stack_push_RegXruleNumInfo:
> stack=0x662CD204; count=0
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_stack_push_RegXruleNumInfo:
> stack=0x662CD204; count=1
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
> number=61001 type=unknown plan=unknown numbertype=ca
>     lling
>     *May 25 11:33:10:
> //-1/12C459A180E7/RXRULE/regxrule_get_RegXrule:
> Invalid translation ruleset tag=0
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_match_internal:
> Error: ruleset for calling number not found
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
> No match: number=61001 type=unknown plan=unknown
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
> number=914165551212 type=unknown plan=unknown number
>     type=called
>     *May 25 11:33:10:
> //-1/12C459A180E7/RXRULE/regxrule_match: Skipping a
> call block rule; number=914165551212 rule
> precedence=1
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_match_internal:
> No match found
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
> No match: number=914165551212 type=unknown plan=unkn
>     own
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
> number= type=unknown plan=unknown numbertype=redirec
>     t-target
>     *May 25 11:33:10:
> //-1/12C459A180E7/RXRULE/regxrule_get_RegXrule:
> Invalid translation ruleset tag=0
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_match_internal:
> Error: ruleset for redirect-target number not found
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
> No match: number= type=unknown plan=unknown
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
> number= type=unknown plan=unknown numbertype=redirec
>     t-called
>     *May 25 11:33:10:
> //-1/12C459A180E7/RXRULE/regxrule_get_RegXrule:
> Invalid translation ruleset tag=0
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_match_internal:
> Error: ruleset for redirect-called number not found
>     *May 25 11:33:10:
>
//-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
> No match: number= type=unknown plan=unknown
>     *May 25 11:33:10:
> //-1/12C459A180E7/RXRULE/regxrule_dp_translate:
> calling_number=61001 calling_octet=0x0
>             called_number=914165551212
> called_octet=0x80
>             redirect_number= redirect_type=0
> redirect_plan=0        redirect_PI=0 redirect_SI=0
>     *May 25 11:33:10:
> //-1/12C459A180E7/RXRULE/regxrule_vp_translate: No
> profile found in voice port or trunk group for
> outgoing directi
>     on
>     *May 25 11:33:10:
> //-1/12C459A180E7/RXRULE/regxrule_vp_translate:
> calling_number=61001 calling_octet=0x0
>             called_number=914165551212
> called_octet=0x80
>             redirect_number= redirect_type=0
> redirect_plan=0
>     *May 25 11:33:11: %ISDN-6-CONNECT: Interface
> Serial0/1/0:22 is now connected to 14165551212 N/A
>     *May 25 11:33:13:
>
//-1/12C459A180E7/RXRULE/regxrule_stack_pop_RegXruleNumInfo:
> stack=0x662CD204; count=2
>     *May 25 11:33:13: %ISDN-6-CONNECT: Interface
> Serial0/1/0:22 is now connected to 14165551212 N/A
>     *May 25 11:33:13: %ISDN-6-CONNECT: Interface
> Serial0/1/0:22 is now connected to 14165551212 N/A
>     *May 25 11:33:13: %ISDN-6-DISCONNECT: Interface
> Serial0/1/0:22  disconnected from 14165551212 , call
> lasted 1 seconds
>     *May 25 11:33:13: %ISDN-6-CONNECT: Interface
> Serial0/1/0:22 is now connected to 14165551212 N/A
>     *May 25 11:33:13: %ISDN-6-DISCONNECT: Interface
> Serial0/1/0:22  disconnected from unknown , call
> lasted 1 seconds
> 
>
------------------------------------------------------------------------------
> 
> 
>
--------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph *
> Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> 
> "Buffalo buffalo Buffalo buffalo buffalo buffalo
> Buffalo buffalo."      WJR
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
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http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  

From asobihoudai at yahoo.com  Fri May 25 11:57:21 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 25 May 2007 08:57:21 -0700 (PDT)
Subject: [cisco-voip] 7970 running SCCP70.8-2-2SR2S resets on CCM
	5.0.4.2000-1
In-Reply-To: <4656E2F5.6070209@dataways.gr>
Message-ID: <153555.75653.qm@web57206.mail.re3.yahoo.com>

Anthony,

Hi, I had a customer with a 7970 that would drop into
SRST mode several times a day when all other phones
would remain registered to the CCM. I upgraded its
firmware and the problem went away.

HTH,
Paul

--- Anthony Kouloglou <akoul at dataways.gr> wrote:

> Hi, i have one CCM 5.0.4.2000-1  and there is an
> 7970 which sometimes 
> resets several times a day!!
> CCM is on a gigabit port of a WS-C2950T-24 and the
> phones on the same 
> switch fast ethernet.
> The phone also sometimes gets registered on the SRST
> which is on the 
> same site. it gets unregistered at the same moment
> and gets registered 
> back to the CCM.
> It gets power seperately from a power cube (2950 has
> no POE)
> Even if it gets registered on the SRST and back
> there is no reason for a 
> full reset.
> Because it has one 7914, it is very noticeable since
> all lights start 
> blinking!
> 
> I took a trace from CCM service but nothing came
> out.
> What are most common reasons for reset of a phone?
> An other 7970 with 7914 also does the same. The
> customer has not noticed 
> if other phones do the same thing but those 7970 do
> it for sure.
> Could 7914 be the problem?
> 7914 has S00105000200  image.
> 
> Thanks
> Anthony
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



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From zohaibshabir at gmail.com  Fri May 25 13:04:03 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Fri, 25 May 2007 22:04:03 +0500
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
Message-ID: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>

Dear All,
Can we use Catalyst 6500 series switch as Voice gateway.

-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and PreSales)
DWP Group, TECH Division
Ph:+92-302-8232689
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From holman at post.harvard.edu  Fri May 25 13:12:24 2007
From: holman at post.harvard.edu (Candace Holman)
Date: Fri, 25 May 2007 13:12:24 -0400
Subject: [cisco-voip] 7970 running SCCP70.8-2-2SR2S resets on CCM
	5.0.4.2000-1
In-Reply-To: <579986.67052.qm@web57213.mail.re3.yahoo.com>
References: <81746c9a0705250739l57122ef2ne615b519a7e6c6d3@mail.gmail.com>
	<579986.67052.qm@web57213.mail.re3.yahoo.com>
Message-ID: <81746c9a0705251012q5961ddf9k28c136f50b246016@mail.gmail.com>

Even better! (although loading a new firmware image the same week it was
posted can give you mixed results....)

Was there a bug id?

On 5/25/07, Paul Choi <asobihoudai at yahoo.com> wrote:
>
> Candace,
>
> Interesting. I'll have to keep your fix in mind the
> next time I run into this issue since I just upgraded
> the customer's phone's firmware and the issue was
> resolved.
>
> Paul
>
>
> --- Candace Holman <holman at post.harvard.edu> wrote:
>
> > We saw this happening a number of times and found it
> > was coincident with a
> > DHCP lease expiration period.  Running Ethereal
> > traces on the phones
> > confirmed that it reset after a DHCP interaction,
> > and hard coding the IP
> > address was the workaround.  You could try that
> > while waiting for a solution
> > or better explanation.
> >
> > You could also try moving the phone or switch port
> > to another network to see
> > the behavior is specific to an underprovisioned
> > network.
> >
> >
> > > Hi, i have one CCM 5.0.4.2000-1  and there is an
> > 7970 which sometimes
> > > resets several times a day!!
> > > CCM is on a gigabit port of a WS-C2950T-24 and the
> > phones on the same
> > > switch fast ethernet.
> > > The phone also sometimes gets registered on the
> > SRST which is on the
> > > same site. it gets unregistered at the same moment
> > and gets registered
> > > back to the CCM.
> > > It gets power seperately from a power cube (2950
> > has no POE)
> > > Even if it gets registered on the SRST and back
> > there is no reason for a
> > > full reset.
> > > Because it has one 7914, it is very noticeable
> > since all lights start
> > > blinking!
> > >
> > > I took a trace from CCM service but nothing came
> > out.
> > > What are most common reasons for reset of a phone?
> > > An other 7970 with 7914 also does the same. The
> > customer has not noticed
> > > if other phones do the same thing but those 7970
> > do it for sure.
> > > Could 7914 be the problem?
> > > 7914 has S00105000200  image.
> > >
> > > Thanks
> > > Anthony
> > >
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
>
> ____________________________________________________________________________________Choose
> the right car based on your needs.  Check out Yahoo! Autos new Car Finder
> tool.
> http://autos.yahoo.com/carfinder/
>
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From jonvoip at gmail.com  Fri May 25 13:15:17 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 25 May 2007 12:15:17 -0500
Subject: [cisco-voip] CCM 5.1 MOH and Conferencing stopped working
Message-ID: <5d093f9a0705251015s5b52c158t21cffaa8bbdfabcc@mail.gmail.com>

Got a three-server cluster, MOH and Conferencing used to work, and
they stopped suddenly.

(for everyone).

IP Voice Media Streaming app is running... this was working yesterday...


Any ideas?



Jonathan

From lelio at uoguelph.ca  Fri May 25 13:17:34 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 13:17:34 -0400
Subject: [cisco-voip] using "reject" rule to reject outbound called
	numbers
References: <88464.29286.qm@web57203.mail.re3.yahoo.com>
Message-ID: <001101c79ef0$95926f10$47196883@cfs.uoguelph.ca>

sorry, forgot to include that.....test says it works great.

vgw-kemptville#test voice translation-rule 1112 914165551212
914165551212 blocked on rule 1

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: Paul Choi 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Friday, May 25, 2007 12:07 PM
  Subject: Re: [cisco-voip] using "reject" rule to reject outbound called numbers


  What does test voice translation rule tell you? Does
  it say your translation rule works according to plan?

  Paul

  --- Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

  > I'm trying to use the reject rule to block outbound
  > called numbers but it's not working. The debugs
  > actually say it's skipping the block rule.
  > 
  > Anyone know what's going on?
  > 
  > 
  > 
  >
  ------------------------------------------------------------------------------
  >   config:
  > 
  >   !
  >   voice translation-profile 999
  >    translate called 1112
  >   !
  >   voice translation-rule 1112
  >    rule 1 reject /914165551212/
  >   !
  >   dial-peer voice 99901 pots
  >    translation-profile outgoing 999 ***tried
  > incoming as well, it failed***
  >    destination-pattern 914165551212
  >    port 0/1/0:23
  >    forward-digits 11
  >   !
  > 
  > 
  > 
  > 
  >
  ------------------------------------------------------------------------------
  >   debug:
  > 
  >     *May 25 11:33:08:
  >
  //-1/xxxxxxxxxxxx/RXRULE/regxrule_get_profile_from_trunkgroup_internal:
  > Voice port 0x673C1D8C does not belong to a
  >     ny trunk group
  >     *May 25 11:33:08:
  >
  //-1/12C459A180E7/RXRULE/regxrule_stack_pop_RegXruleNumInfo:
  > stack=0x662CD204; count=1
  >     *May 25 11:33:08:
  >
  //-1/12C459A180E7/RXRULE/regxrule_stack_push_RegXruleNumInfo:
  > stack=0x662CD204; count=0
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_stack_push_RegXruleNumInfo:
  > stack=0x662CD204; count=1
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
  > number=61001 type=unknown plan=unknown numbertype=ca
  >     lling
  >     *May 25 11:33:10:
  > //-1/12C459A180E7/RXRULE/regxrule_get_RegXrule:
  > Invalid translation ruleset tag=0
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_match_internal:
  > Error: ruleset for calling number not found
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
  > No match: number=61001 type=unknown plan=unknown
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
  > number=914165551212 type=unknown plan=unknown number
  >     type=called
  >     *May 25 11:33:10:
  > //-1/12C459A180E7/RXRULE/regxrule_match: Skipping a
  > call block rule; number=914165551212 rule
  > precedence=1
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_match_internal:
  > No match found
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
  > No match: number=914165551212 type=unknown plan=unkn
  >     own
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
  > number= type=unknown plan=unknown numbertype=redirec
  >     t-target
  >     *May 25 11:33:10:
  > //-1/12C459A180E7/RXRULE/regxrule_get_RegXrule:
  > Invalid translation ruleset tag=0
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_match_internal:
  > Error: ruleset for redirect-target number not found
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
  > No match: number= type=unknown plan=unknown
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
  > number= type=unknown plan=unknown numbertype=redirec
  >     t-called
  >     *May 25 11:33:10:
  > //-1/12C459A180E7/RXRULE/regxrule_get_RegXrule:
  > Invalid translation ruleset tag=0
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_match_internal:
  > Error: ruleset for redirect-called number not found
  >     *May 25 11:33:10:
  >
  //-1/12C459A180E7/RXRULE/regxrule_profile_translate_internal:
  > No match: number= type=unknown plan=unknown
  >     *May 25 11:33:10:
  > //-1/12C459A180E7/RXRULE/regxrule_dp_translate:
  > calling_number=61001 calling_octet=0x0
  >             called_number=914165551212
  > called_octet=0x80
  >             redirect_number= redirect_type=0
  > redirect_plan=0        redirect_PI=0 redirect_SI=0
  >     *May 25 11:33:10:
  > //-1/12C459A180E7/RXRULE/regxrule_vp_translate: No
  > profile found in voice port or trunk group for
  > outgoing directi
  >     on
  >     *May 25 11:33:10:
  > //-1/12C459A180E7/RXRULE/regxrule_vp_translate:
  > calling_number=61001 calling_octet=0x0
  >             called_number=914165551212
  > called_octet=0x80
  >             redirect_number= redirect_type=0
  > redirect_plan=0
  >     *May 25 11:33:11: %ISDN-6-CONNECT: Interface
  > Serial0/1/0:22 is now connected to 14165551212 N/A
  >     *May 25 11:33:13:
  >
  //-1/12C459A180E7/RXRULE/regxrule_stack_pop_RegXruleNumInfo:
  > stack=0x662CD204; count=2
  >     *May 25 11:33:13: %ISDN-6-CONNECT: Interface
  > Serial0/1/0:22 is now connected to 14165551212 N/A
  >     *May 25 11:33:13: %ISDN-6-CONNECT: Interface
  > Serial0/1/0:22 is now connected to 14165551212 N/A
  >     *May 25 11:33:13: %ISDN-6-DISCONNECT: Interface
  > Serial0/1/0:22  disconnected from 14165551212 , call
  > lasted 1 seconds
  >     *May 25 11:33:13: %ISDN-6-CONNECT: Interface
  > Serial0/1/0:22 is now connected to 14165551212 N/A
  >     *May 25 11:33:13: %ISDN-6-DISCONNECT: Interface
  > Serial0/1/0:22  disconnected from unknown , call
  > lasted 1 seconds
  > 
  >
  ------------------------------------------------------------------------------
  > 
  > 
  >
  --------------------------------------------------------------------------------
  > Lelio Fulgenzi, B.A.
  > Senior Analyst (CCS) * University of Guelph *
  > Guelph, Ontario N1G 2W1
  > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  > 
  > "Buffalo buffalo Buffalo buffalo buffalo buffalo
  > Buffalo buffalo."      WJR
  > > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  > 



         
  ____________________________________________________________________________________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.
  http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  
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From lelio at uoguelph.ca  Fri May 25 13:19:39 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 13:19:39 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
Message-ID: <003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca>

sure, you can use the CMM modules with FXO or PRI ports.

take a serious look at comparing the cost of the blades with an ISR though.

you'll be surprised as to how much cheaper it can be.

and you'll get to use the ISR for all sorts of other stuff like SRST. granted you can use the CMM for SRST but it is limited in memory.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: zohaib shabir 
  To: cisco-voip at puck.nether.net ; cisco-voip-request at puck.nether.net 
  Sent: Friday, May 25, 2007 1:04 PM
  Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway


  Dear All,
  Can we use Catalyst 6500 series switch as Voice gateway.

  -- 
  Regards,
  Zohaib Shabir
  Network Engineer(Voice and PreSales)
  DWP Group, TECH Division
  Ph:+92-302-8232689 


------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From asobihoudai at yahoo.com  Fri May 25 13:22:13 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 25 May 2007 10:22:13 -0700 (PDT)
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
Message-ID: <756482.68406.qm@web57209.mail.re3.yahoo.com>

Yes. Just put a fat CCM blade in the box and you can
use it as a VoIP interface.

Paul

--- zohaib shabir <zohaibshabir at gmail.com> wrote:

> Dear All,
> Can we use Catalyst 6500 series switch as Voice
> gateway.
> 
> -- 
> Regards,
> Zohaib Shabir
> Network Engineer(Voice and PreSales)
> DWP Group, TECH Division
> Ph:+92-302-8232689
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________Got a little couch potato? 
Check out fun summer activities for kids.
http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+kids&cs=bz 

From asobihoudai at yahoo.com  Fri May 25 13:23:34 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 25 May 2007 10:23:34 -0700 (PDT)
Subject: [cisco-voip] 7970 running SCCP70.8-2-2SR2S resets on CCM
	5.0.4.2000-1
In-Reply-To: <81746c9a0705251012q5961ddf9k28c136f50b246016@mail.gmail.com>
Message-ID: <177084.99201.qm@web57210.mail.re3.yahoo.com>

There wasn't a bug ID. The client was using CCM 3.3
and  the firmware was quite dated so I just went off
of a colleague's recommendations to update the
firmware.

HTH!
Paul

--- Candace Holman <holman at post.harvard.edu> wrote:

> Even better! (although loading a new firmware image
> the same week it was
> posted can give you mixed results....)
> 
> Was there a bug id?
> 
> On 5/25/07, Paul Choi <asobihoudai at yahoo.com> wrote:
> >
> > Candace,
> >
> > Interesting. I'll have to keep your fix in mind
> the
> > next time I run into this issue since I just
> upgraded
> > the customer's phone's firmware and the issue was
> > resolved.
> >
> > Paul
> >
> >
> > --- Candace Holman <holman at post.harvard.edu>
> wrote:
> >
> > > We saw this happening a number of times and
> found it
> > > was coincident with a
> > > DHCP lease expiration period.  Running Ethereal
> > > traces on the phones
> > > confirmed that it reset after a DHCP
> interaction,
> > > and hard coding the IP
> > > address was the workaround.  You could try that
> > > while waiting for a solution
> > > or better explanation.
> > >
> > > You could also try moving the phone or switch
> port
> > > to another network to see
> > > the behavior is specific to an underprovisioned
> > > network.
> > >
> > >
> > > > Hi, i have one CCM 5.0.4.2000-1  and there is
> an
> > > 7970 which sometimes
> > > > resets several times a day!!
> > > > CCM is on a gigabit port of a WS-C2950T-24 and
> the
> > > phones on the same
> > > > switch fast ethernet.
> > > > The phone also sometimes gets registered on
> the
> > > SRST which is on the
> > > > same site. it gets unregistered at the same
> moment
> > > and gets registered
> > > > back to the CCM.
> > > > It gets power seperately from a power cube
> (2950
> > > has no POE)
> > > > Even if it gets registered on the SRST and
> back
> > > there is no reason for a
> > > > full reset.
> > > > Because it has one 7914, it is very noticeable
> > > since all lights start
> > > > blinking!
> > > >
> > > > I took a trace from CCM service but nothing
> came
> > > out.
> > > > What are most common reasons for reset of a
> phone?
> > > > An other 7970 with 7914 also does the same.
> The
> > > customer has not noticed
> > > > if other phones do the same thing but those
> 7970
> > > do it for sure.
> > > > Could 7914 be the problem?
> > > > 7914 has S00105000200  image.
> > > >
> > > > Thanks
> > > > Anthony
> > > >
> > > >
> _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> >
> >
> >
>
____________________________________________________________________________________Choose
> > the right car based on your needs.  Check out
> Yahoo! Autos new Car Finder
> > tool.
> > http://autos.yahoo.com/carfinder/
> >
> 



 
____________________________________________________________________________________
8:00? 8:25? 8:40? Find a flick in no time 
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news

From Scott.Voll at wesd.org  Fri May 25 13:35:58 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 25 May 2007 10:35:58 -0700
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <756482.68406.qm@web57209.mail.re3.yahoo.com>
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
	<756482.68406.qm@web57209.mail.re3.yahoo.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE801@Apollo.wesd.org>

How many PRI's are you looking for?  An ISR might be cheaper.

Note. I do have a CMM with 6 PRI and ACT card.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Friday, May 25, 2007 10:22 AM
To: zohaib shabir; cisco-voip at puck.nether.net;
cisco-voip-request at puck.nether.net
Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway

Yes. Just put a fat CCM blade in the box and you can
use it as a VoIP interface.

Paul

--- zohaib shabir <zohaibshabir at gmail.com> wrote:

> Dear All,
> Can we use Catalyst 6500 series switch as Voice
> gateway.
> 
> -- 
> Regards,
> Zohaib Shabir
> Network Engineer(Voice and PreSales)
> DWP Group, TECH Division
> Ph:+92-302-8232689
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
________________________________________________________________________
____________Got a little couch potato? 
Check out fun summer activities for kids.
http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+ki
ds&cs=bz 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From zohaibshabir at gmail.com  Fri May 25 13:33:07 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Fri, 25 May 2007 22:33:07 +0500
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca>
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
	<003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca>
Message-ID: <1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com>

Thanks for the reply,
Problem for me is my client wants 60 FXO ports per router i cannot do even
through 3845. Please advise me which way to go

On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  sure, you can use the CMM modules with FXO or PRI ports.
>
> take a serious look at comparing the cost of the blades with an ISR
> though.
>
> you'll be surprised as to how much cheaper it can be.
>
> and you'll get to use the ISR for all sorts of other stuff like SRST.
> granted you can use the CMM for SRST but it is limited in memory.
>
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."
> WJR
>
> ----- Original Message -----
> *From:* zohaib shabir <zohaibshabir at gmail.com>
> *To:* cisco-voip at puck.nether.net ; cisco-voip-request at puck.nether.net
> *Sent:* Friday, May 25, 2007 1:04 PM
> *Subject:* [cisco-voip] Catalyst 6500 switch as Voice Gateway
>
> Dear All,
> Can we use Catalyst 6500 series switch as Voice gateway.
>
> --
> Regards,
> Zohaib Shabir
> Network Engineer(Voice and PreSales)
> DWP Group, TECH Division
> Ph:+92-302-8232689
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and PreSales)
DWP Group, TECH Division
Ph:+92-302-8232689
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From zohaibshabir at gmail.com  Fri May 25 13:41:16 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Fri, 25 May 2007 22:41:16 +0500
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE801@Apollo.wesd.org>
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
	<756482.68406.qm@web57209.mail.re3.yahoo.com>
	<407055A92CECCB499C922A2D35FC19A608DEE801@Apollo.wesd.org>
Message-ID: <1ff12970705251041x62f3943dl8aa6c0a18cff08be@mail.gmail.com>

Well, I have to terminate 60 FXO ports. Please help if there is any platform
in which i can terminate 60 FXO ports.

On 5/25/07, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
> How many PRI's are you looking for?  An ISR might be cheaper.
>
> Note. I do have a CMM with 6 PRI and ACT card.
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
> Sent: Friday, May 25, 2007 10:22 AM
> To: zohaib shabir; cisco-voip at puck.nether.net;
> cisco-voip-request at puck.nether.net
> Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway
>
> Yes. Just put a fat CCM blade in the box and you can
> use it as a VoIP interface.
>
> Paul
>
> --- zohaib shabir <zohaibshabir at gmail.com> wrote:
>
> > Dear All,
> > Can we use Catalyst 6500 series switch as Voice
> > gateway.
> >
> > --
> > Regards,
> > Zohaib Shabir
> > Network Engineer(Voice and PreSales)
> > DWP Group, TECH Division
> > Ph:+92-302-8232689
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
>
> ________________________________________________________________________
> ____________Got a little couch potato?
> Check out fun summer activities for kids.
> http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+ki
> ds&cs=bz
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and PreSales)
DWP Group, TECH Division
Ph:+92-302-8232689
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From Scott.Voll at wesd.org  Fri May 25 13:44:52 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 25 May 2007 10:44:52 -0700
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <1ff12970705251041x62f3943dl8aa6c0a18cff08be@mail.gmail.com>
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
	<756482.68406.qm@web57209.mail.re3.yahoo.com>
	<407055A92CECCB499C922A2D35FC19A608DEE801@Apollo.wesd.org>
	<1ff12970705251041x62f3943dl8aa6c0a18cff08be@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE802@Apollo.wesd.org>

Could you do it with a couple VG248's?  or are they FXS?

 

________________________________

From: zohaib shabir [mailto:zohaibshabir at gmail.com] 
Sent: Friday, May 25, 2007 10:41 AM
To: Voll, Scott
Cc: Paul Choi; cisco-voip at puck.nether.net;
cisco-voip-request at puck.nether.net
Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway

 

Well, I have to terminate 60 FXO ports. Please help if there is any
platform in which i can terminate 60 FXO ports.

On 5/25/07, Voll, Scott < Scott.Voll at wesd.org
<mailto:Scott.Voll at wesd.org> > wrote:

How many PRI's are you looking for?  An ISR might be cheaper. 

Note. I do have a CMM with 6 PRI and ACT card.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto: cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Paul Choi
Sent: Friday, May 25, 2007 10:22 AM
To: zohaib shabir; cisco-voip at puck.nether.net;
cisco-voip-request at puck.nether.net
Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway

Yes. Just put a fat CCM blade in the box and you can
use it as a VoIP interface.

Paul

--- zohaib shabir < zohaibshabir at gmail.com> wrote:

> Dear All,
> Can we use Catalyst 6500 series switch as Voice
> gateway.
>
> --
> Regards,
> Zohaib Shabir 
> Network Engineer(Voice and PreSales)
> DWP Group, TECH Division
> Ph:+92-302-8232689
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>




________________________________________________________________________

____________Got a little couch potato?
Check out fun summer activities for kids.
http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+ki

ds&cs=bz
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and PreSales)
DWP Group, TECH Division
Ph:+92-302-8232689 

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From craig at staffin.org  Fri May 25 13:45:43 2007
From: craig at staffin.org (Craig Staffin)
Date: Fri, 25 May 2007 12:45:43 -0500
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com>
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
	<003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca>
	<1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com>
Message-ID: <9b76d88c0705251045o2fcc1bdag226498722d67603a@mail.gmail.com>

why would anyone need that many FXO ports.  If thats the case you should be
looing at some form of digital communication.

Craig

On 5/25/07, zohaib shabir <zohaibshabir at gmail.com> wrote:
>
> Thanks for the reply,
> Problem for me is my client wants 60 FXO ports per router i cannot do even
> through 3845. Please advise me which way to go
>
> On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote:
> >
> >  sure, you can use the CMM modules with FXO or PRI ports.
> >
> > take a serious look at comparing the cost of the blades with an ISR
> > though.
> >
> > you'll be surprised as to how much cheaper it can be.
> >
> > and you'll get to use the ISR for all sorts of other stuff like SRST.
> > granted you can use the CMM for SRST but it is limited in memory.
> >
> >
> >
> > --------------------------------------------------------------------------------
> > Lelio Fulgenzi, B.A.
> > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."
> > WJR
> >
> > ----- Original Message -----
> >  *From:* zohaib shabir <zohaibshabir at gmail.com>
> > *To:* cisco-voip at puck.nether.net ; cisco-voip-request at puck.nether.net
> > *Sent:* Friday, May 25, 2007 1:04 PM
> > *Subject:* [cisco-voip] Catalyst 6500 switch as Voice Gateway
> >
> > Dear All,
> > Can we use Catalyst 6500 series switch as Voice gateway.
> >
> > --
> > Regards,
> > Zohaib Shabir
> > Network Engineer(Voice and PreSales)
> > DWP Group, TECH Division
> > Ph:+92-302-8232689
> >
> > ------------------------------
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> Regards,
> Zohaib Shabir
> Network Engineer(Voice and PreSales)
> DWP Group, TECH Division
> Ph:+92-302-8232689
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Craig Staffin
Craig at staffin.org
(H) 262-437-7313
(C) 262-613-6003
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From zohaibshabir at gmail.com  Fri May 25 13:50:25 2007
From: zohaibshabir at gmail.com (zohaib shabir)
Date: Fri, 25 May 2007 22:50:25 +0500
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <9b76d88c0705251045o2fcc1bdag226498722d67603a@mail.gmail.com>
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
	<003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca>
	<1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com>
	<9b76d88c0705251045o2fcc1bdag226498722d67603a@mail.gmail.com>
Message-ID: <1ff12970705251050v4d531ea3he72c8375de047891@mail.gmail.com>

Well actually my client does not want the PRI lines he wants FXO ports so
that he gets 60 lines independent of each other.

On 5/25/07, Craig Staffin <craig at staffin.org> wrote:
>
> why would anyone need that many FXO ports.  If thats the case you should
> be looing at some form of digital communication.
>
> Craig
>
> On 5/25/07, zohaib shabir <zohaibshabir at gmail.com> wrote:
> >
> > Thanks for the reply,
> > Problem for me is my client wants 60 FXO ports per router i cannot do
> > even through 3845. Please advise me which way to go
> >
> > On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote:
> > >
> > >  sure, you can use the CMM modules with FXO or PRI ports.
> > >
> > > take a serious look at comparing the cost of the blades with an ISR
> > > though.
> > >
> > > you'll be surprised as to how much cheaper it can be.
> > >
> > > and you'll get to use the ISR for all sorts of other stuff like SRST.
> > > granted you can use the CMM for SRST but it is limited in memory.
> > >
> > >
> > >
> > > --------------------------------------------------------------------------------
> > > Lelio Fulgenzi, B.A.
> > > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> > > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> > > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > >
> > > "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo
> > > buffalo."      WJR
> > >
> > > ----- Original Message -----
> > >  *From:* zohaib shabir <zohaibshabir at gmail.com>
> > > *To:* cisco-voip at puck.nether.net ; cisco-voip-request at puck.nether.net
> > > *Sent:* Friday, May 25, 2007 1:04 PM
> > > *Subject:* [cisco-voip] Catalyst 6500 switch as Voice Gateway
> > >
> > > Dear All,
> > > Can we use Catalyst 6500 series switch as Voice gateway.
> > >
> > > --
> > > Regards,
> > > Zohaib Shabir
> > > Network Engineer(Voice and PreSales)
> > > DWP Group, TECH Division
> > > Ph:+92-302-8232689
> > >
> > > ------------------------------
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> >
> >
> > --
> > Regards,
> > Zohaib Shabir
> > Network Engineer(Voice and PreSales)
> > DWP Group, TECH Division
> > Ph:+92-302-8232689
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>
>
> --
> Craig Staffin
> Craig at staffin.org
> (H) 262-437-7313
> (C) 262-613-6003




-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and PreSales)
DWP Group, TECH Division
Ph:+92-302-8232689
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From WOCAMPO at broward.org  Fri May 25 14:06:02 2007
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Fri, 25 May 2007 14:06:02 -0400
Subject: [cisco-voip] Open Source MCU
In-Reply-To: <4655B41A.7070608@infiltrated.net>
References: <963CE94787005149A52C17F33A028E1207E62003@EVS1B.boco.co.boulder.co.us>
	<4655B41A.7070608@infiltrated.net>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F786024DA34F@UVEXVS01.bc.broward.cty>

We had a Demo in the MCU and works great. 

In the Internal network you can create a Route Pattern and whoever has a
camera will be able to enjoy the conference dialing the number.

The other way, it is trough IP connection. 

 

I also recommend Radvision who not only has MCU capability but also
desktop-base visual communication in the same box.

We had a demo also and the Price I believe is shipper.  

 

RADVISION (Nasdaq: RVSN
<http://www.nasdaq.com/scripts/redir.asp?symbol=RVSN&page=quotes> ) is
the industry's leading provider of products and technologies for unified
visual communications over IP, 3G and emerging next generation networks
- enabling high definition video - conferencing, converged video
telephony services, and scalable desktop-based visual communications. 

 

http://radvision.com/

 

 

 

Walter Ocampo

 

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of J. Oquendo
Sent: Thursday, May 24, 2007 11:50 AM
To: Leetun, Rob
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Open Source MCU

 

Leetun, Rob wrote:

> Hi,

>  

> What is good open source MCU?  Has anyone implemented an open source 

> MCU, and if so, how did it go?

>  

> I am plan on using it in a CCM 4.1.3 environment along with a 2821 

> gatekeeper.

>  

> Thanks.

>  

> Rob

> ----------------------------------------------------------------------

> --

> 

> _______________________________________________

> cisco-voip mailing list

> cisco-voip at puck.nether.net

> https://puck.nether.net/mailman/listinfo/cisco-voip

>   

Don't know about production systems though...

 

http://www.openh323.org/

http://www.voxgratia.org/modules/news/

 

 

--

====================================================

J. Oquendo

http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743

echo infiltrated.net|sed 's/^/sil@/g' 

 

"Wise men talk because they have something to say; fools, because they
have to say something." -- Plato

 

 

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From bcarter at sentinel.com  Fri May 25 14:09:35 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Fri, 25 May 2007 13:09:35 -0500
Subject: [cisco-voip] CM 4.2(3)sr2a - Gone
Message-ID: <6A0744639B86DD459433108DBF101D26016AECC4@MAIL01.sentinel.com>

What happened to it?
 
 
Bill Carter
Sr. Business Communications Analyst
Sentinel Technologies
(O) 217.391.5015
 
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From vances at motivity.ca  Fri May 25 13:55:17 2007
From: vances at motivity.ca (Vance Shipley)
Date: Fri, 25 May 2007 13:55:17 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <1ff12970705251050v4d531ea3he72c8375de047891@mail.gmail.com>
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
	<003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca>
	<1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com>
	<9b76d88c0705251045o2fcc1bdag226498722d67603a@mail.gmail.com>
	<1ff12970705251050v4d531ea3he72c8375de047891@mail.gmail.com>
Message-ID: <20070525175517.GB72351@frogman.motivity.ca>


The cxorrect solution to this problem is to use an external
channel bank to convert the analog lines to digital.  I highly
recommend the Access bank product from Carrier Access.

	-Vance

On Fri, May 25, 2007 at 10:50:25PM +0500, zohaib shabir wrote:
}  Well actually my client does not want the PRI lines he wants FXO ports so
}  that he gets 60 lines independent of each other.

From MLinsemier at apcapital.com  Fri May 25 14:34:39 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Fri, 25 May 2007 14:34:39 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <1ff12970705251050v4d531ea3he72c8375de047891@mail.gmail.com>
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com><003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca><1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com><9b76d88c0705251045o2fcc1bdag226498722d67603a@mail.gmail.com>
	<1ff12970705251050v4d531ea3he72c8375de047891@mail.gmail.com>
Message-ID: <AB0A35F2486F81468971AA68558BE7190658C2D8@smieln01winm01.micoanet.com>

Could I ask what application the customer is planning on using these 60
FXO's for?  It's pretty cost prohibitive in most markets to run so many
analog circuits.  I think we calculated that the cutoff on FXO vs. PRI
was around 12 lines in Michigan.  Costs isn't the only thing to consider
either.  Remember that each line will need to be separately tweaked at
the gateway which could make it a management / troubleshooting
nightmare.  Additionally, there are several "digital" services that
simply aren't available in the analog world.

 

Matt

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of zohaib shabir
Sent: Friday, May 25, 2007 1:50 PM
To: Craig Staffin
Cc: cisco-voip at puck.nether.net; cisco-voip-request at puck.nether.net
Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway

 

Well actually my client does not want the PRI lines he wants FXO ports
so that he gets 60 lines independent of each other.

On 5/25/07, Craig Staffin < craig at staffin.org> wrote:

why would anyone need that many FXO ports.  If thats the case you should
be looing at some form of digital communication. 

Craig

 

On 5/25/07, zohaib shabir <zohaibshabir at gmail.com> wrote:

Thanks for the reply,
Problem for me is my client wants 60 FXO ports per router i cannot do
even through 3845. Please advise me which way to go

 

On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote:

sure, you can use the CMM modules with FXO or PRI ports.

 

take a serious look at comparing the cost of the blades with an ISR
though.

 

you'll be surprised as to how much cheaper it can be.

 

and you'll get to use the ISR for all sorts of other stuff like SRST.
granted you can use the CMM for SRST but it is limited in memory.

 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."
WJR

	----- Original Message ----- 

	From: zohaib shabir <mailto:zohaibshabir at gmail.com>  

	To: cisco-voip at puck.nether.net ;
cisco-voip-request at puck.nether.net 

	Sent: Friday, May 25, 2007 1:04 PM 

	Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway

	 

	Dear All,
	Can we use Catalyst 6500 series switch as Voice gateway.
	
	-- 
	Regards,
	Zohaib Shabir
	Network Engineer(Voice and PreSales)
	DWP Group, TECH Division
	Ph:+92-302-8232689 

________________________________

	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and PreSales)
DWP Group, TECH Division
Ph:+92-302-8232689 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip




-- 

Craig Staffin
Craig at staffin.org
(H) 262-437-7313 
(C) 262-613-6003 




-- 
Regards,
Zohaib Shabir
Network Engineer(Voice and PreSales)
DWP Group, TECH Division
Ph:+92-302-8232689 


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From lelio at uoguelph.ca  Fri May 25 14:35:26 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 14:35:26 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
	<003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca>
	<1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com>
Message-ID: <009401c79efb$76cc7840$47196883@cfs.uoguelph.ca>

Don't think you're gonna get 60 FXO ports on a 3845. 

http://www.cisco.com/en/US/customer/products/ps5855/products_qanda_item0900aecd8016c487.shtml

2 x EVM-HDs with two 6FXOs give 12 FXOs (plus 8FXS/DID if you can use them).

4 NM-HDs with 4FXOs gives 16 FXO

4 HWIC slots with 4 FXOs gives 16FXO

44 FXO at most from what I can see.


You're better off convincing them to go with digital. Worse comes to worse you can use seperate RTM groups for each LDN coming into the trunk so each line can only accept one call.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: zohaib shabir 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net ; cisco-voip-request at puck.nether.net 
  Sent: Friday, May 25, 2007 1:33 PM
  Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway


  Thanks for the reply,
  Problem for me is my client wants 60 FXO ports per router i cannot do even through 3845. Please advise me which way to go


  On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote:
    sure, you can use the CMM modules with FXO or PRI ports.

    take a serious look at comparing the cost of the blades with an ISR though.

    you'll be surprised as to how much cheaper it can be.

    and you'll get to use the ISR for all sorts of other stuff like SRST. granted you can use the CMM for SRST but it is limited in memory.


    --------------------------------------------------------------------------------
    Lelio Fulgenzi, B.A.
    Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
    (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
    "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

      ----- Original Message ----- 
      From: zohaib shabir 
      To: cisco-voip at puck.nether.net ; cisco-voip-request at puck.nether.net 
      Sent: Friday, May 25, 2007 1:04 PM 
      Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway


      Dear All,
      Can we use Catalyst 6500 series switch as Voice gateway.

      -- 
      Regards,
      Zohaib Shabir
      Network Engineer(Voice and PreSales)
      DWP Group, TECH Division
      Ph:+92-302-8232689 


--------------------------------------------------------------------------


      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip





  -- 
  Regards,
  Zohaib Shabir
  Network Engineer(Voice and PreSales)
  DWP Group, TECH Division
  Ph:+92-302-8232689 
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From lelio at uoguelph.ca  Fri May 25 14:37:51 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 14:37:51 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com>
	<003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca>
	<1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com>
	<9b76d88c0705251045o2fcc1bdag226498722d67603a@mail.gmail.com>
	<1ff12970705251050v4d531ea3he72c8375de047891@mail.gmail.com>
Message-ID: <00c501c79efb$cd190f60$47196883@cfs.uoguelph.ca>

Ask the client what their goals are. Work from there. You can do quite a bit with PRI configuration.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: zohaib shabir 
  To: Craig Staffin 
  Cc: Lelio Fulgenzi ; cisco-voip-request at puck.nether.net ; cisco-voip at puck.nether.net 
  Sent: Friday, May 25, 2007 1:50 PM
  Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway


  Well actually my client does not want the PRI lines he wants FXO ports so that he gets 60 lines independent of each other.


  On 5/25/07, Craig Staffin < craig at staffin.org> wrote:
    why would anyone need that many FXO ports.  If thats the case you should be looing at some form of digital communication. 

    Craig



    On 5/25/07, zohaib shabir <zohaibshabir at gmail.com> wrote:
      Thanks for the reply,
      Problem for me is my client wants 60 FXO ports per router i cannot do even through 3845. Please advise me which way to go



      On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote:
        sure, you can use the CMM modules with FXO or PRI ports.

        take a serious look at comparing the cost of the blades with an ISR though.

        you'll be surprised as to how much cheaper it can be.

        and you'll get to use the ISR for all sorts of other stuff like SRST. granted you can use the CMM for SRST but it is limited in memory.


        --------------------------------------------------------------------------------
        Lelio Fulgenzi, B.A.
        Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
        (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
        ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
        "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

          ----- Original Message ----- 
          From: zohaib shabir 
          To: cisco-voip at puck.nether.net ; cisco-voip-request at puck.nether.net 
          Sent: Friday, May 25, 2007 1:04 PM 
          Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway


          Dear All,
          Can we use Catalyst 6500 series switch as Voice gateway.

          -- 
          Regards,
          Zohaib Shabir
          Network Engineer(Voice and PreSales)
          DWP Group, TECH Division
          Ph:+92-302-8232689 


----------------------------------------------------------------------


          _______________________________________________
          cisco-voip mailing list
          cisco-voip at puck.nether.net
          https://puck.nether.net/mailman/listinfo/cisco-voip





      -- 
      Regards,
      Zohaib Shabir
      Network Engineer(Voice and PreSales)
      DWP Group, TECH Division
      Ph:+92-302-8232689 

      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net 
      https://puck.nether.net/mailman/listinfo/cisco-voip





    -- 

    Craig Staffin
    Craig at staffin.org
    (H) 262-437-7313 
    (C) 262-613-6003 



  -- 
  Regards,
  Zohaib Shabir
  Network Engineer(Voice and PreSales)
  DWP Group, TECH Division
  Ph:+92-302-8232689 
-------------- next part --------------
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From jonvoip at gmail.com  Fri May 25 14:45:08 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 25 May 2007 13:45:08 -0500
Subject: [cisco-voip] Yanking Compact Flash while the router is up...
Message-ID: <5d093f9a0705251145k63365279t98cb0bb490aa1360@mail.gmail.com>

So, I am curious, can you pull a CF card from a 2821 while the router
is up and running?

We need to replace flaky flash with a new one, but I would rather not
have to talk a remote tech (low skillset) through a ROMMON TFTP
operation..

If not, we can do it the hard way....but I have heard rumors that it works...

If it does, what are the risks...



Jonathan

From ray at oneunified.net  Fri May 25 14:56:21 2007
From: ray at oneunified.net (Ray Burkholder)
Date: Fri, 25 May 2007 15:56:21 -0300
Subject: [cisco-voip] [c-nsp] Yanking Compact Flash while the router is
	up...
In-Reply-To: <5d093f9a0705251145k63365279t98cb0bb490aa1360@mail.gmail.com>
References: <5d093f9a0705251145k63365279t98cb0bb490aa1360@mail.gmail.com>
Message-ID: <03f201c79efe$642d4a40$680a0a0a@oneunified.local>

Some machines have enough flash built in to accept downgraded IOS.  Swap the
external flash.  Power up, then upgrade the IOS again. 

> -----Original Message-----
> From: cisco-nsp-bounces at puck.nether.net 
> [mailto:cisco-nsp-bounces at puck.nether.net] On Behalf Of 
> Jonathan Charles
> Sent: Friday, May 25, 2007 15:45
> To: ciscovoip; cisco-nsp at puck.nether.net
> Subject: [c-nsp] Yanking Compact Flash while the router is up...
> 
> So, I am curious, can you pull a CF card from a 2821 while 
> the router is up and running?
> 
> We need to replace flaky flash with a new one, but I would 
> rather not have to talk a remote tech (low skillset) through 
> a ROMMON TFTP operation..
> 
> If not, we can do it the hard way....but I have heard rumors 
> that it works...
> 
> If it does, what are the risks...
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-nsp mailing list  cisco-nsp at puck.nether.net 
> https://puck.nether.net/mailman/listinfo/cisco-nsp
> archive at http://puck.nether.net/pipermail/cisco-nsp/
> 
> --
> Scanned for viruses and dangerous content at 
> http://www.oneunified.net and is believed to be clean.
> 
> 


-- 
Scanned for viruses and dangerous content at 
http://www.oneunified.net and is believed to be clean.


From asobihoudai at yahoo.com  Fri May 25 15:03:45 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 25 May 2007 12:03:45 -0700 (PDT)
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <009401c79efb$76cc7840$47196883@cfs.uoguelph.ca>
Message-ID: <78346.85344.qm@web57203.mail.re3.yahoo.com>

Lelio,

What about using multiple high density analog modules?
[is this even possible?]

24x3 = 72 FXO ports. I'm not sure that even a 3845
would be able to have three high density analog ports
on it.

Paul

--- Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

> Don't think you're gonna get 60 FXO ports on a 3845.
> 
> 
>
http://www.cisco.com/en/US/customer/products/ps5855/products_qanda_item0900aecd8016c487.shtml
> 
> 2 x EVM-HDs with two 6FXOs give 12 FXOs (plus
> 8FXS/DID if you can use them).
> 
> 4 NM-HDs with 4FXOs gives 16 FXO
> 
> 4 HWIC slots with 4 FXOs gives 16FXO
> 
> 44 FXO at most from what I can see.
> 
> 
> You're better off convincing them to go with
> digital. Worse comes to worse you can use seperate
> RTM groups for each LDN coming into the trunk so
> each line can only accept one call.
> 
> 
>
--------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph *
> Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> 
> "Buffalo buffalo Buffalo buffalo buffalo buffalo
> Buffalo buffalo."      WJR
> 
>   ----- Original Message ----- 
>   From: zohaib shabir 
>   To: Lelio Fulgenzi 
>   Cc: cisco-voip at puck.nether.net ;
> cisco-voip-request at puck.nether.net 
>   Sent: Friday, May 25, 2007 1:33 PM
>   Subject: Re: [cisco-voip] Catalyst 6500 switch as
> Voice Gateway
> 
> 
>   Thanks for the reply,
>   Problem for me is my client wants 60 FXO ports per
> router i cannot do even through 3845. Please advise
> me which way to go
> 
> 
>   On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca>
> wrote:
>     sure, you can use the CMM modules with FXO or
> PRI ports.
> 
>     take a serious look at comparing the cost of the
> blades with an ISR though.
> 
>     you'll be surprised as to how much cheaper it
> can be.
> 
>     and you'll get to use the ISR for all sorts of
> other stuff like SRST. granted you can use the CMM
> for SRST but it is limited in memory.
> 
> 
>    
>
--------------------------------------------------------------------------------
>     Lelio Fulgenzi, B.A.
>     Senior Analyst (CCS) * University of Guelph *
> Guelph, Ontario N1G 2W1
>     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>    
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> 
>     "Buffalo buffalo Buffalo buffalo buffalo buffalo
> Buffalo buffalo."      WJR
> 
>       ----- Original Message ----- 
>       From: zohaib shabir 
>       To: cisco-voip at puck.nether.net ;
> cisco-voip-request at puck.nether.net 
>       Sent: Friday, May 25, 2007 1:04 PM 
>       Subject: [cisco-voip] Catalyst 6500 switch as
> Voice Gateway
> 
> 
>       Dear All,
>       Can we use Catalyst 6500 series switch as
> Voice gateway.
> 
>       -- 
>       Regards,
>       Zohaib Shabir
>       Network Engineer(Voice and PreSales)
>       DWP Group, TECH Division
>       Ph:+92-302-8232689 
> 
> 
>
--------------------------------------------------------------------------
> 
> 
>      
> _______________________________________________
>       cisco-voip mailing list
>       cisco-voip at puck.nether.net
>      
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> 
> 
>   -- 
>   Regards,
>   Zohaib Shabir
>   Network Engineer(Voice and PreSales)
>   DWP Group, TECH Division
>   Ph:+92-302-8232689 >
_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. 
http://mobile.yahoo.com/go?refer=1GNXIC

From lelio at uoguelph.ca  Fri May 25 15:07:47 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 15:07:47 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
References: <78346.85344.qm@web57203.mail.re3.yahoo.com>
Message-ID: <011101c79eff$fb524370$47196883@cfs.uoguelph.ca>

which highdensity modules are you referring too?

under the relevant interfaces and modules, i only see the EVM-HD. that can only do 2x6FXO


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: Paul Choi 
  To: Lelio Fulgenzi ; zohaib shabir 
  Cc: cisco-voip-request at puck.nether.net ; cisco-voip at puck.nether.net 
  Sent: Friday, May 25, 2007 3:03 PM
  Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway


  Lelio,

  What about using multiple high density analog modules?
  [is this even possible?]

  24x3 = 72 FXO ports. I'm not sure that even a 3845
  would be able to have three high density analog ports
  on it.

  Paul

  --- Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

  > Don't think you're gonna get 60 FXO ports on a 3845.
  > 
  > 
  >
  http://www.cisco.com/en/US/customer/products/ps5855/products_qanda_item0900aecd8016c487.shtml
  > 
  > 2 x EVM-HDs with two 6FXOs give 12 FXOs (plus
  > 8FXS/DID if you can use them).
  > 
  > 4 NM-HDs with 4FXOs gives 16 FXO
  > 
  > 4 HWIC slots with 4 FXOs gives 16FXO
  > 
  > 44 FXO at most from what I can see.
  > 
  > 
  > You're better off convincing them to go with
  > digital. Worse comes to worse you can use seperate
  > RTM groups for each LDN coming into the trunk so
  > each line can only accept one call.
  > 
  > 
  >
  --------------------------------------------------------------------------------
  > Lelio Fulgenzi, B.A.
  > Senior Analyst (CCS) * University of Guelph *
  > Guelph, Ontario N1G 2W1
  > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  > 
  > "Buffalo buffalo Buffalo buffalo buffalo buffalo
  > Buffalo buffalo."      WJR
  > 
  >   ----- Original Message ----- 
  >   From: zohaib shabir 
  >   To: Lelio Fulgenzi 
  >   Cc: cisco-voip at puck.nether.net ;
  > cisco-voip-request at puck.nether.net 
  >   Sent: Friday, May 25, 2007 1:33 PM
  >   Subject: Re: [cisco-voip] Catalyst 6500 switch as
  > Voice Gateway
  > 
  > 
  >   Thanks for the reply,
  >   Problem for me is my client wants 60 FXO ports per
  > router i cannot do even through 3845. Please advise
  > me which way to go
  > 
  > 
  >   On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca>
  > wrote:
  >     sure, you can use the CMM modules with FXO or
  > PRI ports.
  > 
  >     take a serious look at comparing the cost of the
  > blades with an ISR though.
  > 
  >     you'll be surprised as to how much cheaper it
  > can be.
  > 
  >     and you'll get to use the ISR for all sorts of
  > other stuff like SRST. granted you can use the CMM
  > for SRST but it is limited in memory.
  > 
  > 
  >    
  >
  --------------------------------------------------------------------------------
  >     Lelio Fulgenzi, B.A.
  >     Senior Analyst (CCS) * University of Guelph *
  > Guelph, Ontario N1G 2W1
  >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  >    
  >
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  > 
  >     "Buffalo buffalo Buffalo buffalo buffalo buffalo
  > Buffalo buffalo."      WJR
  > 
  >       ----- Original Message ----- 
  >       From: zohaib shabir 
  >       To: cisco-voip at puck.nether.net ;
  > cisco-voip-request at puck.nether.net 
  >       Sent: Friday, May 25, 2007 1:04 PM 
  >       Subject: [cisco-voip] Catalyst 6500 switch as
  > Voice Gateway
  > 
  > 
  >       Dear All,
  >       Can we use Catalyst 6500 series switch as
  > Voice gateway.
  > 
  >       -- 
  >       Regards,
  >       Zohaib Shabir
  >       Network Engineer(Voice and PreSales)
  >       DWP Group, TECH Division
  >       Ph:+92-302-8232689 
  > 
  > 
  >
  --------------------------------------------------------------------------
  > 
  > 
  >      
  > _______________________________________________
  >       cisco-voip mailing list
  >       cisco-voip at puck.nether.net
  >      
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  > 
  > 
  > 
  > 
  > 
  >   -- 
  >   Regards,
  >   Zohaib Shabir
  >   Network Engineer(Voice and PreSales)
  >   DWP Group, TECH Division
  >   Ph:+92-302-8232689 >
  _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  > 



         
  ____________________________________________________________________________________Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. 
  http://mobile.yahoo.com/go?refer=1GNXIC
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From Matt.Slaga at us.didata.com  Fri May 25 15:09:49 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 25 May 2007 15:09:49 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <00c501c79efb$cd190f60$47196883@cfs.uoguelph.ca>
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com><003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca><1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com><9b76d88c0705251045o2fcc1bdag226498722d67603a@mail.gmail.com><1ff12970705251050v4d531ea3he72c8375de047891@mail.gmail.com>
	<00c501c79efb$cd190f60$47196883@cfs.uoguelph.ca>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005AD5EBD@USNAEXCH.na.didata.local>

Unless they are integrating into some kind of antiquated IVR system that
needs FXO inputs, I would strongly suggest you find a different solution
for the customer.  You will have one heck of a time configuring all
those ports, not too mention how much doo-doo you and the customer would
be in if you have any FXO disconnect supervision problems.

 

My 2 cents

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, May 25, 2007 2:38 PM
To: zohaib shabir; Craig Staffin
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway

 

Ask the client what their goals are. Work from there. You can do quite a
bit with PRI configuration.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."
WJR

	----- Original Message ----- 

	From: zohaib shabir <mailto:zohaibshabir at gmail.com>  

	To: Craig Staffin <mailto:craig at staffin.org>  

	Cc: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  ;
cisco-voip-request at puck.nether.net ; cisco-voip at puck.nether.net 

	Sent: Friday, May 25, 2007 1:50 PM

	Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway

	 

	Well actually my client does not want the PRI lines he wants FXO
ports so that he gets 60 lines independent of each other.

	On 5/25/07, Craig Staffin < craig at staffin.org> wrote: 

	why would anyone need that many FXO ports.  If thats the case
you should be looing at some form of digital communication. 
	
	Craig 

	 

	On 5/25/07, zohaib shabir <zohaibshabir at gmail.com> wrote: 

	Thanks for the reply,
	Problem for me is my client wants 60 FXO ports per router i
cannot do even through 3845. Please advise me which way to go 

	 

	On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote: 

	sure, you can use the CMM modules with FXO or PRI ports.

	 

	take a serious look at comparing the cost of the blades with an
ISR though.

	 

	you'll be surprised as to how much cheaper it can be.

	 

	and you'll get to use the ISR for all sorts of other stuff like
SRST. granted you can use the CMM for SRST but it is limited in memory.

	 

	 

	
------------------------------------------------------------------------
--------
	Lelio Fulgenzi, B.A.
	Senior Analyst (CCS) * University of Guelph * Guelph, Ontario
N1G 2W1
	(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
	
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
	"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo
buffalo."      WJR

		----- Original Message ----- 

		From: zohaib shabir <mailto:zohaibshabir at gmail.com>  

		To: cisco-voip at puck.nether.net ;
cisco-voip-request at puck.nether.net 

		Sent: Friday, May 25, 2007 1:04 PM 

		Subject: [cisco-voip] Catalyst 6500 switch as Voice
Gateway

		 

		Dear All,
		Can we use Catalyst 6500 series switch as Voice gateway.
		
		-- 
		Regards,
		Zohaib Shabir
		Network Engineer(Voice and PreSales)
		DWP Group, TECH Division
		Ph:+92-302-8232689 

________________________________

		_______________________________________________
		cisco-voip mailing list
		cisco-voip at puck.nether.net
		https://puck.nether.net/mailman/listinfo/cisco-voip

	
	
	
	-- 
	Regards,
	Zohaib Shabir
	Network Engineer(Voice and PreSales)
	DWP Group, TECH Division
	Ph:+92-302-8232689 

	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net 
	https://puck.nether.net/mailman/listinfo/cisco-voip

	
	
	
	-- 
	
	

	Craig Staffin
	Craig at staffin.org
	(H) 262-437-7313 
	(C) 262-613-6003 

	
	
	
	-- 
	Regards,
	Zohaib Shabir
	Network Engineer(Voice and PreSales)
	DWP Group, TECH Division
	Ph:+92-302-8232689 




-----------------------------------------
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From lelio at uoguelph.ca  Fri May 25 15:20:10 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 15:20:10 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
References: <1ff12970705251004o4473d87eqa8b70f2765a46ea2@mail.gmail.com><003a01c79ef0$e0590ae0$47196883@cfs.uoguelph.ca><1ff12970705251033x64723cabm59d55b142ffd17c7@mail.gmail.com><9b76d88c0705251045o2fcc1bdag226498722d67603a@mail.gmail.com><1ff12970705251050v4d531ea3he72c8375de047891@mail.gmail.com>
	<00c501c79efb$cd190f60$47196883@cfs.uoguelph.ca>
	<C1FE15183DA37645BC0633BC604E44F005AD5EBD@USNAEXCH.na.didata.local>
Message-ID: <015d01c79f01$b671e650$47196883@cfs.uoguelph.ca>

OMG - the CMM does not have FXO ports! 

WTH?


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: Matt Slaga (US) 
  To: Lelio Fulgenzi ; zohaib shabir ; Craig Staffin 
  Cc: cisco-voip at puck.nether.net 
  Sent: Friday, May 25, 2007 3:09 PM
  Subject: RE: [cisco-voip] Catalyst 6500 switch as Voice Gateway


  Unless they are integrating into some kind of antiquated IVR system that needs FXO inputs, I would strongly suggest you find a different solution for the customer.  You will have one heck of a time configuring all those ports, not too mention how much doo-doo you and the customer would be in if you have any FXO disconnect supervision problems.

   

  My 2 cents

   

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Friday, May 25, 2007 2:38 PM
  To: zohaib shabir; Craig Staffin
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway

   

  Ask the client what their goals are. Work from there. You can do quite a bit with PRI configuration.

   

  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

    ----- Original Message ----- 

    From: zohaib shabir 

    To: Craig Staffin 

    Cc: Lelio Fulgenzi ; cisco-voip-request at puck.nether.net ; cisco-voip at puck.nether.net 

    Sent: Friday, May 25, 2007 1:50 PM

    Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway

     

    Well actually my client does not want the PRI lines he wants FXO ports so that he gets 60 lines independent of each other.

    On 5/25/07, Craig Staffin < craig at staffin.org> wrote: 

    why would anyone need that many FXO ports.  If thats the case you should be looing at some form of digital communication. 

    Craig 

     

    On 5/25/07, zohaib shabir <zohaibshabir at gmail.com> wrote: 

    Thanks for the reply,
    Problem for me is my client wants 60 FXO ports per router i cannot do even through 3845. Please advise me which way to go 

     

    On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca> wrote: 

    sure, you can use the CMM modules with FXO or PRI ports.

     

    take a serious look at comparing the cost of the blades with an ISR though.

     

    you'll be surprised as to how much cheaper it can be.

     

    and you'll get to use the ISR for all sorts of other stuff like SRST. granted you can use the CMM for SRST but it is limited in memory.

     

     

    --------------------------------------------------------------------------------
    Lelio Fulgenzi, B.A.
    Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
    (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
    "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

      ----- Original Message ----- 

      From: zohaib shabir 

      To: cisco-voip at puck.nether.net ; cisco-voip-request at puck.nether.net 

      Sent: Friday, May 25, 2007 1:04 PM 

      Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway

       

      Dear All,
      Can we use Catalyst 6500 series switch as Voice gateway.

      -- 
      Regards,
      Zohaib Shabir
      Network Engineer(Voice and PreSales)
      DWP Group, TECH Division
      Ph:+92-302-8232689 


--------------------------------------------------------------------------

      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip




    -- 
    Regards,
    Zohaib Shabir
    Network Engineer(Voice and PreSales)
    DWP Group, TECH Division
    Ph:+92-302-8232689 


    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net 
    https://puck.nether.net/mailman/listinfo/cisco-voip




    -- 



    Craig Staffin
    Craig at staffin.org
    (H) 262-437-7313 
    (C) 262-613-6003 




    -- 
    Regards,
    Zohaib Shabir
    Network Engineer(Voice and PreSales)
    DWP Group, TECH Division
    Ph:+92-302-8232689 



------------------------------------------------------------------------------


  Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. 
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From jackson.tim at gmail.com  Fri May 25 15:25:27 2007
From: jackson.tim at gmail.com (Tim Jackson)
Date: Fri, 25 May 2007 14:25:27 -0500
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <011101c79eff$fb524370$47196883@cfs.uoguelph.ca>
References: <78346.85344.qm@web57203.mail.re3.yahoo.com>
	<011101c79eff$fb524370$47196883@cfs.uoguelph.ca>
Message-ID: <4407932e0705251225j34454e55uc7e01938b56477c3@mail.gmail.com>

Use T1 CAS interfaces and channel banks... Doesn't make sense to try and
throw that many analog lines into a card...

I'd reccomend the CAC Adit600 for it.. You can throw in 6 8port FXO/FXS
cards (a mix) and it has 2x T1 interfaces facing outwards... 3 T1s of Voice
is all that you need to cover this..

I'd still shy away from it as it will suck...

Maybe a couple if IAD2431/T1E1 models to cover it all... That'd probably be
your cheapest route, but even then, CPD and signaling issues will eat you
alive...

--
Tim

On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  which highdensity modules are you referring too?
>
> under the relevant interfaces and modules, i only see the EVM-HD. that can
> only do 2x6FXO
>
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."
> WJR
>
> ----- Original Message -----
> *From:* Paul Choi <asobihoudai at yahoo.com>
> *To:* Lelio Fulgenzi <lelio at uoguelph.ca> ; zohaib shabir<zohaibshabir at gmail.com>
> *Cc:* cisco-voip-request at puck.nether.net ; cisco-voip at puck.nether.net
> *Sent:* Friday, May 25, 2007 3:03 PM
> *Subject:* Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway
>
> Lelio,
>
> What about using multiple high density analog modules?
> [is this even possible?]
>
> 24x3 = 72 FXO ports. I'm not sure that even a 3845
> would be able to have three high density analog ports
> on it.
>
> Paul
>
> --- Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
> > Don't think you're gonna get 60 FXO ports on a 3845.
> >
> >
> >
>
> http://www.cisco.com/en/US/customer/products/ps5855/products_qanda_item0900aecd8016c487.shtml
> >
> > 2 x EVM-HDs with two 6FXOs give 12 FXOs (plus
> > 8FXS/DID if you can use them).
> >
> > 4 NM-HDs with 4FXOs gives 16 FXO
> >
> > 4 HWIC slots with 4 FXOs gives 16FXO
> >
> > 44 FXO at most from what I can see.
> >
> >
> > You're better off convincing them to go with
> > digital. Worse comes to worse you can use seperate
> > RTM groups for each LDN coming into the trunk so
> > each line can only accept one call.
> >
> >
> >
>
> --------------------------------------------------------------------------------
> > Lelio Fulgenzi, B.A.
> > Senior Analyst (CCS) * University of Guelph *
> > Guelph, Ontario N1G 2W1
> > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> >
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> >
> > "Buffalo buffalo Buffalo buffalo buffalo buffalo
> > Buffalo buffalo."      WJR
> >
> >   ----- Original Message -----
> >   From: zohaib shabir
> >   To: Lelio Fulgenzi
> >   Cc: cisco-voip at puck.nether.net ;
> > cisco-voip-request at puck.nether.net
> >   Sent: Friday, May 25, 2007 1:33 PM
> >   Subject: Re: [cisco-voip] Catalyst 6500 switch as
> > Voice Gateway
> >
> >
> >   Thanks for the reply,
> >   Problem for me is my client wants 60 FXO ports per
> > router i cannot do even through 3845. Please advise
> > me which way to go
> >
> >
> >   On 5/25/07, Lelio Fulgenzi < lelio at uoguelph.ca>
> > wrote:
> >     sure, you can use the CMM modules with FXO or
> > PRI ports.
> >
> >     take a serious look at comparing the cost of the
> > blades with an ISR though.
> >
> >     you'll be surprised as to how much cheaper it
> > can be.
> >
> >     and you'll get to use the ISR for all sorts of
> > other stuff like SRST. granted you can use the CMM
> > for SRST but it is limited in memory.
> >
> >
> >
> >
>
> --------------------------------------------------------------------------------
> >     Lelio Fulgenzi, B.A.
> >     Senior Analyst (CCS) * University of Guelph *
> > Guelph, Ontario N1G 2W1
> >     (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> >
> >
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> >
> >     "Buffalo buffalo Buffalo buffalo buffalo buffalo
> > Buffalo buffalo."      WJR
> >
> >       ----- Original Message -----
> >       From: zohaib shabir
> >       To: cisco-voip at puck.nether.net ;
> > cisco-voip-request at puck.nether.net
> >       Sent: Friday, May 25, 2007 1:04 PM
> >       Subject: [cisco-voip] Catalyst 6500 switch as
> > Voice Gateway
> >
> >
> >       Dear All,
> >       Can we use Catalyst 6500 series switch as
> > Voice gateway.
> >
> >       --
> >       Regards,
> >       Zohaib Shabir
> >       Network Engineer(Voice and PreSales)
> >       DWP Group, TECH Division
> >       Ph:+92-302-8232689
> >
> >
> >
> --------------------------------------------------------------------------
> >
> >
> >
> > _______________________________________________
> >       cisco-voip mailing list
> >       cisco-voip at puck.nether.net
> >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >
> >
> >   --
> >   Regards,
> >   Zohaib Shabir
> >   Network Engineer(Voice and PreSales)
> >   DWP Group, TECH Division
> >   Ph:+92-302-8232689 >
> _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
>
> ____________________________________________________________________________________Take
> the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news,
> photos & more.
> http://mobile.yahoo.com/go?refer=1GNXIC
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From Gerrad.Biffle at greensboro-nc.gov  Fri May 25 15:03:49 2007
From: Gerrad.Biffle at greensboro-nc.gov (Biffle, Gerrad)
Date: Fri, 25 May 2007 15:03:49 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <1ff12970705251041x62f3943dl8aa6c0a18cff08be@mail.gmail.com>
Message-ID: <038ECED6C286904CA93C8A76ABC45DF50BD21F50@VENUS.greensboronc.org>

What about (3) VG224s?  This would give you a couple of spares, if
needed.  I'm pretty sure they do either FXO or FXS.  Cost wouldn't be
that bad for three of them.


________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of zohaib shabir
	Sent: Friday, May 25, 2007 1:41 PM
	To: Voll, Scott
	Cc: cisco-voip-request at puck.nether.net;
cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway
	
	
	Well, I have to terminate 60 FXO ports. Please help if there is
any platform in which i can terminate 60 FXO ports.
	
	
	On 5/25/07, Voll, Scott < Scott.Voll at wesd.org
<mailto:Scott.Voll at wesd.org> > wrote: 

		How many PRI's are you looking for?  An ISR might be
cheaper. 
		
		Note. I do have a CMM with 6 PRI and ACT card.
		
		Scott
		
		-----Original Message-----
		From: cisco-voip-bounces at puck.nether.net
		[mailto: cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Paul Choi
		Sent: Friday, May 25, 2007 10:22 AM
		To: zohaib shabir; cisco-voip at puck.nether.net;
		cisco-voip-request at puck.nether.net
		Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice
Gateway
		
		Yes. Just put a fat CCM blade in the box and you can
		use it as a VoIP interface.
		
		Paul
		
		--- zohaib shabir < zohaibshabir at gmail.com> wrote:
		
		> Dear All,
		> Can we use Catalyst 6500 series switch as Voice
		> gateway.
		>
		> --
		> Regards,
		> Zohaib Shabir 
		> Network Engineer(Voice and PreSales)
		> DWP Group, TECH Division
		> Ph:+92-302-8232689
		> > _______________________________________________
		> cisco-voip mailing list
		> cisco-voip at puck.nether.net
		> https://puck.nether.net/mailman/listinfo/cisco-voip
		>
		
		
		
		
	
________________________________________________________________________

		____________Got a little couch potato?
		Check out fun summer activities for kids.
	
http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+ki

		ds&cs=bz
		_______________________________________________
		cisco-voip mailing list
		cisco-voip at puck.nether.net
		https://puck.nether.net/mailman/listinfo/cisco-voip
		




	-- 
	Regards,
	Zohaib Shabir
	Network Engineer(Voice and PreSales)
	DWP Group, TECH Division
	Ph:+92-302-8232689 

=======================================================
Please note that email sent to and from this address is subject 
to the North Carolina Public Records Law and may be disclosed to 
third parties.
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From lelio at uoguelph.ca  Fri May 25 15:39:04 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 15:39:04 -0400
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
References: <038ECED6C286904CA93C8A76ABC45DF50BD21F50@VENUS.greensboronc.org>
Message-ID: <01b401c79f04$5a425290$47196883@cfs.uoguelph.ca>

vg224 can only do fxs as far as i can tell.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: Biffle, Gerrad 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, May 25, 2007 3:03 PM
  Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway


  What about (3) VG224s?  This would give you a couple of spares, if needed.  I'm pretty sure they do either FXO or FXS.  Cost wouldn't be that bad for three of them.



----------------------------------------------------------------------------
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of zohaib shabir
    Sent: Friday, May 25, 2007 1:41 PM
    To: Voll, Scott
    Cc: cisco-voip-request at puck.nether.net; cisco-voip at puck.nether.net
    Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway


    Well, I have to terminate 60 FXO ports. Please help if there is any platform in which i can terminate 60 FXO ports.


    On 5/25/07, Voll, Scott < Scott.Voll at wesd.org> wrote: 
      How many PRI's are you looking for?  An ISR might be cheaper. 

      Note. I do have a CMM with 6 PRI and ACT card.

      Scott

      -----Original Message-----
      From: cisco-voip-bounces at puck.nether.net
      [mailto: cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
      Sent: Friday, May 25, 2007 10:22 AM
      To: zohaib shabir; cisco-voip at puck.nether.net;
      cisco-voip-request at puck.nether.net
      Subject: Re: [cisco-voip] Catalyst 6500 switch as Voice Gateway

      Yes. Just put a fat CCM blade in the box and you can
      use it as a VoIP interface.

      Paul

      --- zohaib shabir < zohaibshabir at gmail.com> wrote:

      > Dear All,
      > Can we use Catalyst 6500 series switch as Voice
      > gateway.
      >
      > --
      > Regards,
      > Zohaib Shabir 
      > Network Engineer(Voice and PreSales)
      > DWP Group, TECH Division
      > Ph:+92-302-8232689
      > > _______________________________________________
      > cisco-voip mailing list
      > cisco-voip at puck.nether.net
      > https://puck.nether.net/mailman/listinfo/cisco-voip
      >




      ________________________________________________________________________ 
      ____________Got a little couch potato?
      Check out fun summer activities for kids.
      http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+ki 
      ds&cs=bz
      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip




    -- 
    Regards,
    Zohaib Shabir
    Network Engineer(Voice and PreSales)
    DWP Group, TECH Division
    Ph:+92-302-8232689 



        =======================================================
        Please note that email sent to and from this address is subject 
        to the North Carolina Public Records Law and may be disclosed to 
        third parties. 



------------------------------------------------------------------------------


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From asobihoudai at yahoo.com  Fri May 25 16:35:16 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Fri, 25 May 2007 13:35:16 -0700 (PDT)
Subject: [cisco-voip] Catalyst 6500 switch as Voice Gateway
In-Reply-To: <01b401c79f04$5a425290$47196883@cfs.uoguelph.ca>
Message-ID: <905342.63635.qm@web57211.mail.re3.yahoo.com>

So far as I can remember, both vg248 and vg224 only
have FXS ports.

Paul

--- Lelio Fulgenzi <lelio at uoguelph.ca> wrote:

> vg224 can only do fxs as far as i can tell.
> 
> 
>
--------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph *
> Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> 
> "Buffalo buffalo Buffalo buffalo buffalo buffalo
> Buffalo buffalo."      WJR
> 
>   ----- Original Message ----- 
>   From: Biffle, Gerrad 
>   To: cisco-voip at puck.nether.net 
>   Sent: Friday, May 25, 2007 3:03 PM
>   Subject: Re: [cisco-voip] Catalyst 6500 switch as
> Voice Gateway
> 
> 
>   What about (3) VG224s?  This would give you a
> couple of spares, if needed.  I'm pretty sure they
> do either FXO or FXS.  Cost wouldn't be that bad for
> three of them.
> 
> 
> 
>
----------------------------------------------------------------------------
>     From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of zohaib shabir
>     Sent: Friday, May 25, 2007 1:41 PM
>     To: Voll, Scott
>     Cc: cisco-voip-request at puck.nether.net;
> cisco-voip at puck.nether.net
>     Subject: Re: [cisco-voip] Catalyst 6500 switch
> as Voice Gateway
> 
> 
>     Well, I have to terminate 60 FXO ports. Please
> help if there is any platform in which i can
> terminate 60 FXO ports.
> 
> 
>     On 5/25/07, Voll, Scott < Scott.Voll at wesd.org>
> wrote: 
>       How many PRI's are you looking for?  An ISR
> might be cheaper. 
> 
>       Note. I do have a CMM with 6 PRI and ACT card.
> 
>       Scott
> 
>       -----Original Message-----
>       From: cisco-voip-bounces at puck.nether.net
>       [mailto: cisco-voip-bounces at puck.nether.net]
> On Behalf Of Paul Choi
>       Sent: Friday, May 25, 2007 10:22 AM
>       To: zohaib shabir; cisco-voip at puck.nether.net;
>       cisco-voip-request at puck.nether.net
>       Subject: Re: [cisco-voip] Catalyst 6500 switch
> as Voice Gateway
> 
>       Yes. Just put a fat CCM blade in the box and
> you can
>       use it as a VoIP interface.
> 
>       Paul
> 
>       --- zohaib shabir < zohaibshabir at gmail.com>
> wrote:
> 
>       > Dear All,
>       > Can we use Catalyst 6500 series switch as
> Voice
>       > gateway.
>       >
>       > --
>       > Regards,
>       > Zohaib Shabir 
>       > Network Engineer(Voice and PreSales)
>       > DWP Group, TECH Division
>       > Ph:+92-302-8232689
>       > >
> _______________________________________________
>       > cisco-voip mailing list
>       > cisco-voip at puck.nether.net
>       >
> https://puck.nether.net/mailman/listinfo/cisco-voip
>       >
> 
> 
> 
> 
>      
>
________________________________________________________________________
> 
>       ____________Got a little couch potato?
>       Check out fun summer activities for kids.
>      
>
http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+ki
> 
>       ds&cs=bz
>      
> _______________________________________________
>       cisco-voip mailing list
>       cisco-voip at puck.nether.net
>      
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> 
>     -- 
>     Regards,
>     Zohaib Shabir
>     Network Engineer(Voice and PreSales)
>     DWP Group, TECH Division
>     Ph:+92-302-8232689 
> 
> 
> 
>        
>
=======================================================
>         Please note that email sent to and from this
> address is subject 
>         to the North Carolina Public Records Law and
> may be disclosed to 
>         third parties. 
> 
> 
> 
>
------------------------------------------------------------------------------
> 
> 
>   _______________________________________________
>   cisco-voip mailing list
>   cisco-voip at puck.nether.net
>  
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. 
http://answers.yahoo.com/dir/?link=list&sid=396545433

From jason.aarons at us.didata.com  Fri May 25 17:32:00 2007
From: jason.aarons at us.didata.com (jason.aarons at us.didata.com)
Date: Fri, 25 May 2007 17:32:00 -0400
Subject: [cisco-voip] 4.1 Callback limitations
Message-ID: <USNAMXWEB01IKOIBAc0000000e5@webmail.us.didata.com>

If someone is member of Line Group will they be able to use Callback softkey? The phone softkey template has callback on ring in and it works for most users but this one. He never gets the softkey....reset thw phone,  etc.

Jason via Treo 700p
904-338-3245 cell


-----------------------------------------
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This e-mail communication and any attachments may contain
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this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
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From jonvoip at gmail.com  Fri May 25 20:34:52 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 25 May 2007 19:34:52 -0500
Subject: [cisco-voip] MGCP Fallback time...
Message-ID: <5d093f9a0705251734n71238a7ch7cbcb676e9de9c7c@mail.gmail.com>

I have a new SRST router (MGCP back to CCM)...

When we yank the WAN, it takes about 13 minutes for the router to
realize that CCM is down and kill the PRI... and revert to H.323 mode.

If you look at the attached log file, you can see that the phones are
registering to the router, and a full 13 minutes later the PRI goes
down...

Is there a way to change this to make it quicker?

IOS is 12.4.3g




Jonathan
-------------- next part --------------
AMLIDAL2811#
*May 26 00:03:48.123: %IPPHONE-6-REG_ALARM: 14: Name=SEP001AA17BE81E Load= SCCP4
1.8-2-1S Last=CM-closed-TCP
*May 26 00:03:48.143: %IPPHONE-6-REGISTER_NEW: ephone-13:SEP001AA17BE81E IP:172.
27.128.23 Socket:18 DeviceType:Phone has registered.
AMLIDAL2811#
*May 26 00:03:51.003: %IPPHONE-6-REG_ALARM: 14: Name=SEP001AA193EE46 Load= SCCP4
1.8-2-1S Last=CM-closed-TCP
*May 26 00:03:51.003: %IPPHONE-6-REGISTER_NEW: ephone-11:SEP001AA193EE46 IP:172.
27.128.30 Socket:19 DeviceType:Phone has registered.
*May 26 00:03:51.255: %IPPHONE-6-REG_ALARM: 14: Name=SEP001AA1B4835B Load= SCCP4
1.8-2-1S Last=CM-closed-TCP
*May 26 00:03:51.255: %IPPHONE-6-REG_ALARM: 14: Name=SEP001AA18DD988 Load= SCCP4
1.8-2-1S Last=CM-closed-TCP
*May 26 00:03:51.255: %IPPHONE-6-REGISTER_NEW: ephone-2:SEP001AA1B4835B IP:172.2
7.128.21 Socket:20 DeviceType:Phone has registered.
AMLIDAL2811#
*May 26 00:03:51.255: %IPPHONE-6-REGISTER_NEW: ephone-22:SEP001AA18DD988 IP:172.
27.128.45 Socket:21 DeviceType:Phone has registered.
*May 26 00:03:51.771: %IPPHONE-6-REG_ALARM: 14: Name=SEP001AA18DE3CB Load= SCCP4
1.8-2-1S Last=CM-closed-TCP
*May 26 00:03:51.771: %IPPHONE-6-REG_ALARM: 14: Name=SEP001AA163B6DC Load= SCCP4
1.8-2-1S Last=CM-closed-TCP
*May 26 00:03:51.779: %IPPHONE-6-REGISTER_NEW: ephone-12:SEP001AA18DE3CB IP:172.
27.128.51 Socket:22 DeviceType:Phone has registered.
*May 26 00:03:52.031: %IPPHONE-6-REGISTER_NEW: ephone-29:SEP001AA163B6DC IP:172.
27.128.44 Socket:23 DeviceType:Phone has registered.
AMLIDAL2811#
*May 26 00:03:52.323: %IPPHONE-6-REG_ALARM: 14: Name=SEP001AA1B48CEB Load= SCCP4
1.8-2-1S Last=CM-closed-TCP
*May 26 00:03:52.623: %IPPHONE-6-REGISTER_NEW: ephone-24:SEP001AA1B48CEB IP:172.
27.128.35 Socket:24 DeviceType:Phone has registered.
*May 26 00:03:53.171: %IPPHONE-6-REG_ALARM: 14: Name=SEP001AA1A804FE Load= SCCP4
1.8-2-1S Last=CM-closed-TCP
*May 26 00:03:53.171: %IPPHONE-6-REGISTER_NEW: ephone-3:SEP001AA1A804FE IP:172.2
7.128.27 Socket:25 DeviceType:Phone has registered.
AMLIDAL2811#
*May 26 00:04:03.083: %IPPHONE-6-REG_ALARM: 14: Name=SEP001AA193EF47 Load= SCCP4
1.8-2-1S Last=CM-closed-TCP
*May 26 00:04:03.083: %IPPHONE-6-REGISTER_NEW: ephone-9:SEP001AA193EF47 IP:172.2
7.128.34 Socket:26 DeviceType:Phone has registered.
AMLIDAL2811#
*May 26 00:05:21.987: %IPPHONE-6-REG_ALARM: Name=ATA001B53F0BABA Load=ATA030203S
CCP051201A.zup Last=Initialized
*May 26 00:05:22.239: %IPPHONE-6-REGISTER_NEW: ephone-4:ATA001B53F0BABA IP:172.2
7.128.50 Socket:27 DeviceType:Phone has registered.
AMLIDAL2811#
*May 26 00:05:26.103: %IPPHONE-6-REG_ALARM: Name=ATA001B53F0BC07 Load=ATA030203S
CCP051201A.zup Last=Initialized
*May 26 00:05:26.103: %IPPHONE-6-REG_ALARM: Name=ATA001B53C64188 Load=ATA030203S
CCP051201A.zup Last=Initialized
*May 26 00:05:26.103: %IPPHONE-6-REGISTER_NEW: ephone-5:ATA001B53F0BC07 IP:172.2
7.128.49 Socket:28 DeviceType:Phone has registered.
*May 26 00:05:26.103: %IPPHONE-6-REGISTER_NEW: ephone-6:ATA001B53C64188 IP:172.2
7.128.54 Socket:29 DeviceType:Phone has registered.
AMLIDAL2811#
*May 26 00:10:51.891: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is
 0x0 0x0, Calling num 6714
*May 26 00:10:51.891: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is
 0x0 0x0, Called num 13126567496
*May 26 00:10:51.891: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref = 0x0086

        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98397
                Exclusive, Channel 23
        Progress Ind i = 0x8183 - Origination address is non-ISDN
AMLIDAL2811#
        Calling Party Number i = 0x0080, '6714'
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0x80, '[removed]'
                Plan:Unknown, Type:Unknown
*May 26 00:10:51.923: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x80
01
        Cause i = 0x80A9 - Temporary failure
*May 26 00:10:51.923: ISDN Se0/0/0:23 **ERROR**: L3_BadPeerMsg: event 0x4D cr 0x
1 callid 0x0
*May 26 00:10:51.923: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8  callref =
 0x8086
        Cause i = 0x8091 - User busy
*May 26 00:10:51.927: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref =
 0x0001
        Cause i = 0x80D1 - Invalid call reference value
AMLIDAL2811#
*May 26 00:13:57.415: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is
 0x0 0x0, Calling num 6788
*May 26 00:13:57.415: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is
 0x0 0x0, Called num 13126567496
*May 26 00:13:57.419: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref = 0x0087

        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98397
                Exclusive, Channel 23
        Progress Ind i = 0x8183 - Origination address is non-ISDN
AMLIDAL2811#
        Calling Party Number i = 0x0080, '6788'
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0x80, '13126567496'
                Plan:Unknown, Type:Unknown
*May 26 00:13:57.495: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8  callref = 0x
8087
        Channel ID i = 0xA98397
                Exclusive, Channel 23
AMLIDAL2811#
*May 26 00:13:59.015: ISDN Se0/0/0:23 Q931: RX <- PROGRESS pd = 8  callref = 0x8
087
        Progress Ind i = 0x8288 - In-band info or appropriate now available
AMLIDAL2811#
*May 26 00:14:04.231: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0
x0087
        Cause i = 0x8090 - Normal call clearing
*May 26 00:14:04.267: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x80
87
*May 26 00:14:04.271: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref =
 0x0087
AMLIDAL2811#
*May 26 00:14:38.011: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is
 0x0 0x0, Calling num 6788
*May 26 00:14:38.015: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0xD is
 0x0 0x0, Called num 16302481953
*May 26 00:14:38.015: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8  callref = 0x0088

        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98397
                Exclusive, Channel 23
        Progress Ind i = 0x8183 - Origination address is non-ISDN
AMLIDAL2811#
        Calling Party Number i = 0x0080, '6788'
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0x80, '[removed]'
                Plan:Unknown, Type:Unknown
*May 26 00:14:38.075: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8  callref = 0x
8088
        Channel ID i = 0xA98397
                Exclusive, Channel 23
AMLIDAL2811#
*May 26 00:14:39.835: ISDN Se0/0/0:23 Q931: RX <- PROGRESS pd = 8  callref = 0x8
088
        Progress Ind i = 0x8288 - In-band info or appropriate now available
AMLIDAL2811#
*May 26 00:14:50.123: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0
x0088
        Cause i = 0x8090 - Normal call clearing
*May 26 00:14:50.171: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x80
88
*May 26 00:14:50.175: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref =
 0x0088
AMLIDAL2811#
*May 26 00:16:34.463: %IPPHONE-6-UNREGISTER_NORMAL: ephone-25:SEP001AA1B4899D IP
:172.27.128.36 Socket:5 DeviceType:Phone has unregistered normally.
*May 26 00:16:34.719: %IPPHONE-6-UNREGISTER_NORMAL: ephone-10:SEP001AA18DE1E1 IP
:172.27.128.52 Socket:14 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:36.735: %IPPHONE-6-UNREGISTER_NORMAL: ephone-19:SEP001AA1B48E8B IP
:172.27.128.32 Socket:6 DeviceType:Phone has unregistered normally.
*May 26 00:16:36.995: %IPPHONE-6-UNREGISTER_NORMAL: ephone-14:SEP001AA1B2B954 IP
:172.27.128.37 Socket:16 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:39.787: [S] mgcp_msg_ack:6059,Updating (*)=172.28.2.5
AMLIDAL2811#
*May 26 00:16:40.271: %IPPHONE-6-UNREGISTER_NORMAL: ephone-18:SEP001AA1B2B91C IP
:172.27.128.39 Socket:13 DeviceType:Phone has unregistered normally.
*May 26 00:16:40.779: %IPPHONE-6-UNREGISTER_NORMAL: ephone-29:SEP001AA163B6DC IP
:172.27.128.44 Socket:23 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:41.539: %IPPHONE-6-UNREGISTER_NORMAL: ephone-17:SEP001AA1B2B9AF IP
:172.27.128.31 Socket:15 DeviceType:Phone has unregistered normally.
*May 26 00:16:42.047: %IPPHONE-6-UNREGISTER_NORMAL: ephone-8:SEP001AA18DE037 IP:
172.27.128.48 Socket:17 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:44.731: ISDN Se0/0/0:23 Q931: L3_ShutDown: Shutting down ISDN Laye
r 3
*May 26 00:16:44.731: %ISDN-6-LAYER2DOWN: Layer 2 for Interface Se0/0/0:23, TEI
0 changed to down
*May 26 00:16:44.735: ISDN Se0/0/0:23 Q931: Ux_DLRelInd: DL_REL_IND received fro
m L2
AMLIDAL2811#
*May 26 00:16:44.823: %IPPHONE-6-UNREGISTER_NORMAL: ephone-2:SEP001AA1B4835B IP:
172.27.128.21 Socket:20 DeviceType:Phone has unregistered normally.
*May 26 00:16:44.827: %IPPHONE-6-UNREGISTER_NORMAL: ephone-22:SEP001AA18DD988 IP
:172.27.128.45 Socket:21 DeviceType:Phone has unregistered normally.
*May 26 00:16:45.583: %IPPHONE-6-UNREGISTER_NORMAL: ephone-12:SEP001AA18DE3CB IP
:172.27.128.51 Socket:22 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:45.839: %IPPHONE-6-UNREGISTER_NORMAL: ephone-28:SEP001AA18DDD39 IP
:172.27.128.25 Socket:9 DeviceType:Phone has unregistered normally.
*May 26 00:16:46.727: %LINK-5-CHANGED: Interface Serial0/0/0:23, changed state t
o administratively down
AMLIDAL2811#
*May 26 00:16:46.851: %IPPHONE-6-UNREGISTER_NORMAL: ephone-9:SEP001AA193EF47 IP:
172.27.128.34 Socket:26 DeviceType:Phone has unregistered normally.
*May 26 00:16:47.615: %IPPHONE-6-UNREGISTER_NORMAL: ephone-3:SEP001AA1A804FE IP:
172.27.128.27 Socket:25 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:50.639: %IPPHONE-6-UNREGISTER_NORMAL: ephone-23:SEP001AA1B48A4F IP
:172.27.128.46 Socket:2 DeviceType:Phone has unregistered normally.
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:23, changed state to
 up
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:0, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:1, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:2, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:3, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:4, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:5, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:6, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:7, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:8, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:9, changed state to
down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:10, changed state to
 down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:11, changed state to
 down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:12, changed state to
 down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:13, changed state to
 down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:14, changed state to
 down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:15, changed state to
 down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:16, changed state to
 down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:17, changed state to
 down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:18, changed state to
 down
*May 26 00:16:51.135: %LINK-3-UPDOWN: Interface Serial0/0/0:19, changed state to
 down
*May 26 00:16:51.139: %LINK-3-UPDOWN: Interface Serial0/0/0:20, changed state to
 down
*May 26 00:16:51.139: %LINK-3-UPDOWN: Interface Serial0/0/0:21, changed state to
 down
*May 26 00:16:51.139: %LINK-3-UPDOWN: Interface Serial0/0/0:22, changed state to
 down
AMLIDAL2811#
*May 26 00:16:51.647: %IPPHONE-6-UNREGISTER_NORMAL: ephone-11:SEP001AA193EE46 IP
:172.27.128.30 Socket:19 DeviceType:Phone has unregistered normally.
*May 26 00:16:51.903: %IPPHONE-6-UNREGISTER_NORMAL: ephone-21:SEP001A6C3117EF IP
:172.27.128.33 Socket:4 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:52.663: %IPPHONE-6-UNREGISTER_NORMAL: ephone-1:SEP001AA1B483DC IP:
172.27.128.28 Socket:12 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:54.431: %IPPHONE-6-UNREGISTER_NORMAL: ephone-15:SEP001AA12736BB IP
:172.27.128.40 Socket:3 DeviceType:Phone has unregistered normally.
*May 26 00:16:54.959: %ISDN-6-LAYER2UP: Layer 2 for Interface Se0/0/0:23, TEI 0
changed to up
*May 26 00:16:55.119: ISDN Se0/0/0:23 **ERROR**: L2_AdvanceVA: TX_ack_queue empt
y
AMLIDAL2811#
*May 26 00:16:55.439: %IPPHONE-6-UNREGISTER_NORMAL: ephone-16:SEP001AA193E981 IP
:172.27.128.29 Socket:8 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:56.703: %IPPHONE-6-UNREGISTER_NORMAL: ephone-20:SEP001AA1B48C5D IP
:172.27.128.22 Socket:10 DeviceType:Phone has unregistered normally.
*May 26 00:16:56.959: %IPPHONE-6-UNREGISTER_NORMAL: ephone-27:SEP001A6C744902 IP
:172.27.128.26 Socket:7 DeviceType:Phone has unregistered normally.
*May 26 00:16:57.215: %IPPHONE-6-UNREGISTER_NORMAL: ephone-7:SEP001AA1B483F1 IP:
172.27.128.24 Socket:11 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:16:57.971: %IPPHONE-6-UNREGISTER_NORMAL: ephone-13:SEP001AA17BE81E IP
:172.27.128.23 Socket:18 DeviceType:Phone has unregistered normally.
*May 26 00:16:58.727: %IPPHONE-6-UNREGISTER_NORMAL: ephone-26:SEP001AA163B83B IP
:172.27.128.53 Socket:1 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:17:16.371: %IPPHONE-6-UNREGISTER_NORMAL: ephone-24:SEP001AA1B48CEB IP
:172.27.128.35 Socket:24 DeviceType:Phone has unregistered normally.
AMLIDAL2811#sh clock
*00:17:41.119 UTC Sat May 26 2007
AMLIDAL2811#
*May 26 00:18:34.503: %IPPHONE-6-UNREGISTER_NORMAL: ephone-5:ATA001B53F0BC07 IP:
172.27.128.49 Socket:28 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:18:35.511: %IPPHONE-6-UNREGISTER_NORMAL: ephone-4:ATA001B53F0BABA IP:
172.27.128.50 Socket:27 DeviceType:Phone has unregistered normally.
AMLIDAL2811#
*May 26 00:18:43.835: %IPPHONE-6-UNREGISTER_NORMAL: ephone-6:ATA001B53C64188 IP:
172.27.128.54 Socket:29 DeviceType:Phone has unregistered normally.
AMLIDAL2811#

From lelio at uoguelph.ca  Fri May 25 20:47:09 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 20:47:09 -0400
Subject: [cisco-voip] MGCP Fallback time...
References: <5d093f9a0705251734n71238a7ch7cbcb676e9de9c7c@mail.gmail.com>
Message-ID: <001101c79f2f$646e0680$c901a8c0@pdp11>

not sure if this is what's happening in your case, but i had a default route 
in my original setup that caused a delay. i had to put some ACLs on it to 
make sure it didn't try to connect to the ccm server over a different route 
(which was blocked somewhere else).

just a thought.

maybe you can post your config.

----- Original Message ----- 
From: "Jonathan Charles" <jonvoip at gmail.com>
To: "ciscovoip" <cisco-voip at puck.nether.net>
Sent: Friday, May 25, 2007 8:34 PM
Subject: [cisco-voip] MGCP Fallback time...


>I have a new SRST router (MGCP back to CCM)...
>
> When we yank the WAN, it takes about 13 minutes for the router to
> realize that CCM is down and kill the PRI... and revert to H.323 mode.
>
> If you look at the attached log file, you can see that the phones are
> registering to the router, and a full 13 minutes later the PRI goes
> down...
>
> Is there a way to change this to make it quicker?
>
> IOS is 12.4.3g
>
>
>
>
> Jonathan
>


--------------------------------------------------------------------------------


> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From jonvoip at gmail.com  Fri May 25 20:57:21 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 25 May 2007 19:57:21 -0500
Subject: [cisco-voip] MGCP Fallback time...
In-Reply-To: <001101c79f2f$646e0680$c901a8c0@pdp11>
References: <5d093f9a0705251734n71238a7ch7cbcb676e9de9c7c@mail.gmail.com>
	<001101c79f2f$646e0680$c901a8c0@pdp11>
Message-ID: <5d093f9a0705251757s13fc34abr3c48fa32c265472f@mail.gmail.com>

Here it is...


Current configuration : 7664 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname PINKY1
!
boot-start-marker
boot-end-marker
!
logging buffered 1000000 warnings
enable secret 5 $1$OGFk$TCpZe0TS9B4anq.Q6MJIQ0
!
no aaa new-model
!
resource policy
!
network-clock-participate wic 0
ip subnet-zero
!
!
ip cef
!
!
no ip domain lookup
ip domain name amli.com
isdn switch-type primary-ni
!
voice-card 0
 no dspfarm
!
!
!
!
!
!
!
!
!
!
!
!
!
application
 global
  service alternate Default
 !
!
!
crypto pki trustpoint TP-self-signed-2413116866
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-2413116866
 revocation-check none
 rsakeypair TP-self-signed-2413116866
!
!
crypto pki certificate chain TP-self-signed-2413116866
 certificate self-signed 01
  3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
  31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
  69666963 6174652D 32343133 31313638 3636301E 170D3037 30353233 31363136
  35385A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
  4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 34313331
  31363836 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
  8100C7DB 31404006 EEE793F7 398F1A35 07509CDB D444F92D 79FF5F4A 73A01D89
  B77EF538 DF19058C CCE57B9C ABF0597B DD60FAE8 3CD96D2E DD46A581 15B15F38
  DC3368D4 074A66F7 9E484DED E36394F6 05EA9FDA F81632FC 31D90520 256D497E
  CE8F91E4 042BD4E6 824C0DE6 F3107769 F0FFA0E0 65321D02 025D7F33 65C6DDE4
  027F0203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603
  551D1104 18301682 14414D4C 4944414C 32383131 2E616D6C 692E636F 6D301F06
  03551D23 04183016 80144221 418BAB16 E9B5F101 F02AA6B0 0E6B618D 83BA301D
  0603551D 0E041604 14422141 8BAB16E9 B5F101F0 2AA6B00E 6B618D83 BA300D06
  092A8648 86F70D01 01040500 03818100 905A4A1F 2A2F0BD5 6F3E7634 C6A122D1
  5EB8D0D3 11AC6A79 6C70001C 3E4D1923 6EDE6337 738A9FE1 3B9574C7 CFE4050B
  67CD8790 BC4F8EA1 5C209CB8 8CC175E7 8E27CFFE E3492ADD 874CEBF1 080244B0
  90D92709 B82A7B48 F3EF1DA5 6A32664E 00794B9A 5DBE2060 FDB89303 E0692D39
  4884D825 FB237370 82321596 A079BB75
  quit
username cisco privilege 15 secret 5 $1$sBAo$/JviEK.cbQuLpFFEKRRsb/
!
!
controller T1 0/0/0
 framing esf
 linecode b8zs
 pri-group timeslots 1-24 service mgcp
!
!
!
interface FastEthernet0/0
 description IP VOICE SUBNET
 ip address 172.27.128.2 255.255.255.0
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0:23
 no ip address
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.27.128.1
!
!
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
access-list 23 permit 10.10.10.0 0.0.0.7
!
!
!
control-plane
!
!
!
voice-port 0/0/0:23
 input gain -3
 output attenuation 3
 cptone C1
!
voice-port 0/1/0
 input gain -3
 cptone C1
 timing hookflash-out 50
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
ccm-manager fallback-mgcp
ccm-manager redundant-host 172.28.2.5
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 172.29.128.4
ccm-manager config
ccm-manager download-tones
!
mgcp
mgcp call-agent 172.29.128.5 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
no mgcp package-capability res-package
mgcp package-capability sst-package
no mgcp package-capability fxr-package
mgcp package-capability pre-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax rate 9600
mgcp rtp payload-type g726r16 static
!
mgcp profile default
!
!
!
dial-peer voice 100 pots
 incoming called-number .
 direct-inward-dial
 port 0/0/0:23
!
dial-peer voice 101 pots
 incoming called-number .
 direct-inward-dial
!
dial-peer voice 103 pots
 preference 2
 destination-pattern 9[2-9]......
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 direct-inward-dial
 port 0/0/0:23
 forward-digits 7
!
dial-peer voice 104 pots
 preference 1
 destination-pattern 91[2-9]..[2-9]......
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 direct-inward-dial
 port 0/0/0:23
 forward-digits 11
!
dial-peer voice 107 pots
 preference 2
 destination-pattern 9[369]11
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 incoming called-number .
 direct-inward-dial
 forward-digits 3
!
dial-peer voice 110 pots
 preference 1
 destination-pattern 9011T
 incoming called-number .
 direct-inward-dial
 port 0/0/0:23
 prefix 011
!
dial-peer voice 111 pots
 preference 2
 destination-pattern 9011T
 incoming called-number .
 direct-inward-dial
 port 0/0/0:23
 prefix 011
!
dial-peer voice 112 pots
 preference 2
 destination-pattern 9972.......
 incoming called-number .
 direct-inward-dial
 port 0/0/0:23
 forward-digits 10
!
dial-peer voice 113 pots
 preference 2
 destination-pattern 9214.......
 incoming called-number .
 direct-inward-dial
 port 0/0/0:23
 forward-digits 10
!
dial-peer voice 114 pots
 preference 2
 destination-pattern 9469.......
 incoming called-number .
 direct-inward-dial
 port 0/0/0:23
 prefix 011
!
dial-peer voice 115 pots
 preference 2
 destination-pattern 9817.......
 incoming called-number .
 direct-inward-dial
 port 0/0/0:23
 forward-digits 10
!
dial-peer voice 999010 pots
 service mgcpapp
 port 0/1/0
!
dial-peer voice 116 pots
 preference 2
 destination-pattern 911
 incoming called-number .
 direct-inward-dial
 port 0/1/0
 forward-digits 3
!
dial-peer voice 117 pots
 preference 2
 destination-pattern 9911
 incoming called-number .
 direct-inward-dial
 port 0/1/0
 forward-digits 3
!
dial-peer voice 102 pots
 preference 1
 destination-pattern 9[2-9]......
 progress_ind alert enable 8
 progress_ind progress enable 8
 progress_ind connect enable 8
 direct-inward-dial
 port 0/0/0:23
 forward-digits 7
!
dial-peer voice 993310 pots
 service mgcpapp
 port 0/0/0:23
!
dial-peer voice 993311 pots
 service mgcpapp
 port 0/1/0
!
!
!
call-manager-fallback
 max-conferences 8 gain -6
 timeouts interdigit 5
 ip source-address 172.27.128.2 port 2000
 max-ephones 42
 max-dn 144
 default-destination 6712
 voicemail 8000
!
banner login ^CCC
-----------------------------------------------------------------------
This system and its data contents are the property of Pinky the Wonder Sperm
 and/or its subsidiaries,and contains confidential and
proprietary information.  Any unauthorized use of this system or data
is strictly prohibited and may subject you to disciplinary action, up
to and including termination, civil penalties, and/or criminal
prosecution and penalties.  Users have no expectation of privacy
relating to the use of the system.  By using the system, you expressly
consent to the monitoring and review of your use of the system and all
data viewed, entered, created, received, stored, or transmitted through
the system.
-----------------------------------------------------------------------
^C
!
line con 0
 exec-timeout 0 0
 password
 logging synchronous
 login
line aux 0


On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> not sure if this is what's happening in your case, but i had a default route
> in my original setup that caused a delay. i had to put some ACLs on it to
> make sure it didn't try to connect to the ccm server over a different route
> (which was blocked somewhere else).
>
> just a thought.
>
> maybe you can post your config.
>
> ----- Original Message -----
> From: "Jonathan Charles" <jonvoip at gmail.com>
> To: "ciscovoip" <cisco-voip at puck.nether.net>
> Sent: Friday, May 25, 2007 8:34 PM
> Subject: [cisco-voip] MGCP Fallback time...
>
>
> >I have a new SRST router (MGCP back to CCM)...
> >
> > When we yank the WAN, it takes about 13 minutes for the router to
> > realize that CCM is down and kill the PRI... and revert to H.323 mode.
> >
> > If you look at the attached log file, you can see that the phones are
> > registering to the router, and a full 13 minutes later the PRI goes
> > down...
> >
> > Is there a way to change this to make it quicker?
> >
> > IOS is 12.4.3g
> >
> >
> >
> >
> > Jonathan
> >
>
>
> --------------------------------------------------------------------------------
>
>
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>

From lelio at uoguelph.ca  Fri May 25 22:10:29 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 25 May 2007 22:10:29 -0400
Subject: [cisco-voip] MGCP Fallback time...
References: <5d093f9a0705251734n71238a7ch7cbcb676e9de9c7c@mail.gmail.com>
	<001101c79f2f$646e0680$c901a8c0@pdp11>
	<5d093f9a0705251757s13fc34abr3c48fa32c265472f@mail.gmail.com>
Message-ID: <000c01c79f3b$089a8660$47196883@cfs.uoguelph.ca>

well, your config is darn close to mine. here's mine to see the difference. you might want to try a few things I have? timeout under the call-manager-fallback?

vgw-kemptville#show running-config
Building configuration...

Current configuration : 9987 bytes
!
version 12.4
service timestamps debug datetime localtime
service timestamps log datetime localtime
service password-encryption
!
hostname vgw-kemptville
!
boot-start-marker
boot system flash:c3845-spservicesk9-mz.124-11.T2.bin
boot-end-marker
!
card type t1 0 0
card type t1 0 1
logging buffered 4096
logging console informational
enable secret <snip>
!
aaa new-model
!
!
aaa authentication login default local group tacacs+ enable
aaa authentication login virt-users group tacacs+ enable
aaa authentication login console group tacacs+ enable
!
!
aaa session-id common
clock timezone EST -5
clock summer-time EDT recurring
network-clock-participate wic 0
network-clock-participate wic 1
no ip dhcp use vrf connected
ip dhcp excluded-address 10.104.126.99
ip dhcp excluded-address 10.104.126.100
!
ip dhcp pool dhcp_vlan826
   network 10.104.126.0 255.255.255.128
   option 150 ip 10.104.21.121
   default-router 10.104.126.1
!
!
ip cef
!
!
multilink bundle-name authenticated
!
isdn switch-type primary-dms100
voice-card 0
 no dspfarm
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
voice translation-rule 613
 rule 1 /\(.+\)/ /9\1/
!
voice translation-rule 900
 rule 1 /^91809......./ /918090000000/
 rule 2 /^91829......./ /918290000000/
 rule 3 /^91900......./ /919000000000/
 rule 4 /^91...976..../ /919760000000/
 rule 5 /^91976..../ /919760000000/
 rule 6 /^9976..../ /919760000000/
!
voice translation-rule 1112
 rule 1 reject /914165551212/
!
!
voice translation-profile 1112
!
voice translation-profile 999
 translate called 1112
!
voice translation-profile kc_did_profile
 translate called 613
!
voice translation-profile uogdev_block_highrisk_900_vtp
 translate called 900
!
!
!
crypto pki trustpoint TP-self-signed-3156109335
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-3156109335
 revocation-check none
 rsakeypair TP-self-signed-3156109335
!
!
crypto pki certificate chain TP-self-signed-3156109335
 certificate self-signed 01
<snip>
  quit
!
!
vtp domain 3845
vtp mode transparent
!
!
controller T1 0/0/0
 framing esf
 linecode b8zs
 pri-group timeslots 1-24 service mgcp
!
controller T1 0/1/0
 framing esf
 linecode b8zs
 pri-group timeslots 1-24 service mgcp
!
vlan 129
!
vlan 725
 name voice_10.104.125.0/25
!
vlan 826
 name voice_10.104.126.0/25
!
translation-rule 1112
!
!
!
!
!
interface GigabitEthernet0/0
 description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
 no ip address
 duplex auto
 speed auto
 media-type rj45
 no keepalive
!
interface GigabitEthernet0/0.129
 encapsulation dot1Q 129 native
 ip address 192.168.129.66 255.255.255.0
 ip access-group vlan129_in in
 ip access-group vlan129_out out
!
interface GigabitEthernet0/0.725
 encapsulation dot1Q 725
 ip address 10.104.125.99 255.255.255.128
!
interface GigabitEthernet0/0.826
 encapsulation dot1Q 826
 ip address 10.104.126.1 255.255.255.128
!
interface Service-Engine0/0
 ip unnumbered GigabitEthernet0/0.826
 service-module ip address 10.104.126.100 255.255.255.128
 service-module ip default-gateway 10.104.126.1
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
 media-type rj45
 no keepalive
!
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-dms100
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable
!
interface Serial0/1/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-dms100
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable
!
ip default-gateway 192.168.129.1
ip route 0.0.0.0 0.0.0.0 192.168.129.1
ip route 10.0.0.0 255.0.0.0 10.104.125.1
ip route 10.104.126.100 255.255.255.255 Service-Engine0/0
!
!
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
ip access-list extended vlan129_in
 deny   ip any 10.0.0.0 0.255.255.255
 deny   ip 10.0.0.0 0.255.255.255 any
 permit ip any any
ip access-list extended vlan129_out
 deny   ip 10.0.0.0 0.255.255.255 any
 deny   ip any 10.0.0.0 0.255.255.255
 permit ip any any
!
logging 192.168.20.5
!
!
tacacs-server host 192.168.20.5
tacacs-server directed-request
tacacs-server key <snip>
!
control-plane
!
!
!
voice-port 0/0/0:23
!
voice-port 0/1/0:23
!
ccm-manager fallback-mgcp
ccm-manager redundant-host 10.104.21.109
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.104.21.109
ccm-manager config
!
mgcp
mgcp call-agent 10.104.21.121 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
no mgcp package-capability res-package
mgcp package-capability sst-package
no mgcp package-capability fxr-package
mgcp package-capability pre-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp rtp payload-type g726r16 static
!
mgcp profile default
!
!
dial-peer cor custom
 name uog_oncampus_pt
 name uog_local_pt
 name uog_freeservices_pt
 name uog_tollfree_pt
 name uog_payservices_pt
 name uog_longdist_pt
 name uog_block_highrisk_pt
!
!
dial-peer cor list uogres-local-css
 member uog_oncampus_pt
 member uog_local_pt
 member uog_tollfree_pt
!
dial-peer cor list uogbus-oncampus-css
 member uog_oncampus_pt
!
dial-peer cor list uogbus-local-css
 member uog_oncampus_pt
 member uog_local_pt
 member uog_freeservices_pt
 member uog_tollfree_pt
!
dial-peer cor list uogbus-longdist-css
 member uog_oncampus_pt
 member uog_local_pt
 member uog_freeservices_pt
 member uog_tollfree_pt
 member uog_payservices_pt
 member uog_longdist_pt
 member uog_block_highrisk_pt
!
dial-peer cor list uogdev-local-css
 member uog_local_pt
!
dial-peer cor list uogdev-freeservices-css
 member uog_freeservices_pt
!
dial-peer cor list uogdev-tollfree-css
 member uog_tollfree_pt
!
dial-peer cor list uogdev-payservices-css
 member uog_payservices_pt
!
dial-peer cor list uogdev-longdist-css
 member uog_longdist_pt
!
dial-peer cor list uogdev-block-highrisk-css
 member uog_block_highrisk_pt
!
!
dial-peer voice 929 pots
 destination-pattern 9[2-9].........
 clid network-number 5198241212
 port 0/1/0:23
 forward-digits 10
!
dial-peer voice 61901 voip
 destination-pattern 61901
 session protocol sipv2
 session target ipv4:10.104.126.100
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!
dial-peer voice 98240022 pots
 destination-pattern 95198240022
 port 0/0/0:23
 forward-digits 10
!
dial-peer voice 1 pots
 translation-profile incoming kc_did_profile
 incoming called-number .
 direct-inward-dial
 port 0/0/0:23
!
dial-peer voice 2 pots
 translation-profile incoming kc_did_profile
 incoming called-number .
 direct-inward-dial
 port 0/1/0:23
!
dial-peer voice 9767 pots
 destination-pattern 95197671060
 port 0/1/0:23
 forward-digits 10
!
dial-peer voice 9129902 pots
 translation-profile outgoing uogdev_block_highrisk_900_vtp
 destination-pattern 91[2-9].........
 forward-digits 11
!
dial-peer voice 99901 pots
 translation-profile incoming 999
 translation-profile outgoing 999
 destination-pattern 914165551212
 port 0/1/0:23
 forward-digits 11
!
!
num-exp 96132589... 61...
num-exp 61000 61901
num-exp 96132581515 61901
num-exp 96132581212 61004
num-exp 96135551414 61901
num-exp 95198400105 61009
!
call-manager-fallback
 secondary-dialtone 9
 max-conferences 4 gain -6
 transfer-system local-consult
 limit-dn 7910 1
 limit-dn 7935 1
 limit-dn 7940 2
 limit-dn 7960 6
 limit-dn 7970 8
 timeouts interdigit 15
 ip source-address 10.104.126.1 port 2000 strict-match
 max-ephones 336
 max-dn 500 dual-line preference 5
 system message primary UofG Link Down. Ltd Service
 keepalive 60
 time-zone 12
 time-format 24
 cor incoming uogres-local-css default
 cor incoming uogbus-longdist-css 1 61001 - 61999
 cor incoming uogbus-oncampus-css 2 61101
!
banner login ^CAccess Restricted to CCS Staff Only. All Access Att

vgw-kemptville#

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: Jonathan Charles 
  To: Lelio Fulgenzi 
  Cc: ciscovoip 
  Sent: Friday, May 25, 2007 8:57 PM
  Subject: Re: [cisco-voip] MGCP Fallback time...


  Here it is...


  Current configuration : 7664 bytes
  !
  version 12.4
  service timestamps debug datetime msec
  service timestamps log datetime msec
  no service password-encryption
  !
  hostname PINKY1
  !
  boot-start-marker
  boot-end-marker
  !
  logging buffered 1000000 warnings
  enable secret 5 $1$OGFk$TCpZe0TS9B4anq.Q6MJIQ0
  !
  no aaa new-model
  !
  resource policy
  !
  network-clock-participate wic 0
  ip subnet-zero
  !
  !
  ip cef
  !
  !
  no ip domain lookup
  ip domain name amli.com
  isdn switch-type primary-ni
  !
  voice-card 0
   no dspfarm
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  !
  application
   global
    service alternate Default
   !
  !
  !
  crypto pki trustpoint TP-self-signed-2413116866
   enrollment selfsigned
   subject-name cn=IOS-Self-Signed-Certificate-2413116866
   revocation-check none
   rsakeypair TP-self-signed-2413116866
  !
  !
  crypto pki certificate chain TP-self-signed-2413116866
   certificate self-signed 01
    3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
    31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
    69666963 6174652D 32343133 31313638 3636301E 170D3037 30353233 31363136
    35385A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
    4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 34313331
    31363836 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
    8100C7DB 31404006 EEE793F7 398F1A35 07509CDB D444F92D 79FF5F4A 73A01D89
    B77EF538 DF19058C CCE57B9C ABF0597B DD60FAE8 3CD96D2E DD46A581 15B15F38
    DC3368D4 074A66F7 9E484DED E36394F6 05EA9FDA F81632FC 31D90520 256D497E
    CE8F91E4 042BD4E6 824C0DE6 F3107769 F0FFA0E0 65321D02 025D7F33 65C6DDE4
    027F0203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603
    551D1104 18301682 14414D4C 4944414C 32383131 2E616D6C 692E636F 6D301F06
    03551D23 04183016 80144221 418BAB16 E9B5F101 F02AA6B0 0E6B618D 83BA301D
    0603551D 0E041604 14422141 8BAB16E9 B5F101F0 2AA6B00E 6B618D83 BA300D06
    092A8648 86F70D01 01040500 03818100 905A4A1F 2A2F0BD5 6F3E7634 C6A122D1
    5EB8D0D3 11AC6A79 6C70001C 3E4D1923 6EDE6337 738A9FE1 3B9574C7 CFE4050B
    67CD8790 BC4F8EA1 5C209CB8 8CC175E7 8E27CFFE E3492ADD 874CEBF1 080244B0
    90D92709 B82A7B48 F3EF1DA5 6A32664E 00794B9A 5DBE2060 FDB89303 E0692D39
    4884D825 FB237370 82321596 A079BB75
    quit
  username cisco privilege 15 secret 5 $1$sBAo$/JviEK.cbQuLpFFEKRRsb/
  !
  !
  controller T1 0/0/0
   framing esf
   linecode b8zs
   pri-group timeslots 1-24 service mgcp
  !
  !
  !
  interface FastEthernet0/0
   description IP VOICE SUBNET
   ip address 172.27.128.2 255.255.255.0
   duplex auto
   speed auto
  !
  interface FastEthernet0/1
   no ip address
   shutdown
   duplex auto
   speed auto
  !
  interface Serial0/0/0:23
   no ip address
   isdn switch-type primary-ni
   isdn incoming-voice voice
   isdn bind-l3 ccm-manager
   no cdp enable
  !
  ip classless
  ip route 0.0.0.0 0.0.0.0 172.27.128.1
  !
  !
  ip http server
  ip http access-class 23
  ip http authentication local
  ip http secure-server
  ip http timeout-policy idle 60 life 86400 requests 10000
  !
  access-list 23 permit 10.10.10.0 0.0.0.7
  !
  !
  !
  control-plane
  !
  !
  !
  voice-port 0/0/0:23
   input gain -3
   output attenuation 3
   cptone C1
  !
  voice-port 0/1/0
   input gain -3
   cptone C1
   timing hookflash-out 50
  !
  voice-port 0/1/1
  !
  voice-port 0/1/2
  !
  voice-port 0/1/3
  !
  ccm-manager fallback-mgcp
  ccm-manager redundant-host 172.28.2.5
  ccm-manager mgcp
  ccm-manager music-on-hold
  ccm-manager config server 172.29.128.4
  ccm-manager config
  ccm-manager download-tones
  !
  mgcp
  mgcp call-agent 172.29.128.5 2427 service-type mgcp version 0.1
  mgcp dtmf-relay voip codec all mode out-of-band
  mgcp rtp unreachable timeout 1000 action notify
  mgcp modem passthrough voip mode nse
  mgcp package-capability rtp-package
  no mgcp package-capability res-package
  mgcp package-capability sst-package
  no mgcp package-capability fxr-package
  mgcp package-capability pre-package
  no mgcp timer receive-rtcp
  mgcp sdp simple
  mgcp fax rate 9600
  mgcp rtp payload-type g726r16 static
  !
  mgcp profile default
  !
  !
  !
  dial-peer voice 100 pots
   incoming called-number .
   direct-inward-dial
   port 0/0/0:23
  !
  dial-peer voice 101 pots
   incoming called-number .
   direct-inward-dial
  !
  dial-peer voice 103 pots
   preference 2
   destination-pattern 9[2-9]......
   progress_ind alert enable 8
   progress_ind progress enable 8
   progress_ind connect enable 8
   direct-inward-dial
   port 0/0/0:23
   forward-digits 7
  !
  dial-peer voice 104 pots
   preference 1
   destination-pattern 91[2-9]..[2-9]......
   progress_ind alert enable 8
   progress_ind progress enable 8
   progress_ind connect enable 8
   direct-inward-dial
   port 0/0/0:23
   forward-digits 11
  !
  dial-peer voice 107 pots
   preference 2
   destination-pattern 9[369]11
   progress_ind alert enable 8
   progress_ind progress enable 8
   progress_ind connect enable 8
   incoming called-number .
   direct-inward-dial
   forward-digits 3
  !
  dial-peer voice 110 pots
   preference 1
   destination-pattern 9011T
   incoming called-number .
   direct-inward-dial
   port 0/0/0:23
   prefix 011
  !
  dial-peer voice 111 pots
   preference 2
   destination-pattern 9011T
   incoming called-number .
   direct-inward-dial
   port 0/0/0:23
   prefix 011
  !
  dial-peer voice 112 pots
   preference 2
   destination-pattern 9972.......
   incoming called-number .
   direct-inward-dial
   port 0/0/0:23
   forward-digits 10
  !
  dial-peer voice 113 pots
   preference 2
   destination-pattern 9214.......
   incoming called-number .
   direct-inward-dial
   port 0/0/0:23
   forward-digits 10
  !
  dial-peer voice 114 pots
   preference 2
   destination-pattern 9469.......
   incoming called-number .
   direct-inward-dial
   port 0/0/0:23
   prefix 011
  !
  dial-peer voice 115 pots
   preference 2
   destination-pattern 9817.......
   incoming called-number .
   direct-inward-dial
   port 0/0/0:23
   forward-digits 10
  !
  dial-peer voice 999010 pots
   service mgcpapp
   port 0/1/0
  !
  dial-peer voice 116 pots
   preference 2
   destination-pattern 911
   incoming called-number .
   direct-inward-dial
   port 0/1/0
   forward-digits 3
  !
  dial-peer voice 117 pots
   preference 2
   destination-pattern 9911
   incoming called-number .
   direct-inward-dial
   port 0/1/0
   forward-digits 3
  !
  dial-peer voice 102 pots
   preference 1
   destination-pattern 9[2-9]......
   progress_ind alert enable 8
   progress_ind progress enable 8
   progress_ind connect enable 8
   direct-inward-dial
   port 0/0/0:23
   forward-digits 7
  !
  dial-peer voice 993310 pots
   service mgcpapp
   port 0/0/0:23
  !
  dial-peer voice 993311 pots
   service mgcpapp
   port 0/1/0
  !
  !
  !
  call-manager-fallback
   max-conferences 8 gain -6
   timeouts interdigit 5
   ip source-address 172.27.128.2 port 2000
   max-ephones 42
   max-dn 144
   default-destination 6712
   voicemail 8000
  !
  banner login ^CCC
  -----------------------------------------------------------------------
  This system and its data contents are the property of Pinky the Wonder Sperm
   and/or its subsidiaries,and contains confidential and
  proprietary information.  Any unauthorized use of this system or data
  is strictly prohibited and may subject you to disciplinary action, up
  to and including termination, civil penalties, and/or criminal
  prosecution and penalties.  Users have no expectation of privacy
  relating to the use of the system.  By using the system, you expressly
  consent to the monitoring and review of your use of the system and all
  data viewed, entered, created, received, stored, or transmitted through
  the system.
  -----------------------------------------------------------------------
  ^C
  !
  line con 0
   exec-timeout 0 0
   password
   logging synchronous
   login
  line aux 0


  On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  > not sure if this is what's happening in your case, but i had a default route
  > in my original setup that caused a delay. i had to put some ACLs on it to
  > make sure it didn't try to connect to the ccm server over a different route
  > (which was blocked somewhere else).
  >
  > just a thought.
  >
  > maybe you can post your config.
  >
  > ----- Original Message -----
  > From: "Jonathan Charles" <jonvoip at gmail.com>
  > To: "ciscovoip" <cisco-voip at puck.nether.net>
  > Sent: Friday, May 25, 2007 8:34 PM
  > Subject: [cisco-voip] MGCP Fallback time...
  >
  >
  > >I have a new SRST router (MGCP back to CCM)...
  > >
  > > When we yank the WAN, it takes about 13 minutes for the router to
  > > realize that CCM is down and kill the PRI... and revert to H.323 mode.
  > >
  > > If you look at the attached log file, you can see that the phones are
  > > registering to the router, and a full 13 minutes later the PRI goes
  > > down...
  > >
  > > Is there a way to change this to make it quicker?
  > >
  > > IOS is 12.4.3g
  > >
  > >
  > >
  > >
  > > Jonathan
  > >
  >
  >
  > --------------------------------------------------------------------------------
  >
  >
  > > _______________________________________________
  > > cisco-voip mailing list
  > > cisco-voip at puck.nether.net
  > > https://puck.nether.net/mailman/listinfo/cisco-voip
  > >
  >
  >
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From keli at carocomp.ro  Sat May 26 02:31:28 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Sat, 26 May 2007 09:31:28 +0300
Subject: [cisco-voip] [c-nsp] Yanking Compact Flash while the router
	is	up...
In-Reply-To: <03f201c79efe$642d4a40$680a0a0a@oneunified.local>
References: <5d093f9a0705251145k63365279t98cb0bb490aa1360@mail.gmail.com>
	<03f201c79efe$642d4a40$680a0a0a@oneunified.local>
Message-ID: <20070526093128.cii4yg3mas88ccc8@mail.carocomp.ro>

I'd definitely go with planned downtime, instead of experimenting, but  
if you have a prepared flash (same IOS, same configs, same files), it  
might just work.

AFAIK a running router shouldn't really use it's flash. [for the short  
time, while you change it] Whatever is used from the IOS should be  
preloaded in memory at boot (I know a *nix kernel behaves like that, I  
suppose something similar from the IOS, based on the boot-up sequence).

But if your router is so critical, that you don't want to power it off  
for the flash change, I'd experiment on a different router first ... ;)

regards,
   Zoltan

Quoting Ray Burkholder <ray at oneunified.net>:

> Some machines have enough flash built in to accept downgraded IOS.  Swap the
> external flash.  Power up, then upgrade the IOS again.
>
>> -----Original Message-----
>> From: cisco-nsp-bounces at puck.nether.net
>> [mailto:cisco-nsp-bounces at puck.nether.net] On Behalf Of
>> Jonathan Charles
>> Sent: Friday, May 25, 2007 15:45
>> To: ciscovoip; cisco-nsp at puck.nether.net
>> Subject: [c-nsp] Yanking Compact Flash while the router is up...
>>
>> So, I am curious, can you pull a CF card from a 2821 while
>> the router is up and running?
>>
>> We need to replace flaky flash with a new one, but I would
>> rather not have to talk a remote tech (low skillset) through
>> a ROMMON TFTP operation..
>>
>> If not, we can do it the hard way....but I have heard rumors
>> that it works...
>>
>> If it does, what are the risks...
>>
>>
>>
>> Jonathan
>> _______________________________________________
>> cisco-nsp mailing list  cisco-nsp at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-nsp
>> archive at http://puck.nether.net/pipermail/cisco-nsp/
>>
>> --
>> Scanned for viruses and dangerous content at
>> http://www.oneunified.net and is believed to be clean.
>>
>>
>
>
> --
> Scanned for viruses and dangerous content at
> http://www.oneunified.net and is believed to be clean.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.


From keli at carocomp.ro  Sat May 26 05:01:21 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Sat, 26 May 2007 12:01:21 +0300
Subject: [cisco-voip] 7911 factory default reset
Message-ID: <20070526120121.o9zvjos0aos8ckwk@mail.carocomp.ro>

Hi!

I have a 7911G phone, with POE, that had problems starting up - got  
stock in the DHCP phase. It seemd like bad cabling, but it behaved the  
same plugged directly into the switch (Cisco POE switch).

So I tried a factory reset, but it won't recover. I'm not sure wether  
I completed the procedure correctly though:
  - reset the phone with the # pressed
  - then press 123456789*#0 in sequence

The prone now enters the "Upgrading" phase, reboots, with a circle  
icon (bullet?) in the lower left corner (where normally a checked  
checkbox resides) then it begins the upgradeing proces all over again.

In the mean time, it recieves an IP adress from DHCP and can be pinged  
for a shor while.

How do I exit this loop?

thanks,
   Zoltan


----------------------------------------------------------------
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From asobihoudai at yahoo.com  Sat May 26 10:55:00 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Sat, 26 May 2007 07:55:00 -0700 (PDT)
Subject: [cisco-voip] 7911 factory default reset
In-Reply-To: <20070526120121.o9zvjos0aos8ckwk@mail.carocomp.ro>
Message-ID: <772285.4342.qm@web57204.mail.re3.yahoo.com>

Can your phone talk to the tftp server properly?


--- keli at carocomp.ro wrote:

> Hi!
> 
> I have a 7911G phone, with POE, that had problems
> starting up - got  
> stock in the DHCP phase. It seemd like bad cabling,
> but it behaved the  
> same plugged directly into the switch (Cisco POE
> switch).
> 
> So I tried a factory reset, but it won't recover.
> I'm not sure wether  
> I completed the procedure correctly though:
>   - reset the phone with the # pressed
>   - then press 123456789*#0 in sequence
> 
> The prone now enters the "Upgrading" phase, reboots,
> with a circle  
> icon (bullet?) in the lower left corner (where
> normally a checked  
> checkbox resides) then it begins the upgradeing
> proces all over again.
> 
> In the mean time, it recieves an IP adress from DHCP
> and can be pinged  
> for a shor while.
> 
> How do I exit this loop?
> 
> thanks,
>    Zoltan
> 



 
____________________________________________________________________________________
Don't pick lemons.
See all the new 2007 cars at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html 

From jonvoip at gmail.com  Sat May 26 11:36:57 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 26 May 2007 10:36:57 -0500
Subject: [cisco-voip] [c-nsp] Yanking Compact Flash while the router is
	up...
In-Reply-To: <20070526093128.cii4yg3mas88ccc8@mail.carocomp.ro>
References: <5d093f9a0705251145k63365279t98cb0bb490aa1360@mail.gmail.com>
	<03f201c79efe$642d4a40$680a0a0a@oneunified.local>
	<20070526093128.cii4yg3mas88ccc8@mail.carocomp.ro>
Message-ID: <5d093f9a0705260836s2bc909c1q497c49cfaac593f1@mail.gmail.com>

Well, we just yanked the flash, stuffed in the new one (it took a good
6 seconds for it to recognize the new flash) and copied the new
code... rebooted... total network downtime, about three minutes.



Jonathan

On 5/26/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
> I'd definitely go with planned downtime, instead of experimenting, but
> if you have a prepared flash (same IOS, same configs, same files), it
> might just work.
>
> AFAIK a running router shouldn't really use it's flash. [for the short
> time, while you change it] Whatever is used from the IOS should be
> preloaded in memory at boot (I know a *nix kernel behaves like that, I
> suppose something similar from the IOS, based on the boot-up sequence).
>
> But if your router is so critical, that you don't want to power it off
> for the flash change, I'd experiment on a different router first ... ;)
>
> regards,
>    Zoltan
>
> Quoting Ray Burkholder <ray at oneunified.net>:
>
> > Some machines have enough flash built in to accept downgraded IOS.  Swap the
> > external flash.  Power up, then upgrade the IOS again.
> >
> >> -----Original Message-----
> >> From: cisco-nsp-bounces at puck.nether.net
> >> [mailto:cisco-nsp-bounces at puck.nether.net] On Behalf Of
> >> Jonathan Charles
> >> Sent: Friday, May 25, 2007 15:45
> >> To: ciscovoip; cisco-nsp at puck.nether.net
> >> Subject: [c-nsp] Yanking Compact Flash while the router is up...
> >>
> >> So, I am curious, can you pull a CF card from a 2821 while
> >> the router is up and running?
> >>
> >> We need to replace flaky flash with a new one, but I would
> >> rather not have to talk a remote tech (low skillset) through
> >> a ROMMON TFTP operation..
> >>
> >> If not, we can do it the hard way....but I have heard rumors
> >> that it works...
> >>
> >> If it does, what are the risks...
> >>
> >>
> >>
> >> Jonathan
> >> _______________________________________________
> >> cisco-nsp mailing list  cisco-nsp at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-nsp
> >> archive at http://puck.nether.net/pipermail/cisco-nsp/
> >>
> >> --
> >> Scanned for viruses and dangerous content at
> >> http://www.oneunified.net and is believed to be clean.
> >>
> >>
> >
> >
> > --
> > Scanned for viruses and dangerous content at
> > http://www.oneunified.net and is believed to be clean.
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Sat May 26 11:46:05 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 26 May 2007 10:46:05 -0500
Subject: [cisco-voip] CallManager 5.2
Message-ID: <5d093f9a0705260846k4ffe1c6cra465694ab14dab9c@mail.gmail.com>

So, anyone upgrade yet?

Just curious if it is going to kill QoS or RTP streams... TAC said it
was just bug fixes...



Jonathan

From jonvoip at gmail.com  Sat May 26 11:52:17 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 26 May 2007 10:52:17 -0500
Subject: [cisco-voip] CallManager 5.2
In-Reply-To: <5d093f9a0705260846k4ffe1c6cra465694ab14dab9c@mail.gmail.com>
References: <5d093f9a0705260846k4ffe1c6cra465694ab14dab9c@mail.gmail.com>
Message-ID: <5d093f9a0705260852i139101d5w7756f41fabac219a@mail.gmail.com>

Correction, 5.1.2... not 5.2



Jonathan

On 5/26/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> So, anyone upgrade yet?
>
> Just curious if it is going to kill QoS or RTP streams... TAC said it
> was just bug fixes...
>
>
>
> Jonathan
>

From jonvoip at gmail.com  Sat May 26 12:01:27 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 26 May 2007 11:01:27 -0500
Subject: [cisco-voip] Upgrade to 5.1(2)
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451F11@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451F11@CSEXCH01.charles-stanley.co.uk>
Message-ID: <5d093f9a0705260901l625f347dnea83faf3187d3122@mail.gmail.com>

I am upgrading a test box right now... It was recommended by TAC last night
(issue with conferencing...)...


Jonathan

On 5/25/07, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote:
>
>  Has anyone upgraded to 5.1(2) yet, just wondered if anyone had any issues
> with it.
>
>
>
>
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From jonvoip at gmail.com  Sat May 26 12:28:19 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 26 May 2007 11:28:19 -0500
Subject: [cisco-voip] CM 4.1(3)SR5
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC031226C4@CH-EXCH-02.intelltech.apptis.inc>
References: <5d093f9a0705241113t11f38317y1a48bd0c69ada7a2@mail.gmail.com>
	<2F09B2162CCDEA4EA05C089B24FBC2BC031226C4@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <5d093f9a0705260928m4bddcdc6id5859bde8a4198ca@mail.gmail.com>

I just had this happen to me two days ago.

The customer had recently moved and wanted to retain their DIDs...
so... they waived 911 service on their PRI, so 911 calls failed.

Instead they were using an analog line for 911.



Jonathan

On 5/24/07, Derek Hover <Derek.Hover at apptis.com> wrote:
>
> Now that it's out, anyone taking a stick to it yet?
>
> Derek
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Thursday, May 24, 2007 2:13 PM
> To: ciscovoip
> Subject: [cisco-voip] Unallocated/Unassigned Number for 911 call...
>
> So, made a 911 test call:
>
> VOICEGW#
> *May 24 18:13:39.451: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8
> callref = 0x004D
>         Bearer Capability i = 0x8090A2
>                 Standard = CCITT
>                 Transfer Capability = Speech
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA98394
>                 Exclusive, Channel 20
>         Calling Party Number i = 0x0181, '[removed]'
>                 Plan:ISDN, Type:Unknown
>         Called Party Number i = 0x80, '911'
>                 Plan:Unknown, Type:Unknown
> *May 24 18:13:39.527: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8
> callref = 0x804D
>         Cause i = 0x8081 - Unallocated/unassigned number
>
>
> Any ideas?
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Sat May 26 12:34:55 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 26 May 2007 11:34:55 -0500
Subject: [cisco-voip] QOS for MPLS
Message-ID: <5d093f9a0705260934t511a1428k295ab44f88455dd2@mail.gmail.com>

I have always understood that QOS settings need to match on both ends
of a circuit... so, if you allocate 800k on one side, you need to do
the same on the other.

I am now working on an MPLS circuit that has DS3s on some sides, DS1s
on others... do I still need to match QOS settings?

Customer wants 800k on the T1s and 2500k on the DS3s... I don't think
it will matter as I am matching these bw limits on the locations...
but how do I throttle calls from a DS3-side location to a DS1
location?

For example... main site is in Chicago, they have a DS3, one remote
site is in Wichita and has a T1... so, I put 800k for Wichita and
2500k for Chicago... what is to stop Chicago from flooding Wichita
with 2.5MB of calls?



Jonathan

From chris.clouse at berbee.com  Sat May 26 13:29:46 2007
From: chris.clouse at berbee.com (Clouse, Chris)
Date: Sat, 26 May 2007 12:29:46 -0500
Subject: [cisco-voip] QOS for MPLS
In-Reply-To: <5d093f9a0705260934t511a1428k295ab44f88455dd2@mail.gmail.com>
References: <5d093f9a0705260934t511a1428k295ab44f88455dd2@mail.gmail.com>
Message-ID: <310F10740485D9469F0D8417950B252D028971E4@CTG-MSNEXC01.staff.berbee.com>

Use CallManager locations to perform Call Admission Control.

Christopher Clouse, CCNP CCDP CCVP MCP Network+


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Saturday, May 26, 2007 11:35 AM
To: ciscovoip
Subject: [cisco-voip] QOS for MPLS

I have always understood that QOS settings need to match on both ends
of a circuit... so, if you allocate 800k on one side, you need to do
the same on the other.

I am now working on an MPLS circuit that has DS3s on some sides, DS1s
on others... do I still need to match QOS settings?

Customer wants 800k on the T1s and 2500k on the DS3s... I don't think
it will matter as I am matching these bw limits on the locations...
but how do I throttle calls from a DS3-side location to a DS1
location?

For example... main site is in Chicago, they have a DS3, one remote
site is in Wichita and has a T1... so, I put 800k for Wichita and
2500k for Chicago... what is to stop Chicago from flooding Wichita
with 2.5MB of calls?



Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From asobihoudai at yahoo.com  Sat May 26 13:35:20 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Sat, 26 May 2007 10:35:20 -0700 (PDT)
Subject: [cisco-voip] QOS for MPLS
In-Reply-To: <310F10740485D9469F0D8417950B252D028971E4@CTG-MSNEXC01.staff.berbee.com>
Message-ID: <378027.42442.qm@web57206.mail.re3.yahoo.com>

He's talking about QoS for MPLS, not CAC.

--- "Clouse, Chris" <chris.clouse at berbee.com> wrote:

> Use CallManager locations to perform Call Admission
> Control.
> 
> Christopher Clouse, CCNP CCDP CCVP MCP Network+
> 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Jonathan
> Charles
> Sent: Saturday, May 26, 2007 11:35 AM
> To: ciscovoip
> Subject: [cisco-voip] QOS for MPLS
> 
> I have always understood that QOS settings need to
> match on both ends
> of a circuit... so, if you allocate 800k on one
> side, you need to do
> the same on the other.
> 
> I am now working on an MPLS circuit that has DS3s on
> some sides, DS1s
> on others... do I still need to match QOS settings?
> 
> Customer wants 800k on the T1s and 2500k on the
> DS3s... I don't think
> it will matter as I am matching these bw limits on
> the locations...
> but how do I throttle calls from a DS3-side location
> to a DS1
> location?
> 
> For example... main site is in Chicago, they have a
> DS3, one remote
> site is in Wichita and has a T1... so, I put 800k
> for Wichita and
> 2500k for Chicago... what is to stop Chicago from
> flooding Wichita
> with 2.5MB of calls?
> 
> 
> 
> Jonathan




       
____________________________________________________________________________________Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.
http://smallbusiness.yahoo.com/webhosting 

From victor.rodriguez at neoris.com  Sat May 26 22:36:00 2007
From: victor.rodriguez at neoris.com (Victor Rodriguez)
Date: Sat, 26 May 2007 21:36:00 -0500
Subject: [cisco-voip] 7911 factory default reset
In-Reply-To: <772285.4342.qm@web57204.mail.re3.yahoo.com>
Message-ID: <0B435E3251BF7741A5A2B860D2D2412E0C5DEE02@MXMTYMXS02.neoris.cxnetworks.net>

I recommend you upgrade the firmware for you 7911 IPphone and then to this procedure (you must use a AC adaptor)

Power off the phone. 

Hold down the # key. 

While holding #, connect an AC adaptor to the phone. 

Continue to hold # until the line buttons blink amber. 

Release the # key. 

Enter 3491672850*# 

The lights should change color from amber to red. When this occurs, the phone is resetting to factory defaults.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: S?bado, 26 de Mayo de 2007 09:55 a.m.
To: keli at carocomp.ro; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7911 factory default reset

Can your phone talk to the tftp server properly?


--- keli at carocomp.ro wrote:

> Hi!
> 
> I have a 7911G phone, with POE, that had problems
> starting up - got  
> stock in the DHCP phase. It seemd like bad cabling,
> but it behaved the  
> same plugged directly into the switch (Cisco POE
> switch).
> 
> So I tried a factory reset, but it won't recover.
> I'm not sure wether  
> I completed the procedure correctly though:
>   - reset the phone with the # pressed
>   - then press 123456789*#0 in sequence
> 
> The prone now enters the "Upgrading" phase, reboots,
> with a circle  
> icon (bullet?) in the lower left corner (where
> normally a checked  
> checkbox resides) then it begins the upgradeing
> proces all over again.
> 
> In the mean time, it recieves an IP adress from DHCP
> and can be pinged  
> for a shor while.
> 
> How do I exit this loop?
> 
> thanks,
>    Zoltan
> 



 
____________________________________________________________________________________
Don't pick lemons.
See all the new 2007 cars at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html 
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From keli at carocomp.ro  Sun May 27 05:49:13 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Sun, 27 May 2007 12:49:13 +0300
Subject: [cisco-voip] 7911 factory default reset
In-Reply-To: <772285.4342.qm@web57204.mail.re3.yahoo.com>
References: <772285.4342.qm@web57204.mail.re3.yahoo.com>
Message-ID: <20070527124913.swe82qv5hcsogw48@mail.carocomp.ro>

In the mean time, this sort-of turned out to be the problem.

The phone is connected to a CCME, and the router did not have the  
proper firmware files loaded. Once I did that, this phone recovered.

However, since I uploaded a newer firmware, that what was on the  
phones, I restarted all the phones, and now I have a different phone  
going crazy.

It can access the tftp server, it can download all the firmware files,  
but then it just keeps rebooting, with "Resetting..." appearing once  
in a while on the phone. It has this loop, where it's powering up once  
with a checkmark, once with an X in that lower-left corner of the  
screen and keeps rebooting continuously.

It has power, it gets DHCP, it's in the proper VLAN, the PC after it  
is working ok etc ...

In the CCME telephony-service config, what is the proper filename for
load 7911 ????
since I've seen various configs on the net, some claiming the jar  
file, some the SCCP and even some the apps file. Does anybody know for  
sure?

thanks,
   Zoltan

Quoting Paul Choi <asobihoudai at yahoo.com>:

> Can your phone talk to the tftp server properly?
>
>
> --- keli at carocomp.ro wrote:
>
>> Hi!
>>
>> I have a 7911G phone, with POE, that had problems
>> starting up - got
>> stock in the DHCP phase. It seemd like bad cabling,
>> but it behaved the
>> same plugged directly into the switch (Cisco POE
>> switch).
>>
>> So I tried a factory reset, but it won't recover.
>> I'm not sure wether
>> I completed the procedure correctly though:
>>   - reset the phone with the # pressed
>>   - then press 123456789*#0 in sequence
>>
>> The prone now enters the "Upgrading" phase, reboots,
>> with a circle
>> icon (bullet?) in the lower left corner (where
>> normally a checked
>> checkbox resides) then it begins the upgradeing
>> proces all over again.
>>
>> In the mean time, it recieves an IP adress from DHCP
>> and can be pinged
>> for a shor while.
>>
>> How do I exit this loop?
>>
>> thanks,
>>    Zoltan
>>
>
>
>
>
> ____________________________________________________________________________________
> Don't pick lemons.
> See all the new 2007 cars at Yahoo! Autos.
> http://autos.yahoo.com/new_cars.html
>



----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.


From keli at carocomp.ro  Sun May 27 05:56:16 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Sun, 27 May 2007 12:56:16 +0300
Subject: [cisco-voip] QOS for MPLS
In-Reply-To: <378027.42442.qm@web57206.mail.re3.yahoo.com>
References: <378027.42442.qm@web57206.mail.re3.yahoo.com>
Message-ID: <20070527125616.i3lt371psgcos8ow@mail.carocomp.ro>

As long as it is about "calls", CAC is the sensitive way to go about  
limiting the calls in the first place. Do QoS "As close to the source,  
as possible", remember? :)

If you do not have a CCM, but other set-up, you should still look in  
this direction first - to limit the calls to a reasonable amount in  
the given directions, before they are hitting your MPLS router.

regards,
   Zoltan

Quoting Paul Choi <asobihoudai at yahoo.com>:

> He's talking about QoS for MPLS, not CAC.
>
> --- "Clouse, Chris" <chris.clouse at berbee.com> wrote:
>
>> Use CallManager locations to perform Call Admission
>> Control.
>>
>> Christopher Clouse, CCNP CCDP CCVP MCP Network+
>>
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On
>> Behalf Of Jonathan
>> Charles
>> Sent: Saturday, May 26, 2007 11:35 AM
>> To: ciscovoip
>> Subject: [cisco-voip] QOS for MPLS
>>
>> I have always understood that QOS settings need to
>> match on both ends
>> of a circuit... so, if you allocate 800k on one
>> side, you need to do
>> the same on the other.
>>
>> I am now working on an MPLS circuit that has DS3s on
>> some sides, DS1s
>> on others... do I still need to match QOS settings?
>>
>> Customer wants 800k on the T1s and 2500k on the
>> DS3s... I don't think
>> it will matter as I am matching these bw limits on
>> the locations...
>> but how do I throttle calls from a DS3-side location
>> to a DS1
>> location?
>>
>> For example... main site is in Chicago, they have a
>> DS3, one remote
>> site is in Wichita and has a T1... so, I put 800k
>> for Wichita and
>> 2500k for Chicago... what is to stop Chicago from
>> flooding Wichita
>> with 2.5MB of calls?
>>
>>
>>
>> Jonathan
>
>
>
>
>
> ____________________________________________________________________________________Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get   
> online.
> http://smallbusiness.yahoo.com/webhosting
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



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From birnit at nait.ca  Sun May 27 10:57:58 2007
From: birnit at nait.ca (Tkachuk,Birni)
Date: Sun, 27 May 2007 08:57:58 -0600
Subject: [cisco-voip] Mailing List
Message-ID: <8BF2352FAC40504B8D83F9403E4CCE311D0CA1@EXGMETRO.nait.ca>

An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20070527/4ed045a1/attachment.html 

From asobihoudai at yahoo.com  Sun May 27 11:27:53 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Sun, 27 May 2007 08:27:53 -0700 (PDT)
Subject: [cisco-voip] Mailing List
In-Reply-To: <8BF2352FAC40504B8D83F9403E4CCE311D0CA1@EXGMETRO.nait.ca>
Message-ID: <306078.42120.qm@web57211.mail.re3.yahoo.com>

Go to the subscription page for this mailing list and
change your settings.

Paul
--- "Tkachuk,Birni" <birnit at nait.ca> wrote:


---------------------------------
I would like to get only a daily digest of e-mails. 
How do I go about requesting that change?  If not
possible then I will have to request to unsubscribe. 
Thanks.
 

Birni Tkachuk
Voice Services Administrator
Northern Alberta Institute of Technology
 
Phone:  780-471-7532; Fax:  780-471-7755

> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________
Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games.
http://sims.yahoo.com/  

From patrick.diener at gmail.com  Sun May 27 13:10:25 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Sun, 27 May 2007 19:10:25 +0200
Subject: [cisco-voip]  QOS for MPLS
In-Reply-To: <c0deb5b80705271009ne4c18adwf2a2319f85d0f151@mail.gmail.com>
References: <5d093f9a0705260934t511a1428k295ab44f88455dd2@mail.gmail.com>
	<c0deb5b80705271009ne4c18adwf2a2319f85d0f151@mail.gmail.com>
Message-ID: <c0deb5b80705271010g3cab96bcg1fa6546a9e1aa0b2@mail.gmail.com>

You could logically limit the available bandwidth from Chicago to
Wichita on the Chicago side using traffic shaping.
This could be used to limit all traffic voice and data. Inside the
traffic shaping queue voice traffic could be prioritized using CBWFQ

http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/products_feature_guide09186a00800800fc.html

Regards
Patrick

On 5/26/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> I have always understood that QOS settings need to match on both ends
> of a circuit... so, if you allocate 800k on one side, you need to do
> the same on the other.
>
> I am now working on an MPLS circuit that has DS3s on some sides, DS1s
> on others... do I still need to match QOS settings?
>
> Customer wants 800k on the T1s and 2500k on the DS3s... I don't think
> it will matter as I am matching these bw limits on the locations...
> but how do I throttle calls from a DS3-side location to a DS1
> location?
>
> For example... main site is in Chicago, they have a DS3, one remote
> site is in Wichita and has a T1... so, I put 800k for Wichita and
> 2500k for Chicago... what is to stop Chicago from flooding Wichita
> with 2.5MB of calls?
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Sun May 27 16:07:00 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 27 May 2007 15:07:00 -0500
Subject: [cisco-voip] QOS for MPLS
In-Reply-To: <20070527125616.i3lt371psgcos8ow@mail.carocomp.ro>
References: <378027.42442.qm@web57206.mail.re3.yahoo.com>
	<20070527125616.i3lt371psgcos8ow@mail.carocomp.ro>
Message-ID: <5d093f9a0705271307q5b3cf923q1d461ebcf2b5ccbc@mail.gmail.com>

My thinking is that I need to set the QOS to be the same as the
maximum number of calls to/from each site.


Jonathan

On 5/27/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
> As long as it is about "calls", CAC is the sensitive way to go about
> limiting the calls in the first place. Do QoS "As close to the source,
> as possible", remember? :)
>
> If you do not have a CCM, but other set-up, you should still look in
> this direction first - to limit the calls to a reasonable amount in
> the given directions, before they are hitting your MPLS router.
>
> regards,
>    Zoltan
>
> Quoting Paul Choi <asobihoudai at yahoo.com>:
>
> > He's talking about QoS for MPLS, not CAC.
> >
> > --- "Clouse, Chris" <chris.clouse at berbee.com> wrote:
> >
> >> Use CallManager locations to perform Call Admission
> >> Control.
> >>
> >> Christopher Clouse, CCNP CCDP CCVP MCP Network+
> >>
> >>
> >> -----Original Message-----
> >> From: cisco-voip-bounces at puck.nether.net
> >> [mailto:cisco-voip-bounces at puck.nether.net] On
> >> Behalf Of Jonathan
> >> Charles
> >> Sent: Saturday, May 26, 2007 11:35 AM
> >> To: ciscovoip
> >> Subject: [cisco-voip] QOS for MPLS
> >>
> >> I have always understood that QOS settings need to
> >> match on both ends
> >> of a circuit... so, if you allocate 800k on one
> >> side, you need to do
> >> the same on the other.
> >>
> >> I am now working on an MPLS circuit that has DS3s on
> >> some sides, DS1s
> >> on others... do I still need to match QOS settings?
> >>
> >> Customer wants 800k on the T1s and 2500k on the
> >> DS3s... I don't think
> >> it will matter as I am matching these bw limits on
> >> the locations...
> >> but how do I throttle calls from a DS3-side location
> >> to a DS1
> >> location?
> >>
> >> For example... main site is in Chicago, they have a
> >> DS3, one remote
> >> site is in Wichita and has a T1... so, I put 800k
> >> for Wichita and
> >> 2500k for Chicago... what is to stop Chicago from
> >> flooding Wichita
> >> with 2.5MB of calls?
> >>
> >>
> >>
> >> Jonathan
> >
> >
> >
> >
> >
> > ____________________________________________________________________________________Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get
> > online.
> > http://smallbusiness.yahoo.com/webhosting
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From keli at carocomp.ro  Sun May 27 16:50:17 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Sun, 27 May 2007 23:50:17 +0300
Subject: [cisco-voip] QOS for MPLS
In-Reply-To: <5d093f9a0705271307q5b3cf923q1d461ebcf2b5ccbc@mail.gmail.com>
References: <378027.42442.qm@web57206.mail.re3.yahoo.com>
	<20070527125616.i3lt371psgcos8ow@mail.carocomp.ro>
	<5d093f9a0705271307q5b3cf923q1d461ebcf2b5ccbc@mail.gmail.com>
Message-ID: <20070527235017.fzfumpwdg08wss4o@mail.carocomp.ro>

That is correct (although you should take some overhead into account  
as well). However, if you do not perform Call Admission Control  
beforehand (you do not limit the maximum number of possible  
simultaneous calls) you've done nothing, or worse.

The classic example being when you have a line that supports 4 calls,  
when the fifth call enters the line, all calls will suffer packet loss  
and delay, not only the fifth one.

I'm not too versed in MPLS QoS, but the idea is always the same.  
Prioritize voice traffic, but don't let it kill your bandwidth, nor  
itself. Thus the need for CAC, then something like CBWFQ/PQ. If  
possible, voice traffic should be beneath shaping/policing limits, so  
that neither would affect it.

regards,
   Zoltan

Quoting Jonathan Charles <jonvoip at gmail.com>:

> My thinking is that I need to set the QOS to be the same as the
> maximum number of calls to/from each site.
>
>
> Jonathan
>
> On 5/27/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
>> As long as it is about "calls", CAC is the sensitive way to go about
>> limiting the calls in the first place. Do QoS "As close to the source,
>> as possible", remember? :)
>>
>> If you do not have a CCM, but other set-up, you should still look in
>> this direction first - to limit the calls to a reasonable amount in
>> the given directions, before they are hitting your MPLS router.
>>
>> regards,
>>   Zoltan
>>
>> Quoting Paul Choi <asobihoudai at yahoo.com>:
>>
>>> He's talking about QoS for MPLS, not CAC.
>>>
>>> --- "Clouse, Chris" <chris.clouse at berbee.com> wrote:
>>>
>>>> Use CallManager locations to perform Call Admission
>>>> Control.
>>>>
>>>> Christopher Clouse, CCNP CCDP CCVP MCP Network+
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: cisco-voip-bounces at puck.nether.net
>>>> [mailto:cisco-voip-bounces at puck.nether.net] On
>>>> Behalf Of Jonathan
>>>> Charles
>>>> Sent: Saturday, May 26, 2007 11:35 AM
>>>> To: ciscovoip
>>>> Subject: [cisco-voip] QOS for MPLS
>>>>
>>>> I have always understood that QOS settings need to
>>>> match on both ends
>>>> of a circuit... so, if you allocate 800k on one
>>>> side, you need to do
>>>> the same on the other.
>>>>
>>>> I am now working on an MPLS circuit that has DS3s on
>>>> some sides, DS1s
>>>> on others... do I still need to match QOS settings?
>>>>
>>>> Customer wants 800k on the T1s and 2500k on the
>>>> DS3s... I don't think
>>>> it will matter as I am matching these bw limits on
>>>> the locations...
>>>> but how do I throttle calls from a DS3-side location
>>>> to a DS1
>>>> location?
>>>>
>>>> For example... main site is in Chicago, they have a
>>>> DS3, one remote
>>>> site is in Wichita and has a T1... so, I put 800k
>>>> for Wichita and
>>>> 2500k for Chicago... what is to stop Chicago from
>>>> flooding Wichita
>>>> with 2.5MB of calls?
>>>>
>>>>
>>>>
>>>> Jonathan
>>>
>>>
>>>
>>>
>>>
>>>
>> ____________________________________________________________________________________Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to   
>> get
>>> online.
>>> http://smallbusiness.yahoo.com/webhosting
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>>
>>
>> ----------------------------------------------------------------
>> This message was sent using IMP, the Internet Messaging Program.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>



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From erickbe at yahoo.com  Sun May 27 16:59:17 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sun, 27 May 2007 13:59:17 -0700 (PDT)
Subject: [cisco-voip] QOS for MPLS
Message-ID: <262147.28980.qm@web37210.mail.mud.yahoo.com>

I've seen one or two setups where the MPLS was fully meshed (all sites could talk to any other site), which makes QoS even trickier. 

----- Original Message ----
From: "keli at carocomp.ro" <keli at carocomp.ro>
To: Jonathan Charles <jonvoip at gmail.com>
Cc: cisco-voip at puck.nether.net
Sent: Sunday, May 27, 2007 3:50:17 PM
Subject: Re: [cisco-voip] QOS for MPLS

That is correct (although you should take some overhead into account  
as well). However, if you do not perform Call Admission Control  
beforehand (you do not limit the maximum number of possible  
simultaneous calls) you've done nothing, or worse.

The classic example being when you have a line that supports 4 calls,  
when the fifth call enters the line, all calls will suffer packet loss  
and delay, not only the fifth one.

I'm not too versed in MPLS QoS, but the idea is always the same.  
Prioritize voice traffic, but don't let it kill your bandwidth, nor  
itself. Thus the need for CAC, then something like CBWFQ/PQ. If  
possible, voice traffic should be beneath shaping/policing limits, so  
that neither would affect it.

regards,
   Zoltan

Quoting Jonathan Charles <jonvoip at gmail.com>:

> My thinking is that I need to set the QOS to be the same as the
> maximum number of calls to/from each site.
>
>
> Jonathan
>
> On 5/27/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
>> As long as it is about "calls", CAC is the sensitive way to go about
>> limiting the calls in the first place. Do QoS "As close to the source,
>> as possible", remember? :)
>>
>> If you do not have a CCM, but other set-up, you should still look in
>> this direction first - to limit the calls to a reasonable amount in
>> the given directions, before they are hitting your MPLS router.
>>
>> regards,
>>   Zoltan
>>
>> Quoting Paul Choi <asobihoudai at yahoo.com>:
>>
>>> He's talking about QoS for MPLS, not CAC.
>>>
>>> --- "Clouse, Chris" <chris.clouse at berbee.com> wrote:
>>>
>>>> Use CallManager locations to perform Call Admission
>>>> Control.
>>>>
>>>> Christopher Clouse, CCNP CCDP CCVP MCP Network+
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: cisco-voip-bounces at puck.nether.net
>>>> [mailto:cisco-voip-bounces at puck.nether.net] On
>>>> Behalf Of Jonathan
>>>> Charles
>>>> Sent: Saturday, May 26, 2007 11:35 AM
>>>> To: ciscovoip
>>>> Subject: [cisco-voip] QOS for MPLS
>>>>
>>>> I have always understood that QOS settings need to
>>>> match on both ends
>>>> of a circuit... so, if you allocate 800k on one
>>>> side, you need to do
>>>> the same on the other.
>>>>
>>>> I am now working on an MPLS circuit that has DS3s on
>>>> some sides, DS1s
>>>> on others... do I still need to match QOS settings?
>>>>
>>>> Customer wants 800k on the T1s and 2500k on the
>>>> DS3s... I don't think
>>>> it will matter as I am matching these bw limits on
>>>> the locations...
>>>> but how do I throttle calls from a DS3-side location
>>>> to a DS1
>>>> location?
>>>>
>>>> For example... main site is in Chicago, they have a
>>>> DS3, one remote
>>>> site is in Wichita and has a T1... so, I put 800k
>>>> for Wichita and
>>>> 2500k for Chicago... what is to stop Chicago from
>>>> flooding Wichita
>>>> with 2.5MB of calls?
>>>>
>>>>
>>>>
>>>> Jonathan
>>>
>>>
>>>
>>>
>>>
>>>
>> ____________________________________________________________________________________Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to   
>> get
>>> online.
>>> http://smallbusiness.yahoo.com/webhosting
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>>
>>
>> ----------------------------------------------------------------
>> This message was sent using IMP, the Internet Messaging Program.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>



----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip





 
____________________________________________________________________________________
We won't tell. Get more on shows you hate to love 
(and love to hate): Yahoo! TV's Guilty Pleasures list.
http://tv.yahoo.com/collections/265 

From netsecuredata at gmail.com  Sun May 27 17:27:22 2007
From: netsecuredata at gmail.com (Jorge Evangelista)
Date: Sun, 27 May 2007 16:27:22 -0500
Subject: [cisco-voip] QOS for MPLS
In-Reply-To: <20070527235017.fzfumpwdg08wss4o@mail.carocomp.ro>
References: <378027.42442.qm@web57206.mail.re3.yahoo.com>
	<20070527125616.i3lt371psgcos8ow@mail.carocomp.ro>
	<5d093f9a0705271307q5b3cf923q1d461ebcf2b5ccbc@mail.gmail.com>
	<20070527235017.fzfumpwdg08wss4o@mail.carocomp.ro>
Message-ID: <de85c96f0705271427h5580cafasdaa42739c56bea4f@mail.gmail.com>

There are a list of EXP [*] values for MPLS that you could use for
marking. I have read that it marks MPLS and IP tunnel packets by
default during some encapsulation operations.

Cisco says "The set mpls experimental topmost command on an input
policy always marks MPLS packets before the node performs all label
forwarding operations (push, swap, or pop). "

There is a book with some examples
Cisco.Press.QoS.for.IP.MPLS.Networks.Jun.2006.chm


Regards,


On 5/27/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
> That is correct (although you should take some overhead into account
> as well). However, if you do not perform Call Admission Control
> beforehand (you do not limit the maximum number of possible
> simultaneous calls) you've done nothing, or worse.
>
> The classic example being when you have a line that supports 4 calls,
> when the fifth call enters the line, all calls will suffer packet loss
> and delay, not only the fifth one.
>
> I'm not too versed in MPLS QoS, but the idea is always the same.
> Prioritize voice traffic, but don't let it kill your bandwidth, nor
> itself. Thus the need for CAC, then something like CBWFQ/PQ. If
> possible, voice traffic should be beneath shaping/policing limits, so
> that neither would affect it.
>
> regards,
>   Zoltan
>
> Quoting Jonathan Charles <jonvoip at gmail.com>:
>
> > My thinking is that I need to set the QOS to be the same as the
> > maximum number of calls to/from each site.
> >
> >
> > Jonathan
> >
> > On 5/27/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
> >> As long as it is about "calls", CAC is the sensitive way to go about
> >> limiting the calls in the first place. Do QoS "As close to the source,
> >> as possible", remember? :)
> >>
> >> If you do not have a CCM, but other set-up, you should still look in
> >> this direction first - to limit the calls to a reasonable amount in
> >> the given directions, before they are hitting your MPLS router.
> >>
> >> regards,
> >>   Zoltan
> >>
> >> Quoting Paul Choi <asobihoudai at yahoo.com>:
> >>
> >>> He's talking about QoS for MPLS, not CAC.
> >>>
> >>> --- "Clouse, Chris" <chris.clouse at berbee.com> wrote:
> >>>
> >>>> Use CallManager locations to perform Call Admission
> >>>> Control.
> >>>>
> >>>> Christopher Clouse, CCNP CCDP CCVP MCP Network+
> >>>>
> >>>>
> >>>> -----Original Message-----
> >>>> From: cisco-voip-bounces at puck.nether.net
> >>>> [mailto:cisco-voip-bounces at puck.nether.net] On
> >>>> Behalf Of Jonathan
> >>>> Charles
> >>>> Sent: Saturday, May 26, 2007 11:35 AM
> >>>> To: ciscovoip
> >>>> Subject: [cisco-voip] QOS for MPLS
> >>>>
> >>>> I have always understood that QOS settings need to
> >>>> match on both ends
> >>>> of a circuit... so, if you allocate 800k on one
> >>>> side, you need to do
> >>>> the same on the other.
> >>>>
> >>>> I am now working on an MPLS circuit that has DS3s on
> >>>> some sides, DS1s
> >>>> on others... do I still need to match QOS settings?
> >>>>
> >>>> Customer wants 800k on the T1s and 2500k on the
> >>>> DS3s... I don't think
> >>>> it will matter as I am matching these bw limits on
> >>>> the locations...
> >>>> but how do I throttle calls from a DS3-side location
> >>>> to a DS1
> >>>> location?
> >>>>
> >>>> For example... main site is in Chicago, they have a
> >>>> DS3, one remote
> >>>> site is in Wichita and has a T1... so, I put 800k
> >>>> for Wichita and
> >>>> 2500k for Chicago... what is to stop Chicago from
> >>>> flooding Wichita
> >>>> with 2.5MB of calls?
> >>>>
> >>>>
> >>>>
> >>>> Jonathan
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >> ____________________________________________________________________________________Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to
> >> get
> >>> online.
> >>> http://smallbusiness.yahoo.com/webhosting
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip at puck.nether.net
> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>
> >>
> >>
> >>
> >> ----------------------------------------------------------------
> >> This message was sent using IMP, the Internet Messaging Program.
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
>
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


-- 
"The network is the computer"

From jonvoip at gmail.com  Sun May 27 17:54:27 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 27 May 2007 16:54:27 -0500
Subject: [cisco-voip] QOS for MPLS
In-Reply-To: <262147.28980.qm@web37210.mail.mud.yahoo.com>
References: <262147.28980.qm@web37210.mail.mud.yahoo.com>
Message-ID: <5d093f9a0705271454u6e0c479cqee972a6fcd9c16a5@mail.gmail.com>

Which is the definition of MPLS, any-to-any communication. All
circuits are point to point into the MPLS cloud and then routing in
the cloud to each site.



Jonathan

On 5/27/07, Erick Bergquist <erickbe at yahoo.com> wrote:
> I've seen one or two setups where the MPLS was fully meshed (all sites could talk to any other site), which makes QoS even trickier.
>
> ----- Original Message ----
> From: "keli at carocomp.ro" <keli at carocomp.ro>
> To: Jonathan Charles <jonvoip at gmail.com>
> Cc: cisco-voip at puck.nether.net
> Sent: Sunday, May 27, 2007 3:50:17 PM
> Subject: Re: [cisco-voip] QOS for MPLS
>
> That is correct (although you should take some overhead into account
> as well). However, if you do not perform Call Admission Control
> beforehand (you do not limit the maximum number of possible
> simultaneous calls) you've done nothing, or worse.
>
> The classic example being when you have a line that supports 4 calls,
> when the fifth call enters the line, all calls will suffer packet loss
> and delay, not only the fifth one.
>
> I'm not too versed in MPLS QoS, but the idea is always the same.
> Prioritize voice traffic, but don't let it kill your bandwidth, nor
> itself. Thus the need for CAC, then something like CBWFQ/PQ. If
> possible, voice traffic should be beneath shaping/policing limits, so
> that neither would affect it.
>
> regards,
>    Zoltan
>
> Quoting Jonathan Charles <jonvoip at gmail.com>:
>
> > My thinking is that I need to set the QOS to be the same as the
> > maximum number of calls to/from each site.
> >
> >
> > Jonathan
> >
> > On 5/27/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
> >> As long as it is about "calls", CAC is the sensitive way to go about
> >> limiting the calls in the first place. Do QoS "As close to the source,
> >> as possible", remember? :)
> >>
> >> If you do not have a CCM, but other set-up, you should still look in
> >> this direction first - to limit the calls to a reasonable amount in
> >> the given directions, before they are hitting your MPLS router.
> >>
> >> regards,
> >>   Zoltan
> >>
> >> Quoting Paul Choi <asobihoudai at yahoo.com>:
> >>
> >>> He's talking about QoS for MPLS, not CAC.
> >>>
> >>> --- "Clouse, Chris" <chris.clouse at berbee.com> wrote:
> >>>
> >>>> Use CallManager locations to perform Call Admission
> >>>> Control.
> >>>>
> >>>> Christopher Clouse, CCNP CCDP CCVP MCP Network+
> >>>>
> >>>>
> >>>> -----Original Message-----
> >>>> From: cisco-voip-bounces at puck.nether.net
> >>>> [mailto:cisco-voip-bounces at puck.nether.net] On
> >>>> Behalf Of Jonathan
> >>>> Charles
> >>>> Sent: Saturday, May 26, 2007 11:35 AM
> >>>> To: ciscovoip
> >>>> Subject: [cisco-voip] QOS for MPLS
> >>>>
> >>>> I have always understood that QOS settings need to
> >>>> match on both ends
> >>>> of a circuit... so, if you allocate 800k on one
> >>>> side, you need to do
> >>>> the same on the other.
> >>>>
> >>>> I am now working on an MPLS circuit that has DS3s on
> >>>> some sides, DS1s
> >>>> on others... do I still need to match QOS settings?
> >>>>
> >>>> Customer wants 800k on the T1s and 2500k on the
> >>>> DS3s... I don't think
> >>>> it will matter as I am matching these bw limits on
> >>>> the locations...
> >>>> but how do I throttle calls from a DS3-side location
> >>>> to a DS1
> >>>> location?
> >>>>
> >>>> For example... main site is in Chicago, they have a
> >>>> DS3, one remote
> >>>> site is in Wichita and has a T1... so, I put 800k
> >>>> for Wichita and
> >>>> 2500k for Chicago... what is to stop Chicago from
> >>>> flooding Wichita
> >>>> with 2.5MB of calls?
> >>>>
> >>>>
> >>>>
> >>>> Jonathan
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >> ____________________________________________________________________________________Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to
> >> get
> >>> online.
> >>> http://smallbusiness.yahoo.com/webhosting
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip at puck.nether.net
> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>
> >>
> >>
> >>
> >> ----------------------------------------------------------------
> >> This message was sent using IMP, the Internet Messaging Program.
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
>
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
> ____________________________________________________________________________________
> We won't tell. Get more on shows you hate to love
> (and love to hate): Yahoo! TV's Guilty Pleasures list.
> http://tv.yahoo.com/collections/265
>

From holman at post.harvard.edu  Sun May 27 21:01:59 2007
From: holman at post.harvard.edu (Candace Holman)
Date: Sun, 27 May 2007 21:01:59 -0400
Subject: [cisco-voip] Mailing List
Message-ID: <81746c9a0705271801x4d6dff3dhb91a4ddf17440cf4@mail.gmail.com>

It's a little unpredictable - unfortunately still sends several "digests"
per day instead of just one - even if you have digest enabled on your
account.  Is it a mailing list setting that can be tweaked?

CH

---------- Forwarded message ----------
From: Paul Choi <asobihoudai at yahoo.com>
To: "Tkachuk,Birni" <birnit at nait.ca>, cisco-voip at puck.nether.net
Date: Sun, 27 May 2007 08:27:53 -0700 (PDT)
Subject: Re: [cisco-voip] Mailing List
Go to the subscription page for this mailing list and
change your settings.

Paul
--- "Tkachuk,Birni" <birnit at nait.ca> wrote:


---------------------------------
I would like to get only a daily digest of e-mails.
How do I go about requesting that change?  If not
possible then I will have to request to unsubscribe.
Thanks.


Birni Tkachuk
Voice Services Administrator
Northern Alberta Institute of Technology
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From daniel.abelski at gmail.com  Sun May 27 23:10:16 2007
From: daniel.abelski at gmail.com (Daniel A. Abelski)
Date: Mon, 28 May 2007 09:10:16 +0600
Subject: [cisco-voip] =?koi8-r?b?79TXxdQ6ICA3OTExIGZhY3RvcnkgZGVmYXVsdCBy?=
	=?koi8-r?b?ZXNldA==?=
In-Reply-To: <20070527124913.swe82qv5hcsogw48@mail.carocomp.ro>
References: <772285.4342.qm@web57204.mail.re3.yahoo.com>
	<20070527124913.swe82qv5hcsogw48@mail.carocomp.ro>
Message-ID: <cd0147ab0705272010y3ab2dd60wf74e4c2a79df14a2@mail.gmail.com>

its the load file

Here http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_installation_guide09186a00807aae21.html

it is said:

 When configuring the load command for  Java-based IP phones, such as
the Cisco Unified IP Phone 7906G, 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971G, configure only the filenames that are marked with an
asterisk (*) in the table below.

and here for different ver. of firmware.	

SCCP11.7-2-1-0S.loads*
term06.default.loads term11.default.loads
cvm11.7-2-0-66.sbn
jar11.7-2-0-66.sbn
dsp11.1-0-0-73.sbn
apps11.1-0-0-72.sbn
cnu11.3-0-0-81.sbn
	

SCCP11.8-0-3S.loads*
term06.default.loads
term11.default.loads
cvm11sccp.8-0-2-25.sbn
jar11sccp.8-0-2-25.sbn
dsp11.1-1-2-26.sbn
apps11.1-1-2-26.sbn
cnu11.3-1-2-26.sbn
	

SIP11.8-2-1S.loads*
term06.default.loads
term11.default.loads
apps11.8-0-2-55.sbn
cnu11.8-2-0-55.sbn
cvm11sip.8-2-0-55.sbn
dsp11.8-2-0-55.sbn
jar11sip.8-0-2-25.sbn

So. You need the SIP11.8-2-1S.loads file and you should add to config

load 7906 SCCP11.7-2-1-0S     for example

2007/5/27, keli at carocomp.ro <keli at carocomp.ro>:
> In the mean time, this sort-of turned out to be the problem.
>
> The phone is connected to a CCME, and the router did not have the
> proper firmware files loaded. Once I did that, this phone recovered.
>
> However, since I uploaded a newer firmware, that what was on the
> phones, I restarted all the phones, and now I have a different phone
> going crazy.
>
> It can access the tftp server, it can download all the firmware files,
> but then it just keeps rebooting, with "Resetting..." appearing once
> in a while on the phone. It has this loop, where it's powering up once
> with a checkmark, once with an X in that lower-left corner of the
> screen and keeps rebooting continuously.
>
> It has power, it gets DHCP, it's in the proper VLAN, the PC after it
> is working ok etc ...
>
> In the CCME telephony-service config, what is the proper filename for
> load 7911 ????
> since I've seen various configs on the net, some claiming the jar
> file, some the SCCP and even some the apps file. Does anybody know for
> sure?
>
> thanks,
>    Zoltan
>
> Quoting Paul Choi <asobihoudai at yahoo.com>:
>
> > Can your phone talk to the tftp server properly?
> >
> >
> > --- keli at carocomp.ro wrote:
> >
> >> Hi!
> >>
> >> I have a 7911G phone, with POE, that had problems
> >> starting up - got
> >> stock in the DHCP phase. It seemd like bad cabling,
> >> but it behaved the
> >> same plugged directly into the switch (Cisco POE
> >> switch).
> >>
> >> So I tried a factory reset, but it won't recover.
> >> I'm not sure wether
> >> I completed the procedure correctly though:
> >>   - reset the phone with the # pressed
> >>   - then press 123456789*#0 in sequence
> >>
> >> The prone now enters the "Upgrading" phase, reboots,
> >> with a circle
> >> icon (bullet?) in the lower left corner (where
> >> normally a checked
> >> checkbox resides) then it begins the upgradeing
> >> proces all over again.
> >>
> >> In the mean time, it recieves an IP adress from DHCP
> >> and can be pinged
> >> for a shor while.
> >>
> >> How do I exit this loop?
> >>
> >> thanks,
> >>    Zoltan
> >>
> >
> >
> >
> >
> >
> ____________________________________________________________________________________
> > Don't pick lemons.
> > See all the new 2007 cars at Yahoo! Autos.
> > http://autos.yahoo.com/new_cars.html
> >
>
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


-- 
Regards,                    |        ? ?????????,
Daniel A. Abelski        |        ????????? ?????? ?????????????

From aelnagar at ACT-EG.COM  Mon May 28 02:38:19 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Mon, 28 May 2007 09:38:19 +0300
Subject: [cisco-voip] PDA or 3rd Party Phone with CM 4.2
Message-ID: <88C8D019DA547C4BABECB9A806ED22EDDB3285@mail.ACT-EG.COM>

Hello all;

 

A customer is asking us to run Avaya H323 phone and a PDA with
callmanager 4.2, for the PDA I found a software that runs SCCP on the
PDA www.ipblue.com <http://www.ipblue.com/>  but the licensee is 115 $
for each PDA (which is very costly for the customer) and for Avaya I
don't know how to install it with the system and if this is available or
not, anyone has information about this??

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From keli at carocomp.ro  Mon May 28 03:15:42 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Mon, 28 May 2007 10:15:42 +0300
Subject: [cisco-voip] 7911 factory default reset
In-Reply-To: <cd0147ab0705272010y3ab2dd60wf74e4c2a79df14a2@mail.gmail.com>
References: <772285.4342.qm@web57204.mail.re3.yahoo.com>
	<20070527124913.swe82qv5hcsogw48@mail.carocomp.ro>
	<cd0147ab0705272010y3ab2dd60wf74e4c2a79df14a2@mail.gmail.com>
Message-ID: <20070528101542.prz8504pc8c8goc4@mail.carocomp.ro>

Then I hope this page has got a typo on it:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_installation_guide09186a00807b806b.html
since I've got CME 4.0(2), but the closest match of firmware I've  
found is something with files having 8-0-3-33 versions

Then again, it MIGHT have been related to network-locale settings,  
that was (accidentally maybe?) set to IT for the first position  
(though US for the rest) - when I verified #sh telephony-service

Anyway, 7906/7911 phones are a PITA, compared to the 7905 series. Who  
says, you need a tftp server for the factory reset of a lousy phone  
anyway? ... [/rant]

thanks for all the answers,
   Zoltan

("Daniel A. Abelski" <daniel.abelski at gmail.com>):

> its the load file
>
> Here
> http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_installation_guide09186a00807aae21.html
>
> it is said:
>
> When configuring the load command for  Java-based IP phones, such as
> the Cisco Unified IP Phone 7906G, 7911G, 7941G, 7941GE, 7961G, 7961GE,
> 7970G, and 7971G, configure only the filenames that are marked with an
> asterisk (*) in the table below.
>
> and here for different ver. of firmware.
>
> SCCP11.7-2-1-0S.loads*
> term06.default.loads term11.default.loads
> cvm11.7-2-0-66.sbn
> jar11.7-2-0-66.sbn
> dsp11.1-0-0-73.sbn
> apps11.1-0-0-72.sbn
> cnu11.3-0-0-81.sbn
>
>
> SCCP11.8-0-3S.loads*
> term06.default.loads
> term11.default.loads
> cvm11sccp.8-0-2-25.sbn
> jar11sccp.8-0-2-25.sbn
> dsp11.1-1-2-26.sbn
> apps11.1-1-2-26.sbn
> cnu11.3-1-2-26.sbn
>
>
> SIP11.8-2-1S.loads*
> term06.default.loads
> term11.default.loads
> apps11.8-0-2-55.sbn
> cnu11.8-2-0-55.sbn
> cvm11sip.8-2-0-55.sbn
> dsp11.8-2-0-55.sbn
> jar11sip.8-0-2-25.sbn
>
> So. You need the SIP11.8-2-1S.loads file and you should add to config
>
> load 7906 SCCP11.7-2-1-0S     for example
>
> 2007/5/27, keli at carocomp.ro <keli at carocomp.ro>:
>> In the mean time, this sort-of turned out to be the problem.
>>
>> The phone is connected to a CCME, and the router did not have the
>> proper firmware files loaded. Once I did that, this phone recovered.
>>
>> However, since I uploaded a newer firmware, that what was on the
>> phones, I restarted all the phones, and now I have a different phone
>> going crazy.
>>
>> It can access the tftp server, it can download all the firmware files,
>> but then it just keeps rebooting, with "Resetting..." appearing once
>> in a while on the phone. It has this loop, where it's powering up once
>> with a checkmark, once with an X in that lower-left corner of the
>> screen and keeps rebooting continuously.
>>
>> It has power, it gets DHCP, it's in the proper VLAN, the PC after it
>> is working ok etc ...
>>
>> In the CCME telephony-service config, what is the proper filename for
>> load 7911 ????
>> since I've seen various configs on the net, some claiming the jar
>> file, some the SCCP and even some the apps file. Does anybody know for
>> sure?
>>
>> thanks,
>>   Zoltan
>>
>> Quoting Paul Choi <asobihoudai at yahoo.com>:
>>
>>> Can your phone talk to the tftp server properly?
>>>
>>>
>>> --- keli at carocomp.ro wrote:
>>>
>>>> Hi!
>>>>
>>>> I have a 7911G phone, with POE, that had problems
>>>> starting up - got
>>>> stock in the DHCP phase. It seemd like bad cabling,
>>>> but it behaved the
>>>> same plugged directly into the switch (Cisco POE
>>>> switch).
>>>>
>>>> So I tried a factory reset, but it won't recover.
>>>> I'm not sure wether
>>>> I completed the procedure correctly though:
>>>>   - reset the phone with the # pressed
>>>>   - then press 123456789*#0 in sequence
>>>>
>>>> The prone now enters the "Upgrading" phase, reboots,
>>>> with a circle
>>>> icon (bullet?) in the lower left corner (where
>>>> normally a checked
>>>> checkbox resides) then it begins the upgradeing
>>>> proces all over again.
>>>>
>>>> In the mean time, it recieves an IP adress from DHCP
>>>> and can be pinged
>>>> for a shor while.
>>>>
>>>> How do I exit this loop?
>>>>
>>>> thanks,
>>>>    Zoltan
>>>>
>>>
>>>
>>>
>>>
>>>
>> ____________________________________________________________________________________
>>> Don't pick lemons.
>>> See all the new 2007 cars at Yahoo! Autos.
>>> http://autos.yahoo.com/new_cars.html
>>>
>>
>>
>>
>> ----------------------------------------------------------------
>> This message was sent using IMP, the Internet Messaging Program.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> -- 
> Regards,                    |        ? ?????????,
> Daniel A. Abelski        |        ????????? ?????? ?????????????



----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.


From asobihoudai at yahoo.com  Mon May 28 03:22:01 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Mon, 28 May 2007 00:22:01 -0700 (PDT)
Subject: [cisco-voip] 7911 factory default reset
In-Reply-To: <20070528101542.prz8504pc8c8goc4@mail.carocomp.ro>
Message-ID: <830549.58471.qm@web57205.mail.re3.yahoo.com>

Without a TFTP server, you'd have a helluva time
getting your phone to do anything after a factory
reset lemme tell ya.

You must setup the tftp-server statements underneath
CCME or you won't get very far.

Paul

--- keli at carocomp.ro wrote:

> Then I hope this page has got a typo on it:
>
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_installation_guide09186a00807b806b.html
> since I've got CME 4.0(2), but the closest match of
> firmware I've  
> found is something with files having 8-0-3-33
> versions
> 
> Then again, it MIGHT have been related to
> network-locale settings,  
> that was (accidentally maybe?) set to IT for the
> first position  
> (though US for the rest) - when I verified #sh
> telephony-service
> 
> Anyway, 7906/7911 phones are a PITA, compared to the
> 7905 series. Who  
> says, you need a tftp server for the factory reset
> of a lousy phone  
> anyway? ... [/rant]
> 
> thanks for all the answers,
>    Zoltan
> 
> ("Daniel A. Abelski" <daniel.abelski at gmail.com>):
> 
> > its the load file
> >
> > Here
> >
>
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_installation_guide09186a00807aae21.html
> >
> > it is said:
> >
> > When configuring the load command for  Java-based
> IP phones, such as
> > the Cisco Unified IP Phone 7906G, 7911G, 7941G,
> 7941GE, 7961G, 7961GE,
> > 7970G, and 7971G, configure only the filenames
> that are marked with an
> > asterisk (*) in the table below.
> >
> > and here for different ver. of firmware.
> >
> > SCCP11.7-2-1-0S.loads*
> > term06.default.loads term11.default.loads
> > cvm11.7-2-0-66.sbn
> > jar11.7-2-0-66.sbn
> > dsp11.1-0-0-73.sbn
> > apps11.1-0-0-72.sbn
> > cnu11.3-0-0-81.sbn
> >
> >
> > SCCP11.8-0-3S.loads*
> > term06.default.loads
> > term11.default.loads
> > cvm11sccp.8-0-2-25.sbn
> > jar11sccp.8-0-2-25.sbn
> > dsp11.1-1-2-26.sbn
> > apps11.1-1-2-26.sbn
> > cnu11.3-1-2-26.sbn
> >
> >
> > SIP11.8-2-1S.loads*
> > term06.default.loads
> > term11.default.loads
> > apps11.8-0-2-55.sbn
> > cnu11.8-2-0-55.sbn
> > cvm11sip.8-2-0-55.sbn
> > dsp11.8-2-0-55.sbn
> > jar11sip.8-0-2-25.sbn
> >
> > So. You need the SIP11.8-2-1S.loads file and you
> should add to config
> >
> > load 7906 SCCP11.7-2-1-0S     for example
> >
> > 2007/5/27, keli at carocomp.ro <keli at carocomp.ro>:
> >> In the mean time, this sort-of turned out to be
> the problem.
> >>
> >> The phone is connected to a CCME, and the router
> did not have the
> >> proper firmware files loaded. Once I did that,
> this phone recovered.
> >>
> >> However, since I uploaded a newer firmware, that
> what was on the
> >> phones, I restarted all the phones, and now I
> have a different phone
> >> going crazy.
> >>
> >> It can access the tftp server, it can download
> all the firmware files,
> >> but then it just keeps rebooting, with
> "Resetting..." appearing once
> >> in a while on the phone. It has this loop, where
> it's powering up once
> >> with a checkmark, once with an X in that
> lower-left corner of the
> >> screen and keeps rebooting continuously.
> >>
> >> It has power, it gets DHCP, it's in the proper
> VLAN, the PC after it
> >> is working ok etc ...
> >>
> >> In the CCME telephony-service config, what is the
> proper filename for
> >> load 7911 ????
> >> since I've seen various configs on the net, some
> claiming the jar
> >> file, some the SCCP and even some the apps file.
> Does anybody know for
> >> sure?
> >>
> >> thanks,
> >>   Zoltan
> >>
> >> Quoting Paul Choi <asobihoudai at yahoo.com>:
> >>
> >>> Can your phone talk to the tftp server properly?
> >>>
> >>>
> >>> --- keli at carocomp.ro wrote:
> >>>
> >>>> Hi!
> >>>>
> >>>> I have a 7911G phone, with POE, that had
> problems
> >>>> starting up - got
> >>>> stock in the DHCP phase. It seemd like bad
> cabling,
> >>>> but it behaved the
> >>>> same plugged directly into the switch (Cisco
> POE
> >>>> switch).
> >>>>
> >>>> So I tried a factory reset, but it won't
> recover.
> >>>> I'm not sure wether
> >>>> I completed the procedure correctly though:
> >>>>   - reset the phone with the # pressed
> >>>>   - then press 123456789*#0 in sequence
> >>>>
> >>>> The prone now enters the "Upgrading" phase,
> reboots,
> >>>> with a circle
> >>>> icon (bullet?) in the lower left corner (where
> >>>> normally a checked
> >>>> checkbox resides) then it begins the upgradeing
> >>>> proces all over again.
> >>>>
> >>>> In the mean time, it recieves an IP adress from
> DHCP
> >>>> and can be pinged
> >>>> for a shor while.
> >>>>
> >>>> How do I exit this loop?
> >>>>
> >>>> thanks,
> >>>>    Zoltan
> >>>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>
>
____________________________________________________________________________________
> >>> Don't pick lemons.
> >>> See all the new 2007 cars at Yahoo! Autos.
> >>> http://autos.yahoo.com/new_cars.html
> >>>
> >>
> >>
> >>
> >>
>
----------------------------------------------------------------
> >> This message was sent using IMP, the Internet
> Messaging Program.
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >
> >
> > -- 
> > Regards,                    |        ??
> ??????????????????,
> > Daniel A. Abelski        |       
> ?????????????????? ????????????
> ??????????????????????????
> 
> 
> 
>
----------------------------------------------------------------
> This message was sent using IMP, the Internet
> Messaging Program.
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



      ____________________________________________________________________________________
Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center.
http://autos.yahoo.com/green_center/ 

From keli at carocomp.ro  Mon May 28 03:40:28 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Mon, 28 May 2007 10:40:28 +0300
Subject: [cisco-voip] 7911 factory default reset
In-Reply-To: <830549.58471.qm@web57205.mail.re3.yahoo.com>
References: <830549.58471.qm@web57205.mail.re3.yahoo.com>
Message-ID: <20070528104028.l2j5o1qbk0w8sgks@mail.carocomp.ro>

You can do a factory reset on a 7905 even without an Ethernet cable  
attached, and it would "just work" next time you plug it in.

Granted, you won't do much with it without an Ethernet cable, but at  
least you sort-of know what is going on, and the phone isn't just  
restarting like a loonatic.

And yes, 7906 does need a tftp server. Kind'a understood that already.

regards,
   Zoltan

(Paul Choi <asobihoudai at yahoo.com>):

> Without a TFTP server, you'd have a helluva time
> getting your phone to do anything after a factory
> reset lemme tell ya.
>
> You must setup the tftp-server statements underneath
> CCME or you won't get very far.
>
> Paul
>
> --- keli at carocomp.ro wrote:
>
>> Then I hope this page has got a typo on it:
>>
> http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_installation_guide09186a00807b806b.html
>> since I've got CME 4.0(2), but the closest match of
>> firmware I've
>> found is something with files having 8-0-3-33
>> versions
>>
>> Then again, it MIGHT have been related to
>> network-locale settings,
>> that was (accidentally maybe?) set to IT for the
>> first position
>> (though US for the rest) - when I verified #sh
>> telephony-service
>>
>> Anyway, 7906/7911 phones are a PITA, compared to the
>> 7905 series. Who
>> says, you need a tftp server for the factory reset
>> of a lousy phone
>> anyway? ... [/rant]
>>
>> thanks for all the answers,
>>    Zoltan
>>
>> ("Daniel A. Abelski" <daniel.abelski at gmail.com>):
>>
>> > its the load file
>> >
>> > Here
>> >
>>
> http://www.cisco.com/en/US/products/sw/voicesw/ps4625/prod_installation_guide09186a00807aae21.html
>> >
>> > it is said:
>> >
>> > When configuring the load command for  Java-based
>> IP phones, such as
>> > the Cisco Unified IP Phone 7906G, 7911G, 7941G,
>> 7941GE, 7961G, 7961GE,
>> > 7970G, and 7971G, configure only the filenames
>> that are marked with an
>> > asterisk (*) in the table below.
>> >
>> > and here for different ver. of firmware.
>> >
>> > SCCP11.7-2-1-0S.loads*
>> > term06.default.loads term11.default.loads
>> > cvm11.7-2-0-66.sbn
>> > jar11.7-2-0-66.sbn
>> > dsp11.1-0-0-73.sbn
>> > apps11.1-0-0-72.sbn
>> > cnu11.3-0-0-81.sbn
>> >
>> >
>> > SCCP11.8-0-3S.loads*
>> > term06.default.loads
>> > term11.default.loads
>> > cvm11sccp.8-0-2-25.sbn
>> > jar11sccp.8-0-2-25.sbn
>> > dsp11.1-1-2-26.sbn
>> > apps11.1-1-2-26.sbn
>> > cnu11.3-1-2-26.sbn
>> >
>> >
>> > SIP11.8-2-1S.loads*
>> > term06.default.loads
>> > term11.default.loads
>> > apps11.8-0-2-55.sbn
>> > cnu11.8-2-0-55.sbn
>> > cvm11sip.8-2-0-55.sbn
>> > dsp11.8-2-0-55.sbn
>> > jar11sip.8-0-2-25.sbn
>> >
>> > So. You need the SIP11.8-2-1S.loads file and you
>> should add to config
>> >
>> > load 7906 SCCP11.7-2-1-0S     for example
>> >
>> > 2007/5/27, keli at carocomp.ro <keli at carocomp.ro>:
>> >> In the mean time, this sort-of turned out to be
>> the problem.
>> >>
>> >> The phone is connected to a CCME, and the router
>> did not have the
>> >> proper firmware files loaded. Once I did that,
>> this phone recovered.
>> >>
>> >> However, since I uploaded a newer firmware, that
>> what was on the
>> >> phones, I restarted all the phones, and now I
>> have a different phone
>> >> going crazy.
>> >>
>> >> It can access the tftp server, it can download
>> all the firmware files,
>> >> but then it just keeps rebooting, with
>> "Resetting..." appearing once
>> >> in a while on the phone. It has this loop, where
>> it's powering up once
>> >> with a checkmark, once with an X in that
>> lower-left corner of the
>> >> screen and keeps rebooting continuously.
>> >>
>> >> It has power, it gets DHCP, it's in the proper
>> VLAN, the PC after it
>> >> is working ok etc ...
>> >>
>> >> In the CCME telephony-service config, what is the
>> proper filename for
>> >> load 7911 ????
>> >> since I've seen various configs on the net, some
>> claiming the jar
>> >> file, some the SCCP and even some the apps file.
>> Does anybody know for
>> >> sure?
>> >>
>> >> thanks,
>> >>   Zoltan
>> >>
>> >> Quoting Paul Choi <asobihoudai at yahoo.com>:
>> >>
>> >>> Can your phone talk to the tftp server properly?
>> >>>
>> >>>
>> >>> --- keli at carocomp.ro wrote:
>> >>>
>> >>>> Hi!
>> >>>>
>> >>>> I have a 7911G phone, with POE, that had
>> problems
>> >>>> starting up - got
>> >>>> stock in the DHCP phase. It seemd like bad
>> cabling,
>> >>>> but it behaved the
>> >>>> same plugged directly into the switch (Cisco
>> POE
>> >>>> switch).
>> >>>>
>> >>>> So I tried a factory reset, but it won't
>> recover.
>> >>>> I'm not sure wether
>> >>>> I completed the procedure correctly though:
>> >>>>   - reset the phone with the # pressed
>> >>>>   - then press 123456789*#0 in sequence
>> >>>>
>> >>>> The prone now enters the "Upgrading" phase,
>> reboots,
>> >>>> with a circle
>> >>>> icon (bullet?) in the lower left corner (where
>> >>>> normally a checked
>> >>>> checkbox resides) then it begins the upgradeing
>> >>>> proces all over again.
>> >>>>
>> >>>> In the mean time, it recieves an IP adress from
>> DHCP
>> >>>> and can be pinged
>> >>>> for a shor while.
>> >>>>
>> >>>> How do I exit this loop?
>> >>>>
>> >>>> thanks,
>> >>>>    Zoltan
>> >>>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>
>>
> ____________________________________________________________________________________
>> >>> Don't pick lemons.
>> >>> See all the new 2007 cars at Yahoo! Autos.
>> >>> http://autos.yahoo.com/new_cars.html
>> >>>
>> >>
>> >>
>> >>
>> >>
>>
> ----------------------------------------------------------------
>> >> This message was sent using IMP, the Internet
>> Messaging Program.
>> >>
>> >> _______________________________________________
>> >> cisco-voip mailing list
>> >> cisco-voip at puck.nether.net
>> >>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> >>
>> >
>> >
>> > --
>> > Regards,                    |        ??
>> ??????????????????,
>> > Daniel A. Abelski        |
>> ?????????????????? ????????????
>> ??????????????????????????
>>
>>
>>
>>
> ----------------------------------------------------------------
>> This message was sent using IMP, the Internet
>> Messaging Program.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
>
>         
> ____________________________________________________________________________________
> Park yourself in front of a world of choices in alternative   
> vehicles. Visit the Yahoo! Auto Green Center.
> http://autos.yahoo.com/green_center/
>



----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.


From aelnagar at ACT-EG.COM  Mon May 28 05:03:54 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Mon, 28 May 2007 12:03:54 +0300
Subject: [cisco-voip] Cisco Unified CallManager Compatibility Matrix Last
	Updated 04/19/2007
Message-ID: <88C8D019DA547C4BABECB9A806ED22EDDB3479@mail.ACT-EG.COM>

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/prod_softwa
re_versions_comparison.html 

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From keli at carocomp.ro  Mon May 28 06:21:33 2007
From: keli at carocomp.ro (keli at carocomp.ro)
Date: Mon, 28 May 2007 13:21:33 +0300
Subject: [cisco-voip] Ringlist.xml / List.xml on CCME?
Message-ID: <20070528132133.5tqxl3efls0wcwww@mail.carocomp.ro>

Hi!

Is it possible to allow Cisco 7911 phones connected to a CCME to  
change backgrounds / ring tones?

I've created the proper List.xml / Ringlist.xml files, and they are  
served through tftp. However, the phones do not seem to ask for them ...

thanks,
  Zoltan


----------------------------------------------------------------
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From jonvoip at gmail.com  Mon May 28 09:57:39 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 28 May 2007 08:57:39 -0500
Subject: [cisco-voip] Upgrade to 5.1(2)
In-Reply-To: <D2BBAE709B4D874DAC556C7EB9D938FC31BC27@exchange.TGASolutions.local>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451F11@CSEXCH01.charles-stanley.co.uk>
	<5d093f9a0705260901l625f347dnea83faf3187d3122@mail.gmail.com>
	<D2BBAE709B4D874DAC556C7EB9D938FC31BC27@exchange.TGASolutions.local>
Message-ID: <5d093f9a0705280657t7a42b743g106429a7e1860c35@mail.gmail.com>

Well, apparently CCM 5 defaults to G.729 for inter-region communications (
G.711 for Intra-region)... This was my first CCM5 multi-site deployment, and
this was the first site that didn't have a CM at it.

We changed the default to 711 for the time being until we can get some
transcoding resources deployed.



Jonathan

On 5/28/07, Jim McBurnett <jim at tgasolutions.com> wrote:
>
>  Jonathan,
>
> What was the conferencing Issue?
>
> Thanks,
>
> jim
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Jonathan Charles
> *Sent:* Saturday, May 26, 2007 12:01 PM
> *To:* Nick Kassel
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Upgrade to 5.1(2)
>
>
>
> I am upgrading a test box right now... It was recommended by TAC last
> night (issue with conferencing...)...
>
>
> Jonathan
>
> On 5/25/07, *Nick Kassel* <Nick.Kassel at charles-stanley.co.uk> wrote:
>
> Has anyone upgraded to 5.1(2) yet, just wondered if anyone had any issues
> with it.
>
>
>
>
>
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From jonvoip at gmail.com  Mon May 28 09:59:03 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 28 May 2007 08:59:03 -0500
Subject: [cisco-voip] Ringlist.xml / List.xml on CCME?
In-Reply-To: <20070528132133.5tqxl3efls0wcwww@mail.carocomp.ro>
References: <20070528132133.5tqxl3efls0wcwww@mail.carocomp.ro>
Message-ID: <5d093f9a0705280659o2e10bcbdk6dfdca353d19b4c3@mail.gmail.com>

You need a DistinctiveRingList.xml for ring tones... (check the file
name spelling)... if you check the phone logs, you will see that it is
getting a file not found for that file.



Jonathan

On 5/28/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
> Hi!
>
> Is it possible to allow Cisco 7911 phones connected to a CCME to
> change backgrounds / ring tones?
>
> I've created the proper List.xml / Ringlist.xml files, and they are
> served through tftp. However, the phones do not seem to ask for them ...
>
> thanks,
>   Zoltan
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From asobihoudai at yahoo.com  Mon May 28 11:34:51 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Mon, 28 May 2007 08:34:51 -0700 (PDT)
Subject: [cisco-voip] Ringlist.xml / List.xml on CCME?
In-Reply-To: <20070528132133.5tqxl3efls0wcwww@mail.carocomp.ro>
Message-ID: <112869.80920.qm@web57207.mail.re3.yahoo.com>

Last I read,

Only 7970's backgrounds can be changed under CCME.
Post your configuration and let's see what's going on
here.


--- keli at carocomp.ro wrote:

> Hi!
> 
> Is it possible to allow Cisco 7911 phones connected
> to a CCME to  
> change backgrounds / ring tones?
> 
> I've created the proper List.xml / Ringlist.xml
> files, and they are  
> served through tftp. However, the phones do not seem
> to ask for them ...
> 
> thanks,
>   Zoltan
> 
> 
>
----------------------------------------------------------------
> This message was sent using IMP, the Internet
> Messaging Program.
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________Got a little couch potato? 
Check out fun summer activities for kids.
http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+kids&cs=bz 

From BRogers at pasadenaisd.org  Mon May 28 12:52:48 2007
From: BRogers at pasadenaisd.org (Brad Rogers)
Date: Mon, 28 May 2007 11:52:48 -0500
Subject: [cisco-voip] Gateway Area Codes
Message-ID: <CB3DDB018C56BB42A32D5C63006B76ED0313BB9B@adminexchg03.pasadenaisd.org>

Is it possible to assign multiple local area codes to a single gateway
in CAR's dial plan config?

 

 

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From keli at carocomp.ro  Mon May 28 15:31:35 2007
From: keli at carocomp.ro (keli)
Date: Mon, 28 May 2007 22:31:35 +0300
Subject: [cisco-voip] Ringlist.xml / List.xml on CCME?
In-Reply-To: <112869.80920.qm@web57207.mail.re3.yahoo.com>
References: <112869.80920.qm@web57207.mail.re3.yahoo.com>
Message-ID: <465B2E17.8000800@carocomp.ro>

It seems, I'm finding things out on my own as well, every time I ask a 
question :-)

I did some tftp debugging, and found out a couple of things:
- The phone does not seem to look for the RingList.xml on startup, but 
only after you enter into the appropriate menu on the phone 
(Settings/User Preferences/Ringtones -> and select a line to change too, 
before it will start looking for ringtones )

- it's similar with the "background" image too. First you need to enter 
the menu, and try to change it, before the phone asks the TFTP server 
for the files.

- *However* List.xml does not go into the TFTP root, but instead into 
Desktop/95x34x1/
( #copy tftp://.../List.xml flash:/Desktop/95x34x1/List.xml ) It says so 
in the docs, but seems like I missed that detail on the first reading.

Disclaimer: well, actually, on 7911G phones they are not "background 
images" as such, only Cisco likes to call them like that. It's that 
monochrome black/transparent logo thingie, that appears on the phone in 
idle mode. It's a nice touch and really easy to upload the client's logo 
there ...

Docs are here, or google "Customizing 7911G": *http://tinyurl.com/36gxtc*

thanks for the ideas,
  Zoltan

Paul Choi wrote:
> Last I read,
>
> Only 7970's backgrounds can be changed under CCME.
> Post your configuration and let's see what's going on
> here.
>
>
> --- keli at carocomp.ro wrote:
>
>   
>> Hi!
>>
>> Is it possible to allow Cisco 7911 phones connected
>> to a CCME to  
>> change backgrounds / ring tones?
>>
>> I've created the proper List.xml / Ringlist.xml
>> files, and they are  
>> served through tftp. However, the phones do not seem
>> to ask for them ...
>>
>> thanks,
>>   Zoltan
>>
>>
>>
>>     
> ----------------------------------------------------------------
>   
>> This message was sent using IMP, the Internet
>> Messaging Program.
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>     
>
>
>
>        
> ____________________________________________________________________________________Got a little couch potato? 
> Check out fun summer activities for kids.
> http://search.yahoo.com/search?fr=oni_on_mail&p=summer+activities+for+kids&cs=bz 
>   


From dhartma5 at optonline.net  Mon May 28 16:23:35 2007
From: dhartma5 at optonline.net (Dennis Hartmann)
Date: Mon, 28 May 2007 16:23:35 -0400
Subject: [cisco-voip] Client / Server Traffic Generator
Message-ID: <004701c7a166$11c8d8e0$1f01a8c0@e1505>

    Does anyone know of any good software that will allow me to generate
traffic between two PCs?  
 
    I'm looking to congest a link running at T-1 speed to bi-directionally
test an LLQ QoS policy.  I'd like to generate as many different application
flows as possible so I can use NBAR and a full 11-class policy like the one
that Cisco recommends in the QoS SRND.  Any advice would be appreciated.
Thanks.
 
Sincerely,
 
Dennis J. Hartmann
White Pine Communications
CCIE# 15651 
CCVP/CCSI/CCIP/CCNP/MCSE
    
 
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From bridge_stone at gmx.net  Mon May 28 17:09:36 2007
From: bridge_stone at gmx.net (Konne)
Date: Mon, 28 May 2007 23:09:36 +0200
Subject: [cisco-voip] Client / Server Traffic Generator
In-Reply-To: <004701c7a166$11c8d8e0$1f01a8c0@e1505>
References: <004701c7a166$11c8d8e0$1f01a8c0@e1505>
Message-ID: <465B4510.50101@gmx.net>

hi dennis,

i use D-ITG, Distributed Internet Traffic Generator.
very nice tool...

bye,
Konrad

Dennis Hartmann wrote:

>     Does anyone know of any good software that will allow me to 
> generate traffic between two PCs? 
>  
>     I'm looking to congest a link running at T-1 speed to 
> bi-directionally test an LLQ QoS policy.  I'd like to generate as many 
> different application flows as possible so I can use NBAR and a full 
> 11-class policy like the one that Cisco recommends in the QoS SRND.  
> Any advice would be appreciated.  Thanks.
>  
> Sincerely,
>  
> *Dennis J. Hartmann
> *White Pine Communications
> *CCIE# 15651 *
> CCVP/CCSI/CCIP/CCNP/MCSE
>    
>  
>
>------------------------------------------------------------------------
>
>_______________________________________________
>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip
>  
>


From lelio at uoguelph.ca  Mon May 28 23:31:18 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 28 May 2007 23:31:18 -0400
Subject: [cisco-voip] IOS vs non-IOS NSE types
Message-ID: <001501c7a1a1$d20f67a0$c901a8c0@pdp11>

Wondering if people have any comments on the NSE type available on 6608 ports. It says IOS or non-IOS. Is one better than the other? I'll likely be using VG224s - so I'm guessing IOS is better.

On the 3845, however, I don't have this sort of option on the CCM config page, and the show running only shows "mgcp modem passthrough voip mode NSE" it doesn't specify IOS or non-IOS type.

Any clues?

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
...there's no such thing as a bad timbit...


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From balbler at nts.at  Tue May 29 02:33:52 2007
From: balbler at nts.at (Bernhard Albler)
Date: Tue, 29 May 2007 08:33:52 +0200
Subject: [cisco-voip] Upgrade to 5.1(2)
In-Reply-To: <5d093f9a0705280657t7a42b743g106429a7e1860c35@mail.gmail.com>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451F11@CSEXCH01.charles-stanley.co.uk><5d093f9a0705260901l625f347dnea83faf3187d3122@mail.gmail.com><D2BBAE709B4D874DAC556C7EB9D938FC31BC27@exchange.TGASolutions.local>
	<5d093f9a0705280657t7a42b743g106429a7e1860c35@mail.gmail.com>
Message-ID: <5e360a2300007727@nts.at>

Hi all,

tried to upgrade a customer from 5.1(1)ES08 to 5.1(2) over the weekend.
I finally had to revert back to the old version, because all pickup
group fields in lines and huntpilots where set to "null".

Most likely I am hitting CSCsi91071, though it seems odd that such a bug
would slip into the release.
Can anyone from cisco enlighten on CM5 version numbers?
Is 5.1(2.9901.2) a version  that comes before or after 5.1(2.1000.11)
(which is the released cm 5.1(2) I believe)?

Regards
bernhard

------------------------------------------
DI (FH) Bernhard Albler, NTS AG
Tel.: +43 (1) 867 67 76 -0, email: balbler at nts.at
________________________________________

From Nick.Kassel at Charles-Stanley.co.uk  Tue May 29 03:41:43 2007
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 29 May 2007 08:41:43 +0100
Subject: [cisco-voip] ciscotsp for CCM 5.04: any How To?
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451F16@CSEXCH01.charles-stanley.co.uk>

We had plenty of problems with TSP and 5.1, the best way to install is
to Pause or Stop the Telephony service then install the TSP and then
install the Wave drivers, if you are lucky it will work, we had about a
50/50 ratio of working against not working, which just doesn't make
sense even the PC's were exactly the same with the same build. Another
thing we found was that if you don't get it working on a PC, then you
never will no matter what we tried to do, either the TSP does not show
up in the Phones and Modems or it does show and the configure button is
greyed out, we had to re build a number of PC's to get this working
which is crazy. My advice is not to use TSP at all, especially if you
are going to be upgrading to 5.1. The trouble was we had given them
dialling from Outlook feature and then we couldn't really take this away
just cause we upgraded to 5.1, but I will not be trying to phase this
out ASAP. 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony
Kouloglou
Sent: 24 May 2007 14:26
To: Mark Turpin
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] ciscotsp for CCM 5.04: any How To?

 

Yes,
i am running XP pro sp2.
Don't i have to do anything with ccm?
Where can i find a guide for this... :-( 

Mark Turpin wrote: 

Have you rebooted?
 
-Mark
 
On 5/24/07 8:01 AM, "Anthony Kouloglou" <akoul at dataways.gr>
<mailto:akoul at dataways.gr>  wrote:
 
  

	Hi,
	1. i have tried to configure it on a ccm 5.0(4) for the first
time so i
	can dial contacts from Outlook.
	2. I have downloaded the plugin from the admin page of the ccm
(version
	5.1.018).
	3. I installed it my PC.
	4. I configure it: ie user, password and ip of ccm
	the user has a phone associated with a phone.
	 
	But, in my outlook i cannot see the cisco tsp or the sepxxxx in
my dial
	lines.
	 
	I looked around but i could not find any guide about this.
	 
	Any ideas? Has anyone dealt with this in the past?
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	    

 
  
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From patrick.diener at gmail.com  Tue May 29 03:47:13 2007
From: patrick.diener at gmail.com (Patrick Diener)
Date: Tue, 29 May 2007 09:47:13 +0200
Subject: [cisco-voip] how to test a t1 cas what are the cmds
In-Reply-To: <ECEA9223F645834FB6A65B7CE377B9CD11C667@mail.adcapnetsvc.local>
References: <ECEA9223F645834FB6A65B7CE377B9CD11C667@mail.adcapnetsvc.local>
Message-ID: <c0deb5b80705290047r372ffc64gfbd3637fd7097e93@mail.gmail.com>

to debug the e&m signaling:
debug vpm signal

to debug the call routing:
debug voice dialpeer

Regards Patrick

On 5/25/07, James Grace <jgrace at adcapnet.com> wrote:
> ?
>
>
> how to test a t1 cas  what are the cmds
> Im trying to test some calls going over the t1 cas wink/start connection.  i
> not sure what cmds i can use
>
>
>
> James Grace
>
> Systems Engineer
> Adcap Network Systems, Inc.
> 6865 Shiloh Road East, Suite 250
> Alpharetta, GA 30005
>
> office: 678-456-6986
> direct: 678-456-6761
> jgrace at adcapnet.com
> www.adcapnet.comDisclaimer For ADCAP Network Systems
>
> "The information transmitted is intended only for the person or entity to
> which it is addressed and may
> contain confidential and/or privileged material. Any review, retransmission,
> dissemination or other use of,
> or taking of any action in reliance upon this information by persons or
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> recipient is prohibited. If you received this message in error, please
> contact the sender and delete the
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> This original email was sent to the internet for delivery at 25 May 2007
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From rafiq at dreamthought.com  Tue May 29 06:03:34 2007
From: rafiq at dreamthought.com (Raf)
Date: Tue, 29 May 2007 11:03:34 +0100 (BST)
Subject: [cisco-voip] CCM 5.1 AXL Testing
Message-ID: <20070529110243.W92176@joshua.dreamthought.com>

Dear all,

I'm trying to verify some software which utilises the AXL soap interface 
on a CCM v5.1.  What I'd ideally like to be able to do is find/write some 
software which provided with an outgoing request message, is able to 
invoke some kind of 'get' method on the remote service to verify that the 
request was currently handled and stored in the call manager.

I'm testing across a range of functions involving different hunt and line 
groups, pickup groups, phone devices and many other features.  I would 
therefore probably like to be able to do is auto-magically, provide the 
outbound set-up request and follow this with a query to confirm that 
applied changes did actually take affect.


I know it's wishful, however I hoped that someone might be able to point 
me to a suitable api or a mapping from AXL device functions to their 
corresponding accessor functions?


Any help would be appreciated.
Kind Regards,

Raf




From balbler at nts.at  Tue May 29 07:45:49 2007
From: balbler at nts.at (Bernhard Albler)
Date: Tue, 29 May 2007 13:45:49 +0200
Subject: [cisco-voip] Upgrade to 5.1(2)
In-Reply-To: <457A257D963C2940B3DB55D0BDBE18397CC7E8@comm01.annese.local>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451F11@CSEXCH01.charles-stanley.co.uk><5d093f9a0705260901l625f347dnea83faf3187d3122@mail.gmail.com><D2BBAE709B4D874DAC556C7EB9D938FC31BC27@exchange.TGASolutions.local><5d093f9a0705280657t7a42b743g106429a7e1860c35@mail.gmail.com>
	<5A8C69748176434F92F1EECD7D21F72CC046E0@srvgraz03.nts.at>
	<457A257D963C2940B3DB55D0BDBE18397CC7E8@comm01.annese.local>
Message-ID: <5f539fa300007db4@nts.at>

Hi all,

well, at least I am not the only one experiencing this issue.

I have a open tac case on this (606091409), I'll keep the list posted on
the outcome.
Thanks god for the 2 partitions on cm5 installations ;-)
Regards
bernhard

------------------------------------------
DI (FH) Bernhard Albler, NTS AG 
Tel.: +43 (1) 867 67 76 -0, email: balbler at nts.at 


From Matt.Slaga at us.didata.com  Tue May 29 08:00:31 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 29 May 2007 08:00:31 -0400
Subject: [cisco-voip] PDA or 3rd Party Phone with CM 4.2
In-Reply-To: <88C8D019DA547C4BABECB9A806ED22EDDB3285@mail.ACT-EG.COM>
References: <88C8D019DA547C4BABECB9A806ED22EDDB3285@mail.ACT-EG.COM>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005AD638F@USNAEXCH.na.didata.local>

Have you tried configuring the avaya phone as an H323 client in CCM?
When you add a new phone, there is an option for H323 client.

 

Just think, when you upgrade to 5.x, in addition to the $115 for ipblue
you will have to buy either three (basic) or six (advanced) DLU's as
well ($150-$300).

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Monday, May 28, 2007 2:38 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PDA or 3rd Party Phone with CM 4.2

 

Hello all;

 

A customer is asking us to run Avaya H323 phone and a PDA with
callmanager 4.2, for the PDA I found a software that runs SCCP on the
PDA www.ipblue.com <http://www.ipblue.com/>  but the licensee is 115 $
for each PDA (which is very costly for the customer) and for Avaya I
don't know how to install it with the system and if this is available or
not, anyone has information about this??




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From axe7961 at hotmail.com  Tue May 29 08:49:23 2007
From: axe7961 at hotmail.com (Axel Wilzinski)
Date: Tue, 29 May 2007 08:49:23 -0400
Subject: [cisco-voip] PRI down after power failure - need to reset MGCP
Message-ID: <BAY103-W119E990B199537561E1715BA2F0@phx.gbl>

Hi,Does anyone know why in order to make a PRI come active I have to manually reset MGCP on a 2851 gateway after either a power failure or power cycle?  If I don't manually do a "no MGCP, MGCP" the PRI stays in "tei_assigned" status.  (The gateway does register with Call Manager after a power cycle and the FXO/FXS ports work fine...)  The IOS version is 12.4(11)T1 and Call Manager is at 4.2.3 with all the latest service releases.Thank You!
_________________________________________________________________
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From bms314 at gmail.com  Tue May 29 10:30:47 2007
From: bms314 at gmail.com (Brian)
Date: Tue, 29 May 2007 09:30:47 -0500
Subject: [cisco-voip] CM 4.2(3)sr2a - Gone
In-Reply-To: <6A0744639B86DD459433108DBF101D26016AECC4@MAIL01.sentinel.com>
References: <Acee99m2BNrNO2M5SmuEyaSjU8pnSw==>
	<6A0744639B86DD459433108DBF101D26016AECC4@MAIL01.sentinel.com>
Message-ID: <7aaaf7f0705290730x1fdda9a1vdb467e5df3d619fa@mail.gmail.com>

Looks like SR2b is there now...anyone know what was wrong with SR2a?

On 5/25/07, Carter, Bill <bcarter at sentinel.com> wrote:
>
>  What happened to it?
>
> **
> *Bill Carter*
> Sr. Business Communications Analyst
> Sentinel Technologies
> (O) 217.391.5015
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From jonvoip at gmail.com  Tue May 29 10:30:49 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 29 May 2007 09:30:49 -0500
Subject: [cisco-voip] MGCP Fallback time...
In-Reply-To: <000c01c79f3b$089a8660$47196883@cfs.uoguelph.ca>
References: <5d093f9a0705251734n71238a7ch7cbcb676e9de9c7c@mail.gmail.com>
	<001101c79f2f$646e0680$c901a8c0@pdp11>
	<5d093f9a0705251757s13fc34abr3c48fa32c265472f@mail.gmail.com>
	<000c01c79f3b$089a8660$47196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0705290730i7b7686cfi5ffb526156884390@mail.gmail.com>

I am debating on setting the switchback time to immediate... but I
don't think that will have any effect.


Jonathan

On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>
> well, your config is darn close to mine. here's mine to see the difference.
> you might want to try a few things I have? timeout under the
> call-manager-fallback?
>
> vgw-kemptville#show running-config
> Building configuration...
>
> Current configuration : 9987 bytes
> !
> version 12.4
> service timestamps debug datetime localtime
> service timestamps log datetime localtime
> service password-encryption
> !
> hostname vgw-kemptville
> !
> boot-start-marker
> boot system flash:c3845-spservicesk9-mz.124-11.T2.bin
> boot-end-marker
> !
> card type t1 0 0
> card type t1 0 1
> logging buffered 4096
> logging console informational
> enable secret <snip>
> !
> aaa new-model
> !
> !
> aaa authentication login default local group tacacs+ enable
> aaa authentication login virt-users group tacacs+ enable
> aaa authentication login console group tacacs+ enable
> !
> !
> aaa session-id common
> clock timezone EST -5
> clock summer-time EDT recurring
> network-clock-participate wic 0
> network-clock-participate wic 1
> no ip dhcp use vrf connected
> ip dhcp excluded-address 10.104.126.99
> ip dhcp excluded-address 10.104.126.100
> !
> ip dhcp pool dhcp_vlan826
>    network 10.104.126.0 255.255.255.128
>    option 150 ip 10.104.21.121
>    default-router 10.104.126.1
> !
> !
> ip cef
> !
> !
> multilink bundle-name authenticated
> !
> isdn switch-type primary-dms100
> voice-card 0
>  no dspfarm
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> voice translation-rule 613
>  rule 1 /\(.+\)/ /9\1/
> !
> voice translation-rule 900
>  rule 1 /^91809......./ /918090000000/
>  rule 2 /^91829......./ /918290000000/
>  rule 3 /^91900......./ /919000000000/
>  rule 4 /^91...976..../ /919760000000/
>  rule 5 /^91976..../ /919760000000/
>  rule 6 /^9976..../ /919760000000/
> !
> voice translation-rule 1112
>  rule 1 reject /914165551212/
> !
> !
> voice translation-profile 1112
> !
> voice translation-profile 999
>  translate called 1112
> !
> voice translation-profile kc_did_profile
>  translate called 613
> !
> voice translation-profile uogdev_block_highrisk_900_vtp
>  translate called 900
> !
> !
> !
> crypto pki trustpoint TP-self-signed-3156109335
>  enrollment selfsigned
>  subject-name cn=IOS-Self-Signed-Certificate-3156109335
>  revocation-check none
>  rsakeypair TP-self-signed-3156109335
> !
> !
> crypto pki certificate chain TP-self-signed-3156109335
>  certificate self-signed 01
> <snip>
>   quit
> !
> !
> vtp domain 3845
> vtp mode transparent
> !
> !
> controller T1 0/0/0
>  framing esf
>  linecode b8zs
>  pri-group timeslots 1-24 service mgcp
> !
> controller T1 0/1/0
>  framing esf
>  linecode b8zs
>  pri-group timeslots 1-24 service mgcp
> !
> vlan 129
> !
> vlan 725
>  name voice_10.104.125.0/25
> !
> vlan 826
>  name voice_10.104.126.0/25
> !
> translation-rule 1112
> !
> !
> !
> !
> !
> interface GigabitEthernet0/0
>  description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
>  no ip address
>  duplex auto
>  speed auto
>  media-type rj45
>  no keepalive
> !
> interface GigabitEthernet0/0.129
>  encapsulation dot1Q 129 native
>  ip address 192.168.129.66 255.255.255.0
>  ip access-group vlan129_in in
>  ip access-group vlan129_out out
> !
> interface GigabitEthernet0/0.725
>  encapsulation dot1Q 725
>  ip address 10.104.125.99 255.255.255.128
> !
> interface GigabitEthernet0/0.826
>  encapsulation dot1Q 826
>  ip address 10.104.126.1 255.255.255.128
> !
> interface Service-Engine0/0
>  ip unnumbered GigabitEthernet0/0.826
>  service-module ip address 10.104.126.100 255.255.255.128
>  service-module ip default-gateway 10.104.126.1
> !
> interface GigabitEthernet0/1
>  no ip address
>  shutdown
>  duplex auto
>  speed auto
>  media-type rj45
>  no keepalive
> !
> interface Serial0/0/0:23
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-dms100
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
> !
> interface Serial0/1/0:23
>  no ip address
>  encapsulation hdlc
>  isdn switch-type primary-dms100
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
> !
> ip default-gateway 192.168.129.1
> ip route 0.0.0.0 0.0.0.0 192.168.129.1
> ip route 10.0.0.0 255.0.0.0 10.104.125.1
> ip route 10.104.126.100 255.255.255.255 Service-Engine0/0
> !
> !
> ip http server
> ip http authentication local
> ip http secure-server
> ip http timeout-policy idle 60 life 86400 requests 10000
> !
> ip access-list extended vlan129_in
>  deny   ip any 10.0.0.0 0.255.255.255
>  deny   ip 10.0.0.0 0.255.255.255 any
>  permit ip any any
> ip access-list extended vlan129_out
>  deny   ip 10.0.0.0 0.255.255.255 any
>  deny   ip any 10.0.0.0 0.255.255.255
>  permit ip any any
> !
> logging 192.168.20.5
> !
> !
> tacacs-server host 192.168.20.5
> tacacs-server directed-request
> tacacs-server key <snip>
> !
> control-plane
> !
> !
> !
> voice-port 0/0/0:23
> !
> voice-port 0/1/0:23
> !
> ccm-manager fallback-mgcp
> ccm-manager redundant-host 10.104.21.109
> ccm-manager mgcp
> ccm-manager music-on-hold
> ccm-manager config server 10.104.21.109
> ccm-manager config
> !
> mgcp
> mgcp call-agent 10.104.21.121 2427 service-type mgcp version 0.1
> mgcp dtmf-relay voip codec all mode out-of-band
> mgcp rtp unreachable timeout 1000 action notify
> mgcp modem passthrough voip mode nse
> mgcp package-capability rtp-package
> no mgcp package-capability res-package
> mgcp package-capability sst-package
> no mgcp package-capability fxr-package
> mgcp package-capability pre-package
> no mgcp timer receive-rtcp
> mgcp sdp simple
> mgcp rtp payload-type g726r16 static
> !
> mgcp profile default
> !
> !
> dial-peer cor custom
>  name uog_oncampus_pt
>  name uog_local_pt
>  name uog_freeservices_pt
>  name uog_tollfree_pt
>  name uog_payservices_pt
>  name uog_longdist_pt
>  name uog_block_highrisk_pt
> !
> !
> dial-peer cor list uogres-local-css
>  member uog_oncampus_pt
>  member uog_local_pt
>  member uog_tollfree_pt
> !
> dial-peer cor list uogbus-oncampus-css
>  member uog_oncampus_pt
> !
> dial-peer cor list uogbus-local-css
>  member uog_oncampus_pt
>  member uog_local_pt
>  member uog_freeservices_pt
>  member uog_tollfree_pt
> !
> dial-peer cor list uogbus-longdist-css
>  member uog_oncampus_pt
>  member uog_local_pt
>  member uog_freeservices_pt
>  member uog_tollfree_pt
>  member uog_payservices_pt
>  member uog_longdist_pt
>  member uog_block_highrisk_pt
> !
> dial-peer cor list uogdev-local-css
>  member uog_local_pt
> !
> dial-peer cor list uogdev-freeservices-css
>  member uog_freeservices_pt
> !
> dial-peer cor list uogdev-tollfree-css
>  member uog_tollfree_pt
> !
> dial-peer cor list uogdev-payservices-css
>  member uog_payservices_pt
> !
> dial-peer cor list uogdev-longdist-css
>  member uog_longdist_pt
> !
> dial-peer cor list uogdev-block-highrisk-css
>  member uog_block_highrisk_pt
> !
> !
> dial-peer voice 929 pots
>  destination-pattern 9[2-9].........
>  clid network-number 5198241212
>  port 0/1/0:23
>  forward-digits 10
> !
> dial-peer voice 61901 voip
>  destination-pattern 61901
>  session protocol sipv2
>  session target ipv4:10.104.126.100
>  dtmf-relay sip-notify
>  codec g711ulaw
>  no vad
> !
> dial-peer voice 98240022 pots
>  destination-pattern 95198240022
>  port 0/0/0:23
>  forward-digits 10
> !
> dial-peer voice 1 pots
>  translation-profile incoming kc_did_profile
>  incoming called-number .
>  direct-inward-dial
>  port 0/0/0:23
> !
> dial-peer voice 2 pots
>  translation-profile incoming kc_did_profile
>  incoming called-number .
>  direct-inward-dial
>  port 0/1/0:23
> !
> dial-peer voice 9767 pots
>  destination-pattern 95197671060
>  port 0/1/0:23
>  forward-digits 10
> !
> dial-peer voice 9129902 pots
>  translation-profile outgoing uogdev_block_highrisk_900_vtp
>  destination-pattern 91[2-9].........
>  forward-digits 11
> !
> dial-peer voice 99901 pots
>  translation-profile incoming 999
>  translation-profile outgoing 999
>  destination-pattern 914165551212
>  port 0/1/0:23
>  forward-digits 11
> !
> !
> num-exp 96132589... 61...
> num-exp 61000 61901
> num-exp 96132581515 61901
> num-exp 96132581212 61004
> num-exp 96135551414 61901
> num-exp 95198400105 61009
> !
> call-manager-fallback
>  secondary-dialtone 9
>  max-conferences 4 gain -6
>  transfer-system local-consult
>  limit-dn 7910 1
>  limit-dn 7935 1
>  limit-dn 7940 2
>  limit-dn 7960 6
>  limit-dn 7970 8
>  timeouts interdigit 15
>  ip source-address 10.104.126.1 port 2000 strict-match
>  max-ephones 336
>  max-dn 500 dual-line preference 5
>  system message primary UofG Link Down. Ltd Service
>  keepalive 60
>  time-zone 12
>  time-format 24
>  cor incoming uogres-local-css default
>  cor incoming uogbus-longdist-css 1 61001 - 61999
>  cor incoming uogbus-oncampus-css 2 61101
> !
> banner login ^CAccess Restricted to CCS Staff Only. All Access Att
>
> vgw-kemptville#
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR
>
>
> ----- Original Message -----
> From: Jonathan Charles
>
> To: Lelio Fulgenzi
> Cc: ciscovoip
> Sent: Friday, May 25, 2007 8:57 PM
> Subject: Re: [cisco-voip] MGCP Fallback time...
>
> Here it is...
>
>
> Current configuration : 7664 bytes
> !
> version 12.4
> service timestamps debug datetime msec
> service timestamps log datetime msec
> no service password-encryption
> !
> hostname PINKY1
> !
> boot-start-marker
> boot-end-marker
> !
> logging buffered 1000000 warnings
> enable secret 5 $1$OGFk$TCpZe0TS9B4anq.Q6MJIQ0
> !
> no aaa new-model
> !
> resource policy
> !
> network-clock-participate wic 0
> ip subnet-zero
> !
> !
> ip cef
> !
> !
> no ip domain lookup
> ip domain name amli.com
> isdn switch-type primary-ni
> !
> voice-card 0
>  no dspfarm
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> !
> application
>  global
>   service alternate Default
>  !
> !
> !
> crypto pki trustpoint TP-self-signed-2413116866
>  enrollment selfsigned
>  subject-name cn=IOS-Self-Signed-Certificate-2413116866
>  revocation-check none
>  rsakeypair TP-self-signed-2413116866
> !
> !
> crypto pki certificate chain TP-self-signed-2413116866
>  certificate self-signed 01
>   3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
>   31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
>   69666963 6174652D 32343133 31313638 3636301E 170D3037 30353233 31363136
>   35385A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
>   4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 34313331
>   31363836 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
>   8100C7DB 31404006 EEE793F7 398F1A35 07509CDB D444F92D 79FF5F4A 73A01D89
>   B77EF538 DF19058C CCE57B9C ABF0597B DD60FAE8 3CD96D2E DD46A581 15B15F38
>   DC3368D4 074A66F7 9E484DED E36394F6 05EA9FDA F81632FC 31D90520 256D497E
>   CE8F91E4 042BD4E6 824C0DE6 F3107769 F0FFA0E0 65321D02 025D7F33 65C6DDE4
>   027F0203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603
>   551D1104 18301682 14414D4C 4944414C 32383131 2E616D6C 692E636F 6D301F06
>   03551D23 04183016 80144221 418BAB16 E9B5F101 F02AA6B0 0E6B618D 83BA301D
>   0603551D 0E041604 14422141 8BAB16E9 B5F101F0 2AA6B00E 6B618D83 BA300D06
>   092A8648 86F70D01 01040500 03818100 905A4A1F 2A2F0BD5 6F3E7634 C6A122D1
>   5EB8D0D3 11AC6A79 6C70001C 3E4D1923 6EDE6337 738A9FE1 3B9574C7 CFE4050B
>   67CD8790 BC4F8EA1 5C209CB8 8CC175E7 8E27CFFE E3492ADD 874CEBF1 080244B0
>   90D92709 B82A7B48 F3EF1DA5 6A32664E 00794B9A 5DBE2060 FDB89303 E0692D39
>   4884D825 FB237370 82321596 A079BB75
>   quit
> username cisco privilege 15 secret 5 $1$sBAo$/JviEK.cbQuLpFFEKRRsb/
> !
> !
> controller T1 0/0/0
>  framing esf
>  linecode b8zs
>  pri-group timeslots 1-24 service mgcp
> !
> !
> !
> interface FastEthernet0/0
>  description IP VOICE SUBNET
>  ip address 172.27.128.2 255.255.255.0
>  duplex auto
>  speed auto
> !
> interface FastEthernet0/1
>  no ip address
>  shutdown
>  duplex auto
>  speed auto
> !
> interface Serial0/0/0:23
>  no ip address
>  isdn switch-type primary-ni
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
> !
> ip classless
> ip route 0.0.0.0 0.0.0.0 172.27.128.1
> !
> !
> ip http server
> ip http access-class 23
> ip http authentication local
> ip http secure-server
> ip http timeout-policy idle 60 life 86400 requests 10000
> !
> access-list 23 permit 10.10.10.0 0.0.0.7
> !
> !
> !
> control-plane
> !
> !
> !
> voice-port 0/0/0:23
>  input gain -3
>  output attenuation 3
>  cptone C1
> !
> voice-port 0/1/0
>  input gain -3
>  cptone C1
>  timing hookflash-out 50
> !
> voice-port 0/1/1
> !
> voice-port 0/1/2
> !
> voice-port 0/1/3
> !
> ccm-manager fallback-mgcp
> ccm-manager redundant-host 172.28.2.5
> ccm-manager mgcp
> ccm-manager music-on-hold
> ccm-manager config server 172.29.128.4
> ccm-manager config
> ccm-manager download-tones
> !
> mgcp
> mgcp call-agent 172.29.128.5 2427 service-type mgcp version 0.1
> mgcp dtmf-relay voip codec all mode out-of-band
> mgcp rtp unreachable timeout 1000 action notify
> mgcp modem passthrough voip mode nse
> mgcp package-capability rtp-package
> no mgcp package-capability res-package
> mgcp package-capability sst-package
> no mgcp package-capability fxr-package
> mgcp package-capability pre-package
> no mgcp timer receive-rtcp
> mgcp sdp simple
> mgcp fax rate 9600
> mgcp rtp payload-type g726r16 static
> !
> mgcp profile default
> !
> !
> !
> dial-peer voice 100 pots
>  incoming called-number .
>  direct-inward-dial
>  port 0/0/0:23
> !
> dial-peer voice 101 pots
>  incoming called-number .
>  direct-inward-dial
> !
> dial-peer voice 103 pots
>  preference 2
>  destination-pattern 9[2-9]......
>  progress_ind alert enable 8
>  progress_ind progress enable 8
>  progress_ind connect enable 8
>  direct-inward-dial
>  port 0/0/0:23
>  forward-digits 7
> !
> dial-peer voice 104 pots
>  preference 1
>  destination-pattern 91[2-9]..[2-9]......
>  progress_ind alert enable 8
>  progress_ind progress enable 8
>  progress_ind connect enable 8
>  direct-inward-dial
>  port 0/0/0:23
>  forward-digits 11
> !
> dial-peer voice 107 pots
>  preference 2
>  destination-pattern 9[369]11
>  progress_ind alert enable 8
>  progress_ind progress enable 8
>  progress_ind connect enable 8
>  incoming called-number .
>  direct-inward-dial
>  forward-digits 3
> !
> dial-peer voice 110 pots
>  preference 1
>  destination-pattern 9011T
>  incoming called-number .
>  direct-inward-dial
>  port 0/0/0:23
>  prefix 011
> !
> dial-peer voice 111 pots
>  preference 2
>  destination-pattern 9011T
>  incoming called-number .
>  direct-inward-dial
>  port 0/0/0:23
>  prefix 011
> !
> dial-peer voice 112 pots
>  preference 2
>  destination-pattern 9972.......
>  incoming called-number .
>  direct-inward-dial
>  port 0/0/0:23
>  forward-digits 10
> !
> dial-peer voice 113 pots
>  preference 2
>  destination-pattern 9214.......
>  incoming called-number .
>  direct-inward-dial
>  port 0/0/0:23
>  forward-digits 10
> !
> dial-peer voice 114 pots
>  preference 2
>  destination-pattern 9469.......
>  incoming called-number .
>  direct-inward-dial
>  port 0/0/0:23
>  prefix 011
> !
> dial-peer voice 115 pots
>  preference 2
>  destination-pattern 9817.......
>  incoming called-number .
>  direct-inward-dial
>  port 0/0/0:23
>  forward-digits 10
> !
> dial-peer voice 999010 pots
>  service mgcpapp
>  port 0/1/0
> !
> dial-peer voice 116 pots
>  preference 2
>  destination-pattern 911
>  incoming called-number .
>  direct-inward-dial
>  port 0/1/0
>  forward-digits 3
> !
> dial-peer voice 117 pots
>  preference 2
>  destination-pattern 9911
>  incoming called-number .
>  direct-inward-dial
>  port 0/1/0
>  forward-digits 3
> !
> dial-peer voice 102 pots
>  preference 1
>  destination-pattern 9[2-9]......
>  progress_ind alert enable 8
>  progress_ind progress enable 8
>  progress_ind connect enable 8
>  direct-inward-dial
>  port 0/0/0:23
>  forward-digits 7
> !
> dial-peer voice 993310 pots
>  service mgcpapp
>  port 0/0/0:23
> !
> dial-peer voice 993311 pots
>  service mgcpapp
>  port 0/1/0
> !
> !
> !
> call-manager-fallback
>  max-conferences 8 gain -6
>  timeouts interdigit 5
>  ip source-address 172.27.128.2 port 2000
>  max-ephones 42
>  max-dn 144
>  default-destination 6712
>  voicemail 8000
> !
> banner login ^CCC
> -----------------------------------------------------------------------
> This system and its data contents are the property of Pinky the Wonder Sperm
>  and/or its subsidiaries,and contains confidential and
> proprietary information.  Any unauthorized use of this system or data
> is strictly prohibited and may subject you to disciplinary action, up
> to and including termination, civil penalties, and/or criminal
> prosecution and penalties.  Users have no expectation of privacy
> relating to the use of the system.  By using the system, you expressly
> consent to the monitoring and review of your use of the system and all
> data viewed, entered, created, received, stored, or transmitted through
> the system.
> -----------------------------------------------------------------------
> ^C
> !
> line con 0
>  exec-timeout 0 0
>  password
>  logging synchronous
>  login
> line aux 0
>
>
> On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > not sure if this is what's happening in your case, but i had a default
> route
> > in my original setup that caused a delay. i had to put some ACLs on it to
> > make sure it didn't try to connect to the ccm server over a different
> route
> > (which was blocked somewhere else).
> >
> > just a thought.
> >
> > maybe you can post your config.
> >
> > ----- Original Message -----
> > From: "Jonathan Charles" <jonvoip at gmail.com>
> > To: "ciscovoip" <cisco-voip at puck.nether.net>
> > Sent: Friday, May 25, 2007 8:34 PM
> > Subject: [cisco-voip] MGCP Fallback time...
> >
> >
> > >I have a new SRST router (MGCP back to CCM)...
> > >
> > > When we yank the WAN, it takes about 13 minutes for the router to
> > > realize that CCM is down and kill the PRI... and revert to H.323 mode.
> > >
> > > If you look at the attached log file, you can see that the phones are
> > > registering to the router, and a full 13 minutes later the PRI goes
> > > down...
> > >
> > > Is there a way to change this to make it quicker?
> > >
> > > IOS is 12.4.3g
> > >
> > >
> > >
> > >
> > > Jonathan
> > >
> >
> >
> >
> --------------------------------------------------------------------------------
> >
> >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
>

From aelnagar at ACT-EG.COM  Tue May 29 10:29:49 2007
From: aelnagar at ACT-EG.COM (Ahmed Elnagar)
Date: Tue, 29 May 2007 17:29:49 +0300
Subject: [cisco-voip] PDA or 3rd Party Phone with CM 4.2
References: <88C8D019DA547C4BABECB9A806ED22EDDB3285@mail.ACT-EG.COM>
	<C1FE15183DA37645BC0633BC604E44F005AD638F@USNAEXCH.na.didata.local>
Message-ID: <88C8D019DA547C4BABECB9A806ED22ED88A089@mail.ACT-EG.COM>

I found a strange statement on Avaya that there H323 phones will only work with there call proecssing server, and even for the SIP is the same. also I dont know why cisco charges for 3rd party phones it is not logic at all.
 
Thanks and Best Regards

Ahmed A. Elnagar
Network Field Engineer
 
Advanced Computer Technology (ACT)
16 Fawzy Ramah St.Off Shehab St.Mohandessin, Giza, Egypt 
Postal Code:12411 Cairo Egypt

Mob: +2010-2833868
Website: www.act-eg.com
E-mail: aelnagar at act-eg.com

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Matt Slaga (US)
Sent: Tue 29-May-07 3:00 PM
To: Ahmed Elnagar; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PDA or 3rd Party Phone with CM 4.2



Have you tried configuring the avaya phone as an H323 client in CCM?  When you add a new phone, there is an option for H323 client.

 

Just think, when you upgrade to 5.x, in addition to the $115 for ipblue you will have to buy either three (basic) or six (advanced) DLU's as well ($150-$300).

 

 

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmed Elnagar
Sent: Monday, May 28, 2007 2:38 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PDA or 3rd Party Phone with CM 4.2

 

Hello all;

 

A customer is asking us to run Avaya H323 phone and a PDA with callmanager 4.2, for the PDA I found a software that runs SCCP on the PDA www.ipblue.com <http://www.ipblue.com/>  but the licensee is 115 $ for each PDA (which is very costly for the customer) and for Avaya I don't know how to install it with the system and if this is available or not, anyone has information about this??

________________________________

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From joerg.wesely at intact-is.com  Tue May 29 10:38:45 2007
From: joerg.wesely at intact-is.com (=?ISO-8859-15?Q?J=F6rg_Wesely?=)
Date: Tue, 29 May 2007 16:38:45 +0200
Subject: [cisco-voip] CM 4.1(3) -> CCME -> CyberData ceiling speaker: Can
	this work?
Message-ID: <f3hdur$a8u$1@sea.gmane.org>

Hello,

we've got a CM 4.1(3) cluster here. Now, there's a need for a simple 
overhead  paging system. Would the following approach work?

- A CCME is connected to the CM using an intercluster trunk.
- A special ephone-dn is configured on the cme using the "feed ip 
<multicast address> <port>
- Several CyberData VOIP ceiling speakers (www.cyberdata.net)are 
configured to the multicast address above.

When dialing the special dn's number is on a phone at the CM and routed 
     through the intercluster trunk, an announcement can be made.

I would be happy if I could avoid a third-party paging application on 
the CM.

best regards

J?rg


From jonvoip at gmail.com  Tue May 29 10:47:25 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 29 May 2007 09:47:25 -0500
Subject: [cisco-voip] CM 4.1(3) -> CCME -> CyberData ceiling speaker:
	Can this work?
In-Reply-To: <f3hdur$a8u$1@sea.gmane.org>
References: <f3hdur$a8u$1@sea.gmane.org>
Message-ID: <5d093f9a0705290747y75c5b71he0b2f61403f284bd@mail.gmail.com>

The paging-dn feature on CME uses a multicast address... if the
speaker really can play-out multicast, it should work fine.

I would get one and test it, and return it if it doesn't work...



Jonathan

On 5/29/07, J?rg Wesely <joerg.wesely at intact-is.com> wrote:
> Hello,
>
> we've got a CM 4.1(3) cluster here. Now, there's a need for a simple
> overhead  paging system. Would the following approach work?
>
> - A CCME is connected to the CM using an intercluster trunk.
> - A special ephone-dn is configured on the cme using the "feed ip
> <multicast address> <port>
> - Several CyberData VOIP ceiling speakers (www.cyberdata.net)are
> configured to the multicast address above.
>
> When dialing the special dn's number is on a phone at the CM and routed
>      through the intercluster trunk, an announcement can be made.
>
> I would be happy if I could avoid a third-party paging application on
> the CM.
>
> best regards
>
> J?rg
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jsteinberg at gmail.com  Tue May 29 11:09:59 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 29 May 2007 11:09:59 -0400
Subject: [cisco-voip] MGCP Fallback time...
In-Reply-To: <5d093f9a0705290730i7b7686cfi5ffb526156884390@mail.gmail.com>
References: <5d093f9a0705251734n71238a7ch7cbcb676e9de9c7c@mail.gmail.com>
	<001101c79f2f$646e0680$c901a8c0@pdp11>
	<5d093f9a0705251757s13fc34abr3c48fa32c265472f@mail.gmail.com>
	<000c01c79f3b$089a8660$47196883@cfs.uoguelph.ca>
	<5d093f9a0705290730i7b7686cfi5ffb526156884390@mail.gmail.com>
Message-ID: <f1b65f880705290809q7bf63e6s4b2eda78d0b4bb73@mail.gmail.com>

switchback timer only affects switchover/switchback between CCM nodes.
 It doesn't alter timing between CCM and SRST.

Have you tried a different IOS rev?  13 minutes seems awfully long.

Justin

On 5/29/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> I am debating on setting the switchback time to immediate... but I
> don't think that will have any effect.
>
>
> Jonathan
>
> On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> >
> >
> > well, your config is darn close to mine. here's mine to see the difference.
> > you might want to try a few things I have? timeout under the
> > call-manager-fallback?
> >
> > vgw-kemptville#show running-config
> > Building configuration...
> >
> > Current configuration : 9987 bytes
> > !
> > version 12.4
> > service timestamps debug datetime localtime
> > service timestamps log datetime localtime
> > service password-encryption
> > !
> > hostname vgw-kemptville
> > !
> > boot-start-marker
> > boot system flash:c3845-spservicesk9-mz.124-11.T2.bin
> > boot-end-marker
> > !
> > card type t1 0 0
> > card type t1 0 1
> > logging buffered 4096
> > logging console informational
> > enable secret <snip>
> > !
> > aaa new-model
> > !
> > !
> > aaa authentication login default local group tacacs+ enable
> > aaa authentication login virt-users group tacacs+ enable
> > aaa authentication login console group tacacs+ enable
> > !
> > !
> > aaa session-id common
> > clock timezone EST -5
> > clock summer-time EDT recurring
> > network-clock-participate wic 0
> > network-clock-participate wic 1
> > no ip dhcp use vrf connected
> > ip dhcp excluded-address 10.104.126.99
> > ip dhcp excluded-address 10.104.126.100
> > !
> > ip dhcp pool dhcp_vlan826
> >    network 10.104.126.0 255.255.255.128
> >    option 150 ip 10.104.21.121
> >    default-router 10.104.126.1
> > !
> > !
> > ip cef
> > !
> > !
> > multilink bundle-name authenticated
> > !
> > isdn switch-type primary-dms100
> > voice-card 0
> >  no dspfarm
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > voice translation-rule 613
> >  rule 1 /\(.+\)/ /9\1/
> > !
> > voice translation-rule 900
> >  rule 1 /^91809......./ /918090000000/
> >  rule 2 /^91829......./ /918290000000/
> >  rule 3 /^91900......./ /919000000000/
> >  rule 4 /^91...976..../ /919760000000/
> >  rule 5 /^91976..../ /919760000000/
> >  rule 6 /^9976..../ /919760000000/
> > !
> > voice translation-rule 1112
> >  rule 1 reject /914165551212/
> > !
> > !
> > voice translation-profile 1112
> > !
> > voice translation-profile 999
> >  translate called 1112
> > !
> > voice translation-profile kc_did_profile
> >  translate called 613
> > !
> > voice translation-profile uogdev_block_highrisk_900_vtp
> >  translate called 900
> > !
> > !
> > !
> > crypto pki trustpoint TP-self-signed-3156109335
> >  enrollment selfsigned
> >  subject-name cn=IOS-Self-Signed-Certificate-3156109335
> >  revocation-check none
> >  rsakeypair TP-self-signed-3156109335
> > !
> > !
> > crypto pki certificate chain TP-self-signed-3156109335
> >  certificate self-signed 01
> > <snip>
> >   quit
> > !
> > !
> > vtp domain 3845
> > vtp mode transparent
> > !
> > !
> > controller T1 0/0/0
> >  framing esf
> >  linecode b8zs
> >  pri-group timeslots 1-24 service mgcp
> > !
> > controller T1 0/1/0
> >  framing esf
> >  linecode b8zs
> >  pri-group timeslots 1-24 service mgcp
> > !
> > vlan 129
> > !
> > vlan 725
> >  name voice_10.104.125.0/25
> > !
> > vlan 826
> >  name voice_10.104.126.0/25
> > !
> > translation-rule 1112
> > !
> > !
> > !
> > !
> > !
> > interface GigabitEthernet0/0
> >  description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
> >  no ip address
> >  duplex auto
> >  speed auto
> >  media-type rj45
> >  no keepalive
> > !
> > interface GigabitEthernet0/0.129
> >  encapsulation dot1Q 129 native
> >  ip address 192.168.129.66 255.255.255.0
> >  ip access-group vlan129_in in
> >  ip access-group vlan129_out out
> > !
> > interface GigabitEthernet0/0.725
> >  encapsulation dot1Q 725
> >  ip address 10.104.125.99 255.255.255.128
> > !
> > interface GigabitEthernet0/0.826
> >  encapsulation dot1Q 826
> >  ip address 10.104.126.1 255.255.255.128
> > !
> > interface Service-Engine0/0
> >  ip unnumbered GigabitEthernet0/0.826
> >  service-module ip address 10.104.126.100 255.255.255.128
> >  service-module ip default-gateway 10.104.126.1
> > !
> > interface GigabitEthernet0/1
> >  no ip address
> >  shutdown
> >  duplex auto
> >  speed auto
> >  media-type rj45
> >  no keepalive
> > !
> > interface Serial0/0/0:23
> >  no ip address
> >  encapsulation hdlc
> >  isdn switch-type primary-dms100
> >  isdn incoming-voice voice
> >  isdn bind-l3 ccm-manager
> >  no cdp enable
> > !
> > interface Serial0/1/0:23
> >  no ip address
> >  encapsulation hdlc
> >  isdn switch-type primary-dms100
> >  isdn incoming-voice voice
> >  isdn bind-l3 ccm-manager
> >  no cdp enable
> > !
> > ip default-gateway 192.168.129.1
> > ip route 0.0.0.0 0.0.0.0 192.168.129.1
> > ip route 10.0.0.0 255.0.0.0 10.104.125.1
> > ip route 10.104.126.100 255.255.255.255 Service-Engine0/0
> > !
> > !
> > ip http server
> > ip http authentication local
> > ip http secure-server
> > ip http timeout-policy idle 60 life 86400 requests 10000
> > !
> > ip access-list extended vlan129_in
> >  deny   ip any 10.0.0.0 0.255.255.255
> >  deny   ip 10.0.0.0 0.255.255.255 any
> >  permit ip any any
> > ip access-list extended vlan129_out
> >  deny   ip 10.0.0.0 0.255.255.255 any
> >  deny   ip any 10.0.0.0 0.255.255.255
> >  permit ip any any
> > !
> > logging 192.168.20.5
> > !
> > !
> > tacacs-server host 192.168.20.5
> > tacacs-server directed-request
> > tacacs-server key <snip>
> > !
> > control-plane
> > !
> > !
> > !
> > voice-port 0/0/0:23
> > !
> > voice-port 0/1/0:23
> > !
> > ccm-manager fallback-mgcp
> > ccm-manager redundant-host 10.104.21.109
> > ccm-manager mgcp
> > ccm-manager music-on-hold
> > ccm-manager config server 10.104.21.109
> > ccm-manager config
> > !
> > mgcp
> > mgcp call-agent 10.104.21.121 2427 service-type mgcp version 0.1
> > mgcp dtmf-relay voip codec all mode out-of-band
> > mgcp rtp unreachable timeout 1000 action notify
> > mgcp modem passthrough voip mode nse
> > mgcp package-capability rtp-package
> > no mgcp package-capability res-package
> > mgcp package-capability sst-package
> > no mgcp package-capability fxr-package
> > mgcp package-capability pre-package
> > no mgcp timer receive-rtcp
> > mgcp sdp simple
> > mgcp rtp payload-type g726r16 static
> > !
> > mgcp profile default
> > !
> > !
> > dial-peer cor custom
> >  name uog_oncampus_pt
> >  name uog_local_pt
> >  name uog_freeservices_pt
> >  name uog_tollfree_pt
> >  name uog_payservices_pt
> >  name uog_longdist_pt
> >  name uog_block_highrisk_pt
> > !
> > !
> > dial-peer cor list uogres-local-css
> >  member uog_oncampus_pt
> >  member uog_local_pt
> >  member uog_tollfree_pt
> > !
> > dial-peer cor list uogbus-oncampus-css
> >  member uog_oncampus_pt
> > !
> > dial-peer cor list uogbus-local-css
> >  member uog_oncampus_pt
> >  member uog_local_pt
> >  member uog_freeservices_pt
> >  member uog_tollfree_pt
> > !
> > dial-peer cor list uogbus-longdist-css
> >  member uog_oncampus_pt
> >  member uog_local_pt
> >  member uog_freeservices_pt
> >  member uog_tollfree_pt
> >  member uog_payservices_pt
> >  member uog_longdist_pt
> >  member uog_block_highrisk_pt
> > !
> > dial-peer cor list uogdev-local-css
> >  member uog_local_pt
> > !
> > dial-peer cor list uogdev-freeservices-css
> >  member uog_freeservices_pt
> > !
> > dial-peer cor list uogdev-tollfree-css
> >  member uog_tollfree_pt
> > !
> > dial-peer cor list uogdev-payservices-css
> >  member uog_payservices_pt
> > !
> > dial-peer cor list uogdev-longdist-css
> >  member uog_longdist_pt
> > !
> > dial-peer cor list uogdev-block-highrisk-css
> >  member uog_block_highrisk_pt
> > !
> > !
> > dial-peer voice 929 pots
> >  destination-pattern 9[2-9].........
> >  clid network-number 5198241212
> >  port 0/1/0:23
> >  forward-digits 10
> > !
> > dial-peer voice 61901 voip
> >  destination-pattern 61901
> >  session protocol sipv2
> >  session target ipv4:10.104.126.100
> >  dtmf-relay sip-notify
> >  codec g711ulaw
> >  no vad
> > !
> > dial-peer voice 98240022 pots
> >  destination-pattern 95198240022
> >  port 0/0/0:23
> >  forward-digits 10
> > !
> > dial-peer voice 1 pots
> >  translation-profile incoming kc_did_profile
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/0/0:23
> > !
> > dial-peer voice 2 pots
> >  translation-profile incoming kc_did_profile
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/1/0:23
> > !
> > dial-peer voice 9767 pots
> >  destination-pattern 95197671060
> >  port 0/1/0:23
> >  forward-digits 10
> > !
> > dial-peer voice 9129902 pots
> >  translation-profile outgoing uogdev_block_highrisk_900_vtp
> >  destination-pattern 91[2-9].........
> >  forward-digits 11
> > !
> > dial-peer voice 99901 pots
> >  translation-profile incoming 999
> >  translation-profile outgoing 999
> >  destination-pattern 914165551212
> >  port 0/1/0:23
> >  forward-digits 11
> > !
> > !
> > num-exp 96132589... 61...
> > num-exp 61000 61901
> > num-exp 96132581515 61901
> > num-exp 96132581212 61004
> > num-exp 96135551414 61901
> > num-exp 95198400105 61009
> > !
> > call-manager-fallback
> >  secondary-dialtone 9
> >  max-conferences 4 gain -6
> >  transfer-system local-consult
> >  limit-dn 7910 1
> >  limit-dn 7935 1
> >  limit-dn 7940 2
> >  limit-dn 7960 6
> >  limit-dn 7970 8
> >  timeouts interdigit 15
> >  ip source-address 10.104.126.1 port 2000 strict-match
> >  max-ephones 336
> >  max-dn 500 dual-line preference 5
> >  system message primary UofG Link Down. Ltd Service
> >  keepalive 60
> >  time-zone 12
> >  time-format 24
> >  cor incoming uogres-local-css default
> >  cor incoming uogbus-longdist-css 1 61001 - 61999
> >  cor incoming uogbus-oncampus-css 2 61101
> > !
> > banner login ^CAccess Restricted to CCS Staff Only. All Access Att
> >
> > vgw-kemptville#
> >
> > --------------------------------------------------------------------------------
> > Lelio Fulgenzi, B.A.
> > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR
> >
> >
> > ----- Original Message -----
> > From: Jonathan Charles
> >
> > To: Lelio Fulgenzi
> > Cc: ciscovoip
> > Sent: Friday, May 25, 2007 8:57 PM
> > Subject: Re: [cisco-voip] MGCP Fallback time...
> >
> > Here it is...
> >
> >
> > Current configuration : 7664 bytes
> > !
> > version 12.4
> > service timestamps debug datetime msec
> > service timestamps log datetime msec
> > no service password-encryption
> > !
> > hostname PINKY1
> > !
> > boot-start-marker
> > boot-end-marker
> > !
> > logging buffered 1000000 warnings
> > enable secret 5 $1$OGFk$TCpZe0TS9B4anq.Q6MJIQ0
> > !
> > no aaa new-model
> > !
> > resource policy
> > !
> > network-clock-participate wic 0
> > ip subnet-zero
> > !
> > !
> > ip cef
> > !
> > !
> > no ip domain lookup
> > ip domain name amli.com
> > isdn switch-type primary-ni
> > !
> > voice-card 0
> >  no dspfarm
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > application
> >  global
> >   service alternate Default
> >  !
> > !
> > !
> > crypto pki trustpoint TP-self-signed-2413116866
> >  enrollment selfsigned
> >  subject-name cn=IOS-Self-Signed-Certificate-2413116866
> >  revocation-check none
> >  rsakeypair TP-self-signed-2413116866
> > !
> > !
> > crypto pki certificate chain TP-self-signed-2413116866
> >  certificate self-signed 01
> >   3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
> >   31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
> >   69666963 6174652D 32343133 31313638 3636301E 170D3037 30353233 31363136
> >   35385A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
> >   4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 34313331
> >   31363836 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
> >   8100C7DB 31404006 EEE793F7 398F1A35 07509CDB D444F92D 79FF5F4A 73A01D89
> >   B77EF538 DF19058C CCE57B9C ABF0597B DD60FAE8 3CD96D2E DD46A581 15B15F38
> >   DC3368D4 074A66F7 9E484DED E36394F6 05EA9FDA F81632FC 31D90520 256D497E
> >   CE8F91E4 042BD4E6 824C0DE6 F3107769 F0FFA0E0 65321D02 025D7F33 65C6DDE4
> >   027F0203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603
> >   551D1104 18301682 14414D4C 4944414C 32383131 2E616D6C 692E636F 6D301F06
> >   03551D23 04183016 80144221 418BAB16 E9B5F101 F02AA6B0 0E6B618D 83BA301D
> >   0603551D 0E041604 14422141 8BAB16E9 B5F101F0 2AA6B00E 6B618D83 BA300D06
> >   092A8648 86F70D01 01040500 03818100 905A4A1F 2A2F0BD5 6F3E7634 C6A122D1
> >   5EB8D0D3 11AC6A79 6C70001C 3E4D1923 6EDE6337 738A9FE1 3B9574C7 CFE4050B
> >   67CD8790 BC4F8EA1 5C209CB8 8CC175E7 8E27CFFE E3492ADD 874CEBF1 080244B0
> >   90D92709 B82A7B48 F3EF1DA5 6A32664E 00794B9A 5DBE2060 FDB89303 E0692D39
> >   4884D825 FB237370 82321596 A079BB75
> >   quit
> > username cisco privilege 15 secret 5 $1$sBAo$/JviEK.cbQuLpFFEKRRsb/
> > !
> > !
> > controller T1 0/0/0
> >  framing esf
> >  linecode b8zs
> >  pri-group timeslots 1-24 service mgcp
> > !
> > !
> > !
> > interface FastEthernet0/0
> >  description IP VOICE SUBNET
> >  ip address 172.27.128.2 255.255.255.0
> >  duplex auto
> >  speed auto
> > !
> > interface FastEthernet0/1
> >  no ip address
> >  shutdown
> >  duplex auto
> >  speed auto
> > !
> > interface Serial0/0/0:23
> >  no ip address
> >  isdn switch-type primary-ni
> >  isdn incoming-voice voice
> >  isdn bind-l3 ccm-manager
> >  no cdp enable
> > !
> > ip classless
> > ip route 0.0.0.0 0.0.0.0 172.27.128.1
> > !
> > !
> > ip http server
> > ip http access-class 23
> > ip http authentication local
> > ip http secure-server
> > ip http timeout-policy idle 60 life 86400 requests 10000
> > !
> > access-list 23 permit 10.10.10.0 0.0.0.7
> > !
> > !
> > !
> > control-plane
> > !
> > !
> > !
> > voice-port 0/0/0:23
> >  input gain -3
> >  output attenuation 3
> >  cptone C1
> > !
> > voice-port 0/1/0
> >  input gain -3
> >  cptone C1
> >  timing hookflash-out 50
> > !
> > voice-port 0/1/1
> > !
> > voice-port 0/1/2
> > !
> > voice-port 0/1/3
> > !
> > ccm-manager fallback-mgcp
> > ccm-manager redundant-host 172.28.2.5
> > ccm-manager mgcp
> > ccm-manager music-on-hold
> > ccm-manager config server 172.29.128.4
> > ccm-manager config
> > ccm-manager download-tones
> > !
> > mgcp
> > mgcp call-agent 172.29.128.5 2427 service-type mgcp version 0.1
> > mgcp dtmf-relay voip codec all mode out-of-band
> > mgcp rtp unreachable timeout 1000 action notify
> > mgcp modem passthrough voip mode nse
> > mgcp package-capability rtp-package
> > no mgcp package-capability res-package
> > mgcp package-capability sst-package
> > no mgcp package-capability fxr-package
> > mgcp package-capability pre-package
> > no mgcp timer receive-rtcp
> > mgcp sdp simple
> > mgcp fax rate 9600
> > mgcp rtp payload-type g726r16 static
> > !
> > mgcp profile default
> > !
> > !
> > !
> > dial-peer voice 100 pots
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/0/0:23
> > !
> > dial-peer voice 101 pots
> >  incoming called-number .
> >  direct-inward-dial
> > !
> > dial-peer voice 103 pots
> >  preference 2
> >  destination-pattern 9[2-9]......
> >  progress_ind alert enable 8
> >  progress_ind progress enable 8
> >  progress_ind connect enable 8
> >  direct-inward-dial
> >  port 0/0/0:23
> >  forward-digits 7
> > !
> > dial-peer voice 104 pots
> >  preference 1
> >  destination-pattern 91[2-9]..[2-9]......
> >  progress_ind alert enable 8
> >  progress_ind progress enable 8
> >  progress_ind connect enable 8
> >  direct-inward-dial
> >  port 0/0/0:23
> >  forward-digits 11
> > !
> > dial-peer voice 107 pots
> >  preference 2
> >  destination-pattern 9[369]11
> >  progress_ind alert enable 8
> >  progress_ind progress enable 8
> >  progress_ind connect enable 8
> >  incoming called-number .
> >  direct-inward-dial
> >  forward-digits 3
> > !
> > dial-peer voice 110 pots
> >  preference 1
> >  destination-pattern 9011T
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/0/0:23
> >  prefix 011
> > !
> > dial-peer voice 111 pots
> >  preference 2
> >  destination-pattern 9011T
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/0/0:23
> >  prefix 011
> > !
> > dial-peer voice 112 pots
> >  preference 2
> >  destination-pattern 9972.......
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/0/0:23
> >  forward-digits 10
> > !
> > dial-peer voice 113 pots
> >  preference 2
> >  destination-pattern 9214.......
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/0/0:23
> >  forward-digits 10
> > !
> > dial-peer voice 114 pots
> >  preference 2
> >  destination-pattern 9469.......
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/0/0:23
> >  prefix 011
> > !
> > dial-peer voice 115 pots
> >  preference 2
> >  destination-pattern 9817.......
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/0/0:23
> >  forward-digits 10
> > !
> > dial-peer voice 999010 pots
> >  service mgcpapp
> >  port 0/1/0
> > !
> > dial-peer voice 116 pots
> >  preference 2
> >  destination-pattern 911
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/1/0
> >  forward-digits 3
> > !
> > dial-peer voice 117 pots
> >  preference 2
> >  destination-pattern 9911
> >  incoming called-number .
> >  direct-inward-dial
> >  port 0/1/0
> >  forward-digits 3
> > !
> > dial-peer voice 102 pots
> >  preference 1
> >  destination-pattern 9[2-9]......
> >  progress_ind alert enable 8
> >  progress_ind progress enable 8
> >  progress_ind connect enable 8
> >  direct-inward-dial
> >  port 0/0/0:23
> >  forward-digits 7
> > !
> > dial-peer voice 993310 pots
> >  service mgcpapp
> >  port 0/0/0:23
> > !
> > dial-peer voice 993311 pots
> >  service mgcpapp
> >  port 0/1/0
> > !
> > !
> > !
> > call-manager-fallback
> >  max-conferences 8 gain -6
> >  timeouts interdigit 5
> >  ip source-address 172.27.128.2 port 2000
> >  max-ephones 42
> >  max-dn 144
> >  default-destination 6712
> >  voicemail 8000
> > !
> > banner login ^CCC
> > -----------------------------------------------------------------------
> > This system and its data contents are the property of Pinky the Wonder Sperm
> >  and/or its subsidiaries,and contains confidential and
> > proprietary information.  Any unauthorized use of this system or data
> > is strictly prohibited and may subject you to disciplinary action, up
> > to and including termination, civil penalties, and/or criminal
> > prosecution and penalties.  Users have no expectation of privacy
> > relating to the use of the system.  By using the system, you expressly
> > consent to the monitoring and review of your use of the system and all
> > data viewed, entered, created, received, stored, or transmitted through
> > the system.
> > -----------------------------------------------------------------------
> > ^C
> > !
> > line con 0
> >  exec-timeout 0 0
> >  password
> >  logging synchronous
> >  login
> > line aux 0
> >
> >
> > On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > not sure if this is what's happening in your case, but i had a default
> > route
> > > in my original setup that caused a delay. i had to put some ACLs on it to
> > > make sure it didn't try to connect to the ccm server over a different
> > route
> > > (which was blocked somewhere else).
> > >
> > > just a thought.
> > >
> > > maybe you can post your config.
> > >
> > > ----- Original Message -----
> > > From: "Jonathan Charles" <jonvoip at gmail.com>
> > > To: "ciscovoip" <cisco-voip at puck.nether.net>
> > > Sent: Friday, May 25, 2007 8:34 PM
> > > Subject: [cisco-voip] MGCP Fallback time...
> > >
> > >
> > > >I have a new SRST router (MGCP back to CCM)...
> > > >
> > > > When we yank the WAN, it takes about 13 minutes for the router to
> > > > realize that CCM is down and kill the PRI... and revert to H.323 mode.
> > > >
> > > > If you look at the attached log file, you can see that the phones are
> > > > registering to the router, and a full 13 minutes later the PRI goes
> > > > down...
> > > >
> > > > Is there a way to change this to make it quicker?
> > > >
> > > > IOS is 12.4.3g
> > > >
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > >
> > >
> > >
> > --------------------------------------------------------------------------------
> > >
> > >
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> > >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From bcarter at sentinel.com  Tue May 29 11:17:46 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Tue, 29 May 2007 10:17:46 -0500
Subject: [cisco-voip] cisco-voip.org Wiki
Message-ID: <6A0744639B86DD459433108DBF101D26016AED45@MAIL01.sentinel.com>

Under Cisco Unified Communications Manager I added a section for SQL
queries.
 
http://cisco-voip.org/viki/index.php/SQL_Queries
 
 
Bill Carter
Sr. Business Communications Analyst
Sentinel Technologies
(O) 217.391.5015
 
-------------- next part --------------
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URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20070529/d0a80977/attachment.html 

From jonvoip at gmail.com  Tue May 29 11:22:26 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 29 May 2007 10:22:26 -0500
Subject: [cisco-voip] MGCP Fallback time...
In-Reply-To: <5d093f9a0705290821p3582f6a3uba555656c9785f95@mail.gmail.com>
References: <5d093f9a0705251734n71238a7ch7cbcb676e9de9c7c@mail.gmail.com>
	<001101c79f2f$646e0680$c901a8c0@pdp11>
	<5d093f9a0705251757s13fc34abr3c48fa32c265472f@mail.gmail.com>
	<000c01c79f3b$089a8660$47196883@cfs.uoguelph.ca>
	<5d093f9a0705290730i7b7686cfi5ffb526156884390@mail.gmail.com>
	<f1b65f880705290809q7bf63e6s4b2eda78d0b4bb73@mail.gmail.com>
	<5d093f9a0705290821p3582f6a3uba555656c9785f95@mail.gmail.com>
Message-ID: <5d093f9a0705290822u16c9e044m9404370392508d81@mail.gmail.com>

Right now, I am planning on 12.4.13b...


Jonathan

On 5/29/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> I am going to for another site, if it works there, I am going to
> upgrade the other.
>
>
>
> Jonathan
>
> On 5/29/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
> > switchback timer only affects switchover/switchback between CCM nodes.
> >  It doesn't alter timing between CCM and SRST.
> >
> > Have you tried a different IOS rev?  13 minutes seems awfully long.
> >
> > Justin
> >
> > On 5/29/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > I am debating on setting the switchback time to immediate... but I
> > > don't think that will have any effect.
> > >
> > >
> > > Jonathan
> > >
> > > On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > >
> > > >
> > > > well, your config is darn close to mine. here's mine to see the difference.
> > > > you might want to try a few things I have? timeout under the
> > > > call-manager-fallback?
> > > >
> > > > vgw-kemptville#show running-config
> > > > Building configuration...
> > > >
> > > > Current configuration : 9987 bytes
> > > > !
> > > > version 12.4
> > > > service timestamps debug datetime localtime
> > > > service timestamps log datetime localtime
> > > > service password-encryption
> > > > !
> > > > hostname vgw-kemptville
> > > > !
> > > > boot-start-marker
> > > > boot system flash:c3845-spservicesk9-mz.124-11.T2.bin
> > > > boot-end-marker
> > > > !
> > > > card type t1 0 0
> > > > card type t1 0 1
> > > > logging buffered 4096
> > > > logging console informational
> > > > enable secret <snip>
> > > > !
> > > > aaa new-model
> > > > !
> > > > !
> > > > aaa authentication login default local group tacacs+ enable
> > > > aaa authentication login virt-users group tacacs+ enable
> > > > aaa authentication login console group tacacs+ enable
> > > > !
> > > > !
> > > > aaa session-id common
> > > > clock timezone EST -5
> > > > clock summer-time EDT recurring
> > > > network-clock-participate wic 0
> > > > network-clock-participate wic 1
> > > > no ip dhcp use vrf connected
> > > > ip dhcp excluded-address 10.104.126.99
> > > > ip dhcp excluded-address 10.104.126.100
> > > > !
> > > > ip dhcp pool dhcp_vlan826
> > > >    network 10.104.126.0 255.255.255.128
> > > >    option 150 ip 10.104.21.121
> > > >    default-router 10.104.126.1
> > > > !
> > > > !
> > > > ip cef
> > > > !
> > > > !
> > > > multilink bundle-name authenticated
> > > > !
> > > > isdn switch-type primary-dms100
> > > > voice-card 0
> > > >  no dspfarm
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > voice translation-rule 613
> > > >  rule 1 /\(.+\)/ /9\1/
> > > > !
> > > > voice translation-rule 900
> > > >  rule 1 /^91809......./ /918090000000/
> > > >  rule 2 /^91829......./ /918290000000/
> > > >  rule 3 /^91900......./ /919000000000/
> > > >  rule 4 /^91...976..../ /919760000000/
> > > >  rule 5 /^91976..../ /919760000000/
> > > >  rule 6 /^9976..../ /919760000000/
> > > > !
> > > > voice translation-rule 1112
> > > >  rule 1 reject /914165551212/
> > > > !
> > > > !
> > > > voice translation-profile 1112
> > > > !
> > > > voice translation-profile 999
> > > >  translate called 1112
> > > > !
> > > > voice translation-profile kc_did_profile
> > > >  translate called 613
> > > > !
> > > > voice translation-profile uogdev_block_highrisk_900_vtp
> > > >  translate called 900
> > > > !
> > > > !
> > > > !
> > > > crypto pki trustpoint TP-self-signed-3156109335
> > > >  enrollment selfsigned
> > > >  subject-name cn=IOS-Self-Signed-Certificate-3156109335
> > > >  revocation-check none
> > > >  rsakeypair TP-self-signed-3156109335
> > > > !
> > > > !
> > > > crypto pki certificate chain TP-self-signed-3156109335
> > > >  certificate self-signed 01
> > > > <snip>
> > > >   quit
> > > > !
> > > > !
> > > > vtp domain 3845
> > > > vtp mode transparent
> > > > !
> > > > !
> > > > controller T1 0/0/0
> > > >  framing esf
> > > >  linecode b8zs
> > > >  pri-group timeslots 1-24 service mgcp
> > > > !
> > > > controller T1 0/1/0
> > > >  framing esf
> > > >  linecode b8zs
> > > >  pri-group timeslots 1-24 service mgcp
> > > > !
> > > > vlan 129
> > > > !
> > > > vlan 725
> > > >  name voice_10.104.125.0/25
> > > > !
> > > > vlan 826
> > > >  name voice_10.104.126.0/25
> > > > !
> > > > translation-rule 1112
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > interface GigabitEthernet0/0
> > > >  description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
> > > >  no ip address
> > > >  duplex auto
> > > >  speed auto
> > > >  media-type rj45
> > > >  no keepalive
> > > > !
> > > > interface GigabitEthernet0/0.129
> > > >  encapsulation dot1Q 129 native
> > > >  ip address 192.168.129.66 255.255.255.0
> > > >  ip access-group vlan129_in in
> > > >  ip access-group vlan129_out out
> > > > !
> > > > interface GigabitEthernet0/0.725
> > > >  encapsulation dot1Q 725
> > > >  ip address 10.104.125.99 255.255.255.128
> > > > !
> > > > interface GigabitEthernet0/0.826
> > > >  encapsulation dot1Q 826
> > > >  ip address 10.104.126.1 255.255.255.128
> > > > !
> > > > interface Service-Engine0/0
> > > >  ip unnumbered GigabitEthernet0/0.826
> > > >  service-module ip address 10.104.126.100 255.255.255.128
> > > >  service-module ip default-gateway 10.104.126.1
> > > > !
> > > > interface GigabitEthernet0/1
> > > >  no ip address
> > > >  shutdown
> > > >  duplex auto
> > > >  speed auto
> > > >  media-type rj45
> > > >  no keepalive
> > > > !
> > > > interface Serial0/0/0:23
> > > >  no ip address
> > > >  encapsulation hdlc
> > > >  isdn switch-type primary-dms100
> > > >  isdn incoming-voice voice
> > > >  isdn bind-l3 ccm-manager
> > > >  no cdp enable
> > > > !
> > > > interface Serial0/1/0:23
> > > >  no ip address
> > > >  encapsulation hdlc
> > > >  isdn switch-type primary-dms100
> > > >  isdn incoming-voice voice
> > > >  isdn bind-l3 ccm-manager
> > > >  no cdp enable
> > > > !
> > > > ip default-gateway 192.168.129.1
> > > > ip route 0.0.0.0 0.0.0.0 192.168.129.1
> > > > ip route 10.0.0.0 255.0.0.0 10.104.125.1
> > > > ip route 10.104.126.100 255.255.255.255 Service-Engine0/0
> > > > !
> > > > !
> > > > ip http server
> > > > ip http authentication local
> > > > ip http secure-server
> > > > ip http timeout-policy idle 60 life 86400 requests 10000
> > > > !
> > > > ip access-list extended vlan129_in
> > > >  deny   ip any 10.0.0.0 0.255.255.255
> > > >  deny   ip 10.0.0.0 0.255.255.255 any
> > > >  permit ip any any
> > > > ip access-list extended vlan129_out
> > > >  deny   ip 10.0.0.0 0.255.255.255 any
> > > >  deny   ip any 10.0.0.0 0.255.255.255
> > > >  permit ip any any
> > > > !
> > > > logging 192.168.20.5
> > > > !
> > > > !
> > > > tacacs-server host 192.168.20.5
> > > > tacacs-server directed-request
> > > > tacacs-server key <snip>
> > > > !
> > > > control-plane
> > > > !
> > > > !
> > > > !
> > > > voice-port 0/0/0:23
> > > > !
> > > > voice-port 0/1/0:23
> > > > !
> > > > ccm-manager fallback-mgcp
> > > > ccm-manager redundant-host 10.104.21.109
> > > > ccm-manager mgcp
> > > > ccm-manager music-on-hold
> > > > ccm-manager config server 10.104.21.109
> > > > ccm-manager config
> > > > !
> > > > mgcp
> > > > mgcp call-agent 10.104.21.121 2427 service-type mgcp version 0.1
> > > > mgcp dtmf-relay voip codec all mode out-of-band
> > > > mgcp rtp unreachable timeout 1000 action notify
> > > > mgcp modem passthrough voip mode nse
> > > > mgcp package-capability rtp-package
> > > > no mgcp package-capability res-package
> > > > mgcp package-capability sst-package
> > > > no mgcp package-capability fxr-package
> > > > mgcp package-capability pre-package
> > > > no mgcp timer receive-rtcp
> > > > mgcp sdp simple
> > > > mgcp rtp payload-type g726r16 static
> > > > !
> > > > mgcp profile default
> > > > !
> > > > !
> > > > dial-peer cor custom
> > > >  name uog_oncampus_pt
> > > >  name uog_local_pt
> > > >  name uog_freeservices_pt
> > > >  name uog_tollfree_pt
> > > >  name uog_payservices_pt
> > > >  name uog_longdist_pt
> > > >  name uog_block_highrisk_pt
> > > > !
> > > > !
> > > > dial-peer cor list uogres-local-css
> > > >  member uog_oncampus_pt
> > > >  member uog_local_pt
> > > >  member uog_tollfree_pt
> > > > !
> > > > dial-peer cor list uogbus-oncampus-css
> > > >  member uog_oncampus_pt
> > > > !
> > > > dial-peer cor list uogbus-local-css
> > > >  member uog_oncampus_pt
> > > >  member uog_local_pt
> > > >  member uog_freeservices_pt
> > > >  member uog_tollfree_pt
> > > > !
> > > > dial-peer cor list uogbus-longdist-css
> > > >  member uog_oncampus_pt
> > > >  member uog_local_pt
> > > >  member uog_freeservices_pt
> > > >  member uog_tollfree_pt
> > > >  member uog_payservices_pt
> > > >  member uog_longdist_pt
> > > >  member uog_block_highrisk_pt
> > > > !
> > > > dial-peer cor list uogdev-local-css
> > > >  member uog_local_pt
> > > > !
> > > > dial-peer cor list uogdev-freeservices-css
> > > >  member uog_freeservices_pt
> > > > !
> > > > dial-peer cor list uogdev-tollfree-css
> > > >  member uog_tollfree_pt
> > > > !
> > > > dial-peer cor list uogdev-payservices-css
> > > >  member uog_payservices_pt
> > > > !
> > > > dial-peer cor list uogdev-longdist-css
> > > >  member uog_longdist_pt
> > > > !
> > > > dial-peer cor list uogdev-block-highrisk-css
> > > >  member uog_block_highrisk_pt
> > > > !
> > > > !
> > > > dial-peer voice 929 pots
> > > >  destination-pattern 9[2-9].........
> > > >  clid network-number 5198241212
> > > >  port 0/1/0:23
> > > >  forward-digits 10
> > > > !
> > > > dial-peer voice 61901 voip
> > > >  destination-pattern 61901
> > > >  session protocol sipv2
> > > >  session target ipv4:10.104.126.100
> > > >  dtmf-relay sip-notify
> > > >  codec g711ulaw
> > > >  no vad
> > > > !
> > > > dial-peer voice 98240022 pots
> > > >  destination-pattern 95198240022
> > > >  port 0/0/0:23
> > > >  forward-digits 10
> > > > !
> > > > dial-peer voice 1 pots
> > > >  translation-profile incoming kc_did_profile
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > > !
> > > > dial-peer voice 2 pots
> > > >  translation-profile incoming kc_did_profile
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/1/0:23
> > > > !
> > > > dial-peer voice 9767 pots
> > > >  destination-pattern 95197671060
> > > >  port 0/1/0:23
> > > >  forward-digits 10
> > > > !
> > > > dial-peer voice 9129902 pots
> > > >  translation-profile outgoing uogdev_block_highrisk_900_vtp
> > > >  destination-pattern 91[2-9].........
> > > >  forward-digits 11
> > > > !
> > > > dial-peer voice 99901 pots
> > > >  translation-profile incoming 999
> > > >  translation-profile outgoing 999
> > > >  destination-pattern 914165551212
> > > >  port 0/1/0:23
> > > >  forward-digits 11
> > > > !
> > > > !
> > > > num-exp 96132589... 61...
> > > > num-exp 61000 61901
> > > > num-exp 96132581515 61901
> > > > num-exp 96132581212 61004
> > > > num-exp 96135551414 61901
> > > > num-exp 95198400105 61009
> > > > !
> > > > call-manager-fallback
> > > >  secondary-dialtone 9
> > > >  max-conferences 4 gain -6
> > > >  transfer-system local-consult
> > > >  limit-dn 7910 1
> > > >  limit-dn 7935 1
> > > >  limit-dn 7940 2
> > > >  limit-dn 7960 6
> > > >  limit-dn 7970 8
> > > >  timeouts interdigit 15
> > > >  ip source-address 10.104.126.1 port 2000 strict-match
> > > >  max-ephones 336
> > > >  max-dn 500 dual-line preference 5
> > > >  system message primary UofG Link Down. Ltd Service
> > > >  keepalive 60
> > > >  time-zone 12
> > > >  time-format 24
> > > >  cor incoming uogres-local-css default
> > > >  cor incoming uogbus-longdist-css 1 61001 - 61999
> > > >  cor incoming uogbus-oncampus-css 2 61101
> > > > !
> > > > banner login ^CAccess Restricted to CCS Staff Only. All Access Att
> > > >
> > > > vgw-kemptville#
> > > >
> > > > --------------------------------------------------------------------------------
> > > > Lelio Fulgenzi, B.A.
> > > > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> > > > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> > > > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > > > "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR
> > > >
> > > >
> > > > ----- Original Message -----
> > > > From: Jonathan Charles
> > > >
> > > > To: Lelio Fulgenzi
> > > > Cc: ciscovoip
> > > > Sent: Friday, May 25, 2007 8:57 PM
> > > > Subject: Re: [cisco-voip] MGCP Fallback time...
> > > >
> > > > Here it is...
> > > >
> > > >
> > > > Current configuration : 7664 bytes
> > > > !
> > > > version 12.4
> > > > service timestamps debug datetime msec
> > > > service timestamps log datetime msec
> > > > no service password-encryption
> > > > !
> > > > hostname PINKY1
> > > > !
> > > > boot-start-marker
> > > > boot-end-marker
> > > > !
> > > > logging buffered 1000000 warnings
> > > > enable secret 5 $1$OGFk$TCpZe0TS9B4anq.Q6MJIQ0
> > > > !
> > > > no aaa new-model
> > > > !
> > > > resource policy
> > > > !
> > > > network-clock-participate wic 0
> > > > ip subnet-zero
> > > > !
> > > > !
> > > > ip cef
> > > > !
> > > > !
> > > > no ip domain lookup
> > > > ip domain name amli.com
> > > > isdn switch-type primary-ni
> > > > !
> > > > voice-card 0
> > > >  no dspfarm
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > !
> > > > application
> > > >  global
> > > >   service alternate Default
> > > >  !
> > > > !
> > > > !
> > > > crypto pki trustpoint TP-self-signed-2413116866
> > > >  enrollment selfsigned
> > > >  subject-name cn=IOS-Self-Signed-Certificate-2413116866
> > > >  revocation-check none
> > > >  rsakeypair TP-self-signed-2413116866
> > > > !
> > > > !
> > > > crypto pki certificate chain TP-self-signed-2413116866
> > > >  certificate self-signed 01
> > > >   3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
> > > >   31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
> > > >   69666963 6174652D 32343133 31313638 3636301E 170D3037 30353233 31363136
> > > >   35385A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
> > > >   4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 34313331
> > > >   31363836 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
> > > >   8100C7DB 31404006 EEE793F7 398F1A35 07509CDB D444F92D 79FF5F4A 73A01D89
> > > >   B77EF538 DF19058C CCE57B9C ABF0597B DD60FAE8 3CD96D2E DD46A581 15B15F38
> > > >   DC3368D4 074A66F7 9E484DED E36394F6 05EA9FDA F81632FC 31D90520 256D497E
> > > >   CE8F91E4 042BD4E6 824C0DE6 F3107769 F0FFA0E0 65321D02 025D7F33 65C6DDE4
> > > >   027F0203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603
> > > >   551D1104 18301682 14414D4C 4944414C 32383131 2E616D6C 692E636F 6D301F06
> > > >   03551D23 04183016 80144221 418BAB16 E9B5F101 F02AA6B0 0E6B618D 83BA301D
> > > >   0603551D 0E041604 14422141 8BAB16E9 B5F101F0 2AA6B00E 6B618D83 BA300D06
> > > >   092A8648 86F70D01 01040500 03818100 905A4A1F 2A2F0BD5 6F3E7634 C6A122D1
> > > >   5EB8D0D3 11AC6A79 6C70001C 3E4D1923 6EDE6337 738A9FE1 3B9574C7 CFE4050B
> > > >   67CD8790 BC4F8EA1 5C209CB8 8CC175E7 8E27CFFE E3492ADD 874CEBF1 080244B0
> > > >   90D92709 B82A7B48 F3EF1DA5 6A32664E 00794B9A 5DBE2060 FDB89303 E0692D39
> > > >   4884D825 FB237370 82321596 A079BB75
> > > >   quit
> > > > username cisco privilege 15 secret 5 $1$sBAo$/JviEK.cbQuLpFFEKRRsb/
> > > > !
> > > > !
> > > > controller T1 0/0/0
> > > >  framing esf
> > > >  linecode b8zs
> > > >  pri-group timeslots 1-24 service mgcp
> > > > !
> > > > !
> > > > !
> > > > interface FastEthernet0/0
> > > >  description IP VOICE SUBNET
> > > >  ip address 172.27.128.2 255.255.255.0
> > > >  duplex auto
> > > >  speed auto
> > > > !
> > > > interface FastEthernet0/1
> > > >  no ip address
> > > >  shutdown
> > > >  duplex auto
> > > >  speed auto
> > > > !
> > > > interface Serial0/0/0:23
> > > >  no ip address
> > > >  isdn switch-type primary-ni
> > > >  isdn incoming-voice voice
> > > >  isdn bind-l3 ccm-manager
> > > >  no cdp enable
> > > > !
> > > > ip classless
> > > > ip route 0.0.0.0 0.0.0.0 172.27.128.1
> > > > !
> > > > !
> > > > ip http server
> > > > ip http access-class 23
> > > > ip http authentication local
> > > > ip http secure-server
> > > > ip http timeout-policy idle 60 life 86400 requests 10000
> > > > !
> > > > access-list 23 permit 10.10.10.0 0.0.0.7
> > > > !
> > > > !
> > > > !
> > > > control-plane
> > > > !
> > > > !
> > > > !
> > > > voice-port 0/0/0:23
> > > >  input gain -3
> > > >  output attenuation 3
> > > >  cptone C1
> > > > !
> > > > voice-port 0/1/0
> > > >  input gain -3
> > > >  cptone C1
> > > >  timing hookflash-out 50
> > > > !
> > > > voice-port 0/1/1
> > > > !
> > > > voice-port 0/1/2
> > > > !
> > > > voice-port 0/1/3
> > > > !
> > > > ccm-manager fallback-mgcp
> > > > ccm-manager redundant-host 172.28.2.5
> > > > ccm-manager mgcp
> > > > ccm-manager music-on-hold
> > > > ccm-manager config server 172.29.128.4
> > > > ccm-manager config
> > > > ccm-manager download-tones
> > > > !
> > > > mgcp
> > > > mgcp call-agent 172.29.128.5 2427 service-type mgcp version 0.1
> > > > mgcp dtmf-relay voip codec all mode out-of-band
> > > > mgcp rtp unreachable timeout 1000 action notify
> > > > mgcp modem passthrough voip mode nse
> > > > mgcp package-capability rtp-package
> > > > no mgcp package-capability res-package
> > > > mgcp package-capability sst-package
> > > > no mgcp package-capability fxr-package
> > > > mgcp package-capability pre-package
> > > > no mgcp timer receive-rtcp
> > > > mgcp sdp simple
> > > > mgcp fax rate 9600
> > > > mgcp rtp payload-type g726r16 static
> > > > !
> > > > mgcp profile default
> > > > !
> > > > !
> > > > !
> > > > dial-peer voice 100 pots
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > > !
> > > > dial-peer voice 101 pots
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > > !
> > > > dial-peer voice 103 pots
> > > >  preference 2
> > > >  destination-pattern 9[2-9]......
> > > >  progress_ind alert enable 8
> > > >  progress_ind progress enable 8
> > > >  progress_ind connect enable 8
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > >  forward-digits 7
> > > > !
> > > > dial-peer voice 104 pots
> > > >  preference 1
> > > >  destination-pattern 91[2-9]..[2-9]......
> > > >  progress_ind alert enable 8
> > > >  progress_ind progress enable 8
> > > >  progress_ind connect enable 8
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > >  forward-digits 11
> > > > !
> > > > dial-peer voice 107 pots
> > > >  preference 2
> > > >  destination-pattern 9[369]11
> > > >  progress_ind alert enable 8
> > > >  progress_ind progress enable 8
> > > >  progress_ind connect enable 8
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  forward-digits 3
> > > > !
> > > > dial-peer voice 110 pots
> > > >  preference 1
> > > >  destination-pattern 9011T
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > >  prefix 011
> > > > !
> > > > dial-peer voice 111 pots
> > > >  preference 2
> > > >  destination-pattern 9011T
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > >  prefix 011
> > > > !
> > > > dial-peer voice 112 pots
> > > >  preference 2
> > > >  destination-pattern 9972.......
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > >  forward-digits 10
> > > > !
> > > > dial-peer voice 113 pots
> > > >  preference 2
> > > >  destination-pattern 9214.......
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > >  forward-digits 10
> > > > !
> > > > dial-peer voice 114 pots
> > > >  preference 2
> > > >  destination-pattern 9469.......
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > >  prefix 011
> > > > !
> > > > dial-peer voice 115 pots
> > > >  preference 2
> > > >  destination-pattern 9817.......
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > >  forward-digits 10
> > > > !
> > > > dial-peer voice 999010 pots
> > > >  service mgcpapp
> > > >  port 0/1/0
> > > > !
> > > > dial-peer voice 116 pots
> > > >  preference 2
> > > >  destination-pattern 911
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/1/0
> > > >  forward-digits 3
> > > > !
> > > > dial-peer voice 117 pots
> > > >  preference 2
> > > >  destination-pattern 9911
> > > >  incoming called-number .
> > > >  direct-inward-dial
> > > >  port 0/1/0
> > > >  forward-digits 3
> > > > !
> > > > dial-peer voice 102 pots
> > > >  preference 1
> > > >  destination-pattern 9[2-9]......
> > > >  progress_ind alert enable 8
> > > >  progress_ind progress enable 8
> > > >  progress_ind connect enable 8
> > > >  direct-inward-dial
> > > >  port 0/0/0:23
> > > >  forward-digits 7
> > > > !
> > > > dial-peer voice 993310 pots
> > > >  service mgcpapp
> > > >  port 0/0/0:23
> > > > !
> > > > dial-peer voice 993311 pots
> > > >  service mgcpapp
> > > >  port 0/1/0
> > > > !
> > > > !
> > > > !
> > > > call-manager-fallback
> > > >  max-conferences 8 gain -6
> > > >  timeouts interdigit 5
> > > >  ip source-address 172.27.128.2 port 2000
> > > >  max-ephones 42
> > > >  max-dn 144
> > > >  default-destination 6712
> > > >  voicemail 8000
> > > > !
> > > > banner login ^CCC
> > > > -----------------------------------------------------------------------
> > > > This system and its data contents are the property of Pinky the Wonder Sperm
> > > >  and/or its subsidiaries,and contains confidential and
> > > > proprietary information.  Any unauthorized use of this system or data
> > > > is strictly prohibited and may subject you to disciplinary action, up
> > > > to and including termination, civil penalties, and/or criminal
> > > > prosecution and penalties.  Users have no expectation of privacy
> > > > relating to the use of the system.  By using the system, you expressly
> > > > consent to the monitoring and review of your use of the system and all
> > > > data viewed, entered, created, received, stored, or transmitted through
> > > > the system.
> > > > -----------------------------------------------------------------------
> > > > ^C
> > > > !
> > > > line con 0
> > > >  exec-timeout 0 0
> > > >  password
> > > >  logging synchronous
> > > >  login
> > > > line aux 0
> > > >
> > > >
> > > > On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > > > not sure if this is what's happening in your case, but i had a default
> > > > route
> > > > > in my original setup that caused a delay. i had to put some ACLs on it to
> > > > > make sure it didn't try to connect to the ccm server over a different
> > > > route
> > > > > (which was blocked somewhere else).
> > > > >
> > > > > just a thought.
> > > > >
> > > > > maybe you can post your config.
> > > > >
> > > > > ----- Original Message -----
> > > > > From: "Jonathan Charles" <jonvoip at gmail.com>
> > > > > To: "ciscovoip" <cisco-voip at puck.nether.net>
> > > > > Sent: Friday, May 25, 2007 8:34 PM
> > > > > Subject: [cisco-voip] MGCP Fallback time...
> > > > >
> > > > >
> > > > > >I have a new SRST router (MGCP back to CCM)...
> > > > > >
> > > > > > When we yank the WAN, it takes about 13 minutes for the router to
> > > > > > realize that CCM is down and kill the PRI... and revert to H.323 mode.
> > > > > >
> > > > > > If you look at the attached log file, you can see that the phones are
> > > > > > registering to the router, and a full 13 minutes later the PRI goes
> > > > > > down...
> > > > > >
> > > > > > Is there a way to change this to make it quicker?
> > > > > >
> > > > > > IOS is 12.4.3g
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > > Jonathan
> > > > > >
> > > > >
> > > > >
> > > > >
> > > > --------------------------------------------------------------------------------
> > > > >
> > > > >
> > > > > > _______________________________________________
> > > > > > cisco-voip mailing list
> > > > > > cisco-voip at puck.nether.net
> > > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > >
> > > > >
> > > > >
> > > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
>

From jonvoip at gmail.com  Tue May 29 11:23:05 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 29 May 2007 10:23:05 -0500
Subject: [cisco-voip] cisco-voip.org Wiki
In-Reply-To: <6A0744639B86DD459433108DBF101D26016AED45@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26016AED45@MAIL01.sentinel.com>
Message-ID: <5d093f9a0705290823x303a521brf639e1c5687ee201@mail.gmail.com>

These are for 3.x and 4.x?

Does anyone have any useful SQL queries for CCM5?



Jonathan

On 5/29/07, Carter, Bill <bcarter at sentinel.com> wrote:
>
>
> Under Cisco Unified Communications Manager I added a section for SQL
> queries.
>
> http://cisco-voip.org/viki/index.php/SQL_Queries
>
>
> Bill Carter
> Sr. Business Communications Analyst
> Sentinel Technologies
> (O) 217.391.5015
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From jonvoip at gmail.com  Tue May 29 11:21:49 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 29 May 2007 10:21:49 -0500
Subject: [cisco-voip] MGCP Fallback time...
In-Reply-To: <f1b65f880705290809q7bf63e6s4b2eda78d0b4bb73@mail.gmail.com>
References: <5d093f9a0705251734n71238a7ch7cbcb676e9de9c7c@mail.gmail.com>
	<001101c79f2f$646e0680$c901a8c0@pdp11>
	<5d093f9a0705251757s13fc34abr3c48fa32c265472f@mail.gmail.com>
	<000c01c79f3b$089a8660$47196883@cfs.uoguelph.ca>
	<5d093f9a0705290730i7b7686cfi5ffb526156884390@mail.gmail.com>
	<f1b65f880705290809q7bf63e6s4b2eda78d0b4bb73@mail.gmail.com>
Message-ID: <5d093f9a0705290821p3582f6a3uba555656c9785f95@mail.gmail.com>

I am going to for another site, if it works there, I am going to
upgrade the other.



Jonathan

On 5/29/07, Justin Steinberg <jsteinberg at gmail.com> wrote:
> switchback timer only affects switchover/switchback between CCM nodes.
>  It doesn't alter timing between CCM and SRST.
>
> Have you tried a different IOS rev?  13 minutes seems awfully long.
>
> Justin
>
> On 5/29/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > I am debating on setting the switchback time to immediate... but I
> > don't think that will have any effect.
> >
> >
> > Jonathan
> >
> > On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > >
> > >
> > > well, your config is darn close to mine. here's mine to see the difference.
> > > you might want to try a few things I have? timeout under the
> > > call-manager-fallback?
> > >
> > > vgw-kemptville#show running-config
> > > Building configuration...
> > >
> > > Current configuration : 9987 bytes
> > > !
> > > version 12.4
> > > service timestamps debug datetime localtime
> > > service timestamps log datetime localtime
> > > service password-encryption
> > > !
> > > hostname vgw-kemptville
> > > !
> > > boot-start-marker
> > > boot system flash:c3845-spservicesk9-mz.124-11.T2.bin
> > > boot-end-marker
> > > !
> > > card type t1 0 0
> > > card type t1 0 1
> > > logging buffered 4096
> > > logging console informational
> > > enable secret <snip>
> > > !
> > > aaa new-model
> > > !
> > > !
> > > aaa authentication login default local group tacacs+ enable
> > > aaa authentication login virt-users group tacacs+ enable
> > > aaa authentication login console group tacacs+ enable
> > > !
> > > !
> > > aaa session-id common
> > > clock timezone EST -5
> > > clock summer-time EDT recurring
> > > network-clock-participate wic 0
> > > network-clock-participate wic 1
> > > no ip dhcp use vrf connected
> > > ip dhcp excluded-address 10.104.126.99
> > > ip dhcp excluded-address 10.104.126.100
> > > !
> > > ip dhcp pool dhcp_vlan826
> > >    network 10.104.126.0 255.255.255.128
> > >    option 150 ip 10.104.21.121
> > >    default-router 10.104.126.1
> > > !
> > > !
> > > ip cef
> > > !
> > > !
> > > multilink bundle-name authenticated
> > > !
> > > isdn switch-type primary-dms100
> > > voice-card 0
> > >  no dspfarm
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > voice translation-rule 613
> > >  rule 1 /\(.+\)/ /9\1/
> > > !
> > > voice translation-rule 900
> > >  rule 1 /^91809......./ /918090000000/
> > >  rule 2 /^91829......./ /918290000000/
> > >  rule 3 /^91900......./ /919000000000/
> > >  rule 4 /^91...976..../ /919760000000/
> > >  rule 5 /^91976..../ /919760000000/
> > >  rule 6 /^9976..../ /919760000000/
> > > !
> > > voice translation-rule 1112
> > >  rule 1 reject /914165551212/
> > > !
> > > !
> > > voice translation-profile 1112
> > > !
> > > voice translation-profile 999
> > >  translate called 1112
> > > !
> > > voice translation-profile kc_did_profile
> > >  translate called 613
> > > !
> > > voice translation-profile uogdev_block_highrisk_900_vtp
> > >  translate called 900
> > > !
> > > !
> > > !
> > > crypto pki trustpoint TP-self-signed-3156109335
> > >  enrollment selfsigned
> > >  subject-name cn=IOS-Self-Signed-Certificate-3156109335
> > >  revocation-check none
> > >  rsakeypair TP-self-signed-3156109335
> > > !
> > > !
> > > crypto pki certificate chain TP-self-signed-3156109335
> > >  certificate self-signed 01
> > > <snip>
> > >   quit
> > > !
> > > !
> > > vtp domain 3845
> > > vtp mode transparent
> > > !
> > > !
> > > controller T1 0/0/0
> > >  framing esf
> > >  linecode b8zs
> > >  pri-group timeslots 1-24 service mgcp
> > > !
> > > controller T1 0/1/0
> > >  framing esf
> > >  linecode b8zs
> > >  pri-group timeslots 1-24 service mgcp
> > > !
> > > vlan 129
> > > !
> > > vlan 725
> > >  name voice_10.104.125.0/25
> > > !
> > > vlan 826
> > >  name voice_10.104.126.0/25
> > > !
> > > translation-rule 1112
> > > !
> > > !
> > > !
> > > !
> > > !
> > > interface GigabitEthernet0/0
> > >  description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
> > >  no ip address
> > >  duplex auto
> > >  speed auto
> > >  media-type rj45
> > >  no keepalive
> > > !
> > > interface GigabitEthernet0/0.129
> > >  encapsulation dot1Q 129 native
> > >  ip address 192.168.129.66 255.255.255.0
> > >  ip access-group vlan129_in in
> > >  ip access-group vlan129_out out
> > > !
> > > interface GigabitEthernet0/0.725
> > >  encapsulation dot1Q 725
> > >  ip address 10.104.125.99 255.255.255.128
> > > !
> > > interface GigabitEthernet0/0.826
> > >  encapsulation dot1Q 826
> > >  ip address 10.104.126.1 255.255.255.128
> > > !
> > > interface Service-Engine0/0
> > >  ip unnumbered GigabitEthernet0/0.826
> > >  service-module ip address 10.104.126.100 255.255.255.128
> > >  service-module ip default-gateway 10.104.126.1
> > > !
> > > interface GigabitEthernet0/1
> > >  no ip address
> > >  shutdown
> > >  duplex auto
> > >  speed auto
> > >  media-type rj45
> > >  no keepalive
> > > !
> > > interface Serial0/0/0:23
> > >  no ip address
> > >  encapsulation hdlc
> > >  isdn switch-type primary-dms100
> > >  isdn incoming-voice voice
> > >  isdn bind-l3 ccm-manager
> > >  no cdp enable
> > > !
> > > interface Serial0/1/0:23
> > >  no ip address
> > >  encapsulation hdlc
> > >  isdn switch-type primary-dms100
> > >  isdn incoming-voice voice
> > >  isdn bind-l3 ccm-manager
> > >  no cdp enable
> > > !
> > > ip default-gateway 192.168.129.1
> > > ip route 0.0.0.0 0.0.0.0 192.168.129.1
> > > ip route 10.0.0.0 255.0.0.0 10.104.125.1
> > > ip route 10.104.126.100 255.255.255.255 Service-Engine0/0
> > > !
> > > !
> > > ip http server
> > > ip http authentication local
> > > ip http secure-server
> > > ip http timeout-policy idle 60 life 86400 requests 10000
> > > !
> > > ip access-list extended vlan129_in
> > >  deny   ip any 10.0.0.0 0.255.255.255
> > >  deny   ip 10.0.0.0 0.255.255.255 any
> > >  permit ip any any
> > > ip access-list extended vlan129_out
> > >  deny   ip 10.0.0.0 0.255.255.255 any
> > >  deny   ip any 10.0.0.0 0.255.255.255
> > >  permit ip any any
> > > !
> > > logging 192.168.20.5
> > > !
> > > !
> > > tacacs-server host 192.168.20.5
> > > tacacs-server directed-request
> > > tacacs-server key <snip>
> > > !
> > > control-plane
> > > !
> > > !
> > > !
> > > voice-port 0/0/0:23
> > > !
> > > voice-port 0/1/0:23
> > > !
> > > ccm-manager fallback-mgcp
> > > ccm-manager redundant-host 10.104.21.109
> > > ccm-manager mgcp
> > > ccm-manager music-on-hold
> > > ccm-manager config server 10.104.21.109
> > > ccm-manager config
> > > !
> > > mgcp
> > > mgcp call-agent 10.104.21.121 2427 service-type mgcp version 0.1
> > > mgcp dtmf-relay voip codec all mode out-of-band
> > > mgcp rtp unreachable timeout 1000 action notify
> > > mgcp modem passthrough voip mode nse
> > > mgcp package-capability rtp-package
> > > no mgcp package-capability res-package
> > > mgcp package-capability sst-package
> > > no mgcp package-capability fxr-package
> > > mgcp package-capability pre-package
> > > no mgcp timer receive-rtcp
> > > mgcp sdp simple
> > > mgcp rtp payload-type g726r16 static
> > > !
> > > mgcp profile default
> > > !
> > > !
> > > dial-peer cor custom
> > >  name uog_oncampus_pt
> > >  name uog_local_pt
> > >  name uog_freeservices_pt
> > >  name uog_tollfree_pt
> > >  name uog_payservices_pt
> > >  name uog_longdist_pt
> > >  name uog_block_highrisk_pt
> > > !
> > > !
> > > dial-peer cor list uogres-local-css
> > >  member uog_oncampus_pt
> > >  member uog_local_pt
> > >  member uog_tollfree_pt
> > > !
> > > dial-peer cor list uogbus-oncampus-css
> > >  member uog_oncampus_pt
> > > !
> > > dial-peer cor list uogbus-local-css
> > >  member uog_oncampus_pt
> > >  member uog_local_pt
> > >  member uog_freeservices_pt
> > >  member uog_tollfree_pt
> > > !
> > > dial-peer cor list uogbus-longdist-css
> > >  member uog_oncampus_pt
> > >  member uog_local_pt
> > >  member uog_freeservices_pt
> > >  member uog_tollfree_pt
> > >  member uog_payservices_pt
> > >  member uog_longdist_pt
> > >  member uog_block_highrisk_pt
> > > !
> > > dial-peer cor list uogdev-local-css
> > >  member uog_local_pt
> > > !
> > > dial-peer cor list uogdev-freeservices-css
> > >  member uog_freeservices_pt
> > > !
> > > dial-peer cor list uogdev-tollfree-css
> > >  member uog_tollfree_pt
> > > !
> > > dial-peer cor list uogdev-payservices-css
> > >  member uog_payservices_pt
> > > !
> > > dial-peer cor list uogdev-longdist-css
> > >  member uog_longdist_pt
> > > !
> > > dial-peer cor list uogdev-block-highrisk-css
> > >  member uog_block_highrisk_pt
> > > !
> > > !
> > > dial-peer voice 929 pots
> > >  destination-pattern 9[2-9].........
> > >  clid network-number 5198241212
> > >  port 0/1/0:23
> > >  forward-digits 10
> > > !
> > > dial-peer voice 61901 voip
> > >  destination-pattern 61901
> > >  session protocol sipv2
> > >  session target ipv4:10.104.126.100
> > >  dtmf-relay sip-notify
> > >  codec g711ulaw
> > >  no vad
> > > !
> > > dial-peer voice 98240022 pots
> > >  destination-pattern 95198240022
> > >  port 0/0/0:23
> > >  forward-digits 10
> > > !
> > > dial-peer voice 1 pots
> > >  translation-profile incoming kc_did_profile
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > > !
> > > dial-peer voice 2 pots
> > >  translation-profile incoming kc_did_profile
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/1/0:23
> > > !
> > > dial-peer voice 9767 pots
> > >  destination-pattern 95197671060
> > >  port 0/1/0:23
> > >  forward-digits 10
> > > !
> > > dial-peer voice 9129902 pots
> > >  translation-profile outgoing uogdev_block_highrisk_900_vtp
> > >  destination-pattern 91[2-9].........
> > >  forward-digits 11
> > > !
> > > dial-peer voice 99901 pots
> > >  translation-profile incoming 999
> > >  translation-profile outgoing 999
> > >  destination-pattern 914165551212
> > >  port 0/1/0:23
> > >  forward-digits 11
> > > !
> > > !
> > > num-exp 96132589... 61...
> > > num-exp 61000 61901
> > > num-exp 96132581515 61901
> > > num-exp 96132581212 61004
> > > num-exp 96135551414 61901
> > > num-exp 95198400105 61009
> > > !
> > > call-manager-fallback
> > >  secondary-dialtone 9
> > >  max-conferences 4 gain -6
> > >  transfer-system local-consult
> > >  limit-dn 7910 1
> > >  limit-dn 7935 1
> > >  limit-dn 7940 2
> > >  limit-dn 7960 6
> > >  limit-dn 7970 8
> > >  timeouts interdigit 15
> > >  ip source-address 10.104.126.1 port 2000 strict-match
> > >  max-ephones 336
> > >  max-dn 500 dual-line preference 5
> > >  system message primary UofG Link Down. Ltd Service
> > >  keepalive 60
> > >  time-zone 12
> > >  time-format 24
> > >  cor incoming uogres-local-css default
> > >  cor incoming uogbus-longdist-css 1 61001 - 61999
> > >  cor incoming uogbus-oncampus-css 2 61101
> > > !
> > > banner login ^CAccess Restricted to CCS Staff Only. All Access Att
> > >
> > > vgw-kemptville#
> > >
> > > --------------------------------------------------------------------------------
> > > Lelio Fulgenzi, B.A.
> > > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> > > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> > > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > > "Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR
> > >
> > >
> > > ----- Original Message -----
> > > From: Jonathan Charles
> > >
> > > To: Lelio Fulgenzi
> > > Cc: ciscovoip
> > > Sent: Friday, May 25, 2007 8:57 PM
> > > Subject: Re: [cisco-voip] MGCP Fallback time...
> > >
> > > Here it is...
> > >
> > >
> > > Current configuration : 7664 bytes
> > > !
> > > version 12.4
> > > service timestamps debug datetime msec
> > > service timestamps log datetime msec
> > > no service password-encryption
> > > !
> > > hostname PINKY1
> > > !
> > > boot-start-marker
> > > boot-end-marker
> > > !
> > > logging buffered 1000000 warnings
> > > enable secret 5 $1$OGFk$TCpZe0TS9B4anq.Q6MJIQ0
> > > !
> > > no aaa new-model
> > > !
> > > resource policy
> > > !
> > > network-clock-participate wic 0
> > > ip subnet-zero
> > > !
> > > !
> > > ip cef
> > > !
> > > !
> > > no ip domain lookup
> > > ip domain name amli.com
> > > isdn switch-type primary-ni
> > > !
> > > voice-card 0
> > >  no dspfarm
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > application
> > >  global
> > >   service alternate Default
> > >  !
> > > !
> > > !
> > > crypto pki trustpoint TP-self-signed-2413116866
> > >  enrollment selfsigned
> > >  subject-name cn=IOS-Self-Signed-Certificate-2413116866
> > >  revocation-check none
> > >  rsakeypair TP-self-signed-2413116866
> > > !
> > > !
> > > crypto pki certificate chain TP-self-signed-2413116866
> > >  certificate self-signed 01
> > >   3082024C 308201B5 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
> > >   31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
> > >   69666963 6174652D 32343133 31313638 3636301E 170D3037 30353233 31363136
> > >   35385A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
> > >   4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 34313331
> > >   31363836 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
> > >   8100C7DB 31404006 EEE793F7 398F1A35 07509CDB D444F92D 79FF5F4A 73A01D89
> > >   B77EF538 DF19058C CCE57B9C ABF0597B DD60FAE8 3CD96D2E DD46A581 15B15F38
> > >   DC3368D4 074A66F7 9E484DED E36394F6 05EA9FDA F81632FC 31D90520 256D497E
> > >   CE8F91E4 042BD4E6 824C0DE6 F3107769 F0FFA0E0 65321D02 025D7F33 65C6DDE4
> > >   027F0203 010001A3 74307230 0F060355 1D130101 FF040530 030101FF 301F0603
> > >   551D1104 18301682 14414D4C 4944414C 32383131 2E616D6C 692E636F 6D301F06
> > >   03551D23 04183016 80144221 418BAB16 E9B5F101 F02AA6B0 0E6B618D 83BA301D
> > >   0603551D 0E041604 14422141 8BAB16E9 B5F101F0 2AA6B00E 6B618D83 BA300D06
> > >   092A8648 86F70D01 01040500 03818100 905A4A1F 2A2F0BD5 6F3E7634 C6A122D1
> > >   5EB8D0D3 11AC6A79 6C70001C 3E4D1923 6EDE6337 738A9FE1 3B9574C7 CFE4050B
> > >   67CD8790 BC4F8EA1 5C209CB8 8CC175E7 8E27CFFE E3492ADD 874CEBF1 080244B0
> > >   90D92709 B82A7B48 F3EF1DA5 6A32664E 00794B9A 5DBE2060 FDB89303 E0692D39
> > >   4884D825 FB237370 82321596 A079BB75
> > >   quit
> > > username cisco privilege 15 secret 5 $1$sBAo$/JviEK.cbQuLpFFEKRRsb/
> > > !
> > > !
> > > controller T1 0/0/0
> > >  framing esf
> > >  linecode b8zs
> > >  pri-group timeslots 1-24 service mgcp
> > > !
> > > !
> > > !
> > > interface FastEthernet0/0
> > >  description IP VOICE SUBNET
> > >  ip address 172.27.128.2 255.255.255.0
> > >  duplex auto
> > >  speed auto
> > > !
> > > interface FastEthernet0/1
> > >  no ip address
> > >  shutdown
> > >  duplex auto
> > >  speed auto
> > > !
> > > interface Serial0/0/0:23
> > >  no ip address
> > >  isdn switch-type primary-ni
> > >  isdn incoming-voice voice
> > >  isdn bind-l3 ccm-manager
> > >  no cdp enable
> > > !
> > > ip classless
> > > ip route 0.0.0.0 0.0.0.0 172.27.128.1
> > > !
> > > !
> > > ip http server
> > > ip http access-class 23
> > > ip http authentication local
> > > ip http secure-server
> > > ip http timeout-policy idle 60 life 86400 requests 10000
> > > !
> > > access-list 23 permit 10.10.10.0 0.0.0.7
> > > !
> > > !
> > > !
> > > control-plane
> > > !
> > > !
> > > !
> > > voice-port 0/0/0:23
> > >  input gain -3
> > >  output attenuation 3
> > >  cptone C1
> > > !
> > > voice-port 0/1/0
> > >  input gain -3
> > >  cptone C1
> > >  timing hookflash-out 50
> > > !
> > > voice-port 0/1/1
> > > !
> > > voice-port 0/1/2
> > > !
> > > voice-port 0/1/3
> > > !
> > > ccm-manager fallback-mgcp
> > > ccm-manager redundant-host 172.28.2.5
> > > ccm-manager mgcp
> > > ccm-manager music-on-hold
> > > ccm-manager config server 172.29.128.4
> > > ccm-manager config
> > > ccm-manager download-tones
> > > !
> > > mgcp
> > > mgcp call-agent 172.29.128.5 2427 service-type mgcp version 0.1
> > > mgcp dtmf-relay voip codec all mode out-of-band
> > > mgcp rtp unreachable timeout 1000 action notify
> > > mgcp modem passthrough voip mode nse
> > > mgcp package-capability rtp-package
> > > no mgcp package-capability res-package
> > > mgcp package-capability sst-package
> > > no mgcp package-capability fxr-package
> > > mgcp package-capability pre-package
> > > no mgcp timer receive-rtcp
> > > mgcp sdp simple
> > > mgcp fax rate 9600
> > > mgcp rtp payload-type g726r16 static
> > > !
> > > mgcp profile default
> > > !
> > > !
> > > !
> > > dial-peer voice 100 pots
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > > !
> > > dial-peer voice 101 pots
> > >  incoming called-number .
> > >  direct-inward-dial
> > > !
> > > dial-peer voice 103 pots
> > >  preference 2
> > >  destination-pattern 9[2-9]......
> > >  progress_ind alert enable 8
> > >  progress_ind progress enable 8
> > >  progress_ind connect enable 8
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > >  forward-digits 7
> > > !
> > > dial-peer voice 104 pots
> > >  preference 1
> > >  destination-pattern 91[2-9]..[2-9]......
> > >  progress_ind alert enable 8
> > >  progress_ind progress enable 8
> > >  progress_ind connect enable 8
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > >  forward-digits 11
> > > !
> > > dial-peer voice 107 pots
> > >  preference 2
> > >  destination-pattern 9[369]11
> > >  progress_ind alert enable 8
> > >  progress_ind progress enable 8
> > >  progress_ind connect enable 8
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  forward-digits 3
> > > !
> > > dial-peer voice 110 pots
> > >  preference 1
> > >  destination-pattern 9011T
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > >  prefix 011
> > > !
> > > dial-peer voice 111 pots
> > >  preference 2
> > >  destination-pattern 9011T
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > >  prefix 011
> > > !
> > > dial-peer voice 112 pots
> > >  preference 2
> > >  destination-pattern 9972.......
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > >  forward-digits 10
> > > !
> > > dial-peer voice 113 pots
> > >  preference 2
> > >  destination-pattern 9214.......
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > >  forward-digits 10
> > > !
> > > dial-peer voice 114 pots
> > >  preference 2
> > >  destination-pattern 9469.......
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > >  prefix 011
> > > !
> > > dial-peer voice 115 pots
> > >  preference 2
> > >  destination-pattern 9817.......
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > >  forward-digits 10
> > > !
> > > dial-peer voice 999010 pots
> > >  service mgcpapp
> > >  port 0/1/0
> > > !
> > > dial-peer voice 116 pots
> > >  preference 2
> > >  destination-pattern 911
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/1/0
> > >  forward-digits 3
> > > !
> > > dial-peer voice 117 pots
> > >  preference 2
> > >  destination-pattern 9911
> > >  incoming called-number .
> > >  direct-inward-dial
> > >  port 0/1/0
> > >  forward-digits 3
> > > !
> > > dial-peer voice 102 pots
> > >  preference 1
> > >  destination-pattern 9[2-9]......
> > >  progress_ind alert enable 8
> > >  progress_ind progress enable 8
> > >  progress_ind connect enable 8
> > >  direct-inward-dial
> > >  port 0/0/0:23
> > >  forward-digits 7
> > > !
> > > dial-peer voice 993310 pots
> > >  service mgcpapp
> > >  port 0/0/0:23
> > > !
> > > dial-peer voice 993311 pots
> > >  service mgcpapp
> > >  port 0/1/0
> > > !
> > > !
> > > !
> > > call-manager-fallback
> > >  max-conferences 8 gain -6
> > >  timeouts interdigit 5
> > >  ip source-address 172.27.128.2 port 2000
> > >  max-ephones 42
> > >  max-dn 144
> > >  default-destination 6712
> > >  voicemail 8000
> > > !
> > > banner login ^CCC
> > > -----------------------------------------------------------------------
> > > This system and its data contents are the property of Pinky the Wonder Sperm
> > >  and/or its subsidiaries,and contains confidential and
> > > proprietary information.  Any unauthorized use of this system or data
> > > is strictly prohibited and may subject you to disciplinary action, up
> > > to and including termination, civil penalties, and/or criminal
> > > prosecution and penalties.  Users have no expectation of privacy
> > > relating to the use of the system.  By using the system, you expressly
> > > consent to the monitoring and review of your use of the system and all
> > > data viewed, entered, created, received, stored, or transmitted through
> > > the system.
> > > -----------------------------------------------------------------------
> > > ^C
> > > !
> > > line con 0
> > >  exec-timeout 0 0
> > >  password
> > >  logging synchronous
> > >  login
> > > line aux 0
> > >
> > >
> > > On 5/25/07, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > > not sure if this is what's happening in your case, but i had a default
> > > route
> > > > in my original setup that caused a delay. i had to put some ACLs on it to
> > > > make sure it didn't try to connect to the ccm server over a different
> > > route
> > > > (which was blocked somewhere else).
> > > >
> > > > just a thought.
> > > >
> > > > maybe you can post your config.
> > > >
> > > > ----- Original Message -----
> > > > From: "Jonathan Charles" <jonvoip at gmail.com>
> > > > To: "ciscovoip" <cisco-voip at puck.nether.net>
> > > > Sent: Friday, May 25, 2007 8:34 PM
> > > > Subject: [cisco-voip] MGCP Fallback time...
> > > >
> > > >
> > > > >I have a new SRST router (MGCP back to CCM)...
> > > > >
> > > > > When we yank the WAN, it takes about 13 minutes for the router to
> > > > > realize that CCM is down and kill the PRI... and revert to H.323 mode.
> > > > >
> > > > > If you look at the attached log file, you can see that the phones are
> > > > > registering to the router, and a full 13 minutes later the PRI goes
> > > > > down...
> > > > >
> > > > > Is there a way to change this to make it quicker?
> > > > >
> > > > > IOS is 12.4.3g
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > >
> > > >
> > > >
> > > --------------------------------------------------------------------------------
> > > >
> > > >
> > > > > _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > >
> > > >
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>

From Matt.Slaga at us.didata.com  Tue May 29 11:51:10 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 29 May 2007 11:51:10 -0400
Subject: [cisco-voip] cisco-voip.org Wiki
In-Reply-To: <5d093f9a0705290823x303a521brf639e1c5687ee201@mail.gmail.com>
References: <6A0744639B86DD459433108DBF101D26016AED45@MAIL01.sentinel.com>
	<5d093f9a0705290823x303a521brf639e1c5687ee201@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B22ABF@USNAEXCH.na.didata.local>

No SQL access in 5.x.  It's a locked out version of IBM Informix.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Tuesday, May 29, 2007 11:23 AM
To: Carter, Bill
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] cisco-voip.org Wiki

These are for 3.x and 4.x?

Does anyone have any useful SQL queries for CCM5?



Jonathan

On 5/29/07, Carter, Bill <bcarter at sentinel.com> wrote:
>
>
> Under Cisco Unified Communications Manager I added a section for SQL
> queries.
>
> http://cisco-voip.org/viki/index.php/SQL_Queries
>
>
> Bill Carter
> Sr. Business Communications Analyst
> Sentinel Technologies
> (O) 217.391.5015
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.

From rratliff at cisco.com  Tue May 29 12:16:50 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 29 May 2007 12:16:50 -0400
Subject: [cisco-voip] Upgrade to 5.1(2)
In-Reply-To: <5e360a2300007727@nts.at>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451F11@CSEXCH01.charles-stanley.co.uk><5d093f9a0705260901l625f347dnea83faf3187d3122@mail.gmail.com><D2BBAE709B4D874DAC556C7EB9D938FC31BC27@exchange.TGASolutions.local>
	<5d093f9a0705280657t7a42b743g106429a7e1860c35@mail.gmail.com>
	<5e360a2300007727@nts.at>
Message-ID: <F9BC9AD0-813A-41AE-B29A-7772318012B7@cisco.com>

5.1(2.9901.2) means it will be in 5.1(3).

If you have a TAC SR then ask them to get it included in a 5.1(2)ES.

-Ryan

On May 29, 2007, at 2:33 AM, Bernhard Albler wrote:

Hi all,

tried to upgrade a customer from 5.1(1)ES08 to 5.1(2) over the weekend.
I finally had to revert back to the old version, because all pickup
group fields in lines and huntpilots where set to "null".

Most likely I am hitting CSCsi91071, though it seems odd that such a bug
would slip into the release.
Can anyone from cisco enlighten on CM5 version numbers?
Is 5.1(2.9901.2) a version  that comes before or after 5.1(2.1000.11)
(which is the released cm 5.1(2) I believe)?

Regards
bernhard

------------------------------------------
DI (FH) Bernhard Albler, NTS AG
Tel.: +43 (1) 867 67 76 -0, email: balbler at nts.at
________________________________________
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jonvoip at gmail.com  Tue May 29 12:18:48 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 29 May 2007 11:18:48 -0500
Subject: [cisco-voip] cisco-voip.org Wiki
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005B22ABF@USNAEXCH.na.didata.local>
References: <6A0744639B86DD459433108DBF101D26016AED45@MAIL01.sentinel.com>
	<5d093f9a0705290823x303a521brf639e1c5687ee201@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F005B22ABF@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0705290918n2d42d661k412b42fd30866bdf@mail.gmail.com>

There is plenty of sql access.

The problem is the formatting of the commands...

go to a CCM5 console and type the following:

run sql ?



Jonathan

On 5/29/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> No SQL access in 5.x.  It's a locked out version of IBM Informix.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Tuesday, May 29, 2007 11:23 AM
> To: Carter, Bill
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] cisco-voip.org Wiki
>
> These are for 3.x and 4.x?
>
> Does anyone have any useful SQL queries for CCM5?
>
>
>
> Jonathan
>
> On 5/29/07, Carter, Bill <bcarter at sentinel.com> wrote:
> >
> >
> > Under Cisco Unified Communications Manager I added a section for SQL
> > queries.
> >
> > http://cisco-voip.org/viki/index.php/SQL_Queries
> >
> >
> > Bill Carter
> > Sr. Business Communications Analyst
> > Sentinel Technologies
> > (O) 217.391.5015
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>

From Matt.Slaga at us.didata.com  Tue May 29 12:29:54 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 29 May 2007 12:29:54 -0400
Subject: [cisco-voip] cisco-voip.org Wiki
In-Reply-To: <5d093f9a0705290918n2d42d661k412b42fd30866bdf@mail.gmail.com>
References: <6A0744639B86DD459433108DBF101D26016AED45@MAIL01.sentinel.com>
	<5d093f9a0705290823x303a521brf639e1c5687ee201@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F005B22ABF@USNAEXCH.na.didata.local>
	<5d093f9a0705290918n2d42d661k412b42fd30866bdf@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B22B64@USNAEXCH.na.didata.local>

I stand corrected!  Didn't know that existed...

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Tuesday, May 29, 2007 12:19 PM
To: Matt Slaga (US)
Cc: Carter, Bill; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] cisco-voip.org Wiki

There is plenty of sql access.

The problem is the formatting of the commands...

go to a CCM5 console and type the following:

run sql ?



Jonathan

On 5/29/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> No SQL access in 5.x.  It's a locked out version of IBM Informix.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Tuesday, May 29, 2007 11:23 AM
> To: Carter, Bill
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] cisco-voip.org Wiki
>
> These are for 3.x and 4.x?
>
> Does anyone have any useful SQL queries for CCM5?
>
>
>
> Jonathan
>
> On 5/29/07, Carter, Bill <bcarter at sentinel.com> wrote:
> >
> >
> > Under Cisco Unified Communications Manager I added a section for SQL
> > queries.
> >
> > http://cisco-voip.org/viki/index.php/SQL_Queries
> >
> >
> > Bill Carter
> > Sr. Business Communications Analyst
> > Sentinel Technologies
> > (O) 217.391.5015
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>

-----------------------------------------
Disclaimer:

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notify us immediately by replying to this message and deleting it
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From jonvoip at gmail.com  Tue May 29 13:13:32 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 29 May 2007 12:13:32 -0500
Subject: [cisco-voip] cisco-voip.org Wiki
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005B22B64@USNAEXCH.na.didata.local>
References: <6A0744639B86DD459433108DBF101D26016AED45@MAIL01.sentinel.com>
	<5d093f9a0705290823x303a521brf639e1c5687ee201@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F005B22ABF@USNAEXCH.na.didata.local>
	<5d093f9a0705290918n2d42d661k412b42fd30866bdf@mail.gmail.com>
	<C1FE15183DA37645BC0633BC604E44F005B22B64@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0705291013v187f1d6hb752b04f2c2e8cc@mail.gmail.com>

Yeah, originally, they weren't going to include it, but TAC mandated
that they be able to query SQL directly...

The question is, I have all of these old SQL things for checking
stuff, and now I do not.



Jonathan

On 5/29/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> I stand corrected!  Didn't know that existed...
>
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Tuesday, May 29, 2007 12:19 PM
> To: Matt Slaga (US)
> Cc: Carter, Bill; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] cisco-voip.org Wiki
>
> There is plenty of sql access.
>
> The problem is the formatting of the commands...
>
> go to a CCM5 console and type the following:
>
> run sql ?
>
>
>
> Jonathan
>
> On 5/29/07, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> > No SQL access in 5.x.  It's a locked out version of IBM Informix.
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > Charles
> > Sent: Tuesday, May 29, 2007 11:23 AM
> > To: Carter, Bill
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] cisco-voip.org Wiki
> >
> > These are for 3.x and 4.x?
> >
> > Does anyone have any useful SQL queries for CCM5?
> >
> >
> >
> > Jonathan
> >
> > On 5/29/07, Carter, Bill <bcarter at sentinel.com> wrote:
> > >
> > >
> > > Under Cisco Unified Communications Manager I added a section for SQL
> > > queries.
> > >
> > > http://cisco-voip.org/viki/index.php/SQL_Queries
> > >
> > >
> > > Bill Carter
> > > Sr. Business Communications Analyst
> > > Sentinel Technologies
> > > (O) 217.391.5015
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain
> > confidential and privileged information and is for use by the
> > designated addressee(s) named above only.  If you are not the
> > intended addressee, you are hereby notified that you have received
> > this communication in error and that any use or reproduction of
> > this email or its contents is strictly prohibited and may be
> > unlawful.  If you have received this communication in error, please
> > notify us immediately by replying to this message and deleting it
> > from your computer. Thank you.
> >
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>

From janderson at enventis.com  Tue May 29 12:49:19 2007
From: janderson at enventis.com (Jeff Anderson (Enventis))
Date: Tue, 29 May 2007 11:49:19 -0500
Subject: [cisco-voip] Unity nearing end of recording limit warning
Message-ID: <C0A504750F3BBE4FADE4ED6D672BA1790C57AD9C@mail.hickorytech.local>

Does anyone know if it is possible to configure a warning to inform a
caller who is leaving a message that they only have 15 seconds left to
wrap up their message? Right now when you hit the limit Unity simply
stops recording.
 
Thanks,
 
Jeff
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From scott.coner at us.didata.com  Tue May 29 13:41:53 2007
From: scott.coner at us.didata.com (Scott Coner (US))
Date: Tue, 29 May 2007 13:41:53 -0400
Subject: [cisco-voip] Cisco Unity, IMAP and Novell GroupWise
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B22CDF@USNAEXCH.na.didata.local>

Anyone successfully configure an IMAP integration with Unity 4.2 and
GroupWise client 6.5.4?

 

We are able to retrieve messages but if the user deletes a message using
the phone the client doesn't seem to get updated.  

 

If I configure the IMAP connection using Outlook or Outlook Express I
can't reproduce the problem.

 

 

 

 




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From MillerS at DicksteinShapiro.COM  Tue May 29 13:51:50 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Tue, 29 May 2007 13:51:50 -0400
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>

We have an odd problem.  In our script on the CRS server, we transfer
calls to an external number for our backup helpdesk vendor.  This works
fine when we insert their regular phone number which is 404-XXX-XXXX,
but when we enter the toll free number 800-XXX-XXXX that is forwarded to
the 404 number by the carrier, we get a busy signal.  Is there anything
that I can change on the IPCC or the Call Manager that might be
preventing this call from transferring?  We can call the 800 number
directly from our Call Manager phones without difficulty.  Please let me
know if you guys have any ideas.  Thank you!
 
Steve Miller

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

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From janderson at enventis.com  Tue May 29 13:59:01 2007
From: janderson at enventis.com (Jeff Anderson (Enventis))
Date: Tue, 29 May 2007 12:59:01 -0500
Subject: [cisco-voip] CME Call-forward question
Message-ID: <C0A504750F3BBE4FADE4ED6D672BA1790C57AE0A@mail.hickorytech.local>

The scenario:
 
CME 3.3
 
x1000 is CFA to x1001. I want the caller who called x1000 to get the
voicemail box of x1001 if x1001 doesn't answer. 
 
Is this possible? I know in call manager you would need to create a
unique VM profile to accomplish this.
 
Thanks,
 
Jeff
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From keli at carocomp.ro  Tue May 29 14:22:33 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Tue, 29 May 2007 21:22:33 +0300
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>
Message-ID: <465C6F69.7070307@carocomp.ro>

Calling Search Space differences, perhaps? Does the call reach the gateway?

Try to check it with dialed number analyzer, perhaps ...

Can you debug the call on the gateway? Does it get there, if so, in what 
form, etc.

regards,
   Zoltan

Miller, Steve wrote:
> We have an odd problem.  In our script on the CRS server, we transfer 
> calls to an external number for our backup helpdesk vendor.  This works 
> fine when we insert their regular phone number which is 404-XXX-XXXX, 
> but when we enter the toll free number 800-XXX-XXXX that is forwarded to 
> the 404 number by the carrier, we get a busy signal.  Is there anything 
> that I can change on the IPCC or the Call Manager that might be 
> preventing this call from transferring?  We can call the 800 number 
> directly from our Call Manager phones without difficulty.  Please let me 
> know if you guys have any ideas.  Thank you!
>  
> Steve Miller
> 
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 
> 
> To reply to our email administrator directly, send an email to postmaster at dicksteinshapiro.com
> 
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
> ==============================================================================
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From bob at smalltime.com  Tue May 29 14:28:59 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Tue, 29 May 2007 13:28:59 -0500
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
In-Reply-To: <465C6F69.7070307@carocomp.ro>
References: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>
	<465C6F69.7070307@carocomp.ro>
Message-ID: <465C70EB.9020805@smalltime.com>

> Miller, Steve wrote:
>> We have an odd problem.  In our script on the CRS server, we
>> transfer calls to an external number for our backup helpdesk
>> vendor.  This works fine when we insert their regular phone number
>> which is 404-XXX-XXXX, but when we enter the toll free number
>> 800-XXX-XXXX that is forwarded to the 404 number by the carrier, we
>> get a busy signal.

Check your Q931 traces; does it look like this?  (I'm also about to 
start troubleshooting this issue, so I don't have an answer).

If the site dials the 800 number directly, it works, but if they use a 
site-wide speed dial that's a CFA on a line that's attached to a 
"SpeedDial" CTI port, it doesn't work.  (*660 is the speed dial)

May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 213928
May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Called num 18005797372
May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8  callref =
0x06A3
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transfer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98382
                 Exclusive, Channel 2
         Calling Party Number i = 0x21A1, '213928'
                 Plan:ISDN, Type:National
         Called Party Number i = 0xA1, '18005797372'
                 Plan:ISDN, Type:National
         Redirecting Number i = '!', 0x008F, '??*660'
                 Plan:ISDN, Type:National

May 29 10:47:41.015: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8
callref = 0x86A3 Channel ID i = 0xA98382 Exclusive, Channel 2

May 29 10:47:41.023: ISDN Se0/1/0:23 Q931: RX <- PROGRESS pd = 8
callref = 0x86A3
         Cause i = 0x80FF - Interworking error; unspecified
         Progress Ind i = 0x8088 - In-band info or appropriate now available

May 29 10:47:44.935: ISDN Se0/1/0:23 Q931: TX -> DISCONNECT pd = 8
callref = 0x06A3 Cause i = 0x8090 - Normal call clearing

May 29 10:47:45.015: ISDN Se0/1/0:23 Q931: RX <- RELEASE pd = 8  callref
= 0x86A3

May 29 10:47:45.019: ISDN Se0/1/0:23 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x06A3

From bcarter at sentinel.com  Tue May 29 14:42:40 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Tue, 29 May 2007 13:42:40 -0500
Subject: [cisco-voip] QOS for MPLS
In-Reply-To: <5d093f9a0705271454u6e0c479cqee972a6fcd9c16a5@mail.gmail.com>
References: <262147.28980.qm@web37210.mail.mud.yahoo.com>
	<5d093f9a0705271454u6e0c479cqee972a6fcd9c16a5@mail.gmail.com>
Message-ID: <6A0744639B86DD459433108DBF101D26016AED73@MAIL01.sentinel.com>

With MPLS and sites with different bandwidths, you should set the QoS
based on the site bandwidth.  So it is ok to have 800k on the T1's and
2500k on the DS3's.  Locations would prevent HQ from flooding a
particular site.  

Because of the mesh possibilities of MPLS, I would set the HQ Location
bandwidth to unlimited and the remote at ~800kb. 

If this is Unified CM 5.X you could use RSVP.  CM locations are more
designed for a Hub and Spoke network.  RSVP accommodates a meshed
topology.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Sunday, May 27, 2007 4:54 PM
To: Erick Bergquist
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] QOS for MPLS

Which is the definition of MPLS, any-to-any communication. All circuits
are point to point into the MPLS cloud and then routing in the cloud to
each site.



Jonathan

On 5/27/07, Erick Bergquist <erickbe at yahoo.com> wrote:
> I've seen one or two setups where the MPLS was fully meshed (all sites
could talk to any other site), which makes QoS even trickier.
>
> ----- Original Message ----
> From: "keli at carocomp.ro" <keli at carocomp.ro>
> To: Jonathan Charles <jonvoip at gmail.com>
> Cc: cisco-voip at puck.nether.net
> Sent: Sunday, May 27, 2007 3:50:17 PM
> Subject: Re: [cisco-voip] QOS for MPLS
>
> That is correct (although you should take some overhead into account 
> as well). However, if you do not perform Call Admission Control 
> beforehand (you do not limit the maximum number of possible 
> simultaneous calls) you've done nothing, or worse.
>
> The classic example being when you have a line that supports 4 calls, 
> when the fifth call enters the line, all calls will suffer packet loss

> and delay, not only the fifth one.
>
> I'm not too versed in MPLS QoS, but the idea is always the same.
> Prioritize voice traffic, but don't let it kill your bandwidth, nor 
> itself. Thus the need for CAC, then something like CBWFQ/PQ. If 
> possible, voice traffic should be beneath shaping/policing limits, so 
> that neither would affect it.
>
> regards,
>    Zoltan
>
> Quoting Jonathan Charles <jonvoip at gmail.com>:
>
> > My thinking is that I need to set the QOS to be the same as the 
> > maximum number of calls to/from each site.
> >
> >
> > Jonathan
> >
> > On 5/27/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
> >> As long as it is about "calls", CAC is the sensitive way to go 
> >> about limiting the calls in the first place. Do QoS "As close to 
> >> the source, as possible", remember? :)
> >>
> >> If you do not have a CCM, but other set-up, you should still look 
> >> in this direction first - to limit the calls to a reasonable amount

> >> in the given directions, before they are hitting your MPLS router.
> >>
> >> regards,
> >>   Zoltan
> >>
> >> Quoting Paul Choi <asobihoudai at yahoo.com>:
> >>
> >>> He's talking about QoS for MPLS, not CAC.
> >>>
> >>> --- "Clouse, Chris" <chris.clouse at berbee.com> wrote:
> >>>
> >>>> Use CallManager locations to perform Call Admission Control.
> >>>>
> >>>> Christopher Clouse, CCNP CCDP CCVP MCP Network+
> >>>>
> >>>>
> >>>> -----Original Message-----
> >>>> From: cisco-voip-bounces at puck.nether.net
> >>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan

> >>>> Charles
> >>>> Sent: Saturday, May 26, 2007 11:35 AM
> >>>> To: ciscovoip
> >>>> Subject: [cisco-voip] QOS for MPLS
> >>>>
> >>>> I have always understood that QOS settings need to match on both 
> >>>> ends of a circuit... so, if you allocate 800k on one side, you 
> >>>> need to do the same on the other.
> >>>>
> >>>> I am now working on an MPLS circuit that has DS3s on some sides, 
> >>>> DS1s on others... do I still need to match QOS settings?
> >>>>
> >>>> Customer wants 800k on the T1s and 2500k on the DS3s... I don't 
> >>>> think it will matter as I am matching these bw limits on the 
> >>>> locations...
> >>>> but how do I throttle calls from a DS3-side location to a DS1 
> >>>> location?
> >>>>
> >>>> For example... main site is in Chicago, they have a DS3, one 
> >>>> remote site is in Wichita and has a T1... so, I put 800k for 
> >>>> Wichita and 2500k for Chicago... what is to stop Chicago from 
> >>>> flooding Wichita with 2.5MB of calls?
> >>>>
> >>>>
> >>>>
> >>>> Jonathan
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >> ___________________________________________________________________
> >> _________________Building a website is a piece of cake. Yahoo! 
> >> Small Business gives you all the tools to get
> >>> online.
> >>> http://smallbusiness.yahoo.com/webhosting
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip at puck.nether.net
> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>
> >>
> >>
> >>
> >> ----------------------------------------------------------------
> >> This message was sent using IMP, the Internet Messaging Program.
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
>
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
> ______________________________________________________________________
> ______________ We won't tell. Get more on shows you hate to love (and 
> love to hate): Yahoo! TV's Guilty Pleasures list.
> http://tv.yahoo.com/collections/265
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From aizhar at ccbootcamp.com  Tue May 29 14:46:50 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Tue, 29 May 2007 11:46:50 -0700
Subject: [cisco-voip] CME Call-forward question
	[bcc][faked-from][html-rem]
References: <C0A504750F3BBE4FADE4ED6D672BA1790C57AE0A@mail.hickorytech.local>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663560FFF@nlimail.nlidomain>

What voice mail system? unity? cue?
 
In unity you can configure the alternate extension to contain 1001 in it.
 
You can also translate the calling number using voice translation rules on the cme. 
 
-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take Cisco Learning credits!
http://www.ccbootcamp.com <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012


________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson (Enventis)
Sent: Tue 5/29/2007 10:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME Call-forward question [bcc][faked-from][html-rem]



The scenario:

CME 3.3

x1000 is CFA to x1001. I want the caller who called x1000 to get the
voicemail box of x1001 if x1001 doesn't answer.

Is this possible? I know in call manager you would need to create a
unique VM profile to accomplish this.

Thanks,

Jeff
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


-------------- next part --------------
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From janderson at enventis.com  Tue May 29 15:20:31 2007
From: janderson at enventis.com (Jeff Anderson (Enventis))
Date: Tue, 29 May 2007 14:20:31 -0500
Subject: [cisco-voip] CME Call-forward question
	[bcc][faked-from][html-rem]
In-Reply-To: <E76ACA8C13EF1847BC69538B522C3663560FFF@nlimail.nlidomain>
Message-ID: <C0A504750F3BBE4FADE4ED6D672BA1790C57AEE4@mail.hickorytech.local>

The voicemail system is CUE. Both of those numbers (1000 and 1001) have
a VM box in unity so I don't know how you would use an alternate
extension.
 
Thanks,
 
________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Tuesday, May 29, 2007 1:47 PM
To: Jeff Anderson (Enventis); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem]
 
What voice mail system? unity? cue?
 
In unity you can configure the alternate extension to contain 1001 in
it.
 
You can also translate the calling number using voice translation rules
on the cme. 
 
-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take
Cisco Learning credits!
http://www.ccbootcamp.com
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012
 
________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
(Enventis)
Sent: Tue 5/29/2007 10:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem]
The scenario:

CME 3.3

x1000 is CFA to x1001. I want the caller who called x1000 to get the
voicemail box of x1001 if x1001 doesn't answer.

Is this possible? I know in call manager you would need to create a
unique VM profile to accomplish this.

Thanks,

Jeff
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From aizhar at ccbootcamp.com  Tue May 29 15:26:02 2007
From: aizhar at ccbootcamp.com (Avner Izhar)
Date: Tue, 29 May 2007 12:26:02 -0700
Subject: [cisco-voip] CME Call-forward question
	[bcc][faked-from][html-rem] [mx][html-rem]
References: <C0A504750F3BBE4FADE4ED6D672BA1790C57AEE4@mail.hickorytech.local>
Message-ID: <E76ACA8C13EF1847BC69538B522C3663561002@nlimail.nlidomain>

Why would you need a voice mail box for x1000, it is CFA to x1001, isn't it? not sure I follow the call flows in this scenario.
 
-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take Cisco Learning credits!
http://www.ccbootcamp.com <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.ccbootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012


________________________________

From: Jeff Anderson (Enventis) [mailto:janderson at enventis.com]
Sent: Tue 5/29/2007 12:20 PM
To: Avner Izhar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CME Call-forward question [bcc][faked-from][html-rem] [mx][html-rem]



The voicemail system is CUE. Both of those numbers (1000 and 1001) have
a VM box in unity so I don't know how you would use an alternate
extension.

Thanks,

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com]
Sent: Tuesday, May 29, 2007 1:47 PM
To: Jeff Anderson (Enventis); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem]

What voice mail system? unity? cue?

In unity you can configure the alternate extension to contain 1001 in
it.

You can also translate the calling number using voice translation rules
on the cme.

-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take
Cisco Learning credits!
http://www.ccbootcamp.com <http://www.ccbootcamp.com/> 
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
(Enventis)
Sent: Tue 5/29/2007 10:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem]
The scenario:

CME 3.3

x1000 is CFA to x1001. I want the caller who called x1000 to get the
voicemail box of x1001 if x1001 doesn't answer.

Is this possible? I know in call manager you would need to create a
unique VM profile to accomplish this.

Thanks,

Jeff
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From janderson at enventis.com  Tue May 29 15:34:06 2007
From: janderson at enventis.com (Jeff Anderson (Enventis))
Date: Tue, 29 May 2007 14:34:06 -0500
Subject: [cisco-voip] CME Call-forward question
	[bcc][faked-from][html-rem] [mx][html-rem]
In-Reply-To: <E76ACA8C13EF1847BC69538B522C3663561002@nlimail.nlidomain>
Message-ID: <C0A504750F3BBE4FADE4ED6D672BA1790C57AF08@mail.hickorytech.local>

The call forward is a temporary thing.
 
For example:
 
Joe at x1000 is out of town for a week. During this time period they
would like all his calls to go to Sally. If sally doesn't answer her
phone they want the caller to leave a VM in Sally's mailbox.
 
I do understand what needs to be done to translate that call coming into
the system. I was hoping for a more user friendly solution.
 
Thanks,
 
________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Tuesday, May 29, 2007 2:26 PM
To: Jeff Anderson (Enventis); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem] [mx][html-rem]
 
Why would you need a voice mail box for x1000, it is CFA to x1001, isn't
it? not sure I follow the call flows in this scenario.
 
-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take
Cisco Learning credits!
http://www.ccbootcamp.com
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012
 
________________________________

From: Jeff Anderson (Enventis) [mailto:janderson at enventis.com]
Sent: Tue 5/29/2007 12:20 PM
To: Avner Izhar; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem] [mx][html-rem]
The voicemail system is CUE. Both of those numbers (1000 and 1001) have
a VM box in unity so I don't know how you would use an alternate
extension.

Thanks,

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com]
Sent: Tuesday, May 29, 2007 1:47 PM
To: Jeff Anderson (Enventis); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem]

What voice mail system? unity? cue?

In unity you can configure the alternate extension to contain 1001 in
it.

You can also translate the calling number using voice translation rules
on the cme.

-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take
Cisco Learning credits!
http://www.ccbootcamp.com <http://www.ccbootcamp.com/> 
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
(Enventis)
Sent: Tue 5/29/2007 10:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem]
The scenario:

CME 3.3

x1000 is CFA to x1001. I want the caller who called x1000 to get the
voicemail box of x1001 if x1001 doesn't answer.

Is this possible? I know in call manager you would need to create a
unique VM profile to accomplish this.

Thanks,

Jeff
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From rylatorr at cisco.com  Tue May 29 15:48:08 2007
From: rylatorr at cisco.com (Ryan LaTorre (rylatorr))
Date: Tue, 29 May 2007 15:48:08 -0400
Subject: [cisco-voip] Cisco Unity, IMAP and Novell GroupWise
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005B22CDF@USNAEXCH.na.didata.local>
Message-ID: <A640D47D9F23E7469C24C96195A7125603D9777F@xmb-rtp-20d.amer.cisco.com>

I'm not familiar with the Groupwise client, but when I use IMAP to
access a regular mailstore from Outlook, I have to press the "Purge
Deleted Messages" command to actually execute the delete.  "Deleting"
the message with IMAP only marks the message for deletion.
 
Maybe there's something similar in Groupwise?
 
-Ryan

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Coner
(US)
Sent: Tuesday, May 29, 2007 1:42 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco Unity, IMAP and Novell GroupWise



Anyone successfully configure an IMAP integration with Unity 4.2 and
GroupWise client 6.5.4?

 

We are able to retrieve messages but if the user deletes a message using
the phone the client doesn't seem to get updated.  

 

If I configure the IMAP connection using Outlook or Outlook Express I
can't reproduce the problem.

 

 

 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 

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From scott.coner at us.didata.com  Tue May 29 15:53:24 2007
From: scott.coner at us.didata.com (Scott Coner (US))
Date: Tue, 29 May 2007 15:53:24 -0400
Subject: [cisco-voip] Cisco Unity, IMAP and Novell GroupWise
In-Reply-To: <A640D47D9F23E7469C24C96195A7125603D9777F@xmb-rtp-20d.amer.cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F005B22CDF@USNAEXCH.na.didata.local>
	<A640D47D9F23E7469C24C96195A7125603D9777F@xmb-rtp-20d.amer.cisco.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B23082@USNAEXCH.na.didata.local>

There is something similar in GroupWise.  My issue is that the message
never leaves the "Inbox" and displays as a new (unread) message.  Of
course, if I've already deleted the message via the phone the message
showing in the Inbox doesn't open which makes sense since it's been
deleted.

 

If I use Outlook, Outlook Express or Mozilla Thunderbird it works as
expected.  My customer is a Novell shop and really wants this working
with GroupWise.  At this point I'm going to contact Novell and see what
happens.

 

Thanks,

Scott

 

________________________________

From: Ryan LaTorre (rylatorr) [mailto:rylatorr at cisco.com] 
Sent: Tuesday, May 29, 2007 3:48 PM
To: Scott Coner (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Cisco Unity, IMAP and Novell GroupWise

 

I'm not familiar with the Groupwise client, but when I use IMAP to
access a regular mailstore from Outlook, I have to press the "Purge
Deleted Messages" command to actually execute the delete.  "Deleting"
the message with IMAP only marks the message for deletion.

 

Maybe there's something similar in Groupwise?

 

-Ryan

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott Coner
(US)
Sent: Tuesday, May 29, 2007 1:42 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco Unity, IMAP and Novell GroupWise

Anyone successfully configure an IMAP integration with Unity 4.2 and
GroupWise client 6.5.4?

 

We are able to retrieve messages but if the user deletes a message using
the phone the client doesn't seem to get updated.  

 

If I configure the IMAP connection using Outlook or Outlook Express I
can't reproduce the problem.

 

 

 

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 




-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.
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From Eric.A.Fretz at L-3Com.com  Tue May 29 15:27:19 2007
From: Eric.A.Fretz at L-3Com.com (Fretz, EA Eric @ IS)
Date: Tue, 29 May 2007 14:27:19 -0500
Subject: [cisco-voip] VoIP trunk calls that will not die!
Message-ID: <F2E442781002EA41B3DB013749748D30063F1928@gvlexch02.is.l-3com.com>

Ok folks, I have a weird one for you.

I have a Cisco 3845 (12.4.5-IPSERVICES) with 8 E&M ports installed.
Each E&M port is setup for a trunk connection to a MeetMe conference
number at our main location.  This works perfectly, except that during a
WAN outage, the trunk calls never go back into "pending" state even
though the conference is unreachable.  I have actually disconnected the
3845 for 48 hours (in a test lab, of course) and the calls showed to be
"S_TRUNKED" the entire time.  This is a problem because the endpoint of
these VoIP trunk calls are DSPs in a WS-SVC-CMM that will timeout after
two hours of RTP inactivity (two hours is the maximum value).  After the
DSPs timeout and WAN connectivity is reestablished, the call will
reconnect to the conference bridge but it will never renegotiate DSP
resources unless the E&M interfaces are manually shutdown and the
conference is deleted.

I have tried various combinations of RTP timeout values on the 3845
router and nothing seems to make those calls die when the WAN connection
is severed.

Any ideas?

From jsteinberg at gmail.com  Tue May 29 16:02:17 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 29 May 2007 16:02:17 -0400
Subject: [cisco-voip] CME Call-forward question
	[bcc][faked-from][html-rem] [mx][html-rem]
In-Reply-To: <C0A504750F3BBE4FADE4ED6D672BA1790C57AF08@mail.hickorytech.local>
References: <E76ACA8C13EF1847BC69538B522C3663561002@nlimail.nlidomain>
	<C0A504750F3BBE4FADE4ED6D672BA1790C57AF08@mail.hickorytech.local>
Message-ID: <f1b65f880705291302h2da2dc6fu71f7bdeda429d3dd@mail.gmail.com>

Last I checked (prior to CME 4), the CME/CUE solution worked of off
'Last Redirected Party" when handling forwarded calls.  This is the
complete opposite from CallManager which operates under the 'First
Redirected Party'.

I don't believe there is any way to change that functionality for CM
or CME.  Lucky for you the CME/CUE solution does exactly what you
want.

Justin

On 5/29/07, Jeff Anderson (Enventis) <janderson at enventis.com> wrote:
>
>
>
>
> The call forward is a temporary thing.
>
>
>
> For example:
>
>
>
> Joe at x1000 is out of town for a week. During this time period they would
> like all his calls to go to Sally. If sally doesn't answer her phone they
> want the caller to leave a VM in Sally's mailbox.
>
>
>
> I do understand what needs to be done to translate that call coming into the
> system. I was hoping for a more user friendly solution.
>
>
>
> Thanks,
>
>
>
>  ________________________________
>
>
> From: Avner Izhar [mailto:aizhar at ccbootcamp.com]
>  Sent: Tuesday, May 29, 2007 2:26 PM
>  To: Jeff Anderson (Enventis); cisco-voip at puck.nether.net
>  Subject: RE: [cisco-voip] CME Call-forward question
> [bcc][faked-from][html-rem] [mx][html-rem]
>
>
>
>
>
>
> Why would you need a voice mail box for x1000, it is CFA to x1001, isn't it?
> not sure I follow the call flows in this scenario.
>
>
>
>  ------------------------------------------- Regards, Avner
> Izhar Technical Instructor CCIE #15999 (Voice) Voice: (702) 968-5100 Fax:
> (702) 446-8012 E-mail: aizhar at ccbootcamp.com Network Learning Inc - A Cisco
> Sponsored Organization(SO) YES! We take Cisco Learning credits!
> http://www.ccbootcamp.com (Cisco Training and Rental Racks)
>  http://www.ccbootcamp.com/groupstudy.html (groupstudy
> member discounts!)
>  Toll Free: 877-NLI-CCIE (877-654-2243)
>  Outside USA: 702-968-5100
>  FAX: 702-446-8012
>
>
>
>  ________________________________
>
>
> From: Jeff Anderson (Enventis) [mailto:janderson at enventis.com]
>  Sent: Tue 5/29/2007 12:20 PM
>  To: Avner Izhar; cisco-voip at puck.nether.net
>  Subject: RE: [cisco-voip] CME Call-forward question
> [bcc][faked-from][html-rem] [mx][html-rem]
>
>
> The voicemail system is CUE. Both of those numbers (1000 and 1001) have
>  a VM box in unity so I don't know how you would use an alternate
>  extension.
>
>  Thanks,
>
>  ________________________________
>
>  From: Avner Izhar [mailto:aizhar at ccbootcamp.com]
>  Sent: Tuesday, May 29, 2007 1:47 PM
>  To: Jeff Anderson (Enventis); cisco-voip at puck.nether.net
>  Subject: RE: [cisco-voip] CME Call-forward question
>  [bcc][faked-from][html-rem]
>
>  What voice mail system? unity? cue?
>
>  In unity you can configure the alternate extension to contain 1001 in
>  it.
>
>  You can also translate the calling number using voice translation rules
>  on the cme.
>
>  -------------------------------------------
>  Regards,
>  Avner Izhar
>  Technical Instructor
>  CCIE #15999 (Voice)
>  Voice: (702) 968-5100
>  Fax: (702) 446-8012
>  E-mail: aizhar at ccbootcamp.com
>  Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take
>  Cisco Learning credits!
>  http://www.ccbootcamp.com
> <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
>  bootcamp.com>  (Cisco Training and Rental Racks)
>  http://www.ccbootcamp.com/groupstudy.html
> <https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
>  bootcamp.com/groupstudy.html>  (groupstudy member discounts!)
>  Toll Free: 877-NLI-CCIE (877-654-2243)
>  Outside USA: 702-968-5100
>  FAX: 702-446-8012
>
>  ________________________________
>
>  From: cisco-voip-bounces at puck.nether.net on behalf of Jeff
> Anderson
>  (Enventis)
>  Sent: Tue 5/29/2007 10:59 AM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] CME Call-forward question
>  [bcc][faked-from][html-rem]
>  The scenario:
>
>  CME 3.3
>
>  x1000 is CFA to x1001. I want the caller who called x1000 to get the
>  voicemail box of x1001 if x1001 doesn't answer.
>
>  Is this possible? I know in call manager you would need to create a
>  unique VM profile to accomplish this.
>
>  Thanks,
>
>  Jeff
>  _______________________________________________
>  cisco-voip mailing list
>  cisco-voip at puck.nether.net
>  https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From CarlosOrtiz at bayviewfinancial.com  Tue May 29 16:04:40 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 29 May 2007 16:04:40 -0400
Subject: [cisco-voip] Enterprise Contact Center
Message-ID: <OF34DF2279.329BA6DD-ON852572EA.006E2B2C-852572EA.006E4AF4@bftg.com>

Anyone have any suggestions for a an Enterprise Contact Center besides 
IPCC.  I would actually like to compare features and functionality with 
IPCC Enterprise.

Carlos
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From nick.jon.griffin at gmail.com  Tue May 29 16:18:46 2007
From: nick.jon.griffin at gmail.com (Nick Griffin)
Date: Tue, 29 May 2007 15:18:46 -0500
Subject: [cisco-voip] Hunt Groups and DND
In-Reply-To: <407055A92CECCB499C922A2D35FC19A608DEE7FB@Apollo.wesd.org>
References: <a2b352c40705241539r28305d68h98ec90615ddd4eee@mail.gmail.com>
	<f1b65f880705241549o7176707dnf11f039b3daf82e1@mail.gmail.com>
	<407055A92CECCB499C922A2D35FC19A608DEE7FB@Apollo.wesd.org>
Message-ID: <a2b352c40705291318o4c92189t9d27dfd4942c5272@mail.gmail.com>

Any other thoughts of suggestions? Thanks

Nick Griffin

On 5/25/07, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
> IPCCx.
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
> Steinberg
> Sent: Thursday, May 24, 2007 3:49 PM
> To: Nick Griffin
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Hunt Groups and DND
>
> This exact feature is in 4.2 or 6.0.
>
> You can hack it in 4.1 or 5.1 by using extension mobility and have
> their user device profile add a second line that is the hunt group
> member.
>
> Others may have different approaches...
>
>
>
> On 5/24/07, Nick Griffin <nick.jon.griffin at gmail.com> wrote:
> > I have a 5.1 install coming up in the near future. On the customers
> existing
> > system they currently use sequential hunt groups in combination with
> DND.
> > Their goal is to route customer calls to reps ASAP, the do not disturb
> > feature keeps the customers wait short by not routing the call to the
> hunt
> > group member while that member is in DND mode. Does anyone know of a
> similar
> > solution that could be accomplish on CCM5.1? I'm thinking Extension
> Mobility
> > with 1 button login, but that seems to take to long with the reset.
> They
> > essentially need to be able to "log out" of a hunt group. Looking for
> any
> > suggestions.
> >
> >  Regards,
> >
> > Nick Griffin
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From CarlosOrtiz at bayviewfinancial.com  Tue May 29 17:04:42 2007
From: CarlosOrtiz at bayviewfinancial.com (CarlosOrtiz at bayviewfinancial.com)
Date: Tue, 29 May 2007 17:04:42 -0400
Subject: [cisco-voip] Enterprise Contact Center
In-Reply-To: <03c501c7a231$e1a2f980$0232a8c0@pwnhome.com>
Message-ID: <OFDDE7877A.DEDA714F-ON852572EA.0073B177-852572EA.0073CA44@bftg.com>

Do these integrate with Call Mgr or require their own PBX to run?



"Philip Walenta" <pwalenta at wi.rr.com> 
05/29/2007 04:42 PM

To
<CarlosOrtiz at bayviewfinancial.com>
cc

Subject
RE: [cisco-voip] Enterprise Contact Center






You can try Interactive Intelligence (www.inin.com I think).  You can also 
check out Symposium from Nortel, and Genesys from Avaya.

From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of 
CarlosOrtiz at bayviewfinancial.com
Sent: Tuesday, May 29, 2007 3:05 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Enterprise Contact Center


Anyone have any suggestions for a an Enterprise Contact Center besides 
IPCC.  I would actually like to compare features and functionality with 
IPCC Enterprise. 

Carlos
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From Nesen.Kisten at za.didata.com  Tue May 29 17:20:56 2007
From: Nesen.Kisten at za.didata.com (Nesen Kisten (ZA))
Date: Tue, 29 May 2007 23:20:56 +0200
Subject: [cisco-voip] MGCP Gateway not registering to Subscriber
References: <DA507CEAA8B853469A9A74AD432B58F41A54BD@ZABRYSVCL17B.af.didata.local>
Message-ID: <DA507CEAA8B853469A9A74AD432B58F41A54BE@ZABRYSVCL17B.af.didata.local>

Hi All,
 
One of my clients currently running CCM 5.1 are currently having a problem, when I stop the publisher CCM service the gateway successfully registers to back-up subscriber, however when I physically shut the port on the switch to which the publisher is connected, the gateway can never register to subscriber.
 
Any thoughts ?
 
Kind Regards
 
Nesen

This email and all contents are subject to the following disclaimer:

"http://www.dimensiondata.com/emaildisclaimer.htm"
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From Nesen.Kisten at za.didata.com  Tue May 29 16:46:44 2007
From: Nesen.Kisten at za.didata.com (Nesen Kisten (ZA))
Date: Tue, 29 May 2007 22:46:44 +0200
Subject: [cisco-voip] MGCP Gateway not registering to Subscriber
Message-ID: <DA507CEAA8B853469A9A74AD432B58F41A54BD@ZABRYSVCL17B.af.didata.local>

Hi All,
 
One of my clients currently running CCM 5.1 are currently having a problem, when I stop the publisher CCM service the gateway successfully registers to back-up subscriber, however when I physically shut the port on the switch to which the publisher is connected, the gateway can never register to subscriber.
 
Any thoughts ?
 
Kind Regards
 
Nesen

This email and all contents are subject to the following disclaimer:

"http://www.dimensiondata.com/emaildisclaimer.htm"
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From pwalenta at wi.rr.com  Tue May 29 18:04:33 2007
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue, 29 May 2007 17:04:33 -0500
Subject: [cisco-voip] Enterprise Contact Center
In-Reply-To: <OFDDE7877A.DEDA714F-ON852572EA.0073B177-852572EA.0073CA44@bftg.com>
Message-ID: <03dc01c7a23d$57364e30$0232a8c0@pwnhome.com>

I know Interactive Intelligence does (or did at one time), unsure about the
others.

  _____  

From: CarlosOrtiz at bayviewfinancial.com
[mailto:CarlosOrtiz at bayviewfinancial.com] 
Sent: Tuesday, May 29, 2007 4:05 PM
To: Philip Walenta
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Enterprise Contact Center



Do these integrate with Call Mgr or require their own PBX to run? 



"Philip Walenta" <pwalenta at wi.rr.com> 


05/29/2007 04:42 PM 


To
<CarlosOrtiz at bayviewfinancial.com> 

cc

Subject
RE: [cisco-voip] Enterprise Contact Center

	




You can try Interactive Intelligence ( <http://www.inin.com/> www.inin.com I
think).  You can also check out Symposium from Nortel, and Genesys from
Avaya. 


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
CarlosOrtiz at bayviewfinancial.com
Sent: Tuesday, May 29, 2007 3:05 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Enterprise Contact Center


Anyone have any suggestions for a an Enterprise Contact Center besides IPCC.
I would actually like to compare features and functionality with IPCC
Enterprise. 

Carlos 

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From smarwah at enetsolutions.net  Tue May 29 18:42:23 2007
From: smarwah at enetsolutions.net (Sumit Marwah)
Date: Tue, 29 May 2007 17:42:23 -0500
Subject: [cisco-voip] CME Call-forward
	question[bcc][faked-from][html-rem]
In-Reply-To: <C0A504750F3BBE4FADE4ED6D672BA1790C57AEE4@mail.hickorytech.local>
References: <E76ACA8C13EF1847BC69538B522C3663560FFF@nlimail.nlidomain>
	<C0A504750F3BBE4FADE4ED6D672BA1790C57AEE4@mail.hickorytech.local>
Message-ID: <FF7F71539DC4E544BA5737C7AA6CEE17369376@ems05.ems-1.net>

You will need to have CME 4.0 with CUE 2.3.1 and there is an option
under voicemail default to play greeting for 'last redirected number".

 

Regards,

 

Sumit Marwah 
Sr. System Engineer
CCIE Voice # 17400

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jeff Anderson
(Enventis)
Sent: Tuesday, May 29, 2007 2:21 PM
To: Avner Izhar; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CME Call-forward
question[bcc][faked-from][html-rem]

 

The voicemail system is CUE. Both of those numbers (1000 and 1001) have
a VM box in unity so I don't know how you would use an alternate
extension.

 

Thanks,

 

________________________________

From: Avner Izhar [mailto:aizhar at ccbootcamp.com] 
Sent: Tuesday, May 29, 2007 1:47 PM
To: Jeff Anderson (Enventis); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem]

 

What voice mail system? unity? cue?

 

In unity you can configure the alternate extension to contain 1001 in
it.

 

You can also translate the calling number using voice translation rules
on the cme. 

 

-------------------------------------------
Regards,
Avner Izhar
Technical Instructor
CCIE #15999 (Voice)
Voice: (702) 968-5100
Fax: (702) 446-8012
E-mail: aizhar at ccbootcamp.com
Network Learning Inc - A Cisco Sponsored Organization(SO) YES! We take
Cisco Learning credits!
http://www.ccbootcamp.com
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com>  (Cisco Training and Rental Racks)
http://www.ccbootcamp.com/groupstudy.html
<https://exchange.ccbootcamp.com/exchweb/bin/redir.asp?URL=http://www.cc
bootcamp.com/groupstudy.html>  (groupstudy member discounts!)
Toll Free: 877-NLI-CCIE (877-654-2243)
Outside USA: 702-968-5100
FAX: 702-446-8012

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
(Enventis)
Sent: Tue 5/29/2007 10:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME Call-forward question
[bcc][faked-from][html-rem]

The scenario:

CME 3.3

x1000 is CFA to x1001. I want the caller who called x1000 to get the
voicemail box of x1001 if x1001 doesn't answer.

Is this possible? I know in call manager you would need to create a
unique VM profile to accomplish this.

Thanks,

Jeff
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From chris.clouse at berbee.com  Tue May 29 21:42:06 2007
From: chris.clouse at berbee.com (Clouse, Chris)
Date: Tue, 29 May 2007 20:42:06 -0500
Subject: [cisco-voip] MGCP Gateway not registering to Subscriber
In-Reply-To: <DA507CEAA8B853469A9A74AD432B58F41A54BE@ZABRYSVCL17B.af.didata.local>
References: <DA507CEAA8B853469A9A74AD432B58F41A54BD@ZABRYSVCL17B.af.didata.local>
	<DA507CEAA8B853469A9A74AD432B58F41A54BE@ZABRYSVCL17B.af.didata.local>
Message-ID: <310F10740485D9469F0D8417950B252D028D3262@CTG-MSNEXC01.staff.berbee.com>

Can you post the gateway configuration?

 

Christopher Clouse, CCNP CCDP CCVP MCP Network+

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nesen Kisten
(ZA)
Sent: Tuesday, May 29, 2007 4:21 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP Gateway not registering to Subscriber

 

Hi All,

 

One of my clients currently running CCM 5.1 are currently having a
problem, when I stop the publisher CCM service the gateway successfully
registers to back-up subscriber, however when I physically shut the port
on the switch to which the publisher is connected, the gateway can never
register to subscriber.

 

Any thoughts ?

 

Kind Regards

 

Nesen

This email and all contents are subject to the following disclaimer:

"http://www.dimensiondata.com/emaildisclaimer.htm"

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From bcarter at sentinel.com  Tue May 29 21:55:36 2007
From: bcarter at sentinel.com (Carter, Bill)
Date: Tue, 29 May 2007 20:55:36 -0500
Subject: [cisco-voip] QOS for MPLS
In-Reply-To: <6A0744639B86DD459433108DBF101D26016AED73@MAIL01.sentinel.com>
References: <262147.28980.qm@web37210.mail.mud.yahoo.com><5d093f9a0705271454u6e0c479cqee972a6fcd9c16a5@mail.gmail.com>
	<6A0744639B86DD459433108DBF101D26016AED73@MAIL01.sentinel.com>
Message-ID: <6A0744639B86DD459433108DBF101D26016AEDC0@MAIL01.sentinel.com>

What about traffic shaping at the head end? 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Tuesday, May 29, 2007 1:43 PM
To: Jonathan Charles; Erick Bergquist
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] QOS for MPLS

With MPLS and sites with different bandwidths, you should set the QoS
based on the site bandwidth.  So it is ok to have 800k on the T1's and
2500k on the DS3's.  Locations would prevent HQ from flooding a
particular site.  

Because of the mesh possibilities of MPLS, I would set the HQ Location
bandwidth to unlimited and the remote at ~800kb. 

If this is Unified CM 5.X you could use RSVP.  CM locations are more
designed for a Hub and Spoke network.  RSVP accommodates a meshed
topology.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Sunday, May 27, 2007 4:54 PM
To: Erick Bergquist
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] QOS for MPLS

Which is the definition of MPLS, any-to-any communication. All circuits
are point to point into the MPLS cloud and then routing in the cloud to
each site.



Jonathan

On 5/27/07, Erick Bergquist <erickbe at yahoo.com> wrote:
> I've seen one or two setups where the MPLS was fully meshed (all sites
could talk to any other site), which makes QoS even trickier.
>
> ----- Original Message ----
> From: "keli at carocomp.ro" <keli at carocomp.ro>
> To: Jonathan Charles <jonvoip at gmail.com>
> Cc: cisco-voip at puck.nether.net
> Sent: Sunday, May 27, 2007 3:50:17 PM
> Subject: Re: [cisco-voip] QOS for MPLS
>
> That is correct (although you should take some overhead into account 
> as well). However, if you do not perform Call Admission Control 
> beforehand (you do not limit the maximum number of possible 
> simultaneous calls) you've done nothing, or worse.
>
> The classic example being when you have a line that supports 4 calls, 
> when the fifth call enters the line, all calls will suffer packet loss

> and delay, not only the fifth one.
>
> I'm not too versed in MPLS QoS, but the idea is always the same.
> Prioritize voice traffic, but don't let it kill your bandwidth, nor 
> itself. Thus the need for CAC, then something like CBWFQ/PQ. If 
> possible, voice traffic should be beneath shaping/policing limits, so 
> that neither would affect it.
>
> regards,
>    Zoltan
>
> Quoting Jonathan Charles <jonvoip at gmail.com>:
>
> > My thinking is that I need to set the QOS to be the same as the 
> > maximum number of calls to/from each site.
> >
> >
> > Jonathan
> >
> > On 5/27/07, keli at carocomp.ro <keli at carocomp.ro> wrote:
> >> As long as it is about "calls", CAC is the sensitive way to go 
> >> about limiting the calls in the first place. Do QoS "As close to 
> >> the source, as possible", remember? :)
> >>
> >> If you do not have a CCM, but other set-up, you should still look 
> >> in this direction first - to limit the calls to a reasonable amount

> >> in the given directions, before they are hitting your MPLS router.
> >>
> >> regards,
> >>   Zoltan
> >>
> >> Quoting Paul Choi <asobihoudai at yahoo.com>:
> >>
> >>> He's talking about QoS for MPLS, not CAC.
> >>>
> >>> --- "Clouse, Chris" <chris.clouse at berbee.com> wrote:
> >>>
> >>>> Use CallManager locations to perform Call Admission Control.
> >>>>
> >>>> Christopher Clouse, CCNP CCDP CCVP MCP Network+
> >>>>
> >>>>
> >>>> -----Original Message-----
> >>>> From: cisco-voip-bounces at puck.nether.net
> >>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan

> >>>> Charles
> >>>> Sent: Saturday, May 26, 2007 11:35 AM
> >>>> To: ciscovoip
> >>>> Subject: [cisco-voip] QOS for MPLS
> >>>>
> >>>> I have always understood that QOS settings need to match on both 
> >>>> ends of a circuit... so, if you allocate 800k on one side, you 
> >>>> need to do the same on the other.
> >>>>
> >>>> I am now working on an MPLS circuit that has DS3s on some sides, 
> >>>> DS1s on others... do I still need to match QOS settings?
> >>>>
> >>>> Customer wants 800k on the T1s and 2500k on the DS3s... I don't 
> >>>> think it will matter as I am matching these bw limits on the 
> >>>> locations...
> >>>> but how do I throttle calls from a DS3-side location to a DS1 
> >>>> location?
> >>>>
> >>>> For example... main site is in Chicago, they have a DS3, one 
> >>>> remote site is in Wichita and has a T1... so, I put 800k for 
> >>>> Wichita and 2500k for Chicago... what is to stop Chicago from 
> >>>> flooding Wichita with 2.5MB of calls?
> >>>>
> >>>>
> >>>>
> >>>> Jonathan
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >> ___________________________________________________________________
> >> _________________Building a website is a piece of cake. Yahoo! 
> >> Small Business gives you all the tools to get
> >>> online.
> >>> http://smallbusiness.yahoo.com/webhosting
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip at puck.nether.net
> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>
> >>
> >>
> >>
> >> ----------------------------------------------------------------
> >> This message was sent using IMP, the Internet Messaging Program.
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip at puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
>
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
> ______________________________________________________________________
> ______________ We won't tell. Get more on shows you hate to love (and 
> love to hate): Yahoo! TV's Guilty Pleasures list.
> http://tv.yahoo.com/collections/265
>
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From jeff at ocjtech.us  Tue May 29 21:59:10 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Tue, 29 May 2007 20:59:10 -0500
Subject: [cisco-voip] CallManager 5.2
In-Reply-To: <5d093f9a0705260852i139101d5w7756f41fabac219a@mail.gmail.com>
References: <5d093f9a0705260846k4ffe1c6cra465694ab14dab9c@mail.gmail.com>
	<5d093f9a0705260852i139101d5w7756f41fabac219a@mail.gmail.com>
Message-ID: <1180490350.5411.4.camel@lt21223.campus.dmacc.edu>

On Sat, 2007-05-26 at 10:52 -0500, Jonathan Charles wrote:
> Correction, 5.1.2... not 5.2
>
> On 5/26/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > So, anyone upgrade yet?
> >
> > Just curious if it is going to kill QoS or RTP streams... TAC said it
> > was just bug fixes...

I upgraded from 5.1.1a to 5.1.2 to fix an issue with CDR processing - no
new issues seen so far.

Jeff

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From nws at fredf.org  Tue May 29 22:29:49 2007
From: nws at fredf.org (Kent Roberts)
Date: Tue, 29 May 2007 20:29:49 -0600
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
In-Reply-To: <465C70EB.9020805@smalltime.com>
References: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>
	<465C6F69.7070307@carocomp.ro> <465C70EB.9020805@smalltime.com>
Message-ID: <3e6f56310705291929y4946b047maab2db9a80fb5d76@mail.gmail.com>

Just a thought,   Do you have dedicated LD circuits?  If so are you routing
sending that 800s via those circuits?   The 800 calls can not be sent out LD
circuits.  They must go out the local.







On 5/29/07, Robert Kulagowski <bob at smalltime.com> wrote:
>
> > Miller, Steve wrote:
> >> We have an odd problem.  In our script on the CRS server, we
> >> transfer calls to an external number for our backup helpdesk
> >> vendor.  This works fine when we insert their regular phone number
> >> which is 404-XXX-XXXX, but when we enter the toll free number
> >> 800-XXX-XXXX that is forwarded to the 404 number by the carrier, we
> >> get a busy signal.
>
> Check your Q931 traces; does it look like this?  (I'm also about to
> start troubleshooting this issue, so I don't have an answer).
>
> If the site dials the 800 number directly, it works, but if they use a
> site-wide speed dial that's a CFA on a line that's attached to a
> "SpeedDial" CTI port, it doesn't work.  (*660 is the speed dial)
>
> May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: Applying typeplan for sw-type
> 0xD is 0x2 0x1, Calling num 213928
> May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: Applying typeplan for sw-type
> 0xD is 0x2 0x1, Called num 18005797372
> May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8  callref =
> 0x06A3
>          Bearer Capability i = 0x8090A2
>                  Standard = CCITT
>                  Transfer Capability = Speech
>                  Transfer Mode = Circuit
>                  Transfer Rate = 64 kbit/s
>          Channel ID i = 0xA98382
>                  Exclusive, Channel 2
>          Calling Party Number i = 0x21A1, '213928'
>                  Plan:ISDN, Type:National
>          Called Party Number i = 0xA1, '18005797372'
>                  Plan:ISDN, Type:National
>          Redirecting Number i = '!', 0x008F, '??*660'
>                  Plan:ISDN, Type:National
>
> May 29 10:47:41.015: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8
> callref = 0x86A3 Channel ID i = 0xA98382 Exclusive, Channel 2
>
> May 29 10:47:41.023: ISDN Se0/1/0:23 Q931: RX <- PROGRESS pd = 8
> callref = 0x86A3
>          Cause i = 0x80FF - Interworking error; unspecified
>          Progress Ind i = 0x8088 - In-band info or appropriate now
> available
>
> May 29 10:47:44.935: ISDN Se0/1/0:23 Q931: TX -> DISCONNECT pd = 8
> callref = 0x06A3 Cause i = 0x8090 - Normal call clearing
>
> May 29 10:47:45.015: ISDN Se0/1/0:23 Q931: RX <- RELEASE pd = 8  callref
> = 0x86A3
>
> May 29 10:47:45.019: ISDN Se0/1/0:23 Q931: TX -> RELEASE_COMP pd = 8
> callref = 0x06A3
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From MillerS at DicksteinShapiro.COM  Wed May 30 08:39:34 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Wed, 30 May 2007 08:39:34 -0400
Subject: [cisco-voip] "Delete failed 0x80043307
	(E_NOTIFIER_USER_NOT_MONITORED)."
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538BE6@DCEX2.DSMO.COM>

When I attempt to delete one of the mailboxes on our system, I receive
the following message: 
 
"Delete failed 0x80043307 (E_NOTIFIER_USER_NOT_MONITORED)."
 
Any insight would be greatly appreciated.  
 
 
Thank you!
 
Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 
 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

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From rduric at uoguelph.ca  Wed May 30 08:53:42 2007
From: rduric at uoguelph.ca (Rasim Duric)
Date: Wed, 30 May 2007 08:53:42 -0400
Subject: [cisco-voip] "Delete failed
	0x80043307(E_NOTIFIER_USER_NOT_MONITORED)."
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538BE6@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490006538BE6@DCEX2.DSMO.COM>
Message-ID: <BD89624EE83BD349BD31A08B453DE244017C1435@uogexb201.cfs.uoguelph.ca>

You might be hitting into
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCsd03143 

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Wednesday, May 30, 2007 8:40 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] "Delete failed
0x80043307(E_NOTIFIER_USER_NOT_MONITORED)."

 

When I attempt to delete one of the mailboxes on our system, I receive
the following message: 

 

"Delete failed 0x80043307 (E_NOTIFIER_USER_NOT_MONITORED)."

 

Any insight would be greatly appreciated.  

 

 

Thank you!

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370| Fax (202) 330-5607
MillerS at dicksteinshapiro.com 

 

--------------------------------------------------------
This e-mail message and any attached files are confidential and are
intended solely for the use of the addressee(s) named above. This
communication may contain material protected by attorney-client, work
product, or other privileges. If you are not the intended recipient or
person responsible for delivering this confidential communication to the
intended recipient, you have received this communication in error, and
any review, use, dissemination, forwarding, printing, copying, or other
distribution of this e-mail message and any attached files is strictly
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communication that is created, received, or sent on its network.  If you
have received this confidential communication in error, please notify
the sender immediately by reply e-mail message and permanently delete
the original message. 
 
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From MLinsemier at apcapital.com  Wed May 30 09:08:48 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Wed, 30 May 2007 09:08:48 -0400
Subject: [cisco-voip] Unity 4.2(1) and MS Exchange 2007
Message-ID: <AB0A35F2486F81468971AA68558BE7190658C9AE@smieln01winm01.micoanet.com>

Maybe this was discussed here before, but I was wondering if anyone knew
if the most current version of Unity 4.2(1) is compatible with MS
Exchange 2007, either officially or unofficially?  I saw some
information on Unity 5.0 for Exchange, but I wasn't sure of its
availability as of yet.  Our company is pushing to do a upgrade within
the next few months and I couldn't find anything on Cisco's site that
spoke of Exchange 2007 compatibility with older versions.  Additionally,
I didn't see any information on which CallManagers Unity 5.0 for
Exchange was compatible with (we are still on 4.1(3)sr5).  Can anyone
provide any input on this?

 

Thanks!

 

Matthew M. Linsemier, CCSP/CCNP/CCDA

Senior Network Engineer

American Physicians

Voice: 517.324.6695

Fax: 517.333.2812

E-Mail: mlinsemier at apcapital.com

 


CONFIDENTIALITY STATEMENT
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are not the intended recipient, any use, disclosure, copying
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From Matt.Slaga at us.didata.com  Wed May 30 09:21:13 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 30 May 2007 09:21:13 -0400
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
In-Reply-To: <3e6f56310705291929y4946b047maab2db9a80fb5d76@mail.gmail.com>
References: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM><465C6F69.7070307@carocomp.ro>
	<465C70EB.9020805@smalltime.com>
	<3e6f56310705291929y4946b047maab2db9a80fb5d76@mail.gmail.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B578D7@USNAEXCH.na.didata.local>

Check the CTI Port's CSS and calling permissions.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kent Roberts
Sent: Tuesday, May 29, 2007 10:30 PM
To: Robert Kulagowski
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Transferring to Outside Lines from the IPCC ?

 

Just a thought,   Do you have dedicated LD circuits?  If so are you
routing sending that 800s via those circuits?   The 800 calls can not be
sent out LD circuits.  They must go out the local.








On 5/29/07, Robert Kulagowski <bob at smalltime.com> wrote:

> Miller, Steve wrote:
>> We have an odd problem.  In our script on the CRS server, we
>> transfer calls to an external number for our backup helpdesk
>> vendor.  This works fine when we insert their regular phone number 
>> which is 404-XXX-XXXX, but when we enter the toll free number
>> 800-XXX-XXXX that is forwarded to the 404 number by the carrier, we
>> get a busy signal.

Check your Q931 traces; does it look like this?  (I'm also about to 
start troubleshooting this issue, so I don't have an answer).

If the site dials the 800 number directly, it works, but if they use a
site-wide speed dial that's a CFA on a line that's attached to a 
"SpeedDial" CTI port, it doesn't work.  (*660 is the speed dial)

May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: Applying typeplan for sw-type
0xD is 0x2 0x1, Calling num 213928
May 29 10:47:40.923 : ISDN Se0/1/0:23 Q931: Applying typeplan for
sw-type
0xD is 0x2 0x1, Called num 18005797372
May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8  callref =
0x06A3
         Bearer Capability i = 0x8090A2 
                 Standard = CCITT
                 Transfer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98382
                 Exclusive, Channel 2 
         Calling Party Number i = 0x21A1, '213928'
                 Plan:ISDN, Type:National
         Called Party Number i = 0xA1, '18005797372'
                 Plan:ISDN, Type:National
         Redirecting Number i = '!', 0x008F, '??*660' 
                 Plan:ISDN, Type:National

May 29 10:47:41.015: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8
callref = 0x86A3 Channel ID i = 0xA98382 Exclusive, Channel 2

May 29 10:47:41.023: ISDN Se0/1/0:23 Q931: RX <- PROGRESS pd = 8 
callref = 0x86A3
         Cause i = 0x80FF - Interworking error; unspecified
         Progress Ind i = 0x8088 - In-band info or appropriate now
available

May 29 10:47:44.935: ISDN Se0/1/0:23 Q931: TX -> DISCONNECT pd = 8 
callref = 0x06A3 Cause i = 0x8090 - Normal call clearing

May 29 10:47:45.015: ISDN Se0/1/0:23 Q931: RX <- RELEASE pd = 8  callref
= 0x86A3

May 29 10:47:45.019: ISDN Se0/1/0:23 Q931: TX -> RELEASE_COMP pd = 8 
callref = 0x06A3
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

 




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From Matt.Slaga at us.didata.com  Wed May 30 09:22:03 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 30 May 2007 09:22:03 -0400
Subject: [cisco-voip] Unity 4.2(1) and MS Exchange 2007
In-Reply-To: <AB0A35F2486F81468971AA68558BE7190658C9AE@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE7190658C9AE@smieln01winm01.micoanet.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B578DA@USNAEXCH.na.didata.local>

Unity 5.0 is the first version compatible with Exchange 2007 as per the
Unity business unit.  Can't say I have ever tried with 4.2 though.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Linsemier,
Matthew
Sent: Wednesday, May 30, 2007 9:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity 4.2(1) and MS Exchange 2007

 

Maybe this was discussed here before, but I was wondering if anyone knew
if the most current version of Unity 4.2(1) is compatible with MS
Exchange 2007, either officially or unofficially?  I saw some
information on Unity 5.0 for Exchange, but I wasn't sure of its
availability as of yet.  Our company is pushing to do a upgrade within
the next few months and I couldn't find anything on Cisco's site that
spoke of Exchange 2007 compatibility with older versions.  Additionally,
I didn't see any information on which CallManagers Unity 5.0 for
Exchange was compatible with (we are still on 4.1(3)sr5).  Can anyone
provide any input on this?

 

Thanks!

 

Matthew M. Linsemier, CCSP/CCNP/CCDA

Senior Network Engineer

American Physicians

Voice: 517.324.6695

Fax: 517.333.2812

E-Mail: mlinsemier at apcapital.com

 

________________________________

CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may be
protected by one or more legal privileges. It is intended solely for the
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communication is UNAUTHORIZED. Neither this information block, the typed
name of the sender, nor anything else in this message is intended to
constitute an electronic signature unless a specific statement to the
contrary is included in this message. If you have received this
communication in error, please immediately contact me and delete this
communication from your computer. Thank you. 

________________________________




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From bob at smalltime.com  Wed May 30 09:38:17 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 30 May 2007 08:38:17 -0500
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
In-Reply-To: <3e6f56310705291929y4946b047maab2db9a80fb5d76@mail.gmail.com>
References: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>	
	<465C6F69.7070307@carocomp.ro> <465C70EB.9020805@smalltime.com>
	<3e6f56310705291929y4946b047maab2db9a80fb5d76@mail.gmail.com>
Message-ID: <465D7E49.6090308@smalltime.com>

>     Check your Q931 traces; does it look like this?  (I'm also about to
>     start troubleshooting this issue, so I don't have an answer).
> 
>     If the site dials the 800 number directly, it works, but if they use a
>     site-wide speed dial that's a CFA on a line that's attached to a
>     "SpeedDial" CTI port, it doesn't work.  (*660 is the speed dial)
> 
>     May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: Applying typeplan for sw-type
>     0xD is 0x2 0x1, Calling num 213928
>     May 29 10:47:40.923 : ISDN Se0/1/0:23 Q931: Applying typeplan for
>     sw-type
>     0xD is 0x2 0x1, Called num 18005797372
>     May 29 10:47:40.923: ISDN Se0/1/0:23 Q931: TX -> SETUP pd = 8  callref =
>     0x06A3
>              Bearer Capability i = 0x8090A2
>                      Standard = CCITT
>                      Transfer Capability = Speech
>                      Transfer Mode = Circuit
>                      Transfer Rate = 64 kbit/s
>              Channel ID i = 0xA98382
>                      Exclusive, Channel 2
>              Calling Party Number i = 0x21A1, '213928'
>                      Plan:ISDN, Type:National
>              Called Party Number i = 0xA1, '18005797372'
>                      Plan:ISDN, Type:National
>              Redirecting Number i = '!', 0x008F, '??*660'
>                      Plan:ISDN, Type:National
> 
>     May 29 10:47:41.015: ISDN Se0/1/0:23 Q931: RX <- CALL_PROC pd = 8
>     callref = 0x86A3 Channel ID i = 0xA98382 Exclusive, Channel 2
> 
>     May 29 10:47:41.023: ISDN Se0/1/0:23 Q931: RX <- PROGRESS pd = 8
>     callref = 0x86A3
>              Cause i = 0x80FF - Interworking error; unspecified
>              Progress Ind i = 0x8088 - In-band info or appropriate now
>     available

Kent Roberts wrote:

> Just a thought,   Do you have dedicated LD circuits?  If so are you
> routing sending that 800s via those circuits?   The 800 calls can not be
> sent out LD circuits.  They must go out the local.

We do have dedicated LD, but 0/1/0 is the local circuit; dialing 
manually from the phone (instead of using the "speed dial") sends the 
call out the same PRI.  The only difference is that "Redirecting Number" 
  stuff which is hosing things up.

From MLinsemier at apcapital.com  Wed May 30 11:38:06 2007
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Wed, 30 May 2007 11:38:06 -0400
Subject: [cisco-voip] Unity 4.2(1) and MS Exchange 2007
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005B578DA@USNAEXCH.na.didata.local>
References: <AB0A35F2486F81468971AA68558BE7190658C9AE@smieln01winm01.micoanet.com>
	<C1FE15183DA37645BC0633BC604E44F005B578DA@USNAEXCH.na.didata.local>
Message-ID: <AB0A35F2486F81468971AA68558BE7190658CB21@smieln01winm01.micoanet.com>

Any idea if Unity 5.0 is backwards compatible with CCM 4.1(3)Sr5?

 

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Wednesday, May 30, 2007 9:22 AM
To: Linsemier, Matthew; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity 4.2(1) and MS Exchange 2007

 

Unity 5.0 is the first version compatible with Exchange 2007 as per the
Unity business unit.  Can't say I have ever tried with 4.2 though.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Linsemier,
Matthew
Sent: Wednesday, May 30, 2007 9:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity 4.2(1) and MS Exchange 2007

 

Maybe this was discussed here before, but I was wondering if anyone knew
if the most current version of Unity 4.2(1) is compatible with MS
Exchange 2007, either officially or unofficially?  I saw some
information on Unity 5.0 for Exchange, but I wasn't sure of its
availability as of yet.  Our company is pushing to do a upgrade within
the next few months and I couldn't find anything on Cisco's site that
spoke of Exchange 2007 compatibility with older versions.  Additionally,
I didn't see any information on which CallManagers Unity 5.0 for
Exchange was compatible with (we are still on 4.1(3)sr5).  Can anyone
provide any input on this?

 

Thanks!

 

Matthew M. Linsemier, CCSP/CCNP/CCDA

Senior Network Engineer

American Physicians

Voice: 517.324.6695

Fax: 517.333.2812

E-Mail: mlinsemier at apcapital.com

 

________________________________

CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may be
protected by one or more legal privileges. It is intended solely for the
use of the addressee identified above. If you are not the intended
recipient, any use, disclosure, copying or distribution of this
communication is UNAUTHORIZED. Neither this information block, the typed
name of the sender, nor anything else in this message is intended to
constitute an electronic signature unless a specific statement to the
contrary is included in this message. If you have received this
communication in error, please immediately contact me and delete this
communication from your computer. Thank you. 

________________________________

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
communication in error, please notify us immediately by replying to this
message and deleting it from your computer. Thank you. 


CONFIDENTIALITY STATEMENT
This communication and any attachments are CONFIDENTIAL and may
be protected by one or more legal privileges. It is intended
solely for the use of the addressee identified above. If you
are not the intended recipient, any use, disclosure, copying
or distribution of this communication is UNAUTHORIZED. Neither
this information block, the typed name of the sender, nor
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From jonvoip at gmail.com  Wed May 30 12:01:27 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 30 May 2007 11:01:27 -0500
Subject: [cisco-voip] CallManager 5.2
In-Reply-To: <1180490350.5411.4.camel@lt21223.campus.dmacc.edu>
References: <5d093f9a0705260846k4ffe1c6cra465694ab14dab9c@mail.gmail.com>
	<5d093f9a0705260852i139101d5w7756f41fabac219a@mail.gmail.com>
	<1180490350.5411.4.camel@lt21223.campus.dmacc.edu>
Message-ID: <5d093f9a0705300901q5e3eb76w222be51bb7b592c8@mail.gmail.com>

Well, you are the only one apparently...

It supposedly nukes out Pickup groups...



Jonathan

On 5/29/07, Jeffrey C. Ollie <jeff at ocjtech.us> wrote:
> On Sat, 2007-05-26 at 10:52 -0500, Jonathan Charles wrote:
> > Correction, 5.1.2... not 5.2
> >
> > On 5/26/07, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > So, anyone upgrade yet?
> > >
> > > Just curious if it is going to kill QoS or RTP streams... TAC said it
> > > was just bug fixes...
>
> I upgraded from 5.1.1a to 5.1.2 to fix an issue with CDR processing - no
> new issues seen so far.
>
> Jeff
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From monavy at gmail.com  Wed May 30 12:42:51 2007
From: monavy at gmail.com (Mohammed Naviwala)
Date: Wed, 30 May 2007 20:42:51 +0400
Subject: [cisco-voip] (no subject)
Message-ID: <136208490705300942w70132d22m90783c0a295c6be2@mail.gmail.com>

-- 
Mohammed Abdul Razzaq
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From jeff at ocjtech.us  Wed May 30 12:53:35 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Wed, 30 May 2007 11:53:35 -0500
Subject: [cisco-voip] CallManager 5.2
In-Reply-To: <5d093f9a0705300901q5e3eb76w222be51bb7b592c8@mail.gmail.com>
References: <5d093f9a0705260846k4ffe1c6cra465694ab14dab9c@mail.gmail.com>
	<5d093f9a0705260852i139101d5w7756f41fabac219a@mail.gmail.com>
	<1180490350.5411.4.camel@lt21223.campus.dmacc.edu>
	<5d093f9a0705300901q5e3eb76w222be51bb7b592c8@mail.gmail.com>
Message-ID: <1180544015.3874.8.camel@lt21223.campus.dmacc.edu>

On Wed, 2007-05-30 at 11:01 -0500, Jonathan Charles wrote:
> Well, you are the only one apparently...
> 
> It supposedly nukes out Pickup groups...

Yeah, it looks like it does... We don't make extensive use of pickup
groups so I hadn't noticed before.

Jeff

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From cisco.voip at verizon.net  Wed May 30 13:12:35 2007
From: cisco.voip at verizon.net (cisco.voip)
Date: Wed, 30 May 2007 13:12:35 -0400
Subject: [cisco-voip] ccmuser / speed dials
References: <mailman.5.1180540805.44883.cisco-voip@puck.nether.net>
Message-ID: <000b01c7a2dd$b85d41a0$661b1b0a@cs.myharris.net>

Why are there so many speed dial buttons on the ccmuser page - up to 99?
How do I access them on the phone?

I am setting up the personal fast dials and can only get the entries through
the personal address book?




From asobihoudai at yahoo.com  Wed May 30 13:16:19 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 30 May 2007 10:16:19 -0700 (PDT)
Subject: [cisco-voip] ccmuser / speed dials
In-Reply-To: <000b01c7a2dd$b85d41a0$661b1b0a@cs.myharris.net>
Message-ID: <74441.81063.qm@web57212.mail.re3.yahoo.com>

Speed dials NOT associated with a button are accessed
via the AbbrDial softkey. Fast dials are dialed using
the services key so long as the user has subscribed
and configured the service correctly.

You may enter fast dials without a directory entry.
Just enter a DIRECTORY NUMBER on the right side when
assigning fast dials.

Paul

--- "cisco.voip" <cisco.voip at verizon.net> wrote:

> Why are there so many speed dial buttons on the
> ccmuser page - up to 99?
> How do I access them on the phone?
> 
> I am setting up the personal fast dials and can only
> get the entries through
> the personal address book?



      ____________________________________________________________________________________
Park yourself in front of a world of choices in alternative vehicles. Visit the Yahoo! Auto Green Center.
http://autos.yahoo.com/green_center/ 

From lelio at uoguelph.ca  Wed May 30 13:16:37 2007
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 30 May 2007 13:16:37 -0400
Subject: [cisco-voip] ccmuser / speed dials
References: <mailman.5.1180540805.44883.cisco-voip@puck.nether.net>
	<000b01c7a2dd$b85d41a0$661b1b0a@cs.myharris.net>
Message-ID: <022201c7a2de$482cbf40$47196883@cfs.uoguelph.ca>

abbreviated dialing.

while the handset is on hook, type  the index number of the speed dial and then press abbrdial softkey
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"Buffalo buffalo Buffalo buffalo buffalo buffalo Buffalo buffalo."      WJR

  ----- Original Message ----- 
  From: cisco.voip 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, May 30, 2007 1:12 PM
  Subject: [cisco-voip] ccmuser / speed dials


  Why are there so many speed dial buttons on the ccmuser page - up to 99?
  How do I access them on the phone?

  I am setting up the personal fast dials and can only get the entries through
  the personal address book?



  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From MillerS at DicksteinShapiro.COM  Wed May 30 13:17:53 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Wed, 30 May 2007 13:17:53 -0400
Subject: [cisco-voip] ccmuser / speed dials
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538C0C@DCEX2.DSMO.COM>

You can access and program all the speed dials from the same interface where you program everything else on the phone.  No one uses the 99, but what the heck.

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 
Fax (202)-330-5607
millers at dicksteinshapiro.com 

----- Original Message -----
From: cisco-voip-bounces at puck.nether.net <cisco-voip-bounces at puck.nether.net>
To: cisco-voip at puck.nether.net <cisco-voip at puck.nether.net>
Sent: Wed May 30 13:12:35 2007
Subject: [cisco-voip] ccmuser / speed dials

Why are there so many speed dial buttons on the ccmuser page - up to 99?
How do I access them on the phone?

I am setting up the personal fast dials and can only get the entries through
the personal address book?



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

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From cisco.voip at verizon.net  Wed May 30 13:54:54 2007
From: cisco.voip at verizon.net (cisco.voip)
Date: Wed, 30 May 2007 13:54:54 -0400
Subject: [cisco-voip] ccmuser / speed dials
References: <74441.81063.qm@web57212.mail.re3.yahoo.com>
Message-ID: <001601c7a2e3$a1c3d430$661b1b0a@cs.myharris.net>

In the Personal Address Book, I configure the Fast Dials and the service is
configured correctly.
However, each entry is preceded by a H, W, or M on the fast dial Menu (for
Home, Work or Mobile).
My situation is a local setup, therefore is there any way to have the Fast
Dial entry on the phone NOT show the
H, W or M preceding the name.  All my numbers are work numbers and the users
are complaining.

Thanks

----- Original Message ----- 
From: "Paul Choi" <asobihoudai at yahoo.com>
To: "cisco.voip" <cisco.voip at verizon.net>; <cisco-voip at puck.nether.net>
Sent: Wednesday, May 30, 2007 1:16 PM
Subject: Re: [cisco-voip] ccmuser / speed dials


> Speed dials NOT associated with a button are accessed
> via the AbbrDial softkey. Fast dials are dialed using
> the services key so long as the user has subscribed
> and configured the service correctly.
>
> You may enter fast dials without a directory entry.
> Just enter a DIRECTORY NUMBER on the right side when
> assigning fast dials.
>
> Paul
>
> --- "cisco.voip" <cisco.voip at verizon.net> wrote:
>
> > Why are there so many speed dial buttons on the
> > ccmuser page - up to 99?
> > How do I access them on the phone?
> >
> > I am setting up the personal fast dials and can only
> > get the entries through
> > the personal address book?
>
>
>
>
____________________________________________________________________________
________
> Park yourself in front of a world of choices in alternative vehicles.
Visit the Yahoo! Auto Green Center.
> http://autos.yahoo.com/green_center/
>



From asobihoudai at yahoo.com  Wed May 30 13:57:16 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 30 May 2007 10:57:16 -0700 (PDT)
Subject: [cisco-voip] ccmuser / speed dials
In-Reply-To: <001601c7a2e3$a1c3d430$661b1b0a@cs.myharris.net>
Message-ID: <935033.21992.qm@web57213.mail.re3.yahoo.com>

Enter the phone number without using the personal
address book. It won't have a label but it won't have
a letter next to it either.

If the users don't like either way, tell them to stuff
it.

--- "cisco.voip" <cisco.voip at verizon.net> wrote:

> In the Personal Address Book, I configure the Fast
> Dials and the service is
> configured correctly.
> However, each entry is preceded by a H, W, or M on
> the fast dial Menu (for
> Home, Work or Mobile).
> My situation is a local setup, therefore is there
> any way to have the Fast
> Dial entry on the phone NOT show the
> H, W or M preceding the name.  All my numbers are
> work numbers and the users
> are complaining.
> 
> Thanks
> 
> ----- Original Message ----- 
> From: "Paul Choi" <asobihoudai at yahoo.com>
> To: "cisco.voip" <cisco.voip at verizon.net>;
> <cisco-voip at puck.nether.net>
> Sent: Wednesday, May 30, 2007 1:16 PM
> Subject: Re: [cisco-voip] ccmuser / speed dials
> 
> 
> > Speed dials NOT associated with a button are
> accessed
> > via the AbbrDial softkey. Fast dials are dialed
> using
> > the services key so long as the user has
> subscribed
> > and configured the service correctly.
> >
> > You may enter fast dials without a directory
> entry.
> > Just enter a DIRECTORY NUMBER on the right side
> when
> > assigning fast dials.
> >
> > Paul
> >
> > --- "cisco.voip" <cisco.voip at verizon.net> wrote:
> >
> > > Why are there so many speed dial buttons on the
> > > ccmuser page - up to 99?
> > > How do I access them on the phone?
> > >
> > > I am setting up the personal fast dials and can
> only
> > > get the entries through
> > > the personal address book?
> >
> >
> >
> >
>
____________________________________________________________________________
> ________
> > Park yourself in front of a world of choices in
> alternative vehicles.
> Visit the Yahoo! Auto Green Center.
> > http://autos.yahoo.com/green_center/
> >
> 
> 
> 



       
____________________________________________________________________________________Get the Yahoo! toolbar and be alerted to new email wherever you're surfing.
http://new.toolbar.yahoo.com/toolbar/features/mail/index.php

From dpriebe at theriverbank.com  Wed May 30 14:05:49 2007
From: dpriebe at theriverbank.com (Duane Priebe)
Date: Wed, 30 May 2007 13:05:49 -0500
Subject: [cisco-voip] Cisco IP Phone Ringers...
Message-ID: <43D799CBE4331D44999AB7C9CD67F302033B99AE@NTEXCHANGE.riverbank.com>

Can anyone provide me with a link as to where I can download additional
ringers for 7941, 7961, & 7970 IP phones.  Also if I were to create my
own ringers is anyone aware of the audio format the file needs to be in
order for the system to accept and play them back correctly?  Thanks for
your help!

Duane
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From Scott.Voll at wesd.org  Wed May 30 14:10:45 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 30 May 2007 11:10:45 -0700
Subject: [cisco-voip] Cisco IP Phone Ringers...
In-Reply-To: <43D799CBE4331D44999AB7C9CD67F302033B99AE@NTEXCHANGE.riverbank.com>
References: <43D799CBE4331D44999AB7C9CD67F302033B99AE@NTEXCHANGE.riverbank.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE81B@Apollo.wesd.org>

http://www.ccmstuff.net/

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Duane Priebe
Sent: Wednesday, May 30, 2007 11:06 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco IP Phone Ringers...

 

Can anyone provide me with a link as to where I can download additional
ringers for 7941, 7961, & 7970 IP phones.  Also if I were to create my
own ringers is anyone aware of the audio format the file needs to be in
order for the system to accept and play them back correctly?  Thanks for
your help!

Duane 

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From Roconnell at unislumin.com  Wed May 30 15:40:27 2007
From: Roconnell at unislumin.com (Ryan O'Connell)
Date: Wed, 30 May 2007 15:40:27 -0400
Subject: [cisco-voip] Unified messaging Domino in French
In-Reply-To: <18082B35CF0EBD4C8D265874508A615C05D63BE3@hermes.unislumin.com>
References: <18082B35CF0EBD4C8D265874508A615C05D63BE3@hermes.unislumin.com>
Message-ID: <18082B35CF0EBD4C8D265874508A615C05F4F0F5@hermes.unislumin.com>

Hello,

 

We are in the process of added a French site to a functioning Unified
Messaging Dominion Environment. We have all the languages functioning
correctly but we still have 2 outstanding issues to resolve. In the
attached Diagram the top portion shows what users are seeing in the
"Subject" heading of their inbox. They would like to have this in
French. The lower part of the diagram is the other issue, where the
media client itself appears to be English. Is there a French version
from IBM or Cisco that can be downloaded?

 

Ryan


 





Attention:

Privileged/Confidential Information may be contained in this message.  Disclosure to any person other than the named recipient is unauthorized.  If you are not the intended recipient, please delete all copies of this information and kindly notify the sender by reply email.  Opinions, conclusions and other information in this message that do not relate to the official business of UNIS LUMIN Inc. shall be understood as neither given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries reserve the right to monitor all e-mail communications through its networks.  Thank you.
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From duncanw at otc.edu  Wed May 30 16:03:17 2007
From: duncanw at otc.edu (DUNCAN, W B.)
Date: Wed, 30 May 2007 15:03:17 -0500
Subject: [cisco-voip] PoE Phone Problem
Message-ID: <06D1B6D4926222458F803D0D3EDCCB7E0201D391@EXM1.otc.edu>

I have had two phones that wont power up with PoE.  I can plug these
into a power brick and they work fine.  Has anyone experienced this and
more so is there a way to repair the phone?

 

Thanks,

W. Brian Duncan

Coordinator of Telecommunications

Ozarks Technical Community College

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From asobihoudai at yahoo.com  Wed May 30 16:53:57 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 30 May 2007 13:53:57 -0700 (PDT)
Subject: [cisco-voip] PoE Phone Problem
In-Reply-To: <06D1B6D4926222458F803D0D3EDCCB7E0201D391@EXM1.otc.edu>
Message-ID: <922858.39928.qm@web57206.mail.re3.yahoo.com>

What fixes have you attempted so far?


--- "DUNCAN, W B." <duncanw at otc.edu> wrote:

> I have had two phones that wont power up with PoE. 
> I can plug these
> into a power brick and they work fine.  Has anyone
> experienced this and
> more so is there a way to repair the phone?
> 
>  
> 
> Thanks,
> 
> W. Brian Duncan
> 
> Coordinator of Telecommunications
> 
> Ozarks Technical Community College
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
____________________________________________________________________________________Yahoo! oneSearch: Finally, mobile search 
that gives answers, not web links. 
http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC

From jonvoip at gmail.com  Wed May 30 16:57:01 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 30 May 2007 15:57:01 -0500
Subject: [cisco-voip] ISDN PRI - Message not compatible with call state...
Message-ID: <5d093f9a0705301357o46e084cbx9f1171d38d47d5d8@mail.gmail.com>

Getting a message on PRI debugs, just wondering if it is something to
be concerned about:

      Cause i = 0x80E56E - Message not compatible with call state

Cisco's website says this for this code:

The remote equipment receives an expected message that does not
correspond to the current state of the connection.

This issue occurs due to a D-channel error.


Is this bad???

the call connects...



Jonathan

From Matt.Slaga at us.didata.com  Wed May 30 17:03:25 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 30 May 2007 17:03:25 -0400
Subject: [cisco-voip] PoE Phone Problem
In-Reply-To: <06D1B6D4926222458F803D0D3EDCCB7E0201D391@EXM1.otc.edu>
References: <06D1B6D4926222458F803D0D3EDCCB7E0201D391@EXM1.otc.edu>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B58515@USNAEXCH.na.didata.local>

What are the phone models?

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of DUNCAN, W B.
Sent: Wednesday, May 30, 2007 4:03 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PoE Phone Problem

 

I have had two phones that wont power up with PoE.  I can plug these
into a power brick and they work fine.  Has anyone experienced this and
more so is there a way to repair the phone?

 

Thanks,

W. Brian Duncan

Coordinator of Telecommunications

Ozarks Technical Community College




-----------------------------------------
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From newlook110 at yahoo.com  Wed May 30 17:47:19 2007
From: newlook110 at yahoo.com (new look)
Date: Wed, 30 May 2007 14:47:19 -0700 (PDT)
Subject: [cisco-voip] E1 Problem
Message-ID: <897115.29512.qm@web61321.mail.yahoo.com>

Dear All,
I'm appling a new Cisco IPTelephony system my voice geatway is Cisco router and I'm using E1 Interface 
the problem is the destance between my company and the Telecom exchange is about 5Km so the max bandwidth I have is 512K
So I need to know can I use this channels only?
what is the router configuration ?
By the way I'm using paradyne two units 7996 (DSL/G703)
Thanks


       
____________________________________________________________________________________Ready for the edge of your seat? 
Check out tonight's top picks on Yahoo! TV. 
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From jonvoip at gmail.com  Wed May 30 17:53:08 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 30 May 2007 16:53:08 -0500
Subject: [cisco-voip] E1 Problem
In-Reply-To: <897115.29512.qm@web61321.mail.yahoo.com>
References: <897115.29512.qm@web61321.mail.yahoo.com>
Message-ID: <5d093f9a0705301453y3ff0c3cdh7902210cdb6c4539@mail.gmail.com>

E1 CAS or E1 PRI?

If E1 CAS, it shouldn't be an issue, just allocate just those
timeslots... (maybe).

If E1 PRI, then no, cuz you need channels 15 and 30 as D-channels, and
I don't see how this could work.



Jonathan

On 5/30/07, new look <newlook110 at yahoo.com> wrote:
>
>
>
> Dear All,
> I'm appling a new Cisco IPTelephony system my voice geatway is Cisco router
> and I'm using E1 Interface
> the problem is the destance between my company and the Telecom exchange is
> about 5Km so the max bandwidth I have is 512K
> So I need to know can I use this channels only?
> what is the router configuration ?
> By the way I'm using paradyne two units 7996 (DSL/G703)
> Thanks
>
>  ________________________________
> Be a better Globetrotter. Get better travel answers from someone who knows.
> Yahoo! Answers - Check it out.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From erickbe at yahoo.com  Wed May 30 17:53:29 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 30 May 2007 14:53:29 -0700 (PDT)
Subject: [cisco-voip] Unity Connections IMAP port number for outbound
	connections
Message-ID: <564227.56146.qm@web37213.mail.mud.yahoo.com>

Hi,

I've searched cisco for this, but not able to find it yet.  Am needing to know what ports to open on firewall so users with blackberrys, etc can get their Unity voicemail using IMAP from outside.  When they are on the internal network it works fine. 

Thanks, Erick



      ____________________________________________________________________________________Shape Yahoo! in your own image.  Join our Network Research Panel today!   http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 



From jonvoip at gmail.com  Wed May 30 17:55:31 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 30 May 2007 16:55:31 -0500
Subject: [cisco-voip] PoE Phone Problem
In-Reply-To: <06D1B6D4926222458F803D0D3EDCCB7E0201D391@EXM1.otc.edu>
References: <06D1B6D4926222458F803D0D3EDCCB7E0201D391@EXM1.otc.edu>
Message-ID: <5d093f9a0705301455m7d91b53sdf7f1dbcf38c6ae6@mail.gmail.com>

I am presuming you have tried placing these phones in the locations of
existing, working phones to ensure it is not a cabling issue. If not,
then do so. We need to find out if it is a phone or cabling issue.

If they do not power up at the other position, then it is a dead phone, RMA it.

If they do, then we have a cabling issue of some sort. Is it possible
we have a non-POE switch between there and the MDF?

We could also have a crushed cable, a really long run, etc... If it is
just a power issue (good ethernet, bad power), then plug it in there
and use a power brick to power up the phone...



Jonathan

On 5/30/07, DUNCAN, W B. <duncanw at otc.edu> wrote:
>
>
>
>
> I have had two phones that wont power up with PoE.  I can plug these into a
> power brick and they work fine.  Has anyone experienced this and more so is
> there a way to repair the phone?
>
>
>
> Thanks,
>
> W. Brian Duncan
>
> Coordinator of Telecommunications
>
> Ozarks Technical Community College
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From jonvoip at gmail.com  Wed May 30 18:00:10 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 30 May 2007 17:00:10 -0500
Subject: [cisco-voip] Unity 4.2(1) and MS Exchange 2007
In-Reply-To: <AB0A35F2486F81468971AA68558BE7190658C9AE@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE7190658C9AE@smieln01winm01.micoanet.com>
Message-ID: <5d093f9a0705301500o7f363bd5p87f333d2258baed5@mail.gmail.com>

Dunno, but I have set up Exchange 2007 UM with CCM 5.1 via SIP (No
Unity)... and it works pretty well, the Outlook Voice Access is pretty
nifty...

The only downside is no auto-attendant... yet.



Jonathan

On 5/30/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
>
> Maybe this was discussed here before, but I was wondering if anyone knew if
> the most current version of Unity 4.2(1) is compatible with MS Exchange
> 2007, either officially or unofficially?  I saw some information on Unity
> 5.0 for Exchange, but I wasn't sure of its availability as of yet.  Our
> company is pushing to do a upgrade within the next few months and I couldn't
> find anything on Cisco's site that spoke of Exchange 2007 compatibility with
> older versions.  Additionally, I didn't see any information on which
> CallManagers Unity 5.0 for Exchange was compatible with (we are still on
> 4.1(3)sr5).  Can anyone provide any input on this?
>
>
>
> Thanks!
>
>
>
> Matthew M. Linsemier, CCSP/CCNP/CCDA
>
> Senior Network Engineer
>
> American Physicians
>
> Voice: 517.324.6695
>
> Fax: 517.333.2812
>
> E-Mail: mlinsemier at apcapital.com
>
>
>
>  ________________________________
>
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be protected
> by one or more legal privileges. It is intended solely for the use of the
> addressee identified above. If you are not the intended recipient, any use,
> disclosure, copying or distribution of this communication is UNAUTHORIZED.
> Neither this information block, the typed name of the sender, nor anything
> else in this message is intended to constitute an electronic signature
> unless a specific statement to the contrary is included in this message. If
> you have received this communication in error, please immediately contact me
> and delete this communication from your computer. Thank you.
>
>  ________________________________
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From PRuttman at foley.com  Wed May 30 18:01:57 2007
From: PRuttman at foley.com (Ruttman, Pete G.)
Date: Wed, 30 May 2007 17:01:57 -0500
Subject: [cisco-voip] Multiple callcenters on IPCC4.0(4)
Message-ID: <5DB17A1560B7EE4186747DF43A6D4C6C6EC2A6@WIMKEEX07.foleylaw.com>

I was wondering if it was possible to house two distinct callcenters on
a single IPCC 4.0(4) solution.  We would like to create a design that
would have each call center segregated as much as possible.  We already
are segregating the centers with different apps, scripts and queues as
well as skills.  The area where we are still seeing everything overlap
is with the recordings.  How do we make sure that callcenter 1's
supervisor doesn't see the recordings for callcenter 2's agents.  Also,
from the server perspective all of our recordings end up C:\Program
Files\Cisco\Desktop_Audio and we would want to separate those so it
hides them from the call center they do not belong to.

Has anyone accomplished this?

Pete Ruttman
Foley & Lardner, LLP

The preceding email message may be confidential or protected by the attorney-client privilege. It is not intended for transmission to, or receipt by, any unauthorized persons.  If you have received this message in error, please (i) do not read it, (ii) reply to the sender that you received the message in error, and (iii) erase or destroy the message.  Legal advice contained in the preceding message is solely for the benefit of the Foley & Lardner LLP client(s) represented by the Firm in the particular matter that is the subject of this message, and may not be relied upon by any other party.      

  
Internal Revenue Service regulations require that certain types of written advice include a disclaimer. To the extent the preceding message contains advice relating to a Federal tax issue, unless expressly stated otherwise the advice is not intended or written to be used, and it cannot be used by the recipient or any other taxpayer, for the purpose of avoiding Federal tax penalties, and was not written to support the promotion or marketing of any transaction or matter discussed herein.

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From Scott.Voll at wesd.org  Wed May 30 18:04:36 2007
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 30 May 2007 15:04:36 -0700
Subject: [cisco-voip] Unity 4.2(1) and MS Exchange 2007
In-Reply-To: <5d093f9a0705301500o7f363bd5p87f333d2258baed5@mail.gmail.com>
References: <AB0A35F2486F81468971AA68558BE7190658C9AE@smieln01winm01.micoanet.com>
	<5d093f9a0705301500o7f363bd5p87f333d2258baed5@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A608DEE820@Apollo.wesd.org>

If you have both Unity 5.0 and Exchange 2007 could you use both for VM?
Be able to use both interfaces to work with your VM.  Since Exchange is
SIP and Unity is TSP could you set it up to call one number for the
Exchange interface to use the Voice Access but use the normal unity way
to do it the normal way?

I'm just thinking of those of us that are going to upgrade unity and
exchange soon.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Wednesday, May 30, 2007 3:00 PM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unity 4.2(1) and MS Exchange 2007

Dunno, but I have set up Exchange 2007 UM with CCM 5.1 via SIP (No
Unity)... and it works pretty well, the Outlook Voice Access is pretty
nifty...

The only downside is no auto-attendant... yet.



Jonathan

On 5/30/07, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
>
> Maybe this was discussed here before, but I was wondering if anyone
knew if
> the most current version of Unity 4.2(1) is compatible with MS
Exchange
> 2007, either officially or unofficially?  I saw some information on
Unity
> 5.0 for Exchange, but I wasn't sure of its availability as of yet.
Our
> company is pushing to do a upgrade within the next few months and I
couldn't
> find anything on Cisco's site that spoke of Exchange 2007
compatibility with
> older versions.  Additionally, I didn't see any information on which
> CallManagers Unity 5.0 for Exchange was compatible with (we are still
on
> 4.1(3)sr5).  Can anyone provide any input on this?
>
>
>
> Thanks!
>
>
>
> Matthew M. Linsemier, CCSP/CCNP/CCDA
>
> Senior Network Engineer
>
> American Physicians
>
> Voice: 517.324.6695
>
> Fax: 517.333.2812
>
> E-Mail: mlinsemier at apcapital.com
>
>
>
>  ________________________________
>
>
> CONFIDENTIALITY STATEMENT
> This communication and any attachments are CONFIDENTIAL and may be
protected
> by one or more legal privileges. It is intended solely for the use of
the
> addressee identified above. If you are not the intended recipient, any
use,
> disclosure, copying or distribution of this communication is
UNAUTHORIZED.
> Neither this information block, the typed name of the sender, nor
anything
> else in this message is intended to constitute an electronic signature
> unless a specific statement to the contrary is included in this
message. If
> you have received this communication in error, please immediately
contact me
> and delete this communication from your computer. Thank you.
>
>  ________________________________
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From peter.perreault at thomson.com  Wed May 30 18:04:24 2007
From: peter.perreault at thomson.com (peter.perreault at thomson.com)
Date: Wed, 30 May 2007 18:04:24 -0400
Subject: [cisco-voip] PoE Phone Problem
In-Reply-To: <5d093f9a0705301455m7d91b53sdf7f1dbcf38c6ae6@mail.gmail.com>
Message-ID: <0D69AD1D5EBD87479C49A58990E8FC549C9348@tfusnjpscmbx03.ERF.THOMSON.COM>

It could also be an issue with the port.  (Sorry if you've already ruled
this out)  I've experienced switches turning POE off on individual ports
once a non POE device was connected to the port.  You have to manually
enable POE on these ports to bring it back up.  There's a show command
that will indicate if POE is on or off on switch ports.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Wednesday, May 30, 2007 5:56 PM
To: DUNCAN, W B.
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PoE Phone Problem

I am presuming you have tried placing these phones in the locations of
existing, working phones to ensure it is not a cabling issue. If not,
then do so. We need to find out if it is a phone or cabling issue.

If they do not power up at the other position, then it is a dead phone,
RMA it.

If they do, then we have a cabling issue of some sort. Is it possible
we have a non-POE switch between there and the MDF?

We could also have a crushed cable, a really long run, etc... If it is
just a power issue (good ethernet, bad power), then plug it in there
and use a power brick to power up the phone...



Jonathan

On 5/30/07, DUNCAN, W B. <duncanw at otc.edu> wrote:
>
>
>
>
> I have had two phones that wont power up with PoE.  I can plug these
into a
> power brick and they work fine.  Has anyone experienced this and more
so is
> there a way to repair the phone?
>
>
>
> Thanks,
>
> W. Brian Duncan
>
> Coordinator of Telecommunications
>
> Ozarks Technical Community College
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From keli at carocomp.ro  Wed May 30 18:45:19 2007
From: keli at carocomp.ro (Kelemen Zoltan)
Date: Thu, 31 May 2007 01:45:19 +0300
Subject: [cisco-voip] Unity Connections IMAP port number for outbound
 connections
In-Reply-To: <564227.56146.qm@web37213.mail.mud.yahoo.com>
References: <564227.56146.qm@web37213.mail.mud.yahoo.com>
Message-ID: <465DFE7F.1060204@carocomp.ro>

IMAP standard port is tcp/143, imap over ssl (imaps) is 993/tcp. I doubt 
  Cisco is using it any differently.

Regards,
   Zoltan

Erick Bergquist wrote:
> Hi,
> 
> I've searched cisco for this, but not able to find it yet.  Am needing to know what ports to open on firewall so users with blackberrys, etc can get their Unity voicemail using IMAP from outside.  When they are on the internal network it works fine. 
> 
> Thanks, Erick
> 
> 
> 
>       ____________________________________________________________________________________Shape Yahoo! in your own image.  Join our Network Research Panel today!   http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From rratliff at cisco.com  Wed May 30 19:39:40 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 30 May 2007 19:39:40 -0400
Subject: [cisco-voip] ISDN PRI - Message not compatible with call
	state...
In-Reply-To: <5d093f9a0705301357o46e084cbx9f1171d38d47d5d8@mail.gmail.com>
References: <5d093f9a0705301357o46e084cbx9f1171d38d47d5d8@mail.gmail.com>
Message-ID: <68C56A16-CBAC-4470-BF17-30F4F31099CF@cisco.com>

Depending on the switch the message will either be ignored or may  
cause the call to drop.  I suspect this has a great deal to do with  
what state the call was in when the erroneous message was received.

It could signal a PRI protocol-type mismatch.

-Ryan

On May 30, 2007, at 4:57 PM, Jonathan Charles wrote:

Getting a message on PRI debugs, just wondering if it is something to
be concerned about:

       Cause i = 0x80E56E - Message not compatible with call state

Cisco's website says this for this code:

The remote equipment receives an expected message that does not
correspond to the current state of the connection.

This issue occurs due to a D-channel error.


Is this bad???

the call connects...



Jonathan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
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From johnstr at sunysuffolk.edu  Wed May 30 18:21:58 2007
From: johnstr at sunysuffolk.edu (Rich Johnston)
Date: Wed, 30 May 2007 18:21:58 -0400
Subject: [cisco-voip] PoE Phone Problem
In-Reply-To: <0D69AD1D5EBD87479C49A58990E8FC549C9348@tfusnjpscmbx03.ERF.THOMSON.COM>
Message-ID: <12F145949C0D8F45910F79FF22CB30DC06C6890E@central5.sccc.a>


Also, it you should review the number of ports with active PoE. You
might have exceeded the switches power supply capability. Turn off PoE
on ports that don't require power.

Richard Johnston, RCDD
Director. Of Network & Telecomm.
Suffolk County Community College 
533 College Road, Selden, NY  11784 
Tel:631-451-4190 Fax:631-451-4970
Email: johnstr at sunysuffolk.edu


This e-mail is intended solely for the person or entity to which it is
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Any review, dissemination, copying, printing or other use of this e-mail
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-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
peter.perreault at thomson.com
Sent: Wednesday, May 30, 2007 6:04 PM
To: jonvoip at gmail.com; duncanw at otc.edu
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PoE Phone Problem

It could also be an issue with the port.  (Sorry if you've already ruled
this out)  I've experienced switches turning POE off on individual ports
once a non POE device was connected to the port.  You have to manually
enable POE on these ports to bring it back up.  There's a show command
that will indicate if POE is on or off on switch ports.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Wednesday, May 30, 2007 5:56 PM
To: DUNCAN, W B.
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PoE Phone Problem

I am presuming you have tried placing these phones in the locations of
existing, working phones to ensure it is not a cabling issue. If not,
then do so. We need to find out if it is a phone or cabling issue.

If they do not power up at the other position, then it is a dead phone,
RMA it.

If they do, then we have a cabling issue of some sort. Is it possible
we have a non-POE switch between there and the MDF?

We could also have a crushed cable, a really long run, etc... If it is
just a power issue (good ethernet, bad power), then plug it in there
and use a power brick to power up the phone...



Jonathan

On 5/30/07, DUNCAN, W B. <duncanw at otc.edu> wrote:
>
>
>
>
> I have had two phones that wont power up with PoE.  I can plug these
into a
> power brick and they work fine.  Has anyone experienced this and more
so is
> there a way to repair the phone?
>
>
>
> Thanks,
>
> W. Brian Duncan
>
> Coordinator of Telecommunications
>
> Ozarks Technical Community College
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
_______________________________________________
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From jsteinberg at gmail.com  Thu May 31 00:01:11 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 30 May 2007 23:01:11 -0500
Subject: [cisco-voip] CallManager 5.2
In-Reply-To: <1180544015.3874.8.camel@lt21223.campus.dmacc.edu>
References: <5d093f9a0705260846k4ffe1c6cra465694ab14dab9c@mail.gmail.com>
	<5d093f9a0705260852i139101d5w7756f41fabac219a@mail.gmail.com>
	<1180490350.5411.4.camel@lt21223.campus.dmacc.edu>
	<5d093f9a0705300901q5e3eb76w222be51bb7b592c8@mail.gmail.com>
	<1180544015.3874.8.camel@lt21223.campus.dmacc.edu>
Message-ID: <f1b65f880705302101r24b2d30eya62d2cc63ae22b7b@mail.gmail.com>

Is the pickup group bug a comestic bug in the ccmadmin web UI or does
it actually break the pickup group functionality?


On 5/30/07, Jeffrey C. Ollie <jeff at ocjtech.us> wrote:
> On Wed, 2007-05-30 at 11:01 -0500, Jonathan Charles wrote:
> > Well, you are the only one apparently...
> >
> > It supposedly nukes out Pickup groups...
>
> Yeah, it looks like it does... We don't make extensive use of pickup
> groups so I hadn't noticed before.
>
> Jeff
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From prashant.k at tvisha.com  Thu May 31 01:44:00 2007
From: prashant.k at tvisha.com (prashant kirpane)
Date: Thu, 31 May 2007 11:14:00 +0530
Subject: [cisco-voip] How to Downgrade Cisco CallManager 4.1 to 4.0 ?
Message-ID: <200705310544.l4V5iBE3051332@puck.nether.net>

Hi all, 

 

I m running a Cisco CallManager 4.1(2) as a Publisher, which is upgraded
from version 4.0. I tried to install a Subscriber callmanager with version
4.0, but its showing an error that it couldn't identify Publisher server
version ! 

Shall I downgrade the Publisher server to 4.0? Is it possible?

 

Thanks and Regards,

PRASHANT

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From asobihoudai at yahoo.com  Thu May 31 01:59:36 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 30 May 2007 22:59:36 -0700 (PDT)
Subject: [cisco-voip] How to Downgrade Cisco CallManager 4.1 to 4.0 ?
In-Reply-To: <200705310544.l4V5iBE3051332@puck.nether.net>
Message-ID: <599775.29210.qm@web57205.mail.re3.yahoo.com>

I'm very confused.

1 - you upgraded the publisher to 4.1(2)
2 - you installed a subscriber with 4.0
3 - subscriber cannot talk to publisher running 4.1(2)

What's the problem again?

I'm fairly positive that you're trying to do something
that's not supported and impossible *if* you're trying
to get a 4.0 subscriber to talk to a 4.1(2) publisher.

Paul


--- prashant kirpane <prashant.k at tvisha.com> wrote:

> Hi all, 
> 
>  
> 
> I m running a Cisco CallManager 4.1(2) as a
> Publisher, which is upgraded
> from version 4.0. I tried to install a Subscriber
> callmanager with version
> 4.0, but its showing an error that it couldn't
> identify Publisher server
> version ! 
> 
> Shall I downgrade the Publisher server to 4.0? Is it
> possible?
> 
>  
> 
> Thanks and Regards,
> 
> PRASHANT
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
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From smarwah at enetsolutions.net  Thu May 31 02:07:14 2007
From: smarwah at enetsolutions.net (Sumit Marwah)
Date: Thu, 31 May 2007 01:07:14 -0500
Subject: [cisco-voip] How to Downgrade Cisco CallManager 4.1 to 4.0 ?
In-Reply-To: <599775.29210.qm@web57205.mail.re3.yahoo.com>
References: <200705310544.l4V5iBE3051332@puck.nether.net>
	<599775.29210.qm@web57205.mail.re3.yahoo.com>
Message-ID: <FF7F71539DC4E544BA5737C7AA6CEE1747706D@ems05.ems-1.net>

Hi Prashant,

If you are installing a brand new Subscriber you would need the same as
Publisher 4.1(2) version CD set to complete the installation.

I don't know the process of downgrading a Call Manager version, you will
have to contact TAC for that...But your best bet is to get the same
version CDs for installing a new subscriber.

If you are upgrading the subscriber from 4.0 to 4.1(2) and it's giving
error message then it could have something to do with the sql
replication. Try using DBHelper.

Regards,

Sumit Marwah 
Sr. System Engineer
CCIE Voice # 17400
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Thursday, May 31, 2007 1:00 AM
To: prashant kirpane; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] How to Downgrade Cisco CallManager 4.1 to 4.0
?

I'm very confused.

1 - you upgraded the publisher to 4.1(2)
2 - you installed a subscriber with 4.0
3 - subscriber cannot talk to publisher running 4.1(2)

What's the problem again?

I'm fairly positive that you're trying to do something
that's not supported and impossible *if* you're trying
to get a 4.0 subscriber to talk to a 4.1(2) publisher.

Paul


--- prashant kirpane <prashant.k at tvisha.com> wrote:

> Hi all, 
> 
>  
> 
> I m running a Cisco CallManager 4.1(2) as a
> Publisher, which is upgraded
> from version 4.0. I tried to install a Subscriber
> callmanager with version
> 4.0, but its showing an error that it couldn't
> identify Publisher server
> version ! 
> 
> Shall I downgrade the Publisher server to 4.0? Is it
> possible?
> 
>  
> 
> Thanks and Regards,
> 
> PRASHANT
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



       
________________________________________________________________________
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From asobihoudai at yahoo.com  Thu May 31 02:37:53 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Wed, 30 May 2007 23:37:53 -0700 (PDT)
Subject: [cisco-voip] Called Number=(TON=Unknown, NPI=Unknown)
In-Reply-To: <FF7F71539DC4E544BA5737C7AA6CEE1747706D@ems05.ems-1.net>
Message-ID: <896412.40704.qm@web57205.mail.re3.yahoo.com>

I have a very odd problem on a 2600 gateway running
12.4(7d). With TAC's assistance to fix ANOTHER very
odd problem where the gateway would interpret "*" as
44 or "D" thereby not letting users sign into Unity, I
upgraded the IOS on this gateway. 

Now I have another problem...I have a 7940 with 4000
on one line and 4003 on another. 

When I dial the 4000 number, the call goes through
fine. When I call 4003, the gateway's dest= field
under debug voice ccapi inout is blank and I get a
fast busy.

dialing 4003...
   Interface=0x8489B314, Call Info(
   Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type
Str=RegularLine, FinalDestinationFlag=FALSE,
   Incoming Dial-peer=100, Progress
Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE
Present=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route
Label=, CLID Transparent=FALSE), Call Id=-1


dialing 4000...
   Interface=0x846A8394, Interface Type=1,
Destination=, Mode=0x0,
   Call Params(Calling Number=(TON=Unknown,
NPI=Unknown, Screening=Not Screened,
Presentation=Allowed),
   Called Number=9964000(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE,
   Subscriber Type Str=RegularLine,
FinalDestinationFlag=FALSE, Outgoing Dial-peer=1, Call
Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route
Label=, tg_label_flag=0, Application Call Id=)

Have any of you encountered this very odd issue?

All other phones at this office are functioning
properly and *internal* calls from within the office
can dial 4003 without issue. It's calls external to
the cluster coming from the PSTN that don't work at
all.

Paul


       
____________________________________________________________________________________Ready for the edge of your seat? 
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From dhooper at emerge.net.au  Thu May 31 02:22:19 2007
From: dhooper at emerge.net.au (Daniel Hooper)
Date: Thu, 31 May 2007 14:22:19 +0800
Subject: [cisco-voip] Cisco video conferencing
Message-ID: <AF85579EB84D4F45A0CBB63DC6747F0202614C@md.team.emerge.net.au>

Hi,

 

I'm interested in deploying a video conferencing solution between 4 offices with approximately 10 users at each.

 

Reading through the Cisco sales guff, I've figured out I have no idea which product I'm actually looking for, it looks as though things have changed names recently and I'm hoping to get away with an ?xpress' setup rather then the full blown call manager + servers setup.

 

Currently the network is serviced by a mixture of 2811s & 2821s routers, Cat 3550 switches & 7940 / 7960 handsets, the phones are all hanging off an asterisk installation .. Bandwidth between offices and the network infrastructure isn't really an issue.

 

Does Cisco offer a SMB product to enable video conferencing between users? 

Do I need a server or appliance at each site to cater for the video mixing / transcoding?

Can webcams on PC's be utilized?

 

Any information or a kick in the right direction is appreciated.

 

-Daniel Hooper

 

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From asobihoudai at yahoo.com  Thu May 31 03:07:55 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 31 May 2007 00:07:55 -0700 (PDT)
Subject: [cisco-voip] Cisco video conferencing
In-Reply-To: <AF85579EB84D4F45A0CBB63DC6747F0202614C@md.team.emerge.net.au>
Message-ID: <628772.38834.qm@web57212.mail.re3.yahoo.com>

Hi Daniel,

If VT Advantage is good enough for you, you're in luck
as CCME 4.0 supports it.

Paul


--- Daniel Hooper <dhooper at emerge.net.au> wrote:

> Hi,
> 
>  
> 
> I'm interested in deploying a video conferencing
> solution between 4 offices with approximately 10
> users at each.
> 
>  
> 
> Reading through the Cisco sales guff, I've figured
> out I have no idea which product I'm actually
> looking for, it looks as though things have changed
> names recently and I'm hoping to get away with an
> ?xpress' setup rather then the full blown call
> manager + servers setup.
> 
>  
> 
> Currently the network is serviced by a mixture of
> 2811s & 2821s routers, Cat 3550 switches & 7940 /
> 7960 handsets, the phones are all hanging off an
> asterisk installation .. Bandwidth between offices
> and the network infrastructure isn't really an
> issue.
> 
>  
> 
> Does Cisco offer a SMB product to enable video
> conferencing between users? 
> 
> Do I need a server or appliance at each site to
> cater for the video mixing / transcoding?
> 
> Can webcams on PC's be utilized?
> 
>  
> 
> Any information or a kick in the right direction is
> appreciated.
> 



 
____________________________________________________________________________________
Food fight? Enjoy some healthy debate 
in the Yahoo! Answers Food & Drink Q&A.
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From smarwah at enetsolutions.net  Thu May 31 03:12:21 2007
From: smarwah at enetsolutions.net (Sumit Marwah)
Date: Thu, 31 May 2007 02:12:21 -0500
Subject: [cisco-voip] Called Number=(TON=Unknown, NPI=Unknown)
In-Reply-To: <896412.40704.qm@web57205.mail.re3.yahoo.com>
References: <FF7F71539DC4E544BA5737C7AA6CEE1747706D@ems05.ems-1.net>
	<896412.40704.qm@web57205.mail.re3.yahoo.com>
Message-ID: <FF7F71539DC4E544BA5737C7AA6CEE1747706E@ems05.ems-1.net>

Hi Paul,

Can you send the complete debug for voice ccapi inout and if you can add
debug isdn q931 log that would be great? Also, adding the configuration
for the gateway would be helpful.

What version was suggested by TAC for the issue you were running into
before?

Thanks,

Sumit Marwah 
Sr. System Engineer
CCIE Voice # 17400
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Paul Choi
Sent: Thursday, May 31, 2007 1:38 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Called Number=(TON=Unknown, NPI=Unknown)

I have a very odd problem on a 2600 gateway running
12.4(7d). With TAC's assistance to fix ANOTHER very
odd problem where the gateway would interpret "*" as
44 or "D" thereby not letting users sign into Unity, I
upgraded the IOS on this gateway. 

Now I have another problem...I have a 7940 with 4000
on one line and 4003 on another. 

When I dial the 4000 number, the call goes through
fine. When I call 4003, the gateway's dest= field
under debug voice ccapi inout is blank and I get a
fast busy.

dialing 4003...
   Interface=0x8489B314, Call Info(
   Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type
Str=RegularLine, FinalDestinationFlag=FALSE,
   Incoming Dial-peer=100, Progress
Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE
Present=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route
Label=, CLID Transparent=FALSE), Call Id=-1


dialing 4000...
   Interface=0x846A8394, Interface Type=1,
Destination=, Mode=0x0,
   Call Params(Calling Number=(TON=Unknown,
NPI=Unknown, Screening=Not Screened,
Presentation=Allowed),
   Called Number=9964000(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE,
   Subscriber Type Str=RegularLine,
FinalDestinationFlag=FALSE, Outgoing Dial-peer=1, Call
Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route
Label=, tg_label_flag=0, Application Call Id=)

Have any of you encountered this very odd issue?

All other phones at this office are functioning
properly and *internal* calls from within the office
can dial 4003 without issue. It's calls external to
the cluster coming from the PSTN that don't work at
all.

Paul


       
________________________________________________________________________
____________Ready for the edge of your seat? 
Check out tonight's top picks on Yahoo! TV. 
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_______________________________________________
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From tele at plexialab.it  Thu May 31 03:33:12 2007
From: tele at plexialab.it (tele)
Date: Thu, 31 May 2007 09:33:12 +0200
Subject: [cisco-voip] Media Gateway issue with NTE
Message-ID: <1180596792.26166.92.camel@mda>

Hi all,

I have a issue with a Cisco AS5400 and NTE, the problem is simple:

the first CRCX:

CRCX 122138580 S1/DS1-0/8 at mgmi03 MGCP 1.0
C:308
L:a:G729;PCMA;PCMU;telephone-event, fxr/fx:t38-loose,
fmtp:"telephone-event 0-15", p:20
M:recvonly
R:fxr/t38
X:1

contains a fmtp with tone from 0 to 15, when a call goes in progress the
MGC send a MDCX to the MG.

MDCX 122138581 S1/DS1-0/8 at mgmi03 MGCP 1.0
C:308
L:a:PCMA;telephone-event, fxr/fx:t38-loose, fmtp:"telephone-event 0-16",
p:20
I: 17703
M:sendrecv
R:fxr/t38
X:1

v=0
o=- 1180529725 1180529725 IN IP4 myip
s=-
c=IN IP4 myip
t=0 0
m=audio 12388 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

As you can seen MGC want to use tone from 0-16, 16 is the "flash".
Now the mediagateway should be able to change the session but an error
occur during NTE negotiation:

May 30 14:56:02.287 CEST: mgcp_modify_connection: calling
mgcp_select_nte
May 30 14:56:02.287 CEST: S1/DS1-0/8
<VOIP>mgcp_nte_select_rfc2833_dtmf_relay -IN: mode/codec = 5/0
May 30 14:56:02.287 CEST: S1/DS1-0/8
<VOIP>mgcp_nte_select_rfc2833_dtmf_relay -OUT: mode/codec = 5/0
May 30 14:56:02.287 CEST: mgcp_nte_select_payload: No NTE convergence:
local nte=(16)/(0000FFFF), sdp nte=(15)/(0000FFFF)
May 30 14:56:02.287 CEST: mgcp_modify_connection - Conflicting NTE
information

and the MG return to the MGC a 502 error:

May 30 14:56:02.287 CEST: 502 122138581 Cannot select NTE

I've read about the MF package for the support of fmtp, i've tried
actived it "mgcp package-capability mf" but nothing change

Can i do something on the MG to solve this issue? or i've to work on the
MGC side?

thank you very much

:tele


From Matt.Slaga at us.didata.com  Thu May 31 08:40:45 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 31 May 2007 08:40:45 -0400
Subject: [cisco-voip] Cisco video conferencing
In-Reply-To: <AF85579EB84D4F45A0CBB63DC6747F0202614C@md.team.emerge.net.au>
References: <AF85579EB84D4F45A0CBB63DC6747F0202614C@md.team.emerge.net.au>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B85A25@USNAEXCH.na.didata.local>

Cisco does not really have a SMB SIP video conferencing product.  I would suggest looking into the Tandberg 1700MXP or something similar.  The Tandberg unit will allow point to point conference as well as multipoint MCUs built into each unit.  I am not sure of the webcam interop with these units however.

 

You can't use VTA with Asterisk as right now it is only SCCP based.  

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Daniel Hooper
Sent: Thursday, May 31, 2007 2:22 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco video conferencing

 

Hi,

 

I'm interested in deploying a video conferencing solution between 4 offices with approximately 10 users at each.

 

Reading through the Cisco sales guff, I've figured out I have no idea which product I'm actually looking for, it looks as though things have changed names recently and I'm hoping to get away with an ?xpress' setup rather then the full blown call manager + servers setup.

 

Currently the network is serviced by a mixture of 2811s & 2821s routers, Cat 3550 switches & 7940 / 7960 handsets, the phones are all hanging off an asterisk installation .. Bandwidth between offices and the network infrastructure isn't really an issue.

 

Does Cisco offer a SMB product to enable video conferencing between users? 

Do I need a server or appliance at each site to cater for the video mixing / transcoding?

Can webcams on PC's be utilized?

 

Any information or a kick in the right direction is appreciated.

 

-Daniel Hooper

 




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From bob at smalltime.com  Thu May 31 08:48:09 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Thu, 31 May 2007 07:48:09 -0500
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
In-Reply-To: <3e6f56310705302136s214a807h88665d2651d71d26@mail.gmail.com>
References: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>	
	<465C6F69.7070307@carocomp.ro> <465C70EB.9020805@smalltime.com>	
	<3e6f56310705291929y4946b047maab2db9a80fb5d76@mail.gmail.com>	
	<465D7E49.6090308@smalltime.com>
	<3e6f56310705302136s214a807h88665d2651d71d26@mail.gmail.com>
Message-ID: <465EC409.30905@smalltime.com>

Kent Roberts wrote:
> If all the perms are correct, cti ports, css, route paterns, etc.  then 
> its fun time...  :)

My most unfavorite words!

All the CSS, etc, is correct; there are other "virtual lines" that do 
work (that are also CFAs), so I'm trying to isolate if it's just the 
800's that have a problem, or all of them.



From jsteinberg at gmail.com  Thu May 31 09:57:41 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 31 May 2007 09:57:41 -0400
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
In-Reply-To: <465EC409.30905@smalltime.com>
References: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>
	<465C6F69.7070307@carocomp.ro> <465C70EB.9020805@smalltime.com>
	<3e6f56310705291929y4946b047maab2db9a80fb5d76@mail.gmail.com>
	<465D7E49.6090308@smalltime.com>
	<3e6f56310705302136s214a807h88665d2651d71d26@mail.gmail.com>
	<465EC409.30905@smalltime.com>
Message-ID: <f1b65f880705310657q27eb1c37u496c9aada7cc6270@mail.gmail.com>

robert,

you definately have a rdnis problem.  i can think of two ways to fix this.

with mgcp you can uncheck the 'enable outbound rdnis' on the t1 pri
mgcp ccmadmin page.

with h323 you need to use a voice translation profile on your gateway
to strip the redirected number info before you sent your q931 setup
msg.

justin


On 5/31/07, Robert Kulagowski <bob at smalltime.com> wrote:
> Kent Roberts wrote:
> > If all the perms are correct, cti ports, css, route paterns, etc.  then
> > its fun time...  :)
>
> My most unfavorite words!
>
> All the CSS, etc, is correct; there are other "virtual lines" that do
> work (that are also CFAs), so I'm trying to isolate if it's just the
> 800's that have a problem, or all of them.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From bob at smalltime.com  Thu May 31 10:22:54 2007
From: bob at smalltime.com (Robert Kulagowski)
Date: Thu, 31 May 2007 09:22:54 -0500
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
In-Reply-To: <f1b65f880705310657q27eb1c37u496c9aada7cc6270@mail.gmail.com>
References: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>	
	<465C6F69.7070307@carocomp.ro> <465C70EB.9020805@smalltime.com>	
	<3e6f56310705291929y4946b047maab2db9a80fb5d76@mail.gmail.com>	
	<465D7E49.6090308@smalltime.com>	
	<3e6f56310705302136s214a807h88665d2651d71d26@mail.gmail.com>	
	<465EC409.30905@smalltime.com>
	<f1b65f880705310657q27eb1c37u496c9aada7cc6270@mail.gmail.com>
Message-ID: <465EDA3E.20205@smalltime.com>

> you definately have a rdnis problem.  i can think of two ways to fix this.

> with h323 you need to use a voice translation profile on your gateway
> to strip the redirected number info before you sent your q931 setup
> msg.

Third way - uncheck "Redirecting Number IE Delivery - Outbound" on the 
gateway configuration page?

Because I've got it checked now - I'll have to wait until things quiet 
down and try it out.

Thanks for the clue.

From rratliff at cisco.com  Thu May 31 10:37:03 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 31 May 2007 10:37:03 -0400
Subject: [cisco-voip] CM 4.1(3)SR5/5a
Message-ID: <5E7E28D1-C5B3-4FA4-AF73-91E167EF87AC@cisco.com>

Anyone running it in muti-node cluster environments?

Any issues seen so far, particularly related to removing CFA on a  
subscriber?

-Ryan

From duncanw at otc.edu  Thu May 31 10:06:44 2007
From: duncanw at otc.edu (DUNCAN, W B.)
Date: Thu, 31 May 2007 09:06:44 -0500
Subject: [cisco-voip] PoE Phone Problem
In-Reply-To: <06D1B6D4926222458F803D0D3EDCCB7E0201D391@EXM1.otc.edu>
References: <06D1B6D4926222458F803D0D3EDCCB7E0201D391@EXM1.otc.edu>
Message-ID: <06D1B6D4926222458F803D0D3EDCCB7E0201D533@EXM1.otc.edu>

I probably should have elaborated more.  The trouble with both of these
phones seems to be the power module.  I plugged a new phone into where
the bad phone was removed and it works fine.  I took the bad phone to a
known good port it does not work.   I plug in a power brick it works.
The phones seem to not be able to sense PoE and I was hoping perhaps
someone else might have experienced this and have some solution.  My
assumption is that the phone is just no longer any good without a power
brick, but I thought I would ask.  

 

Thanks all,

W. Brian Duncan

Coordinator of Telecommunications

Ozarks Technical Community College

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of DUNCAN, W B.
Sent: Wednesday, May 30, 2007 3:03 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PoE Phone Problem

 

I have had two phones that wont power up with PoE.  I can plug these
into a power brick and they work fine.  Has anyone experienced this and
more so is there a way to repair the phone?

 

Thanks,

W. Brian Duncan

Coordinator of Telecommunications

Ozarks Technical Community College

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From jsteinberg at gmail.com  Thu May 31 11:13:17 2007
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 31 May 2007 11:13:17 -0400
Subject: [cisco-voip] Transferring to Outside Lines from the IPCC ?
In-Reply-To: <465EDA3E.20205@smalltime.com>
References: <B9B5B75DF9145F44A21111DD92CC490006538BC6@DCEX2.DSMO.COM>
	<465C6F69.7070307@carocomp.ro> <465C70EB.9020805@smalltime.com>
	<3e6f56310705291929y4946b047maab2db9a80fb5d76@mail.gmail.com>
	<465D7E49.6090308@smalltime.com>
	<3e6f56310705302136s214a807h88665d2651d71d26@mail.gmail.com>
	<465EC409.30905@smalltime.com>
	<f1b65f880705310657q27eb1c37u496c9aada7cc6270@mail.gmail.com>
	<465EDA3E.20205@smalltime.com>
Message-ID: <f1b65f880705310813t7ec9ab79n5e180f07143f45ee@mail.gmail.com>

ya thats what i meant.

make sure you arent using rdnis for centralized vm with voice over
pstn as it will break that functionality.

On 5/31/07, Robert Kulagowski <bob at smalltime.com> wrote:
> > you definately have a rdnis problem.  i can think of two ways to fix this.
>
> > with h323 you need to use a voice translation profile on your gateway
> > to strip the redirected number info before you sent your q931 setup
> > msg.
>
> Third way - uncheck "Redirecting Number IE Delivery - Outbound" on the
> gateway configuration page?
>
> Because I've got it checked now - I'll have to wait until things quiet
> down and try it out.
>
> Thanks for the clue.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From arlen.nascimento at gmail.com  Thu May 31 11:17:51 2007
From: arlen.nascimento at gmail.com (Arlen Nascimento)
Date: Thu, 31 May 2007 11:17:51 -0400
Subject: [cisco-voip] Trunk
Message-ID: <51461ca60705310817k65b2c9f5k1578189712276160@mail.gmail.com>

Hi all,
i set up a trunk between asterisk and callmanager, i can put calls in both
directions but the call on ccme side is just noise.
i set g711alaw codec in both sides, but the noise persists

Regards,

-- 
Arlen Nascimento
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From prashant.k at tvisha.com  Thu May 31 10:37:44 2007
From: prashant.k at tvisha.com (PRASHANT K)
Date: Thu, 31 May 2007 20:07:44 +0530
Subject: [cisco-voip] Error message while installing CCM 4.0 Subscriber
Message-ID: <200705311537.l4VFbu1F067315@puck.nether.net>

Hi all, 

 

This is the actual error message im getting while installing the subscriber:
"THE INSTALLATION FAILED TO VERIFY THE CISCO CALLMANAGER VERSION THAT RUNS
ON THE PUBLISHER DATABASE SERVER. CANCEL THE INSTALLTION, AND REVIEW THE LOG
FILE AT C:\PROGRAM FILES\COMMON FILES\CISCO\LOGS\CCMUINST.LOG " 

 

05/31/2007 16:23:03.140 MSIApi.cpp MSICheckPubVersion          Function
Started 

05/31/2007 16:23:03.156 MSIApi.cpp MSICheckPubVersion          Successful in
setting property: PubVersionMatch to false 

05/31/2007 16:23:03.156 MSIApi.cpp MSICheckPubVersion          Successful in
getting property: PUBSVRNAME value: CCM1 

05/31/2007 16:23:03.156 MSIApi.cpp MSICheckPubVersion          Successful in
getting property: BUILDVERSION value: 4.0(1) 

05/31/2007 16:23:03.218 MSIApi.cpp MSICheckPubVersion          Error failed
to get registry key: SOFTWARE\Cisco Systems, Inc.\CallManager value: Build 

05/31/2007 16:23:03.218 MSIApi.cpp MSICheckPubVersion          Error code :
1603 

05/31/2007 16:23:03.218 MSIApi.cpp MSICheckPubVersion          Message
displayed to user: The installation failed to verify the Cisco CallManager
version that runs on the publisher database server.  Cancel the
installation, and review the log file at C:\Program Files\Common
Files\Cisco\Logs\CCMUIInst.log

 

The Publisher is working fine. I checked the logs, registry, but im unable
to solve this.

 

Thanks and Regards,

Prashant

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From rratliff at cisco.com  Thu May 31 11:46:58 2007
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 31 May 2007 11:46:58 -0400
Subject: [cisco-voip] Error message while installing CCM 4.0 Subscriber
In-Reply-To: <200705311537.l4VFbu1F067315@puck.nether.net>
References: <200705311537.l4VFbu1F067315@puck.nether.net>
Message-ID: <F0D8683A-FC46-415D-BE3F-F57A322B865C@cisco.com>

On the subscriber run regedit, then see if you can connect to the  
publisher via hostname (ccm1) and browse to HKLM\SOFTWARE\Cisco  
Systems, Inc.\CallManager\Build

-Ryan

On May 31, 2007, at 10:37 AM, PRASHANT K wrote:

Hi all,



This is the actual error message im getting while installing the  
subscriber: ?THE INSTALLATION FAILED TO VERIFY THE CISCO CALLMANAGER  
VERSION THAT RUNS ON THE PUBLISHER DATABASE SERVER. CANCEL THE  
INSTALLTION, AND REVIEW THE LOG FILE AT C:\PROGRAM FILES\COMMON FILES 
\CISCO\LOGS\CCMUINST.LOG ?



05/31/2007 16:23:03.140 MSIApi.cpp MSICheckPubVersion           
Function Started

05/31/2007 16:23:03.156 MSIApi.cpp MSICheckPubVersion           
Successful in setting property: PubVersionMatch to false

05/31/2007 16:23:03.156 MSIApi.cpp MSICheckPubVersion           
Successful in getting property: PUBSVRNAME value: CCM1

05/31/2007 16:23:03.156 MSIApi.cpp MSICheckPubVersion           
Successful in getting property: BUILDVERSION value: 4.0(1)

05/31/2007 16:23:03.218 MSIApi.cpp MSICheckPubVersion          Error  
failed to get registry key: SOFTWARE\Cisco Systems, Inc.\CallManager  
value: Build

05/31/2007 16:23:03.218 MSIApi.cpp MSICheckPubVersion          Error  
code : 1603

05/31/2007 16:23:03.218 MSIApi.cpp MSICheckPubVersion           
Message displayed to user: The installation failed to verify the  
Cisco CallManager version that runs on the publisher database  
server.  Cancel the installation, and review the log file at C: 
\Program Files\Common Files\Cisco\Logs\CCMUIInst.log



The Publisher is working fine. I checked the logs, registry, but im  
unable to solve this?



Thanks and Regards,

Prashant

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From matt at saskin.net  Thu May 31 12:05:56 2007
From: matt at saskin.net (Matthew Saskin)
Date: Thu, 31 May 2007 12:05:56 -0400
Subject: [cisco-voip] Yanking Compact Flash while the router is up...
In-Reply-To: <5d093f9a0705251145k63365279t98cb0bb490aa1360@mail.gmail.com>
References: <5d093f9a0705251145k63365279t98cb0bb490aa1360@mail.gmail.com>
Message-ID: <465EF264.6030608@saskin.net>

Works fine in my experience - not to say it won't crash the router, just 
that I haven't seen it crash a router :)

-matt

Jonathan Charles wrote:
> So, I am curious, can you pull a CF card from a 2821 while the router
> is up and running?
> 
> We need to replace flaky flash with a new one, but I would rather not
> have to talk a remote tech (low skillset) through a ROMMON TFTP
> operation..
> 
> If not, we can do it the hard way....but I have heard rumors that it works...
> 
> If it does, what are the risks...
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From MillerS at DicksteinShapiro.COM  Thu May 31 12:13:38 2007
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Thu, 31 May 2007 12:13:38 -0400
Subject: [cisco-voip]  Any Experience with OpenView ?..
Message-ID: <B9B5B75DF9145F44A21111DD92CC490006538C2A@DCEX2.DSMO.COM>

Most of our network servers are monitored by something called "OpenView".  It notifies engineers if there is a problem.  Does anyone have experience with this program for CCM or Unity?

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 
Fax (202)-330-5607
millers at dicksteinshapiro.com 

----- Original Message -----
From: cisco-voip-bounces at puck.nether.net <cisco-voip-bounces at puck.nether.net>
To: Jonathan Charles <jonvoip at gmail.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>; cisco-nsp at puck.nether.net <cisco-nsp at puck.nether.net>
Sent: Thu May 31 12:05:56 2007
Subject: Re: [cisco-voip] Yanking Compact Flash while the router is up...

Works fine in my experience - not to say it won't crash the router, just 
that I haven't seen it crash a router :)

-matt

Jonathan Charles wrote:
> So, I am curious, can you pull a CF card from a 2821 while the router
> is up and running?
> 
> We need to replace flaky flash with a new one, but I would rather not
> have to talk a remote tech (low skillset) through a ROMMON TFTP
> operation..
> 
> If not, we can do it the hard way....but I have heard rumors that it works...
> 
> If it does, what are the risks...
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


--------------------------------------------------------
This e-mail message and any attached files are confidential and are intended solely for the use of the addressee(s) named above. This communication may contain material protected by attorney-client, work product, or other privileges. If you are not the intended recipient or person responsible for delivering this confidential communication to the intended recipient, you have received this communication in error, and any review, use, dissemination, forwarding, printing, copying, or other distribution of this e-mail message and any attached files is strictly prohibited. Dickstein Shapiro reserves the right to monitor any communication that is created, received, or sent on its network.  If you have received this confidential communication in error, please notify the sender immediately by reply e-mail message and permanently delete the original message. 

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From jonvoip at gmail.com  Thu May 31 12:19:08 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 31 May 2007 11:19:08 -0500
Subject: [cisco-voip] Any Experience with OpenView ?..
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538C2A@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490006538C2A@DCEX2.DSMO.COM>
Message-ID: <5d093f9a0705310919v738c5518o95a926ccc3f9c19@mail.gmail.com>

I used to work for a company that monitored hundreds of customers
using Openview, we ran the OVO agent on CCM, Unity, IPCC, CER... all
Windows...

Worked great and it is Cisco approved.



Jonathan

On 5/31/07, Miller, Steve <MillerS at dicksteinshapiro.com> wrote:
>
>
>
> Most of our network servers are monitored by something called "OpenView".
> It notifies engineers if there is a problem.  Does anyone have experience
> with this program for CCM or Unity?
>
>  Steve Miller
>  Telecom Engineer
>  Dickstein Shapiro LLP
>  1825 Eye Street NW | Washington, DC 20006
>  Tel (202) 420-3370
>  Fax (202)-330-5607
>  millers at dicksteinshapiro.com
>
>  ----- Original Message -----
>  From: cisco-voip-bounces at puck.nether.net
> <cisco-voip-bounces at puck.nether.net>
>  To: Jonathan Charles <jonvoip at gmail.com>
>  Cc: ciscovoip <cisco-voip at puck.nether.net>; cisco-nsp at puck.nether.net
> <cisco-nsp at puck.nether.net>
>  Sent: Thu May 31 12:05:56 2007
>  Subject: Re: [cisco-voip] Yanking Compact Flash while the router is up...
>
>  Works fine in my experience - not to say it won't crash the router, just
>  that I haven't seen it crash a router :)
>
>  -matt
>
>  Jonathan Charles wrote:
>  > So, I am curious, can you pull a CF card from a 2821 while the router
>  > is up and running?
>  >
>  > We need to replace flaky flash with a new one, but I would rather not
>  > have to talk a remote tech (low skillset) through a ROMMON TFTP
>  > operation..
>  >
>  > If not, we can do it the hard way....but I have heard rumors that it
> works...
>  >
>  > If it does, what are the risks...
>  >
>  >
>  >
>  > Jonathan
>  > _______________________________________________
>  > cisco-voip mailing list
>  > cisco-voip at puck.nether.net
>  > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>  _______________________________________________
>  cisco-voip mailing list
>  cisco-voip at puck.nether.net
>  https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> --------------------------------------------------------
> This e-mail message and any attached files are confidential and are intended
> solely for the use of the addressee(s) named above. This communication may
> contain material protected by attorney-client, work product, or other
> privileges. If you are not the intended recipient or person responsible for
> delivering this confidential communication to the intended recipient, you
> have received this communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or other distribution of this
> e-mail message and any attached files is strictly prohibited. Dickstein
> Shapiro reserves the right to monitor any communication that is created,
> received, or sent on its network. If you have received this confidential
> communication in error, please notify the sender immediately by reply e-mail
> message and permanently delete the original message.
>
> To reply to our email administrator directly, send an email to
> postmaster at dicksteinshapiro.com
>
> Dickstein Shapiro LLP
> http://www.DicksteinShapiro.com
> ==============================================================================
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From erickbe at yahoo.com  Thu May 31 12:23:36 2007
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu, 31 May 2007 09:23:36 -0700 (PDT)
Subject: [cisco-voip] Yanking Compact Flash while the router is up...
Message-ID: <893415.84904.qm@web37210.mail.mud.yahoo.com>

I have seen it cause a router to crash when the router was configured for SRST MOH and it could no longer read the audio file from flash.... if i recall this may have been due to a bug also in earlier IOS versions. What I did to get around the issue the one time I had to deal with an urgent issue and had to do this, was briefly unconfigure the SRST MOH and reboot the router so it wasn't accessing the flash all the time. 


----- Original Message ----
From: Matthew Saskin <matt at saskin.net>
To: Jonathan Charles <jonvoip at gmail.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>; cisco-nsp at puck.nether.net
Sent: Thursday, May 31, 2007 11:05:56 AM
Subject: Re: [cisco-voip] Yanking Compact Flash while the router is up...

Works fine in my experience - not to say it won't crash the router, just 
that I haven't seen it crash a router :)

-matt

Jonathan Charles wrote:
> So, I am curious, can you pull a CF card from a 2821 while the router
> is up and running?
> 
> We need to replace flaky flash with a new one, but I would rather not
> have to talk a remote tech (low skillset) through a ROMMON TFTP
> operation..
> 
> If not, we can do it the hard way....but I have heard rumors that it works...
> 
> If it does, what are the risks...
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip





       
____________________________________________________________________________________
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Comedy with an Edge to see what's on, when. 
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From jason.aarons at us.didata.com  Thu May 31 12:25:15 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 31 May 2007 12:25:15 -0400
Subject: [cisco-voip] Any Experience with OpenView ?..
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538C2A@DCEX2.DSMO.COM>
References: <B9B5B75DF9145F44A21111DD92CC490006538C2A@DCEX2.DSMO.COM>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B86006@USNAEXCH.na.didata.local>

OpenView is a family of products, while Operations Managers does a great job, the HP Systems Insight Manager product knows the SNMP MIBs and Agents loaded on HP boxes. I?ve used both as they are just SNMP tools, but the HP Insight Manager required less customization and did a great job of monitoring all my HP-DL3XX servers. If I was using IBM I would look at IBM Director, if using Dell servers then Dell OpenManage.  By default the Win2k OS build from Cisco has the HP Insight Agents already installed.

 

http://h18013.www1.hp.com/products/servers/management/hpsim/index.html

 

http://h20229.www2.hp.com/

 

 

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Miller, Steve
Sent: Thursday, May 31, 2007 12:14 PM
Cc: cisco-voip at puck.nether.net; cisco-nsp at puck.nether.net
Subject: [cisco-voip] Any Experience with OpenView ?..

 

Most of our network servers are monitored by something called "OpenView".  It notifies engineers if there is a problem.  Does anyone have experience with this program for CCM or Unity?

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370
Fax (202)-330-5607
millers at dicksteinshapiro.com

----- Original Message -----
From: cisco-voip-bounces at puck.nether.net <cisco-voip-bounces at puck.nether.net>
To: Jonathan Charles <jonvoip at gmail.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>; cisco-nsp at puck.nether.net <cisco-nsp at puck.nether.net>
Sent: Thu May 31 12:05:56 2007
Subject: Re: [cisco-voip] Yanking Compact Flash while the router is up...

Works fine in my experience - not to say it won't crash the router, just
that I haven't seen it crash a router :)

-matt

Jonathan Charles wrote:
> So, I am curious, can you pull a CF card from a 2821 while the router
> is up and running?
>
> We need to replace flaky flash with a new one, but I would rather not
> have to talk a remote tech (low skillset) through a ROMMON TFTP
> operation..
>
> If not, we can do it the hard way....but I have heard rumors that it works...
>
> If it does, what are the risks...
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


_______________________________________________
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From asobihoudai at yahoo.com  Thu May 31 13:16:31 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 31 May 2007 10:16:31 -0700 (PDT)
Subject: [cisco-voip] Called Number=(TON=Unknown, NPI=Unknown)
In-Reply-To: <FF7F71539DC4E544BA5737C7AA6CEE1747706E@ems05.ems-1.net>
Message-ID: <168866.71769.qm@web57214.mail.re3.yahoo.com>

--- Sumit Marwah <smarwah at enetsolutions.net> wrote:

> Hi Paul,
> 
> Can you send the complete debug for voice ccapi
> inout and if you can add
> debug isdn q931 log that would be great? Also,
> adding the configuration
> for the gateway would be helpful.
> 
> What version was suggested by TAC for the issue you
> were running into
> before?

Sumit,

The IOS version recommended by TAC is currently on the
gateway. There are no PRI lines on this system, only
E&M DIOD lines.

Here is the output from debug voice ccapi inout.

Deleting profileTable[0x84838C34]
May 30 22:39:14.539:
//-1/19BFBABF8D85/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0
May 30 22:39:14.543:
//-1/19BFBABF8D85/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x8489B314, Call Info(
   Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type
Str=RegularLine, FinalDestinationFlag=FALSE,
   Incoming Dial-peer=100, Progress
Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE
Present=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route
Label=, CLID Transparent=FALSE), Call Id=-1
May 30 22:39:14.543:
//-1/19BFBABF8D85/CCAPI/cc_api_call_setup_ind_common:
   (calling is null, answer_addr=,
dest_pattern=[469]11,
   e164_ans_addr=0, e164_dest_pattern=0)
May 30 22:39:14.547:
//-1/19BFBABF8D85/CCAPI/cc_api_call_setup_ind_common:
   Interface Type=1, Protocol=0
May 30 22:39:14.547:
//-1/19BFBABF8D85/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed)
May 30 22:39:14.547:
//-1/19BFBABF8D85/CCAPI/ccCheckClipClir:
   Calling Party Number Is User Provided
May 30 22:39:14.547:
//-1/19BFBABF8D85/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed)
May 30 22:39:14.547:
//3448/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:
   Total Call Count=0, Call Entry(Call Count On=FALSE,
Incoming Call=TRUE)
May 30 22:39:14.547:
//3448/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:
   Total Call Count=1
May 30 22:39:14.547:
//3448/19BFBABF8D85/CCAPI/cc_insert_guid_pod_entry:
   Incoming=TRUE, Call Id=3448
May 30 22:39:14.551:
//3448/19BFBABF8D85/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown))
May 30 22:39:14.551:
//3448/19BFBABF8D85/CCAPI/cc_process_call_setup_ind:
   Event=0x84865FE0
May 30 22:39:14.555:
//-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:
   Matching Parameters; Called Number=, Call Transfer
Consult Id=
May 30 22:39:14.555:
//-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Searching Node;
   Called Number=, Call Transfer Consult Id=
May 30 22:39:14.555:
//-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:
   No Matching Node
May 30 22:39:14.555:
//3448/19BFBABF8D85/CCAPI/ccCallSetContext:
   Context=0x83D38F68
May 30 22:39:14.555:
//3448/19BFBABF8D85/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 3448 with tag 100 to app
"_ManagedAppProcess_Default"
May 30 22:39:14.559:
//3448/19BFBABF8D85/CCAPI/ccCallSetupAck:
   Call Id=3448
May 30 22:39:14.559:
//3448/19BFBABF8D85/CCAPI/cc_api_set_transfer_info:
   Transfer Number=, Transfer Reason=0x0
May 30 22:39:14.559:
//3448/19BFBABF8D85/CCAPI/cc_api_set_transfer_info:
   Call Transfer Reset
May 30 22:39:14.559:
//3448/19BFBABF8D85/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=TRUE, Tone=Dial Tone,
   Tone Direction=Network, Params=0x0, Call Id=3448
May 30 22:39:14.559:
//3448/19BFBABF8D85/CCAPI/ccSetDigitTimeouts:
   Initial Digit Timeout=-1000(ms), Inter Digit
Timeout=-1000(ms)
May 30 22:39:14.563:
//3448/19BFBABF8D85/CCAPI/ccSetDigitTimeouts:
   Call Entry(Inter Digit Timeout=10000(ms), Initial
Digit Timeout=10000(ms))
May 30 22:39:14.563:
//3448/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0xD78, digit_event=0x1, enable=TRUE,
consume=FALSE)
May 30 22:39:14.563:
//3448/19BFBABF8D85/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=3448
May 30 22:39:14.563:
//3448/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x8489B314, callID=0xD78, disp=0,
digit_event=0x1, enable=TRUE, consume=FALSE)
May 30 22:39:14.563:
//3448/19BFBABF8D85/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0x0,
Interface=0x8489B314, Call Id=3448
May 30 22:39:14.563:
//3448/19BFBABF8D85/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=10000(ms), Inter
Digit Timeout=10000(ms))
May 30 22:39:15.804:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=9, DigitBeginFlags=0x1,
   Rtp Timestamp=0xFFFF0008, Rtp Expiration=0x0
May 30 22:39:15.824:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=9, Duration=4,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone
Mode=DTMF
May 30 22:39:15.828:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
May 30 22:39:15.924:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=9, DigitBeginFlags=0x1,
   Rtp Timestamp=0xFFFF0008, Rtp Expiration=0x0
May 30 22:39:15.944:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=9, Duration=4,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone
Mode=DTMF
May 30 22:39:15.944:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
May 30 22:39:16.044:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=6, DigitBeginFlags=0x1,
   Rtp Timestamp=0xFFFF0008, Rtp Expiration=0x0
May 30 22:39:16.084:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=6, Duration=24,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone
Mode=DTMF
May 30 22:39:16.084:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
May 30 22:39:16.176:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=4, DigitBeginFlags=0x1,
   Rtp Timestamp=0xFFFF0008, Rtp Expiration=0x0
May 30 22:39:16.216:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=4, Duration=24,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone
Mode=DTMF
May 30 22:39:16.216:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
May 30 22:39:16.304:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=0, DigitBeginFlags=0x1,
   Rtp Timestamp=0xFFFF0008, Rtp Expiration=0x0
May 30 22:39:16.356:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=0, Duration=24,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone
Mode=DTMF
May 30 22:39:16.356:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
May 30 22:39:16.436:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=0, DigitBeginFlags=0x1,
   Rtp Timestamp=0xFFFF0008, Rtp Expiration=0x0
May 30 22:39:16.492:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=0, Duration=24,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone
Mode=DTMF
May 30 22:39:16.492:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
May 30 22:39:16.564:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=0, DigitBeginFlags=0x1,
   Rtp Timestamp=0xFFFF0008, Rtp Expiration=0x0
May 30 22:39:16.584:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3448, Digit=0, Duration=4,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone
Mode=DTMF
May 30 22:39:16.584:
//3448/19BFBABF8D85/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
May 30 22:39:16.588:
//3448/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0xD78, digit_event=0x0, enable=FALSE,
consume=FALSE)
May 30 22:39:16.588:
//3448/19BFBABF8D85/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=3448
May 30 22:39:16.588:
//3448/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x8489B314, callID=0xD78, disp=0,
digit_event=0x0, enable=FALSE, consume=FALSE)
May 30 22:39:16.588:
//3448/19BFBABF8D85/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0x0,
Interface=0x8489B314, Call Id=3448
May 30 22:39:16.588:
//3448/19BFBABF8D85/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=10000(ms), Inter
Digit Timeout=10000(ms))
May 30 22:39:16.592:
//3448/19BFBABF8D85/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
May 30 22:39:16.596:
//3448/19BFBABF8D85/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=FALSE, Mode=0,
   Outgoing Dial-peer=1, Params=0x8580D794, Progress
Indication=ORIGINATING SIDE IS NON ISDN(3)
May 30 22:39:16.596:
//3448/19BFBABF8D85/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed)
May 30 22:39:16.600:
//3448/19BFBABF8D85/CCAPI/ccCheckClipClir:
   Calling Party Number Is User Provided
May 30 22:39:16.600:
//3448/19BFBABF8D85/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed)
May 30 22:39:16.600:
//3448/19BFBABF8D85/CCAPI/ccCallSetupRequest:
   Destination Pattern=99640.., Called Number=9964000,
Digit Strip=FALSE
May 30 22:39:16.600:
//3448/19BFBABF8D85/CCAPI/ccCallSetupRequest:
   Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed),
   Called Number=9964000(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=, Final Destination Flag=FALSE,
   Guid=19BFBABF-0E70-11DC-8D85-97C7BB78F5A4, Outgoing
Dial-peer=1
May 30 22:39:16.600:
//3448/19BFBABF8D85/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=9964000
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0
May 30 22:39:16.604:
//3448/19BFBABF8D85/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x846A8394, Interface Type=1,
Destination=, Mode=0x0,
   Call Params(Calling Number=(TON=Unknown,
NPI=Unknown, Screening=Not Screened,
Presentation=Allowed),
   Called Number=9964000(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE,
   Subscriber Type Str=RegularLine,
FinalDestinationFlag=FALSE, Outgoing Dial-peer=1, Call
Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route
Label=, tg_label_flag=0, Application Call Id=)
May 30 22:39:16.604:
//3449/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:
   Total Call Count=1, Call Entry(Call Count On=FALSE,
Incoming Call=FALSE)
May 30 22:39:16.604:
//3448/19BFBABF8D85/CCAPI/cc_peer_bind:
   Bind=TRUE, Binder Call Id=3448, Bindee Call Id=3449
May 30 22:39:16.604:
//3449/19BFBABF8D85/CCAPI/cc_insert_guid_pod_entry:
   Incoming=FALSE, Call Id=3449
May 30 22:39:16.608:
//3449/19BFBABF8D85/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface
Type=1, FlowMode=1
May 30 22:39:16.608:
//3448/19BFBABF8D85/CCAPI/ccCallSetupRequest:exit at 2111
May 30 22:39:16.608:
//3449/19BFBABF8D85/CCAPI/ccCallSetContext:
   Context=0x8580D744
May 30 22:39:16.608:
//3448/19BFBABF8D85/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=1
May 30 22:39:16.612:
//-1/xxxxxxxxxxxx/CCAPI/cc_api_supported_data:
   data_mode=0x10082
May 30 22:39:16.616:
//3449/19BFBABF8D85/CCAPI/cc_incr_if_call_volume:
   Remote IP Address=199.142.129.7, Hwidb=Serial0/0:0
May 30 22:39:16.616:
//3449/19BFBABF8D85/CCAPI/cc_incr_if_call_volume:
   Total Call Count=1, Voip Call Count=1, MMoip Call
Count=0
May 30 22:39:16.752:
//3449/19BFBABF8D85/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 1)
May 30 22:39:16.752:
//3449/19BFBABF8D85/CCAPI/cc_api_call_proceeding:
   Interface=0x846A8394, Progress Indication=NULL(0)
May 30 22:39:16.752:
//3449/19BFBABF8D85/CCAPI/cc_api_call_proceeding:exit at 6160
May 30 22:39:16.760:
//3449/19BFBABF8D85/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 1)
May 30 22:39:16.760:
//3449/19BFBABF8D85/CCAPI/cc_api_set_delay_xport:
   CallInfo(delay xport=TRUE)
May 30 22:39:16.764:
//3449/19BFBABF8D85/CCAPI/cc_api_call_alert:
   Interface=0x846A8394, Progress Indication=NULL(0),
Signal Indication=SIGNAL RINGBACK(1)
May 30 22:39:16.764:
//3449/19BFBABF8D85/CCAPI/cc_api_call_alert:
   Call Entry(Retry Count=0, Responsed=TRUE)
May 30 22:39:16.764:
//3449/19BFBABF8D85/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 1)
May 30 22:39:16.764:
//3449/19BFBABF8D85/CCAPI/cc_api_call_notify:
   Data Bitmask=0x5, Interface=0x846A8394, Call
Id=3449
May 30 22:39:16.768:
//3449/19BFBABF8D85/CCAPI/cc_api_call_notify:exit at 6934
May 30 22:39:16.768:
//3448/19BFBABF8D85/CCAPI/ccCallAlert:
   Progress Indication=NULL(0), Signal
Indication=SIGNAL RINGBACK(1)
May 30 22:39:16.768:
//3448/19BFBABF8D85/CCAPI/ccCallAlert:
   Call Entry(Responsed=TRUE, AlertSent=TRUE)
May 30 22:39:16.772:
//3449/19BFBABF8D85/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
   CallInfo(ssCTreRoutingNotSupported=FALSE)
May 30 22:39:16.772:
//3449/19BFBABF8D85/CCAPI/cc_api_get_ccm_detected:
   CallInfo(ccm detected=TRUE)
May 30 22:39:16.772:
//3448/19BFBABF8D85/CCAPI/ccCallNotify:
   Data Bitmask=0x5, Call Id=3448
May 30 22:39:20.083:
//3448/19BFBABF8D85/CCAPI/cc_api_call_disconnected:
   Cause Value=16, Interface=0x8489B314, Call Id=3448
May 30 22:39:20.083:
//3448/19BFBABF8D85/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=16, Retry
Count=0)
May 30 22:39:20.083:
//3448/19BFBABF8D85/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=FALSE, Tone=Null,
   Tone Direction=Network, Params=0x0, Call Id=3448
May 30 22:39:20.087:
//3449/19BFBABF8D85/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous
Disconnect Cause=0, Disconnect Cause=0)
May 30 22:39:20.087:
//3449/19BFBABF8D85/CCAPI/ccCallDisconnect:
   Start Calling Accounting;
   Call Entry(Incoming=FALSE)
May 30 22:39:20.087:
//3449/19BFBABF8D85/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Disconnect Cause=0)
May 30 22:39:20.087:
//3449/19BFBABF8D85/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause
Value=16)
May 30 22:39:20.087:
//3449/19BFBABF8D85/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
May 30 22:39:20.091:
//3448/19BFBABF8D85/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous
Disconnect Cause=0, Disconnect Cause=16)
May 30 22:39:20.091:
//3448/19BFBABF8D85/CCAPI/ccCallDisconnect:
   Start Calling Accounting;
   Call Entry(Incoming=TRUE)
May 30 22:39:20.091:
//3448/19BFBABF8D85/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Disconnect Cause=16)
May 30 22:39:20.091:
//3448/19BFBABF8D85/CCAPI/ccCallDisconnect:
   Call Entry(Disconnect Cause=16)
May 30 22:39:20.091:
//3448/19BFBABF8D85/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause
Value=16)
May 30 22:39:20.091:
//3448/19BFBABF8D85/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
May 30 22:39:20.095:
//-1/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:
   CALL_ERROR_INFORMATIONAL; Call Id Is Invalid=-1
May 30 22:39:20.095:
//3448/xxxxxxxxxxxx/CCAPI/cc_api_request_rt_packet_stats_done:
   CALL_ERROR_INFORMATIONAL; tag not properly returned
from SPI
May 30 22:39:20.499:
//3448/19BFBABF8D85/CCAPI/ccCallGetVoipFlag:
   Data Bitmask=0x1, Call Id=3448
May 30 22:39:20.499:
//3448/19BFBABF8D85/CCAPI/ccCallGetVoipFlag:
   Flag=FALSE
May 30 22:39:20.503:
//3448/19BFBABF8D85/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x8489B314, Tag=0x0, Call
Id=3448,
   Call Entry(Disconnect Cause=16, Voice Class Cause
Code=0, Retry Count=0)
May 30 22:39:20.503:
//3448/19BFBABF8D85/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 22:39:20.503:
//3448/19BFBABF8D85/CCAPI/cc_delete_guid_pod_entry:
   Incoming=TRUE
May 30 22:39:20.507:
//3448/19BFBABF8D85/CCAPI/cc_delete_call_entry:
   ccFreeRawMsgInfo=0x8486850C
May 30 22:39:20.507:
//3448/19BFBABF8D85/CCAPI/cc_delete_call_entry:
   Total Call Count=1, Call Entry(Call Count On=FALSE,
Incoming Call=TRUE)
May 30 22:39:20.507:
//3448/19BFBABF8D85/CCAPI/cc_delete_call_entry:
   Total Call Count=0
May 30 22:39:20.507:
//3448/19BFBABF8D85/CCAPI/cc_delete_call_entry:
   Deleting profileTable[0x848398DC]
May 30 22:39:22.100:
//-1/xxxxxxxxxxxx/CCAPI/cc_api_icpif:
   ExpectFactor=0xA
May 30 22:39:22.100:
//3449/19BFBABF8D85/CCAPI/ccCallGetVoipFlag:
   Data Bitmask=0x1, Call Id=3449
May 30 22:39:22.100:
//3449/19BFBABF8D85/CCAPI/ccCallGetVoipFlag:
   Flag=FALSE
May 30 22:39:22.100:
//3449/19BFBABF8D85/CCAPI/cc_decr_if_call_volume:
   Remote IP Address=199.142.129.7, Hwidb=Serial0/0:0
May 30 22:39:22.100:
//3449/19BFBABF8D85/CCAPI/cc_decr_if_call_volume:
   Total Call Count=0, Voip Call Count=0, MMoip Call
Count=0
May 30 22:39:22.104:
//3449/19BFBABF8D85/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x846A8394, Tag=0x0, Call
Id=3449,
   Call Entry(Disconnect Cause=16, Voice Class Cause
Code=0, Retry Count=0)
May 30 22:39:22.104:
//3449/19BFBABF8D85/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 22:39:22.104:
//3449/19BFBABF8D85/CCAPI/cc_delete_guid_pod_entry:
   Incoming=FALSE
May 30 22:39:22.104:
//3449/19BFBABF8D85/CCAPI/cc_delete_call_entry:
   Total Call Count=0, Call Entry(Call Count On=FALSE,
Incoming Call=FALSE)
May 30 22:39:22.108:
//3449/19BFBABF8D85/CCAPI/cc_delete_call_entry:
   Deleting profileTable[0x84838ACC]
May 30 22:41:09.819:
//-1/5E756AA28D88/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0
May 30 22:41:09.819:
//-1/5E756AA28D88/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x8489B314, Call Info(
   Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type
Str=RegularLine, FinalDestinationFlag=FALSE,
   Incoming Dial-peer=100, Progress
Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE
Present=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route
Label=, CLID Transparent=FALSE), Call Id=-1
May 30 22:41:09.823:
//-1/5E756AA28D88/CCAPI/cc_api_call_setup_ind_common:
   (calling is null, answer_addr=,
dest_pattern=[469]11,
   e164_ans_addr=0, e164_dest_pattern=0)
May 30 22:41:09.823:
//-1/5E756AA28D88/CCAPI/cc_api_call_setup_ind_common:
   Interface Type=1, Protocol=0
May 30 22:41:09.823:
//-1/5E756AA28D88/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed)
May 30 22:41:09.823:
//-1/5E756AA28D88/CCAPI/ccCheckClipClir:
   Calling Party Number Is User Provided
May 30 22:41:09.823:
//-1/5E756AA28D88/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed)
May 30 22:41:09.827:
//3450/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:
   Total Call Count=0, Call Entry(Call Count On=FALSE,
Incoming Call=TRUE)
May 30 22:41:09.827:
//3450/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:
   Total Call Count=1
May 30 22:41:09.827:
//3450/5E756AA28D88/CCAPI/cc_insert_guid_pod_entry:
   Incoming=TRUE, Call Id=3450
May 30 22:41:09.827:
//3450/5E756AA28D88/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=(TON=Unknown, NPI=Unknown,
Screening=Not Screened, Presentation=Allowed),
   Called Number=(TON=Unknown, NPI=Unknown))
May 30 22:41:09.831:
//3450/5E756AA28D88/CCAPI/cc_process_call_setup_ind:
   Event=0x848657F0
May 30 22:41:09.831:
//-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:
   Matching Parameters; Called Number=, Call Transfer
Consult Id=
May 30 22:41:09.831:
//-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Searching Node;
   Called Number=, Call Transfer Consult Id=
May 30 22:41:09.831:
//-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:
   No Matching Node
May 30 22:41:09.831:
//3450/5E756AA28D88/CCAPI/ccCallSetContext:
   Context=0x85811484
May 30 22:41:09.835:
//3450/5E756AA28D88/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 3450 with tag 100 to app
"_ManagedAppProcess_Default"
May 30 22:41:09.835:
//3450/5E756AA28D88/CCAPI/ccCallSetupAck:
   Call Id=3450
May 30 22:41:09.835:
//3450/5E756AA28D88/CCAPI/cc_api_set_transfer_info:
   Transfer Number=, Transfer Reason=0x0
May 30 22:41:09.835:
//3450/5E756AA28D88/CCAPI/cc_api_set_transfer_info:
   Call Transfer Reset
May 30 22:41:09.839:
//3450/5E756AA28D88/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=TRUE, Tone=Dial Tone,
   Tone Direction=Network, Params=0x0, Call Id=3450
May 30 22:41:09.839:
//3450/5E756AA28D88/CCAPI/ccSetDigitTimeouts:
   Initial Digit Timeout=-1000(ms), Inter Digit
Timeout=-1000(ms)
May 30 22:41:09.839:
//3450/5E756AA28D88/CCAPI/ccSetDigitTimeouts:
   Call Entry(Inter Digit Timeout=10000(ms), Initial
Digit Timeout=10000(ms))
May 30 22:41:09.839:
//3450/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0xD7A, digit_event=0x1, enable=TRUE,
consume=FALSE)
May 30 22:41:09.839:
//3450/5E756AA28D88/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=3450
May 30 22:41:09.839:
//3450/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x8489B314, callID=0xD7A, disp=0,
digit_event=0x1, enable=TRUE, consume=FALSE)
May 30 22:41:09.843:
//3450/5E756AA28D88/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0x0,
Interface=0x8489B314, Call Id=3450
May 30 22:41:09.843:
//3450/5E756AA28D88/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=10000(ms), Inter
Digit Timeout=10000(ms))
May 30 22:41:11.083:
//3450/5E756AA28D88/CCAPI/cc_api_call_digit_begin:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3450, Digit=D, DigitBeginFlags=0x1,
   Rtp Timestamp=0xFFFF0008, Rtp Expiration=0x0
May 30 22:41:11.163:
//3450/5E756AA28D88/CCAPI/cc_api_call_digit_end:
   Destination Interface=0x0, Destination Mask=0x1,
Destination Call Id=-1,
   Source Call Id=3450, Digit=D, Duration=64,
   Xrule Calling Tag=0, Xrule Called Tag=0, Digit Tone
Mode=DTMF
May 30 22:41:11.163:
//3450/5E756AA28D88/CCAPI/cc_api_call_digit_end:
   Call Entry(Handoff Depth=0)
May 30 22:41:11.167:
//3450/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
   (callID=0xD7A, digit_event=0x0, enable=FALSE,
consume=FALSE)
May 30 22:41:11.167:
//3450/5E756AA28D88/CCAPI/ccCallReportDigits:
   Enabled=TRUE, Call Id=3450
May 30 22:41:11.167:
//3450/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
   (vdbPtr=0x8489B314, callID=0xD7A, disp=0,
digit_event=0x0, enable=FALSE, consume=FALSE)
May 30 22:41:11.167:
//3450/5E756AA28D88/CCAPI/cc_api_call_report_digits_done:
   Enabled=TRUE, Disposition=0x0,
Interface=0x8489B314, Call Id=3450
May 30 22:41:11.167:
//3450/5E756AA28D88/CCAPI/cc_api_call_report_digits_done:
   Call Entry(Initial Digit Timeout=10000(ms), Inter
Digit Timeout=10000(ms))
May 30 22:41:11.171:
//3450/5E756AA28D88/CCAPI/ccCallDisconnect:
   Cause Value=28, Tag=0x0, Call Entry(Previous
Disconnect Cause=0, Disconnect Cause=0)
May 30 22:41:11.171:
//3450/5E756AA28D88/CCAPI/ccCallDisconnect:
   Start Calling Accounting;
   Call Entry(Incoming=TRUE)
May 30 22:41:11.171:
//3450/5E756AA28D88/CCAPI/ccCallDisconnect:
   Cause Value=28, Call Entry(Disconnect Cause=0)
May 30 22:41:11.171:
//3450/5E756AA28D88/CCAPI/ccCallDisconnect:
   Cause Value=28, Call Entry(Responsed=TRUE, Cause
Value=28)
May 30 22:41:11.175:
//3450/5E756AA28D88/CCAPI/cc_api_get_transfer_info:
   Transfer Number Is Null
May 30 22:41:11.175:
//3450/5E756AA28D88/CCAPI/cc_api_call_disc_cause_update:
   Cause Value=16
May 30 22:41:11.175:
//3450/5E756AA28D88/CCAPI/cc_api_call_disc_cause_update:
   Call Entry(Disconnect Cause=16)
May 30 22:41:16.216:
//3450/5E756AA28D88/CCAPI/ccCallGetVoipFlag:
   Data Bitmask=0x1, Call Id=3450
May 30 22:41:16.216:
//3450/5E756AA28D88/CCAPI/ccCallGetVoipFlag:
   Flag=FALSE
May 30 22:41:16.220:
//3450/5E756AA28D88/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x8489B314, Tag=0x0, Call
Id=3450,
   Call Entry(Disconnect Cause=16, Voice Class Cause
Code=0, Retry Count=0)
May 30 22:41:16.220:
//3450/5E756AA28D88/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 22:41:16.224:
//3450/5E756AA28D88/CCAPI/cc_delete_guid_pod_entry:
   Incoming=TRUE
May 30 22:41:16.224:
//3450/5E756AA28D88/CCAPI/cc_delete_call_entry:
   ccFreeRawMsgInfo=0x84869474
May 30 22:41:16.224:
//3450/5E756AA28D88/CCAPI/cc_delete_call_entry:
   Total Call Count=1, Call Entry(Call Count On=FALSE,
Incoming Call=TRUE)
May 30 22:41:16.224:
//3450/5E756AA28D88/CCAPI/cc_delete_call_entry:
   Total Call Count=0
May 30 22:41:16.224:
//3450/5E756AA28D88/CCAPI/cc_delete_call_entry:
   Deleting profileTable[0x848391B0]
May 30 22:44:15.967:
//-1/xxxxxxxxxxxx/CCAPI/ccAppShutdownMode:
   ccAppShutdownMode: remove it from the queue


       
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From asobihoudai at yahoo.com  Thu May 31 13:23:26 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 31 May 2007 10:23:26 -0700 (PDT)
Subject: [cisco-voip] Any Experience with OpenView ?..
In-Reply-To: <B9B5B75DF9145F44A21111DD92CC490006538C2A@DCEX2.DSMO.COM>
Message-ID: <370821.19527.qm@web57203.mail.re3.yahoo.com>

HP OpenView is a common system/network monitoring
program like CiscoWorks or Nagios. I am not sure if
SNMP monitoring is supported with CallManager but I'm
fairly certain it does as it would be incredibly
cumbersome to monitor individual CCM boxes.

Paul

--- "Miller, Steve" <MillerS at DicksteinShapiro.COM>
wrote:

> Most of our network servers are monitored by
> something called "OpenView".  It notifies engineers
> if there is a problem.  Does anyone have experience
> with this program for CCM or Unity?
> 
> Steve Miller
> Telecom Engineer
> Dickstein Shapiro LLP
> 1825 Eye Street NW | Washington, DC 20006
> Tel (202) 420-3370 
> Fax (202)-330-5607
> millers at dicksteinshapiro.com 
> 
> ----- Original Message -----
> From: cisco-voip-bounces at puck.nether.net
> <cisco-voip-bounces at puck.nether.net>
> To: Jonathan Charles <jonvoip at gmail.com>
> Cc: ciscovoip <cisco-voip at puck.nether.net>;
> cisco-nsp at puck.nether.net
> <cisco-nsp at puck.nether.net>
> Sent: Thu May 31 12:05:56 2007
> Subject: Re: [cisco-voip] Yanking Compact Flash
> while the router is up...
> 
> Works fine in my experience - not to say it won't
> crash the router, just 
> that I haven't seen it crash a router :)
> 
> -matt
> 
> Jonathan Charles wrote:
> > So, I am curious, can you pull a CF card from a
> 2821 while the router
> > is up and running?
> > 
> > We need to replace flaky flash with a new one, but
> I would rather not
> > have to talk a remote tech (low skillset) through
> a ROMMON TFTP
> > operation..
> > 
> > If not, we can do it the hard way....but I have
> heard rumors that it works...
> > 
> > If it does, what are the risks...
> > 
> > 
> > 
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
>
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> product, or other privileges. If you are not the
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> delivering this confidential communication to the
> intended recipient, you have received this
> communication in error, and any review, use,
> dissemination, forwarding, printing, copying, or
> other distribution of this e-mail message and any
> attached files is strictly prohibited. Dickstein
> Shapiro reserves the right to monitor any
> communication that is created, received, or sent on
> its network.  If you have received this confidential
> communication in error, please notify the sender
> immediately by reply e-mail message and permanently
> delete the original message. 
> 
> To reply to our email administrator directly, send
> an email to postmaster at dicksteinshapiro.com
> 
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> http://www.DicksteinShapiro.com
>
==============================================================================
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



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From jonvoip at gmail.com  Thu May 31 13:27:48 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 31 May 2007 12:27:48 -0500
Subject: [cisco-voip] Yanking Compact Flash while the router is up...
In-Reply-To: <893415.84904.qm@web37210.mail.mud.yahoo.com>
References: <893415.84904.qm@web37210.mail.mud.yahoo.com>
Message-ID: <5d093f9a0705311027w45c30692v39fc570e823f966e@mail.gmail.com>

Well, I have done it twice in the last two weeks, no problems.

Neither of the routers was a voice gateway, they were just boring WAN routers...


Jonathan

On 5/31/07, Erick Bergquist <erickbe at yahoo.com> wrote:
> I have seen it cause a router to crash when the router was configured for SRST MOH and it could no longer read the audio file from flash.... if i recall this may have been due to a bug also in earlier IOS versions. What I did to get around the issue the one time I had to deal with an urgent issue and had to do this, was briefly unconfigure the SRST MOH and reboot the router so it wasn't accessing the flash all the time.
>
>
> ----- Original Message ----
> From: Matthew Saskin <matt at saskin.net>
> To: Jonathan Charles <jonvoip at gmail.com>
> Cc: ciscovoip <cisco-voip at puck.nether.net>; cisco-nsp at puck.nether.net
> Sent: Thursday, May 31, 2007 11:05:56 AM
> Subject: Re: [cisco-voip] Yanking Compact Flash while the router is up...
>
> Works fine in my experience - not to say it won't crash the router, just
> that I haven't seen it crash a router :)
>
> -matt
>
> Jonathan Charles wrote:
> > So, I am curious, can you pull a CF card from a 2821 while the router
> > is up and running?
> >
> > We need to replace flaky flash with a new one, but I would rather not
> > have to talk a remote tech (low skillset) through a ROMMON TFTP
> > operation..
> >
> > If not, we can do it the hard way....but I have heard rumors that it works...
> >
> > If it does, what are the risks...
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
>
> ____________________________________________________________________________________
> Sick sense of humor? Visit Yahoo! TV's
> Comedy with an Edge to see what's on, when.
> http://tv.yahoo.com/collections/222
>

From jonvoip at gmail.com  Thu May 31 13:30:37 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 31 May 2007 12:30:37 -0500
Subject: [cisco-voip] Error message while installing CCM 4.0 Subscriber
In-Reply-To: <200705311537.l4VFbu1F067315@puck.nether.net>
References: <200705311537.l4VFbu1F067315@puck.nether.net>
Message-ID: <5d093f9a0705311030l5a455b45w46ad0beea5721cec@mail.gmail.com>

Is the subscriber running the same version as the publisher?



Jonathan

On 5/31/07, PRASHANT K <prashant.k at tvisha.com> wrote:
>
>
>
>
> Hi all,
>
>
>
> This is the actual error message im getting while installing the subscriber:
> "THE INSTALLATION FAILED TO VERIFY THE CISCO CALLMANAGER VERSION THAT RUNS
> ON THE PUBLISHER DATABASE SERVER. CANCEL THE INSTALLTION, AND REVIEW THE LOG
> FILE AT C:\PROGRAM FILES\COMMON FILES\CISCO\LOGS\CCMUINST.LOG "
>
>
>
> 05/31/2007 16:23:03.140 MSIApi.cpp MSICheckPubVersion          Function
> Started
>
> 05/31/2007 16:23:03.156 MSIApi.cpp MSICheckPubVersion          Successful in
> setting property: PubVersionMatch to false
>
> 05/31/2007 16:23:03.156 MSIApi.cpp MSICheckPubVersion          Successful in
> getting property: PUBSVRNAME value: CCM1
>
> 05/31/2007 16:23:03.156 MSIApi.cpp MSICheckPubVersion          Successful in
> getting property: BUILDVERSION value: 4.0(1)
>
> 05/31/2007 16:23:03.218 MSIApi.cpp MSICheckPubVersion          Error failed
> to get registry key: SOFTWARE\Cisco Systems, Inc.\CallManager value: Build
>
> 05/31/2007 16:23:03.218 MSIApi.cpp MSICheckPubVersion          Error code :
> 1603
>
> 05/31/2007 16:23:03.218 MSIApi.cpp MSICheckPubVersion          Message
> displayed to user: The installation failed to verify the Cisco CallManager
> version that runs on the publisher database server.  Cancel the
> installation, and review the log file at C:\Program Files\Common
> Files\Cisco\Logs\CCMUIInst.log
>
>
>
> The Publisher is working fine. I checked the logs, registry, but im unable
> to solve this?
>
>
>
> Thanks and Regards,
>
> Prashant
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From jason.aarons at us.didata.com  Thu May 31 13:30:37 2007
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 31 May 2007 13:30:37 -0400
Subject: [cisco-voip] CCM 4.1 > CCMUser limit what services they can
	subscribe to?
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B8613A@USNAEXCH.na.didata.local>

Customer has 15 services configured in CCMAdmin > Feature > Cisco IP
Phone Services, but via CCMUser they only want the end-user to see about
6 services. Any way to limit this to the 6 services they want? Currently
CCMUser shows all 15 services.. I don't want to remove all access to
Services in CCMUser. There is some stuff we need end users to update
themselves.

 

 

Jason Aarons

Consultant

http://www.dimensiondata.com/na <http://www.dimensiondata.com/na> 

904-338-3245 cell

 

For urgent issues notify your Project Manager, for 24x7 support contact
the Dimension Data NOC at 800-974-6584.

 




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From axe7961 at hotmail.com  Thu May 31 13:39:56 2007
From: axe7961 at hotmail.com (Axel Wilzinski)
Date: Thu, 31 May 2007 13:39:56 -0400
Subject: [cisco-voip] Rebuilding a Subscriber after 4.2.(3) maintenance
	upgrade
Message-ID: <BAY103-W4FF6512490B13A65F633EBA2D0@phx.gbl>

We installed 4.2.(1) pub and sub and upgraded to 4.2.(3) via a maintenance release download.My question is, if my subscriber dies, what do I do to rebuild it?  I can't install 4.2.(1) from the CD's because it can't talk to my 4.2.(3) publisher.  Is this really the case?   Do I have to ask our Cisco rep if we can order 4.2.(3) without licenses?  This seems rather silly, especially if I have to do it after every maintenance release.
_________________________________________________________________
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From Matt.Slaga at us.didata.com  Thu May 31 13:44:43 2007
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 31 May 2007 13:44:43 -0400
Subject: [cisco-voip] Rebuilding a Subscriber after 4.2.(3)
	maintenanceupgrade
In-Reply-To: <BAY103-W4FF6512490B13A65F633EBA2D0@phx.gbl>
References: <BAY103-W4FF6512490B13A65F633EBA2D0@phx.gbl>
Message-ID: <C1FE15183DA37645BC0633BC604E44F005B86194@USNAEXCH.na.didata.local>

According to Cisco, you should only be upgrading versions if you have
SAU, otherwise there is an upgrade charge.  To get these CDs, go to
www.cisco.com/upgrade and enter your CCO number to order the 4.2(3)
discs.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Axel Wilzinski
Sent: Thursday, May 31, 2007 1:40 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Rebuilding a Subscriber after 4.2.(3)
maintenanceupgrade

 

We installed 4.2.(1) pub and sub and upgraded to 4.2.(3) via a
maintenance release download.

My question is, if my subscriber dies, what do I do to rebuild it?  I
can't install 4.2.(1) from the CD's because it can't talk to my 4.2.(3)
publisher.  Is this really the case?   Do I have to ask our Cisco rep if
we can order 4.2.(3) without licenses?  This seems rather silly,
especially if I have to do it after every maintenance release.

 

________________________________

Add some color. Personalize your inbox with your favorite colors. Try
it!




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From asobihoudai at yahoo.com  Thu May 31 14:05:25 2007
From: asobihoudai at yahoo.com (Paul Choi)
Date: Thu, 31 May 2007 11:05:25 -0700 (PDT)
Subject: [cisco-voip] CCM 4.1 > CCMUser limit what services they can
	subscribe to?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F005B8613A@USNAEXCH.na.didata.local>
Message-ID: <404826.33354.qm@web57203.mail.re3.yahoo.com>

I think it might be possible to do this by customizing
the web interface for certain users......direct them
to another URL and other users to the full access URL.

I'm not sure on this, I'm talking out of my nose here.

--- "Jason Aarons (US)" <jason.aarons at us.didata.com>
wrote:

> Customer has 15 services configured in CCMAdmin >
> Feature > Cisco IP
> Phone Services, but via CCMUser they only want the
> end-user to see about
> 6 services. Any way to limit this to the 6 services
> they want? Currently
> CCMUser shows all 15 services.. I don't want to
> remove all access to
> Services in CCMUser. There is some stuff we need end
> users to update
> themselves.
> 
>  
> 
>  
> 
> Jason Aarons
> 
> Consultant
> 
> http://www.dimensiondata.com/na
> <http://www.dimensiondata.com/na> 
> 
> 904-338-3245 cell
> 
>  
> 
> For urgent issues notify your Project Manager, for
> 24x7 support contact
> the Dimension Data NOC at 800-974-6584.
> 
>  
> 
> 
> 
> 
> -----------------------------------------
> Disclaimer:
> 
> This e-mail communication and any attachments may
> contain
> confidential and privileged information and is for
> use by the
> designated addressee(s) named above only.  If you
> are not the
> intended addressee, you are hereby notified that you
> have received
> this communication in error and that any use or
> reproduction of
> this email or its contents is strictly prohibited
> and may be
> unlawful.  If you have received this communication
> in error, please
> notify us immediately by replying to this message
> and deleting it
> from your computer. Thank you.>
_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



 
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From jeff at ocjtech.us  Thu May 31 15:08:32 2007
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Thu, 31 May 2007 14:08:32 -0500
Subject: [cisco-voip] kipvmsMixer01?
Message-ID: <1180638512.3904.9.camel@lt21223.campus.dmacc.edu>

Realtime monitoring on our 5.1.2 cluster started spitting out the
following messages.  I'm unable to find any information about what the
kipvmsMixer01 process is/does.  Can anyone shed some light on this
process?

        Processor load over 90 Percent.
        kipvmsMixer01 (50 percent) uses most of the CPU. 
         
        Processor_Info: 
        
        
         For processor instance 1: %CPU= 99, %User= 0, %System= 99, %Nice= 0, %Idle= 0, %IOWait= 0, %softirq= 0, %irq= 0. 
        
        
         For processor instance _Total: %CPU= 99, %User= 36, %System= 60, %Nice= 0, %Idle= 1, %IOWait= 1, %softirq= 3, %irq= 0. 
        
        
         For processor instance 0: %CPU= 98, %User= 72, %System= 20, %Nice= 0, %Idle= 2, %IOWait= 1, %softirq= 6, %irq= 0. 
        
        
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From tele at plexialab.it  Thu May 31 15:43:27 2007
From: tele at plexialab.it (tele)
Date: Thu, 31 May 2007 21:43:27 +0200
Subject: [cisco-voip] Media Gateway issue with NTE
In-Reply-To: <C2846355.2366%mark.turpin@calence.com>
References: <C2846355.2366%mark.turpin@calence.com>
Message-ID: <465F255F.2000206@plexialab.it>

yes.

!
mgcp
mgcp call-agent IP_MGC 2427 service-type mgcp version 1.0
mgcp dtmf-relay voip codec all mode nte-ca
mgcp codec clear-channel packetization-period 10
mgcp ip qos dscp cs3 signaling
mgcp package-capability dtmf-package
mgcp package-capability mf-package
mgcp package-capability rtp-package
mgcp default-package fxr-package
mgcp timer nse-response t38 250
mgcp fax t38 ls_redundancy 5
mgcp fax t38 hs_redundancy 2
mgcp rtp payload-type clear-channel static
!
mgcp profile default
!


Mark Turpin wrote:
> Can you paste your present MGCP config?
>
> Thanks,
> -Mark
>
>
> On 5/31/07 2:33 AM, "tele" <tele at plexialab.it> wrote:
>
>   
>> Hi all,
>>
>> I have a issue with a Cisco AS5400 and NTE, the problem is simple:
>>
>> the first CRCX:
>>
>> CRCX 122138580 S1/DS1-0/8 at mgmi03 MGCP 1.0
>> C:308
>> L:a:G729;PCMA;PCMU;telephone-event, fxr/fx:t38-loose,
>> fmtp:"telephone-event 0-15", p:20
>> M:recvonly
>> R:fxr/t38
>> X:1
>>
>> contains a fmtp with tone from 0 to 15, when a call goes in progress the
>> MGC send a MDCX to the MG.
>>
>> MDCX 122138581 S1/DS1-0/8 at mgmi03 MGCP 1.0
>> C:308
>> L:a:PCMA;telephone-event, fxr/fx:t38-loose, fmtp:"telephone-event 0-16",
>> p:20
>> I: 17703
>> M:sendrecv
>> R:fxr/t38
>> X:1
>>
>> v=0
>> o=- 1180529725 1180529725 IN IP4 myip
>> s=-
>> c=IN IP4 myip
>> t=0 0
>> m=audio 12388 RTP/AVP 8 101
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>>
>> As you can seen MGC want to use tone from 0-16, 16 is the "flash".
>> Now the mediagateway should be able to change the session but an error
>> occur during NTE negotiation:
>>
>> May 30 14:56:02.287 CEST: mgcp_modify_connection: calling
>> mgcp_select_nte
>> May 30 14:56:02.287 CEST: S1/DS1-0/8
>> <VOIP>mgcp_nte_select_rfc2833_dtmf_relay -IN: mode/codec = 5/0
>> May 30 14:56:02.287 CEST: S1/DS1-0/8
>> <VOIP>mgcp_nte_select_rfc2833_dtmf_relay -OUT: mode/codec = 5/0
>> May 30 14:56:02.287 CEST: mgcp_nte_select_payload: No NTE convergence:
>> local nte=(16)/(0000FFFF), sdp nte=(15)/(0000FFFF)
>> May 30 14:56:02.287 CEST: mgcp_modify_connection - Conflicting NTE
>> information
>>
>> and the MG return to the MGC a 502 error:
>>
>> May 30 14:56:02.287 CEST: 502 122138581 Cannot select NTE
>>
>> I've read about the MF package for the support of fmtp, i've tried
>> actived it "mgcp package-capability mf" but nothing change
>>
>> Can i do something on the MG to solve this issue? or i've to work on the
>> MGC side?
>>
>> thank you very much
>>
>> :tele
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>     
>
> --
>
> Mark Turpin
> Senior Consultant - Unified Communications
> Calence, LLC
> IP Phone: 314.656.2315
> Mobile: 314.629.8714
> www.calence.com
> 2005 Cisco Global Channel Partner of the Year ? US/Canada
>
> _________________________
>
>
> The information contained in this message and any attachment may contain
> privileged or confidential information protected from disclosure. If you are
> not the intended recipient, or an employee or agent responsible for
> delivering this message to the intended recipient, you are hereby notified
> that any review, dissemination, distribution or copying of this information
> is strictly prohibited.  If you have received this transmission in error,
> please notify the sender immediately by replying to this message and
> destroying the original and all copies.  Thank you.
>
>
>   


From tele at plexialab.it  Thu May 31 16:41:08 2007
From: tele at plexialab.it (tele)
Date: Thu, 31 May 2007 22:41:08 +0200
Subject: [cisco-voip] Media Gateway issue with NTE
In-Reply-To: <465F255F.2000206@plexialab.it>
References: <C2846355.2366%mark.turpin@calence.com>
	<465F255F.2000206@plexialab.it>
Message-ID: <465F32E4.7040608@plexialab.it>

some additional info about my issue:

The RFC3660 talks about the MF package

2.9 - Media Format Parameter Package

   Package Name: FM

The package is called FM.

In the Cisco media gateway i have only a "MF" package and not a "FM" 
package.
So i think there is no FM package for Cisco MGCP.

*mf*-*package*?MF package. Events and signals for multifrequency tones 
(MF) relay.

i don't know if with a FM package support i can solve my problem.

:tele


tele wrote:
> yes.
>
> !
> mgcp
> mgcp call-agent IP_MGC 2427 service-type mgcp version 1.0
> mgcp dtmf-relay voip codec all mode nte-ca
> mgcp codec clear-channel packetization-period 10
> mgcp ip qos dscp cs3 signaling
> mgcp package-capability dtmf-package
> mgcp package-capability mf-package
> mgcp package-capability rtp-package
> mgcp default-package fxr-package
> mgcp timer nse-response t38 250
> mgcp fax t38 ls_redundancy 5
> mgcp fax t38 hs_redundancy 2
> mgcp rtp payload-type clear-channel static
> !
> mgcp profile default
> !
>
>
> Mark Turpin wrote:
>   
>> Can you paste your present MGCP config?
>>
>> Thanks,
>> -Mark
>>
>>
>> On 5/31/07 2:33 AM, "tele" <tele at plexialab.it> wrote:
>>
>>   
>>     
>>> Hi all,
>>>
>>> I have a issue with a Cisco AS5400 and NTE, the problem is simple:
>>>
>>> the first CRCX:
>>>
>>> CRCX 122138580 S1/DS1-0/8 at mgmi03 MGCP 1.0
>>> C:308
>>> L:a:G729;PCMA;PCMU;telephone-event, fxr/fx:t38-loose,
>>> fmtp:"telephone-event 0-15", p:20
>>> M:recvonly
>>> R:fxr/t38
>>> X:1
>>>
>>> contains a fmtp with tone from 0 to 15, when a call goes in progress the
>>> MGC send a MDCX to the MG.
>>>
>>> MDCX 122138581 S1/DS1-0/8 at mgmi03 MGCP 1.0
>>> C:308
>>> L:a:PCMA;telephone-event, fxr/fx:t38-loose, fmtp:"telephone-event 0-16",
>>> p:20
>>> I: 17703
>>> M:sendrecv
>>> R:fxr/t38
>>> X:1
>>>
>>> v=0
>>> o=- 1180529725 1180529725 IN IP4 myip
>>> s=-
>>> c=IN IP4 myip
>>> t=0 0
>>> m=audio 12388 RTP/AVP 8 101
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>>
>>> As you can seen MGC want to use tone from 0-16, 16 is the "flash".
>>> Now the mediagateway should be able to change the session but an error
>>> occur during NTE negotiation:
>>>
>>> May 30 14:56:02.287 CEST: mgcp_modify_connection: calling
>>> mgcp_select_nte
>>> May 30 14:56:02.287 CEST: S1/DS1-0/8
>>> <VOIP>mgcp_nte_select_rfc2833_dtmf_relay -IN: mode/codec = 5/0
>>> May 30 14:56:02.287 CEST: S1/DS1-0/8
>>> <VOIP>mgcp_nte_select_rfc2833_dtmf_relay -OUT: mode/codec = 5/0
>>> May 30 14:56:02.287 CEST: mgcp_nte_select_payload: No NTE convergence:
>>> local nte=(16)/(0000FFFF), sdp nte=(15)/(0000FFFF)
>>> May 30 14:56:02.287 CEST: mgcp_modify_connection - Conflicting NTE
>>> information
>>>
>>> and the MG return to the MGC a 502 error:
>>>
>>> May 30 14:56:02.287 CEST: 502 122138581 Cannot select NTE
>>>
>>> I've read about the MF package for the support of fmtp, i've tried
>>> actived it "mgcp package-capability mf" but nothing change
>>>
>>> Can i do something on the MG to solve this issue? or i've to work on the
>>> MGC side?
>>>
>>> thank you very much
>>>
>>> :tele
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>     
>>>       
>> --
>>
>> Mark Turpin
>> Senior Consultant - Unified Communications
>> Calence, LLC
>> IP Phone: 314.656.2315
>> Mobile: 314.629.8714
>> www.calence.com
>> 2005 Cisco Global Channel Partner of the Year ? US/Canada
>>
>> _________________________
>>
>>
>> The information contained in this message and any attachment may contain
>> privileged or confidential information protected from disclosure. If you are
>> not the intended recipient, or an employee or agent responsible for
>> delivering this message to the intended recipient, you are hereby notified
>> that any review, dissemination, distribution or copying of this information
>> is strictly prohibited.  If you have received this transmission in error,
>> please notify the sender immediately by replying to this message and
>> destroying the original and all copies.  Thank you.
>>
>>
>>   
>>     
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From VLoschiavo at data-corporation.com  Thu May 31 17:08:52 2007
From: VLoschiavo at data-corporation.com (Vince Loschiavo)
Date: Thu, 31 May 2007 17:08:52 -0400
Subject: [cisco-voip] mcs-7816
In-Reply-To: <465F32E4.7040608@plexialab.it>
Message-ID: <CB4BC8C24EF6E149AED9FDB9B9AF5713416F7C@MAIL02.hq.data-corporation.com>

I know this has been discussed, but I can't seem to find the email or
the link....

Does anyone have a link to a CCO page that shows equivelant hardware for
the new 7816's.  

This is in preparation for CCM 6.0, Unity 5.0, and MPE 2.0.

Thanks,
V



____________________________________________________________
Vincent Loschiavo
Senior Consultant
Datacorp
18503 Pines Blvd, Suite 212, Pembroke Pines, FL, 33029
"Cisco Advanced Unified Communications Partner"
Cell Phone:        786-282-1164
Office Phone:    305-594-6933
____________________________________________________________ 

From jonvoip at gmail.com  Thu May 31 23:44:43 2007
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 31 May 2007 22:44:43 -0500
Subject: [cisco-voip] SQL query for CCM5.X to see 911 calls
Message-ID: <5d093f9a0705312044v75b386d6n1201efe201517520@mail.gmail.com>

Anyone have one? My current SQL query for this is for SQL 2000, not informix...



Jonathan

