From bababooey at cox.net  Thu Dec  1 00:56:00 2005
From: bababooey at cox.net (Bob A. Bowie)
Date: Thu Dec  1 00:56:15 2005
Subject: [cisco-voip] CME to CM 4.1 SCCP Call info - more questions
References: <000801c5f572$0ed38ff0$0f0110ac@corporation2>
	<63592da9c51ae913c95c43d9aefc1a3d@cisco.com>
Message-ID: <000a01c5f63b$e9dd6070$0f0110ac@corporation2>

If I initiate a conference from the Call Manager IP Phones to a two phones on the CME system,
does this result in two streams to the CME router or a single stream to the MTP?

Also, what if the conference is generated from the CME phone and conference phones on CME and the Call Manager
are there multiple streams?

Thanks

  ----- Original Message ----- 
  From: Kevin Thorngren 
  To: Bob A. Bowie 
  Cc: cisco-voip at puck.nether.net 
  Sent: Wednesday, November 30, 2005 11:01 AM
  Subject: Re: [cisco-voip] CME to CM 4.1 SCCP Call info


  Hi Bob,

  Yes, that is the expected behavior. A call between two phones on the same CME will have RTP streams directly between the two phones. If the call is to another destination through the CME then the CME is used as an MTP device for the call.

  Kevin
  On Nov 30, 2005, at 12:51 AM, Bob A. Bowie wrote:


    Hello, I have a remote site with CME 3.3 and my Call Manager Cluster 4.1.3 configured as H323 trunk
    through the WAN.  I have a question about call setup, as I am viewing the CME side with a packet analyzer,
    I see the CM IP phone stream to the CME router, and the CME router to the CME IP  Phone.
    QUestion: since these are both sccp phones on a WAN, I thought the media stream should be direct, once
    call setup is completed?  Is what I am seeing correct?
     
    Tx
     _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip
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From bababooey at cox.net  Thu Dec  1 01:33:10 2005
From: bababooey at cox.net (Bob A. Bowie)
Date: Thu Dec  1 01:33:22 2005
Subject: [cisco-voip] CME to CM 4.1 Conference Call info
References: <000801c5f572$0ed38ff0$0f0110ac@corporation2>
	<63592da9c51ae913c95c43d9aefc1a3d@cisco.com>
Message-ID: <001501c5f641$1ae639d0$0f0110ac@corporation2>

I have CME and CM4.1 installed.  When I create a conference from a CME IP Phone to other CME phones and CM IP Phones,
it works correctly.  However, when I conference from CM IP phones and conference in CME IP Phones, it connects, then immediately
drops the entire call and says "can not join calls".  Is there something onCall Manager I need to enable?  I am only using 2 CME phones and 1 CM IP Phone.

Thanks
  ----- Original Message ----- 
  From: Kevin Thorngren 
  To: Bob A. Bowie 
  Cc: cisco-voip at puck.nether.net 
  Sent: Wednesday, November 30, 2005 11:01 AM
  Subject: Re: [cisco-voip] CME to CM 4.1 SCCP Call info


  Hi Bob,

  Yes, that is the expected behavior. A call between two phones on the same CME will have RTP streams directly between the two phones. If the call is to another destination through the CME then the CME is used as an MTP device for the call.

  Kevin
  On Nov 30, 2005, at 12:51 AM, Bob A. Bowie wrote:


    Hello, I have a remote site with CME 3.3 and my Call Manager Cluster 4.1.3 configured as H323 trunk
    through the WAN.  I have a question about call setup, as I am viewing the CME side with a packet analyzer,
    I see the CM IP phone stream to the CME router, and the CME router to the CME IP  Phone.
    QUestion: since these are both sccp phones on a WAN, I thought the media stream should be direct, once
    call setup is completed?  Is what I am seeing correct?
     
    Tx
     _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip
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From kostas.anagnopoulos at oteglobe.net  Thu Dec  1 04:29:45 2005
From: kostas.anagnopoulos at oteglobe.net (kostas anagnopoulos)
Date: Thu Dec  1 04:30:40 2005
Subject: [cisco-voip] Sigtran sua question
Message-ID: <NJELIKCKHGFOCCAIFLJHAEIPEKAA.kostas.anagnopoulos@oteglobe.net>

Is it possible to have two Cisco ITPs running SUA 
between two SS7 STPs like below

SS7 STP <---tdm link---> ITP <---sua---> ITP <---tdm link---> SS7 STP 

thanks
Kostas





-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


From Ruben.Montes at eu.didata.com  Thu Dec  1 07:13:07 2005
From: Ruben.Montes at eu.didata.com (Ruben Montes)
Date: Thu Dec  1 07:13:11 2005
Subject: [cisco-voip] Callback over a Q.SIG
Message-ID: <D305A0848689344BB98F14A3F13161B710AFB0@eubebrusvex.eu.didata.local>

	Hello,

	 

	Does anybody know if it possible to do a callback between a CCM
4.1(3) and a PBX MD-110 with BC10?

	I suppose it may depend on the configuration of the QSIG in the
PBX, am I right?

	 

	Thanks,

	 

	Ruben

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From tgeorgiev at is-bg.net  Thu Dec  1 07:53:27 2005
From: tgeorgiev at is-bg.net (Teodor Georgiev)
Date: Thu Dec  1 07:52:52 2005
Subject: [cisco-voip] Callback over a Q.SIG
In-Reply-To: <D305A0848689344BB98F14A3F13161B710AFB0@eubebrusvex.eu.didata.local>
References: <D305A0848689344BB98F14A3F13161B710AFB0@eubebrusvex.eu.didata.local>
Message-ID: <200512011453.27168.tgeorgiev@is-bg.net>



 Correct. In general that is the major advantage of QSIG - a lot of 
supplementary services like CNIP, CCBS, CCNA and etc... 

MD110 supports both call-back on busy and call-back on no-answer with QSIG 
(ECMA) only.

Hint, at the CCM side, set the A/B-numbers to NPI = Unknown or Private, 
otherwise the callback might not work! You know, callback is logically to 
work only in a closed internal 'phone network' ;)

On Thursday 01 December 2005 14:13, Ruben Montes wrote:
> 	Hello,
>
>
>
> 	Does anybody know if it possible to do a callback between a CCM
> 4.1(3) and a PBX MD-110 with BC10?
>
> 	I suppose it may depend on the configuration of the QSIG in the
> PBX, am I right?
>
>
>
> 	Thanks,
>
>
>
> 	Ruben

From tgeorgiev at is-bg.net  Thu Dec  1 07:53:27 2005
From: tgeorgiev at is-bg.net (Teodor Georgiev)
Date: Thu Dec  1 07:52:52 2005
Subject: [cisco-voip] Callback over a Q.SIG
In-Reply-To: <D305A0848689344BB98F14A3F13161B710AFB0@eubebrusvex.eu.didata.local>
References: <D305A0848689344BB98F14A3F13161B710AFB0@eubebrusvex.eu.didata.local>
Message-ID: <200512011453.27168.tgeorgiev@is-bg.net>



 Correct. In general that is the major advantage of QSIG - a lot of 
supplementary services like CNIP, CCBS, CCNA and etc... 

MD110 supports both call-back on busy and call-back on no-answer with QSIG 
(ECMA) only.

Hint, at the CCM side, set the A/B-numbers to NPI = Unknown or Private, 
otherwise the callback might not work! You know, callback is logically to 
work only in a closed internal 'phone network' ;)

On Thursday 01 December 2005 14:13, Ruben Montes wrote:
> 	Hello,
>
>
>
> 	Does anybody know if it possible to do a callback between a CCM
> 4.1(3) and a PBX MD-110 with BC10?
>
> 	I suppose it may depend on the configuration of the QSIG in the
> PBX, am I right?
>
>
>
> 	Thanks,
>
>
>
> 	Ruben

From Ruben.Montes at eu.didata.com  Thu Dec  1 07:56:31 2005
From: Ruben.Montes at eu.didata.com (Ruben Montes)
Date: Thu Dec  1 07:56:39 2005
Subject: [cisco-voip] Callback over a Q.SIG
Message-ID: <D305A0848689344BB98F14A3F13161B710AFDB@eubebrusvex.eu.didata.local>

Hello,

What do you mean when you say "set the A/B-numbers to NPI = Unknown or
Private"?
How can I set this?

Regards,

Ruben
-----Mensaje original-----
De: Teodor Georgiev [mailto:tgeorgiev at is-bg.net] 
Enviado el: jueves, 01 de diciembre de 2005 13:53
Para: cisco-voip at puck.nether.net
CC: Ruben Montes; cisco-voip at puck-nether.net
Asunto: Re: [cisco-voip] Callback over a Q.SIG



 Correct. In general that is the major advantage of QSIG - a lot of 
supplementary services like CNIP, CCBS, CCNA and etc... 

MD110 supports both call-back on busy and call-back on no-answer with
QSIG 
(ECMA) only.

Hint, at the CCM side, set the A/B-numbers to NPI = Unknown or Private, 
otherwise the callback might not work! You know, callback is logically
to 
work only in a closed internal 'phone network' ;)

On Thursday 01 December 2005 14:13, Ruben Montes wrote:
> 	Hello,
>
>
>
> 	Does anybody know if it possible to do a callback between a CCM
> 4.1(3) and a PBX MD-110 with BC10?
>
> 	I suppose it may depend on the configuration of the QSIG in the
> PBX, am I right?
>
>
>
> 	Thanks,
>
>
>
> 	Ruben



From Brian.Henry at apptis.com  Thu Dec  1 07:59:46 2005
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu Dec  1 08:00:13 2005
Subject: [cisco-voip] Callback over a Q.SIG
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC03D75E@CH-EXCH-02.intelltech.apptis.inc>

Did you look at this link?
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080237271.shtml
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ruben Montes
Sent: Thu 12/1/2005 7:56 AM
To: tgeorgiev at is-bg.net; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Callback over a Q.SIG



Hello,

What do you mean when you say "set the A/B-numbers to NPI = Unknown or
Private"?
How can I set this?

Regards,

Ruben
-----Mensaje original-----
De: Teodor Georgiev [mailto:tgeorgiev at is-bg.net]
Enviado el: jueves, 01 de diciembre de 2005 13:53
Para: cisco-voip at puck.nether.net
CC: Ruben Montes; cisco-voip at puck-nether.net
Asunto: Re: [cisco-voip] Callback over a Q.SIG



 Correct. In general that is the major advantage of QSIG - a lot of
supplementary services like CNIP, CCBS, CCNA and etc...

MD110 supports both call-back on busy and call-back on no-answer with
QSIG
(ECMA) only.

Hint, at the CCM side, set the A/B-numbers to NPI = Unknown or Private,
otherwise the callback might not work! You know, callback is logically
to
work only in a closed internal 'phone network' ;)

On Thursday 01 December 2005 14:13, Ruben Montes wrote:
>       Hello,
>
>
>
>       Does anybody know if it possible to do a callback between a CCM
> 4.1(3) and a PBX MD-110 with BC10?
>
>       I suppose it may depend on the configuration of the QSIG in the
> PBX, am I right?
>
>
>
>       Thanks,
>
>
>
>       Ruben


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


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From retts at harman.com  Thu Dec  1 08:53:29 2005
From: retts at harman.com (Etts, Russell)
Date: Thu Dec  1 08:53:37 2005
Subject: [cisco-voip] H323 vs MGCP
Message-ID: <896DB57E2683B248936F5B50A197A624051AC2C8@hcgwdx2.hcg.ad.harman.com>

Thank you Jim 
 
I put everything back to H323 and everything is working now.  I
appreciate all the advice!
 
Russ

  _____  

From: Jim McBurnett [mailto:jim at tgasolutions.com] 
Sent: Wednesday, November 30, 2005 7:20 PM
To: Etts, Russell; Carter, Bill; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] H323 vs MGCP


Russell,
Check check very carefully what is and is not supported on the MGCP and
2801..
There are some VWIC combo's that require H323 on the 2801 with CCM...
 
J

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Etts, Russell
Sent: Wednesday, November 30, 2005 12:43 PM
To: Carter, Bill; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] H323 vs MGCP


Hi Bill
 
Thank you so much for your answer!  I just want to be clear on this - if
I use H323, I'm not hurting anything, but I get better monitoring with
MGCP.  The reason I'm asking is this:
 
We had an H323 enabled before and everything seemed to be working
properly.  I enabled MGCP and now nothing works - the calls just get
kicked back to the operator at the remote location.  I think this might
be a configuration issue with the remote PBX, but I can't be sure.
 
Thanks
 
Russ

  _____  

From: Carter, Bill [mailto:bcarter at sentinel.com] 
Sent: Wednesday, November 30, 2005 12:30 PM
To: Etts, Russell; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] H323 vs MGCP


Usually more feature available with H323 (CallerID), but MGCP is managed
via CallManager.  Because CCM manages calls to the MGCP gateway, you get
better monitoring statistics (current # of Calls, channels in use,
etc.).
 
My rule of thumb is, use MGCP when you can, H323 when you have to.


  _____  

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Etts, Russell
	Sent: Wednesday, November 30, 2005 11:21 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] H323 vs MGCP
	
	

	Hi everyone 

	I'm in the process of setting up a CCM (4.1) and 2801s.  What is
the advantage/ disadvantage between using a trunk between the CCM and
the 2801 (uses MGCP) and a regular H323 gateway?

	Thanks 
	
	Russ 



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From kthorngr at cisco.com  Thu Dec  1 08:59:24 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Dec  1 08:59:57 2005
Subject: [cisco-voip] CME to CM 4.1 SCCP Call info - more questions
In-Reply-To: <000a01c5f63b$e9dd6070$0f0110ac@corporation2>
References: <000801c5f572$0ed38ff0$0f0110ac@corporation2>
	<63592da9c51ae913c95c43d9aefc1a3d@cisco.com>
	<000a01c5f63b$e9dd6070$0f0110ac@corporation2>
Message-ID: <cfb75a517776a1cd94b968fccfcbd63e@cisco.com>

Hi Bob,

I haven't tried this, don't have it setup at the moment, but will try 
to walk through what happens.

CCM IP Phone (Phone A) conferences two CME IP Phones (Phone B and Phone 
C):
- Phone A and Phone B are connected
RTP Stream: Phone A------CME (MTP)-----Phone B
- Phone A hits Confrn and connects Phone C
RTP Stream: Phone A------CME (MTP)     Phone B
                                +-------Phone C
- Phone A hits Confrn again to connect B
RTP Stream: Phone A---CFB---CME (MTP)-----Phone B
                        +----CME (MTP)-----Phone C
For this case the CME will still be used as an MTP and will send 
individual streams to the CCM CFB.  From a CCM perspective the CFB 
needs to stream to each endpoint in the conference and CME is signaling 
the endpoint address as the MTP for each CME phone.

CME Phone B conferences Phone A and Phone C
- Phone A and Phone B are connected
RTP Stream: Phone A------CME (MTP)-----Phone B
- Phone B hits Confrn and connects Phone C
RTP Stream: Phone A      CME (MTP)     Phone B
                                           |
                                        Phone C
- Phone B hits Confrn again to connect Phone A
RTP Stream: Phone A------CME (CFB)-----Phone B
                                +-------Phone C
In this case the two CME phones will stream to the CME CFB (on UDP port 
2000 I believe).  The CME CFB will mix the streams and send one stream 
to the CCM phone.

Again, I don't have this setup to confirm but believe this is the way 
it will work.  You can web into the phones (http://<phone IP Address>) 
and look at the Stream 1 statistics to see where the phones are 
streaming to at any point in the call.

Please let me if you find anything different or have any questions.

Kevin
On Dec 1, 2005, at 12:56 AM, Bob A. Bowie wrote:

> If I initiate a conference from the Call Manager IP Phones?to a two 
> phones on the CME system,
> does this result in two streams to the CME router or a single stream 
> to the MTP?
> ?
> Also, what if the conference is generated from the CME phone and 
> conference phones on CME and the Call Manager
> are there multiple streams?
> ?
> Thanks
> ?
>> ----- Original Message -----
>> From: Kevin Thorngren
>> To: Bob A. Bowie
>> Cc: cisco-voip at puck.nether.net
>> Sent: Wednesday, November 30, 2005 11:01 AM
>> Subject: Re: [cisco-voip] CME to CM 4.1 SCCP Call info
>>
>> Hi Bob,
>>
>> Yes, that is the expected behavior. A call between two phones on the 
>> same CME will have RTP streams directly between the two phones. If 
>> the call is to another destination through the CME then the CME is 
>> used as an MTP device for the call.
>>
>> Kevin
>> On Nov 30, 2005, at 12:51 AM, Bob A. Bowie wrote:
>>
>>> Hello, I have a remote site with CME 3.3 and my Call Manager Cluster 
>>> 4.1.3 configured as H323 trunk
>>> through the WAN.? I have a question about call setup, as I am 
>>> viewing the CME side with a packet analyzer,
>>> I see the CM IP phone stream to the CME router, and the CME router 
>>> to the CME IP? Phone.
>>> QUestion: since these are both sccp phones on a WAN, I thought the 
>>> media stream should be direct, once
>>> call setup is completed?? Is what I am seeing?correct?
>>> ?
>>> Tx
>>> ?_______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
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From kthorngr at cisco.com  Thu Dec  1 09:20:45 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Dec  1 09:21:08 2005
Subject: [cisco-voip] Re: CME to CM 4.1 Conference Call info
In-Reply-To: <001501c5f641$1ae639d0$0f0110ac@corporation2>
References: <000801c5f572$0ed38ff0$0f0110ac@corporation2>
	<63592da9c51ae913c95c43d9aefc1a3d@cisco.com>
	<001501c5f641$1ae639d0$0f0110ac@corporation2>
Message-ID: <6342fc8450b51b7f6ae15d838f987bdd@cisco.com>

Hi Bob,

Did you select the "MTP Required" option on the H.323 Trunk in CCM?

If not then please read this URL for more information:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_implementation_design_guide_chapter09186a008044751b.html#wp1045 
089

Kevin
On Dec 1, 2005, at 1:33 AM, Bob A. Bowie wrote:

> I have CME and CM4.1 installed.? When I create a conference from a CME  
> IP Phone to other CME phones and CM IP Phones,
> it works correctly.? However, when I conference from CM IP phones and  
> conference in CME IP Phones, it connects, then immediately
> drops the entire call and says "can not join calls".? Is there  
> something onCall Manager I need to enable?? I am only using 2 CME  
> phones and 1 CM IP Phone.
> ?
> Thanks
>> ----- Original Message -----
>> From: Kevin Thorngren
>> To: Bob A. Bowie
>> Cc: cisco-voip at puck.nether.net
>> Sent: Wednesday, November 30, 2005 11:01 AM
>> Subject: Re: [cisco-voip] CME to CM 4.1 SCCP Call info
>>
>> Hi Bob,
>>
>> Yes, that is the expected behavior. A call between two phones on the  
>> same CME will have RTP streams directly between the two phones. If  
>> the call is to another destination through the CME then the CME is  
>> used as an MTP device for the call.
>>
>> Kevin
>> On Nov 30, 2005, at 12:51 AM, Bob A. Bowie wrote:
>>
>>> Hello, I have a remote site with CME 3.3 and my Call Manager Cluster  
>>> 4.1.3 configured as H323 trunk
>>> through the WAN.? I have a question about call setup, as I am  
>>> viewing the CME side with a packet analyzer,
>>> I see the CM IP phone stream to the CME router, and the CME router  
>>> to the CME IP? Phone.
>>> QUestion: since these are both sccp phones on a WAN, I thought the  
>>> media stream should be direct, once
>>> call setup is completed?? Is what I am seeing?correct?
>>> ?
>>> Tx
>>> ?_______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
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From balbler at nts.at  Thu Dec  1 09:49:58 2005
From: balbler at nts.at (Bernhard Albler)
Date: Thu Dec  1 09:50:08 2005
Subject: [cisco-voip] HTTP API for TCL IVR
Message-ID: <5A8C69748176434F92F1EECD7D21F72C353709@srvgraz03.nts.at>

Hi,

 

I know this isn't a List for Developer questions per se but anyway:
Does anyone have a example of using the tcl http api inside tcl ivr 2.0?
I'm somewhat at loss at how to use it (package imports etc.?).

Can't seem to find anything on CCO also.

Best regards

bernhard

 

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From zephy316 at gmail.com  Thu Dec  1 12:05:19 2005
From: zephy316 at gmail.com (Israel Lang)
Date: Thu Dec  1 12:05:22 2005
Subject: [cisco-voip] Another CME Question (or two)
Message-ID: <91e236d30512010905w20acfe47u3366f548c2b03fad@mail.gmail.com>

1) Anyway to create an intercom that autoanswers without the mute button
engaged? In other words I intercom Patty and she can directly respond
without turning the mute button off.

2) I have a few 7914 side carriages for the receptionists. Anyway to have a
single line both monitor a line and serve as a intercom line? That way
before the receptionist tries to intercom a person they can see whether they
are on the phone or not.

Thanks

--
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol
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From helder.guarda at gs.com  Thu Dec  1 14:17:22 2005
From: helder.guarda at gs.com (Guarda, Helder)
Date: Thu Dec  1 14:17:31 2005
Subject: [cisco-voip] Gatekeeper
Message-ID: <C3357D07220AB64DB09C205967E4B3A31F04A783@gsnmp16es.firmwide.corp.gs.com>

All:

Was wondering if anyone has implemented Gatekeeper between 2 CCM clusters.
I need to test this out and am looking for  a sample configuration.

Thanks,

Helder Guarda
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From tim.reimers at asheville.k12.nc.us  Thu Dec  1 14:54:31 2005
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Thu Dec  1 14:54:39 2005
Subject: [cisco-voip] Loading SIP firmware onto a 7940...
Message-ID: <050944EE48FF384F9C663F7048C47DA701237D70@webmail.asheville.k12.nc.us>

Hi -

 

I'm getting 'invalid load ID' trying to load a phone with a SIP image..

 

 

This is the current load on that phone - P00306000400

 

OS79XX.txt - contains the string "P0S307400"

 

The following files are in the 

C:\\Program Files\Cisco\TFTPPath

 

P0S307400.loads

P0S307400.sb2

P0S307400.bin

 

Originally they had the name P0S3-07-4-00.*

But upon reading somewhere that that didn't work, I went and changed
everything..

 

 

On the phone, in the 'load' field, I've tried putting "P0S307400.bin"
and "P0S307400"

To no avail....

 

What am I mis-doing???

 

Thanks, Tim

 

 

 

 

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From jonc at ftnc.net  Thu Dec  1 16:59:20 2005
From: jonc at ftnc.net (jonc)
Date: Thu Dec  1 16:59:36 2005
Subject: [cisco-voip] Loading SIP firmware onto a 7940...
In-Reply-To: <050944EE48FF384F9C663F7048C47DA701237D70@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA701237D70@webmail.asheville.k12.nc.us>
Message-ID: <1133474360.25057.23.camel@localhost.localdomain>

On Thu, 2005-12-01 at 14:54, Tim Reimers wrote:
> Hi ?
> 
>  
> 
> I?m getting ?invalid load ID? trying to load a phone with a SIP
> image..
> 
>  
> 
>  
> 
> This is the current load on that phone - P00306000400
> 
>  
> 
> OS79XX.txt ? contains the string ?P0S307400?
> 

Just for S&G's try creating a file in the TFTP directory called:
SIPDefault.cnf

Put the following line in the file: 
image_version: P0S307400


When I first started burning new/old images onto these Cisco phones it
seemed like a black box, but now I've seen just about every conceivable
combo that can exist on these buggers.  

Keep reading those docs, you'll get it loaded eventually... and good
luck!

Jon Carnes
FeatureTel 


From gary.d.brinkman.jr at census.gov  Thu Dec  1 17:16:15 2005
From: gary.d.brinkman.jr at census.gov (gary.d.brinkman.jr@census.gov)
Date: Thu Dec  1 17:16:28 2005
Subject: [cisco-voip] CDR question..
In-Reply-To: <050944EE48FF384F9C663F7048C47DA701237D50@webmail.asheville.k12.nc.us>
Message-ID: <OF6B73D7D3.B262AFEA-ON852570CA.0079D14D-852570CA.007A5674@census.gov>

Not sure about 4.0 but on 3.3 you can run this query against the CDR
database and get what you need.  Just change the originalCalledPartyNumber
to the number you need.  And while 1128142800 is equivalent to 10/1/2005
12:00 AM  EST you may need to get a Epoch time converter to get a different
time.


SELECT     duration
FROM         CallDetailRecord
WHERE     (dateTimeOrigination > '1128142800') and
(originalCalledPartyNumber like '%8005551212')

-----

Gary D. Brinkman
U.S. Census Bureau
Telecommunications Office



                                                                           
             "Tim Reimers"                                                 
             <tim.reimers at ashe                                             
             ville.k12.nc.us>                                           To 
             Sent by:                  <cisco-voip at puck.nether.net>        
             cisco-voip-bounce                                          cc 
             s at puck.nether.net                                             
                                                                   Subject 
                                       [cisco-voip] CDR question..         
             11/30/05 11:26 AM                                             
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           




Hi everyone---

I have a CDR newbie kind of question---

Can someone give me the steps needed to determine the following:  (CCM 4.0)

All calls to a given 800 number in say, the last two months?

Number of calls
Duration of each call

If I can get it into Excel I can munge around the data to get the right
things I want in terms of averages, etc?

But I?ve not done any CDR stuff as yet..
 _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jgrace at digitelusa.net  Thu Dec  1 18:13:46 2005
From: jgrace at digitelusa.net (James)
Date: Thu Dec  1 18:13:51 2005
Subject: [cisco-voip] Voice record
Message-ID: <1199305432-1133478823-cardhu_blackberry.rim.net-14434-@engine20-cell02>

Is there a out of box voice record feature in cm 4.1.3
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From dh4578 at sbc.com  Thu Dec  1 18:21:14 2005
From: dh4578 at sbc.com (HIERS, DAVID (AIT))
Date: Thu Dec  1 18:21:33 2005
Subject: [cisco-voip] Voice record
Message-ID: <07A5FFB8936EA0478953583E57591C0A0158D94B@CAFRFD1MSGUSR09.ITServices.sbc.com>

Nope.


David Hiers

CCIE 10734, CISSP

-###-

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net]On Behalf Of James
Sent: Thursday, December 01, 2005 3:14 PM
To: CiscosupportUpuck
Subject: [cisco-voip] Voice record


Is there a out of box voice record feature in cm 4.1.3
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From bandwidth.user at gmail.com  Fri Dec  2 00:35:30 2005
From: bandwidth.user at gmail.com (Roy)
Date: Fri Dec  2 02:29:26 2005
Subject: [cisco-voip] CCM/IPIPGW Device Config: phone vs. gateway
Message-ID: <1133501730.3543.106.camel@roy>

Hi Guys,

PSTN--AS5400--[IP Cloud]---IPIPGW--SJPhone
                  |
                 CCM


Would there be pros/cons if I'd configure the IPIPGW as a "phone" on the
CCM rather than as h.323 device?

Any thoughts?

TIA,

Roy


From tgeorgiev at is-bg.net  Fri Dec  2 03:59:47 2005
From: tgeorgiev at is-bg.net (Teodor Georgiev)
Date: Fri Dec  2 03:59:13 2005
Subject: [cisco-voip] Callback over a Q.SIG
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC03D75E@CH-EXCH-02.intelltech.apptis.inc>
References: <2F09B2162CCDEA4EA05C089B24FBC2BC03D75E@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <200512021059.48133.tgeorgiev@is-bg.net>


Does he have to?

This page speaks of something different - sending/receiving calling name 
(CNIP) in different manners =)

On Thursday 01 December 2005 14:59, Brian Henry wrote:
> Did you look at this link?
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0918
>6a0080237271.shtml
>
> Brian
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net on behalf of Ruben Montes
> Sent: Thu 12/1/2005 7:56 AM
> To: tgeorgiev at is-bg.net; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Callback over a Q.SIG
>
>
>
> Hello,
>
> What do you mean when you say "set the A/B-numbers to NPI = Unknown or
> Private"?
> How can I set this?
>
> Regards,
>
> Ruben
> -----Mensaje original-----
> De: Teodor Georgiev [mailto:tgeorgiev at is-bg.net]
> Enviado el: jueves, 01 de diciembre de 2005 13:53
> Para: cisco-voip at puck.nether.net
> CC: Ruben Montes; cisco-voip at puck-nether.net
> Asunto: Re: [cisco-voip] Callback over a Q.SIG
>
>
>
>  Correct. In general that is the major advantage of QSIG - a lot of
> supplementary services like CNIP, CCBS, CCNA and etc...
>
> MD110 supports both call-back on busy and call-back on no-answer with
> QSIG
> (ECMA) only.
>
> Hint, at the CCM side, set the A/B-numbers to NPI = Unknown or Private,
> otherwise the callback might not work! You know, callback is logically
> to
> work only in a closed internal 'phone network' ;)
>
> On Thursday 01 December 2005 14:13, Ruben Montes wrote:
> >       Hello,
> >
> >
> >
> >       Does anybody know if it possible to do a callback between a CCM
> > 4.1(3) and a PBX MD-110 with BC10?
> >
> >       I suppose it may depend on the configuration of the QSIG in the
> > PBX, am I right?
> >
> >
> >
> >       Thanks,
> >
> >
> >
> >       Ruben
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From Aaron.Derooy at xerox.com  Fri Dec  2 09:44:20 2005
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Fri Dec  2 09:45:07 2005
Subject: [cisco-voip] Voice Quality issue; Echo
Message-ID: <E0EF33751DEA634DA817A51024A8E3F205407577@usa0300ms01.na.xerox.net>

The customer is complaining of echo. If I change the echo canceller
coverage to 32ms and the ERL to 4db, what effect will that have? Will it
cause the echo canceller to take up to 3 seconds to converge or maybe
result in sentences being chopped off? Looking for any guidance on
making echo canceller coverage and/or erl changes.

Aaron 




From henri.herscher at gmail.com  Fri Dec  2 09:45:53 2005
From: henri.herscher at gmail.com (Henri Herscher)
Date: Fri Dec  2 09:45:56 2005
Subject: [cisco-voip] Voice record
In-Reply-To: <1199305432-1133478823-cardhu_blackberry.rim.net-14434-@engine20-cell02>
References: <1199305432-1133478823-cardhu_blackberry.rim.net-14434-@engine20-cell02>
Message-ID: <d2b7a1670512020645o6afc3445u@mail.gmail.com>

Hi James,

There isn't but you can use the open source VoIP recorder oreka to do
this: http://www.oreka.org (it supports Cisco Skinny VoIP protocol)
Oreka works by packet sniffing so if you can setup traffic mirroring
(SPAN) to a windows or Linux box, it'll do the job.

Cheers
Bruno

On 01/12/05, James <jgrace at digitelusa.net> wrote:
> Is there a out of box voice record feature in cm 4.1.3
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From highspeedsnow at gmail.com  Fri Dec  2 10:06:04 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Fri Dec  2 10:06:48 2005
Subject: [cisco-voip] Voice Quality issue; Echo
In-Reply-To: <E0EF33751DEA634DA817A51024A8E3F205407577@usa0300ms01.na.xerox.net>
Message-ID: <439062ff.6d317b98.0a62.ffffb432@mx.gmail.com>

Aaron,

An ERL plan must have at least a 6db difference between the Tx and Rx sides
of the ECAN to converge. Without this difference between the two, the ECAN
will never even engage ....

And to answer your question (and it's not the preferred answer I'd like to
give) but yes, an ECAN can take up to 3 seconds to fully converge, though a
lot of times it is much quicker than that ...

What version of code on what box are you running?
Thought of running the VIC2 with the Extended 128ms ECAN daughter card?


-Mark
CCIE Voice #14073



> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of DeRooy, Aaron H
> Sent: Friday, December 02, 2005 8:44 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Voice Quality issue; Echo
> 
> The customer is complaining of echo. If I change the echo canceller
> coverage to 32ms and the ERL to 4db, what effect will that have? Will it
> cause the echo canceller to take up to 3 seconds to converge or maybe
> result in sentences being chopped off? Looking for any guidance on
> making echo canceller coverage and/or erl changes.
> 
> Aaron
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From Aaron.Derooy at xerox.com  Fri Dec  2 10:40:46 2005
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Fri Dec  2 10:42:03 2005
Subject: [cisco-voip] SQL commands
Message-ID: <E0EF33751DEA634DA817A51024A8E3F2054075FA@usa0300ms01.na.xerox.net>

Is anybody familiar with DB commands that control SQL within Call
Manager, say version 3.3 or 4.0? I'm preparing for the IPTT exam and
want to understand how to resync or repair pub and sub db's that may be
out of sync. 

Aaron 




From rratliff at cisco.com  Fri Dec  2 11:06:20 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Dec  2 11:06:19 2005
Subject: [cisco-voip] SQL commands
In-Reply-To: <E0EF33751DEA634DA817A51024A8E3F2054075FA@usa0300ms01.na.xerox.net>
References: <E0EF33751DEA634DA817A51024A8E3F2054075FA@usa0300ms01.na.xerox.net>
Message-ID: <59968D2A-11C6-4C0C-8845-5FF037293249@cisco.com>

The manual process doesn't include any SQL "commands" though it is  
done via Enterprise Manager.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_tech_note09186a0080094aeb.shtml

The preferred way is to use DBLHelper.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_tech_note09186a00801e7ddf.shtml

-Ryan

On Dec 2, 2005, at 10:40 AM, DeRooy, Aaron H wrote:

Is anybody familiar with DB commands that control SQL within Call
Manager, say version 3.3 or 4.0? I'm preparing for the IPTT exam and
want to understand how to resync or repair pub and sub db's that may be
out of sync.

Aaron



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


From rratliff at cisco.com  Fri Dec  2 11:07:17 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Dec  2 11:07:15 2005
Subject: [cisco-voip] Voice Quality issue; Echo
In-Reply-To: <E0EF33751DEA634DA817A51024A8E3F205407577@usa0300ms01.na.xerox.net>
References: <E0EF33751DEA634DA817A51024A8E3F205407577@usa0300ms01.na.xerox.net>
Message-ID: <F01F619B-CDC7-4C4D-AF87-7380AE942EB1@cisco.com>

Check out http://www.cisco.com/en/US/tech/tk652/tk698/ 
technologies_tech_note09186a0080149a1f.shtml.

-Ryan

On Dec 2, 2005, at 9:44 AM, DeRooy, Aaron H wrote:

The customer is complaining of echo. If I change the echo canceller
coverage to 32ms and the ERL to 4db, what effect will that have? Will it
cause the echo canceller to take up to 3 seconds to converge or maybe
result in sentences being chopped off? Looking for any guidance on
making echo canceller coverage and/or erl changes.

Aaron



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


From jburton at NETechCorp.com  Fri Dec  2 13:58:42 2005
From: jburton at NETechCorp.com (Jason Burton)
Date: Fri Dec  2 13:59:01 2005
Subject: [cisco-voip] Voice record
Message-ID: <FDC137145729AD438CDCA0F9FC6F7236337ED6@mail.netechcorp.com>

Does it work with CME?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Henri Herscher
Sent: Friday, December 02, 2005 9:46 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Voice record

Hi James,

There isn't but you can use the open source VoIP recorder oreka to do
this: http://www.oreka.org (it supports Cisco Skinny VoIP protocol)
Oreka works by packet sniffing so if you can setup traffic mirroring
(SPAN) to a windows or Linux box, it'll do the job.

Cheers
Bruno

On 01/12/05, James <jgrace at digitelusa.net> wrote:
> Is there a out of box voice record feature in cm 4.1.3
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> 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 henri.herscher at gmail.com  Fri Dec  2 14:18:48 2005
From: henri.herscher at gmail.com (Henri Herscher)
Date: Fri Dec  2 14:18:51 2005
Subject: [cisco-voip] Voice record
In-Reply-To: <FDC137145729AD438CDCA0F9FC6F7236337ED6@mail.netechcorp.com>
References: <FDC137145729AD438CDCA0F9FC6F7236337ED6@mail.netechcorp.com>
Message-ID: <d2b7a1670512021118n18d33dfdp@mail.gmail.com>

Never tried it against CME specifically but I'm pretty sure that for simple
Skinny metadata extraction and RTP capturing there is no substantial
difference between CM and CME.

Henri

On 02/12/05, Jason Burton <jburton at netechcorp.com> wrote:
>
> Does it work with CME?
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Henri Herscher
> Sent: Friday, December 02, 2005 9:46 AM
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Voice record
>
> Hi James,
>
> There isn't but you can use the open source VoIP recorder oreka to do
> this: http://www.oreka.org (it supports Cisco Skinny VoIP protocol)
> Oreka works by packet sniffing so if you can setup traffic mirroring
> (SPAN) to a windows or Linux box, it'll do the job.
>
> Cheers
> Bruno
>
> On 01/12/05, James <jgrace at digitelusa.net> wrote:
> > Is there a out of box voice record feature in cm 4.1.3
> > James D. Grace
> > System Engineer
> > CCNP CCNA MCSE MCDBA
> > Digitel Corp.
> > _______________________________________________
> > 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  Fri Dec  2 14:44:48 2005
From: fwn at feasible.net (Fred Nielsen)
Date: Fri Dec  2 14:44:48 2005
Subject: [cisco-voip] Voice record
In-Reply-To: <FDC137145729AD438CDCA0F9FC6F7236337ED6@mail.netechcorp.com>
Message-ID: <005401c5f778$dbfdb2a0$6401a8c0@ws1>

It does indeed.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Friday, December 02, 2005 10:59 AM
To: Henri Herscher; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Voice record

Does it work with CME?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Henri Herscher
Sent: Friday, December 02, 2005 9:46 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Voice record

Hi James,

There isn't but you can use the open source VoIP recorder oreka to do
this: http://www.oreka.org (it supports Cisco Skinny VoIP protocol)
Oreka works by packet sniffing so if you can setup traffic mirroring
(SPAN) to a windows or Linux box, it'll do the job.

Cheers
Bruno

On 01/12/05, James <jgrace at digitelusa.net> wrote:
> Is there a out of box voice record feature in cm 4.1.3
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> 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



From erickbe at yahoo.com  Fri Dec  2 16:40:22 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Dec  2 16:40:30 2005
Subject: [cisco-voip] SDL Link OOS errors
In-Reply-To: <F01F619B-CDC7-4C4D-AF87-7380AE942EB1@cisco.com>
Message-ID: <20051202214022.79583.qmail@web34301.mail.mud.yahoo.com>

Hi, 

I am getting some SDL Link OOS errors once or twice a
day on a 3 node CCM cluster and the SDLs occur at same
time on each server. 1 server is at remote site across
a WAN (3 meg pipe). The network underneath is good and
routes are stable. No one is complaining or
experiencing problems other then this. 

When the SDL errors happen, the route lists also
unregister. What is weird about this unregistration is
that SiteA's routelist normally register to SiteA CCM
but when the SDL error happens there is no event log
error about it unregistering on SiteA. I see the
routelist unregister on the SiteB server though. The
same thing happens for other route list homed to the
other servers as primary. Wouldn't one see it
unregister from the primary CCM before it unregisters
from the secondary server in the callmanager-group
settings for the device pool? 

This is on 4.0(2a) SR1a


		
__________________________________ 
Start your day with Yahoo! - Make it your home page! 
http://www.yahoo.com/r/hs

From andy at dignans.com  Fri Dec  2 17:31:22 2005
From: andy at dignans.com (Andrew Dignan)
Date: Fri Dec  2 17:31:24 2005
Subject: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication for
	IP Phones
Message-ID: <60259.12.158.2.254.1133562682.squirrel@www.dignans.com>

I understand all the drawbacks/limitatations to setting this up but I was
just curious to see if anyone out there has deployed this and if its been
stable.  I am doing my best to convince the customer otherwise because it
just scares me that the fundamental operations of call setup/signalling
and registration is so different.  That being said I may still end up
having to deploy it and atleast if its being widely deployed and its
stable I will be able to sleep better.

TIA,

Andy

From it at cimgroup.com  Fri Dec  2 18:57:51 2005
From: it at cimgroup.com (IT)
Date: Fri Dec  2 18:54:00 2005
Subject: [cisco-voip] PRI Alternative
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02575FB2@exchange.cimgroup.com>

Hi,

I have a CallManager 4.0 installation at my headquarters office, and I
have a branch office with a 2801 connected to HQ with a point to point
T1. Currently, we have filled all 4 slots on the 2801.
T1- for point to point connectivity
FXO- for outbound 911 calling (we do not have a E911 server)
9-port POE switch- for all devices (which takes up two slots)

This has worked well, as have been call forwarding all their local
numbers to our Headquarters PRI.  Since they are in a different area
code, this has been quite expensive. We don't want to install a PRI on
the 2801, since it seems like to much of an expense for only 6 people. I
think the solution is to have an FXS and FXO in this 2801.  But, here
are my questions:
* I have been told an FXS card cannot handle caller-id, is this true?
* Is there an FXO/FXS on a single card, since I have run out of room on
my 2801 chassis?
* How many lines would an office of 6 people require?
* Does anyone have any other ideas?

Thanks,
Avidan

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 



From pwalenta at wi.rr.com  Fri Dec  2 19:16:43 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri Dec  2 19:16:40 2005
Subject: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication
	for	IP Phones
In-Reply-To: <60259.12.158.2.254.1133562682.squirrel@www.dignans.com>
Message-ID: <200512030013.jB30Dldu016667@ms-smtp-02.rdc-kc.rr.com>

Well, let's hope this customer isn't going to run IPCC (either flavor) as
neither support media encryption.

Other than that I've deployed it for one small customer (about 50 phones)
and I didn't see any real hassle with it.   

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
Sent: Friday, December 02, 2005 4:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication for
IP Phones

I understand all the drawbacks/limitatations to setting this up but I was
just curious to see if anyone out there has deployed this and if its been
stable.  I am doing my best to convince the customer otherwise because it
just scares me that the fundamental operations of call setup/signalling and
registration is so different.  That being said I may still end up having to
deploy it and atleast if its being widely deployed and its stable I will be
able to sleep better.

TIA,

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


From ml at nesys.it  Fri Dec  2 20:52:46 2005
From: ml at nesys.it (Andrea Riela)
Date: Fri Dec  2 20:52:52 2005
Subject: [cisco-voip] Voice record
In-Reply-To: <005401c5f778$dbfdb2a0$6401a8c0@ws1>
References: <005401c5f778$dbfdb2a0$6401a8c0@ws1>
Message-ID: <caa9e13f7e626464bdcbe8e05d661ee3@nesys.it>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

I'm interested ... how does it work?
It's a sniffer? when could I use it if all phones have dynamic ip?
I've already seen the doc, but maybe a practical experience will be 
better ...

Thanks for your support
Andrea
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (Darwin)

iD8DBQFDkPpuMakHrsrHP9wRAn/SAKCi+/VAlu+VeQrQMbkWjujVPfRiQACfaq2N
9Klcl+NiLXpkGM3K4mjfYdc=
=txqs
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From henri.herscher at gmail.com  Fri Dec  2 21:35:46 2005
From: henri.herscher at gmail.com (Henri Herscher)
Date: Fri Dec  2 21:35:49 2005
Subject: [cisco-voip] Voice record
In-Reply-To: <caa9e13f7e626464bdcbe8e05d661ee3@nesys.it>
References: <005401c5f778$dbfdb2a0$6401a8c0@ws1>
	<caa9e13f7e626464bdcbe8e05d661ee3@nesys.it>
Message-ID: <d2b7a1670512021835k9289d9fv@mail.gmail.com>

Hi Andrea,

Yes, it's a sniffer that captures all RTP packets for voice and all Skinny
packets for call control traffic (or SIP for that matter). Having dynamic
IPs does not hurt because ultimately, the call will be tagged with local
party and remote party (i.e. extension numbers) extracted from the skinny
messages. As long as the IP stays the same during the duration of a call,
we're fine (which has to be the case).

If you give it a try, please let me know how it goes.

Thanks
Henri

On 02/12/05, Andrea Riela <ml at nesys.it> wrote:
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> I'm interested ... how does it work?
> It's a sniffer? when could I use it if all phones have dynamic ip?
> I've already seen the doc, but maybe a practical experience will be
> better ...
>
> Thanks for your support
> Andrea
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.1 (Darwin)
>
> iD8DBQFDkPpuMakHrsrHP9wRAn/SAKCi+/VAlu+VeQrQMbkWjujVPfRiQACfaq2N
> 9Klcl+NiLXpkGM3K4mjfYdc=
> =txqs
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From andy at dignans.com  Fri Dec  2 23:14:12 2005
From: andy at dignans.com (Andrew Dignan)
Date: Fri Dec  2 23:14:15 2005
Subject: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication 
	for	IP Phones
In-Reply-To: <200512030013.jB30Dldu016667@ms-smtp-02.rdc-kc.rr.com>
References: <60259.12.158.2.254.1133562682.squirrel@www.dignans.com>
	<200512030013.jB30Dldu016667@ms-smtp-02.rdc-kc.rr.com>
Message-ID: <1862.64.81.226.84.1133583252.squirrel@www.dignans.com>

They aren't and understand that CTI Apps do not support it.  However,
wouldn't you be able to configure the Agent phones as "non-secure" and be
able to use encryption but just not to agent phones?

> Well, let's hope this customer isn't going to run IPCC (either flavor) as
> neither support media encryption.
>
> Other than that I've deployed it for one small customer (about 50 phones)
> and I didn't see any real hassle with it.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
> Sent: Friday, December 02, 2005 4:31 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication for
> IP Phones
>
> I understand all the drawbacks/limitatations to setting this up but I was
> just curious to see if anyone out there has deployed this and if its been
> stable.  I am doing my best to convince the customer otherwise because it
> just scares me that the fundamental operations of call setup/signalling
> and
> registration is so different.  That being said I may still end up having
> to
> deploy it and atleast if its being widely deployed and its stable I will
> be
> able to sleep better.
>
> TIA,
>
> Andy
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


From bababooey at cox.net  Sat Dec  3 00:27:16 2005
From: bababooey at cox.net (Bob A. Bowie)
Date: Sat Dec  3 00:27:27 2005
Subject: [cisco-voip] Cisco CME conference limits
Message-ID: <000901c5f7ca$3aa8e990$0f0110ac@corporation2>

Is the phones on a cme system limited to 3 party conferference?

Do I need dsp resources on a cme system that is on the ip wan? or are they only needed to transcode pstn lines?

thanks

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From lelio at uoguelph.ca  Sat Dec  3 00:39:44 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat Dec  3 00:39:54 2005
Subject: [cisco-voip] CDR question..
References: <OF6B73D7D3.B262AFEA-ON852570CA.0079D14D-852570CA.007A5674@census.gov>
Message-ID: <01fa01c5f7cb$f73e1570$ca01a8c0@deadfish>

I use this tiny awk script:

[hostname]$ more cdate
#!/bin/sh
echo $1 | /bin/awk '{ print strftime("%m/%d/%Y %H:%M:%S",$1), $2, $3, $4, $5, $6}'

and I call it with "cdate epochtime"


  ----- Original Message ----- 
  From: gary.d.brinkman.jr at census.gov 
  To: Tim Reimers 
  Cc: cisco-voip-bounces at puck.nether.net ; cisco-voip at puck.nether.net 
  Sent: Thursday, December 01, 2005 5:16 PM
  Subject: Re: [cisco-voip] CDR question..


  Not sure about 4.0 but on 3.3 you can run this query against the CDR
  database and get what you need.  Just change the originalCalledPartyNumber
  to the number you need.  And while 1128142800 is equivalent to 10/1/2005
  12:00 AM  EST you may need to get a Epoch time converter to get a different
  time.


  SELECT     duration
  FROM         CallDetailRecord
  WHERE     (dateTimeOrigination > '1128142800') and
  (originalCalledPartyNumber like '%8005551212')

  -----

  Gary D. Brinkman
  U.S. Census Bureau
  Telecommunications Office



                                                                             
               "Tim Reimers"                                                 
               <tim.reimers at ashe                                             
               ville.k12.nc.us>                                           To 
               Sent by:                  <cisco-voip at puck.nether.net>        
               cisco-voip-bounce                                          cc 
               s at puck.nether.net                                             
                                                                     Subject 
                                         [cisco-voip] CDR question..         
               11/30/05 11:26 AM                                             
                                                                             
                                                                             
                                                                             
                                                                             
                                                                             




  Hi everyone---

  I have a CDR newbie kind of question---

  Can someone give me the steps needed to determine the following:  (CCM 4.0)

  All calls to a given 800 number in say, the last two months?

  Number of calls
  Duration of each call

  If I can get it into Excel I can munge around the data to get the right
  things I want in terms of averages, etc?

  But I?ve not done any CDR stuff as yet..
   _______________________________________________
  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 pwalenta at wi.rr.com  Sat Dec  3 02:22:28 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Sat Dec  3 02:22:25 2005
Subject: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication
	forIP Phones
In-Reply-To: <1862.64.81.226.84.1133583252.squirrel@www.dignans.com>
Message-ID: <200512030722.jB37MI9u008424@ms-smtp-04.rdc-kc.rr.com>

Well, there's also the gateways to consider, and if memory serves (I could
be wrong), if security is on, it's on for gateways globally.  There's always
mixed mode so secure and non-secure devices can both register, but I've not
played with it so I cannot vouch for how well it works :-)

-----Original Message-----
From: Andrew Dignan [mailto:andy at dignans.com] 
Sent: Friday, December 02, 2005 10:14 PM
To: Philip Walenta
Cc: andy at dignans.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication
forIP Phones

They aren't and understand that CTI Apps do not support it.  However,
wouldn't you be able to configure the Agent phones as "non-secure" and be
able to use encryption but just not to agent phones?

> Well, let's hope this customer isn't going to run IPCC (either flavor) 
> as neither support media encryption.
>
> Other than that I've deployed it for one small customer (about 50 
> phones) and I didn't see any real hassle with it.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
> Sent: Friday, December 02, 2005 4:31 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication 
> for IP Phones
>
> I understand all the drawbacks/limitatations to setting this up but I 
> was just curious to see if anyone out there has deployed this and if 
> its been stable.  I am doing my best to convince the customer 
> otherwise because it just scares me that the fundamental operations of 
> call setup/signalling and registration is so different.  That being 
> said I may still end up having to deploy it and atleast if its being 
> widely deployed and its stable I will be able to sleep better.
>
> TIA,
>
> Andy
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


From highspeedsnow at gmail.com  Sat Dec  3 11:12:13 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sat Dec  3 11:12:49 2005
Subject: [cisco-voip] Cisco CME conference limits
In-Reply-To: <000901c5f7ca$3aa8e990$0f0110ac@corporation2>
Message-ID: <4391c3ff.31912b65.39c9.6010@mx.gmail.com>

The phones are limited to a 3 party conference if you are using the CME
software to enable the conferencing.

Conference Express is also available for your use on a CME system, enabling
you conference ID's and more members in the conf bridge.

 

-Mark Snow

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bob A. Bowie
Sent: Friday, December 02, 2005 11:27 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco CME conference limits

 

Is the phones on a cme system limited to 3 party conferference?

 

Do I need dsp resources on a cme system that is on the ip wan? or are they
only needed to transcode pstn lines?

 

thanks

 

 

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From lelio at uoguelph.ca  Sat Dec  3 12:00:25 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat Dec  3 12:00:34 2005
Subject: [cisco-voip] CDR question..
References: <050944EE48FF384F9C663F7048C47DA70117DCD7@webmail.asheville.k12.nc.us>
Message-ID: <00c401c5f82b$0e4d0d20$c901a8c0@EATIUSBIRDIUS>

I should have thought out my response a little more clearly. That converter I use is a command line converter when I'm doing work and troubleshooting. I use a similar awk statement when I'm doing my CDR processing to save the epoch date for our admin staff so they CDR records have a human readable format.

As far as the SQL query goes, we run one every day at 12:15am using a scheduled job and use the following query. This gives us everything from the time the job is run to two weeks prior.

SELECT [CallDetailRecord].[dateTimeOrigination], 
       [CallDetailRecord].[callingPartyNumber], 
       [CallDetailRecord].[originalCalledPartyNumber], 
       [CallDetailRecord].[finalCalledPartyNumber], 
       [CallDetailRecord].[duration] 
FROM   [CallDetailRecord] 
WHERE  (dateTimeOrigination > DATEDIFF([SECOND], '1/1/1970', GETDATE()) - 1209600)




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


From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Sat 12/3/2005 12:39 AM
To: Tim Reimers; gary.d.brinkman.jr at census.gov
Cc: cisco-voip-bounces at puck.nether.net; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR question..



  I use this tiny awk script:

  [hostname]$ more cdate
  #!/bin/sh
  echo $1 | /bin/awk '{ print strftime("%m/%d/%Y %H:%M:%S",$1), $2, $3, $4, $5, $6}'

  and I call it with "cdate epochtime"


    ----- Original Message ----- 
    From: gary.d.brinkman.jr at census.gov 
    To: Tim Reimers 
    Cc: cisco-voip-bounces at puck.nether.net ; cisco-voip at puck.nether.net 
    Sent: Thursday, December 01, 2005 5:16 PM
    Subject: Re: [cisco-voip] CDR question..


    Not sure about 4.0 but on 3.3 you can run this query against the CDR
    database and get what you need.  Just change the originalCalledPartyNumber
    to the number you need.  And while 1128142800 is equivalent to 10/1/2005
    12:00 AM  EST you may need to get a Epoch time converter to get a different
    time.


    SELECT     duration
    FROM         CallDetailRecord
    WHERE     (dateTimeOrigination > '1128142800') and
    (originalCalledPartyNumber like '%8005551212')

    -----

    Gary D. Brinkman
    U.S. Census Bureau
    Telecommunications Office



                                                                               
                 "Tim Reimers"                                                 
                 <tim.reimers at ashe                                             
                 ville.k12.nc.us>                                           To 
                 Sent by:                  <cisco-voip at puck.nether.net>        
                 cisco-voip-bounce                                          cc 
                 s at puck.nether.net                                             
                                                                       Subject 
                                           [cisco-voip] CDR question..         
                 11/30/05 11:26 AM                                             
                                                                               
                                                                               
                                                                               
                                                                               
                                                                               




    Hi everyone---

    I have a CDR newbie kind of question---

    Can someone give me the steps needed to determine the following:  (CCM 4.0)

    All calls to a given 800 number in say, the last two months.

    Number of calls
    Duration of each call

    If I can get it into Excel I can munge around the data to get the right
    things I want in terms of averages, etc.

    But I've not done any CDR stuff as yet..
     _______________________________________________
    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 bababooey at cox.net  Sat Dec  3 13:21:16 2005
From: bababooey at cox.net (Bob A. Bowie)
Date: Sat Dec  3 13:21:27 2005
Subject: [cisco-voip] Cisco CME conference limits
References: <4391c3ff.31912b65.39c9.6010@mx.gmail.com>
Message-ID: <000901c5f836$5b34b330$0f0110ac@corporation2>

Mark, Do you have a link to 'Conference Express' desdcription?
Is it a feature under the IOS or a separate hardware solution?
  ----- Original Message ----- 
  From: Mark Snow 
  To: 'Bob A. Bowie' ; cisco-voip at puck.nether.net 
  Sent: Saturday, December 03, 2005 11:12 AM
  Subject: RE: [cisco-voip] Cisco CME conference limits


  The phones are limited to a 3 party conference if you are using the CME software to enable the conferencing.

  Conference Express is also available for your use on a CME system, enabling you conference ID's and more members in the conf bridge.

   

  -Mark Snow

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bob A. Bowie
  Sent: Friday, December 02, 2005 11:27 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Cisco CME conference limits

   

  Is the phones on a cme system limited to 3 party conferference?

   

  Do I need dsp resources on a cme system that is on the ip wan? or are they only needed to transcode pstn lines?

   

  thanks

   

   
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From highspeedsnow at gmail.com  Sat Dec  3 22:14:41 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sat Dec  3 22:14:44 2005
Subject: [cisco-voip] Cisco CME conference limits
In-Reply-To: <000901c5f836$5b34b330$0f0110ac@corporation2>
References: <4391c3ff.31912b65.39c9.6010@mx.gmail.com>
	<000901c5f836$5b34b330$0f0110ac@corporation2>
Message-ID: <367fad90512031914v65e60707g8e11cea339d39a7c@mail.gmail.com>

Bob, I am not on my primary computer right now, but I will email you the
document on how to set this up, tomorrow.


On 12/3/05, Bob A. Bowie <bababooey at cox.net> wrote:
>
> Mark, Do you have a link to 'Conference Express' desdcription?
> Is it a feature under the IOS or a separate hardware solution?
>
> ----- Original Message -----
> *From:* Mark Snow <highspeedsnow at gmail.com>
> *To:* 'Bob A. Bowie' <bababooey at cox.net> ; cisco-voip at puck.nether.net
> *Sent:* Saturday, December 03, 2005 11:12 AM
> *Subject:* RE: [cisco-voip] Cisco CME conference limits
>
>  The phones are limited to a 3 party conference if you are using the CME
> software to enable the conferencing.
>
> Conference Express is also available for your use on a CME system,
> enabling you conference ID's and more members in the conf bridge.
>
>
>
> -Mark Snow
>
>
>   ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Bob A. Bowie
> *Sent:* Friday, December 02, 2005 11:27 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Cisco CME conference limits
>
>
>
> Is the phones on a cme system limited to 3 party conferference?
>
>
>
> Do I need dsp resources on a cme system that is on the ip wan? or are they
> only needed to transcode pstn lines?
>
>
>
> thanks
>
>
>
>
>
>


--
Snow
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From andy at dignans.com  Sat Dec  3 22:54:13 2005
From: andy at dignans.com (Andrew Dignan)
Date: Sat Dec  3 22:54:16 2005
Subject: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication 
	forIP Phones
In-Reply-To: <200512030722.jB37MI9u008424@ms-smtp-04.rdc-kc.rr.com>
References: <1862.64.81.226.84.1133583252.squirrel@www.dignans.com>
	<200512030722.jB37MI9u008424@ms-smtp-04.rdc-kc.rr.com>
Message-ID: <1663.64.81.226.84.1133668453.squirrel@www.dignans.com>

The install guide says that you need two Security Tokens (USB Keys) the
first time you configure the CTL Client.  If I want to install multiple
CTL Clients in a cluster, do I need 2 per client or can I use the same two
usb keys for each CTL Client?  If I do only have one CTL client (if its
installed on the publisher for example) what issues will I run into if the
CTL Client server is offline?  Will reset phones not be able to register
to the subscriber?  What other issues might I run into?

tia,

andy

> Well, there's also the gateways to consider, and if memory serves (I could
> be wrong), if security is on, it's on for gateways globally.  There's
> always
> mixed mode so secure and non-secure devices can both register, but I've
> not
> played with it so I cannot vouch for how well it works :-)
>
> -----Original Message-----
> From: Andrew Dignan [mailto:andy at dignans.com]
> Sent: Friday, December 02, 2005 10:14 PM
> To: Philip Walenta
> Cc: andy at dignans.com; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication
> forIP Phones
>
> They aren't and understand that CTI Apps do not support it.  However,
> wouldn't you be able to configure the Agent phones as "non-secure" and be
> able to use encryption but just not to agent phones?
>
>> Well, let's hope this customer isn't going to run IPCC (either flavor)
>> as neither support media encryption.
>>
>> Other than that I've deployed it for one small customer (about 50
>> phones) and I didn't see any real hassle with it.
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
>> Sent: Friday, December 02, 2005 4:31 PM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] CallManager 4.1(3) and Encryption/Authentication
>> for IP Phones
>>
>> I understand all the drawbacks/limitatations to setting this up but I
>> was just curious to see if anyone out there has deployed this and if
>> its been stable.  I am doing my best to convince the customer
>> otherwise because it just scares me that the fundamental operations of
>> call setup/signalling and registration is so different.  That being
>> said I may still end up having to deploy it and atleast if its being
>> widely deployed and its stable I will be able to sleep better.
>>
>> TIA,
>>
>> Andy
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>


From nkhwaja at gmail.com  Sun Dec  4 00:43:37 2005
From: nkhwaja at gmail.com (Nasser Khwaja)
Date: Sun Dec  4 00:43:39 2005
Subject: [cisco-voip] How to change Exchange server for Unity 4.0(3)?
Message-ID: <23435d3d0512032143x16c7affcrcf16c76776b98559@mail.gmail.com>

Hi,
I need to change My Unity 4.0(3) SR 1server that is currently pointing to an
Exchange 2000 server SP3 to another Exchange 2000 server SP3 in the same
existing domain. What is the way to accomplish this?
Thanks
-Tennisman.
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From voipforum at apptis.com  Sun Dec  4 01:51:46 2005
From: voipforum at apptis.com (VoIP Forum)
Date: Sun Dec  4 01:52:04 2005
Subject: [cisco-voip] How to change Exchange server for Unity 4.0(3)?
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BCF05D9C@CH-EXCH-02.intelltech.apptis.inc>

Follow this link...

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_upgrade_g
uides_chapter09186a0080205a7f.html#wp1126338

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nasser Khwaja
Sent: Sunday, December 04, 2005 12:44 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How to change Exchange server for Unity 4.0(3)?

 

Hi,
I need to change My Unity 4.0(3) SR 1server that is currently pointing
to an Exchange 2000 server SP3 to another Exchange 2000 server SP3 in
the same existing domain. What is the way to accomplish this?
Thanks
-Tennisman.

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From SPackett at fenwick.com  Sun Dec  4 02:44:17 2005
From: SPackett at fenwick.com (Stu Packett)
Date: Sun Dec  4 02:44:27 2005
Subject: [cisco-voip] CDR question..
Message-ID: <965C0255FD5828438E34F9A68ECA76E7233E18@svcexch02.firm.fenwick.llp>

I do similar CDR queries as well when someone inadvertantly dials 911.
I have an Excel spreadsheet that makes it easier for me.  The date that
the CDR reports is the number of seconds (or minutes?) from January 1,
1970.  So I have a field in Excel that says:  =A1/86400+DATE(1970,1,1)
that converts the numbers into dates.  If this isn't that clear, I can
send my CDR template for Excel that I always use.
 
The CDR query I would run in SQL Query Analzyer is:
select *
from CallDetailRecord
where finalCalledPartyNumber LIKE '91800%'
 
Hope this helps.


________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, November 30, 2005 8:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CDR question..



Hi everyone---

 

I have a CDR newbie kind of question---

 

Can someone give me the steps needed to determine the following:  (CCM
4.0)

 

All calls to a given 800 number in say, the last two months...

 

Number of calls

Duration of each call

 

If I can get it into Excel I can munge around the data to get the right
things I want in terms of averages, etc...

 

But I've not done any CDR stuff as yet..
--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
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From SPackett at fenwick.com  Sun Dec  4 02:47:23 2005
From: SPackett at fenwick.com (Stu Packett)
Date: Sun Dec  4 02:47:33 2005
Subject: [cisco-voip] CDR question..
Message-ID: <965C0255FD5828438E34F9A68ECA76E7233E19@svcexch02.firm.fenwick.llp>

I just read Lelio's email after I sent my reply to Tim.  Lelio's SQL
query is actually something I would prefer to run.  It converts it in
the query which is one less step than what I had replied to.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Saturday, December 03, 2005 9:00 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR question..


I should have thought out my response a little more clearly. That
converter I use is a command line converter when I'm doing work and
troubleshooting. I use a similar awk statement when I'm doing my CDR
processing to save the epoch date for our admin staff so they CDR
records have a human readable format.
 
As far as the SQL query goes, we run one every day at 12:15am using a
scheduled job and use the following query. This gives us everything from
the time the job is run to two weeks prior.
 

SELECT [CallDetailRecord].[dateTimeOrigination], 
       [CallDetailRecord].[callingPartyNumber], 
       [CallDetailRecord].[originalCalledPartyNumber], 
       [CallDetailRecord].[finalCalledPartyNumber], 
       [CallDetailRecord].[duration] 
FROM   [CallDetailRecord] 
WHERE  (dateTimeOrigination > DATEDIFF([SECOND], '1/1/1970', GETDATE())
- 1209600)

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Sat 12/3/2005 12:39 AM
To: Tim Reimers; gary.d.brinkman.jr at census.gov
Cc: cisco-voip-bounces at puck.nether.net; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR question..



	I use this tiny awk script:
	 
	[hostname]$ more cdate
	#!/bin/sh
	echo $1 | /bin/awk '{ print strftime("%m/%d/%Y %H:%M:%S",$1),
$2, $3, $4, $5, $6}'
	
	and I call it with "cdate epochtime"
	 
	 

		----- Original Message ----- 
		From: gary.d.brinkman.jr at census.gov 
		To: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>

		Cc: cisco-voip-bounces at puck.nether.net ;
cisco-voip at puck.nether.net 
		Sent: Thursday, December 01, 2005 5:16 PM
		Subject: Re: [cisco-voip] CDR question..

		Not sure about 4.0 but on 3.3 you can run this query
against the CDR
		database and get what you need.  Just change the
originalCalledPartyNumber
		to the number you need.  And while 1128142800 is
equivalent to 10/1/2005
		12:00 AM  EST you may need to get a Epoch time converter
to get a different
		time.
		
		
		SELECT     duration
		FROM         CallDetailRecord
		WHERE     (dateTimeOrigination > '1128142800') and
		(originalCalledPartyNumber like '%8005551212')
		
		-----
		
		Gary D. Brinkman
		U.S. Census Bureau
		Telecommunications Office
		
		
		
	

		             "Tim Reimers"

		             <tim.reimers at ashe

		             ville.k12.nc.us>
To 
		             Sent by:
<cisco-voip at puck.nether.net>        
		             cisco-voip-bounce
cc 
		             s at puck.nether.net

	
Subject 
		                                       [cisco-voip] CDR
question..         
		             11/30/05 11:26 AM

	

	

	

	

	

		
		
		
		
		Hi everyone---
		
		I have a CDR newbie kind of question---
		
		Can someone give me the steps needed to determine the
following:  (CCM 4.0)
		
		All calls to a given 800 number in say, the last two
months...
		
		Number of calls
		Duration of each call
		
		If I can get it into Excel I can munge around the data
to get the right
		things I want in terms of averages, etc...
		
		But I've not done any CDR stuff as yet..
		 _______________________________________________
		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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--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
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From lelio at uoguelph.ca  Sun Dec  4 02:49:13 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun Dec  4 02:49:19 2005
Subject: [cisco-voip] CDR question..
References: <965C0255FD5828438E34F9A68ECA76E7233E18@svcexch02.firm.fenwick.llp>
Message-ID: <001901c5f8a7$38b0e0b0$ca01a8c0@deadfish>

Something to worry about with Excel is the conversion of daylight saving time correctly. Our LD billing office uses Excel and they have to do some manual modifications to account for it.

Just a thought.

  ----- Original Message ----- 
  From: Stu Packett 
  To: Tim Reimers ; cisco-voip at puck.nether.net 
  Sent: Sunday, December 04, 2005 2:44 AM
  Subject: RE: [cisco-voip] CDR question..


  I do similar CDR queries as well when someone inadvertantly dials 911.  I have an Excel spreadsheet that makes it easier for me.  The date that the CDR reports is the number of seconds (or minutes?) from January 1, 1970.  So I have a field in Excel that says:  =A1/86400+DATE(1970,1,1) that converts the numbers into dates.  If this isn't that clear, I can send my CDR template for Excel that I always use.

  The CDR query I would run in SQL Query Analzyer is:
  select *
  from CallDetailRecord
  where finalCalledPartyNumber LIKE '91800%'

  Hope this helps.



------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
  Sent: Wednesday, November 30, 2005 8:26 AM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] CDR question..


  Hi everyone---



  I have a CDR newbie kind of question---



  Can someone give me the steps needed to determine the following:  (CCM 4.0)



  All calls to a given 800 number in say, the last two months.



  Number of calls

  Duration of each call



  If I can get it into Excel I can munge around the data to get the right things I want in terms of averages, etc.



  But I've not done any CDR stuff as yet..




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

  IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein. 

------------------------------------------------------------------------------
  ATTENTION:
  The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited. 
   
  If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message. 


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


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From jgrace at digitelusa.net  Sun Dec  4 12:35:47 2005
From: jgrace at digitelusa.net (James Grace)
Date: Sun Dec  4 12:38:04 2005
Subject: [cisco-voip] problem with part and css 
Message-ID: <000f01c5f8f9$29df4390$030a0f0a@digitelcorp.com>

we have a cm 4.1.2 es 41   
i was just told by a tac rep that 4.1. and higher uses both css on line and device.
what does this mean????

i have my device as unrestricted and all of the part in it.  and then i have my line configure with a customize css, ie maybe no int calls or LD calls.  
the problem im having is that when i make a ld or int call,  uses the device.  and when i make a ipphone call ( 4 dgit dialing between sites) it uses the line. I dont understand this.  nore do i understand  the statement   IT USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.  Thanks in advance
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From lelio at uoguelph.ca  Sun Dec  4 13:13:40 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun Dec  4 13:13:48 2005
Subject: [cisco-voip] problem with part and css 
References: <000f01c5f8f9$29df4390$030a0f0a@digitelcorp.com>
Message-ID: <001a01c5f8fe$747cba40$c901a8c0@EATIUSBIRDIUS>

Think of it this way, a calling search space is simply an ordered list of partitions which the line or device searches through for a match of the dialed digits. If you look through the traces, you will see the term PSS which I believe stands for Phone Search Space. This is made up of the line's ordered list of partitions and then the device's ordered list of partitions. For example:

CSS1=PartA,PartB,PartC
CSS2=PartX,PartY,PartZ
Device assigned CSS1
Line assigned CSS2
the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC

So the idea you propose is a valid one. We use it here. Assign the device access to everything and then deny the line access to particular services. We isolate the 'everything' to PSTN access and actually allow oncampus access through the line's partition.

You have to make sure that the denies are at least as specific as the allows. And by denies, I mean, creating a route pattern that has the 'block' option checked.

Post back if you need more...



  ----- Original Message ----- 
  From: James Grace 
  To: cisco-voip at puck.nether.net 
  Sent: Sunday, December 04, 2005 12:35 PM
  Subject: [cisco-voip] problem with part and css 


  we have a cm 4.1.2 es 41   
  i was just told by a tac rep that 4.1. and higher uses both css on line and device.
  what does this mean????

  i have my device as unrestricted and all of the part in it.  and then i have my line configure with a customize css, ie maybe no int calls or LD calls.  
  the problem im having is that when i make a ld or int call,  uses the device.  and when i make a ipphone call ( 4 dgit dialing between sites) it uses the line. I dont understand this.  nore do i understand  the statement   IT USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.  Thanks in advance


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


  _______________________________________________
  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  Sun Dec  4 13:19:10 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun Dec  4 13:19:19 2005
Subject: [cisco-voip] problem with part and css (the none partition)
References: <000f01c5f8f9$29df4390$030a0f0a@digitelcorp.com>
	<001a01c5f8fe$747cba40$c901a8c0@EATIUSBIRDIUS>
Message-ID: <003301c5f8ff$39b688e0$c901a8c0@EATIUSBIRDIUS>

Oh yeah, and don't forget the <None> partition. It's not really a partition, but it encompasses as DNs and patterns that have not been assigned to a partition. There is no easy way to block access to these DNs unless you create duplicate entries in another partition and block those (which we have done). I can't remember the search algorithm and whether the <None> partition is simply placed last in the ordered list and searched accordingly or if it is searched only after no match is made in the other partitions. A search on CCO docs could tell you this probably.

We've moved towards a dialplan model that has next to no DNs in the <None> partition. The only ones we have in there are our campus emergency number and our helpline number. In the event that a phone goes out misconfigured, chances are very high they will be able to call these numbers.

  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: James Grace ; cisco-voip at puck.nether.net 
  Sent: Sunday, December 04, 2005 1:13 PM
  Subject: Re: [cisco-voip] problem with part and css 


  Think of it this way, a calling search space is simply an ordered list of partitions which the line or device searches through for a match of the dialed digits. If you look through the traces, you will see the term PSS which I believe stands for Phone Search Space. This is made up of the line's ordered list of partitions and then the device's ordered list of partitions. For example:

  CSS1=PartA,PartB,PartC
  CSS2=PartX,PartY,PartZ
  Device assigned CSS1
  Line assigned CSS2
  the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC

  So the idea you propose is a valid one. We use it here. Assign the device access to everything and then deny the line access to particular services. We isolate the 'everything' to PSTN access and actually allow oncampus access through the line's partition.

  You have to make sure that the denies are at least as specific as the allows. And by denies, I mean, creating a route pattern that has the 'block' option checked.

  Post back if you need more...



    ----- Original Message ----- 
    From: James Grace 
    To: cisco-voip at puck.nether.net 
    Sent: Sunday, December 04, 2005 12:35 PM
    Subject: [cisco-voip] problem with part and css 


    we have a cm 4.1.2 es 41   
    i was just told by a tac rep that 4.1. and higher uses both css on line and device.
    what does this mean????

    i have my device as unrestricted and all of the part in it.  and then i have my line configure with a customize css, ie maybe no int calls or LD calls.  
    the problem im having is that when i make a ld or int call,  uses the device.  and when i make a ipphone call ( 4 dgit dialing between sites) it uses the line. I dont understand this.  nore do i understand  the statement   IT USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.  Thanks in advance


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


    _______________________________________________
    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 highspeedsnow at gmail.com  Sun Dec  4 14:13:47 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sun Dec  4 14:14:23 2005
Subject: [cisco-voip] problem with part and css (the none partition)
In-Reply-To: <003301c5f8ff$39b688e0$c901a8c0@EATIUSBIRDIUS>
Message-ID: <4393400c.06d97ac8.6751.ffffca5d@mx.gmail.com>

Guys,

 

The <none> partition is actually a real partition, it is there by default
and is assigned to the very bottom of every CSS.

 

So if you had a RP of '91X' in the INTERAL_PT belonging to the HQ_911_CSS
and Phone 1 at HQ had this CSS assigned,  but you also had a RP of '911' in
in the <none> Partition (PT) and the user at Phone 1 at HQ dialed 911, he
would actually be routed to the 911 RP in the <none> PT due to the fact that
it appears last in the HQ_911_CSS and is a longer match.

 

It is similar to the <null> MRG, which belongs at the very bottom of every
MRGL.

 

MTC and HTH some :-)

 

-Mark Snow

CCIE Voice # 14073

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Sunday, December 04, 2005 12:19 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] problem with part and css (the none partition)

 

Oh yeah, and don't forget the <None> partition. It's not really a partition,
but it encompasses as DNs and patterns that have not been assigned to a
partition. There is no easy way to block access to these DNs unless you
create duplicate entries in another partition and block those (which we have
done). I can't remember the search algorithm and whether the <None>
partition is simply placed last in the ordered list and searched accordingly
or if it is searched only after no match is made in the other partitions. A
search on CCO docs could tell you this probably.

 

We've moved towards a dialplan model that has next to no DNs in the <None>
partition. The only ones we have in there are our campus emergency number
and our helpline number. In the event that a phone goes out misconfigured,
chances are very high they will be able to call these numbers.

 

----- Original Message ----- 

From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  

To: James Grace <mailto:jgrace at digitelusa.net>  ; cisco-voip at puck.nether.net


Sent: Sunday, December 04, 2005 1:13 PM

Subject: Re: [cisco-voip] problem with part and css 

 

Think of it this way, a calling search space is simply an ordered list of
partitions which the line or device searches through for a match of the
dialed digits. If you look through the traces, you will see the term PSS
which I believe stands for Phone Search Space. This is made up of the line's
ordered list of partitions and then the device's ordered list of partitions.
For example:

 

CSS1=PartA,PartB,PartC

CSS2=PartX,PartY,PartZ

Device assigned CSS1

Line assigned CSS2

the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC

 

So the idea you propose is a valid one. We use it here. Assign the device
access to everything and then deny the line access to particular services.
We isolate the 'everything' to PSTN access and actually allow oncampus
access through the line's partition.

 

You have to make sure that the denies are at least as specific as the
allows. And by denies, I mean, creating a route pattern that has the 'block'
option checked.

 

Post back if you need more...

 

 

 

----- Original Message ----- 

From: James Grace <mailto:jgrace at digitelusa.net>  

To: cisco-voip at puck.nether.net 

Sent: Sunday, December 04, 2005 12:35 PM

Subject: [cisco-voip] problem with part and css 

 

we have a cm 4.1.2 es 41   

i was just told by a tac rep that 4.1. and higher uses both css on line and
device.

what does this mean????

 

i have my device as unrestricted and all of the part in it.  and then i have
my line configure with a customize css, ie maybe no int calls or LD calls.  

the problem im having is that when i make a ld or int call,  uses the
device.  and when i make a ipphone call ( 4 dgit dialing between sites) it
uses the line. I dont understand this.  nore do i understand  the statement
IT USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.
Thanks in advance


  _____  


_______________________________________________
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 Michael_Davis at eLoyalty.com  Sun Dec  4 14:35:54 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Sun Dec  4 14:36:06 2005
Subject: [cisco-voip] problem with part and css (the none partition)
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E323@elhqex01.eloyaltyco.com>

Also,
Prior to CCM 3.1, Device Level CSS were placed ahead of Line Level CSS.  As of 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT for CTI devices (Softphone, CTI Ports, Route Points, etc,) which use the old method.
 
Mike
 

	-----Original Message----- 
	From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow 
	Sent: Sun 12/4/2005 1:13 PM 
	To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net 
	Cc: 
	Subject: RE: [cisco-voip] problem with part and css (the none partition)
	
	

	Guys,

	 

	The <none> partition is actually a real partition, it is there by default and is assigned to the very bottom of every CSS.

	 

	So if you had a RP of ?91X? in the INTERAL_PT belonging to the HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you also had a RP of ?911? in in the <none> Partition (PT) and the user at Phone 1 at HQ dialed 911, he would actually be routed to the 911 RP in the <none> PT due to the fact that it appears last in the HQ_911_CSS and is a longer match.

	 

	It is similar to the <null> MRG, which belongs at the very bottom of every MRGL.

	 

	MTC and HTH some :-)

	 

	-Mark Snow

	CCIE Voice # 14073

	 

	
  _____  


	From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	Sent: Sunday, December 04, 2005 12:19 PM
	To: cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] problem with part and css (the none partition)

	 

	Oh yeah, and don't forget the <None> partition. It's not really a partition, but it encompasses as DNs and patterns that have not been assigned to a partition. There is no easy way to block access to these DNs unless you create duplicate entries in another partition and block those (which we have done). I can't remember the search algorithm and whether the <None> partition is simply placed last in the ordered list and searched accordingly or if it is searched only after no match is made in the other partitions. A search on CCO docs could tell you this probably.

	 

	We've moved towards a dialplan model that has next to no DNs in the <None> partition. The only ones we have in there are our campus emergency number and our helpline number. In the event that a phone goes out misconfigured, chances are very high they will be able to call these numbers.

	 

		----- Original Message ----- 

		From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  

		To: James Grace <mailto:jgrace at digitelusa.net>  ; cisco-voip at puck.nether.net 

		Sent: Sunday, December 04, 2005 1:13 PM

		Subject: Re: [cisco-voip] problem with part and css 

		 

		Think of it this way, a calling search space is simply an ordered list of partitions which the line or device searches through for a match of the dialed digits. If you look through the traces, you will see the term PSS which I believe stands for Phone Search Space. This is made up of the line's ordered list of partitions and then the device's ordered list of partitions. For example:

		 

		CSS1=PartA,PartB,PartC

		CSS2=PartX,PartY,PartZ

		Device assigned CSS1

		Line assigned CSS2

		the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC

		 

		So the idea you propose is a valid one. We use it here. Assign the device access to everything and then deny the line access to particular services. We isolate the 'everything' to PSTN access and actually allow oncampus access through the line's partition.

		 

		You have to make sure that the denies are at least as specific as the allows. And by denies, I mean, creating a route pattern that has the 'block' option checked.

		 

		Post back if you need more...

		 

		 

		 

			----- Original Message ----- 

			From: James Grace <mailto:jgrace at digitelusa.net>  

			To: cisco-voip at puck.nether.net 

			Sent: Sunday, December 04, 2005 12:35 PM

			Subject: [cisco-voip] problem with part and css 

			 

			we have a cm 4.1.2 es 41   

			i was just told by a tac rep that 4.1. and higher uses both css on line and device.

			what does this mean????

			 

			i have my device as unrestricted and all of the part in it.  and then i have my line configure with a customize css, ie maybe no int calls or LD calls.  

			the problem im having is that when i make a ld or int call,  uses the device.  and when i make a ipphone call ( 4 dgit dialing between sites) it uses the line. I dont understand this.  nore do i understand  the statement   IT USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.  Thanks in advance

			
  _____  


			_______________________________________________
			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 highspeedsnow at gmail.com  Sun Dec  4 14:47:39 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sun Dec  4 14:48:15 2005
Subject: [cisco-voip] problem with part and css (the none partition)
In-Reply-To: <037594FB5AFC354FA38653EB3FF3242105A8E323@elhqex01.eloyaltyco.com>
Message-ID: <439347fd.0c8a6f2c.3bfb.ffffda7a@mx.gmail.com>

Good Point!

For those of you out there running versions prior to 3.1 when it was still
'Call Mangler'.

Also if you are running those previous versions, try Forwarding your phone
to it's own DN and then calling it from another phone - just for fun! ;)

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Davis, Michael
> Sent: Sunday, December 04, 2005 1:36 PM
> To: cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] problem with part and css (the none partition)
> 
> Also,
> Prior to CCM 3.1, Device Level CSS were placed ahead of Line Level CSS.
As of
> 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT for CTI
devices
> (Softphone, CTI Ports, Route Points, etc,) which use the old method.
> 
> Mike
> 
> 
> 	-----Original Message-----
> 	From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow
> 	Sent: Sun 12/4/2005 1:13 PM
> 	To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
> 	Cc:
> 	Subject: RE: [cisco-voip] problem with part and css (the none
partition)
> 
> 
> 
> 	Guys,
> 
> 
> 
> 	The <none> partition is actually a real partition, it is there by
default and
> is assigned to the very bottom of every CSS.
> 
> 
> 
> 	So if you had a RP of '91X' in the INTERAL_PT belonging to the
> HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you also had a RP
of
> '911' in in the <none> Partition (PT) and the user at Phone 1 at HQ dialed
911, he
> would actually be routed to the 911 RP in the <none> PT due to the fact
that it
> appears last in the HQ_911_CSS and is a longer match.
> 
> 
> 
> 	It is similar to the <null> MRG, which belongs at the very bottom of
every
> MRGL.
> 
> 
> 
> 	MTC and HTH some :-)
> 
> 
> 
> 	-Mark Snow
> 
> 	CCIE Voice # 14073
> 
> 
> 
> 
>   _____
> 
> 
> 	From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
> 	Sent: Sunday, December 04, 2005 12:19 PM
> 	To: cisco-voip at puck.nether.net
> 	Subject: Re: [cisco-voip] problem with part and css (the none
partition)
> 
> 
> 
> 	Oh yeah, and don't forget the <None> partition. It's not really a
partition,
> but it encompasses as DNs and patterns that have not been assigned to a
> partition. There is no easy way to block access to these DNs unless you
create
> duplicate entries in another partition and block those (which we have
done). I
> can't remember the search algorithm and whether the <None> partition is
simply
> placed last in the ordered list and searched accordingly or if it is
searched only
> after no match is made in the other partitions. A search on CCO docs could
tell
> you this probably.
> 
> 
> 
> 	We've moved towards a dialplan model that has next to no DNs in the
> <None> partition. The only ones we have in there are our campus emergency
> number and our helpline number. In the event that a phone goes out
> misconfigured, chances are very high they will be able to call these
numbers.
> 
> 
> 
> 		----- Original Message -----
> 
> 		From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
> 
> 		To: James Grace <mailto:jgrace at digitelusa.net>  ; cisco-
> voip at puck.nether.net
> 
> 		Sent: Sunday, December 04, 2005 1:13 PM
> 
> 		Subject: Re: [cisco-voip] problem with part and css
> 
> 
> 
> 		Think of it this way, a calling search space is simply an
ordered
> list of partitions which the line or device searches through for a match
of the
> dialed digits. If you look through the traces, you will see the term PSS
which I
> believe stands for Phone Search Space. This is made up of the line's
ordered list of
> partitions and then the device's ordered list of partitions. For example:
> 
> 
> 
> 		CSS1=PartA,PartB,PartC
> 
> 		CSS2=PartX,PartY,PartZ
> 
> 		Device assigned CSS1
> 
> 		Line assigned CSS2
> 
> 		the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC
> 
> 
> 
> 		So the idea you propose is a valid one. We use it here.
Assign the
> device access to everything and then deny the line access to particular
services.
> We isolate the 'everything' to PSTN access and actually allow oncampus
access
> through the line's partition.
> 
> 
> 
> 		You have to make sure that the denies are at least as
specific as
> the allows. And by denies, I mean, creating a route pattern that has the
'block'
> option checked.
> 
> 
> 
> 		Post back if you need more...
> 
> 
> 
> 
> 
> 
> 
> 			----- Original Message -----
> 
> 			From: James Grace <mailto:jgrace at digitelusa.net>
> 
> 			To: cisco-voip at puck.nether.net
> 
> 			Sent: Sunday, December 04, 2005 12:35 PM
> 
> 			Subject: [cisco-voip] problem with part and css
> 
> 
> 
> 			we have a cm 4.1.2 es 41
> 
> 			i was just told by a tac rep that 4.1. and higher
uses both
> css on line and device.
> 
> 			what does this mean????
> 
> 
> 
> 			i have my device as unrestricted and all of the part
in it.
> and then i have my line configure with a customize css, ie maybe no int
calls or LD
> calls.
> 
> 			the problem im having is that when i make a ld or
int call,
> uses the device.  and when i make a ipphone call ( 4 dgit dialing between
sites) it
> uses the line. I dont understand this.  nore do i understand  the
statement   IT
> USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.
> Thanks in advance
> 
> 
>   _____
> 
> 
> 			_______________________________________________
> 			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


From Michael_Davis at eLoyalty.com  Sun Dec  4 15:02:33 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Sun Dec  4 15:02:46 2005
Subject: [cisco-voip] problem with part and css (the none partition)
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E324@elhqex01.eloyaltyco.com>

I don't have a whole lot of experience with 3.1, but I have recently been "burnt" by forgetting the CTI vs non-CTI search order. :-)

	-----Original Message----- 
	From: Mark Snow [mailto:highspeedsnow at gmail.com] 
	Sent: Sun 12/4/2005 1:47 PM 
	To: Davis, Michael; cisco-voip at puck.nether.net 
	Cc: 
	Subject: RE: [cisco-voip] problem with part and css (the none partition)
	
	

	Good Point!
	
	For those of you out there running versions prior to 3.1 when it was still
	'Call Mangler'.
	
	Also if you are running those previous versions, try Forwarding your phone
	to it's own DN and then calling it from another phone - just for fun! ;)
	
	> -----Original Message-----
	> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
	> bounces at puck.nether.net] On Behalf Of Davis, Michael
	> Sent: Sunday, December 04, 2005 1:36 PM
	> To: cisco-voip at puck.nether.net
	> Subject: RE: [cisco-voip] problem with part and css (the none partition)
	>
	> Also,
	> Prior to CCM 3.1, Device Level CSS were placed ahead of Line Level CSS.
	As of
	> 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT for CTI
	devices
	> (Softphone, CTI Ports, Route Points, etc,) which use the old method.
	>
	> Mike
	>
	>
	>       -----Original Message-----
	>       From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow
	>       Sent: Sun 12/4/2005 1:13 PM
	>       To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
	>       Cc:
	>       Subject: RE: [cisco-voip] problem with part and css (the none
	partition)
	>
	>
	>
	>       Guys,
	>
	>
	>
	>       The <none> partition is actually a real partition, it is there by
	default and
	> is assigned to the very bottom of every CSS.
	>
	>
	>
	>       So if you had a RP of '91X' in the INTERAL_PT belonging to the
	> HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you also had a RP
	of
	> '911' in in the <none> Partition (PT) and the user at Phone 1 at HQ dialed
	911, he
	> would actually be routed to the 911 RP in the <none> PT due to the fact
	that it
	> appears last in the HQ_911_CSS and is a longer match.
	>
	>
	>
	>       It is similar to the <null> MRG, which belongs at the very bottom of
	every
	> MRGL.
	>
	>
	>
	>       MTC and HTH some :-)
	>
	>
	>
	>       -Mark Snow
	>
	>       CCIE Voice # 14073
	>
	>
	>
	>
	>   _____
	>
	>
	>       From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
	> bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	>       Sent: Sunday, December 04, 2005 12:19 PM
	>       To: cisco-voip at puck.nether.net
	>       Subject: Re: [cisco-voip] problem with part and css (the none
	partition)
	>
	>
	>
	>       Oh yeah, and don't forget the <None> partition. It's not really a
	partition,
	> but it encompasses as DNs and patterns that have not been assigned to a
	> partition. There is no easy way to block access to these DNs unless you
	create
	> duplicate entries in another partition and block those (which we have
	done). I
	> can't remember the search algorithm and whether the <None> partition is
	simply
	> placed last in the ordered list and searched accordingly or if it is
	searched only
	> after no match is made in the other partitions. A search on CCO docs could
	tell
	> you this probably.
	>
	>
	>
	>       We've moved towards a dialplan model that has next to no DNs in the
	> <None> partition. The only ones we have in there are our campus emergency
	> number and our helpline number. In the event that a phone goes out
	> misconfigured, chances are very high they will be able to call these
	numbers.
	>
	>
	>
	>               ----- Original Message -----
	>
	>               From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
	>
	>               To: James Grace <mailto:jgrace at digitelusa.net>  ; cisco-
	> voip at puck.nether.net
	>
	>               Sent: Sunday, December 04, 2005 1:13 PM
	>
	>               Subject: Re: [cisco-voip] problem with part and css
	>
	>
	>
	>               Think of it this way, a calling search space is simply an
	ordered
	> list of partitions which the line or device searches through for a match
	of the
	> dialed digits. If you look through the traces, you will see the term PSS
	which I
	> believe stands for Phone Search Space. This is made up of the line's
	ordered list of
	> partitions and then the device's ordered list of partitions. For example:
	>
	>
	>
	>               CSS1=PartA,PartB,PartC
	>
	>               CSS2=PartX,PartY,PartZ
	>
	>               Device assigned CSS1
	>
	>               Line assigned CSS2
	>
	>               the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC
	>
	>
	>
	>               So the idea you propose is a valid one. We use it here.
	Assign the
	> device access to everything and then deny the line access to particular
	services.
	> We isolate the 'everything' to PSTN access and actually allow oncampus
	access
	> through the line's partition.
	>
	>
	>
	>               You have to make sure that the denies are at least as
	specific as
	> the allows. And by denies, I mean, creating a route pattern that has the
	'block'
	> option checked.
	>
	>
	>
	>               Post back if you need more...
	>
	>
	>
	>
	>
	>
	>
	>                       ----- Original Message -----
	>
	>                       From: James Grace <mailto:jgrace at digitelusa.net>
	>
	>                       To: cisco-voip at puck.nether.net
	>
	>                       Sent: Sunday, December 04, 2005 12:35 PM
	>
	>                       Subject: [cisco-voip] problem with part and css
	>
	>
	>
	>                       we have a cm 4.1.2 es 41
	>
	>                       i was just told by a tac rep that 4.1. and higher
	uses both
	> css on line and device.
	>
	>                       what does this mean????
	>
	>
	>
	>                       i have my device as unrestricted and all of the part
	in it.
	> and then i have my line configure with a customize css, ie maybe no int
	calls or LD
	> calls.
	>
	>                       the problem im having is that when i make a ld or
	int call,
	> uses the device.  and when i make a ipphone call ( 4 dgit dialing between
	sites) it
	> uses the line. I dont understand this.  nore do i understand  the
	statement   IT
	> USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.
	> Thanks in advance
	>
	>
	>   _____
	>
	>
	>                       _______________________________________________
	>                       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
	



From lelio at uoguelph.ca  Sun Dec  4 15:04:20 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun Dec  4 15:04:36 2005
Subject: [cisco-voip] problem with part and css (the none partition)
References: <037594FB5AFC354FA38653EB3FF3242105A8E324@elhqex01.eloyaltyco.com>
Message-ID: <002601c5f90d$ea4cf410$c901a8c0@EATIUSBIRDIUS>

Is there any particular reason why they've gone this way with the CTIs? Are they eventually going to change it?

  ----- Original Message ----- 
  From: Davis, Michael 
  To: cisco-voip at puck.nether.net 
  Sent: Sunday, December 04, 2005 3:02 PM
  Subject: RE: [cisco-voip] problem with part and css (the none partition)


  I don't have a whole lot of experience with 3.1, but I have recently been "burnt" by forgetting the CTI vs non-CTI search order. :-)

  -----Original Message----- 
  From: Mark Snow [mailto:highspeedsnow at gmail.com] 
  Sent: Sun 12/4/2005 1:47 PM 
  To: Davis, Michael; cisco-voip at puck.nether.net 
  Cc: 
  Subject: RE: [cisco-voip] problem with part and css (the none partition)



  Good Point!

  For those of you out there running versions prior to 3.1 when it was still
  'Call Mangler'.

  Also if you are running those previous versions, try Forwarding your phone
  to it's own DN and then calling it from another phone - just for fun! ;)

  > -----Original Message-----
  > From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
  > bounces at puck.nether.net] On Behalf Of Davis, Michael
  > Sent: Sunday, December 04, 2005 1:36 PM
  > To: cisco-voip at puck.nether.net
  > Subject: RE: [cisco-voip] problem with part and css (the none partition)
  >
  > Also,
  > Prior to CCM 3.1, Device Level CSS were placed ahead of Line Level CSS.
  As of
  > 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT for CTI
  devices
  > (Softphone, CTI Ports, Route Points, etc,) which use the old method.
  >
  > Mike
  >
  >
  >       -----Original Message-----
  >       From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow
  >       Sent: Sun 12/4/2005 1:13 PM
  >       To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
  >       Cc:
  >       Subject: RE: [cisco-voip] problem with part and css (the none
  partition)
  >
  >
  >
  >       Guys,
  >
  >
  >
  >       The <none> partition is actually a real partition, it is there by
  default and
  > is assigned to the very bottom of every CSS.
  >
  >
  >
  >       So if you had a RP of '91X' in the INTERAL_PT belonging to the
  > HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you also had a RP
  of
  > '911' in in the <none> Partition (PT) and the user at Phone 1 at HQ dialed
  911, he
  > would actually be routed to the 911 RP in the <none> PT due to the fact
  that it
  > appears last in the HQ_911_CSS and is a longer match.
  >
  >
  >
  >       It is similar to the <null> MRG, which belongs at the very bottom of
  every
  > MRGL.
  >
  >
  >
  >       MTC and HTH some :-)
  >
  >
  >
  >       -Mark Snow
  >
  >       CCIE Voice # 14073
  >
  >
  >
  >
  >   _____
  >
  >
  >       From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
  > bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  >       Sent: Sunday, December 04, 2005 12:19 PM
  >       To: cisco-voip at puck.nether.net
  >       Subject: Re: [cisco-voip] problem with part and css (the none
  partition)
  >
  >
  >
  >       Oh yeah, and don't forget the <None> partition. It's not really a
  partition,
  > but it encompasses as DNs and patterns that have not been assigned to a
  > partition. There is no easy way to block access to these DNs unless you
  create
  > duplicate entries in another partition and block those (which we have
  done). I
  > can't remember the search algorithm and whether the <None> partition is
  simply
  > placed last in the ordered list and searched accordingly or if it is
  searched only
  > after no match is made in the other partitions. A search on CCO docs could
  tell
  > you this probably.
  >
  >
  >
  >       We've moved towards a dialplan model that has next to no DNs in the
  > <None> partition. The only ones we have in there are our campus emergency
  > number and our helpline number. In the event that a phone goes out
  > misconfigured, chances are very high they will be able to call these
  numbers.
  >
  >
  >
  >               ----- Original Message -----
  >
  >               From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
  >
  >               To: James Grace <mailto:jgrace at digitelusa.net>  ; cisco-
  > voip at puck.nether.net
  >
  >               Sent: Sunday, December 04, 2005 1:13 PM
  >
  >               Subject: Re: [cisco-voip] problem with part and css
  >
  >
  >
  >               Think of it this way, a calling search space is simply an
  ordered
  > list of partitions which the line or device searches through for a match
  of the
  > dialed digits. If you look through the traces, you will see the term PSS
  which I
  > believe stands for Phone Search Space. This is made up of the line's
  ordered list of
  > partitions and then the device's ordered list of partitions. For example:
  >
  >
  >
  >               CSS1=PartA,PartB,PartC
  >
  >               CSS2=PartX,PartY,PartZ
  >
  >               Device assigned CSS1
  >
  >               Line assigned CSS2
  >
  >               the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC
  >
  >
  >
  >               So the idea you propose is a valid one. We use it here.
  Assign the
  > device access to everything and then deny the line access to particular
  services.
  > We isolate the 'everything' to PSTN access and actually allow oncampus
  access
  > through the line's partition.
  >
  >
  >
  >               You have to make sure that the denies are at least as
  specific as
  > the allows. And by denies, I mean, creating a route pattern that has the
  'block'
  > option checked.
  >
  >
  >
  >               Post back if you need more...
  >
  >
  >
  >
  >
  >
  >
  >                       ----- Original Message -----
  >
  >                       From: James Grace <mailto:jgrace at digitelusa.net>
  >
  >                       To: cisco-voip at puck.nether.net
  >
  >                       Sent: Sunday, December 04, 2005 12:35 PM
  >
  >                       Subject: [cisco-voip] problem with part and css
  >
  >
  >
  >                       we have a cm 4.1.2 es 41
  >
  >                       i was just told by a tac rep that 4.1. and higher
  uses both
  > css on line and device.
  >
  >                       what does this mean????
  >
  >
  >
  >                       i have my device as unrestricted and all of the part
  in it.
  > and then i have my line configure with a customize css, ie maybe no int
  calls or LD
  > calls.
  >
  >                       the problem im having is that when i make a ld or
  int call,
  > uses the device.  and when i make a ipphone call ( 4 dgit dialing between
  sites) it
  > uses the line. I dont understand this.  nore do i understand  the
  statement   IT
  > USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.
  > Thanks in advance
  >
  >
  >   _____
  >
  >
  >                       _______________________________________________
  >                       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
-------------- next part --------------
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From highspeedsnow at gmail.com  Sun Dec  4 16:20:03 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sun Dec  4 16:20:45 2005
Subject: [cisco-voip] problem with part and css (the none partition)
In-Reply-To: <002601c5f90d$ea4cf410$c901a8c0@EATIUSBIRDIUS>
Message-ID: <43935daa.4c47d377.323a.fffffb62@mx.gmail.com>

The answer probably lies with the PM, but I don't know why or when .

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Sunday, December 04, 2005 2:04 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] problem with part and css (the none partition)

 

Is there any particular reason why they've gone this way with the CTIs? Are
they eventually going to change it?

 

----- Original Message ----- 

From: Davis, <mailto:Michael_Davis at eLoyalty.com>  Michael 

To: cisco-voip at puck.nether.net 

Sent: Sunday, December 04, 2005 3:02 PM

Subject: RE: [cisco-voip] problem with part and css (the none partition)

 

I don't have a whole lot of experience with 3.1, but I have recently been
"burnt" by forgetting the CTI vs non-CTI search order. :-)

-----Original Message----- 
From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: Sun 12/4/2005 1:47 PM 
To: Davis, Michael; cisco-voip at puck.nether.net 
Cc: 
Subject: RE: [cisco-voip] problem with part and css (the none partition)



Good Point!

For those of you out there running versions prior to 3.1 when it was still
'Call Mangler'.

Also if you are running those previous versions, try Forwarding your phone
to it's own DN and then calling it from another phone - just for fun! ;)

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Davis, Michael
> Sent: Sunday, December 04, 2005 1:36 PM
> To: cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] problem with part and css (the none partition)
>
> Also,
> Prior to CCM 3.1, Device Level CSS were placed ahead of Line Level CSS.
As of
> 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT for CTI
devices
> (Softphone, CTI Ports, Route Points, etc,) which use the old method.
>
> Mike
>
>
>       -----Original Message-----
>       From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow
>       Sent: Sun 12/4/2005 1:13 PM
>       To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
>       Cc:
>       Subject: RE: [cisco-voip] problem with part and css (the none
partition)
>
>
>
>       Guys,
>
>
>
>       The <none> partition is actually a real partition, it is there by
default and
> is assigned to the very bottom of every CSS.
>
>
>
>       So if you had a RP of '91X' in the INTERAL_PT belonging to the
> HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you also had a RP
of
> '911' in in the <none> Partition (PT) and the user at Phone 1 at HQ dialed
911, he
> would actually be routed to the 911 RP in the <none> PT due to the fact
that it
> appears last in the HQ_911_CSS and is a longer match.
>
>
>
>       It is similar to the <null> MRG, which belongs at the very bottom of
every
> MRGL.
>
>
>
>       MTC and HTH some :-)
>
>
>
>       -Mark Snow
>
>       CCIE Voice # 14073
>
>
>
>
>   _____
>
>
>       From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
>       Sent: Sunday, December 04, 2005 12:19 PM
>       To: cisco-voip at puck.nether.net
>       Subject: Re: [cisco-voip] problem with part and css (the none
partition)
>
>
>
>       Oh yeah, and don't forget the <None> partition. It's not really a
partition,
> but it encompasses as DNs and patterns that have not been assigned to a
> partition. There is no easy way to block access to these DNs unless you
create
> duplicate entries in another partition and block those (which we have
done). I
> can't remember the search algorithm and whether the <None> partition is
simply
> placed last in the ordered list and searched accordingly or if it is
searched only
> after no match is made in the other partitions. A search on CCO docs could
tell
> you this probably.
>
>
>
>       We've moved towards a dialplan model that has next to no DNs in the
> <None> partition. The only ones we have in there are our campus emergency
> number and our helpline number. In the event that a phone goes out
> misconfigured, chances are very high they will be able to call these
numbers.
>
>
>
>               ----- Original Message -----
>
>               From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
>
>               To: James Grace <mailto:jgrace at digitelusa.net>  ; cisco-
> voip at puck.nether.net
>
>               Sent: Sunday, December 04, 2005 1:13 PM
>
>               Subject: Re: [cisco-voip] problem with part and css
>
>
>
>               Think of it this way, a calling search space is simply an
ordered
> list of partitions which the line or device searches through for a match
of the
> dialed digits. If you look through the traces, you will see the term PSS
which I
> believe stands for Phone Search Space. This is made up of the line's
ordered list of
> partitions and then the device's ordered list of partitions. For example:
>
>
>
>               CSS1=PartA,PartB,PartC
>
>               CSS2=PartX,PartY,PartZ
>
>               Device assigned CSS1
>
>               Line assigned CSS2
>
>               the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC
>
>
>
>               So the idea you propose is a valid one. We use it here.
Assign the
> device access to everything and then deny the line access to particular
services.
> We isolate the 'everything' to PSTN access and actually allow oncampus
access
> through the line's partition.
>
>
>
>               You have to make sure that the denies are at least as
specific as
> the allows. And by denies, I mean, creating a route pattern that has the
'block'
> option checked.
>
>
>
>               Post back if you need more...
>
>
>
>
>
>
>
>                       ----- Original Message -----
>
>                       From: James Grace <mailto:jgrace at digitelusa.net>
>
>                       To: cisco-voip at puck.nether.net
>
>                       Sent: Sunday, December 04, 2005 12:35 PM
>
>                       Subject: [cisco-voip] problem with part and css
>
>
>
>                       we have a cm 4.1.2 es 41
>
>                       i was just told by a tac rep that 4.1. and higher
uses both
> css on line and device.
>
>                       what does this mean????
>
>
>
>                       i have my device as unrestricted and all of the part
in it.
> and then i have my line configure with a customize css, ie maybe no int
calls or LD
> calls.
>
>                       the problem im having is that when i make a ld or
int call,
> uses the device.  and when i make a ipphone call ( 4 dgit dialing between
sites) it
> uses the line. I dont understand this.  nore do i understand  the
statement   IT
> USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.
> Thanks in advance
>
>
>   _____
>
>
>                       _______________________________________________
>                       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 rares at raisoftware.ro  Sun Dec  4 16:25:52 2005
From: rares at raisoftware.ro (Rares Ispas)
Date: Sun Dec  4 16:26:26 2005
Subject: [cisco-voip] Voice record
In-Reply-To: <1199305432-1133478823-cardhu_blackberry.rim.net-14434-@engine20-cell02>
References: <1199305432-1133478823-cardhu_blackberry.rim.net-14434-@engine20-cell02>
Message-ID: <op.s0925ewuyg9d0q@relay.raisoftware.ro>

Hello,

Cisco Contact Center has recording and monitoring capabilities. But it can  
record only on demand, and is intended only for low volumes.

For a centralized call recorder scalable to hundreds of phones, you might  
want to try CallReplay (http://www.call-replay.com). It also has call  
monitoring & whispering (the ability to listen to an ongoing call and to  
say something which only one party can hear) and supports extension  
mobility at no extra cost.

-- 

Regards,
Rares Ispas

RAI Software SRL, Bucharest, Romania
YM: rares_ispas2001
MSN: irares at hotmail.com

On Fri, 02 Dec 2005 01:13:46 +0200, James <jgrace at digitelusa.net> wrote:

> Is there a out of box voice record feature in cm 4.1.3
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip




From lelio at uoguelph.ca  Sun Dec  4 17:06:43 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun Dec  4 17:09:13 2005
Subject: [cisco-voip] problem with part and css (the none partition)
References: <43935daa.4c47d377.323a.fffffb62@mx.gmail.com>
Message-ID: <000001c5f91f$576e8390$c901a8c0@EATIUSBIRDIUS>

And the other question is....I'm assuming this doesn't affect IPCommunicator? I ran a test in the lab and it created the PSS according to line+device. Is this because IP communicator is not a CTI application like Softphone? Are my results what you would expect?
  ----- Original Message ----- 
  From: Mark Snow 
  To: 'Lelio Fulgenzi' ; cisco-voip at puck.nether.net 
  Sent: Sunday, December 04, 2005 4:20 PM
  Subject: RE: [cisco-voip] problem with part and css (the none partition)


  The answer probably lies with the PM, but I don't know why or when .

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Sunday, December 04, 2005 2:04 PM
  To: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] problem with part and css (the none partition)

   

  Is there any particular reason why they've gone this way with the CTIs? Are they eventually going to change it?

   

    ----- Original Message ----- 

    From: Davis, Michael 

    To: cisco-voip at puck.nether.net 

    Sent: Sunday, December 04, 2005 3:02 PM

    Subject: RE: [cisco-voip] problem with part and css (the none partition)

     

    I don't have a whole lot of experience with 3.1, but I have recently been "burnt" by forgetting the CTI vs non-CTI search order. :-)

    -----Original Message----- 
    From: Mark Snow [mailto:highspeedsnow at gmail.com] 
    Sent: Sun 12/4/2005 1:47 PM 
    To: Davis, Michael; cisco-voip at puck.nether.net 
    Cc: 
    Subject: RE: [cisco-voip] problem with part and css (the none partition)



    Good Point!

    For those of you out there running versions prior to 3.1 when it was still
    'Call Mangler'.

    Also if you are running those previous versions, try Forwarding your phone
    to it's own DN and then calling it from another phone - just for fun! ;)

    > -----Original Message-----
    > From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
    > bounces at puck.nether.net] On Behalf Of Davis, Michael
    > Sent: Sunday, December 04, 2005 1:36 PM
    > To: cisco-voip at puck.nether.net
    > Subject: RE: [cisco-voip] problem with part and css (the none partition)
    >
    > Also,
    > Prior to CCM 3.1, Device Level CSS were placed ahead of Line Level CSS.
    As of
    > 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT for CTI
    devices
    > (Softphone, CTI Ports, Route Points, etc,) which use the old method.
    >
    > Mike
    >
    >
    >       -----Original Message-----
    >       From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow
    >       Sent: Sun 12/4/2005 1:13 PM
    >       To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
    >       Cc:
    >       Subject: RE: [cisco-voip] problem with part and css (the none
    partition)
    >
    >
    >
    >       Guys,
    >
    >
    >
    >       The <none> partition is actually a real partition, it is there by
    default and
    > is assigned to the very bottom of every CSS.
    >
    >
    >
    >       So if you had a RP of '91X' in the INTERAL_PT belonging to the
    > HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you also had a RP
    of
    > '911' in in the <none> Partition (PT) and the user at Phone 1 at HQ dialed
    911, he
    > would actually be routed to the 911 RP in the <none> PT due to the fact
    that it
    > appears last in the HQ_911_CSS and is a longer match.
    >
    >
    >
    >       It is similar to the <null> MRG, which belongs at the very bottom of
    every
    > MRGL.
    >
    >
    >
    >       MTC and HTH some :-)
    >
    >
    >
    >       -Mark Snow
    >
    >       CCIE Voice # 14073
    >
    >
    >
    >
    >   _____
    >
    >
    >       From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
    > bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
    >       Sent: Sunday, December 04, 2005 12:19 PM
    >       To: cisco-voip at puck.nether.net
    >       Subject: Re: [cisco-voip] problem with part and css (the none
    partition)
    >
    >
    >
    >       Oh yeah, and don't forget the <None> partition. It's not really a
    partition,
    > but it encompasses as DNs and patterns that have not been assigned to a
    > partition. There is no easy way to block access to these DNs unless you
    create
    > duplicate entries in another partition and block those (which we have
    done). I
    > can't remember the search algorithm and whether the <None> partition is
    simply
    > placed last in the ordered list and searched accordingly or if it is
    searched only
    > after no match is made in the other partitions. A search on CCO docs could
    tell
    > you this probably.
    >
    >
    >
    >       We've moved towards a dialplan model that has next to no DNs in the
    > <None> partition. The only ones we have in there are our campus emergency
    > number and our helpline number. In the event that a phone goes out
    > misconfigured, chances are very high they will be able to call these
    numbers.
    >
    >
    >
    >               ----- Original Message -----
    >
    >               From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
    >
    >               To: James Grace <mailto:jgrace at digitelusa.net>  ; cisco-
    > voip at puck.nether.net
    >
    >               Sent: Sunday, December 04, 2005 1:13 PM
    >
    >               Subject: Re: [cisco-voip] problem with part and css
    >
    >
    >
    >               Think of it this way, a calling search space is simply an
    ordered
    > list of partitions which the line or device searches through for a match
    of the
    > dialed digits. If you look through the traces, you will see the term PSS
    which I
    > believe stands for Phone Search Space. This is made up of the line's
    ordered list of
    > partitions and then the device's ordered list of partitions. For example:
    >
    >
    >
    >               CSS1=PartA,PartB,PartC
    >
    >               CSS2=PartX,PartY,PartZ
    >
    >               Device assigned CSS1
    >
    >               Line assigned CSS2
    >
    >               the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC
    >
    >
    >
    >               So the idea you propose is a valid one. We use it here.
    Assign the
    > device access to everything and then deny the line access to particular
    services.
    > We isolate the 'everything' to PSTN access and actually allow oncampus
    access
    > through the line's partition.
    >
    >
    >
    >               You have to make sure that the denies are at least as
    specific as
    > the allows. And by denies, I mean, creating a route pattern that has the
    'block'
    > option checked.
    >
    >
    >
    >               Post back if you need more...
    >
    >
    >
    >
    >
    >
    >
    >                       ----- Original Message -----
    >
    >                       From: James Grace <mailto:jgrace at digitelusa.net>
    >
    >                       To: cisco-voip at puck.nether.net
    >
    >                       Sent: Sunday, December 04, 2005 12:35 PM
    >
    >                       Subject: [cisco-voip] problem with part and css
    >
    >
    >
    >                       we have a cm 4.1.2 es 41
    >
    >                       i was just told by a tac rep that 4.1. and higher
    uses both
    > css on line and device.
    >
    >                       what does this mean????
    >
    >
    >
    >                       i have my device as unrestricted and all of the part
    in it.
    > and then i have my line configure with a customize css, ie maybe no int
    calls or LD
    > calls.
    >
    >                       the problem im having is that when i make a ld or
    int call,
    > uses the device.  and when i make a ipphone call ( 4 dgit dialing between
    sites) it
    > uses the line. I dont understand this.  nore do i understand  the
    statement   IT
    > USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.
    > Thanks in advance
    >
    >
    >   _____
    >
    >
    >                       _______________________________________________
    >                       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
-------------- next part --------------
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From highspeedsnow at gmail.com  Sun Dec  4 17:51:07 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sun Dec  4 17:51:54 2005
Subject: [cisco-voip] problem with part and css (the none partition)
In-Reply-To: <000001c5f91f$576e8390$c901a8c0@EATIUSBIRDIUS>
Message-ID: <43937306.54b1178d.323a.048e@mx.gmail.com>

You got it, that is correct - it looks like a telecaster phone to the CCM
system.

 

  _____  

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Sunday, December 04, 2005 4:07 PM
To: Mark Snow; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] problem with part and css (the none partition)

 

And the other question is....I'm assuming this doesn't affect
IPCommunicator? I ran a test in the lab and it created the PSS according to
line+device. Is this because IP communicator is not a CTI application like
Softphone? Are my results what you would expect?

----- Original Message ----- 

From: Mark Snow <mailto:highspeedsnow at gmail.com>  

To: 'Lelio Fulgenzi' <mailto:lelio at uoguelph.ca>  ;
cisco-voip at puck.nether.net 

Sent: Sunday, December 04, 2005 4:20 PM

Subject: RE: [cisco-voip] problem with part and css (the none partition)

 

The answer probably lies with the PM, but I don't know why or when .

 


  _____  


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Sunday, December 04, 2005 2:04 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] problem with part and css (the none partition)

 

Is there any particular reason why they've gone this way with the CTIs? Are
they eventually going to change it?

 

----- Original Message ----- 

From: Davis, <mailto:Michael_Davis at eLoyalty.com>  Michael 

To: cisco-voip at puck.nether.net 

Sent: Sunday, December 04, 2005 3:02 PM

Subject: RE: [cisco-voip] problem with part and css (the none partition)

 

I don't have a whole lot of experience with 3.1, but I have recently been
"burnt" by forgetting the CTI vs non-CTI search order. :-)

-----Original Message----- 
From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: Sun 12/4/2005 1:47 PM 
To: Davis, Michael; cisco-voip at puck.nether.net 
Cc: 
Subject: RE: [cisco-voip] problem with part and css (the none partition)



Good Point!

For those of you out there running versions prior to 3.1 when it was still
'Call Mangler'.

Also if you are running those previous versions, try Forwarding your phone
to it's own DN and then calling it from another phone - just for fun! ;)

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Davis, Michael
> Sent: Sunday, December 04, 2005 1:36 PM
> To: cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] problem with part and css (the none partition)
>
> Also,
> Prior to CCM 3.1, Device Level CSS were placed ahead of Line Level CSS.
As of
> 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT for CTI
devices
> (Softphone, CTI Ports, Route Points, etc,) which use the old method.
>
> Mike
>
>
>       -----Original Message-----
>       From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow
>       Sent: Sun 12/4/2005 1:13 PM
>       To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
>       Cc:
>       Subject: RE: [cisco-voip] problem with part and css (the none
partition)
>
>
>
>       Guys,
>
>
>
>       The <none> partition is actually a real partition, it is there by
default and
> is assigned to the very bottom of every CSS.
>
>
>
>       So if you had a RP of '91X' in the INTERAL_PT belonging to the
> HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you also had a RP
of
> '911' in in the <none> Partition (PT) and the user at Phone 1 at HQ dialed
911, he
> would actually be routed to the 911 RP in the <none> PT due to the fact
that it
> appears last in the HQ_911_CSS and is a longer match.
>
>
>
>       It is similar to the <null> MRG, which belongs at the very bottom of
every
> MRGL.
>
>
>
>       MTC and HTH some :-)
>
>
>
>       -Mark Snow
>
>       CCIE Voice # 14073
>
>
>
>
>   _____
>
>
>       From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
>       Sent: Sunday, December 04, 2005 12:19 PM
>       To: cisco-voip at puck.nether.net
>       Subject: Re: [cisco-voip] problem with part and css (the none
partition)
>
>
>
>       Oh yeah, and don't forget the <None> partition. It's not really a
partition,
> but it encompasses as DNs and patterns that have not been assigned to a
> partition. There is no easy way to block access to these DNs unless you
create
> duplicate entries in another partition and block those (which we have
done). I
> can't remember the search algorithm and whether the <None> partition is
simply
> placed last in the ordered list and searched accordingly or if it is
searched only
> after no match is made in the other partitions. A search on CCO docs could
tell
> you this probably.
>
>
>
>       We've moved towards a dialplan model that has next to no DNs in the
> <None> partition. The only ones we have in there are our campus emergency
> number and our helpline number. In the event that a phone goes out
> misconfigured, chances are very high they will be able to call these
numbers.
>
>
>
>               ----- Original Message -----
>
>               From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
>
>               To: James Grace <mailto:jgrace at digitelusa.net>  ; cisco-
> voip at puck.nether.net
>
>               Sent: Sunday, December 04, 2005 1:13 PM
>
>               Subject: Re: [cisco-voip] problem with part and css
>
>
>
>               Think of it this way, a calling search space is simply an
ordered
> list of partitions which the line or device searches through for a match
of the
> dialed digits. If you look through the traces, you will see the term PSS
which I
> believe stands for Phone Search Space. This is made up of the line's
ordered list of
> partitions and then the device's ordered list of partitions. For example:
>
>
>
>               CSS1=PartA,PartB,PartC
>
>               CSS2=PartX,PartY,PartZ
>
>               Device assigned CSS1
>
>               Line assigned CSS2
>
>               the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC
>
>
>
>               So the idea you propose is a valid one. We use it here.
Assign the
> device access to everything and then deny the line access to particular
services.
> We isolate the 'everything' to PSTN access and actually allow oncampus
access
> through the line's partition.
>
>
>
>               You have to make sure that the denies are at least as
specific as
> the allows. And by denies, I mean, creating a route pattern that has the
'block'
> option checked.
>
>
>
>               Post back if you need more...
>
>
>
>
>
>
>
>                       ----- Original Message -----
>
>                       From: James Grace <mailto:jgrace at digitelusa.net>
>
>                       To: cisco-voip at puck.nether.net
>
>                       Sent: Sunday, December 04, 2005 12:35 PM
>
>                       Subject: [cisco-voip] problem with part and css
>
>
>
>                       we have a cm 4.1.2 es 41
>
>                       i was just told by a tac rep that 4.1. and higher
uses both
> css on line and device.
>
>                       what does this mean????
>
>
>
>                       i have my device as unrestricted and all of the part
in it.
> and then i have my line configure with a customize css, ie maybe no int
calls or LD
> calls.
>
>                       the problem im having is that when i make a ld or
int call,
> uses the device.  and when i make a ipphone call ( 4 dgit dialing between
sites) it
> uses the line. I dont understand this.  nore do i understand  the
statement   IT
> USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.
> Thanks in advance
>
>
>   _____
>
>
>                       _______________________________________________
>                       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 Michael_Davis at eLoyalty.com  Sun Dec  4 17:55:06 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Sun Dec  4 17:55:19 2005
Subject: [cisco-voip] problem with part and css (the none partition)
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E326@elhqex01.eloyaltyco.com>

IIRC,  the IP Communicator was at some point based on the old Selsius Cisco 30 SP+ rather than CTI. Someone will correct me if I'm wrong.  I know this is definitely the case for CAD Media Termination clients, but am not certain about IP Communicator.
 
Either way, IP Communicator is configured as a hardware device and uses SCCP rather than JTAPI for control.  So in my mind it is a Software only IP Phone, not a "true" JTAPI controlled CTI device. So your testing is consistent with what I've seen.  However, things like IP Blue, Cisco Softphone,  CTI Ports/CTI Route Points and the JTAPI apps such as IPCC/IPCCX, etc) that control them are going to be affected.
 
I know the original post was specific to IP Communicator, but with all the discussion around how CSSes are processed, I thought I should add the caveat.
 
HTH clarify,
 
Mike
 

	-----Original Message----- 
	From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi 
	Sent: Sun 12/4/2005 4:06 PM 
	To: Mark Snow; cisco-voip at puck.nether.net 
	Cc: 
	Subject: Re: [cisco-voip] problem with part and css (the none partition)
	
	
	And the other question is....I'm assuming this doesn't affect IPCommunicator? I ran a test in the lab and it created the PSS according to line+device. Is this because IP communicator is not a CTI application like Softphone? Are my results what you would expect?

		----- Original Message ----- 
		From: Mark Snow <mailto:highspeedsnow at gmail.com>  
		To: 'Lelio Fulgenzi' <mailto:lelio at uoguelph.ca>  ; cisco-voip at puck.nether.net 
		Sent: Sunday, December 04, 2005 4:20 PM
		Subject: RE: [cisco-voip] problem with part and css (the none partition)


		The answer probably lies with the PM, but I don?t know why or when ?

		 

		
  _____  


		From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
		Sent: Sunday, December 04, 2005 2:04 PM
		To: cisco-voip at puck.nether.net
		Subject: Re: [cisco-voip] problem with part and css (the none partition)

		 

		Is there any particular reason why they've gone this way with the CTIs? Are they eventually going to change it?

		 

			----- Original Message ----- 

			From: Davis, Michael <mailto:Michael_Davis at eLoyalty.com>  

			To: cisco-voip at puck.nether.net 

			Sent: Sunday, December 04, 2005 3:02 PM

			Subject: RE: [cisco-voip] problem with part and css (the none partition)

			 

			I don't have a whole lot of experience with 3.1, but I have recently been "burnt" by forgetting the CTI vs non-CTI search order. :-)
			
			-----Original Message----- 
			From: Mark Snow [mailto:highspeedsnow at gmail.com] 
			Sent: Sun 12/4/2005 1:47 PM 
			To: Davis, Michael; cisco-voip at puck.nether.net 
			Cc: 
			Subject: RE: [cisco-voip] problem with part and css (the none partition)
			
			
			
			Good Point!
			
			For those of you out there running versions prior to 3.1 when it was still
			'Call Mangler'.
			
			Also if you are running those previous versions, try Forwarding your phone
			to it's own DN and then calling it from another phone - just for fun! ;)
			
			> -----Original Message-----
			> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
			> bounces at puck.nether.net] On Behalf Of Davis, Michael
			> Sent: Sunday, December 04, 2005 1:36 PM
			> To: cisco-voip at puck.nether.net
			> Subject: RE: [cisco-voip] problem with part and css (the none partition)
			>
			> Also,
			> Prior to CCM 3.1, Device Level CSS were placed ahead of Line Level CSS.
			As of
			> 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT for CTI
			devices
			> (Softphone, CTI Ports, Route Points, etc,) which use the old method.
			>
			> Mike
			>
			>
			>       -----Original Message-----
			>       From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow
			>       Sent: Sun 12/4/2005 1:13 PM
			>       To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
			>       Cc:
			>       Subject: RE: [cisco-voip] problem with part and css (the none
			partition)
			>
			>
			>
			>       Guys,
			>
			>
			>
			>       The <none> partition is actually a real partition, it is there by
			default and
			> is assigned to the very bottom of every CSS.
			>
			>
			>
			>       So if you had a RP of '91X' in the INTERAL_PT belonging to the
			> HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you also had a RP
			of
			> '911' in in the <none> Partition (PT) and the user at Phone 1 at HQ dialed
			911, he
			> would actually be routed to the 911 RP in the <none> PT due to the fact
			that it
			> appears last in the HQ_911_CSS and is a longer match.
			>
			>
			>
			>       It is similar to the <null> MRG, which belongs at the very bottom of
			every
			> MRGL.
			>
			>
			>
			>       MTC and HTH some :-)
			>
			>
			>
			>       -Mark Snow
			>
			>       CCIE Voice # 14073
			>
			>
			>
			>
			>   _____
			>
			>
			>       From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
			> bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
			>       Sent: Sunday, December 04, 2005 12:19 PM
			>       To: cisco-voip at puck.nether.net
			>       Subject: Re: [cisco-voip] problem with part and css (the none
			partition)
			>
			>
			>
			>       Oh yeah, and don't forget the <None> partition. It's not really a
			partition,
			> but it encompasses as DNs and patterns that have not been assigned to a
			> partition. There is no easy way to block access to these DNs unless you
			create
			> duplicate entries in another partition and block those (which we have
			done). I
			> can't remember the search algorithm and whether the <None> partition is
			simply
			> placed last in the ordered list and searched accordingly or if it is
			searched only
			> after no match is made in the other partitions. A search on CCO docs could
			tell
			> you this probably.
			>
			>
			>
			>       We've moved towards a dialplan model that has next to no DNs in the
			> <None> partition. The only ones we have in there are our campus emergency
			> number and our helpline number. In the event that a phone goes out
			> misconfigured, chances are very high they will be able to call these
			numbers.
			>
			>
			>
			>               ----- Original Message -----
			>
			>               From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
			>
			>               To: James Grace <mailto:jgrace at digitelusa.net>  ; cisco-
			> voip at puck.nether.net
			>
			>               Sent: Sunday, December 04, 2005 1:13 PM
			>
			>               Subject: Re: [cisco-voip] problem with part and css
			>
			>
			>
			>               Think of it this way, a calling search space is simply an
			ordered
			> list of partitions which the line or device searches through for a match
			of the
			> dialed digits. If you look through the traces, you will see the term PSS
			which I
			> believe stands for Phone Search Space. This is made up of the line's
			ordered list of
			> partitions and then the device's ordered list of partitions. For example:
			>
			>
			>
			>               CSS1=PartA,PartB,PartC
			>
			>               CSS2=PartX,PartY,PartZ
			>
			>               Device assigned CSS1
			>
			>               Line assigned CSS2
			>
			>               the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC
			>
			>
			>
			>               So the idea you propose is a valid one. We use it here.
			Assign the
			> device access to everything and then deny the line access to particular
			services.
			> We isolate the 'everything' to PSTN access and actually allow oncampus
			access
			> through the line's partition.
			>
			>
			>
			>               You have to make sure that the denies are at least as
			specific as
			> the allows. And by denies, I mean, creating a route pattern that has the
			'block'
			> option checked.
			>
			>
			>
			>               Post back if you need more...
			>
			>
			>
			>
			>
			>
			>
			>                       ----- Original Message -----
			>
			>                       From: James Grace <mailto:jgrace at digitelusa.net>
			>
			>                       To: cisco-voip at puck.nether.net
			>
			>                       Sent: Sunday, December 04, 2005 12:35 PM
			>
			>                       Subject: [cisco-voip] problem with part and css
			>
			>
			>
			>                       we have a cm 4.1.2 es 41
			>
			>                       i was just told by a tac rep that 4.1. and higher
			uses both
			> css on line and device.
			>
			>                       what does this mean????
			>
			>
			>
			>                       i have my device as unrestricted and all of the part
			in it.
			> and then i have my line configure with a customize css, ie maybe no int
			calls or LD
			> calls.
			>
			>                       the problem im having is that when i make a ld or
			int call,
			> uses the device.  and when i make a ipphone call ( 4 dgit dialing between
			sites) it
			> uses the line. I dont understand this.  nore do i understand  the
			statement   IT
			> USES BOTH LINE AND DEVICE.   can someone Please clear this up for me.
			> Thanks in advance
			>
			>
			>   _____
			>
			>
			>                       _______________________________________________
			>                       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



From jburton at NETechCorp.com  Sun Dec  4 18:16:18 2005
From: jburton at NETechCorp.com (Jason Burton)
Date: Sun Dec  4 18:16:25 2005
Subject: [cisco-voip] Not a technical question.
Message-ID: <FDC137145729AD438CDCA0F9FC6F7236339557@mail.netechcorp.com>

Does anyone have a Visio stencil for VIC2-4FXO?  I couldn't find it on Cisco's stencil page.
 
Thanks!


From mfa at crec.ifas.ufl.edu  Sun Dec  4 19:27:33 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Sun Dec  4 19:27:39 2005
Subject: [cisco-voip] CDR question
References: <200512041700.jB4H04fu013054@puck.nether.net>
Message-ID: <001701c5f932$b040b980$0402a8c0@crec.ifas.ufl.edu>

Good point.  I do almost everything in Excel, and still don't have a nice 
way to deal with DST.  Of course, with Congress screwing around with the 
algorithm, very few programs are going to be able to deal with it nicely... 
:~).

Happy holidays all,

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

> Date: Sun, 4 Dec 2005 02:49:13 -0500
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: Re: [cisco-voip] CDR question..
> To: "Stu Packett" <SPackett at fenwick.com>, "Tim Reimers"
> <tim.reimers at asheville.k12.nc.us>, <cisco-voip at puck.nether.net>
> Message-ID: <001901c5f8a7$38b0e0b0$ca01a8c0 at deadfish>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Something to worry about with Excel is the conversion of daylight saving 
> time correctly. Our LD billing office uses Excel and they have to do some 
> manual modifications to account for it.
>
> Just a thought.
>
>  ----- Original Message ----- 
>  From: Stu Packett
>  To: Tim Reimers ; cisco-voip at puck.nether.net
>  Sent: Sunday, December 04, 2005 2:44 AM
>  Subject: RE: [cisco-voip] CDR question..
>
>
>  I do similar CDR queries as well when someone inadvertantly dials 911.  I 
> have an Excel spreadsheet that makes it easier for me.  The date that the 
> CDR reports is the number of seconds (or minutes?) from January 1, 1970. 
> So I have a field in Excel that says:  =A1/86400+DATE(1970,1,1) that 
> converts the numbers into dates.  If this isn't that clear, I can send my 
> CDR template for Excel that I always use.
>
>  The CDR query I would run in SQL Query Analzyer is:
>  select *
>  from CallDetailRecord
>  where finalCalledPartyNumber LIKE '91800%'
>
>  Hope this helps.
>
>
>
> ------------------------------------------------------------------------------
>  From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
>  Sent: Wednesday, November 30, 2005 8:26 AM
>  To: cisco-voip at puck.nether.net
>  Subject: [cisco-voip] CDR question..
>
>
>  Hi everyone---
>
>
>
>  I have a CDR newbie kind of question---
>
>
>
>  Can someone give me the steps needed to determine the following:  (CCM 
> 4.0)
>
>
>
>  All calls to a given 800 number in say, the last two months.
>
>
>
>  Number of calls
>
>  Duration of each call
>
>
>
>  If I can get it into Excel I can munge around the data to get the right 
> things I want in terms of averages, etc.
>
>
>
>  But I've not done any CDR stuff as yet..
>
>
>
>
> ------------------------------------------------------------------------------


From Andrius.Kislas at elsis.lt  Mon Dec  5 01:38:51 2005
From: Andrius.Kislas at elsis.lt (Andrius Kislas)
Date: Mon Dec  5 01:38:09 2005
Subject: [cisco-voip] One CM and several identical networks
Message-ID: <4393E07B.2060905@elsis.lt>

Hello,

I have a question about CallManager working as a SCCP signaling provider
for several networks using VPN over Internet. It so happens that two
networks are with the same IP adresses (192.168.0.x/24). What is the
"best practice" in such situation for CM to distinguish these networks?
I thought about two solutions:

1) Put a secondary IP address on CM for the second identical network.
Then the make policy routing on a router, so that packets FROM one CM
address goes to the first network, and packets FROM the second address
goes to the second network.

2) Configure NAT on VPN tunnels, so that packets from first network
reaches CM as one IP address and packets from second networks reaches CM
with second address.

I haven't tried neither of these solutions. Maybe there is more
intelligent way of solving the problem?

Andrius

From sean at northrock.bm  Mon Dec  5 08:07:18 2005
From: sean at northrock.bm (Sean Watkins)
Date: Mon Dec  5 08:07:27 2005
Subject: [cisco-voip] Cisco CME conference limits
Message-ID: <B146219F50F53348A6EF98F14B4202C635D4F2@sumo2.northrock.bm>

Is there any such thing as a "CME User FAQ" or something somewhere. There is
CME info scattered all over CCO; but nothing concrete on real world
setups/examples
 
Sean
 

  _____  

From: Bob A. Bowie [mailto:bababooey at cox.net] 
Sent: Saturday, December 03, 2005 2:21 PM
To: Mark Snow; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Cisco CME conference limits


Mark, Do you have a link to 'Conference Express' desdcription?
Is it a feature under the IOS or a separate hardware solution?

----- Original Message ----- 
From: Mark  <mailto:highspeedsnow at gmail.com> Snow 
To: 'Bob A. Bowie' <mailto:bababooey at cox.net>  ; cisco-voip at puck.nether.net
<mailto:cisco-voip at puck.nether.net>  
Sent: Saturday, December 03, 2005 11:12 AM
Subject: RE: [cisco-voip] Cisco CME conference limits


The phones are limited to a 3 party conference if you are using the CME
software to enable the conferencing.

Conference Express is also available for your use on a CME system, enabling
you conference ID's and more members in the conf bridge.

 

-Mark Snow

 


  _____  


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 Bob A. Bowie
Sent: Friday, December 02, 2005 11:27 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco CME conference limits

 

Is the phones on a cme system limited to 3 party conferference?

 

Do I need dsp resources on a cme system that is on the ip wan? or are they
only needed to transcode pstn lines?

 

thanks

 

 

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From rratliff at cisco.com  Mon Dec  5 10:03:59 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Dec  5 10:04:03 2005
Subject: [cisco-voip] problem with part and css (the none partition)
In-Reply-To: <037594FB5AFC354FA38653EB3FF3242105A8E326@elhqex01.eloyaltyco.com>
References: <037594FB5AFC354FA38653EB3FF3242105A8E326@elhqex01.eloyaltyco.com>
Message-ID: <DC01E5B9-46DD-433A-89DE-1DCA0E924440@cisco.com>

IPC is actually based on the 7970 IP phone.  You are correct it talks  
skinny, not CTI/Jtapi.


-Ryan

On Dec 4, 2005, at 5:55 PM, Davis, Michael wrote:

IIRC,  the IP Communicator was at some point based on the old Selsius  
Cisco 30 SP+ rather than CTI. Someone will correct me if I'm wrong.   
I know this is definitely the case for CAD Media Termination clients,  
but am not certain about IP Communicator.

Either way, IP Communicator is configured as a hardware device and  
uses SCCP rather than JTAPI for control.  So in my mind it is a  
Software only IP Phone, not a "true" JTAPI controlled CTI device. So  
your testing is consistent with what I've seen.  However, things like  
IP Blue, Cisco Softphone,  CTI Ports/CTI Route Points and the JTAPI  
apps such as IPCC/IPCCX, etc) that control them are going to be  
affected.

I know the original post was specific to IP Communicator, but with  
all the discussion around how CSSes are processed, I thought I should  
add the caveat.

HTH clarify,

Mike


	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
	Sent: Sun 12/4/2005 4:06 PM
	To: Mark Snow; cisco-voip at puck.nether.net
	Cc:
	Subject: Re: [cisco-voip] problem with part and css (the none  
partition)
	
	
	And the other question is....I'm assuming this doesn't affect  
IPCommunicator? I ran a test in the lab and it created the PSS  
according to line+device. Is this because IP communicator is not a  
CTI application like Softphone? Are my results what you would expect?

		----- Original Message -----
		From: Mark Snow <mailto:highspeedsnow at gmail.com>
		To: 'Lelio Fulgenzi' <mailto:lelio at uoguelph.ca>  ; cisco- 
voip at puck.nether.net
		Sent: Sunday, December 04, 2005 4:20 PM
		Subject: RE: [cisco-voip] problem with part and css (the none  
partition)


		The answer probably lies with the PM, but I don?t know why or when ?

		

		
   _____


		From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
		Sent: Sunday, December 04, 2005 2:04 PM
		To: cisco-voip at puck.nether.net
		Subject: Re: [cisco-voip] problem with part and css (the none  
partition)

		

		Is there any particular reason why they've gone this way with the  
CTIs? Are they eventually going to change it?

		

			----- Original Message -----

			From: Davis, Michael <mailto:Michael_Davis at eLoyalty.com>

			To: cisco-voip at puck.nether.net

			Sent: Sunday, December 04, 2005 3:02 PM

			Subject: RE: [cisco-voip] problem with part and css (the none  
partition)

			

			I don't have a whole lot of experience with 3.1, but I have  
recently been "burnt" by forgetting the CTI vs non-CTI search order. :-)
			
			-----Original Message-----
			From: Mark Snow [mailto:highspeedsnow at gmail.com]
			Sent: Sun 12/4/2005 1:47 PM
			To: Davis, Michael; cisco-voip at puck.nether.net
			Cc:
			Subject: RE: [cisco-voip] problem with part and css (the none  
partition)
			
			
			
			Good Point!
			
			For those of you out there running versions prior to 3.1 when it  
was still
			'Call Mangler'.
			
			Also if you are running those previous versions, try Forwarding  
your phone
			to it's own DN and then calling it from another phone - just for  
fun! ;)
			
			> -----Original Message-----
			> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
			> bounces at puck.nether.net] On Behalf Of Davis, Michael
			> Sent: Sunday, December 04, 2005 1:36 PM
			> To: cisco-voip at puck.nether.net
			> Subject: RE: [cisco-voip] problem with part and css (the none  
partition)
			>
			> Also,
			> Prior to CCM 3.1, Device Level CSS were placed ahead of Line  
Level CSS.
			As of
			> 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT  
for CTI
			devices
			> (Softphone, CTI Ports, Route Points, etc,) which use the old  
method.
			>
			> Mike
			>
			>
			>       -----Original Message-----
			>       From: cisco-voip-bounces at puck.nether.net on behalf of Mark  
Snow
			>       Sent: Sun 12/4/2005 1:13 PM
			>       To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
			>       Cc:
			>       Subject: RE: [cisco-voip] problem with part and css (the none
			partition)
			>
			>
			>
			>       Guys,
			>
			>
			>
			>       The <none> partition is actually a real partition, it is  
there by
			default and
			> is assigned to the very bottom of every CSS.
			>
			>
			>
			>       So if you had a RP of '91X' in the INTERAL_PT belonging to  
the
			> HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you  
also had a RP
			of
			> '911' in in the <none> Partition (PT) and the user at Phone 1 at  
HQ dialed
			911, he
			> would actually be routed to the 911 RP in the <none> PT due to  
the fact
			that it
			> appears last in the HQ_911_CSS and is a longer match.
			>
			>
			>
			>       It is similar to the <null> MRG, which belongs at the very  
bottom of
			every
			> MRGL.
			>
			>
			>
			>       MTC and HTH some :-)
			>
			>
			>
			>       -Mark Snow
			>
			>       CCIE Voice # 14073
			>
			>
			>
			>
			>   _____
			>
			>
			>       From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
			> bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
			>       Sent: Sunday, December 04, 2005 12:19 PM
			>       To: cisco-voip at puck.nether.net
			>       Subject: Re: [cisco-voip] problem with part and css (the none
			partition)
			>
			>
			>
			>       Oh yeah, and don't forget the <None> partition. It's not  
really a
			partition,
			> but it encompasses as DNs and patterns that have not been  
assigned to a
			> partition. There is no easy way to block access to these DNs  
unless you
			create
			> duplicate entries in another partition and block those (which we  
have
			done). I
			> can't remember the search algorithm and whether the <None>  
partition is
			simply
			> placed last in the ordered list and searched accordingly or if  
it is
			searched only
			> after no match is made in the other partitions. A search on CCO  
docs could
			tell
			> you this probably.
			>
			>
			>
			>       We've moved towards a dialplan model that has next to no  
DNs in the
			> <None> partition. The only ones we have in there are our campus  
emergency
			> number and our helpline number. In the event that a phone goes out
			> misconfigured, chances are very high they will be able to call  
these
			numbers.
			>
			>
			>
			>               ----- Original Message -----
			>
			>               From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
			>
			>               To: James Grace <mailto:jgrace at digitelusa.net>  ;  
cisco-
			> voip at puck.nether.net
			>
			>               Sent: Sunday, December 04, 2005 1:13 PM
			>
			>               Subject: Re: [cisco-voip] problem with part and css
			>
			>
			>
			>               Think of it this way, a calling search space is  
simply an
			ordered
			> list of partitions which the line or device searches through for  
a match
			of the
			> dialed digits. If you look through the traces, you will see the  
term PSS
			which I
			> believe stands for Phone Search Space. This is made up of the  
line's
			ordered list of
			> partitions and then the device's ordered list of partitions. For  
example:
			>
			>
			>
			>               CSS1=PartA,PartB,PartC
			>
			>               CSS2=PartX,PartY,PartZ
			>
			>               Device assigned CSS1
			>
			>               Line assigned CSS2
			>
			>               the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC
			>
			>
			>
			>               So the idea you propose is a valid one. We use it  
here.
			Assign the
			> device access to everything and then deny the line access to  
particular
			services.
			> We isolate the 'everything' to PSTN access and actually allow  
oncampus
			access
			> through the line's partition.
			>
			>
			>
			>               You have to make sure that the denies are at least as
			specific as
			> the allows. And by denies, I mean, creating a route pattern that  
has the
			'block'
			> option checked.
			>
			>
			>
			>               Post back if you need more...
			>
			>
			>
			>
			>
			>
			>
			>                       ----- Original Message -----
			>
			>                       From: James Grace  
<mailto:jgrace at digitelusa.net>
			>
			>                       To: cisco-voip at puck.nether.net
			>
			>                       Sent: Sunday, December 04, 2005 12:35 PM
			>
			>                       Subject: [cisco-voip] problem with part  
and css
			>
			>
			>
			>                       we have a cm 4.1.2 es 41
			>
			>                       i was just told by a tac rep that 4.1. and  
higher
			uses both
			> css on line and device.
			>
			>                       what does this mean????
			>
			>
			>
			>                       i have my device as unrestricted and all  
of the part
			in it.
			> and then i have my line configure with a customize css, ie maybe  
no int
			calls or LD
			> calls.
			>
			>                       the problem im having is that when i make  
a ld or
			int call,
			> uses the device.  and when i make a ipphone call ( 4 dgit  
dialing between
			sites) it
			> uses the line. I dont understand this.  nore do i understand  the
			statement   IT
			> USES BOTH LINE AND DEVICE.   can someone Please clear this up  
for me.
			> Thanks in advance
			>
			>
			>   _____
			>
			>
			>                        
_______________________________________________
			>                       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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			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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From rratliff at cisco.com  Mon Dec  5 10:06:55 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Dec  5 10:07:00 2005
Subject: [cisco-voip] One CM and several identical networks
In-Reply-To: <4393E07B.2060905@elsis.lt>
References: <4393E07B.2060905@elsis.lt>
Message-ID: <CB7389F8-ECCF-4656-B2E6-9CE4173517F1@cisco.com>

Your CM server should never have multiple IP addresses.   I've  
certainly no experience deploying IPT but it sounds to me like NAT  
with skinny/H.323 fixup is what you need.

Did I mention to absolutely NEVER put multiple IP addresses on your  
CM?   Well, at least if you want it to work...

-Ryan

On Dec 5, 2005, at 1:38 AM, Andrius Kislas wrote:

Hello,

I have a question about CallManager working as a SCCP signaling provider
for several networks using VPN over Internet. It so happens that two
networks are with the same IP adresses (192.168.0.x/24). What is the
"best practice" in such situation for CM to distinguish these networks?
I thought about two solutions:

1) Put a secondary IP address on CM for the second identical network.
Then the make policy routing on a router, so that packets FROM one CM
address goes to the first network, and packets FROM the second address
goes to the second network.

2) Configure NAT on VPN tunnels, so that packets from first network
reaches CM as one IP address and packets from second networks reaches CM
with second address.

I haven't tried neither of these solutions. Maybe there is more
intelligent way of solving the problem?

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


From ml at nesys.it  Mon Dec  5 10:29:41 2005
From: ml at nesys.it (Andrea Riela)
Date: Mon Dec  5 10:29:55 2005
Subject: [cisco-voip] about SIP UA
Message-ID: <f7994dc4af812a75515d9842a35ad568@nesys.it>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

hi folks,

on CCME I could configure only one sip-ua ... the question is: I don't 
really know CCM, but it permits multiple sip-ua registrations?
Thanks for your support
Regards
Andrea
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Version: GnuPG v1.4.1 (Darwin)

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QI2qv/ic9ZlfcapG8cdMQXI=
=hbFm
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From techguy at gmail.com  Mon Dec  5 10:51:29 2005
From: techguy at gmail.com (TechGuy)
Date: Mon Dec  5 10:51:32 2005
Subject: [cisco-voip] ATA versus VG224
Message-ID: <a1b07ea60512050751v1273342fx2496553b001dbcc7@mail.gmail.com>

Can someone share with me thoughts on ATA's versus VG224.  I have heard that
the VG224 is a little more reliable and less problamatic then the ATA's

I believe the VG224 is a little more flexible in how you can configure it?
Mainly looking for something to handle a bunch of fax machines.  Looking for
the most reliable MGCP solution.
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From CORTIZ at broward.org  Mon Dec  5 11:07:11 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon Dec  5 11:07:18 2005
Subject: FW: [cisco-voip] ATA versus VG224
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2AF5@MREXVS01.bc.broward.cty>

 

 

________________________________

From: Ortiz, Carlos 
Sent: Monday, December 05, 2005 11:06 AM
To: 'TechGuy'
Subject: RE: [cisco-voip] ATA versus VG224

 

The VG224's work really well with MGCP.  The only issue I have had is
that there is a bug with AAA on the VG224 that is addressed in newer
versions of code, but that would require a memory upgrade.  The best
thing I can tell you about them is that I installed one about 1 year ago
and another about 9 months ago.  Haven't touched them since..... :-)

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Monday, December 05, 2005 10:51 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ATA versus VG224

 

Can someone share with me thoughts on ATA's versus VG224.  I have heard
that the VG224 is a little more reliable and less problamatic then the
ATA's

 

I believe the VG224 is a little more flexible in how you can configure
it?  Mainly looking for something to handle a bunch of fax machines.
Looking for the most reliable MGCP solution.

 

 

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From john.lange at open-it.ca  Mon Dec  5 11:36:39 2005
From: john.lange at open-it.ca (John Lange)
Date: Mon Dec  5 11:36:48 2005
Subject: [cisco-voip] 7960 hijacks the user interface when calls come in
Message-ID: <1133800599.12048.45.camel@ws50.darkcore.net>

We have a problem with our Cisco 7960s using the SIP load.

While trying to transfer a call, if another call comes in, the entire
user interface is "hijacked" and you can not complete the transfer
process.

The only option available on the softkeys is "accept" (for the new
call), so you can not complete the transfer.

This is a serious problem especially when trying to use the 7960 as a
switchboard phone causing many calls to be lost in "transfer limbo".

As a side issue, is there any way to reprogram the layout of the
softkeys on the 7960? Having to press "more" to get to "transfer" is
just silly.

-- 
John Lange



From techguy at gmail.com  Mon Dec  5 11:47:53 2005
From: techguy at gmail.com (TechGuy)
Date: Mon Dec  5 11:47:56 2005
Subject: [cisco-voip] ATA versus VG224
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2AF4@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2AF4@MREXVS01.bc.broward.cty>
Message-ID: <a1b07ea60512050847p26bea5efn88f5658fd3154463@mail.gmail.com>

Can anyone share with me some benefits of using a VG224 over a bunch of
ATA's?  Particularly interested in the pro's of a VG224 for fax machines
over the use of ATA's?




On 12/5/05, Ortiz, Carlos <CORTIZ at broward.org> wrote:
>
>  The VG224's work really well with MGCP.  The only issue I have had is
> that there is a bug with AAA on the VG224 that is addressed in newer
> versions of code, but that would require a memory upgrade.  The best thing I
> can tell you about them is that I installed one about 1 year ago and another
> about 9 months ago.  Haven't touched them since?.. J
>
>
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *TechGuy
> *Sent:* Monday, December 05, 2005 10:51 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] ATA versus VG224
>
>
>
> Can someone share with me thoughts on ATA's versus VG224.  I have heard
> that the VG224 is a little more reliable and less problamatic then the ATA's
>
>
>
> I believe the VG224 is a little more flexible in how you can configure
> it?  Mainly looking for something to handle a bunch of fax machines.
> Looking for the most reliable MGCP solution.
>
>
>
>
>
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From Court.Schuett at us.millwardbrown.com  Mon Dec  5 11:49:55 2005
From: Court.Schuett at us.millwardbrown.com (Schuett, Court (MBNAP it))
Date: Mon Dec  5 11:50:07 2005
Subject: [cisco-voip] 7960 hijacks the user interface when calls come in
Message-ID: <90B962BEA7D42F40BF0DF00044A5A1B60B32F56F@x-np5.am.group.local>

Device -> Device Settings -> Softkey Template

Under there, find the softkey template assigned to the phone in question.
Then, click 'Configure Softkey Layout'.  You should be able to change
everything you need in there.  There is a button for 'Ring In' which is
where you'll probably need to add 'Transfer' to.

Hope that helps.

 


Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Lange
Sent: Monday, December 05, 2005 10:37 AM
To: Cisco-VoIP
Subject: [cisco-voip] 7960 hijacks the user interface when calls come in

We have a problem with our Cisco 7960s using the SIP load.

While trying to transfer a call, if another call comes in, the entire user
interface is "hijacked" and you can not complete the transfer process.

The only option available on the softkeys is "accept" (for the new call), so
you can not complete the transfer.

This is a serious problem especially when trying to use the 7960 as a
switchboard phone causing many calls to be lost in "transfer limbo".

As a side issue, is there any way to reprogram the layout of the softkeys on
the 7960? Having to press "more" to get to "transfer" is just silly.

--
John Lange


_______________________________________________
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  Mon Dec  5 12:08:55 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec  5 12:08:54 2005
Subject: [cisco-voip] ATA versus VG224
Message-ID: <D713462ED535184D830F6F6301753E7E01A11D8A@VISHNU.wesd.org>

ATA's can become problematic.  They don't support anything but fax
passthrough.  If you have any super G3 (33.6) fax machines you have to
dumb them down to make them work and ECM has to be disable.  If you can
afford the VG go VG.  IF you can afford Pots lines, go Pots lines.

 

Just my 2 cents.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Monday, December 05, 2005 8:48 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ATA versus VG224

 

Can anyone share with me some benefits of using a VG224 over a bunch of
ATA's?  Particularly interested in the pro's of a VG224 for fax machines
over the use of ATA's?

 



 

On 12/5/05, Ortiz, Carlos <CORTIZ at broward.org> wrote: 

The VG224's work really well with MGCP.  The only issue I have had is
that there is a bug with AAA on the VG224 that is addressed in newer
versions of code, but that would require a memory upgrade.  The best
thing I can tell you about them is that I installed one about 1 year ago
and another about 9 months ago.  Haven't touched them since..... :-) 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Monday, December 05, 2005 10:51 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ATA versus VG224

 

Can someone share with me thoughts on ATA's versus VG224.  I have heard
that the VG224 is a little more reliable and less problamatic then the
ATA's 

 

I believe the VG224 is a little more flexible in how you can configure
it?  Mainly looking for something to handle a bunch of fax machines.
Looking for the most reliable MGCP solution. 

 

 

 

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From highspeedsnow at gmail.com  Mon Dec  5 12:09:50 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Mon Dec  5 12:10:23 2005
Subject: [cisco-voip] ATA versus VG224
In-Reply-To: <a1b07ea60512050847p26bea5efn88f5658fd3154463@mail.gmail.com>
Message-ID: <4394747c.4441e16c.623e.ffffa349@mx.gmail.com>

The ATA is a very low-end device. It will support Faxing, but only up to
9600 baud (need to hardcode your Fax machine down to this speed, and turn
ECM off). The ATA will not support a modem (though folks have reported
credit card machines working with an ATA).

The VG248 supports faster Fax speeds of 14.4 and low end modems.

The VG224 is the newest device, has the newest DSP's and supports higher
speed faxing and higher speed modems - though neither at V.90 speeds or
fully supporting the SuperG3 faxing speeds.

 

Make sure that you are running all your gateways with a IOS type NSE payload
(100) and that you are running Fax-Passthrough on every device, and not fax
relay.

 

-Mark Snow

CCIE Voice #14073

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Monday, December 05, 2005 11:48 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ATA versus VG224

 

Can anyone share with me some benefits of using a VG224 over a bunch of
ATA's?  Particularly interested in the pro's of a VG224 for fax machines
over the use of ATA's?

 



 

On 12/5/05, Ortiz, Carlos <CORTIZ at broward.org> wrote: 

The VG224's work really well with MGCP.  The only issue I have had is that
there is a bug with AAA on the VG224 that is addressed in newer versions of
code, but that would require a memory upgrade.  The best thing I can tell
you about them is that I installed one about 1 year ago and another about 9
months ago.  Haven't touched them since... :-) 

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Monday, December 05, 2005 10:51 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ATA versus VG224

 

Can someone share with me thoughts on ATA's versus VG224.  I have heard that
the VG224 is a little more reliable and less problamatic then the ATA's 

 

I believe the VG224 is a little more flexible in how you can configure it?
Mainly looking for something to handle a bunch of fax machines.  Looking for
the most reliable MGCP solution. 

 

 

 

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From davidp at bostonpost.com  Mon Dec  5 12:15:36 2005
From: davidp at bostonpost.com (David Pitkin)
Date: Mon Dec  5 12:15:40 2005
Subject: [cisco-voip] TBCT with Unity / CM3.1
Message-ID: <4DEBDDAFBB23C04BA17EFE391444211301547884@exchange.bostonpost.com>

Hi,


Anyone ever setup call reflection using the ISDN feature Two B Channel
Transfer on their PRI using unity and call manager?

 

I am worried about having the appropriate CDR records get created. 

 

Thanks,

 

David

 

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From techguy at gmail.com  Mon Dec  5 12:18:50 2005
From: techguy at gmail.com (TechGuy)
Date: Mon Dec  5 12:18:53 2005
Subject: [cisco-voip] User unable to log into Attendant Console
Message-ID: <a1b07ea60512050918kec3bebet9ef1f8656935503d@mail.gmail.com>

I have a user who is unable to login to attendant console, it is reporting
back "invalid username, please try again".  However the user can log into
CCMUser

Any ideas where to start?  I dont know much about attendant console.
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From Michael_Davis at eLoyalty.com  Mon Dec  5 12:24:12 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Mon Dec  5 12:24:27 2005
Subject: [cisco-voip] problem with part and css (the none partition)
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E32B@elhqex01.eloyaltyco.com>

That makes sense.  Thanks for the clarification.

	-----Original Message----- 
	From: Ryan Ratliff [mailto:rratliff at cisco.com] 
	Sent: Mon 12/5/2005 9:03 AM 
	To: Davis, Michael 
	Cc: Lelio Fulgenzi; Mark Snow; cisco-voip at puck.nether.net 
	Subject: Re: [cisco-voip] problem with part and css (the none partition)
	
	

	IPC is actually based on the 7970 IP phone.  You are correct it talks 
	skinny, not CTI/Jtapi.
	
	
	-Ryan
	
	On Dec 4, 2005, at 5:55 PM, Davis, Michael wrote:
	
	IIRC,  the IP Communicator was at some point based on the old Selsius 
	Cisco 30 SP+ rather than CTI. Someone will correct me if I'm wrong.  
	I know this is definitely the case for CAD Media Termination clients, 
	but am not certain about IP Communicator.
	
	Either way, IP Communicator is configured as a hardware device and 
	uses SCCP rather than JTAPI for control.  So in my mind it is a 
	Software only IP Phone, not a "true" JTAPI controlled CTI device. So 
	your testing is consistent with what I've seen.  However, things like 
	IP Blue, Cisco Softphone,  CTI Ports/CTI Route Points and the JTAPI 
	apps such as IPCC/IPCCX, etc) that control them are going to be 
	affected.
	
	I know the original post was specific to IP Communicator, but with 
	all the discussion around how CSSes are processed, I thought I should 
	add the caveat.
	
	HTH clarify,
	
	Mike
	
	
	        -----Original Message-----
	        From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
	        Sent: Sun 12/4/2005 4:06 PM
	        To: Mark Snow; cisco-voip at puck.nether.net
	        Cc:
	        Subject: Re: [cisco-voip] problem with part and css (the none 
	partition)
	       
	       
	        And the other question is....I'm assuming this doesn't affect 
	IPCommunicator? I ran a test in the lab and it created the PSS 
	according to line+device. Is this because IP communicator is not a 
	CTI application like Softphone? Are my results what you would expect?
	
	                ----- Original Message -----
	                From: Mark Snow <mailto:highspeedsnow at gmail.com>
	                To: 'Lelio Fulgenzi' <mailto:lelio at uoguelph.ca>  ; cisco-
	voip at puck.nether.net
	                Sent: Sunday, December 04, 2005 4:20 PM
	                Subject: RE: [cisco-voip] problem with part and css (the none 
	partition)
	
	
	                The answer probably lies with the PM, but I don?t know why or when ?
	
	               
	
	               
	   _____
	
	
	                From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
	bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	                Sent: Sunday, December 04, 2005 2:04 PM
	                To: cisco-voip at puck.nether.net
	                Subject: Re: [cisco-voip] problem with part and css (the none 
	partition)
	
	               
	
	                Is there any particular reason why they've gone this way with the 
	CTIs? Are they eventually going to change it?
	
	               
	
	                        ----- Original Message -----
	
	                        From: Davis, Michael <mailto:Michael_Davis at eLoyalty.com>
	
	                        To: cisco-voip at puck.nether.net
	
	                        Sent: Sunday, December 04, 2005 3:02 PM
	
	                        Subject: RE: [cisco-voip] problem with part and css (the none 
	partition)
	
	                       
	
	                        I don't have a whole lot of experience with 3.1, but I have 
	recently been "burnt" by forgetting the CTI vs non-CTI search order. :-)
	                       
	                        -----Original Message-----
	                        From: Mark Snow [mailto:highspeedsnow at gmail.com]
	                        Sent: Sun 12/4/2005 1:47 PM
	                        To: Davis, Michael; cisco-voip at puck.nether.net
	                        Cc:
	                        Subject: RE: [cisco-voip] problem with part and css (the none 
	partition)
	                       
	                       
	                       
	                        Good Point!
	                       
	                        For those of you out there running versions prior to 3.1 when it 
	was still
	                        'Call Mangler'.
	                       
	                        Also if you are running those previous versions, try Forwarding 
	your phone
	                        to it's own DN and then calling it from another phone - just for 
	fun! ;)
	                       
	                        > -----Original Message-----
	                        > From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
	                        > bounces at puck.nether.net] On Behalf Of Davis, Michael
	                        > Sent: Sunday, December 04, 2005 1:36 PM
	                        > To: cisco-voip at puck.nether.net
	                        > Subject: RE: [cisco-voip] problem with part and css (the none 
	partition)
	                        >
	                        > Also,
	                        > Prior to CCM 3.1, Device Level CSS were placed ahead of Line 
	Level CSS.
	                        As of
	                        > 3.1, Line Level CSSes are placed ahead of Device Level, EXCEPT 
	for CTI
	                        devices
	                        > (Softphone, CTI Ports, Route Points, etc,) which use the old 
	method.
	                        >
	                        > Mike
	                        >
	                        >
	                        >       -----Original Message-----
	                        >       From: cisco-voip-bounces at puck.nether.net on behalf of Mark 
	Snow
	                        >       Sent: Sun 12/4/2005 1:13 PM
	                        >       To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
	                        >       Cc:
	                        >       Subject: RE: [cisco-voip] problem with part and css (the none
	                        partition)
	                        >
	                        >
	                        >
	                        >       Guys,
	                        >
	                        >
	                        >
	                        >       The <none> partition is actually a real partition, it is 
	there by
	                        default and
	                        > is assigned to the very bottom of every CSS.
	                        >
	                        >
	                        >
	                        >       So if you had a RP of '91X' in the INTERAL_PT belonging to 
	the
	                        > HQ_911_CSS and Phone 1 at HQ had this CSS assigned,  but you 
	also had a RP
	                        of
	                        > '911' in in the <none> Partition (PT) and the user at Phone 1 at 
	HQ dialed
	                        911, he
	                        > would actually be routed to the 911 RP in the <none> PT due to 
	the fact
	                        that it
	                        > appears last in the HQ_911_CSS and is a longer match.
	                        >
	                        >
	                        >
	                        >       It is similar to the <null> MRG, which belongs at the very 
	bottom of
	                        every
	                        > MRGL.
	                        >
	                        >
	                        >
	                        >       MTC and HTH some :-)
	                        >
	                        >
	                        >
	                        >       -Mark Snow
	                        >
	                        >       CCIE Voice # 14073
	                        >
	                        >
	                        >
	                        >
	                        >   _____
	                        >
	                        >
	                        >       From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
	                        > bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	                        >       Sent: Sunday, December 04, 2005 12:19 PM
	                        >       To: cisco-voip at puck.nether.net
	                        >       Subject: Re: [cisco-voip] problem with part and css (the none
	                        partition)
	                        >
	                        >
	                        >
	                        >       Oh yeah, and don't forget the <None> partition. It's not 
	really a
	                        partition,
	                        > but it encompasses as DNs and patterns that have not been 
	assigned to a
	                        > partition. There is no easy way to block access to these DNs 
	unless you
	                        create
	                        > duplicate entries in another partition and block those (which we 
	have
	                        done). I
	                        > can't remember the search algorithm and whether the <None> 
	partition is
	                        simply
	                        > placed last in the ordered list and searched accordingly or if 
	it is
	                        searched only
	                        > after no match is made in the other partitions. A search on CCO 
	docs could
	                        tell
	                        > you this probably.
	                        >
	                        >
	                        >
	                        >       We've moved towards a dialplan model that has next to no 
	DNs in the
	                        > <None> partition. The only ones we have in there are our campus 
	emergency
	                        > number and our helpline number. In the event that a phone goes out
	                        > misconfigured, chances are very high they will be able to call 
	these
	                        numbers.
	                        >
	                        >
	                        >
	                        >               ----- Original Message -----
	                        >
	                        >               From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>
	                        >
	                        >               To: James Grace <mailto:jgrace at digitelusa.net>  ; 
	cisco-
	                        > voip at puck.nether.net
	                        >
	                        >               Sent: Sunday, December 04, 2005 1:13 PM
	                        >
	                        >               Subject: Re: [cisco-voip] problem with part and css
	                        >
	                        >
	                        >
	                        >               Think of it this way, a calling search space is 
	simply an
	                        ordered
	                        > list of partitions which the line or device searches through for 
	a match
	                        of the
	                        > dialed digits. If you look through the traces, you will see the 
	term PSS
	                        which I
	                        > believe stands for Phone Search Space. This is made up of the 
	line's
	                        ordered list of
	                        > partitions and then the device's ordered list of partitions. For 
	example:
	                        >
	                        >
	                        >
	                        >               CSS1=PartA,PartB,PartC
	                        >
	                        >               CSS2=PartX,PartY,PartZ
	                        >
	                        >               Device assigned CSS1
	                        >
	                        >               Line assigned CSS2
	                        >
	                        >               the PSS would be: PartX,PartY,PartZ,PartA,PartB,PartC
	                        >
	                        >
	                        >
	                        >               So the idea you propose is a valid one. We use it 
	here.
	                        Assign the
	                        > device access to everything and then deny the line access to 
	particular
	                        services.
	                        > We isolate the 'everything' to PSTN access and actually allow 
	oncampus
	                        access
	                        > through the line's partition.
	                        >
	                        >
	                        >
	                        >               You have to make sure that the denies are at least as
	                        specific as
	                        > the allows. And by denies, I mean, creating a route pattern that 
	has the
	                        'block'
	                        > option checked.
	                        >
	                        >
	                        >
	                        >               Post back if you need more...
	                        >
	                        >
	                        >
	                        >
	                        >
	                        >
	                        >
	                        >                       ----- Original Message -----
	                        >
	                        >                       From: James Grace 
	<mailto:jgrace at digitelusa.net>
	                        >
	                        >                       To: cisco-voip at puck.nether.net
	                        >
	                        >                       Sent: Sunday, December 04, 2005 12:35 PM
	                        >
	                        >                       Subject: [cisco-voip] problem with part 
	and css
	                        >
	                        >
	                        >
	                        >                       we have a cm 4.1.2 es 41
	                        >
	                        >                       i was just told by a tac rep that 4.1. and 
	higher
	                        uses both
	                        > css on line and device.
	                        >
	                        >                       what does this mean????
	                        >
	                        >
	                        >
	                        >                       i have my device as unrestricted and all 
	of the part
	                        in it.
	                        > and then i have my line configure with a customize css, ie maybe 
	no int
	                        calls or LD
	                        > calls.
	                        >
	                        >                       the problem im having is that when i make 
	a ld or
	                        int call,
	                        > uses the device.  and when i make a ipphone call ( 4 dgit 
	dialing between
	                        sites) it
	                        > uses the line. I dont understand this.  nore do i understand  the
	                        statement   IT
	                        > USES BOTH LINE AND DEVICE.   can someone Please clear this up 
	for me.
	                        > Thanks in advance
	                        >
	                        >
	                        >   _____
	                        >
	                        >
	                        >                       
	_______________________________________________
	                        >                       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
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	



From davidp at bostonpost.com  Mon Dec  5 12:28:06 2005
From: davidp at bostonpost.com (David Pitkin)
Date: Mon Dec  5 12:28:08 2005
Subject: [cisco-voip] ATA versus VG224
Message-ID: <4DEBDDAFBB23C04BA17EFE391444211301547885@exchange.bostonpost.com>

That was true of the ATA-L. They would only support up to 9600.

 

The newer ATA186-I1 and ATA186-I2 support 14.4

 

http://www.cisco.com/warp/public/770/fn17440.shtml

 

David

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Snow
Sent: Monday, December 05, 2005 12:10 PM
To: 'TechGuy'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ATA versus VG224

 

The ATA is a very low-end device. It will support Faxing, but only up to
9600 baud (need to hardcode your Fax machine down to this speed, and
turn ECM off). The ATA will not support a modem (though folks have
reported credit card machines working with an ATA).

The VG248 supports faster Fax speeds of 14.4 and low end modems.

The VG224 is the newest device, has the newest DSP's and supports higher
speed faxing and higher speed modems - though neither at V.90 speeds or
fully supporting the SuperG3 faxing speeds.

 

Make sure that you are running all your gateways with a IOS type NSE
payload (100) and that you are running Fax-Passthrough on every device,
and not fax relay.

 

-Mark Snow

CCIE Voice #14073

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Monday, December 05, 2005 11:48 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ATA versus VG224

 

Can anyone share with me some benefits of using a VG224 over a bunch of
ATA's?  Particularly interested in the pro's of a VG224 for fax machines
over the use of ATA's?

 



 

On 12/5/05, Ortiz, Carlos <CORTIZ at broward.org> wrote: 

The VG224's work really well with MGCP.  The only issue I have had is
that there is a bug with AAA on the VG224 that is addressed in newer
versions of code, but that would require a memory upgrade.  The best
thing I can tell you about them is that I installed one about 1 year ago
and another about 9 months ago.  Haven't touched them since..... :-) 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Monday, December 05, 2005 10:51 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ATA versus VG224

 

Can someone share with me thoughts on ATA's versus VG224.  I have heard
that the VG224 is a little more reliable and less problamatic then the
ATA's 

 

I believe the VG224 is a little more flexible in how you can configure
it?  Mainly looking for something to handle a bunch of fax machines.
Looking for the most reliable MGCP solution. 

 

 

 

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From Scott.Voll at wesd.org  Mon Dec  5 12:33:01 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec  5 12:32:58 2005
Subject: [cisco-voip] Ciptug Unity workshop
Message-ID: <D713462ED535184D830F6F6301753E7E01A11D8E@VISHNU.wesd.org>

I went to the Unity troubleshooting workshop and they talked about
checking the Eadmin mailbox and possibility moving the Eadmin to a local
user (Me)  How do I change it to me rather then the Eadmin.  I currently
have the Eadmin disabled for security reasons.

 

Scott

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From rratliff at cisco.com  Mon Dec  5 12:59:42 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Dec  5 12:59:42 2005
Subject: [cisco-voip] User unable to log into Attendant Console
In-Reply-To: <a1b07ea60512050918kec3bebet9ef1f8656935503d@mail.gmail.com>
References: <a1b07ea60512050918kec3bebet9ef1f8656935503d@mail.gmail.com>
Message-ID: <E4B5D338-3833-42D0-A898-B8DBF2D19CAF@cisco.com>

AC uses a different username/password than CM directory.

In CCMADmin go to Service->Cisco CM Attendant Console->Cisco CM  
Attendant Console User.
 From there it's pretty self explanatory.

-Ryan

On Dec 5, 2005, at 12:18 PM, TechGuy wrote:

I have a user who is unable to login to attendant console, it is  
reporting back "invalid username, please try again".  However the  
user can log into CCMUser

Any ideas where to start?  I dont know much about attendant console.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From highspeedsnow at gmail.com  Mon Dec  5 13:03:29 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Mon Dec  5 13:04:03 2005
Subject: [cisco-voip] ATA versus VG224
In-Reply-To: <4DEBDDAFBB23C04BA17EFE391444211301547885@exchange.bostonpost.com>
Message-ID: <43948110.3bbd6b95.5fdf.61ac@mx.gmail.com>

True, thanks!

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of David Pitkin
Sent: Monday, December 05, 2005 12:28 PM
To: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ATA versus VG224

 

That was true of the ATA-L. They would only support up to 9600.

 

The newer ATA186-I1 and ATA186-I2 support 14.4

 

http://www.cisco.com/warp/public/770/fn17440.shtml

 

David

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Snow
Sent: Monday, December 05, 2005 12:10 PM
To: 'TechGuy'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ATA versus VG224

 

The ATA is a very low-end device. It will support Faxing, but only up to
9600 baud (need to hardcode your Fax machine down to this speed, and turn
ECM off). The ATA will not support a modem (though folks have reported
credit card machines working with an ATA).

The VG248 supports faster Fax speeds of 14.4 and low end modems.

The VG224 is the newest device, has the newest DSP's and supports higher
speed faxing and higher speed modems - though neither at V.90 speeds or
fully supporting the SuperG3 faxing speeds.

 

Make sure that you are running all your gateways with a IOS type NSE payload
(100) and that you are running Fax-Passthrough on every device, and not fax
relay.

 

-Mark Snow

CCIE Voice #14073

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Monday, December 05, 2005 11:48 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ATA versus VG224

 

Can anyone share with me some benefits of using a VG224 over a bunch of
ATA's?  Particularly interested in the pro's of a VG224 for fax machines
over the use of ATA's?

 



 

On 12/5/05, Ortiz, Carlos <CORTIZ at broward.org> wrote: 

The VG224's work really well with MGCP.  The only issue I have had is that
there is a bug with AAA on the VG224 that is addressed in newer versions of
code, but that would require a memory upgrade.  The best thing I can tell
you about them is that I installed one about 1 year ago and another about 9
months ago.  Haven't touched them since... :-) 

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Monday, December 05, 2005 10:51 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ATA versus VG224

 

Can someone share with me thoughts on ATA's versus VG224.  I have heard that
the VG224 is a little more reliable and less problamatic then the ATA's 

 

I believe the VG224 is a little more flexible in how you can configure it?
Mainly looking for something to handle a bunch of fax machines.  Looking for
the most reliable MGCP solution. 

 

 

 

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From techguy at gmail.com  Mon Dec  5 13:46:54 2005
From: techguy at gmail.com (TechGuy)
Date: Mon Dec  5 13:46:57 2005
Subject: [cisco-voip] User unable to log into Attendant Console
In-Reply-To: <E4B5D338-3833-42D0-A898-B8DBF2D19CAF@cisco.com>
References: <a1b07ea60512050918kec3bebet9ef1f8656935503d@mail.gmail.com>
	<E4B5D338-3833-42D0-A898-B8DBF2D19CAF@cisco.com>
Message-ID: <a1b07ea60512051046h2f9a82cdqc4e9f65eb8e1824f@mail.gmail.com>

Got it, stupid me.. I thought it was simply a matter of adding a user to the
Global Directory and then setting the CTI Applications usage.  We only have
one other person who uses it and well I have not touched setting anyone up
for a very long time.  My bad!  Thanks for the help.




On 12/5/05, Ryan Ratliff <rratliff at cisco.com> wrote:
>
> AC uses a different username/password than CM directory.
>
> In CCMADmin go to Service->Cisco CM Attendant Console->Cisco CM
> Attendant Console User.
> From there it's pretty self explanatory.
>
> -Ryan
>
> On Dec 5, 2005, at 12:18 PM, TechGuy wrote:
>
> I have a user who is unable to login to attendant console, it is
> reporting back "invalid username, please try again".  However the
> user can log into CCMUser
>
> Any ideas where to start?  I dont know much about attendant console.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From it at cimgroup.com  Mon Dec  5 14:17:02 2005
From: it at cimgroup.com (IT)
Date: Mon Dec  5 14:13:10 2005
Subject: [cisco-voip] FXO/FXS on a single card
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02576305@exchange.cimgroup.com>

Hi,

I have a CallManager 4.0 installation at my headquarters office, and I
have a branch office with a 2801 connected to HQ with a point to point
T1. Currently, we have filled all 4 slots on the 2801.
T1- for point to point connectivity
FXO- for outbound 911 calling (we do not have a E911 server) 9-port POE
switch- for all devices (which takes up two slots)

This has worked well, as have been call forwarding all their local
numbers to our Headquarters PRI.  Since they are in a different area
code, this has been quite expensive. We don't want to install a PRI on
the 2801, since it seems like to much of an expense for only 6 people. I
think the solution is to have an FXS and FXO in this 2801.  But, here
are my questions:
* I have been told an FXS card cannot handle caller-id, is this true?
* Is there an FXO/FXS on a single card, since I have run out of room on
my 2801 chassis?
* How many lines would an office of 6 people require?
* Does anyone have any other ideas?

Thanks,
Avidan

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 



From CORTIZ at broward.org  Mon Dec  5 14:22:24 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon Dec  5 14:22:32 2005
Subject: [cisco-voip] Caller ID
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2AFB@MREXVS01.bc.broward.cty>

Can someone confirm that caller id is not available on FXO ports? I just
replaced an old Norstar system with a local SRST gateway and FXO ports
but no caller ID is coming through.  I thought I have seen a few people
mention that here.  If that is the case any prospects that it may become
available in the future?

 

Carlos Ortiz

Senior Information Technology Specialist

Broward County Office of Information Technology

Data Communications

954-357-7981

cortiz at broward.org

 

 

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From john.lange at open-it.ca  Mon Dec  5 14:37:30 2005
From: john.lange at open-it.ca (John Lange)
Date: Mon Dec  5 14:37:39 2005
Subject: [cisco-voip] 7960 hijacks the user interface when calls come in
In-Reply-To: <90B962BEA7D42F40BF0DF00044A5A1B60B32F56F@x-np5.am.group.local>
References: <90B962BEA7D42F40BF0DF00044A5A1B60B32F56F@x-np5.am.group.local>
Message-ID: <1133811451.4643.2.camel@laptop.darkcore.net>

Thanks Court.

You must be referring to the Cisco Call Manager web interface? I don't
see any similar settings on the phone itself.

Assuming you are talking about Call manager, does call manager allow you
to change the soft-key settings on the SIP load of the phones?

Aside from the key-map settings, is there a way to prevent the
hijacking?

Thanks

John

On Mon, 2005-12-05 at 10:49 -0600, Schuett, Court (MBNAP it) wrote:
> Device -> Device Settings -> Softkey Template
> 
> Under there, find the softkey template assigned to the phone in question.
> Then, click 'Configure Softkey Layout'.  You should be able to change
> everything you need in there.  There is a button for 'Ring In' which is
> where you'll probably need to add 'Transfer' to.
> 
> Hope that helps.
> 
>  
> 
> 
> Court Schuett
>  
> Millward Brown
> Network Systems Engineer
> 630-955-8983
> court.schuett at us.millwardbrown.com
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Lange
> Sent: Monday, December 05, 2005 10:37 AM
> To: Cisco-VoIP
> Subject: [cisco-voip] 7960 hijacks the user interface when calls come in
> 
> We have a problem with our Cisco 7960s using the SIP load.
> 
> While trying to transfer a call, if another call comes in, the entire user
> interface is "hijacked" and you can not complete the transfer process.
> 
> The only option available on the softkeys is "accept" (for the new call), so
> you can not complete the transfer.
> 
> This is a serious problem especially when trying to use the 7960 as a
> switchboard phone causing many calls to be lost in "transfer limbo".
> 
> As a side issue, is there any way to reprogram the layout of the softkeys on
> the 7960? Having to press "more" to get to "transfer" is just silly.
> 
> --
> John Lange
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> ==================================================== 
> This email is confidential and intended solely for the use of the 
> individual or organization to whom it is addressed. Any opinions or 
> advice presented are solely those of the author and do not necessarily 
> represent those of the Millward Brown Group of Companies.  If you are 
> not the intended recipient of this email, you should not copy, modify, 
> distribute or take any action in reliance on it. If you have received 
> this email in error please notify the sender and delete this email 
> from your system. Although this email has been checked for viruses 
>  and other defects, no responsibility can be accepted for any loss or 
> damage arising from its receipt or use. 
> ==================================================== 
> 


From Michael_Davis at eLoyalty.com  Mon Dec  5 14:52:43 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Mon Dec  5 14:52:57 2005
Subject: [cisco-voip] Caller ID
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E32C@elhqex01.eloyaltyco.com>

Carlos in Broward county - my old stomping grounds (grew up in Davie). :-)
 
 
You need an FXO with a "M1" designation for Caller-ID. For example: VIC-4FXO-M1.
 
 DNIS for DID calls, obviously is not supported.
 
http://www.cisco.com/en/US/products/hw/voiceapp/ps967/products_data_sheet09186a0080107b55.html

HTH,
 
Mike
 
 
-----Original Message----- 
From: cisco-voip-bounces at puck.nether.net on behalf of Ortiz, Carlos 
Sent: Mon 12/5/2005 1:22 PM 
To: cisco-voip at puck.nether.net 
Cc: 
Subject: [cisco-voip] Caller ID



	Can someone confirm that caller id is not available on FXO ports? I just replaced an old Norstar system with a local SRST gateway and FXO ports but no caller ID is coming through.  I thought I have seen a few people mention that here.  If that is the case any prospects that it may become available in the future?

	 

	Carlos Ortiz

	Senior Information Technology Specialist

	Broward County Office of Information Technology

	Data Communications

	954-357-7981

	cortiz at broward.org

	 

	 



From Michael_Davis at eLoyalty.com  Mon Dec  5 14:56:37 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Mon Dec  5 14:56:51 2005
Subject: [cisco-voip] FXO/FXS on a single card
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E32D@elhqex01.eloyaltyco.com>

Caller-id, though supported on FXO cards with "M1" vintage, won't help you here. You need the card to see DNIS for direct-inward-dial support. Is this an MGCP or H323 gateway? Off the top of my head, you have a couple of possibilities.  1) Fractional T1 2) FXO-M1  that hits an AutoAttendant TCL script on the gateway.  
 
 
 
 
 
 

	-----Original Message----- 
	From: cisco-voip-bounces at puck.nether.net on behalf of IT 
	Sent: Mon 12/5/2005 1:17 PM 
	To: cisco-voip at puck.nether.net 
	Cc: 
	Subject: [cisco-voip] FXO/FXS on a single card
	
	

	Hi,
	
	I have a CallManager 4.0 installation at my headquarters office, and I
	have a branch office with a 2801 connected to HQ with a point to point
	T1. Currently, we have filled all 4 slots on the 2801.
	T1- for point to point connectivity
	FXO- for outbound 911 calling (we do not have a E911 server) 9-port POE
	switch- for all devices (which takes up two slots)
	
	This has worked well, as have been call forwarding all their local
	numbers to our Headquarters PRI.  Since they are in a different area
	code, this has been quite expensive. We don't want to install a PRI on
	the 2801, since it seems like to much of an expense for only 6 people. I
	think the solution is to have an FXS and FXO in this 2801.  But, here
	are my questions:
	* I have been told an FXS card cannot handle caller-id, is this true?
	* Is there an FXO/FXS on a single card, since I have run out of room on
	my 2801 chassis?
	* How many lines would an office of 6 people require?
	* Does anyone have any other ideas?
	
	Thanks,
	Avidan
	
	******************************************************
	This email may contain trade secrets and/or other information that is
	proprietary and/or confidential. Use of this by anyone other than the
	intended recipient(s) is prohibited and may be unlawful. If you are not
	the intended recipient, delete the e-mail and notify us immediately.
	******************************************************
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	



From pwalenta at wi.rr.com  Mon Dec  5 15:17:34 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Mon Dec  5 15:17:33 2005
Subject: [cisco-voip] Caller ID
In-Reply-To: <037594FB5AFC354FA38653EB3FF3242105A8E32C@elhqex01.eloyaltyco.com>
Message-ID: <200512052014.jB5KEUdu026247@ms-smtp-02.rdc-kc.rr.com>

Additionally, you must run H.323, Caller ID on FXO is not supported with
MGCP.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Davis, Michael
Sent: Monday, December 05, 2005 1:53 PM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Caller ID

Carlos in Broward county - my old stomping grounds (grew up in Davie). :-)
 
 
You need an FXO with a "M1" designation for Caller-ID. For example:
VIC-4FXO-M1.
 
 DNIS for DID calls, obviously is not supported.
 
http://www.cisco.com/en/US/products/hw/voiceapp/ps967/products_data_sheet091
86a0080107b55.html

HTH,
 
Mike
 
 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net on behalf of Ortiz, Carlos
Sent: Mon 12/5/2005 1:22 PM
To: cisco-voip at puck.nether.net
Cc: 
Subject: [cisco-voip] Caller ID



	Can someone confirm that caller id is not available on FXO ports? I
just replaced an old Norstar system with a local SRST gateway and FXO ports
but no caller ID is coming through.  I thought I have seen a few people
mention that here.  If that is the case any prospects that it may become
available in the future?

	 

	Carlos Ortiz

	Senior Information Technology Specialist

	Broward County Office of Information Technology

	Data Communications

	954-357-7981

	cortiz at broward.org

	 

	 


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


From it at cimgroup.com  Mon Dec  5 20:53:57 2005
From: it at cimgroup.com (IT)
Date: Mon Dec  5 20:50:06 2005
Subject: [cisco-voip] FXO/FXS on a single card
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0257666D@exchange.cimgroup.com>

It is an MGCP gateway. How would the FXO-M1 hitting an AutoAttendant TCL
work?
Thanks,
Avidan


-----Original Message-----
From: Davis, Michael [mailto:Michael_Davis at eLoyalty.com] 
Sent: Monday, December 05, 2005 11:57 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] FXO/FXS on a single card

Caller-id, though supported on FXO cards with "M1" vintage, won't help
you here. You need the card to see DNIS for direct-inward-dial support.
Is this an MGCP or H323 gateway? Off the top of my head, you have a
couple of possibilities.  1) Fractional T1 2) FXO-M1  that hits an
AutoAttendant TCL script on the gateway.  
 
 
 
 
 
 

	-----Original Message----- 
	From: cisco-voip-bounces at puck.nether.net on behalf of IT 
	Sent: Mon 12/5/2005 1:17 PM 
	To: cisco-voip at puck.nether.net 
	Cc: 
	Subject: [cisco-voip] FXO/FXS on a single card
	
	

	Hi,
	
	I have a CallManager 4.0 installation at my headquarters office,
and I
	have a branch office with a 2801 connected to HQ with a point to
point
	T1. Currently, we have filled all 4 slots on the 2801.
	T1- for point to point connectivity
	FXO- for outbound 911 calling (we do not have a E911 server)
9-port POE
	switch- for all devices (which takes up two slots)
	
	This has worked well, as have been call forwarding all their
local
	numbers to our Headquarters PRI.  Since they are in a different
area
	code, this has been quite expensive. We don't want to install a
PRI on
	the 2801, since it seems like to much of an expense for only 6
people. I
	think the solution is to have an FXS and FXO in this 2801.  But,
here
	are my questions:
	* I have been told an FXS card cannot handle caller-id, is this
true?
	* Is there an FXO/FXS on a single card, since I have run out of
room on
	my 2801 chassis?
	* How many lines would an office of 6 people require?
	* Does anyone have any other ideas?
	
	Thanks,
	Avidan
	
	******************************************************
	This email may contain trade secrets and/or other information
that is
	proprietary and/or confidential. Use of this by anyone other
than the
	intended recipient(s) is prohibited and may be unlawful. If you
are not
	the intended recipient, delete the e-mail and notify us
immediately.
	******************************************************
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	



From Michael_Davis at eLoyalty.com  Mon Dec  5 21:04:39 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Mon Dec  5 21:04:53 2005
Subject: [cisco-voip] FXO/FXS on a single card
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E332@elhqex01.eloyaltyco.com>

It won't work for MGCP gateways, only H.323.  You would create an inbound dial-peer that executes the TCL application.  The application askes the caller for the extension desired.  Then, an outbound leg to that extension (residing on CCM) is established.
 
Ok. Other possibilities - send those calls to a Unity directory handler, Cisco AutoAttendant (link below)  or perhaps IPCC Express.
 
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00801ed106.html
 
 

	-----Original Message----- 
	From: IT [mailto:it at cimgroup.com] 
	Sent: Mon 12/5/2005 7:53 PM 
	To: Davis, Michael; IT; cisco-voip at puck.nether.net 
	Cc: 
	Subject: RE: [cisco-voip] FXO/FXS on a single card
	
	

	It is an MGCP gateway. How would the FXO-M1 hitting an AutoAttendant TCL
	work?
	Thanks,
	Avidan
	
	
	-----Original Message-----
	From: Davis, Michael [mailto:Michael_Davis at eLoyalty.com]
	Sent: Monday, December 05, 2005 11:57 AM
	To: IT; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] FXO/FXS on a single card
	
	Caller-id, though supported on FXO cards with "M1" vintage, won't help
	you here. You need the card to see DNIS for direct-inward-dial support.
	Is this an MGCP or H323 gateway? Off the top of my head, you have a
	couple of possibilities.  1) Fractional T1 2) FXO-M1  that hits an
	AutoAttendant TCL script on the gateway. 
	
	
	
	
	
	
	
	        -----Original Message-----
	        From: cisco-voip-bounces at puck.nether.net on behalf of IT
	        Sent: Mon 12/5/2005 1:17 PM
	        To: cisco-voip at puck.nether.net
	        Cc:
	        Subject: [cisco-voip] FXO/FXS on a single card
	       
	       
	
	        Hi,
	       
	        I have a CallManager 4.0 installation at my headquarters office,
	and I
	        have a branch office with a 2801 connected to HQ with a point to
	point
	        T1. Currently, we have filled all 4 slots on the 2801.
	        T1- for point to point connectivity
	        FXO- for outbound 911 calling (we do not have a E911 server)
	9-port POE
	        switch- for all devices (which takes up two slots)
	       
	        This has worked well, as have been call forwarding all their
	local
	        numbers to our Headquarters PRI.  Since they are in a different
	area
	        code, this has been quite expensive. We don't want to install a
	PRI on
	        the 2801, since it seems like to much of an expense for only 6
	people. I
	        think the solution is to have an FXS and FXO in this 2801.  But,
	here
	        are my questions:
	        * I have been told an FXS card cannot handle caller-id, is this
	true?
	        * Is there an FXO/FXS on a single card, since I have run out of
	room on
	        my 2801 chassis?
	        * How many lines would an office of 6 people require?
	        * Does anyone have any other ideas?
	       
	        Thanks,
	        Avidan
	       
	        ******************************************************
	        This email may contain trade secrets and/or other information
	that is
	        proprietary and/or confidential. Use of this by anyone other
	than the
	        intended recipient(s) is prohibited and may be unlawful. If you
	are not
	        the intended recipient, delete the e-mail and notify us
	immediately.
	        ******************************************************
	       
	       
	        _______________________________________________
	        cisco-voip mailing list
	        cisco-voip at puck.nether.net
	        https://puck.nether.net/mailman/listinfo/cisco-voip
	       
	



From jgrace at digitelusa.net  Mon Dec  5 21:32:22 2005
From: jgrace at digitelusa.net (James Grace)
Date: Mon Dec  5 21:34:39 2005
Subject: [cisco-voip] error on ipphone "http request canceled" 
Message-ID: <003801c5fa0d$4a5d8950$030a0f0a@digitelcorp.com>

all our ip phone are getting this error.  we are not performing any special task when it comes up. it when the phones are in idle.  has any body send this.
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From rratliff at cisco.com  Mon Dec  5 21:40:00 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Dec  5 21:40:00 2005
Subject: [cisco-voip] error on ipphone "http request canceled" 
In-Reply-To: <003801c5fa0d$4a5d8950$030a0f0a@digitelcorp.com>
References: <003801c5fa0d$4a5d8950$030a0f0a@digitelcorp.com>
Message-ID: <3A636965-104B-4D09-8786-AF3909DE4AC3@cisco.com>

Do you have an idle URL configured?

-Ryan

On Dec 5, 2005, at 9:32 PM, James Grace wrote:

all our ip phone are getting this error.  we are not performing any  
special task when it comes up. it when the phones are in idle.  has  
any body send this.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From it at cimgroup.com  Mon Dec  5 21:47:49 2005
From: it at cimgroup.com (IT)
Date: Mon Dec  5 21:43:57 2005
Subject: [cisco-voip] FXO/FXS on a single card
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02576698@exchange.cimgroup.com>

But that's all assuming I have room for the card. All 4 slots are taken.

2 by the 9port poe switch. 1 by the FXS card. 1 by the T1 card...
How could I even begin to accomplish this in a 2801?

-avidan

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Davis, Michael
Sent: Monday, December 05, 2005 6:05 PM
To: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] FXO/FXS on a single card

It won't work for MGCP gateways, only H.323.  You would create an
inbound dial-peer that executes the TCL application.  The application
askes the caller for the extension desired.  Then, an outbound leg to
that extension (residing on CCM) is established.
 
Ok. Other possibilities - send those calls to a Unity directory handler,
Cisco AutoAttendant (link below)  or perhaps IPCC Express.
 
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_ad
ministration_guide_chapter09186a00801ed106.html
 
 

	-----Original Message----- 
	From: IT [mailto:it at cimgroup.com] 
	Sent: Mon 12/5/2005 7:53 PM 
	To: Davis, Michael; IT; cisco-voip at puck.nether.net 
	Cc: 
	Subject: RE: [cisco-voip] FXO/FXS on a single card
	
	

	It is an MGCP gateway. How would the FXO-M1 hitting an
AutoAttendant TCL
	work?
	Thanks,
	Avidan
	
	
	-----Original Message-----
	From: Davis, Michael [mailto:Michael_Davis at eLoyalty.com]
	Sent: Monday, December 05, 2005 11:57 AM
	To: IT; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] FXO/FXS on a single card
	
	Caller-id, though supported on FXO cards with "M1" vintage,
won't help
	you here. You need the card to see DNIS for direct-inward-dial
support.
	Is this an MGCP or H323 gateway? Off the top of my head, you
have a
	couple of possibilities.  1) Fractional T1 2) FXO-M1  that hits
an
	AutoAttendant TCL script on the gateway. 
	
	
	
	
	
	
	
	        -----Original Message-----
	        From: cisco-voip-bounces at puck.nether.net on behalf of IT
	        Sent: Mon 12/5/2005 1:17 PM
	        To: cisco-voip at puck.nether.net
	        Cc:
	        Subject: [cisco-voip] FXO/FXS on a single card
	       
	       
	
	        Hi,
	       
	        I have a CallManager 4.0 installation at my headquarters
office,
	and I
	        have a branch office with a 2801 connected to HQ with a
point to
	point
	        T1. Currently, we have filled all 4 slots on the 2801.
	        T1- for point to point connectivity
	        FXO- for outbound 911 calling (we do not have a E911
server)
	9-port POE
	        switch- for all devices (which takes up two slots)
	       
	        This has worked well, as have been call forwarding all
their
	local
	        numbers to our Headquarters PRI.  Since they are in a
different
	area
	        code, this has been quite expensive. We don't want to
install a
	PRI on
	        the 2801, since it seems like to much of an expense for
only 6
	people. I
	        think the solution is to have an FXS and FXO in this
2801.  But,
	here
	        are my questions:
	        * I have been told an FXS card cannot handle caller-id,
is this
	true?
	        * Is there an FXO/FXS on a single card, since I have run
out of
	room on
	        my 2801 chassis?
	        * How many lines would an office of 6 people require?
	        * Does anyone have any other ideas?
	       
	        Thanks,
	        Avidan
	       
	        ******************************************************
	        This email may contain trade secrets and/or other
information
	that is
	        proprietary and/or confidential. Use of this by anyone
other
	than the
	        intended recipient(s) is prohibited and may be unlawful.
If you
	are not
	        the intended recipient, delete the e-mail and notify us
	immediately.
	        ******************************************************
	       
	       
	        _______________________________________________
	        cisco-voip mailing list
	        cisco-voip at puck.nether.net
	        https://puck.nether.net/mailman/listinfo/cisco-voip
	       
	


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From andy at dignans.com  Mon Dec  5 21:48:02 2005
From: andy at dignans.com (Andrew Dignan)
Date: Mon Dec  5 21:48:05 2005
Subject: [cisco-voip] IPCC 7.0 System Install - Web Config Tool/Internet
	Script Editor
Message-ID: <4416.64.81.226.84.1133837282.squirrel@www.dignans.com>

Planning a 7.0 "System Install" and I am having
trouble finding some information.  If anyone has access to a 7.0 System
Install and could answer these that would be great.

1. Using the "System Install" Web Configuration Tool (replaces the
Configuration Manager) is there Bulk Imports/Exports for Agents, Call types,
network vru scripts, services, skill groups, and dialed numbers?

2. In the Internet Script Editor for System Install is their an Import/Export
for the scripts?

3. In the Internet Script Editor for System Install is there a
"Label" node still (probably the ability to have a
"dynamic" label) so calls can be redirected within the ICM script?

From jgrace at digitelusa.net  Tue Dec  6 00:24:52 2005
From: jgrace at digitelusa.net (James Grace)
Date: Tue Dec  6 00:27:08 2005
Subject: [cisco-voip] how to add a second cm server to ldap dir sever for crs
Message-ID: <008601c5fa25$6361b990$030a0f0a@digitelcorp.com>

we want to set up redundancey for our crs app.  we added the second ip address for the other cm in the crs administrator.  but when we orignally installed the crs server, we did not fill in server B info.  So someone said we have to add the server to the cti manager list.  but we could not add with dsbrowser /editable
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From Scott.Voll at wesd.org  Tue Dec  6 00:36:13 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec  6 00:37:10 2005
Subject: [cisco-voip] how to add a second cm server to ldap dir sever for
	crs
Message-ID: <D713462ED535184D830F6F6301753E7E018B3176@VISHNU.wesd.org>

what version?  3.x you should be able to rerun the install and add it.  4.x not sure.
 
Scott

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of James Grace
Sent: Mon 12/5/2005 9:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to add a second cm server to ldap dir sever for crs


we want to set up redundancey for our crs app.  we added the second ip address for the other cm in the crs administrator.  but when we orignally installed the crs server, we did not fill in server B info.  So someone said we have to add the server to the cti manager list.  but we could not add with dsbrowser /editable
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From ml at iridescent.org  Tue Dec  6 00:56:57 2005
From: ml at iridescent.org (Lawrence E. Bakst)
Date: Tue Dec  6 00:57:20 2005
Subject: [cisco-voip] FXO/FXS on a single card
In-Reply-To: <037594FB5AFC354FA38653EB3FF3242105A8E32D@elhqex01.eloyaltyco.com>
References: <037594FB5AFC354FA38653EB3FF3242105A8E32D@elhqex01.eloyaltyco.com>
Message-ID: <p06230903bfbad28d3804@mail.iridescent.org>

Sorry I don't see how adding FXS is going to help you.

I don't know what GEO you are in but I think 1 or 2 ISDN BRI lines might work well in this situation. I have one here at home and it works well for me. You can replace your current card 2FXO card with a VIC2-2BRI-NT/TE and get twice the number of lines. You also need two NT1's. You get both ANI and DNIS and it has full supervision almost like a PRI. 911 should be OK as well.

Depending upon where you are you might be able to get ISDN MSN (Multiple Subscriber Numbering) which allows multiple numbers (more than the two SPIDs) to be assigned to an ISDN line. It's like DID for ISDN.

I guess the only issue is what the rates are for ISDN.

As far as how many lines you need it depends. 6 sales persons will probably need  more than 6 lines. 6 engineers might only need 2 lines. I would start with 2 BRI which is 4 lines.

You also might be able to do something with SIP and the new CME 3.4. SIP rates are pretty cheap these days.

Actually why don't you use the ISDN for incoming and SIP for outgoing?

Best,

leb


At 1:56 PM -0600 12/5/05, Davis, Michael wrote:
>Caller-id, though supported on FXO cards with "M1" vintage, won't help you here. You need the card to see DNIS for direct-inward-dial support. Is this an MGCP or H323 gateway? Off the top of my head, you have a couple of possibilities.  1) Fractional T1 2) FXO-M1  that hits an AutoAttendant TCL script on the gateway. 
> 
> 
> 
> 
> 
> 
>
>	-----Original Message-----
>	From: cisco-voip-bounces at puck.nether.net on behalf of IT
>	Sent: Mon 12/5/2005 1:17 PM
>	To: cisco-voip at puck.nether.net
>	Cc:
>	Subject: [cisco-voip] FXO/FXS on a single card
>	
>	
>
>	Hi,
>	
>	I have a CallManager 4.0 installation at my headquarters office, and I
>	have a branch office with a 2801 connected to HQ with a point to point
>	T1. Currently, we have filled all 4 slots on the 2801.
>	T1- for point to point connectivity
>	FXO- for outbound 911 calling (we do not have a E911 server) 9-port POE
>	switch- for all devices (which takes up two slots)
>	
>	This has worked well, as have been call forwarding all their local
>	numbers to our Headquarters PRI.  Since they are in a different area
>	code, this has been quite expensive. We don't want to install a PRI on
>	the 2801, since it seems like to much of an expense for only 6 people. I
>	think the solution is to have an FXS and FXO in this 2801.  But, here
>	are my questions:
>	* I have been told an FXS card cannot handle caller-id, is this true?
>	* Is there an FXO/FXS on a single card, since I have run out of room on
>	my 2801 chassis?
>	* How many lines would an office of 6 people require?
>	* Does anyone have any other ideas?
>	
>	Thanks,
>	Avidan
>	
>	******************************************************
>	This email may contain trade secrets and/or other information that is
>	proprietary and/or confidential. Use of this by anyone other than the
>	intended recipient(s) is prohibited and may be unlawful. If you are not
>	the intended recipient, delete the e-mail and notify us immediately.
>	******************************************************
>	
>	
>	_______________________________________________
>	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 mody_mody165 at yahoo.com  Tue Dec  6 02:38:10 2005
From: mody_mody165 at yahoo.com (Mohamed Ahmed)
Date: Tue Dec  6 02:38:20 2005
Subject: [cisco-voip] Hey ;)
Message-ID: <2081231330.1133854701661.JavaMail.root@rainer>

Mohamed Ahmed has invited you to join hi5. By joining hi5, you will be connected to Mohamed and all of Mohamed's friends.
 
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This invitation was sent to cisco-voip at puck.nether.net on behalf of Mohamed Ahmed (mody_mody165 at yahoo.com).

If you do not wish to receive invitations from hi5 members, click on the link below:
http://www.hi5.com/friend/displayBlockInvite.do?inviteId=11KYRS47CU49391933c0
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From commo_ssg_31f at yahoo.com  Tue Dec  6 06:28:27 2005
From: commo_ssg_31f at yahoo.com (ash AD)
Date: Tue Dec  6 06:28:34 2005
Subject: [cisco-voip] One CM and several identical networks
In-Reply-To: <CB7389F8-ECCF-4656-B2E6-9CE4173517F1@cisco.com>
Message-ID: <20051206112827.91737.qmail@web50401.mail.yahoo.com>

Andrius,
  
I have to echo what Ryan wrote. We tried to implement the second interface on the MCS server and killed our CCM (CCM 3.3(4)) in Iraq. Is there a reason you couldn't use more private address space i.e..:
   
  Site 1 Voice Net/VLAN: 192.168.0.0/24
  Site 2 Voice Net/VLAN: 192.168.1.0/24
  Site 3 Voice Net/VLAN: 192.168.2.0/24
   
  I know this is much like a private VLAN approach but, it seems simple and heck, the IPs are free. 
   
  V/R,
  PETER CASANAVE
  WAN / VoIP Network Engineer
  U.S. Army
  
Ryan Ratliff <rratliff at cisco.com> wrote:
  Your CM server should never have multiple IP addresses. I've 
certainly no experience deploying IPT but it sounds to me like NAT 
with skinny/H.323 fixup is what you need.

Did I mention to absolutely NEVER put multiple IP addresses on your 
CM? Well, at least if you want it to work...

-Ryan

On Dec 5, 2005, at 1:38 AM, Andrius Kislas wrote:

Hello,

I have a question about CallManager working as a SCCP signaling provider
for several networks using VPN over Internet. It so happens that two
networks are with the same IP adresses (192.168.0.x/24). What is the
"best practice" in such situation for CM to distinguish these networks?
I thought about two solutions:

1) Put a secondary IP address on CM for the second identical network.
Then the make policy routing on a router, so that packets FROM one CM
address goes to the first network, and packets FROM the second address
goes to the second network.

2) Configure NAT on VPN tunnels, so that packets from first network
reaches CM as one IP address and packets from second networks reaches CM
with second address.

I haven't tried neither of these solutions. Maybe there is more
intelligent way of solving the problem?

Andrius
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From rleetun at co.boulder.co.us  Tue Dec  6 08:50:55 2005
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Tue Dec  6 08:51:03 2005
Subject: [cisco-voip] problem with part and css (the none partition)
Message-ID: <963CE94787005149A52C17F33A028E12051D40E4@EVS1B.boco.co.boulder.co.us>

Quick question on IPC; has anyone heard when Cisco is going to deploy
the video hook to IPC?  We could really use this technology now.

-Rob

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Monday, December 05, 2005 8:04 AM
To: Davis, Michael
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] problem with part and css (the none partition)

IPC is actually based on the 7970 IP phone.  You are correct it talks  
skinny, not CTI/Jtapi.


-Ryan

On Dec 4, 2005, at 5:55 PM, Davis, Michael wrote:

IIRC,  the IP Communicator was at some point based on the old Selsius  
Cisco 30 SP+ rather than CTI. Someone will correct me if I'm wrong.   
I know this is definitely the case for CAD Media Termination clients,  
but am not certain about IP Communicator.

Either way, IP Communicator is configured as a hardware device and  
uses SCCP rather than JTAPI for control.  So in my mind it is a  
Software only IP Phone, not a "true" JTAPI controlled CTI device. So  
your testing is consistent with what I've seen.  However, things like  
IP Blue, Cisco Softphone,  CTI Ports/CTI Route Points and the JTAPI  
apps such as IPCC/IPCCX, etc) that control them are going to be  
affected.

I know the original post was specific to IP Communicator, but with  
all the discussion around how CSSes are processed, I thought I should  
add the caveat.

HTH clarify,

Mike


	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net on behalf of Lelio
Fulgenzi
	Sent: Sun 12/4/2005 4:06 PM
	To: Mark Snow; cisco-voip at puck.nether.net
	Cc:
	Subject: Re: [cisco-voip] problem with part and css (the none  
partition)
	
	
	And the other question is....I'm assuming this doesn't affect  
IPCommunicator? I ran a test in the lab and it created the PSS  
according to line+device. Is this because IP communicator is not a  
CTI application like Softphone? Are my results what you would expect?

		----- Original Message -----
		From: Mark Snow <mailto:highspeedsnow at gmail.com>
		To: 'Lelio Fulgenzi' <mailto:lelio at uoguelph.ca>  ;
cisco- 
voip at puck.nether.net
		Sent: Sunday, December 04, 2005 4:20 PM
		Subject: RE: [cisco-voip] problem with part and css (the
none  
partition)


		The answer probably lies with the PM, but I don't know
why or when ...

		

		
   _____


		From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
		Sent: Sunday, December 04, 2005 2:04 PM
		To: cisco-voip at puck.nether.net
		Subject: Re: [cisco-voip] problem with part and css (the
none  
partition)

		

		Is there any particular reason why they've gone this way
with the  
CTIs? Are they eventually going to change it?

		

			----- Original Message -----

			From: Davis, Michael
<mailto:Michael_Davis at eLoyalty.com>

			To: cisco-voip at puck.nether.net

			Sent: Sunday, December 04, 2005 3:02 PM

			Subject: RE: [cisco-voip] problem with part and
css (the none  
partition)

			

			I don't have a whole lot of experience with 3.1,
but I have  
recently been "burnt" by forgetting the CTI vs non-CTI search order. :-)
			
			-----Original Message-----
			From: Mark Snow [mailto:highspeedsnow at gmail.com]
			Sent: Sun 12/4/2005 1:47 PM
			To: Davis, Michael; cisco-voip at puck.nether.net
			Cc:
			Subject: RE: [cisco-voip] problem with part and
css (the none  
partition)
			
			
			
			Good Point!
			
			For those of you out there running versions
prior to 3.1 when it  
was still
			'Call Mangler'.
			
			Also if you are running those previous versions,
try Forwarding  
your phone
			to it's own DN and then calling it from another
phone - just for  
fun! ;)
			
			> -----Original Message-----
			> From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-
			> bounces at puck.nether.net] On Behalf Of Davis,
Michael
			> Sent: Sunday, December 04, 2005 1:36 PM
			> To: cisco-voip at puck.nether.net
			> Subject: RE: [cisco-voip] problem with part
and css (the none  
partition)
			>
			> Also,
			> Prior to CCM 3.1, Device Level CSS were placed
ahead of Line  
Level CSS.
			As of
			> 3.1, Line Level CSSes are placed ahead of
Device Level, EXCEPT  
for CTI
			devices
			> (Softphone, CTI Ports, Route Points, etc,)
which use the old  
method.
			>
			> Mike
			>
			>
			>       -----Original Message-----
			>       From: cisco-voip-bounces at puck.nether.net
on behalf of Mark  
Snow
			>       Sent: Sun 12/4/2005 1:13 PM
			>       To: 'Lelio Fulgenzi';
cisco-voip at puck.nether.net
			>       Cc:
			>       Subject: RE: [cisco-voip] problem with
part and css (the none
			partition)
			>
			>
			>
			>       Guys,
			>
			>
			>
			>       The <none> partition is actually a real
partition, it is  
there by
			default and
			> is assigned to the very bottom of every CSS.
			>
			>
			>
			>       So if you had a RP of '91X' in the
INTERAL_PT belonging to  
the
			> HQ_911_CSS and Phone 1 at HQ had this CSS
assigned,  but you  
also had a RP
			of
			> '911' in in the <none> Partition (PT) and the
user at Phone 1 at  
HQ dialed
			911, he
			> would actually be routed to the 911 RP in the
<none> PT due to  
the fact
			that it
			> appears last in the HQ_911_CSS and is a longer
match.
			>
			>
			>
			>       It is similar to the <null> MRG, which
belongs at the very  
bottom of
			every
			> MRGL.
			>
			>
			>
			>       MTC and HTH some :-)
			>
			>
			>
			>       -Mark Snow
			>
			>       CCIE Voice # 14073
			>
			>
			>
			>
			>   _____
			>
			>
			>       From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-
			> bounces at puck.nether.net] On Behalf Of Lelio
Fulgenzi
			>       Sent: Sunday, December 04, 2005 12:19 PM
			>       To: cisco-voip at puck.nether.net
			>       Subject: Re: [cisco-voip] problem with
part and css (the none
			partition)
			>
			>
			>
			>       Oh yeah, and don't forget the <None>
partition. It's not  
really a
			partition,
			> but it encompasses as DNs and patterns that
have not been  
assigned to a
			> partition. There is no easy way to block
access to these DNs  
unless you
			create
			> duplicate entries in another partition and
block those (which we  
have
			done). I
			> can't remember the search algorithm and
whether the <None>  
partition is
			simply
			> placed last in the ordered list and searched
accordingly or if  
it is
			searched only
			> after no match is made in the other
partitions. A search on CCO  
docs could
			tell
			> you this probably.
			>
			>
			>
			>       We've moved towards a dialplan model
that has next to no  
DNs in the
			> <None> partition. The only ones we have in
there are our campus  
emergency
			> number and our helpline number. In the event
that a phone goes out
			> misconfigured, chances are very high they will
be able to call  
these
			numbers.
			>
			>
			>
			>               ----- Original Message -----
			>
			>               From: Lelio Fulgenzi
<mailto:lelio at uoguelph.ca>
			>
			>               To: James Grace
<mailto:jgrace at digitelusa.net>  ;  
cisco-
			> voip at puck.nether.net
			>
			>               Sent: Sunday, December 04, 2005
1:13 PM
			>
			>               Subject: Re: [cisco-voip]
problem with part and css
			>
			>
			>
			>               Think of it this way, a calling
search space is  
simply an
			ordered
			> list of partitions which the line or device
searches through for  
a match
			of the
			> dialed digits. If you look through the traces,
you will see the  
term PSS
			which I
			> believe stands for Phone Search Space. This is
made up of the  
line's
			ordered list of
			> partitions and then the device's ordered list
of partitions. For  
example:
			>
			>
			>
			>               CSS1=PartA,PartB,PartC
			>
			>               CSS2=PartX,PartY,PartZ
			>
			>               Device assigned CSS1
			>
			>               Line assigned CSS2
			>
			>               the PSS would be:
PartX,PartY,PartZ,PartA,PartB,PartC
			>
			>
			>
			>               So the idea you propose is a
valid one. We use it  
here.
			Assign the
			> device access to everything and then deny the
line access to  
particular
			services.
			> We isolate the 'everything' to PSTN access and
actually allow  
oncampus
			access
			> through the line's partition.
			>
			>
			>
			>               You have to make sure that the
denies are at least as
			specific as
			> the allows. And by denies, I mean, creating a
route pattern that  
has the
			'block'
			> option checked.
			>
			>
			>
			>               Post back if you need more...
			>
			>
			>
			>
			>
			>
			>
			>                       ----- Original Message
-----
			>
			>                       From: James Grace  
<mailto:jgrace at digitelusa.net>
			>
			>                       To:
cisco-voip at puck.nether.net
			>
			>                       Sent: Sunday, December
04, 2005 12:35 PM
			>
			>                       Subject: [cisco-voip]
problem with part  
and css
			>
			>
			>
			>                       we have a cm 4.1.2 es 41
			>
			>                       i was just told by a tac
rep that 4.1. and  
higher
			uses both
			> css on line and device.
			>
			>                       what does this mean????
			>
			>
			>
			>                       i have my device as
unrestricted and all  
of the part
			in it.
			> and then i have my line configure with a
customize css, ie maybe  
no int
			calls or LD
			> calls.
			>
			>                       the problem im having is
that when i make  
a ld or
			int call,
			> uses the device.  and when i make a ipphone
call ( 4 dgit  
dialing between
			sites) it
			> uses the line. I dont understand this.  nore
do i understand  the
			statement   IT
			> USES BOTH LINE AND DEVICE.   can someone
Please clear this up  
for me.
			> Thanks in advance
			>
			>
			>   _____
			>
			>
			>                        
_______________________________________________
			>                       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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			cisco-voip mailing list
			cisco-voip at puck.nether.net
	
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From highspeedsnow at gmail.com  Tue Dec  6 09:38:50 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Tue Dec  6 09:40:23 2005
Subject: [cisco-voip] how to add a second cm server to ldap dir sever
	forcrs
In-Reply-To: <D713462ED535184D830F6F6301753E7E018B3176@VISHNU.wesd.org>
Message-ID: <4395a2d1.23274e41.289f.71a3@mx.gmail.com>

Ver 3.x there is no CRS app redundancy (cold standby only - totally manual).

Ver 4.0 adds a quasi-hot standby for apps and failover.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, December 06, 2005 12:36 AM
To: James Grace; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] how to add a second cm server to ldap dir sever
forcrs

 

what version?  3.x you should be able to rerun the install and add it.  4.x
not sure.

 

Scott

 

  _____  

From: cisco-voip-bounces at puck.nether.net on behalf of James Grace
Sent: Mon 12/5/2005 9:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to add a second cm server to ldap dir sever for
crs

we want to set up redundancey for our crs app.  we added the second ip
address for the other cm in the crs administrator.  but when we orignally
installed the crs server, we did not fill in server B info.  So someone said
we have to add the server to the cti manager list.  but we could not add
with dsbrowser /editable

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From jgrace at digitelusa.net  Tue Dec  6 10:24:00 2005
From: jgrace at digitelusa.net (James)
Date: Tue Dec  6 10:23:57 2005
Subject: [cisco-voip] Out palse a name
Message-ID: <1952665177-1133882635-cardhu_blackberry.rim.net-758-@engine11-cell02>

Is there anyway for me to out pulse a company name on cm.  I was told the the carrier has to do name and we can do number if we wanted to
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From Andrius.Kislas at elsis.lt  Tue Dec  6 10:30:50 2005
From: Andrius.Kislas at elsis.lt (Andrius Kislas)
Date: Tue Dec  6 10:31:10 2005
Subject: [cisco-voip] One CM and several identical networks
In-Reply-To: <20051206112827.91737.qmail@web50401.mail.yahoo.com>
References: <20051206112827.91737.qmail@web50401.mail.yahoo.com>
Message-ID: <4395AEAA.7080709@elsis.lt>

Hi,

my situation was that client _already_ has some IP addresses and do not
want to change them. And it so happens that one CM has to reach two
networks that actually has same address space.

If we cannot put secondary IP address, we still virtually can make this
secondary and so on IP addresses using NAT with SCCP/H323 fixup as
offered Ryan.

Andrius

ash AD wrote:
> Andrius,
> 
> I have to echo what Ryan wrote. We tried to implement the second
> interface on the MCS server and killed our CCM (CCM 3.3(4)) in Iraq. Is
> there a reason you couldn't use more private address space i.e..:
>  
> Site 1 Voice Net/VLAN: 192.168.0.0/24
> Site 2 Voice Net/VLAN: 192.168.1.0/24
> Site 3 Voice Net/VLAN: 192.168.2.0/24
>  
> I know this is much like a private VLAN approach but, it seems simple
> and heck, the IPs are free.
>  
> V/R,
> PETER CASANAVE
> WAN / VoIP Network Engineer
> U.S. Army
> 
> */Ryan Ratliff <rratliff at cisco.com>/* wrote:
> 
>     Your CM server should never have multiple IP addresses. I've
>     certainly no experience deploying IPT but it sounds to me like NAT
>     w! ith skinny/H.323 fixup is what you need.
> 
>     Did I mention to absolutely NEVER put multiple IP addresses on your
>     CM? Well, at least if you want it to work...
> 
>     -Ryan
> 
>     On Dec 5, 2005, at 1:38 AM, Andrius Kislas wrote:
> 
>     Hello,
> 
>     I have a question about CallManager working as a SCCP signaling provider
>     for several networks using VPN over Internet. It so happens that two
>     networks are with the same IP adresses (192.168.0.x/24). What is the
>     "best practice" in such situation for CM to distinguish these networks?
>     I thought about two solutions:
> 
>     1) Put a secondary IP address on CM for the second identical network.
>     Then the make policy routing on a router, so that packets FROM one CM
>     address goes to the first network, and packets FROM the second address
>     goes to the second network.
> 
>     2) Configure NAT on VPN tunnels, so that packets from first network
>     reaches CM as one IP address and packets from second networks reaches CM
>     w! ith second address.
> 
>     I haven't tried neither of these solutions. Maybe there is more
>     intelligent way of solving the problem?
> 
>     Andrius
>     _______________________________________________
>     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
> 
> 
> ------------------------------------------------------------------------
> *Yahoo! Personals*
> Skip the bars and set-ups and start using Yahoo! Personals for free
> <http://us.rd.yahoo.com/evt=36107/*http://personals.yahoo.com/us/reg/free7days
>>
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From jfaulhaber at avodaq.com  Tue Dec  6 10:31:27 2005
From: jfaulhaber at avodaq.com (Jens Faulhaber)
Date: Tue Dec  6 10:31:37 2005
Subject: [cisco-voip] RASAggregator trunk issue
Message-ID: <CBE25A9701EB75489AF31FE565DD3F9FBF0F5A@sta-msg-1.sta.avodaq.com>

Hi Guys mabe one of you can help us.....

 

We have the following issue:



We are using CCM 4.1.3 SR 1 and we have the issue that the RASAggregator trunk is not working properly.

It is setup as described in the SRND but our NetMeeting clients are not able to call any client registered on CCM. They can receive calls but cannot set up the call by themselves.

 

The configuration looks like that.

 

H323TRM01 - gatekeeper controlled - this is the RASAggregator trunk to our gatekeeper - and successfully registered on the GK. With the DN 1101 

H323TRM03 - registered on CCM via IP with the DN 1002

 

Gatekeeper on 2611 with IOS  2600-ix-mz.123-14.T5.bin and the GK config.

zone local sj-zone cisco.com invia sj-zone outvia sj-zone enable-intrazone

 gw-type-prefix 1#* default-technology

 no use-proxy sj-zone default inbound-to terminal

 no use-proxy sj-zone default outbound-from terminal

 no shutdown

 endpoint ttl 60

 

the GK show command show that everything should be fine....

 

GK01#sh gatekeeper endpoints

                    GATEKEEPER ENDPOINT REGISTRATION

                    ================================

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name         Type    Flags

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

10.1.1.1        49875 10.1.1.1        49821 sj-zone           H323-GW

    H323-ID: RasAggregator16_sj-zone_1

    Voice Capacity Max.=  Avail.=  Current.= 0

10.128.128.65   1720  10.128.128.65   2295  sj-zone           TERM

    E164-ID: 1103

10.128.128.113  1720  10.128.128.113  2363  sj-zone           TERM

    E164-ID: 1101

Total number of active registrations = 3

 

GK01#sh gatekeeper gw-type-prefix

GATEWAY TYPE PREFIX TABLE

=========================

Prefix: 1#*    (Default gateway-technology)

  Zone sj-zone master gateway list:

    10.1.1.1:49875 RasAggregator16_sj-zone_1

 

 

 

The CallManager can process the call from H323TRM03 (non GK controlled) to the H323TRM01 (GK controlled)

In the CCM trace we find the info that the CCM cannot process the call because he does not find the DN or GW

 

We find that CC trace output for every unsucessfull call:

 

}|<CLID::StandAloneCluster><NID::10.1.1.1>

12/06/2005 10:56:42.942 CCM|}|<CLID::StandAloneCluster><NID::10.1.1.1>

12/06/2005 10:56:42.958 CCM|H225Handler::getSourceE164Address ERROR Could not find e164 address|<CLID::StandAloneCluster><NID::10.1.1.1><CT::1,100,132,143.1><IP::10.128.128.113><DEV::>

12/06/2005 10:56:42.958 CCM|e164 H225Handler::parseAndFwdH225uuie ERROR Incoming H225 call, Could not find H225D (client or gateway is missing)|<CLID::StandAloneCluster><NID::10.1.1.1><CT::1,100,132,143.1><IP::10.128.128.113><DEV::>

12/06/2005 10:56:42.958 CCM|value H323-UserInformation ::= |<CLID::StandAloneCluster><NID::10.1.1.1>

 

And now the  Gatekeeper gets a DRQ from the CCM and the call fails.... 

 

Do you have any ideas where our issues are? 

 

 

___________________

Jens Faulhaber

Senior IT-Consultant

 

avodaq AG

Moosstra?e 16

82319 Starnberg

 

Tel.:   +49 8151 36858347

Mobil: +49 163 7788247

Fax:   +49 8151 91 8382

http://www.avodaq.com <http://www.avodaq.com/> 

 

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From rratliff at cisco.com  Tue Dec  6 10:56:48 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue Dec  6 10:56:49 2005
Subject: [cisco-voip] RASAggregator trunk issue
In-Reply-To: <CBE25A9701EB75489AF31FE565DD3F9FBF0F5A@sta-msg-1.sta.avodaq.com>
References: <CBE25A9701EB75489AF31FE565DD3F9FBF0F5A@sta-msg-1.sta.avodaq.com>
Message-ID: <6D0B5443-0B9B-4C3E-A877-DD3B0C06EA86@cisco.com>

Is H323TRM03 added as an H.323 gateway in the CCMAdmin?  By default  
CM will not accept calls from H.323 endpoints that are not in the  
database.

-Ryan

On Dec 6, 2005, at 10:31 AM, Jens Faulhaber wrote:

Hi Guys mabe one of you can help us.....



We have the following issue:


We are using CCM 4.1.3 SR 1 and we have the issue that the  
RASAggregator trunk is not working properly.

It is setup as described in the SRND but our NetMeeting clients are  
not able to call any client registered on CCM. They can receive calls  
but cannot set up the call by themselves.



The configuration looks like that.



H323TRM01 ? gatekeeper controlled ? this is the RASAggregator trunk  
to our gatekeeper ? and successfully registered on the GK. With the  
DN 1101

H323TRM03 ? registered on CCM via IP with the DN 1002



Gatekeeper on 2611 with IOS  2600-ix-mz.123-14.T5.bin and the GK config.

zone local sj-zone cisco.com invia sj-zone outvia sj-zone enable- 
intrazone

  gw-type-prefix 1#* default-technology

  no use-proxy sj-zone default inbound-to terminal

  no use-proxy sj-zone default outbound-from terminal

  no shutdown

  endpoint ttl 60



the GK show command show that everything should be fine....



GK01#sh gatekeeper endpoints

                     GATEKEEPER ENDPOINT REGISTRATION

                     ================================

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name         Type     
Flags

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

10.1.1.1        49875 10.1.1.1        49821 sj-zone           H323-GW

     H323-ID: RasAggregator16_sj-zone_1

     Voice Capacity Max.=  Avail.=  Current.= 0

10.128.128.65   1720  10.128.128.65   2295  sj-zone           TERM

     E164-ID: 1103

10.128.128.113  1720  10.128.128.113  2363  sj-zone           TERM

     E164-ID: 1101

Total number of active registrations = 3



GK01#sh gatekeeper gw-type-prefix

GATEWAY TYPE PREFIX TABLE

=========================

Prefix: 1#*    (Default gateway-technology)

   Zone sj-zone master gateway list:

     10.1.1.1:49875 RasAggregator16_sj-zone_1







The CallManager can process the call from H323TRM03 (non GK  
controlled) to the H323TRM01 (GK controlled)

In the CCM trace we find the info that the CCM cannot process the  
call because he does not find the DN or GW



We find that CC trace output for every unsucessfull call:



}|<CLID::StandAloneCluster><NID::10.1.1.1>

12/06/2005 10:56:42.942 CCM|}|<CLID::StandAloneCluster><NID::10.1.1.1>

12/06/2005 10:56:42.958 CCM|H225Handler::getSourceE164Address ERROR  
Could not find e164 address|<CLID::StandAloneCluster><NID:: 
10.1.1.1><CT::1,100,132,143.1><IP::10.128.128.113><DEV::>

12/06/2005 10:56:42.958 CCM|e164 H225Handler::parseAndFwdH225uuie  
ERROR Incoming H225 call, Could not find H225D (client or gateway is  
missing)|<CLID::StandAloneCluster><NID::10.1.1.1><CT:: 
1,100,132,143.1><IP::10.128.128.113><DEV::>

12/06/2005 10:56:42.958 CCM|value H323-UserInformation ::= | 
<CLID::StandAloneCluster><NID::10.1.1.1>



And now the  Gatekeeper gets a DRQ from the CCM and the call fails....



Do you have any ideas where our issues are?





___________________

Jens Faulhaber

Senior IT-Consultant



avodaq AG

Moosstra?e 16

82319 Starnberg



Tel.:   +49 8151 36858347

Mobil: +49 163 7788247

Fax:   +49 8151 91 8382

http://www.avodaq.com



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



From jfaulhaber at avodaq.com  Tue Dec  6 11:00:54 2005
From: jfaulhaber at avodaq.com (Jens Faulhaber)
Date: Tue Dec  6 11:01:03 2005
Subject: AW: [cisco-voip] RASAggregator trunk issue
Message-ID: <CBE25A9701EB75489AF31FE565DD3F9FBF0F63@sta-msg-1.sta.avodaq.com>

Its in configured in the CM as a H.323 Client - but gatekeeper controlled. -> in the NetMeeting we entered the GK address and E.164 info.



-----Urspr?ngliche Nachricht-----
Von: Ryan Ratliff [mailto:rratliff at cisco.com] 
Gesendet: Dienstag, 6. Dezember 2005 16:57
An: Jens Faulhaber
Cc: cisco-voip at puck.nether.net
Betreff: Re: [cisco-voip] RASAggregator trunk issue

Is H323TRM03 added as an H.323 gateway in the CCMAdmin?  By default  
CM will not accept calls from H.323 endpoints that are not in the  
database.

-Ryan

On Dec 6, 2005, at 10:31 AM, Jens Faulhaber wrote:

Hi Guys mabe one of you can help us.....



We have the following issue:


We are using CCM 4.1.3 SR 1 and we have the issue that the  
RASAggregator trunk is not working properly.

It is setup as described in the SRND but our NetMeeting clients are  
not able to call any client registered on CCM. They can receive calls  
but cannot set up the call by themselves.



The configuration looks like that.



H323TRM01 - gatekeeper controlled - this is the RASAggregator trunk  
to our gatekeeper - and successfully registered on the GK. With the  
DN 1101

H323TRM03 - registered on CCM via IP with the DN 1002



Gatekeeper on 2611 with IOS  2600-ix-mz.123-14.T5.bin and the GK config.

zone local sj-zone cisco.com invia sj-zone outvia sj-zone enable- 
intrazone

  gw-type-prefix 1#* default-technology

  no use-proxy sj-zone default inbound-to terminal

  no use-proxy sj-zone default outbound-from terminal

  no shutdown

  endpoint ttl 60



the GK show command show that everything should be fine....



GK01#sh gatekeeper endpoints

                     GATEKEEPER ENDPOINT REGISTRATION

                     ================================

CallSignalAddr  Port  RASSignalAddr   Port  Zone Name         Type     
Flags

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

10.1.1.1        49875 10.1.1.1        49821 sj-zone           H323-GW

     H323-ID: RasAggregator16_sj-zone_1

     Voice Capacity Max.=  Avail.=  Current.= 0

10.128.128.65   1720  10.128.128.65   2295  sj-zone           TERM

     E164-ID: 1103

10.128.128.113  1720  10.128.128.113  2363  sj-zone           TERM

     E164-ID: 1101

Total number of active registrations = 3



GK01#sh gatekeeper gw-type-prefix

GATEWAY TYPE PREFIX TABLE

=========================

Prefix: 1#*    (Default gateway-technology)

   Zone sj-zone master gateway list:

     10.1.1.1:49875 RasAggregator16_sj-zone_1







The CallManager can process the call from H323TRM03 (non GK  
controlled) to the H323TRM01 (GK controlled)

In the CCM trace we find the info that the CCM cannot process the  
call because he does not find the DN or GW



We find that CC trace output for every unsucessfull call:



}|<CLID::StandAloneCluster><NID::10.1.1.1>

12/06/2005 10:56:42.942 CCM|}|<CLID::StandAloneCluster><NID::10.1.1.1>

12/06/2005 10:56:42.958 CCM|H225Handler::getSourceE164Address ERROR  
Could not find e164 address|<CLID::StandAloneCluster><NID:: 
10.1.1.1><CT::1,100,132,143.1><IP::10.128.128.113><DEV::>

12/06/2005 10:56:42.958 CCM|e164 H225Handler::parseAndFwdH225uuie  
ERROR Incoming H225 call, Could not find H225D (client or gateway is  
missing)|<CLID::StandAloneCluster><NID::10.1.1.1><CT:: 
1,100,132,143.1><IP::10.128.128.113><DEV::>

12/06/2005 10:56:42.958 CCM|value H323-UserInformation ::= | 
<CLID::StandAloneCluster><NID::10.1.1.1>



And now the  Gatekeeper gets a DRQ from the CCM and the call fails....



Do you have any ideas where our issues are?





___________________

Jens Faulhaber

Senior IT-Consultant



avodaq AG

Moosstra?e 16

82319 Starnberg



Tel.:   +49 8151 36858347

Mobil: +49 163 7788247

Fax:   +49 8151 91 8382

http://www.avodaq.com



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




From jeff at ocjtech.us  Tue Dec  6 11:26:54 2005
From: jeff at ocjtech.us (Jeffrey C. Ollie)
Date: Tue Dec  6 11:27:02 2005
Subject: [cisco-voip] CDR Reporting Tool
Message-ID: <1133886414.3355.13.camel@lt16585.campus.dmacc.edu>

Can anyone point me to a CDR reporting tool that:

1) Will directly access the Cisco CDR database (we've used Infortel but
it discards a lot of the information in the Cisco CDR records).
2) Provides more advanced reporting capabilities than ART provides.
3) Specifically targeted at CDR reports, not some generic SQL reporting
tool.

Jeff Ollie



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From hermann.baiter at epson-electronics.de  Tue Dec  6 11:28:26 2005
From: hermann.baiter at epson-electronics.de (Baiter, Hermann)
Date: Tue Dec  6 11:28:32 2005
Subject: [cisco-voip] 7960 hijacks the user interface when calls come in
Message-ID: <0287F73B115DD311948A00902799138507981DE0@eegexch.epson-electronics.de>

I face the same problem here. But when I try to add "Transfer" at "Ring In"
via CCM Web then I can add only following items:
Undefined
Answer
Immediate Divert

How can I add the Transfer button?

Thanks

Hermann Baiter
EPSON EUROPE ELECTRONICS GmbH
Riesstr. 15
80992 Muenchen - Germany
Tel: +49-(0)89-14 00 5-330
Fax: +49-(0)89-14 00 5-4330
Mobile: +49-(0)170-8 54 99 66
e-mail:Hermann.Baiter at epson-electronics.de
Please visit us at www.epson-electronics.de



> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Lange
> Sent: Monday, December 05, 2005 8:38 PM
> To: Schuett, Court (MBNAP it)
> Cc: Cisco-VoIP
> Subject: RE: [cisco-voip] 7960 hijacks the user interface 
> when calls come in
> 
> 
> Thanks Court.
> 
> You must be referring to the Cisco Call Manager web 
> interface? I don't see any similar settings on the phone itself.
> 
> Assuming you are talking about Call manager, does call 
> manager allow you to change the soft-key settings on the SIP 
> load of the phones?
> 
> Aside from the key-map settings, is there a way to prevent 
> the hijacking?
> 
> Thanks
> 
> John
> 
> On Mon, 2005-12-05 at 10:49 -0600, Schuett, Court (MBNAP it) wrote:
> > Device -> Device Settings -> Softkey Template
> > 
> > Under there, find the softkey template assigned to the phone in 
> > question. Then, click 'Configure Softkey Layout'.  You 
> should be able 
> > to change everything you need in there.  There is a button 
> for 'Ring 
> > In' which is where you'll probably need to add 'Transfer' to.
> > 
> > Hope that helps.
> > 
> >  
> > 
> > 
> > Court Schuett
> >  
> > Millward Brown
> > Network Systems Engineer
> > 630-955-8983
> > court.schuett at us.millwardbrown.com
> > 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of John Lange
> > Sent: Monday, December 05, 2005 10:37 AM
> > To: Cisco-VoIP
> > Subject: [cisco-voip] 7960 hijacks the user interface when 
> calls come 
> > in
> > 
> > We have a problem with our Cisco 7960s using the SIP load.
> > 
> > While trying to transfer a call, if another call comes in, 
> the entire 
> > user interface is "hijacked" and you can not complete the transfer 
> > process.
> > 
> > The only option available on the softkeys is "accept" (for the new 
> > call), so you can not complete the transfer.
> > 
> > This is a serious problem especially when trying to use the 
> 7960 as a 
> > switchboard phone causing many calls to be lost in "transfer limbo".
> > 
> > As a side issue, is there any way to reprogram the layout of the 
> > softkeys on the 7960? Having to press "more" to get to 
> "transfer" is 
> > just silly.
> > 
> > --
> > John Lange
> > 
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net 
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > ====================================================
> > This email is confidential and intended solely for the use of the 
> > individual or organization to whom it is addressed. Any opinions or 
> > advice presented are solely those of the author and do not 
> necessarily 
> > represent those of the Millward Brown Group of Companies.  
> If you are 
> > not the intended recipient of this email, you should not 
> copy, modify, 
> > distribute or take any action in reliance on it. If you 
> have received 
> > this email in error please notify the sender and delete this email 
> > from your system. Although this email has been checked for viruses 
> >  and other defects, no responsibility can be accepted for 
> any loss or 
> > damage arising from its receipt or use. 
> > ==================================================== 
> > 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

From lelio at uoguelph.ca  Tue Dec  6 11:32:12 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Dec  6 11:32:19 2005
Subject: [cisco-voip] CDR Reporting Tool
References: <1133886414.3355.13.camel@lt16585.campus.dmacc.edu>
Message-ID: <00b901c5fa82$9c4c5d70$30196883@cfs.uoguelph.ca>

I'd also like to hear about such a beast.

But I'd also like to hear about people's experience with setting up their own reports using a custom SQL report tool. I've looked at a few out there that seem quite good and allow you to create SQL reports with user input and are web based.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Jeffrey C. Ollie 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, December 06, 2005 11:26 AM
  Subject: [cisco-voip] CDR Reporting Tool

  Can anyone point me to a CDR reporting tool that:

  1) Will directly access the Cisco CDR database (we've used Infortel but
  it discards a lot of the information in the Cisco CDR records).
  2) Provides more advanced reporting capabilities than ART provides.
  3) Specifically targeted at CDR reports, not some generic SQL reporting
  tool.

  Jeff Ollie


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From jgrace at digitelusa.net  Tue Dec  6 12:06:10 2005
From: jgrace at digitelusa.net (James)
Date: Tue Dec  6 12:06:08 2005
Subject: [cisco-voip] Corp directory. Has all my service user acct listed
Message-ID: <1306481486-1133888765-cardhu_blackberry.rim.net-376-@engine18-cell02>

When I goto my corp dir on a ipphone .  It display all user configured ie jtapi admins.   How do I remove these user from corp dir
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From rratliff at cisco.com  Tue Dec  6 12:12:33 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue Dec  6 12:12:33 2005
Subject: [cisco-voip] Corp directory. Has all my service user acct listed
In-Reply-To: <1306481486-1133888765-cardhu_blackberry.rim.net-376-@engine18-cell02>
References: <1306481486-1133888765-cardhu_blackberry.rim.net-376-@engine18-cell02>
Message-ID: <8B63918D-7F80-4949-B131-475EF0E659D1@cisco.com>

http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_tech_note09186a00804d2087.shtml

-Ryan

On Dec 6, 2005, at 12:06 PM, James wrote:

When I goto my corp dir on a ipphone .  It display all user  
configured ie jtapi admins.   How do I remove these user from corp dir
James D. Grace
System Engineer
CCNP CCNA MCSE MCDBA
Digitel Corp.
_______________________________________________
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 Dec  6 12:12:32 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec  6 12:14:25 2005
Subject: [cisco-voip] CDR Reporting Tool
Message-ID: <D713462ED535184D830F6F6301753E7E018B3178@VISHNU.wesd.org>

we used our in house programmers and told them what we needed and they came up with a custom app for us.  Not web based but it interfaces with our billing system to automagically send bills each month based on total toll charges.  took him about 2.5 hours to do.
 
Scott

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
Sent: Tue 12/6/2005 8:32 AM
To: Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR Reporting Tool


I'd also like to hear about such a beast.
 
But I'd also like to hear about people's experience with setting up their own reports using a custom SQL report tool. I've looked at a few out there that seem quite good and allow you to create SQL reports with user input and are web based.
 
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ

	----- Original Message ----- 
	From: Jeffrey C. Ollie <mailto:jeff at ocjtech.us>  
	To: cisco-voip at puck.nether.net 
	Sent: Tuesday, December 06, 2005 11:26 AM
	Subject: [cisco-voip] CDR Reporting Tool
	 
	Can anyone point me to a CDR reporting tool that:
	 
	1) Will directly access the Cisco CDR database (we've used Infortel but
	it discards a lot of the information in the Cisco CDR records).
	2) Provides more advanced reporting capabilities than ART provides.
	3) Specifically targeted at CDR reports, not some generic SQL reporting
	tool.
	 
	Jeff Ollie
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	

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From rleetun at co.boulder.co.us  Tue Dec  6 12:26:49 2005
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Tue Dec  6 12:26:53 2005
Subject: [cisco-voip] CDR Reporting Tool
Message-ID: <963CE94787005149A52C17F33A028E12051D40EF@EVS1B.boco.co.boulder.co.us>

Dido here.  Our Oracle developers built a package in Crystal Reports for
management, which spits out about 10 to 15 different reports each month.


 

Rob

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, December 06, 2005 10:13 AM
To: Lelio Fulgenzi; Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CDR Reporting Tool

 

we used our in house programmers and told them what we needed and they
came up with a custom app for us.  Not web based but it interfaces with
our billing system to automagically send bills each month based on total
toll charges.  took him about 2.5 hours to do.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
Sent: Tue 12/6/2005 8:32 AM
To: Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR Reporting Tool

I'd also like to hear about such a beast.

 

But I'd also like to hear about people's experience with setting up
their own reports using a custom SQL report tool. I've looked at a few
out there that seem quite good and allow you to create SQL reports with
user input and are web based.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Jeffrey C. Ollie <mailto:jeff at ocjtech.us>  

	To: cisco-voip at puck.nether.net 

	Sent: Tuesday, December 06, 2005 11:26 AM

	Subject: [cisco-voip] CDR Reporting Tool

	 

	Can anyone point me to a CDR reporting tool that:

	 

	1) Will directly access the Cisco CDR database (we've used
Infortel but
	it discards a lot of the information in the Cisco CDR records).
	2) Provides more advanced reporting capabilities than ART
provides.
	3) Specifically targeted at CDR reports, not some generic SQL
reporting
	tool.

	 

	Jeff Ollie

	_______________________________________________
	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 Dec  6 12:48:56 2005
From: fwn at feasible.net (Fred Nielsen)
Date: Tue Dec  6 12:49:01 2005
Subject: [cisco-voip] Out palse a name
In-Reply-To: <1952665177-1133882635-cardhu_blackberry.rim.net-758-@engine11-cell02>
Message-ID: <005e01c5fa8d$5582b7d0$6401a8c0@ws1>

Depends on your carrier and trunk type.  Many carriers will allow you to
send whatever digits you wanted for calling party (caller ID or ANI) for an
outbound call on PRI or BRI.  Some restrict you to sending only those
numbers that are associated with your trunk.

No U.S. carriers I'm aware of support customer provided calling party name
fields however.  You can send them on something like a PRI, but they will
almost surely be ignored.  Many carriers now provision a static name (i.e.
"Company X") that those with Caller ID will see when establishing a customer
trunk these days.

Fred Nielsen
AT&T Inc.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James
Sent: Tuesday, December 06, 2005 7:24 AM
To: CiscosupportUpuck
Subject: [cisco-voip] Out palse a name

Is there anyway for me to out pulse a company name on cm.  I was told the
the carrier has to do name and we can do number if we wanted to
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.
_______________________________________________
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 Dec  6 15:20:43 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec  6 15:20:39 2005
Subject: [cisco-voip] phone to phone paging
Message-ID: <D713462ED535184D830F6F6301753E7E01A11E17@VISHNU.wesd.org>

I already have Berbee InformaCast for paging between phones.  So I know
about paging software. BUT I have an assistant on a PBX phone and the
admin just got a 7940 phone.  One DID on the 7940.  I know I could setup
a second extension on the admin's phone and set it to auto pick up on
speaker phone, but I was wondering if Cisco has made any kind of code
dialing to do this.  

 

What I'm thinking would be nice is to make a translation pattern of
3.xxxx would set the call pick up to speaker phone on that extension.
This way you don't have to burn a second DID / extension.  Does anyone
else think that would be a benefit?  Are current PBX does this with a
7+extension.

 

Scott

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From Scott.Voll at wesd.org  Tue Dec  6 17:02:03 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec  6 17:02:02 2005
Subject: [cisco-voip] CDR
Message-ID: <D713462ED535184D830F6F6301753E7E01A11E25@VISHNU.wesd.org>

I load my CDR each night at Midnight.  I'm getting alerts that my call
processing node CPU is pegging.  Any idea how to resolve this?  I also
got an alert that I have exceeded my row limit.  I decreased from 120
days to 90.  I'm hoping that corrects the second issue.

 

CM 4.1.3sr1.

 

Thanks

 

Scott

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From Hans-Peter.Walter at tds.de  Tue Dec  6 18:10:57 2005
From: Hans-Peter.Walter at tds.de (Hans-Peter Walter)
Date: Tue Dec  6 18:07:42 2005
Subject: [cisco-voip] Inline-Power / PoE: HWIC-D-9ESW-POE and  NM-16ESW-PWR
Message-ID: <OFFE638722.DDA723A8-ONC12570CF.007D39A6-C12570CF.007F0E30@de.tds-global.com>

just played around with the "dynamic configuration tool" 
(http://www.cisco.com/appcontent/apollo/configureHomeGuest.html)
I configured an 2811 with some inline power stuff.
When it checks the config, it reports some warnings (see below). Why? I 
think everything needed is there?!?
Anybody experience with the HWIC-D-9ESW-POE or  NM-16ESW-PWR? 
Recommendations? Differences?
And maybe somebody can explain the difference between PoE and Inline-Power 
in this case?
I think its the same, just want to be sure...

Thanks,
HP

Product             Description      Quantity 
CISCO2811-AC-IP     2811 w/ AC+POE,2FE,4HWICs,2PVDMs,1NME,2AIMS,IP 
BASE,64F/256D1 
S28NAESK9-12401     Cisco 2800 IOS ADVANCED ENTERPRISE SERVICES      1  
NM-16ESW-PWR        1 16 port 10/100  EtherSwitch NM with In-Line Power 
support 1 
PVDM2-16            16-Channel Packet Voice/Fax DSP Module      1 
HWIC-4ESW-POE       4-Port Ethernet Switch HWIC with Power Over Ethernet   
  1 
HWIC-D-9ESW-POE     9-Port Ethernet Switch HWIC with Power Over Ethernet   
  1 
VIC2-2BRI-NT/TE     Two-port Voice Interface Card - BRI (NT and TE)      1 
 

FL-SRST-SMALL       Feat Lic Survivable Remote Site Telephony up to 24 
phones   1 
PPWR-DCARD-16ESW    1 Power daughtercard for 16 port Etherswitch NM      1 
 
CAB-AC              Power Cord,110V      1 
PWR-2811-AC-IP      Cisco 2811 AC/IP power supply      1 
ROUTER-SDM          Device manager for routers      1 
MEM2800-256D-INC    256MB DDR DRAM Memory factory default for the Cisco 
2800    1 
MEM2800-64CF-INC    64MB CF default for Cisco 2800 Series      1 
CON-OSP-2811ACIP    ONSITE 24X7X4 2811 w/ AC+POE,2FE,4HWICs,2PVDMs,1NME,2  
    1 

==================================================================================================================
Cisco Configuration Tool Configuration Warnings 
1.  Integrated internal inline power supply (such as PWR-28XX-AC-IP) or an 
external power supply (such as PWR-CHASSIS-360W) is recommended for the 
NM-XXESW-PWR selection made. 
2.  Power Over Ethernet (POE) requires an AC plus IP system power supply 
(such as PWR-28XX-AC-IP). 
3.  Do you realize you can enable security services like Firewall/IPS, VPN 
and Network Admission Control(NAC) on this router for an incremental cost? 
Please order appropriate security bundle or image 
==================================================================================================================
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From lelio at uoguelph.ca  Tue Dec  6 18:23:10 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Dec  6 18:23:10 2005
Subject: [cisco-voip] Inline-Power / PoE: HWIC-D-9ESW-POE and NM-16ESW-PWR
References: <OFFE638722.DDA723A8-ONC12570CF.007D39A6-C12570CF.007F0E30@de.tds-global.com>
Message-ID: <003301c5fabc$06187930$ca01a8c0@deadfish>

According to the following documents, inline power is the Cisco pre-standard for power over ethernet and PoE is the new standard.

Inline Power:
http://www.cisco.com/en/US/netsol/ns340/ns394/ns165/networking_solutions_audience_business_benefit09186a0080154647.html

PoE
http://www.cisco.com/en/US/netsol/ns340/ns394/ns147/ns412/netbr09186a00801f4329.html

The best would be to read the specifications of each product to see exactly what you are getting and whether the product is 802.3af compatible or not. 

Then again, you may not need 802.3af. ;)

  ----- Original Message ----- 
  From: Hans-Peter Walter 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, December 06, 2005 6:10 PM
  Subject: [cisco-voip] Inline-Power / PoE: HWIC-D-9ESW-POE and NM-16ESW-PWR



  just played around with the "dynamic configuration tool" (http://www.cisco.com/appcontent/apollo/configureHomeGuest.html) 
  I configured an 2811 with some inline power stuff. 
  When it checks the config, it reports some warnings (see below). Why? I think everything needed is there?!? 
  Anybody experience with the HWIC-D-9ESW-POE or  NM-16ESW-PWR? Recommendations? Differences? 
  And maybe somebody can explain the difference between PoE and Inline-Power in this case? 
  I think its the same, just want to be sure... 

  Thanks, 
  HP 

  Product             Description                                                 Quantity  
  CISCO2811-AC-IP     2811 w/ AC+POE,2FE,4HWICs,2PVDMs,1NME,2AIMS,IP BASE,64F/256D1         
  S28NAESK9-12401     Cisco 2800 IOS ADVANCED ENTERPRISE SERVICES                 1         
  NM-16ESW-PWR        1 16 port 10/100  EtherSwitch NM with In-Line Power support 1         
  PVDM2-16            16-Channel Packet Voice/Fax DSP Module                      1         
  HWIC-4ESW-POE       4-Port Ethernet Switch HWIC with Power Over Ethernet        1         
  HWIC-D-9ESW-POE     9-Port Ethernet Switch HWIC with Power Over Ethernet        1         
  VIC2-2BRI-NT/TE     Two-port Voice Interface Card - BRI (NT and TE)             1         

  FL-SRST-SMALL       Feat Lic Survivable Remote Site Telephony up to 24 phones   1         
  PPWR-DCARD-16ESW    1 Power daughtercard for 16 port Etherswitch NM             1         
  CAB-AC              Power Cord,110V                                             1         
  PWR-2811-AC-IP      Cisco 2811 AC/IP power supply                               1         
  ROUTER-SDM          Device manager for routers                                  1         
  MEM2800-256D-INC    256MB DDR DRAM Memory factory default for the Cisco 2800    1         
  MEM2800-64CF-INC    64MB CF default for Cisco 2800 Series                       1         
  CON-OSP-2811ACIP    ONSITE 24X7X4 2811 w/ AC+POE,2FE,4HWICs,2PVDMs,1NME,2       1         

  ================================================================================================================== 
  Cisco Configuration Tool Configuration Warnings 
  1.  Integrated internal inline power supply (such as PWR-28XX-AC-IP) or an external power supply (such as PWR-CHASSIS-360W) is recommended for the NM-XXESW-PWR selection made. 
  2.  Power Over Ethernet (POE) requires an AC plus IP system power supply (such as PWR-28XX-AC-IP). 
  3.  Do you realize you can enable security services like Firewall/IPS, VPN and Network Admission Control(NAC) on this router for an incremental cost? Please order appropriate security bundle or image   
  ==================================================================================================================


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


  _______________________________________________
  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 Dec  6 18:49:14 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Dec  6 18:49:13 2005
Subject: [cisco-voip] CDR
References: <D713462ED535184D830F6F6301753E7E01A11E25@VISHNU.wesd.org>
Message-ID: <005801c5fabf$aa7f3c90$ca01a8c0@deadfish>

Don't feel bad - I export only 14 days and I get CPU alerts. We just ignore them. I'm sure there is a way to filter them with the mgmt tools, I think these CPU alarms are generated by ITEM, so you can ignore them from a specific time range.

Not sure about the row limit though. I think that has to do with the maximum number of rows a query will return. Why are you loading 120 days every night? 

  ----- Original Message ----- 
  From: Voll, Scott 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, December 06, 2005 5:02 PM
  Subject: [cisco-voip] CDR


  I load my CDR each night at Midnight.  I'm getting alerts that my call processing node CPU is pegging.  Any idea how to resolve this?  I also got an alert that I have exceeded my row limit.  I decreased from 120 days to 90.  I'm hoping that corrects the second issue.

   

  CM 4.1.3sr1.

   

  Thanks

   

  Scott



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


  _______________________________________________
  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 Dec  6 21:12:40 2005
From: fwn at feasible.net (Fred Nielsen)
Date: Tue Dec  6 21:12:44 2005
Subject: [cisco-voip] phone to phone paging
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A11E17@VISHNU.wesd.org>
Message-ID: <009801c5fad3$b417a8a0$6401a8c0@ws1>

It doesn't necessarily have to be a DID. but you are right you do need to
burn an extension the way it works today. But frankly, the extension isn't
really a big deal, as you can mask it from the end-users completely and use
something really obscure that would never conflict (i.e make the real
extension something like 1111xxxx or whatever lead-in digits that won't
overlay with your dialing plan.. and translate appropriately).

 

What bothers me the most is that you have to burn a line appearance.  Those
can be precious when deploying 7940s as your standard.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, December 06, 2005 12:21 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] phone to phone paging

 

I already have Berbee InformaCast for paging between phones.  So I know
about paging software. BUT I have an assistant on a PBX phone and the admin
just got a 7940 phone.  One DID on the 7940.  I know I could setup a second
extension on the admin's phone and set it to auto pick up on speaker phone,
but I was wondering if Cisco has made any kind of code dialing to do this.  

 

What I'm thinking would be nice is to make a translation pattern of 3.xxxx
would set the call pick up to speaker phone on that extension.  This way you
don't have to burn a second DID / extension.  Does anyone else think that
would be a benefit?  Are current PBX does this with a 7+extension.

 

Scott

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From jgrace at digitelusa.net  Tue Dec  6 21:37:50 2005
From: jgrace at digitelusa.net (James Grace)
Date: Tue Dec  6 21:40:08 2005
Subject: [cisco-voip] is there any known issues with installing Mcafee on
	callmanagers and unity 
Message-ID: <006601c5fad7$38218280$030a0f0a@digitelcorp.com>

our customer wants us to install mcafee virus protection. on all the avvid servers CM, unity and crs.  is there any known issues or gotchas a need to look out for.
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From pnowicki at neo.rr.com  Tue Dec  6 23:24:06 2005
From: pnowicki at neo.rr.com (pnowicki@neo.rr.com)
Date: Tue Dec  6 23:24:11 2005
Subject: [cisco-voip] is there any known issues with installing Mcafee on
	callmanagers and unity
Message-ID: <48add8489285.48928548add8@columbus.rr.com>

I have used it a bunch of times before with no problems


For CallManager
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_bulletin0900a
ecd800f617e.html

I can not find the doc right now, but I know that it has been tested 
with Unity 4.0 as well.
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our customer wants us to install mcafee virus protection. on all the avvid servers CM, unity and crs.  is there any known issues or gotchas a need to look out for.
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From mihi at ludens.elte.hu  Wed Dec  7 03:28:32 2005
From: mihi at ludens.elte.hu (MIHI)
Date: Wed Dec  7 03:28:38 2005
Subject: [cisco-voip] Inline-Power / PoE: HWIC-D-9ESW-POE and NM-16ESW-PWR
In-Reply-To: <OFFE638722.DDA723A8-ONC12570CF.007D39A6-C12570CF.007F0E30@de.tds-global.com>
Message-ID: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAA4TU9w2JpAEaGBBSVmG+/fsKAAAAQAAAAkQVwKB/6k0iUAhgupt3I3QEAAAAA@ludens.elte.hu>

Hello Hans-Peter,
 
The HWIC cards can supply PoE but also support Cisco's pre-standard inline
power. This is fine for all Cisco phones.
 
The NM-16ESW is an earlier generation EtherSwitch module supporting only
Cisco's pre-standard inline power. This is fine for most Cisco phones, only
some of the latest GE models require 802.3af PoE.
 
If you use a new generation NME-16ES-1G-P instead of NM-16ESW to get PoE on
all ports, more powerful switching and an additional 10/100/1000 port.
 
You might get the warning because you put too many PoE ports into the
chassis. You included the appropriate power supply, which is fine, but the
2811
<http://www.cisco.com/en/US/products/ps5855/products_data_sheet0900aecd8028d
15f.html#wp9000542> has 160W. 
 
160 / (4 + 8 + 16) results with 5.7 which seems to be less than what is
required by an IP phone, so take care.
 
Marton

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Hans-Peter Walter
Sent: Wednesday, December 07, 2005 12:11 AM
To: cisco-voip at puck.nether.net
Subject: **SPAM** [cisco-voip] Inline-Power / PoE: HWIC-D-9ESW-POE and
NM-16ESW-PWR



just played around with the "dynamic configuration tool"
(http://www.cisco.com/appcontent/apollo/configureHomeGuest.html) 
I configured an 2811 with some inline power stuff. 
When it checks the config, it reports some warnings (see below). Why? I
think everything needed is there?!? 
Anybody experience with the HWIC-D-9ESW-POE or  NM-16ESW-PWR?
Recommendations? Differences? 
And maybe somebody can explain the difference between PoE and Inline-Power
in this case? 
I think its the same, just want to be sure... 

Thanks, 
HP 

Product             Description
Quantity  
CISCO2811-AC-IP     2811 w/ AC+POE,2FE,4HWICs,2PVDMs,1NME,2AIMS,IP
BASE,64F/256D1         
S28NAESK9-12401     Cisco 2800 IOS ADVANCED ENTERPRISE SERVICES
1         
NM-16ESW-PWR        1 16 port 10/100  EtherSwitch NM with In-Line Power
support 1         
PVDM2-16            16-Channel Packet Voice/Fax DSP Module
1         
HWIC-4ESW-POE       4-Port Ethernet Switch HWIC with Power Over Ethernet
1         
HWIC-D-9ESW-POE     9-Port Ethernet Switch HWIC with Power Over Ethernet
1         
VIC2-2BRI-NT/TE     Two-port Voice Interface Card - BRI (NT and TE)
1         

FL-SRST-SMALL       Feat Lic Survivable Remote Site Telephony up to 24
phones   1         
PPWR-DCARD-16ESW    1 Power daughtercard for 16 port Etherswitch NM
1         
CAB-AC              Power Cord,110V
1         
PWR-2811-AC-IP      Cisco 2811 AC/IP power supply
1         
ROUTER-SDM          Device manager for routers
1         
MEM2800-256D-INC    256MB DDR DRAM Memory factory default for the Cisco 2800
1         
MEM2800-64CF-INC    64MB CF default for Cisco 2800 Series
1         
CON-OSP-2811ACIP    ONSITE 24X7X4 2811 w/ AC+POE,2FE,4HWICs,2PVDMs,1NME,2
1         

============================================================================
====================================== 
Cisco Configuration Tool Configuration Warnings 
1.  Integrated internal inline power supply (such as PWR-28XX-AC-IP) or an
external power supply (such as PWR-CHASSIS-360W) is recommended for the
NM-XXESW-PWR selection made. 
2.  Power Over Ethernet (POE) requires an AC plus IP system power supply
(such as PWR-28XX-AC-IP). 
3.  Do you realize you can enable security services like Firewall/IPS, VPN
and Network Admission Control(NAC) on this router for an incremental cost?
Please order appropriate security bundle or image   
============================================================================
======================================
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From su1droot at gmail.com  Wed Dec  7 10:49:13 2005
From: su1droot at gmail.com (su1droot)
Date: Wed Dec  7 10:49:15 2005
Subject: [cisco-voip] Phone conversations delayed
Message-ID: <f83dffc80512070749n23bd847dge4bb3bdb5b674d67@mail.gmail.com>

Hello,
We are having an issues with some IP Phone to IP phones calls across our WAN
where the conversation will become delayed. It is best described where the 2
parties will become out of sync from each other each other by about a second
or 2. This doesn't appear to be every call just some calls.

There are no visible issues on the WAN interfaces involved and ping times
are pretty good between the sites.

Does anyone have any advise on good ways to troubleshoot this and resolve
it?  My CCO searches haven't been too successful.
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From kthorngr at cisco.com  Wed Dec  7 10:54:48 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed Dec  7 10:54:58 2005
Subject: [cisco-voip] Phone conversations delayed
In-Reply-To: <f83dffc80512070749n23bd847dge4bb3bdb5b674d67@mail.gmail.com>
References: <f83dffc80512070749n23bd847dge4bb3bdb5b674d67@mail.gmail.com>
Message-ID: <b4860b18625d5d2d97006ff37034f525@cisco.com>

Sounds like a QoS issue in your WAN.  I would recommend you start with 
the Enterprise QoS SRND that can be found at 
http://www.cisco.com/go/srnd.

Kevin
On Dec 7, 2005, at 10:49 AM, su1droot wrote:

> Hello,
> We are having an issues with some IP Phone to IP phones calls across 
> our WAN where the conversation will become delayed. It is best 
> described where the 2 parties will become out of sync from each other 
> each other by about a second or 2. This doesn't appear to be every 
> call just some calls.
>
> There are no visible issues on the WAN interfaces involved and ping 
> times are pretty good between the sites.
>
> Does anyone have any advise on good ways to troubleshoot this and 
> resolve it?? My CCO searches haven't been too successful.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From jgrace at digitelusa.net  Wed Dec  7 13:36:37 2005
From: jgrace at digitelusa.net (James Grace)
Date: Wed Dec  7 13:37:16 2005
Subject: [cisco-voip] what ver of csa is compadiable with unity 4.0.5
Message-ID: <200512071336.AA1924661458@mail.digitelusa.net>

what ver of csa is compadiable with unity 4.0.5
---



From gboda at eircom.ie  Wed Dec  7 13:42:54 2005
From: gboda at eircom.ie (gboda@eircom.ie)
Date: Wed Dec  7 13:43:07 2005
Subject: [cisco-voip] SCCP to H.323 calls
Message-ID: <1BF7574B8535D5119F790002B3304FE312EE8E4E@EIRCOMMSG02>


Hi All,

How would the H.245 channel be set up in the following call scenario:

7960->CCM->Intercluster Trunk (H.323)->H.323 softswitch->PSTN->analog phone

Would CCM negotiate the codec and other media parameters with the H.323
gateway on behalf of the phone? How would the phone know what codec to use
and where to send the RTP packets? Would we need an MTP?

Thanks.

Greg



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From andy at dignans.com  Wed Dec  7 14:07:03 2005
From: andy at dignans.com (Andrew Dignan)
Date: Wed Dec  7 14:07:06 2005
Subject: [cisco-voip] is there any known issues with installing Mcafee 
	on callmanagers and unity
In-Reply-To: <48add8489285.48928548add8@columbus.rr.com>
References: <48add8489285.48928548add8@columbus.rr.com>
Message-ID: <2891.64.81.226.84.1133982423.squirrel@www.dignans.com>

CallManager/CRS

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/sec_vir/mca_ids/index.htm

This still applies for Unity 4.x
http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_configuration_example09186a00801abf25.shtml#k6

> I have used it a bunch of times before with no problems
>
>
> For CallManager
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_bulletin0900a
> ecd800f617e.html
>
> I can not find the doc right now, but I know that it has been tested
> with Unity 4.0 as well.
> our customer wants us to install mcafee virus protection. on all the avvid
> servers CM, unity and crs.  is there any known issues or gotchas a need to
> look out for.
> _______________________________________________
> 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 andy at dignans.com  Wed Dec  7 14:09:22 2005
From: andy at dignans.com (Andrew Dignan)
Date: Wed Dec  7 14:09:24 2005
Subject: [cisco-voip] what ver of csa is compadiable with unity 4.0.5
In-Reply-To: <200512071336.AA1924661458@mail.digitelusa.net>
References: <200512071336.AA1924661458@mail.digitelusa.net>
Message-ID: <2933.64.81.226.84.1133982562.squirrel@www.dignans.com>

http://www.cisco.com/cgi-bin/tablebuild.pl/unity3d

CiscoUnity-CSA-4.5.1.639-2.0.2-K9.exe

That is the most recent version.

> what ver of csa is compadiable with unity 4.0.5
> ---
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From it at cimgroup.com  Wed Dec  7 18:27:08 2005
From: it at cimgroup.com (IT)
Date: Wed Dec  7 18:23:17 2005
Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
Message-ID: <B487D4C1FE6D04439061018C6EBC109B025C3640@exchange.cimgroup.com>

Hi,

My PRI on a 2801 just died, and SBC said they were able to bring the
line back up just by running a test. They said that this "rebooted" the
line, and when they first logged in they saw that the lines were
"locked" out, and this was generally caused by my equipment. What can I
do to verify if this is correct? What did they mean? It's a serial card,
and here is the show intereface serial 0/2/0:23 output:

show interfaces serial 0/2/0:23
Serial0/2/0:23 is up, line protocol is up (spoofing)
  Hardware is DSX1
  MTU 1500 bytes, BW 64 Kbit, DLY 20000 usec,
     reliability 255/255, txload 1/255, rxload 1/255
  Encapsulation HDLC, loopback not set
  Last input 00:00:15, output 00:00:15, output hang never
  Last clearing of "show interface" counters never
  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
  Queueing strategy: weighted fair
  Output queue: 0/1000/64/0 (size/max total/threshold/drops)
     Conversations  0/1/256 (active/max active/max total)
     Reserved Conversations 0/0 (allocated/max allocated)
     Available Bandwidth 48 kilobits/sec
  5 minute input rate 0 bits/sec, 0 packets/sec
  5 minute output rate 0 bits/sec, 0 packets/sec
     204146 packets input, 973103 bytes, 0 no buffer
     Received 0 broadcasts, 0 runts, 2 giants, 0 throttles
     115437 input errors, 115437 CRC, 51503 frame, 30018 overrun, 0
ignored, 85696 abort
     204467 packets output, 978336 bytes, 0 underruns
     0 output errors, 0 collisions, 12 interface resets
     0 output buffer failures, 0 output buffers swapped out
     30 carrier transitions
  Timeslot(s) Used:24, SCC: 0, Transmitter delay is 0 flags

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
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From bababooey at cox.net  Wed Dec  7 21:04:54 2005
From: bababooey at cox.net (Bob A. Bowie)
Date: Wed Dec  7 21:05:04 2005
Subject: [cisco-voip] Phone number display CME
References: <200512071700.jB7H09JS071480@puck.nether.net>
Message-ID: <000501c5fb9b$c990f470$0f0110ac@corporation2>

Is there a way to remove the phone number from the top line of the IP Phone 
display
in Cisco CME?




From highspeedsnow at gmail.com  Wed Dec  7 22:18:54 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Wed Dec  7 22:19:36 2005
Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B025C3640@exchange.cimgroup.com>
Message-ID: <4397a646.15ac7442.5866.5ee5@mx.gmail.com>

Do a
'sh isdn status'
and a
'sh isdn service'

And send us the results ....

-Mark

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of IT
> Sent: Wednesday, December 07, 2005 6:27 PM
> To: Jose Quesada; cisco-voip at puck.nether.net
> Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
> 
> Hi,
> 
> My PRI on a 2801 just died, and SBC said they were able to bring the
> line back up just by running a test. They said that this "rebooted" the
> line, and when they first logged in they saw that the lines were
> "locked" out, and this was generally caused by my equipment. What can I
> do to verify if this is correct? What did they mean? It's a serial card,
> and here is the show intereface serial 0/2/0:23 output:
> 
> show interfaces serial 0/2/0:23
> Serial0/2/0:23 is up, line protocol is up (spoofing)
>   Hardware is DSX1
>   MTU 1500 bytes, BW 64 Kbit, DLY 20000 usec,
>      reliability 255/255, txload 1/255, rxload 1/255
>   Encapsulation HDLC, loopback not set
>   Last input 00:00:15, output 00:00:15, output hang never
>   Last clearing of "show interface" counters never
>   Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
>   Queueing strategy: weighted fair
>   Output queue: 0/1000/64/0 (size/max total/threshold/drops)
>      Conversations  0/1/256 (active/max active/max total)
>      Reserved Conversations 0/0 (allocated/max allocated)
>      Available Bandwidth 48 kilobits/sec
>   5 minute input rate 0 bits/sec, 0 packets/sec
>   5 minute output rate 0 bits/sec, 0 packets/sec
>      204146 packets input, 973103 bytes, 0 no buffer
>      Received 0 broadcasts, 0 runts, 2 giants, 0 throttles
>      115437 input errors, 115437 CRC, 51503 frame, 30018 overrun, 0
> ignored, 85696 abort
>      204467 packets output, 978336 bytes, 0 underruns
>      0 output errors, 0 collisions, 12 interface resets
>      0 output buffer failures, 0 output buffers swapped out
>      30 carrier transitions
>   Timeslot(s) Used:24, SCC: 0, Transmitter delay is 0 flags
> 
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From jgrace at digitelusa.net  Wed Dec  7 23:05:46 2005
From: jgrace at digitelusa.net (James Grace)
Date: Wed Dec  7 23:06:23 2005
Subject: [cisco-voip] how to setup agent-initiated recording 
Message-ID: <200512072305.AA1343815904@mail.digitelusa.net>

how to setup agent-initiated recording 
---



From it at cimgroup.com  Wed Dec  7 23:17:50 2005
From: it at cimgroup.com (IT)
Date: Wed Dec  7 23:14:54 2005
Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
Message-ID: <B487D4C1FE6D04439061018C6EBC109B025C37E8@exchange.cimgroup.com>

The line is back up now, but here are the results of the two commands.
Thanks,
Avidan


cimgroupsanfran#show isdn status
Global ISDN Switchtype = primary-4ess

%Q.931 is backhauled to CCM-MANAGER on DSL 0. Layer 3 output may not
apply

ISDN Serial0/2/0:23 interface
        dsl 0, interface ISDN Switchtype = primary-ni
        L2 Protocol = Q.921  L3 Protocol(s) = CCM-MANAGER
    Layer 1 Status:
        ACTIVE
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
    Layer 3 Status:
        0 Active Layer 3 Call(s)
    Active dsl 0 CCBs = 0
    The Free Channel Mask:  0x807FFFFF
    Number of L2 Discards = 0, L2 Session ID = 117
    Total Allocated ISDN CCBs = 0


cimgroupsanfran#show isdn service
PRI Channel Statistics:

%Q.931 is backhauled to CCM-MANAGER on DSL 0. Layer 3 output may not
apply

ISDN Se0/2/0:23, Channel [1-24]
  Configured Isdn Interface (dsl) 0
   Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart
5=Maint_Pend)
    Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
    State   :  0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3
   Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend
9=OOSPend)
    Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
    State   :  0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2

-----Original Message-----
From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: Wednesday, December 07, 2005 7:19 PM
To: IT; Jose Quesada; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] PRI serial "lockout"? what is SBC talking
about

Do a
'sh isdn status'
and a
'sh isdn service'

And send us the results ....

-Mark

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of IT
> Sent: Wednesday, December 07, 2005 6:27 PM
> To: Jose Quesada; cisco-voip at puck.nether.net
> Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
> 
> Hi,
> 
> My PRI on a 2801 just died, and SBC said they were able to bring the
> line back up just by running a test. They said that this "rebooted"
the
> line, and when they first logged in they saw that the lines were
> "locked" out, and this was generally caused by my equipment. What can
I
> do to verify if this is correct? What did they mean? It's a serial
card,
> and here is the show intereface serial 0/2/0:23 output:
> 
> show interfaces serial 0/2/0:23
> Serial0/2/0:23 is up, line protocol is up (spoofing)
>   Hardware is DSX1
>   MTU 1500 bytes, BW 64 Kbit, DLY 20000 usec,
>      reliability 255/255, txload 1/255, rxload 1/255
>   Encapsulation HDLC, loopback not set
>   Last input 00:00:15, output 00:00:15, output hang never
>   Last clearing of "show interface" counters never
>   Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops:
0
>   Queueing strategy: weighted fair
>   Output queue: 0/1000/64/0 (size/max total/threshold/drops)
>      Conversations  0/1/256 (active/max active/max total)
>      Reserved Conversations 0/0 (allocated/max allocated)
>      Available Bandwidth 48 kilobits/sec
>   5 minute input rate 0 bits/sec, 0 packets/sec
>   5 minute output rate 0 bits/sec, 0 packets/sec
>      204146 packets input, 973103 bytes, 0 no buffer
>      Received 0 broadcasts, 0 runts, 2 giants, 0 throttles
>      115437 input errors, 115437 CRC, 51503 frame, 30018 overrun, 0
> ignored, 85696 abort
>      204467 packets output, 978336 bytes, 0 underruns
>      0 output errors, 0 collisions, 12 interface resets
>      0 output buffer failures, 0 output buffers swapped out
>      30 carrier transitions
>   Timeslot(s) Used:24, SCC: 0, Transmitter delay is 0 flags
> 
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are
not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From it at cimgroup.com  Wed Dec  7 23:36:56 2005
From: it at cimgroup.com (IT)
Date: Wed Dec  7 23:33:04 2005
Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
Message-ID: <B487D4C1FE6D04439061018C6EBC109B025C37F3@exchange.cimgroup.com>

Perhaps this is of some use as well:

cimgroupsanfran#show controllers serial 0/2/0:23
Interface Serial0/2/0:23
Hardware is GT96Kidb at 0x64B5E154, driver data structure at 0x64C47628
wic_info 0x0
Physical Port 3, SCC Num 3
MPSC Registers:
MMCR_L=0x000304C0, MMCR_H=0x00000000, MPCR=0x00000000
CHR1=0x00FE007E, CHR2=0x00000000, CHR3=0x00000648, CHR4=0x00000000
CHR5=0x00000000, CHR6=0x00000000, CHR7=0x00000000, CHR8=0x00000000
CHR9=0x00000000, CHR10=0x00003008
SDMA Registers:
SDC=0x00002201, SDCM=0x00000080, SGC=0x0000C000
CRDP=0x0ECFDD50, CTDP=0x0ED22E90, FTDB=0x0ED22E90
Main Routing Register=0x0003F5C7 BRG Conf Register=0x00480000
Rx Clk Routing Register=0x76543280 Tx Clk Routing Register=0x76543290
GPP Registers:
Conf=0x7F00    , Io=0x500     , Data=0x7F7F477F, Level=0x0
Conf0=0x3000002 , Io0=0x6000600 , Data0=0x5B7FFDDF, Level0=0x18000000
FTDM Registers:
&TCR=0x24018B08, TCR=0x8A81A500
&tdpram[0]=0x24010B00, tdpram[0]=0xC0000
&tdpram[1]=0x24010B04, tdpram[1]=0xC0000
&tdpram[2]=0x24010B08, tdpram[2]=0xC0000
&tdpram[3]=0x24010B0C, tdpram[3]=0xC0000
&tdpram[4]=0x24010B10, tdpram[4]=0xC0000
&rdpram[0]=0x24014B00, rdpram[0]=0xC0000
&rdpram[1]=0x24014B04, rdpram[1]=0xC0000
&rdpram[2]=0x24014B08, rdpram[2]=0xC0000
&rdpram[3]=0x24014B0C, rdpram[3]=0xC0000
&rdpram[4]=0x24014B10, rdpram[4]=0xC0000
TDM FPGA Registers:
TDM FPGA Version: 20.0
vmcr[0] = 0x604 @ 0x4B000010, vmcr[1] = 0x40010 @ 0x4B000014,
 vmcr[2] = 0x80 @ 0x4B000018 vmcr[3]= 0x0 @ 0x4B00001C
vtcr[0] = 0x0 @ 0x4B000020, vtcr[1] = 0x0 @ 0x4B000024,
 vtcr[2] = 0x0 @ 0x4B000028, vtcr[3]= 0x0 @ 0x4B00002C
nhr = 0x3D00 @ 0x4B00004C, isr = 0x600 & 0x4B000050, imr = 0x0 @
0x4B000054
 atcr = 0x0 @ 0x4B000030
prcr = 0x10404 @ 0x4B000040, psr = 0xA0000E @ 0x4B000044, gtcr = 0x2 @
0x4B000034
rcr = 0x1F0011 @ 0x4B000004
pphr = 0x1 @ 0x4B000058
tshr = 0x1 @ 0x4B00005C
sr1 = 0x669 @ 0x4B00006C
sr2 = 0x52 @ 0x4B000070
sr3 = 0x5D @ 0x4B000074
chpir = 0x110011 @ 0x4B000064
drcr = 0x0 @ 0x4B000060
miscr = 0x416BFF @ 0x4B000000
d_ihpr = 0xEBDC690 @ 0x4B000080
d_ehpr[i] = 0xEBD4520 @ 0x4B000090
d_ehpr[i] = 0xEBD8000 @ 0x4B000094
d_ehpr[i] = 0xEBDC000 @ 0x4B000098
d_ehpr[i] = 0xEBE0000 @ 0x4B00009C
d_ehpr[i] = 0xEBD65D0 @ 0x4B0000A0
d_ehpr[i] = 0xEBE8000 @ 0x4B0000A4
d_ehpr[i] = 0xEBEC000 @ 0x4B0000A8
d_ehpr[i] = 0xEBF0000 @ 0x4B0000AC
85696 input aborts on receiving flag sequence
0 throttles, 0 enables
30018 overruns
0 transmitter underruns
0 transmitter CTS losts
320273 rxintr, 205204 txintr, 1 rxerr, 0 txerr
713588 mpsc_rx, 0 mpsc_rxerr, 892 mpsc_rlsc, 61222 mpsc_rhnt, 654341
mpsc_rfsc
43 mpsc_rcsc, 0 mpsc_rovr, 0 mpsc_rcdl, 0 mpsc_rckg, 0 mpsc_bper
0 mpsc_txerr, 0 mpsc_teidl, 0 mpsc_tudr, 0 mpsc_tctsl, 0 mpsc_tckg
0 sdma_rx_sf, 2 sdma_rx_mfl, 30018 sdma_rx_or, 85696 sdma_rx_abr, 51503
sdma_rx_no
0 sdma_rx_de, 0 sdma_rx_cdl, 115437 sdma_rx_ce, 0 sdma_tx_rl, 0
sdma_tx_ur
1 sdma_rx_reserr, 0 sdma_tx_reserr
0 rx_bogus_pkts, rx_bogus_flag FALSE
0 sdma_tx_ur_processed
fport 1, 0 ftdm_rint, 0 ftdm_rsl, 0 ftdm_tint, 0 ftdm_tsl

tx_limited = 1(2), errata19 count1 - 0, count2 - 0
Receive Ring
rxr head (5)(0x0ECFDD50), rxr tail (0)(0x0ECFDD00)
  rmd(ECFDD00): nbd ECFDD10 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED313E8
  rmd(ECFDD10): nbd ECFDD20 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2B448
  rmd(ECFDD20): nbd ECFDD30 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED30728
  rmd(ECFDD30): nbd ECFDD40 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2E0E8
  rmd(ECFDD40): nbd ECFDD50 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED247E8
  rmd(ECFDD50): nbd ECFDD60 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2CDC8
  rmd(ECFDD60): nbd ECFDD70 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2C768
  rmd(ECFDD70): nbd ECFDD80 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2F408
  rmd(ECFDD80): nbd ECFDD90 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2EDA8
  rmd(ECFDD90): nbd ECFDDA0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2FA68
  rmd(ECFDDA0): nbd ECFDDB0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED29468
  rmd(ECFDDB0): nbd ECFDDC0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED26E28
  rmd(ECFDDC0): nbd ECFDDD0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED234C8
  rmd(ECFDDD0): nbd ECFDDE0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2E748
  rmd(ECFDDE0): nbd ECFDDF0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED24E48
  rmd(ECFDDF0): nbd ECFDE00 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED23B28
  rmd(ECFDE00): nbd ECFDE10 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED26168
  rmd(ECFDE10): nbd ECFDE20 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2D428
  rmd(ECFDE20): nbd ECFDE30 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED30D88
  rmd(ECFDE30): nbd ECFDE40 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED27488
  rmd(ECFDE40): nbd ECFDE50 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED25B08
  rmd(ECFDE50): nbd ECFDE60 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED24188
  rmd(ECFDE60): nbd ECFDE70 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED27AE8
  rmd(ECFDE70): nbd ECFDE80 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED28148
  rmd(ECFDE80): nbd ECFDE90 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED254A8
  rmd(ECFDE90): nbd ECFDEA0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2A128
  rmd(ECFDEA0): nbd ECFDEB0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED267C8
  rmd(ECFDEB0): nbd ECFDEC0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED29AC8
  rmd(ECFDEC0): nbd ECFDED0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2A788
  rmd(ECFDED0): nbd ECFDEE0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2C108
  rmd(ECFDEE0): nbd ECFDEF0 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED2BAA8
  rmd(ECFDEF0): nbd ECFDD00 cmd_sts 80800000 buf_sz 06000000 buf_ptr
ED28E08
Transmit Ring
txr head (37)(0x0ED22E90), txr tail (37)(0x0ED22E90)
  tmd(ED22C40): nbd ED22C50 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAF98
  tmd(ED22C50): nbd ED22C60 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAE58
  tmd(ED22C60): nbd ED22C70 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAE58
  tmd(ED22C70): nbd ED22C80 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3EC58
  tmd(ED22C80): nbd ED22C90 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED08718
  tmd(ED22C90): nbd ED22CA0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3F158
  tmd(ED22CA0): nbd ED22CB0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14018
  tmd(ED22CB0): nbd ED22CC0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED037B8
  tmd(ED22CC0): nbd ED22CD0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAE58
  tmd(ED22CD0): nbd ED22CE0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED22CE0): nbd ED22CF0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED20F38
  tmd(ED22CF0): nbd ED22D00 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3EC58
  tmd(ED22D00): nbd ED22D10 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECD8298
  tmd(ED22D10): nbd ED22D20 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED08718
  tmd(ED22D20): nbd ED22D30 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED4A7D8
  tmd(ED22D30): nbd ED22D40 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14018
  tmd(ED22D40): nbd ED22D50 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C458
  tmd(ED22D50): nbd ED22D60 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED03C18
  tmd(ED22D60): nbd ED22D70 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED20F38
  tmd(ED22D70): nbd ED22D80 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED03C18
  tmd(ED22D80): nbd ED22D90 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C458
  tmd(ED22D90): nbd ED22DA0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECFE218
  tmd(ED22DA0): nbd ED22DB0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED22DB0): nbd ED22DC0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C138
  tmd(ED22DC0): nbd ED22DD0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECD9558
  tmd(ED22DD0): nbd ED22DE0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14BF8
  tmd(ED22DE0): nbd ED22DF0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED20F38
  tmd(ED22DF0): nbd ED22E00 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDB718
  tmd(ED22E00): nbd ED22E10 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14BF8
  tmd(ED22E10): nbd ED22E20 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3EC58
  tmd(ED22E20): nbd ED22E30 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECFE218
  tmd(ED22E30): nbd ED22E40 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAF98
  tmd(ED22E40): nbd ED22E50 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0D358
  tmd(ED22E50): nbd ED22E60 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED03C18
  tmd(ED22E60): nbd ED22E70 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14BF8
  tmd(ED22E70): nbd ED22E80 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED22E80): nbd ED22E90 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED20F38
  tmd(ED22E90): nbd ED22EA0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3F158
  tmd(ED22EA0): nbd ED22EB0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED08718
  tmd(ED22EB0): nbd ED22EC0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDA598
  tmd(ED22EC0): nbd ED22ED0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C458
  tmd(ED22ED0): nbd ED22EE0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED03C18
  tmd(ED22EE0): nbd ED22EF0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14018
  tmd(ED22EF0): nbd ED22F00 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED22F00): nbd ED22F10 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14BF8
  tmd(ED22F10): nbd ED22F20 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED03C18
  tmd(ED22F20): nbd ED22F30 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDA598
  tmd(ED22F30): nbd ED22F40 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3F158
  tmd(ED22F40): nbd ED22F50 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED22F50): nbd ED22F60 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C458
  tmd(ED22F60): nbd ED22F70 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECD9558
  tmd(ED22F70): nbd ED22F80 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDB718
  tmd(ED22F80): nbd ED22F90 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED22F90): nbd ED22FA0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAF98
  tmd(ED22FA0): nbd ED22FB0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED03C18
  tmd(ED22FB0): nbd ED22FC0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED22FC0): nbd ED22FD0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECD8298
  tmd(ED22FD0): nbd ED22FE0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED4A7D8
  tmd(ED22FE0): nbd ED22FF0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3EC58
  tmd(ED22FF0): nbd ED23000 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0D358
  tmd(ED23000): nbd ED23010 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDB718
  tmd(ED23010): nbd ED23020 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED03C18
  tmd(ED23020): nbd ED23030 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAE58
  tmd(ED23030): nbd ED23040 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECD9558
  tmd(ED23040): nbd ED23050 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED23050): nbd ED23060 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED23060): nbd ED23070 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDB718
  tmd(ED23070): nbd ED23080 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C458
  tmd(ED23080): nbd ED23090 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED037B8
  tmd(ED23090): nbd ED230A0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C138
  tmd(ED230A0): nbd ED230B0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED20F38
  tmd(ED230B0): nbd ED230C0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAE58
  tmd(ED230C0): nbd ED230D0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C598
  tmd(ED230D0): nbd ED230E0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECD9C38
  tmd(ED230E0): nbd ED230F0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECFE0D8
  tmd(ED230F0): nbd ED23100 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED4BEF8
  tmd(ED23100): nbd ED23110 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAF98
  tmd(ED23110): nbd ED23120 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECFE218
  tmd(ED23120): nbd ED23130 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED23130): nbd ED23140 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDA598
  tmd(ED23140): nbd ED23150 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3F158
  tmd(ED23150): nbd ED23160 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED119F8
  tmd(ED23160): nbd ED23170 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDB718
  tmd(ED23170): nbd ED23180 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14BF8
  tmd(ED23180): nbd ED23190 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14BF8
  tmd(ED23190): nbd ED231A0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECD8298
  tmd(ED231A0): nbd ED231B0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED037B8
  tmd(ED231B0): nbd ED231C0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED20F38
  tmd(ED231C0): nbd ED231D0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECD9558
  tmd(ED231D0): nbd ED231E0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C458
  tmd(ED231E0): nbd ED231F0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED037B8
  tmd(ED231F0): nbd ED23200 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED4A7D8
  tmd(ED23200): nbd ED23210 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED20F38
  tmd(ED23210): nbd ED23220 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C598
  tmd(ED23220): nbd ED23230 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDA598
  tmd(ED23230): nbd ED23240 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECFE0D8
  tmd(ED23240): nbd ED23250 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED08718
  tmd(ED23250): nbd ED23260 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECFE0D8
  tmd(ED23260): nbd ED23270 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDA598
  tmd(ED23270): nbd ED23280 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14BF8
  tmd(ED23280): nbd ED23290 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0D0D8
  tmd(ED23290): nbd ED232A0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED14E78
  tmd(ED232A0): nbd ED232B0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDB718
  tmd(ED232B0): nbd ED232C0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECFE0D8
  tmd(ED232C0): nbd ED232D0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3F158
  tmd(ED232D0): nbd ED232E0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECFE218
  tmd(ED232E0): nbd ED232F0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED099D8
  tmd(ED232F0): nbd ED23300 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDA598
  tmd(ED23300): nbd ED23310 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3FA18
  tmd(ED23310): nbd ED23320 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAE58
  tmd(ED23320): nbd ED23330 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C138
  tmd(ED23330): nbd ED23340 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C458
  tmd(ED23340): nbd ED23350 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECFE0D8
  tmd(ED23350): nbd ED23360 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3F158
  tmd(ED23360): nbd ED23370 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDB718
  tmd(ED23370): nbd ED23380 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED08718
  tmd(ED23380): nbd ED23390 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C278
  tmd(ED23390): nbd ED233A0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED099D8
  tmd(ED233A0): nbd ED233B0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C458
  tmd(ED233B0): nbd ED233C0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0D0D8
  tmd(ED233C0): nbd ED233D0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDAF98
  tmd(ED233D0): nbd ED233E0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3EED8
  tmd(ED233E0): nbd ED233F0 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED0C278
  tmd(ED233F0): nbd ED23400 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDB718
  tmd(ED23400): nbd ED23410 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ECDB718
  tmd(ED23410): nbd ED23420 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED3F158
  tmd(ED23420): nbd ED23430 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED4BEF8
  tmd(ED23430): nbd ED22C40 cmd_sts 00C30000 byt_cnt ABCDABCD buf_ptr
ED099D8

buffer size 1524

-----Original Message-----
From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: Wednesday, December 07, 2005 7:19 PM
To: IT; Jose Quesada; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] PRI serial "lockout"? what is SBC talking
about

Do a
'sh isdn status'
and a
'sh isdn service'

And send us the results ....

-Mark

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of IT
> Sent: Wednesday, December 07, 2005 6:27 PM
> To: Jose Quesada; cisco-voip at puck.nether.net
> Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
> 
> Hi,
> 
> My PRI on a 2801 just died, and SBC said they were able to bring the
> line back up just by running a test. They said that this "rebooted"
the
> line, and when they first logged in they saw that the lines were
> "locked" out, and this was generally caused by my equipment. What can
I
> do to verify if this is correct? What did they mean? It's a serial
card,
> and here is the show intereface serial 0/2/0:23 output:
> 
> show interfaces serial 0/2/0:23
> Serial0/2/0:23 is up, line protocol is up (spoofing)
>   Hardware is DSX1
>   MTU 1500 bytes, BW 64 Kbit, DLY 20000 usec,
>      reliability 255/255, txload 1/255, rxload 1/255
>   Encapsulation HDLC, loopback not set
>   Last input 00:00:15, output 00:00:15, output hang never
>   Last clearing of "show interface" counters never
>   Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops:
0
>   Queueing strategy: weighted fair
>   Output queue: 0/1000/64/0 (size/max total/threshold/drops)
>      Conversations  0/1/256 (active/max active/max total)
>      Reserved Conversations 0/0 (allocated/max allocated)
>      Available Bandwidth 48 kilobits/sec
>   5 minute input rate 0 bits/sec, 0 packets/sec
>   5 minute output rate 0 bits/sec, 0 packets/sec
>      204146 packets input, 973103 bytes, 0 no buffer
>      Received 0 broadcasts, 0 runts, 2 giants, 0 throttles
>      115437 input errors, 115437 CRC, 51503 frame, 30018 overrun, 0
> ignored, 85696 abort
>      204467 packets output, 978336 bytes, 0 underruns
>      0 output errors, 0 collisions, 12 interface resets
>      0 output buffer failures, 0 output buffers swapped out
>      30 carrier transitions
>   Timeslot(s) Used:24, SCC: 0, Transmitter delay is 0 flags
> 
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are
not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 



From jgrace at digitelusa.net  Wed Dec  7 23:03:04 2005
From: jgrace at digitelusa.net (James Grace)
Date: Thu Dec  8 00:06:06 2005
Subject: [cisco-voip] how to setup agent-initiated recording 
Message-ID: <009001c5fbb4$c78897f0$030a0f0a@digitelcorp.com>

how to setup agent-initiated recording 
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From akoul at dataways.gr  Thu Dec  8 03:11:28 2005
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Dec  8 03:11:44 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
Message-ID: <4397EAB0.3010407@dataways.gr>

Hi all,
i have ordered ccm 4.0(2)a and the whole package of the CDs came to my 
office. I installed the CCM with no problem and the Extended Services CD 
that was in the package (i wanted to use the 4 port of IVR for the Auto 
Attendant). The engine got installed easylly but i could not make it work.
When i looked in the compatibility matrix of ccm, i saw that CRS 3.5.2 
is needed with CCM 4.0(2)a!!
The whole issue with the CRS engine is getting annoying since it is not 
the first time that this is happening.
Is it possible to download it and make it work from tha CCO site or i 
need to order the CDs? Can i upgrade CRS 3.1(1) to 3.5(2) or i have to 
unistall it and reinstall the new one?

Thanks
Anthony Kouloglou

From mderriennic at activeo.fr  Thu Dec  8 04:39:08 2005
From: mderriennic at activeo.fr (Marc DERRIENNIC)
Date: Thu Dec  8 04:39:33 2005
Subject: [cisco-voip] IPCC & DB2 UDB iSeries
Message-ID: <001101c5fbdb$3d5bc080$4e01a8c0@activox.net>

Hi,

Has any of you configure a Cisco IPCC Express 3.5 or 4.0 to use an ODBC
connection with a DB2 UDB iSeries database ?

Many thanks,



Marc Derriennic
Pre Sales Consultant
Tel : +33 1 47 86 58 90
Mobile : +33 6 07 29 65 73
mderriennic at activeo.fr


www.activeo.fr
Limebridge global alliance partner

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From highspeedsnow at gmail.com  Thu Dec  8 08:42:19 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Thu Dec  8 08:43:38 2005
Subject: [cisco-voip] Phone number display CME
In-Reply-To: <000501c5fb9b$c990f470$0f0110ac@corporation2>
Message-ID: <43983887.73b6bd8e.2ba9.2163@mx.gmail.com>

Are you talking about the upper right hand corner?
If so, use configure 'descripton' under the ephone-dn for the first line.
If not, configure the 'label' under the ephone-dn for the first line.

HTH

-Mark Snow

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Bob A. Bowie
> Sent: Wednesday, December 07, 2005 9:05 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Phone number display CME
> 
> Is there a way to remove the phone number from the top line of the IP
Phone
> display
> in Cisco CME?
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From highspeedsnow at gmail.com  Thu Dec  8 09:06:05 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Thu Dec  8 09:07:22 2005
Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B025C37E8@exchange.cimgroup.com>
Message-ID: <43983e16.1187cb9f.2ba9.26fd@mx.gmail.com>

These two look exactly as they should - but then you say the PRI is back up
- so you probably already know that!

-Mark Snow
CCIE Voice #14073

> -----Original Message-----
> From: IT [mailto:it at cimgroup.com]
> Sent: Wednesday, December 07, 2005 11:18 PM
> To: Mark Snow; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] PRI serial "lockout"? what is SBC talking about
> 
> The line is back up now, but here are the results of the two commands.
> Thanks,
> Avidan
> 
> 
> cimgroupsanfran#show isdn status
> Global ISDN Switchtype = primary-4ess
> 
> %Q.931 is backhauled to CCM-MANAGER on DSL 0. Layer 3 output may not
> apply
> 
> ISDN Serial0/2/0:23 interface
>         dsl 0, interface ISDN Switchtype = primary-ni
>         L2 Protocol = Q.921  L3 Protocol(s) = CCM-MANAGER
>     Layer 1 Status:
>         ACTIVE
>     Layer 2 Status:
>         TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
>     Layer 3 Status:
>         0 Active Layer 3 Call(s)
>     Active dsl 0 CCBs = 0
>     The Free Channel Mask:  0x807FFFFF
>     Number of L2 Discards = 0, L2 Session ID = 117
>     Total Allocated ISDN CCBs = 0
> 
> 
> cimgroupsanfran#show isdn service
> PRI Channel Statistics:
> 
> %Q.931 is backhauled to CCM-MANAGER on DSL 0. Layer 3 output may not
> apply
> 
> ISDN Se0/2/0:23, Channel [1-24]
>   Configured Isdn Interface (dsl) 0
>    Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart
> 5=Maint_Pend)
>     Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
>     State   :  0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3
>    Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend
> 9=OOSPend)
>     Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
>     State   :  0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2
> 
> -----Original Message-----
> From: Mark Snow [mailto:highspeedsnow at gmail.com]
> Sent: Wednesday, December 07, 2005 7:19 PM
> To: IT; Jose Quesada; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] PRI serial "lockout"? what is SBC talking
> about
> 
> Do a
> 'sh isdn status'
> and a
> 'sh isdn service'
> 
> And send us the results ....
> 
> -Mark
> 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> > bounces at puck.nether.net] On Behalf Of IT
> > Sent: Wednesday, December 07, 2005 6:27 PM
> > To: Jose Quesada; cisco-voip at puck.nether.net
> > Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
> >
> > Hi,
> >
> > My PRI on a 2801 just died, and SBC said they were able to bring the
> > line back up just by running a test. They said that this "rebooted"
> the
> > line, and when they first logged in they saw that the lines were
> > "locked" out, and this was generally caused by my equipment. What can
> I
> > do to verify if this is correct? What did they mean? It's a serial
> card,
> > and here is the show intereface serial 0/2/0:23 output:
> >
> > show interfaces serial 0/2/0:23
> > Serial0/2/0:23 is up, line protocol is up (spoofing)
> >   Hardware is DSX1
> >   MTU 1500 bytes, BW 64 Kbit, DLY 20000 usec,
> >      reliability 255/255, txload 1/255, rxload 1/255
> >   Encapsulation HDLC, loopback not set
> >   Last input 00:00:15, output 00:00:15, output hang never
> >   Last clearing of "show interface" counters never
> >   Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops:
> 0
> >   Queueing strategy: weighted fair
> >   Output queue: 0/1000/64/0 (size/max total/threshold/drops)
> >      Conversations  0/1/256 (active/max active/max total)
> >      Reserved Conversations 0/0 (allocated/max allocated)
> >      Available Bandwidth 48 kilobits/sec
> >   5 minute input rate 0 bits/sec, 0 packets/sec
> >   5 minute output rate 0 bits/sec, 0 packets/sec
> >      204146 packets input, 973103 bytes, 0 no buffer
> >      Received 0 broadcasts, 0 runts, 2 giants, 0 throttles
> >      115437 input errors, 115437 CRC, 51503 frame, 30018 overrun, 0
> > ignored, 85696 abort
> >      204467 packets output, 978336 bytes, 0 underruns
> >      0 output errors, 0 collisions, 12 interface resets
> >      0 output buffer failures, 0 output buffers swapped out
> >      30 carrier transitions
> >   Timeslot(s) Used:24, SCC: 0, Transmitter delay is 0 flags
> >
> > ******************************************************
> > This email may contain trade secrets and/or other information that is
> > proprietary and/or confidential. Use of this by anyone other than the
> > intended recipient(s) is prohibited and may be unlawful. If you are
> not
> > the intended recipient, delete the e-mail and notify us immediately.
> > ******************************************************
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip


From Michael_Davis at eLoyalty.com  Thu Dec  8 09:13:28 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Thu Dec  8 09:13:44 2005
Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E33D@elhqex01.eloyaltyco.com>

Some things I've seen cause that behavior are 1) voice-port shut down while serial and controller were up - no B-channels available 'though D-channel was happy.  2) Mis-configured NFAS backup D-channel - one side thinks there's a backup D-channel and the other thinks that channel is a B-channel so it appears locked.  
 
Yeah, hard to say in this particular case. 

	-----Original Message----- 
	From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow 
	Sent: Thu 12/8/2005 8:06 AM 
	To: 'IT'; cisco-voip at puck.nether.net 
	Cc: 
	Subject: RE: [cisco-voip] PRI serial "lockout"? what is SBC talking about
	
	

	These two look exactly as they should - but then you say the PRI is back up
	- so you probably already know that!
	
	-Mark Snow
	CCIE Voice #14073
	
	> -----Original Message-----
	> From: IT [mailto:it at cimgroup.com]
	> Sent: Wednesday, December 07, 2005 11:18 PM
	> To: Mark Snow; cisco-voip at puck.nether.net
	> Subject: RE: [cisco-voip] PRI serial "lockout"? what is SBC talking about
	>
	> The line is back up now, but here are the results of the two commands.
	> Thanks,
	> Avidan
	>
	>
	> cimgroupsanfran#show isdn status
	> Global ISDN Switchtype = primary-4ess
	>
	> %Q.931 is backhauled to CCM-MANAGER on DSL 0. Layer 3 output may not
	> apply
	>
	> ISDN Serial0/2/0:23 interface
	>         dsl 0, interface ISDN Switchtype = primary-ni
	>         L2 Protocol = Q.921  L3 Protocol(s) = CCM-MANAGER
	>     Layer 1 Status:
	>         ACTIVE
	>     Layer 2 Status:
	>         TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
	>     Layer 3 Status:
	>         0 Active Layer 3 Call(s)
	>     Active dsl 0 CCBs = 0
	>     The Free Channel Mask:  0x807FFFFF
	>     Number of L2 Discards = 0, L2 Session ID = 117
	>     Total Allocated ISDN CCBs = 0
	>
	>
	> cimgroupsanfran#show isdn service
	> PRI Channel Statistics:
	>
	> %Q.931 is backhauled to CCM-MANAGER on DSL 0. Layer 3 output may not
	> apply
	>
	> ISDN Se0/2/0:23, Channel [1-24]
	>   Configured Isdn Interface (dsl) 0
	>    Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart
	> 5=Maint_Pend)
	>     Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
	>     State   :  0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3
	>    Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend
	> 9=OOSPend)
	>     Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
	>     State   :  0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2
	>
	> -----Original Message-----
	> From: Mark Snow [mailto:highspeedsnow at gmail.com]
	> Sent: Wednesday, December 07, 2005 7:19 PM
	> To: IT; Jose Quesada; cisco-voip at puck.nether.net
	> Subject: RE: [cisco-voip] PRI serial "lockout"? what is SBC talking
	> about
	>
	> Do a
	> 'sh isdn status'
	> and a
	> 'sh isdn service'
	>
	> And send us the results ....
	>
	> -Mark
	>
	> > -----Original Message-----
	> > From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
	> > bounces at puck.nether.net] On Behalf Of IT
	> > Sent: Wednesday, December 07, 2005 6:27 PM
	> > To: Jose Quesada; cisco-voip at puck.nether.net
	> > Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
	> >
	> > Hi,
	> >
	> > My PRI on a 2801 just died, and SBC said they were able to bring the
	> > line back up just by running a test. They said that this "rebooted"
	> the
	> > line, and when they first logged in they saw that the lines were
	> > "locked" out, and this was generally caused by my equipment. What can
	> I
	> > do to verify if this is correct? What did they mean? It's a serial
	> card,
	> > and here is the show intereface serial 0/2/0:23 output:
	> >
	> > show interfaces serial 0/2/0:23
	> > Serial0/2/0:23 is up, line protocol is up (spoofing)
	> >   Hardware is DSX1
	> >   MTU 1500 bytes, BW 64 Kbit, DLY 20000 usec,
	> >      reliability 255/255, txload 1/255, rxload 1/255
	> >   Encapsulation HDLC, loopback not set
	> >   Last input 00:00:15, output 00:00:15, output hang never
	> >   Last clearing of "show interface" counters never
	> >   Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops:
	> 0
	> >   Queueing strategy: weighted fair
	> >   Output queue: 0/1000/64/0 (size/max total/threshold/drops)
	> >      Conversations  0/1/256 (active/max active/max total)
	> >      Reserved Conversations 0/0 (allocated/max allocated)
	> >      Available Bandwidth 48 kilobits/sec
	> >   5 minute input rate 0 bits/sec, 0 packets/sec
	> >   5 minute output rate 0 bits/sec, 0 packets/sec
	> >      204146 packets input, 973103 bytes, 0 no buffer
	> >      Received 0 broadcasts, 0 runts, 2 giants, 0 throttles
	> >      115437 input errors, 115437 CRC, 51503 frame, 30018 overrun, 0
	> > ignored, 85696 abort
	> >      204467 packets output, 978336 bytes, 0 underruns
	> >      0 output errors, 0 collisions, 12 interface resets
	> >      0 output buffer failures, 0 output buffers swapped out
	> >      30 carrier transitions
	> >   Timeslot(s) Used:24, SCC: 0, Transmitter delay is 0 flags
	> >
	> > ******************************************************
	> > This email may contain trade secrets and/or other information that is
	> > proprietary and/or confidential. Use of this by anyone other than the
	> > intended recipient(s) is prohibited and may be unlawful. If you are
	> not
	> > the intended recipient, delete the e-mail and notify us immediately.
	> > ******************************************************
	> >
	> >
	> > _______________________________________________
	> > 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 BillS at tns.its.psu.edu  Thu Dec  8 09:18:36 2005
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Thu Dec  8 09:18:44 2005
Subject: [cisco-voip] how to setup agent-initiated recording 
Message-ID: <07A14D6987A5F647B2415D72EDF63B5602DC86EA@tnsex01.tns.its.psu.edu>

Is this the title of a book you're writing?  A how-to document?  Ehhh??


 


  _____  

From: James Grace [mailto:jgrace at digitelusa.net] 
Sent: Wednesday, December 07, 2005 11:03 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to setup agent-initiated recording 


how to setup agent-initiated recording 

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From bcarter at sentinel.com  Thu Dec  8 10:09:01 2005
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu Dec  8 10:09:08 2005
Subject: [cisco-voip] CallPark Orbit?
Message-ID: <6A0744639B86DD459433108DBF101D26A43DA7@MAIL01.sentinel.com>

We have a customer who would like to have a receptionist park calls, and
other users be able to see a list of parked calls on the display of the
IP Phones.  The users would be able to selectively pickup the parked
calls.  Is this possible?
 


From helder.guarda at gs.com  Thu Dec  8 10:13:56 2005
From: helder.guarda at gs.com (Guarda, Helder)
Date: Thu Dec  8 10:14:15 2005
Subject: [cisco-voip] (no subject)
Message-ID: <C3357D07220AB64DB09C205967E4B3A31F04A79E@gsnmp16es.firmwide.corp.gs.com>

Trying to set up an H323 trunk between call manager and a 3745- anyone ever
do this?  Any help would be appreciated.

Thanks,

helder
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From rares at raisoftware.ro  Thu Dec  8 10:21:44 2005
From: rares at raisoftware.ro (Rares Ispas)
Date: Thu Dec  8 10:23:36 2005
Subject: [cisco-voip] phone to phone paging
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A11E17@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A11E17@VISHNU.wesd.org>
Message-ID: <op.s1g0yiwuyg9d0q@relay.raisoftware.ro>

Scott,

Syn-Apps SA-Announce (http://www.syn-apps.net) has this. Actually, there  
are two features doing this. One is feature called "Hoot-N-Holler" where  
you dial a page group, then enter the number you want to autoanswer on  
speaker.

The second feature is a kind of walkie talkie were you do a page to  
preprogrammed numbers, and the destinations can press a softkey and "talk  
back".

Regards,
Rares Ispas

RAI Software SRL, Bucharest, Romania
site: www.raisoftware.ro
YM: rares_ispas2001
MSN: irares at hotmail.com

On Tue, 06 Dec 2005 22:20:43 +0200, Voll, Scott <Scott.Voll at wesd.org>  
wrote:

> I already have Berbee InformaCast for paging between phones.  So I know
> about paging software. BUT I have an assistant on a PBX phone and the
> admin just got a 7940 phone.  One DID on the 7940.  I know I could setup
> a second extension on the admin's phone and set it to auto pick up on
> speaker phone, but I was wondering if Cisco has made any kind of code
> dialing to do this.
>
>
> What I'm thinking would be nice is to make a translation pattern of
> 3.xxxx would set the call pick up to speaker phone on that extension.
> This way you don't have to burn a second DID / extension.  Does anyone
> else think that would be a benefit?  Are current PBX does this with a
> 7+extension.
>
>
> Scott
>




From Scott.Voll at wesd.org  Thu Dec  8 10:57:03 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec  8 10:56:59 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
Message-ID: <D713462ED535184D830F6F6301753E7E01A11E7E@VISHNU.wesd.org>

Yes and Yes.

You can upgrade from 3.1 to 3.5.  Very easy.  Took me ~ an hour.

You do need to PUT for the CD's.  there is NO download.  What a pain.  I
hate being in your shoes.  Because it takes a couple weeks any time I
talk to cisco.

Maybe you could open a TAC case and they could overnight it if you're
down.

Lesson learned.  Always look at all your IPT before upgrades.  Most
everything is dependent in some way weather a simple TSP upgrade or a
major release.

Read those docs.

Scott



-----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, December 08, 2005 12:11 AM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] Inconsistent CCM version and CRS

Hi all,
i have ordered ccm 4.0(2)a and the whole package of the CDs came to my 
office. I installed the CCM with no problem and the Extended Services CD

that was in the package (i wanted to use the 4 port of IVR for the Auto 
Attendant). The engine got installed easylly but i could not make it
work.
When i looked in the compatibility matrix of ccm, i saw that CRS 3.5.2 
is needed with CCM 4.0(2)a!!
The whole issue with the CRS engine is getting annoying since it is not 
the first time that this is happening.
Is it possible to download it and make it work from tha CCO site or i 
need to order the CDs? Can i upgrade CRS 3.1(1) to 3.5(2) or i have to 
unistall it and reinstall the new one?

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


From Scott.Voll at wesd.org  Thu Dec  8 11:05:36 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec  8 11:05:30 2005
Subject: [cisco-voip] CallPark Orbit?
Message-ID: <D713462ED535184D830F6F6301753E7E01A11E81@VISHNU.wesd.org>

I currently don't know of an app that does this. (not that it doesn't)
But you should be able to make an app like this.

You might check with IPcelrate or cistera.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Thursday, December 08, 2005 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallPark Orbit?

We have a customer who would like to have a receptionist park calls, and
other users be able to see a list of parked calls on the display of the
IP Phones.  The users would be able to selectively pickup the parked
calls.  Is this possible?
 

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


From akoul at dataways.gr  Thu Dec  8 11:35:56 2005
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Dec  8 11:36:11 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A11E7E@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A11E7E@VISHNU.wesd.org>
Message-ID: <439860EC.7050201@dataways.gr>

Thanks Scott,
you have been very helpfull. That is all i needed to hear. I will try to 
purchase the cd's.

Thanks again

Voll, Scott wrote:

>Yes and Yes.
>
>You can upgrade from 3.1 to 3.5.  Very easy.  Took me ~ an hour.
>
>You do need to PUT for the CD's.  there is NO download.  What a pain.  I
>hate being in your shoes.  Because it takes a couple weeks any time I
>talk to cisco.
>
>Maybe you could open a TAC case and they could overnight it if you're
>down.
>
>Lesson learned.  Always look at all your IPT before upgrades.  Most
>everything is dependent in some way weather a simple TSP upgrade or a
>major release.
>
>Read those docs.
>
>Scott
>
>
>
>-----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, December 08, 2005 12:11 AM
>To: cisco-voip at puck-nether.net
>Subject: [cisco-voip] Inconsistent CCM version and CRS
>
>Hi all,
>i have ordered ccm 4.0(2)a and the whole package of the CDs came to my 
>office. I installed the CCM with no problem and the Extended Services CD
>
>that was in the package (i wanted to use the 4 port of IVR for the Auto 
>Attendant). The engine got installed easylly but i could not make it
>work.
>When i looked in the compatibility matrix of ccm, i saw that CRS 3.5.2 
>is needed with CCM 4.0(2)a!!
>The whole issue with the CRS engine is getting annoying since it is not 
>the first time that this is happening.
>Is it possible to download it and make it work from tha CCO site or i 
>need to order the CDs? Can i upgrade CRS 3.1(1) to 3.5(2) or i have to 
>unistall it and reinstall the new one?
>
>Thanks
>Anthony Kouloglou
>_______________________________________________
>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip
>
>  
>

From Scott.Voll at wesd.org  Thu Dec  8 11:41:03 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec  8 11:40:58 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
Message-ID: <D713462ED535184D830F6F6301753E7E01A11E86@VISHNU.wesd.org>

Can't you get it through the Product Upgrade Tool?  Don't you have the
SAU contract?



-----Original Message-----
From: Anthony Kouloglou [mailto:akoul at dataways.gr] 
Sent: Thursday, December 08, 2005 8:36 AM
To: Voll, Scott
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] Inconsistent CCM version and CRS

Thanks Scott,
you have been very helpfull. That is all i needed to hear. I will try to

purchase the cd's.

Thanks again

Voll, Scott wrote:

>Yes and Yes.
>
>You can upgrade from 3.1 to 3.5.  Very easy.  Took me ~ an hour.
>
>You do need to PUT for the CD's.  there is NO download.  What a pain.
I
>hate being in your shoes.  Because it takes a couple weeks any time I
>talk to cisco.
>
>Maybe you could open a TAC case and they could overnight it if you're
>down.
>
>Lesson learned.  Always look at all your IPT before upgrades.  Most
>everything is dependent in some way weather a simple TSP upgrade or a
>major release.
>
>Read those docs.
>
>Scott
>
>
>
>-----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, December 08, 2005 12:11 AM
>To: cisco-voip at puck-nether.net
>Subject: [cisco-voip] Inconsistent CCM version and CRS
>
>Hi all,
>i have ordered ccm 4.0(2)a and the whole package of the CDs came to my 
>office. I installed the CCM with no problem and the Extended Services
CD
>
>that was in the package (i wanted to use the 4 port of IVR for the Auto

>Attendant). The engine got installed easylly but i could not make it
>work.
>When i looked in the compatibility matrix of ccm, i saw that CRS 3.5.2 
>is needed with CCM 4.0(2)a!!
>The whole issue with the CRS engine is getting annoying since it is not

>the first time that this is happening.
>Is it possible to download it and make it work from tha CCO site or i 
>need to order the CDs? Can i upgrade CRS 3.1(1) to 3.5(2) or i have to 
>unistall it and reinstall the new one?
>
>Thanks
>Anthony Kouloglou
>_______________________________________________
>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip
>
>  
>


From akoul at dataways.gr  Thu Dec  8 11:45:27 2005
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Dec  8 11:45:39 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A11E86@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A11E86@VISHNU.wesd.org>
Message-ID: <43986327.8020801@dataways.gr>

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From Scott.Voll at wesd.org  Thu Dec  8 11:57:24 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec  8 11:57:18 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
Message-ID: <D713462ED535184D830F6F6301753E7E01A11E89@VISHNU.wesd.org>

Go to http://tools.cisco.com/gct/Upgrade/jsp/index.jsp and enter your
SAU contract and look for the software your looking for.

 

Scott

 

________________________________

From: Anthony Kouloglou [mailto:akoul at dataways.gr] 
Sent: Thursday, December 08, 2005 8:45 AM
To: Voll, Scott
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] Inconsistent CCM version and CRS

 

Voll,
i am just an engineer and i am not aware of the ways a company purchase
products (especially from Cisco!). I am sure we have the SAU contract
but i have never used the Product Upgrade Tool and i don't know what you
can do with it.

Cheers,
Anthony

Voll, Scott wrote: 

Can't you get it through the Product Upgrade Tool?  Don't you have the
SAU contract?
 
 
 
-----Original Message-----
From: Anthony Kouloglou [mailto:akoul at dataways.gr] 
Sent: Thursday, December 08, 2005 8:36 AM
To: Voll, Scott
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] Inconsistent CCM version and CRS
 
Thanks Scott,
you have been very helpfull. That is all i needed to hear. I will try to
 
purchase the cd's.
 
Thanks again
 
Voll, Scott wrote:
 
  

	Yes and Yes.
	 
	You can upgrade from 3.1 to 3.5.  Very easy.  Took me ~ an hour.
	 
	You do need to PUT for the CD's.  there is NO download.  What a
pain.
	    

I
  

	hate being in your shoes.  Because it takes a couple weeks any
time I
	talk to cisco.
	 
	Maybe you could open a TAC case and they could overnight it if
you're
	down.
	 
	Lesson learned.  Always look at all your IPT before upgrades.
Most
	everything is dependent in some way weather a simple TSP upgrade
or a
	major release.
	 
	Read those docs.
	 
	Scott
	 
	 
	 
	-----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, December 08, 2005 12:11 AM
	To: cisco-voip at puck-nether.net
	Subject: [cisco-voip] Inconsistent CCM version and CRS
	 
	Hi all,
	i have ordered ccm 4.0(2)a and the whole package of the CDs came
to my 
	office. I installed the CCM with no problem and the Extended
Services
	    

CD
  

	that was in the package (i wanted to use the 4 port of IVR for
the Auto
	    

 
  

	Attendant). The engine got installed easylly but i could not
make it
	work.
	When i looked in the compatibility matrix of ccm, i saw that CRS
3.5.2 
	is needed with CCM 4.0(2)a!!
	The whole issue with the CRS engine is getting annoying since it
is not
	    

 
  

	the first time that this is happening.
	Is it possible to download it and make it work from tha CCO site
or i 
	need to order the CDs? Can i upgrade CRS 3.1(1) to 3.5(2) or i
have to 
	unistall it and reinstall the new one?
	 
	Thanks
	Anthony Kouloglou
	_______________________________________________
	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 Dec  8 12:08:16 2005
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu Dec  8 12:08:30 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A11E89@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A11E89@VISHNU.wesd.org>
Message-ID: <43986880.3090206@dataways.gr>

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From jgrace at digitelusa.net  Thu Dec  8 12:49:39 2005
From: jgrace at digitelusa.net (James Grace)
Date: Thu Dec  8 12:50:17 2005
Subject: [cisco-voip] few questions about cm and unity
Message-ID: <200512081249.AA1299644594@mail.digitelusa.net>

whats the default mB size in unity 
and where do i change it 
---



From jgrace at digitelusa.net  Thu Dec  8 12:51:37 2005
From: jgrace at digitelusa.net (James Grace)
Date: Thu Dec  8 12:52:15 2005
Subject: [cisco-voip] unity question 
Message-ID: <200512081251.AA1653997780@mail.digitelusa.net>

how to change the amount of days unity saves messages
---



From jsteinberg at gmail.com  Thu Dec  8 14:16:39 2005
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu Dec  8 14:16:41 2005
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 31
In-Reply-To: <200512080434.jB84X6DB041481@puck.nether.net>
References: <200512080434.jB84X6DB041481@puck.nether.net>
Message-ID: <f1b65f880512081116q4aab4eay@mail.gmail.com>

Your line has alot of CRC input errors.  Do you also have errors in
the current interval when you issue 'show controller T1 0/2/0' ?  Try
clearing the errors and see whether they come back.  If they come
back, you might have a wiring issue, have SBC loop your CSU or
smartjacks to see where the problem is.

On the other hand, since SBC was able to resolve the issue without
your input it was probably a problem on their network and they fixed
it by taking a loop down or something like that.  Telco's always blame
CPE first - i think it is in their handbooks.

Justin

> Message: 5
> Date: Wed, 7 Dec 2005 15:27:08 -0800
> From: "IT" <it at cimgroup.com>
> Subject: [cisco-voip] PRI serial "lockout"? what is SBC talking about
> To: "Jose Quesada" <jquesada at cimgroup.com>,
>        <cisco-voip at puck.nether.net>
> Message-ID:
>        <B487D4C1FE6D04439061018C6EBC109B025C3640 at exchange.cimgroup.com>
> Content-Type: text/plain;       charset="us-ascii"
>
> Hi,
>
> My PRI on a 2801 just died, and SBC said they were able to bring the
> line back up just by running a test. They said that this "rebooted" the
> line, and when they first logged in they saw that the lines were
> "locked" out, and this was generally caused by my equipment. What can I
> do to verify if this is correct? What did they mean? It's a serial card,
> and here is the show intereface serial 0/2/0:23 output:
>
> show interfaces serial 0/2/0:23
> Serial0/2/0:23 is up, line protocol is up (spoofing)
>  Hardware is DSX1
>  MTU 1500 bytes, BW 64 Kbit, DLY 20000 usec,
>     reliability 255/255, txload 1/255, rxload 1/255
>  Encapsulation HDLC, loopback not set
>  Last input 00:00:15, output 00:00:15, output hang never
>  Last clearing of "show interface" counters never
>  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0
>  Queueing strategy: weighted fair
>  Output queue: 0/1000/64/0 (size/max total/threshold/drops)
>     Conversations  0/1/256 (active/max active/max total)
>     Reserved Conversations 0/0 (allocated/max allocated)
>     Available Bandwidth 48 kilobits/sec
>  5 minute input rate 0 bits/sec, 0 packets/sec
>  5 minute output rate 0 bits/sec, 0 packets/sec
>     204146 packets input, 973103 bytes, 0 no buffer
>     Received 0 broadcasts, 0 runts, 2 giants, 0 throttles
>     115437 input errors, 115437 CRC, 51503 frame, 30018 overrun, 0
> ignored, 85696 abort
>     204467 packets output, 978336 bytes, 0 underruns
>     0 output errors, 0 collisions, 12 interface resets
>     0 output buffer failures, 0 output buffers swapped out
>     30 carrier transitions
>  Timeslot(s) Used:24, SCC: 0, Transmitter delay is 0 flags
>
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
>
>
>
>


From jonvoip at gmail.com  Thu Dec  8 16:19:33 2005
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Dec  8 16:19:37 2005
Subject: [cisco-voip] Transfer from shared line?
Message-ID: <5d093f9a0512081319k67d3a178oa67e9e83e2d85ee5@mail.gmail.com>

Got a client, has a shared line on 9 phones, says the transfer soft-key is
greyed out. Is there an issue with transferring from a shared line? He can
transfer fine from the other lines on these phones.



Jonathan
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From Scott.Voll at wesd.org  Thu Dec  8 16:50:06 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec  8 16:50:03 2005
Subject: [cisco-voip] Transfer from shared line?
Message-ID: <D713462ED535184D830F6F6301753E7E01A11EAA@VISHNU.wesd.org>

How many lines are assigned to the phone?  If you don't have enough
lines I've seen this.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Thursday, December 08, 2005 1:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Transfer from shared line?

 

Got a client, has a shared line on 9 phones, says the transfer soft-key
is greyed out. Is there an issue with transferring from a shared line?
He can transfer fine from the other lines on these phones.



Jonathan

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From lelio at uoguelph.ca  Thu Dec  8 16:57:24 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Dec  8 16:57:31 2005
Subject: [cisco-voip] Transfer from shared line?
References: <D713462ED535184D830F6F6301753E7E01A11EAA@VISHNU.wesd.org>
Message-ID: <002601c5fc42$608ea790$30196883@cfs.uoguelph.ca>

We have this happening at our residence desks. Pre-v4 has no control over the number of calls assigned to each line, so if you are on a call and you get another call through call waiting and try to transfer that second call, you will not be able to.

In version 4 and above I believe, you are able to configure (as Scott suggests), the number of max incoming calls (i.e. call waiting and fwd busy trigger) and the max outgoing calls (for transfer etc). these are per line settings.

here's a snapshot and help file snippet. 


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


      Multiple Call / Call Waiting Settings - Changes affect only this device 
      Maximum Number of Calls*  (1 - 200) 
      Busy Trigger*  (<= Max. Calls) 


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


      Multiple Call/Call Waiting Settings 
      Maximum Number of Calls 
     You can configure up to 200 calls for a line on a device, with the limiting factor being the total number of calls that are configured on the device. As you configure the number of calls for one line, the calls that are available for another line decrease. 

      The default specifies 4. If the phone does not allow multiple calls for each line, the default specifies 2. 

      For CTI route points, you can configure up to 10,000 calls for each port. The default specifies 5000 calls. Use this field in conjunction with the Busy Trigger field. 

      Note : Although the default specifies 5000 calls for maximum number of active calls that can be configured on a CTI route point, Cisco recommends that you set the maximum number of calls to no more than 200 per route point. This will prevent system performance degradation. If the CTI application needs more than 200 calls, Cisco recommends that you configure multiple CTI route points. 

      Tip : To review how this setting works for devices with shared line appearances, refer to Shared Line Appearance in the Cisco CallManager System Guide. 
     
      Busy Trigger 
     This setting, which works in conjunction with Maximum Number of Calls and Call Forward Busy, determines the maximum number of calls to be presented at the line. If maximum number of calls is set for 50 and the busy trigger is set to 40, then incoming call 41 gets rejected with a busy cause (and will get forwarded if Call Forward Busy is set). If this line is shared, all the lines must be busy before incoming calls get rejected. 

      Use this field in conjunction with Maximum Number of Calls for CTI route points. The default specifies 4500 calls. 

      Tip : To review how this setting works for devices with shared line appearances, refer to Shared Line Appearance in the Cisco CallManager System Guide. 
     


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Voll, Scott 
  To: Jonathan Charles ; cisco-voip at puck.nether.net 
  Sent: Thursday, December 08, 2005 4:50 PM
  Subject: RE: [cisco-voip] Transfer from shared line?


  How many lines are assigned to the phone?  If you don't have enough lines I've seen this.

   

  Scott

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
  Sent: Thursday, December 08, 2005 1:20 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Transfer from shared line?

   

  Got a client, has a shared line on 9 phones, says the transfer soft-key is greyed out. Is there an issue with transferring from a shared line? He can transfer fine from the other lines on these phones.



  Jonathan



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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From BillS at tns.its.psu.edu  Thu Dec  8 17:26:24 2005
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Thu Dec  8 17:26:28 2005
Subject: [cisco-voip] Cannot use Unity media master from Windows 2003
	terminal server c lient
Message-ID: <07A14D6987A5F647B2415D72EDF63B5602DC877C@tnsex01.tns.its.psu.edu>

The title explains most of it.  We have a Windows 2003 Terminal Server being
used as a "gateway" where people log in and run the Unity admin site (and
other VoIP management tools).  The MediaMaster control loads and appears to
work correctly, but it does not.  If I set it to phone, the phone rings, I
pick up, and nothing happens.  At first it seemed like an access list issue
but I can log in directly to Unity and use the MediaMaster in this way and
also from an XP machine on the Unity subnet.

Any thoughts... I've been round and round with a couple fellows from TAC and
not gotten anywhere.  Has anyone seen and corrected an issue like this
before?  
 

---
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/

From lelio at uoguelph.ca  Thu Dec  8 17:39:54 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Dec  8 17:39:59 2005
Subject: [cisco-voip] Cannot use Unity media master from Windows
	2003terminal server c lient
References: <07A14D6987A5F647B2415D72EDF63B5602DC877C@tnsex01.tns.its.psu.edu>
Message-ID: <000c01c5fc48$4f5f7cf0$30196883@cfs.uoguelph.ca>

We had something similar. Well two things really. The first was a small red X where the media master bar was, and this required logging in as administrator and loading the page again and downloading the app.

The second thing was exactly as you describe, and for the life of me I can't remember what we did to fix it. How long do you wait before you hang up? I remember waiting a while, like a minute or two and eventually, the audio stream played.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Simon, Bill 
  To: 'cisco-voip at puck.nether.net' 
  Sent: Thursday, December 08, 2005 5:26 PM
  Subject: [cisco-voip] Cannot use Unity media master from Windows 2003terminal server c lient


  The title explains most of it.  We have a Windows 2003 Terminal Server being
  used as a "gateway" where people log in and run the Unity admin site (and
  other VoIP management tools).  The MediaMaster control loads and appears to
  work correctly, but it does not.  If I set it to phone, the phone rings, I
  pick up, and nothing happens.  At first it seemed like an access list issue
  but I can log in directly to Unity and use the MediaMaster in this way and
  also from an XP machine on the Unity subnet.

  Any thoughts... I've been round and round with a couple fellows from TAC and
  not gotten anywhere.  Has anyone seen and corrected an issue like this
  before?  
   

  ---
  Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
  http://tns.its.psu.edu/
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From BillS at tns.its.psu.edu  Thu Dec  8 20:13:35 2005
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Thu Dec  8 20:13:39 2005
Subject: [cisco-voip] Cannot use Unity media master from Windows 2003t
	erminal server c lient
Message-ID: <07A14D6987A5F647B2415D72EDF63B56F789CC@tnsex01.tns.its.psu.edu>

I'll try again but I don't think it ever plays.  Basically I wait until
Unity hangs up on me.  No audio ever plays.

-----Original Message-----
From: Lelio Fulgenzi
To: Simon, Bill; cisco-voip at puck.nether.net
Sent: 12/8/2005 5:39 PM
Subject: Re: [cisco-voip] Cannot use Unity media master from Windows
2003terminal server c lient

We had something similar. Well two things really. The first was a small
red X where the media master bar was, and this required logging in as
administrator and loading the page again and downloading the app.
 
The second thing was exactly as you describe, and for the life of me I
can't remember what we did to fix it. How long do you wait before you
hang up? I remember waiting a while, like a minute or two and
eventually, the audio stream played.
 
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

----- Original Message ----- 
From: Simon,  <mailto:BillS at tns.its.psu.edu> Bill 
To: 'cisco-voip at puck.nether.net' <mailto:'cisco-voip at puck.nether.net'>  
Sent: Thursday, December 08, 2005 5:26 PM
Subject: [cisco-voip] Cannot use Unity media master from Windows
2003terminal server c lient

The title explains most of it.  We have a Windows 2003 Terminal Server
being
used as a "gateway" where people log in and run the Unity admin site
(and
other VoIP management tools).  The MediaMaster control loads and appears
to
work correctly, but it does not.  If I set it to phone, the phone rings,
I
pick up, and nothing happens.  At first it seemed like an access list
issue
but I can log in directly to Unity and use the MediaMaster in this way
and
also from an XP machine on the Unity subnet.

Any thoughts... I've been round and round with a couple fellows from TAC
and
not gotten anywhere.  Has anyone seen and corrected an issue like this
before?  
 

---
Bill Simon - bills at tns.its.psu.edu <mailto:bills at tns.its.psu.edu>  -
(814) 865-2270
http://tns.its.psu.edu/ <http://tns.its.psu.edu/> 
_______________________________________________
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> 



From rares at raisoftware.ro  Fri Dec  9 05:49:51 2005
From: rares at raisoftware.ro (Rares Ispas)
Date: Fri Dec  9 05:50:31 2005
Subject: [cisco-voip] CallPark Orbit?
In-Reply-To: <6A0744639B86DD459433108DBF101D26A43DA7@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26A43DA7@MAIL01.sentinel.com>
Message-ID: <op.s1ii1dg8yg9d0q@relay.raisoftware.ro>

Bill,

You might want to try Syn-Apps Park-N-Page (http://www.syn-apps.net). It  
is a bit smarter than that, it announces using automated voice that a call  
has been parked and for what number it was originally intended. And it  
also rings the original destination.

http://www.syn-apps.net/Park-N-Page.aspx

-- 

Regards,
Rares Ispas

RAI Software SRL, Bucharest, Romania
cell: +40 722 475 870
site: www.raisoftware.ro
YM: rares_ispas2001
MSN: irares at hotmail.com

From leozicovich at yahoo.co.uk  Fri Dec  9 06:27:36 2005
From: leozicovich at yahoo.co.uk (Leo Zicovich)
Date: Fri Dec  9 06:27:44 2005
Subject: [cisco-voip] RTP packets in a VoIP network.
Message-ID: <20051209112736.99384.qmail@web26301.mail.ukl.yahoo.com>

Hello all,

I am working in a VoIP network and I have a question
using tethereal to get the RTP packets to saved them
in a file to be played later. 

I use cisco AS5400 with an IVR (just to do a transfer
in a  Bridge transfer mode. The IVR adds the callee to
the connection and remains in the connection for the
duration of the transferred call):  GW1/2 --> IVR -->
GW1/2, and I am trying to use tethereal command line
on the IVR, but I get only the UDP/RTP packet?s for
the first connection (GW1 --> IVR), as soon as the
transfer is done (IVR --> GW1/2) and the communication
established, nothing is captured. was it enough clear?
I hope so.

Anyone know why is it happening? And how can I get all
the communication UDP/RTP packet?s?

Many thank in advance

regards,

Leo



		
___________________________________________________________ 
How much free photo storage do you get? Store your holiday 
snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com

From ciscozest at gmail.com  Fri Dec  9 07:42:39 2005
From: ciscozest at gmail.com (Robin H)
Date: Fri Dec  9 07:42:42 2005
Subject: [cisco-voip] Automatic Sql DB copied from Subscriber to Publisher
Message-ID: <f99cc3f60512090442p591c69b3sa2d6b3bfd9257832@mail.gmail.com>

I need to setup two IVR servers and I plan to store a special-time database
(a new table) on CallManager Servers (using Sql). Normally I only create the
script to query DB from single server ie: subscriber. But if the subscriber
server down, the ivr won't able to give correct prompts. Is there a way I
can make that sql db/table to be copied over to another server so that I
still have redundancy. Thank you.
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From balbler at nts.at  Fri Dec  9 08:05:23 2005
From: balbler at nts.at (Bernhard Albler)
Date: Fri Dec  9 08:04:40 2005
Subject: [cisco-voip] Experiences with Walker / Plantronics Push to Talk
	Handsets
Message-ID: <5A8C69748176434F92F1EECD7D21F72C353A30@srvgraz03.nts.at>

Hi,

 

We have a prospective Customer who needs functionality like provided by
these handsets: http://www.headsetzone.com/ws2620.html

Basically they only transmit Audio when a button is pressed. The
customer was told by someone that the Walker / Plantronics Handset
produces a interference noise when the button is pressed. Has anyone
experience with these? How well to they work, are there any
alternatives?

 

Many thanks in advance

Best regards

bernhard

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From lelio at uoguelph.ca  Fri Dec  9 08:55:32 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Dec  9 08:55:37 2005
Subject: [cisco-voip] Automatic Sql DB copied from Subscriber to Publisher
References: <f99cc3f60512090442p591c69b3sa2d6b3bfd9257832@mail.gmail.com>
Message-ID: <001201c5fcc8$38c328e0$30196883@cfs.uoguelph.ca>

We used to do something like this and take advantage of the database applications on the CallManager servers. However, since they introduced the concept of the shared key between the servers, we were not able to get this working again. No amount of troubleshooting with our local SQL expert was able to help. Eventually we had to drop the idea of using the push/pull subscription and use export/import scripts to replicate the table on the other server. If your database is not that large and doesn't see that many changes, you could probably get away with running the script every hour.

That being said, the TAC does not like you adding your own database on CallManager servers and may infact as you to delete it if they see any problems arise. So were were told. 

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Robin H 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, December 09, 2005 7:42 AM
  Subject: [cisco-voip] Automatic Sql DB copied from Subscriber to Publisher


  I need to setup two IVR servers and I plan to store a special-time database (a new table) on CallManager Servers (using Sql). Normally I only create the script to query DB from single server ie: subscriber. But if the subscriber server down, the ivr won't able to give correct prompts. Is there a way I can make that sql db/table to be copied over to another server so that I still have redundancy. 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 Nick.Kassel at Charles-Stanley.co.uk  Fri Dec  9 09:30:52 2005
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri Dec  9 09:31:03 2005
Subject: [cisco-voip] FXS port problems
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC1839AF@CSEXCH01.charles-stanley.co.uk>

I am configuring a 4 port fxs card for analogue lines to be used in a
branch office, I have incoming working fine but the outgoing calls, it
will ring once on the destination phone and then cut out. 

 

I have the following dial peer and then a route pattern of 81250 on the
call manager pointing to the gateway.

 

dial-peer voice 1250 pots

 destination-pattern 81250

 progress_ind alert enable 8

 fax rate disable

 port 0/1/0

 

Any help would be appreciated.

 

 

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

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets 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 andy at dignans.com  Fri Dec  9 09:57:26 2005
From: andy at dignans.com (Andrew Dignan)
Date: Fri Dec  9 09:57:30 2005
Subject: [cisco-voip] UK E1/PRI - Caller ID
Message-ID: <2977.64.81.226.84.1134140246.squirrel@www.dignans.com>

Anyone on the list located in the UK?  I am having an issue with incoming
caller id on an E1 PRI.  The caller id I am receiving does not have the
"0" prefixed for national calls and the international calls do not have
the "00" prefixed.  They come in for all calls with ISDN/National as the
plan/type so I don't have anything I can use in a voice translation rule
to make a decision on whether or not to prefix a 0 or 00.

Andy

From Nick.Kassel at Charles-Stanley.co.uk  Fri Dec  9 10:09:21 2005
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Fri Dec  9 10:09:36 2005
Subject: [cisco-voip] UK E1/PRI - Caller ID
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC1839B4@CSEXCH01.charles-stanley.co.uk>

Try adding these this is what we have on our gateways in the UK.

translation-rule 1
 Rule 0 ^0. 90
 Rule 1 ^1. 901
 Rule 2 ^2. 902
 Rule 7 ^7. 907

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
Sent: 09 December 2005 14:57
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] UK E1/PRI - Caller ID

Anyone on the list located in the UK?  I am having an issue with
incoming
caller id on an E1 PRI.  The caller id I am receiving does not have the
"0" prefixed for national calls and the international calls do not have
the "00" prefixed.  They come in for all calls with ISDN/National as the
plan/type so I don't have anything I can use in a voice translation rule
to make a decision on whether or not to prefix a 0 or 00.

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

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets 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.



From ciscozest at gmail.com  Fri Dec  9 10:37:15 2005
From: ciscozest at gmail.com (Ciscozest)
Date: Fri Dec  9 10:36:51 2005
Subject: [cisco-voip] Automatic Sql DB copied from Subscriber to Publisher
References: <f99cc3f60512090442p591c69b3sa2d6b3bfd9257832@mail.gmail.com>
	<001201c5fcc8$38c328e0$30196883@cfs.uoguelph.ca>
Message-ID: <00b701c5fcd6$70b0c100$6301a8c0@windfall99>

Thanks for your sharing Lelio.
So sad to hear no redundancy

regards,
Robin
  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: Robin H ; cisco-voip at puck.nether.net 
  Sent: Friday, December 09, 2005 9:55 PM
  Subject: Re: [cisco-voip] Automatic Sql DB copied from Subscriber to Publisher


  We used to do something like this and take advantage of the database applications on the CallManager servers. However, since they introduced the concept of the shared key between the servers, we were not able to get this working again. No amount of troubleshooting with our local SQL expert was able to help. Eventually we had to drop the idea of using the push/pull subscription and use export/import scripts to replicate the table on the other server. If your database is not that large and doesn't see that many changes, you could probably get away with running the script every hour.

  That being said, the TAC does not like you adding your own database on CallManager servers and may infact as you to delete it if they see any problems arise. So were were told. 

  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
  IP phone on a long haul 10base2 connection"                                           LFJ
    ----- Original Message ----- 
    From: Robin H 
    To: cisco-voip at puck.nether.net 
    Sent: Friday, December 09, 2005 7:42 AM
    Subject: [cisco-voip] Automatic Sql DB copied from Subscriber to Publisher


    I need to setup two IVR servers and I plan to store a special-time database (a new table) on CallManager Servers (using Sql). Normally I only create the script to query DB from single server ie: subscriber. But if the subscriber server down, the ivr won't able to give correct prompts. Is there a way I can make that sql db/table to be copied over to another server so that I still have redundancy. 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 Scott.Voll at wesd.org  Fri Dec  9 11:13:16 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Dec  9 11:13:11 2005
Subject: [cisco-voip] RTP packets in a VoIP network.
Message-ID: <D713462ED535184D830F6F6301753E7E01A11EDD@VISHNU.wesd.org>

If I'm understanding you correctly the call setup is setting up the first leg (which you can get) and then the second leg brakes it.  The RTP stream starts from GW1 to IVR then after the transfer the RTP stream is from GW1 to GW2.

I'm not sure how your are getting your Capture but look for the RTP stream from GW1 to GW2.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Leo Zicovich
Sent: Friday, December 09, 2005 3:28 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] RTP packets in a VoIP network.

Hello all,

I am working in a VoIP network and I have a question
using tethereal to get the RTP packets to saved them
in a file to be played later. 

I use cisco AS5400 with an IVR (just to do a transfer
in a  Bridge transfer mode. The IVR adds the callee to
the connection and remains in the connection for the
duration of the transferred call):  GW1/2 --> IVR -->
GW1/2, and I am trying to use tethereal command line
on the IVR, but I get only the UDP/RTP packet's for
the first connection (GW1 --> IVR), as soon as the
transfer is done (IVR --> GW1/2) and the communication
established, nothing is captured. was it enough clear?
I hope so.

Anyone know why is it happening? And how can I get all
the communication UDP/RTP packet?s?

Many thank in advance

regards,

Leo



		
___________________________________________________________ 
How much free photo storage do you get? Store your holiday 
snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.com
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jburton at NETechCorp.com  Fri Dec  9 11:18:39 2005
From: jburton at NETechCorp.com (Jason Burton)
Date: Fri Dec  9 11:18:59 2005
Subject: [cisco-voip] Virus Scanners
Message-ID: <FDC137145729AD438CDCA0F9FC6F723633854B@mail.netechcorp.com>

Does someone have the link for 'approved' virus scanners that can run on
CallManager?

 

 

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From andy at dignans.com  Fri Dec  9 11:26:57 2005
From: andy at dignans.com (Andrew Dignan)
Date: Fri Dec  9 11:27:02 2005
Subject: [cisco-voip] UK E1/PRI - Caller ID
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC1839B4@CSEXCH01.charles-stanley.co.uk
	>
References: <253FAE7A06B11B49A3A29770B2B8BAFC1839B4@CSEXCH01.charles-stanley.co.uk>
Message-ID: <3572.64.81.226.84.1134145617.squirrel@www.dignans.com>

This would work I believe for National calls but for International we are
not receiving "00".  If someone from the US calls your site what does your
caller-id look like?

Andy

> Try adding these this is what we have on our gateways in the UK.
>
> translation-rule 1
>  Rule 0 ^0. 90
>  Rule 1 ^1. 901
>  Rule 2 ^2. 902
>  Rule 7 ^7. 907
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
> Sent: 09 December 2005 14:57
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] UK E1/PRI - Caller ID
>
> Anyone on the list located in the UK?  I am having an issue with
> incoming
> caller id on an E1 PRI.  The caller id I am receiving does not have the
> "0" prefixed for national calls and the international calls do not have
> the "00" prefixed.  They come in for all calls with ISDN/National as the
> plan/type so I don't have anything I can use in a voice translation rule
> to make a decision on whether or not to prefix a 0 or 00.
>
> Andy
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> ***********************************************************************************************
>
> The information contained in this e-mail is strictly confidential, some or
> all
> of which may be legally privileged. It is for the intended recipient
> only.
> Access to this e-mail by any other person is prohibited. If you are not
> the
> intended recipient, 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 and may be unlawful. Please contact the sender
> immediately
> should this e-mail have been incorrectly addressed or transmitted.
>
> 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 accept that
> there
> may be a delay in processing the instructions received from e-mails after
> Charles Stanley has received them. You are advised that urgent, time
> sensitive
> and confidential communications should not be sent by e-mail.
>
> You acknowledge that e-mails are not secure and you accept the risk of
> malfunction, viruses, unauthorised interference, mis-delivery or delay.
> ************************************************************************************************
>
>
> 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 Tozer Wingate are divisions of Charles
> Stanley & Co. Ltd
>
> Authorised and Regulated by the Financial Services Authority, Member of
> the
> London Stock Exchange, International Securities Markets 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.
>
>


From lelio at uoguelph.ca  Fri Dec  9 11:28:56 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Dec  9 11:29:03 2005
Subject: [cisco-voip] Virus Scanners
References: <FDC137145729AD438CDCA0F9FC6F723633854B@mail.netechcorp.com>
Message-ID: <004801c5fcdd$a68c6d40$30196883@cfs.uoguelph.ca>

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_bulletin0900aecd800f8572.html

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Jason Burton 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, December 09, 2005 11:18 AM
  Subject: [cisco-voip] Virus Scanners


  Does someone have the link for 'approved' virus scanners that can run on CallManager?

   

   



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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From ciscozest at gmail.com  Fri Dec  9 11:36:22 2005
From: ciscozest at gmail.com (Ciscozest)
Date: Fri Dec  9 11:35:57 2005
Subject: [cisco-voip] FXOLS_GUARD_OUT message
References: <253FAE7A06B11B49A3A29770B2B8BAFC1839B4@CSEXCH01.charles-stanley.co.uk>
Message-ID: <007f01c5fcde$b2e8aee0$6301a8c0@windfall99>

I have Unity with call handler (CH) feature to transfer call to external number. When I make a call from an IP Phone that is registered to CallManager, the Unity CallHandler can perform call transfer smoothly. But if i make call from CME site (voip) to central site where unity and CCM reside, the transfer always failed and I notice my FXO interface status went to connect state then went to FXOLS_GUARD_OUT status before it went on-hook again and the caller was stuck in a kind of music on hold. Anyone has experience this before? I use VIC-FXO (for PSTN link) and also have used command "supervision disconnect signal" under the voice-port. thanks

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From jburton at NETechCorp.com  Fri Dec  9 11:50:18 2005
From: jburton at NETechCorp.com (Jason Burton)
Date: Fri Dec  9 11:50:38 2005
Subject: [cisco-voip] Virus Scanners
Message-ID: <FDC137145729AD438CDCA0F9FC6F7236338551@mail.netechcorp.com>

Thanks!

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Friday, December 09, 2005 11:29 AM
To: Jason Burton; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Virus Scanners

 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_bulletin0900ae
cd800f8572.html

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Jason Burton <mailto:jburton at NETechCorp.com>  

	To: cisco-voip at puck.nether.net 

	Sent: Friday, December 09, 2005 11:18 AM

	Subject: [cisco-voip] Virus Scanners

	 

	Does someone have the link for 'approved' virus scanners that
can run on CallManager?

	 

	 

	
________________________________


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

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From andy at dignans.com  Fri Dec  9 12:19:50 2005
From: andy at dignans.com (Andrew Dignan)
Date: Fri Dec  9 12:19:53 2005
Subject: [cisco-voip] UK E1/PRI - Caller ID
In-Reply-To: <3572.64.81.226.84.1134145617.squirrel@www.dignans.com>
References: <253FAE7A06B11B49A3A29770B2B8BAFC1839B4@CSEXCH01.charles-stanley.co.uk>
	<3572.64.81.226.84.1134145617.squirrel@www.dignans.com>
Message-ID: <3705.64.81.226.84.1134148790.squirrel@www.dignans.com>

I ended up going with:

---------------------------
voice translation-rule 1
 rule 1 /\(^[127].........$\)/ /90\1/
 rule 2 /\(^[1-9]*\)/ /900\1/
!
!
voice translation-profile Prefix_9
 translate calling 1
---------------------------

Basically any call that comes in beginning with 1, 2, or 7 and has 10
digits, thats a local/national call.  If it is == to 10 digit or > and got
past rule 1 I will prefix 00.  So if a call came from Africa (Cisco TAC)
for example which would be "3227045555" it wouldn't match rule 1 but would
match rule 2 and prefix 900.  From what I can gather no UK > UK calls will
ever come in with anything other than 1,2, or 7 so as long as that holds
true I think I will be ok.

> This would work I believe for National calls but for International we are
> not receiving "00".  If someone from the US calls your site what does your
> caller-id look like?
>
> Andy
>
>> Try adding these this is what we have on our gateways in the UK.
>>
>> translation-rule 1
>>  Rule 0 ^0. 90
>>  Rule 1 ^1. 901
>>  Rule 2 ^2. 902
>>  Rule 7 ^7. 907
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
>> Sent: 09 December 2005 14:57
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] UK E1/PRI - Caller ID
>>
>> Anyone on the list located in the UK?  I am having an issue with
>> incoming
>> caller id on an E1 PRI.  The caller id I am receiving does not have the
>> "0" prefixed for national calls and the international calls do not have
>> the "00" prefixed.  They come in for all calls with ISDN/National as the
>> plan/type so I don't have anything I can use in a voice translation rule
>> to make a decision on whether or not to prefix a 0 or 00.
>>
>> Andy
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> ***********************************************************************************************
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>
>


From leozicovich at yahoo.co.uk  Fri Dec  9 12:28:29 2005
From: leozicovich at yahoo.co.uk (Leo Zicovich)
Date: Fri Dec  9 12:28:37 2005
Subject: [cisco-voip] RTP packets in a VoIP network.
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A11EDD@VISHNU.wesd.org>
Message-ID: <20051209172829.9449.qmail@web26310.mail.ukl.yahoo.com>

Hi Scott,

Basically, the scenario is: PSTN ---> GW1---> IVR
(transfer in a Bridge mode) --->GW1 ---> PSTN. Using
H323.

And what I need is to get the RTP packets for a single
conversation in order to be saved in a file to be
played later. So, what I am trying to do is use a
protocol analyzer on the IVR to get that.
Unfortunately, I just get the RTP traffic for  PSTN
---> GW1---> IVR, after the transfer all the RTP
traffic is intra GW1.
Is there any way to configure the cisco in the way to
remain the IVR in the connection for the duration of
the call, so all the RTP traffic? Do you know another
way to do that?

Many thank,

Regards,
Leo


--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> If I'm understanding you correctly the call setup is
> setting up the first leg (which you can get) and
> then the second leg brakes it.  The RTP stream
> starts from GW1 to IVR then after the transfer the
> RTP stream is from GW1 to GW2.
> 
> I'm not sure how your are getting your Capture but
> look for the RTP stream from GW1 to GW2.
> 
> Scott
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Leo Zicovich
> Sent: Friday, December 09, 2005 3:28 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] RTP packets in a VoIP network.
> 
> Hello all,
> 
> I am working in a VoIP network and I have a question
> using tethereal to get the RTP packets to saved them
> in a file to be played later. 
> 
> I use cisco AS5400 with an IVR (just to do a
> transfer
> in a  Bridge transfer mode. The IVR adds the callee
> to
> the connection and remains in the connection for the
> duration of the transferred call):  GW1/2 --> IVR
> -->
> GW1/2, and I am trying to use tethereal command line
> on the IVR, but I get only the UDP/RTP packet's for
> the first connection (GW1 --> IVR), as soon as the
> transfer is done (IVR --> GW1/2) and the
> communication
> established, nothing is captured. was it enough
> clear?
> I hope so.
> 
> Anyone know why is it happening? And how can I get
> all
> the communication UDP/RTP packet?s?
> 
> Many thank in advance
> 
> regards,
> 
> Leo
> 
> 
> 
> 		
>
___________________________________________________________
> 
> How much free photo storage do you get? Store your
> holiday 
> snaps for FREE with Yahoo! Photos
> http://uk.photos.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



		
___________________________________________________________ 
To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com

From Scott.Voll at wesd.org  Fri Dec  9 12:32:54 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Dec  9 12:32:47 2005
Subject: [cisco-voip] RTP packets in a VoIP network.
Message-ID: <D713462ED535184D830F6F6301753E7E01A11EE6@VISHNU.wesd.org>

Any way to use the protocol analyzer on GW1?

Span the port on GW1.  this would give you the whole RTP stream.  It will have to different destination streams but you could piece it back together.

Scott

-----Original Message-----
From: Leo Zicovich [mailto:leozicovich at yahoo.co.uk] 
Sent: Friday, December 09, 2005 9:28 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] RTP packets in a VoIP network.

Hi Scott,

Basically, the scenario is: PSTN ---> GW1---> IVR
(transfer in a Bridge mode) --->GW1 ---> PSTN. Using
H323.

And what I need is to get the RTP packets for a single
conversation in order to be saved in a file to be
played later. So, what I am trying to do is use a
protocol analyzer on the IVR to get that.
Unfortunately, I just get the RTP traffic for  PSTN
---> GW1---> IVR, after the transfer all the RTP
traffic is intra GW1.
Is there any way to configure the cisco in the way to
remain the IVR in the connection for the duration of
the call, so all the RTP traffic? Do you know another
way to do that?

Many thank,

Regards,
Leo


--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> If I'm understanding you correctly the call setup is
> setting up the first leg (which you can get) and
> then the second leg brakes it.  The RTP stream
> starts from GW1 to IVR then after the transfer the
> RTP stream is from GW1 to GW2.
> 
> I'm not sure how your are getting your Capture but
> look for the RTP stream from GW1 to GW2.
> 
> Scott
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Leo Zicovich
> Sent: Friday, December 09, 2005 3:28 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] RTP packets in a VoIP network.
> 
> Hello all,
> 
> I am working in a VoIP network and I have a question
> using tethereal to get the RTP packets to saved them
> in a file to be played later. 
> 
> I use cisco AS5400 with an IVR (just to do a
> transfer
> in a  Bridge transfer mode. The IVR adds the callee
> to
> the connection and remains in the connection for the
> duration of the transferred call):  GW1/2 --> IVR
> -->
> GW1/2, and I am trying to use tethereal command line
> on the IVR, but I get only the UDP/RTP packet's for
> the first connection (GW1 --> IVR), as soon as the
> transfer is done (IVR --> GW1/2) and the
> communication
> established, nothing is captured. was it enough
> clear?
> I hope so.
> 
> Anyone know why is it happening? And how can I get
> all
> the communication UDP/RTP packet?s?
> 
> Many thank in advance
> 
> regards,
> 
> Leo
> 
> 
> 
> 		
>
___________________________________________________________
> 
> How much free photo storage do you get? Store your
> holiday 
> snaps for FREE with Yahoo! Photos
> http://uk.photos.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



		
___________________________________________________________ 
To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com


From timfrazee at gmail.com  Fri Dec  9 13:08:18 2005
From: timfrazee at gmail.com (Timothy Frazee)
Date: Fri Dec  9 13:08:24 2005
Subject: [cisco-voip] remote 7960 behind a 501 tweaks
In-Reply-To: <005e01c5fa8d$5582b7d0$6401a8c0@ws1>
Message-ID: <01d001c5fceb$890ea680$6a777e8b@ds.ad.adp.com>

Does anybody have any tips/tricks to improve voice quality on my 501

 

I know I cant load 7 to officially get support for voice but this is what I
have so I gots to make it work.... you know.

 

Attached is my running config off my 501

 

interface ethernet0 auto

interface ethernet1 100full

nameif ethernet0 outside security0

nameif ethernet1 inside security100

enable password xxxxxxxxxxxxxxxxx encrypted

passwd xxxxxxxxxxxxxxxxxxxx encrypted

hostname xxxxxxxxxxxxxxxxxx

domain-name xxxxxxxxxxxxxxxxxxx

clock timezone CST -6

clock summer-time CDT recurring

fixup protocol dns maximum-length 512

fixup protocol ftp 21

fixup protocol h323 h225 1720

fixup protocol h323 ras 1718-1719

fixup protocol http 80

fixup protocol rsh 514

fixup protocol rtsp 554

no fixup protocol sip 5060

fixup protocol sip udp 5060

no fixup protocol skinny 2000

fixup protocol smtp 25

fixup protocol sqlnet 1521

fixup protocol tftp 69

no names

access-list inbound permit tcp any any eq 8662 

access-list inbound permit udp any any eq 8672 

access-list inbound permit tcp any any eq 6881 

access-list inbound permit tcp any any eq 8881 

access-list inbound permit udp any any eq 8881 

access-list inbound permit icmp any any 

pager lines 24

logging on

logging timestamp

logging trap notifications

logging host inside 10.100.0.150 format emblem

icmp permit any inside

mtu outside 1500

mtu inside 1500

ip address outside dhcp setroute

ip address inside 139.126.119.110 255.255.255.248

ip audit info action alarm

ip audit attack action alarm

pdm logging debugging 100

pdm history enable

arp timeout 14400

global (outside) 1 interface

nat (inside) 1 0.0.0.0 0.0.0.0 0 0

static (inside,outside) tcp interface 8662 10.100.0.10 8662 netmask
255.255.255.255 0 0 

static (inside,outside) udp interface 8672 10.100.0.10 8672 netmask
255.255.255.255 0 0 

static (inside,outside) tcp interface 8881 10.100.0.10 8881 netmask
255.255.255.255 0 0 

static (inside,outside) udp interface 8881 10.100.0.10 8881 netmask
255.255.255.255 0 0 

access-group inbound in interface outside

route inside 10.100.0.0 255.255.255.0 139.126.119.109 1

route inside 10.200.200.0 255.255.255.0 139.126.119.109 1

timeout xlate 3:00:00

timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 rpc 0:10:00 h225
1:00:00

timeout h323 0:05:00 mgcp 0:05:00 sip 0:30:00 sip_media 0:02:00

timeout uauth 0:05:00 absolute

aaa-server TACACS+ protocol tacacs+ 

aaa-server TACACS+ max-failed-attempts 3 

aaa-server TACACS+ deadtime 10 

aaa-server RADIUS protocol radius 

aaa-server RADIUS max-failed-attempts 3 

aaa-server RADIUS deadtime 10 

aaa-server LOCAL protocol local 

ntp server 192.5.41.41 source outside

ntp server 192.5.41.40 source outside

http server enable

http 10.200.0.0 255.255.255.0 inside

snmp-server location Frazee Data Center

snmp-server contact Timothy Frazee

snmp-server community public

no snmp-server enable traps

floodguard enable

sysopt connection permit-ipsec

isakmp enable outside

isakmp nat-traversal 20

telnet 192.168.0.0 255.255.255.0 inside

telnet 139.126.119.0 255.255.255.0 inside

telnet 139.126.174.0 255.255.255.0 inside

telnet 10.100.0.0 255.255.255.0 inside

telnet timeout 60

ssh xxxxxxxxxxxxxxxxxxxxxxxx outside

ssh 10.100.0.0 255.255.255.0 inside

ssh timeout 60

management-access inside

console timeout 60

dhcpd address 139.126.119.105-139.126.119.109 inside

dhcpd dns 139.126.174.22 139.126.12.13

dhcpd wins 139.126.174.22 139.126.12.113

dhcpd lease 600

dhcpd ping_timeout 750

dhcpd domain ds.ad.adp.com

dhcpd option 150 ip 139.126.174.51

vpnclient server 63.144.123.169

vpnclient mode network-extension-mode

vpnclient vpngroup xxxxxxxxxxxxxxxxxxxx password ********

vpnclient username xxxxxxxxxxxxxxxxxxxx password ********

vpnclient enable

terminal width 80

 

 

 

 

 

 

The setup is

 

Charter communications home cable internet with 3 down and 512 up to a cisco
501 to a 2950 with qos and vlans setup

 

My 7940 hangs inside of one of the vlans off the 2950

 

 

 

Any help would be great, thanks

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From David.Sullivan at barnet.ac.uk  Fri Dec  9 16:10:56 2005
From: David.Sullivan at barnet.ac.uk (David Sullivan)
Date: Fri Dec  9 16:11:32 2005
Subject: [cisco-voip] UK E1/PRI - Caller ID
In-Reply-To: <3705.64.81.226.84.1134148790.squirrel@www.dignans.com>
References: <3572.64.81.226.84.1134145617.squirrel@www.dignans.com>
Message-ID: <E1EkpWL-0001W7-00@delta.barnet.ac.uk>

On 9 Dec 2005 at 11:19, Andrew Dignan wrote:

> I ended up going with:
> 
> ---------------------------
> voice translation-rule 1
>  rule 1 /\(^[127].........$\)/ /90\1/
>  rule 2 /\(^[1-9]*\)/ /900\1/
> !
> !
> voice translation-profile Prefix_9
>  translate calling 1
> ---------------------------
> 
> Basically any call that comes in beginning with 1, 2, or 7 and has 10
> digits, thats a local/national call.  If it is == to 10 digit or > and
> got past rule 1 I will prefix 00.  So if a call came from Africa
> (Cisco TAC) for example which would be "3227045555" it wouldn't match
> rule 1 but would match rule 2 and prefix 900.  From what I can gather
> no UK > UK calls will ever come in with anything other than 1,2, or 7
> so as long as that holds true I think I will be ok.

It's possible to set a UK CLI to be several other numbers such as an 
08 or 05 which might be as long as 11 digits.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_not
e09186a008028af3a.shtml provides a useful readable summary on this.

I'm pretty sure I've seen International CLIs presented as 
"International" on our resold BT Wholesale PRIs  before now.
If anyone not in the UK feels like placing a call to the non-
connected DDI +442083625000 this should show up in the isdn debug 
log.

Regards.

David.
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From erickbe at yahoo.com  Sat Dec 10 00:34:36 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sat Dec 10 00:34:45 2005
Subject: [cisco-voip] Cannot use Unity media master from Windows 2003
	terminal server c lient
In-Reply-To: <07A14D6987A5F647B2415D72EDF63B5602DC877C@tnsex01.tns.its.psu.edu>
Message-ID: <20051210053436.56674.qmail@web34310.mail.mud.yahoo.com>

Is your Unity server running on Windows 2003 also or
just the Terminal-server jumpbox people are using? 

Windows 2003 terminal server when people connect in
console mode and/or have the use audio option checked
redirects the servers audio/media to the remote
desktop user. Perhaps that is causing a problem. I
know that is a issue on unity servers on windows 2003
and remote desktop users.

Can you have a user connect to the managment server
with VNC and see if it works better?

--- "Simon, Bill" <BillS at tns.its.psu.edu> wrote:

> The title explains most of it.  We have a Windows
> 2003 Terminal Server being
> used as a "gateway" where people log in and run the
> Unity admin site (and
> other VoIP management tools).  The MediaMaster
> control loads and appears to
> work correctly, but it does not.  If I set it to
> phone, the phone rings, I
> pick up, and nothing happens.  At first it seemed
> like an access list issue
> but I can log in directly to Unity and use the
> MediaMaster in this way and
> also from an XP machine on the Unity subnet.
> 
> Any thoughts... I've been round and round with a
> couple fellows from TAC and
> not gotten anywhere.  Has anyone seen and corrected
> an issue like this
> before?  
>  
> 
> ---
> Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
> http://tns.its.psu.edu/
> _______________________________________________
> 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 andy at carse.demon.co.uk  Sat Dec 10 02:28:27 2005
From: andy at carse.demon.co.uk (Andy Carse)
Date: Sat Dec 10 02:29:08 2005
Subject: [cisco-voip] UK E1/PRI - Caller ID
In-Reply-To: <3705.64.81.226.84.1134148790.squirrel@www.dignans.com>
References: <253FAE7A06B11B49A3A29770B2B8BAFC1839B4@CSEXCH01.charles-stanley.co.uk>	<3572.64.81.226.84.1134145617.squirrel@www.dignans.com>
	<3705.64.81.226.84.1134148790.squirrel@www.dignans.com>
Message-ID: <439A839B.80603@carse.demon.co.uk>

This doc from cisco website seems to fit what you are trying to achieve.

http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml 


<snip>

Example 1

In this example, if a number starts with "4" and the type is "national", 
the rule adds "90" as a prefix. If the type is "international", the rule 
adds "900" as the prefix.

*voice translation-rule 7*
* rule 1 /^4/ /904/ type national national*
* rule 2 /^4/ /9004/ type international international*
 

router# *test voice translation-rule 7 493456567 type national*
Matched with rule 1
Original number: 493456567      Translated number: 90493456567
Original number type: national  Translated number type: national
Original number plan: none      Translated number plan: none
    
router# *test voice translation-rule 7 493456567 type international*
Matched with rule 2
Original number: 493456567              Translated number: 900493456567
Original number type: international     Translated number type: 
international
Original number plan: none              Translated number plan: none

This is useful when telephone companies (Telcos) remove access codes on 
national and international numbers. You can add the correct prefix with 
the number type as a basis.

</snip>

>I ended up going with:
>
>---------------------------
>voice translation-rule 1
> rule 1 /\(^[127].........$\)/ /90\1/
> rule 2 /\(^[1-9]*\)/ /900\1/
>!
>!
>voice translation-profile Prefix_9
> translate calling 1
>---------------------------
>
>Basically any call that comes in beginning with 1, 2, or 7 and has 10
>digits, thats a local/national call.  If it is == to 10 digit or > and got
>past rule 1 I will prefix 00.  So if a call came from Africa (Cisco TAC)
>for example which would be "3227045555" it wouldn't match rule 1 but would
>match rule 2 and prefix 900.  From what I can gather no UK > UK calls will
>ever come in with anything other than 1,2, or 7 so as long as that holds
>true I think I will be ok.
>
>  
>
>>This would work I believe for National calls but for International we are
>>not receiving "00".  If someone from the US calls your site what does your
>>caller-id look like?
>>
>>Andy
>>
>>    
>>
>>>Try adding these this is what we have on our gateways in the UK.
>>>
>>>translation-rule 1
>>> Rule 0 ^0. 90
>>> Rule 1 ^1. 901
>>> Rule 2 ^2. 902
>>> Rule 7 ^7. 907
>>>
>>>-----Original Message-----
>>>From: cisco-voip-bounces at puck.nether.net
>>>[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
>>>Sent: 09 December 2005 14:57
>>>To: cisco-voip at puck.nether.net
>>>Subject: [cisco-voip] UK E1/PRI - Caller ID
>>>
>>>Anyone on the list located in the UK?  I am having an issue with
>>>incoming
>>>caller id on an E1 PRI.  The caller id I am receiving does not have the
>>>"0" prefixed for national calls and the international calls do not have
>>>the "00" prefixed.  They come in for all calls with ISDN/National as the
>>>plan/type so I don't have anything I can use in a voice translation rule
>>>to make a decision on whether or not to prefix a 0 or 00.
>>>
>>>Andy
>>>_______________________________________________
>>>cisco-voip mailing list
>>>cisco-voip at puck.nether.net
>>>https://puck.nether.net/mailman/listinfo/cisco-voip
>>>***********************************************************************************************
>>>
>>>The information contained in this e-mail is strictly confidential, some
>>>or
>>>all
>>>of which may be legally privileged. It is for the intended recipient
>>>only.
>>>Access to this e-mail by any other person is prohibited. If you are not
>>>the
>>>intended recipient, 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 and may be unlawful. Please contact the sender
>>>immediately
>>>should this e-mail have been incorrectly addressed or transmitted.
>>>
>>>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 accept that
>>>there
>>>may be a delay in processing the instructions received from e-mails
>>>after
>>>Charles Stanley has received them. You are advised that urgent, time
>>>sensitive
>>>and confidential communications should not be sent by e-mail.
>>>
>>>You acknowledge that e-mails are not secure and you accept the risk of
>>>malfunction, viruses, unauthorised interference, mis-delivery or delay.
>>>************************************************************************************************
>>>
>>>
>>>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 Tozer Wingate are divisions of Charles
>>>Stanley & Co. Ltd
>>>
>>>Authorised and Regulated by the Financial Services Authority, Member of
>>>the
>>>London Stock Exchange, International Securities Markets 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
>
>  
>


From andy at dignans.com  Sat Dec 10 09:44:01 2005
From: andy at dignans.com (Andrew Dignan)
Date: Sat Dec 10 09:44:04 2005
Subject: [cisco-voip] UK E1/PRI - Caller ID
In-Reply-To: <439A839B.80603@carse.demon.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC1839B4@CSEXCH01.charles-stanley.co.uk>	<3572.64.81.226.84.1134145617.squirrel@www.dignans.com>
	<3705.64.81.226.84.1134148790.squirrel@www.dignans.com>
	<439A839B.80603@carse.demon.co.uk>
Message-ID: <4968.64.81.226.84.1134225841.squirrel@www.dignans.com>

Well, the problem is that British Telecomm is not seting the type as
"International" on any calls.  All calls come as type "National" so I
can't distinguish between national/local and international calls on the
type, only by what digits I am getting.  Does BT send your International
calls inbound marked with the correct type?

> This doc from cisco website seems to fit what you are trying to achieve.
>
> http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
>
>
> <snip>
>
> Example 1
>
> In this example, if a number starts with "4" and the type is "national",
> the rule adds "90" as a prefix. If the type is "international", the rule
> adds "900" as the prefix.
>
> *voice translation-rule 7*
> * rule 1 /^4/ /904/ type national national*
> * rule 2 /^4/ /9004/ type international international*
>
>
> router# *test voice translation-rule 7 493456567 type national*
> Matched with rule 1
> Original number: 493456567      Translated number: 90493456567
> Original number type: national  Translated number type: national
> Original number plan: none      Translated number plan: none
>
> router# *test voice translation-rule 7 493456567 type international*
> Matched with rule 2
> Original number: 493456567              Translated number: 900493456567
> Original number type: international     Translated number type:
> international
> Original number plan: none              Translated number plan: none
>
> This is useful when telephone companies (Telcos) remove access codes on
> national and international numbers. You can add the correct prefix with
> the number type as a basis.
>
> </snip>
>
>>I ended up going with:
>>
>>---------------------------
>>voice translation-rule 1
>> rule 1 /\(^[127].........$\)/ /90\1/
>> rule 2 /\(^[1-9]*\)/ /900\1/
>>!
>>!
>>voice translation-profile Prefix_9
>> translate calling 1
>>---------------------------
>>
>>Basically any call that comes in beginning with 1, 2, or 7 and has 10
>>digits, thats a local/national call.  If it is == to 10 digit or > and
>> got
>>past rule 1 I will prefix 00.  So if a call came from Africa (Cisco TAC)
>>for example which would be "3227045555" it wouldn't match rule 1 but
>> would
>>match rule 2 and prefix 900.  From what I can gather no UK > UK calls
>> will
>>ever come in with anything other than 1,2, or 7 so as long as that holds
>>true I think I will be ok.
>>
>>
>>
>>>This would work I believe for National calls but for International we
>>> are
>>>not receiving "00".  If someone from the US calls your site what does
>>> your
>>>caller-id look like?
>>>
>>>Andy
>>>
>>>
>>>
>>>>Try adding these this is what we have on our gateways in the UK.
>>>>
>>>>translation-rule 1
>>>> Rule 0 ^0. 90
>>>> Rule 1 ^1. 901
>>>> Rule 2 ^2. 902
>>>> Rule 7 ^7. 907
>>>>
>>>>-----Original Message-----
>>>>From: cisco-voip-bounces at puck.nether.net
>>>>[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andrew Dignan
>>>>Sent: 09 December 2005 14:57
>>>>To: cisco-voip at puck.nether.net
>>>>Subject: [cisco-voip] UK E1/PRI - Caller ID
>>>>
>>>>Anyone on the list located in the UK?  I am having an issue with
>>>>incoming
>>>>caller id on an E1 PRI.  The caller id I am receiving does not have the
>>>>"0" prefixed for national calls and the international calls do not have
>>>>the "00" prefixed.  They come in for all calls with ISDN/National as
>>>> the
>>>>plan/type so I don't have anything I can use in a voice translation
>>>> rule
>>>>to make a decision on whether or not to prefix a 0 or 00.
>>>>
>>>>Andy
>>>>_______________________________________________
>>>>cisco-voip mailing list
>>>>cisco-voip at puck.nether.net
>>>>https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>***********************************************************************************************
>>>>
>>>>The information contained in this e-mail is strictly confidential, some
>>>>or
>>>>all
>>>>of which may be legally privileged. It is for the intended recipient
>>>>only.
>>>>Access to this e-mail by any other person is prohibited. If you are not
>>>>the
>>>>intended recipient, 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 and may be unlawful. Please contact the sender
>>>>immediately
>>>>should this e-mail have been incorrectly addressed or transmitted.
>>>>
>>>>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 accept
>>>> that
>>>>there
>>>>may be a delay in processing the instructions received from e-mails
>>>>after
>>>>Charles Stanley has received them. You are advised that urgent, time
>>>>sensitive
>>>>and confidential communications should not be sent by e-mail.
>>>>
>>>>You acknowledge that e-mails are not secure and you accept the risk of
>>>>malfunction, viruses, unauthorised interference, mis-delivery or delay.
>>>>************************************************************************************************
>>>>
>>>>
>>>>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 Tozer Wingate are divisions of Charles
>>>>Stanley & Co. Ltd
>>>>
>>>>Authorised and Regulated by the Financial Services Authority, Member of
>>>>the
>>>>London Stock Exchange, International Securities Markets 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
>>
>>
>>
>
>


From jgrace at digitelusa.net  Sat Dec 10 15:24:27 2005
From: jgrace at digitelusa.net (James)
Date: Sat Dec 10 15:24:30 2005
Subject: [cisco-voip] What the default unity vm box size
Message-ID: <943591778-1134246267-cardhu_blackberry.rim.net-17597-@engine12-cell02>

Is there a way to change the time allow to leave a vm message in unity.  And what is the default time set to
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From kthorngr at cisco.com  Sat Dec 10 16:21:54 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Sat Dec 10 16:22:03 2005
Subject: [cisco-voip] What the default unity vm box size
In-Reply-To: <943591778-1134246267-cardhu_blackberry.rim.net-17597-@engine12-cell02>
References: <943591778-1134246267-cardhu_blackberry.rim.net-17597-@engine12-cell02>
Message-ID: <15fd0775dfd4cf6da4bd2e1ed1b2940f@cisco.com>

Take a look here:
http://www.cisco.com/en/US/products/sw/voicesw/ps2237/ 
products_administration_guide_chapter09186a008022d19b.html

Kevin
On Dec 10, 2005, at 3:24 PM, James wrote:

> Is there a way to change the time allow to leave a vm message in  
> unity.  And what is the default time set to
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From nkhwaja at gmail.com  Sun Dec 11 02:59:43 2005
From: nkhwaja at gmail.com (Nasser Khwaja)
Date: Sun Dec 11 02:59:45 2005
Subject: [cisco-voip] Cat 5500 CatOS & IOS upgd
Message-ID: <23435d3d0512102359t3b967e59t680f4a1dfa8e6adc@mail.gmail.com>

Hi,
On a Cat 5500, the Sup II CatOS is ver 4.5 & on RSM IOS is ver. 11.2. I want
to upgrade BOTH of the above. Is their a compatibility matrix?
Thanks - Tennisman.
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From Brian.Henry at apptis.com  Sun Dec 11 09:01:15 2005
From: Brian.Henry at apptis.com (Brian Henry)
Date: Sun Dec 11 09:01:48 2005
Subject: [cisco-voip] Cat 5500 CatOS & IOS upgd
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC03D777@CH-EXCH-02.intelltech.apptis.inc>

Did you try this link already?
http://tools.cisco.com/Support/Fusion/FusionHome.do
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Nasser Khwaja
Sent: Sun 12/11/2005 2:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cat 5500 CatOS & IOS upgd



Hi, 
On a Cat 5500, the Sup II CatOS is ver 4.5 & on RSM IOS is ver. 11.2. I want to upgrade BOTH of the above. Is their a compatibility matrix? 
Thanks - Tennisman.


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From Nick.Kassel at Charles-Stanley.co.uk  Sun Dec 11 09:27:22 2005
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Sun Dec 11 09:27:36 2005
Subject: [cisco-voip] FXS port problems
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC1839C3@CSEXCH01.charles-stanley.co.uk>

I am able to dial out of the ports that are not configured for incoming
calls but as soon as I associate it with a dial-peer the outgoing calls
stop and I get the fast busy signal, the incoming will the work fine. Am
I missing a command to allow the ports to act as both incoming and
outgoing.

 

below is my voice ports and dial peer config

 

 

!

!

!

voice-port 0/0/0:15

!

voice-port 0/1/0

 cptone GB

 description Connected to 0131 550 1250

!

voice-port 0/1/1

 cptone GB

!

voice-port 0/1/2

 cptone GB

!

voice-port 0/1/3

 cptone GB

!

mgcp

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

!

!

!

dial-peer voice 100 voip

 incoming called-number .

 codec g711ulaw

 no vad

!

dial-peer voice 9 pots

 destination-pattern 9T

 progress_ind alert enable 8

 direct-inward-dial

 port 0/0/0:15

!

dial-peer voice 2 voip

 destination-pattern 5012..

 progress_ind setup enable 3

 translate-outgoing calling 1

 session target ipv4:172.30.0.2

 dtmf-relay h245-alphanumeric

 codec g711ulaw

 no vad

!

dial-peer voice 3 voip

 preference 2

 destination-pattern 5012..

 progress_ind setup enable 3

 translate-outgoing calling 1

 session target ipv4:172.20.100.2

 dtmf-relay h245-alphanumeric

 codec g711ulaw

 no vad

!

dial-peer voice 1250 pots

 destination-pattern 81250

 fax rate disable

 port 0/1/0

!

gateway

 timer receive-rtp 1200

!

!

!

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: 09 December 2005 14:31
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] FXS port problems

 

I am configuring a 4 port fxs card for analogue lines to be used in a
branch office, I have incoming working fine but the outgoing calls, it
will ring once on the destination phone and then cut out. 

 

I have the following dial peer and then a route pattern of 81250 on the
call manager pointing to the gateway.

 

dial-peer voice 1250 pots

 destination-pattern 81250

 progress_ind alert enable 8

 fax rate disable

 port 0/1/0

 

Any help would be appreciated.

 

 

 


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

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets 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.

 

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

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets 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.
-------------- next part --------------
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From gmaiorana at ncmail.com.au  Sun Dec 11 18:58:39 2005
From: gmaiorana at ncmail.com.au (Greg Maiorana)
Date: Sun Dec 11 19:01:58 2005
Subject: [cisco-voip] Setting up a constant voice link
Message-ID: <439CBD2F.2020204@ncmail.com.au>

Hello All,
        I trying to work out a solution with a customer to bring their 
current pots leased line system to a VoIP solution. The customer 
currently has a PSTN leased line from site A to B, which then plugs into 
some type of intercom system each end(think of an analog phone on each 
end with it off hook all the time).

Is there anyway to do this over VoIP ?
That is have an open link between say 2 x 1721 or 1751 cisco routers 
with FXS ports to have an non stop open stream between the FXS port at 
each location. So when customer A presses the intercom button to connect 
to site B its already open & there is no dialling between the 2 systems.
Any suggestions or idea's would be largely appreciated.

Cheers,

Greg Maiorana.

From jgrace at digitelusa.net  Sun Dec 11 19:07:22 2005
From: jgrace at digitelusa.net (James)
Date: Sun Dec 11 19:07:17 2005
Subject: [cisco-voip] Vt advantage
Message-ID: <1452648987-1134346035-cardhu_blackberry.rim.net-24689-@engine09-cell02>

I'm trying to set up vtadvantage.  But I'm not having luck on enabling video on my 7940.   I don't see how to make the vidoe icon appear. And not sure if this is related but when I open vtadvantage it does no detect my ip phone.  I am connected to the ipphone swith 
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From jkoniecki at comcast.net  Sun Dec 11 19:37:12 2005
From: jkoniecki at comcast.net (Jim Koniecki)
Date: Sun Dec 11 19:37:23 2005
Subject: [cisco-voip] Vt advantage
In-Reply-To: <1452648987-1134346035-cardhu_blackberry.rim.net-24689-@engine09-cell02>
Message-ID: <003001c5feb4$31dfdc10$c800a8c0@Jalbert1>

James,

This is how I have it working 


VTadv---(USB)----LAPTOP----IPPhone1001-----SWITCH
VTadv---(USB)----LAPTOP----IPPhone1002-----SWITCH

Then just follow this
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a00802345f4.shtml

Components Used 
The information in this document is based on these software and hardware
versions: 

Cisco CallManager 4.01 with SR2a or later 

Cisco IP phones 7960G and 7940G with phone loads 6.0(4) or later



Jim Koniecki 


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James
Sent: Sunday, December 11, 2005 7:07 PM
To: CiscosupportUpuck
Subject: [cisco-voip] Vt advantage

I'm trying to set up vtadvantage.  But I'm not having luck on enabling video
on my 7940.   I don't see how to make the vidoe icon appear. And not sure if
this is related but when I open vtadvantage it does no detect my ip phone.
I am connected to the ipphone swith 
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From gmaiorana at ncmail.com.au  Sun Dec 11 22:38:14 2005
From: gmaiorana at ncmail.com.au (Greg Maiorana)
Date: Sun Dec 11 22:56:30 2005
Subject: [cisco-voip] Setting up a constant voice link
Message-ID: <439CF0A6.6030302@ncmail.com.au>

Hello All,
       Im trying to work out a solution with a customer to bring their 
current pots leased line system to a VoIP solution. The customer 
currently has a PSTN leased line from site A to B, which then plugs into 
some type of intercom system each end(think of an analog phone on each 
end with it off hook all the time).

Is there anyway to do this over VoIP ?
That is have an open link between say 2 x 1721 or 1751 cisco routers 
with FXS ports to have an non stop open stream between the FXS port at 
each location. So when customer A presses the intercom button to connect 
to site B its already open & there is no dialling between the 2 systems.
Any suggestions or idea's would be largely appreciated.

Cheers,

Greg Maiorana.

From rratliff at cisco.com  Sun Dec 11 23:33:09 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Sun Dec 11 23:33:23 2005
Subject: [cisco-voip] Setting up a constant voice link
In-Reply-To: <439CF0A6.6030302@ncmail.com.au>
References: <439CF0A6.6030302@ncmail.com.au>
Message-ID: <30069D7C-F3A2-4F8A-A3ED-2DFA67EEBE0F@cisco.com>

Isn't this hoot and holler?

12.2T docs:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/ 
products_feature_guide09186a0080087cfc.html

-Ryan

On Dec 11, 2005, at 10:38 PM, Greg Maiorana wrote:

Hello All,
        Im trying to work out a solution with a customer to bring their
current pots leased line system to a VoIP solution. The customer
currently has a PSTN leased line from site A to B, which then plugs into
some type of intercom system each end(think of an analog phone on each
end with it off hook all the time).

Is there anyway to do this over VoIP ?
That is have an open link between say 2 x 1721 or 1751 cisco routers
with FXS ports to have an non stop open stream between the FXS port at
each location. So when customer A presses the intercom button to connect
to site B its already open & there is no dialling between the 2 systems.
Any suggestions or idea's would be largely appreciated.

Cheers,

Greg Maiorana.
_______________________________________________
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  Mon Dec 12 06:09:08 2005
From: Ruben.Montes at eu.didata.com (Ruben Montes)
Date: Mon Dec 12 06:09:13 2005
Subject: [cisco-voip] Slips errors in a Q.SIG connection
Message-ID: <D305A0848689344BB98F14A3F13161B710B497@eubebrusvex.eu.didata.local>

Hello,

 

I have a connection between a PBX MD110 and a Cisco router C2851 running
MGCP. I have Slips errors all the time, counter is increasing every few
seconds. Calls are working well but I have a lot of errors. The MD110 is
configured to supply the clock and this is the config in the router:

 

controller E1 0/0/0

 clock source line primary

 pri-group timeslots 1-31 service mgcp

!

controller E1 0/0/1

 pri-group timeslots 1-31 service mgcp

 

I configured the 'clock source line primary' in one of them but both
QSiGs have the same Slip errors:

 

r2851ctp01#sh controllers e1

E1 0/0/0 is up.

  Applique type is Channelized E1 - balanced

  No alarms detected.

  alarm-trigger is not set

  Version info Firmware: 20050620, FPGA: 16, spm_count = 0

  Framing is CRC4, Line Code is HDB3, Clock Source is Line Primary.

  CRC Threshold is 320. Reported from firmware  is 320.

  Data in current interval (422 seconds elapsed):

     0 Line Code Violations, 0 Path Code Violations

     48 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

     48 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail
Secs

E1 0/0/1 is up.

  Applique type is Channelized E1 - balanced

  No alarms detected.

  alarm-trigger is not set

  Version info Firmware: 20050620, FPGA: 16, spm_count = 0

  Framing is CRC4, Line Code is HDB3, Clock Source is Line.

  CRC Threshold is 320. Reported from firmware  is 320.

  Data in current interval (423 seconds elapsed):

     0 Line Code Violations, 0 Path Code Violations

     48 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

     48 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail
Secs

 

Any idea? The config in the CCM side is almost the default...

 

Regards,

 

Ruben

 

 

 

  _____  

De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de Leetun, Rob
Enviado el: martes, 06 de diciembre de 2005 18:27
Para: Voll, Scott; Lelio Fulgenzi; Jeffrey C. Ollie;
cisco-voip at puck.nether.net
Asunto: RE: [cisco-voip] CDR Reporting Tool

 

Dido here.  Our Oracle developers built a package in Crystal Reports for
management, which spits out about 10 to 15 different reports each month.


 

Rob

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, December 06, 2005 10:13 AM
To: Lelio Fulgenzi; Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CDR Reporting Tool

 

we used our in house programmers and told them what we needed and they
came up with a custom app for us.  Not web based but it interfaces with
our billing system to automagically send bills each month based on total
toll charges.  took him about 2.5 hours to do.

 

Scott

 

  _____  

From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
Sent: Tue 12/6/2005 8:32 AM
To: Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR Reporting Tool

I'd also like to hear about such a beast.

 

But I'd also like to hear about people's experience with setting up
their own reports using a custom SQL report tool. I've looked at a few
out there that seem quite good and allow you to create SQL reports with
user input and are web based.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Jeffrey C. Ollie <mailto:jeff at ocjtech.us>  

	To: cisco-voip at puck.nether.net 

	Sent: Tuesday, December 06, 2005 11:26 AM

	Subject: [cisco-voip] CDR Reporting Tool

	 

	Can anyone point me to a CDR reporting tool that:

	 

	1) Will directly access the Cisco CDR database (we've used
Infortel but
	it discards a lot of the information in the Cisco CDR records).
	2) Provides more advanced reporting capabilities than ART
provides.
	3) Specifically targeted at CDR reports, not some generic SQL
reporting
	tool.

	 

	Jeff Ollie

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

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From Ruben.Montes at eu.didata.com  Mon Dec 12 06:20:46 2005
From: Ruben.Montes at eu.didata.com (Ruben Montes)
Date: Mon Dec 12 06:20:52 2005
Subject: [cisco-voip] Slips errors in a Q.SIG connection
Message-ID: <D305A0848689344BB98F14A3F13161B710B4A2@eubebrusvex.eu.didata.local>

Yes, I have it.

 

I attach the config related with the voice part:

 

 

aaa session-id common

!

resource policy

!

clock timezone GMT+1 1

network-clock-participate wic 0 

ip subnet-zero

!

!

ip cef

no ip dhcp use vrf connected

!

!

no ip domain lookup

no ftp-server write-enable

isdn switch-type primary-qsig

!

voice-card 0

 no dspfarm

!

controller E1 0/0/0

 clock source line primary

 pri-group timeslots 1-31 service mgcp

!

controller E1 0/0/1

 pri-group timeslots 1-31 service mgcp

!

!

!

interface GigabitEthernet0/0

 ip address 47.193.64.3 255.255.255.0

 duplex full

 speed 100

!

interface GigabitEthernet0/1

 no ip address

 shutdown

 duplex auto

 speed auto

!

interface Serial0/0/0:15

 no ip address

 isdn switch-type primary-qsig

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

interface Serial0/0/1:15

 no ip address

 isdn switch-type primary-qsig

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

ip classless

ip route 0.0.0.0 0.0.0.0 47.193.64.1

!

!

i

!

control-plane

!

!

!

voice-port 0/0/0:15

 cptone ES

!

voice-port 0/0/1:15

 cptone ES

!

ccm-manager switchback immediate

ccm-manager redundant-host 47.222.64.2

ccm-manager mgcp

!

mgcp

mgcp call-agent 47.193.64.2 service-type mgcp version 0.1

mgcp bind control source-interface GigabitEthernet0/0

mgcp bind media source-interface GigabitEthernet0/0

!

mgcp profile default

!

!

!

dial-peer voice 1 pots

 service mgcpapp

 port 0/0/0:15

!

dial-peer voice 2 pots

 service mgcpapp

 port 0/0/1:15

!

!

!

 

End

 

Regards,

 

Ruben

 

  _____  

De: Brian Henry [mailto:Brian.Henry at apptis.com] 
Enviado el: lunes, 12 de diciembre de 2005 12:18
Para: Ruben Montes
Asunto: Re: [cisco-voip] Slips errors in a Q.SIG connection

 

Do you have the.command network clock participate wic x?
--------------------------
Sent from my BlackBerry Wireless Handheld


-----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: Mon Dec 12 06:09:08 2005
Subject: [cisco-voip] Slips errors in a Q.SIG connection

Hello,



I have a connection between a PBX MD110 and a Cisco router C2851 running
MGCP. I have Slips errors all the time, counter is increasing every few
seconds. Calls are working well but I have a lot of errors. The MD110 is
configured to supply the clock and this is the config in the router:



controller E1 0/0/0

 clock source line primary

 pri-group timeslots 1-31 service mgcp

!

controller E1 0/0/1

 pri-group timeslots 1-31 service mgcp



I configured the 'clock source line primary' in one of them but both
QSiGs have the same Slip errors:



r2851ctp01#sh controllers e1

E1 0/0/0 is up.

  Applique type is Channelized E1 - balanced

  No alarms detected.

  alarm-trigger is not set

  Version info Firmware: 20050620, FPGA: 16, spm_count = 0

  Framing is CRC4, Line Code is HDB3, Clock Source is Line Primary.

  CRC Threshold is 320. Reported from firmware  is 320.

  Data in current interval (422 seconds elapsed):

     0 Line Code Violations, 0 Path Code Violations

     48 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

     48 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail
Secs

E1 0/0/1 is up.

  Applique type is Channelized E1 - balanced

  No alarms detected.

  alarm-trigger is not set

  Version info Firmware: 20050620, FPGA: 16, spm_count = 0

  Framing is CRC4, Line Code is HDB3, Clock Source is Line.

  CRC Threshold is 320. Reported from firmware  is 320.

  Data in current interval (423 seconds elapsed):

     0 Line Code Violations, 0 Path Code Violations

     48 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

     48 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail
Secs



Any idea? The config in the CCM side is almost the default...



Regards,



Ruben







________________________________

De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de Leetun, Rob
Enviado el: martes, 06 de diciembre de 2005 18:27
Para: Voll, Scott; Lelio Fulgenzi; Jeffrey C. Ollie;
cisco-voip at puck.nether.net
Asunto: RE: [cisco-voip] CDR Reporting Tool



Dido here.  Our Oracle developers built a package in Crystal Reports for
management, which spits out about 10 to 15 different reports each month.




Rob



________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, December 06, 2005 10:13 AM
To: Lelio Fulgenzi; Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CDR Reporting Tool



we used our in house programmers and told them what we needed and they
came up with a custom app for us.  Not web based but it interfaces with
our billing system to automagically send bills each month based on total
toll charges.  took him about 2.5 hours to do.



Scott



________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
Sent: Tue 12/6/2005 8:32 AM
To: Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR Reporting Tool

I'd also like to hear about such a beast.



But I'd also like to hear about people's experience with setting up
their own reports using a custom SQL report tool. I've looked at a few
out there that seem quite good and allow you to create SQL reports with
user input and are web based.



------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

        ----- Original Message -----

        From: Jeffrey C. Ollie <mailto:jeff at ocjtech.us> 

        To: cisco-voip at puck.nether.net

        Sent: Tuesday, December 06, 2005 11:26 AM

        Subject: [cisco-voip] CDR Reporting Tool

        

        Can anyone point me to a CDR reporting tool that:

        

        1) Will directly access the Cisco CDR database (we've used
Infortel but
        it discards a lot of the information in the Cisco CDR records).
        2) Provides more advanced reporting capabilities than ART
provides.
        3) Specifically targeted at CDR reports, not some generic SQL
reporting
        tool.

        

        Jeff Ollie

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

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From gboda at eircom.ie  Mon Dec 12 06:36:38 2005
From: gboda at eircom.ie (gboda@eircom.ie)
Date: Mon Dec 12 06:36:52 2005
Subject: [cisco-voip] Slips errors in a Q.SIG connection
Message-ID: <1BF7574B8535D5119F790002B3304FE312EE8E64@EIRCOMMSG02>

Hi Ruben,
 
you need to set the following:
 
network-clock-participate slot x 

network-clock-select x E1 x/x/x 

 

Best regards,

 

Greg

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net]On Behalf Of Ruben Montes
Sent: 12 December 2005 11:21
To: Brian Henry
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Slips errors in a Q.SIG connection



Yes, I have it.

 

I attach the config related with the voice part:

 

 

aaa session-id common

!

resource policy

!

clock timezone GMT+1 1

network-clock-participate wic 0 

ip subnet-zero

!

!

ip cef

no ip dhcp use vrf connected

!

!

no ip domain lookup

no ftp-server write-enable

isdn switch-type primary-qsig

!

voice-card 0

 no dspfarm

!

controller E1 0/0/0

 clock source line primary

 pri-group timeslots 1-31 service mgcp

!

controller E1 0/0/1

 pri-group timeslots 1-31 service mgcp

!

!

!

interface GigabitEthernet0/0

 ip address 47.193.64.3 255.255.255.0

 duplex full

 speed 100

!

interface GigabitEthernet0/1

 no ip address

 shutdown

 duplex auto

 speed auto

!

interface Serial0/0/0:15

 no ip address

 isdn switch-type primary-qsig

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

interface Serial0/0/1:15

 no ip address

 isdn switch-type primary-qsig

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

ip classless

ip route 0.0.0.0 0.0.0.0 47.193.64.1

!

!

i

!

control-plane

!

!

!

voice-port 0/0/0:15

 cptone ES

!

voice-port 0/0/1:15

 cptone ES

!

ccm-manager switchback immediate

ccm-manager redundant-host 47.222.64.2

ccm-manager mgcp

!

mgcp

mgcp call-agent 47.193.64.2 service-type mgcp version 0.1

mgcp bind control source-interface GigabitEthernet0/0

mgcp bind media source-interface GigabitEthernet0/0

!

mgcp profile default

!

!

!

dial-peer voice 1 pots

 service mgcpapp

 port 0/0/0:15

!

dial-peer voice 2 pots

 service mgcpapp

 port 0/0/1:15

!

!

!

 

End

 

Regards,

 

Ruben

 


  _____  


De: Brian Henry [mailto:Brian.Henry at apptis.com] 
Enviado el: lunes, 12 de diciembre de 2005 12:18
Para: Ruben Montes
Asunto: Re: [cisco-voip] Slips errors in a Q.SIG connection

 

Do you have the.command network clock participate wic x?
--------------------------
Sent from my BlackBerry Wireless Handheld


-----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: Mon Dec 12 06:09:08 2005
Subject: [cisco-voip] Slips errors in a Q.SIG connection

Hello,



I have a connection between a PBX MD110 and a Cisco router C2851 running
MGCP. I have Slips errors all the time, counter is increasing every few
seconds. Calls are working well but I have a lot of errors. The MD110 is
configured to supply the clock and this is the config in the router:



controller E1 0/0/0

 clock source line primary

 pri-group timeslots 1-31 service mgcp

!

controller E1 0/0/1

 pri-group timeslots 1-31 service mgcp



I configured the 'clock source line primary' in one of them but both QSiGs
have the same Slip errors:



r2851ctp01#sh controllers e1

E1 0/0/0 is up.

  Applique type is Channelized E1 - balanced

  No alarms detected.

  alarm-trigger is not set

  Version info Firmware: 20050620, FPGA: 16, spm_count = 0

  Framing is CRC4, Line Code is HDB3, Clock Source is Line Primary.

  CRC Threshold is 320. Reported from firmware  is 320.

  Data in current interval (422 seconds elapsed):

     0 Line Code Violations, 0 Path Code Violations

     48 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

     48 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs

E1 0/0/1 is up.

  Applique type is Channelized E1 - balanced

  No alarms detected.

  alarm-trigger is not set

  Version info Firmware: 20050620, FPGA: 16, spm_count = 0

  Framing is CRC4, Line Code is HDB3, Clock Source is Line.

  CRC Threshold is 320. Reported from firmware  is 320.

  Data in current interval (423 seconds elapsed):

     0 Line Code Violations, 0 Path Code Violations

     48 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

     48 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs



Any idea? The config in the CCM side is almost the default...



Regards,



Ruben







________________________________

De: cisco-voip-bounces at puck.nether.net [
mailto:cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] En nombre de Leetun, Rob
Enviado el: martes, 06 de diciembre de 2005 18:27
Para: Voll, Scott; Lelio Fulgenzi; Jeffrey C. Ollie;
cisco-voip at puck.nether.net
Asunto: RE: [cisco-voip] CDR Reporting Tool



Dido here.  Our Oracle developers built a package in Crystal Reports for
management, which spits out about 10 to 15 different reports each month. 



Rob



________________________________

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 Voll, Scott
Sent: Tuesday, December 06, 2005 10:13 AM
To: Lelio Fulgenzi; Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CDR Reporting Tool



we used our in house programmers and told them what we needed and they came
up with a custom app for us.  Not web based but it interfaces with our
billing system to automagically send bills each month based on total toll
charges.  took him about 2.5 hours to do.



Scott



________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
Sent: Tue 12/6/2005 8:32 AM
To: Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR Reporting Tool

I'd also like to hear about such a beast.



But I'd also like to hear about people's experience with setting up their
own reports using a custom SQL report tool. I've looked at a few out there
that seem quite good and allow you to create SQL reports with user input and
are web based.



----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more jitter than
an
IP phone on a long haul 10base2 connection"
LFJ

        ----- Original Message -----

        From: Jeffrey C. Ollie < mailto:jeff at ocjtech.us
<mailto:jeff at ocjtech.us> > 

        To: cisco-voip at puck.nether.net

        Sent: Tuesday, December 06, 2005 11:26 AM

        Subject: [cisco-voip] CDR Reporting Tool

        

        Can anyone point me to a CDR reporting tool that:

        

        1) Will directly access the Cisco CDR database (we've used Infortel
but
        it discards a lot of the information in the Cisco CDR records).
        2) Provides more advanced reporting capabilities than ART provides.
        3) Specifically targeted at CDR reports, not some generic SQL
reporting
        tool.

        

        Jeff Ollie

        _______________________________________________
        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 mfa at crec.ifas.ufl.edu  Mon Dec 12 07:42:43 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Mon Dec 12 07:42:47 2005
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 42
References: <200512121111.jBCB9DLc023945@puck.nether.net>
Message-ID: <004501c5ff19$8d219220$0402a8c0@crec.ifas.ufl.edu>

You've also got to enable video capabilities for the phone in CallManager, 
down toward the bottom of the phone cnfiguration page..  If the little video 
icon doesn't appear on the display, it's not video enabled.  I remember the 
setup being a little confusing and not entirely conforming to the 
documentation, but I just followed the directions, ad-libbing where 
required, and it worked OK.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL


> Date: Mon, 12 Dec 2005 00:07:22 +0000 GMT
> From: "James" <jgrace at digitelusa.net>
> Subject: [cisco-voip] Vt advantage
> To: "CiscosupportUpuck" <cisco-voip at puck.nether.net>
> Message-ID:
> <1452648987-1134346035-cardhu_blackberry.rim.net-24689- at engine09-cell02>
>
> Content-Type: text/plain
>
> I'm trying to set up vtadvantage.  But I'm not having luck on enabling 
> video on my 7940.   I don't see how to make the vidoe icon appear. And not 
> sure if this is related but when I open vtadvantage it does no detect my 
> ip phone.  I am connected to the ipphone swith
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.


From Spencer.Wood at dot.state.oh.us  Mon Dec 12 07:55:39 2005
From: Spencer.Wood at dot.state.oh.us (Spencer.Wood@dot.state.oh.us)
Date: Mon Dec 12 07:55:48 2005
Subject: [cisco-voip] Setting up a constant voice link
In-Reply-To: <439CF0A6.6030302@ncmail.com.au>
Message-ID: <OF6E753165.2001707F-ON852570D5.00469673-852570D5.00470496@dot.state.oh.us>

A Pair of E&M Interface's should do the trick...

Here is a Cisco Design Guide on LMR (Land Mobile Radio):

http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps5207/products_implementation_design_guide_book09186a0080347c1b.html

While what you are looking for isn't exactly LMR, the overall concept 
should still work.  Your implementation should actually be easier if there 
isn't any Push-To-Talk Interfaces.

Spencer

************************************************************
Spencer Wood, Network Manager
Ohio Department Of Transportation
1320 Arthur E. Adams Drive
Columbus, Ohio 43221 
E-Mail: Spencer.Wood at dot.state.oh.us
Phone: 614.644.5422/Fax: 614.887.4021/Cell: 614.774.3123 
************************************************************* 



Greg Maiorana <gmaiorana at ncmail.com.au> 
Sent by: cisco-voip-bounces at puck.nether.net
12/11/2005 10:38 PM

To
cisco-voip at puck.nether.net
cc

Subject
[cisco-voip] Setting up a constant voice link






Hello All,
       Im trying to work out a solution with a customer to bring their 
current pots leased line system to a VoIP solution. The customer 
currently has a PSTN leased line from site A to B, which then plugs into 
some type of intercom system each end(think of an analog phone on each 
end with it off hook all the time).

Is there anyway to do this over VoIP ?
That is have an open link between say 2 x 1721 or 1751 cisco routers 
with FXS ports to have an non stop open stream between the FXS port at 
each location. So when customer A presses the intercom button to connect 
to site B its already open & there is no dialling between the 2 systems.
Any suggestions or idea's would be largely appreciated.

Cheers,

Greg Maiorana.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
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From syed.abbas at eecu.org  Mon Dec 12 09:06:47 2005
From: syed.abbas at eecu.org (Syed Abbas)
Date: Mon Dec 12 09:07:07 2005
Subject: [cisco-voip] VT Advantage Issue
Message-ID: <46599BB550C81741993E3765F670707A5C982B@exchbe01.EECU.DOM>

 Hello James,

Have you enabled the Video Capabilities for that specific phone on Call
Manager? 

Regards,

Syed

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
cisco-voip-request at puck.nether.net
Sent: Monday, December 12, 2005 5:12 AM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 34, Issue 42

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Today's Topics:

   1. Setting up a constant voice link (Greg Maiorana)
   2. Vt advantage (James)
   3. RE: Vt advantage (Jim Koniecki)
   4. Setting up a constant voice link (Greg Maiorana)
   5. Re: Setting up a constant voice link (Ryan Ratliff)
   6. Slips errors in a Q.SIG connection (Ruben Montes)


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

Message: 1
Date: Mon, 12 Dec 2005 10:58:39 +1100
From: Greg Maiorana <gmaiorana at ncmail.com.au>
Subject: [cisco-voip] Setting up a constant voice link
To: cisco-voip at puck.nether.net, gmaiorana at ncmail.com.au
Message-ID: <439CBD2F.2020204 at ncmail.com.au>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hello All,
        I trying to work out a solution with a customer to bring their
current pots leased line system to a VoIP solution. The customer
currently has a PSTN leased line from site A to B, which then plugs into
some type of intercom system each end(think of an analog phone on each
end with it off hook all the time).

Is there anyway to do this over VoIP ?
That is have an open link between say 2 x 1721 or 1751 cisco routers
with FXS ports to have an non stop open stream between the FXS port at
each location. So when customer A presses the intercom button to connect
to site B its already open & there is no dialling between the 2 systems.
Any suggestions or idea's would be largely appreciated.

Cheers,

Greg Maiorana.


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

Message: 2
Date: Mon, 12 Dec 2005 00:07:22 +0000 GMT
From: "James" <jgrace at digitelusa.net>
Subject: [cisco-voip] Vt advantage
To: "CiscosupportUpuck" <cisco-voip at puck.nether.net>
Message-ID:
	
<1452648987-1134346035-cardhu_blackberry.rim.net-24689- at engine09-cell02>
	
Content-Type: text/plain

I'm trying to set up vtadvantage.  But I'm not having luck on enabling
video on my 7940.   I don't see how to make the vidoe icon appear. And
not sure if this is related but when I open vtadvantage it does no
detect my ip phone.  I am connected to the ipphone swith 
James D. Grace
System Engineer
CCNP CCNA MCSE MCDBA
Digitel Corp.


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

Message: 3
Date: Sun, 11 Dec 2005 19:37:12 -0500
From: "Jim Koniecki" <jkoniecki at comcast.net>
Subject: RE: [cisco-voip] Vt advantage
To: "'James'" <jgrace at digitelusa.net>,	"'CiscosupportUpuck'"
	<cisco-voip at puck.nether.net>
Message-ID: <003001c5feb4$31dfdc10$c800a8c0 at Jalbert1>
Content-Type: text/plain;	charset="us-ascii"

James,

This is how I have it working 


VTadv---(USB)----LAPTOP----IPPhone1001-----SWITCH
VTadv---(USB)----LAPTOP----IPPhone1002-----SWITCH

Then just follow this
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186
a00802345f4.shtml

Components Used
The information in this document is based on these software and hardware
versions: 

Cisco CallManager 4.01 with SR2a or later 

Cisco IP phones 7960G and 7940G with phone loads 6.0(4) or later



Jim Koniecki 


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James
Sent: Sunday, December 11, 2005 7:07 PM
To: CiscosupportUpuck
Subject: [cisco-voip] Vt advantage

I'm trying to set up vtadvantage.  But I'm not having luck on enabling
video
on my 7940.   I don't see how to make the vidoe icon appear. And not
sure if
this is related but when I open vtadvantage it does no detect my ip
phone.
I am connected to the ipphone swith
James D. Grace
System Engineer
CCNP CCNA MCSE MCDBA
Digitel Corp.
_______________________________________________
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cisco-voip at puck.nether.net
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------------------------------

Message: 4
Date: Mon, 12 Dec 2005 14:38:14 +1100
From: Greg Maiorana <gmaiorana at ncmail.com.au>
Subject: [cisco-voip] Setting up a constant voice link
To: cisco-voip at puck.nether.net
Message-ID: <439CF0A6.6030302 at ncmail.com.au>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hello All,
       Im trying to work out a solution with a customer to bring their
current pots leased line system to a VoIP solution. The customer
currently has a PSTN leased line from site A to B, which then plugs into
some type of intercom system each end(think of an analog phone on each
end with it off hook all the time).

Is there anyway to do this over VoIP ?
That is have an open link between say 2 x 1721 or 1751 cisco routers
with FXS ports to have an non stop open stream between the FXS port at
each location. So when customer A presses the intercom button to connect
to site B its already open & there is no dialling between the 2 systems.
Any suggestions or idea's would be largely appreciated.

Cheers,

Greg Maiorana.


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

Message: 5
Date: Sun, 11 Dec 2005 23:33:09 -0500
From: Ryan Ratliff <rratliff at cisco.com>
Subject: Re: [cisco-voip] Setting up a constant voice link
To: Greg Maiorana <gmaiorana at ncmail.com.au>
Cc: cisco-voip at puck.nether.net
Message-ID: <30069D7C-F3A2-4F8A-A3ED-2DFA67EEBE0F at cisco.com>
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed

Isn't this hoot and holler?

12.2T docs:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/
products_feature_guide09186a0080087cfc.html

-Ryan

On Dec 11, 2005, at 10:38 PM, Greg Maiorana wrote:

Hello All,
        Im trying to work out a solution with a customer to bring their
current pots leased line system to a VoIP solution. The customer
currently has a PSTN leased line from site A to B, which then plugs into
some type of intercom system each end(think of an analog phone on each
end with it off hook all the time).

Is there anyway to do this over VoIP ?
That is have an open link between say 2 x 1721 or 1751 cisco routers
with FXS ports to have an non stop open stream between the FXS port at
each location. So when customer A presses the intercom button to connect
to site B its already open & there is no dialling between the 2 systems.
Any suggestions or idea's would be largely appreciated.

Cheers,

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



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

Message: 6
Date: Mon, 12 Dec 2005 12:09:08 +0100
From: "Ruben Montes" <Ruben.Montes at eu.didata.com>
Subject: [cisco-voip] Slips errors in a Q.SIG connection
To: <cisco-voip at puck.nether.net>
Message-ID:
	
<D305A0848689344BB98F14A3F13161B710B497 at eubebrusvex.eu.didata.local>
Content-Type: text/plain; charset="us-ascii"

Hello,

 

I have a connection between a PBX MD110 and a Cisco router C2851 running
MGCP. I have Slips errors all the time, counter is increasing every few
seconds. Calls are working well but I have a lot of errors. The MD110 is
configured to supply the clock and this is the config in the router:

 

controller E1 0/0/0

 clock source line primary

 pri-group timeslots 1-31 service mgcp

!

controller E1 0/0/1

 pri-group timeslots 1-31 service mgcp

 

I configured the 'clock source line primary' in one of them but both
QSiGs have the same Slip errors:

 

r2851ctp01#sh controllers e1

E1 0/0/0 is up.

  Applique type is Channelized E1 - balanced

  No alarms detected.

  alarm-trigger is not set

  Version info Firmware: 20050620, FPGA: 16, spm_count = 0

  Framing is CRC4, Line Code is HDB3, Clock Source is Line Primary.

  CRC Threshold is 320. Reported from firmware  is 320.

  Data in current interval (422 seconds elapsed):

     0 Line Code Violations, 0 Path Code Violations

     48 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

     48 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail
Secs

E1 0/0/1 is up.

  Applique type is Channelized E1 - balanced

  No alarms detected.

  alarm-trigger is not set

  Version info Firmware: 20050620, FPGA: 16, spm_count = 0

  Framing is CRC4, Line Code is HDB3, Clock Source is Line.

  CRC Threshold is 320. Reported from firmware  is 320.

  Data in current interval (423 seconds elapsed):

     0 Line Code Violations, 0 Path Code Violations

     48 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins

     48 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail
Secs

 

Any idea? The config in the CCM side is almost the default...

 

Regards,

 

Ruben

 

 

 

  _____  

De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de Leetun, Rob
Enviado el: martes, 06 de diciembre de 2005 18:27
Para: Voll, Scott; Lelio Fulgenzi; Jeffrey C. Ollie;
cisco-voip at puck.nether.net
Asunto: RE: [cisco-voip] CDR Reporting Tool

 

Dido here.  Our Oracle developers built a package in Crystal Reports for
management, which spits out about 10 to 15 different reports each month.


 

Rob

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, December 06, 2005 10:13 AM
To: Lelio Fulgenzi; Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CDR Reporting Tool

 

we used our in house programmers and told them what we needed and they
came up with a custom app for us.  Not web based but it interfaces with
our billing system to automagically send bills each month based on total
toll charges.  took him about 2.5 hours to do.

 

Scott

 

  _____  

From: cisco-voip-bounces at puck.nether.net on behalf of Lelio Fulgenzi
Sent: Tue 12/6/2005 8:32 AM
To: Jeffrey C. Ollie; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR Reporting Tool

I'd also like to hear about such a beast.

 

But I'd also like to hear about people's experience with setting up
their own reports using a custom SQL report tool. I've looked at a few
out there that seem quite good and allow you to create SQL reports with
user input and are web based.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Jeffrey C. Ollie <mailto:jeff at ocjtech.us>  

	To: cisco-voip at puck.nether.net 

	Sent: Tuesday, December 06, 2005 11:26 AM

	Subject: [cisco-voip] CDR Reporting Tool

	 

	Can anyone point me to a CDR reporting tool that:

	 

	1) Will directly access the Cisco CDR database (we've used
Infortel but
	it discards a lot of the information in the Cisco CDR records).
	2) Provides more advanced reporting capabilities than ART
provides.
	3) Specifically targeted at CDR reports, not some generic SQL
reporting
	tool.

	 

	Jeff Ollie

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

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

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End of cisco-voip Digest, Vol 34, Issue 42
******************************************






From sean at northrock.bm  Mon Dec 12 09:17:48 2005
From: sean at northrock.bm (Sean Watkins)
Date: Mon Dec 12 09:17:54 2005
Subject: [cisco-voip] Connect PLAR on ISDN Voice Ports?
Message-ID: <B146219F50F53348A6EF98F14B4202C635D996@sumo2.northrock.bm>

Does anyone know if the connection plar commands works on ISDN Voice-ports?

I want to connect incoming calls on an ISDN line to a number outside a
gateway via sip... It seems to ignore the connect plar command however.

Sean

From jgrace at digitelusa.net  Mon Dec 12 10:07:04 2005
From: jgrace at digitelusa.net (James)
Date: Mon Dec 12 10:06:58 2005
Subject: [cisco-voip] Unity message are cut short
Message-ID: <1657789890-1134400016-cardhu_blackberry.rim.net-21752-@engine09-cell02>

We r using unit 4.0.5 and our when someone leave amessage its getting cut off.  Its about 40 to 45 second.  Now the subcriber mb says 300sec. And the caller get the warning from unity leting them know there time is up
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From JKing at thegranitegroup.com  Mon Dec 12 10:28:47 2005
From: JKing at thegranitegroup.com (King, Jesse)
Date: Mon Dec 12 10:28:54 2005
Subject: [cisco-voip] Unity message are cut short
Message-ID: <80FDD0062521CE4084591182C4DDB6BE011084E3@papaya.thegranitegroup.com>

If you are using MGCP gateways, you need to add the command:

no mgcp timer receive-rtcp

Jesse 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James
Sent: Monday, December 12, 2005 10:07 AM
To: CiscosupportUpuck
Subject: [cisco-voip] Unity message are cut short

We r using unit 4.0.5 and our when someone leave amessage its getting
cut off.  Its about 40 to 45 second.  Now the subcriber mb says 300sec.
And the caller get the warning from unity leting them know there time is
up James D. Grace System Engineer CCNP CCNA MCSE MCDBA Digitel Corp.
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jose.mujica at netxar.com  Mon Dec 12 10:29:51 2005
From: jose.mujica at netxar.com (Jose Mujica)
Date: Mon Dec 12 10:29:54 2005
Subject: [cisco-voip] CallManager 4.1(x) and Equitrac Integration?
Message-ID: <20051212152838.M68801@netxar.com>

Hi.

I have a customer that is currently using the equitrac software for accounting
and billing purposes. It is already integrated with a Legacy PBX using FAC and
CMC codes. We are going to replace the PBX with a CallManager cluster and I
would like to know if anyone out there has done an integration with CCM and
Equitrac and if so how?

The equitrac engineers are asking us to provide a flat file of the CDR
database in order for them to extract the FAC and CMC codes for their
application. Is it possible? If so how?

Thanks in advanced,
-Jose

From mfa at crec.ifas.ufl.edu  Mon Dec 12 10:35:21 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Mon Dec 12 10:35:24 2005
Subject: [cisco-voip] Configuring VoIP on 3750s
References: <200512121139.jBCBat2Z026032@puck.nether.net>
Message-ID: <005701c5ff31$a98eb6f0$400cfb0a@crec.ifas.ufl.edu>

We just installed a WS-C3750-48P and transferred the port configuration from 
two 3524s to it.  One 7940 refuses to come up, stopping with an "Ethernet 
disconnected" message.  We replaced it with another 7940 and all is well.

While investigating, I actually read the 3750 Software Configuration Guide 
for 12.2(25)SED, and it it says "...voice VLAN is not supported on trunk 
ports."  (p. 15-3.) Their example configuration of a 7960 shows only:

  mls qos trust cos
  switchport voice vlan dot1p

whereas we've been using something like

 switchport trunk encapsulation dot1q
 switchport trunk native vlan 103
 switchport trunk allowed vlan 1,2,103,1002-1005
 switchport mode trunk
 switchport voice vlan 2
 spanning-tree portfast

on the 3524s, and in fact this seems to work on all the other phones we've 
connected to the 3750 so far.  I'm willing to accept that this particular 
phone might need some TAC attention, but my main question at this point is: 
If you can't configue the voice VLAN on a trunk port, how do you set the 
VLAN for the switchport on the phone?  We have no intention of extending 
trust past the phone, and have to set different VLANs for several different 
applications.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 


From CORTIZ at broward.org  Mon Dec 12 11:02:32 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon Dec 12 11:02:40 2005
Subject: [cisco-voip] Configuring VoIP on 3750s
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B48@MREXVS01.bc.broward.cty>

Looks like you are using the old style configuration that required the
encapsulation commands, etc. Are you upgrading from the old 3524XL
series? My 3750 ports look like this....


switchport access vlan 10
 switchport mode access
 switchport voice vlan 100
 switchport port-security
 switchport port-security maximum 3
 no ip address
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0 
 mls qos trust device cisco-phone
 mls qos trust cos
 no mdix auto
 auto qos voip cisco-phone 
 spanning-tree portfast


Would this work for you?  

Carlos
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Monday, December 12, 2005 10:35 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Configuring VoIP on 3750s

We just installed a WS-C3750-48P and transferred the port configuration
from 
two 3524s to it.  One 7940 refuses to come up, stopping with an
"Ethernet 
disconnected" message.  We replaced it with another 7940 and all is
well.

While investigating, I actually read the 3750 Software Configuration
Guide 
for 12.2(25)SED, and it it says "...voice VLAN is not supported on trunk

ports."  (p. 15-3.) Their example configuration of a 7960 shows only:

  mls qos trust cos
  switchport voice vlan dot1p

whereas we've been using something like

 switchport trunk encapsulation dot1q
 switchport trunk native vlan 103
 switchport trunk allowed vlan 1,2,103,1002-1005
 switchport mode trunk
 switchport voice vlan 2
 spanning-tree portfast

on the 3524s, and in fact this seems to work on all the other phones
we've 
connected to the 3750 so far.  I'm willing to accept that this
particular 
phone might need some TAC attention, but my main question at this point
is: 
If you can't configue the voice VLAN on a trunk port, how do you set the

VLAN for the switchport on the phone?  We have no intention of extending

trust past the phone, and have to set different VLANs for several
different 
applications.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

_______________________________________________
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From Scott.Voll at wesd.org  Mon Dec 12 11:15:07 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec 12 11:14:54 2005
Subject: [cisco-voip] CM -- Norton Corp 10
Message-ID: <D713462ED535184D830F6F6301753E7E01A11F2D@VISHNU.wesd.org>

OK.  The doc's do NOT include CM support for Norton corp 10.  But my AM
Check and said this, 

"This has passed testing, the web page has not been updated yet
according to one of our Product Managers."

 

Any Cisco guys care to confirm?  Or is anyone using it?

 

THX

 

Scott

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From akoul at dataways.gr  Mon Dec 12 11:57:54 2005
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Mon Dec 12 11:58:17 2005
Subject: [cisco-voip] Attendant Console message "Initialization of Call
	Control Failed..."
Message-ID: <439DAC12.50708@dataways.gr>

Hi All,
i have a CCM 4.1(3) which, according to the Compatibility Matrix, should 
have Attendant Console 1.4(1)ES1. Instead the CCM had 1.4(1)ES2 and i 
have problems in getting the line status unavailable and the message 
"Initialization of Call Control Failed, Retrying". I have loaded the 
latest firmaware on the 7960 phone.
What could be possible wrong?
P.S. I have never manage to install the attendant console at once!!!!!

From pwalenta at wi.rr.com  Mon Dec 12 12:10:37 2005
From: pwalenta at wi.rr.com (pwalenta@wi.rr.com)
Date: Mon Dec 12 12:10:42 2005
Subject: [cisco-voip] Attendant Console message
	"Initialization of Call	Control Failed..."
Message-ID: <e0009e6870.e6870e0009@rdc-kc.rr.com>

The biggest thing to remember when dealing with AC is to create the user
"AC" in DC directory, and then add the phones that need call control to
"AC"'s controlled devices.  Without this, AC cannot access any phones. 
Full instructions on this process can be found here:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803f3a8b.html#wp29392

----- Original Message -----
From: Anthony Kouloglou <akoul at dataways.gr>
Date: Monday, December 12, 2005 10:57 am
Subject: [cisco-voip] Attendant Console message "Initialization of Call
Control Failed..."

> Hi All,
> i have a CCM 4.1(3) which, according to the Compatibility Matrix, 
> should 
> have Attendant Console 1.4(1)ES1. Instead the CCM had 1.4(1)ES2 and 
> i 
> have problems in getting the line status unavailable and the 
> message 
> "Initialization of Call Control Failed, Retrying". I have loaded 
> the 
> latest firmaware on the 7960 phone.
> What could be possible wrong?
> P.S. I have never manage to install the attendant console at once!!!!!
> _______________________________________________
> 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 Dec 12 12:13:47 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec 12 12:13:34 2005
Subject: [cisco-voip] Attendant Console message "Initialization of
	CallControl Failed..."
Message-ID: <D713462ED535184D830F6F6301753E7E01A11F34@VISHNU.wesd.org>


 
1) Check the user called 'ac' and make sure that this user is associated
with the mac address of the
phones that you are using for those users.
2) Make sure that those users have accounts for web attendant
3) Stop and start the TCD service.
4) Make sure that the TCD is started and running properly on all the
call managers
5) Make sure that the IP Phones and the Attendant Console client are
talking to the primary Call manager

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony
Kouloglou
Sent: Monday, December 12, 2005 8:58 AM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] Attendant Console message "Initialization of
CallControl Failed..."

Hi All,
i have a CCM 4.1(3) which, according to the Compatibility Matrix, should

have Attendant Console 1.4(1)ES1. Instead the CCM had 1.4(1)ES2 and i 
have problems in getting the line status unavailable and the message 
"Initialization of Call Control Failed, Retrying". I have loaded the 
latest firmaware on the 7960 phone.
What could be possible wrong?
P.S. I have never manage to install the attendant console at once!!!!!
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From pwalenta at wi.rr.com  Mon Dec 12 12:13:53 2005
From: pwalenta at wi.rr.com (pwalenta@wi.rr.com)
Date: Mon Dec 12 12:13:58 2005
Subject: [cisco-voip] CallManager 4.1(x) and Equitrac Integration?
Message-ID: <e2e3ae3ab5.e3ab5e2e3a@rdc-kc.rr.com>

I have one customer that does this.

There are two available methods for creating a flat-file.

One, you can go into enterprise parameters and tell the CallManager
publisher to keep/generate flat files.  If this is turned on, it is your
responsbility to delete these files otherwise they will fill up the hard
drive.  By default, files are created in 1 minute increments.

The second method involves you pulling CDR records right out of SQL and
generating your own flat file on whatever schedule you'd like.  If you'd
want to go this route, be wary of saturating the publisher CPU with SQL
requests.  Full information on CM 4.1.3 CDR access and formats can be
found here: 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_programming_usage_guide09186a0080405548.html

----- Original Message -----
From: Jose Mujica <jose.mujica at netxar.com>
Date: Monday, December 12, 2005 9:29 am
Subject: [cisco-voip] CallManager 4.1(x) and Equitrac Integration?

> Hi.
> 
> I have a customer that is currently using the equitrac software for 
> accountingand billing purposes. It is already integrated with a 
> Legacy PBX using FAC and
> CMC codes. We are going to replace the PBX with a CallManager 
> cluster and I
> would like to know if anyone out there has done an integration with 
> CCM and
> Equitrac and if so how?
> 
> The equitrac engineers are asking us to provide a flat file of the CDR
> database in order for them to extract the FAC and CMC codes for their
> application. Is it possible? If so how?
> 
> Thanks in advanced,
> -Jose
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

From akoul at dataways.gr  Mon Dec 12 12:15:25 2005
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Mon Dec 12 12:15:47 2005
Subject: [cisco-voip] Attendant Console message "Initialization of Call
	Control Failed..."
In-Reply-To: <e0009e6870.e6870e0009@rdc-kc.rr.com>
References: <e0009e6870.e6870e0009@rdc-kc.rr.com>
Message-ID: <439DB02D.3030109@dataways.gr>

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From mfa at crec.ifas.ufl.edu  Mon Dec 12 13:32:17 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Mon Dec 12 13:32:14 2005
Subject: [cisco-voip] Re: Configuring VoIP on 3750s
References: <200512121702.jBCH1JeZ056418@puck.nether.net>
Message-ID: <00d601c5ff4a$619d51d0$400cfb0a@crec.ifas.ufl.edu>

Yes, our older switches are all either 3524-PWR-XLs or 2900XLs.  I'll try 
your suggestion; thanks.

Mike

> Date: Mon, 12 Dec 2005 11:02:32 -0500
> From: "Ortiz, Carlos" <CORTIZ at broward.org>
> Subject: RE: [cisco-voip] Configuring VoIP on 3750s
> To: "Mike Armstrong" <mfa at ufl.edu>
> Cc: cisco-voip at puck.nether.net
> Message-ID:
> <26B357EF9FB7D04A989C9E82A9E5436E01BC2B48 at MREXVS01.bc.broward.cty>
> Content-Type: text/plain; charset="us-ascii"
>
> Looks like you are using the old style configuration that required the
> encapsulation commands, etc. Are you upgrading from the old 3524XL
> series? My 3750 ports look like this....
>
>
> switchport access vlan 10
> switchport mode access
> switchport voice vlan 100
> switchport port-security
> switchport port-security maximum 3
> no ip address
> srr-queue bandwidth share 10 10 60 20
> srr-queue bandwidth shape  10  0  0  0
> mls qos trust device cisco-phone
> mls qos trust cos
> no mdix auto
> auto qos voip cisco-phone
> spanning-tree portfast
>
>
> Would this work for you?
>
> Carlos
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
> Sent: Monday, December 12, 2005 10:35 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Configuring VoIP on 3750s
>
> We just installed a WS-C3750-48P and transferred the port configuration
> from
> two 3524s to it.  One 7940 refuses to come up, stopping with an
> "Ethernet
> disconnected" message.  We replaced it with another 7940 and all is
> well.
>
> While investigating, I actually read the 3750 Software Configuration
> Guide
> for 12.2(25)SED, and it it says "...voice VLAN is not supported on trunk
>
> ports."  (p. 15-3.) Their example configuration of a 7960 shows only:
>
>  mls qos trust cos
>  switchport voice vlan dot1p
>
> whereas we've been using something like
>
> switchport trunk encapsulation dot1q
> switchport trunk native vlan 103
> switchport trunk allowed vlan 1,2,103,1002-1005
> switchport mode trunk
> switchport voice vlan 2
> spanning-tree portfast
>
> on the 3524s, and in fact this seems to work on all the other phones
> we've
> connected to the 3750 so far.  I'm willing to accept that this
> particular
> phone might need some TAC attention, but my main question at this point
> is:
> If you can't configue the voice VLAN on a trunk port, how do you set the
>
> VLAN for the switchport on the phone?  We have no intention of extending
>
> trust past the phone, and have to set different VLANs for several
> different
> applications.
>
> Mike Armstrong
> UF/IFAS CREC
> Lake Alfred, FL


From tim.reimers at asheville.k12.nc.us  Mon Dec 12 14:46:40 2005
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Mon Dec 12 14:46:16 2005
Subject: [cisco-voip] Symantec antivirus and CCM/Unity 4.0??
Message-ID: <050944EE48FF384F9C663F7048C47DA701237E13@webmail.asheville.k12.nc.us>

Hi folks---
 
I'm hoping someone out there has bookmarks to the 'how to configure'
Symantec Corporate Edition Enterprise (client) 
correctly for both Unity 4.0 and CCM 4.0?
 
I'm not finding even any discussion other than that it's 'supported' in
3.3.....
 
much less anything where Cisco is stating where the Exchange mailbox
store is located in Unity, or the SQL databases are in CCM???
 
Surely there's some fairly comprehensive document that tells what to
exclude from scans, etc, and WHERE those files are?
 
thanks , Tim
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From CORTIZ at broward.org  Mon Dec 12 16:25:14 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon Dec 12 16:25:24 2005
Subject: [cisco-voip] H.323 Dial Peer config
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B5F@MREXVS01.bc.broward.cty>

I have never configured my FXO PSTN router connections with H.323, only
MGCP.  I did configure H.323 for PRI interfaces. Going by that and what
I have refreshed with CCO docs, I have come up with a starting config to
do FXO with H.323. Dial peer 1 would be for the incoming call leg on all
calls.  This assumes a 4 port FXO card.  Does this appear correct?  Have
there been any improvements on the "hunting" for outbound calls beyond
using the preference command? Any suggestions or help would be
appreciated.

 

dial-peer voice 1 pots

 description ** Inbound Call leg all calls **

 incoming called-number .

 direct-inward-dial

 port 0/1/0

!

dial-peer voice 954 pots

 description ** Local 954,754 Calls **

 destination-pattern 9[7,9]54.......

 progress_ind setup enable 3

 progress_ind alert enable 8

 progress_ind progress enable 8

 progress_ind connect enable 8

 direct-inward-dial

 port 0/1/0

 forward-digits all

 

dial-peer voice 9542 pots

 description ** Local 954,754 Calls **

 destination-pattern 9[7,9]54.......

 preference 1

 progress_ind setup enable 3

 progress_ind alert enable 8

 progress_ind progress enable 8

 progress_ind connect enable 8

 direct-inward-dial

 port 0/1/1

 forward-digits all

 

dial-peer voice 9543 pots

 description ** Local 954,754 Calls **

 destination-pattern 9[7,9]54.......

 preference 2

 progress_ind setup enable 3

 progress_ind alert enable 8

 progress_ind progress enable 8

 progress_ind connect enable 8

 direct-inward-dial

 port 0/1/2

 forward-digits all

 

dial-peer voice 9544 pots

 description ** Local 954,754 Calls **

 destination-pattern 9[7,9]54.......

 preference 3

 progress_ind setup enable 3

 progress_ind alert enable 8

 progress_ind progress enable 8

 progress_ind connect enable 8

 direct-inward-dial

 port 0/1/3

 forward-digits all

 

dial-peer voice 9999 voip

 description ** Inbound 357 nums to mr-cm-sub1 **

 destination-pattern [5-8]...

 progress_ind setup enable 3

 session target ipv4:10.100.200.21

 incoming called-number .

 dtmf-relay h245-alphanumeric

 codec g711ulaw

!

dial-peer voice 99992 voip

 description ** Inbound 357 nums to mr-cm-pub **

 preference 2

 destination-pattern [5-8]...

 progress_ind setup enable 3

 session target ipv4:10.100.200.20

 incoming called-number .

 dtmf-relay h245-alphanumeric

 codec g711ulaw

 

 

Carlos Ortiz

Senior Information Technology Specialist

Broward County Office of Information Technology

Data Communications

954-357-7981

cortiz at broward.org

 

 

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From jgrace at digitelusa.net  Mon Dec 12 18:53:52 2005
From: jgrace at digitelusa.net (James)
Date: Mon Dec 12 18:53:46 2005
Subject: [cisco-voip] Reinstalling cra don't ask me for monitor
Message-ID: <1895169109-1134431624-cardhu_blackberry.rim.net-29083-@engine09-cell02>

I was told to reinstall crs to config the monitor server.  But the install did not ask me for the ip. Of the second port 
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From erickbe at yahoo.com  Mon Dec 12 22:37:52 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon Dec 12 22:37:59 2005
Subject: [cisco-voip] H.323 Dial Peer config
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B5F@MREXVS01.bc.broward.cty>
Message-ID: <20051213033752.99278.qmail@web34305.mail.mud.yahoo.com>

On the dial-peers, you are forwarding all digits so
that will also send the 9 that starts all your
destination patterns. 

Usually with FXO, there is no inbound caller id to
tell it where to route call so on the voice-port you
need to use connection plar to route call somewhere
(This is same as Attendant DN setting on the endpoint
in CCM when using MGCP). 

They have added trunk groups to IOS, so you can assign
voice ports to a trunkgroup, then have one dial-peer
pointing to the trunk group. You can then assign
preferences to voice-port that is a member of the
trunk group, etc. This requires a trunk group to be
configured globally. Under the trunkgroup config you
can configure hunt method, etc. 

Erick

--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> I have never configured my FXO PSTN router
> connections with H.323, only
> MGCP.  I did configure H.323 for PRI interfaces.
> Going by that and what
> I have refreshed with CCO docs, I have come up with
> a starting config to
> do FXO with H.323. Dial peer 1 would be for the
> incoming call leg on all
> calls.  This assumes a 4 port FXO card.  Does this
> appear correct?  Have
> there been any improvements on the "hunting" for
> outbound calls beyond
> using the preference command? Any suggestions or
> help would be
> appreciated.
> 
>  
> 
> dial-peer voice 1 pots
> 
>  description ** Inbound Call leg all calls **
> 
>  incoming called-number .
> 
>  direct-inward-dial
> 
>  port 0/1/0
> 
> !
> 
> dial-peer voice 954 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/0
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9542 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  preference 1
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/1
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9543 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  preference 2
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/2
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9544 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  preference 3
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/3
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9999 voip
> 
>  description ** Inbound 357 nums to mr-cm-sub1 **
> 
>  destination-pattern [5-8]...
> 
>  progress_ind setup enable 3
> 
>  session target ipv4:10.100.200.21
> 
>  incoming called-number .
> 
>  dtmf-relay h245-alphanumeric
> 
>  codec g711ulaw
> 
> !
> 
> dial-peer voice 99992 voip
> 
>  description ** Inbound 357 nums to mr-cm-pub **
> 
>  preference 2
> 
>  destination-pattern [5-8]...
> 
>  progress_ind setup enable 3
> 
>  session target ipv4:10.100.200.20
> 
>  incoming called-number .
> 
>  dtmf-relay h245-alphanumeric
> 
>  codec g711ulaw
> 
>  
> 
>  
> 
> Carlos Ortiz
> 
> Senior Information Technology Specialist
> 
> Broward County Office of Information Technology
> 
> Data Communications
> 
> 954-357-7981
> 
> cortiz at broward.org
> 
>  
> 
>  
> 
> > _______________________________________________
> 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 ciscozest at gmail.com  Mon Dec 12 23:19:14 2005
From: ciscozest at gmail.com (Robin H)
Date: Mon Dec 12 23:19:17 2005
Subject: [cisco-voip] Function of VM Pilot CSS
Message-ID: <f99cc3f60512122019j6827d8d9hee807a3b78113d1f@mail.gmail.com>

Hi, Can anyone explain to me the function of CSS on VM Pilot configuration
page? I set my VM ports and internal phones in a unique partition and CSS.
So far has no problem accessing VM. Thanks

regards,
Robin
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From lelio at uoguelph.ca  Mon Dec 12 23:38:05 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon Dec 12 23:38:12 2005
Subject: [cisco-voip] Function of VM Pilot CSS
References: <f99cc3f60512122019j6827d8d9hee807a3b78113d1f@mail.gmail.com>
Message-ID: <004d01c5ff9f$028daf50$c901a8c0@EATIUSBIRDIUS>

I won't take credit for this one because Wes straightened it out when I was having a problem. I hope he'll correct me if I've got it wrong.

The VM Pilot CCS is used when you check the VM box on the line configuration page. If you put anything in the destination field, then it uses the CSS you've selected.

You'll have to check to see what gets checked if you push FwdAll and then the messages button and/or your vm pilot number.

  ----- Original Message ----- 
  From: Robin H 
  To: cisco-voip at puck.nether.net 
  Sent: Monday, December 12, 2005 11:19 PM
  Subject: [cisco-voip] Function of VM Pilot CSS


  Hi, Can anyone explain to me the function of CSS on VM Pilot configuration page? I set my VM ports and internal phones in a unique partition and CSS. So far has no problem accessing VM. Thanks

  regards,
  Robin


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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Joe_Bell at adp.com  Tue Dec 13 00:41:50 2005
From: Joe_Bell at adp.com (Bell, Joe)
Date: Tue Dec 13 00:49:19 2005
Subject: [cisco-voip] CRA JTAPI redirect clarification
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3B2B611C@EXCHANGEDS01.ds.ad.adp.com>

Hello All,

 

I was wondering if anyone could explain, or send me a link, that
properly describes the three options listed on the redirection property
of the JTAPI Call Control Group Configuration.  At the bottom of this
page is a selection titled "Calling Search Space for ReDirect".  Here
are the three types and the Help page definitions.

 

*	Call manager default-The redirect uses the search space that is
the default for the implementation.   ??
*	Calling party-The redirect uses the search space of the calling
address. 
*	Redirecting party-The redirect uses the search space of the
redirect controller's address. 

I am sure the Redirecting party is the CTI ports css.   The other two I
am not completely sure.  I believe the Calling party is a dn that is
registered to CCM (not external to the system), or is it a gateways' css
were the call enters the system?  And I believe the Call Manager Default
is the cti route points css (not the ports).  

 

Thanks a lot!,

 

Joe Bell



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 robin.inderberg at candidator.se  Tue Dec 13 05:43:55 2005
From: robin.inderberg at candidator.se (Robin Inderberg)
Date: Tue Dec 13 05:38:40 2005
Subject: [cisco-voip] Installation of Subscriber
Message-ID: <39EB1FE59758D9448A5D0B348857C727394E8E@cannt04.candidator.se>

Good morning,

 

I've had a bit of trouble this night with the installation of a
subscriber. 

 

The publisher has CCM 4.1(3), and now it has MSSQL2k SP4, the same as
the subscriber has.

But the installation stops when It comes to the CCM installation.

 

I thought the problem were that the CM01 didn't allow mixed mode
connections, and I guess the installation tries to connect using the SA
account?

 

When I check the logfile called, CCMDBSetup 12-13-2005 10.51.47.log, it
says like this.

 

 

12-13-2005 10:51:47 Starting CallManager Installation

12-13-2005 10:51:47 Setup Registry for the install status = 1

12-13-2005 10:51:47 Checking SQL version...

12-13-2005 10:51:50 Executing CReplaceString function...

12-13-2005 10:51:52 Return Code = 0

12-13-2005 10:51:52 Starting MSSQLServer service...

12-13-2005 10:52:20 Return Code = 0

12-13-2005 10:52:20 Starting SQLServerAgent service...

12-13-2005 10:52:24 Return Code = 0

12-13-2005 10:52:24 Starting MSDTC service...

12-13-2005 10:52:25 Return Code = 0

12-13-2005 10:52:25 Getting Local host name from the registry...

12-13-2005 10:52:25 Reading data from response file for Publisher and
Database...

12-13-2005 10:52:25 gbPublishGlassHouse: [0]

12-13-2005 10:52:25 gbPullSubscription: [1]

12-13-2005 10:52:25 szGlassServer: [SINV-IPLA-CM-01]

12-13-2005 10:52:25 szGlassDB [CCM0301]

12-13-2005 10:52:25 Checking the Database Connection....

12-13-2005 10:52:25 _DBGetVersion: SINV-IPLA-CM-01 CCM0301  ******** 

12-13-2005 10:52:42 Subscriber has problem to connect to the Publisher
Database.

 

 

I have tested to connect using the Enterprise Manager, and it works both
with SA and Windows Authentication. 

Should I manually configure anything with ODBC? 

 

 

The SQL version is on both servers, Microsoft SQL Server  2000 -
8.00.2039 (Intel X86)   May  3 2005 23:18:38.

 

 

 

Best Regards

 

 

Robin Inderberg

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From lex at sandy.ru  Tue Dec 13 05:46:41 2005
From: lex at sandy.ru (Alex A. Pavlenko)
Date: Tue Dec 13 05:46:46 2005
Subject: [cisco-voip] Connect PLAR on ISDN Voice Ports?
References: <B146219F50F53348A6EF98F14B4202C635D996@sumo2.northrock.bm>
Message-ID: <02c101c5ffd2$8054e6f0$19477dc1@corporate.sandy.ru>

Hello

I use it on BRI port. Can't say anything about PRI. But I think
connection plar on digital ports is ugly approach. Use
direct-inward-dial in conjuction with voip dial-peer instead.
I have to use connection plar since my telco does not send
called number in the incoming Q.931 Setup message :(

--
Alex Pavlenko
System Administrator
Sandy Info

----- Original Message ----- 
From: "Sean Watkins" <sean at northrock.bm>
To: <cisco-voip at puck.nether.net>
Sent: Monday, December 12, 2005 5:17 PM
Subject: [cisco-voip] Connect PLAR on ISDN Voice Ports?


> Does anyone know if the connection plar commands works on ISDN 
> Voice-ports?
>
> I want to connect incoming calls on an ISDN line to a number outside a
> gateway via sip... It seems to ignore the connect plar command however.
>
> Sean
> _______________________________________________
> 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 Dec 13 06:33:16 2005
From: Ruben.Montes at eu.didata.com (Ruben Montes)
Date: Tue Dec 13 06:33:20 2005
Subject: [cisco-voip] QoS in distributed cluster
Message-ID: <D305A0848689344BB98F14A3F13161B710B65A@eubebrusvex.eu.didata.local>

	Hello,

	 

	I have a distributed cluster. The publisher is located in one
city and the subscriber is in another one. The network is a LAN-to-LAN
with all L2 interfaces.

	In the best case, there are two 100MB hops between the
CallManagers, and in the worst case (redundant path) two 10MB hops. I'm
a little bit worried about that and I would like to configure QoS. We
have a bunch of devices between the CCMs:

	-         C6500 with sup720

	-         C4500 with 4232L3

	-         C3548

	-         C4506

	 

	My question is if it enough to mark the interesting traffic and
map it to a high priority egress queue in all the path, or if it is
necessary to agreggate policiers to limit the traffc.

	 

	If I had a WAN I would configure CWFQ, but in this case, which
do you think is the best option?

	 

	Regards,

	 

	Ruben

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From Ruben.Montes at eu.didata.com  Tue Dec 13 07:17:19 2005
From: Ruben.Montes at eu.didata.com (Ruben Montes)
Date: Tue Dec 13 07:17:40 2005
Subject: [cisco-voip] QoS in distributed cluster
Message-ID: <D305A0848689344BB98F14A3F13161B710B686@eubebrusvex.eu.didata.local>

Thanks for you response.

Yes, I'm using L3 in the devices: MSFC3 in the 6500, the 4232L3 in a
4006 with CatOs and a C4506 with L3.
Now, my RTT aprox is 12ms, but I would like to configure QoS in all the
network to have all the voice and intra-cluster traffic controlled.

I would use autoqos in the C45006 because this is the only device that
supports that functionality. I would do a similar mapping in the 4000
and the 6500. Do you think it would be enough?

Regards,

Ruben

-----Mensaje original-----
De: Ray Burkholder [mailto:ray at oneunified.net] 
Enviado el: martes, 13 de diciembre de 2005 13:10
Para: Ruben Montes
Asunto: Re: [cisco-voip] QoS in distributed cluster

Hi There,

As an AVVID practitioner, I would highly recommend you review Cisco's
DQOS
course notes and text book (if you havn't already).  They explain the
queueing
structures available in switches and how to configure them in light of
prioritizing prec 3 and prec 5 traffic.  In summary though, with
properly
configured switch QOS, you would be making use of 2 or 4 or more queues
in each
switch type, and they take care of queueing requirements.  I find the
auto-qos
functions in each switch type very handy in figuring out the
peculiarities of
each switch.

In addition, for the WAN or MAN or WLAN that you have, are you using
layer 3
switching in the devices?  Recent Cisco best practices indicate that
alot of
segmentation is a good thing.

One more thing, for distributed clusters, Cisco requires that one way
between
callmanagers must not be more than 20ms, or 40ms round trip.

Many of the Cisco devices have SLA/SAA/RTT capabilities so you can
monitor this
in real time.

In addition, by using some Cricket addons, you can monitor your network
wide QOS
statistics quite readily for delays, jitter, drops, and losses.

Quoting Ruben Montes <Ruben.Montes at eu.didata.com>:

> 	Hello,
> 
> 	 
> 
> 	I have a distributed cluster. The publisher is located in one
> city and the subscriber is in another one. The network is a LAN-to-LAN
> with all L2 interfaces.
> 
> 	In the best case, there are two 100MB hops between the
> CallManagers, and in the worst case (redundant path) two 10MB hops.
I'm
> a little bit worried about that and I would like to configure QoS. We
> have a bunch of devices between the CCMs:
> 
> 	-         C6500 with sup720
> 
> 	-         C4500 with 4232L3
> 
> 	-         C3548
> 
> 	-         C4506
> 
> 	 
> 
> 	My question is if it enough to mark the interesting traffic and
> map it to a high priority egress queue in all the path, or if it is
> necessary to agreggate policiers to limit the traffc.
> 
> 	 
> 
> 	If I had a WAN I would configure CWFQ, but in this case, which
> do you think is the best option?
> 
> 	 
> 
> 	Regards,
> 
> 	 
> 
> 	Ruben
> 
> 
> -- 
> Scanned for viruses and dangerous content at 
> http://www.oneunified.net and is believed to be clean.
> 
> 


-- 
Ray Burkholder
http://www.oneunified.net
ray at oneunified.net
441 505 7293

-------------------------------------------------
Sent from http://www.oneunified.net via IMP: http://horde.org/imp/

-- 
Scanned for viruses and dangerous content at 
http://www.oneunified.net and is believed to be clean.




From Brian.Henry at apptis.com  Tue Dec 13 07:18:15 2005
From: Brian.Henry at apptis.com (Brian Henry)
Date: Tue Dec 13 07:21:27 2005
Subject: [cisco-voip] QoS in distributed cluster
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC03D784@CH-EXCH-02.intelltech.apptis.inc>

I would mark your traffic like you say, of course have priority for voice and set the rest of the classes by Cisco docs when it comes to control traffic and video.  The rest like SMTP, SQL, telnet, etc.. you will have to determine that.  Then just make sure you have the proper queuing depths configured on your egress and ingress.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ruben Montes
Sent: Tue 12/13/2005 6:33 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] QoS in distributed cluster



	Hello,

	 

	I have a distributed cluster. The publisher is located in one city and the subscriber is in another one. The network is a LAN-to-LAN with all L2 interfaces.

	In the best case, there are two 100MB hops between the CallManagers, and in the worst case (redundant path) two 10MB hops. I'm a little bit worried about that and I would like to configure QoS. We have a bunch of devices between the CCMs:

	-         C6500 with sup720

	-         C4500 with 4232L3

	-         C3548

	-         C4506

	 

	My question is if it enough to mark the interesting traffic and map it to a high priority egress queue in all the path, or if it is necessary to agreggate policiers to limit the traffc.

	 

	If I had a WAN I would configure CWFQ, but in this case, which do you think is the best option?

	 

	Regards,

	 

	Ruben

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From pwalenta at wi.rr.com  Tue Dec 13 07:44:07 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue Dec 13 07:44:02 2005
Subject: [cisco-voip] QoS in distributed cluster
In-Reply-To: <D305A0848689344BB98F14A3F13161B710B65A@eubebrusvex.eu.didata.local>
Message-ID: <200512131243.jBDChs9u019014@ms-smtp-04.rdc-kc.rr.com>

I would highly recommend reading the CallManager 4.0/4.1 SRND here:
 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation
_design_guide_book09186a008044714e.html
 
and the QoS SRND here:
 
http://www.cisco.com/univercd/cc/td/doc/solution/esm/qossrnd.pdf
 
They go over many of the questions you ask.
 
The base configuration for CallManager recommends 900k between CallManager
pub and sub per 10,000 bhca.
 
The one thing that concerns me about your setup listed below is that it is
all L2.  One spanning tree problem (or even simple reconvergence) could
seriously mess up your voice network.  I'd consider some readdressing and a
move to L3 between your "WAN" links to help prevent problems, and you
definitely have the hardware to handle this.
 
I don't recommend using policing for any voice traffic (particularly
CallManager).  Marking voice and CallManager traffic according to the SRND
has been enough for me in the past.  In your case with an all-switch
network, the recommended queue depths on the 100MB links should be
sufficient (assuming latency is currently good) .


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruben Montes
Sent: Tuesday, December 13, 2005 5:33 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] QoS in distributed cluster



Hello,

 

I have a distributed cluster. The publisher is located in one city and the
subscriber is in another one. The network is a LAN-to-LAN with all L2
interfaces.

In the best case, there are two 100MB hops between the CallManagers, and in
the worst case (redundant path) two 10MB hops. I'm a little bit worried
about that and I would like to configure QoS. We have a bunch of devices
between the CCMs:

-         C6500 with sup720

-         C4500 with 4232L3

-         C3548

-         C4506

 

My question is if it enough to mark the interesting traffic and map it to a
high priority egress queue in all the path, or if it is necessary to
agreggate policiers to limit the traffc.

 

If I had a WAN I would configure CWFQ, but in this case, which do you think
is the best option?

 

Regards,

 

Ruben

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From ciscozest at gmail.com  Tue Dec 13 07:48:28 2005
From: ciscozest at gmail.com (Ciscozest)
Date: Tue Dec 13 07:48:32 2005
Subject: [cisco-voip] Function of VM Pilot CSS
References: <f99cc3f60512122019j6827d8d9hee807a3b78113d1f@mail.gmail.com>
	<004d01c5ff9f$028daf50$c901a8c0@EATIUSBIRDIUS>
Message-ID: <00d901c5ffe3$86930bd0$6301a8c0@windfall99>

Hi,

Here is my setting detail:
Phone A: Partition = "internal", CSS = "Site1-all", CFB/CFNA to VM.
VM ports: Partition = "internal", CSS = "Site1-all".
VM Pilot: CSS = "none"

Analysis:
- I can access VM via message button
- call from phone B to A can be diverted to VM when is not answered. 
- Unity can dialout to internal and external number.

in summary I don;t see the funtionality of CSS in VM Pilot Conifguration page. I thought setting "none" CSS would restrict it from dialing a configured partition(s). I use CM 3.3(5)sr1. Anyone can enlighten me? thanks

regards,
Robin
  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: Robin H ; cisco-voip at puck.nether.net 
  Sent: Tuesday, December 13, 2005 12:38 PM
  Subject: Re: [cisco-voip] Function of VM Pilot CSS


  I won't take credit for this one because Wes straightened it out when I was having a problem. I hope he'll correct me if I've got it wrong.

  The VM Pilot CCS is used when you check the VM box on the line configuration page. If you put anything in the destination field, then it uses the CSS you've selected.

  You'll have to check to see what gets checked if you push FwdAll and then the messages button and/or your vm pilot number.

    ----- Original Message ----- 
    From: Robin H 
    To: cisco-voip at puck.nether.net 
    Sent: Monday, December 12, 2005 11:19 PM
    Subject: [cisco-voip] Function of VM Pilot CSS


    Hi, Can anyone explain to me the function of CSS on VM Pilot configuration page? I set my VM ports and internal phones in a unique partition and CSS. So far has no problem accessing VM. Thanks

    regards,
    Robin


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


    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip
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From ray at oneunified.net  Tue Dec 13 07:51:10 2005
From: ray at oneunified.net (Ray Burkholder)
Date: Tue Dec 13 07:52:52 2005
Subject: [cisco-voip] QoS in distributed cluster
In-Reply-To: <D305A0848689344BB98F14A3F13161B710B686@eubebrusvex.eu.didata.local>
References: <D305A0848689344BB98F14A3F13161B710B686@eubebrusvex.eu.didata.local>
Message-ID: <1134478270.439ec3beb2e45@sp01.oneunified.net>

As they say, 'it depends'.  If your links are saturated, you have to pay extra
attention to how you play with your traffic.  If not, then it isn't quite so
bad.  Are you running more than 10 mbps thorugh your primary links?  If so,
then you really need to pay extra attention to the
class-map/policy-map/service-policy configurations for bandwidth, policing, and
shaping on the devices in the 10mbps links.  Again Cisco's DQOS books go into a
fair amount of detail on how to do this, especially with how to deal with
congestion in networks.  As experiementation, you can even rate-limit your
100mbps links to 10mbps to simulate what will happen on your 10mbps links.

You've also checked that your WAN links are all full-duplex and are configured
as such, and are working with zero errors?

The idea though is that you can not 'set and forget'.  You have to monitor and
test your configurations.  Commands like 'sho policy-map interface ..' help you
identify what is happening with the various queues.

When setting priority queues, various switches have various number of egress
queues, with the priorty queue not always being the same.  You'll have to did
through Cisco's documentation to work out the specifics.  Some of the DQOS
course notes help with a number of your switch types.

Using NBAR and SLA/SAA in conjunction with Cricket and some third party config
help you monitor in-depth the QoS portions of your configs to see what is
getting through and what is getting dropped.


Quoting Ruben Montes <Ruben.Montes at eu.didata.com>:

> Thanks for you response.
> 
> Yes, I'm using L3 in the devices: MSFC3 in the 6500, the 4232L3 in a
> 4006 with CatOs and a C4506 with L3.
> Now, my RTT aprox is 12ms, but I would like to configure QoS in all the
> network to have all the voice and intra-cluster traffic controlled.
> 
> I would use autoqos in the C45006 because this is the only device that
> supports that functionality. I would do a similar mapping in the 4000
> and the 6500. Do you think it would be enough?
> 
> Regards,
> 
> Ruben
> 
> -----Mensaje original-----
> De: Ray Burkholder [mailto:ray at oneunified.net] 
> Enviado el: martes, 13 de diciembre de 2005 13:10
> Para: Ruben Montes
> Asunto: Re: [cisco-voip] QoS in distributed cluster
> 
> Hi There,
> 
> As an AVVID practitioner, I would highly recommend you review Cisco's
> DQOS
> course notes and text book (if you havn't already).  They explain the
> queueing
> structures available in switches and how to configure them in light of
> prioritizing prec 3 and prec 5 traffic.  In summary though, with
> properly
> configured switch QOS, you would be making use of 2 or 4 or more queues
> in each
> switch type, and they take care of queueing requirements.  I find the
> auto-qos
> functions in each switch type very handy in figuring out the
> peculiarities of
> each switch.
> 
> In addition, for the WAN or MAN or WLAN that you have, are you using
> layer 3
> switching in the devices?  Recent Cisco best practices indicate that
> alot of
> segmentation is a good thing.
> 
> One more thing, for distributed clusters, Cisco requires that one way
> between
> callmanagers must not be more than 20ms, or 40ms round trip.
> 
> Many of the Cisco devices have SLA/SAA/RTT capabilities so you can
> monitor this
> in real time.
> 
> In addition, by using some Cricket addons, you can monitor your network
> wide QOS
> statistics quite readily for delays, jitter, drops, and losses.
> 
> Quoting Ruben Montes <Ruben.Montes at eu.didata.com>:
> 
> > 	Hello,
> > 
> > 	 
> > 
> > 	I have a distributed cluster. The publisher is located in one
> > city and the subscriber is in another one. The network is a LAN-to-LAN
> > with all L2 interfaces.
> > 
> > 	In the best case, there are two 100MB hops between the
> > CallManagers, and in the worst case (redundant path) two 10MB hops.
> I'm
> > a little bit worried about that and I would like to configure QoS. We
> > have a bunch of devices between the CCMs:
> > 
> > 	-         C6500 with sup720
> > 
> > 	-         C4500 with 4232L3
> > 
> > 	-         C3548
> > 
> > 	-         C4506
> > 
> > 	 
> > 
> > 	My question is if it enough to mark the interesting traffic and
> > map it to a high priority egress queue in all the path, or if it is
> > necessary to agreggate policiers to limit the traffc.
> > 
> > 	 
> > 
> > 	If I had a WAN I would configure CWFQ, but in this case, which
> > do you think is the best option?
> > 
> > 	 
> > 
> > 	Regards,
> > 
> > 	 
> > 
> > 	Ruben
> > 
> > 
> > -- 
> > Scanned for viruses and dangerous content at 
> > http://www.oneunified.net and is believed to be clean.
> > 
> > 
> 
> 
> -- 
> Ray Burkholder
> http://www.oneunified.net
> ray at oneunified.net
> 441 505 7293
> 
> -------------------------------------------------
> Sent from http://www.oneunified.net via IMP: http://horde.org/imp/
> 
> -- 
> 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.
> 
> 


-- 
Ray Burkholder
http://www.oneunified.net
ray at oneunified.net
441 505 7293

-------------------------------------------------
Sent from http://www.oneunified.net via IMP: http://horde.org/imp/

-- 
Scanned for viruses and dangerous content at 
http://www.oneunified.net and is believed to be clean.


From hsahady at free.fr  Tue Dec 13 08:43:44 2005
From: hsahady at free.fr (sahady)
Date: Tue Dec 13 08:44:02 2005
Subject: [cisco-voip] debit card call application: number of digits doesn't
	match cardlen 
Message-ID: <20051213134351.E623D20002B9@mwinf1003.wanadoo.fr>

Hello

I trying to setup a prepaid card application on cisco AS5350, and I got a
played message ''you have entered an invalid number of digits please
re-enter your card number.."  once the digits was collected.

 

By looking the debit card call sequence the error is related to the number
of collected digit is not match with the card length digit. I setup
different length with parameter call application mycard uid-len without
success. I don't know where the application looking for the card length.

 

Hard AS5350

IOS     12.3.13a

Debitcard: app_debitcard.2.0.2.8.tcl

 

Ps: debug voip ivr digit: talk about vcr disable?

 

 

Regards.

hamdouna

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From lelio at uoguelph.ca  Tue Dec 13 08:51:33 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Dec 13 08:51:38 2005
Subject: [cisco-voip] Function of VM Pilot CSS
References: <f99cc3f60512122019j6827d8d9hee807a3b78113d1f@mail.gmail.com>
	<004d01c5ff9f$028daf50$c901a8c0@EATIUSBIRDIUS>
	<00d901c5ffe3$86930bd0$6301a8c0@windfall99>
Message-ID: <003b01c5ffec$53f46ee0$30196883@cfs.uoguelph.ca>

Wes also mentioned this: 
  When you check the 'Voicemail' box for CFB/CFNA, we attempt to use the CSS of the Voicemail Pilot by default.  If that is blank then we revert to using the CSS of the originating device.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Ciscozest 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Tuesday, December 13, 2005 7:48 AM
  Subject: Re: [cisco-voip] Function of VM Pilot CSS


  Hi,

  Here is my setting detail:
  Phone A: Partition = "internal", CSS = "Site1-all", CFB/CFNA to VM.
  VM ports: Partition = "internal", CSS = "Site1-all".
  VM Pilot: CSS = "none"

  Analysis:
  - I can access VM via message button
  - call from phone B to A can be diverted to VM when is not answered. 
  - Unity can dialout to internal and external number.

  in summary I don;t see the funtionality of CSS in VM Pilot Conifguration page. I thought setting "none" CSS would restrict it from dialing a configured partition(s). I use CM 3.3(5)sr1. Anyone can enlighten me? thanks

  regards,
  Robin
    ----- Original Message ----- 
    From: Lelio Fulgenzi 
    To: Robin H ; cisco-voip at puck.nether.net 
    Sent: Tuesday, December 13, 2005 12:38 PM
    Subject: Re: [cisco-voip] Function of VM Pilot CSS


    I won't take credit for this one because Wes straightened it out when I was having a problem. I hope he'll correct me if I've got it wrong.

    The VM Pilot CCS is used when you check the VM box on the line configuration page. If you put anything in the destination field, then it uses the CSS you've selected.

    You'll have to check to see what gets checked if you push FwdAll and then the messages button and/or your vm pilot number.

      ----- Original Message ----- 
      From: Robin H 
      To: cisco-voip at puck.nether.net 
      Sent: Monday, December 12, 2005 11:19 PM
      Subject: [cisco-voip] Function of VM Pilot CSS


      Hi, Can anyone explain to me the function of CSS on VM Pilot configuration page? I set my VM ports and internal phones in a unique partition and CSS. So far has no problem accessing VM. Thanks

      regards,
      Robin


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


      _______________________________________________
      cisco-voip mailing list
      cisco-voip at puck.nether.net
      https://puck.nether.net/mailman/listinfo/cisco-voip
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From james.dust at charles-stanley.co.uk  Tue Dec 13 09:03:17 2005
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Tue Dec 13 09:03:28 2005
Subject: [cisco-voip] Unity Out of office message
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AE2A72F2@CSEXCH01.charles-stanley.co.uk>

We have a branch office that is using the Unity Voicemail app, they
would like the ability to be able to have a recorded announcement saying
that they are are out of the office over the Christmas holidays but
would stop the caller leaving a message.
 
Has anyone done this?
 
Regards.
 
James Dust
Network & Communications
Charles Stanley & Co.
25 Luke Street, London EC2A 4AR
Tel: 020-7953 2314 - Fax: 020-7150 9314
www.charles-stanley.co.uk <http://www.charles-stanley.co.uk/>  
 
***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk 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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
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This footnote also confirms that this email message has been swept by McAfee
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From lelio at uoguelph.ca  Tue Dec 13 09:08:29 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Dec 13 09:08:34 2005
Subject: [cisco-voip] Unity Out of office message
References: <FDEEDC07A7DA6E41BCB819487E7C85AE2A72F2@CSEXCH01.charles-stanley.co.uk>
Message-ID: <006101c5ffee$b1be1150$30196883@cfs.uoguelph.ca>

As far as I know there is no way to configure a mailbox to stop accepting messages. You can modify the after greeting action to say goodbye or something like that, but that does not prevent subscribers from sending messages to this mailbox. You can also investigate modifying the quotas to prevent delivery, but I faintly recall there was a side effect to that as well.
 
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: James Dust 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, December 13, 2005 9:03 AM
  Subject: [cisco-voip] Unity Out of office message


  We have a branch office that is using the Unity Voicemail app, they would like the ability to be able to have a recorded announcement saying that they are are out of the office over the Christmas holidays but would stop the caller leaving a message.

  Has anyone done this?

  Regards.

  James Dust
  Network & Communications
  Charles Stanley & Co.
  25 Luke Street, London EC2A 4AR
  Tel: 020-7953 2314 - Fax: 020-7150 9314
  www.charles-stanley.co.uk 




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

        The information contained in this e-mail is strictly confidential, some or all
        of which may be legally privileged. It is for the intended recipient only.
        Access to this e-mail by any other person is prohibited. If you are not the
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        replying to or any action taken or omitted to be taken in reliance on this
        e-mail, is prohibited and may be unlawful. Please contact the sender immediately
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        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 accept that there
        may be a delay in processing the instructions received from e-mails after
        Charles Stanley has received them. You are advised that urgent, time sensitive
        and confidential communications should not be sent by e-mail. 

        You acknowledge that e-mails are not secure and you accept the risk of
        malfunction, viruses, unauthorised interference, mis-delivery or delay.
        ************************************************************************************************


        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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

        Authorised and Regulated by the Financial Services Authority, Member of the
        London Stock Exchange, International Securities Markets 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
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From Scott.Voll at wesd.org  Tue Dec 13 09:13:52 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 13 09:13:41 2005
Subject: [cisco-voip] Unity Out of office message
Message-ID: <D713462ED535184D830F6F6301753E7E01A11F71@VISHNU.wesd.org>

Under greetings --> After Greeting can't you -->send caller to -->  hang
up?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Dust
Sent: Tuesday, December 13, 2005 6:03 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Out of office message
Importance: High

 

We have a branch office that is using the Unity Voicemail app, they
would like the ability to be able to have a recorded announcement saying
that they are are out of the office over the Christmas holidays but
would stop the caller leaving a message.

 

Has anyone done this?

 

Regards.

 

James Dust
Network & Communications
Charles Stanley & Co.
25 Luke Street, London EC2A 4AR
Tel: 020-7953 2314 - Fax: 020-7150 9314
www.charles-stanley.co.uk <http://www.charles-stanley.co.uk/>  

 

 


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

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets 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 JHorton at jackhenry.com  Tue Dec 13 09:35:15 2005
From: JHorton at jackhenry.com (Josh Horton)
Date: Tue Dec 13 09:37:02 2005
Subject: [cisco-voip] Unity Out of office message
Message-ID: <13591CB4B2359F4D96E2606DD6B840A00211B96A@smoxchg.jhacorp.com>

Is the branch using DIDs, a receptionist or a combination of both?

 

Josh

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, December 13, 2005 8:14 AM
To: James Dust; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity Out of office message

 

Under greetings --> After Greeting can't you -->send caller to -->  hang
up?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Dust
Sent: Tuesday, December 13, 2005 6:03 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Out of office message
Importance: High

 

We have a branch office that is using the Unity Voicemail app, they
would like the ability to be able to have a recorded announcement saying
that they are are out of the office over the Christmas holidays but
would stop the caller leaving a message.

 

Has anyone done this?

 

Regards.

 

James Dust
Network & Communications
Charles Stanley & Co.
25 Luke Street, London EC2A 4AR
Tel: 020-7953 2314 - Fax: 020-7150 9314
www.charles-stanley.co.uk <http://www.charles-stanley.co.uk/>  

 

 


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

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets 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.

 




NOTICE: This electronic mail message and any files transmitted with it are intended exclusively for the individual or entity to which it is addressed. The message, together with any attachment, may contain confidential and/or privileged information. Any unauthorized review, use, printing, saving, copying, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies.



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From james.dust at charles-stanley.co.uk  Tue Dec 13 09:48:22 2005
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Tue Dec 13 09:48:35 2005
Subject: [cisco-voip] Unity Out of office message
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AE2A72F4@CSEXCH01.charles-stanley.co.uk>

Combination of both, it is however the main number they would like the
recorded message on.
 
What Scott suggested worked though.
 
James.

________________________________

From: Josh Horton [mailto:JHorton at jackhenry.com] 
Sent: 13 December 2005 14:35
To: Voll, Scott; James Dust; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity Out of office message



Is the branch using DIDs, a receptionist or a combination of both?

 

Josh

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, December 13, 2005 8:14 AM
To: James Dust; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity Out of office message

 

Under greetings --> After Greeting can't you -->send caller to -->  hang
up?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Dust
Sent: Tuesday, December 13, 2005 6:03 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Out of office message
Importance: High

 

We have a branch office that is using the Unity Voicemail app, they
would like the ability to be able to have a recorded announcement saying
that they are are out of the office over the Christmas holidays but
would stop the caller leaving a message.

 

Has anyone done this?

 

Regards.

 

James Dust
Network & Communications
Charles Stanley & Co.
25 Luke Street, London EC2A 4AR
Tel: 020-7953 2314 - Fax: 020-7150 9314
www.charles-stanley.co.uk <http://www.charles-stanley.co.uk/>  

 

 


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

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets 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.

 

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

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets 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.
-------------- next part --------------
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From rratliff at cisco.com  Tue Dec 13 10:20:30 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue Dec 13 10:20:31 2005
Subject: [cisco-voip] Installation of Subscriber
In-Reply-To: <39EB1FE59758D9448A5D0B348857C727394E8E@cannt04.candidator.se>
References: <39EB1FE59758D9448A5D0B348857C727394E8E@cannt04.candidator.se>
Message-ID: <ADDFDBAE-6F57-497D-9424-9BCCFA8E76C8@cisco.com>

I'd first verify hosts and lmhosts entries are added for SINV-IPLA- 
CM-01 on the subscriber server.

Are you doing a new install of a subscriber or an upgrade?  For a new  
install SQL should not already be installed on the subscriber server.

-Ryan

On Dec 13, 2005, at 5:43 AM, Robin Inderberg wrote:

Good morning,



I?ve had a bit of trouble this night with the installation of a  
subscriber.



The publisher has CCM 4.1(3), and now it has MSSQL2k SP4, the same as  
the subscriber has.

But the installation stops when It comes to the CCM installation.



I thought the problem were that the CM01 didn?t allow mixed mode  
connections, and I guess the installation tries to connect using the  
SA account?



When I check the logfile called, CCMDBSetup 12-13-2005 10.51.47.log,  
it says like this.





12-13-2005 10:51:47 Starting CallManager Installation

12-13-2005 10:51:47 Setup Registry for the install status = 1

12-13-2005 10:51:47 Checking SQL version...

12-13-2005 10:51:50 Executing CReplaceString function...

12-13-2005 10:51:52 Return Code = 0

12-13-2005 10:51:52 Starting MSSQLServer service...

12-13-2005 10:52:20 Return Code = 0

12-13-2005 10:52:20 Starting SQLServerAgent service...

12-13-2005 10:52:24 Return Code = 0

12-13-2005 10:52:24 Starting MSDTC service...

12-13-2005 10:52:25 Return Code = 0

12-13-2005 10:52:25 Getting Local host name from the registry...

12-13-2005 10:52:25 Reading data from response file for Publisher and  
Database...

12-13-2005 10:52:25 gbPublishGlassHouse: [0]

12-13-2005 10:52:25 gbPullSubscription: [1]

12-13-2005 10:52:25 szGlassServer: [SINV-IPLA-CM-01]

12-13-2005 10:52:25 szGlassDB [CCM0301]

12-13-2005 10:52:25 Checking the Database Connection....

12-13-2005 10:52:25 _DBGetVersion: SINV-IPLA-CM-01 CCM0301  ********

12-13-2005 10:52:42 Subscriber has problem to connect to the  
Publisher Database.





I have tested to connect using the Enterprise Manager, and it works  
both with SA and Windows Authentication.

Should I manually configure anything with ODBC?





The SQL version is on both servers, Microsoft SQL Server  2000 -  
8.00.2039 (Intel X86)   May  3 2005 23:18:38.







Best Regards





Robin Inderberg

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



From Aaron.Derooy at xerox.com  Tue Dec 13 10:39:58 2005
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Tue Dec 13 10:41:07 2005
Subject: [cisco-voip] Which g/w is my call using
Message-ID: <E0EF33751DEA634DA817A51024A8E3F2055101F0@usa0300ms01.na.xerox.net>

Without doing a debug or show command on the g/w itself, does anyone
know how I can find out which g/w a particular call is using? I tried
looking at the Event Viewer but it did now update with an entry when I
made  a PSTN call but it did not update.
Thanks,
Aaron





From lelio at uoguelph.ca  Tue Dec 13 10:46:01 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Dec 13 10:46:04 2005
Subject: [cisco-voip] Which g/w is my call using
References: <E0EF33751DEA634DA817A51024A8E3F2055101F0@usa0300ms01.na.xerox.net>
Message-ID: <00e001c5fffc$51c629f0$30196883@cfs.uoguelph.ca>

If you go to the webpage of the phone and go to streaming statistics, you should see something like this:

      Remote Address   <ipaddr>/<port> 
      Local Address  <ipaddr>/<port> 


This is what I use to see what gateway it is using. We use 6008 T1/PRI MGCP and it works well.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: DeRooy, Aaron H 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, December 13, 2005 10:39 AM
  Subject: [cisco-voip] Which g/w is my call using


  Without doing a debug or show command on the g/w itself, does anyone
  know how I can find out which g/w a particular call is using? I tried
  looking at the Event Viewer but it did now update with an entry when I
  made  a PSTN call but it did not update.
  Thanks,
  Aaron




  _______________________________________________
  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 Dec 13 11:08:05 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 13 11:07:51 2005
Subject: [cisco-voip] Which g/w is my call using
Message-ID: <D713462ED535184D830F6F6301753E7E01A11F7E@VISHNU.wesd.org>

https://cm/dna

 

Dial number analyzer works well if you have it setup also.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, December 13, 2005 7:46 AM
To: DeRooy, Aaron H; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Which g/w is my call using

 

If you go to the webpage of the phone and go to streaming statistics,
you should see something like this:

 

Remote Address 

 

<ipaddr>/<port>

Local Address

 

<ipaddr>/<port>

 

This is what I use to see what gateway it is using. We use 6008 T1/PRI
MGCP and it works well.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: DeRooy, Aaron H <mailto:Aaron.Derooy at xerox.com>  

	To: cisco-voip at puck.nether.net 

	Sent: Tuesday, December 13, 2005 10:39 AM

	Subject: [cisco-voip] Which g/w is my call using

	 

	Without doing a debug or show command on the g/w itself, does
anyone
	know how I can find out which g/w a particular call is using? I
tried
	looking at the Event Viewer but it did now update with an entry
when I
	made  a PSTN call but it did not update.
	Thanks,
	Aaron
	
	
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

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From robin.inderberg at candidator.se  Tue Dec 13 11:17:38 2005
From: robin.inderberg at candidator.se (Robin Inderberg)
Date: Tue Dec 13 11:12:20 2005
Subject: SV: SV: [cisco-voip] Installation of Subscriber
Message-ID: <39EB1FE59758D9448A5D0B348857C7273E52F2@cannt04.candidator.se>

Okey, thanks for the quick answers Ryan!

I will try that this evening.. 

I can not see the CFA in the CCMAdmin or the CCMUser for that users page.

Best regards


Robin Inderberg

-----Ursprungligt meddelande-----
Fr?n: Ryan Ratliff [mailto:rratliff at cisco.com] 
Skickat: den 13 december 2005 16:56
Till: Robin Inderberg
?mne: Re: SV: [cisco-voip] Installation of Subscriber

lmhosts is used for wins resolution, and hosts for dns.  I know  
Windows will take hosts over dns and I assume lmhosts over any wins  
servers configured, if it is enabled in the nic settings.

For the CFA issue if you look at the person's phone in the CCMAdmin  
does it show the CFA set?  If so, you need to restart the Database  
Layer Monitor server on the publisher and then change the CFA on the  
phone via the CCMAdmin web page to synch it up.

If the CFA is not present on the CCMAdmin web page but the ccm where  
the phone is registered is still forwarding, try the above but if it  
does not fix the issue you may need to restart the CCM process on  
that node.

-Ryan

On Dec 13, 2005, at 10:56 AM, Robin Inderberg wrote:

Hi there, sorry for taking your time..

I solved it some hour ago, and it was like you said.
I did test the connection with the SQL Enterprise Manager, and it  
worked to connect to the DB. BUT, it took a lot of time. That is why  
I were so confused why it didn't work with the CM 4.1(3) installation.

Then I guess the timeout in the installation is lower then in the  
Enterprise Manager.

Is It right that Windows uses hosts, dns, lmhosts, in that order?  
Must be something like that.

I got a quick one too, one of the users did enabled CFA on his DN.  
And this morning he should disable it, everything seemed to work, but  
it didn't take away the CFA.

I can't find the number he have forwarded it too in the SQL either, I  
checked under CCM0301 -> Tables -> NumPlan.



Thanks for your time Ryan!


Best regards


Robin Inderberg


-----Ursprungligt meddelande-----
Fr?n: Ryan Ratliff [mailto:rratliff at cisco.com]
Skickat: den 13 december 2005 16:21
Till: Robin Inderberg
Kopia: cisco-voip at puck.nether.net
?mne: Re: [cisco-voip] Installation of Subscriber

I'd first verify hosts and lmhosts entries are added for SINV-IPLA-
CM-01 on the subscriber server.

Are you doing a new install of a subscriber or an upgrade?  For a new
install SQL should not already be installed on the subscriber server.

-Ryan

On Dec 13, 2005, at 5:43 AM, Robin Inderberg wrote:

Good morning,



I've had a bit of trouble this night with the installation of a
subscriber.



The publisher has CCM 4.1(3), and now it has MSSQL2k SP4, the same as
the subscriber has.

But the installation stops when It comes to the CCM installation.



I thought the problem were that the CM01 didn't allow mixed mode
connections, and I guess the installation tries to connect using the
SA account?



When I check the logfile called, CCMDBSetup 12-13-2005 10.51.47.log,
it says like this.





12-13-2005 10:51:47 Starting CallManager Installation

12-13-2005 10:51:47 Setup Registry for the install status = 1

12-13-2005 10:51:47 Checking SQL version...

12-13-2005 10:51:50 Executing CReplaceString function...

12-13-2005 10:51:52 Return Code = 0

12-13-2005 10:51:52 Starting MSSQLServer service...

12-13-2005 10:52:20 Return Code = 0

12-13-2005 10:52:20 Starting SQLServerAgent service...

12-13-2005 10:52:24 Return Code = 0

12-13-2005 10:52:24 Starting MSDTC service...

12-13-2005 10:52:25 Return Code = 0

12-13-2005 10:52:25 Getting Local host name from the registry...

12-13-2005 10:52:25 Reading data from response file for Publisher and
Database...

12-13-2005 10:52:25 gbPublishGlassHouse: [0]

12-13-2005 10:52:25 gbPullSubscription: [1]

12-13-2005 10:52:25 szGlassServer: [SINV-IPLA-CM-01]

12-13-2005 10:52:25 szGlassDB [CCM0301]

12-13-2005 10:52:25 Checking the Database Connection....

12-13-2005 10:52:25 _DBGetVersion: SINV-IPLA-CM-01 CCM0301  ********

12-13-2005 10:52:42 Subscriber has problem to connect to the
Publisher Database.





I have tested to connect using the Enterprise Manager, and it works
both with SA and Windows Authentication.

Should I manually configure anything with ODBC?





The SQL version is on both servers, Microsoft SQL Server  2000 -
8.00.2039 (Intel X86)   May  3 2005 23:18:38.







Best Regards





Robin Inderberg

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




From jgrace at digitelusa.net  Tue Dec 13 13:05:28 2005
From: jgrace at digitelusa.net (James)
Date: Tue Dec 13 13:05:23 2005
Subject: [cisco-voip] Crs realtime report no outbound call
Message-ID: <1576700104-1134497120-cardhu_blackberry.rim.net-30988-@engine09-cell02>

On the realtime report is crs we are not recording outbound calls. Inbound is fine but outbnd. Is always 0
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From Allan.J.Cleaveland at L-3com.com  Tue Dec 13 13:20:51 2005
From: Allan.J.Cleaveland at L-3com.com (Cleaveland, AJ Allan @ IS)
Date: Tue Dec 13 13:21:13 2005
Subject: [cisco-voip] JTAPI Question: Use of Cisco Route Points and CTI Ports
Message-ID: <7736DD1D5D6EDD4081B590872B56DCCC0C29F7D9@gvlexch01.gvl.is.l-3com.com>

I'm working with JTAPI and CallManager 4.1.
 
I wish to use route points or CTI ports (I really don't care which as long
as it will act like a phone) to programmatically make calls and conference
calls.  I can successfully register a CiscoMediaTerminal but even after it's
registered I always get the message Address is out of service.  So my
question is, how can I put it into service so I can make a call?
 
I've tried the same thing with route points but don't get nearly as far,
since I can't get the CiscoRouteTerminal in order to register it (I use
CiscoProvider.getTerminals to get all the terminals and then loop through
them to find the one I want).
 
I have set up both the CTI port and route point in CallManager, I've
assigned both numbers, and I've made sure my user has access to them both.
 
Thank you,
Allan
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From kthorngr at cisco.com  Tue Dec 13 13:48:02 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Tue Dec 13 13:48:19 2005
Subject: [cisco-voip] JTAPI Question: Use of Cisco Route Points and CTI
	Ports
In-Reply-To: <7736DD1D5D6EDD4081B590872B56DCCC0C29F7D9@gvlexch01.gvl.is.l-3com.com>
References: <7736DD1D5D6EDD4081B590872B56DCCC0C29F7D9@gvlexch01.gvl.is.l-3com.com>
Message-ID: <6f2981959a4ea4cf6b58abc8558eea77@cisco.com>

Hi Allen,

You will want to use CTI Ports to use phone type features.  The 
username you have configured needs to be associated to the CTI Port 
(not sure if that is what you mean by the user has access).  You need 
to make sure you are using the correct version of the CiscoTSP by going 
to the Install Plugins page from CCMAdmin.  If your application still 
doesn't work after checking these steps then you might want to try 
phone.exe, found at the below web site, to see if you can register the 
CTI Port.

http://www.julmar.com/samples.htm

Kevin
On Dec 13, 2005, at 1:20 PM, Cleaveland, AJ Allan @ IS wrote:

> I'm working with JTAPI and CallManager 4.1.
> ?
> I wish to use route points or CTI ports (I really don't care which as 
> long as it will act like a phone) to programmatically make calls and 
> conference calls.? I can successfully register a CiscoMediaTerminal 
> but even after it's registered I always get the message Address is out 
> of service.? So my question is, how can I put it into service so I can 
> make a call?
> ?
> I've tried the same thing with route points but don't get nearly as 
> far, since I can't get the CiscoRouteTerminal in order to register it 
> (I use CiscoProvider.getTerminals to get all the terminals and then 
> loop through them to find the one I want).
> ?
> I have set up both the CTI port and route point in CallManager, I've 
> assigned both numbers, and I've made sure my user has access to them 
> both.
> ?
> Thank you,
> Allan_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From jburton at NETechCorp.com  Tue Dec 13 13:50:46 2005
From: jburton at NETechCorp.com (Jason Burton)
Date: Tue Dec 13 13:50:54 2005
Subject: [cisco-voip] Unity connection product
Message-ID: <FDC137145729AD438CDCA0F9FC6F72363A2FDC@mail.netechcorp.com>

Does anyone have any experience using the live record with this product?
I'm assuming it works like unity where you have to conference in a dn
used for recording.  Is there anyway to use unity to record without
having the end user involved?

 

Thanks again!

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From jose.mujica at netxar.com  Tue Dec 13 14:09:28 2005
From: jose.mujica at netxar.com (Jose Mujica)
Date: Tue Dec 13 14:09:32 2005
Subject: [cisco-voip] CallManager 4.1 and ACT Card on CMM Module
Message-ID: <20051213190111.M8761@netxar.com>

Hi.

 I'm having a problem and would like your inputs. I have an ACT Module in a
Cat6k CMM Module running 12.3(8)XY2. I have configured the mediacard for
conference resources for G711 and G729 codecs. I have also configured the SCCP
and DSPFARM for the conference resources and associate all the configuration
all together.

 On the CCM I have created a hardware conference resource using the IOS
Enhanced Conference Resource and the device shows up as register on
callmanager and connected on the CMM Module. The client currently has a
central site and a remote site and regions has been configured to use G729
over the WAN. The CMM module is at the central site. When a conference bridge
is initiated using the ACT dsps I don't have 2-way voice on all 3 IP Phones (2
central site and  1 at the remote site). On the CAT6k I can see the sessions
with the show mediacard connected command but I only have 1-way audio. 

Any ideas? Below is the config for the ACT card and show commands:

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

6509-CMM-GWY1#sh ver
Cisco IOS Software, Cat6K-lc Software (wscmm-I6S-M), Version 12.3(8)XY2, RELEASE
SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2004 by Cisco Systems, Inc.
Compiled Tue 16-Nov-04 12:19 by cmong

ROM: System Bootstrap, Version 12.2(1r)T2, RELEASE SOFTWARE (fc1)

6509-CMM-GWY1 uptime is 10 weeks, 4 days, 59 minutes
System returned to ROM by power-on
System restarted at 11:41:30 AST Fri Sep 30 2005
System image file is "bootflash:wscmm-i6s-mz.123-8.XY2.bin"

cisco WS-SVC-CMM Cat6k Voice Linecard (R7000) processor (revision 0xFF) with 196
608K/65536K bytes of memory.
Processor board ID SAD09010E57
R7000 CPU at 400MHz, Implementation 39, Rev 3.3, 256KB L2 Cache
Last reset from power-on
2 FastEthernet interfaces
1 Gigabit Ethernet interface
96 Serial interfaces
6 Channelized T1/PRI ports
32768K bytes of processor board bootflash (Read/Write)

Configuration register is 0x2102

6509-CMM-GWY1#


Total: 0
6509-CMM-GWY1#
6509-CMM-GWY1#sh mediacard connections
Id Type Slot/ RPort SPort RxPkts TxPkts Remote-Ip
DSP/Ch
1 xcode 4/1/1 19482 22392 203 0 172.16.56.34
2 xcode 4/1/2 31390 23256 198 0 172.16.55.12
Total: 2
6509-CMM-GWY1#

6509-CMM-GWY1#sh sccp conn
6509-CMM-GWY1#sh sccp connections
sess_id conn_id stype mode codec ripaddr rport sport

33560307 36347521 xcode sendrecv g711u 172.16.55.12 31390 23256
33560307 36347489 xcode sendrecv g729b 172.16.56.34 19482 22392

Total number of active session(s) 1, and connection(s) 2

6509-CMM-GWY1#


The IP Address 172.16.56.34 is the IP Phone across the WAN and the
172.16.55.12 is the IP Address of the CallManager. I'm using the IPVM service
for conference right now and using the ACT for transcoding. The conference was
initiated by the remote ip phone and 2 IP Phones from the central site were in
the conference. They were able to talk to each other but i was unable to hear
them and viceversa. I also tried to configured the ACT for conference
resources but when I initate the conference from the remote ip phone I
received a fast busy signal.

Here is the relevant config on the CMM for the ACT card:

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

interface GigabitEthernet1/0
ip address 172.16.57.5 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
no negotiation auto
no keepalive
!
interface FastEthernet3/0
description ** Ad-hoc Confr and Tcoding PA Mod 4 **
ip address 172.16.57.10 255.255.255.255
no ip redirects
no ip unreachables
no ip proxy-arp
half-duplex
!
mediacard 4
resource-pool cfb dsps 4
!
!
sccp local GigabitEthernet1/0
sccp ccm 172.16.55.12 identifier 2 version 4.1
sccp ccm 172.16.55.11 identifier 1 version 4.1
sccp
!
sccp ccm group 1
associate ccm 2 priority 1
associate ccm 1 priority 2
associate profile 10 register CFB00033235c614

!
dspfarm
!

!
dspfarm profile 10 conference adhoc
codec g711ulaw packetization-period 30
codec g711alaw packetization-period 30
codec g729r8 packetization-period 30
codec g729ar8 packetization-period 30
codec g723r63 packetization-period 30
codec g723r53 packetization-period 30
associate resource-pool cfb
!
! 

Thanks in advanced,
-Jose

From Eric.A.Fretz at L-3com.com  Tue Dec 13 14:33:37 2005
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Tue Dec 13 14:34:11 2005
Subject: [cisco-voip] CallManager 4.1 and ACT Card on CMM Module
Message-ID: <51A2C3ED40A3584281444F65BC9026270A8844EC@gvlexch02.gvl.is.l-3com.com>

What type of software are you running on the host switch: CatOS or IOS?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jose Mujica
Sent: Tuesday, December 13, 2005 1:09 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager 4.1 and ACT Card on CMM Module


Hi.

 I'm having a problem and would like your inputs. I have an ACT Module in a
Cat6k CMM Module running 12.3(8)XY2. I have configured the mediacard for
conference resources for G711 and G729 codecs. I have also configured the
SCCP and DSPFARM for the conference resources and associate all the
configuration all together.

 On the CCM I have created a hardware conference resource using the IOS
Enhanced Conference Resource and the device shows up as register on
callmanager and connected on the CMM Module. The client currently has a
central site and a remote site and regions has been configured to use G729
over the WAN. The CMM module is at the central site. When a conference
bridge is initiated using the ACT dsps I don't have 2-way voice on all 3 IP
Phones (2 central site and  1 at the remote site). On the CAT6k I can see
the sessions with the show mediacard connected command but I only have 1-way
audio. 

Any ideas? Below is the config for the ACT card and show commands:

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

6509-CMM-GWY1#sh ver
Cisco IOS Software, Cat6K-lc Software (wscmm-I6S-M), Version 12.3(8)XY2,
RELEASE SOFTWARE (fc1) Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2004 by Cisco Systems, Inc. Compiled Tue 16-Nov-04 12:19
by cmong

ROM: System Bootstrap, Version 12.2(1r)T2, RELEASE SOFTWARE (fc1)

6509-CMM-GWY1 uptime is 10 weeks, 4 days, 59 minutes
System returned to ROM by power-on
System restarted at 11:41:30 AST Fri Sep 30 2005
System image file is "bootflash:wscmm-i6s-mz.123-8.XY2.bin"

cisco WS-SVC-CMM Cat6k Voice Linecard (R7000) processor (revision 0xFF) with
196 608K/65536K bytes of memory. Processor board ID SAD09010E57 R7000 CPU at
400MHz, Implementation 39, Rev 3.3, 256KB L2 Cache Last reset from power-on
2 FastEthernet interfaces 1 Gigabit Ethernet interface 96 Serial interfaces
6 Channelized T1/PRI ports 32768K bytes of processor board bootflash
(Read/Write)

Configuration register is 0x2102

6509-CMM-GWY1#


Total: 0
6509-CMM-GWY1#
6509-CMM-GWY1#sh mediacard connections
Id Type Slot/ RPort SPort RxPkts TxPkts Remote-Ip
DSP/Ch
1 xcode 4/1/1 19482 22392 203 0 172.16.56.34
2 xcode 4/1/2 31390 23256 198 0 172.16.55.12
Total: 2
6509-CMM-GWY1#

6509-CMM-GWY1#sh sccp conn
6509-CMM-GWY1#sh sccp connections
sess_id conn_id stype mode codec ripaddr rport sport

33560307 36347521 xcode sendrecv g711u 172.16.55.12 31390 23256 33560307
36347489 xcode sendrecv g729b 172.16.56.34 19482 22392

Total number of active session(s) 1, and connection(s) 2

6509-CMM-GWY1#


The IP Address 172.16.56.34 is the IP Phone across the WAN and the
172.16.55.12 is the IP Address of the CallManager. I'm using the IPVM
service for conference right now and using the ACT for transcoding. The
conference was initiated by the remote ip phone and 2 IP Phones from the
central site were in the conference. They were able to talk to each other
but i was unable to hear them and viceversa. I also tried to configured the
ACT for conference resources but when I initate the conference from the
remote ip phone I received a fast busy signal.

Here is the relevant config on the CMM for the ACT card:

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

interface GigabitEthernet1/0
ip address 172.16.57.5 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
no negotiation auto
no keepalive
!
interface FastEthernet3/0
description ** Ad-hoc Confr and Tcoding PA Mod 4 **
ip address 172.16.57.10 255.255.255.255
no ip redirects
no ip unreachables
no ip proxy-arp
half-duplex
!
mediacard 4
resource-pool cfb dsps 4
!
!
sccp local GigabitEthernet1/0
sccp ccm 172.16.55.12 identifier 2 version 4.1
sccp ccm 172.16.55.11 identifier 1 version 4.1
sccp
!
sccp ccm group 1
associate ccm 2 priority 1
associate ccm 1 priority 2
associate profile 10 register CFB00033235c614

!
dspfarm
!

!
dspfarm profile 10 conference adhoc
codec g711ulaw packetization-period 30
codec g711alaw packetization-period 30
codec g729r8 packetization-period 30
codec g729ar8 packetization-period 30
codec g723r63 packetization-period 30
codec g723r53 packetization-period 30
associate resource-pool cfb
!
! 

Thanks in advanced,
-Jose
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From CORTIZ at broward.org  Tue Dec 13 15:01:07 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue Dec 13 15:01:18 2005
Subject: [cisco-voip] H.323 Dial Peer config
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B69@MREXVS01.bc.broward.cty>

We currently use Centrex analog lines and Centrex PRI lines and they
both require you to pass the leading 9 to the PSTN.  Does my dial peer 1
look correct for incoming calls? Do I just specify any port there?  I
will definitely look into the Trunk group config.

Someone replied to me a couple of weeks back and said if I use an FXO
card with M1 designation (part VIC-4FXO-M1) and H.323 gateway I could
get caller ID.  I just checked and this card reached EOL Jan 15, 2004.
I have the newer replacement cards VIC2-4FXO. I would hope these would
support that and more.  Any thoughts on the caller ID issue would be
appreciated.

Carlos

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com] 
Sent: Monday, December 12, 2005 10:38 PM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H.323 Dial Peer config

On the dial-peers, you are forwarding all digits so
that will also send the 9 that starts all your
destination patterns. 

Usually with FXO, there is no inbound caller id to
tell it where to route call so on the voice-port you
need to use connection plar to route call somewhere
(This is same as Attendant DN setting on the endpoint
in CCM when using MGCP). 

They have added trunk groups to IOS, so you can assign
voice ports to a trunkgroup, then have one dial-peer
pointing to the trunk group. You can then assign
preferences to voice-port that is a member of the
trunk group, etc. This requires a trunk group to be
configured globally. Under the trunkgroup config you
can configure hunt method, etc. 

Erick

--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> I have never configured my FXO PSTN router
> connections with H.323, only
> MGCP.  I did configure H.323 for PRI interfaces.
> Going by that and what
> I have refreshed with CCO docs, I have come up with
> a starting config to
> do FXO with H.323. Dial peer 1 would be for the
> incoming call leg on all
> calls.  This assumes a 4 port FXO card.  Does this
> appear correct?  Have
> there been any improvements on the "hunting" for
> outbound calls beyond
> using the preference command? Any suggestions or
> help would be
> appreciated.
> 
>  
> 
> dial-peer voice 1 pots
> 
>  description ** Inbound Call leg all calls **
> 
>  incoming called-number .
> 
>  direct-inward-dial
> 
>  port 0/1/0
> 
> !
> 
> dial-peer voice 954 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/0
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9542 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  preference 1
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/1
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9543 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  preference 2
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/2
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9544 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  preference 3
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/3
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9999 voip
> 
>  description ** Inbound 357 nums to mr-cm-sub1 **
> 
>  destination-pattern [5-8]...
> 
>  progress_ind setup enable 3
> 
>  session target ipv4:10.100.200.21
> 
>  incoming called-number .
> 
>  dtmf-relay h245-alphanumeric
> 
>  codec g711ulaw
> 
> !
> 
> dial-peer voice 99992 voip
> 
>  description ** Inbound 357 nums to mr-cm-pub **
> 
>  preference 2
> 
>  destination-pattern [5-8]...
> 
>  progress_ind setup enable 3
> 
>  session target ipv4:10.100.200.20
> 
>  incoming called-number .
> 
>  dtmf-relay h245-alphanumeric
> 
>  codec g711ulaw
> 
>  
> 
>  
> 
> Carlos Ortiz
> 
> Senior Information Technology Specialist
> 
> Broward County Office of Information Technology
> 
> Data Communications
> 
> 954-357-7981
> 
> cortiz at broward.org
> 
>  
> 
>  
> 
> > _______________________________________________
> 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 Scott.Voll at wesd.org  Tue Dec 13 15:18:57 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 13 15:18:46 2005
Subject: [cisco-voip] CallManager 4.1 and ACT Card on CMM Module
Message-ID: <D713462ED535184D830F6F6301753E7E01A11FB4@VISHNU.wesd.org>

More of a neat thing but why is your FE configured as half-duplex?

What is IPVM?  Why not try the meetme conferencing?

Why are you having to transcode with this conference resource?

Have you thought about upgrading to xy6.  It has been very stable for
us.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jose Mujica
Sent: Tuesday, December 13, 2005 11:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager 4.1 and ACT Card on CMM Module

Hi.

 I'm having a problem and would like your inputs. I have an ACT Module
in a
Cat6k CMM Module running 12.3(8)XY2. I have configured the mediacard for
conference resources for G711 and G729 codecs. I have also configured
the SCCP
and DSPFARM for the conference resources and associate all the
configuration
all together.

 On the CCM I have created a hardware conference resource using the IOS
Enhanced Conference Resource and the device shows up as register on
callmanager and connected on the CMM Module. The client currently has a
central site and a remote site and regions has been configured to use
G729
over the WAN. The CMM module is at the central site. When a conference
bridge
is initiated using the ACT dsps I don't have 2-way voice on all 3 IP
Phones (2
central site and  1 at the remote site). On the CAT6k I can see the
sessions
with the show mediacard connected command but I only have 1-way audio. 

Any ideas? Below is the config for the ACT card and show commands:

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

6509-CMM-GWY1#sh ver
Cisco IOS Software, Cat6K-lc Software (wscmm-I6S-M), Version 12.3(8)XY2,
RELEASE
SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2004 by Cisco Systems, Inc.
Compiled Tue 16-Nov-04 12:19 by cmong

ROM: System Bootstrap, Version 12.2(1r)T2, RELEASE SOFTWARE (fc1)

6509-CMM-GWY1 uptime is 10 weeks, 4 days, 59 minutes
System returned to ROM by power-on
System restarted at 11:41:30 AST Fri Sep 30 2005
System image file is "bootflash:wscmm-i6s-mz.123-8.XY2.bin"

cisco WS-SVC-CMM Cat6k Voice Linecard (R7000) processor (revision 0xFF)
with 196
608K/65536K bytes of memory.
Processor board ID SAD09010E57
R7000 CPU at 400MHz, Implementation 39, Rev 3.3, 256KB L2 Cache
Last reset from power-on
2 FastEthernet interfaces
1 Gigabit Ethernet interface
96 Serial interfaces
6 Channelized T1/PRI ports
32768K bytes of processor board bootflash (Read/Write)

Configuration register is 0x2102

6509-CMM-GWY1#


Total: 0
6509-CMM-GWY1#
6509-CMM-GWY1#sh mediacard connections
Id Type Slot/ RPort SPort RxPkts TxPkts Remote-Ip
DSP/Ch
1 xcode 4/1/1 19482 22392 203 0 172.16.56.34
2 xcode 4/1/2 31390 23256 198 0 172.16.55.12
Total: 2
6509-CMM-GWY1#

6509-CMM-GWY1#sh sccp conn
6509-CMM-GWY1#sh sccp connections
sess_id conn_id stype mode codec ripaddr rport sport

33560307 36347521 xcode sendrecv g711u 172.16.55.12 31390 23256
33560307 36347489 xcode sendrecv g729b 172.16.56.34 19482 22392

Total number of active session(s) 1, and connection(s) 2

6509-CMM-GWY1#


The IP Address 172.16.56.34 is the IP Phone across the WAN and the
172.16.55.12 is the IP Address of the CallManager. I'm using the IPVM
service
for conference right now and using the ACT for transcoding. The
conference was
initiated by the remote ip phone and 2 IP Phones from the central site
were in
the conference. They were able to talk to each other but i was unable to
hear
them and viceversa. I also tried to configured the ACT for conference
resources but when I initate the conference from the remote ip phone I
received a fast busy signal.

Here is the relevant config on the CMM for the ACT card:

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

interface GigabitEthernet1/0
ip address 172.16.57.5 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
no negotiation auto
no keepalive
!
interface FastEthernet3/0
description ** Ad-hoc Confr and Tcoding PA Mod 4 **
ip address 172.16.57.10 255.255.255.255
no ip redirects
no ip unreachables
no ip proxy-arp
half-duplex
!
mediacard 4
resource-pool cfb dsps 4
!
!
sccp local GigabitEthernet1/0
sccp ccm 172.16.55.12 identifier 2 version 4.1
sccp ccm 172.16.55.11 identifier 1 version 4.1
sccp
!
sccp ccm group 1
associate ccm 2 priority 1
associate ccm 1 priority 2
associate profile 10 register CFB00033235c614

!
dspfarm
!

!
dspfarm profile 10 conference adhoc
codec g711ulaw packetization-period 30
codec g711alaw packetization-period 30
codec g729r8 packetization-period 30
codec g729ar8 packetization-period 30
codec g723r63 packetization-period 30
codec g723r53 packetization-period 30
associate resource-pool cfb
!
! 

Thanks in advanced,
-Jose
_______________________________________________
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 Dec 13 15:23:50 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 13 15:23:35 2005
Subject: [cisco-voip] CallManager 4.1 and ACT Card on CMM Module
Message-ID: <D713462ED535184D830F6F6301753E7E01A11FB8@VISHNU.wesd.org>

After thoughts.

Are you trying to use the ACT card for Xcoding or Conferencing?

Xcoding should be a MTP resource not a CFB resource.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jose Mujica
Sent: Tuesday, December 13, 2005 11:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager 4.1 and ACT Card on CMM Module

Hi.

 I'm having a problem and would like your inputs. I have an ACT Module
in a
Cat6k CMM Module running 12.3(8)XY2. I have configured the mediacard for
conference resources for G711 and G729 codecs. I have also configured
the SCCP
and DSPFARM for the conference resources and associate all the
configuration
all together.

 On the CCM I have created a hardware conference resource using the IOS
Enhanced Conference Resource and the device shows up as register on
callmanager and connected on the CMM Module. The client currently has a
central site and a remote site and regions has been configured to use
G729
over the WAN. The CMM module is at the central site. When a conference
bridge
is initiated using the ACT dsps I don't have 2-way voice on all 3 IP
Phones (2
central site and  1 at the remote site). On the CAT6k I can see the
sessions
with the show mediacard connected command but I only have 1-way audio. 

Any ideas? Below is the config for the ACT card and show commands:

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

6509-CMM-GWY1#sh ver
Cisco IOS Software, Cat6K-lc Software (wscmm-I6S-M), Version 12.3(8)XY2,
RELEASE
SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2004 by Cisco Systems, Inc.
Compiled Tue 16-Nov-04 12:19 by cmong

ROM: System Bootstrap, Version 12.2(1r)T2, RELEASE SOFTWARE (fc1)

6509-CMM-GWY1 uptime is 10 weeks, 4 days, 59 minutes
System returned to ROM by power-on
System restarted at 11:41:30 AST Fri Sep 30 2005
System image file is "bootflash:wscmm-i6s-mz.123-8.XY2.bin"

cisco WS-SVC-CMM Cat6k Voice Linecard (R7000) processor (revision 0xFF)
with 196
608K/65536K bytes of memory.
Processor board ID SAD09010E57
R7000 CPU at 400MHz, Implementation 39, Rev 3.3, 256KB L2 Cache
Last reset from power-on
2 FastEthernet interfaces
1 Gigabit Ethernet interface
96 Serial interfaces
6 Channelized T1/PRI ports
32768K bytes of processor board bootflash (Read/Write)

Configuration register is 0x2102

6509-CMM-GWY1#


Total: 0
6509-CMM-GWY1#
6509-CMM-GWY1#sh mediacard connections
Id Type Slot/ RPort SPort RxPkts TxPkts Remote-Ip
DSP/Ch
1 xcode 4/1/1 19482 22392 203 0 172.16.56.34
2 xcode 4/1/2 31390 23256 198 0 172.16.55.12
Total: 2
6509-CMM-GWY1#

6509-CMM-GWY1#sh sccp conn
6509-CMM-GWY1#sh sccp connections
sess_id conn_id stype mode codec ripaddr rport sport

33560307 36347521 xcode sendrecv g711u 172.16.55.12 31390 23256
33560307 36347489 xcode sendrecv g729b 172.16.56.34 19482 22392

Total number of active session(s) 1, and connection(s) 2

6509-CMM-GWY1#


The IP Address 172.16.56.34 is the IP Phone across the WAN and the
172.16.55.12 is the IP Address of the CallManager. I'm using the IPVM
service
for conference right now and using the ACT for transcoding. The
conference was
initiated by the remote ip phone and 2 IP Phones from the central site
were in
the conference. They were able to talk to each other but i was unable to
hear
them and viceversa. I also tried to configured the ACT for conference
resources but when I initate the conference from the remote ip phone I
received a fast busy signal.

Here is the relevant config on the CMM for the ACT card:

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

interface GigabitEthernet1/0
ip address 172.16.57.5 255.255.255.0
no ip redirects
no ip unreachables
no ip proxy-arp
no negotiation auto
no keepalive
!
interface FastEthernet3/0
description ** Ad-hoc Confr and Tcoding PA Mod 4 **
ip address 172.16.57.10 255.255.255.255
no ip redirects
no ip unreachables
no ip proxy-arp
half-duplex
!
mediacard 4
resource-pool cfb dsps 4
!
!
sccp local GigabitEthernet1/0
sccp ccm 172.16.55.12 identifier 2 version 4.1
sccp ccm 172.16.55.11 identifier 1 version 4.1
sccp
!
sccp ccm group 1
associate ccm 2 priority 1
associate ccm 1 priority 2
associate profile 10 register CFB00033235c614

!
dspfarm
!

!
dspfarm profile 10 conference adhoc
codec g711ulaw packetization-period 30
codec g711alaw packetization-period 30
codec g729r8 packetization-period 30
codec g729ar8 packetization-period 30
codec g723r63 packetization-period 30
codec g723r53 packetization-period 30
associate resource-pool cfb
!
! 

Thanks in advanced,
-Jose
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Tue Dec 13 16:39:07 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Dec 13 16:39:11 2005
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How about
	3.5 and 4.x?
Message-ID: <005701c6002d$a5466460$30196883@cfs.uoguelph.ca>

We have some issues with IPCCX v3.0 where the calling party info is only displayed on the phone when you pick up the line and not while it is ringing. While it is ringing it shows the queue DN (or route point, I can't remember now). Similarly, in the directories, it does not show the actual calling party info but again, the queue DN (or route point).

Is this the same behaviour in 3.5 and 4? Just wondering before I put in a feature request.

Lelio

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From CORTIZ at broward.org  Tue Dec 13 16:43:19 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue Dec 13 16:43:29 2005
Subject: [cisco-voip] H.323 Dial Peer config
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B6D@MREXVS01.bc.broward.cty>

Erick,

All the information I am finding on Trunk Groups refers to PRI's.  I can
create a trunk group at the global level, but I don't see an option that
would allow me to add FXO voice ports to it. See below:

PESPC(config)#trunk group  FXO-G1             
PESPC(config-trunk-group)#?
Trunk Group configuration commands:
  capacity             capacity update timer config
  carrier-id           Name of the carrier associated with this trunk
group
  default              Set a command to its defaults
  description          Trunk group description
  exit                 Exit from trunk group configuration mode
  help                 Description of the interactive help system
  hunt-scheme          The hunt scheme in trunk group
  isdn                 Specify isdn supplementary service
  max-calls            Maximum number of calls allowed
  max-retry            Maximum number of outgoing attempts
  no                   Negate a command or set its defaults
  tgrep                TGREP config
  translation-profile  Translation profile
  voice-class          Set voice class control parameters


-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com] 
Sent: Monday, December 12, 2005 10:38 PM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H.323 Dial Peer config

On the dial-peers, you are forwarding all digits so
that will also send the 9 that starts all your
destination patterns. 

Usually with FXO, there is no inbound caller id to
tell it where to route call so on the voice-port you
need to use connection plar to route call somewhere
(This is same as Attendant DN setting on the endpoint
in CCM when using MGCP). 

They have added trunk groups to IOS, so you can assign
voice ports to a trunkgroup, then have one dial-peer
pointing to the trunk group. You can then assign
preferences to voice-port that is a member of the
trunk group, etc. This requires a trunk group to be
configured globally. Under the trunkgroup config you
can configure hunt method, etc. 

Erick

--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> I have never configured my FXO PSTN router
> connections with H.323, only
> MGCP.  I did configure H.323 for PRI interfaces.
> Going by that and what
> I have refreshed with CCO docs, I have come up with
> a starting config to
> do FXO with H.323. Dial peer 1 would be for the
> incoming call leg on all
> calls.  This assumes a 4 port FXO card.  Does this
> appear correct?  Have
> there been any improvements on the "hunting" for
> outbound calls beyond
> using the preference command? Any suggestions or
> help would be
> appreciated.
> 
>  
> 
> dial-peer voice 1 pots
> 
>  description ** Inbound Call leg all calls **
> 
>  incoming called-number .
> 
>  direct-inward-dial
> 
>  port 0/1/0
> 
> !
> 
> dial-peer voice 954 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/0
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9542 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  preference 1
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/1
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9543 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  preference 2
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/2
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9544 pots
> 
>  description ** Local 954,754 Calls **
> 
>  destination-pattern 9[7,9]54.......
> 
>  preference 3
> 
>  progress_ind setup enable 3
> 
>  progress_ind alert enable 8
> 
>  progress_ind progress enable 8
> 
>  progress_ind connect enable 8
> 
>  direct-inward-dial
> 
>  port 0/1/3
> 
>  forward-digits all
> 
>  
> 
> dial-peer voice 9999 voip
> 
>  description ** Inbound 357 nums to mr-cm-sub1 **
> 
>  destination-pattern [5-8]...
> 
>  progress_ind setup enable 3
> 
>  session target ipv4:10.100.200.21
> 
>  incoming called-number .
> 
>  dtmf-relay h245-alphanumeric
> 
>  codec g711ulaw
> 
> !
> 
> dial-peer voice 99992 voip
> 
>  description ** Inbound 357 nums to mr-cm-pub **
> 
>  preference 2
> 
>  destination-pattern [5-8]...
> 
>  progress_ind setup enable 3
> 
>  session target ipv4:10.100.200.20
> 
>  incoming called-number .
> 
>  dtmf-relay h245-alphanumeric
> 
>  codec g711ulaw
> 
>  
> 
>  
> 
> Carlos Ortiz
> 
> Senior Information Technology Specialist
> 
> Broward County Office of Information Technology
> 
> Data Communications
> 
> 954-357-7981
> 
> cortiz at broward.org
> 
>  
> 
>  
> 
> > _______________________________________________
> 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 CORTIZ at broward.org  Tue Dec 13 16:48:43 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue Dec 13 16:48:51 2005
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How
	about3.5 and 4.x?
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B6E@MREXVS01.bc.broward.cty>

Just checked ours (3.5) and it does show the DN of the Route Port.  I'm
sure Scott Voll can let us know all about 4.0.  :-)

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, December 13, 2005 4:39 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How
about3.5 and 4.x?

 

We have some issues with IPCCX v3.0 where the calling party info is only
displayed on the phone when you pick up the line and not while it is
ringing. While it is ringing it shows the queue DN (or route point, I
can't remember now). Similarly, in the directories, it does not show the
actual calling party info but again, the queue DN (or route point).

 

Is this the same behaviour in 3.5 and 4? Just wondering before I put in
a feature request.

 

Lelio

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

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From pwalenta at wi.rr.com  Tue Dec 13 16:57:11 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue Dec 13 16:57:01 2005
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How
	about	3.5 and 4.x?
In-Reply-To: <005701c6002d$a5466460$30196883@cfs.uoguelph.ca>
Message-ID: <200512132153.jBDLrjdu004188@ms-smtp-02.rdc-kc.rr.com>

This has been an issue for quite some time.  If I remember correctly, this
is a CTI limitation.  I though the CAD desktop (3.0 and higher) displayed
this information, but I could be wrong.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, December 13, 2005 3:39 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How about
3.5 and 4.x?


We have some issues with IPCCX v3.0 where the calling party info is only
displayed on the phone when you pick up the line and not while it is
ringing. While it is ringing it shows the queue DN (or route point, I can't
remember now). Similarly, in the directories, it does not show the actual
calling party info but again, the queue DN (or route point).
 
Is this the same behaviour in 3.5 and 4? Just wondering before I put in a
feature request.
 
Lelio
 
----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than
an
IP phone on a long haul 10base2 connection"
LFJ

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From Scott.Voll at wesd.org  Tue Dec 13 17:03:21 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 13 17:03:07 2005
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 -
	Howabout	3.5 and 4.x?
Message-ID: <D713462ED535184D830F6F6301753E7E01A11FCB@VISHNU.wesd.org>

Yes. In the little windows that come with the agent you can see the ANI
as the original calling number.  But as far as the display on the phone,
the CTI Port is what you see.  This is good for us as we need to know
which contact center they are trying to reach.  IE> the internal users
or a regional user.  We run two different numbers / IPCC scripts and
they go to the same agents. But the CTI port name is different so the
agents know which number they called.

 

Scott

 

PS.  So If cisco's listening, make it optional not mandatory please.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Tuesday, December 13, 2005 1:57 PM
To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 -
Howabout 3.5 and 4.x?

 

This has been an issue for quite some time.  If I remember correctly,
this is a CTI limitation.  I though the CAD desktop (3.0 and higher)
displayed this information, but I could be wrong.

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	Sent: Tuesday, December 13, 2005 3:39 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 -
How about 3.5 and 4.x?

	We have some issues with IPCCX v3.0 where the calling party info
is only displayed on the phone when you pick up the line and not while
it is ringing. While it is ringing it shows the queue DN (or route
point, I can't remember now). Similarly, in the directories, it does not
show the actual calling party info but again, the queue DN (or route
point).

	 

	Is this the same behaviour in 3.5 and 4? Just wondering before I
put in a feature request.

	 

	Lelio

	 

	
------------------------------------------------------------------------
--------
	Lelio Fulgenzi, B.A.
	Network Analyst (CCS) * University of Guelph * Guelph, Ontario
N1G 2W1
	(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
	
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
	"I had a coffee and Coke at lunch today...and now, I've got more
jitter than an
	IP phone on a long haul 10base2 connection"
LFJ

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From lelio at uoguelph.ca  Tue Dec 13 17:07:02 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Dec 13 17:07:08 2005
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How
	about	3.5 and 4.x?
References: <200512132153.jBDLrjdu004188@ms-smtp-02.rdc-kc.rr.com>
Message-ID: <002c01c60031$8d1ca4e0$30196883@cfs.uoguelph.ca>

Yes - the computer agent desktop shows this on the screen, but there is no historical information on the screen (or the phone). :(
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Philip Walenta 
  To: 'Lelio Fulgenzi' ; cisco-voip at puck.nether.net 
  Sent: Tuesday, December 13, 2005 4:57 PM
  Subject: RE: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How about 3.5 and 4.x?


  This has been an issue for quite some time.  If I remember correctly, this is a CTI limitation.  I though the CAD desktop (3.0 and higher) displayed this information, but I could be wrong.



----------------------------------------------------------------------------
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
    Sent: Tuesday, December 13, 2005 3:39 PM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How about 3.5 and 4.x?


    We have some issues with IPCCX v3.0 where the calling party info is only displayed on the phone when you pick up the line and not while it is ringing. While it is ringing it shows the queue DN (or route point, I can't remember now). Similarly, in the directories, it does not show the actual calling party info but again, the queue DN (or route point).

    Is this the same behaviour in 3.5 and 4? Just wondering before I put in a feature request.

    Lelio

    --------------------------------------------------------------------------------
    Lelio Fulgenzi, B.A.
    Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
    (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
    "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
    IP phone on a long haul 10base2 connection"                                           LFJ
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From lelio at uoguelph.ca  Tue Dec 13 17:10:46 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue Dec 13 17:10:52 2005
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 -
	Howabout	3.5 and 4.x?
References: <D713462ED535184D830F6F6301753E7E01A11FCB@VISHNU.wesd.org>
Message-ID: <005301c60032$12779730$30196883@cfs.uoguelph.ca>

I remember our PSTN carrier mentioning that with analog 800 service (why on Earth would you expect it with a more advanced digital PRI service) you could get both the calling line ID and called 800 number. That way you're call centre knew how to answer the phone. Cisco should do something similar where the both the calling party and the original called party are sent to the line. 

I agree Scott, having the queue information is important for some people so they know what line they are answering.

I'd give up the ID on the incoming call as long as it appeared in the call history.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Voll, Scott 
  To: Philip Walenta ; Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Tuesday, December 13, 2005 5:03 PM
  Subject: RE: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - Howabout 3.5 and 4.x?


  Yes. In the little windows that come with the agent you can see the ANI as the original calling number.  But as far as the display on the phone, the CTI Port is what you see.  This is good for us as we need to know which contact center they are trying to reach.  IE> the internal users or a regional user.  We run two different numbers / IPCC scripts and they go to the same agents. But the CTI port name is different so the agents know which number they called.

   

  Scott

   

  PS.  So If cisco's listening, make it optional not mandatory please.

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
  Sent: Tuesday, December 13, 2005 1:57 PM
  To: 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
  Subject: RE: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - Howabout 3.5 and 4.x?

   

  This has been an issue for quite some time.  If I remember correctly, this is a CTI limitation.  I though the CAD desktop (3.0 and higher) displayed this information, but I could be wrong.

     


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

    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
    Sent: Tuesday, December 13, 2005 3:39 PM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How about 3.5 and 4.x?

    We have some issues with IPCCX v3.0 where the calling party info is only displayed on the phone when you pick up the line and not while it is ringing. While it is ringing it shows the queue DN (or route point, I can't remember now). Similarly, in the directories, it does not show the actual calling party info but again, the queue DN (or route point).

     

    Is this the same behaviour in 3.5 and 4? Just wondering before I put in a feature request.

     

    Lelio

     

    --------------------------------------------------------------------------------
    Lelio Fulgenzi, B.A.
    Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
    (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
    "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
    IP phone on a long haul 10base2 connection"                                           LFJ
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From pwalenta at wi.rr.com  Tue Dec 13 17:16:21 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue Dec 13 17:16:11 2005
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How
	about3.5 and 4.x?
In-Reply-To: <002c01c60031$8d1ca4e0$30196883@cfs.uoguelph.ca>
Message-ID: <200512132212.jBDMCtdu000145@ms-smtp-02.rdc-kc.rr.com>

Actually, depending on the level of CAD you've purchased (I think everything
above "Standard" would work, there is a call history (perhaps just
outbound)?  I don't remember where it's tucked, but I thought there was
"Agent stats" or some such thing.


  _____  

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Tuesday, December 13, 2005 4:07 PM
To: Philip Walenta; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How
about3.5 and 4.x?


Yes - the computer agent desktop shows this on the screen, but there is no
historical information on the screen (or the phone). :(
----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than
an
IP phone on a long haul 10base2 connection"
LFJ

----- Original Message ----- 
From: Philip  <mailto:pwalenta at wi.rr.com> Walenta 
To: 'Lelio Fulgenzi' <mailto:lelio at uoguelph.ca>  ;
cisco-voip at puck.nether.net 
Sent: Tuesday, December 13, 2005 4:57 PM
Subject: RE: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How
about 3.5 and 4.x?

This has been an issue for quite some time.  If I remember correctly, this
is a CTI limitation.  I though the CAD desktop (3.0 and higher) displayed
this information, but I could be wrong.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, December 13, 2005 3:39 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Calling ID Issues with IPCCExpress v3.0 - How about
3.5 and 4.x?


We have some issues with IPCCX v3.0 where the calling party info is only
displayed on the phone when you pick up the line and not while it is
ringing. While it is ringing it shows the queue DN (or route point, I can't
remember now). Similarly, in the directories, it does not show the actual
calling party info but again, the queue DN (or route point).
 
Is this the same behaviour in 3.5 and 4? Just wondering before I put in a
feature request.
 
Lelio
 
----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than
an
IP phone on a long haul 10base2 connection"
LFJ

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From erickbe at yahoo.com  Tue Dec 13 23:29:17 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue Dec 13 23:29:24 2005
Subject: [cisco-voip] H.323 Dial Peer config
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B6D@MREXVS01.bc.broward.cty>
Message-ID: <20051214042918.75183.qmail@web34306.mail.mud.yahoo.com>

Carlos, Here is a sample...


trunk group  pstn

voice-port 0/2/1
 trunk-group pstn
!
voice-port 0/2/2
 trunk-group pstn
!
voice-port 0/2/3
 trunk-group pstn
!
voice-port 0/3/0
 trunk-group pstn
!
voice-port 0/3/1
 trunk-group pstn


dial-peer voice 1800 pots
 trunkgroup pstn
 destination-pattern 9T


--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> Erick,
> 
> All the information I am finding on Trunk Groups
> refers to PRI's.  I can
> create a trunk group at the global level, but I
> don't see an option that
> would allow me to add FXO voice ports to it. See
> below:
> 
> PESPC(config)#trunk group  FXO-G1             
> PESPC(config-trunk-group)#?
> Trunk Group configuration commands:
>   capacity             capacity update timer config
>   carrier-id           Name of the carrier
> associated with this trunk
> group
>   default              Set a command to its defaults
>   description          Trunk group description
>   exit                 Exit from trunk group
> configuration mode
>   help                 Description of the
> interactive help system
>   hunt-scheme          The hunt scheme in trunk
> group
>   isdn                 Specify isdn supplementary
> service
>   max-calls            Maximum number of calls
> allowed
>   max-retry            Maximum number of outgoing
> attempts
>   no                   Negate a command or set its
> defaults
>   tgrep                TGREP config
>   translation-profile  Translation profile
>   voice-class          Set voice class control
> parameters
> 
> 
> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> Sent: Monday, December 12, 2005 10:38 PM
> To: Ortiz, Carlos; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] H.323 Dial Peer config
> 
> On the dial-peers, you are forwarding all digits so
> that will also send the 9 that starts all your
> destination patterns. 
> 
> Usually with FXO, there is no inbound caller id to
> tell it where to route call so on the voice-port you
> need to use connection plar to route call somewhere
> (This is same as Attendant DN setting on the
> endpoint
> in CCM when using MGCP). 
> 
> They have added trunk groups to IOS, so you can
> assign
> voice ports to a trunkgroup, then have one dial-peer
> pointing to the trunk group. You can then assign
> preferences to voice-port that is a member of the
> trunk group, etc. This requires a trunk group to be
> configured globally. Under the trunkgroup config you
> can configure hunt method, etc. 
> 
> Erick
> 
> --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
> 
> > I have never configured my FXO PSTN router
> > connections with H.323, only
> > MGCP.  I did configure H.323 for PRI interfaces.
> > Going by that and what
> > I have refreshed with CCO docs, I have come up
> with
> > a starting config to
> > do FXO with H.323. Dial peer 1 would be for the
> > incoming call leg on all
> > calls.  This assumes a 4 port FXO card.  Does this
> > appear correct?  Have
> > there been any improvements on the "hunting" for
> > outbound calls beyond
> > using the preference command? Any suggestions or
> > help would be
> > appreciated.
> > 
> >  
> > 
> > dial-peer voice 1 pots
> > 
> >  description ** Inbound Call leg all calls **
> > 
> >  incoming called-number .
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/0
> > 
> > !
> > 
> > dial-peer voice 954 pots
> > 
> >  description ** Local 954,754 Calls **
> > 
> >  destination-pattern 9[7,9]54.......
> > 
> >  progress_ind setup enable 3
> > 
> >  progress_ind alert enable 8
> > 
> >  progress_ind progress enable 8
> > 
> >  progress_ind connect enable 8
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/0
> > 
> >  forward-digits all
> > 
> >  
> > 
> > dial-peer voice 9542 pots
> > 
> >  description ** Local 954,754 Calls **
> > 
> >  destination-pattern 9[7,9]54.......
> > 
> >  preference 1
> > 
> >  progress_ind setup enable 3
> > 
> >  progress_ind alert enable 8
> > 
> >  progress_ind progress enable 8
> > 
> >  progress_ind connect enable 8
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/1
> > 
> >  forward-digits all
> > 
> >  
> > 
> > dial-peer voice 9543 pots
> > 
> >  description ** Local 954,754 Calls **
> > 
> >  destination-pattern 9[7,9]54.......
> > 
> >  preference 2
> > 
> >  progress_ind setup enable 3
> > 
> >  progress_ind alert enable 8
> > 
> >  progress_ind progress enable 8
> > 
> >  progress_ind connect enable 8
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/2
> > 
> >  forward-digits all
> > 
> >  
> > 
> > dial-peer voice 9544 pots
> > 
> >  description ** Local 954,754 Calls **
> > 
> >  destination-pattern 9[7,9]54.......
> > 
> >  preference 3
> > 
> >  progress_ind setup enable 3
> > 
> >  progress_ind alert enable 8
> > 
> >  progress_ind progress enable 8
> > 
> >  progress_ind connect enable 8
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/3
> > 
> >  forward-digits all
> > 
> >  
> > 
> > dial-peer voice 9999 voip
> > 
> >  description ** Inbound 357 nums to mr-cm-sub1 **
> > 
> >  destination-pattern [5-8]...
> > 
> >  progress_ind setup enable 3
> > 
> >  session target ipv4:10.100.200.21
> > 
> >  incoming called-number .
> > 
> >  dtmf-relay h245-alphanumeric
> > 
> >  codec g711ulaw
> > 
> > !
> > 
> > dial-peer voice 99992 voip
> > 
> >  description ** Inbound 357 nums to mr-cm-pub **
> > 
> >  preference 2
> > 
> >  destination-pattern [5-8]...
> > 
> >  progress_ind setup enable 3
> > 
> >  session target ipv4:10.100.200.20
> > 
> >  incoming called-number .
> > 
> >  dtmf-relay h245-alphanumeric
> > 
> >  codec g711ulaw
> > 
> >  
> > 
> >  
> > 
> > Carlos Ortiz
> > 
> > Senior Information Technology Specialist
> > 
> > Broward County Office of Information Technology
> > 
> > Data Communications
> > 
> > 954-357-7981
> > 
> > cortiz at broward.org
> > 
> >  
> > 
> >  
> > 
> > > _______________________________________________
> > 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 
> 


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

From PRuttman at foley.com  Wed Dec 14 09:52:49 2005
From: PRuttman at foley.com (Ruttman, Peter G.)
Date: Wed Dec 14 09:52:52 2005
Subject: [cisco-voip] Caller ID names and 4.x
Message-ID: <7E3064B7EFC16C41B185C41D141008B7BBBC35@foleyex97.foleylaw.com>

We have two separate clusters (4.02 and 4.13) both that have issues
where we mask our outbound caller id numbers with our main number.  For
the name we would like it to be the name of our company.  For these two
offices we get occasional reports that a person that was once with our
company or is still with the company is showing up on the caller ID for
people receiving calls from us.  In the case of the retired person that
person is not even listed on our Callmanager.  He also only seems to
show up when the call ends up at certain carriers like Vonage.  

Does Callmanager control anything with caller ID name?

Pete

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 davidp at bostonpost.com  Wed Dec 14 10:05:08 2005
From: davidp at bostonpost.com (David Pitkin)
Date: Wed Dec 14 10:05:10 2005
Subject: [cisco-voip] Caller ID names and 4.x
Message-ID: <4DEBDDAFBB23C04BA17EFE391444211301547AA7@exchange.bostonpost.com>

My understanding is that CallManager "could" but that US Carriers do not
do it. Cell Phones and other corporate PBX's also do not receive the
text.


When I need to modify the text I first get the numbers right and then
work with our carrier to get the text right and confirm it by calling a
Vonage number (and then checking the nice web page which shows the text)

 

I believe that even my CLEC has to make the request to the ILEC (Verizon
here in the NE)

 

David [ p. 603-669-8553 x6908 e. davidp at bostonpost.com ]

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter
G.
Sent: Wednesday, December 14, 2005 9:53 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Caller ID names and 4.x

 

We have two separate clusters (4.02 and 4.13) both that have issues
where we mask our outbound caller id numbers with our main number.  For
the name we would like it to be the name of our company.  For these two
offices we get occasional reports that a person that was once with our
company or is still with the company is showing up on the caller ID for
people receiving calls from us.  In the case of the retired person that
person is not even listed on our Callmanager.  He also only seems to
show up when the call ends up at certain carriers like Vonage.  

Does Callmanager control anything with caller ID name? 

Pete 


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 erickbe at yahoo.com  Wed Dec 14 10:16:43 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Dec 14 10:16:51 2005
Subject: [cisco-voip] Caller ID names and 4.x
In-Reply-To: <7E3064B7EFC16C41B185C41D141008B7BBBC35@foleyex97.foleylaw.com>
Message-ID: <20051214151643.58542.qmail@web34309.mail.mud.yahoo.com>

If you have display IE checked on the gateway, call
manager will send the name out in setup message. Most
all carriers ignore that though. The carriers look up
the calling party number to a name in the database on
the carriers network and thats where the name comes
from passed to party being called. I would uncheck
Display IE on links to the public network. I have also
seen that confuse some carriers switches in a few
cases resulting in the calling party # to name
database lookup not being done right.

But the last carrier (where the call terminates) could
use some other database then another carrier so thats
why names may show up differently sometimes depending
on where you call. No real easy way to fix this that I
know of except to push your carrier to make sure the
name is right for your numbers or call the carrier
that the person you're calling is using and see if you
can get them to update their records.

The other thing is, make sure if using ISDN , that the
numbering plan and type is set correctly for calling
party number and/or called party number. If you're in
the US, setting the calling party number to ISDN and
National is usually a good starting point. You may
have to set different numbering plans and types for
different numbers called though as some carriers like
to see different settings , and other carriers down
the line may want something else then another carrier
to pass caller id right so it shows up. Setting the
called party number type and plan to Unknown/Unknown
is a good starting point also, but same thing applies
-- it depends on carriers, types of switches used, and
numbers called and what they want to see.

In most cases, the call will connect and work fine,
but caller id and other stuff may not work as expected
if there is a numbering plan/type not liked by someone
along the way.


--- "Ruttman, Peter G." <PRuttman at foley.com> wrote:

> We have two separate clusters (4.02 and 4.13) both
> that have issues
> where we mask our outbound caller id numbers with
> our main number.  For
> the name we would like it to be the name of our
> company.  For these two
> offices we get occasional reports that a person that
> was once with our
> company or is still with the company is showing up
> on the caller ID for
> people receiving calls from us.  In the case of the
> retired person that
> person is not even listed on our Callmanager.  He
> also only seems to
> show up when the call ends up at certain carriers
> like Vonage.  
> 
> Does Callmanager control anything with caller ID
> name?
> 
> Pete
> 
> 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
> 


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

From pwalenta at wi.rr.com  Wed Dec 14 10:20:54 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Wed Dec 14 10:20:45 2005
Subject: [cisco-voip] Caller ID names and 4.x
In-Reply-To: <7E3064B7EFC16C41B185C41D141008B7BBBC35@foleyex97.foleylaw.com>
Message-ID: <200512141517.jBEFHTgu022122@ms-smtp-03.rdc-kc.rr.com>

Has their phone in the system been deleted?  If so, has their directory
number also been deleted?  Is it possible (possibly far fetched) that a
deleted number is somehow "leaking" out the gateway?


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter G.
Sent: Wednesday, December 14, 2005 8:53 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Caller ID names and 4.x



We have two separate clusters (4.02 and 4.13) both that have issues where we
mask our outbound caller id numbers with our main number.  For the name we
would like it to be the name of our company.  For these two offices we get
occasional reports that a person that was once with our company or is still
with the company is showing up on the caller ID for people receiving calls
from us.  In the case of the retired person that person is not even listed
on our Callmanager.  He also only seems to show up when the call ends up at
certain carriers like Vonage.  

Does Callmanager control anything with caller ID name? 

Pete 


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 jkoniecki at comcast.net  Wed Dec 14 10:20:47 2005
From: jkoniecki at comcast.net (Jim Koniecki)
Date: Wed Dec 14 10:20:58 2005
Subject: [cisco-voip] Caller ID names and 4.x- Same Problem in the past.
In-Reply-To: <7E3064B7EFC16C41B185C41D141008B7BBBC35@foleyex97.foleylaw.com>
Message-ID: <000a01c600c1$f6358ab0$c800a8c0@Jalbert1>

Peter, 

 

I had a similar problem just the other week. In my case the "retired" user
had been with the company for 15 years and had the same DID number for his
employment duration. His calling name and number were still configured from
the old Centrex service they used to have before replacing it with an AVVID
system.  We had to call the carried and have them clear out this entry on
the trunk.

 

I hope this helps. 

 

Jim K 

 

 

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter G.
Sent: Wednesday, December 14, 2005 9:53 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Caller ID names and 4.x

 

We have two separate clusters (4.02 and 4.13) both that have issues where we
mask our outbound caller id numbers with our main number.  For the name we
would like it to be the name of our company.  For these two offices we get
occasional reports that a person that was once with our company or is still
with the company is showing up on the caller ID for people receiving calls
from us.  In the case of the retired person that person is not even listed
on our Callmanager.  He also only seems to show up when the call ends up at
certain carriers like Vonage.  

Does Callmanager control anything with caller ID name? 

Pete 


The preceding email message may be confidential or protected by the
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From Nick.Kassel at Charles-Stanley.co.uk  Wed Dec 14 10:28:31 2005
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed Dec 14 10:28:41 2005
Subject: [cisco-voip] ISDN Simulator
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC0C0166@CSEXCH01.charles-stanley.co.uk>

I'm looking to incorporate an ISDN simulator into our test lab, does any
body use these in their labs and if so can you recommend one worth
looking at for PRI use.

 

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malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
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From lelio at uoguelph.ca  Wed Dec 14 10:34:16 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Dec 14 10:34:22 2005
Subject: [cisco-voip] ISDN Simulator
References: <253FAE7A06B11B49A3A29770B2B8BAFC0C0166@CSEXCH01.charles-stanley.co.uk>
Message-ID: <007901c600c3$d8013ba0$30196883@cfs.uoguelph.ca>

I've looked at Adtran a few times. But after one of our SE's pointed me to T1 crossover cables, we went to using two 6608 ports for our testing - one registered to our production cluster and one registered to our test cluster. Saved us a few dollars.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Nick Kassel 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, December 14, 2005 10:28 AM
  Subject: [cisco-voip] ISDN Simulator


  I'm looking to incorporate an ISDN simulator into our test lab, does any body use these in their labs and if so can you recommend one worth looking at for PRI use.






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

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        Registered in England No. 1903304

        Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles Stanley & Co. Ltd

        Authorised and Regulated by the Financial Services Authority, Member of the
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From highspeedsnow at gmail.com  Wed Dec 14 10:33:26 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Wed Dec 14 10:34:34 2005
Subject: [cisco-voip] ISDN Simulator
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC0C0166@CSEXCH01.charles-stanley.co.uk>
Message-ID: <43a03b85.2bef00eb.3b0c.3fbb@mx.gmail.com>

Nick,

 

The Madge 60 or the Atlas 500 are always good bets.

 

-Mark Snow 

CCIE Voice # 14073

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Wednesday, December 14, 2005 10:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ISDN Simulator

 

I'm looking to incorporate an ISDN simulator into our test lab, does any
body use these in their labs and if so can you recommend one worth looking
at for PRI use.

 

 



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

The information contained in this e-mail is strictly confidential, some or
all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
****************************************************************************
********************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets 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 highspeedsnow at gmail.com  Wed Dec 14 10:41:34 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Wed Dec 14 10:42:45 2005
Subject: [cisco-voip] Caller ID names and 4.x
In-Reply-To: <20051214151643.58542.qmail@web34309.mail.mud.yahoo.com>
Message-ID: <43a03d71.571d8e10.7ca9.393d@mx.gmail.com>

Peter,

Erick put it as best it can be put - I am confident this is exactly what is
happening to you with the Carrier's DB names being incorrect - and probably
the terminating carrier's DB.


-Mark Snow 
CCIE Voice # 14073

> -----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, December 14, 2005 10:17 AM
> To: Ruttman, Peter G.; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Caller ID names and 4.x
> 
> If you have display IE checked on the gateway, call
> manager will send the name out in setup message. Most
> all carriers ignore that though. The carriers look up
> the calling party number to a name in the database on
> the carriers network and thats where the name comes
> from passed to party being called. I would uncheck
> Display IE on links to the public network. I have also
> seen that confuse some carriers switches in a few
> cases resulting in the calling party # to name
> database lookup not being done right.
> 
> But the last carrier (where the call terminates) could
> use some other database then another carrier so thats
> why names may show up differently sometimes depending
> on where you call. No real easy way to fix this that I
> know of except to push your carrier to make sure the
> name is right for your numbers or call the carrier
> that the person you're calling is using and see if you
> can get them to update their records.
> 
> The other thing is, make sure if using ISDN , that the
> numbering plan and type is set correctly for calling
> party number and/or called party number. If you're in
> the US, setting the calling party number to ISDN and
> National is usually a good starting point. You may
> have to set different numbering plans and types for
> different numbers called though as some carriers like
> to see different settings , and other carriers down
> the line may want something else then another carrier
> to pass caller id right so it shows up. Setting the
> called party number type and plan to Unknown/Unknown
> is a good starting point also, but same thing applies
> -- it depends on carriers, types of switches used, and
> numbers called and what they want to see.
> 
> In most cases, the call will connect and work fine,
> but caller id and other stuff may not work as expected
> if there is a numbering plan/type not liked by someone
> along the way.
> 
> 
> --- "Ruttman, Peter G." <PRuttman at foley.com> wrote:
> 
> > We have two separate clusters (4.02 and 4.13) both
> > that have issues
> > where we mask our outbound caller id numbers with
> > our main number.  For
> > the name we would like it to be the name of our
> > company.  For these two
> > offices we get occasional reports that a person that
> > was once with our
> > company or is still with the company is showing up
> > on the caller ID for
> > people receiving calls from us.  In the case of the
> > retired person that
> > person is not even listed on our Callmanager.  He
> > also only seems to
> > show up when the call ends up at certain carriers
> > like Vonage.
> >
> > Does Callmanager control anything with caller ID
> > name?
> >
> > Pete
> >
> > 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
> >
> 
> 
> __________________________________________________
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> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip


From highspeedsnow at gmail.com  Wed Dec 14 10:42:52 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Wed Dec 14 10:44:02 2005
Subject: [cisco-voip] ISDN Simulator
In-Reply-To: <007901c600c3$d8013ba0$30196883@cfs.uoguelph.ca>
Message-ID: <43a03dbf.61261d77.7bc2.531e@mx.gmail.com>

Yes actually using cisco gateways in the middle with a CCM is the best way
to prepare for the exam.

 

 

-Mark Snow 

CCIE Voice # 14073

 

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, December 14, 2005 10:34 AM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ISDN Simulator

 

I've looked at Adtran a few times. But after one of our SE's pointed me to
T1 crossover cables, we went to using two 6608 ports for our testing - one
registered to our production cluster and one registered to our test cluster.
Saved us a few dollars.

 

----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than
an
IP phone on a long haul 10base2 connection"
LFJ

----- Original Message ----- 

From: Nick Kassel <mailto:Nick.Kassel at charles-stanley.co.uk>  

To: cisco-voip at puck.nether.net 

Sent: Wednesday, December 14, 2005 10:28 AM

Subject: [cisco-voip] ISDN Simulator

 

I'm looking to incorporate an ISDN simulator into our test lab, does any
body use these in their labs and if so can you recommend one worth looking
at for PRI use.

 

 



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

The information contained in this e-mail is strictly confidential, some or
all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
****************************************************************************
********************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets 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 Nick.Kassel at Charles-Stanley.co.uk  Wed Dec 14 10:50:43 2005
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed Dec 14 10:50:55 2005
Subject: [cisco-voip] ISDN Simulator
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC0C0167@CSEXCH01.charles-stanley.co.uk>

Thanks any idea on the rough cost of the Adtran Atlas 550

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Wednesday, December 14, 2005 3:34 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ISDN Simulator

 

I've looked at Adtran a few times. But after one of our SE's pointed me
to T1 crossover cables, we went to using two 6608 ports for our testing
- one registered to our production cluster and one registered to our
test cluster. Saved us a few dollars.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Nick Kassel <mailto:Nick.Kassel at charles-stanley.co.uk>  

	To: cisco-voip at puck.nether.net 

	Sent: Wednesday, December 14, 2005 10:28 AM

	Subject: [cisco-voip] ISDN Simulator

	 

	I'm looking to incorporate an ISDN simulator into our test lab,
does any body use these in their labs and if so can you recommend one
worth looking at for PRI use.

	 

	 


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

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets 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

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

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets 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 lelio at uoguelph.ca  Wed Dec 14 10:51:24 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Dec 14 10:51:27 2005
Subject: [cisco-voip] Caller ID names and 4.x
References: <7E3064B7EFC16C41B185C41D141008B7BBBC35@foleyex97.foleylaw.com>
Message-ID: <00d401c600c6$3c9d3c60$30196883@cfs.uoguelph.ca>

Caller ID names and 4.xI'm not sure if I've mentioned this before, my apologies if I have. We ran into this problem a while back and my research found out some interesting things. In the US, I believe it is up to the called party's provider to do the directory lookup and provide the calling name/number ID. In Canada, however, it is up to the calling party provider to send the calling name/number ID. What caused us the problem was that our new digital PRI service does not have the switch parameter enabled for our provider to populate so we send no calling name whatsoever. What compounded the issue was that we had a number of franchises on campus that listed themselves in the directory with our main number. Typically the last person to list in the directory with a number is the one that is displayed in the US, so we were having calls appear as coming from a local hamburger chain!

As other's have mentioned, what has likely happened, is there is a DID with a person's name in the directory that is being caught somewhere.

For those Canadians out there with Bell PRI service, I've put a feature request into CIPTUG - so please vote! ;)

Lelio


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Ruttman, Peter G. 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, December 14, 2005 9:52 AM
  Subject: [cisco-voip] Caller ID names and 4.x


  We have two separate clusters (4.02 and 4.13) both that have issues where we mask our outbound caller id numbers with our main number.  For the name we would like it to be the name of our company.  For these two offices we get occasional reports that a person that was once with our company or is still with the company is showing up on the caller ID for people receiving calls from us.  In the case of the retired person that person is not even listed on our Callmanager.  He also only seems to show up when the call ends up at certain carriers like Vonage.  

  Does Callmanager control anything with caller ID name? 

  Pete 


  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 lelio at uoguelph.ca  Wed Dec 14 10:54:46 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Dec 14 10:54:51 2005
Subject: [cisco-voip] ISDN Simulator
References: <253FAE7A06B11B49A3A29770B2B8BAFC0C0167@CSEXCH01.charles-stanley.co.uk>
Message-ID: <00f001c600c6$b52a4a10$30196883@cfs.uoguelph.ca>

http://www.epinions.com/enth-Multislot_Chassis-All-ATLAS_550_T1_TO_PRI_CONVRT

2300 new 1500 refurb?

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Nick Kassel 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Wednesday, December 14, 2005 10:50 AM
  Subject: RE: [cisco-voip] ISDN Simulator


  Thanks any idea on the rough cost of the Adtran Atlas 550




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

  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
  Sent: Wednesday, December 14, 2005 3:34 PM
  To: Nick Kassel; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] ISDN Simulator



  I've looked at Adtran a few times. But after one of our SE's pointed me to T1 crossover cables, we went to using two 6608 ports for our testing - one registered to our production cluster and one registered to our test cluster. Saved us a few dollars.



  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
  IP phone on a long haul 10base2 connection"                                           LFJ

    ----- Original Message ----- 

    From: Nick Kassel 

    To: cisco-voip at puck.nether.net 

    Sent: Wednesday, December 14, 2005 10:28 AM

    Subject: [cisco-voip] ISDN Simulator



    I'm looking to incorporate an ISDN simulator into our test lab, does any body use these in their labs and if so can you recommend one worth looking at for PRI use.






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        You acknowledge that e-mails are not secure and you accept the risk 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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

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From tim.reimers at asheville.k12.nc.us  Wed Dec 14 11:17:02 2005
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed Dec 14 11:17:10 2005
Subject: [cisco-voip] MOH audio source..
Message-ID: <050944EE48FF384F9C663F7048C47DA701237E3A@webmail.asheville.k12.nc.us>

hi --
 
for those who may be interested, I've now succeeded in configuring a MOH
audio source on CCM 4.0
 
If you are like me, your servers do not have audio cards in them--
 
Get a Griffin "iMic" USB audio adapter
http://www.cdwg.com/shop/products/default.aspx?EDC=422596
CDW Part:  	 422596	
Mfg. Part:  	 2020-IMIC	
 
Mine just plugged right into the CM - no drivers needed.
 
I did have to reboot the CCM to get it to work as an audio device-- 
 
After doing that,
 
In CCMAdmin  -- Service >> Media Resources >> MOH Audio Source 
Option 51 - Fixed Audio Source
 
All you have to do is add the MOH Audio Source Device  -- which for the
Griffin, is "USB Audio Source"
 
Update, Reset MOH Servers
 
 
 
Now --- as to the question I have....
 
What is the best way to 'group' change all extensions to using the 51
Fixed Audio source instead of, say '2 - All In the Mind'
 
I don't _really_ want to try using BAT or something--- there has to be a
way to select a whole group of phones and make this one change on all of
them at once...
 
anyone?
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From Court.Schuett at us.millwardbrown.com  Wed Dec 14 11:20:46 2005
From: Court.Schuett at us.millwardbrown.com (Schuett, Court (MBNAP it))
Date: Wed Dec 14 11:20:55 2005
Subject: [cisco-voip] MOH audio source..
Message-ID: <90B962BEA7D42F40BF0DF00044A5A1B60B32F5CC@x-np5.am.group.local>

Ideally, they aren't configured on each device or line.  The best place to
configure them is in the Device Pool.  Configure them there, then leave them
blank in each phone.  A change to the Device Pool will change all of them.
However, if there is already something in the device/line, it will override
the Device Pool settings.  That's why they should all be left blank unless
they are an exception.
 
Hope that helps.
 
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
<mailto:court.schuett at us.millwardbrown.com> 
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, December 14, 2005 10:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MOH audio source..


hi --
 
for those who may be interested, I've now succeeded in configuring a MOH
audio source on CCM 4.0
 
If you are like me, your servers do not have audio cards in them--
 
Get a Griffin "iMic" USB audio adapter
http://www.cdwg.com/shop/products/default.aspx?EDC=422596
<http://www.cdwg.com/shop/products/default.aspx?EDC=422596> 
CDW Part:  	 422596	
Mfg. Part:  	 2020-IMIC	
 
Mine just plugged right into the CM - no drivers needed.
 
I did have to reboot the CCM to get it to work as an audio device-- 
 
After doing that,
 
In CCMAdmin  -- Service >> Media Resources >> MOH Audio Source 
Option 51 - Fixed Audio Source
 
All you have to do is add the MOH Audio Source Device  -- which for the
Griffin, is "USB Audio Source"
 
Update, Reset MOH Servers
 
 
 
Now --- as to the question I have....
 
What is the best way to 'group' change all extensions to using the 51 Fixed
Audio source instead of, say '2 - All In the Mind'
 
I don't _really_ want to try using BAT or something--- there has to be a way
to select a whole group of phones and make this one change on all of them at
once...
 
anyone?
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From Scott.Voll at wesd.org  Wed Dec 14 11:31:54 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 14 11:31:41 2005
Subject: [cisco-voip] ethernet disconnected.
Message-ID: <D713462ED535184D830F6F6301753E7E01A12004@VISHNU.wesd.org>

Call me weird or something but how does a 7940 phone boot up off PoE and
then go to Ethernet disconnected?  Still has power to the phone.  Any
ideas as to what could be the problem?  Is this a Switch problem?  I
tested the cable and it seems to be good.  Is this a IOS bug?  

Hope someone can give me a hand.

Scott



From davidp at bostonpost.com  Wed Dec 14 11:31:57 2005
From: davidp at bostonpost.com (David Pitkin)
Date: Wed Dec 14 11:31:58 2005
Subject: [cisco-voip] IP Communicator and IPCC Express
Message-ID: <4DEBDDAFBB23C04BA17EFE391444211301547AAD@exchange.bostonpost.com>

Has anyone successfully used an IP Communicator as an IPCC express
agent? I am unable to pick the IP Communicator as an ICD extension in
the User Directory.

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From Court.Schuett at us.millwardbrown.com  Wed Dec 14 11:35:08 2005
From: Court.Schuett at us.millwardbrown.com (Schuett, Court (MBNAP it))
Date: Wed Dec 14 11:35:19 2005
Subject: [cisco-voip] MOH audio source..
Message-ID: <90B962BEA7D42F40BF0DF00044A5A1B60B32F5CD@x-np5.am.group.local>

I'm going through the exact same thing.  Got about 300 devices to set right.
There are a lot of things you shouldn't set on the Device/Line.  Check out
the Device Pool settings and pretty much anything in there, don't put on the
Device/Line.  It'll make it way easier to make changes down the road.
 
Might want to check on something else while you're doing this.  (I ran into
the same problem).  On each phone, check to see if they've got a second line
configured with a slightly altered extension.  For example, extension 8983
for Line 1 and 9983 for Line 2.  A lot of times, the vendors place a second
line on every phone and then have the CallForwardBusy roll to that.  What
you should do is just make sure that Maximum Number of calls is set (I use
4) and the Busy trigger is set (I use 2).  The 8983/9983 is an older way of
doing the same thing.  It chews up extensions (probably uses a seperate
partition too) and can interfere with CallBack features.  Just something
I've run into a lot with vendors.  
 
Good luck fixing the mess! ;)
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
<mailto:court.schuett at us.millwardbrown.com> 
 

  _____  

From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
Sent: Wednesday, December 14, 2005 10:30 AM
To: Schuett, Court (MBNAP it)
Subject: RE: [cisco-voip] MOH audio source..


argh... the vendor already configured it on each of the lines in each
phone... sigh...
 
 
Well, I'll add it to the Device pools... that way, I can slowly remove from
each line... <muttering comments about the vendor..>
 
Tim

  _____  

From: Schuett, Court (MBNAP it) [mailto:Court.Schuett at us.millwardbrown.com] 
Sent: Wednesday, December 14, 2005 11:21 AM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..


Ideally, they aren't configured on each device or line.  The best place to
configure them is in the Device Pool.  Configure them there, then leave them
blank in each phone.  A change to the Device Pool will change all of them.
However, if there is already something in the device/line, it will override
the Device Pool settings.  That's why they should all be left blank unless
they are an exception.
 
Hope that helps.
 
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
<mailto:court.schuett at us.millwardbrown.com> 
 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, December 14, 2005 10:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MOH audio source..


hi --
 
for those who may be interested, I've now succeeded in configuring a MOH
audio source on CCM 4.0
 
If you are like me, your servers do not have audio cards in them--
 
Get a Griffin "iMic" USB audio adapter
http://www.cdwg.com/shop/products/default.aspx?EDC=422596
<http://www.cdwg.com/shop/products/default.aspx?EDC=422596> 
CDW Part:  	 422596	
Mfg. Part:  	 2020-IMIC	
 
Mine just plugged right into the CM - no drivers needed.
 
I did have to reboot the CCM to get it to work as an audio device-- 
 
After doing that,
 
In CCMAdmin  -- Service >> Media Resources >> MOH Audio Source 
Option 51 - Fixed Audio Source
 
All you have to do is add the MOH Audio Source Device  -- which for the
Griffin, is "USB Audio Source"
 
Update, Reset MOH Servers
 
 
 
Now --- as to the question I have....
 
What is the best way to 'group' change all extensions to using the 51 Fixed
Audio source instead of, say '2 - All In the Mind'
 
I don't _really_ want to try using BAT or something--- there has to be a way
to select a whole group of phones and make this one change on all of them at
once...
 
anyone?
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From lelio at uoguelph.ca  Wed Dec 14 11:40:46 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Dec 14 11:40:50 2005
Subject: [cisco-voip] ethernet disconnected.
References: <D713462ED535184D830F6F6301753E7E01A12004@VISHNU.wesd.org>
Message-ID: <016401c600cd$221d7790$30196883@cfs.uoguelph.ca>

Is the Ethernet disconnected message coming from the phone or the PC? 

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Voll, Scott 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, December 14, 2005 11:31 AM
  Subject: [cisco-voip] ethernet disconnected.


  Call me weird or something but how does a 7940 phone boot up off PoE and
  then go to Ethernet disconnected?  Still has power to the phone.  Any
  ideas as to what could be the problem?  Is this a Switch problem?  I
  tested the cable and it seems to be good.  Is this a IOS bug?  

  Hope someone can give me a hand.

  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  Wed Dec 14 11:42:20 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 14 11:42:04 2005
Subject: [cisco-voip] ethernet disconnected.
Message-ID: <D713462ED535184D830F6F6301753E7E01A12006@VISHNU.wesd.org>

Phone.

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Wednesday, December 14, 2005 8:41 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ethernet disconnected.

 

Is the Ethernet disconnected message coming from the phone or the PC? 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Voll, Scott <mailto:Scott.Voll at wesd.org>  

	To: cisco-voip at puck.nether.net 

	Sent: Wednesday, December 14, 2005 11:31 AM

	Subject: [cisco-voip] ethernet disconnected.

	 

	Call me weird or something but how does a 7940 phone boot up off
PoE and
	then go to Ethernet disconnected?  Still has power to the phone.
Any
	ideas as to what could be the problem?  Is this a Switch
problem?  I
	tested the cable and it seems to be good.  Is this a IOS bug?  
	
	Hope someone can give me a hand.
	
	Scott
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

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From Michael_Davis at eLoyalty.com  Wed Dec 14 11:51:20 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Wed Dec 14 11:51:45 2005
Subject: [cisco-voip] MOH audio source..
Message-ID: <037594FB5AFC354FA38653EB3FF324210385F15F@elhqex01.eloyaltyco.com>

I agree with you regarding the device pools, but why wouldn't you use BAT to modify phone MRGLs?
 
Michael
CCIE #7303 (Voice/R&S)
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net]On Behalf Of Schuett, Court (MBNAP it)
Sent: Wednesday, December 14, 2005 10:35 AM
To: 'Tim Reimers'
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..


I'm going through the exact same thing.  Got about 300 devices to set right.  There are a lot of things you shouldn't set on the Device/Line.  Check out the Device Pool settings and pretty much anything in there, don't put on the Device/Line.  It'll make it way easier to make changes down the road.
 
Might want to check on something else while you're doing this.  (I ran into the same problem).  On each phone, check to see if they've got a second line configured with a slightly altered extension.  For example, extension 8983 for Line 1 and 9983 for Line 2.  A lot of times, the vendors place a second line on every phone and then have the CallForwardBusy roll to that.  What you should do is just make sure that Maximum Number of calls is set (I use 4) and the Busy trigger is set (I use 2).  The 8983/9983 is an older way of doing the same thing.  It chews up extensions (probably uses a seperate partition too) and can interfere with CallBack features.  Just something I've run into a lot with vendors.  
 
Good luck fixing the mess! ;)
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
 

  _____  

From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
Sent: Wednesday, December 14, 2005 10:30 AM
To: Schuett, Court (MBNAP it)
Subject: RE: [cisco-voip] MOH audio source..


argh... the vendor already configured it on each of the lines in each phone... sigh...
 
 
Well, I'll add it to the Device pools... that way, I can slowly remove from each line... <muttering comments about the vendor..>
 
Tim

  _____  

From: Schuett, Court (MBNAP it) [mailto:Court.Schuett at us.millwardbrown.com] 
Sent: Wednesday, December 14, 2005 11:21 AM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..


Ideally, they aren't configured on each device or line.  The best place to configure them is in the Device Pool.  Configure them there, then leave them blank in each phone.  A change to the Device Pool will change all of them.  However, if there is already something in the device/line, it will override the Device Pool settings.  That's why they should all be left blank unless they are an exception.
 
Hope that helps.
 
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
 

  _____  

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, December 14, 2005 10:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MOH audio source..


hi --
 
for those who may be interested, I've now succeeded in configuring a MOH audio source on CCM 4.0
 
If you are like me, your servers do not have audio cards in them--
 
Get a Griffin "iMic" USB audio adapter
http://www.cdwg.com/shop/products/default.aspx?EDC=422596

CDW Part:  	 422596	
Mfg. Part:  	 2020-IMIC	
 
Mine just plugged right into the CM - no drivers needed.
 
I did have to reboot the CCM to get it to work as an audio device-- 
 
After doing that,
 
In CCMAdmin  -- Service >> Media Resources >> MOH Audio Source 
Option 51 - Fixed Audio Source
 
All you have to do is add the MOH Audio Source Device  -- which for the Griffin, is "USB Audio Source"
 
Update, Reset MOH Servers
 
 
 
Now --- as to the question I have....
 
What is the best way to 'group' change all extensions to using the 51 Fixed Audio source instead of, say '2 - All In the Mind'
 
I don't _really_ want to try using BAT or something--- there has to be a way to select a whole group of phones and make this one change on all of them at once...
 
anyone?

-------------- next part --------------
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From jose.mujica at netxar.com  Wed Dec 14 12:02:25 2005
From: jose.mujica at netxar.com (Jose Mujica)
Date: Wed Dec 14 12:02:27 2005
Subject: [cisco-voip] CCM 4.1 and AAR Problem.. Configuration?
Message-ID: <20051214165444.M43263@netxar.com>

Hi.

 I have a problem in making AAR to work. I have a central site and a remote
site. I have an AAR Group with Prefix '91'. All External Phone Number Mask
have been configure properly. I also created a CSS and Partitions for AAR
functionality and I add a route pattern to it that will match the prefix + the
external number mask and is routed to the remote site branch gateway in order
to reach the central site user using the PSTN.

THe problem is that the setup of the call never get to the router and the
display of the remote ip phones says Network Congestion. Rerouting and after
that the display change to 'Enter Number' and stays there... The remote ip
phone and the central ip phone are in the same AAR Group and the remote ip
phone has the CSS for AAR set, also the central site has the CSS AAR set as
well as the branch voip gateway. The gateway is an MGCP gateway using FXO lines.


Attached is the CCM trace when AAR kicks in.. Just wondering if anyone has the
same problem before or if it is a misconfiguration in my part.

Remote IP Phone DN: 1811
Central Site IP Phone 2227
AAR Group - Prefix '91'
External Phone Number Mask of 2227 is 7877588585


Thanks in advanced,
-Jose

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From Scott.Voll at wesd.org  Wed Dec 14 12:16:00 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 14 12:15:45 2005
Subject: [cisco-voip] MOH audio source..
Message-ID: <D713462ED535184D830F6F6301753E7E01A1200D@VISHNU.wesd.org>

I have not used BAT in the past but It looks like I should.  I'm going
through the BAT tool and it shows (CM4.1) I can change the hold audio
sources both user and network.  Question: does this do in on both the
device and line or only the device?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Davis, Michael
Sent: Wednesday, December 14, 2005 8:51 AM
To: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..

 

I agree with you regarding the device pools, but why wouldn't you use
BAT to modify phone MRGLs?

 

Michael

CCIE #7303 (Voice/R&S)

 

	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net]On Behalf Of Schuett, Court
(MBNAP it)
	Sent: Wednesday, December 14, 2005 10:35 AM
	To: 'Tim Reimers'
	Cc: cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] MOH audio source..

	I'm going through the exact same thing.  Got about 300 devices
to set right.  There are a lot of things you shouldn't set on the
Device/Line.  Check out the Device Pool settings and pretty much
anything in there, don't put on the Device/Line.  It'll make it way
easier to make changes down the road.

	 

	Might want to check on something else while you're doing this.
(I ran into the same problem).  On each phone, check to see if they've
got a second line configured with a slightly altered extension.  For
example, extension 8983 for Line 1 and 9983 for Line 2.  A lot of times,
the vendors place a second line on every phone and then have the
CallForwardBusy roll to that.  What you should do is just make sure that
Maximum Number of calls is set (I use 4) and the Busy trigger is set (I
use 2).  The 8983/9983 is an older way of doing the same thing.  It
chews up extensions (probably uses a seperate partition too) and can
interfere with CallBack features.  Just something I've run into a lot
with vendors.  

	 

	Good luck fixing the mess! ;)

	 

	Court Schuett

	 

	Millward Brown

	Network Systems Engineer

	630-955-8983

	court.schuett at us.millwardbrown.com

	 

	 

	
________________________________


	From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
	Sent: Wednesday, December 14, 2005 10:30 AM
	To: Schuett, Court (MBNAP it)
	Subject: RE: [cisco-voip] MOH audio source..

	argh... the vendor already configured it on each of the lines in
each phone... sigh...

	 

	 

	Well, I'll add it to the Device pools... that way, I can slowly
remove from each line... <muttering comments about the vendor..>

	 

	Tim

	 

	
________________________________


	From: Schuett, Court (MBNAP it)
[mailto:Court.Schuett at us.millwardbrown.com] 
	Sent: Wednesday, December 14, 2005 11:21 AM
	To: Tim Reimers; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] MOH audio source..

	Ideally, they aren't configured on each device or line.  The
best place to configure them is in the Device Pool.  Configure them
there, then leave them blank in each phone.  A change to the Device Pool
will change all of them.  However, if there is already something in the
device/line, it will override the Device Pool settings.  That's why they
should all be left blank unless they are an exception.

	 

	Hope that helps.

	 

	 

	Court Schuett

	 

	Millward Brown

	Network Systems Engineer

	630-955-8983

	court.schuett at us.millwardbrown.com

	 

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
	Sent: Wednesday, December 14, 2005 10:17 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] MOH audio source..

	hi --

	 

	for those who may be interested, I've now succeeded in
configuring a MOH audio source on CCM 4.0

	 

	If you are like me, your servers do not have audio cards in
them--

	 

	Get a Griffin "iMic" USB audio adapter

	http://www.cdwg.com/shop/products/default.aspx?EDC=422596

CDW Part:  

422596

Mfg. Part:  

2020-IMIC

	 

	Mine just plugged right into the CM - no drivers needed.

	 

	I did have to reboot the CCM to get it to work as an audio
device-- 

	 

	After doing that,

	 

	In CCMAdmin  -- Service >> Media Resources >> MOH Audio Source 

	Option 51 - Fixed Audio Source

	 

	All you have to do is add the MOH Audio Source Device  -- which
for the Griffin, is "USB Audio Source"

	 

	Update, Reset MOH Servers

	 

	 

	 

	Now --- as to the question I have....

	 

	What is the best way to 'group' change all extensions to using
the 51 Fixed Audio source instead of, say '2 - All In the Mind'

	 

	I don't _really_ want to try using BAT or something--- there has
to be a way to select a whole group of phones and make this one change
on all of them at once...

	 

	anyone?

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From rratliff at cisco.com  Wed Dec 14 12:26:59 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Dec 14 12:26:58 2005
Subject: [cisco-voip] MOH audio source..
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A1200D@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A1200D@VISHNU.wesd.org>
Message-ID: <659B3792-5A83-4F6F-B638-CAFA1A3EBC9D@cisco.com>

When you go into BAT and choose Configure->Phones are you then  
selecting "Update Phones" or "Update Lines"?

-Ryan

On Dec 14, 2005, at 12:16 PM, Voll, Scott wrote:

I have not used BAT in the past but It looks like I should.  I?m  
going through the BAT tool and it shows (CM4.1) I can change the hold  
audio sources both user and network.  Question: does this do in on  
both the device and line or only the device?



Scott



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Davis, Michael
Sent: Wednesday, December 14, 2005 8:51 AM
To: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..



I agree with you regarding the device pools, but why wouldn't you use  
BAT to modify phone MRGLs?



Michael

CCIE #7303 (Voice/R&S)



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net]On Behalf Of Schuett, Court (MBNAP it)
Sent: Wednesday, December 14, 2005 10:35 AM
To: 'Tim Reimers'
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..

I'm going through the exact same thing.  Got about 300 devices to set  
right.  There are a lot of things you shouldn't set on the Device/ 
Line.  Check out the Device Pool settings and pretty much anything in  
there, don't put on the Device/Line.  It'll make it way easier to  
make changes down the road.



Might want to check on something else while you're doing this.  (I  
ran into the same problem).  On each phone, check to see if they've  
got a second line configured with a slightly altered extension.  For  
example, extension 8983 for Line 1 and 9983 for Line 2.  A lot of  
times, the vendors place a second line on every phone and then have  
the CallForwardBusy roll to that.  What you should do is just make  
sure that Maximum Number of calls is set (I use 4) and the Busy  
trigger is set (I use 2).  The 8983/9983 is an older way of doing the  
same thing.  It chews up extensions (probably uses a seperate  
partition too) and can interfere with CallBack features.  Just  
something I've run into a lot with vendors.



Good luck fixing the mess! ;)



Court Schuett



Millward Brown

Network Systems Engineer

630-955-8983

court.schuett at us.millwardbrown.com





From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us]
Sent: Wednesday, December 14, 2005 10:30 AM
To: Schuett, Court (MBNAP it)
Subject: RE: [cisco-voip] MOH audio source..

argh... the vendor already configured it on each of the lines in each  
phone... sigh...





Well, I'll add it to the Device pools... that way, I can slowly  
remove from each line... <muttering comments about the vendor..>



Tim



From: Schuett, Court (MBNAP it)  
[mailto:Court.Schuett at us.millwardbrown.com]
Sent: Wednesday, December 14, 2005 11:21 AM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..

Ideally, they aren't configured on each device or line.  The best  
place to configure them is in the Device Pool.  Configure them there,  
then leave them blank in each phone.  A change to the Device Pool  
will change all of them.  However, if there is already something in  
the device/line, it will override the Device Pool settings.  That's  
why they should all be left blank unless they are an exception.



Hope that helps.





Court Schuett



Millward Brown

Network Systems Engineer

630-955-8983

court.schuett at us.millwardbrown.com





From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, December 14, 2005 10:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MOH audio source..

hi --



for those who may be interested, I've now succeeded in configuring a  
MOH audio source on CCM 4.0



If you are like me, your servers do not have audio cards in them--



Get a Griffin "iMic" USB audio adapter

http://www.cdwg.com/shop/products/default.aspx?EDC=422596

CDW Part:

422596

Mfg. Part:

2020-IMIC



Mine just plugged right into the CM - no drivers needed.



I did have to reboot the CCM to get it to work as an audio device--



After doing that,



In CCMAdmin  -- Service >> Media Resources >> MOH Audio Source

Option 51 - Fixed Audio Source



All you have to do is add the MOH Audio Source Device  -- which for  
the Griffin, is "USB Audio Source"



Update, Reset MOH Servers







Now --- as to the question I have....



What is the best way to 'group' change all extensions to using the 51  
Fixed Audio source instead of, say '2 - All In the Mind'



I don't _really_ want to try using BAT or something--- there has to  
be a way to select a whole group of phones and make this one change  
on all of them at once...



anyone?

_______________________________________________
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 Dec 14 12:34:17 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 14 12:34:01 2005
Subject: [cisco-voip] MOH audio source..
Message-ID: <D713462ED535184D830F6F6301753E7E01A1200E@VISHNU.wesd.org>

Thanks Ryan--

That answered my question. :-)

Scott

Open mouth, insert foot, say bye bye ;-)

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Wednesday, December 14, 2005 9:27 AM
To: Voll, Scott
Cc: Davis, Michael; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MOH audio source..

When you go into BAT and choose Configure->Phones are you then  
selecting "Update Phones" or "Update Lines"?

-Ryan

On Dec 14, 2005, at 12:16 PM, Voll, Scott wrote:

I have not used BAT in the past but It looks like I should.  I'm  
going through the BAT tool and it shows (CM4.1) I can change the hold  
audio sources both user and network.  Question: does this do in on  
both the device and line or only the device?



Scott



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Davis, Michael
Sent: Wednesday, December 14, 2005 8:51 AM
To: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..



I agree with you regarding the device pools, but why wouldn't you use  
BAT to modify phone MRGLs?



Michael

CCIE #7303 (Voice/R&S)



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net]On Behalf Of Schuett, Court (MBNAP it)
Sent: Wednesday, December 14, 2005 10:35 AM
To: 'Tim Reimers'
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..

I'm going through the exact same thing.  Got about 300 devices to set  
right.  There are a lot of things you shouldn't set on the Device/ 
Line.  Check out the Device Pool settings and pretty much anything in  
there, don't put on the Device/Line.  It'll make it way easier to  
make changes down the road.



Might want to check on something else while you're doing this.  (I  
ran into the same problem).  On each phone, check to see if they've  
got a second line configured with a slightly altered extension.  For  
example, extension 8983 for Line 1 and 9983 for Line 2.  A lot of  
times, the vendors place a second line on every phone and then have  
the CallForwardBusy roll to that.  What you should do is just make  
sure that Maximum Number of calls is set (I use 4) and the Busy  
trigger is set (I use 2).  The 8983/9983 is an older way of doing the  
same thing.  It chews up extensions (probably uses a seperate  
partition too) and can interfere with CallBack features.  Just  
something I've run into a lot with vendors.



Good luck fixing the mess! ;)



Court Schuett



Millward Brown

Network Systems Engineer

630-955-8983

court.schuett at us.millwardbrown.com





From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us]
Sent: Wednesday, December 14, 2005 10:30 AM
To: Schuett, Court (MBNAP it)
Subject: RE: [cisco-voip] MOH audio source..

argh... the vendor already configured it on each of the lines in each  
phone... sigh...





Well, I'll add it to the Device pools... that way, I can slowly  
remove from each line... <muttering comments about the vendor..>



Tim



From: Schuett, Court (MBNAP it)  
[mailto:Court.Schuett at us.millwardbrown.com]
Sent: Wednesday, December 14, 2005 11:21 AM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MOH audio source..

Ideally, they aren't configured on each device or line.  The best  
place to configure them is in the Device Pool.  Configure them there,  
then leave them blank in each phone.  A change to the Device Pool  
will change all of them.  However, if there is already something in  
the device/line, it will override the Device Pool settings.  That's  
why they should all be left blank unless they are an exception.



Hope that helps.





Court Schuett



Millward Brown

Network Systems Engineer

630-955-8983

court.schuett at us.millwardbrown.com





From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, December 14, 2005 10:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MOH audio source..

hi --



for those who may be interested, I've now succeeded in configuring a  
MOH audio source on CCM 4.0



If you are like me, your servers do not have audio cards in them--



Get a Griffin "iMic" USB audio adapter

http://www.cdwg.com/shop/products/default.aspx?EDC=422596

CDW Part:

422596

Mfg. Part:

2020-IMIC



Mine just plugged right into the CM - no drivers needed.



I did have to reboot the CCM to get it to work as an audio device--



After doing that,



In CCMAdmin  -- Service >> Media Resources >> MOH Audio Source

Option 51 - Fixed Audio Source



All you have to do is add the MOH Audio Source Device  -- which for  
the Griffin, is "USB Audio Source"



Update, Reset MOH Servers







Now --- as to the question I have....



What is the best way to 'group' change all extensions to using the 51  
Fixed Audio source instead of, say '2 - All In the Mind'



I don't _really_ want to try using BAT or something--- there has to  
be a way to select a whole group of phones and make this one change  
on all of them at once...



anyone?

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



From mfa at crec.ifas.ufl.edu  Wed Dec 14 13:13:47 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Wed Dec 14 13:13:52 2005
Subject: [cisco-voip] ethernet disconnected
References: <200512141651.jBEGpjxX036188@puck.nether.net>
Message-ID: <012f01c600da$20baee70$400cfb0a@crec.ifas.ufl.edu>

I've got a TAC case open on this (602611867).  7940 works OK on a 3524, 
won't work on a 3750.  I think it's power-related, since a debug interface 
shows absolutely no IP traffic.  The switch sees a powered device, offers 
power, and then the phone shows the "Ethernet disconnected" message.  It's 
peculiar to this phone; another of the identical hardware level (and 
firmware) works OK.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

> Date: Wed, 14 Dec 2005 08:42:20 -0800
> From: "Voll, Scott" <Scott.Voll at wesd.org>
> Subject: RE: [cisco-voip] ethernet disconnected.
> To: "Lelio Fulgenzi" <lelio at uoguelph.ca>, <cisco-voip at puck.nether.net>
> Message-ID: <D713462ED535184D830F6F6301753E7E01A12006 at VISHNU.wesd.org>
> Content-Type: text/plain; charset="us-ascii"
>
> Phone.
>
>
>
> ________________________________
>
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Wednesday, December 14, 2005 8:41 AM
> To: Voll, Scott; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] ethernet disconnected.
>
>
>
> Is the Ethernet disconnected message coming from the phone or the PC?
>
>
>
> ------------------------------------------------------------------------
> --------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more jitter
> than an
> IP phone on a long haul 10base2 connection"
> LFJ
>
> ----- Original Message ----- 
>
> From: Voll, Scott <mailto:Scott.Voll at wesd.org>
>
> To: cisco-voip at puck.nether.net
>
> Sent: Wednesday, December 14, 2005 11:31 AM
>
> Subject: [cisco-voip] ethernet disconnected.
>
>
>
> Call me weird or something but how does a 7940 phone boot up off
> PoE and
> then go to Ethernet disconnected?  Still has power to the phone.
> Any
> ideas as to what could be the problem?  Is this a Switch
> problem?  I
> tested the cable and it seems to be good.  Is this a IOS bug?
>
> Hope someone can give me a hand.
>
> Scott


From JKing at thegranitegroup.com  Wed Dec 14 13:14:57 2005
From: JKing at thegranitegroup.com (King, Jesse)
Date: Wed Dec 14 13:15:07 2005
Subject: [cisco-voip] ethernet disconnected.
Message-ID: <80FDD0062521CE4084591182C4DDB6BE011084FC@papaya.thegranitegroup.com>

Scott,
 
Check -
 
http://www.cisco.com/en/US/customer/products/hw/phones/ps379/products_fi
eld_notice09186a00801cfa59.shtml
 
I think there was a recall or something on a few phones for this issue.
 
Jess

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, December 14, 2005 11:42 AM
To: Lelio Fulgenzi; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ethernet disconnected.



Phone.

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Wednesday, December 14, 2005 8:41 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ethernet disconnected.

 

Is the Ethernet disconnected message coming from the phone or the PC? 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Voll, Scott <mailto:Scott.Voll at wesd.org>  

	To: cisco-voip at puck.nether.net 

	Sent: Wednesday, December 14, 2005 11:31 AM

	Subject: [cisco-voip] ethernet disconnected.

	 

	Call me weird or something but how does a 7940 phone boot up off
PoE and
	then go to Ethernet disconnected?  Still has power to the phone.
Any
	ideas as to what could be the problem?  Is this a Switch
problem?  I
	tested the cable and it seems to be good.  Is this a IOS bug?  
	
	Hope someone can give me a hand.
	
	Scott
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip

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From kthorngr at cisco.com  Wed Dec 14 13:24:13 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed Dec 14 13:24:29 2005
Subject: [cisco-voip] CCM 4.1 and AAR Problem.. Configuration?
In-Reply-To: <20051214165444.M43263@netxar.com>
References: <20051214165444.M43263@netxar.com>
Message-ID: <3f61fb17401e8c0e052683895e72419e@cisco.com>

Hi Jose,

Here is what I see in the trace snippet:

=== AAR is invoked with 91 being prepended to 7877588585.  Strange the 
CSS is listed as a GUID and not a list of partitions.
12/14/2005 12:44:22.299 CCM|AAR invoked: cgpn = "1811"  cdpn = "2227"  
prefix digits = "91"  alternate num = "7877588585"  AAR CSS= 
"{BD492F17-4CDC-42D2-A7B1-330869C6446B}"  cdcc=269148, mAARHopCount=1

=== Digit Analysis is invoked on the reroute
12/14/2005 12:44:22.315 CCM|Digit analysis: wait_DaReq - cepn=[] 
BlockFlag=[1]
12/14/2005 12:44:22.315 CCM|Digit Analysis: getDaRes - 
voiceMailCallingSearchSpace=[]

=== DA lists the CSS as Null
12/14/2005 12:44:22.315 CCM|Digit analysis: 
match(pi="0",fqcn="7878345355", cn="1811", plv="5", pss="NULL", 
TodFilteredPss="NULL", dd="917877588585",dac="0")

=== Since it is Null, DA does not match any patterns
12/14/2005 12:44:22.315 CCM|Digit analysis: 
potentialMatches=NoPotentialMatchesExist

Sounds like you have everything set correctly.  Need to figure out why 
the phone does not contain any partitions for the AAR CSS.

Did you reset the phone (1811) after setting the ARR CSS?

Is 1811 a shared line and if so are you using the phone that the AAR 
CSS is assigned to?

Kevin

On Dec 14, 2005, at 12:02 PM, Jose Mujica wrote:

> Hi.
>
>  I have a problem in making AAR to work. I have a central site and a 
> remote
> site. I have an AAR Group with Prefix '91'. All External Phone Number 
> Mask
> have been configure properly. I also created a CSS and Partitions for 
> AAR
> functionality and I add a route pattern to it that will match the 
> prefix + the
> external number mask and is routed to the remote site branch gateway 
> in order
> to reach the central site user using the PSTN.
>
> THe problem is that the setup of the call never get to the router and 
> the
> display of the remote ip phones says Network Congestion. Rerouting and 
> after
> that the display change to 'Enter Number' and stays there... The 
> remote ip
> phone and the central ip phone are in the same AAR Group and the 
> remote ip
> phone has the CSS for AAR set, also the central site has the CSS AAR 
> set as
> well as the branch voip gateway. The gateway is an MGCP gateway using 
> FXO lines.
>
>
> Attached is the CCM trace when AAR kicks in.. Just wondering if anyone 
> has the
> same problem before or if it is a misconfiguration in my part.
>
> Remote IP Phone DN: 1811
> Central Site IP Phone 2227
> AAR Group - Prefix '91'
> External Phone Number Mask of 2227 is 7877588585
>
>
> Thanks in advanced,
> -Jose
>
> <AAR-Problems.txt>_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From CORTIZ at broward.org  Wed Dec 14 14:35:07 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed Dec 14 14:35:17 2005
Subject: [cisco-voip] H.323 Dial Peer config
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B71@MREXVS01.bc.broward.cty>

Ok, not how I imagined it (Logically) but I'll give it a shot.  One
other dumb question with the 9T destination pattern.  Is the leading 9
dropped if you do not specify forward-digits-all.  I found a CCO example
the other day that indicated this, but looking for confirmation.  

Carlos  

-----Original Message-----
From: Erick Bergquist [mailto:erickbe at yahoo.com] 
Sent: Tuesday, December 13, 2005 11:29 PM
To: Ortiz, Carlos
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] H.323 Dial Peer config

Carlos, Here is a sample...


trunk group  pstn

voice-port 0/2/1
 trunk-group pstn
!
voice-port 0/2/2
 trunk-group pstn
!
voice-port 0/2/3
 trunk-group pstn
!
voice-port 0/3/0
 trunk-group pstn
!
voice-port 0/3/1
 trunk-group pstn


dial-peer voice 1800 pots
 trunkgroup pstn
 destination-pattern 9T


--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> Erick,
> 
> All the information I am finding on Trunk Groups refers to PRI's.  I 
> can create a trunk group at the global level, but I don't see an 
> option that would allow me to add FXO voice ports to it. See
> below:
> 
> PESPC(config)#trunk group  FXO-G1             
> PESPC(config-trunk-group)#?
> Trunk Group configuration commands:
>   capacity             capacity update timer config
>   carrier-id           Name of the carrier
> associated with this trunk
> group
>   default              Set a command to its defaults
>   description          Trunk group description
>   exit                 Exit from trunk group
> configuration mode
>   help                 Description of the
> interactive help system
>   hunt-scheme          The hunt scheme in trunk
> group
>   isdn                 Specify isdn supplementary
> service
>   max-calls            Maximum number of calls
> allowed
>   max-retry            Maximum number of outgoing
> attempts
>   no                   Negate a command or set its
> defaults
>   tgrep                TGREP config
>   translation-profile  Translation profile
>   voice-class          Set voice class control
> parameters
> 
> 
> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com]
> Sent: Monday, December 12, 2005 10:38 PM
> To: Ortiz, Carlos; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] H.323 Dial Peer config
> 
> On the dial-peers, you are forwarding all digits so that will also 
> send the 9 that starts all your destination patterns.
> 
> Usually with FXO, there is no inbound caller id to tell it where to 
> route call so on the voice-port you need to use connection plar to 
> route call somewhere (This is same as Attendant DN setting on the 
> endpoint in CCM when using MGCP).
> 
> They have added trunk groups to IOS, so you can assign voice ports to 
> a trunkgroup, then have one dial-peer pointing to the trunk group. You

> can then assign preferences to voice-port that is a member of the 
> trunk group, etc. This requires a trunk group to be configured 
> globally. Under the trunkgroup config you can configure hunt method, 
> etc.
> 
> Erick
> 
> --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
> 
> > I have never configured my FXO PSTN router connections with H.323, 
> > only MGCP.  I did configure H.323 for PRI interfaces.
> > Going by that and what
> > I have refreshed with CCO docs, I have come up
> with
> > a starting config to
> > do FXO with H.323. Dial peer 1 would be for the incoming call leg on

> > all calls.  This assumes a 4 port FXO card.  Does this appear 
> > correct?  Have there been any improvements on the "hunting" for 
> > outbound calls beyond using the preference command? Any suggestions 
> > or help would be appreciated.
> > 
> >  
> > 
> > dial-peer voice 1 pots
> > 
> >  description ** Inbound Call leg all calls **
> > 
> >  incoming called-number .
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/0
> > 
> > !
> > 
> > dial-peer voice 954 pots
> > 
> >  description ** Local 954,754 Calls **
> > 
> >  destination-pattern 9[7,9]54.......
> > 
> >  progress_ind setup enable 3
> > 
> >  progress_ind alert enable 8
> > 
> >  progress_ind progress enable 8
> > 
> >  progress_ind connect enable 8
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/0
> > 
> >  forward-digits all
> > 
> >  
> > 
> > dial-peer voice 9542 pots
> > 
> >  description ** Local 954,754 Calls **
> > 
> >  destination-pattern 9[7,9]54.......
> > 
> >  preference 1
> > 
> >  progress_ind setup enable 3
> > 
> >  progress_ind alert enable 8
> > 
> >  progress_ind progress enable 8
> > 
> >  progress_ind connect enable 8
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/1
> > 
> >  forward-digits all
> > 
> >  
> > 
> > dial-peer voice 9543 pots
> > 
> >  description ** Local 954,754 Calls **
> > 
> >  destination-pattern 9[7,9]54.......
> > 
> >  preference 2
> > 
> >  progress_ind setup enable 3
> > 
> >  progress_ind alert enable 8
> > 
> >  progress_ind progress enable 8
> > 
> >  progress_ind connect enable 8
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/2
> > 
> >  forward-digits all
> > 
> >  
> > 
> > dial-peer voice 9544 pots
> > 
> >  description ** Local 954,754 Calls **
> > 
> >  destination-pattern 9[7,9]54.......
> > 
> >  preference 3
> > 
> >  progress_ind setup enable 3
> > 
> >  progress_ind alert enable 8
> > 
> >  progress_ind progress enable 8
> > 
> >  progress_ind connect enable 8
> > 
> >  direct-inward-dial
> > 
> >  port 0/1/3
> > 
> >  forward-digits all
> > 
> >  
> > 
> > dial-peer voice 9999 voip
> > 
> >  description ** Inbound 357 nums to mr-cm-sub1 **
> > 
> >  destination-pattern [5-8]...
> > 
> >  progress_ind setup enable 3
> > 
> >  session target ipv4:10.100.200.21
> > 
> >  incoming called-number .
> > 
> >  dtmf-relay h245-alphanumeric
> > 
> >  codec g711ulaw
> > 
> > !
> > 
> > dial-peer voice 99992 voip
> > 
> >  description ** Inbound 357 nums to mr-cm-pub **
> > 
> >  preference 2
> > 
> >  destination-pattern [5-8]...
> > 
> >  progress_ind setup enable 3
> > 
> >  session target ipv4:10.100.200.20
> > 
> >  incoming called-number .
> > 
> >  dtmf-relay h245-alphanumeric
> > 
> >  codec g711ulaw
> > 
> >  
> > 
> >  
> > 
> > Carlos Ortiz
> > 
> > Senior Information Technology Specialist
> > 
> > Broward County Office of Information Technology
> > 
> > Data Communications
> > 
> > 954-357-7981
> > 
> > cortiz at broward.org
> > 
> >  
> > 
> >  
> > 
> > > _______________________________________________
> > 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
> 


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


From erickbe at yahoo.com  Wed Dec 14 14:38:59 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Dec 14 14:39:06 2005
Subject: [cisco-voip] H.323 Dial Peer config
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B71@MREXVS01.bc.broward.cty>
Message-ID: <20051214193859.34919.qmail@web34314.mail.mud.yahoo.com>

Yea, it is dropped without a forward-digits statement.
You could also do prefix 9. 

--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> Ok, not how I imagined it (Logically) but I'll give
> it a shot.  One
> other dumb question with the 9T destination pattern.
>  Is the leading 9
> dropped if you do not specify forward-digits-all.  I
> found a CCO example
> the other day that indicated this, but looking for
> confirmation.  
> 
> Carlos  
> 
> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> Sent: Tuesday, December 13, 2005 11:29 PM
> To: Ortiz, Carlos
> Cc: cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] H.323 Dial Peer config
> 
> Carlos, Here is a sample...
> 
> 
> trunk group  pstn
> 
> voice-port 0/2/1
>  trunk-group pstn
> !
> voice-port 0/2/2
>  trunk-group pstn
> !
> voice-port 0/2/3
>  trunk-group pstn
> !
> voice-port 0/3/0
>  trunk-group pstn
> !
> voice-port 0/3/1
>  trunk-group pstn
> 
> 
> dial-peer voice 1800 pots
>  trunkgroup pstn
>  destination-pattern 9T
> 
> 
> --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
> 
> > Erick,
> > 
> > All the information I am finding on Trunk Groups
> refers to PRI's.  I 
> > can create a trunk group at the global level, but
> I don't see an 
> > option that would allow me to add FXO voice ports
> to it. See
> > below:
> > 
> > PESPC(config)#trunk group  FXO-G1             
> > PESPC(config-trunk-group)#?
> > Trunk Group configuration commands:
> >   capacity             capacity update timer
> config
> >   carrier-id           Name of the carrier
> > associated with this trunk
> > group
> >   default              Set a command to its
> defaults
> >   description          Trunk group description
> >   exit                 Exit from trunk group
> > configuration mode
> >   help                 Description of the
> > interactive help system
> >   hunt-scheme          The hunt scheme in trunk
> > group
> >   isdn                 Specify isdn supplementary
> > service
> >   max-calls            Maximum number of calls
> > allowed
> >   max-retry            Maximum number of outgoing
> > attempts
> >   no                   Negate a command or set its
> > defaults
> >   tgrep                TGREP config
> >   translation-profile  Translation profile
> >   voice-class          Set voice class control
> > parameters
> > 
> > 
> > -----Original Message-----
> > From: Erick Bergquist [mailto:erickbe at yahoo.com]
> > Sent: Monday, December 12, 2005 10:38 PM
> > To: Ortiz, Carlos; cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] H.323 Dial Peer config
> > 
> > On the dial-peers, you are forwarding all digits
> so that will also 
> > send the 9 that starts all your destination
> patterns.
> > 
> > Usually with FXO, there is no inbound caller id to
> tell it where to 
> > route call so on the voice-port you need to use
> connection plar to 
> > route call somewhere (This is same as Attendant DN
> setting on the 
> > endpoint in CCM when using MGCP).
> > 
> > They have added trunk groups to IOS, so you can
> assign voice ports to 
> > a trunkgroup, then have one dial-peer pointing to
> the trunk group. You
> 
> > can then assign preferences to voice-port that is
> a member of the 
> > trunk group, etc. This requires a trunk group to
> be configured 
> > globally. Under the trunkgroup config you can
> configure hunt method, 
> > etc.
> > 
> > Erick
> > 
> > --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
> > 
> > > I have never configured my FXO PSTN router
> connections with H.323, 
> > > only MGCP.  I did configure H.323 for PRI
> interfaces.
> > > Going by that and what
> > > I have refreshed with CCO docs, I have come up
> > with
> > > a starting config to
> > > do FXO with H.323. Dial peer 1 would be for the
> incoming call leg on
> 
> > > all calls.  This assumes a 4 port FXO card. 
> Does this appear 
> > > correct?  Have there been any improvements on
> the "hunting" for 
> > > outbound calls beyond using the preference
> command? Any suggestions 
> > > or help would be appreciated.
> > > 
> > >  
> > > 
> > > dial-peer voice 1 pots
> > > 
> > >  description ** Inbound Call leg all calls **
> > > 
> > >  incoming called-number .
> > > 
> > >  direct-inward-dial
> > > 
> > >  port 0/1/0
> > > 
> > > !
> > > 
> > > dial-peer voice 954 pots
> > > 
> > >  description ** Local 954,754 Calls **
> > > 
> > >  destination-pattern 9[7,9]54.......
> > > 
> > >  progress_ind setup enable 3
> > > 
> > >  progress_ind alert enable 8
> > > 
> > >  progress_ind progress enable 8
> > > 
> > >  progress_ind connect enable 8
> > > 
> > >  direct-inward-dial
> > > 
> > >  port 0/1/0
> > > 
> > >  forward-digits all
> > > 
> > >  
> > > 
> > > dial-peer voice 9542 pots
> > > 
> > >  description ** Local 954,754 Calls **
> > > 
> > >  destination-pattern 9[7,9]54.......
> > > 
> > >  preference 1
> > > 
> > >  progress_ind setup enable 3
> > > 
> > >  progress_ind alert enable 8
> > > 
> > >  progress_ind progress enable 8
> > > 
> > >  progress_ind connect enable 8
> > > 
> > >  direct-inward-dial
> > > 
> > >  port 0/1/1
> > > 
> > >  forward-digits all
> > > 
> > >  
> > > 
> > > dial-peer voice 9543 pots
> > > 
> > >  description ** Local 954,754 Calls **
> > > 
> > >  destination-pattern 9[7,9]54.......
> > > 
> > >  preference 2
> > > 
> > >  progress_ind setup enable 3
> > > 
> > >  progress_ind alert enable 8
> > > 
> > >  progress_ind progress enable 8
> > > 
> > >  progress_ind connect enable 8
> > > 
> > >  direct-inward-dial
> > > 
> > >  port 0/1/2
> > > 
> > >  forward-digits all
> > > 
> > >  
> > > 
> > > dial-peer voice 9544 pots
> > > 
> > >  description ** Local 954,754 Calls **
> > > 
> > >  destination-pattern 9[7,9]54.......
> > > 
> > >  preference 3
> > > 
> > >  progress_ind setup enable 3
> > > 
> > >  progress_ind alert enable 8
> > > 
> > >  progress_ind progress enable 8
> > > 
> > >  progress_ind connect enable 8
> > > 
> > >  direct-inward-dial
> > > 
> > >  port 0/1/3
> > > 
> > >  forward-digits all
> > > 
> > >  
> > > 
> > > dial-peer voice 9999 voip
> > > 
> > >  description ** Inbound 357 nums to mr-cm-sub1
> **
> > > 
> > >  destination-pattern [5-8]...
> > > 
> > >  progress_ind setup enable 3
> > > 
> > >  session target ipv4:10.100.200.21
> > > 
> > >  incoming called-number .
> > > 
> > >  dtmf-relay h245-alphanumeric
> > > 
> > >  codec g711ulaw
> > > 
> > > !
> > > 
> > > dial-peer voice 99992 voip
> > > 
> > >  description ** Inbound 357 nums to mr-cm-pub **
> > > 
> > >  preference 2
> > > 
> > >  destination-pattern [5-8]...
> > > 
> > >  progress_ind setup enable 3
> > > 
> > >  session target ipv4:10.100.200.20
> > > 
> > >  incoming called-number .
> > > 
> > >  dtmf-relay h245-alphanumeric
> > > 
> > >  codec g711ulaw
> > > 
> > >  
> > > 
> > >  
> > > 
> > > Carlos Ortiz
> > > 
> > > Senior Information Technology Specialist
> > > 
> > > Broward County Office of Information Technology
> > > 
> > > Data Communications
> > > 
> > > 954-357-7981
> > > 
> > > cortiz at broward.org
> > > 
> > >  
> > > 
> > >  
> > > 
> > > >
> _______________________________________________
> > > 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
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com 
> 


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

From mfa at crec.ifas.ufl.edu  Wed Dec 14 15:26:46 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Wed Dec 14 15:26:49 2005
Subject: [cisco-voip] ethernet disconnected.
References: <200512141935.jBEJZJDK059355@puck.nether.net>
Message-ID: <015301c600ec$b494f2a0$400cfb0a@crec.ifas.ufl.edu>

You got my hopes up briefly, but our failing phone's s/n is outside the 
range of those in the FN.  TAC asked for a bunch of debugging information 
from the switch, but I expect an RMA phone replacement is in the wings.

mike

> Date: Wed, 14 Dec 2005 13:14:57 -0500
> From: "King, Jesse" <JKing at thegranitegroup.com>
> Subject: RE: [cisco-voip] ethernet disconnected.
> To: "Voll, Scott" <Scott.Voll at wesd.org>, "Lelio Fulgenzi"
> <lelio at uoguelph.ca>, <cisco-voip at puck.nether.net>
> Message-ID:
> <80FDD0062521CE4084591182C4DDB6BE011084FC at papaya.thegranitegroup.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Scott,
>
> Check -
>
> http://www.cisco.com/en/US/customer/products/hw/phones/ps379/products_fi
> eld_notice09186a00801cfa59.shtml
>
> I think there was a recall or something on a few phones for this issue.
>
> Jess


From Scott.Voll at wesd.org  Wed Dec 14 15:29:17 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 14 15:29:01 2005
Subject: [cisco-voip] ethernet disconnected.
Message-ID: <D713462ED535184D830F6F6301753E7E01A12030@VISHNU.wesd.org>

Interesting thing is that it followed the port not the phone for us.

Scott 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Wednesday, December 14, 2005 12:27 PM
To: cisco-voip at puck.nether.net; JKing at thegranitegroup.com
Subject: RE: [cisco-voip] ethernet disconnected.

You got my hopes up briefly, but our failing phone's s/n is outside the 
range of those in the FN.  TAC asked for a bunch of debugging
information 
from the switch, but I expect an RMA phone replacement is in the wings.

mike

> Date: Wed, 14 Dec 2005 13:14:57 -0500
> From: "King, Jesse" <JKing at thegranitegroup.com>
> Subject: RE: [cisco-voip] ethernet disconnected.
> To: "Voll, Scott" <Scott.Voll at wesd.org>, "Lelio Fulgenzi"
> <lelio at uoguelph.ca>, <cisco-voip at puck.nether.net>
> Message-ID:
> <80FDD0062521CE4084591182C4DDB6BE011084FC at papaya.thegranitegroup.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Scott,
>
> Check -
>
>
http://www.cisco.com/en/US/customer/products/hw/phones/ps379/products_fi
> eld_notice09186a00801cfa59.shtml
>
> I think there was a recall or something on a few phones for this
issue.
>
> Jess

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


From candace_holman at harvard.edu  Wed Dec 14 15:28:53 2005
From: candace_holman at harvard.edu (Candace Holman)
Date: Wed Dec 14 15:29:02 2005
Subject: [cisco-voip] Re: ethernet disconnected.
Message-ID: <43A08085.20303@harvard.edu>

Does it make sense that the port has MAC address authentication on it?  
Does the MAC address authentication shut down PoE, or just the ethernet 
traffic?

Candace


> Subject:
> [cisco-voip] ethernet disconnected.
> From:
> "Voll, Scott" <Scott.Voll at wesd.org>
> Date:
> Wed, 14 Dec 2005 08:31:54 -0800
>
> To:
> <cisco-voip at puck.nether.net>
>
>
>Call me weird or something but how does a 7940 phone boot up off PoE and
>then go to Ethernet disconnected?  Still has power to the phone.  Any
>ideas as to what could be the problem?  Is this a Switch problem?  I
>tested the cable and it seems to be good.  Is this a IOS bug?  
>
>Hope someone can give me a hand.
>
>Scott
>
>
>
>  
>


From mike.c at altatechnologies.com  Wed Dec 14 15:33:31 2005
From: mike.c at altatechnologies.com (Mike Carter)
Date: Wed Dec 14 15:36:26 2005
Subject: [cisco-voip] need help moving off sip to mgcp
Message-ID: <BFEHKOIOENIIPHGJOCHDAEANFBAA.mike.c@altatechnologies.com>

Hi all

Customer has 7940G which were upgraded to SIP -- they need to use MGCP.  How
do they switch Protocols???

they've already unlocked the cofig and tried to override with no luck...any
idea would be greatly appreciated

Anyone that has the time for a tech call to assist please contact me asap

TIA

Michael Carter
Account Manager
Alta Technologies Inc
763-475-5327



From lelio at uoguelph.ca  Wed Dec 14 16:00:17 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Dec 14 16:00:22 2005
Subject: [cisco-voip] Unity & BellHop
Message-ID: <01ac01c600f1$62e23cb0$30196883@cfs.uoguelph.ca>

Has anyone else used Unity with BellHop? I raised the issue of annual resets and was told to look at this.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From SPackett at fenwick.com  Wed Dec 14 16:58:13 2005
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Dec 14 16:58:31 2005
Subject: [cisco-voip] ISDN Simulator
Message-ID: <965C0255FD5828438E34F9A68ECA76E740A61B@svcexch02.firm.fenwick.llp>

Mark:
I was looking into the Adtran Atlas 550 or 830, but you mentioned you
use Cisco Gateways in the middle.  I'm about to replace 7 of our VG200
gateways for 2811 routers.  My goal is to replicate as much as possible
our production setup.  Could I use those old VG200s and have it function
like an ISDN simulator?  Could you explain more of your physical setup?
 
Thanks.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Snow
Sent: Wednesday, December 14, 2005 7:43 AM
To: 'Lelio Fulgenzi'; 'Nick Kassel'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ISDN Simulator



Yes actually using cisco gateways in the middle with a CCM is the best
way to prepare for the exam.

 

 

-Mark Snow 

CCIE Voice # 14073

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, December 14, 2005 10:34 AM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ISDN Simulator

 

I've looked at Adtran a few times. But after one of our SE's pointed me
to T1 crossover cables, we went to using two 6608 ports for our testing
- one registered to our production cluster and one registered to our
test cluster. Saved us a few dollars.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 

	From: Nick Kassel <mailto:Nick.Kassel at charles-stanley.co.uk>  

	To: cisco-voip at puck.nether.net 

	Sent: Wednesday, December 14, 2005 10:28 AM

	Subject: [cisco-voip] ISDN Simulator

	 

	I'm looking to incorporate an ISDN simulator into our test lab,
does any body use these in their labs and if so can you recommend one
worth looking at for PRI use.

	 

	 


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From rratliff at cisco.com  Wed Dec 14 17:06:43 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Dec 14 17:07:12 2005
Subject: [cisco-voip] need help moving off sip to mgcp
In-Reply-To: <BFEHKOIOENIIPHGJOCHDAEANFBAA.mike.c@altatechnologies.com>
References: <BFEHKOIOENIIPHGJOCHDAEANFBAA.mike.c@altatechnologies.com>
Message-ID: <15B79E92-BF39-45B2-93E0-67523B86B105@cisco.com>

Check out http://www.cisco.com/en/US/tech/tk652/tk701/ 
technologies_tech_note09186a0080094584.shtml.  It describes moving  
from sip to sccp, but for MGCP the process should be similar.

In general you need a tftp server with OS79XX.TXT configured with the  
image you wish to load.  At boot the phone will compare the first  
several characters of that image with what it is currently running  
and if the protocol is different (ie sccp, sip, mgcp) then it will  
load the new firmware.

-Ryan

On Dec 14, 2005, at 3:33 PM, Mike Carter wrote:

Hi all

Customer has 7940G which were upgraded to SIP -- they need to use  
MGCP.  How
do they switch Protocols???

they've already unlocked the cofig and tried to override with no  
luck...any
idea would be greatly appreciated

Anyone that has the time for a tech call to assist please contact me  
asap

TIA

Michael Carter
Account Manager
Alta Technologies Inc
763-475-5327


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From highspeedsnow at gmail.com  Wed Dec 14 17:08:08 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Wed Dec 14 17:09:29 2005
Subject: [cisco-voip] ISDN Simulator
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E740A61B@svcexch02.firm.fenwick.llp>
Message-ID: <43a09815.63d4e973.4546.fffff08e@mx.gmail.com>

Sure, I actuall have to run and will explain (read draw out) later - but if
those VG200's had a PRI card in them - then you could have them speak either
H323 to each other - or have them controlled by MGCP back to a separate CCM
that acted as the PSTN.

 

Does that make sense?

 

HTH,

 

Mark Snow

 

  _____  

From: Stu Packett [mailto:SPackett at fenwick.com] 
Sent: Wednesday, December 14, 2005 4:58 PM
To: Mark Snow; Lelio Fulgenzi; Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ISDN Simulator

 

Mark:

I was looking into the Adtran Atlas 550 or 830, but you mentioned you use
Cisco Gateways in the middle.  I'm about to replace 7 of our VG200 gateways
for 2811 routers.  My goal is to replicate as much as possible our
production setup.  Could I use those old VG200s and have it function like an
ISDN simulator?  Could you explain more of your physical setup?

 

Thanks.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Snow
Sent: Wednesday, December 14, 2005 7:43 AM
To: 'Lelio Fulgenzi'; 'Nick Kassel'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ISDN Simulator

Yes actually using cisco gateways in the middle with a CCM is the best way
to prepare for the exam.

 

 

-Mark Snow 

CCIE Voice # 14073

 

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Wednesday, December 14, 2005 10:34 AM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ISDN Simulator

 

I've looked at Adtran a few times. But after one of our SE's pointed me to
T1 crossover cables, we went to using two 6608 ports for our testing - one
registered to our production cluster and one registered to our test cluster.
Saved us a few dollars.

 

----------------------------------------------------------------------------
----
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than
an
IP phone on a long haul 10base2 connection"
LFJ

----- Original Message ----- 

From: Nick Kassel <mailto:Nick.Kassel at charles-stanley.co.uk>  

To: cisco-voip at puck.nether.net 

Sent: Wednesday, December 14, 2005 10:28 AM

Subject: [cisco-voip] ISDN Simulator

 

I'm looking to incorporate an ISDN simulator into our test lab, does any
body use these in their labs and if so can you recommend one worth looking
at for PRI use.

 

 



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Registered Office: 25 Luke Street London EC2A 4AR

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Registered in England No. 1903304

Charles Stanley Sutherlands and Tozer Wingate are divisions of Charles
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From nikola at worldnet.att.net  Wed Dec 14 19:15:24 2005
From: nikola at worldnet.att.net (Nikola Stojsin)
Date: Wed Dec 14 19:15:27 2005
Subject: [cisco-voip] ethernet disconnected.
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A12030@VISHNU.wesd.org>
Message-ID: <4k6l9v$47jlnr@smtp01.mrf.mail.rcn.net>

Two VERY silly suggestions - but both of them did happen to me:

1) I would try eliminating the cable as a potential cause (i.e. connecting
the offending phone to its switch port using a short known-to-be-good patch
cable). I had once one of those
'good-enough-to-turn-the-tester-light-on-but-not-to-connect' cables. (Gotta
love them.) Also, if you did not already, I would try swapping ports and/or
phones. (Meaning plugging a different phone into the offending switch port
and offending phone into another switch port.) 

2) My second occurrence of that lovely message was caused by hardware
failure of phone's built-in switch (it was a 7960, I think).

HTH, 
Nikola


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, December 14, 2005 3:29 PM
To: Mike Armstrong; cisco-voip at puck.nether.net; JKing at thegranitegroup.com
Subject: RE: [cisco-voip] ethernet disconnected.

Interesting thing is that it followed the port not the phone for us.

Scott 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Wednesday, December 14, 2005 12:27 PM
To: cisco-voip at puck.nether.net; JKing at thegranitegroup.com
Subject: RE: [cisco-voip] ethernet disconnected.

You got my hopes up briefly, but our failing phone's s/n is outside the 
range of those in the FN.  TAC asked for a bunch of debugging
information 
from the switch, but I expect an RMA phone replacement is in the wings.

mike

> Date: Wed, 14 Dec 2005 13:14:57 -0500
> From: "King, Jesse" <JKing at thegranitegroup.com>
> Subject: RE: [cisco-voip] ethernet disconnected.
> To: "Voll, Scott" <Scott.Voll at wesd.org>, "Lelio Fulgenzi"
> <lelio at uoguelph.ca>, <cisco-voip at puck.nether.net>
> Message-ID:
> <80FDD0062521CE4084591182C4DDB6BE011084FC at papaya.thegranitegroup.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Scott,
>
> Check -
>
>
http://www.cisco.com/en/US/customer/products/hw/phones/ps379/products_fi
> eld_notice09186a00801cfa59.shtml
>
> I think there was a recall or something on a few phones for this
issue.
>
> Jess

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From it at cimgroup.com  Wed Dec 14 19:35:51 2005
From: it at cimgroup.com (IT)
Date: Wed Dec 14 19:32:03 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0260D981@exchange.cimgroup.com>

Hello,

I have inherited a Cisco 7935 and I am running CM 4.0
I am having trouble getting the 7935 to pick up new firmware. I have a
tftp server running on my CM Publisher, and the device defaults are set
correctly for the firmware version. However, the screen only displays:
Station IP: 10.96.3.58
Contacting: 10.96.3.11 (my tftp server)

When I go to http://10.96.3.58 I get:
Wind Web Server Error Report:
Server Error: 501 Not Implemented
No RPM for this combination of URL and method
/dynaform/index_Dyna.htm

I am able to telnet to 10.96.3.58, but I get a username & password
prompt, but I do not know the password, and cisco/cisco doesn't work.

Does anyone know how else I can force flash this device?

On another note, I had to explicitly set the VLAN on my switch, as the
device was picking up the data vlan when connecting, shouldn't this
device auto register on the voice vlan?

Thanks,
Avidan

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This email may contain trade secrets and/or other information that is 
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From nkhwaja at gmail.com  Wed Dec 14 23:18:20 2005
From: nkhwaja at gmail.com (Nasser Khwaja)
Date: Wed Dec 14 23:18:23 2005
Subject: [cisco-voip] Cat 6509 memory commands request
Message-ID: <23435d3d0512142018r3606b94gb2476bc06875df7d@mail.gmail.com>

Hi,
On a Catalyst 6509, please let me know the commands to find the RAM on each
of the following?

1)Catalyst 6000 supervisor 2 (WS-X6K-SUP2-2GE)

2)Policy Feature Card 2 (WS-F6K-PFC2)

3)Cat6k MSFC 2 daughterboard  (WS-F6K-MSFC2)

Thanks
-Tennisman.
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From Ruben.Montes at eu.didata.com  Thu Dec 15 05:16:08 2005
From: Ruben.Montes at eu.didata.com (Ruben Montes)
Date: Thu Dec 15 05:16:57 2005
Subject: [cisco-voip] CFwAll error through Q.SIG
Message-ID: <D305A0848689344BB98F14A3F13161B705B144@eubebrusvex.eu.didata.local>

Hello,

I have a QSIG connection between a PBX MD110 and a C2851 MGCP gateway (CCM4.1(3) cluster).

I have ip phone A (DN 48810) forwarded to ip phone B through CfwAll softkey (DN 8001). If I try to call the number of phone A from an analog phone that is in the PBX side, call should be forwarded to phone B.

The problem is that the call is forwarded but it only rings in phone B during 0,5 seconds and then the call is hung. 

If I do a 'debug isdn q931' in the router I have this message:

   Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18396
                Preferred, Channel 22
        Progress Ind i = 0xA190 - Reserved value
        Calling Party Number i = 0x0181, '659457184'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0xC9, '48810'
                Plan:Private, Type:Subscriber(local)
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B4
        Shift to Codeset 5
        Codeset 5 IE 0x31  i = 0x81
        Shift to Codeset 5
        Codeset 5 IE 0x32  i = 0x80
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8  callref = 0xABBE

        Channel ID i = 0xA98396
                Exclusive, Channel 22
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> FACILITY pd = 8  callref = 0xABBE
        Facility i = 0x9FAA068001008201008B0100A114020101020114300C0A01010A0102800438303031
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> ALERTING pd = 8  callref = 0xABBE
        Progress Ind i = 0x8088 - In-band info or appropriate now available
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: RX <- STATUS pd = 8  callref = 0x2BBE
        Cause i = 0x81E41C - Invalid information element contents
        Call State i = 0x03
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> DISCONNECT pd = 8  callref = 0xABBE

        Cause i = 0x80E2 - Message not compatible with call state or not implemented
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: RX <- RELEASE pd = 8  callref = 0x2BBE
        Cause i = 0x81E2 - Message not compatible with call state or not implemented
        User-User i = 0x00FEB4110180
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0xABBE

Does anybody know which is the problem?

Thanks in advance,

Ruben

 

 

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From ciscozest at gmail.com  Thu Dec 15 05:27:36 2005
From: ciscozest at gmail.com (Robin H)
Date: Thu Dec 15 05:27:39 2005
Subject: [cisco-voip] IVR transfer to Voicemail
Message-ID: <f99cc3f60512150227q1c2a04d5q472acc2de83386f1@mail.gmail.com>

 Hi,
I need to setup an IVR script that allows transfer to subscriber's voicemail
directly. I have created necessary transfer from CCM to unity and it only
works for direct dialing from internal phones. From IVR script, the transfer
always failed and the caller got multiple beeps before hearing an invalid
extension prompt. Anyone experience this before? Plz enlighten me. Thanks

Regards,
Robin
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From Nick.Kassel at Charles-Stanley.co.uk  Thu Dec 15 05:29:30 2005
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu Dec 15 05:29:47 2005
Subject: [cisco-voip] CFwAll error through Q.SIG
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC0C016F@CSEXCH01.charles-stanley.co.uk>

check out the document

 

http://www.cisco.com/warp/public/129/isdn_disc_code.html 

 

Cause i = 0x80E2 - Message not compatible with call state or not
implemented

 

 

E2

Message not compatible with call state or not implemented 

The remote equipment receives an invalid message with no standard cause.

This cause indicates that the equipment receives a message that is not
permissible in the call state according to the procedures. This cause
can also indicate that the equipment receives a STATUS message to
indicate an incompatible call state.

The issue occurs due to a D-channel error. If the error recurs, report
the error to your ISDN service provider.

 

 

 

 

 

 

 

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruben Montes
Sent: Thursday, December 15, 2005 10:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CFwAll error through Q.SIG

 

Hello,

I have a QSIG connection between a PBX MD110 and a C2851 MGCP gateway
(CCM4.1(3) cluster).

I have ip phone A (DN 48810) forwarded to ip phone B through CfwAll
softkey (DN 8001). If I try to call the number of phone A from an analog
phone that is in the PBX side, call should be forwarded to phone B.

The problem is that the call is forwarded but it only rings in phone B
during 0,5 seconds and then the call is hung. 

If I do a 'debug isdn q931' in the router I have this message:

   Sending Complete
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18396
                Preferred, Channel 22
        Progress Ind i = 0xA190 - Reserved value
        Calling Party Number i = 0x0181, '659457184'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0xC9, '48810'
                Plan:Private, Type:Subscriber(local)
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B4
        Shift to Codeset 5
        Codeset 5 IE 0x31  i = 0x81
        Shift to Codeset 5
        Codeset 5 IE 0x32  i = 0x80
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8  callref =
0xABBE

        Channel ID i = 0xA98396
                Exclusive, Channel 22
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> FACILITY pd = 8  callref =
0xABBE
        Facility i =
0x9FAA068001008201008B0100A114020101020114300C0A01010A0102800438303031
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> ALERTING pd = 8  callref =
0xABBE
        Progress Ind i = 0x8088 - In-band info or appropriate now
available
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: RX <- STATUS pd = 8  callref =
0x2BBE
        Cause i = 0x81E41C - Invalid information element contents
        Call State i = 0x03
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> DISCONNECT pd = 8  callref
= 0xABBE

        Cause i = 0x80E2 - Message not compatible with call state or not
implemented
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: RX <- RELEASE pd = 8  callref =
0x2BBE
        Cause i = 0x81E2 - Message not compatible with call state or not
implemented
        User-User i = 0x00FEB4110180
Dec 15 11:09:15: ISDN Se0/0/1:15 Q931: TX -> RELEASE_COMP pd = 8
callref = 0xABBE

Does anybody know which is the problem?

Thanks in advance,

Ruben

 

 

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Registered in England No. 1903304

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From Nick.Kassel at Charles-Stanley.co.uk  Thu Dec 15 06:21:44 2005
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu Dec 15 06:21:56 2005
Subject: [cisco-voip] IOS for1760
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC0C0172@CSEXCH01.charles-stanley.co.uk>

I'm looking for a good stable version of IP voice code for a 1760, can
any one recommend one for this model.

 

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The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
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may be a delay in processing the instructions received from e-mails after
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and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

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

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From Nick.Kassel at Charles-Stanley.co.uk  Thu Dec 15 06:39:53 2005
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu Dec 15 06:40:08 2005
Subject: [cisco-voip] Call Manager Serviceability
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC0C0173@CSEXCH01.charles-stanley.co.uk>

On the Call Manager serviceability, I am unable to view any H323
gateways we have a number of them at remote sites and only 1 MGCP
gateway, the MGCP gateway displays on the device summary fine. Is this
because the status on the Call manager shows as unknown and is there any
way of getting the H323 gateways to show as we want to monitor amount of
calls each remote site is placing.

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

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

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

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From ciscozest at gmail.com  Thu Dec 15 08:31:28 2005
From: ciscozest at gmail.com (Ciscozest)
Date: Thu Dec 15 08:31:26 2005
Subject: [cisco-voip] Function of VM Pilot CSS
References: <f99cc3f60512122019j6827d8d9hee807a3b78113d1f@mail.gmail.com>
	<004d01c5ff9f$028daf50$c901a8c0@EATIUSBIRDIUS>
	<00d901c5ffe3$86930bd0$6301a8c0@windfall99>
	<003b01c5ffec$53f46ee0$30196883@cfs.uoguelph.ca>
Message-ID: <006401c6017b$dc9d9d80$6301a8c0@windfall99>

I got it now. Thanks Lelio.
  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: Ciscozest ; cisco-voip at puck.nether.net 
  Sent: Tuesday, December 13, 2005 9:51 PM
  Subject: Re: [cisco-voip] Function of VM Pilot CSS


  Wes also mentioned this: 
    When you check the 'Voicemail' box for CFB/CFNA, we attempt to use the CSS of the Voicemail Pilot by default.  If that is blank then we revert to using the CSS of the originating device.

  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
  IP phone on a long haul 10base2 connection"                                           LFJ
    ----- Original Message ----- 
    From: Ciscozest 
    To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
    Sent: Tuesday, December 13, 2005 7:48 AM
    Subject: Re: [cisco-voip] Function of VM Pilot CSS


    Hi,

    Here is my setting detail:
    Phone A: Partition = "internal", CSS = "Site1-all", CFB/CFNA to VM.
    VM ports: Partition = "internal", CSS = "Site1-all".
    VM Pilot: CSS = "none"

    Analysis:
    - I can access VM via message button
    - call from phone B to A can be diverted to VM when is not answered. 
    - Unity can dialout to internal and external number.

    in summary I don;t see the funtionality of CSS in VM Pilot Conifguration page. I thought setting "none" CSS would restrict it from dialing a configured partition(s). I use CM 3.3(5)sr1. Anyone can enlighten me? thanks

    regards,
    Robin
      ----- Original Message ----- 
      From: Lelio Fulgenzi 
      To: Robin H ; cisco-voip at puck.nether.net 
      Sent: Tuesday, December 13, 2005 12:38 PM
      Subject: Re: [cisco-voip] Function of VM Pilot CSS


      I won't take credit for this one because Wes straightened it out when I was having a problem. I hope he'll correct me if I've got it wrong.

      The VM Pilot CCS is used when you check the VM box on the line configuration page. If you put anything in the destination field, then it uses the CSS you've selected.

      You'll have to check to see what gets checked if you push FwdAll and then the messages button and/or your vm pilot number.

        ----- Original Message ----- 
        From: Robin H 
        To: cisco-voip at puck.nether.net 
        Sent: Monday, December 12, 2005 11:19 PM
        Subject: [cisco-voip] Function of VM Pilot CSS


        Hi, Can anyone explain to me the function of CSS on VM Pilot configuration page? I set my VM ports and internal phones in a unique partition and CSS. So far has no problem accessing VM. Thanks

        regards,
        Robin


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


        _______________________________________________
        cisco-voip mailing list
        cisco-voip at puck.nether.net
        https://puck.nether.net/mailman/listinfo/cisco-voip
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From techguy at gmail.com  Thu Dec 15 08:38:27 2005
From: techguy at gmail.com (TechGuy)
Date: Thu Dec 15 08:38:29 2005
Subject: [cisco-voip] Anyone using Cistera Call Center Recording ?
Message-ID: <a1b07ea60512150538y2469f0c5r6588f4119032785a@mail.gmail.com>

Just thought I would ask and see who else might be using the Cistera Call
Center Recording?  We just got ours in and setup yesterday, so far so good.
Seems rather easy and simple, however I have questions on maintenance and
managing the device from an administration stand point and want to talk with
others who use one and have experience administering one.

Thanks!
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From pwalenta at wi.rr.com  Thu Dec 15 08:44:48 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu Dec 15 08:44:36 2005
Subject: [cisco-voip] IVR transfer to Voicemail
In-Reply-To: <f99cc3f60512150227q1c2a04d5q472acc2de83386f1@mail.gmail.com>
Message-ID: <200512151341.jBFDfJgu020262@ms-smtp-03.rdc-kc.rr.com>

Cisco's IVR (I'm assuming IPCC express) doesn't support the * if that's what
you've used for your CCM -> Unity transfer.  I've always had to invent an
extension for the IVR to dial and use a special mask on a voicemail profile
to get this to work.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robin H
Sent: Thursday, December 15, 2005 4:28 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IVR transfer to Voicemail



Hi,
I need to setup an IVR script that allows transfer to subscriber's voicemail
directly. I have created necessary transfer from CCM to unity and it only
works for direct dialing from internal phones. From IVR script, the transfer
always failed and the caller got multiple beeps before hearing an invalid
extension prompt. Anyone experience this before? Plz enlighten me. Thanks 
 
Regards,
Robin

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From pwalenta at wi.rr.com  Thu Dec 15 08:47:47 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu Dec 15 08:47:35 2005
Subject: [cisco-voip] Call Manager Serviceability
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC0C0173@CSEXCH01.charles-stanley.co.uk>
Message-ID: <200512151345.jBFDjmZ7014785@ms-smtp-01.rdc-kc.rr.com>

H.323 gateways never "register" with CallManager because H.323 is not a
completely statefull protocol.  There are keepalives when a call is in
progress, but that's it.
 
CCM Serviceability however can show calls active to H.323 devices.  If you
poke around in Serviceability monitor they can be found under call
processing I believe.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Thursday, December 15, 2005 5:40 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call Manager Serviceability



On the Call Manager serviceability, I am unable to view any H323 gateways we
have a number of them at remote sites and only 1 MGCP gateway, the MGCP
gateway displays on the device summary fine. Is this because the status on
the Call manager shows as unknown and is there any way of getting the H323
gateways to show as we want to monitor amount of calls each remote site is
placing.




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

The information contained in this e-mail is strictly confidential, some or
all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender
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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 accept that
there
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Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
****************************************************************************
********************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

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

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From Nick.Kassel at Charles-Stanley.co.uk  Thu Dec 15 08:57:19 2005
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu Dec 15 08:57:32 2005
Subject: [cisco-voip] Call Manager Serviceability
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC0C0176@CSEXCH01.charles-stanley.co.uk>

Thanks, I was hoping to be able to see the active calls per Gateway
though.

 

________________________________

From: Philip Walenta [mailto:pwalenta at wi.rr.com] 
Sent: Thursday, December 15, 2005 1:48 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Call Manager Serviceability

 

H.323 gateways never "register" with CallManager because H.323 is not a
completely statefull protocol.  There are keepalives when a call is in
progress, but that's it.

 

CCM Serviceability however can show calls active to H.323 devices.  If
you poke around in Serviceability monitor they can be found under call
processing I believe.

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
	Sent: Thursday, December 15, 2005 5:40 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Call Manager Serviceability

	On the Call Manager serviceability, I am unable to view any H323
gateways we have a number of them at remote sites and only 1 MGCP
gateway, the MGCP gateway displays on the device summary fine. Is this
because the status on the Call manager shows as unknown and is there any
way of getting the H323 gateways to show as we want to monitor amount of
calls each remote site is placing.

	 


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

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets 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.

	 

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

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
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should this e-mail have been incorrectly addressed or transmitted.

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 accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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 Tozer Wingate are divisions of Charles Stanley & Co. Ltd

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

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From pwalenta at wi.rr.com  Thu Dec 15 09:04:12 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu Dec 15 09:04:00 2005
Subject: [cisco-voip] Call Manager Serviceability
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC0C0176@CSEXCH01.charles-stanley.co.uk>
Message-ID: <200512151402.jBFE2DZ7004569@ms-smtp-01.rdc-kc.rr.com>

If this is 4.0 or 4.1, I believe you can monitor specific gateways by going
into the call manager performance monitoring details and selecting specific
gateways (which I think is listed at the bottom of the RTMT tool).


  _____  

From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Thursday, December 15, 2005 7:57 AM
To: Philip Walenta; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Call Manager Serviceability



Thanks, I was hoping to be able to see the active calls per Gateway though.

 


  _____  


From: Philip Walenta [mailto:pwalenta at wi.rr.com] 
Sent: Thursday, December 15, 2005 1:48 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Call Manager Serviceability

 

H.323 gateways never "register" with CallManager because H.323 is not a
completely statefull protocol.  There are keepalives when a call is in
progress, but that's it.

 

CCM Serviceability however can show calls active to H.323 devices.  If you
poke around in Serviceability monitor they can be found under call
processing I believe.

 


  _____  


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Thursday, December 15, 2005 5:40 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call Manager Serviceability

On the Call Manager serviceability, I am unable to view any H323 gateways we
have a number of them at remote sites and only 1 MGCP gateway, the MGCP
gateway displays on the device summary fine. Is this because the status on
the Call manager shows as unknown and is there any way of getting the H323
gateways to show as we want to monitor amount of calls each remote site is
placing.

 



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

The information contained in this e-mail is strictly confidential, some or
all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
****************************************************************************
********************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets 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.

 




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

The information contained in this e-mail is strictly confidential, some or
all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, 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 and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

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 accept that
there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
****************************************************************************
********************


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 Tozer Wingate are divisions of Charles
Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, International Securities Markets 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 nkhwaja at gmail.com  Thu Dec 15 09:10:24 2005
From: nkhwaja at gmail.com (Nasser Khwaja)
Date: Thu Dec 15 09:10:27 2005
Subject: [cisco-voip] Cat 6509 memory commands request
In-Reply-To: <23435d3d0512142018r3606b94gb2476bc06875df7d@mail.gmail.com>
References: <23435d3d0512142018r3606b94gb2476bc06875df7d@mail.gmail.com>
Message-ID: <23435d3d0512150610s5f0b3173g77148311c049ce7a@mail.gmail.com>

Hi,
On a Catalyst 6509, please let me know the commands to find the RAM on each
of the following?

1)Catalyst 6000 supervisor 2 (WS-X6K-SUP2-2GE)

2)Policy Feature Card 2 (WS-F6K-PFC2)

3)Cat6k MSFC 2 daughterboard  (WS-F6K-MSFC2)

Thanks
-Tennisman.
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From ciscozest at gmail.com  Thu Dec 15 09:30:53 2005
From: ciscozest at gmail.com (Ciscozest)
Date: Thu Dec 15 09:30:49 2005
Subject: [cisco-voip] IVR transfer to Voicemail
References: <200512151341.jBFDfJgu020262@ms-smtp-03.rdc-kc.rr.com>
Message-ID: <003101c60184$29c36560$6301a8c0@windfall99>

Yes, you are right about my setting. If that's the case I will use "#" or prefix digit as last resort then :( 
Thanks for your confirmation.  
  ----- Original Message ----- 
  From: Philip Walenta 
  To: 'Robin H' ; cisco-voip at puck.nether.net 
  Sent: Thursday, December 15, 2005 9:44 PM
  Subject: RE: [cisco-voip] IVR transfer to Voicemail


  Cisco's IVR (I'm assuming IPCC express) doesn't support the * if that's what you've used for your CCM -> Unity transfer.  I've always had to invent an extension for the IVR to dial and use a special mask on a voicemail profile to get this to work.



----------------------------------------------------------------------------
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robin H
    Sent: Thursday, December 15, 2005 4:28 AM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] IVR transfer to Voicemail


    Hi,
    I need to setup an IVR script that allows transfer to subscriber's voicemail directly. I have created necessary transfer from CCM to unity and it only works for direct dialing from internal phones. From IVR script, the transfer always failed and the caller got multiple beeps before hearing an invalid extension prompt. Anyone experience this before? Plz enlighten me. Thanks 

    Regards,
    Robin
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From paland at gmail.com  Thu Dec 15 10:04:59 2005
From: paland at gmail.com (Patrick Aland)
Date: Thu Dec 15 10:05:01 2005
Subject: [cisco-voip] Which g/w is my call using
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A11F7E@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A11F7E@VISHNU.wesd.org>
Message-ID: <960c6a4d0512150704ree0fd4bp42d0f441c9306932@mail.gmail.com>

Jus tremember that DNA doesn't take into account actual gateway resources so
while it will tell you the route-list, etc that you might possibly using it
won't tell you which gateway will actually end up processing the call, using
the web page  on the phone is the easiest way (I've found) to determine what
gateway you are actually using for a particular call.

--Patrick

On 12/13/05, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>  https://cm/dna
>
>
>
> Dial number analyzer works well if you have it setup also.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Lelio Fulgenzi
> *Sent:* Tuesday, December 13, 2005 7:46 AM
> *To:* DeRooy, Aaron H; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Which g/w is my call using
>
>
>
> If you go to the webpage of the phone and go to streaming statistics, you
> should see something like this:
>
>
>
> *Remote Address *
>
>
>
> *<ipaddr>/<port>*
>
> *Local Address*
>
>
>
> *<ipaddr>/<port>*
>
>
>
> This is what I use to see what gateway it is using. We use 6008 T1/PRI
> MGCP and it works well.
>
>
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I had a coffee and Coke at lunch today...and now, I've got more jitter
> than an
> IP phone on a long haul 10base2
> connection"                                           LFJ
>
>  ----- Original Message -----
>
> *From:* DeRooy, Aaron H <Aaron.Derooy at xerox.com>
>
> *To:* cisco-voip at puck.nether.net
>
> *Sent:* Tuesday, December 13, 2005 10:39 AM
>
> *Subject:* [cisco-voip] Which g/w is my call using
>
>
>
> Without doing a debug or show command on the g/w itself, does anyone
> know how I can find out which g/w a particular call is using? I tried
> looking at the Event Viewer but it did now update with an entry when I
> made  a PSTN call but it did not update.
> Thanks,
> Aaron
>
>
>
>
> _______________________________________________
> 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
>
>
>


--
--Patrick
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From PRuttman at foley.com  Thu Dec 15 10:21:22 2005
From: PRuttman at foley.com (Ruttman, Peter G.)
Date: Thu Dec 15 10:21:38 2005
Subject: [cisco-voip] Need help Verifying QoS configuration
Message-ID: <7E3064B7EFC16C41B185C41D141008B7BBBFC3@foleyex97.foleylaw.com>

Okay, I installed a new office and it has a separate building with a WAN
link to the larger office.  There is no PRI connection there so they
rely on the WAN to get to the larger office to get to the PRI to make
outbound and receive inbound calls.  I have enabled QoS but we are still
receiving reports of robotic or choppy voice.  I decided I wanted to
move up the chain created a verify policy to make sure that on each
router I was seeing marked packets.  Surprisingly, I have been unable to
do this.  I see with the command 'show policy-map int' that I am getting
packets marked with dscp of EF at the router but when I try to do this
on the switches I am unable to create a proper class/policy design to
see the packets.

I found a doc for the 3550 to do QoS at the wiring closet and modified
that with no luck.  Here's the design that isn't working:

class-map match-any VOICE
  match access-group 1
!
!
policy-map VERIFY
  class VOICE
    set ip dscp 40 (I ultimately want to use dscp 46 but I saw some
errors when I didn't have something defined in the class in the policy
so I put this here.  I've defined my router to watch for dscp 40 as well
so I should see the counters grow there as well if I get this working.)


access-list 1 permit 10.33.109.30


interface GigabitEthernet0/1
 switchport trunk encapsulation dot1q
 switchport mode trunk
 no ip address
 speed nonegotiate
 service-policy input VERIFY
 auto qos voip trust
 wrr-queue bandwidth 20 1 80 1
 wrr-queue queue-limit 80 1 20 1
 wrr-queue cos-map 1 0 1 2 4
 wrr-queue cos-map 3 3 6 7
 wrr-queue cos-map 4 5
 priority-queue out
!
!
interface GigabitEthernet0/2
 switchport trunk encapsulation dot1q
 switchport mode trunk
 no ip address
 service-policy input VERIFY
 auto qos voip trust
 wrr-queue bandwidth 20 1 80 1
 wrr-queue queue-limit 80 1 20 1
 wrr-queue cos-map 1 0 1 2 4
 wrr-queue cos-map 3 3 6 7
 wrr-queue cos-map 4 5
 priority-queue out
!
For the port with the phone that is listed in the access-list I have
tried the following:

interface FastEthernet0/11
 switchport trunk encapsulation dot1q
 switchport trunk native vlan 9
 switchport mode trunk
 switchport voice vlan 109
 no ip address
 spanning-tree portfast

Or:

interface FastEthernet0/11
 switchport trunk encapsulation dot1q
 switchport trunk native vlan 9
 switchport mode trunk
 switchport voice vlan 109
 no ip address
 mls qos trust device cisco-phone
 mls qos trust cos
 auto qos voip cisco-phone
 wrr-queue bandwidth 20 1 80 1
 wrr-queue min-reserve 1 5
 wrr-queue min-reserve 2 6
 wrr-queue min-reserve 3 7
 wrr-queue min-reserve 4 8
 wrr-queue cos-map 1 0 1 2 4
 wrr-queue cos-map 3 3 6 7
 wrr-queue cos-map 4 5
 priority-queue out
 spanning-tree portfast

Or:

interface FastEthernet0/11
 switchport trunk encapsulation dot1q
 switchport trunk native vlan 9
 switchport mode trunk
 switchport voice vlan 109
 no ip address
 mls qos trust device cisco-phone
 mls qos trust dscp
 auto qos voip cisco-phone
 wrr-queue bandwidth 20 1 80 1
 wrr-queue min-reserve 1 5
 wrr-queue min-reserve 2 6
 wrr-queue min-reserve 3 7
 wrr-queue min-reserve 4 8
 wrr-queue cos-map 1 0 1 2 4
 wrr-queue cos-map 3 3 6 7
 wrr-queue cos-map 4 5
 priority-queue out
 spanning-tree portfast

Here is the output to show policy-map int:

 GigabitEthernet0/1

  service-policy input: VERIFY

    class-map: VOICE (match-any)
      0 packets, 0 bytes
      5 minute offered rate 0 bps, drop rate 0 bps
      match: access-group 1
        0 packets, 0 bytes
        5 minute rate 0 bps

    class-map: class-default (match-any)
      0 packets, 0 bytes
      5 minute offered rate 0 bps, drop rate 0 bps
      match: any
        0 packets, 0 bytes
        5 minute rate 0 bps

 GigabitEthernet0/2

  service-policy input: VERIFY

    class-map: VOICE (match-any)
      0 packets, 0 bytes
      5 minute offered rate 0 bps, drop rate 0 bps
      match: access-group 1
        0 packets, 0 bytes
        5 minute rate 0 bps

    class-map: class-default (match-any)
      0 packets, 0 bytes
      5 minute offered rate 0 bps, drop rate 0 bps
      match: any
        0 packets, 0 bytes
        5 minute rate 0 bps





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From Scott.Voll at wesd.org  Thu Dec 15 10:57:06 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 15 10:56:49 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
Message-ID: <D713462ED535184D830F6F6301753E7E01A1205E@VISHNU.wesd.org>

I had some problems with a 7935 and ended up RMA'ing it.  But after I
RMA'd it Wes told me how to reset it to Factory defaults.  Might be
worth a try.

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

Do a **1 from the phone. A dialog will appear asking if you want to
delete the phone configuration. Press Yes to clear the complete setting.

If you press  NO, . it will  put you in the phone setting  menu screen,
which also gives  the option in  changing the  passwd.



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

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, December 14, 2005 4:36 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco 7935 firmware issues

Hello,

I have inherited a Cisco 7935 and I am running CM 4.0
I am having trouble getting the 7935 to pick up new firmware. I have a
tftp server running on my CM Publisher, and the device defaults are set
correctly for the firmware version. However, the screen only displays:
Station IP: 10.96.3.58
Contacting: 10.96.3.11 (my tftp server)

When I go to http://10.96.3.58 I get:
Wind Web Server Error Report:
Server Error: 501 Not Implemented
No RPM for this combination of URL and method
/dynaform/index_Dyna.htm

I am able to telnet to 10.96.3.58, but I get a username & password
prompt, but I do not know the password, and cisco/cisco doesn't work.

Does anyone know how else I can force flash this device?

On another note, I had to explicitly set the VLAN on my switch, as the
device was picking up the data vlan when connecting, shouldn't this
device auto register on the voice vlan?

Thanks,
Avidan

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 


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


From Scott.Voll at wesd.org  Thu Dec 15 11:05:54 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 15 11:05:37 2005
Subject: [cisco-voip] IOS for1760
Message-ID: <D713462ED535184D830F6F6301753E7E01A1205F@VISHNU.wesd.org>

I have a 1760 running 12.3(4)T1 that has been running 1 year 47 weeks.
Just has fxo ports / t1(WAN) / Ethernet.

 

sh ver

Cisco IOS Software, C1700 Software (C1700-IPVOICE-M), Version 12.3(4)T1,
RELEASE SOFTWARE (fc2)

TAC Support: http://www.cisco.com/tac

Copyright (c) 1986-2003 by Cisco Systems, Inc.

Compiled Tue 25-Nov-03 05:06 by hqluong

 

ROM: System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1)

 

Portland_Rd uptime is 1 year, 47 weeks, 21 hours, 34 minutes

System returned to ROM by power-on

System restarted at 10:23:15 PST Tue Jan 20 2004

System image file is "flash:c1700-ipvoice-mz.123-4.T1.bin"

 

Cisco 1760 (MPC860P) processor (revision 0x500) with 114688K/16384K
bytes of memory.

Processor board ID FOC073610JF (2523343492), with hardware revision 0000

MPC860P processor: part number 5, mask 2

1 FastEthernet interface

1 Serial interface

2 Voice FXO interfaces

WIC T1-DSU

32K bytes of NVRAM.

32768K bytes of processor board System flash (Read/Write)

 

Configuration register is 0x2102

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Thursday, December 15, 2005 3:22 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IOS for1760

 

I'm looking for a good stable version of IP voice code for a 1760, can
any one recommend one for this model.

 

 


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From BRogers at pasadenaisd.org  Thu Dec 15 11:36:59 2005
From: BRogers at pasadenaisd.org (Brad Rogers)
Date: Thu Dec 15 11:37:14 2005
Subject: [cisco-voip] Anyone using Cistera Call Center Recording ?
Message-ID: <CB3DDB018C56BB42A32D5C63006B76ED5A53C9@adminexchg03.pasadenaisd.org>

We've been running one for about a year now.  It works pretty well.  It
pretty much runs itself.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Thursday, December 15, 2005 7:38 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Anyone using Cistera Call Center Recording ?

 

Just thought I would ask and see who else might be using the Cistera
Call Center Recording?  We just got ours in and setup yesterday, so far
so good.  Seems rather easy and simple, however I have questions on
maintenance and managing the device from an administration stand point
and want to talk with others who use one and have experience
administering one. 

 

Thanks!

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From jonvoip at gmail.com  Thu Dec 15 11:39:39 2005
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Dec 15 11:39:49 2005
Subject: [cisco-voip] MGCP fails to start after reload
Message-ID: <5d093f9a0512150839i7d3aba2fm5dd60e4e834184f0@mail.gmail.com>

Have a 2620XM, 12.3.7T, router took a power hit, afterwards MGCP would not
come up until a no mgcp, mgcp had been entered.

I have advised customer to upgrade to 12.3.11T7, but he wants documentation
of bug first.

Checked the bug toolkit, nothing specific is popping up.

Any ideas?
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From Dean.Sarff at mms.gov  Thu Dec 15 11:55:49 2005
From: Dean.Sarff at mms.gov (Sarff, Dean)
Date: Thu Dec 15 11:55:57 2005
Subject: [cisco-voip] re: QoS Toolsets
Message-ID: <37D05FC7972DF14DB5F6ABE256D0C36B01E3FCB5@IMSHEXPRI03.service.agency.mms.pri>

I'd like feedback from those users of either NET IQ's VoIP tools or
Integrated Research's PROGNOSIS.  How do you like your product
selection?  

Dean Sarff, MMS A&B IMD, 703-787-1272

Merry Christmas!  Happy Birthday Jesus!
May you an your family have a Blessed Holiday.

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From voipforum at apptis.com  Thu Dec 15 11:59:36 2005
From: voipforum at apptis.com (VoIP Forum)
Date: Thu Dec 15 11:59:45 2005
Subject: [cisco-voip] re: QoS Toolsets
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BCF05E8D@CH-EXCH-02.intelltech.apptis.inc>

Qovia is the way to go!

www.qovia.com <http://www.qovia.com/> 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sarff, Dean
Sent: Thursday, December 15, 2005 11:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] re: QoS Toolsets

 

I'd like feedback from those users of either NET IQ's VoIP tools or
Integrated Research's PROGNOSIS.  How do you like your product
selection?  

Dean Sarff, MMS A&B IMD, 703-787-1272

Merry Christmas!  Happy Birthday Jesus!

May you an your family have a Blessed Holiday.

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From Scott.Voll at wesd.org  Thu Dec 15 12:04:48 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 15 12:04:31 2005
Subject: [cisco-voip] re: QoS Toolsets
Message-ID: <D713462ED535184D830F6F6301753E7E01A12065@VISHNU.wesd.org>

Dean-

 

I'm just throwing this out there.

 

I went to the CIPTUG Conference 2 weeks ago and they had all the vendors
there.  I looked at both plus a couple others.  What are you looking
for?

 

I went down with the focus of finding a VoIP Management suite and came
back knowing I just wanted a really good troubleshooting suite.  What I
have decided is that I will most likely go with Qovia's solution.

 

The price break is much better and it puts sensors at remote sites that
return live information to the console and also gets real time snmp info
from the switches, routers and VGW's.  It will basically give you where
the problems are on a particular call.

 

Just my 2 cents

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sarff, Dean
Sent: Thursday, December 15, 2005 8:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] re: QoS Toolsets

 

I'd like feedback from those users of either NET IQ's VoIP tools or
Integrated Research's PROGNOSIS.  How do you like your product
selection?  

Dean Sarff, MMS A&B IMD, 703-787-1272

Merry Christmas!  Happy Birthday Jesus!

May you an your family have a Blessed Holiday.

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From Scott.Voll at wesd.org  Thu Dec 15 12:07:16 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 15 12:07:00 2005
Subject: [cisco-voip] change to Qovia  re: QoS Toolsets
Message-ID: <D713462ED535184D830F6F6301753E7E01A12066@VISHNU.wesd.org>

Are you using it?  I saw it at CIPTUG and am trying to get a quote to
show management.  Can you give me your opinion and overall impression?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of VoIP Forum
Sent: Thursday, December 15, 2005 9:00 AM
To: Sarff, Dean; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] re: QoS Toolsets

 

Qovia is the way to go!

www.qovia.com <http://www.qovia.com/> 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sarff, Dean
Sent: Thursday, December 15, 2005 11:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] re: QoS Toolsets

 

I'd like feedback from those users of either NET IQ's VoIP tools or
Integrated Research's PROGNOSIS.  How do you like your product
selection?  

Dean Sarff, MMS A&B IMD, 703-787-1272

Merry Christmas!  Happy Birthday Jesus!

May you an your family have a Blessed Holiday.

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From Scott.Voll at wesd.org  Thu Dec 15 12:09:00 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 15 12:08:43 2005
Subject: [cisco-voip] re: QoS Toolsets
Message-ID: <D713462ED535184D830F6F6301753E7E01A12068@VISHNU.wesd.org>

Also it is the Qovia sensor that is repackaged into a Cisco 1040.  

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, December 15, 2005 9:05 AM
To: Sarff, Dean; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] re: QoS Toolsets

 

Dean-

 

I'm just throwing this out there.

 

I went to the CIPTUG Conference 2 weeks ago and they had all the vendors
there.  I looked at both plus a couple others.  What are you looking
for?

 

I went down with the focus of finding a VoIP Management suite and came
back knowing I just wanted a really good troubleshooting suite.  What I
have decided is that I will most likely go with Qovia's solution.

 

The price break is much better and it puts sensors at remote sites that
return live information to the console and also gets real time snmp info
from the switches, routers and VGW's.  It will basically give you where
the problems are on a particular call.

 

Just my 2 cents

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sarff, Dean
Sent: Thursday, December 15, 2005 8:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] re: QoS Toolsets

 

I'd like feedback from those users of either NET IQ's VoIP tools or
Integrated Research's PROGNOSIS.  How do you like your product
selection?  

Dean Sarff, MMS A&B IMD, 703-787-1272

Merry Christmas!  Happy Birthday Jesus!

May you an your family have a Blessed Holiday.

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From tennille.spence at gsa.gov  Thu Dec 15 12:14:21 2005
From: tennille.spence at gsa.gov (tennille.spence@gsa.gov)
Date: Thu Dec 15 12:14:37 2005
Subject: [cisco-voip] re: QoS Toolsets
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A12065@VISHNU.wesd.org>
Message-ID: <OFE6255039.A714924C-ON862570D8.005DF87E-862570D8.005EB2A7@gsa.gov>


Hi,

We use qovia to monitor our 350 Cisco phone system and have been happy with
it. We looked at prognosis and NetIQ as well and decided on Qovia as it
gave us as much as other at a half the cost. We have two sites and three
probes. If you want to use the "active call testing" feature, which allows
you to simulate calls across two points, you must have a monitoring device
at each site if you want to test connectivity across sites. The more probes
you have the easier it will be to pinpoint the location on your network
where problems are occurring. If you just have one probe you basically will
be allowed to determine if problems are occurring while arriving or leaving
that monitoring point only.

Tennille Spence
IT Business Liaison
PBS Technology Branch
816-823-1357 Office
816-806-6342 Cell
816-823-5526 Fax

"What lies behind us and what lies before us are tiny matters compared to
what lies within us." - Ralph Waldo Emerson

"Opportunity is missed by most people because it is dressed in overalls and
looks like work." - Thomas A. Edison



                                                                           
                                                                           
             "Voll, Scott"                                                 
             <Scott.Voll at wesd.                                          To 
             org>                      "Sarff, Dean" <Dean.Sarff at mms.gov>, 
             Sent by:                  cisco-voip at puck.nether.net          
             cisco-voip-bounce                                          cc 
             s at puck.nether.net                                             
                                                                   Subject 
                                       RE: [cisco-voip] re: QoS Toolsets   
             12/15/2005 11:04                                              
             AM                                                            
                                                                           
                                                                           
                                                                           
                                                                           




Dean?

I?m just throwing this out there.

I went to the CIPTUG Conference 2 weeks ago and they had all the vendors
there.  I looked at both plus a couple others.  What are you looking for?

I went down with the focus of finding a VoIP Management suite and came back
knowing I just wanted a really good troubleshooting suite.  What I have
decided is that I will most likely go with Qovia?s solution.

The price break is much better and it puts sensors at remote sites that
return live information to the console and also gets real time snmp info
from the switches, routers and VGW?s.  It will basically give you where the
problems are on a particular call.

Just my 2 cents

Scott


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sarff, Dean
Sent: Thursday, December 15, 2005 8:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] re: QoS Toolsets



I?d like feedback from those users of either NET IQ?s VoIP tools or
Integrated Research?s PROGNOSIS.  How do you like your product selection?


Dean Sarff, MMS A&B IMD, 703-787-1272


Merry Christmas!  Happy Birthday Jesus!


May you an your family have a Blessed Holiday.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




From Court.Schuett at us.millwardbrown.com  Thu Dec 15 12:15:49 2005
From: Court.Schuett at us.millwardbrown.com (Schuett, Court (MBNAP it))
Date: Thu Dec 15 12:15:39 2005
Subject: [cisco-voip] FYI 4.1(2) to 4.1(3) Upgrade and Question
Message-ID: <90B962BEA7D42F40BF0DF00044A5A1B60B32F5E1@x-np5.am.group.local>

Last night, we upgraded from 4.1(2) to 4.1(3).  Most things ran very smooth,
but immediately after the upgrade, all calls in and out of our main location
were down.  The trick is is that our main location runs into a PBX instead
of a telco off of our PRI.  It seems that during the upgrade, several
settings on the MGCP gateway (VG200) were lost.  We are running into  Nortel
PBX (DMS-100), and these settings were lost:
 
Send Extra Leading Character in DislplayIE
MCDN Channgel Number Extension Bit Set to Zero
Interface Identifier Present
 
The calls were coming in, but were immediatly dropping from the phone
because the PBX was not getting the proper signalling.  This was the case
for both inbound and outbound.  Once they were reset, everything worked
fine.  Just in case anyone might run across something similar.
 
Which brings me to my mostly unrelated question.  We've got the equivalent
of a 7835 with 4 Gigs of RAM in it.  I'm looking that the Task Manager on it
right now, at it is showing:
 

 
 
However, my RTMT is constantly giving LowAvailableVirtualMemory errors.  
 
This only started after the upgrade and has remained even after an extra
reboot.  Anyone seen this before?
 
Thanks!
 
 
Court Schuett
 
Millward Brown
Network Systems Engineer
630-955-8983
court.schuett at us.millwardbrown.com
<mailto:court.schuett at us.millwardbrown.com> 
 

 

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From jose.mujica at netxar.com  Thu Dec 15 12:26:12 2005
From: jose.mujica at netxar.com (Jose Mujica)
Date: Thu Dec 15 12:26:15 2005
Subject: [cisco-voip] SRST VM-Integration with FXO Lines
Message-ID: <20051215172109.M58764@netxar.com>

Hi. 

 I already configured the vm-integration for sending dtmf tones through FXO
lines connected to the local srst router in order to access the Unity Server
at central site that is connected to the PSTN using T1/PRI circuits. The
problem that I'm having is that the srst router is sending all digits (i.e.
Telephone Number + #FDN#2 to the PSTN and the generic message is saying that
the call cannot be completed as dialed.

 I was thinking that the router will send the extra dtmf tones after the call
has been established. Am I correct? I'm using SRST 3.3 in an IOS 12.3(14)T5.

Below are the vm-integration commands:

vm-integration
 pattern direct * CGN
 pattern ext-to-ext no-answer # FDN #2
 pattern ext-to-ext busy # FDN #2
 pattern trunk-to-ext no-answer # FDN #2
 pattern trunk-to-ext busy # FDN #2
!

Any idea?
-Jose


From voipforum at apptis.com  Thu Dec 15 12:29:29 2005
From: voipforum at apptis.com (VoIP Forum)
Date: Thu Dec 15 12:32:48 2005
Subject: [cisco-voip] RE: change to Qovia  re: QoS Toolsets
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BCF05E95@CH-EXCH-02.intelltech.apptis.inc>

This is not a sales pitch, and I get nothing for this...

 

Qovia simply blows Cisco's current ITEM release out of the water, and
leaves the awkwardness of Prognosis in the dark. 

Cost wise, it is far less expensive than NetIQ in most installations and
can do so much more.

 

I my opinion it is the best Cisco VoIP Real-Time management tool there
is. We use it throughout our entire enterprise VoIP Network and have
been able to resolve many issues in minutes where we had spent hours in
the past troubleshooting. The best thing I could say that you may win
Mgmt over with... "We use it for SLA and QoS mgmt on remote sites. Since
the probes can simulate traffic to each other across the WAN. Due to
mitigation of lost call or poor call quality, Qovia pays for itself
within months instead of years. We have reduced time to roll out new
sites as well, since we can take a few days worth of testing and put it
into a few hours worth of active tests using the probes in conjunction
with Qovia Central.", and you can quote me on that.

 

It is 100% off box so there are not compliance and SMARTnet issues, plus
it utilizes its own DB and does not rely on the CDR which is not
accessible until after the call has ended. We have it integrated to our
NMS it sends e-mail pages to us when there is an issue. That let's us
actually see the MOS of the active call and on each leg of the call
while the user is on the call. This is great for real-time circuit
maintenance; for example, we can tweak llq (QoS) settings while the call
is up to see if that corrects the issue or not. We can identify bad
circuits and codec issues, etc... We can look at any services and
process utilization on the CCMs real-time or historically. We also use
Qovia for capacity planning.

 

Another awesome feature is the device inventory reports that will
actually show you comparisons to see who is moving their phone and from
where to where. You can run hourly, nightly, weekly, monthly reports and
they are all useful. We had visiting users grabbing phones off desks and
moving them wherever, and we could find them with a simple report as
well as what day it moved to track the culprits down.

 

 

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Thursday, December 15, 2005 12:07 PM
To: VoIP Forum
Cc: cisco-voip at puck.nether.net
Subject: change to Qovia re: QoS Toolsets

 

Are you using it?  I saw it at CIPTUG and am trying to get a quote to
show management.  Can you give me your opinion and overall impression?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of VoIP Forum
Sent: Thursday, December 15, 2005 9:00 AM
To: Sarff, Dean; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] re: QoS Toolsets

 

Qovia is the way to go!

www.qovia.com <http://www.qovia.com/> 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sarff, Dean
Sent: Thursday, December 15, 2005 11:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] re: QoS Toolsets

 

I'd like feedback from those users of either NET IQ's VoIP tools or
Integrated Research's PROGNOSIS.  How do you like your product
selection?  

Dean Sarff, MMS A&B IMD, 703-787-1272

Merry Christmas!  Happy Birthday Jesus!

May you an your family have a Blessed Holiday.

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From Scott.Voll at wesd.org  Thu Dec 15 12:37:53 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 15 12:37:36 2005
Subject: [cisco-voip] RE: change to Qovia  re: QoS Toolsets
Message-ID: <D713462ED535184D830F6F6301753E7E01A1206E@VISHNU.wesd.org>

Thanks... the more info I can sent to management the better.

 

Thanks again

 

Scott

 

________________________________

From: VoIP Forum [mailto:voipforum at apptis.com] 
Sent: Thursday, December 15, 2005 9:29 AM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: RE: change to Qovia re: QoS Toolsets

 

This is not a sales pitch, and I get nothing for this...

 

Qovia simply blows Cisco's current ITEM release out of the water, and
leaves the awkwardness of Prognosis in the dark. 

Cost wise, it is far less expensive than NetIQ in most installations and
can do so much more.

 

I my opinion it is the best Cisco VoIP Real-Time management tool there
is. We use it throughout our entire enterprise VoIP Network and have
been able to resolve many issues in minutes where we had spent hours in
the past troubleshooting. The best thing I could say that you may win
Mgmt over with... "We use it for SLA and QoS mgmt on remote sites. Since
the probes can simulate traffic to each other across the WAN. Due to
mitigation of lost call or poor call quality, Qovia pays for itself
within months instead of years. We have reduced time to roll out new
sites as well, since we can take a few days worth of testing and put it
into a few hours worth of active tests using the probes in conjunction
with Qovia Central.", and you can quote me on that.

 

It is 100% off box so there are not compliance and SMARTnet issues, plus
it utilizes its own DB and does not rely on the CDR which is not
accessible until after the call has ended. We have it integrated to our
NMS it sends e-mail pages to us when there is an issue. That let's us
actually see the MOS of the active call and on each leg of the call
while the user is on the call. This is great for real-time circuit
maintenance; for example, we can tweak llq (QoS) settings while the call
is up to see if that corrects the issue or not. We can identify bad
circuits and codec issues, etc... We can look at any services and
process utilization on the CCMs real-time or historically. We also use
Qovia for capacity planning.

 

Another awesome feature is the device inventory reports that will
actually show you comparisons to see who is moving their phone and from
where to where. You can run hourly, nightly, weekly, monthly reports and
they are all useful. We had visiting users grabbing phones off desks and
moving them wherever, and we could find them with a simple report as
well as what day it moved to track the culprits down.

 

 

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Thursday, December 15, 2005 12:07 PM
To: VoIP Forum
Cc: cisco-voip at puck.nether.net
Subject: change to Qovia re: QoS Toolsets

 

Are you using it?  I saw it at CIPTUG and am trying to get a quote to
show management.  Can you give me your opinion and overall impression?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of VoIP Forum
Sent: Thursday, December 15, 2005 9:00 AM
To: Sarff, Dean; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] re: QoS Toolsets

 

Qovia is the way to go!

www.qovia.com <http://www.qovia.com/> 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sarff, Dean
Sent: Thursday, December 15, 2005 11:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] re: QoS Toolsets

 

I'd like feedback from those users of either NET IQ's VoIP tools or
Integrated Research's PROGNOSIS.  How do you like your product
selection?  

Dean Sarff, MMS A&B IMD, 703-787-1272

Merry Christmas!  Happy Birthday Jesus!

May you an your family have a Blessed Holiday.

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From candace_holman at harvard.edu  Thu Dec 15 13:07:23 2005
From: candace_holman at harvard.edu (Candace Holman)
Date: Thu Dec 15 13:07:31 2005
Subject: [cisco-voip] re: QoS Toolsets
Message-ID: <43A1B0DB.4030303@harvard.edu>

What version of Qovia are you referring to? I don't have some of these 
features in my installation.

I've tested all three of the products under discussion, and best 
all-around alerting tool so far is NetIQ. Qovia needs a little work to 
provide a way to turn off alerts when the listener decides, not just 
when Qovia sees the condition go away. I get spammed by my own tool 
because it is not adjustable enough (at least not in the version I use). 
We should be able to acknowledge a problem to stop the alerts, 
consolidate alerts, and choose the delivery media. Qovia can only send 
emails or snmp traps. Also the granularity of the MoS metrics is a 
little under par. Qovia doesn't do more than NetIQ, except in one area - 
providing quick, easy and multi-tenant/multi-privilege real-time access 
to phones and call data. This is a hardware solution with a centralized 
server, so you have to cart it out and set it up, and have available a 
span port for each probe.

IMO, NetIQ does more than Qovia, and monitors all kinds of information 
on the CallManagers and gateways, not just the phones. Qovia only seems 
to monitor phones and quite limited data on the gateways and switches. 
NetIQ can send alerts to a variety of media: pages, emails, snmp traps, 
or just log the data.

Candace

Subject:
[cisco-voip] RE: change to Qovia re: QoS Toolsets
From:
"VoIP Forum" <voipforum at apptis.com>
Date:
Thu, 15 Dec 2005 12:29:29 -0500

To:
"Voll, Scott" <Scott.Voll at wesd.org>
CC:
cisco-voip at puck.nether.net


*/This is not a sales pitch, and I get nothing for this?/*

Qovia simply blows Cisco?s current ITEM release out of the water, and 
leaves the awkwardness of Prognosis in the dark.

Cost wise, it is far less expensive than NetIQ in most installations and 
can do so much more.

I my opinion it is the best Cisco VoIP Real-Time management tool there 
is. We use it throughout our entire enterprise VoIP Network and have 
been able to resolve many issues in minutes where we had spent hours in 
the past troubleshooting. The best thing I could say that you may win 
Mgmt over with? ?We use it for SLA and QoS mgmt on remote sites. Since 
the probes can simulate traffic to each other across the WAN. Due to 
mitigation of lost call or poor call quality, Qovia pays for itself 
within months instead of years. We have reduced time to roll out new 
sites as well, since we can take a few days worth of testing and put it 
into a few hours worth of active tests using the probes in conjunction 
with Qovia Central.?, and you can quote me on that.

It is 100% off box so there are not compliance and SMARTnet issues, plus 
it utilizes its own DB and does not rely on the CDR which is not 
accessible until after the call has ended. We have it integrated to our 
NMS it sends e-mail pages to us when there is an issue. That let?s us 
actually see the MOS of the active call and on each leg of the call 
while the user is on the call. This is great for real-time circuit 
maintenance; for example, we can tweak llq (QoS) settings while the call 
is up to see if that corrects the issue or not. We can identify bad 
circuits and codec issues, etc? We can look at any services and process 
utilization on the CCMs real-time or historically. We also use Qovia for 
capacity planning.

Another awesome feature is the device inventory reports that will 
actually show you comparisons to see who is moving their phone and from 
where to where. You can run hourly, nightly, weekly, monthly reports and 
they are all useful. We had visiting users grabbing phones off desks and 
moving them wherever, and we could find them with a simple report as 
well as what day it moved to track the culprits down.



From CORTIZ at broward.org  Thu Dec 15 13:44:34 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Thu Dec 15 13:44:43 2005
Subject: [cisco-voip] Citrix Click to call
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B79@MREXVS01.bc.broward.cty>

1) If anyone here went to CIPTUG and has a contact at Citrix please
Unicast me their info (I am in Florida).  I would like to see how much
the click to call application goes for.

 

2) Anyone out there using this yet? It seems to me instead of only
having the option to dial from Outlook, dialing from any web page and
Email client would be better.

 

Thanks

 

Carlos Ortiz

Senior Information Technology Specialist

Broward County Office of Information Technology

Data Communications

954-357-7981

cortiz at broward.org

 

 

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From it at cimgroup.com  Thu Dec 15 14:11:07 2005
From: it at cimgroup.com (IT)
Date: Thu Dec 15 14:07:09 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0265C33B@exchange.cimgroup.com>

I have tried reseting to factory defaults, but the issue remains. Is
there a default vxWorks username and password? I feel like if I can get
in via SSH, I can force a firmware upgrade...

Thanks,
Avidan

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Thursday, December 15, 2005 7:57 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Cisco 7935 firmware issues

I had some problems with a 7935 and ended up RMA'ing it.  But after I
RMA'd it Wes told me how to reset it to Factory defaults.  Might be
worth a try.

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

Do a **1 from the phone. A dialog will appear asking if you want to
delete the phone configuration. Press Yes to clear the complete setting.

If you press  NO, . it will  put you in the phone setting  menu screen,
which also gives  the option in  changing the  passwd.



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

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, December 14, 2005 4:36 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco 7935 firmware issues

Hello,

I have inherited a Cisco 7935 and I am running CM 4.0
I am having trouble getting the 7935 to pick up new firmware. I have a
tftp server running on my CM Publisher, and the device defaults are set
correctly for the firmware version. However, the screen only displays:
Station IP: 10.96.3.58
Contacting: 10.96.3.11 (my tftp server)

When I go to http://10.96.3.58 I get:
Wind Web Server Error Report:
Server Error: 501 Not Implemented
No RPM for this combination of URL and method
/dynaform/index_Dyna.htm

I am able to telnet to 10.96.3.58, but I get a username & password
prompt, but I do not know the password, and cisco/cisco doesn't work.

Does anyone know how else I can force flash this device?

On another note, I had to explicitly set the VLAN on my switch, as the
device was picking up the data vlan when connecting, shouldn't this
device auto register on the voice vlan?

Thanks,
Avidan

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 


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


From jsteinberg at gmail.com  Thu Dec 15 14:16:52 2005
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu Dec 15 14:16:56 2005
Subject: [cisco-voip] deciphering 7960 Phone's internal webpages
Message-ID: <f1b65f880512151116n46d4b9a3g@mail.gmail.com>

Does anyone have any detailed information on what's available on the
7960's internal webpages?

Some things are obvious, but then there are some sections such as
'debug display' and 'stack statistics' that can be a little more
cryptic.
Example:
2005-12-12 15:07:45.220 Code:8108, P:0, S:12300900

There is also the streaming statistics section - can anyone explain
those timestamps and the jitter values.  I'm thinking the jitter value
might be:
average,max

TIA

Justin


From Scott.Voll at wesd.org  Thu Dec 15 14:20:34 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 15 14:20:16 2005
Subject: [cisco-voip] deciphering 7960 Phone's internal webpages
Message-ID: <D713462ED535184D830F6F6301753E7E01A1207E@VISHNU.wesd.org>

Don't know about the debug an stack but the stream stats are the
previous call and the jitter values are average and max.

Scott

-----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, December 15, 2005 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] deciphering 7960 Phone's internal webpages

Does anyone have any detailed information on what's available on the
7960's internal webpages?

Some things are obvious, but then there are some sections such as
'debug display' and 'stack statistics' that can be a little more
cryptic.
Example:
2005-12-12 15:07:45.220 Code:8108, P:0, S:12300900

There is also the streaming statistics section - can anyone explain
those timestamps and the jitter values.  I'm thinking the jitter value
might be:
average,max

TIA

Justin

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


From bfeeny at mac.com  Thu Dec 15 14:22:46 2005
From: bfeeny at mac.com (Brian Feeny)
Date: Thu Dec 15 14:22:49 2005
Subject: [cisco-voip] Caller Name ID possible via Facility IE with H.323?
Message-ID: <3324066.1134674566831.JavaMail.bfeeny@mac.com>


I have read the thread:

https://puck.nether.net/pipermail/cisco-voip/2005-March/003509.html

And was curious as to the outcome.  I have PRI's coming in from a 5ESS, and the carrier is sending the ID in the facility IE (inbound calls to us), and of course CM appears to want it in the display IE.  I have read of ways CM would take it via the Facility ID if using MGCP AND NI2/3 or MGCP AND QSIG.  Does any solution at all exist for H.323?  I have tried to set my switch type for NI and QSIG, both work as far as placing/receiving calls, but I still cannot get inbound name id.  I have tried to set "isdn supp-service name calling" on the gateway as well but it doesn't seem to fix it.  Gateways are running late 12.3T code, CM is 4.1(3)SR2.

Sure I can try to battle it out with Bellsouth and ask them to change to display IE, but I am sure they will not do it for me, I'll try though.  But I am hoping there is a solution for those of us with H.323 links.  I could possibly get them to switch the circuit to NI2 or QSIG.  Like I said switching the switch type on the gateway works, without having the carrier do anything probably because its just compatible, but not sure if formally having them change the switch emulation would do anything.  The q.931 output is definitely more "friendly" with QSIG/NI, as it decodes the facility IE.  With 5ESS it doesn't like to receive the name id facility ie stuff, it errors on it (but of course call completes).

Brian

From jgrace at digitelusa.net  Thu Dec 15 15:39:51 2005
From: jgrace at digitelusa.net (James)
Date: Thu Dec 15 15:39:45 2005
Subject: [cisco-voip] Speed dial on softkey
Message-ID: <62570315-1134679182-cardhu_blackberry.rim.net-3710-@engine21-cell02>

   I have a question.    Is it posible to create a speed dial on the softkey button
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From voipforum at apptis.com  Thu Dec 15 16:24:37 2005
From: voipforum at apptis.com (VoIP Forum)
Date: Thu Dec 15 16:24:44 2005
Subject: [cisco-voip] CRS Application Trigger Question
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BCF05EAD@CH-EXCH-02.intelltech.apptis.inc>

What happens to an incoming call when the Maximum Sessions limit is
reached on a trigger? The help docs don't help too much.

 

Max Sessions 

Maximum number of simultaneous instances of the associated application
that can run when triggered by this trigger

 

I need it to send a busy tone back to the caller, but currently it just
rings and rings.

 

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From mfa at crec.ifas.ufl.edu  Thu Dec 15 17:45:58 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Thu Dec 15 17:46:02 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
References: <200512152124.jBFLOi1e053011@puck.nether.net>
Message-ID: <01f401c601c9$50dfdb20$400cfb0a@crec.ifas.ufl.edu>

I had a similar issue with a 7935 a few months ago.  In my case, we RMAed 
the 7935, but it turned out to be the switch.  Don't know if this will help, 
since it's a long convoluted story, but here's the summary I sent to TAC 
after I eventually found the real problem.  I never received a reply to my 
question in the last paragraph.

The problem was/is, we are using 3524-PWR-XL switches, and port security
behaves a little differently on these switches than it does on "real" IOS
switches, as documented in the TAC Case Collection at
http://www.ciscotaccc.com/lanswitching/showcase?case=K96362614.  (We had
been using port security to protect public access jacks where the conference
phones were used.)  One of the effects of this is that if Port Security is
used, once a device (e.g., the 7935 in this case) is plugged into a port on
a given switch, it can't be plugged into any other port on the switch.  In
our case, it looked as if the 7935 was broken, since it refused to give up
its old DHCP-assigned address, and could not be reset to its default
settings.

Eliminating Port Security from the configuration solved the apparent "device
failure" problem and allowed us to move the phone freely among all the 3524
ports.  However, this doesn't explain why requesting the phone reset to
device defaults didn't work either -- if you could find out why this is, I'd
appreciate it.

Good luck,

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL


> Date: Thu, 15 Dec 2005 11:11:07 -0800
> From: "IT" <it at cimgroup.com>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
> To: "Voll, Scott" <Scott.Voll at wesd.org>, "IT" <it at cimgroup.com>,
> <cisco-voip at puck.nether.net>
> Message-ID:
> <B487D4C1FE6D04439061018C6EBC109B0265C33B at exchange.cimgroup.com>
> Content-Type: text/plain; charset="us-ascii"
>
> I have tried reseting to factory defaults, but the issue remains. Is
> there a default vxWorks username and password? I feel like if I can get
> in via SSH, I can force a firmware upgrade...
>
> Thanks,
> Avidan
>
> -----Original Message-----
> From: Voll, Scott [mailto:Scott.Voll at wesd.org]
> Sent: Thursday, December 15, 2005 7:57 AM
> To: IT; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
>
> I had some problems with a 7935 and ended up RMA'ing it.  But after I
> RMA'd it Wes told me how to reset it to Factory defaults.  Might be
> worth a try.
>
> *********************
>
> Do a **1 from the phone. A dialog will appear asking if you want to
> delete the phone configuration. Press Yes to clear the complete setting.
>
> If you press  NO, . it will  put you in the phone setting  menu screen,
> which also gives  the option in  changing the  passwd.
>
>
>
> **********************
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
> Sent: Wednesday, December 14, 2005 4:36 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Cisco 7935 firmware issues
>
> Hello,
>
> I have inherited a Cisco 7935 and I am running CM 4.0
> I am having trouble getting the 7935 to pick up new firmware. I have a
> tftp server running on my CM Publisher, and the device defaults are set
> correctly for the firmware version. However, the screen only displays:
> Station IP: 10.96.3.58
> Contacting: 10.96.3.11 (my tftp server)
>
> When I go to http://10.96.3.58 I get:
> Wind Web Server Error Report:
> Server Error: 501 Not Implemented
> No RPM for this combination of URL and method
> /dynaform/index_Dyna.htm
>
> I am able to telnet to 10.96.3.58, but I get a username & password
> prompt, but I do not know the password, and cisco/cisco doesn't work.
>
> Does anyone know how else I can force flash this device?
>
> On another note, I had to explicitly set the VLAN on my switch, as the
> device was picking up the data vlan when connecting, shouldn't this
> device auto register on the voice vlan?
>
> Thanks,
> Avidan


From lelio at uoguelph.ca  Thu Dec 15 17:50:31 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Dec 15 17:50:35 2005
Subject: [cisco-voip] Speed dial on softkey
References: <62570315-1134679182-cardhu_blackberry.rim.net-3710-@engine21-cell02>
Message-ID: <015d01c601c9$f3e56830$c901a8c0@EATIUSBIRDIUS>

as far as I know, softkeys are not userdefinable nor definable with speed dials. too bad, a user defined softkey would be great.
  ----- Original Message ----- 
  From: James 
  To: CiscosupportUpuck 
  Sent: Thursday, December 15, 2005 3:39 PM
  Subject: [cisco-voip] Speed dial on softkey


     I have a question.    Is it posible to create a speed dial on the softkey button
  James D. Grace 
  System Engineer
  CCNP CCNA MCSE MCDBA 
  Digitel Corp.
  _______________________________________________
  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  Thu Dec 15 17:52:51 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Dec 15 17:52:55 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
References: <D713462ED535184D830F6F6301753E7E01A1205E@VISHNU.wesd.org>
Message-ID: <018d01c601ca$476f83f0$c901a8c0@EATIUSBIRDIUS>

will this work if you don't have the password? 
  ----- Original Message ----- 
  From: Voll, Scott 
  To: IT ; cisco-voip at puck.nether.net 
  Sent: Thursday, December 15, 2005 10:57 AM
  Subject: RE: [cisco-voip] Cisco 7935 firmware issues


  I had some problems with a 7935 and ended up RMA'ing it.  But after I
  RMA'd it Wes told me how to reset it to Factory defaults.  Might be
  worth a try.

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

  Do a **1 from the phone. A dialog will appear asking if you want to
  delete the phone configuration. Press Yes to clear the complete setting.

  If you press  NO, . it will  put you in the phone setting  menu screen,
  which also gives  the option in  changing the  passwd.



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

  Scott

  -----Original Message-----
  From: cisco-voip-bounces at puck.nether.net
  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
  Sent: Wednesday, December 14, 2005 4:36 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Cisco 7935 firmware issues

  Hello,

  I have inherited a Cisco 7935 and I am running CM 4.0
  I am having trouble getting the 7935 to pick up new firmware. I have a
  tftp server running on my CM Publisher, and the device defaults are set
  correctly for the firmware version. However, the screen only displays:
  Station IP: 10.96.3.58
  Contacting: 10.96.3.11 (my tftp server)

  When I go to http://10.96.3.58 I get:
  Wind Web Server Error Report:
  Server Error: 501 Not Implemented
  No RPM for this combination of URL and method
  /dynaform/index_Dyna.htm

  I am able to telnet to 10.96.3.58, but I get a username & password
  prompt, but I do not know the password, and cisco/cisco doesn't work.

  Does anyone know how else I can force flash this device?

  On another note, I had to explicitly set the VLAN on my switch, as the
  device was picking up the data vlan when connecting, shouldn't this
  device auto register on the voice vlan?

  Thanks,
  Avidan

  ******************************************************
  This email may contain trade secrets and/or other information that is 
  proprietary and/or confidential. Use of this by anyone other than the 
  intended recipient(s) is prohibited and may be unlawful. If you are not 
  the intended recipient, delete the e-mail and notify us immediately.
  ****************************************************** 


  _______________________________________________
  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 Scott.Voll at wesd.org  Thu Dec 15 18:09:55 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 15 18:09:43 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
Message-ID: <D713462ED535184D830F6F6301753E7E01A1208F@VISHNU.wesd.org>

That was my problem.... So **1 and see what happens.

 

Scott

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Thursday, December 15, 2005 2:53 PM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Cisco 7935 firmware issues

 

will this work if you don't have the password? 

	----- Original Message ----- 

	From: Voll, Scott <mailto:Scott.Voll at wesd.org>  

	To: IT <mailto:it at cimgroup.com>  ; cisco-voip at puck.nether.net 

	Sent: Thursday, December 15, 2005 10:57 AM

	Subject: RE: [cisco-voip] Cisco 7935 firmware issues

	 

	I had some problems with a 7935 and ended up RMA'ing it.  But
after I
	RMA'd it Wes told me how to reset it to Factory defaults.  Might
be
	worth a try.
	
	*********************
	
	Do a **1 from the phone. A dialog will appear asking if you want
to
	delete the phone configuration. Press Yes to clear the complete
setting.
	
	If you press  NO, . it will  put you in the phone setting  menu
screen,
	which also gives  the option in  changing the  passwd.
	
	
	
	**********************
	
	Scott
	
	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
	[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
	Sent: Wednesday, December 14, 2005 4:36 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Cisco 7935 firmware issues
	
	Hello,
	
	I have inherited a Cisco 7935 and I am running CM 4.0
	I am having trouble getting the 7935 to pick up new firmware. I
have a
	tftp server running on my CM Publisher, and the device defaults
are set
	correctly for the firmware version. However, the screen only
displays:
	Station IP: 10.96.3.58
	Contacting: 10.96.3.11 (my tftp server)
	
	When I go to http://10.96.3.58 I get:
	Wind Web Server Error Report:
	Server Error: 501 Not Implemented
	No RPM for this combination of URL and method
	/dynaform/index_Dyna.htm
	
	I am able to telnet to 10.96.3.58, but I get a username &
password
	prompt, but I do not know the password, and cisco/cisco doesn't
work.
	
	Does anyone know how else I can force flash this device?
	
	On another note, I had to explicitly set the VLAN on my switch,
as the
	device was picking up the data vlan when connecting, shouldn't
this
	device auto register on the voice vlan?
	
	Thanks,
	Avidan
	
	******************************************************
	This email may contain trade secrets and/or other information
that is 
	proprietary and/or confidential. Use of this by anyone other
than the 
	intended recipient(s) is prohibited and may be unlawful. If you
are not 
	the intended recipient, delete the e-mail and notify us
immediately.
	****************************************************** 
	
	
	_______________________________________________
	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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From lelio at uoguelph.ca  Thu Dec 15 18:40:09 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Dec 15 18:40:14 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
References: <D713462ED535184D830F6F6301753E7E01A1208F@VISHNU.wesd.org>
Message-ID: <000e01c601d0$e333b580$c901a8c0@EATIUSBIRDIUS>

argh. a TAC engineer said I had to RMA and only if I had a contract. grrrr. I'll have to try a **1 and see if it ask for a password.
  ----- Original Message ----- 
  From: Voll, Scott 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Thursday, December 15, 2005 6:09 PM
  Subject: RE: [cisco-voip] Cisco 7935 firmware issues


  That was my problem.. So **1 and see what happens.

   

  Scott

   


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

  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
  Sent: Thursday, December 15, 2005 2:53 PM
  To: Voll, Scott; IT; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Cisco 7935 firmware issues

   

  will this work if you don't have the password? 

    ----- Original Message ----- 

    From: Voll, Scott 

    To: IT ; cisco-voip at puck.nether.net 

    Sent: Thursday, December 15, 2005 10:57 AM

    Subject: RE: [cisco-voip] Cisco 7935 firmware issues

     

    I had some problems with a 7935 and ended up RMA'ing it.  But after I
    RMA'd it Wes told me how to reset it to Factory defaults.  Might be
    worth a try.

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

    Do a **1 from the phone. A dialog will appear asking if you want to
    delete the phone configuration. Press Yes to clear the complete setting.

    If you press  NO, . it will  put you in the phone setting  menu screen,
    which also gives  the option in  changing the  passwd.



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

    Scott

    -----Original Message-----
    From: cisco-voip-bounces at puck.nether.net
    [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
    Sent: Wednesday, December 14, 2005 4:36 PM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] Cisco 7935 firmware issues

    Hello,

    I have inherited a Cisco 7935 and I am running CM 4.0
    I am having trouble getting the 7935 to pick up new firmware. I have a
    tftp server running on my CM Publisher, and the device defaults are set
    correctly for the firmware version. However, the screen only displays:
    Station IP: 10.96.3.58
    Contacting: 10.96.3.11 (my tftp server)

    When I go to http://10.96.3.58 I get:
    Wind Web Server Error Report:
    Server Error: 501 Not Implemented
    No RPM for this combination of URL and method
    /dynaform/index_Dyna.htm

    I am able to telnet to 10.96.3.58, but I get a username & password
    prompt, but I do not know the password, and cisco/cisco doesn't work.

    Does anyone know how else I can force flash this device?

    On another note, I had to explicitly set the VLAN on my switch, as the
    device was picking up the data vlan when connecting, shouldn't this
    device auto register on the voice vlan?

    Thanks,
    Avidan

    ******************************************************
    This email may contain trade secrets and/or other information that is 
    proprietary and/or confidential. Use of this by anyone other than the 
    intended recipient(s) is prohibited and may be unlawful. If you are not 
    the intended recipient, delete the e-mail and notify us immediately.
    ****************************************************** 


    _______________________________________________
    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 Joe_Bell at adp.com  Thu Dec 15 19:20:38 2005
From: Joe_Bell at adp.com (Bell, Joe)
Date: Thu Dec 15 19:21:01 2005
Subject: [cisco-voip] Caller Name ID possible via Facility IE with H.323?
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3B2D76C2@EXCHANGEDS01.ds.ad.adp.com>

Brian,

Unless someone knows any pfm, there is no way to accomplish caller-id on
a PRI with H.323 unless your provider switches the data into Display IE
(good luck with that).  Recently (using 12.4(4T), I opened a TAC case
about this issue to see if there was any way to accomplish this or if it
had been addressed in the new 12.4 train.  TAC sent me a clear answer
saying H.323 and caller-id are unsupported and in order to get it, I
needed to go to MGCP.

Joe Bell

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Feeny
Sent: Thursday, December 15, 2005 11:23 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Caller Name ID possible via Facility IE with
H.323?


I have read the thread:

https://puck.nether.net/pipermail/cisco-voip/2005-March/003509.html

And was curious as to the outcome.  I have PRI's coming in from a 5ESS,
and the carrier is sending the ID in the facility IE (inbound calls to
us), and of course CM appears to want it in the display IE.  I have read
of ways CM would take it via the Facility ID if using MGCP AND NI2/3 or
MGCP AND QSIG.  Does any solution at all exist for H.323?  I have tried
to set my switch type for NI and QSIG, both work as far as
placing/receiving calls, but I still cannot get inbound name id.  I have
tried to set "isdn supp-service name calling" on the gateway as well but
it doesn't seem to fix it.  Gateways are running late 12.3T code, CM is
4.1(3)SR2.

Sure I can try to battle it out with Bellsouth and ask them to change to
display IE, but I am sure they will not do it for me, I'll try though.
But I am hoping there is a solution for those of us with H.323 links.  I
could possibly get them to switch the circuit to NI2 or QSIG.  Like I
said switching the switch type on the gateway works, without having the
carrier do anything probably because its just compatible, but not sure
if formally having them change the switch emulation would do anything.
The q.931 output is definitely more "friendly" with QSIG/NI, as it
decodes the facility IE.  With 5ESS it doesn't like to receive the name
id facility ie stuff, it errors on it (but of course call completes).

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


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.


From bfeeny at mac.com  Thu Dec 15 21:07:09 2005
From: bfeeny at mac.com (Brian Feeny)
Date: Thu Dec 15 21:07:13 2005
Subject: [cisco-voip] Caller Name ID possible via Facility IE with H.323?
In-Reply-To: <472D0CBC7B31C1499B77C1DAA3308B3B2D76C2@EXCHANGEDS01.ds.ad.adp.com>
References: <472D0CBC7B31C1499B77C1DAA3308B3B2D76C2@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <12544185.1134698829510.JavaMail.bfeeny@mac.com>

Thanks, I have a case open, but I am wanting a clear explanation of the "isdn supp-service name calling", and how it applies, if at all to H.323.

I am going to try to get Hellsouth to switch to Display IE, but I am not holding my breath.

Brian
 
On Thursday, December 15, 2005, at 07:22PM, Bell, Joe <Joe_Bell at adp.com> wrote:

>Brian,
>
>Unless someone knows any pfm, there is no way to accomplish caller-id on
>a PRI with H.323 unless your provider switches the data into Display IE
>(good luck with that).  Recently (using 12.4(4T), I opened a TAC case
>about this issue to see if there was any way to accomplish this or if it
>had been addressed in the new 12.4 train.  TAC sent me a clear answer
>saying H.323 and caller-id are unsupported and in order to get it, I
>needed to go to MGCP.
>
>Joe Bell
>
>-----Original Message-----
>From: cisco-voip-bounces at puck.nether.net
>[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Feeny
>Sent: Thursday, December 15, 2005 11:23 AM
>To: cisco-voip at puck.nether.net
>Subject: [cisco-voip] Caller Name ID possible via Facility IE with
>H.323?
>
>
>I have read the thread:
>
>https://puck.nether.net/pipermail/cisco-voip/2005-March/003509.html
>
>And was curious as to the outcome.  I have PRI's coming in from a 5ESS,
>and the carrier is sending the ID in the facility IE (inbound calls to
>us), and of course CM appears to want it in the display IE.  I have read
>of ways CM would take it via the Facility ID if using MGCP AND NI2/3 or
>MGCP AND QSIG.  Does any solution at all exist for H.323?  I have tried
>to set my switch type for NI and QSIG, both work as far as
>placing/receiving calls, but I still cannot get inbound name id.  I have
>tried to set "isdn supp-service name calling" on the gateway as well but
>it doesn't seem to fix it.  Gateways are running late 12.3T code, CM is
>4.1(3)SR2.
>
>Sure I can try to battle it out with Bellsouth and ask them to change to
>display IE, but I am sure they will not do it for me, I'll try though.
>But I am hoping there is a solution for those of us with H.323 links.  I
>could possibly get them to switch the circuit to NI2 or QSIG.  Like I
>said switching the switch type on the gateway works, without having the
>carrier do anything probably because its just compatible, but not sure
>if formally having them change the switch emulation would do anything.
>The q.931 output is definitely more "friendly" with QSIG/NI, as it
>decodes the facility IE.  With 5ESS it doesn't like to receive the name
>id facility ie stuff, it errors on it (but of course call completes).
>
>Brian
>_______________________________________________
>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>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 it at cimgroup.com  Thu Dec 15 21:14:02 2005
From: it at cimgroup.com (IT)
Date: Thu Dec 15 21:10:04 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0265C5AF@exchange.cimgroup.com>

I am able to factory reset, since the default password on the device is
**#

However, the device just isn't hitting the TFTP server. I am not seeing
anything in the TFTP log, but the screen of the 7935 says "Contacting
10.96.3.11" which is my tftp server.

Any ideas on how I can monitor the traffic on the switch to which this
is connected?

 

Thanks,

Avidan

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Thursday, December 15, 2005 3:40 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Cisco 7935 firmware issues

 

argh. a TAC engineer said I had to RMA and only if I had a contract.
grrrr. I'll have to try a **1 and see if it ask for a password.

	----- 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 

	Sent: Thursday, December 15, 2005 6:09 PM

	Subject: RE: [cisco-voip] Cisco 7935 firmware issues

	 

	That was my problem.... So **1 and see what happens.

	 

	Scott

	 

	
________________________________


	From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
	Sent: Thursday, December 15, 2005 2:53 PM
	To: Voll, Scott; IT; cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Cisco 7935 firmware issues

	 

	will this work if you don't have the password? 

		----- Original Message ----- 

		From: Voll, Scott <mailto:Scott.Voll at wesd.org>  

		To: IT <mailto:it at cimgroup.com>  ;
cisco-voip at puck.nether.net 

		Sent: Thursday, December 15, 2005 10:57 AM

		Subject: RE: [cisco-voip] Cisco 7935 firmware issues

		 

		I had some problems with a 7935 and ended up RMA'ing it.
But after I
		RMA'd it Wes told me how to reset it to Factory
defaults.  Might be
		worth a try.
		
		*********************
		
		Do a **1 from the phone. A dialog will appear asking if
you want to
		delete the phone configuration. Press Yes to clear the
complete setting.
		
		If you press  NO, . it will  put you in the phone
setting  menu screen,
		which also gives  the option in  changing the  passwd.
		
		
		
		**********************
		
		Scott
		
		-----Original Message-----
		From: cisco-voip-bounces at puck.nether.net
		[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
IT
		Sent: Wednesday, December 14, 2005 4:36 PM
		To: cisco-voip at puck.nether.net
		Subject: [cisco-voip] Cisco 7935 firmware issues
		
		Hello,
		
		I have inherited a Cisco 7935 and I am running CM 4.0
		I am having trouble getting the 7935 to pick up new
firmware. I have a
		tftp server running on my CM Publisher, and the device
defaults are set
		correctly for the firmware version. However, the screen
only displays:
		Station IP: 10.96.3.58
		Contacting: 10.96.3.11 (my tftp server)
		
		When I go to http://10.96.3.58 I get:
		Wind Web Server Error Report:
		Server Error: 501 Not Implemented
		No RPM for this combination of URL and method
		/dynaform/index_Dyna.htm
		
		I am able to telnet to 10.96.3.58, but I get a username
& password
		prompt, but I do not know the password, and cisco/cisco
doesn't work.
		
		Does anyone know how else I can force flash this device?
		
		On another note, I had to explicitly set the VLAN on my
switch, as the
		device was picking up the data vlan when connecting,
shouldn't this
		device auto register on the voice vlan?
		
		Thanks,
		Avidan
		
		******************************************************
		This email may contain trade secrets and/or other
information that is 
		proprietary and/or confidential. Use of this by anyone
other than the 
		intended recipient(s) is prohibited and may be unlawful.
If you are not 
		the intended recipient, delete the e-mail and notify us
immediately.
		****************************************************** 
		
		
		_______________________________________________
		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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From Michael_Davis at eLoyalty.com  Thu Dec 15 21:39:54 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Thu Dec 15 21:40:19 2005
Subject: [cisco-voip] RE: change to Qovia  re: QoS Toolsets
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E36C@elhqex01.eloyaltyco.com>

I second that.  In fact recently, after Qovia did their dog and pony for us, we decided to both buy and resell Qovia.  We use it to augment the IPT side of our ICM/IPCC contact center management services.
 
Very good stuff, indeed. BTW, I get no commission or compensation nor benefit for saying so. Though I probably should speak with mgmt about that! ;-)
 
Michael
CCIE #7303 (Voice/R&S)
 
 

	-----Original Message----- 
	From: cisco-voip-bounces at puck.nether.net on behalf of VoIP Forum 
	Sent: Thu 12/15/2005 11:29 AM 
	To: Voll, Scott 
	Cc: cisco-voip at puck.nether.net 
	Subject: [cisco-voip] RE: change to Qovia re: QoS Toolsets
	
	

	This is not a sales pitch, and I get nothing for this?

	 

	Qovia simply blows Cisco?s current ITEM release out of the water, and leaves the awkwardness of Prognosis in the dark. 

	Cost wise, it is far less expensive than NetIQ in most installations and can do so much more.

	 

	I my opinion it is the best Cisco VoIP Real-Time management tool there is. We use it throughout our entire enterprise VoIP Network and have been able to resolve many issues in minutes where we had spent hours in the past troubleshooting. The best thing I could say that you may win Mgmt over with? ?We use it for SLA and QoS mgmt on remote sites. Since the probes can simulate traffic to each other across the WAN. Due to mitigation of lost call or poor call quality, Qovia pays for itself within months instead of years. We have reduced time to roll out new sites as well, since we can take a few days worth of testing and put it into a few hours worth of active tests using the probes in conjunction with Qovia Central.?, and you can quote me on that.

	 

	It is 100% off box so there are not compliance and SMARTnet issues, plus it utilizes its own DB and does not rely on the CDR which is not accessible until after the call has ended. We have it integrated to our NMS it sends e-mail pages to us when there is an issue. That let?s us actually see the MOS of the active call and on each leg of the call while the user is on the call. This is great for real-time circuit maintenance; for example, we can tweak llq (QoS) settings while the call is up to see if that corrects the issue or not. We can identify bad circuits and codec issues, etc? We can look at any services and process utilization on the CCMs real-time or historically. We also use Qovia for capacity planning.

	 

	Another awesome feature is the device inventory reports that will actually show you comparisons to see who is moving their phone and from where to where. You can run hourly, nightly, weekly, monthly reports and they are all useful. We had visiting users grabbing phones off desks and moving them wherever, and we could find them with a simple report as well as what day it moved to track the culprits down.

	 

	 

	 

	
  _____  


	From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
	Sent: Thursday, December 15, 2005 12:07 PM
	To: VoIP Forum
	Cc: cisco-voip at puck.nether.net
	Subject: change to Qovia re: QoS Toolsets

	 

	Are you using it?  I saw it at CIPTUG and am trying to get a quote to show management.  Can you give me your opinion and overall impression?

	 

	Scott

	 

	
  _____  


	From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of VoIP Forum
	Sent: Thursday, December 15, 2005 9:00 AM
	To: Sarff, Dean; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] re: QoS Toolsets

	 

	Qovia is the way to go!

	www.qovia.com <http://www.qovia.com/> 

	 

	 

	
  _____  


	From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sarff, Dean
	Sent: Thursday, December 15, 2005 11:56 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] re: QoS Toolsets

	 

	I?d like feedback from those users of either NET IQ?s VoIP tools or Integrated Research?s PROGNOSIS.  How do you like your product selection?  

	Dean Sarff, MMS A&B IMD, 703-787-1272

	Merry Christmas!  Happy Birthday Jesus!

	May you an your family have a Blessed Holiday.



From mfa at crec.ifas.ufl.edu  Thu Dec 15 22:01:32 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Thu Dec 15 22:01:40 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
References: <200512160210.jBG2A4g4083653@puck.nether.net>
Message-ID: <005101c601ed$05cee580$0402a8c0@crec.ifas.ufl.edu>

Depends on the switch, but try

switch# term mon
switch# debug interface FAx/y

or set up a SPAN session to a monitoring port

mike

> Date: Thu, 15 Dec 2005 18:14:02 -0800
> From: "IT" <it at cimgroup.com>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
> To: "Lelio Fulgenzi" <lelio at uoguelph.ca>, "Voll, Scott"
> <Scott.Voll at wesd.org>, <cisco-voip at puck.nether.net>
> Message-ID:
> <B487D4C1FE6D04439061018C6EBC109B0265C5AF at exchange.cimgroup.com>
> Content-Type: text/plain; charset="us-ascii"
>
> I am able to factory reset, since the default password on the device is
> **#
>
> However, the device just isn't hitting the TFTP server. I am not seeing
> anything in the TFTP log, but the screen of the 7935 says "Contacting
> 10.96.3.11" which is my tftp server.
>
> Any ideas on how I can monitor the traffic on the switch to which this
> is connected?
>
>
>
> Thanks,
>
> Avidan
>
>
>
>
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
> Sent: Thursday, December 15, 2005 3:40 PM
> To: Voll, Scott; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> argh. a TAC engineer said I had to RMA and only if I had a contract.
> grrrr. I'll have to try a **1 and see if it ask for a password.
>
> ----- 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
>
> Sent: Thursday, December 15, 2005 6:09 PM
>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> That was my problem.... So **1 and see what happens.
>
>
>
> Scott
>
>
>
>
> ________________________________
>
>
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Thursday, December 15, 2005 2:53 PM
> To: Voll, Scott; IT; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> will this work if you don't have the password?
>
> ----- Original Message ----- 
>
> From: Voll, Scott <mailto:Scott.Voll at wesd.org>
>
> To: IT <mailto:it at cimgroup.com>  ;
> cisco-voip at puck.nether.net
>
> Sent: Thursday, December 15, 2005 10:57 AM
>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> I had some problems with a 7935 and ended up RMA'ing it.
> But after I
> RMA'd it Wes told me how to reset it to Factory
> defaults.  Might be
> worth a try.
>
> *********************
>
> Do a **1 from the phone. A dialog will appear asking if
> you want to
> delete the phone configuration. Press Yes to clear the
> complete setting.
>
> If you press  NO, . it will  put you in the phone
> setting  menu screen,
> which also gives  the option in  changing the  passwd.
>
>
>
> **********************
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> IT
> Sent: Wednesday, December 14, 2005 4:36 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Cisco 7935 firmware issues
>
> Hello,
>
> I have inherited a Cisco 7935 and I am running CM 4.0
> I am having trouble getting the 7935 to pick up new
> firmware. I have a
> tftp server running on my CM Publisher, and the device
> defaults are set
> correctly for the firmware version. However, the screen
> only displays:
> Station IP: 10.96.3.58
> Contacting: 10.96.3.11 (my tftp server)
>
> When I go to http://10.96.3.58 I get:
> Wind Web Server Error Report:
> Server Error: 501 Not Implemented
> No RPM for this combination of URL and method
> /dynaform/index_Dyna.htm
>
> I am able to telnet to 10.96.3.58, but I get a username
> & password
> prompt, but I do not know the password, and cisco/cisco
> doesn't work.
>
> Does anyone know how else I can force flash this device?
>
> On another note, I had to explicitly set the VLAN on my
> switch, as the
> device was picking up the data vlan when connecting,
> shouldn't this
> device auto register on the voice vlan?
>
> Thanks,
> Avidan
>
> ******************************************************
> This email may contain trade secrets and/or other
> information that is
> proprietary and/or confidential. Use of this by anyone
> other than the
> intended recipient(s) is prohibited and may be unlawful.
> If you are not
> the intended recipient, delete the e-mail and notify us
> immediately.
> ******************************************************
>
>
> _______________________________________________
> 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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> URL: 
> https://puck.nether.net/pipermail/cisco-voip/attachments/20051215/6eef9b48/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 34, Issue 81
> ****************************************** 


From Michael_Davis at eLoyalty.com  Thu Dec 15 22:04:13 2005
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Thu Dec 15 22:04:37 2005
Subject: [cisco-voip] re: QoS Toolsets
Message-ID: <037594FB5AFC354FA38653EB3FF3242105A8E36D@elhqex01.eloyaltyco.com>

Hi Candace,
 
I think the two are sort of apples to oranges. If you want to manage the health of a CCM server, NetIQ with its software agent, deep Microsoft perfmon hooks etc, has a strong play.  But it should be compared against NetCool or HPOV, Unicenter, or the like. But that's not really Qovia's strength.
 
Seems to me the problem with NetIQ, at least when I last looked at it, is that like so many frameworks, it relies on stats reported from a server (CCM for example) rather than an actual packet capture.  I love that when a quality threshold is broken, or if I decide I want to, I can play a .wav from end to end and hear what the callers hear. And I can get R-factor in both directions, which I cannot with with NetIQ and others. I can also verify that my DSCP is not re-marked on the far end of an MPLS cloud.
 
Architecturally, the two look somewhat similar - centralized server(s) with remote agents (Qovia - hardware probe, NetIQ - software agent).  
 
Please correct me if any this is no longer valid.
 
Qovia in some respects, is more like an adjunct for a framework. It has hooks for Netcool, or HPOV, or directly to your DBMS to augment the framework rather than replace it. I can also use the probe to collect traps and/or mibwalk, or remote access over a secured connection.  No small issue for a multi-tenant / multi-customer managed service provider.
 
I'm not sure what you mean by MOS granularity being sub par. Given that Qovia can see both sides of both RTP streams, record them play them back, measure them, send a wav between them, etc, and NetIQ can't. Please elaborate.
 
I think your point about server managment is well taken. NetIQ is certainly better for that. But while that information is necessary for us to manage an environment,  our SLAs are concerned directly with voice quality and uptime.  It makes sense to have both server and voice sides of the network covered.
 
Michael
 
 
 
 
 
 
 

	-----Original Message----- 
	From: cisco-voip-bounces at puck.nether.net on behalf of Candace Holman 
	Sent: Thu 12/15/2005 12:07 PM 
	To: cisco-voip at puck.nether.net 
	Cc: 
	Subject: [cisco-voip] re: QoS Toolsets
	
	

	What version of Qovia are you referring to? I don't have some of these
	features in my installation.
	
	I've tested all three of the products under discussion, and best
	all-around alerting tool so far is NetIQ. Qovia needs a little work to
	provide a way to turn off alerts when the listener decides, not just
	when Qovia sees the condition go away. I get spammed by my own tool
	because it is not adjustable enough (at least not in the version I use).
	We should be able to acknowledge a problem to stop the alerts,
	consolidate alerts, and choose the delivery media. Qovia can only send
	emails or snmp traps. Also the granularity of the MoS metrics is a
	little under par. Qovia doesn't do more than NetIQ, except in one area -
	providing quick, easy and multi-tenant/multi-privilege real-time access
	to phones and call data. This is a hardware solution with a centralized
	server, so you have to cart it out and set it up, and have available a
	span port for each probe.
	
	IMO, NetIQ does more than Qovia, and monitors all kinds of information
	on the CallManagers and gateways, not just the phones. Qovia only seems
	to monitor phones and quite limited data on the gateways and switches.
	NetIQ can send alerts to a variety of media: pages, emails, snmp traps,
	or just log the data.
	
	Candace
	
	Subject:
	[cisco-voip] RE: change to Qovia re: QoS Toolsets
	From:
	"VoIP Forum" <voipforum at apptis.com>
	Date:
	Thu, 15 Dec 2005 12:29:29 -0500
	
	To:
	"Voll, Scott" <Scott.Voll at wesd.org>
	CC:
	cisco-voip at puck.nether.net
	
	
	*/This is not a sales pitch, and I get nothing for this?/*
	
	Qovia simply blows Cisco?s current ITEM release out of the water, and
	leaves the awkwardness of Prognosis in the dark.
	
	Cost wise, it is far less expensive than NetIQ in most installations and
	can do so much more.
	
	I my opinion it is the best Cisco VoIP Real-Time management tool there
	is. We use it throughout our entire enterprise VoIP Network and have
	been able to resolve many issues in minutes where we had spent hours in
	the past troubleshooting. The best thing I could say that you may win
	Mgmt over with? ?We use it for SLA and QoS mgmt on remote sites. Since
	the probes can simulate traffic to each other across the WAN. Due to
	mitigation of lost call or poor call quality, Qovia pays for itself
	within months instead of years. We have reduced time to roll out new
	sites as well, since we can take a few days worth of testing and put it
	into a few hours worth of active tests using the probes in conjunction
	with Qovia Central.?, and you can quote me on that.
	
	It is 100% off box so there are not compliance and SMARTnet issues, plus
	it utilizes its own DB and does not rely on the CDR which is not
	accessible until after the call has ended. We have it integrated to our
	NMS it sends e-mail pages to us when there is an issue. That let?s us
	actually see the MOS of the active call and on each leg of the call
	while the user is on the call. This is great for real-time circuit
	maintenance; for example, we can tweak llq (QoS) settings while the call
	is up to see if that corrects the issue or not. We can identify bad
	circuits and codec issues, etc? We can look at any services and process
	utilization on the CCMs real-time or historically. We also use Qovia for
	capacity planning.
	
	Another awesome feature is the device inventory reports that will
	actually show you comparisons to see who is moving their phone and from
	where to where. You can run hourly, nightly, weekly, monthly reports and
	they are all useful. We had visiting users grabbing phones off desks and
	moving them wherever, and we could find them with a simple report as
	well as what day it moved to track the culprits down.
	
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	



From erickbe at yahoo.com  Thu Dec 15 22:51:04 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Dec 15 22:51:12 2005
Subject: [cisco-voip] Caller Name ID possible via Facility IE with H.323?
In-Reply-To: <12544185.1134698829510.JavaMail.bfeeny@mac.com>
Message-ID: <20051216035104.21619.qmail@web34309.mail.mud.yahoo.com>

I think I recall some discussion on this list or in
the cisco forums about this being changed in a future
release of CME (and thus IOS hopefully). I'll have to
do some digging to see if I can find that again. 

--- Brian Feeny <bfeeny at mac.com> wrote:

> Thanks, I have a case open, but I am wanting a clear
> explanation of the "isdn supp-service name calling",
> and how it applies, if at all to H.323.
> 
> I am going to try to get Hellsouth to switch to
> Display IE, but I am not holding my breath.
> 
> Brian
>  
> On Thursday, December 15, 2005, at 07:22PM, Bell,
> Joe <Joe_Bell at adp.com> wrote:
> 
> >Brian,
> >
> >Unless someone knows any pfm, there is no way to
> accomplish caller-id on
> >a PRI with H.323 unless your provider switches the
> data into Display IE
> >(good luck with that).  Recently (using 12.4(4T), I
> opened a TAC case
> >about this issue to see if there was any way to
> accomplish this or if it
> >had been addressed in the new 12.4 train.  TAC sent
> me a clear answer
> >saying H.323 and caller-id are unsupported and in
> order to get it, I
> >needed to go to MGCP.
> >
> >Joe Bell
> >
> >-----Original Message-----
> >From: cisco-voip-bounces at puck.nether.net
> >[mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Brian Feeny
> >Sent: Thursday, December 15, 2005 11:23 AM
> >To: cisco-voip at puck.nether.net
> >Subject: [cisco-voip] Caller Name ID possible via
> Facility IE with
> >H.323?
> >
> >
> >I have read the thread:
> >
>
>https://puck.nether.net/pipermail/cisco-voip/2005-March/003509.html
> >
> >And was curious as to the outcome.  I have PRI's
> coming in from a 5ESS,
> >and the carrier is sending the ID in the facility
> IE (inbound calls to
> >us), and of course CM appears to want it in the
> display IE.  I have read
> >of ways CM would take it via the Facility ID if
> using MGCP AND NI2/3 or
> >MGCP AND QSIG.  Does any solution at all exist for
> H.323?  I have tried
> >to set my switch type for NI and QSIG, both work as
> far as
> >placing/receiving calls, but I still cannot get
> inbound name id.  I have
> >tried to set "isdn supp-service name calling" on
> the gateway as well but
> >it doesn't seem to fix it.  Gateways are running
> late 12.3T code, CM is
> >4.1(3)SR2.
> >
> >Sure I can try to battle it out with Bellsouth and
> ask them to change to
> >display IE, but I am sure they will not do it for
> me, I'll try though.
> >But I am hoping there is a solution for those of us
> with H.323 links.  I
> >could possibly get them to switch the circuit to
> NI2 or QSIG.  Like I
> >said switching the switch type on the gateway
> works, without having the
> >carrier do anything probably because its just
> compatible, but not sure
> >if formally having them change the switch emulation
> would do anything.
> >The q.931 output is definitely more "friendly" with
> QSIG/NI, as it
> >decodes the facility IE.  With 5ESS it doesn't like
> to receive the name
> >id facility ie stuff, it errors on it (but of
> course call completes).
> >
> >Brian
> >_______________________________________________
> >cisco-voip mailing list
> >cisco-voip at puck.nether.net
> >https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >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.
> >
> >_______________________________________________
> >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 erickbe at yahoo.com  Thu Dec 15 23:55:07 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Dec 15 23:55:15 2005
Subject: [cisco-voip] Caller Name ID possible via Facility IE with H.323?
In-Reply-To: <20051216035104.21619.qmail@web34309.mail.mud.yahoo.com>
Message-ID: <20051216045507.28070.qmail@web34311.mail.mud.yahoo.com>

Here's what I found from previous discussion on this
topic on this list and at forums.

http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=IP%20Communications%20and%20Video&topic=IP%20Telephony&CommCmd=MB%3Fcmd%3Ddisplay_location%26location%3D.1dd93785/2

The cisco forum has someone from cisco indicating
support for facility IE will be there next year, but
no way to confirm or if it'll be a CME-only feature.
If it were in IOS, then it would need to be passed to
CCM in messages CCM could understand also. 

The following is a snippet from a previous post here I
had saved, not sure who it was from but was regarding
the isdn supp-service command. Maybe is only effective
with SIP? I have not had need to use this command
myself yet. (Sorry for formatting below).

> [...]
> > 1) Caller ID name data comes in on the PRI, but
> doesn't appear to get
> >    handed off to the Asterisk server via SIP, at
> least not in any
> >    format that Asterisk understands.  Caller ID
> _number_ works fine.
> 
> So, for everybody following along at home, it
> appears as though one
> must configure "isdn supp-service calling name"
> under the sX/Y:23
> interface config in order for the Access Server to
> recognize caller ID
> name in the ISDN facility messages.  This feature is
> available in
> 12.3(14)T and 12.3(11)T (and likely earlier builds),
> but not in
> 12.3(13) mainline.
> 
> Problem is, when running T train code, the Cisco has
> developed a habit
> of continuing to ring a SIP channel, even when a
> PSTN caller has
> disconnected.  This appears to be a result of
> something having changed
> in how the box sends out SIP BYE/CANCEL messages,
> though I'm a bit
> confused as to the particulars.  



--- Erick Bergquist <erickbe at yahoo.com> wrote:

> I think I recall some discussion on this list or in
> the cisco forums about this being changed in a
> future
> release of CME (and thus IOS hopefully). I'll have
> to
> do some digging to see if I can find that again. 
> 
> --- Brian Feeny <bfeeny at mac.com> wrote:
> 
> > Thanks, I have a case open, but I am wanting a
> clear
> > explanation of the "isdn supp-service name
> calling",
> > and how it applies, if at all to H.323.
> > 
> > I am going to try to get Hellsouth to switch to
> > Display IE, but I am not holding my breath.
> > 
> > Brian
> >  
> > On Thursday, December 15, 2005, at 07:22PM, Bell,
> > Joe <Joe_Bell at adp.com> wrote:
> > 
> > >Brian,
> > >
> > >Unless someone knows any pfm, there is no way to
> > accomplish caller-id on
> > >a PRI with H.323 unless your provider switches
> the
> > data into Display IE
> > >(good luck with that).  Recently (using 12.4(4T),
> I
> > opened a TAC case
> > >about this issue to see if there was any way to
> > accomplish this or if it
> > >had been addressed in the new 12.4 train.  TAC
> sent
> > me a clear answer
> > >saying H.323 and caller-id are unsupported and in
> > order to get it, I
> > >needed to go to MGCP.
> > >
> > >Joe Bell
> > >
> > >-----Original Message-----
> > >From: cisco-voip-bounces at puck.nether.net
> > >[mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Brian Feeny
> > >Sent: Thursday, December 15, 2005 11:23 AM
> > >To: cisco-voip at puck.nether.net
> > >Subject: [cisco-voip] Caller Name ID possible via
> > Facility IE with
> > >H.323?
> > >
> > >
> > >I have read the thread:
> > >
> >
>
>https://puck.nether.net/pipermail/cisco-voip/2005-March/003509.html
> > >
> > >And was curious as to the outcome.  I have PRI's
> > coming in from a 5ESS,
> > >and the carrier is sending the ID in the facility
> > IE (inbound calls to
> > >us), and of course CM appears to want it in the
> > display IE.  I have read
> > >of ways CM would take it via the Facility ID if
> > using MGCP AND NI2/3 or
> > >MGCP AND QSIG.  Does any solution at all exist
> for
> > H.323?  I have tried
> > >to set my switch type for NI and QSIG, both work
> as
> > far as
> > >placing/receiving calls, but I still cannot get
> > inbound name id.  I have
> > >tried to set "isdn supp-service name calling" on
> > the gateway as well but
> > >it doesn't seem to fix it.  Gateways are running
> > late 12.3T code, CM is
> > >4.1(3)SR2.
> > >
> > >Sure I can try to battle it out with Bellsouth
> and
> > ask them to change to
> > >display IE, but I am sure they will not do it for
> > me, I'll try though.
> > >But I am hoping there is a solution for those of
> us
> > with H.323 links.  I
> > >could possibly get them to switch the circuit to
> > NI2 or QSIG.  Like I
> > >said switching the switch type on the gateway
> > works, without having the
> > >carrier do anything probably because its just
> > compatible, but not sure
> > >if formally having them change the switch
> emulation
> > would do anything.
> > >The q.931 output is definitely more "friendly"
> with
> > QSIG/NI, as it
> > >decodes the facility IE.  With 5ESS it doesn't
> like
> > to receive the name
> > >id facility ie stuff, it errors on it (but of
> > course call completes).
> > >
> > >Brian
> > >_______________________________________________
> > >cisco-voip mailing list
> > >cisco-voip at puck.nether.net
> >
> >https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >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.


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

From rmontejano at earthlink.net  Fri Dec 16 00:06:12 2005
From: rmontejano at earthlink.net (Ross Montejano)
Date: Fri Dec 16 00:06:15 2005
Subject: [cisco-voip] h323 set-up for Fax and POS
Message-ID: <3471087.1134709573337.JavaMail.root@elwamui-mouette.atl.sa.earthlink.net>

Hi,

Have a customer that wishes to move their Fax and POS off of analog CO lines onto PRI. PRI terminates into H323 gateway. Fax and POS are connected to FXS ports on gateway.

Am looking for configuration examples to make this work. Any links or examples would be greatly appreciated.

Thank You,

Ross

From trey.howland at gtri.gatech.edu  Fri Dec 16 00:34:46 2005
From: trey.howland at gtri.gatech.edu (Trey Howland)
Date: Fri Dec 16 00:36:36 2005
Subject: [cisco-voip] Caller Name ID possible via Facility IE with H.323?
In-Reply-To: <20051216045507.28070.qmail@web34311.mail.mud.yahoo.com>
References: <20051216045507.28070.qmail@web34311.mail.mud.yahoo.com>
Message-ID: <160ff111ec8f5c6834cf7a7a118aa273@gtri.gatech.edu>

CSCdu81533 is in a closed state and no mention of any work to add the  
feature.  TAC's only suggestion was to move to MGCP.

CSCdu81533 - Caller name display in q931 facility msg is not processed  
and passed Release-note: Name display received in Q931 facility msgs is  
not processed and passed onto h225 by IOS gateways. There is no  
workaround.




On Dec 15, 2005, at 11:55 PM, Erick Bergquist wrote:

> Here's what I found from previous discussion on this
> topic on this list and at forums.
>
> http://forum.cisco.com/eforum/servlet/NetProf? 
> page=netprof&forum=IP%20Communications%20and%20Video&topic=IP%20Telepho 
> ny&CommCmd=MB%3Fcmd%3Ddisplay_location%26location%3D.1dd93785/2
>
> The cisco forum has someone from cisco indicating
> support for facility IE will be there next year, but
> no way to confirm or if it'll be a CME-only feature.
> If it were in IOS, then it would need to be passed to
> CCM in messages CCM could understand also.
>
> The following is a snippet from a previous post here I
> had saved, not sure who it was from but was regarding
> the isdn supp-service command. Maybe is only effective
> with SIP? I have not had need to use this command
> myself yet. (Sorry for formatting below).
>
>
>
Georgia Institute of Technology     Atlanta, Georgia
Georgia Tech Research Institute
Electronic Systems Laboratory

Public Key ID: 0x894F4419
[7338 D848 8BEE FF6D 23D1  4355 2F6C DE0C 894F 4419]


From Scott.Voll at wesd.org  Fri Dec 16 10:14:42 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Dec 16 10:14:27 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
Message-ID: <D713462ED535184D830F6F6301753E7E01A12098@VISHNU.wesd.org>

Can you ping the 7935? 

Is there any ACLs or FWs between it and the DNS, TFTP, CM, etc.?

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Thursday, December 15, 2005 7:02 PM
To: cisco-voip at puck.nether.net; it at cimgroup.com
Subject: Re:[cisco-voip] Cisco 7935 firmware issues

Depends on the switch, but try

switch# term mon
switch# debug interface FAx/y

or set up a SPAN session to a monitoring port

mike

> Date: Thu, 15 Dec 2005 18:14:02 -0800
> From: "IT" <it at cimgroup.com>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
> To: "Lelio Fulgenzi" <lelio at uoguelph.ca>, "Voll, Scott"
> <Scott.Voll at wesd.org>, <cisco-voip at puck.nether.net>
> Message-ID:
> <B487D4C1FE6D04439061018C6EBC109B0265C5AF at exchange.cimgroup.com>
> Content-Type: text/plain; charset="us-ascii"
>
> I am able to factory reset, since the default password on the device
is
> **#
>
> However, the device just isn't hitting the TFTP server. I am not
seeing
> anything in the TFTP log, but the screen of the 7935 says "Contacting
> 10.96.3.11" which is my tftp server.
>
> Any ideas on how I can monitor the traffic on the switch to which this
> is connected?
>
>
>
> Thanks,
>
> Avidan
>
>
>
>
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio
Fulgenzi
> Sent: Thursday, December 15, 2005 3:40 PM
> To: Voll, Scott; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> argh. a TAC engineer said I had to RMA and only if I had a contract.
> grrrr. I'll have to try a **1 and see if it ask for a password.
>
> ----- 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
>
> Sent: Thursday, December 15, 2005 6:09 PM
>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> That was my problem.... So **1 and see what happens.
>
>
>
> Scott
>
>
>
>
> ________________________________
>
>
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Thursday, December 15, 2005 2:53 PM
> To: Voll, Scott; IT; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> will this work if you don't have the password?
>
> ----- Original Message ----- 
>
> From: Voll, Scott <mailto:Scott.Voll at wesd.org>
>
> To: IT <mailto:it at cimgroup.com>  ;
> cisco-voip at puck.nether.net
>
> Sent: Thursday, December 15, 2005 10:57 AM
>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> I had some problems with a 7935 and ended up RMA'ing it.
> But after I
> RMA'd it Wes told me how to reset it to Factory
> defaults.  Might be
> worth a try.
>
> *********************
>
> Do a **1 from the phone. A dialog will appear asking if
> you want to
> delete the phone configuration. Press Yes to clear the
> complete setting.
>
> If you press  NO, . it will  put you in the phone
> setting  menu screen,
> which also gives  the option in  changing the  passwd.
>
>
>
> **********************
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> IT
> Sent: Wednesday, December 14, 2005 4:36 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Cisco 7935 firmware issues
>
> Hello,
>
> I have inherited a Cisco 7935 and I am running CM 4.0
> I am having trouble getting the 7935 to pick up new
> firmware. I have a
> tftp server running on my CM Publisher, and the device
> defaults are set
> correctly for the firmware version. However, the screen
> only displays:
> Station IP: 10.96.3.58
> Contacting: 10.96.3.11 (my tftp server)
>
> When I go to http://10.96.3.58 I get:
> Wind Web Server Error Report:
> Server Error: 501 Not Implemented
> No RPM for this combination of URL and method
> /dynaform/index_Dyna.htm
>
> I am able to telnet to 10.96.3.58, but I get a username
> & password
> prompt, but I do not know the password, and cisco/cisco
> doesn't work.
>
> Does anyone know how else I can force flash this device?
>
> On another note, I had to explicitly set the VLAN on my
> switch, as the
> device was picking up the data vlan when connecting,
> shouldn't this
> device auto register on the voice vlan?
>
> Thanks,
> Avidan
>
> ******************************************************
> This email may contain trade secrets and/or other
> information that is
> proprietary and/or confidential. Use of this by anyone
> other than the
> intended recipient(s) is prohibited and may be unlawful.
> If you are not
> the intended recipient, delete the e-mail and notify us
> immediately.
> ******************************************************
>
>
> _______________________________________________
> 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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> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> End of cisco-voip Digest, Vol 34, Issue 81
> ****************************************** 

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From zephy316 at gmail.com  Fri Dec 16 10:25:13 2005
From: zephy316 at gmail.com (Israel Lang)
Date: Fri Dec 16 10:25:17 2005
Subject: [cisco-voip] CME Question
Message-ID: <91e236d30512160725g22e76fffr28a0055effcdb18c@mail.gmail.com>

I have a client who has outgrown their key trunk system. We have installed a
CME system for them. They currently have 6 POTS lines and are going to
expand that to 9 or 10 come the first of the year.

The question I have is what is the best way to set-up call coverage for
those lines in CME?

Currently I have created a 4 line hunt group:

ephone-hunt 1 sequential
 pilot 5001
 list 1100, 1101, 1102, 1103
 final 6001
 timeout 8
 statistics collect

Call are covered by three receptionists and they all have the four
extensions listed on their phones. 6001 goes to the AA should no one answer.
Normally on 2-3 inbound calls every happen at once so I am fairly confident
the 4 lines are enough to do what we want.

Anybody have a better solution or a different way that they do it that may
be worth looking at?

Additionally, the client is used to having the phone ring only 4 times after
hours and then going directly to the AA. Anyway given the way I have it
currently setup that I could accomplish the same thing in CME?

Thanks,
--
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol
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From bcarter at sentinel.com  Fri Dec 16 11:47:02 2005
From: bcarter at sentinel.com (Carter, Bill)
Date: Fri Dec 16 11:47:10 2005
Subject: [cisco-voip] CME support for 7941G/7961G/7961G-GE
Message-ID: <6A0744639B86DD459433108DBF101D26D206A9@MAIL01.sentinel.com>

When will these IP Phones be supported in CallManager Express? 



From tennille.spence at gsa.gov  Fri Dec 16 12:24:39 2005
From: tennille.spence at gsa.gov (tennille.spence@gsa.gov)
Date: Fri Dec 16 12:24:45 2005
Subject: [cisco-voip] Unity - Messages left are not being recorded
Message-ID: <OF74DE5059.9375E578-ON862570D9.005F4724-862570D9.005FA51F@gsa.gov>


I have a user who has been reporting that she occasionally does not receive
her voicemails. This has been happening intermittently over the past year.
This morning she logged into voicemail and it said she had one new message.
When she pressed 1, it played nothing and immediately to repeat this
message press 1, to save it press 2..etc. She continuously pressed 1 and
nothing happened. In the past, I thought it was just people calling and
listening to the greeting and then not leaving a message. In this case, the
guy called back later and was upset that she had not returned his call. He
had left a long message. Has anyone had this happen. Any ideas on how to go
about pinpointing this issue or what would cause this to happen? We are
running 3.15 unity and CCM 4.0(2) sr 1a. The call happened around 8:30 am
this morning from an external phone number. I sis look at the subscriber
message activity and it showed unknown caller called at 8:29 which
triggered the MWI light to come on but the message was actually blank and
lasted about 1 second.

Tennille Spence
IT Business Liaison
PBS Technology Branch
816-823-1357 Office
816-806-6342 Cell
816-823-5526 Fax

"What lies behind us and what lies before us are tiny matters compared to
what lies within us." - Ralph Waldo Emerson

"Opportunity is missed by most people because it is dressed in overalls and
looks like work." - Thomas A. Edison



From mike.c at altatechnologies.com  Fri Dec 16 13:21:34 2005
From: mike.c at altatechnologies.com (Mike Carter)
Date: Fri Dec 16 13:24:29 2005
Subject: [cisco-voip] phone password issue 
Message-ID: <BFEHKOIOENIIPHGJOCHDAELJFBAA.mike.c@altatechnologies.com>

HELP!!!!!!!

not sure how but somehow one of our phones got locked----none of the default
passwords work

ideas????

Michael Carter
Account Manager
Alta Technologies Inc
763-475-5327
800-546-2582 (327)
AIM mikec at alta
yahoo IM cartematalta
mike.c at altatechnologies.com
Sun IBM HP Cisco Nortel Juniper Adtran
D-Link Emulex Qlogic Overland Storage
We buy and sell all of the above...please email your offers or requests.



From lelio at uoguelph.ca  Fri Dec 16 14:02:03 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Dec 16 14:01:56 2005
Subject: [cisco-voip] getting translation patterns in CDRs
Message-ID: <002701c60273$339efd70$ca01a8c0@deadfish>

I'd like to get translations in my CDRs somehow. Whether they are two entries or one entry or whatever. Can anyone tell me if the Digit Analysis Complexity service parameter affects CDRs at all? And whether it puts too much of a strain on CallManager?

Thanks, Lelio


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

      Digit Analysis Complexity : This parameter allows the user to toggle between two modes; that is, StandardAnalysis and TranslationAndAlternatePatternAnalysis. The TranslationAndAlternatePatternAnalysis mode gives information about the Translation pattern and the Alternate matches in the CCM Traces while finding a match for a pattern. The valid values for this field follow: 
      -- StandardAnalysis 
      -- TranslationAndAlternatePatternAnalysis 

      This is a required field.
      Default: StandardAnalysis.
     


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


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
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From it at cimgroup.com  Fri Dec 16 14:33:09 2005
From: it at cimgroup.com (IT)
Date: Fri Dec 16 14:29:10 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0265C701@exchange.cimgroup.com>

Nope, nothing is being filtered with FW or ACL's. In fact, I plugged in
another 7935 and it loaded the new firmware no problem. I ran a port
scan on the device, and telnet and www are open.
How else can I force this device to upgrade?
-Avidan

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Friday, December 16, 2005 7:15 AM
To: Mike Armstrong; cisco-voip at puck.nether.net; IT
Subject: RE: [cisco-voip] Cisco 7935 firmware issues

Can you ping the 7935? 

Is there any ACLs or FWs between it and the DNS, TFTP, CM, etc.?

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Armstrong
Sent: Thursday, December 15, 2005 7:02 PM
To: cisco-voip at puck.nether.net; it at cimgroup.com
Subject: Re:[cisco-voip] Cisco 7935 firmware issues

Depends on the switch, but try

switch# term mon
switch# debug interface FAx/y

or set up a SPAN session to a monitoring port

mike

> Date: Thu, 15 Dec 2005 18:14:02 -0800
> From: "IT" <it at cimgroup.com>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
> To: "Lelio Fulgenzi" <lelio at uoguelph.ca>, "Voll, Scott"
> <Scott.Voll at wesd.org>, <cisco-voip at puck.nether.net>
> Message-ID:
> <B487D4C1FE6D04439061018C6EBC109B0265C5AF at exchange.cimgroup.com>
> Content-Type: text/plain; charset="us-ascii"
>
> I am able to factory reset, since the default password on the device
is
> **#
>
> However, the device just isn't hitting the TFTP server. I am not
seeing
> anything in the TFTP log, but the screen of the 7935 says "Contacting
> 10.96.3.11" which is my tftp server.
>
> Any ideas on how I can monitor the traffic on the switch to which this
> is connected?
>
>
>
> Thanks,
>
> Avidan
>
>
>
>
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio
Fulgenzi
> Sent: Thursday, December 15, 2005 3:40 PM
> To: Voll, Scott; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> argh. a TAC engineer said I had to RMA and only if I had a contract.
> grrrr. I'll have to try a **1 and see if it ask for a password.
>
> ----- 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
>
> Sent: Thursday, December 15, 2005 6:09 PM
>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> That was my problem.... So **1 and see what happens.
>
>
>
> Scott
>
>
>
>
> ________________________________
>
>
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Thursday, December 15, 2005 2:53 PM
> To: Voll, Scott; IT; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> will this work if you don't have the password?
>
> ----- Original Message ----- 
>
> From: Voll, Scott <mailto:Scott.Voll at wesd.org>
>
> To: IT <mailto:it at cimgroup.com>  ;
> cisco-voip at puck.nether.net
>
> Sent: Thursday, December 15, 2005 10:57 AM
>
> Subject: RE: [cisco-voip] Cisco 7935 firmware issues
>
>
>
> I had some problems with a 7935 and ended up RMA'ing it.
> But after I
> RMA'd it Wes told me how to reset it to Factory
> defaults.  Might be
> worth a try.
>
> *********************
>
> Do a **1 from the phone. A dialog will appear asking if
> you want to
> delete the phone configuration. Press Yes to clear the
> complete setting.
>
> If you press  NO, . it will  put you in the phone
> setting  menu screen,
> which also gives  the option in  changing the  passwd.
>
>
>
> **********************
>
> Scott
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> IT
> Sent: Wednesday, December 14, 2005 4:36 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Cisco 7935 firmware issues
>
> Hello,
>
> I have inherited a Cisco 7935 and I am running CM 4.0
> I am having trouble getting the 7935 to pick up new
> firmware. I have a
> tftp server running on my CM Publisher, and the device
> defaults are set
> correctly for the firmware version. However, the screen
> only displays:
> Station IP: 10.96.3.58
> Contacting: 10.96.3.11 (my tftp server)
>
> When I go to http://10.96.3.58 I get:
> Wind Web Server Error Report:
> Server Error: 501 Not Implemented
> No RPM for this combination of URL and method
> /dynaform/index_Dyna.htm
>
> I am able to telnet to 10.96.3.58, but I get a username
> & password
> prompt, but I do not know the password, and cisco/cisco
> doesn't work.
>
> Does anyone know how else I can force flash this device?
>
> On another note, I had to explicitly set the VLAN on my
> switch, as the
> device was picking up the data vlan when connecting,
> shouldn't this
> device auto register on the voice vlan?
>
> Thanks,
> Avidan
>
> ******************************************************
> This email may contain trade secrets and/or other
> information that is
> proprietary and/or confidential. Use of this by anyone
> other than the
> intended recipient(s) is prohibited and may be unlawful.
> If you are not
> the intended recipient, delete the e-mail and notify us
> immediately.
> ******************************************************
>
>
> _______________________________________________
> 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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> https://puck.nether.net/mailman/listinfo/cisco-voip
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>
> End of cisco-voip Digest, Vol 34, Issue 81
> ****************************************** 

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From rratliff at cisco.com  Fri Dec 16 14:48:15 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Dec 16 14:48:24 2005
Subject: [cisco-voip] Caller Name ID possible via Facility IE with H.323?
In-Reply-To: <160ff111ec8f5c6834cf7a7a118aa273@gtri.gatech.edu>
References: <20051216045507.28070.qmail@web34311.mail.mud.yahoo.com>
	<160ff111ec8f5c6834cf7a7a118aa273@gtri.gatech.edu>
Message-ID: <6C51423C-315B-42BF-A0C4-48AA31A1B794@cisco.com>

Unfortunately this is correct.

The bug mentioned below references CSCdt03385 which is where the  
support for this was built into MGCP (as of CM 3.3.3).  Currently  
there is no support for H.323 with CallManager.   I can't speak to CME.

-Ryan

On Dec 16, 2005, at 12:34 AM, Trey Howland wrote:

CSCdu81533 is in a closed state and no mention of any work to add the
feature.  TAC's only suggestion was to move to MGCP.

CSCdu81533 - Caller name display in q931 facility msg is not processed
and passed Release-note: Name display received in Q931 facility msgs is
not processed and passed onto h225 by IOS gateways. There is no
workaround.




On Dec 15, 2005, at 11:55 PM, Erick Bergquist wrote:

> Here's what I found from previous discussion on this
> topic on this list and at forums.
>
> http://forum.cisco.com/eforum/servlet/NetProf?
> page=netprof&forum=IP%20Communications%20and%20Video&topic=IP% 
> 20Telepho
> ny&CommCmd=MB%3Fcmd%3Ddisplay_location%26location%3D.1dd93785/2
>
> The cisco forum has someone from cisco indicating
> support for facility IE will be there next year, but
> no way to confirm or if it'll be a CME-only feature.
> If it were in IOS, then it would need to be passed to
> CCM in messages CCM could understand also.
>
> The following is a snippet from a previous post here I
> had saved, not sure who it was from but was regarding
> the isdn supp-service command. Maybe is only effective
> with SIP? I have not had need to use this command
> myself yet. (Sorry for formatting below).
>
>
>
Georgia Institute of Technology     Atlanta, Georgia
Georgia Tech Research Institute
Electronic Systems Laboratory

Public Key ID: 0x894F4419
[7338 D848 8BEE FF6D 23D1  4355 2F6C DE0C 894F 4419]

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From pyago at adomo.com  Fri Dec 16 14:46:16 2005
From: pyago at adomo.com (Paul Yago)
Date: Fri Dec 16 14:48:40 2005
Subject: [cisco-voip] Attendant Console Errors
Message-ID: <170BC64B51964949A28FE8A48EF1D4EBFBEF08@corpexchsrv01.adomo.com>

Hello, does anyone with Attendant Console experience know why I'm seeing
this error (below)?

 

I'm trying to configure a hunt group with 2 DNs and 1 pilot point. I
created an "ac" user as both an Attendant Console User and CCM user,
with association to both DNs and the Pilot Point. In the documentation,
it's not clear to me how the "ac" user should be used. Does he need to
be part of the AC hunt group? 

 

Any help is very much appreciated.

Thanks.

Paul

 

 

 

50621 12/16 11:35:22.806 ERROR HuntGroups:

pkid: 0xAE5672C0728C86458CAE73F87025E57E, name: New_JTAPI, dn: 6500,
fkDevice: 0x8FCBD68266333C4D98325062886C69DB, fkCSS:
0x5563A64BB3195843A5E2A5FBD113AFE6, huntingAlgorithm: 1,
dbHuntingAlgorithm: 1, owner: 0, active: 2, queueEnabled: false,
queueSize: 32, holdTime: 0, CMs: CM_LABCM01- 

Lines: 

     DN: 5750, AlwaysRoute: true, Online: true, LineActive: true,
RoutingCall: false, Waiting: false, LS: Active, Idle Time:
1134697714514, RouteUnknown: true

     DN: 7017, AlwaysRoute: true, Online: true, LineActive: true,
RoutingCall: false, Waiting: false, LS: Idle, Idle Time: 1134697737624,
RouteUnknown: true

HuntGroupsUsers:

pkid: 0xAF326A961CFAEF419870B74D4BB27478, userid: ac, passwd: *****

Sessions:

Active Sessions:

Deactive Sessions:

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From rratliff at cisco.com  Fri Dec 16 15:08:39 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Dec 16 15:08:39 2005
Subject: [cisco-voip] Attendant Console Errors
In-Reply-To: <170BC64B51964949A28FE8A48EF1D4EBFBEF08@corpexchsrv01.adomo.com>
References: <170BC64B51964949A28FE8A48EF1D4EBFBEF08@corpexchsrv01.adomo.com>
Message-ID: <3C916BD0-79A8-4218-B337-2B8CE9EBA23A@cisco.com>

The AC user is used by Telephony Call Dispatcher to logon to CTI.  It  
does not need to be configured other than be in the directory with  
"CTI Use" and "Call Park Retrieval Allowed" check boxes ticked.  The  
password and PIN should be 12345 and the device associations should  
be any AC hunt pilot and any phone to be controlled by the people  
using the AC desktop client.

AC does not need to be an Attendant Console user.

-Ryan

On Dec 16, 2005, at 2:46 PM, Paul Yago wrote:

Hello, does anyone with Attendant Console experience know why I?m  
seeing this error (below)?



I?m trying to configure a hunt group with 2 DNs and 1 pilot point. I  
created an ?ac? user as both an Attendant Console User and CCM user,  
with association to both DNs and the Pilot Point. In the  
documentation, it?s not clear to me how the ?ac? user should be used.  
Does he need to be part of the AC hunt group?



Any help is very much appreciated.

Thanks.

Paul







50621 12/16 11:35:22.806 ERROR HuntGroups:

pkid: 0xAE5672C0728C86458CAE73F87025E57E, name: New_JTAPI, dn: 6500,  
fkDevice: 0x8FCBD68266333C4D98325062886C69DB, fkCSS:  
0x5563A64BB3195843A5E2A5FBD113AFE6, huntingAlgorithm: 1,  
dbHuntingAlgorithm: 1, owner: 0, active: 2, queueEnabled: false,  
queueSize: 32, holdTime: 0, CMs: CM_LABCM01-

Lines:

      DN: 5750, AlwaysRoute: true, Online: true, LineActive: true,  
RoutingCall: false, Waiting: false, LS: Active, Idle Time:  
1134697714514, RouteUnknown: true

      DN: 7017, AlwaysRoute: true, Online: true, LineActive: true,  
RoutingCall: false, Waiting: false, LS: Idle, Idle Time:  
1134697737624, RouteUnknown: true

HuntGroupsUsers:

pkid: 0xAF326A961CFAEF419870B74D4BB27478, userid: ac, passwd: *****

Sessions:

Active Sessions:

Deactive Sessions:

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



From tennille.spence at gsa.gov  Fri Dec 16 16:31:26 2005
From: tennille.spence at gsa.gov (tennille.spence@gsa.gov)
Date: Fri Dec 16 16:31:42 2005
Subject: [cisco-voip] Setting Up Direct Transfer to Voicemail
Message-ID: <OF60668F4F.B21D7A63-ON862570D9.007615F3-862570D9.00763BF5@gsa.gov>


Can anyone offer some assistance / advice on setting up a direct transfer
to voicemail?

Tennille Spence
IT Business Liaison
PBS Technology Branch
816-823-1357 Office
816-806-6342 Cell
816-823-5526 Fax

"What lies behind us and what lies before us are tiny matters compared to
what lies within us." - Ralph Waldo Emerson

"Opportunity is missed by most people because it is dressed in overalls and
looks like work." - Thomas A. Edison



From lelio at uoguelph.ca  Fri Dec 16 16:38:24 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Dec 16 16:38:19 2005
Subject: [cisco-voip] Setting Up Direct Transfer to Voicemail
References: <OF60668F4F.B21D7A63-ON862570D9.007615F3-862570D9.00763BF5@gsa.gov>
Message-ID: <000901c60289$0b9a8950$ca01a8c0@deadfish>

Basically, you transfer a call to a route point that has a voicemail profile mask that removes the leading digits.

This document explains how to do it. It works well, but assumes you only have one voicemail pilot. If you have more, you will need more route points.

http://www.cisco.com/warp/customer/788/AVVID/transfer_voicemail.html


  ----- Original Message ----- 
  From: tennille.spence at gsa.gov 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, December 16, 2005 4:31 PM
  Subject: [cisco-voip] Setting Up Direct Transfer to Voicemail



  Can anyone offer some assistance / advice on setting up a direct transfer
  to voicemail?

  Tennille Spence
  IT Business Liaison
  PBS Technology Branch
  816-823-1357 Office
  816-806-6342 Cell
  816-823-5526 Fax

  "What lies behind us and what lies before us are tiny matters compared to
  what lies within us." - Ralph Waldo Emerson

  "Opportunity is missed by most people because it is dressed in overalls and
  looks like work." - Thomas A. Edison


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From bababooey at cox.net  Fri Dec 16 16:58:37 2005
From: bababooey at cox.net (Bob A. Bowie)
Date: Fri Dec 16 16:58:47 2005
Subject: [cisco-voip] 7914/7970 - CME
References: <200512161700.jBGH05EV090375@puck.nether.net>
Message-ID: <000801c6028b$df07d520$0f0110ac@corporation2>

Does anyone know if the 7914 exp module is compatible with the 7970 ip 
phones
use with CM Express?

Tx 



From pyago at adomo.com  Fri Dec 16 18:06:57 2005
From: pyago at adomo.com (Paul Yago)
Date: Fri Dec 16 18:09:20 2005
Subject: [cisco-voip] Attendant Console Errors
Message-ID: <170BC64B51964949A28FE8A48EF1D4EBFBEF62@corpexchsrv01.adomo.com>

Thanks for the response Ryan.

I have a user named "ac" with the following checked.
1) Enable CTI Application Use
2) Enable CTI Super Provider
3) Call Park

And the pilot point and 2 hunt group members are all associated with
"ac". Password and Pin are 12345. 

I'm still getting the fast busy though when calling the pilot point. Any
other suggestions?

Thanks,
Paul



-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Friday, December 16, 2005 12:09 PM
To: Paul Yago
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant Console Errors

The AC user is used by Telephony Call Dispatcher to logon to CTI.  It  
does not need to be configured other than be in the directory with  
"CTI Use" and "Call Park Retrieval Allowed" check boxes ticked.  The  
password and PIN should be 12345 and the device associations should  
be any AC hunt pilot and any phone to be controlled by the people  
using the AC desktop client.

AC does not need to be an Attendant Console user.

-Ryan

On Dec 16, 2005, at 2:46 PM, Paul Yago wrote:

Hello, does anyone with Attendant Console experience know why I'm  
seeing this error (below)?



I'm trying to configure a hunt group with 2 DNs and 1 pilot point. I  
created an "ac" user as both an Attendant Console User and CCM user,  
with association to both DNs and the Pilot Point. In the  
documentation, it's not clear to me how the "ac" user should be used.  
Does he need to be part of the AC hunt group?



Any help is very much appreciated.

Thanks.

Paul







50621 12/16 11:35:22.806 ERROR HuntGroups:

pkid: 0xAE5672C0728C86458CAE73F87025E57E, name: New_JTAPI, dn: 6500,  
fkDevice: 0x8FCBD68266333C4D98325062886C69DB, fkCSS:  
0x5563A64BB3195843A5E2A5FBD113AFE6, huntingAlgorithm: 1,  
dbHuntingAlgorithm: 1, owner: 0, active: 2, queueEnabled: false,  
queueSize: 32, holdTime: 0, CMs: CM_LABCM01-

Lines:

      DN: 5750, AlwaysRoute: true, Online: true, LineActive: true,  
RoutingCall: false, Waiting: false, LS: Active, Idle Time:  
1134697714514, RouteUnknown: true

      DN: 7017, AlwaysRoute: true, Online: true, LineActive: true,  
RoutingCall: false, Waiting: false, LS: Idle, Idle Time:  
1134697737624, RouteUnknown: true

HuntGroupsUsers:

pkid: 0xAF326A961CFAEF419870B74D4BB27478, userid: ac, passwd: *****

Sessions:

Active Sessions:

Deactive Sessions:

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From candace_holman at harvard.edu  Fri Dec 16 18:22:49 2005
From: candace_holman at harvard.edu (Candace Holman)
Date: Fri Dec 16 18:22:57 2005
Subject: [cisco-voip] References for shared lines 3.3
Message-ID: <43A34C49.3040405@harvard.edu>

Does anyone have a good reference for implementing shared lines with 
ring delay in CM 3.3?  We want the phone to ring on device A, but only 
flash (at first) on device B where the second shared line appearance 
lives.  User B can pick it up on flash or on ring.  Is this a job for IPMA?

Candace



From lelio at uoguelph.ca  Fri Dec 16 18:55:44 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Dec 16 18:55:37 2005
Subject: [cisco-voip] References for shared lines 3.3
References: <43A34C49.3040405@harvard.edu>
Message-ID: <000901c6029c$3abd3530$ca01a8c0@deadfish>

I'm not aware of any two-stage ringing options with either the phones or with IPMA. Perhaps IPMA has undergone some changes, it has been a while since I've played with it.

You can get what you are looking for if you are willing to use another line appearance on the phones. The first line appearance on phone A rings, the first line appearance on phone B flashes. The FwdNoAnswer target is another line on each phone with different ring behaviour.


  ----- Original Message ----- 
  From: Candace Holman 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, December 16, 2005 6:22 PM
  Subject: [cisco-voip] References for shared lines 3.3


  Does anyone have a good reference for implementing shared lines with 
  ring delay in CM 3.3?  We want the phone to ring on device A, but only 
  flash (at first) on device B where the second shared line appearance 
  lives.  User B can pick it up on flash or on ring.  Is this a job for IPMA?

  Candace


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Brian.Henry at apptis.com  Fri Dec 16 18:56:24 2005
From: Brian.Henry at apptis.com (Brian Henry)
Date: Fri Dec 16 18:57:57 2005
Subject: [cisco-voip] 7914/7970 - CME
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC03D789@CH-EXCH-02.intelltech.apptis.inc>

Cisco has not developed yet nor do I know if they have it slated for a 7970 as a feature request as of today in either CCM or CME.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Bob A. Bowie
Sent: Fri 12/16/2005 4:58 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7914/7970 - CME



Does anyone know if the 7914 exp module is compatible with the 7970 ip
phones
use with CM Express?

Tx


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From Scott.Voll at wesd.org  Fri Dec 16 19:00:43 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Dec 16 19:00:27 2005
Subject: [cisco-voip] 7914/7970 - CME
Message-ID: <D713462ED535184D830F6F6301753E7E01A120D2@VISHNU.wesd.org>

Are you saying I can't use a 7970 with 7914 with CM 4.1?

 

I just purchased both.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Henry
Sent: Friday, December 16, 2005 3:56 PM
To: Bob A. Bowie; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] 7914/7970 - CME

 

Cisco has not developed yet nor do I know if they have it slated for a
7970 as a feature request as of today in either CCM or CME.

 

Brian

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Bob A. Bowie
Sent: Fri 12/16/2005 4:58 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7914/7970 - CME

Does anyone know if the 7914 exp module is compatible with the 7970 ip
phones
use with CM Express?

Tx


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cisco-voip at puck.nether.net
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From lelio at uoguelph.ca  Fri Dec 16 19:04:27 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Dec 16 19:04:20 2005
Subject: [cisco-voip] 7914/7970 - CME
References: <D713462ED535184D830F6F6301753E7E01A120D2@VISHNU.wesd.org>
Message-ID: <002e01c6029d$72977f50$ca01a8c0@deadfish>

[cisco-voip] 7914/7970 - CMEYou should be OK Scott. If we can trust the docs. ;) Still unclear why they won't support it on a 7940. :(

>From the Cisco IP Phone Expansion Module 7914 data sheet:
SYSTEM REQUIREMENTS

Cisco CallManager Version 3.1(2c) or later
HARDWARE REQUIREMENTS

. Cisco IP Phones 7960G (CP-7960G), 7961G (CP-7961G) , 7970G (CP-7970G) and 7971G-GE (CP-7971G-GE)

. Cisco IP Phone Expansion Module 7914 (CP-7914=) (maximum of two modules) (Connecting cable supplied with each Cisco 7914)

. New footstand is required, single (CP-SINGLFOOTSTAND=) (Figure 2) or double (CP-DOUBLFOOTSTAND=) (Figure 3) as appropriate

. Power cube (CP-PWR-CUBE-3=) and cord (CP-PWR-CORD-xx=) to supply local power





  ----- Original Message ----- 
  From: Voll, Scott 
  To: Brian Henry ; Bob A. Bowie ; cisco-voip at puck.nether.net 
  Sent: Friday, December 16, 2005 7:00 PM
  Subject: RE: [cisco-voip] 7914/7970 - CME


  Are you saying I can't use a 7970 with 7914 with CM 4.1?

   

  I just purchased both.

   

  Scott

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Henry
  Sent: Friday, December 16, 2005 3:56 PM
  To: Bob A. Bowie; cisco-voip at puck.nether.net
  Subject: RE: [cisco-voip] 7914/7970 - CME

   

  Cisco has not developed yet nor do I know if they have it slated for a 7970 as a feature request as of today in either CCM or CME.

   

  Brian

   


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

  From: cisco-voip-bounces at puck.nether.net on behalf of Bob A. Bowie
  Sent: Fri 12/16/2005 4:58 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] 7914/7970 - CME

  Does anyone know if the 7914 exp module is compatible with the 7970 ip
  phones
  use with CM Express?

  Tx


  _______________________________________________
  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 subs-cisco at cbvcollections.com  Fri Dec 16 19:24:09 2005
From: subs-cisco at cbvcollections.com (Cisco)
Date: Fri Dec 16 19:24:23 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
Message-ID: <9AEF8665DE74FA43B3B987FB931E92AE0F1002@CBVVANEM01.cbv.dns>

I'm trying to use route filters (area code == 204 OR etc.) with a "9.@"
pattern instead of the more difficult to manage 9.1204XXXXXXX to route
long distance calls.  
 
The pattern/filter work fine, the problem is that when the number is
matched using 9.@ pattern (verified with the Cisco the dialled number
analyser), by the time it reaches the local provider somehow the "1" is
dropped and the caller is presented with a message "you must dial 1 to
access a long distance number" from the local provider.  If I change the
pattern back to 9.1XXXXXXXXXX then the number is dialled as expected.
 
I've used the dialled number analyser to ensure that no transformations
are taking place at any point along the route to the PEST gateway.
According to the analyser the dialled number that is sent to the PEST
gateway in either case is the same.  However, one works and the other
does not.
 
I should also note that the 9.@ pattern with a filter of (local area
code == 604 OR etc.) for local 10 digit dialling works just fine as
either 9.@ or 9.XXXXXXXXXX.  Both are routed as expected.
 
Ideas?
________________________________

 <http://www.cbvcollections.com/cbvImages/ecardlogo.jpg> 

Frank Wakelin
Senior Network/Security Analyst


100 - 814 Richards St.
Vancouver, BC
V6B 3A7


Telephone: (604) 661-7906
fwakelin at cbvcollections.com


Confidentiality Warning:
This message and any attachments are intended only for the use of the intended recipients (cisco-voip at puck.nether.net), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender (mailto:subs-cisco at cbvcollections.com) immediately by return e-mail, and delete this message and any attachments from your system. Thank you.

Information Confidentielle:
Le present message, ainsi que tout fichier qui y est joint, est envoye a l'intention exclusive de son ou de ses destinataires (cisco-voip at puck.nether.net); il est de nature confidentielle et peut constituer une information privilegiee. Nous avertissons toute personne autre que le destinataire prevu que tout examen, reacheminement, impression, copie, distribution ou autre utilisation de ce message et de tout fichier qui y est joint est strictement interdit. Si vous n'etes pas le destinataire prevu, veuillez en aviser immediatement l'expediteur (mailto:subs-cisco at cbvcollections.com) par retour de courriel et supprimer ce message et tout document joint de votre systeme. Merci


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From jgrace at digitelusa.net  Fri Dec 16 19:44:03 2005
From: jgrace at digitelusa.net (James)
Date: Fri Dec 16 19:43:58 2005
Subject: [cisco-voip] Ccm forward call
Message-ID: <1015019729-1134780236-cardhu_blackberry.rim.net-5216-@engine20-cell02>

I have dids configured on our cm that r configured to forward to an outside line. My question is if I call into that did from my cell and it forwards me to a ld.outside number. will that call tie up a port as long as the call is connected.  And will the charges go on our pri 
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From highspeedsnow at gmail.com  Fri Dec 16 19:55:15 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Fri Dec 16 19:56:35 2005
Subject: [cisco-voip] Ccm forward call
In-Reply-To: <1015019729-1134780236-cardhu_blackberry.rim.net-5216-@engine20-cell02>
Message-ID: <43a36241.65375949.7bc2.5d08@mx.gmail.com>

That call will actually tie up two ports (incoming and outgoing) and yes
incur a charge.

Wouldn't it be nice if the Carrier's supported Q.SIG Anti-Trumboning ;)  


-Mark Snow
CCIE Voice # 14073

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of James
> Sent: Friday, December 16, 2005 7:44 PM
> To: CiscosupportUpuck
> Subject: [cisco-voip] Ccm forward call
> 
> I have dids configured on our cm that r configured to forward to an
outside line.
> My question is if I call into that did from my cell and it forwards me to
a ld.outside
> number. will that call tie up a port as long as the call is connected.
And will the
> charges go on our pri
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From fwn at feasible.net  Fri Dec 16 19:59:29 2005
From: fwn at feasible.net (Fred Nielsen)
Date: Fri Dec 16 19:59:31 2005
Subject: [cisco-voip] Ccm forward call
In-Reply-To: <43a36241.65375949.7bc2.5d08@mx.gmail.com>
Message-ID: <01db01c602a5$230298f0$6401a8c0@ws1>

Of course path pruning would bring up obvious questions about who pays toll
rates once call control is released ;)

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Snow
Sent: Friday, December 16, 2005 4:55 PM
To: 'James'; 'CiscosupportUpuck'
Subject: RE: [cisco-voip] Ccm forward call

That call will actually tie up two ports (incoming and outgoing) and yes
incur a charge.

Wouldn't it be nice if the Carrier's supported Q.SIG Anti-Trumboning ;)  


-Mark Snow
CCIE Voice # 14073

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of James
> Sent: Friday, December 16, 2005 7:44 PM
> To: CiscosupportUpuck
> Subject: [cisco-voip] Ccm forward call
> 
> I have dids configured on our cm that r configured to forward to an
outside line.
> My question is if I call into that did from my cell and it forwards me to
a ld.outside
> number. will that call tie up a port as long as the call is connected.
And will the
> charges go on our pri
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> 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 highspeedsnow at gmail.com  Fri Dec 16 19:58:58 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Fri Dec 16 20:00:22 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
In-Reply-To: <9AEF8665DE74FA43B3B987FB931E92AE0F1002@CBVVANEM01.cbv.dns>
Message-ID: <43a36322.745d40f3.0858.1e2b@mx.gmail.com>

Is it a PRI that the call is going out of on an IOS GW?

If so, have you done a 'debug isdn q931' to ensure that in fact the CCM is
dropping the 1?

 

-Mark Snow

CCIE Voice # 14073

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
Sent: Friday, December 16, 2005 7:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

I'm trying to use route filters (area code == 204 OR etc.) with a "9.@"
pattern instead of the more difficult to manage 9.1204XXXXXXX to route long
distance calls.  

 

The pattern/filter work fine, the problem is that when the number is matched
using 9.@ pattern (verified with the Cisco the dialled number analyser), by
the time it reaches the local provider somehow the "1" is dropped and the
caller is presented with a message "you must dial 1 to access a long
distance number" from the local provider.  If I change the pattern back to
9.1XXXXXXXXXX then the number is dialled as expected.

 

I've used the dialled number analyser to ensure that no transformations are
taking place at any point along the route to the PEST gateway.  According to
the analyser the dialled number that is sent to the PEST gateway in either
case is the same.  However, one works and the other does not.

 

I should also note that the 9.@ pattern with a filter of (local area code ==
604 OR etc.) for local 10 digit dialling works just fine as either 9.@ or
9.XXXXXXXXXX.  Both are routed as expected.

 

Ideas?

  _____  



  <http://www.cbvcollections.com/cbvImages/ecardlogo.jpg> 

Frank Wakelin
Senior Network/Security Analyst


100 - 814 Richards St.
Vancouver, BC
V6B 3A7


Telephone: (604) 661-7906
fwakelin at cbvcollections.com

 

Confidentiality Warning:
This message and any attachments are intended only for the use of the
intended recipients (cisco-voip at puck.nether.net), are confidential, and may
be privileged. If you are not the intended recipient, you are hereby
notified that any review, retransmission, conversion to hard copy, copying,
circulation or other use of this message and any attachments is strictly
prohibited. If you are not the intended recipient, please notify the sender
(Cisco <mailto:subs-cisco at cbvcollections.com> ) immediately by return
e-mail, and delete this message and any attachments from your system. Thank
you. 

Information Confidentielle:
Le present message, ainsi que tout fichier qui y est joint, est envoye a
l'intention exclusive de son ou de ses destinataires
(cisco-voip at puck.nether.net); il est de nature confidentielle et peut
constituer une information privilegiee. Nous avertissons toute personne
autre que le destinataire prevu que tout examen, reacheminement, impression,
copie, distribution ou autre utilisation de ce message et de tout fichier
qui y est joint est strictement interdit. Si vous n'etes pas le destinataire
prevu, veuillez en aviser immediatement l'expediteur (Cisco
<mailto:subs-cisco at cbvcollections.com> ) par retour de courriel et supprimer
ce message et tout document joint de votre systeme. Merci

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From jgrace at digitelusa.net  Fri Dec 16 20:07:46 2005
From: jgrace at digitelusa.net (James)
Date: Fri Dec 16 20:07:39 2005
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 75
Message-ID: <640204586-1134781657-cardhu_blackberry.rim.net-11810-@engine18-cell02>

Hello. I found out that the soft keys for cisco ipphones can not be configured to do a speed dial or short cut to call the upstairs operator.  The function of the softkey are preconfigured in the software and cannot be modified. The only changes will be the layout of the keys
-----Original Message-----
From: cisco-voip-request at puck.nether.net
Date: Thu, 15 Dec 2005 12:00:09 
To:cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 34, Issue 75

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: QoS Toolsets (Sarff, Dean)
   2. RE: re: QoS Toolsets (VoIP Forum)


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

Message: 1
Date: Thu, 15 Dec 2005 11:55:49 -0500
From: "Sarff, Dean" <Dean.Sarff at mms.gov>
Subject: [cisco-voip] re: QoS Toolsets
To: <cisco-voip at puck.nether.net>
Message-ID:
	<37D05FC7972DF14DB5F6ABE256D0C36B01E3FCB5 at IMSHEXPRI03.service.agency.mms.pri>
	
Content-Type: text/plain; charset="us-ascii"

I'd like feedback from those users of either NET IQ's VoIP tools or
Integrated Research's PROGNOSIS.  How do you like your product
selection?  

Dean Sarff, MMS A&B IMD, 703-787-1272

Merry Christmas!  Happy Birthday Jesus!
May you an your family have a Blessed Holiday.

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

Message: 2
Date: Thu, 15 Dec 2005 11:59:36 -0500
From: "VoIP Forum" <voipforum at apptis.com>
Subject: RE: [cisco-voip] re: QoS Toolsets
To: "Sarff, Dean" <Dean.Sarff at mms.gov>, <cisco-voip at puck.nether.net>
Message-ID:
	<2F09B2162CCDEA4EA05C089B24FBC2BCF05E8D at CH-EXCH-02.intelltech.apptis.inc>
	
Content-Type: text/plain; charset="us-ascii"

Qovia is the way to go!

www.qovia.com <http://www.qovia.com/> 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sarff, Dean
Sent: Thursday, December 15, 2005 11:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] re: QoS Toolsets

 

I'd like feedback from those users of either NET IQ's VoIP tools or
Integrated Research's PROGNOSIS.  How do you like your product
selection?  

Dean Sarff, MMS A&B IMD, 703-787-1272

Merry Christmas!  Happy Birthday Jesus!

May you an your family have a Blessed Holiday.

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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 34, Issue 75
******************************************
---




James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From subs-cisco at cbvcollections.com  Fri Dec 16 20:17:01 2005
From: subs-cisco at cbvcollections.com (Cisco)
Date: Fri Dec 16 20:17:10 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
Message-ID: <9AEF8665DE74FA43B3B987FB931E92AE0F1004@CBVVANEM01.cbv.dns>

Well I'm far from the telephony expert I want to be (at this point -
give me a few months) but it would appear that you may be on to
something.  I took a look at the gateway configurations within
CallManager.  Each of the gateways within the PSTN group (Product :
Cisco Catalyst 6000 T1 VoIP Gateway - Device Protocol: Digital Access
PRI) have the same configuration.  The following settings are currently
used from the Call Routing Information (Outbound):

*	
	Calling Line ID Presentation*: Restricted
*	
	Calling Party Selection*: Originator
*	
	Called party IE number type unknown*: Cisco CallManager
*	
	Calling party IE number type unknown*: National
*	
	Called Numbering Plan*: Cisco CallManager
*	
	Calling Numbering Plan*: ISDN
*	
	Number of digits to strip*: 0
*	
	Caller ID DN:
*	
	SMDI Base Port*:

Do you think the underlined (above) is causing the issue?  Is this
something I can change (to "Cisco CallManager") or do I need to involve
our provider?

________________________________

From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: December 16, 2005 4:59 PM
To: Cisco; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...



Is it a PRI that the call is going out of on an IOS GW?

If so, have you done a 'debug isdn q931' to ensure that in fact the CCM
is dropping the 1?

 

-Mark Snow

CCIE Voice # 14073

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
Sent: Friday, December 16, 2005 7:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

I'm trying to use route filters (area code == 204 OR etc.) with a "9.@"
pattern instead of the more difficult to manage 9.1204XXXXXXX to route
long distance calls.  

 

The pattern/filter work fine, the problem is that when the number is
matched using 9.@ pattern (verified with the Cisco the dialled number
analyser), by the time it reaches the local provider somehow the "1" is
dropped and the caller is presented with a message "you must dial 1 to
access a long distance number" from the local provider.  If I change the
pattern back to 9.1XXXXXXXXXX then the number is dialled as expected.

 

I've used the dialled number analyser to ensure that no transformations
are taking place at any point along the route to the PEST gateway.
According to the analyser the dialled number that is sent to the PEST
gateway in either case is the same.  However, one works and the other
does not.

 

I should also note that the 9.@ pattern with a filter of (local area
code == 604 OR etc.) for local 10 digit dialling works just fine as
either 9.@ or 9.XXXXXXXXXX.  Both are routed as expected.

 

Ideas?

________________________________

 <http://www.cbvcollections.com/cbvImages/ecardlogo.jpg> 

Frank Wakelin
Senior Network/Security Analyst

100 - 814 Richards St.
Vancouver, BC
V6B 3A7

Telephone: (604) 661-7906
fwakelin at cbvcollections.com

 

Confidentiality Warning:
This message and any attachments are intended only for the use of the
intended recipients (cisco-voip at puck.nether.net), are confidential, and
may be privileged. If you are not the intended recipient, you are hereby
notified that any review, retransmission, conversion to hard copy,
copying, circulation or other use of this message and any attachments is
strictly prohibited. If you are not the intended recipient, please
notify the sender (Cisco <mailto:subs-cisco at cbvcollections.com> )
immediately by return e-mail, and delete this message and any
attachments from your system. Thank you. 

Information Confidentielle:
Le present message, ainsi que tout fichier qui y est joint, est envoye a
l'intention exclusive de son ou de ses destinataires
(cisco-voip at puck.nether.net); il est de nature confidentielle et peut
constituer une information privilegiee. Nous avertissons toute personne
autre que le destinataire prevu que tout examen, reacheminement,
impression, copie, distribution ou autre utilisation de ce message et de
tout fichier qui y est joint est strictement interdit. Si vous n'etes
pas le destinataire prevu, veuillez en aviser immediatement l'expediteur
(Cisco <mailto:subs-cisco at cbvcollections.com> ) par retour de courriel
et supprimer ce message et tout document joint de votre systeme. Merci

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From jgrace at digitelusa.net  Fri Dec 16 20:42:02 2005
From: jgrace at digitelusa.net (James)
Date: Fri Dec 16 20:41:55 2005
Subject: [cisco-voip] Sync address book plugin
Message-ID: <2130867875-1134783712-cardhu_blackberry.rim.net-11521-@engine21-cell02>

What would be the best way to sync cm with outlook address book.  I ran the install and did the sync. But it sync the window address book.  And this do not get populated unless the user were using outlook express. Just looking for some deployment ideals
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From kthorngr at cisco.com  Fri Dec 16 21:10:16 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri Dec 16 21:10:29 2005
Subject: [cisco-voip] Sync address book plugin
In-Reply-To: <2130867875-1134783712-cardhu_blackberry.rim.net-11521-@engine21-cell02>
References: <2130867875-1134783712-cardhu_blackberry.rim.net-11521-@engine21-cell02>
Message-ID: <0f3dc82a27ad5544de8a0e07a8e93c74@cisco.com>

The tabsync that is installed from the CCM plugins will only sync with 
the Windows Address Book.  You will need to you Personal Assistant or 
PPS to sync with the adddress book stored in Exchange.

Kevin
On Dec 16, 2005, at 8:42 PM, James wrote:

> What would be the best way to sync cm with outlook address book.  I 
> ran the install and did the sync. But it sync the window address book. 
>  And this do not get populated unless the user were using outlook 
> express. Just looking for some deployment ideals
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From lelio at uoguelph.ca  Fri Dec 16 21:14:32 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Dec 16 21:14:25 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
References: <9AEF8665DE74FA43B3B987FB931E92AE0F1002@CBVVANEM01.cbv.dns>
Message-ID: <004101c602af$9ebc7100$ca01a8c0@deadfish>

There are a few spots that digit manipulation occurs that you should look at. The first, is the route pattern. Near the bottom you have the called party transformations. Here you can drop pre-At for example so you don't send the 9 to the PSTN. The second is the route list and the configuration of each route group within the route list. Again, take a look at the called party transformations. If you don't have route groups in the route list, I think you can have the trunks listed, but I'm not 100% sure.

That's the first two places I would look. If the two route patterns are using the same gateways, I don't think it would be your gateway configuration.

  ----- Original Message ----- 
  From: Cisco 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, December 16, 2005 7:24 PM
  Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...


  I'm trying to use route filters (area code == 204 OR etc.) with a "9.@" pattern instead of the more difficult to manage 9.1204XXXXXXX to route long distance calls.  

  The pattern/filter work fine, the problem is that when the number is matched using 9.@ pattern (verified with the Cisco the dialled number analyser), by the time it reaches the local provider somehow the "1" is dropped and the caller is presented with a message "you must dial 1 to access a long distance number" from the local provider.  If I change the pattern back to 9.1XXXXXXXXXX then the number is dialled as expected.

  I've used the dialled number analyser to ensure that no transformations are taking place at any point along the route to the PEST gateway.  According to the analyser the dialled number that is sent to the PEST gateway in either case is the same.  However, one works and the other does not.

  I should also note that the 9.@ pattern with a filter of (local area code == 604 OR etc.) for local 10 digit dialling works just fine as either 9.@ or 9.XXXXXXXXXX.  Both are routed as expected.

  Ideas?

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

             Frank Wakelin
              Senior Network/Security Analyst

             
              100 - 814 Richards St.
              Vancouver, BC
              V6B 3A7

             
              Telephone: (604) 661-7906
              fwakelin at cbvcollections.com
             
       
  Confidentiality Warning:
  This message and any attachments are intended only for the use of the intended recipients (cisco-voip at puck.nether.net), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender (Cisco) immediately by return e-mail, and delete this message and any attachments from your system. Thank you. 
  Information Confidentielle:
  Le present message, ainsi que tout fichier qui y est joint, est envoye a l'intention exclusive de son ou de ses destinataires (cisco-voip at puck.nether.net); il est de nature confidentielle et peut constituer une information privilegiee. Nous avertissons toute personne autre que le destinataire prevu que tout examen, reacheminement, impression, copie, distribution ou autre utilisation de ce message et de tout fichier qui y est joint est strictement interdit. Si vous n'etes pas le destinataire prevu, veuillez en aviser immediatement l'expediteur (Cisco) par retour de courriel et supprimer ce message et tout document joint de votre systeme. Merci



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


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  https://puck.nether.net/mailman/listinfo/cisco-voip
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From highspeedsnow at gmail.com  Fri Dec 16 23:41:22 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Fri Dec 16 23:42:06 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
In-Reply-To: <9AEF8665DE74FA43B3B987FB931E92AE0F1004@CBVVANEM01.cbv.dns>
Message-ID: <43a3971b.067bec8e.021e.77ea@mx.gmail.com>

Lelio is correct, make sure that digit manipulation is not occurring at the
Route Pattern, OR within the Route Groups inside the Route Lists.

 

If those are not stripping anything but PreDot then try this:

Try changing:

*	Called Numbering Plan*: unknown
*	Calling Numbering Plan*: unknown

 

Then test your calls - see if they work - you will have to do a reset on the
6608 gateway first, so make sure that there are no calls going on at the
time.

 

Also, since you are running a 6608 blade on a 6500 Cat, you cannot do the
'debug isdn q931' command I asked about previously.

 

LMK if that helps .

 

And hey, if you do want to become a Voice Expert soon, check out my class
sometime:

http://www.ipexpert.com/products_services/product.asp?sku=ip0012

 

-Mark Snow

CCIE Voice # 14073

 

 

  _____  

From: Cisco [mailto:subs-cisco at cbvcollections.com] 
Sent: Friday, December 16, 2005 8:17 PM
To: Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

Well I'm far from the telephony expert I want to be (at this point - give me
a few months) but it would appear that you may be on to something.  I took a
look at the gateway configurations within CallManager.  Each of the gateways
within the PSTN group (Product : Cisco Catalyst 6000 T1 VoIP Gateway -
Device Protocol: Digital Access PRI) have the same configuration.  The
following settings are currently used from the Call Routing Information
(Outbound):

*	Calling Line ID Presentation*: Restricted
*	Calling Party Selection*: Originator
*	Called party IE number type unknown*: Cisco CallManager
*	Calling party IE number type unknown*: National
*	Called Numbering Plan*: Cisco CallManager
*	Calling Numbering Plan*: ISDN
*	Number of digits to strip*: 0
*	Caller ID DN:
*	SMDI Base Port*:

Do you think the underlined (above) is causing the issue?  Is this something
I can change (to "Cisco CallManager") or do I need to involve our provider?

 

  _____  

From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: December 16, 2005 4:59 PM
To: Cisco; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

Is it a PRI that the call is going out of on an IOS GW?

If so, have you done a 'debug isdn q931' to ensure that in fact the CCM is
dropping the 1?

 

-Mark Snow

CCIE Voice # 14073

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
Sent: Friday, December 16, 2005 7:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

I'm trying to use route filters (area code == 204 OR etc.) with a "9.@"
pattern instead of the more difficult to manage 9.1204XXXXXXX to route long
distance calls.  

 

The pattern/filter work fine, the problem is that when the number is matched
using 9.@ pattern (verified with the Cisco the dialled number analyser), by
the time it reaches the local provider somehow the "1" is dropped and the
caller is presented with a message "you must dial 1 to access a long
distance number" from the local provider.  If I change the pattern back to
9.1XXXXXXXXXX then the number is dialled as expected.

 

I've used the dialled number analyser to ensure that no transformations are
taking place at any point along the route to the PEST gateway.  According to
the analyser the dialled number that is sent to the PEST gateway in either
case is the same.  However, one works and the other does not.

 

I should also note that the 9.@ pattern with a filter of (local area code ==
604 OR etc.) for local 10 digit dialling works just fine as either 9.@ or
9.XXXXXXXXXX.  Both are routed as expected.

 

Ideas?

  _____  



  <http://www.cbvcollections.com/cbvImages/ecardlogo.jpg> 

Frank Wakelin
Senior Network/Security Analyst


100 - 814 Richards St.
Vancouver, BC
V6B 3A7


Telephone: (604) 661-7906
fwakelin at cbvcollections.com

 

Confidentiality Warning:
This message and any attachments are intended only for the use of the
intended recipients (cisco-voip at puck.nether.net), are confidential, and may
be privileged. If you are not the intended recipient, you are hereby
notified that any review, retransmission, conversion to hard copy, copying,
circulation or other use of this message and any attachments is strictly
prohibited. If you are not the intended recipient, please notify the sender
(Cisco <mailto:subs-cisco at cbvcollections.com> ) immediately by return
e-mail, and delete this message and any attachments from your system. Thank
you. 

Information Confidentielle:
Le present message, ainsi que tout fichier qui y est joint, est envoye a
l'intention exclusive de son ou de ses destinataires
(cisco-voip at puck.nether.net); il est de nature confidentielle et peut
constituer une information privilegiee. Nous avertissons toute personne
autre que le destinataire prevu que tout examen, reacheminement, impression,
copie, distribution ou autre utilisation de ce message et de tout fichier
qui y est joint est strictement interdit. Si vous n'etes pas le destinataire
prevu, veuillez en aviser immediatement l'expediteur (Cisco
<mailto:subs-cisco at cbvcollections.com> ) par retour de courriel et supprimer
ce message et tout document joint de votre systeme. Merci

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From Brian.Henry at apptis.com  Fri Dec 16 23:41:42 2005
From: Brian.Henry at apptis.com (Brian Henry)
Date: Fri Dec 16 23:44:50 2005
Subject: [cisco-voip] 7914/7970 - CME
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC03D78A@CH-EXCH-02.intelltech.apptis.inc>

I stand corrected I did not know that they had made that happen.  A nice to know

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Fri 12/16/2005 7:04 PM
To: Voll, Scott; Brian Henry; Bob A. Bowie; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7914/7970 - CME


You should be OK Scott. If we can trust the docs. ;) Still unclear why they won't support it on a 7940. :(
 
>From the Cisco IP Phone Expansion Module 7914 data sheet <http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html> :

SYSTEM REQUIREMENTS

Cisco CallManager Version 3.1(2c) or later
HARDWARE REQUIREMENTS

* Cisco IP Phones 7960G (CP-7960G), 7961G (CP-7961G) , 7970G (CP-7970G) and 7971G-GE (CP-7971G-GE)

* Cisco IP Phone Expansion Module 7914 (CP-7914=) (maximum of two modules) (Connecting cable supplied with each Cisco 7914)

* New footstand is required, single (CP-SINGLFOOTSTAND=) (Figure 2) or double (CP-DOUBLFOOTSTAND=) (Figure 3) as appropriate

* Power cube (CP-PWR-CUBE-3=) and cord (CP-PWR-CORD-xx=) to supply local power

 

 

	----- Original Message ----- 
	From: Voll, Scott <mailto:Scott.Voll at wesd.org>  
	To: Brian Henry <mailto:Brian.Henry at apptis.com>  ; Bob A. Bowie <mailto:bababooey at cox.net>  ; cisco-voip at puck.nether.net 
	Sent: Friday, December 16, 2005 7:00 PM
	Subject: RE: [cisco-voip] 7914/7970 - CME


	Are you saying I can't use a 7970 with 7914 with CM 4.1?

	 

	I just purchased both.

	 

	Scott

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Henry
	Sent: Friday, December 16, 2005 3:56 PM
	To: Bob A. Bowie; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] 7914/7970 - CME

	 

	Cisco has not developed yet nor do I know if they have it slated for a 7970 as a feature request as of today in either CCM or CME.

	 

	Brian

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net on behalf of Bob A. Bowie
	Sent: Fri 12/16/2005 4:58 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] 7914/7970 - CME

	Does anyone know if the 7914 exp module is compatible with the 7970 ip
	phones
	use with CM Express?
	
	Tx
	
	
	_______________________________________________
	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 highspeedsnow at gmail.com  Fri Dec 16 23:50:28 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Fri Dec 16 23:51:14 2005
Subject: [cisco-voip] Ccm forward call
In-Reply-To: <01db01c602a5$230298f0$6401a8c0@ws1>
Message-ID: <43a3993f.4665195a.4546.67ef@mx.gmail.com>

Right, of course, *sigh*

> -----Original Message-----
> From: Fred Nielsen [mailto:fwn at feasible.net]
> Sent: Friday, December 16, 2005 7:59 PM
> To: 'Mark Snow'; 'James'; 'CiscosupportUpuck'
> Subject: RE: [cisco-voip] Ccm forward call
> 
> Of course path pruning would bring up obvious questions about who pays
toll
> rates once call control is released ;)
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Snow
> Sent: Friday, December 16, 2005 4:55 PM
> To: 'James'; 'CiscosupportUpuck'
> Subject: RE: [cisco-voip] Ccm forward call
> 
> That call will actually tie up two ports (incoming and outgoing) and yes
> incur a charge.
> 
> Wouldn't it be nice if the Carrier's supported Q.SIG Anti-Trumboning ;)
> 
> 
> -Mark Snow
> CCIE Voice # 14073
> 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> > bounces at puck.nether.net] On Behalf Of James
> > Sent: Friday, December 16, 2005 7:44 PM
> > To: CiscosupportUpuck
> > Subject: [cisco-voip] Ccm forward call
> >
> > I have dids configured on our cm that r configured to forward to an
> outside line.
> > My question is if I call into that did from my cell and it forwards me
to
> a ld.outside
> > number. will that call tie up a port as long as the call is connected.
> And will the
> > charges go on our pri
> > James D. Grace
> > System Engineer
> > CCNP CCNA MCSE MCDBA
> > Digitel Corp.
> > _______________________________________________
> > 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 Brian.Henry at apptis.com  Fri Dec 16 23:48:34 2005
From: Brian.Henry at apptis.com (Brian Henry)
Date: Fri Dec 16 23:53:10 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC03D78B@CH-EXCH-02.intelltech.apptis.inc>

>From my experience and I do believe from a few other places that I read.  If you are going to use Route Filters with a 9.@ or 9 at . then you can drop the pre@, this is a given.  As far as your dial plan you can usually make it CallManager because it knows what it should be from its own tables, like ISDN, National, etc..
If you stick with 9.1XX... then that is when you might have to do the unknown to let your provider determine what plan it is.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Mark Snow
Sent: Fri 12/16/2005 11:41 PM
To: 'Cisco'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...



Lelio is correct, make sure that digit manipulation is not occurring at the Route Pattern, OR within the Route Groups inside the Route Lists.

 

If those are not stripping anything but PreDot then try this:

Try changing:

*	Called Numbering Plan*: unknown 
*	Calling Numbering Plan*: unknown 

 

Then test your calls - see if they work - you will have to do a reset on the 6608 gateway first, so make sure that there are no calls going on at the time.

 

Also, since you are running a 6608 blade on a 6500 Cat, you cannot do the 'debug isdn q931' command I asked about previously.

 

LMK if that helps ...

 

And hey, if you do want to become a Voice Expert soon, check out my class sometime:

http://www.ipexpert.com/products_services/product.asp?sku=ip0012

 

-Mark Snow

CCIE Voice # 14073

 

 

________________________________

From: Cisco [mailto:subs-cisco at cbvcollections.com] 
Sent: Friday, December 16, 2005 8:17 PM
To: Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

Well I'm far from the telephony expert I want to be (at this point - give me a few months) but it would appear that you may be on to something.  I took a look at the gateway configurations within CallManager.  Each of the gateways within the PSTN group (Product : Cisco Catalyst 6000 T1 VoIP Gateway - Device Protocol: Digital Access PRI) have the same configuration.  The following settings are currently used from the Call Routing Information (Outbound):

*	Calling Line ID Presentation*: Restricted 
*	Calling Party Selection*: Originator 
*	Called party IE number type unknown*: Cisco CallManager 
*	Calling party IE number type unknown*: National 
*	Called Numbering Plan*: Cisco CallManager 
*	Calling Numbering Plan*: ISDN 
*	Number of digits to strip*: 0 
*	Caller ID DN: 
*	SMDI Base Port*: 

Do you think the underlined (above) is causing the issue?  Is this something I can change (to "Cisco CallManager") or do I need to involve our provider?

 

________________________________

From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: December 16, 2005 4:59 PM
To: Cisco; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

Is it a PRI that the call is going out of on an IOS GW?

If so, have you done a 'debug isdn q931' to ensure that in fact the CCM is dropping the 1?

 

-Mark Snow

CCIE Voice # 14073

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
Sent: Friday, December 16, 2005 7:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

I'm trying to use route filters (area code == 204 OR etc.) with a "9.@" pattern instead of the more difficult to manage 9.1204XXXXXXX to route long distance calls.  

 

The pattern/filter work fine, the problem is that when the number is matched using 9.@ pattern (verified with the Cisco the dialled number analyser), by the time it reaches the local provider somehow the "1" is dropped and the caller is presented with a message "you must dial 1 to access a long distance number" from the local provider.  If I change the pattern back to 9.1XXXXXXXXXX then the number is dialled as expected.

 

I've used the dialled number analyser to ensure that no transformations are taking place at any point along the route to the PEST gateway.  According to the analyser the dialled number that is sent to the PEST gateway in either case is the same.  However, one works and the other does not.

 

I should also note that the 9.@ pattern with a filter of (local area code == 604 OR etc.) for local 10 digit dialling works just fine as either 9.@ or 9.XXXXXXXXXX.  Both are routed as expected.

 

Ideas?

________________________________



Frank Wakelin
Senior Network/Security Analyst

100 - 814 Richards St.
Vancouver, BC
V6B 3A7

Telephone: (604) 661-7906
fwakelin at cbvcollections.com

 

Confidentiality Warning:
This message and any attachments are intended only for the use of the intended recipients (cisco-voip at puck.nether.net), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender (Cisco <mailto:subs-cisco at cbvcollections.com> ) immediately by return e-mail, and delete this message and any attachments from your system. Thank you. 

Information Confidentielle:
Le present message, ainsi que tout fichier qui y est joint, est envoye a l'intention exclusive de son ou de ses destinataires (cisco-voip at puck.nether.net); il est de nature confidentielle et peut constituer une information privilegiee. Nous avertissons toute personne autre que le destinataire prevu que tout examen, reacheminement, impression, copie, distribution ou autre utilisation de ce message et de tout fichier qui y est joint est strictement interdit. Si vous n'etes pas le destinataire prevu, veuillez en aviser immediatement l'expediteur (Cisco <mailto:subs-cisco at cbvcollections.com> ) par retour de courriel et supprimer ce message et tout document joint de votre systeme. Merci

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From lelio at uoguelph.ca  Sat Dec 17 00:29:09 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat Dec 17 00:29:01 2005
Subject: [cisco-voip] Ccm forward call
References: <43a3993f.4665195a.4546.67ef@mx.gmail.com>
Message-ID: <001a01c602ca$ce9f3ef0$ca01a8c0@deadfish>

Personally, I think it should be up to the last redirecting party that should pay the toll. Let's say a client calls me, and I've forwarded my office phone to my cottage. Why should the client pay the toll? It was me who made the decision to play hookie. The same goes for transfering calls. 

We've had a request for people to forward their phones off-campus and have had to restrict that to local calls only because of the ways CDRs are cut. If we have more requests we're going to have to work with our CDR billing firm to make sure they follow through on the CDR and catch those. A bit of a pain if you ask me.

  ----- Original Message ----- 
  From: Mark Snow 
  To: 'Fred Nielsen' ; 'James' ; 'CiscosupportUpuck' 
  Sent: Friday, December 16, 2005 11:50 PM
  Subject: RE: [cisco-voip] Ccm forward call


  Right, of course, *sigh*

  > -----Original Message-----
  > From: Fred Nielsen [mailto:fwn at feasible.net]
  > Sent: Friday, December 16, 2005 7:59 PM
  > To: 'Mark Snow'; 'James'; 'CiscosupportUpuck'
  > Subject: RE: [cisco-voip] Ccm forward call
  > 
  > Of course path pruning would bring up obvious questions about who pays
  toll
  > rates once call control is released ;)
  > 
  > -----Original Message-----
  > From: cisco-voip-bounces at puck.nether.net
  > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mark Snow
  > Sent: Friday, December 16, 2005 4:55 PM
  > To: 'James'; 'CiscosupportUpuck'
  > Subject: RE: [cisco-voip] Ccm forward call
  > 
  > That call will actually tie up two ports (incoming and outgoing) and yes
  > incur a charge.
  > 
  > Wouldn't it be nice if the Carrier's supported Q.SIG Anti-Trumboning ;)
  > 
  > 
  > -Mark Snow
  > CCIE Voice # 14073
  > 
  > > -----Original Message-----
  > > From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
  > > bounces at puck.nether.net] On Behalf Of James
  > > Sent: Friday, December 16, 2005 7:44 PM
  > > To: CiscosupportUpuck
  > > Subject: [cisco-voip] Ccm forward call
  > >
  > > I have dids configured on our cm that r configured to forward to an
  > outside line.
  > > My question is if I call into that did from my cell and it forwards me
  to
  > a ld.outside
  > > number. will that call tie up a port as long as the call is connected.
  > And will the
  > > charges go on our pri
  > > James D. Grace
  > > System Engineer
  > > CCNP CCNA MCSE MCDBA
  > > Digitel Corp.
  > > _______________________________________________
  > > 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 lelio at uoguelph.ca  Sat Dec 17 00:35:18 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat Dec 17 00:35:13 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
References: <43a3971b.067bec8e.021e.77ea@mx.gmail.com>
Message-ID: <002901c602cb$ab108100$ca01a8c0@deadfish>

You cannot do a debug isdn, but you can import the CCM trace files into the q931 debug tool and read the output that way. It's a pretty good tool and I've used it a few times. That's the standalone tool, not the one that is built into CallManager.

And our numbering plans are set to CallManager.

Frank - are you in Canada? What is your provider? Bell? I can provide with all of our 6608 parameters if that would help.
  ----- Original Message ----- 
  From: Mark Snow 
  To: 'Cisco' ; cisco-voip at puck.nether.net 
  Sent: Friday, December 16, 2005 11:41 PM
  Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...


  Lelio is correct, make sure that digit manipulation is not occurring at the Route Pattern, OR within the Route Groups inside the Route Lists.

   

  If those are not stripping anything but PreDot then try this:

  Try changing:

    a.. Called Numbering Plan*: unknown 
    b.. Calling Numbering Plan*: unknown 
   

  Then test your calls - see if they work - you will have to do a reset on the 6608 gateway first, so make sure that there are no calls going on at the time.

   

  Also, since you are running a 6608 blade on a 6500 Cat, you cannot do the 'debug isdn q931' command I asked about previously.

   

  LMK if that helps .

   

  And hey, if you do want to become a Voice Expert soon, check out my class sometime:

  http://www.ipexpert.com/products_services/product.asp?sku=ip0012

   

  -Mark Snow

  CCIE Voice # 14073

   

   


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

  From: Cisco [mailto:subs-cisco at cbvcollections.com] 
  Sent: Friday, December 16, 2005 8:17 PM
  To: Mark Snow; cisco-voip at puck.nether.net
  Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

   

  Well I'm far from the telephony expert I want to be (at this point - give me a few months) but it would appear that you may be on to something.  I took a look at the gateway configurations within CallManager.  Each of the gateways within the PSTN group (Product : Cisco Catalyst 6000 T1 VoIP Gateway - Device Protocol: Digital Access PRI) have the same configuration.  The following settings are currently used from the Call Routing Information (Outbound):

    a.. Calling Line ID Presentation*: Restricted 
    b.. Calling Party Selection*: Originator 
    c.. Called party IE number type unknown*: Cisco CallManager 
    d.. Calling party IE number type unknown*: National 
    e.. Called Numbering Plan*: Cisco CallManager 
    f.. Calling Numbering Plan*: ISDN 
    g.. Number of digits to strip*: 0 
    h.. Caller ID DN: 
    i.. SMDI Base Port*: 
  Do you think the underlined (above) is causing the issue?  Is this something I can change (to "Cisco CallManager") or do I need to involve our provider?

   


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

  From: Mark Snow [mailto:highspeedsnow at gmail.com] 
  Sent: December 16, 2005 4:59 PM
  To: Cisco; cisco-voip at puck.nether.net
  Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

  Is it a PRI that the call is going out of on an IOS GW?

  If so, have you done a 'debug isdn q931' to ensure that in fact the CCM is dropping the 1?

   

  -Mark Snow

  CCIE Voice # 14073

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
  Sent: Friday, December 16, 2005 7:24 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...

   

  I'm trying to use route filters (area code == 204 OR etc.) with a "9.@" pattern instead of the more difficult to manage 9.1204XXXXXXX to route long distance calls.  

   

  The pattern/filter work fine, the problem is that when the number is matched using 9.@ pattern (verified with the Cisco the dialled number analyser), by the time it reaches the local provider somehow the "1" is dropped and the caller is presented with a message "you must dial 1 to access a long distance number" from the local provider.  If I change the pattern back to 9.1XXXXXXXXXX then the number is dialled as expected.

   

  I've used the dialled number analyser to ensure that no transformations are taking place at any point along the route to the PEST gateway.  According to the analyser the dialled number that is sent to the PEST gateway in either case is the same.  However, one works and the other does not.

   

  I should also note that the 9.@ pattern with a filter of (local area code == 604 OR etc.) for local 10 digit dialling works just fine as either 9.@ or 9.XXXXXXXXXX.  Both are routed as expected.

   

  Ideas?


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


             Frank Wakelin
              Senior Network/Security Analyst
             
              100 - 814 Richards St.
              Vancouver, BC
              V6B 3A7
             
              Telephone: (604) 661-7906
              fwakelin at cbvcollections.com
             

       
   

  Confidentiality Warning:
  This message and any attachments are intended only for the use of the intended recipients (cisco-voip at puck.nether.net), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender (Cisco) immediately by return e-mail, and delete this message and any attachments from your system. Thank you. 

  Information Confidentielle:
  Le present message, ainsi que tout fichier qui y est joint, est envoye a l'intention exclusive de son ou de ses destinataires (cisco-voip at puck.nether.net); il est de nature confidentielle et peut constituer une information privilegiee. Nous avertissons toute personne autre que le destinataire prevu que tout examen, reacheminement, impression, copie, distribution ou autre utilisation de ce message et de tout fichier qui y est joint est strictement interdit. Si vous n'etes pas le destinataire prevu, veuillez en aviser immediatement l'expediteur (Cisco) par retour de courriel et supprimer ce message et tout document joint de votre systeme. Merci



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


  _______________________________________________
  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 subs-cisco at cbvcollections.com  Sat Dec 17 01:58:28 2005
From: subs-cisco at cbvcollections.com (Cisco)
Date: Sat Dec 17 01:59:09 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
Message-ID: <9AEF8665DE74FA43B3B987FB931E92AE0ADA47@CBVVANEM01.cbv.dns>

Yes, I'm with Bell (West) in Vancouver.  We have another CM cluster in
the east (Toronto) who is also with Bell.  The cluster in Toronto is
using a different PRI protocol (DMS-100) then the Vancouver is using
(NI2).
 
Well it is late at night so I've been playing with the gateway configs
and trying your suggestion.  After changing them to unknown - it worked!
Can you tell me why?  Also what should the setting for "Called party IE
number type unknown*" be?
 
P.S.  I will definitely look into your site/course offering!

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Friday, December 16, 2005 9:35 PM
To: Mark Snow; Cisco; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager Route Pattern/Filter Problem...


You cannot do a debug isdn, but you can import the CCM trace files into
the q931 debug tool and read the output that way. It's a pretty good
tool and I've used it a few times. That's the standalone tool, not the
one that is built into CallManager.
 
And our numbering plans are set to CallManager.
 
Frank - are you in Canada? What is your provider? Bell? I can provide
with all of our 6608 parameters if that would help.

	----- Original Message ----- 
	From: Mark Snow <mailto:highspeedsnow at gmail.com>  
	To: 'Cisco' <mailto:subs-cisco at cbvcollections.com>  ;
cisco-voip at puck.nether.net 
	Sent: Friday, December 16, 2005 11:41 PM
	Subject: RE: [cisco-voip] CallManager Route Pattern/Filter
Problem...


	Lelio is correct, make sure that digit manipulation is not
occurring at the Route Pattern, OR within the Route Groups inside the
Route Lists.

	 

	If those are not stripping anything but PreDot then try this:

	Try changing:

	*	Called Numbering Plan*: unknown 
	*	Calling Numbering Plan*: unknown 

	 

	Then test your calls - see if they work - you will have to do a
reset on the 6608 gateway first, so make sure that there are no calls
going on at the time.

	 

	Also, since you are running a 6608 blade on a 6500 Cat, you
cannot do the 'debug isdn q931' command I asked about previously.

	 

	LMK if that helps ...

	 

	And hey, if you do want to become a Voice Expert soon, check out
my class sometime:

	http://www.ipexpert.com/products_services/product.asp?sku=ip0012

	 

	-Mark Snow

	CCIE Voice # 14073

	 

	 

	
________________________________


	From: Cisco [mailto:subs-cisco at cbvcollections.com] 
	Sent: Friday, December 16, 2005 8:17 PM
	To: Mark Snow; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	 

	Well I'm far from the telephony expert I want to be (at this
point - give me a few months) but it would appear that you may be on to
something.  I took a look at the gateway configurations within
CallManager.  Each of the gateways within the PSTN group (Product :
Cisco Catalyst 6000 T1 VoIP Gateway - Device Protocol: Digital Access
PRI) have the same configuration.  The following settings are currently
used from the Call Routing Information (Outbound):

	*	Calling Line ID Presentation*: Restricted 
	*	Calling Party Selection*: Originator 
	*	Called party IE number type unknown*: Cisco CallManager 
	*	Calling party IE number type unknown*: National 
	*	Called Numbering Plan*: Cisco CallManager 
	*	Calling Numbering Plan*: ISDN 
	*	Number of digits to strip*: 0 
	*	Caller ID DN: 
	*	SMDI Base Port*: 

	Do you think the underlined (above) is causing the issue?  Is
this something I can change (to "Cisco CallManager") or do I need to
involve our provider?

	 

	
________________________________


	From: Mark Snow [mailto:highspeedsnow at gmail.com] 
	Sent: December 16, 2005 4:59 PM
	To: Cisco; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	Is it a PRI that the call is going out of on an IOS GW?

	If so, have you done a 'debug isdn q931' to ensure that in fact
the CCM is dropping the 1?

	 

	-Mark Snow

	CCIE Voice # 14073

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
	Sent: Friday, December 16, 2005 7:24 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	 

	I'm trying to use route filters (area code == 204 OR etc.) with
a "9.@" pattern instead of the more difficult to manage 9.1204XXXXXXX to
route long distance calls.  

	 

	The pattern/filter work fine, the problem is that when the
number is matched using 9.@ pattern (verified with the Cisco the dialled
number analyser), by the time it reaches the local provider somehow the
"1" is dropped and the caller is presented with a message "you must dial
1 to access a long distance number" from the local provider.  If I
change the pattern back to 9.1XXXXXXXXXX then the number is dialled as
expected.

	 

	I've used the dialled number analyser to ensure that no
transformations are taking place at any point along the route to the
PEST gateway.  According to the analyser the dialled number that is sent
to the PEST gateway in either case is the same.  However, one works and
the other does not.

	 

	I should also note that the 9.@ pattern with a filter of (local
area code == 604 OR etc.) for local 10 digit dialling works just fine as
either 9.@ or 9.XXXXXXXXXX.  Both are routed as expected.

	 

	Ideas?

	
________________________________

 <http://www.cbvcollections.com/cbvImages/ecardlogo.jpg> 

Frank Wakelin
Senior Network/Security Analyst

100 - 814 Richards St.
Vancouver, BC
V6B 3A7

Telephone: (604) 661-7906
fwakelin at cbvcollections.com

	 

	Confidentiality Warning:
	This message and any attachments are intended only for the use
of the intended recipients (cisco-voip at puck.nether.net), are
confidential, and may be privileged. If you are not the intended
recipient, you are hereby notified that any review, retransmission,
conversion to hard copy, copying, circulation or other use of this
message and any attachments is strictly prohibited. If you are not the
intended recipient, please notify the sender (Cisco
<mailto:subs-cisco at cbvcollections.com> ) immediately by return e-mail,
and delete this message and any attachments from your system. Thank you.


	Information Confidentielle:
	Le present message, ainsi que tout fichier qui y est joint, est
envoye a l'intention exclusive de son ou de ses destinataires
(cisco-voip at puck.nether.net); il est de nature confidentielle et peut
constituer une information privilegiee. Nous avertissons toute personne
autre que le destinataire prevu que tout examen, reacheminement,
impression, copie, distribution ou autre utilisation de ce message et de
tout fichier qui y est joint est strictement interdit. Si vous n'etes
pas le destinataire prevu, veuillez en aviser immediatement l'expediteur
(Cisco <mailto:subs-cisco at cbvcollections.com> ) par retour de courriel
et supprimer ce message et tout document joint de votre systeme. Merci

	
________________________________


	

	_______________________________________________
	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 subs-cisco at cbvcollections.com  Sat Dec 17 03:23:25 2005
From: subs-cisco at cbvcollections.com (Cisco)
Date: Sat Dec 17 03:26:13 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
Message-ID: <9AEF8665DE74FA43B3B987FB931E92AE0ADA48@CBVVANEM01.cbv.dns>

Argh - one more thing.  We have a satellite office that uses the same CM
cluster which experienced the same problem - with called numbering plan
on CallManager all calls matching route pattern 9.@ that included 1
played the carrier message (you must dial 1 first...).  
 
I tried adjusting the called/calling party numbering plans to unknown as
I did the local 6608 gateways.  Once the gateway was adjusted all
dialled numbers called that are matched 9.@ that include the "1" were
routed to the gateway, but then dropped.  The main different there is
that the gateway is a 2691. 
 
Thoughts - ideas?

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
Sent: Friday, December 16, 2005 10:58 PM
To: Lelio Fulgenzi; Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...


Yes, I'm with Bell (West) in Vancouver.  We have another CM cluster in
the east (Toronto) who is also with Bell.  The cluster in Toronto is
using a different PRI protocol (DMS-100) then the Vancouver is using
(NI2).
 
Well it is late at night so I've been playing with the gateway configs
and trying your suggestion.  After changing them to unknown - it worked!
Can you tell me why?  Also what should the setting for "Called party IE
number type unknown*" be?
 
P.S.  I will definitely look into your site/course offering!

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Friday, December 16, 2005 9:35 PM
To: Mark Snow; Cisco; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager Route Pattern/Filter Problem...


You cannot do a debug isdn, but you can import the CCM trace files into
the q931 debug tool and read the output that way. It's a pretty good
tool and I've used it a few times. That's the standalone tool, not the
one that is built into CallManager.
 
And our numbering plans are set to CallManager.
 
Frank - are you in Canada? What is your provider? Bell? I can provide
with all of our 6608 parameters if that would help.

	----- Original Message ----- 
	From: Mark Snow <mailto:highspeedsnow at gmail.com>  
	To: 'Cisco' <mailto:subs-cisco at cbvcollections.com>  ;
cisco-voip at puck.nether.net 
	Sent: Friday, December 16, 2005 11:41 PM
	Subject: RE: [cisco-voip] CallManager Route Pattern/Filter
Problem...


	Lelio is correct, make sure that digit manipulation is not
occurring at the Route Pattern, OR within the Route Groups inside the
Route Lists.

	 

	If those are not stripping anything but PreDot then try this:

	Try changing:

	*	Called Numbering Plan*: unknown 
	*	Calling Numbering Plan*: unknown 

	 

	Then test your calls - see if they work - you will have to do a
reset on the 6608 gateway first, so make sure that there are no calls
going on at the time.

	 

	Also, since you are running a 6608 blade on a 6500 Cat, you
cannot do the 'debug isdn q931' command I asked about previously.

	 

	LMK if that helps ...

	 

	And hey, if you do want to become a Voice Expert soon, check out
my class sometime:

	http://www.ipexpert.com/products_services/product.asp?sku=ip0012

	 

	-Mark Snow

	CCIE Voice # 14073

	 

	 

	
________________________________


	From: Cisco [mailto:subs-cisco at cbvcollections.com] 
	Sent: Friday, December 16, 2005 8:17 PM
	To: Mark Snow; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	 

	Well I'm far from the telephony expert I want to be (at this
point - give me a few months) but it would appear that you may be on to
something.  I took a look at the gateway configurations within
CallManager.  Each of the gateways within the PSTN group (Product :
Cisco Catalyst 6000 T1 VoIP Gateway - Device Protocol: Digital Access
PRI) have the same configuration.  The following settings are currently
used from the Call Routing Information (Outbound):

	*	Calling Line ID Presentation*: Restricted 
	*	Calling Party Selection*: Originator 
	*	Called party IE number type unknown*: Cisco CallManager 
	*	Calling party IE number type unknown*: National 
	*	Called Numbering Plan*: Cisco CallManager 
	*	Calling Numbering Plan*: ISDN 
	*	Number of digits to strip*: 0 
	*	Caller ID DN: 
	*	SMDI Base Port*: 

	Do you think the underlined (above) is causing the issue?  Is
this something I can change (to "Cisco CallManager") or do I need to
involve our provider?

	 

	
________________________________


	From: Mark Snow [mailto:highspeedsnow at gmail.com] 
	Sent: December 16, 2005 4:59 PM
	To: Cisco; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	Is it a PRI that the call is going out of on an IOS GW?

	If so, have you done a 'debug isdn q931' to ensure that in fact
the CCM is dropping the 1?

	 

	-Mark Snow

	CCIE Voice # 14073

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
	Sent: Friday, December 16, 2005 7:24 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	 

	I'm trying to use route filters (area code == 204 OR etc.) with
a "9.@" pattern instead of the more difficult to manage 9.1204XXXXXXX to
route long distance calls.  

	 

	The pattern/filter work fine, the problem is that when the
number is matched using 9.@ pattern (verified with the Cisco the dialled
number analyser), by the time it reaches the local provider somehow the
"1" is dropped and the caller is presented with a message "you must dial
1 to access a long distance number" from the local provider.  If I
change the pattern back to 9.1XXXXXXXXXX then the number is dialled as
expected.

	 

	I've used the dialled number analyser to ensure that no
transformations are taking place at any point along the route to the
PEST gateway.  According to the analyser the dialled number that is sent
to the PEST gateway in either case is the same.  However, one works and
the other does not.

	 

	I should also note that the 9.@ pattern with a filter of (local
area code == 604 OR etc.) for local 10 digit dialling works just fine as
either 9.@ or 9.XXXXXXXXXX.  Both are routed as expected.

	 

	Ideas?

	
________________________________

 <http://www.cbvcollections.com/cbvImages/ecardlogo.jpg> 

Frank Wakelin
Senior Network/Security Analyst

100 - 814 Richards St.
Vancouver, BC
V6B 3A7

Telephone: (604) 661-7906
fwakelin at cbvcollections.com

	 

	Confidentiality Warning:
	This message and any attachments are intended only for the use
of the intended recipients (cisco-voip at puck.nether.net), are
confidential, and may be privileged. If you are not the intended
recipient, you are hereby notified that any review, retransmission,
conversion to hard copy, copying, circulation or other use of this
message and any attachments is strictly prohibited. If you are not the
intended recipient, please notify the sender (Cisco
<mailto:subs-cisco at cbvcollections.com> ) immediately by return e-mail,
and delete this message and any attachments from your system. Thank you.


	Information Confidentielle:
	Le present message, ainsi que tout fichier qui y est joint, est
envoye a l'intention exclusive de son ou de ses destinataires
(cisco-voip at puck.nether.net); il est de nature confidentielle et peut
constituer une information privilegiee. Nous avertissons toute personne
autre que le destinataire prevu que tout examen, reacheminement,
impression, copie, distribution ou autre utilisation de ce message et de
tout fichier qui y est joint est strictement interdit. Si vous n'etes
pas le destinataire prevu, veuillez en aviser immediatement l'expediteur
(Cisco <mailto:subs-cisco at cbvcollections.com> ) par retour de courriel
et supprimer ce message et tout document joint de votre systeme. Merci

	
________________________________


	

	_______________________________________________
	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 highspeedsnow at gmail.com  Sat Dec 17 08:42:23 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sat Dec 17 08:46:47 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
In-Reply-To: <9AEF8665DE74FA43B3B987FB931E92AE0ADA47@CBVVANEM01.cbv.dns>
Message-ID: <43a416c5.20fdbfdb.41a0.ffffead6@mx.gmail.com>

The problem lies with the fact that some carriers do not want to see a 1 and
the Number Plan/Type ISDN/National - same goes with International calls -
some switches just puke if you send them both, however send them one or the
other and they are just fine.

 

Basically it is a programming problem in your local 5ESS - But the carrier
would never admit that ;)

 

  _____  

From: Cisco [mailto:subs-cisco at cbvcollections.com] 
Sent: Saturday, December 17, 2005 1:58 AM
To: Lelio Fulgenzi; Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

Yes, I'm with Bell (West) in Vancouver.  We have another CM cluster in the
east (Toronto) who is also with Bell.  The cluster in Toronto is using a
different PRI protocol (DMS-100) then the Vancouver is using (NI2).

 

Well it is late at night so I've been playing with the gateway configs and
trying your suggestion.  After changing them to unknown - it worked!  Can
you tell me why?  Also what should the setting for "Called party IE number
type unknown*" be?

 

P.S.  I will definitely look into your site/course offering!

 

  _____  

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Friday, December 16, 2005 9:35 PM
To: Mark Snow; Cisco; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager Route Pattern/Filter Problem...

You cannot do a debug isdn, but you can import the CCM trace files into the
q931 debug tool and read the output that way. It's a pretty good tool and
I've used it a few times. That's the standalone tool, not the one that is
built into CallManager.

 

And our numbering plans are set to CallManager.

 

Frank - are you in Canada? What is your provider? Bell? I can provide with
all of our 6608 parameters if that would help.

----- Original Message ----- 

From: Mark Snow <mailto:highspeedsnow at gmail.com>  

To: 'Cisco' <mailto:subs-cisco at cbvcollections.com>  ;
cisco-voip at puck.nether.net 

Sent: Friday, December 16, 2005 11:41 PM

Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

Lelio is correct, make sure that digit manipulation is not occurring at the
Route Pattern, OR within the Route Groups inside the Route Lists.

 

If those are not stripping anything but PreDot then try this:

Try changing:

*	Called Numbering Plan*: unknown 
*	Calling Numbering Plan*: unknown 

 

Then test your calls - see if they work - you will have to do a reset on the
6608 gateway first, so make sure that there are no calls going on at the
time.

 

Also, since you are running a 6608 blade on a 6500 Cat, you cannot do the
'debug isdn q931' command I asked about previously.

 

LMK if that helps .

 

And hey, if you do want to become a Voice Expert soon, check out my class
sometime:

http://www.ipexpert.com/products_services/product.asp?sku=ip0012

 

-Mark Snow

CCIE Voice # 14073

 

 


  _____  


From: Cisco [mailto:subs-cisco at cbvcollections.com] 
Sent: Friday, December 16, 2005 8:17 PM
To: Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

Well I'm far from the telephony expert I want to be (at this point - give me
a few months) but it would appear that you may be on to something.  I took a
look at the gateway configurations within CallManager.  Each of the gateways
within the PSTN group (Product : Cisco Catalyst 6000 T1 VoIP Gateway -
Device Protocol: Digital Access PRI) have the same configuration.  The
following settings are currently used from the Call Routing Information
(Outbound):

*	Calling Line ID Presentation*: Restricted 
*	Calling Party Selection*: Originator 
*	Called party IE number type unknown*: Cisco CallManager 
*	Calling party IE number type unknown*: National 
*	Called Numbering Plan*: Cisco CallManager 
*	Calling Numbering Plan*: ISDN 
*	Number of digits to strip*: 0 
*	Caller ID DN: 
*	SMDI Base Port*: 

Do you think the underlined (above) is causing the issue?  Is this something
I can change (to "Cisco CallManager") or do I need to involve our provider?

 


  _____  


From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: December 16, 2005 4:59 PM
To: Cisco; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

Is it a PRI that the call is going out of on an IOS GW?

If so, have you done a 'debug isdn q931' to ensure that in fact the CCM is
dropping the 1?

 

-Mark Snow

CCIE Voice # 14073

 


  _____  


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
Sent: Friday, December 16, 2005 7:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

I'm trying to use route filters (area code == 204 OR etc.) with a "9.@"
pattern instead of the more difficult to manage 9.1204XXXXXXX to route long
distance calls.  

 

The pattern/filter work fine, the problem is that when the number is matched
using 9.@ pattern (verified with the Cisco the dialled number analyser), by
the time it reaches the local provider somehow the "1" is dropped and the
caller is presented with a message "you must dial 1 to access a long
distance number" from the local provider.  If I change the pattern back to
9.1XXXXXXXXXX then the number is dialled as expected.

 

I've used the dialled number analyser to ensure that no transformations are
taking place at any point along the route to the PEST gateway.  According to
the analyser the dialled number that is sent to the PEST gateway in either
case is the same.  However, one works and the other does not.

 

I should also note that the 9.@ pattern with a filter of (local area code ==
604 OR etc.) for local 10 digit dialling works just fine as either 9.@ or
9.XXXXXXXXXX.  Both are routed as expected.

 

Ideas?


  _____  



  <http://www.cbvcollections.com/cbvImages/ecardlogo.jpg> 

Frank Wakelin
Senior Network/Security Analyst


100 - 814 Richards St.
Vancouver, BC
V6B 3A7


Telephone: (604) 661-7906
fwakelin at cbvcollections.com

 

Confidentiality Warning:
This message and any attachments are intended only for the use of the
intended recipients (cisco-voip at puck.nether.net), are confidential, and may
be privileged. If you are not the intended recipient, you are hereby
notified that any review, retransmission, conversion to hard copy, copying,
circulation or other use of this message and any attachments is strictly
prohibited. If you are not the intended recipient, please notify the sender
(Cisco <mailto:subs-cisco at cbvcollections.com> ) immediately by return
e-mail, and delete this message and any attachments from your system. Thank
you. 

Information Confidentielle:
Le present message, ainsi que tout fichier qui y est joint, est envoye a
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(cisco-voip at puck.nether.net); il est de nature confidentielle et peut
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<mailto:subs-cisco at cbvcollections.com> ) par retour de courriel et supprimer
ce message et tout document joint de votre systeme. Merci


  _____  


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From highspeedsnow at gmail.com  Sat Dec 17 08:44:25 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sat Dec 17 08:48:51 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
In-Reply-To: <9AEF8665DE74FA43B3B987FB931E92AE0ADA48@CBVVANEM01.cbv.dns>
Message-ID: <43a41740.6324e822.7bc2.ffff8e8c@mx.gmail.com>

That sattelilte most likely uses a different telco POP and the telly's
switch wants to see both - switch the Plan/Type back to ISDN/National or
CallManager/CallManager but for that GW only and you should be fine.

 

HTH

 

-Mark Snow

 

  _____  

From: Cisco [mailto:subs-cisco at cbvcollections.com] 
Sent: Saturday, December 17, 2005 3:23 AM
To: Lelio Fulgenzi; Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

Argh - one more thing.  We have a satellite office that uses the same CM
cluster which experienced the same problem - with called numbering plan on
CallManager all calls matching route pattern 9.@ that included 1 played the
carrier message (you must dial 1 first...).  

 

I tried adjusting the called/calling party numbering plans to unknown as I
did the local 6608 gateways.  Once the gateway was adjusted all dialled
numbers called that are matched 9.@ that include the "1" were routed to the
gateway, but then dropped.  The main different there is that the gateway is
a 2691. 

 

Thoughts - ideas?

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
Sent: Friday, December 16, 2005 10:58 PM
To: Lelio Fulgenzi; Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

Yes, I'm with Bell (West) in Vancouver.  We have another CM cluster in the
east (Toronto) who is also with Bell.  The cluster in Toronto is using a
different PRI protocol (DMS-100) then the Vancouver is using (NI2).

 

Well it is late at night so I've been playing with the gateway configs and
trying your suggestion.  After changing them to unknown - it worked!  Can
you tell me why?  Also what should the setting for "Called party IE number
type unknown*" be?

 

P.S.  I will definitely look into your site/course offering!

 

  _____  

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Friday, December 16, 2005 9:35 PM
To: Mark Snow; Cisco; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager Route Pattern/Filter Problem...

You cannot do a debug isdn, but you can import the CCM trace files into the
q931 debug tool and read the output that way. It's a pretty good tool and
I've used it a few times. That's the standalone tool, not the one that is
built into CallManager.

 

And our numbering plans are set to CallManager.

 

Frank - are you in Canada? What is your provider? Bell? I can provide with
all of our 6608 parameters if that would help.

----- Original Message ----- 

From: Mark Snow <mailto:highspeedsnow at gmail.com>  

To: 'Cisco' <mailto:subs-cisco at cbvcollections.com>  ;
cisco-voip at puck.nether.net 

Sent: Friday, December 16, 2005 11:41 PM

Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

Lelio is correct, make sure that digit manipulation is not occurring at the
Route Pattern, OR within the Route Groups inside the Route Lists.

 

If those are not stripping anything but PreDot then try this:

Try changing:

*	Called Numbering Plan*: unknown 
*	Calling Numbering Plan*: unknown 

 

Then test your calls - see if they work - you will have to do a reset on the
6608 gateway first, so make sure that there are no calls going on at the
time.

 

Also, since you are running a 6608 blade on a 6500 Cat, you cannot do the
'debug isdn q931' command I asked about previously.

 

LMK if that helps .

 

And hey, if you do want to become a Voice Expert soon, check out my class
sometime:

http://www.ipexpert.com/products_services/product.asp?sku=ip0012

 

-Mark Snow

CCIE Voice # 14073

 

 


  _____  


From: Cisco [mailto:subs-cisco at cbvcollections.com] 
Sent: Friday, December 16, 2005 8:17 PM
To: Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

Well I'm far from the telephony expert I want to be (at this point - give me
a few months) but it would appear that you may be on to something.  I took a
look at the gateway configurations within CallManager.  Each of the gateways
within the PSTN group (Product : Cisco Catalyst 6000 T1 VoIP Gateway -
Device Protocol: Digital Access PRI) have the same configuration.  The
following settings are currently used from the Call Routing Information
(Outbound):

*	Calling Line ID Presentation*: Restricted 
*	Calling Party Selection*: Originator 
*	Called party IE number type unknown*: Cisco CallManager 
*	Calling party IE number type unknown*: National 
*	Called Numbering Plan*: Cisco CallManager 
*	Calling Numbering Plan*: ISDN 
*	Number of digits to strip*: 0 
*	Caller ID DN: 
*	SMDI Base Port*: 

Do you think the underlined (above) is causing the issue?  Is this something
I can change (to "Cisco CallManager") or do I need to involve our provider?

 


  _____  


From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: December 16, 2005 4:59 PM
To: Cisco; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

Is it a PRI that the call is going out of on an IOS GW?

If so, have you done a 'debug isdn q931' to ensure that in fact the CCM is
dropping the 1?

 

-Mark Snow

CCIE Voice # 14073

 


  _____  


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
Sent: Friday, December 16, 2005 7:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

I'm trying to use route filters (area code == 204 OR etc.) with a "9.@"
pattern instead of the more difficult to manage 9.1204XXXXXXX to route long
distance calls.  

 

The pattern/filter work fine, the problem is that when the number is matched
using 9.@ pattern (verified with the Cisco the dialled number analyser), by
the time it reaches the local provider somehow the "1" is dropped and the
caller is presented with a message "you must dial 1 to access a long
distance number" from the local provider.  If I change the pattern back to
9.1XXXXXXXXXX then the number is dialled as expected.

 

I've used the dialled number analyser to ensure that no transformations are
taking place at any point along the route to the PEST gateway.  According to
the analyser the dialled number that is sent to the PEST gateway in either
case is the same.  However, one works and the other does not.

 

I should also note that the 9.@ pattern with a filter of (local area code ==
604 OR etc.) for local 10 digit dialling works just fine as either 9.@ or
9.XXXXXXXXXX.  Both are routed as expected.

 

Ideas?


  _____  



  <http://www.cbvcollections.com/cbvImages/ecardlogo.jpg> 

Frank Wakelin
Senior Network/Security Analyst


100 - 814 Richards St.
Vancouver, BC
V6B 3A7


Telephone: (604) 661-7906
fwakelin at cbvcollections.com

 

Confidentiality Warning:
This message and any attachments are intended only for the use of the
intended recipients (cisco-voip at puck.nether.net), are confidential, and may
be privileged. If you are not the intended recipient, you are hereby
notified that any review, retransmission, conversion to hard copy, copying,
circulation or other use of this message and any attachments is strictly
prohibited. If you are not the intended recipient, please notify the sender
(Cisco <mailto:subs-cisco at cbvcollections.com> ) immediately by return
e-mail, and delete this message and any attachments from your system. Thank
you. 

Information Confidentielle:
Le present message, ainsi que tout fichier qui y est joint, est envoye a
l'intention exclusive de son ou de ses destinataires
(cisco-voip at puck.nether.net); il est de nature confidentielle et peut
constituer une information privilegiee. Nous avertissons toute personne
autre que le destinataire prevu que tout examen, reacheminement, impression,
copie, distribution ou autre utilisation de ce message et de tout fichier
qui y est joint est strictement interdit. Si vous n'etes pas le destinataire
prevu, veuillez en aviser immediatement l'expediteur (Cisco
<mailto:subs-cisco at cbvcollections.com> ) par retour de courriel et supprimer
ce message et tout document joint de votre systeme. Merci


  _____  


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From techguy at gmail.com  Sat Dec 17 14:33:40 2005
From: techguy at gmail.com (TechGuy)
Date: Sat Dec 17 14:33:43 2005
Subject: [cisco-voip] Call rings phones once and then drops
Message-ID: <a1b07ea60512171133i79723eecs2d95643eaa5f034f@mail.gmail.com>

Calls ring in on local PRI trunks on 3845 router, makes its way to
callmanager, makes its way to phone and rings once on the phone.  But that
is it, calls instantly drop after first ring of the phone.  Does not give
person chance to answer just drops without any user intervention.

Using MGCP between 3845 and callmanager.

Any ideas?
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From techguy at gmail.com  Sat Dec 17 15:19:32 2005
From: techguy at gmail.com (TechGuy)
Date: Sat Dec 17 15:19:36 2005
Subject: [cisco-voip] Call rings phones once and then drops
Message-ID: <a1b07ea60512171219p288d1552qddda2d4820ebba47@mail.gmail.com>

Here is the debug:

emb3845#debug isdn q931
debug isdn q931 is              ON.
emb3845#
*Dec 17 20:07:23.806: ISDN Se0/1/1:23 Q931: RX <- SETUP pd = 8  callref =
0x00D2
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98381
                Exclusive, Channel 1
        Calling Party Number i = 0x2183, '3375015555'
                Plan:ISDN, Type:National
        Called Party Number i = 0x80, '4080'
                Plan:Unknown, Type:Unknown
*Dec 17 20:07:24.830: ISDN Se0/1/1:23 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x80D2
        Cause i = 0x80A9 - Temporary failure
emb3845#


More info on the problem.  This router has two local PRI trunks coming in to
a 2MFT card.  It also has two CAS (E&M Wink) trunks coming into a 2MFT card
for long distance and incoming toll free.

The two CAS trunks work fine.  We are able to get incoming calls via the CAS
trunks mentioned above with no problems.  Also, we have another gateway
configured on the same callmanager in a similiar fashion and all works
fine.  This was all working earlier this morning.  No known changes occured
between when it worked and stopped working.

Quite an interesting problem, we have TAC on the line now but thought I
would see what others think.
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From highspeedsnow at gmail.com  Sat Dec 17 15:55:50 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sat Dec 17 16:00:43 2005
Subject: [cisco-voip] Call rings phones once and then drops
In-Reply-To: <a1b07ea60512171219p288d1552qddda2d4820ebba47@mail.gmail.com>
Message-ID: <43a47c79.7ea88ad1.4117.ffffa263@mx.gmail.com>

Could very well be an ISND Timer issue - try adjusting your T304 and T310
timers up a bit - also maybe (far stretch) but maybe adjust your T309 timer
up..

 

-Mark Snow

CCIE Voice# 14073

 

 

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Saturday, December 17, 2005 3:20 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Call rings phones once and then drops

 

Here is the debug:

 

emb3845#debug isdn q931
debug isdn q931 is              ON.
emb3845#
*Dec 17 20:07:23.806: ISDN Se0/1/1:23 Q931: RX <- SETUP pd = 8  callref =
0x00D2 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA98381 
                Exclusive, Channel 1 
        Calling Party Number i = 0x2183, '3375015555' 
                Plan:ISDN, Type:National 
        Called Party Number i = 0x80, '4080' 
                Plan:Unknown, Type:Unknown
*Dec 17 20:07:24.830: ISDN Se0/1/1:23 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x80D2 
        Cause i = 0x80A9 - Temporary failure
emb3845#

 

 

More info on the problem.  This router has two local PRI trunks coming in to
a 2MFT card.  It also has two CAS (E&M Wink) trunks coming into a 2MFT card
for long distance and incoming toll free.

 

The two CAS trunks work fine.  We are able to get incoming calls via the CAS
trunks mentioned above with no problems.  Also, we have another gateway
configured on the same callmanager in a similiar fashion and all works fine.
This was all working earlier this morning.  No known changes occured between
when it worked and stopped working. 

 

Quite an interesting problem, we have TAC on the line now but thought I
would see what others think.

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From ciscozest at gmail.com  Sun Dec 18 05:54:33 2005
From: ciscozest at gmail.com (Ciscozest)
Date: Sun Dec 18 05:54:40 2005
Subject: [cisco-voip] Tech Prefix from ATA186
References: <43A1B0DB.4030303@harvard.edu>
Message-ID: <000b01c603c1$70d00410$6301a8c0@windfall99>

Hi,
I want to register my ATA to GK with tech-refix 1#. I had used "pwd0" = "1#", but when I did "show gatekeper 
gw-type-prefix", it didn;t shown the tech prefix from ATA.
Can anyone help? Thanks

regards,
Robin 


From tgeorgiev at is-bg.net  Sun Dec 18 09:24:00 2005
From: tgeorgiev at is-bg.net (Teodor Georgiev)
Date: Sun Dec 18 09:24:09 2005
Subject: [cisco-voip] Call rings phones once and then drops
References: <a1b07ea60512171133i79723eecs2d95643eaa5f034f@mail.gmail.com>
Message-ID: <002f01c603de$b1356f50$978e4bd9@tg>


do a debug and see the codec negotation. Looks like the devices can not negotiate on the codecs used.

  ----- Original Message ----- 
  From: TechGuy 
  To: cisco-voip at puck.nether.net 
  Sent: Saturday, December 17, 2005 9:33 PM
  Subject: [cisco-voip] Call rings phones once and then drops


  Calls ring in on local PRI trunks on 3845 router, makes its way to callmanager, makes its way to phone and rings once on the phone.  But that is it, calls instantly drop after first ring of the phone.  Does not give person chance to answer just drops without any user intervention. 

  Using MGCP between 3845 and callmanager.

  Any ideas?


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


  _______________________________________________
  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  Sun Dec 18 12:27:31 2005
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Sun Dec 18 12:27:34 2005
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 95
In-Reply-To: <200512181700.jBIH07KQ093009@puck.nether.net>
References: <200512181700.jBIH07KQ093009@puck.nether.net>
Message-ID: <f1b65f880512180927l10179306w@mail.gmail.com>

Try reregistering the PRI via the gateway page in CCMADMIN.  I had a
very similiar problem albeit it was occurring on a T1 CAS but
resetting the gateway resolved the problem.

Are you running CCM3.3(3)sr4a ?

> Message: 1
> Date: Sat, 17 Dec 2005 13:33:40 -0600
> From: TechGuy <techguy at gmail.com>
> Subject: [cisco-voip] Call rings phones once and then drops
> To: cisco-voip at puck.nether.net
> Message-ID:
>         <a1b07ea60512171133i79723eecs2d95643eaa5f034f at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Calls ring in on local PRI trunks on 3845 router, makes its way to
> callmanager, makes its way to phone and rings once on the phone.  But that
> is it, calls instantly drop after first ring of the phone.  Does not give
> person chance to answer just drops without any user intervention.
>
> Using MGCP between 3845 and callmanager.
>
> Any ideas?


From bababooey at cox.net  Sun Dec 18 18:45:28 2005
From: bababooey at cox.net (Bob A. Bowie)
Date: Sun Dec 18 18:45:38 2005
Subject: [cisco-voip] CME - unknown umber
References: <200512181700.jBIH07KG093009@puck.nether.net>
Message-ID: <000301c6042d$2131d070$0f0110ac@corporation2>

When I reboot my CME router and after the IP Phones have their load and
are active, I dial a number as defined by a dial peer, I receive an unknown 
number message on the phone.
After another minute or two the number works.  What causes the delay, is 
there any way to improve this?

Thanks,
Bob




From rratliff at cisco.com  Sun Dec 18 20:46:42 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Sun Dec 18 20:46:44 2005
Subject: [cisco-voip] Attendant Console Errors
In-Reply-To: <170BC64B51964949A28FE8A48EF1D4EBFBEF62@corpexchsrv01.adomo.com>
References: <170BC64B51964949A28FE8A48EF1D4EBFBEF62@corpexchsrv01.adomo.com>
Message-ID: <5789E04A-BFB5-4B2C-91FD-6E56D5631A7F@cisco.com>

Is your hunt pilot an AC or CM controlled pilot?
The AC pilot is created via Service->Callmanager Attendant Console- 
 >Pilot Point.   The CM pilot is done via Route Plan->Route/Hunt- 
 >Hunt Pilot.

You don't need CTI Super Provider, might as well take it off the user.

What are your hunt group members?  The AC hunt group will either have  
AC user line numbers or dn/partition combinations.    If it only has  
AC user line numbers then you'll need to have an AC user with the  
console up and running.

-Ryan

On Dec 16, 2005, at 6:06 PM, Paul Yago wrote:

Thanks for the response Ryan.

I have a user named "ac" with the following checked.
1) Enable CTI Application Use
2) Enable CTI Super Provider
3) Call Park

And the pilot point and 2 hunt group members are all associated with
"ac". Password and Pin are 12345.

I'm still getting the fast busy though when calling the pilot point. Any
other suggestions?

Thanks,
Paul



-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Friday, December 16, 2005 12:09 PM
To: Paul Yago
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant Console Errors

The AC user is used by Telephony Call Dispatcher to logon to CTI.  It
does not need to be configured other than be in the directory with
"CTI Use" and "Call Park Retrieval Allowed" check boxes ticked.  The
password and PIN should be 12345 and the device associations should
be any AC hunt pilot and any phone to be controlled by the people
using the AC desktop client.

AC does not need to be an Attendant Console user.

-Ryan

On Dec 16, 2005, at 2:46 PM, Paul Yago wrote:

Hello, does anyone with Attendant Console experience know why I'm
seeing this error (below)?



I'm trying to configure a hunt group with 2 DNs and 1 pilot point. I
created an "ac" user as both an Attendant Console User and CCM user,
with association to both DNs and the Pilot Point. In the
documentation, it's not clear to me how the "ac" user should be used.
Does he need to be part of the AC hunt group?



Any help is very much appreciated.

Thanks.

Paul







50621 12/16 11:35:22.806 ERROR HuntGroups:

pkid: 0xAE5672C0728C86458CAE73F87025E57E, name: New_JTAPI, dn: 6500,
fkDevice: 0x8FCBD68266333C4D98325062886C69DB, fkCSS:
0x5563A64BB3195843A5E2A5FBD113AFE6, huntingAlgorithm: 1,
dbHuntingAlgorithm: 1, owner: 0, active: 2, queueEnabled: false,
queueSize: 32, holdTime: 0, CMs: CM_LABCM01-

Lines:

       DN: 5750, AlwaysRoute: true, Online: true, LineActive: true,
RoutingCall: false, Waiting: false, LS: Active, Idle Time:
1134697714514, RouteUnknown: true

       DN: 7017, AlwaysRoute: true, Online: true, LineActive: true,
RoutingCall: false, Waiting: false, LS: Idle, Idle Time:
1134697737624, RouteUnknown: true

HuntGroupsUsers:

pkid: 0xAF326A961CFAEF419870B74D4BB27478, userid: ac, passwd: *****

Sessions:

Active Sessions:

Deactive Sessions:

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


From it at cimgroup.com  Sun Dec 18 21:34:54 2005
From: it at cimgroup.com (IT)
Date: Sun Dec 18 21:30:54 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <B487D4C1FE6D04439061018C6EBC109B01EB0E14@exchange.cimgroup.com>

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their video conference needs. I currently have a Polycom Viewstation EX, but with the upcoming IP/VC installation, I didnt know if buying 3 more of those was the right idea. What is everyone using out there for video conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting MeetingPlace really help in small conferences?

thanks for the advice.
avidan

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
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From highspeedsnow at gmail.com  Sun Dec 18 22:12:50 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Sun Dec 18 22:12:54 2005
Subject: [cisco-voip] Tech Prefix from ATA186
In-Reply-To: <000b01c603c1$70d00410$6301a8c0@windfall99>
Message-ID: <43a62534.1d098763.2245.ffffb0f0@mx.gmail.com>

You also need to set the following field: 

AutMethod = 0x00000004

-Mark Snow
CCIE Voice# 14073

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Ciscozest
> Sent: Sunday, December 18, 2005 5:55 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Tech Prefix from ATA186
> 
> Hi,
> I want to register my ATA to GK with tech-refix 1#. I had used "pwd0" =
"1#",
> but when I did "show gatekeper
> gw-type-prefix", it didn;t shown the tech prefix from ATA.
> Can anyone help? Thanks
> 
> regards,
> Robin
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From pnowicki at neo.rr.com  Mon Dec 19 01:00:26 2005
From: pnowicki at neo.rr.com (pnowicki@neo.rr.com)
Date: Mon Dec 19 01:00:32 2005
Subject: [cisco-voip] Attendant Console Errors
Message-ID: <bc010ebbb442.bbb442bc010e@columbus.rr.com>

A lot of times, I'll also have to stop and restart the TCD service in 
order to get it to kick in...

----- Original Message -----
From: Ryan Ratliff <rratliff at cisco.com>
Date: Sunday, December 18, 2005 8:46 pm
Subject: Re: [cisco-voip] Attendant Console Errors

> Is your hunt pilot an AC or CM controlled pilot?
> The AC pilot is created via Service->Callmanager Attendant Console-
> 
> >Pilot Point.   The CM pilot is done via Route Plan->Route/Hunt- 
> >Hunt Pilot.
> 
> You don't need CTI Super Provider, might as well take it off the 
user.
> 
> What are your hunt group members?  The AC hunt group will either 
> have  
> AC user line numbers or dn/partition combinations.    If it only 
> has  
> AC user line numbers then you'll need to have an AC user with the  
> console up and running.
> 
> -Ryan
> 
> On Dec 16, 2005, at 6:06 PM, Paul Yago wrote:
> 
> Thanks for the response Ryan.
> 
> I have a user named "ac" with the following checked.
> 1) Enable CTI Application Use
> 2) Enable CTI Super Provider
> 3) Call Park
> 
> And the pilot point and 2 hunt group members are all associated with
> "ac". Password and Pin are 12345.
> 
> I'm still getting the fast busy though when calling the pilot 
> point. Any
> other suggestions?
> 
> Thanks,
> Paul
> 
> 
> 
> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> Sent: Friday, December 16, 2005 12:09 PM
> To: Paul Yago
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Attendant Console Errors
> 
> The AC user is used by Telephony Call Dispatcher to logon to CTI.  It
> does not need to be configured other than be in the directory with
> "CTI Use" and "Call Park Retrieval Allowed" check boxes ticked.  The
> password and PIN should be 12345 and the device associations should
> be any AC hunt pilot and any phone to be controlled by the people
> using the AC desktop client.
> 
> AC does not need to be an Attendant Console user.
> 
> -Ryan
> 
> On Dec 16, 2005, at 2:46 PM, Paul Yago wrote:
> 
> Hello, does anyone with Attendant Console experience know why I'm
> seeing this error (below)?
> 
> 
> 
> I'm trying to configure a hunt group with 2 DNs and 1 pilot point. I
> created an "ac" user as both an Attendant Console User and CCM user,
> with association to both DNs and the Pilot Point. In the
> documentation, it's not clear to me how the "ac" user should be used.
> Does he need to be part of the AC hunt group?
> 
> 
> 
> Any help is very much appreciated.
> 
> Thanks.
> 
> Paul
> 
> 
> 
> 
> 
> 
> 
> 50621 12/16 11:35:22.806 ERROR HuntGroups:
> 
> pkid: 0xAE5672C0728C86458CAE73F87025E57E, name: New_JTAPI, dn: 6500,
> fkDevice: 0x8FCBD68266333C4D98325062886C69DB, fkCSS:
> 0x5563A64BB3195843A5E2A5FBD113AFE6, huntingAlgorithm: 1,
> dbHuntingAlgorithm: 1, owner: 0, active: 2, queueEnabled: false,
> queueSize: 32, holdTime: 0, CMs: CM_LABCM01-
> 
> Lines:
> 
>       DN: 5750, AlwaysRoute: true, Online: true, LineActive: true,
> RoutingCall: false, Waiting: false, LS: Active, Idle Time:
> 1134697714514, RouteUnknown: true
> 
>       DN: 7017, AlwaysRoute: true, Online: true, LineActive: true,
> RoutingCall: false, Waiting: false, LS: Idle, Idle Time:
> 1134697737624, RouteUnknown: true
> 
> HuntGroupsUsers:
> 
> pkid: 0xAF326A961CFAEF419870B74D4BB27478, userid: ac, passwd: *****
> 
> Sessions:
> 
> Active Sessions:
> 
> Deactive Sessions:
> 
> _______________________________________________
> 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 CORTIZ at broward.org  Mon Dec 19 07:49:05 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon Dec 19 07:49:13 2005
Subject: [cisco-voip] Call rings phones once and then drops
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B8E@MREXVS01.bc.broward.cty>

Have seen this when the inbound CSS for the gateway is not defined or
correct.  You would think it would never even make it to the phone, but
this happened to me with a 3745.

 

Carlos 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Saturday, December 17, 2005 2:34 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call rings phones once and then drops

 

Calls ring in on local PRI trunks on 3845 router, makes its way to
callmanager, makes its way to phone and rings once on the phone.  But
that is it, calls instantly drop after first ring of the phone.  Does
not give person chance to answer just drops without any user
intervention. 

 

Using MGCP between 3845 and callmanager.

 

Any ideas?

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From CORTIZ at broward.org  Mon Dec 19 07:53:54 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon Dec 19 07:54:01 2005
Subject: [cisco-voip] Setting Up Direct Transfer to Voicemail
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B8F@MREXVS01.bc.broward.cty>

This link may help....

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2237/products_
tech_note09186a008015b963.shtml


-----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: Friday, December 16, 2005 4:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Setting Up Direct Transfer to Voicemail


Can anyone offer some assistance / advice on setting up a direct
transfer
to voicemail?

Tennille Spence
IT Business Liaison
PBS Technology Branch
816-823-1357 Office
816-806-6342 Cell
816-823-5526 Fax

"What lies behind us and what lies before us are tiny matters compared
to
what lies within us." - Ralph Waldo Emerson

"Opportunity is missed by most people because it is dressed in overalls
and
looks like work." - Thomas A. Edison


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


From mody_mody165 at yahoo.com  Mon Dec 19 10:05:08 2005
From: mody_mody165 at yahoo.com (eng_mohamed_dems)
Date: Mon Dec 19 10:05:22 2005
Subject: [cisco-voip] neeeeeeeeeeeeeeed heeeeeeeeeelp
Message-ID: <20051219150508.35540.qmail@web31002.mail.mud.yahoo.com>

hello everybody
  I'm a voip beginner and I need your help 
  I've an 2620 Router(16MB flash/48MBDRAM) and i want to install a nm-v2 but I don't know the ios version to install on the router 
  please send me the release name and i'll download it from the cisco site
  please help me
  thanks in advance

__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
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From smgustafson at gmail.com  Mon Dec 19 10:10:19 2005
From: smgustafson at gmail.com (Steve G)
Date: Mon Dec 19 10:10:22 2005
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 96
In-Reply-To: <200512191254.jBJCs3M8021255@puck.nether.net>
References: <200512191254.jBJCs3M8021255@puck.nether.net>
Message-ID: <8a1f34b50512190710h21bce42fv79c11968f65c32d8@mail.gmail.com>

Avidan,
     I am planning a slightly larger setup using the 3544 chassis with the
2PRI gateway blade.  I have 12 videoconferencing systems spread out over our
WAN.  90 percent of them are Polycom VSX7000 models.  I have heard the
Tandbergs will integrate pretty seemlessly with CallManager, but I do not
have any experience using them.  I have setup a gatekeeper which allows the
H.323 Video endpoints (Polycom stuff) to call Skinny endpoints (Cisco VTA,
7985...etc).  It's a pretty slick setup and not too complicated to set-up,
as long as you have control over the CCM and gatekeepers.  The most
difficult part might be coordinating with other groups if that is required.

We are also looking at the MCU blade for the 3540 chassis.

Steve


>
> ---------- Forwarded message ----------
> From: "IT" <it at cimgroup.com>
> To: <cisco-voip at puck.nether.net>
> Date: Sun, 18 Dec 2005 18:34:54 -0800
> Subject: [cisco-voip] Video conf. equipment for IP/VC
>
> I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
> video conference needs. I currently have a Polycom Viewstation EX, but with
> the upcoming IP/VC installation, I didnt know if buying 3 more of those was
> the right idea. What is everyone using out there for video conferencing with
> their IP/VC? Tandbergs? Polycoms?  Does getting MeetingPlace really help in
> small conferences?
>
> thanks for the advice.
> avidan
>
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
>
>
>
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From Marcin_Nowacki at sevenet.pl  Mon Dec 19 10:20:12 2005
From: Marcin_Nowacki at sevenet.pl (Marcin Nowacki)
Date: Mon Dec 19 10:20:34 2005
Subject: [cisco-voip] neeeeeeeeeeeeeeed heeeeeeeeeelp
Message-ID: <AF7AA9F0562A514FBB1FEF74A53971B45F8670@serwer11.sevenet.local>

Hello

 

If you hale Access to CCO ( and I suppose you have if you want to
download software ) you may check your hardware-software compatibility
on his site :

 

http://tools.cisco.com/Support/Fusion/FusionHCHome.do

 

Greetings

Marcin

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
eng_mohamed_dems
Sent: Monday, December 19, 2005 4:05 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] neeeeeeeeeeeeeeed heeeeeeeeeelp

 

hello everybody

I'm a voip beginner and I need your help 

I've an 2620 Router(16MB flash/48MBDRAM) and i want to install a nm-v2
but I don't know the ios version to install on the router 

please send me the release name and i'll download it from the cisco site

please help me

thanks in advance

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

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From pdiener at ita.hsr.ch  Mon Dec 19 10:22:31 2005
From: pdiener at ita.hsr.ch (patrick.diener@ita.hsr.ch)
Date: Mon Dec 19 10:22:39 2005
Subject: [cisco-voip] neeeeeeeeeeeeeeed heeeeeeeeeelp
In-Reply-To: <20051219150508.35540.qmail@web31002.mail.mud.yahoo.com>
Message-ID: <20051219152231.2B8C3E8036@mail.ita.hsr.ch>

Et voila.

 

 
<http://www.cisco.com/en/US/products/hw/modules/ps2617/products_tech_note091
86a0080094ac0.shtml>
http://www.cisco.com/en/US/products/hw/modules/ps2617/products_tech_note0918
6a0080094ac0.shtml

 

Cheers

Patrick

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of eng_mohamed_dems
Sent: Montag, 19. Dezember 2005 16:05
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] neeeeeeeeeeeeeeed heeeeeeeeeelp

 

hello everybody

I'm a voip beginner and I need your help 

I've an 2620 Router(16MB flash/48MBDRAM) and i want to install a nm-v2 but I
don't know the ios version to install on the router 

please send me the release name and i'll download it from the cisco site

please help me

thanks in advance

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

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From rratliff at cisco.com  Mon Dec 19 10:24:24 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Dec 19 10:24:23 2005
Subject: [cisco-voip] neeeeeeeeeeeeeeed heeeeeeeeeelp
In-Reply-To: <20051219150508.35540.qmail@web31002.mail.mud.yahoo.com>
References: <20051219150508.35540.qmail@web31002.mail.mud.yahoo.com>
Message-ID: <FE86AAA3-9B0E-4F5B-8FC1-710BF19EA59F@cisco.com>

I believe you mean NM-2V.   If that's the case check out http:// 
www.cisco.com/en/US/products/hw/modules/ps2617/ 
products_tech_note09186a0080094ac0.shtml (first hit on www.cisco.com  
searching for nm-2v).

-Ryan

On Dec 19, 2005, at 10:05 AM, eng_mohamed_dems wrote:

hello everybody
I'm a voip beginner and I need your help
I've an 2620 Router(16MB flash/48MBDRAM) and i want to install a nm- 
v2 but I don't know the ios version to install on the router
please send me the release name and i'll download it from the cisco site
please help me
thanks in advance
__________________________________________________
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 CORTIZ at broward.org  Mon Dec 19 10:29:02 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon Dec 19 10:29:11 2005
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 96
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B93@MREXVS01.bc.broward.cty>

Steve,

 

Would you mind pointing me out to the documentation you used for this
setup?  Specifically the Gatekeeper and MCU setup.  We are not doing
this yet, but I can see it on the roadmap.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Steve G
Sent: Monday, December 19, 2005 10:10 AM
To: cisco-voip at puck.nether.net; it at cimgroup.com
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 96

 

Avidan,

     I am planning a slightly larger setup using the 3544 chassis with
the 2PRI gateway blade.  I have 12 videoconferencing systems spread out
over our WAN.  90 percent of them are Polycom VSX7000 models.  I have
heard the Tandbergs will integrate pretty seemlessly with CallManager,
but I do not have any experience using them.  I have setup a gatekeeper
which allows the H.323 Video endpoints (Polycom stuff) to call Skinny
endpoints (Cisco VTA, 7985...etc).  It's a pretty slick setup and not
too complicated to set-up, as long as you have control over the CCM and
gatekeepers.  The most difficult part might be coordinating with other
groups if that is required. 

 

We are also looking at the MCU blade for the 3540 chassis.  

 

Steve
 

	
	---------- Forwarded message ----------
	From: "IT" <it at cimgroup.com >
	To: <cisco-voip at puck.nether.net>
	Date: Sun, 18 Dec 2005 18:34:54 -0800
	Subject: [cisco-voip] Video conf. equipment for IP/VC

	I am planning to setup a 3511 IP/VC to handle 4 branch offices
and their video conference needs. I currently have a Polycom Viewstation
EX, but with the upcoming IP/VC installation, I didnt know if buying 3
more of those was the right idea. What is everyone using out there for
video conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences? 
	
	thanks for the advice.
	avidan 

	******************************************************
	This email may contain trade secrets and/or other information
that is 
	proprietary and/or confidential. Use of this by anyone other
than the 
	intended recipient(s) is prohibited and may be unlawful. If you
are not 
	the intended recipient, delete the e-mail and notify us
immediately.
	****************************************************** 

	 

 

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From monika.kapil at evalueserve.com  Mon Dec 19 10:26:16 2005
From: monika.kapil at evalueserve.com (Monika Kapil)
Date: Mon Dec 19 10:32:42 2005
Subject: [cisco-voip] need help
Message-ID: <A99CFB142B0B45489B4123C28ED33E3C06CCDCF9@EVSEMAIL.Evalueserve.com>

Hi All,

 

I am facing an issue in logging in to cisco desktop agent the error is 

"unable to log agent in"

 

The error log shows :

 

ASL10013  Set agent state call failed (267).

20:42:01 12/19/2005 MAJOR  PhoneDev PD2066 AgentLogin: Agent state login
request failed: Unable to log agent in

 

I have checked that the particular IP phone is associated with the rm
user.

 

Kindly suggest.

 

Regards

Monika

 

 

 

Monika Kapil

________________________________

Network Administrator, Information Technology
Evalueserve 
Tel: +91 124 251 54400(3122)

 Mobile (+91 9810434611)
www.evalueserve.com

 

 

  

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From Scott.Voll at wesd.org  Mon Dec 19 10:46:21 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec 19 10:46:26 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <D713462ED535184D830F6F6301753E7E01A120E1@VISHNU.wesd.org>

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 

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From CORTIZ at broward.org  Mon Dec 19 10:47:57 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon Dec 19 10:48:07 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B98@MREXVS01.bc.broward.cty>

Are you going to be doing Multipoint conferencing?  IHave you set your
stuff up yet?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:46 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 

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From lelio at uoguelph.ca  Mon Dec 19 10:52:13 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon Dec 19 10:52:17 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
References: <D713462ED535184D830F6F6301753E7E01A120E1@VISHNU.wesd.org>
Message-ID: <008001c604b4$2dec8ee0$ca01a8c0@deadfish>

Video conf. equipment for IP/VCAt CIPTUG, I found out that Sony has a skinny unit as well that registers to CallManager (or so they said). Take a look at a google search for Sony + CallManager for lots of links.

The only thing I didn't like was that they didn't support CDP, so they can't negotiate VLAN access on the port. The data network for that port has to allow access to your CallManager network.
  ----- Original Message ----- 
  From: Voll, Scott 
  To: IT ; cisco-voip at puck.nether.net 
  Sent: Monday, December 19, 2005 10:46 AM
  Subject: RE: [cisco-voip] Video conf. equipment for IP/VC


  We currently have a couple of Polycoms but will be getting a Tandberg shortly.  Both seem to be good.  It really comes down to features.  I'm hoping I can get rid of the old Vtel units we have soon.  Stay away from VTEL!!!!!!!!!!!!!

   

  Nice thing with the Tandberg we demo'd the other day is it can register directly with CM.

   

  Scott

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
  Sent: Sunday, December 18, 2005 6:35 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Video conf. equipment for IP/VC

   

  I am planning to setup a 3511 IP/VC to handle 4 branch offices and their video conference needs. I currently have a Polycom Viewstation EX, but with the upcoming IP/VC installation, I didnt know if buying 3 more of those was the right idea. What is everyone using out there for video conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting MeetingPlace really help in small conferences?

  thanks for the advice.
  avidan 

  ******************************************************
  This email may contain trade secrets and/or other information that is 
  proprietary and/or confidential. Use of this by anyone other than the 
  intended recipient(s) is prohibited and may be unlawful. If you are not 
  the intended recipient, delete the e-mail and notify us immediately.
  ****************************************************** 



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


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  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Scott.Voll at wesd.org  Mon Dec 19 10:58:21 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec 19 10:58:29 2005
Subject: [cisco-voip] need help
Message-ID: <D713462ED535184D830F6F6301753E7E01A120E2@VISHNU.wesd.org>

Did you check the user is associated to the phone and the phone as the
ICD checked?

 

Are they apart of a queue?

 

scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Monika Kapil
Sent: Monday, December 19, 2005 7:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] need help

 

Hi All,

 

I am facing an issue in logging in to cisco desktop agent the error is 

"unable to log agent in"

 

The error log shows :

 

ASL10013  Set agent state call failed (267).

20:42:01 12/19/2005 MAJOR  PhoneDev PD2066 AgentLogin: Agent state login
request failed: Unable to log agent in

 

I have checked that the particular IP phone is associated with the rm
user.

 

Kindly suggest.

 

Regards

Monika

 

 

 

Monika Kapil

________________________________

Network Administrator, Information Technology
Evalueserve 
Tel: +91 124 251 54400(3122)

 Mobile (+91 9810434611)
www.evalueserve.com

 

 

  

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From PBrunnen at bccnetworks.net  Mon Dec 19 11:15:15 2005
From: PBrunnen at bccnetworks.net (PBrunnen@bccnetworks.net)
Date: Mon Dec 19 11:15:22 2005
Subject: [cisco-voip] Newbie Question
Message-ID: <OF749F26ED.5C1F6E26-ON852570DC.00591208-852570DC.0059498E@bccnetworks.net>

Hello all!
   Sorry in advance for this question... but I was not sure where else to 
ask it.
I had heard that you can setup a router, such as a 2621, as a VoIP call 
center.
If anyone could provide any further details as to how this is 
accomplished, I would greatly appreciate it.
I have been searching around the net with no success except to run across 
this list.

Thank you again for any help you can provide.  -Cheers, Peter.


====================
 In specification, Murphy's
Law supersedes Ohm's Law.
       --Unknown
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From monika.kapil at evalueserve.com  Mon Dec 19 11:25:59 2005
From: monika.kapil at evalueserve.com (Monika Kapil)
Date: Mon Dec 19 11:25:56 2005
Subject: [cisco-voip] need help
Message-ID: <A99CFB142B0B45489B4123C28ED33E3C06CCDDA3@EVSEMAIL.Evalueserve.com>

Yes,I have checked and the user is associated to the phone abd the ICD
is checked.

 

Thanks

Monika

 

 

 

 

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, December 19, 2005 9:28 PM
To: Monika Kapil; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] need help

 

Did you check the user is associated to the phone and the phone as the
ICD checked?

 

Are they apart of a queue?

 

scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Monika Kapil
Sent: Monday, December 19, 2005 7:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] need help

 

Hi All,

 

I am facing an issue in logging in to cisco desktop agent the error is 

"unable to log agent in"

 

The error log shows :

 

ASL10013  Set agent state call failed (267).

20:42:01 12/19/2005 MAJOR  PhoneDev PD2066 AgentLogin: Agent state login
request failed: Unable to log agent in

 

I have checked that the particular IP phone is associated with the rm
user.

 

Kindly suggest.

 

Regards

Monika

 

 

 

Monika Kapil

________________________________

Network Administrator, Information Technology
Evalueserve 
Tel: +91 124 251 54400(3122)

 Mobile (+91 9810434611)
www.evalueserve.com

 

 

  

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From bpeyton at sover.net  Mon Dec 19 11:28:33 2005
From: bpeyton at sover.net (Brandon Peyton)
Date: Mon Dec 19 11:28:53 2005
Subject: [cisco-voip] need help
In-Reply-To: <A99CFB142B0B45489B4123C28ED33E3C06CCDCF9@EVSEMAIL.Evalueserve.com>
Message-ID: <200512191628.jBJGSXjl018673@mailgate4.sover.net>

Might want to make sure that you've got  Enable CTI Application Use selected
under the user.
 
Regards,
Brandon

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Monika Kapil
Sent: Monday, December 19, 2005 10:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] need help



Hi All,

 

I am facing an issue in logging in to cisco desktop agent the error is 

"unable to log agent in"

 

The error log shows :

 

ASL10013  Set agent state call failed (267).

20:42:01 12/19/2005 MAJOR  PhoneDev PD2066 AgentLogin: Agent state login
request failed: Unable to log agent in

 

I have checked that the particular IP phone is associated with the rm user.

 

Kindly suggest.

 

Regards

Monika

 

 




Monika Kapil

  _____  

Network Administrator, Information Technology
Evalueserve 
Tel: +91 124 251 54400(3122)

 Mobile (+91 9810434611)
www.evalueserve.com


 

 

  

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From monika.kapil at evalueserve.com  Mon Dec 19 11:37:14 2005
From: monika.kapil at evalueserve.com (Monika Kapil)
Date: Mon Dec 19 11:37:11 2005
Subject: [cisco-voip] need help
Message-ID: <A99CFB142B0B45489B4123C28ED33E3C06CCDDC4@EVSEMAIL.Evalueserve.com>

Yes, that has been selected. Then even I am getting the same error.

 

Regards,

Monika

 

 

 

 

 

 

________________________________

From: Brandon Peyton [mailto:bpeyton at sover.net] 
Sent: Monday, December 19, 2005 9:59 PM
To: Monika Kapil; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] need help

 

Might want to make sure that you've got  Enable CTI Application Use
selected under the user.

 

Regards,

Brandon

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Monika Kapil
Sent: Monday, December 19, 2005 10:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] need help

Hi All,

 

I am facing an issue in logging in to cisco desktop agent the error is 

"unable to log agent in"

 

The error log shows :

 

ASL10013  Set agent state call failed (267).

20:42:01 12/19/2005 MAJOR  PhoneDev PD2066 AgentLogin: Agent state login
request failed: Unable to log agent in

 

I have checked that the particular IP phone is associated with the rm
user.

 

Kindly suggest.

 

Regards

Monika

 

 

 

Monika Kapil

________________________________

Network Administrator, Information Technology
Evalueserve 
Tel: +91 124 251 54400(3122)

 Mobile (+91 9810434611)
www.evalueserve.com

 

 

  

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From bpeyton at sover.net  Mon Dec 19 12:04:41 2005
From: bpeyton at sover.net (Brandon Peyton)
Date: Mon Dec 19 12:06:54 2005
Subject: [cisco-voip] need help
In-Reply-To: <A99CFB142B0B45489B4123C28ED33E3C06CCDDC4@EVSEMAIL.Evalueserve.com>
Message-ID: <200512191704.jBJH4fop020070@mailgate5.sover.net>

Do they have the proper share to Desktop_cfg on their machine?
 
Brandon

  _____  

From: Monika Kapil [mailto:monika.kapil at evalueserve.com] 
Sent: Monday, December 19, 2005 11:37 AM
To: Brandon Peyton; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] need help



Yes, that has been selected. Then even I am getting the same error.

 

Regards,

Monika

 

 

 

  _____  

From: Brandon Peyton [mailto:bpeyton at sover.net] 
Sent: Monday, December 19, 2005 9:59 PM
To: Monika Kapil; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] need help

 

Might want to make sure that you've got  Enable CTI Application Use selected
under the user.

 

Regards,

Brandon

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Monika Kapil
Sent: Monday, December 19, 2005 10:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] need help

Hi All,

 

I am facing an issue in logging in to cisco desktop agent the error is 

"unable to log agent in"

 

The error log shows :

 

ASL10013  Set agent state call failed (267).

20:42:01 12/19/2005 MAJOR  PhoneDev PD2066 AgentLogin: Agent state login
request failed: Unable to log agent in

 

I have checked that the particular IP phone is associated with the rm user.

 

Kindly suggest.

 

Regards

Monika

 

 




Monika Kapil

  _____  

Network Administrator, Information Technology
Evalueserve 
Tel: +91 124 251 54400(3122)

 Mobile (+91 9810434611)
www.evalueserve.com


 

 

  

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From Scott.Voll at wesd.org  Mon Dec 19 12:34:55 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec 19 12:35:02 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <D713462ED535184D830F6F6301753E7E01A120E3@VISHNU.wesd.org>

One of our School Districts bought some Sony's a year or so ago.  Didn't
like them.  They had some issues with other devices / MCU's.  Mic was
built on to the camera which causes a lot of echo.

 

IMHO stick with the Tandberg or polycom.

 

Scott

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Monday, December 19, 2005 7:52 AM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Video conf. equipment for IP/VC

 

At CIPTUG, I found out that Sony has a skinny unit as well that
registers to CallManager (or so they said). Take a look at a google
search for Sony + CallManager
<http://www.google.com/search?sourceid=navclient&ie=UTF-8&rls=GGLG,GGLG:
2005-40,GGLG:en&q=sony+callmanager>  for lots of links.

 

The only thing I didn't like was that they didn't support CDP, so they
can't negotiate VLAN access on the port. The data network for that port
has to allow access to your CallManager network.

	----- Original Message ----- 

	From: Voll, Scott <mailto:Scott.Voll at wesd.org>  

	To: IT <mailto:it at cimgroup.com>  ; cisco-voip at puck.nether.net 

	Sent: Monday, December 19, 2005 10:46 AM

	Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

	 

	We currently have a couple of Polycoms but will be getting a
Tandberg shortly.  Both seem to be good.  It really comes down to
features.  I'm hoping I can get rid of the old Vtel units we have soon.
Stay away from VTEL!!!!!!!!!!!!!

	 

	Nice thing with the Tandberg we demo'd the other day is it can
register directly with CM.

	 

	Scott

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
	Sent: Sunday, December 18, 2005 6:35 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Video conf. equipment for IP/VC

	 

	I am planning to setup a 3511 IP/VC to handle 4 branch offices
and their video conference needs. I currently have a Polycom Viewstation
EX, but with the upcoming IP/VC installation, I didnt know if buying 3
more of those was the right idea. What is everyone using out there for
video conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?
	
	thanks for the advice.
	avidan 

	******************************************************
	This email may contain trade secrets and/or other information
that is 
	proprietary and/or confidential. Use of this by anyone other
than the 
	intended recipient(s) is prohibited and may be unlawful. If you
are not 
	the intended recipient, delete the e-mail and notify us
immediately.
	****************************************************** 

	
________________________________


	_______________________________________________
	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  Mon Dec 19 12:39:40 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec 19 12:39:49 2005
Subject: [cisco-voip] Newbie Question
Message-ID: <D713462ED535184D830F6F6301753E7E01A120E4@VISHNU.wesd.org>

CCM Express might give you a hunt group (not sure as I only use CM) or
IVR but I haven't heard of ICD on a router.  IF I'm wrong, I'm sure I
will be corrected.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
PBrunnen at bccnetworks.net
Sent: Monday, December 19, 2005 8:15 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Newbie Question

 


Hello all! 
   Sorry in advance for this question... but I was not sure where else
to ask it. 
I had heard that you can setup a router, such as a 2621, as a VoIP call
center. 
If anyone could provide any further details as to how this is
accomplished, I would greatly appreciate it. 
I have been searching around the net with no success except to run
across this list. 

Thank you again for any help you can provide.  -Cheers, Peter.


====================
In specification, Murphy's
Law supersedes Ohm's Law.
      --Unknown

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From b.turnbow at twt.it  Mon Dec 19 12:56:48 2005
From: b.turnbow at twt.it (Brian Turnbow)
Date: Mon Dec 19 12:56:57 2005
Subject: [cisco-voip] Analog line extension
Message-ID: <52964DBFB59FA34F98E96921BE21C906B235B3@twt-exch.milano.twt>

Hello all,
I have a need to extend 32 analog lines over an IP network with fxs on
one side and fxo on the other side.
I need to set up hotline for each port so that when the phone goes off
hook it connects automatically to the specified port on the other end. 
I can't change to ISDN and need to maintain the actual analog
configuration. (I've tried to convice them to change architecture but
it's a no go)
That being said I thought of the cisco IAD, but it seems to only offer
high port density with FXS lines not FXO... 
Any suggestions on euipment ? 

Thanks
Brian


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From it at cimgroup.com  Mon Dec 19 13:27:03 2005
From: it at cimgroup.com (IT)
Date: Mon Dec 19 13:23:03 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0265CA75@exchange.cimgroup.com>

I intend on doing multi-point conferencing. Right now, I just have a
Viewstation EX, so I only do point-to-point conferences with other
companies. I am putting in an order for a 3511 and 2 more video
conference cameras, so my 2 biggest branch offices. I am hoping to have
a 3-way video conference happening by the end of January.  However, I am
having a very hard time just connecting my Polycom Viewstation to
CallManager. Is it even connectable? It seems the viewstation only has a
"gatekeeper" IP setting for what I assume will be the IP of the 3511.
Where do I set the IP of CallManager on this polycom?

 

Thanks,

Avidan

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Monday, December 19, 2005 7:48 AM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

Are you going to be doing Multipoint conferencing?  IHave you set your
stuff up yet?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:46 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 

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From jgrace at digitelusa.net  Mon Dec 19 13:33:35 2005
From: jgrace at digitelusa.net (James)
Date: Mon Dec 19 13:33:26 2005
Subject: [cisco-voip] Reporting from cm
Message-ID: <822606664-1135017204-cardhu_blackberry.rim.net-18404-@engine08-cell02>

I've been asked what kind of reporting can cm do. I seen the plugin for I guess real time diag. 
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From Scott.Voll at wesd.org  Mon Dec 19 14:36:03 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec 19 14:36:11 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <D713462ED535184D830F6F6301753E7E01A120E9@VISHNU.wesd.org>

 

We have the polycom sp and it does NOT have an interface with CM.  you
will most likely have to use the MCU to connect to a CM video conference
via a gatekeeper.

 

Some MCU do have a built in Gatekeeper.  I suggest just getting your
self an old 26xx series router and the IP Plus IOS and doing it with one
of these.

 

The MCU will just be a e.164 off the gatekeeper.  Each h323 video unit
will just dial the e164 to get into the conference.

 

Just as an FYI you will most likely have to exchange gatekeeper info
with other company's if you are plan to have them dial into your
conferences.

 

Scott

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 10:27 AM
To: Ortiz, Carlos; Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

I intend on doing multi-point conferencing. Right now, I just have a
Viewstation EX, so I only do point-to-point conferences with other
companies. I am putting in an order for a 3511 and 2 more video
conference cameras, so my 2 biggest branch offices. I am hoping to have
a 3-way video conference happening by the end of January.  However, I am
having a very hard time just connecting my Polycom Viewstation to
CallManager. Is it even connectable? It seems the viewstation only has a
"gatekeeper" IP setting for what I assume will be the IP of the 3511.
Where do I set the IP of CallManager on this polycom?

 

Thanks,

Avidan

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Monday, December 19, 2005 7:48 AM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

Are you going to be doing Multipoint conferencing?  IHave you set your
stuff up yet?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:46 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 

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From Scott.Voll at wesd.org  Mon Dec 19 14:37:52 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec 19 14:38:00 2005
Subject: [cisco-voip] Reporting from cm
Message-ID: <D713462ED535184D830F6F6301753E7E01A120EA@VISHNU.wesd.org>

What kind of reporting are you looking for?

Call detail reporting, traces, dial plans, etc?

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James
Sent: Monday, December 19, 2005 10:34 AM
To: CiscosupportUpuck
Subject: [cisco-voip] Reporting from cm

I've been asked what kind of reporting can cm do. I seen the plugin for
I guess real time diag. 
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From BurwellJ at firstcharter.com  Mon Dec 19 16:39:37 2005
From: BurwellJ at firstcharter.com (Jason Burwell)
Date: Mon Dec 19 16:39:47 2005
Subject: [cisco-voip] Inbound Calling Name
Message-ID: <697161D0107BAD40AFE9A86B06509AC90643E17F@FCEXCLUSTER.FirstCharter.com>

On CCM 3.3.4 can you get Inbound Calling Name when using 4ESS Protocol?
Using 6608 as a Gateway.
 
Thanks!
Jason


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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,
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this information by persons or entities other than the intended recipient is prohibited.
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From Scott.Voll at wesd.org  Mon Dec 19 16:52:49 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon Dec 19 16:52:58 2005
Subject: [cisco-voip] ART/CDR/CAR
Message-ID: <D713462ED535184D830F6F6301753E7E01A120F5@VISHNU.wesd.org>

Has anyone else notices if you have MLA in CM 4.1 that you have to use
the Administrator username and password to get to the log in for ART/CDR
and then you have to use your MLA account to access the CDR/CAR
information?  Is this a bug?

 

Scott

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From jsteinberg at gmail.com  Mon Dec 19 16:55:31 2005
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Mon Dec 19 16:55:35 2005
Subject: [cisco-voip] Export Fast Dials and Address Book
Message-ID: <f1b65f880512191355w4a50ec7cv@mail.gmail.com>

I have two CallManager clusters that I'm combining into one cluster. 
On both clusters I have users who aggresively use the fast dial and
address book IP phone services.  I'm looking for a way to export this
data so that I can take the contents of one cluster and add them to
the other cluster without overwriting what is currently there.

I can see the data when I browse via the DC Admin application but do
not see any options in the menu's to backup or export this data.

I also did not see this available in BAT.

Any help is greatly appreciated.

Thanks,

Justin


From jburton at NETechCorp.com  Mon Dec 19 17:04:22 2005
From: jburton at NETechCorp.com (Jason Burton)
Date: Mon Dec 19 17:04:32 2005
Subject: [cisco-voip] need help
Message-ID: <FDC137145729AD438CDCA0F9FC6F72363A3552@mail.netechcorp.com>

What version are you running?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:58 AM
To: Monika Kapil; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] need help

 

Did you check the user is associated to the phone and the phone as the
ICD checked?

 

Are they apart of a queue?

 

scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Monika Kapil
Sent: Monday, December 19, 2005 7:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] need help

 

Hi All,

 

I am facing an issue in logging in to cisco desktop agent the error is 

"unable to log agent in"

 

The error log shows :

 

ASL10013  Set agent state call failed (267).

20:42:01 12/19/2005 MAJOR  PhoneDev PD2066 AgentLogin: Agent state login
request failed: Unable to log agent in

 

I have checked that the particular IP phone is associated with the rm
user.

 

Kindly suggest.

 

Regards

Monika

 

 

 

Monika Kapil

________________________________

Network Administrator, Information Technology
Evalueserve 
Tel: +91 124 251 54400(3122)

 Mobile (+91 9810434611)
www.evalueserve.com

 

 

  

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From rratliff at cisco.com  Mon Dec 19 20:26:09 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon Dec 19 20:26:17 2005
Subject: [cisco-voip] ART/CDR/CAR
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A120F5@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A120F5@VISHNU.wesd.org>
Message-ID: <F1122E99-20C8-4CD4-A3D6-0DFDA76632E3@cisco.com>

100% WAD.

Actually the CAR/ART logon info is set the first time you login with  
the default user/pass.  It makes you choose the user(s) that will  
have access to the ART web page.
Doing this creates the group in the ldap directory that any member of  
which can access ART.

MLA is only for CCMAdmin.  ART is not CCMAdmin :)

-Ryan

On Dec 19, 2005, at 4:52 PM, Voll, Scott wrote:

Has anyone else notices if you have MLA in CM 4.1 that you have to  
use the Administrator username and password to get to the log in for  
ART/CDR and then you have to use your MLA account to access the CDR/ 
CAR information?  Is this a bug?



Scott

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


From smgustafson at gmail.com  Mon Dec 19 21:57:25 2005
From: smgustafson at gmail.com (Steve G)
Date: Mon Dec 19 21:57:27 2005
Subject: [cisco-voip] CallManager 4.1 Video Telephony SRND
Message-ID: <8a1f34b50512191857i3274f8cema8489050cd9606cb@mail.gmail.com>

Hey guys, check this out.
http://www.cisco.com/application/pdf/en/us/guest/netsol/ns268/c649/ccmigration_09186a00804ff6ba.pdf

H.323 clients can communicate with Skinny clients using e.164 as Scott
mentioned.  H.323 clients require a gatekeeper to do this.  I have tested it
and it works with a plain old 3600 series router (it will work with a 2600XM
also).
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From it at cimgroup.com  Mon Dec 19 23:43:25 2005
From: it at cimgroup.com (IT)
Date: Mon Dec 19 23:39:26 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0265CE24@exchange.cimgroup.com>

This sounds like more work than I am interested in. Is there a better
Video conference option to use with the 3511?

Anyone have experience with the Tandberg?

Thanks,

Avidan

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, December 19, 2005 11:36 AM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

 

We have the polycom sp and it does NOT have an interface with CM.  you
will most likely have to use the MCU to connect to a CM video conference
via a gatekeeper.

 

Some MCU do have a built in Gatekeeper.  I suggest just getting your
self an old 26xx series router and the IP Plus IOS and doing it with one
of these.

 

The MCU will just be a e.164 off the gatekeeper.  Each h323 video unit
will just dial the e164 to get into the conference.

 

Just as an FYI you will most likely have to exchange gatekeeper info
with other company's if you are plan to have them dial into your
conferences.

 

Scott

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 10:27 AM
To: Ortiz, Carlos; Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

I intend on doing multi-point conferencing. Right now, I just have a
Viewstation EX, so I only do point-to-point conferences with other
companies. I am putting in an order for a 3511 and 2 more video
conference cameras, so my 2 biggest branch offices. I am hoping to have
a 3-way video conference happening by the end of January.  However, I am
having a very hard time just connecting my Polycom Viewstation to
CallManager. Is it even connectable? It seems the viewstation only has a
"gatekeeper" IP setting for what I assume will be the IP of the 3511.
Where do I set the IP of CallManager on this polycom?

 

Thanks,

Avidan

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Monday, December 19, 2005 7:48 AM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

Are you going to be doing Multipoint conferencing?  IHave you set your
stuff up yet?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:46 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
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From rleetun at co.boulder.co.us  Tue Dec 20 08:30:40 2005
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Tue Dec 20 08:30:46 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <963CE94787005149A52C17F33A028E12051D4147@EVS1B.boco.co.boulder.co.us>

We have used a Tandberg, but we chose to purchase two Sony PCS-TL50s
instead.
 
Rob

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Monday, December 19, 2005 9:43 PM
To: Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC



This sounds like more work than I am interested in. Is there a better
Video conference option to use with the 3511?

Anyone have experience with the Tandberg?

Thanks,

Avidan

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, December 19, 2005 11:36 AM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

 

We have the polycom sp and it does NOT have an interface with CM.  you
will most likely have to use the MCU to connect to a CM video conference
via a gatekeeper.

 

Some MCU do have a built in Gatekeeper.  I suggest just getting your
self an old 26xx series router and the IP Plus IOS and doing it with one
of these.

 

The MCU will just be a e.164 off the gatekeeper.  Each h323 video unit
will just dial the e164 to get into the conference.

 

Just as an FYI you will most likely have to exchange gatekeeper info
with other company's if you are plan to have them dial into your
conferences.

 

Scott

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 10:27 AM
To: Ortiz, Carlos; Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

I intend on doing multi-point conferencing. Right now, I just have a
Viewstation EX, so I only do point-to-point conferences with other
companies. I am putting in an order for a 3511 and 2 more video
conference cameras, so my 2 biggest branch offices. I am hoping to have
a 3-way video conference happening by the end of January.  However, I am
having a very hard time just connecting my Polycom Viewstation to
CallManager. Is it even connectable? It seems the viewstation only has a
"gatekeeper" IP setting for what I assume will be the IP of the 3511.
Where do I set the IP of CallManager on this polycom?

 

Thanks,

Avidan

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Monday, December 19, 2005 7:48 AM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

Are you going to be doing Multipoint conferencing?  IHave you set your
stuff up yet?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:46 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
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From Kevin.Hart at brooks.com  Tue Dec 20 11:14:50 2005
From: Kevin.Hart at brooks.com (Hart, Kevin)
Date: Tue Dec 20 11:17:27 2005
Subject: [cisco-voip] Inbound Calling Name
Message-ID: <6FA79BD0B67DD411AAC400306E00B08C04E35835@exchange1.helixtech.com>

Hi Jason,

 

We are running CCM 4.1 with the 6608 blades using PRI 4ESS protocol and the
Send Calling Name In Facility IE option is grayed out in the PRI Protocol
Type Specific Information screen. Based on this fact, I would assume that
inbound CID with name is not possible using this protocol. If your ISDN
vendor supports NI2 then this will probably work.

 

Kevin

 

 

  _____  

From: Jason Burwell [mailto:BurwellJ at firstcharter.com] 
Sent: Monday, December 19, 2005 4:40 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Inbound Calling Name

 

On CCM 3.3.4 can you get Inbound Calling Name when using 4ESS Protocol?
Using 6608 as a Gateway.

 

Thanks!

Jason



**********************************************************************
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 in error, please contact the sender and delete the
material from
any computer. 
**********************************************************************

_____________________________________________________________________
This email has been scanned by the MessageLabs Email Security System.



_____________________________________________________________________
This email message, including any attachments, may contain confidential and proprietary information for the sole use of the intended recipient.  If you are not the intended recipient, you are hereby notified that any use, copying or dissemination of this message is strictly prohibited.  If you received this message in error, please notify Brooks Automation, Inc. immediately by reply email or by calling Brooks US Headquarters at +1 978-262-2400. Then delete this message from your system, without making any copy or distribution.  Thank you.
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From rratliff at cisco.com  Tue Dec 20 11:36:20 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue Dec 20 11:36:18 2005
Subject: [cisco-voip] Inbound Calling Name
In-Reply-To: <6FA79BD0B67DD411AAC400306E00B08C04E35835@exchange1.helixtech.com>
References: <6FA79BD0B67DD411AAC400306E00B08C04E35835@exchange1.helixtech.com>
Message-ID: <2A21C2B2-29AD-471E-AFA1-E2747F090E15@cisco.com>

If the protocol supports caller name in a Display IE rather than  
Facility then it should work just fine.  MGCP NI2 is the only  
protocol type we support caller name in Facility IE though.

-Ryan

On Dec 20, 2005, at 11:14 AM, Hart, Kevin wrote:

Hi Jason,



We are running CCM 4.1 with the 6608 blades using PRI 4ESS protocol  
and the Send Calling Name In Facility IE option is grayed out in the  
PRI Protocol Type Specific Information screen. Based on this fact, I  
would assume that inbound CID with name is not possible using this  
protocol. If your ISDN vendor supports NI2 then this will probably work.



Kevin





From: Jason Burwell [mailto:BurwellJ at firstcharter.com]
Sent: Monday, December 19, 2005 4:40 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Inbound Calling Name



On CCM 3.3.4 can you get Inbound Calling Name when using 4ESS  
Protocol? Using 6608 as a Gateway.



Thanks!

Jason



**********************************************************************
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 in error, please contact the sender and delete  
the material from
any computer.
**********************************************************************

_____________________________________________________________________
This email has been scanned by the MessageLabs Email Security System.


_____________________________________________________________________
This email message, including any attachments, may contain  
confidential and proprietary information for the sole use of the  
intended recipient. If you are not the intended recipient, you are  
hereby notified that any use, copying or dissemination of this  
message is strictly prohibited. If you received this message in  
error, please notify Brooks Automation, Inc. immediately by reply  
email or by calling Brooks US Headquarters at +1 978-262-2400. Then  
delete this message from your system, without making any copy or  
distribution. Thank you.
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From Scott.Voll at wesd.org  Tue Dec 20 12:13:33 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 20 12:13:45 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <D713462ED535184D830F6F6301753E7E01A12113@VISHNU.wesd.org>

All we have done is demo'd the Tandberg so far, but it does have an MCU
built in.  A lot of polycoms do also.  The nice thing about the Tandberg
is that you can also interface with CM.

 

Scott

 

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 8:43 PM
To: Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

This sounds like more work than I am interested in. Is there a better
Video conference option to use with the 3511?

Anyone have experience with the Tandberg?

Thanks,

Avidan

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, December 19, 2005 11:36 AM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

 

We have the polycom sp and it does NOT have an interface with CM.  you
will most likely have to use the MCU to connect to a CM video conference
via a gatekeeper.

 

Some MCU do have a built in Gatekeeper.  I suggest just getting your
self an old 26xx series router and the IP Plus IOS and doing it with one
of these.

 

The MCU will just be a e.164 off the gatekeeper.  Each h323 video unit
will just dial the e164 to get into the conference.

 

Just as an FYI you will most likely have to exchange gatekeeper info
with other company's if you are plan to have them dial into your
conferences.

 

Scott

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 10:27 AM
To: Ortiz, Carlos; Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

I intend on doing multi-point conferencing. Right now, I just have a
Viewstation EX, so I only do point-to-point conferences with other
companies. I am putting in an order for a 3511 and 2 more video
conference cameras, so my 2 biggest branch offices. I am hoping to have
a 3-way video conference happening by the end of January.  However, I am
having a very hard time just connecting my Polycom Viewstation to
CallManager. Is it even connectable? It seems the viewstation only has a
"gatekeeper" IP setting for what I assume will be the IP of the 3511.
Where do I set the IP of CallManager on this polycom?

 

Thanks,

Avidan

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Monday, December 19, 2005 7:48 AM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

Are you going to be doing Multipoint conferencing?  IHave you set your
stuff up yet?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:46 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 

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From akoul at dataways.gr  Tue Dec 20 12:14:33 2005
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Tue Dec 20 12:14:41 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A11E7E@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A11E7E@VISHNU.wesd.org>
Message-ID: <43A83BF9.4030604@dataways.gr>

Hi Scott,
i Had 3.1(1) and i managed to find the Extended services AA cd version 
3.5(2). Since i started the upgrade, 6,5 hours passed and it is still in 
job 5 out of 6!!! What can be possibly wrong?

Anthony

Voll, Scott wrote:

>Yes and Yes.
>
>You can upgrade from 3.1 to 3.5.  Very easy.  Took me ~ an hour.
>
>You do need to PUT for the CD's.  there is NO download.  What a pain.  I
>hate being in your shoes.  Because it takes a couple weeks any time I
>talk to cisco.
>
>Maybe you could open a TAC case and they could overnight it if you're
>down.
>
>Lesson learned.  Always look at all your IPT before upgrades.  Most
>everything is dependent in some way weather a simple TSP upgrade or a
>major release.
>
>Read those docs.
>
>Scott
>
>
>
>-----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, December 08, 2005 12:11 AM
>To: cisco-voip at puck-nether.net
>Subject: [cisco-voip] Inconsistent CCM version and CRS
>
>Hi all,
>i have ordered ccm 4.0(2)a and the whole package of the CDs came to my 
>office. I installed the CCM with no problem and the Extended Services CD
>
>that was in the package (i wanted to use the 4 port of IVR for the Auto 
>Attendant). The engine got installed easylly but i could not make it
>work.
>When i looked in the compatibility matrix of ccm, i saw that CRS 3.5.2 
>is needed with CCM 4.0(2)a!!
>The whole issue with the CRS engine is getting annoying since it is not 
>the first time that this is happening.
>Is it possible to download it and make it work from tha CCO site or i 
>need to order the CDs? Can i upgrade CRS 3.1(1) to 3.5(2) or i have to 
>unistall it and reinstall the new one?
>
>Thanks
>Anthony Kouloglou
>_______________________________________________
>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 Dec 20 12:34:27 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 20 12:34:36 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
Message-ID: <D713462ED535184D830F6F6301753E7E01A12117@VISHNU.wesd.org>

I don't think the Extended services is what you need.  I think it should
be IPCC or IVR cd's

Scott

-----Original Message-----
From: Anthony Kouloglou [mailto:akoul at dataways.gr] 
Sent: Tuesday, December 20, 2005 9:15 AM
To: Voll, Scott
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] Inconsistent CCM version and CRS

Hi Scott,
i Had 3.1(1) and i managed to find the Extended services AA cd version 
3.5(2). Since i started the upgrade, 6,5 hours passed and it is still in

job 5 out of 6!!! What can be possibly wrong?

Anthony

Voll, Scott wrote:

>Yes and Yes.
>
>You can upgrade from 3.1 to 3.5.  Very easy.  Took me ~ an hour.
>
>You do need to PUT for the CD's.  there is NO download.  What a pain.
I
>hate being in your shoes.  Because it takes a couple weeks any time I
>talk to cisco.
>
>Maybe you could open a TAC case and they could overnight it if you're
>down.
>
>Lesson learned.  Always look at all your IPT before upgrades.  Most
>everything is dependent in some way weather a simple TSP upgrade or a
>major release.
>
>Read those docs.
>
>Scott
>
>
>
>-----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, December 08, 2005 12:11 AM
>To: cisco-voip at puck-nether.net
>Subject: [cisco-voip] Inconsistent CCM version and CRS
>
>Hi all,
>i have ordered ccm 4.0(2)a and the whole package of the CDs came to my 
>office. I installed the CCM with no problem and the Extended Services
CD
>
>that was in the package (i wanted to use the 4 port of IVR for the Auto

>Attendant). The engine got installed easylly but i could not make it
>work.
>When i looked in the compatibility matrix of ccm, i saw that CRS 3.5.2 
>is needed with CCM 4.0(2)a!!
>The whole issue with the CRS engine is getting annoying since it is not

>the first time that this is happening.
>Is it possible to download it and make it work from tha CCO site or i 
>need to order the CDs? Can i upgrade CRS 3.1(1) to 3.5(2) or i have to 
>unistall it and reinstall the new one?
>
>Thanks
>Anthony Kouloglou
>_______________________________________________
>cisco-voip mailing list
>cisco-voip at puck.nether.net
>https://puck.nether.net/mailman/listinfo/cisco-voip
>
>  
>


From bababooey at cox.net  Tue Dec 20 13:21:37 2005
From: bababooey at cox.net (bababooey@cox.net)
Date: Tue Dec 20 13:24:12 2005
Subject: [cisco-voip] CM 4.1 to PBX GW troublshsooting
Message-ID: <20051220182216.YNBN29285.eastrmmtao03.cox.net@[172.18.52.8]>

I have a Call Manager 4.1.3 and a router interface to a pbx with a T1 trunk.  It is configured for H.323.
I find that I must reset the T1 controller almost daily to restore the PBX access.  I believe the problem is on my side not the PBX, because PBX users can still ring ip phones on the Call Manager.

How do I begin to troubleshoot this problem?
Which debug would be helpful?

Thanks - 
Bob



From Scott.Voll at wesd.org  Tue Dec 20 14:48:35 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 20 14:48:43 2005
Subject: [cisco-voip] What's the difference
Message-ID: <D713462ED535184D830F6F6301753E7E01A12122@VISHNU.wesd.org>

What is the difference between a device with SEP and ADP prefixes?  CM
4.1 but I think I remember it from 3.3.

 

TIA

 

Scott

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From pwalenta at wi.rr.com  Tue Dec 20 14:54:19 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue Dec 20 14:54:23 2005
Subject: [cisco-voip] What's the difference
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A12122@VISHNU.wesd.org>
Message-ID: <200512201954.jBKJsF9u029634@ms-smtp-04.rdc-kc.rr.com>

SEP are phone devices (7960 etc).  ADP indicates device profile for
extension mobility.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, December 20, 2005 1:49 PM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] What's the difference



What is the difference between a device with SEP and ADP prefixes?  CM 4.1
but I think I remember it from 3.3.

 

TIA

 

Scott

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From it at cimgroup.com  Tue Dec 20 15:15:51 2005
From: it at cimgroup.com (IT)
Date: Tue Dec 20 15:11:49 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0265D017@exchange.cimgroup.com>

And if it has an MCU built-in, do I still need a IP/VC 3511?

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, December 20, 2005 9:14 AM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

All we have done is demo'd the Tandberg so far, but it does have an MCU
built in.  A lot of polycoms do also.  The nice thing about the Tandberg
is that you can also interface with CM.

 

Scott

 

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 8:43 PM
To: Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

This sounds like more work than I am interested in. Is there a better
Video conference option to use with the 3511?

Anyone have experience with the Tandberg?

Thanks,

Avidan

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, December 19, 2005 11:36 AM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

 

We have the polycom sp and it does NOT have an interface with CM.  you
will most likely have to use the MCU to connect to a CM video conference
via a gatekeeper.

 

Some MCU do have a built in Gatekeeper.  I suggest just getting your
self an old 26xx series router and the IP Plus IOS and doing it with one
of these.

 

The MCU will just be a e.164 off the gatekeeper.  Each h323 video unit
will just dial the e164 to get into the conference.

 

Just as an FYI you will most likely have to exchange gatekeeper info
with other company's if you are plan to have them dial into your
conferences.

 

Scott

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 10:27 AM
To: Ortiz, Carlos; Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

I intend on doing multi-point conferencing. Right now, I just have a
Viewstation EX, so I only do point-to-point conferences with other
companies. I am putting in an order for a 3511 and 2 more video
conference cameras, so my 2 biggest branch offices. I am hoping to have
a 3-way video conference happening by the end of January.  However, I am
having a very hard time just connecting my Polycom Viewstation to
CallManager. Is it even connectable? It seems the viewstation only has a
"gatekeeper" IP setting for what I assume will be the IP of the 3511.
Where do I set the IP of CallManager on this polycom?

 

Thanks,

Avidan

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Monday, December 19, 2005 7:48 AM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

Are you going to be doing Multipoint conferencing?  IHave you set your
stuff up yet?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:46 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 

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From Scott.Voll at wesd.org  Tue Dec 20 15:15:32 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 20 15:15:41 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <D713462ED535184D830F6F6301753E7E01A12123@VISHNU.wesd.org>

The answer is, it depends.  If the unit that has the built in MCU is
always in the video conference you can probably get away with out a 3511
so long as all the units are h323. now if you are wanting to use CM VT
advantage you will most likely still need a MCU (35xx)  I don't' know
about the Tandberg as it does have a built in MCU and can register to
CM.

 

Scott

 

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Tuesday, December 20, 2005 12:16 PM
To: Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

And if it has an MCU built-in, do I still need a IP/VC 3511?

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, December 20, 2005 9:14 AM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

All we have done is demo'd the Tandberg so far, but it does have an MCU
built in.  A lot of polycoms do also.  The nice thing about the Tandberg
is that you can also interface with CM.

 

Scott

 

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 8:43 PM
To: Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

This sounds like more work than I am interested in. Is there a better
Video conference option to use with the 3511?

Anyone have experience with the Tandberg?

Thanks,

Avidan

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, December 19, 2005 11:36 AM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

 

We have the polycom sp and it does NOT have an interface with CM.  you
will most likely have to use the MCU to connect to a CM video conference
via a gatekeeper.

 

Some MCU do have a built in Gatekeeper.  I suggest just getting your
self an old 26xx series router and the IP Plus IOS and doing it with one
of these.

 

The MCU will just be a e.164 off the gatekeeper.  Each h323 video unit
will just dial the e164 to get into the conference.

 

Just as an FYI you will most likely have to exchange gatekeeper info
with other company's if you are plan to have them dial into your
conferences.

 

Scott

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 10:27 AM
To: Ortiz, Carlos; Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

I intend on doing multi-point conferencing. Right now, I just have a
Viewstation EX, so I only do point-to-point conferences with other
companies. I am putting in an order for a 3511 and 2 more video
conference cameras, so my 2 biggest branch offices. I am hoping to have
a 3-way video conference happening by the end of January.  However, I am
having a very hard time just connecting my Polycom Viewstation to
CallManager. Is it even connectable? It seems the viewstation only has a
"gatekeeper" IP setting for what I assume will be the IP of the 3511.
Where do I set the IP of CallManager on this polycom?

 

Thanks,

Avidan

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Monday, December 19, 2005 7:48 AM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

Are you going to be doing Multipoint conferencing?  IHave you set your
stuff up yet?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:46 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 

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From it at cimgroup.com  Tue Dec 20 15:36:11 2005
From: it at cimgroup.com (IT)
Date: Tue Dec 20 15:32:11 2005
Subject: [cisco-voip] Video conf. equipment for IP/VC
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0265D03D@exchange.cimgroup.com>

Well, I would like to have both VT Advantages, and h323 video conference
units all in a multipoint conference. If I am getting a 3511, I assume I
do not need a video camera with an MCU built-in, right? can some people
post their setups, and what hardware they used, and a brief summary of
their experiences? It seems most people on here have been using
Tandbergs and Sonys....any polycoms?

 

Thanks,

Avidan

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, December 20, 2005 12:16 PM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

The answer is, it depends.  If the unit that has the built in MCU is
always in the video conference you can probably get away with out a 3511
so long as all the units are h323. now if you are wanting to use CM VT
advantage you will most likely still need a MCU (35xx)  I don't' know
about the Tandberg as it does have a built in MCU and can register to
CM.

 

Scott

 

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Tuesday, December 20, 2005 12:16 PM
To: Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

And if it has an MCU built-in, do I still need a IP/VC 3511?

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, December 20, 2005 9:14 AM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

All we have done is demo'd the Tandberg so far, but it does have an MCU
built in.  A lot of polycoms do also.  The nice thing about the Tandberg
is that you can also interface with CM.

 

Scott

 

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 8:43 PM
To: Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

This sounds like more work than I am interested in. Is there a better
Video conference option to use with the 3511?

Anyone have experience with the Tandberg?

Thanks,

Avidan

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, December 19, 2005 11:36 AM
To: IT; Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

 

We have the polycom sp and it does NOT have an interface with CM.  you
will most likely have to use the MCU to connect to a CM video conference
via a gatekeeper.

 

Some MCU do have a built in Gatekeeper.  I suggest just getting your
self an old 26xx series router and the IP Plus IOS and doing it with one
of these.

 

The MCU will just be a e.164 off the gatekeeper.  Each h323 video unit
will just dial the e164 to get into the conference.

 

Just as an FYI you will most likely have to exchange gatekeeper info
with other company's if you are plan to have them dial into your
conferences.

 

Scott

________________________________

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, December 19, 2005 10:27 AM
To: Ortiz, Carlos; Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

I intend on doing multi-point conferencing. Right now, I just have a
Viewstation EX, so I only do point-to-point conferences with other
companies. I am putting in an order for a 3511 and 2 more video
conference cameras, so my 2 biggest branch offices. I am hoping to have
a 3-way video conference happening by the end of January.  However, I am
having a very hard time just connecting my Polycom Viewstation to
CallManager. Is it even connectable? It seems the viewstation only has a
"gatekeeper" IP setting for what I assume will be the IP of the 3511.
Where do I set the IP of CallManager on this polycom?

 

Thanks,

Avidan

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Monday, December 19, 2005 7:48 AM
To: Voll, Scott; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

Are you going to be doing Multipoint conferencing?  IHave you set your
stuff up yet?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, December 19, 2005 10:46 AM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Video conf. equipment for IP/VC

 

We currently have a couple of Polycoms but will be getting a Tandberg
shortly.  Both seem to be good.  It really comes down to features.  I'm
hoping I can get rid of the old Vtel units we have soon.  Stay away from
VTEL!!!!!!!!!!!!!

 

Nice thing with the Tandberg we demo'd the other day is it can register
directly with CM.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Sunday, December 18, 2005 6:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video conf. equipment for IP/VC

 

I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
video conference needs. I currently have a Polycom Viewstation EX, but
with the upcoming IP/VC installation, I didnt know if buying 3 more of
those was the right idea. What is everyone using out there for video
conferencing with their IP/VC? Tandbergs? Polycoms?  Does getting
MeetingPlace really help in small conferences?

thanks for the advice.
avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 

-------------- next part --------------
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From c719562 at gmail.com  Tue Dec 20 18:37:02 2005
From: c719562 at gmail.com (Serg)
Date: Tue Dec 20 18:37:04 2005
Subject: [cisco-voip] Re: Call rings phones once and then drops
Message-ID: <f3e392860512201537k3d288a1t2eeae0514e8a151d@mail.gmail.com>

Hi TechGuy,
                   Not sure if you have already rectified this problem but i
can let you know that I had exactly the same problem about 6 month ago with
a Cisco 3845 + NM-HDV2 + PVDM-32 x 2.

At the time I was running IOS 12.3.14T2 IP-voice Feature set.  Looking at
your "debug isdn q931" debug, the cause code is exactly the same as what I
had.

Basically, the problem is to do with the E1/T1 controller's clocking getting
out of sync with the Carrier's T1/E1.  If you do a "show controllers e1" and
check whether you have excessive  Line Code Violations (LCVs), Path Code
Violations (PCV),  Slip Secs, etc.  These should be minimal.

Cisco may point you to the following BugID but I can assure that the work
around did nothing to fix my problems.

http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCef89288

Make sure that you follow the "Clocking Configurations On Voice-Capable
IOS-Based Platforms" (
http://www.cisco.com/en/US/products/hw/routers/ps259/products_tech_note09186a008031a072.shtml
).

Also make sure you follows the "Configuring MGCP PRI Backhaul and T1 CAS
Support for Cisco CallManager" (
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/ccm_c/intcnf3.htm)
and that you do not add the "application mgcpapp" command to dial peers that
support PRI backhaul.

Lastly, the most important step that I undertook was to roll-back (yes,
beleive it or not) to IOS 12.3.11T6 IP-Voice or above and I've never had any
problems since.

Hope this helps, other Cisco Bugs of interest are:

http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCEG90199

http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCEG89017

Good Luck,

Serg


On 12/19/05, cisco-voip-request at puck.nether.net <
cisco-voip-request at puck.nether.net> wrote:
>
> 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. Call rings phones once and then drops (TechGuy)
>    2. Re: Call rings phones once and then drops (TechGuy)
>    3. RE: Call rings phones once and then drops (Mark Snow)
>    4. Tech Prefix from ATA186 (Ciscozest)
>    5. Re: Call rings phones once and then drops (Teodor Georgiev)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sat, 17 Dec 2005 13:33:40 -0600
> From: TechGuy <techguy at gmail.com>
> Subject: [cisco-voip] Call rings phones once and then drops
> To: cisco-voip at puck.nether.net
> Message-ID:
>         <a1b07ea60512171133i79723eecs2d95643eaa5f034f at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Calls ring in on local PRI trunks on 3845 router, makes its way to
> callmanager, makes its way to phone and rings once on the phone.  But that
> is it, calls instantly drop after first ring of the phone.  Does not give
> person chance to answer just drops without any user intervention.
>
> Using MGCP between 3845 and callmanager.
>
> Any ideas?
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL:
> https://puck.nether.net/pipermail/cisco-voip/attachments/20051217/a63be268/attachment-0001.html
>
> ------------------------------
>
> Message: 2
> Date: Sat, 17 Dec 2005 14:19:32 -0600
> From: TechGuy <techguy at gmail.com>
> Subject: Re: [cisco-voip] Call rings phones once and then drops
> To: cisco-voip at puck.nether.net
> Message-ID:
>         <a1b07ea60512171219p288d1552qddda2d4820ebba47 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Here is the debug:
>
> emb3845#debug isdn q931
> debug isdn q931 is              ON.
> emb3845#
> *Dec 17 20:07:23.806: ISDN Se0/1/1:23 Q931: RX <- SETUP pd = 8  callref =
> 0x00D2
>         Bearer Capability i = 0x8090A2
>                 Standard = CCITT
>                 Transer Capability = Speech
>                 Transfer Mode = Circuit
>                 Transfer Rate = 64 kbit/s
>         Channel ID i = 0xA98381
>                 Exclusive, Channel 1
>         Calling Party Number i = 0x2183, '3375015555'
>                 Plan:ISDN, Type:National
>         Called Party Number i = 0x80, '4080'
>                 Plan:Unknown, Type:Unknown
> *Dec 17 20:07:24.830: ISDN Se0/1/1:23 Q931: TX -> RELEASE_COMP pd = 8
> callref = 0x80D2
>         Cause i = 0x80A9 - Temporary failure
> emb3845#
>
>
> More info on the problem.  This router has two local PRI trunks coming in
> to
> a 2MFT card.  It also has two CAS (E&M Wink) trunks coming into a 2MFT
> card
> for long distance and incoming toll free.
>
> The two CAS trunks work fine.  We are able to get incoming calls via the
> CAS
> trunks mentioned above with no problems.  Also, we have another gateway
> configured on the same callmanager in a similiar fashion and all works
> fine.  This was all working earlier this morning.  No known changes
> occured
> between when it worked and stopped working.
>
> Quite an interesting problem, we have TAC on the line now but thought I
> would see what others think.
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL:
> https://puck.nether.net/pipermail/cisco-voip/attachments/20051217/3bcb8972/attachment-0001.html
>
> ------------------------------
>
>
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From erickbe at yahoo.com  Tue Dec 20 23:40:16 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue Dec 20 23:40:23 2005
Subject: [cisco-voip] Re: Call rings phones once and then drops
In-Reply-To: <f3e392860512201537k3d288a1t2eeae0514e8a151d@mail.gmail.com>
Message-ID: <20051221044016.21263.qmail@web34304.mail.mud.yahoo.com>


I also have seen problems related to MGCP in 12.3(14)T
and 12.4 versions before 12.4(5) - haven't got much
exposure to 12.4(5) yet to see how well it is over
time. I find 12.3(11)T to be pretty stable right now
(T6 and T7) compared to newer versions. I did have big
issues with 12.3(11)T8 and analog FXO ports w/MGCP on
multiple routers but 12.3(11)T7 doesn't appear to have
that problem. 12.3(11)T8 also seems to be okay with
PRIs and MGCP. 

Just my 2 cents...

--- Serg <c719562 at gmail.com> wrote:

> Hi TechGuy,
>                    Not sure if you have already
> rectified this problem but i
> can let you know that I had exactly the same problem
> about 6 month ago with
> a Cisco 3845 + NM-HDV2 + PVDM-32 x 2.
> 
> At the time I was running IOS 12.3.14T2 IP-voice
> Feature set.  Looking at
> your "debug isdn q931" debug, the cause code is
> exactly the same as what I
> had.
> 
> Basically, the problem is to do with the E1/T1
> controller's clocking getting
> out of sync with the Carrier's T1/E1.  If you do a
> "show controllers e1" and
> check whether you have excessive  Line Code
> Violations (LCVs), Path Code
> Violations (PCV),  Slip Secs, etc.  These should be
> minimal.
> 
> Cisco may point you to the following BugID but I can
> assure that the work
> around did nothing to fix my problems.
> 
>
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCef89288
> 
> Make sure that you follow the "Clocking
> Configurations On Voice-Capable
> IOS-Based Platforms" (
>
http://www.cisco.com/en/US/products/hw/routers/ps259/products_tech_note09186a008031a072.shtml
> ).
> 
> Also make sure you follows the "Configuring MGCP PRI
> Backhaul and T1 CAS
> Support for Cisco CallManager" (
>
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/ccm_c/intcnf3.htm)
> and that you do not add the "application mgcpapp"
> command to dial peers that
> support PRI backhaul.
> 
> Lastly, the most important step that I undertook was
> to roll-back (yes,
> beleive it or not) to IOS 12.3.11T6 IP-Voice or
> above and I've never had any
> problems since.
> 
> Hope this helps, other Cisco Bugs of interest are:
> 
>
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCEG90199
> 
>
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCEG89017
> 
> Good Luck,
> 
> Serg
> 
> 
> On 12/19/05, cisco-voip-request at puck.nether.net <
> cisco-voip-request at puck.nether.net> wrote:
> >
> > 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. Call rings phones once and then drops
> (TechGuy)
> >    2. Re: Call rings phones once and then drops
> (TechGuy)
> >    3. RE: Call rings phones once and then drops
> (Mark Snow)
> >    4. Tech Prefix from ATA186 (Ciscozest)
> >    5. Re: Call rings phones once and then drops
> (Teodor Georgiev)
> >
> >
> >
>
----------------------------------------------------------------------
> >
> > Message: 1
> > Date: Sat, 17 Dec 2005 13:33:40 -0600
> > From: TechGuy <techguy at gmail.com>
> > Subject: [cisco-voip] Call rings phones once and
> then drops
> > To: cisco-voip at puck.nether.net
> > Message-ID:
> >        
>
<a1b07ea60512171133i79723eecs2d95643eaa5f034f at mail.gmail.com>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Calls ring in on local PRI trunks on 3845 router,
> makes its way to
> > callmanager, makes its way to phone and rings once
> on the phone.  But that
> > is it, calls instantly drop after first ring of
> the phone.  Does not give
> > person chance to answer just drops without any
> user intervention.
> >
> > Using MGCP between 3845 and callmanager.
> >
> > Any ideas?
> > -------------- next part --------------
> > An HTML attachment was scrubbed...
> > URL:
> >
>
https://puck.nether.net/pipermail/cisco-voip/attachments/20051217/a63be268/attachment-0001.html
> >
> > ------------------------------
> >
> > Message: 2
> > Date: Sat, 17 Dec 2005 14:19:32 -0600
> > From: TechGuy <techguy at gmail.com>
> > Subject: Re: [cisco-voip] Call rings phones once
> and then drops
> > To: cisco-voip at puck.nether.net
> > Message-ID:
> >        
>
<a1b07ea60512171219p288d1552qddda2d4820ebba47 at mail.gmail.com>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Here is the debug:
> >
> > emb3845#debug isdn q931
> > debug isdn q931 is              ON.
> > emb3845#
> > *Dec 17 20:07:23.806: ISDN Se0/1/1:23 Q931: RX <-
> SETUP pd = 8  callref =
> > 0x00D2
> >         Bearer Capability i = 0x8090A2
> >                 Standard = CCITT
> >                 Transer Capability = Speech
> >                 Transfer Mode = Circuit
> >                 Transfer Rate = 64 kbit/s
> >         Channel ID i = 0xA98381
> >                 Exclusive, Channel 1
> >         Calling Party Number i = 0x2183,
> '3375015555'
> >                 Plan:ISDN, Type:National
> >         Called Party Number i = 0x80, '4080'
> >                 Plan:Unknown, Type:Unknown
> > *Dec 17 20:07:24.830: ISDN Se0/1/1:23 Q931: TX ->
> RELEASE_COMP pd = 8
> > callref = 0x80D2
> >         Cause i = 0x80A9 - Temporary failure
> > emb3845#
> >
> >
> > More info on the problem.  This router has two
> local PRI trunks coming in
> > to
> > a 2MFT card.  It also has two CAS (E&M Wink)
> trunks coming into a 2MFT
> > card
> > for long distance and incoming toll free.
> >
> > The two CAS trunks work fine.  We are able to get
> incoming calls via the
> > CAS
> > trunks mentioned above with no problems.  Also, we
> have another gateway
> > configured on the same callmanager in a similiar
> fashion and all works
> > fine.  This was all working earlier this morning. 
> No known changes
> > occured
> > between when it worked and stopped working.
> >
> > Quite an interesting problem, we have TAC on the
> line now but thought I
> > would see what others think.
> > -------------- next part --------------
> > An HTML attachment was scrubbed...
> > URL:
> >
>
https://puck.nether.net/pipermail/cisco-voip/attachments/20051217/3bcb8972/attachment-0001.html
> >
> > ------------------------------
> >
> >
> > _______________________________________________
> 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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From it at cimgroup.com  Wed Dec 21 00:10:33 2005
From: it at cimgroup.com (IT)
Date: Wed Dec 21 00:06:36 2005
Subject: [cisco-voip] CDR Error
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0265D266@exchange.cimgroup.com>

It appears that none of my reports contain any data when I use CAR, even
though I enabled the CDR loading.

Here is the error I have in ARTSchError.log

DirUser.OperationError

            at DirUser.LoadComClass.retAupairClass(LoadComClass.java)

            at DirUser.DirUser.<init>(DirUser.java)

            at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java:137)

            at
com.cisco.art.controller.CtrLoader.run(CtrLoader.java:1356)

            at com.cisco.art.scheduler.SchThread.run(SchThread.java:550)

DirUser.OperationError

            at DirUser.LoadComClass.retAupairClass(LoadComClass.java)

            at DirUser.DirUser.<init>(DirUser.java)

            at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java:137)

            at
com.cisco.art.controller.CtrLoader.run(CtrLoader.java:1356)

            at com.cisco.art.scheduler.SchThread.run(SchThread.java:550)

 

I have Active Directory integration installed with CM4.0. Has anyone
seen this issue before?

 

Thanks,

Avidan 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
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From rratliff at cisco.com  Wed Dec 21 00:23:07 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Dec 21 00:23:10 2005
Subject: [cisco-voip] CDR Error
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B0265D266@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B0265D266@exchange.cimgroup.com>
Message-ID: <A1BB653D-A834-44FA-A83F-EFF248A67B77@cisco.com>

DirUser is the API ART uses for ldap access.

Are you even able to login to the ART web page?

Many times we see issues where the CDRs are being loaded just not  
associated the correct user.
Are your reports user or department based?

If so, try doing a CDR search on your extension to see if it finds  
calls.

There is a known issue with ART where when it goes to retrieve the  
user a device is associated to if the directory returns multiple  
users ART will take the first one.  This can cause issues with IPCC  
agents, AC users, etc where you have lots of devices associated with  
application users.  The "owner userid" field on the phone device page  
was created to workaround this.

Of course none of this may be relevant to your issue but it's a  
common problem.

-Ryan

On Dec 21, 2005, at 12:10 AM, IT wrote:

It appears that none of my reports contain any data when I use CAR,  
even though I enabled the CDR loading.

Here is the error I have in ARTSchError.log

DirUser.OperationError

             at DirUser.LoadComClass.retAupairClass(LoadComClass.java)

             at DirUser.DirUser.<init>(DirUser.java)

             at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java: 
137)

             at com.cisco.art.controller.CtrLoader.run(CtrLoader.java: 
1356)

             at com.cisco.art.scheduler.SchThread.run(SchThread.java: 
550)

DirUser.OperationError

             at DirUser.LoadComClass.retAupairClass(LoadComClass.java)

             at DirUser.DirUser.<init>(DirUser.java)

             at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java: 
137)

             at com.cisco.art.controller.CtrLoader.run(CtrLoader.java: 
1356)

             at com.cisco.art.scheduler.SchThread.run(SchThread.java: 
550)



I have Active Directory integration installed with CM4.0. Has anyone  
seen this issue before?



Thanks,

Avidan

******************************************************
This email may contain trade secrets and/or other information that is
proprietary and/or confidential. Use of this by anyone other than the
intended recipient(s) is prohibited and may be unlawful. If you are not
the intended recipient, delete the e-mail and notify us immediately.
******************************************************

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


From akoul at dataways.gr  Wed Dec 21 02:21:02 2005
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Wed Dec 21 02:21:14 2005
Subject: [cisco-voip] Inconsistent CCM version and CRS
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A12117@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A12117@VISHNU.wesd.org>
Message-ID: <43A9025E.20709@dataways.gr>

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From subs-cisco at cbvcollections.com  Wed Dec 21 02:59:15 2005
From: subs-cisco at cbvcollections.com (Cisco)
Date: Wed Dec 21 02:59:31 2005
Subject: [cisco-voip] CallManager Route Pattern/Filter Problem...
Message-ID: <9AEF8665DE74FA43B3B987FB931E92AE0ADAB6@CBVVANEM01.cbv.dns>

I forgot to write and thank the two of you (and the list) for your help.
All gateways in each site are now routing calls using the 9.@ route
patterns/filters.  The satellite problem was due to a lack of calling
party ID, not called.  Setting plan types to unknown for all gateways
worked just fine.
 
P.S. Mark - I'll definitely look into your course offerings...


________________________________

From: Mark Snow [mailto:highspeedsnow at gmail.com] 
Sent: Saturday, December 17, 2005 5:44 AM
To: Cisco; 'Lelio Fulgenzi'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...



That sattelilte most likely uses a different telco POP and the telly's
switch wants to see both - switch the Plan/Type back to ISDN/National or
CallManager/CallManager but for that GW only and you should be fine.

 

HTH

 

-Mark Snow

 

________________________________

From: Cisco [mailto:subs-cisco at cbvcollections.com] 
Sent: Saturday, December 17, 2005 3:23 AM
To: Lelio Fulgenzi; Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

 

Argh - one more thing.  We have a satellite office that uses the same CM
cluster which experienced the same problem - with called numbering plan
on CallManager all calls matching route pattern 9.@ that included 1
played the carrier message (you must dial 1 first...).  

 

I tried adjusting the called/calling party numbering plans to unknown as
I did the local 6608 gateways.  Once the gateway was adjusted all
dialled numbers called that are matched 9.@ that include the "1" were
routed to the gateway, but then dropped.  The main different there is
that the gateway is a 2691. 

 

Thoughts - ideas?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
Sent: Friday, December 16, 2005 10:58 PM
To: Lelio Fulgenzi; Mark Snow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager Route Pattern/Filter Problem...

Yes, I'm with Bell (West) in Vancouver.  We have another CM cluster in
the east (Toronto) who is also with Bell.  The cluster in Toronto is
using a different PRI protocol (DMS-100) then the Vancouver is using
(NI2).

 

Well it is late at night so I've been playing with the gateway configs
and trying your suggestion.  After changing them to unknown - it worked!
Can you tell me why?  Also what should the setting for "Called party IE
number type unknown*" be?

 

P.S.  I will definitely look into your site/course offering!

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Friday, December 16, 2005 9:35 PM
To: Mark Snow; Cisco; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager Route Pattern/Filter Problem...

You cannot do a debug isdn, but you can import the CCM trace files into
the q931 debug tool and read the output that way. It's a pretty good
tool and I've used it a few times. That's the standalone tool, not the
one that is built into CallManager.

 

And our numbering plans are set to CallManager.

 

Frank - are you in Canada? What is your provider? Bell? I can provide
with all of our 6608 parameters if that would help.

	----- Original Message ----- 

	From: Mark Snow <mailto:highspeedsnow at gmail.com>  

	To: 'Cisco' <mailto:subs-cisco at cbvcollections.com>  ;
cisco-voip at puck.nether.net 

	Sent: Friday, December 16, 2005 11:41 PM

	Subject: RE: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	 

	Lelio is correct, make sure that digit manipulation is not
occurring at the Route Pattern, OR within the Route Groups inside the
Route Lists.

	 

	If those are not stripping anything but PreDot then try this:

	Try changing:

	*	Called Numbering Plan*: unknown 
	*	Calling Numbering Plan*: unknown 

	 

	Then test your calls - see if they work - you will have to do a
reset on the 6608 gateway first, so make sure that there are no calls
going on at the time.

	 

	Also, since you are running a 6608 blade on a 6500 Cat, you
cannot do the 'debug isdn q931' command I asked about previously.

	 

	LMK if that helps ...

	 

	And hey, if you do want to become a Voice Expert soon, check out
my class sometime:

	http://www.ipexpert.com/products_services/product.asp?sku=ip0012

	 

	-Mark Snow

	CCIE Voice # 14073

	 

	 

	
________________________________


	From: Cisco [mailto:subs-cisco at cbvcollections.com] 
	Sent: Friday, December 16, 2005 8:17 PM
	To: Mark Snow; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	 

	Well I'm far from the telephony expert I want to be (at this
point - give me a few months) but it would appear that you may be on to
something.  I took a look at the gateway configurations within
CallManager.  Each of the gateways within the PSTN group (Product :
Cisco Catalyst 6000 T1 VoIP Gateway - Device Protocol: Digital Access
PRI) have the same configuration.  The following settings are currently
used from the Call Routing Information (Outbound):

	*	Calling Line ID Presentation*: Restricted 
	*	Calling Party Selection*: Originator 
	*	Called party IE number type unknown*: Cisco CallManager 
	*	Calling party IE number type unknown*: National 
	*	Called Numbering Plan*: Cisco CallManager 
	*	Calling Numbering Plan*: ISDN 
	*	Number of digits to strip*: 0 
	*	Caller ID DN: 
	*	SMDI Base Port*: 

	Do you think the underlined (above) is causing the issue?  Is
this something I can change (to "Cisco CallManager") or do I need to
involve our provider?

	 

	
________________________________


	From: Mark Snow [mailto:highspeedsnow at gmail.com] 
	Sent: December 16, 2005 4:59 PM
	To: Cisco; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	Is it a PRI that the call is going out of on an IOS GW?

	If so, have you done a 'debug isdn q931' to ensure that in fact
the CCM is dropping the 1?

	 

	-Mark Snow

	CCIE Voice # 14073

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Cisco
	Sent: Friday, December 16, 2005 7:24 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] CallManager Route Pattern/Filter
Problem...

	 

	I'm trying to use route filters (area code == 204 OR etc.) with
a "9.@" pattern instead of the more difficult to manage 9.1204XXXXXXX to
route long distance calls.  

	 

	The pattern/filter work fine, the problem is that when the
number is matched using 9.@ pattern (verified with the Cisco the dialled
number analyser), by the time it reaches the local provider somehow the
"1" is dropped and the caller is presented with a message "you must dial
1 to access a long distance number" from the local provider.  If I
change the pattern back to 9.1XXXXXXXXXX then the number is dialled as
expected.

	 

	I've used the dialled number analyser to ensure that no
transformations are taking place at any point along the route to the
PEST gateway.  According to the analyser the dialled number that is sent
to the PEST gateway in either case is the same.  However, one works and
the other does not.

	 

	I should also note that the 9.@ pattern with a filter of (local
area code == 604 OR etc.) for local 10 digit dialling works just fine as
either 9.@ or 9.XXXXXXXXXX.  Both are routed as expected.

	 

	Ideas?

	
________________________________

 <http://www.cbvcollections.com/cbvImages/ecardlogo.jpg> 

Frank Wakelin
Senior Network/Security Analyst

100 - 814 Richards St.
Vancouver, BC
V6B 3A7

Telephone: (604) 661-7906
fwakelin at cbvcollections.com

	 

	Confidentiality Warning:
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	Information Confidentielle:
	Le present message, ainsi que tout fichier qui y est joint, est
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et supprimer ce message et tout document joint de votre systeme. Merci

	
________________________________


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From robin.inderberg at candidator.se  Wed Dec 21 05:10:11 2005
From: robin.inderberg at candidator.se (Robin Inderberg)
Date: Wed Dec 21 05:04:48 2005
Subject: [cisco-voip] Changes not being activated
Message-ID: <39EB1FE59758D9448A5D0B348857C7273E5755@cannt04.candidator.se>

Hi,

 

I cant manage this one by my self, don?t realy know what could be wrong. And I think that one of you has the answer (as always) :-).

 

I have a DN that a customer often uses CFA on, he often sets CFA to his cellphone, everything worked well, he could add a CFA number and disable it. But now the change does not take any effect, he can disable it, everything looks fine in the phone. But the CFA doesn't get disabled. Now I have checked the Enterprise Manager and looked for a CFA on his DN, but could not find anything. I tested to remove his DN, and just see if the CFA or the change did get power, but no. The DN is there, and forwards to his cellphone. 

 

 

I have tried to restart the DBL monitor, but it didn't help.

 

Anyone who has an idea of what could cause this one?

 

 

Best Regards

 

 

 

 

Robin Inderberg

robin.inderberg at candidator.se

 

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From robin.inderberg at candidator.se  Wed Dec 21 05:32:32 2005
From: robin.inderberg at candidator.se (Robin Inderberg)
Date: Wed Dec 21 05:27:08 2005
Subject: VB: [cisco-voip] Changes not being activated
Message-ID: <39EB1FE59758D9448A5D0B348857C7273E575F@cannt04.candidator.se>

Hi once again,

 

Just a little update on the case, no changes is getting understood by the CCM. The changes is visible on the ccmadmin, and I can add phones, change things on DN, but the changes does not get active. 

 

Regards once again

 

 

/Robin

________________________________

Fr?n: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] F?r Robin Inderberg
Skickat: den 21 december 2005 11:10
Till: cisco-voip at puck.nether.net
?mne: [cisco-voip] Changes not being activated

 

Hi,

 

I cant manage this one by my self, don?t realy know what could be wrong. And I think that one of you has the answer (as always) :-).

 

I have a DN that a customer often uses CFA on, he often sets CFA to his cellphone, everything worked well, he could add a CFA number and disable it. But now the change does not take any effect, he can disable it, everything looks fine in the phone. But the CFA doesn't get disabled. Now I have checked the Enterprise Manager and looked for a CFA on his DN, but could not find anything. I tested to remove his DN, and just see if the CFA or the change did get power, but no. The DN is there, and forwards to his cellphone. 

 

 

I have tried to restart the DBL monitor, but it didn't help.

 

Anyone who has an idea of what could cause this one?

 

 

Best Regards

 

 

 

 

Robin Inderberg

robin.inderberg at candidator.se

 

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From rleetun at co.boulder.co.us  Wed Dec 21 08:33:23 2005
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Wed Dec 21 08:33:30 2005
Subject: [cisco-voip] Re: Call rings phones once and then drops
Message-ID: <963CE94787005149A52C17F33A028E12051D416B@EVS1B.boco.co.boulder.co.us>

We have been working with Cisco to resolve DSP issues.  This ver might
help your situation?  12.2(8r) T2 and we are running h.323/MGCP with
this ver 12.0(6r)T with no problems.

And, my 2 cents too... 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
Sent: Tuesday, December 20, 2005 9:40 PM
To: Serg; techguy at gmail.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Re: Call rings phones once and then drops


I also have seen problems related to MGCP in 12.3(14)T and 12.4 versions
before 12.4(5) - haven't got much exposure to 12.4(5) yet to see how
well it is over time. I find 12.3(11)T to be pretty stable right now
(T6 and T7) compared to newer versions. I did have big issues with
12.3(11)T8 and analog FXO ports w/MGCP on multiple routers but
12.3(11)T7 doesn't appear to have that problem. 12.3(11)T8 also seems to
be okay with PRIs and MGCP. 

Just my 2 cents...

--- Serg <c719562 at gmail.com> wrote:

> Hi TechGuy,
>                    Not sure if you have already rectified this problem

> but i can let you know that I had exactly the same problem about 6 
> month ago with a Cisco 3845 + NM-HDV2 + PVDM-32 x 2.
> 
> At the time I was running IOS 12.3.14T2 IP-voice Feature set.  Looking

> at your "debug isdn q931" debug, the cause code is exactly the same as

> what I had.
> 
> Basically, the problem is to do with the E1/T1 controller's clocking 
> getting out of sync with the Carrier's T1/E1.  If you do a "show 
> controllers e1" and check whether you have excessive  Line Code 
> Violations (LCVs), Path Code Violations (PCV),  Slip Secs, etc.  These

> should be minimal.
> 
> Cisco may point you to the following BugID but I can assure that the 
> work around did nothing to fix my problems.
> 
>
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCef89288
> 
> Make sure that you follow the "Clocking Configurations On 
> Voice-Capable IOS-Based Platforms" (
>
http://www.cisco.com/en/US/products/hw/routers/ps259/products_tech_note0
9186a008031a072.shtml
> ).
> 
> Also make sure you follows the "Configuring MGCP PRI Backhaul and T1 
> CAS Support for Cisco CallManager" (
>
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/
vvfax_c/callc_c/ccm_c/intcnf3.htm)
> and that you do not add the "application mgcpapp"
> command to dial peers that
> support PRI backhaul.
> 
> Lastly, the most important step that I undertook was to roll-back 
> (yes, beleive it or not) to IOS 12.3.11T6 IP-Voice or above and I've 
> never had any problems since.
> 
> Hope this helps, other Cisco Bugs of interest are:
> 
>
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCEG90199
> 
>
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCEG89017
> 
> Good Luck,
> 
> Serg
> 
> 
> On 12/19/05, cisco-voip-request at puck.nether.net < 
> cisco-voip-request at puck.nether.net> wrote:
> >
> > 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. Call rings phones once and then drops
> (TechGuy)
> >    2. Re: Call rings phones once and then drops
> (TechGuy)
> >    3. RE: Call rings phones once and then drops
> (Mark Snow)
> >    4. Tech Prefix from ATA186 (Ciscozest)
> >    5. Re: Call rings phones once and then drops
> (Teodor Georgiev)
> >
> >
> >
>
----------------------------------------------------------------------
> >
> > Message: 1
> > Date: Sat, 17 Dec 2005 13:33:40 -0600
> > From: TechGuy <techguy at gmail.com>
> > Subject: [cisco-voip] Call rings phones once and
> then drops
> > To: cisco-voip at puck.nether.net
> > Message-ID:
> >        
>
<a1b07ea60512171133i79723eecs2d95643eaa5f034f at mail.gmail.com>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Calls ring in on local PRI trunks on 3845 router,
> makes its way to
> > callmanager, makes its way to phone and rings once
> on the phone.  But that
> > is it, calls instantly drop after first ring of
> the phone.  Does not give
> > person chance to answer just drops without any
> user intervention.
> >
> > Using MGCP between 3845 and callmanager.
> >
> > Any ideas?
> > -------------- next part -------------- An HTML attachment was 
> > scrubbed...
> > URL:
> >
>
https://puck.nether.net/pipermail/cisco-voip/attachments/20051217/a63be2
68/attachment-0001.html
> >
> > ------------------------------
> >
> > Message: 2
> > Date: Sat, 17 Dec 2005 14:19:32 -0600
> > From: TechGuy <techguy at gmail.com>
> > Subject: Re: [cisco-voip] Call rings phones once
> and then drops
> > To: cisco-voip at puck.nether.net
> > Message-ID:
> >        
>
<a1b07ea60512171219p288d1552qddda2d4820ebba47 at mail.gmail.com>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Here is the debug:
> >
> > emb3845#debug isdn q931
> > debug isdn q931 is              ON.
> > emb3845#
> > *Dec 17 20:07:23.806: ISDN Se0/1/1:23 Q931: RX <-
> SETUP pd = 8  callref =
> > 0x00D2
> >         Bearer Capability i = 0x8090A2
> >                 Standard = CCITT
> >                 Transer Capability = Speech
> >                 Transfer Mode = Circuit
> >                 Transfer Rate = 64 kbit/s
> >         Channel ID i = 0xA98381
> >                 Exclusive, Channel 1
> >         Calling Party Number i = 0x2183,
> '3375015555'
> >                 Plan:ISDN, Type:National
> >         Called Party Number i = 0x80, '4080'
> >                 Plan:Unknown, Type:Unknown *Dec 17 20:07:24.830: 
> > ISDN Se0/1/1:23 Q931: TX ->
> RELEASE_COMP pd = 8
> > callref = 0x80D2
> >         Cause i = 0x80A9 - Temporary failure emb3845#
> >
> >
> > More info on the problem.  This router has two
> local PRI trunks coming in
> > to
> > a 2MFT card.  It also has two CAS (E&M Wink)
> trunks coming into a 2MFT
> > card
> > for long distance and incoming toll free.
> >
> > The two CAS trunks work fine.  We are able to get
> incoming calls via the
> > CAS
> > trunks mentioned above with no problems.  Also, we
> have another gateway
> > configured on the same callmanager in a similiar
> fashion and all works
> > fine.  This was all working earlier this morning. 
> No known changes
> > occured
> > between when it worked and stopped working.
> >
> > Quite an interesting problem, we have TAC on the
> line now but thought I
> > would see what others think.
> > -------------- next part --------------
> > An HTML attachment was scrubbed...
> > URL:
> >
>
https://puck.nether.net/pipermail/cisco-voip/attachments/20051217/3bcb89
72/attachment-0001.html
> >
> > ------------------------------
> >
> >
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


__________________________________________________
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From rleetun at co.boulder.co.us  Wed Dec 21 08:34:32 2005
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Wed Dec 21 08:34:37 2005
Subject: [cisco-voip] Re: Call rings phones once and then drops
Message-ID: <963CE94787005149A52C17F33A028E12051D416C@EVS1B.boco.co.boulder.co.us>

By the way, we were running 12.4 and tried 12.3(14)T and had many
issues. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
Sent: Tuesday, December 20, 2005 9:40 PM
To: Serg; techguy at gmail.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Re: Call rings phones once and then drops


I also have seen problems related to MGCP in 12.3(14)T and 12.4 versions
before 12.4(5) - haven't got much exposure to 12.4(5) yet to see how
well it is over time. I find 12.3(11)T to be pretty stable right now
(T6 and T7) compared to newer versions. I did have big issues with
12.3(11)T8 and analog FXO ports w/MGCP on multiple routers but
12.3(11)T7 doesn't appear to have that problem. 12.3(11)T8 also seems to
be okay with PRIs and MGCP. 

Just my 2 cents...

--- Serg <c719562 at gmail.com> wrote:

> Hi TechGuy,
>                    Not sure if you have already rectified this problem

> but i can let you know that I had exactly the same problem about 6 
> month ago with a Cisco 3845 + NM-HDV2 + PVDM-32 x 2.
> 
> At the time I was running IOS 12.3.14T2 IP-voice Feature set.  Looking

> at your "debug isdn q931" debug, the cause code is exactly the same as

> what I had.
> 
> Basically, the problem is to do with the E1/T1 controller's clocking 
> getting out of sync with the Carrier's T1/E1.  If you do a "show 
> controllers e1" and check whether you have excessive  Line Code 
> Violations (LCVs), Path Code Violations (PCV),  Slip Secs, etc.  These

> should be minimal.
> 
> Cisco may point you to the following BugID but I can assure that the 
> work around did nothing to fix my problems.
> 
>
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCef89288
> 
> Make sure that you follow the "Clocking Configurations On 
> Voice-Capable IOS-Based Platforms" (
>
http://www.cisco.com/en/US/products/hw/routers/ps259/products_tech_note0
9186a008031a072.shtml
> ).
> 
> Also make sure you follows the "Configuring MGCP PRI Backhaul and T1 
> CAS Support for Cisco CallManager" (
>
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/
vvfax_c/callc_c/ccm_c/intcnf3.htm)
> and that you do not add the "application mgcpapp"
> command to dial peers that
> support PRI backhaul.
> 
> Lastly, the most important step that I undertook was to roll-back 
> (yes, beleive it or not) to IOS 12.3.11T6 IP-Voice or above and I've 
> never had any problems since.
> 
> Hope this helps, other Cisco Bugs of interest are:
> 
>
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCEG90199
> 
>
http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCEG89017
> 
> Good Luck,
> 
> Serg
> 
> 
> On 12/19/05, cisco-voip-request at puck.nether.net < 
> cisco-voip-request at puck.nether.net> wrote:
> >
> > 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. Call rings phones once and then drops
> (TechGuy)
> >    2. Re: Call rings phones once and then drops
> (TechGuy)
> >    3. RE: Call rings phones once and then drops
> (Mark Snow)
> >    4. Tech Prefix from ATA186 (Ciscozest)
> >    5. Re: Call rings phones once and then drops
> (Teodor Georgiev)
> >
> >
> >
>
----------------------------------------------------------------------
> >
> > Message: 1
> > Date: Sat, 17 Dec 2005 13:33:40 -0600
> > From: TechGuy <techguy at gmail.com>
> > Subject: [cisco-voip] Call rings phones once and
> then drops
> > To: cisco-voip at puck.nether.net
> > Message-ID:
> >        
>
<a1b07ea60512171133i79723eecs2d95643eaa5f034f at mail.gmail.com>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Calls ring in on local PRI trunks on 3845 router,
> makes its way to
> > callmanager, makes its way to phone and rings once
> on the phone.  But that
> > is it, calls instantly drop after first ring of
> the phone.  Does not give
> > person chance to answer just drops without any
> user intervention.
> >
> > Using MGCP between 3845 and callmanager.
> >
> > Any ideas?
> > -------------- next part -------------- An HTML attachment was 
> > scrubbed...
> > URL:
> >
>
https://puck.nether.net/pipermail/cisco-voip/attachments/20051217/a63be2
68/attachment-0001.html
> >
> > ------------------------------
> >
> > Message: 2
> > Date: Sat, 17 Dec 2005 14:19:32 -0600
> > From: TechGuy <techguy at gmail.com>
> > Subject: Re: [cisco-voip] Call rings phones once
> and then drops
> > To: cisco-voip at puck.nether.net
> > Message-ID:
> >        
>
<a1b07ea60512171219p288d1552qddda2d4820ebba47 at mail.gmail.com>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > Here is the debug:
> >
> > emb3845#debug isdn q931
> > debug isdn q931 is              ON.
> > emb3845#
> > *Dec 17 20:07:23.806: ISDN Se0/1/1:23 Q931: RX <-
> SETUP pd = 8  callref =
> > 0x00D2
> >         Bearer Capability i = 0x8090A2
> >                 Standard = CCITT
> >                 Transer Capability = Speech
> >                 Transfer Mode = Circuit
> >                 Transfer Rate = 64 kbit/s
> >         Channel ID i = 0xA98381
> >                 Exclusive, Channel 1
> >         Calling Party Number i = 0x2183,
> '3375015555'
> >                 Plan:ISDN, Type:National
> >         Called Party Number i = 0x80, '4080'
> >                 Plan:Unknown, Type:Unknown *Dec 17 20:07:24.830: 
> > ISDN Se0/1/1:23 Q931: TX ->
> RELEASE_COMP pd = 8
> > callref = 0x80D2
> >         Cause i = 0x80A9 - Temporary failure emb3845#
> >
> >
> > More info on the problem.  This router has two
> local PRI trunks coming in
> > to
> > a 2MFT card.  It also has two CAS (E&M Wink)
> trunks coming into a 2MFT
> > card
> > for long distance and incoming toll free.
> >
> > The two CAS trunks work fine.  We are able to get
> incoming calls via the
> > CAS
> > trunks mentioned above with no problems.  Also, we
> have another gateway
> > configured on the same callmanager in a similiar
> fashion and all works
> > fine.  This was all working earlier this morning. 
> No known changes
> > occured
> > between when it worked and stopped working.
> >
> > Quite an interesting problem, we have TAC on the
> line now but thought I
> > would see what others think.
> > -------------- next part --------------
> > An HTML attachment was scrubbed...
> > URL:
> >
>
https://puck.nether.net/pipermail/cisco-voip/attachments/20051217/3bcb89
72/attachment-0001.html
> >
> > ------------------------------
> >
> >
> > _______________________________________________
> 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  Wed Dec 21 09:19:15 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Dec 21 09:19:16 2005
Subject: VB: [cisco-voip] Changes not being activated
In-Reply-To: <39EB1FE59758D9448A5D0B348857C7273E575F@cannt04.candidator.se>
References: <39EB1FE59758D9448A5D0B348857C7273E575F@cannt04.candidator.se>
Message-ID: <5F9EDC6A-1C8A-4ED2-8B28-A6C844AA0A37@cisco.com>

What version of CM are you running?

There are a few ways change notification can break in CM 4.x.  The  
first thing to try would be to restart the Cisco Database Layer  
Monitor service on the publisher and then try to update a phone via  
the CCMAdmin.  If you can even reset a phone via CCMAdmin then change  
notification is working.

-Ryan

On Dec 21, 2005, at 5:32 AM, Robin Inderberg wrote:

Hi once again,



Just a little update on the case, no changes is getting understood by  
the CCM. The changes is visible on the ccmadmin, and I can add  
phones, change things on DN, but the changes does not get active.



Regards once again





/Robin

Fr?n: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] F?r Robin Inderberg
Skickat: den 21 december 2005 11:10
Till: cisco-voip at puck.nether.net
?mne: [cisco-voip] Changes not being activated



Hi,



I cant manage this one by my self, don?t realy know what could be  
wrong. And I think that one of you has the answer (as always) J.



I have a DN that a customer often uses CFA on, he often sets CFA to  
his cellphone, everything worked well, he could add a CFA number and  
disable it. But now the change does not take any effect, he can  
disable it, everything looks fine in the phone. But the CFA doesn?t  
get disabled. Now I have checked the Enterprise Manager and looked  
for a CFA on his DN, but could not find anything. I tested to remove  
his DN, and just see if the CFA or the change did get power, but no.  
The DN is there, and forwards to his cellphone.





I have tried to restart the DBL monitor, but it didn?t help.



Anyone who has an idea of what could cause this one?





Best Regards









Robin Inderberg

robin.inderberg at candidator.se



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



From PRuttman at foley.com  Wed Dec 21 09:43:57 2005
From: PRuttman at foley.com (Ruttman, Peter G.)
Date: Wed Dec 21 09:43:55 2005
Subject: [cisco-voip] Busy out channels on an mgcp gateway
Message-ID: <7E3064B7EFC16C41B185C41D141008B7C3CE83@foleyex97.foleylaw.com>

I need to reset an mgcp gateway PRI that has inbound DIDs associated
with it.  Is it possible to busy out the channels on it so that new
calls do not get pinned up on it?  Then I can safely reset the gateway
during work hours.

Pete

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 lelio at uoguelph.ca  Wed Dec 21 09:55:55 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Dec 21 09:56:00 2005
Subject: [cisco-voip] Busy out channels on an mgcp gateway
References: <7E3064B7EFC16C41B185C41D141008B7C3CE83@foleyex97.foleylaw.com>
Message-ID: <008f01c6063e$a56c1220$30196883@cfs.uoguelph.ca>

Busy out channels on an mgcp gatewayI've heard people use the following service parameter to do what you are asking. There are five of them available I think.


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


      Change B-Channel Maintenance Status 1 : This parameter allows Cisco CallManager to change individual B-Channel maintenance status for PRI interface in real time for troubleshooting. Input Format:
      Device Name = B-channel Maintenance Status
      Example:
      S0/DS1-0 at SDA123456789ABC = 0000 1111 0000 0000 0000 0001(24 bits for T1. Channel number begins with 1 to 24 from left to right. The last one specifies the D-Channel, which does not get affected. In this example, 5th-8th B-Channels are out of service).
      or,
      S1/DS1-0 at DLS2-CM136-VG200-2.SELSIUSCMLAB.CISCO.COM = 0000 0000 0000 0000 0000 0000 1111 1111(32 bits for E1. B-Channel number begins with 1 to 32 from left to right. The 16th channel specifies D-Channel. The last bit does not affect any channel. In this example, 25th-31st channels are marked out of service).
      Explanation:
      The DEVICE NAME MUST exactly match the gateway name on the top of the gateway configuration web page. Copy and paste that field into this service parameter to avoid any manual inputting mistake. Ensure '=' is mandatory and unique to distinguish the DEVICE NAME and BCHANNEL MAINTENANCE STATUS fields. BCHANNEL MAINTENANCE STATUS = xxxx xxxx xxxx xxxx xxxx xxxx xxxx xxxx 
      Where x can be: 
      0 -- In service 
      1 -- Graceful out of service (Change channel status until active call ends if an active call exists on that channel) 
      2 -- Forceful out of service (Tear down active call first; then, change channel status immediately if an active call exists on that channel).
      The system treats any other value as invalid and the value does not take effect. Make sure that the total number of 'x' is either 24 for T1 or 32 for E1. Any other invalid length or mismatch (for example, 24 for E1) gets treated as error, and no action gets taken for that device. 
      SPACE: Optional.
      The spaces between device name and '=', between '=' and BCHANNEL MAINTENANCE STATUS, and between x and x, are all optional. You cannot use any space inside the DEVICE NAME field. 
      To use this feature, the "Enable Status Poll" flag on the gateway configuration page must be checked and updated, then the gateway must be reset for the first time.After this, changing service parameter will take effect without resetting the gateway. 
      Maximum length: 255.
     

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

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Ruttman, Peter G. 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, December 21, 2005 9:43 AM
  Subject: [cisco-voip] Busy out channels on an mgcp gateway


  I need to reset an mgcp gateway PRI that has inbound DIDs associated with it.  Is it possible to busy out the channels on it so that new calls do not get pinned up on it?  Then I can safely reset the gateway during work hours.

  Pete 


  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 smgustafson at gmail.com  Wed Dec 21 10:17:21 2005
From: smgustafson at gmail.com (Steve G)
Date: Wed Dec 21 10:17:27 2005
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 110
In-Reply-To: <200512202032.jBKKWF0I068418@puck.nether.net>
References: <200512202032.jBKKWF0I068418@puck.nether.net>
Message-ID: <8a1f34b50512210717y689b9d94g923adf4602ba1b25@mail.gmail.com>

>
> Not to keep dragging this thing out.  This I feel is important to know if
> you are going to use a 35XX MCU.

"The IP/VC 3511 can be configured to run in either SCCP mode or H.323 mode,
while the IP/VC 3540 model can be configured to run both protocols
simultaneously and divide the total number of available MP resources between
the two."

from:
http://www.cisco.com/application/pdf/en/us/guest/netsol/ns268/c649/ccmigration_09186a00804ff6ba.pdf

This is important if you are using tandberg, VTA, or the 7985 video phone in
conjunction with Polycom or other H.323 endpoints.

Steve


---------- Forwarded message ----------
> From: "Voll, Scott" <Scott.Voll at wesd.org>
> To: "IT" <it at cimgroup.com>, "Ortiz, Carlos" <CORTIZ at broward.org>, <
> cisco-voip at puck.nether.net>
> Date: Tue, 20 Dec 2005 12:15:32 -0800
> Subject: RE: [cisco-voip] Video conf. equipment for IP/VC
>
> The answer is, it depends.  If the unit that has the built in MCU is
> always in the video conference you can probably get away with out a 3511 so
> long as all the units are h323. now if you are wanting to use CM VT
> advantage you will most likely still need a MCU (35xx)  I don't' know about
> the Tandberg as it does have a built in MCU and can register to CM.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* IT [mailto:it at cimgroup.com]
> *Sent:* Tuesday, December 20, 2005 12:16 PM
> *To:* Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> And if it has an MCU built-in, do I still need a IP/VC 3511?
>
>
>  ------------------------------
>
> *From:* Voll, Scott [mailto:Scott.Voll at wesd.org]
> *Sent:* Tuesday, December 20, 2005 9:14 AM
> *To:* IT; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> All we have done is demo'd the Tandberg so far, but it does have an MCU
> built in.  A lot of polycoms do also.  The nice thing about the Tandberg is
> that you can also interface with CM.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* IT [mailto:it at cimgroup.com]
> *Sent:* Monday, December 19, 2005 8:43 PM
> *To:* Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> This sounds like more work than I am interested in. Is there a better
> Video conference option to use with the 3511?
>
> Anyone have experience with the Tandberg?
>
> Thanks,
>
> Avidan
>
>
>  ------------------------------
>
> *From:* Voll, Scott [mailto:Scott.Voll at wesd.org]
> *Sent:* Monday, December 19, 2005 11:36 AM
> *To:* IT; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
>
>
> We have the polycom sp and it does NOT have an interface with CM.  you
> will most likely have to use the MCU to connect to a CM video conference via
> a gatekeeper.
>
>
>
> Some MCU do have a built in Gatekeeper.  I suggest just getting your self
> an old 26xx series router and the IP Plus IOS and doing it with one of
> these.
>
>
>
> The MCU will just be a e.164 off the gatekeeper.  Each h323 video unit
> will just dial the e164 to get into the conference.
>
>
>
> Just as an FYI you will most likely have to exchange gatekeeper info with
> other company's if you are plan to have them dial into your conferences.
>
>
>
> Scott
>  ------------------------------
>
> *From:* IT [mailto:it at cimgroup.com]
> *Sent:* Monday, December 19, 2005 10:27 AM
> *To:* Ortiz, Carlos; Voll, Scott; IT; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> I intend on doing multi-point conferencing. Right now, I just have a
> Viewstation EX, so I only do point-to-point conferences with other
> companies. I am putting in an order for a 3511 and 2 more video conference
> cameras, so my 2 biggest branch offices. I am hoping to have a 3-way video
> conference happening by the end of January.  However, I am having a very
> hard time just connecting my Polycom Viewstation to CallManager. Is it even
> connectable? It seems the viewstation only has a "gatekeeper" IP setting for
> what I assume will be the IP of the 3511.  Where do I set the IP of
> CallManager on this polycom?
>
>
>
> Thanks,
>
> Avidan
>
>
>  ------------------------------
>
> *From:* Ortiz, Carlos [mailto:CORTIZ at broward.org]
> *Sent:* Monday, December 19, 2005 7:48 AM
> *To:* Voll, Scott; IT; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> Are you going to be doing Multipoint conferencing?  IHave you set your
> stuff up yet?
>
>
>
> Carlos
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Voll, Scott
> *Sent:* Monday, December 19, 2005 10:46 AM
> *To:* IT; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> We currently have a couple of Polycoms but will be getting a Tandberg
> shortly.  Both seem to be good.  It really comes down to features.  I'm
> hoping I can get rid of the old Vtel units we have soon.  Stay away from
> VTEL!!!!!!!!!!!!!
>
>
>
> Nice thing with the Tandberg we demo'd the other day is it can register
> directly with CM.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *IT
> *Sent:* Sunday, December 18, 2005 6:35 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
> video conference needs. I currently have a Polycom Viewstation EX, but with
> the upcoming IP/VC installation, I didnt know if buying 3 more of those was
> the right idea. What is everyone using out there for video conferencing with
> their IP/VC? Tandbergs? Polycoms?  Does getting MeetingPlace really help in
> small conferences?
>
> thanks for the advice.
> avidan
>
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
>
>
> ---------- Forwarded message ----------
> From: "IT" <it at cimgroup.com>
> To: "Voll, Scott" <Scott.Voll at wesd.org>, "IT" <it at cimgroup.com>, "Ortiz,
> Carlos" <CORTIZ at broward.org>, <cisco-voip at puck.nether.net>
> Date: Tue, 20 Dec 2005 12:36:11 -0800
> Subject: RE: [cisco-voip] Video conf. equipment for IP/VC
>
> Well, I would like to have both VT Advantages, and h323 video conference
> units all in a multipoint conference. If I am getting a 3511, I assume I do
> not need a video camera with an MCU built-in, right? can some people post
> their setups, and what hardware they used, and a brief summary of their
> experiences? It seems most people on here have been using Tandbergs and
> Sonys?.any polycoms?
>
>
>
> Thanks,
>
> Avidan
>
>
>  ------------------------------
>
> *From:* Voll, Scott [mailto:Scott.Voll at wesd.org]
> *Sent:* Tuesday, December 20, 2005 12:16 PM
> *To:* IT; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> The answer is, it depends.  If the unit that has the built in MCU is
> always in the video conference you can probably get away with out a 3511 so
> long as all the units are h323. now if you are wanting to use CM VT
> advantage you will most likely still need a MCU (35xx)  I don't' know about
> the Tandberg as it does have a built in MCU and can register to CM.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* IT [mailto:it at cimgroup.com]
> *Sent:* Tuesday, December 20, 2005 12:16 PM
> *To:* Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> And if it has an MCU built-in, do I still need a IP/VC 3511?
>
>
>  ------------------------------
>
> *From:* Voll, Scott [mailto:Scott.Voll at wesd.org]
> *Sent:* Tuesday, December 20, 2005 9:14 AM
> *To:* IT; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> All we have done is demo'd the Tandberg so far, but it does have an MCU
> built in.  A lot of polycoms do also.  The nice thing about the Tandberg is
> that you can also interface with CM.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* IT [mailto:it at cimgroup.com]
> *Sent:* Monday, December 19, 2005 8:43 PM
> *To:* Voll, Scott; IT; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> This sounds like more work than I am interested in. Is there a better
> Video conference option to use with the 3511?
>
> Anyone have experience with the Tandberg?
>
> Thanks,
>
> Avidan
>
>
>  ------------------------------
>
> *From:* Voll, Scott [mailto:Scott.Voll at wesd.org]
> *Sent:* Monday, December 19, 2005 11:36 AM
> *To:* IT; Ortiz, Carlos; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
>
>
> We have the polycom sp and it does NOT have an interface with CM.  you
> will most likely have to use the MCU to connect to a CM video conference via
> a gatekeeper.
>
>
>
> Some MCU do have a built in Gatekeeper.  I suggest just getting your self
> an old 26xx series router and the IP Plus IOS and doing it with one of
> these.
>
>
>
> The MCU will just be a e.164 off the gatekeeper.  Each h323 video unit
> will just dial the e164 to get into the conference.
>
>
>
> Just as an FYI you will most likely have to exchange gatekeeper info with
> other company's if you are plan to have them dial into your conferences.
>
>
>
> Scott
>  ------------------------------
>
> *From:* IT [mailto:it at cimgroup.com]
> *Sent:* Monday, December 19, 2005 10:27 AM
> *To:* Ortiz, Carlos; Voll, Scott; IT; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> I intend on doing multi-point conferencing. Right now, I just have a
> Viewstation EX, so I only do point-to-point conferences with other
> companies. I am putting in an order for a 3511 and 2 more video conference
> cameras, so my 2 biggest branch offices. I am hoping to have a 3-way video
> conference happening by the end of January.  However, I am having a very
> hard time just connecting my Polycom Viewstation to CallManager. Is it even
> connectable? It seems the viewstation only has a "gatekeeper" IP setting for
> what I assume will be the IP of the 3511.  Where do I set the IP of
> CallManager on this polycom?
>
>
>
> Thanks,
>
> Avidan
>
>
>  ------------------------------
>
> *From:* Ortiz, Carlos [mailto:CORTIZ at broward.org]
> *Sent:* Monday, December 19, 2005 7:48 AM
> *To:* Voll, Scott; IT; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> Are you going to be doing Multipoint conferencing?  IHave you set your
> stuff up yet?
>
>
>
> Carlos
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Voll, Scott
> *Sent:* Monday, December 19, 2005 10:46 AM
> *To:* IT; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> We currently have a couple of Polycoms but will be getting a Tandberg
> shortly.  Both seem to be good.  It really comes down to features.  I'm
> hoping I can get rid of the old Vtel units we have soon.  Stay away from
> VTEL!!!!!!!!!!!!!
>
>
>
> Nice thing with the Tandberg we demo'd the other day is it can register
> directly with CM.
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *IT
> *Sent:* Sunday, December 18, 2005 6:35 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Video conf. equipment for IP/VC
>
>
>
> I am planning to setup a 3511 IP/VC to handle 4 branch offices and their
> video conference needs. I currently have a Polycom Viewstation EX, but with
> the upcoming IP/VC installation, I didnt know if buying 3 more of those was
> the right idea. What is everyone using out there for video conferencing with
> their IP/VC? Tandbergs? Polycoms?  Does getting MeetingPlace really help in
> small conferences?
>
> thanks for the advice.
> avidan
>
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
>
> _______________________________________________
> 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  Wed Dec 21 10:23:20 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Dec 21 10:23:24 2005
Subject: [cisco-voip] Busy out channels on an mgcp gateway
In-Reply-To: <008f01c6063e$a56c1220$30196883@cfs.uoguelph.ca>
References: <7E3064B7EFC16C41B185C41D141008B7C3CE83@foleyex97.foleylaw.com>
	<008f01c6063e$a56c1220$30196883@cfs.uoguelph.ca>
Message-ID: <625836AC-B4CB-4865-A18E-132DC617252C@cisco.com>

Those will work but require the gateway be reset to take affect.

-Ryan

On Dec 21, 2005, at 9:55 AM, Lelio Fulgenzi wrote:

I've heard people use the following service parameter to do what you  
are asking. There are five of them available I think.


Change B-Channel Maintenance Status 1 :This parameter allows Cisco  
CallManager to change individual B-Channel maintenance status for PRI  
interface in real time for troubleshooting. Input Format:
Device Name = B-channel Maintenance Status
Example:
S0/DS1-0 at SDA123456789ABC = 0000 1111 0000 0000 0000 0001(24 bits for  
T1. Channel number begins with 1 to 24 from left to right. The last  
one specifies the D-Channel, which does not get affected. In this  
example, 5th-8th B-Channels are out of service).
or,
S1/DS1-0 at DLS2-CM136-VG200-2.SELSIUSCMLAB.CISCO.COM = 0000 0000 0000  
0000 0000 0000 1111 1111(32 bits for E1. B-Channel number begins with  
1 to 32 from left to right. The 16th channel specifies D-Channel. The  
last bit does not affect any channel. In this example, 25th-31st  
channels are marked out of service).
Explanation:
The DEVICE NAME MUST exactly match the gateway name on the top of the  
gateway configuration web page. Copy and paste that field into this  
service parameter to avoid any manual inputting mistake. Ensure '='  
is mandatory and unique to distinguish the DEVICE NAME and BCHANNEL  
MAINTENANCE STATUS fields. BCHANNEL MAINTENANCE STATUS = xxxx xxxx  
xxxx xxxx xxxx xxxx xxxx xxxx
Where x can be:
0 -- In service
1 -- Graceful out of service (Change channel status until active call  
ends if an active call exists on that channel)
2 -- Forceful out of service (Tear down active call first; then,  
change channel status immediately if an active call exists on that  
channel).
The system treats any other value as invalid and the value does not  
take effect. Make sure that the total number of ?x? is either 24 for  
T1 or 32 for E1. Any other invalid length or mismatch (for example,  
24 for E1) gets treated as error, and no action gets taken for that  
device.
SPACE: Optional.
The spaces between device name and '=', between ?=? and BCHANNEL  
MAINTENANCE STATUS, and between x and x, are all optional. You cannot  
use any space inside the DEVICE NAME field.
To use this feature, the "Enable Status Poll" flag on the gateway  
configuration page must be checked and updated, then the gateway must  
be reset for the first time.After this, changing service parameter  
will take effect without resetting the gateway.
Maximum length: 255.
------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: Ruttman, Peter G.
To: cisco-voip at puck.nether.net
Sent: Wednesday, December 21, 2005 9:43 AM
Subject: [cisco-voip] Busy out channels on an mgcp gateway

I need to reset an mgcp gateway PRI that has inbound DIDs associated  
with it.  Is it possible to busy out the channels on it so that new  
calls do not get pinned up on it?  Then I can safely reset the  
gateway during work hours.

Pete


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



From lelio at uoguelph.ca  Wed Dec 21 10:35:00 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Dec 21 10:35:30 2005
Subject: [cisco-voip] Busy out channels on an mgcp gateway
References: <7E3064B7EFC16C41B185C41D141008B7C3CE83@foleyex97.foleylaw.com>
	<008f01c6063e$a56c1220$30196883@cfs.uoguelph.ca>
	<625836AC-B4CB-4865-A18E-132DC617252C@cisco.com>
Message-ID: <00ba01c60644$1b3b9430$30196883@cfs.uoguelph.ca>

so much for the graceful option, eh? ;)

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Ryan Ratliff 
  To: Lelio Fulgenzi 
  Cc: Ruttman, Peter G. ; cisco-voip at puck.nether.net 
  Sent: Wednesday, December 21, 2005 10:23 AM
  Subject: Re: [cisco-voip] Busy out channels on an mgcp gateway


  Those will work but require the gateway be reset to take affect.

  -Ryan

  On Dec 21, 2005, at 9:55 AM, Lelio Fulgenzi wrote:

  I've heard people use the following service parameter to do what you  
  are asking. There are five of them available I think.


  Change B-Channel Maintenance Status 1 :This parameter allows Cisco  
  CallManager to change individual B-Channel maintenance status for PRI  
  interface in real time for troubleshooting. Input Format:
  Device Name = B-channel Maintenance Status
  Example:
  S0/DS1-0 at SDA123456789ABC = 0000 1111 0000 0000 0000 0001(24 bits for  
  T1. Channel number begins with 1 to 24 from left to right. The last  
  one specifies the D-Channel, which does not get affected. In this  
  example, 5th-8th B-Channels are out of service).
  or,
  S1/DS1-0 at DLS2-CM136-VG200-2.SELSIUSCMLAB.CISCO.COM = 0000 0000 0000  
  0000 0000 0000 1111 1111(32 bits for E1. B-Channel number begins with  
  1 to 32 from left to right. The 16th channel specifies D-Channel. The  
  last bit does not affect any channel. In this example, 25th-31st  
  channels are marked out of service).
  Explanation:
  The DEVICE NAME MUST exactly match the gateway name on the top of the  
  gateway configuration web page. Copy and paste that field into this  
  service parameter to avoid any manual inputting mistake. Ensure '='  
  is mandatory and unique to distinguish the DEVICE NAME and BCHANNEL  
  MAINTENANCE STATUS fields. BCHANNEL MAINTENANCE STATUS = xxxx xxxx  
  xxxx xxxx xxxx xxxx xxxx xxxx
  Where x can be:
  0 -- In service
  1 -- Graceful out of service (Change channel status until active call  
  ends if an active call exists on that channel)
  2 -- Forceful out of service (Tear down active call first; then,  
  change channel status immediately if an active call exists on that  
  channel).
  The system treats any other value as invalid and the value does not  
  take effect. Make sure that the total number of ?x? is either 24 for  
  T1 or 32 for E1. Any other invalid length or mismatch (for example,  
  24 for E1) gets treated as error, and no action gets taken for that  
  device.
  SPACE: Optional.
  The spaces between device name and '=', between ?=? and BCHANNEL  
  MAINTENANCE STATUS, and between x and x, are all optional. You cannot  
  use any space inside the DEVICE NAME field.
  To use this feature, the "Enable Status Poll" flag on the gateway  
  configuration page must be checked and updated, then the gateway must  
  be reset for the first time.After this, changing service parameter  
  will take effect without resetting the gateway.
  Maximum length: 255.
  ------------------------------------------------------------------------ 
  --------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  "I had a coffee and Coke at lunch today...and now, I've got more  
  jitter than an
  IP phone on a long haul 10base2  
  connection"                                           LFJ
  ----- Original Message -----
  From: Ruttman, Peter G.
  To: cisco-voip at puck.nether.net
  Sent: Wednesday, December 21, 2005 9:43 AM
  Subject: [cisco-voip] Busy out channels on an mgcp gateway

  I need to reset an mgcp gateway PRI that has inbound DIDs associated  
  with it.  Is it possible to busy out the channels on it so that new  
  calls do not get pinned up on it?  Then I can safely reset the  
  gateway during work hours.

  Pete


  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
  _______________________________________________
  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  Wed Dec 21 10:47:28 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Dec 21 10:47:29 2005
Subject: [cisco-voip] Busy out channels on an mgcp gateway
In-Reply-To: <00ba01c60644$1b3b9430$30196883@cfs.uoguelph.ca>
References: <7E3064B7EFC16C41B185C41D141008B7C3CE83@foleyex97.foleylaw.com>
	<008f01c6063e$a56c1220$30196883@cfs.uoguelph.ca>
	<625836AC-B4CB-4865-A18E-132DC617252C@cisco.com>
	<00ba01c60644$1b3b9430$30196883@cfs.uoguelph.ca>
Message-ID: <B1A2F17C-6CD7-42F3-B79F-10927CE3067C@cisco.com>

I was debating internally whether the reset of the MGCP gateway would  
drop active calls or not.

I know for sure that when you reset a phone (ccmadmin reset) it will  
not actually reset until the active call is torn down.
I'd like to think an MGCP gateway would behave the same.  Even if it  
does so, would the change to inactive channels take immediately or  
would all of them go to graceful out of service only when all active  
calls have been torn down?

I just did a quick test with a CM 4.1 mgcp gateway with an FXS port.   
The reset did indeed knock the FXS end of the call off when the  
gateway unregistered and Temp Fail showed on the IP phone.
I don't have a gateway with a PRI handy to test with unfortunately.

-Ryan

On Dec 21, 2005, at 10:35 AM, Lelio Fulgenzi wrote:

so much for the graceful option, eh? ;)

------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: Ryan Ratliff
To: Lelio Fulgenzi
Cc: Ruttman, Peter G. ; cisco-voip at puck.nether.net
Sent: Wednesday, December 21, 2005 10:23 AM
Subject: Re: [cisco-voip] Busy out channels on an mgcp gateway

Those will work but require the gateway be reset to take affect.

-Ryan

On Dec 21, 2005, at 9:55 AM, Lelio Fulgenzi wrote:

I've heard people use the following service parameter to do what you
are asking. There are five of them available I think.


Change B-Channel Maintenance Status 1 :This parameter allows Cisco
CallManager to change individual B-Channel maintenance status for PRI
interface in real time for troubleshooting. Input Format:
Device Name = B-channel Maintenance Status
Example:
S0/DS1-0 at SDA123456789ABC = 0000 1111 0000 0000 0000 0001(24 bits for
T1. Channel number begins with 1 to 24 from left to right. The last
one specifies the D-Channel, which does not get affected. In this
example, 5th-8th B-Channels are out of service).
or,
S1/DS1-0 at DLS2-CM136-VG200-2.SELSIUSCMLAB.CISCO.COM = 0000 0000 0000
0000 0000 0000 1111 1111(32 bits for E1. B-Channel number begins with
1 to 32 from left to right. The 16th channel specifies D-Channel. The
last bit does not affect any channel. In this example, 25th-31st
channels are marked out of service).
Explanation:
The DEVICE NAME MUST exactly match the gateway name on the top of the
gateway configuration web page. Copy and paste that field into this
service parameter to avoid any manual inputting mistake. Ensure '='
is mandatory and unique to distinguish the DEVICE NAME and BCHANNEL
MAINTENANCE STATUS fields. BCHANNEL MAINTENANCE STATUS = xxxx xxxx
xxxx xxxx xxxx xxxx xxxx xxxx
Where x can be:
0 -- In service
1 -- Graceful out of service (Change channel status until active call
ends if an active call exists on that channel)
2 -- Forceful out of service (Tear down active call first; then,
change channel status immediately if an active call exists on that
channel).
The system treats any other value as invalid and the value does not
take effect. Make sure that the total number of ?x? is either 24 for
T1 or 32 for E1. Any other invalid length or mismatch (for example,
24 for E1) gets treated as error, and no action gets taken for that
device.
SPACE: Optional.
The spaces between device name and '=', between ?=? and BCHANNEL
MAINTENANCE STATUS, and between x and x, are all optional. You cannot
use any space inside the DEVICE NAME field.
To use this feature, the "Enable Status Poll" flag on the gateway
configuration page must be checked and updated, then the gateway must
be reset for the first time.After this, changing service parameter
will take effect without resetting the gateway.
Maximum length: 255.
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more
jitter than an
IP phone on a long haul 10base2
connection"                                           LFJ
----- Original Message -----
From: Ruttman, Peter G.
To: cisco-voip at puck.nether.net
Sent: Wednesday, December 21, 2005 9:43 AM
Subject: [cisco-voip] Busy out channels on an mgcp gateway

I need to reset an mgcp gateway PRI that has inbound DIDs associated
with it.  Is it possible to busy out the channels on it so that new
calls do not get pinned up on it?  Then I can safely reset the
gateway during work hours.

Pete


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





From lelio at uoguelph.ca  Wed Dec 21 10:48:56 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Dec 21 10:49:03 2005
Subject: [cisco-voip] Busy out channels on an mgcp gateway
References: <7E3064B7EFC16C41B185C41D141008B7C3CE83@foleyex97.foleylaw.com>
	<008f01c6063e$a56c1220$30196883@cfs.uoguelph.ca>
	<625836AC-B4CB-4865-A18E-132DC617252C@cisco.com>
	<00ba01c60644$1b3b9430$30196883@cfs.uoguelph.ca>
	<B1A2F17C-6CD7-42F3-B79F-10927CE3067C@cisco.com>
Message-ID: <00d401c60646$0d6e7d20$30196883@cfs.uoguelph.ca>

I'd like to say I could test it on one of our test PRI ports - but I've seen wierd things with our 6608 that ends up resetting all our PRI ports. :(

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Ryan Ratliff 
  To: Lelio Fulgenzi 
  Cc: Ruttman, Peter G. ; cisco-voip at puck.nether.net 
  Sent: Wednesday, December 21, 2005 10:47 AM
  Subject: Re: [cisco-voip] Busy out channels on an mgcp gateway


  I was debating internally whether the reset of the MGCP gateway would  
  drop active calls or not.

  I know for sure that when you reset a phone (ccmadmin reset) it will  
  not actually reset until the active call is torn down.
  I'd like to think an MGCP gateway would behave the same.  Even if it  
  does so, would the change to inactive channels take immediately or  
  would all of them go to graceful out of service only when all active  
  calls have been torn down?

  I just did a quick test with a CM 4.1 mgcp gateway with an FXS port.   
  The reset did indeed knock the FXS end of the call off when the  
  gateway unregistered and Temp Fail showed on the IP phone.
  I don't have a gateway with a PRI handy to test with unfortunately.

  -Ryan

  On Dec 21, 2005, at 10:35 AM, Lelio Fulgenzi wrote:

  so much for the graceful option, eh? ;)

  ------------------------------------------------------------------------ 
  --------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  "I had a coffee and Coke at lunch today...and now, I've got more  
  jitter than an
  IP phone on a long haul 10base2  
  connection"                                           LFJ
  ----- Original Message -----
  From: Ryan Ratliff
  To: Lelio Fulgenzi
  Cc: Ruttman, Peter G. ; cisco-voip at puck.nether.net
  Sent: Wednesday, December 21, 2005 10:23 AM
  Subject: Re: [cisco-voip] Busy out channels on an mgcp gateway

  Those will work but require the gateway be reset to take affect.

  -Ryan

  On Dec 21, 2005, at 9:55 AM, Lelio Fulgenzi wrote:

  I've heard people use the following service parameter to do what you
  are asking. There are five of them available I think.


  Change B-Channel Maintenance Status 1 :This parameter allows Cisco
  CallManager to change individual B-Channel maintenance status for PRI
  interface in real time for troubleshooting. Input Format:
  Device Name = B-channel Maintenance Status
  Example:
  S0/DS1-0 at SDA123456789ABC = 0000 1111 0000 0000 0000 0001(24 bits for
  T1. Channel number begins with 1 to 24 from left to right. The last
  one specifies the D-Channel, which does not get affected. In this
  example, 5th-8th B-Channels are out of service).
  or,
  S1/DS1-0 at DLS2-CM136-VG200-2.SELSIUSCMLAB.CISCO.COM = 0000 0000 0000
  0000 0000 0000 1111 1111(32 bits for E1. B-Channel number begins with
  1 to 32 from left to right. The 16th channel specifies D-Channel. The
  last bit does not affect any channel. In this example, 25th-31st
  channels are marked out of service).
  Explanation:
  The DEVICE NAME MUST exactly match the gateway name on the top of the
  gateway configuration web page. Copy and paste that field into this
  service parameter to avoid any manual inputting mistake. Ensure '='
  is mandatory and unique to distinguish the DEVICE NAME and BCHANNEL
  MAINTENANCE STATUS fields. BCHANNEL MAINTENANCE STATUS = xxxx xxxx
  xxxx xxxx xxxx xxxx xxxx xxxx
  Where x can be:
  0 -- In service
  1 -- Graceful out of service (Change channel status until active call
  ends if an active call exists on that channel)
  2 -- Forceful out of service (Tear down active call first; then,
  change channel status immediately if an active call exists on that
  channel).
  The system treats any other value as invalid and the value does not
  take effect. Make sure that the total number of ?x? is either 24 for
  T1 or 32 for E1. Any other invalid length or mismatch (for example,
  24 for E1) gets treated as error, and no action gets taken for that
  device.
  SPACE: Optional.
  The spaces between device name and '=', between ?=? and BCHANNEL
  MAINTENANCE STATUS, and between x and x, are all optional. You cannot
  use any space inside the DEVICE NAME field.
  To use this feature, the "Enable Status Poll" flag on the gateway
  configuration page must be checked and updated, then the gateway must
  be reset for the first time.After this, changing service parameter
  will take effect without resetting the gateway.
  Maximum length: 255.
  ------------------------------------------------------------------------
  --------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  "I had a coffee and Coke at lunch today...and now, I've got more
  jitter than an
  IP phone on a long haul 10base2
  connection"                                           LFJ
  ----- Original Message -----
  From: Ruttman, Peter G.
  To: cisco-voip at puck.nether.net
  Sent: Wednesday, December 21, 2005 9:43 AM
  Subject: [cisco-voip] Busy out channels on an mgcp gateway

  I need to reset an mgcp gateway PRI that has inbound DIDs associated
  with it.  Is it possible to busy out the channels on it so that new
  calls do not get pinned up on it?  Then I can safely reset the
  gateway during work hours.

  Pete


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



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From cenders at homesbyavi.com  Tue Dec 20 14:28:40 2005
From: cenders at homesbyavi.com (Corbett Enders)
Date: Wed Dec 21 11:06:53 2005
Subject: [cisco-voip] Directory Integration
Message-ID: <24DFD2417491E547AC80AE140706C4857F2F@EX001.homesbyavi.local>

Is there a general best practice around integration with Windows Active Directory?  I'm thinking it is a good idea, but I'm wondering if anyone has precautions or reasons NOT to use this model?

Regards, 
Corbett Enders. 

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From cenders at homesbyavi.com  Tue Dec 20 14:28:35 2005
From: cenders at homesbyavi.com (Corbett Enders)
Date: Wed Dec 21 11:06:54 2005
Subject: [cisco-voip] Call Recording...
Message-ID: <24DFD2417491E547AC80AE140706C4857F2E@EX001.homesbyavi.local>

I'm looking at implementing CallRex from TelRex. Looks like a decent solution. Does anyone have experience with it? 
I'm also looking at using the open source (free) product called Oreka. Does anyone have experience with this product as well? ( <http://oreka.sourceforge.net/> http://oreka.sourceforge.net/)

Regards,
Corbett Enders 

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From kthorngr at cisco.com  Wed Dec 21 11:10:44 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed Dec 21 11:10:52 2005
Subject: [cisco-voip] Directory Integration
In-Reply-To: <24DFD2417491E547AC80AE140706C4857F2F@EX001.homesbyavi.local>
References: <24DFD2417491E547AC80AE140706C4857F2F@EX001.homesbyavi.local>
Message-ID: <8757545a750ac8b47628c5d3d0f22014@cisco.com>

The CallManager design guide has some good information.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_implementation_design_guide_chapter09186a0080447505.html

Kevin
On Dec 20, 2005, at 2:28 PM, Corbett Enders wrote:

> Is there a general best practice around integration with Windows  
> Active Directory?? I'm thinking it is a good idea, but I'm wondering  
> if anyone has precautions or reasons NOT to use this model?
>
> Regards,
> Corbett Enders.
> _______________________________________________
> 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  Wed Dec 21 12:00:24 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Dec 21 12:00:36 2005
Subject: [cisco-voip] AC Pilot Point ringing busy
In-Reply-To: <8757545a750ac8b47628c5d3d0f22014@cisco.com>
Message-ID: <20051221170024.2060.qmail@web34311.mail.mud.yahoo.com>

Hi, Have CCM 4.1(3)sr1 in lab and trying to get a
attendant console pilot point going and have hunt
group with 2-3 numbers in it (call directory numbers)
and whenever I call pilot point # it is busy. It is
not in a partition, etc and there are no matching
numbers in dialplan. I reset the server many times,
restarted TCD, etc, and the thing is still busy.

DNA shows the number is blocked on it's analysis. In
CCM Detailed traces it shows theres NoPotentialMatches
when the number is called. 

Going through some TCD traces now to figure this out,
but thought I'd see if anyone had seen this before.
Not much to the setup and only a few phones on this
lab setup. The ac user exists and has right password
(12345) as I'm able to login with AC fine, etc. 

The problem I'm trying to work on is when calls come
into a AC pilot for a client, when they park the call
when it reverts off of park it rings back to the pilot
# and not the users phone who parked the call and
someone else in hunt group gets the call. I'm thinking
it may be one of the service parameter settings but
wanted to confirm it on my lab setup but having issues
with AC. 


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
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From erickbe at yahoo.com  Wed Dec 21 12:16:30 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Dec 21 12:16:38 2005
Subject: [cisco-voip] AC Pilot Point ringing busy
In-Reply-To: <20051221170024.2060.qmail@web34311.mail.mud.yahoo.com>
Message-ID: <20051221171630.32993.qmail@web34308.mail.mud.yahoo.com>

Nevermind.. got it working. Deleted the pilot point
and added it back. Ugh.

--- Erick Bergquist <erickbe at yahoo.com> wrote:

> Hi, Have CCM 4.1(3)sr1 in lab and trying to get a
> attendant console pilot point going and have hunt
> group with 2-3 numbers in it (call directory
> numbers)
> and whenever I call pilot point # it is busy. It is
> not in a partition, etc and there are no matching
> numbers in dialplan. I reset the server many times,
> restarted TCD, etc, and the thing is still busy.
> 
> DNA shows the number is blocked on it's analysis. In
> CCM Detailed traces it shows theres
> NoPotentialMatches
> when the number is called. 
> 
> Going through some TCD traces now to figure this
> out,
> but thought I'd see if anyone had seen this before.
> Not much to the setup and only a few phones on this
> lab setup. The ac user exists and has right password
> (12345) as I'm able to login with AC fine, etc. 
> 
> The problem I'm trying to work on is when calls come
> into a AC pilot for a client, when they park the
> call
> when it reverts off of park it rings back to the
> pilot
> # and not the users phone who parked the call and
> someone else in hunt group gets the call. I'm
> thinking
> it may be one of the service parameter settings but
> wanted to confirm it on my lab setup but having
> issues
> with AC. 
> 
> 
> __________________________________________________
> 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
> 


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http://mail.yahoo.com 

From erickbe at yahoo.com  Wed Dec 21 12:37:42 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Dec 21 12:37:49 2005
Subject: [cisco-voip] Hunt Pilot/List and Parked call reversion
In-Reply-To: <20051221170024.2060.qmail@web34311.mail.mud.yahoo.com>
Message-ID: <20051221173742.81581.qmail@web34313.mail.mud.yahoo.com>

Hi,

Does anyone know if calls to a hunt pilot, when a
member of hunt list/linegroup answers call and parks
call for the call to revert to the users phone who
parked the call? 

right now members of the line group who park call made
to the hunt pilot, when the parked call reverts it
rings another member of the line group.

I've dug in service parameters, etc but not finding
any knobs. Is this working as designed? 


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

From mike.c at altatechnologies.com  Wed Dec 21 13:29:19 2005
From: mike.c at altatechnologies.com (Mike Carter)
Date: Wed Dec 21 13:32:21 2005
Subject: [cisco-voip] HELP---PASSWORD IS LOCKED 
Message-ID: <BFEHKOIOENIIPHGJOCHDOEAPFDAA.mike.c@altatechnologies.com>

HELP!!!!!!!

not sure how but somehow one of our phones got locked----none of the default
passwords work

ideas????

Michael Carter
Account Manager
Alta Technologies Inc
763-475-5327
800-546-2582 (327)
AIM mikec at alta
yahoo IM cartematalta
mike.c at altatechnologies.com
Sun IBM HP Cisco Nortel Juniper Adtran
D-Link Emulex Qlogic Overland Storage
We buy and sell all of the above...please email your offers or requests.



From jgrace at digitelusa.net  Wed Dec 21 13:42:13 2005
From: jgrace at digitelusa.net (James)
Date: Wed Dec 21 13:42:21 2005
Subject: [cisco-voip] Unity sending message
Message-ID: <364521873-1135190538-cardhu_blackberry.rim.net-23637-@engine18-cell02>

I have a question.  In unity when you crreate an vm and it give u the options like 4 to send message. And it give you the dial by name or ##  for ext.   Is there a way to flip it around to ask for ext and then name
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From wael733 at hotmail.com  Wed Dec 21 14:03:41 2005
From: wael733 at hotmail.com (wael ahmed el mezain)
Date: Wed Dec 21 14:03:44 2005
Subject: [cisco-voip] - PGW220, SLT (2651) & AS5400
Message-ID: <BAY105-F244463F15E8BC7F498F361E7310@phx.gbl>

Hello Guys,

Is there anyone install this system before to give me a hand [PGW220, SLT 
(2651) & AS5400].

I want to configure it to make interface between PSTN & Cisco VoIP system 
(MGCP server, HSI server, BAMS server & 2651 as SLT and AS5400 as Access 
Media) and how to config. the CCM to go to PGW gateway through the 3745 
gatekeeper.

You can add me on your MSN to make a chat (wael733 at hotmail.com).

Thanks,
Wael Ahmed



From jkoniecki at comcast.net  Wed Dec 21 14:34:51 2005
From: jkoniecki at comcast.net (Jim Koniecki)
Date: Wed Dec 21 14:34:58 2005
Subject: [cisco-voip] HELP---PASSWORD IS LOCKED 
In-Reply-To: <BFEHKOIOENIIPHGJOCHDOEAPFDAA.mike.c@altatechnologies.com>
Message-ID: <000801c60665$9cfe1a80$c800a8c0@Jalbert1>

Mike, 

Please explain what you mean by locked? What are you trying to do or
perform?

What is the default password you are using ?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Carter
Sent: Wednesday, December 21, 2005 1:29 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] HELP---PASSWORD IS LOCKED 

HELP!!!!!!!

not sure how but somehow one of our phones got locked----none of the default
passwords work

ideas????

Michael Carter
Account Manager
Alta Technologies Inc
763-475-5327
800-546-2582 (327)
AIM mikec at alta
yahoo IM cartematalta
mike.c at altatechnologies.com
Sun IBM HP Cisco Nortel Juniper Adtran
D-Link Emulex Qlogic Overland Storage
We buy and sell all of the above...please email your offers or requests.


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



From jgrace at digitelusa.net  Wed Dec 21 15:50:20 2005
From: jgrace at digitelusa.net (James)
Date: Wed Dec 21 15:50:28 2005
Subject: [cisco-voip] New gw missing t1 controller
Message-ID: <1940869722-1135198224-cardhu_blackberry.rim.net-9259-@engine19-cell02>

We r installing a new gateway. And for somereason the serial show up in the config. But the t1 interface do not. I ran into this some time ago and I think we had to addACmd.  
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From it at cimgroup.com  Wed Dec 21 16:04:17 2005
From: it at cimgroup.com (IT)
Date: Wed Dec 21 16:00:16 2005
Subject: [cisco-voip] CDR Error
Message-ID: <B487D4C1FE6D04439061018C6EBC109B026B1C19@exchange.cimgroup.com>

It appears the problem is larger than that. I installed the ART on 11/8,
and the Max_date for any query has not moved from 11/8. I manually
changed the max_date, but all the reports are just filled with 0's. So I
changed the max_date back.
I have setup the Load Scheduler to run on default of midnight. When I
check the Tbl_Event_Log in SQL, the CDRLoad Jobs are taking only a
second or so (start->end time), and most of the utilization jobs are
returning a status of -2. The Error_log table is filled with
CtrPregenReports[generate] errors with Error_code 1020.
Where else can I look?

Thanks,
Avidan

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, December 20, 2005 9:23 PM
To: IT
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR Error

DirUser is the API ART uses for ldap access.

Are you even able to login to the ART web page?

Many times we see issues where the CDRs are being loaded just not  
associated the correct user.
Are your reports user or department based?

If so, try doing a CDR search on your extension to see if it finds  
calls.

There is a known issue with ART where when it goes to retrieve the  
user a device is associated to if the directory returns multiple  
users ART will take the first one.  This can cause issues with IPCC  
agents, AC users, etc where you have lots of devices associated with  
application users.  The "owner userid" field on the phone device page  
was created to workaround this.

Of course none of this may be relevant to your issue but it's a  
common problem.

-Ryan

On Dec 21, 2005, at 12:10 AM, IT wrote:

It appears that none of my reports contain any data when I use CAR,  
even though I enabled the CDR loading.

Here is the error I have in ARTSchError.log

DirUser.OperationError

             at DirUser.LoadComClass.retAupairClass(LoadComClass.java)

             at DirUser.DirUser.<init>(DirUser.java)

             at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java: 
137)

             at com.cisco.art.controller.CtrLoader.run(CtrLoader.java: 
1356)

             at com.cisco.art.scheduler.SchThread.run(SchThread.java: 
550)

DirUser.OperationError

             at DirUser.LoadComClass.retAupairClass(LoadComClass.java)

             at DirUser.DirUser.<init>(DirUser.java)

             at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java: 
137)

             at com.cisco.art.controller.CtrLoader.run(CtrLoader.java: 
1356)

             at com.cisco.art.scheduler.SchThread.run(SchThread.java: 
550)



I have Active Directory integration installed with CM4.0. Has anyone  
seen this issue before?



Thanks,

Avidan

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From SPackett at fenwick.com  Wed Dec 21 16:35:29 2005
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Dec 21 16:35:37 2005
Subject: [cisco-voip] Unity Permissions Wizard - Not Taking Effect
Message-ID: <965C0255FD5828438E34F9A68ECA76E740A686@svcexch02.firm.fenwick.llp>

I brought up a new Exchange server and a few Exchange stores on existing
servers, but after running the Unity Permissions Wizard, the permissions
are not taking effect.  When I looked at the Permissions Wizard logs, it
shows that UnityMsgSvc (Unity Message Store Service account) as
'Succeeded'.  I've waited over 15 minutes for the AD replication to
occur and I've restarted Unity.  I even restarted the Exchange
Information Store service on the new Exchange server, but I still cannot
get the permissions to carry over.  Because the permissions are not
taking effect on Exchange, any user I enable on the new Exchange server
fails.

Unity 4.1.(1) ES11 on Windows 2003
Exchange 2003 SP1

Thanks in advance.
--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
--------------------------------------------------------

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The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
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From nikos at oteglobe.net  Wed Dec 21 16:52:21 2005
From: nikos at oteglobe.net (Nikos Leontsinis)
Date: Wed Dec 21 16:52:58 2005
Subject: [cisco-voip] New gw missing t1 controller
In-Reply-To: <1940869722-1135198224-cardhu_blackberry.rim.net-9259-@engine19-cell02>
Message-ID: <006301c60678$d2fd84f0$031010ac@nikoslaptop>

Did you add the below commands?

service-module T1 clock source line
service-module T1 framing esf
service-module T1 linecode b8zs
service-module T1 timeslots ? speed ?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James
Sent: Wednesday, December 21, 2005 10:50 PM
To: CiscosupportUpuck
Subject: [cisco-voip] New gw missing t1 controller

We r installing a new gateway. And for somereason the serial show up in the
config. But the t1 interface do not. I ran into this some time ago and I
think we had to addACmd.  
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-- 
This message has been scanned for viruses and
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From SPackett at fenwick.com  Wed Dec 21 16:53:57 2005
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Dec 21 16:54:06 2005
Subject: [cisco-voip] New gw missing t1 controller
Message-ID: <965C0255FD5828438E34F9A68ECA76E740A688@svcexch02.firm.fenwick.llp>

I think I ran into this last week when I brought up two new 2811s.  Try
the 'card type t1' command then the 'controller t1 ?' command. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James
Sent: Wednesday, December 21, 2005 12:50 PM
To: CiscosupportUpuck
Subject: [cisco-voip] New gw missing t1 controller

We r installing a new gateway. And for somereason the serial show up in
the config. But the t1 interface do not. I ran into this some time ago
and I think we had to addACmd.  
James D. Grace
System Engineer
CCNP CCNA MCSE MCDBA
Digitel Corp.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
--------------------------------------------------------

IRS Circular 230  Disclosure:  To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice in this communication (including attachments) is not intended or written by Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) avoiding penalties under the Internal Revenue Code or (ii) promoting, marketing, or recommending to another party any transaction or matter addressed herein.
--------------------------------------------------------

ATTENTION:
The information contained in this message may be legally privileged and confidential.  It is intended to be read only by the individual or entity to whom it is addressed or by their designee. If the reader of this message is not the intended recipient, you are on notice that any distribution of this message, in any form, is strictly prohibited.
 
If you have received this message in error, please immediately notify the sender and/or Fenwick & West LLP by telephone at (650) 988-8500 and delete or destroy any copy of this message.


From mfa at crec.ifas.ufl.edu  Wed Dec 21 17:03:39 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Wed Dec 21 17:03:43 2005
Subject: [cisco-voip] Saving IP phone directories
References: <200512212135.jBLLZcpd037476@puck.nether.net>
Message-ID: <015c01c6067a$66233d00$400cfb0a@crec.ifas.ufl.edu>

When replacing an IP phone, we usually just assign the old IP to the new 
phone and make the substitution.  However, all the stored directories 
(Missed Calls, etc.) are lost across a power reset.  Is there any way to 
save those and reload them into the new phone, probably via some XML magic?

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

----- Original Message ----- 
From: <cisco-voip-request at puck.nether.net>
To: <cisco-voip at puck.nether.net>
Sent: Wednesday, December 21, 2005 4:35 PM
Subject: cisco-voip Digest, Vol 34, Issue 117


> 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. AC Pilot Point ringing busy (Erick Bergquist)
>   2. Re: AC Pilot Point ringing busy (Erick Bergquist)
>   3. Hunt Pilot/List and Parked call reversion (Erick Bergquist)
>   4. HELP---PASSWORD IS LOCKED  (Mike Carter)
>   5. Unity sending message (James)
>   6. - PGW220, SLT (2651) & AS5400 (wael ahmed el mezain)
>   7. RE: HELP---PASSWORD IS LOCKED  (Jim Koniecki)
>   8. New gw missing t1 controller (James)
>   9. RE: CDR Error (IT)
>  10. Unity Permissions Wizard - Not Taking Effect (Stu Packett)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 21 Dec 2005 09:00:24 -0800 (PST)
> From: Erick Bergquist <erickbe at yahoo.com>
> Subject: [cisco-voip] AC Pilot Point ringing busy
> To: cisco-voip at puck.nether.net
> Message-ID: <20051221170024.2060.qmail at web34311.mail.mud.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi, Have CCM 4.1(3)sr1 in lab and trying to get a
> attendant console pilot point going and have hunt
> group with 2-3 numbers in it (call directory numbers)
> and whenever I call pilot point # it is busy. It is
> not in a partition, etc and there are no matching
> numbers in dialplan. I reset the server many times,
> restarted TCD, etc, and the thing is still busy.
>
> DNA shows the number is blocked on it's analysis. In
> CCM Detailed traces it shows theres NoPotentialMatches
> when the number is called.
>
> Going through some TCD traces now to figure this out,
> but thought I'd see if anyone had seen this before.
> Not much to the setup and only a few phones on this
> lab setup. The ac user exists and has right password
> (12345) as I'm able to login with AC fine, etc.
>
> The problem I'm trying to work on is when calls come
> into a AC pilot for a client, when they park the call
> when it reverts off of park it rings back to the pilot
> # and not the users phone who parked the call and
> someone else in hunt group gets the call. I'm thinking
> it may be one of the service parameter settings but
> wanted to confirm it on my lab setup but having issues
> with AC.
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
>
> ------------------------------
>
> Message: 2
> Date: Wed, 21 Dec 2005 09:16:30 -0800 (PST)
> From: Erick Bergquist <erickbe at yahoo.com>
> Subject: Re: [cisco-voip] AC Pilot Point ringing busy
> To: erickbe at yahoo.com, cisco-voip at puck.nether.net
> Message-ID: <20051221171630.32993.qmail at web34308.mail.mud.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Nevermind.. got it working. Deleted the pilot point
> and added it back. Ugh.
>
> --- Erick Bergquist <erickbe at yahoo.com> wrote:
>
>> Hi, Have CCM 4.1(3)sr1 in lab and trying to get a
>> attendant console pilot point going and have hunt
>> group with 2-3 numbers in it (call directory
>> numbers)
>> and whenever I call pilot point # it is busy. It is
>> not in a partition, etc and there are no matching
>> numbers in dialplan. I reset the server many times,
>> restarted TCD, etc, and the thing is still busy.
>>
>> DNA shows the number is blocked on it's analysis. In
>> CCM Detailed traces it shows theres
>> NoPotentialMatches
>> when the number is called.
>>
>> Going through some TCD traces now to figure this
>> out,
>> but thought I'd see if anyone had seen this before.
>> Not much to the setup and only a few phones on this
>> lab setup. The ac user exists and has right password
>> (12345) as I'm able to login with AC fine, etc.
>>
>> The problem I'm trying to work on is when calls come
>> into a AC pilot for a client, when they park the
>> call
>> when it reverts off of park it rings back to the
>> pilot
>> # and not the users phone who parked the call and
>> someone else in hunt group gets the call. I'm
>> thinking
>> it may be one of the service parameter settings but
>> wanted to confirm it on my lab setup but having
>> issues
>> with AC.
>>
>>
>> __________________________________________________
>> 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
>>
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 21 Dec 2005 09:37:42 -0800 (PST)
> From: Erick Bergquist <erickbe at yahoo.com>
> Subject: [cisco-voip] Hunt Pilot/List and Parked call reversion
> To: cisco-voip at puck.nether.net
> Message-ID: <20051221173742.81581.qmail at web34313.mail.mud.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi,
>
> Does anyone know if calls to a hunt pilot, when a
> member of hunt list/linegroup answers call and parks
> call for the call to revert to the users phone who
> parked the call?
>
> right now members of the line group who park call made
> to the hunt pilot, when the parked call reverts it
> rings another member of the line group.
>
> I've dug in service parameters, etc but not finding
> any knobs. Is this working as designed?
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
>
> ------------------------------
>
> Message: 4
> Date: Wed, 21 Dec 2005 12:29:19 -0600
> From: "Mike Carter" <mike.c at altatechnologies.com>
> Subject: [cisco-voip] HELP---PASSWORD IS LOCKED
> To: <cisco-voip at puck.nether.net>
> Message-ID: <BFEHKOIOENIIPHGJOCHDOEAPFDAA.mike.c at altatechnologies.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> HELP!!!!!!!
>
> not sure how but somehow one of our phones got locked----none of the 
> default
> passwords work
>
> ideas????
>
> Michael Carter
> Account Manager
> Alta Technologies Inc
> 763-475-5327
> 800-546-2582 (327)
> AIM mikec at alta
> yahoo IM cartematalta
> mike.c at altatechnologies.com
> Sun IBM HP Cisco Nortel Juniper Adtran
> D-Link Emulex Qlogic Overland Storage
> We buy and sell all of the above...please email your offers or requests.
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Wed, 21 Dec 2005 18:42:13 +0000 GMT
> From: "James" <jgrace at digitelusa.net>
> Subject: [cisco-voip] Unity sending message
> To: "CiscosupportUpuck" <cisco-voip at puck.nether.net>
> Message-ID:
> <364521873-1135190538-cardhu_blackberry.rim.net-23637- at engine18-cell02>
>
> Content-Type: text/plain
>
> I have a question.  In unity when you crreate an vm and it give u the 
> options like 4 to send message. And it give you the dial by name or ## 
> for ext.   Is there a way to flip it around to ask for ext and then name
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 21 Dec 2005 19:03:41 +0000
> From: "wael ahmed el mezain" <wael733 at hotmail.com>
> Subject: [cisco-voip] - PGW220, SLT (2651) & AS5400
> To: cisco-voip at puck.nether.net
> Message-ID: <BAY105-F244463F15E8BC7F498F361E7310 at phx.gbl>
> Content-Type: text/plain; format=flowed
>
> Hello Guys,
>
> Is there anyone install this system before to give me a hand [PGW220, SLT
> (2651) & AS5400].
>
> I want to configure it to make interface between PSTN & Cisco VoIP system
> (MGCP server, HSI server, BAMS server & 2651 as SLT and AS5400 as Access
> Media) and how to config. the CCM to go to PGW gateway through the 3745
> gatekeeper.
>
> You can add me on your MSN to make a chat (wael733 at hotmail.com).
>
> Thanks,
> Wael Ahmed
>
>
>
>
> ------------------------------
>
> Message: 7
> Date: Wed, 21 Dec 2005 14:34:51 -0500
> From: "Jim Koniecki" <jkoniecki at comcast.net>
> Subject: RE: [cisco-voip] HELP---PASSWORD IS LOCKED
> To: "'Mike Carter'" <mike.c at altatechnologies.com>,
> <cisco-voip at puck.nether.net>
> Message-ID: <000801c60665$9cfe1a80$c800a8c0 at Jalbert1>
> Content-Type: text/plain; charset="us-ascii"
>
> Mike,
>
> Please explain what you mean by locked? What are you trying to do or
> perform?
>
> What is the default password you are using ?
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Carter
> Sent: Wednesday, December 21, 2005 1:29 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] HELP---PASSWORD IS LOCKED
>
> HELP!!!!!!!
>
> not sure how but somehow one of our phones got locked----none of the 
> default
> passwords work
>
> ideas????
>
> Michael Carter
> Account Manager
> Alta Technologies Inc
> 763-475-5327
> 800-546-2582 (327)
> AIM mikec at alta
> yahoo IM cartematalta
> mike.c at altatechnologies.com
> Sun IBM HP Cisco Nortel Juniper Adtran
> D-Link Emulex Qlogic Overland Storage
> We buy and sell all of the above...please email your offers or requests.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> ------------------------------
>
> Message: 8
> Date: Wed, 21 Dec 2005 20:50:20 +0000 GMT
> From: "James" <jgrace at digitelusa.net>
> Subject: [cisco-voip] New gw missing t1 controller
> To: "CiscosupportUpuck" <cisco-voip at puck.nether.net>
> Message-ID:
> <1940869722-1135198224-cardhu_blackberry.rim.net-9259- at engine19-cell02>
>
> Content-Type: text/plain
>
> We r installing a new gateway. And for somereason the serial show up in 
> the config. But the t1 interface do not. I ran into this some time ago and 
> I think we had to addACmd.
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
>
>
> ------------------------------
>
> Message: 9
> Date: Wed, 21 Dec 2005 13:04:17 -0800
> From: "IT" <it at cimgroup.com>
> Subject: RE: [cisco-voip] CDR Error
> To: "Ryan Ratliff" <rratliff at cisco.com>, "IT" <it at cimgroup.com>
> Cc: cisco-voip at puck.nether.net
> Message-ID:
> <B487D4C1FE6D04439061018C6EBC109B026B1C19 at exchange.cimgroup.com>
> Content-Type: text/plain; charset="us-ascii"
>
> It appears the problem is larger than that. I installed the ART on 11/8,
> and the Max_date for any query has not moved from 11/8. I manually
> changed the max_date, but all the reports are just filled with 0's. So I
> changed the max_date back.
> I have setup the Load Scheduler to run on default of midnight. When I
> check the Tbl_Event_Log in SQL, the CDRLoad Jobs are taking only a
> second or so (start->end time), and most of the utilization jobs are
> returning a status of -2. The Error_log table is filled with
> CtrPregenReports[generate] errors with Error_code 1020.
> Where else can I look?
>
> Thanks,
> Avidan
>
> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> Sent: Tuesday, December 20, 2005 9:23 PM
> To: IT
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CDR Error
>
> DirUser is the API ART uses for ldap access.
>
> Are you even able to login to the ART web page?
>
> Many times we see issues where the CDRs are being loaded just not
> associated the correct user.
> Are your reports user or department based?
>
> If so, try doing a CDR search on your extension to see if it finds
> calls.
>
> There is a known issue with ART where when it goes to retrieve the
> user a device is associated to if the directory returns multiple
> users ART will take the first one.  This can cause issues with IPCC
> agents, AC users, etc where you have lots of devices associated with
> application users.  The "owner userid" field on the phone device page
> was created to workaround this.
>
> Of course none of this may be relevant to your issue but it's a
> common problem.
>
> -Ryan
>
> On Dec 21, 2005, at 12:10 AM, IT wrote:
>
> It appears that none of my reports contain any data when I use CAR,
> even though I enabled the CDR loading.
>
> Here is the error I have in ARTSchError.log
>
> DirUser.OperationError
>
>             at DirUser.LoadComClass.retAupairClass(LoadComClass.java)
>
>             at DirUser.DirUser.<init>(DirUser.java)
>
>             at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java:
> 137)
>
>             at com.cisco.art.controller.CtrLoader.run(CtrLoader.java:
> 1356)
>
>             at com.cisco.art.scheduler.SchThread.run(SchThread.java:
> 550)
>
> DirUser.OperationError
>
>             at DirUser.LoadComClass.retAupairClass(LoadComClass.java)
>
>             at DirUser.DirUser.<init>(DirUser.java)
>
>             at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java:
> 137)
>
>             at com.cisco.art.controller.CtrLoader.run(CtrLoader.java:
> 1356)
>
>             at com.cisco.art.scheduler.SchThread.run(SchThread.java:
> 550)
>
>
>
> I have Active Directory integration installed with CM4.0. Has anyone
> seen this issue before?
>
>
>
> Thanks,
>
> Avidan
>
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> ------------------------------
>
> Message: 10
> Date: Wed, 21 Dec 2005 13:35:29 -0800
> From: "Stu Packett" <SPackett at fenwick.com>
> Subject: [cisco-voip] Unity Permissions Wizard - Not Taking Effect
> To: <cisco-voip at puck.nether.net>
> Message-ID:
> <965C0255FD5828438E34F9A68ECA76E740A686 at svcexch02.firm.fenwick.llp>
> Content-Type: text/plain; charset="us-ascii"
>
> I brought up a new Exchange server and a few Exchange stores on existing
> servers, but after running the Unity Permissions Wizard, the permissions
> are not taking effect.  When I looked at the Permissions Wizard logs, it
> shows that UnityMsgSvc (Unity Message Store Service account) as
> 'Succeeded'.  I've waited over 15 minutes for the AD replication to
> occur and I've restarted Unity.  I even restarted the Exchange
> Information Store service on the new Exchange server, but I still cannot
> get the permissions to carry over.  Because the permissions are not
> taking effect on Exchange, any user I enable on the new Exchange server
> fails.
>
> Unity 4.1.(1) ES11 on Windows 2003
> Exchange 2003 SP1
>
> Thanks in advance.
> --------------------------------------------------------
>
> IRS Circular 230  Disclosure:  To ensure compliance with requirements 
> imposed by the IRS, we inform you that any U.S. federal tax advice in this 
> communication (including attachments) is not intended or written by 
> Fenwick & West LLP to be used, and cannot be used, for the purpose of (i) 
> avoiding penalties under the Internal Revenue Code or (ii) promoting, 
> marketing, or recommending to another party any transaction or matter 
> addressed herein.
> --------------------------------------------------------
>
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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 34, Issue 117
> ******************************************* 


From Scott.Voll at wesd.org  Wed Dec 21 18:11:47 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 21 18:11:57 2005
Subject: [cisco-voip] Directory Integration
Message-ID: <D713462ED535184D830F6F6301753E7E01A1213F@VISHNU.wesd.org>

Pros with AD integration.  One username and password.

 

Con's: if your AD goes down, no one can log in.

 

You're very reliant on a stable AD.  If you have lots of AD instability,
don't integrate.

 

Works fine for us.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Corbett Enders
Sent: Tuesday, December 20, 2005 11:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Directory Integration

 

Is there a general best practice around integration with Windows Active
Directory?  I'm thinking it is a good idea, but I'm wondering if anyone
has precautions or reasons NOT to use this model?

Regards, 
Corbett Enders. 

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From lelio at uoguelph.ca  Wed Dec 21 18:15:09 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed Dec 21 18:15:16 2005
Subject: [cisco-voip] Directory Integration
References: <D713462ED535184D830F6F6301753E7E01A1213F@VISHNU.wesd.org>
Message-ID: <009401c60684$62db3260$30196883@cfs.uoguelph.ca>

Directory IntegrationI really wish that there was more to offer with AD integration than just that. perhaps display name for a DN?

I also think AD integration gives you centralized PIN.

Chris Adams is on the list - Chris, can you remind me of that one problem you had where you had to populate a different field for something to work b/c the integration broke it?
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Voll, Scott 
  To: Corbett Enders ; cisco-voip at puck.nether.net 
  Sent: Wednesday, December 21, 2005 6:11 PM
  Subject: RE: [cisco-voip] Directory Integration


  Pros with AD integration.  One username and password.

   

  Con's: if your AD goes down, no one can log in.

   

  You're very reliant on a stable AD.  If you have lots of AD instability, don't integrate.

   

  Works fine for us.

   

  Scott

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Corbett Enders
  Sent: Tuesday, December 20, 2005 11:29 AM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Directory Integration

   

  Is there a general best practice around integration with Windows Active Directory?  I'm thinking it is a good idea, but I'm wondering if anyone has precautions or reasons NOT to use this model?

  Regards, 
  Corbett Enders. 



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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From cenders at homesbyavi.com  Wed Dec 21 18:31:45 2005
From: cenders at homesbyavi.com (Corbett Enders)
Date: Wed Dec 21 18:31:54 2005
Subject: [cisco-voip] Directory Integration
Message-ID: <24DFD2417491E547AC80AE140706C4857F6D@EX001.homesbyavi.local>

I read something to that effect regarding hiding accounts from the Call
Manager, by setting a custom field entry.
 
I have also noted an issue with regards to Exchange 2000, not sure yet
if it applies to Exchange 2003 (iNetOrgPerson class).
 
Slight twist on my original question:  Have you performed the AD
Integration and regretted it?  Did it break something you were trying to
do elsewhere?  And on the flip side, did anyone not perform the AD
Integration and now wish that you had?
 
Pros:  Single directory (the model everyone pitches, yet not often
used), Simplified Management (AD Users & Computers?)
Cons: Increased dependency on AD domain controllers, possible latency
for some of the call flow (lookups).
 
 
________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Wednesday, December 21, 2005 4:15 PM
To: Voll, Scott; Corbett Enders; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Directory Integration


I really wish that there was more to offer with AD integration than just
that. perhaps display name for a DN?
 
I also think AD integration gives you centralized PIN.
 
Chris Adams is on the list - Chris, can you remind me of that one
problem you had where you had to populate a different field for
something to work b/c the integration broke it?
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an
IP phone on a long haul 10base2 connection"
LFJ

	----- Original Message ----- 
	From: Voll, Scott <mailto:Scott.Voll at wesd.org>  
	To: Corbett Enders <mailto:cenders at homesbyavi.com>  ;
cisco-voip at puck.nether.net 
	Sent: Wednesday, December 21, 2005 6:11 PM
	Subject: RE: [cisco-voip] Directory Integration


	Pros with AD integration.  One username and password.

	 

	Con's: if your AD goes down, no one can log in.

	 

	You're very reliant on a stable AD.  If you have lots of AD
instability, don't integrate.

	 

	Works fine for us.

	 

	Scott

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Corbett Enders
	Sent: Tuesday, December 20, 2005 11:29 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Directory Integration

	 

	Is there a general best practice around integration with Windows
Active Directory?  I'm thinking it is a good idea, but I'm wondering if
anyone has precautions or reasons NOT to use this model?

	Regards, 
	Corbett Enders. 

	
________________________________


	

	_______________________________________________
	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  Wed Dec 21 22:35:01 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Dec 21 22:35:03 2005
Subject: [cisco-voip] Directory Integration
In-Reply-To: <24DFD2417491E547AC80AE140706C4857F6D@EX001.homesbyavi.local>
References: <24DFD2417491E547AC80AE140706C4857F6D@EX001.homesbyavi.local>
Message-ID: <C6B96068-EFB7-4672-8ECE-C66D6FA30C85@cisco.com>

Biggest issues I see with AD integrations are referrals and DNS  
issues, and the two go hand-in-hand a lot of the time.  The scripts  
we use for CCMAdmin and various web pages are sensitive to delays and  
referral hop count.

Lots of people have to point CM to the root of the domain and with AD  
2003 this leaves you to the whim of DNS for forestdnszones and  
domaindnszones lookups.  Inevitably it seems if one DC in your forest  
is down that is the one that CM will pick out the ones returned by  
DNS to contact for your referral, and next thing you know it takes 5  
minutes to search your global directory and EM just plain doesn't work.

Regarding the centralized PIN if you mean between CM and Unity this  
isn't true.   You do get the advantage of synched CCMUser and AD  
passwords for users.

-Ryan

On Dec 21, 2005, at 6:31 PM, Corbett Enders wrote:

I read something to that effect regarding hiding accounts from the  
Call Manager, by setting a custom field entry.

I have also noted an issue with regards to Exchange 2000, not sure  
yet if it applies to Exchange 2003 (iNetOrgPerson class).

Slight twist on my original question:  Have you performed the AD  
Integration and regretted it?  Did it break something you were trying  
to do elsewhere?  And on the flip side, did anyone not perform the AD  
Integration and now wish that you had?

Pros:  Single directory (the model everyone pitches, yet not often  
used), Simplified Management (AD Users & Computers?)
Cons: Increased dependency on AD domain controllers, possible latency  
for some of the call flow (lookups).


From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Wednesday, December 21, 2005 4:15 PM
To: Voll, Scott; Corbett Enders; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Directory Integration

I really wish that there was more to offer with AD integration than  
just that. perhaps display name for a DN?

I also think AD integration gives you centralized PIN.

Chris Adams is on the list - Chris, can you remind me of that one  
problem you had where you had to populate a different field for  
something to work b/c the integration broke it?
------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: Voll, Scott
To: Corbett Enders ; cisco-voip at puck.nether.net
Sent: Wednesday, December 21, 2005 6:11 PM
Subject: RE: [cisco-voip] Directory Integration

Pros with AD integration.  One username and password.



Con?s: if your AD goes down, no one can log in.



You?re very reliant on a stable AD.  If you have lots of AD  
instability, don?t integrate.



Works fine for us.



Scott



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Corbett Enders
Sent: Tuesday, December 20, 2005 11:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Directory Integration



Is there a general best practice around integration with Windows  
Active Directory?  I'm thinking it is a good idea, but I'm wondering  
if anyone has precautions or reasons NOT to use this model?

Regards,
Corbett Enders.



_______________________________________________
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 cenders at homesbyavi.com  Wed Dec 21 22:49:18 2005
From: cenders at homesbyavi.com (Corbett Enders)
Date: Wed Dec 21 22:49:28 2005
Subject: [cisco-voip] Directory Integration
Message-ID: <24DFD2417491E547AC80AE140706C4857F73@EX001.homesbyavi.local>

In the case of adding users or removing users, I assume the AD
Integration offers some benefit there as well?  Does the assignment of
user extensions, etc, all occur within AD Users and Computers or do we
still have to resort to the web interface (I think the guide spoke
somewhat to this).

We're a simple AD domain, and a single OU for our "normal" users
(non-admin type of accounts) so we'd probably point it at that.

But it sounds like I'm hearing that there aren't enough advantages
(reasons) to go with an Integrated design? 

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Wednesday, December 21, 2005 8:35 PM
To: Corbett Enders
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Directory Integration

Biggest issues I see with AD integrations are referrals and DNS issues,
and the two go hand-in-hand a lot of the time.  The scripts we use for
CCMAdmin and various web pages are sensitive to delays and referral hop
count.

Lots of people have to point CM to the root of the domain and with AD
2003 this leaves you to the whim of DNS for forestdnszones and
domaindnszones lookups.  Inevitably it seems if one DC in your forest is
down that is the one that CM will pick out the ones returned by DNS to
contact for your referral, and next thing you know it takes 5 minutes to
search your global directory and EM just plain doesn't work.

Regarding the centralized PIN if you mean between CM and Unity this  
isn't true.   You do get the advantage of synched CCMUser and AD  
passwords for users.

-Ryan

On Dec 21, 2005, at 6:31 PM, Corbett Enders wrote:

I read something to that effect regarding hiding accounts from the Call
Manager, by setting a custom field entry.

I have also noted an issue with regards to Exchange 2000, not sure yet
if it applies to Exchange 2003 (iNetOrgPerson class).

Slight twist on my original question:  Have you performed the AD
Integration and regretted it?  Did it break something you were trying to
do elsewhere?  And on the flip side, did anyone not perform the AD
Integration and now wish that you had?

Pros:  Single directory (the model everyone pitches, yet not often
used), Simplified Management (AD Users & Computers?)
Cons: Increased dependency on AD domain controllers, possible latency
for some of the call flow (lookups).


From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Wednesday, December 21, 2005 4:15 PM
To: Voll, Scott; Corbett Enders; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Directory Integration

I really wish that there was more to offer with AD integration than just
that. perhaps display name for a DN?

I also think AD integration gives you centralized PIN.

Chris Adams is on the list - Chris, can you remind me of that one
problem you had where you had to populate a different field for
something to work b/c the integration broke it?
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: Voll, Scott
To: Corbett Enders ; cisco-voip at puck.nether.net
Sent: Wednesday, December 21, 2005 6:11 PM
Subject: RE: [cisco-voip] Directory Integration

Pros with AD integration.  One username and password.



Con's: if your AD goes down, no one can log in.



You're very reliant on a stable AD.  If you have lots of AD instability,
don't integrate.



Works fine for us.



Scott



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Corbett Enders
Sent: Tuesday, December 20, 2005 11:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Directory Integration



Is there a general best practice around integration with Windows  
Active Directory?  I'm thinking it is a good idea, but I'm wondering  
if anyone has precautions or reasons NOT to use this model?

Regards,
Corbett Enders.



_______________________________________________
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  Wed Dec 21 23:51:29 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Dec 21 23:51:40 2005
Subject: [cisco-voip] Directory Integration
In-Reply-To: <24DFD2417491E547AC80AE140706C4857F73@EX001.homesbyavi.local>
References: <24DFD2417491E547AC80AE140706C4857F73@EX001.homesbyavi.local>
Message-ID: <EDBDEF22-8894-4552-96C7-EFA901DB3982@cisco.com>

By default you aren't allowed to add or delete users via CCMAdmin web  
page with an AD integration.

If you look at a global directory page for a user everything above  
user locale (except for pin) is pulled from AD.  The first time you  
modify a user via CCMAdmin it will create a profile and CCN-profile  
in a sub-container of the Cisco ou you create in AD.  These profiles  
are where all of the Cisco specific information is stored.

-Ryan

On Dec 21, 2005, at 10:49 PM, Corbett Enders wrote:

In the case of adding users or removing users, I assume the AD
Integration offers some benefit there as well?  Does the assignment of
user extensions, etc, all occur within AD Users and Computers or do we
still have to resort to the web interface (I think the guide spoke
somewhat to this).

We're a simple AD domain, and a single OU for our "normal" users
(non-admin type of accounts) so we'd probably point it at that.

But it sounds like I'm hearing that there aren't enough advantages
(reasons) to go with an Integrated design?

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Wednesday, December 21, 2005 8:35 PM
To: Corbett Enders
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Directory Integration

Biggest issues I see with AD integrations are referrals and DNS issues,
and the two go hand-in-hand a lot of the time.  The scripts we use for
CCMAdmin and various web pages are sensitive to delays and referral hop
count.

Lots of people have to point CM to the root of the domain and with AD
2003 this leaves you to the whim of DNS for forestdnszones and
domaindnszones lookups.  Inevitably it seems if one DC in your forest is
down that is the one that CM will pick out the ones returned by DNS to
contact for your referral, and next thing you know it takes 5 minutes to
search your global directory and EM just plain doesn't work.

Regarding the centralized PIN if you mean between CM and Unity this
isn't true.   You do get the advantage of synched CCMUser and AD
passwords for users.

-Ryan

On Dec 21, 2005, at 6:31 PM, Corbett Enders wrote:

I read something to that effect regarding hiding accounts from the Call
Manager, by setting a custom field entry.

I have also noted an issue with regards to Exchange 2000, not sure yet
if it applies to Exchange 2003 (iNetOrgPerson class).

Slight twist on my original question:  Have you performed the AD
Integration and regretted it?  Did it break something you were trying to
do elsewhere?  And on the flip side, did anyone not perform the AD
Integration and now wish that you had?

Pros:  Single directory (the model everyone pitches, yet not often
used), Simplified Management (AD Users & Computers?)
Cons: Increased dependency on AD domain controllers, possible latency
for some of the call flow (lookups).


From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Wednesday, December 21, 2005 4:15 PM
To: Voll, Scott; Corbett Enders; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Directory Integration

I really wish that there was more to offer with AD integration than just
that. perhaps display name for a DN?

I also think AD integration gives you centralized PIN.

Chris Adams is on the list - Chris, can you remind me of that one
problem you had where you had to populate a different field for
something to work b/c the integration broke it?
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more jitter
than an IP phone on a long haul 10base2
connection"                                           LFJ
----- Original Message -----
From: Voll, Scott
To: Corbett Enders ; cisco-voip at puck.nether.net
Sent: Wednesday, December 21, 2005 6:11 PM
Subject: RE: [cisco-voip] Directory Integration

Pros with AD integration.  One username and password.



Con's: if your AD goes down, no one can log in.



You're very reliant on a stable AD.  If you have lots of AD instability,
don't integrate.



Works fine for us.



Scott



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
bounces at puck.nether.net] On Behalf Of Corbett Enders
Sent: Tuesday, December 20, 2005 11:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Directory Integration



Is there a general best practice around integration with Windows
Active Directory?  I'm thinking it is a good idea, but I'm wondering
if anyone has precautions or reasons NOT to use this model?

Regards,
Corbett Enders.



_______________________________________________
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


From robin.inderberg at candidator.se  Thu Dec 22 02:18:34 2005
From: robin.inderberg at candidator.se (Robin Inderberg)
Date: Thu Dec 22 02:13:10 2005
Subject: SV: VB: [cisco-voip] Changes not being activated
Message-ID: <39EB1FE59758D9448A5D0B348857C7273E57C2@cannt04.candidator.se>

Goodmorning,

I am running CM 4.1(3), just have updated it and added a publisher.
I think it must be after these changes everything started to make some problems. I restarted CM01 and CM02 yesterday, and after that my changes took effect. Tried to restart the DBL Monitor yesterday too, and I could not control the phones. Could not make a restart and no updates were taking power.

Thank you very much for your time, Rasmin and Ryan!


Best regards


/Robin Inderberg

-----Ursprungligt meddelande-----
Fr?n: Ryan Ratliff [mailto:rratliff at cisco.com] 
Skickat: den 21 december 2005 15:19
Till: Robin Inderberg
Kopia: cisco-voip at puck.nether.net
?mne: Re: VB: [cisco-voip] Changes not being activated

What version of CM are you running?

There are a few ways change notification can break in CM 4.x.  The  
first thing to try would be to restart the Cisco Database Layer  
Monitor service on the publisher and then try to update a phone via  
the CCMAdmin.  If you can even reset a phone via CCMAdmin then change  
notification is working.

-Ryan

On Dec 21, 2005, at 5:32 AM, Robin Inderberg wrote:

Hi once again,



Just a little update on the case, no changes is getting understood by  
the CCM. The changes is visible on the ccmadmin, and I can add  
phones, change things on DN, but the changes does not get active.



Regards once again





/Robin

Fr?n: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] F?r Robin Inderberg
Skickat: den 21 december 2005 11:10
Till: cisco-voip at puck.nether.net
?mne: [cisco-voip] Changes not being activated



Hi,



I cant manage this one by my self, don?t realy know what could be  
wrong. And I think that one of you has the answer (as always) J.



I have a DN that a customer often uses CFA on, he often sets CFA to  
his cellphone, everything worked well, he could add a CFA number and  
disable it. But now the change does not take any effect, he can  
disable it, everything looks fine in the phone. But the CFA doesn't  
get disabled. Now I have checked the Enterprise Manager and looked  
for a CFA on his DN, but could not find anything. I tested to remove  
his DN, and just see if the CFA or the change did get power, but no.  
The DN is there, and forwards to his cellphone.





I have tried to restart the DBL monitor, but it didn't help.



Anyone who has an idea of what could cause this one?





Best Regards









Robin Inderberg

robin.inderberg at candidator.se



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




From acm at netuse.de  Thu Dec 22 04:49:23 2005
From: acm at netuse.de (Ahmad Cheikh Moussa)
Date: Thu Dec 22 04:49:34 2005
Subject: [cisco-voip] callforwardall on 4.1(3)
Message-ID: <43AA76A3.5030406@netuse.de>

Hi!

I have a question to the callforwardall feature.
Is there a  way to bypass the callforwardall, if the
call is from an internal phone ? I mean without using a second line.
Something like the callforward on no answer. There you can define,
how the callmanager reacts, if the incoming call is an internal
or an external call.

Best regards,
  Ahmad

-- 
Ahmad Cheikh-Moussa
NetUSE AG
Dr.-Hell-Stra?e, 24107 Kiel, Germany
Telefon: +49 431 2390 400 --  Telefax: +49 431 2390 499
Service: Service at NetUSE.DE --  http://NetUSE.DE/



From kthorngr at cisco.com  Thu Dec 22 06:50:09 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Dec 22 06:50:21 2005
Subject: [cisco-voip] callforwardall on 4.1(3)
In-Reply-To: <43AA76A3.5030406@netuse.de>
References: <43AA76A3.5030406@netuse.de>
Message-ID: <c45c9ff7402c57de4913cebb8de31d85@cisco.com>

Hi Ahmad,

Within CallManager there is no option to change the behavior of Call  
Forward All.  You can use Personal Assistant to allow the users to  
setup routing rules that will allow them to have internal calls ring  
their phone and external calls to ring one or more destinations.

http://www.cisco.com/en/US/products/sw/voicesw/ps2026/ 
products_data_sheet0900aecd801c39dd.html

Kevin
On Dec 22, 2005, at 4:49 AM, Ahmad Cheikh Moussa wrote:

> Hi!
>
> I have a question to the callforwardall feature.
> Is there a  way to bypass the callforwardall, if the
> call is from an internal phone ? I mean without using a second line.
> Something like the callforward on no answer. There you can define,
> how the callmanager reacts, if the incoming call is an internal
> or an external call.
>
> Best regards,
>   Ahmad
>
> -- 
> Ahmad Cheikh-Moussa
> NetUSE AG
> Dr.-Hell-Stra?e, 24107 Kiel, Germany
> Telefon: +49 431 2390 400 --  Telefax: +49 431 2390 499
> Service: Service at NetUSE.DE --  http://NetUSE.DE/
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



From ealeatherman at gmail.com  Thu Dec 22 11:58:38 2005
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu Dec 22 11:58:41 2005
Subject: [cisco-voip] IOS MGCP question
Message-ID: <94a1afde0512220858w3b0a1925x1375ef9afd305925@mail.gmail.com>

Hi folks,

I'm pretty new to setting up gateways with callmanager, up until now
all i've had the chance to mess with has been 6608 cards.. have a
question about some older hardware if anyone has any ideas:

I have 2 clusters configured in the lab, one is "WVU", the other is
"Potomac State". WVU has a 2611XM router with a NM-HDV/1MFT-T1 VWIC,
configured as a MGCP gateway and setup for PRI. everything looks good
on it from what I can tell. register on CCM, PRI Backhaul connection
is up, it sees the mgcp enpoints, etc etc.

"Potomac State" has a VG200 with a NM-HDV/1MFT-T1 VWIC also. i'm
running IOS 12.2(16a) on it right now. have the gateway T1 interfaces
connected with a T1 crossover cable, have a green CD light and the sho
isdn command indicates that the d-channels are up.

"Potomac State" CCM shows the VG200 as registered, and sho ccm-manager
confirms that. but, IOS will not take the command "isdn bind-l3
ccm-manager" on the serial interface.. which is what the documentation
tells me to do to get the PRI backhaul to callmanager working. also I
see errors in trace files saying something about tcp connection not
open, which is making me think the PRI backhaul isnt working. I see
minimal messages when i turn on debug mgcp packets, one or two intial
messages and then the vgw just keeps updating CCM with "nothing is
happening" messages, but nothing else, no messages from CCM back to
the gateway other than "200 OK" type messages. Trying to call out from
Potomac State to WVU using a route pattern pointed at the gateway just
gets me reorder, nothing even seems to show up on the VG200 debug.

Any ideas? neither "isdn bind-l3 ccm-manager" nor "ccm-manager config"
seem to work on this VG200 no matter which IOS i put on it, I also
tried the latest IOS but I couldnt even get the mgcp endpoints to come
up with that version. probably something very obvious that im missing
here.

Happy Holidays,

Ed Leatherman
WVU Telecommunications


From jgrace at digitelusa.net  Thu Dec 22 12:01:08 2005
From: jgrace at digitelusa.net (James)
Date: Thu Dec 22 12:01:15 2005
Subject: [cisco-voip] When I'm configuring the p2p t1
Message-ID: <1829684144-1135270872-cardhu_blackberry.rim.net-16331-@engine12-cell02>

On confuring my p2p t1 for the remote srst site, I have my serial int but no controller int. Isint this how it suppose to be 
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From Scott.Voll at wesd.org  Thu Dec 22 12:04:09 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 22 12:04:17 2005
Subject: [cisco-voip] When I'm configuring the p2p t1
Message-ID: <D713462ED535184D830F6F6301753E7E01A1214F@VISHNU.wesd.org>

Controller would only be on a PSTN PRI / CAS.  Not a point to point wan
link.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James
Sent: Thursday, December 22, 2005 9:01 AM
To: CiscosupportUpuck
Subject: [cisco-voip] When I'm configuring the p2p t1

On confuring my p2p t1 for the remote srst site, I have my serial int
but no controller int. Isint this how it suppose to be 
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From kthorngr at cisco.com  Thu Dec 22 12:18:21 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Dec 22 12:18:46 2005
Subject: [cisco-voip] IOS MGCP question
In-Reply-To: <94a1afde0512220858w3b0a1925x1375ef9afd305925@mail.gmail.com>
References: <94a1afde0512220858w3b0a1925x1375ef9afd305925@mail.gmail.com>
Message-ID: <f85c6ac8e36009d6b85a94a1688bdc81@cisco.com>

Hi Ed,

According to the Feature Navigator 12.2(16a) on the VG200 does not 
support MGCP PRI backhaul to CCM.  You will need to use a 12.2T version 
or a 12.3 version on the VG200.

You can search by feature from here:
http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp

Look for the feature "MGCP PRI backhaul and T1-CAS support for Call 
Manager (IP-PBX)".  Then select the VG200.  This will give you all your 
options and you can determine which IOS will best fit your needs.

Also for a back to back config don't forget to but the "isdn 
protocol-emulate network" command on the serial interface of one of the 
routers to bring up layer 2.

HTH,
Kevin

On Dec 22, 2005, at 11:58 AM, Ed Leatherman wrote:

> Hi folks,
>
> I'm pretty new to setting up gateways with callmanager, up until now
> all i've had the chance to mess with has been 6608 cards.. have a
> question about some older hardware if anyone has any ideas:
>
> I have 2 clusters configured in the lab, one is "WVU", the other is
> "Potomac State". WVU has a 2611XM router with a NM-HDV/1MFT-T1 VWIC,
> configured as a MGCP gateway and setup for PRI. everything looks good
> on it from what I can tell. register on CCM, PRI Backhaul connection
> is up, it sees the mgcp enpoints, etc etc.
>
> "Potomac State" has a VG200 with a NM-HDV/1MFT-T1 VWIC also. i'm
> running IOS 12.2(16a) on it right now. have the gateway T1 interfaces
> connected with a T1 crossover cable, have a green CD light and the sho
> isdn command indicates that the d-channels are up.
>
> "Potomac State" CCM shows the VG200 as registered, and sho ccm-manager
> confirms that. but, IOS will not take the command "isdn bind-l3
> ccm-manager" on the serial interface.. which is what the documentation
> tells me to do to get the PRI backhaul to callmanager working. also I
> see errors in trace files saying something about tcp connection not
> open, which is making me think the PRI backhaul isnt working. I see
> minimal messages when i turn on debug mgcp packets, one or two intial
> messages and then the vgw just keeps updating CCM with "nothing is
> happening" messages, but nothing else, no messages from CCM back to
> the gateway other than "200 OK" type messages. Trying to call out from
> Potomac State to WVU using a route pattern pointed at the gateway just
> gets me reorder, nothing even seems to show up on the VG200 debug.
>
> Any ideas? neither "isdn bind-l3 ccm-manager" nor "ccm-manager config"
> seem to work on this VG200 no matter which IOS i put on it, I also
> tried the latest IOS but I couldnt even get the mgcp endpoints to come
> up with that version. probably something very obvious that im missing
> here.
>
> Happy Holidays,
>
> Ed Leatherman
> WVU Telecommunications
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From voipforum at apptis.com  Thu Dec 22 14:45:48 2005
From: voipforum at apptis.com (VoIP Forum)
Date: Thu Dec 22 14:45:55 2005
Subject: [cisco-voip] VM Pilot or FwdNoAns Issue
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BCF05F44@CH-EXCH-02.intelltech.apptis.inc>

Has anyone recently upgraded OS from the 2000.2.7 train to 2000.4.2?

 

After an upgrade from 2000.2.7sr8 to 2000.4.2sr2 we have been getting
dispersed reports that calls are not rolling to VM after the NoAns
timeout. Does this sound familiar to anyone?

Do we need an ES to resolve this?

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From highspeedsnow at gmail.com  Thu Dec 22 14:56:34 2005
From: highspeedsnow at gmail.com (Mark Snow)
Date: Thu Dec 22 14:57:05 2005
Subject: [cisco-voip] When I'm configuring the p2p t1
In-Reply-To: <1829684144-1135270872-cardhu_blackberry.rim.net-16331-@engine12-cell02>
Message-ID: <43ab050c.1e5494a0.5d88.40d6@mx.gmail.com>

It all depends on the type of WIC/VWIC card that you are using ....
What part number are you using?

If it is something like a WIC-2T then no, you will not (though you can still
do a 'sh controllers s0/0/0') - if it is something like a VWIC-1T1 then yes,
you will have a controller interface in which you make the T1 anything you
want (CAS, PRI, Channel-Group).

-Mark Snow
CCIE Voice #14073

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of James
> Sent: Thursday, December 22, 2005 12:01 PM
> To: CiscosupportUpuck
> Subject: [cisco-voip] When I'm configuring the p2p t1
> 
> On confuring my p2p t1 for the remote srst site, I have my serial int but
no
> controller int. Isint this how it suppose to be
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From CORTIZ at broward.org  Thu Dec 22 15:00:16 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Thu Dec 22 15:00:24 2005
Subject: [cisco-voip] VM Pilot or FwdNoAns Issue
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2BB9@MREXVS01.bc.broward.cty>

That's not good news.  Is it mandatory to move to this train?

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of VoIP Forum
Sent: Thursday, December 22, 2005 2:46 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VM Pilot or FwdNoAns Issue

 

Has anyone recently upgraded OS from the 2000.2.7 train to 2000.4.2?

 

After an upgrade from 2000.2.7sr8 to 2000.4.2sr2 we have been getting
dispersed reports that calls are not rolling to VM after the NoAns
timeout. Does this sound familiar to anyone?

Do we need an ES to resolve this?

-------------- next part --------------
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From ealeatherman at gmail.com  Thu Dec 22 15:41:30 2005
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu Dec 22 15:41:33 2005
Subject: [cisco-voip] IOS MGCP question
In-Reply-To: <f85c6ac8e36009d6b85a94a1688bdc81@cisco.com>
References: <94a1afde0512220858w3b0a1925x1375ef9afd305925@mail.gmail.com>
	<f85c6ac8e36009d6b85a94a1688bdc81@cisco.com>
Message-ID: <94a1afde0512221241r264a1c0dw7d02354833ae71e9@mail.gmail.com>

Kevin,

Turns out none of the VG200's I have came with enough flash memory to
upgrade to any versions supporting MGCP PRI backhaul.. they are all
8MB flash and the lowest version of 12.2T that hasnt been removed from
CCO (that I could find) is just a bit bigger than 8MB. All the 8MB
versions have been removed, too old I guess. I'll try to get H323
working for this test instead.

I hadn't used the feature navigator before, thanks for showing me.

Ed

On 12/22/05, Kevin Thorngren <kthorngr at cisco.com> wrote:
> Hi Ed,
>
> According to the Feature Navigator 12.2(16a) on the VG200 does not
> support MGCP PRI backhaul to CCM.  You will need to use a 12.2T version
> or a 12.3 version on the VG200.
>
> You can search by feature from here:
> http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
>
> Look for the feature "MGCP PRI backhaul and T1-CAS support for Call
> Manager (IP-PBX)".  Then select the VG200.  This will give you all your
> options and you can determine which IOS will best fit your needs.
>
> Also for a back to back config don't forget to but the "isdn
> protocol-emulate network" command on the serial interface of one of the
> routers to bring up layer 2.
>
> HTH,
> Kevin
>


From CORTIZ at broward.org  Thu Dec 22 15:46:52 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Thu Dec 22 15:47:00 2005
Subject: [cisco-voip] IOS MGCP question
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2BBC@MREXVS01.bc.broward.cty>

Hey Ed, I remember you from CIPTUG! The newer VG224's work really well
with MGCP.  The only issue I currently have is that my IOS version has a
AAA bug in it.  I need to upgrade my memory to fix this.  Really hasn't
impacted me though because the boxes run and I never touch them....

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Thursday, December 22, 2005 3:42 PM
To: Kevin Thorngren
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IOS MGCP question

Kevin,

Turns out none of the VG200's I have came with enough flash memory to
upgrade to any versions supporting MGCP PRI backhaul.. they are all
8MB flash and the lowest version of 12.2T that hasnt been removed from
CCO (that I could find) is just a bit bigger than 8MB. All the 8MB
versions have been removed, too old I guess. I'll try to get H323
working for this test instead.

I hadn't used the feature navigator before, thanks for showing me.

Ed

On 12/22/05, Kevin Thorngren <kthorngr at cisco.com> wrote:
> Hi Ed,
>
> According to the Feature Navigator 12.2(16a) on the VG200 does not
> support MGCP PRI backhaul to CCM.  You will need to use a 12.2T
version
> or a 12.3 version on the VG200.
>
> You can search by feature from here:
> http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
>
> Look for the feature "MGCP PRI backhaul and T1-CAS support for Call
> Manager (IP-PBX)".  Then select the VG200.  This will give you all
your
> options and you can determine which IOS will best fit your needs.
>
> Also for a back to back config don't forget to but the "isdn
> protocol-emulate network" command on the serial interface of one of
the
> routers to bring up layer 2.
>
> HTH,
> Kevin
>

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


From lelio at uoguelph.ca  Thu Dec 22 15:51:06 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Dec 22 15:51:14 2005
Subject: [cisco-voip] IOS MGCP question
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2BBC@MREXVS01.bc.broward.cty>
Message-ID: <008d01c60739$6db72fd0$30196883@cfs.uoguelph.ca>

I'm going to miss the vg248 *sniff*

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Ortiz, Carlos 
  To: Ed Leatherman ; Kevin Thorngren 
  Cc: cisco-voip at puck.nether.net 
  Sent: Thursday, December 22, 2005 3:46 PM
  Subject: RE: [cisco-voip] IOS MGCP question


  Hey Ed, I remember you from CIPTUG! The newer VG224's work really well
  with MGCP.  The only issue I currently have is that my IOS version has a
  AAA bug in it.  I need to upgrade my memory to fix this.  Really hasn't
  impacted me though because the boxes run and I never touch them....

  Carlos

  -----Original Message-----
  From: cisco-voip-bounces at puck.nether.net
  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
  Sent: Thursday, December 22, 2005 3:42 PM
  To: Kevin Thorngren
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] IOS MGCP question

  Kevin,

  Turns out none of the VG200's I have came with enough flash memory to
  upgrade to any versions supporting MGCP PRI backhaul.. they are all
  8MB flash and the lowest version of 12.2T that hasnt been removed from
  CCO (that I could find) is just a bit bigger than 8MB. All the 8MB
  versions have been removed, too old I guess. I'll try to get H323
  working for this test instead.

  I hadn't used the feature navigator before, thanks for showing me.

  Ed

  On 12/22/05, Kevin Thorngren <kthorngr at cisco.com> wrote:
  > Hi Ed,
  >
  > According to the Feature Navigator 12.2(16a) on the VG200 does not
  > support MGCP PRI backhaul to CCM.  You will need to use a 12.2T
  version
  > or a 12.3 version on the VG200.
  >
  > You can search by feature from here:
  > http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
  >
  > Look for the feature "MGCP PRI backhaul and T1-CAS support for Call
  > Manager (IP-PBX)".  Then select the VG200.  This will give you all
  your
  > options and you can determine which IOS will best fit your needs.
  >
  > Also for a back to back config don't forget to but the "isdn
  > protocol-emulate network" command on the serial interface of one of
  the
  > routers to bring up layer 2.
  >
  > HTH,
  > Kevin
  >

  _______________________________________________
  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 rratliff at cisco.com  Thu Dec 22 16:05:54 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu Dec 22 16:05:54 2005
Subject: [cisco-voip] VM Pilot or FwdNoAns Issue
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BCF05F44@CH-EXCH-02.intelltech.apptis.inc>
References: <2F09B2162CCDEA4EA05C089B24FBC2BCF05F44@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <08E9FF90-CC66-4F41-9ECF-1CDD2519EDCD@cisco.com>

Have you gotten CCM and SDL traces of these calls?   An OS update  
*should* have no impact on how CM call processing works.

If it's getting widespread and you have data for it please open a TAC  
case.

-Ryan

On Dec 22, 2005, at 2:45 PM, VoIP Forum wrote:

Has anyone recently upgraded OS from the 2000.2.7 train to 2000.4.2?



After an upgrade from 2000.2.7sr8 to 2000.4.2sr2 we have been getting  
dispersed reports that calls are not rolling to VM after the NoAns  
timeout. Does this sound familiar to anyone?

Do we need an ES to resolve this?

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


From jonvoip at gmail.com  Thu Dec 22 18:18:45 2005
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu Dec 22 18:18:48 2005
Subject: [cisco-voip] Difference between National, CallManager,
	ISDN for Caller ID?
Message-ID: <5d093f9a0512221518t71192c26pb4aad2979661744f@mail.gmail.com>

What is it?

Is this telco dependent, or do are they specific to customer needs?

Also, what is the difference of manipulations between:

 Outbound Calls Calling Line ID Presentation* --- Not Selected ---Default
AllowedRestricted Calling Party Selection* --- Not Selected ---OriginatorFirst
Redirect NumberLast Redirect NumberFirst Redirect Number (External)Last
Redirect Number (External) Called party IE number type unknown* --- Not
Selected ---Cisco CallManagerUnknownNationalInternationalSubscriber Calling
party IE number type unknown* --- Not Selected ---Cisco CallManagerUnknown
NationalInternationalSubscriber Called Numbering Plan* --- Not
Selected ---Cisco
CallManagerISDNNational StandardPrivateUnknown Calling Numbering Plan* ---
Not Selected ---Cisco CallManagerISDNNational StandardPrivateUnknown Number
of digits to strip* --- Not Selected ---012345678910111213141516171819202122
23242526272829303132 Caller ID DN  SMDI Base Port*
In other words, how can you manipulate what is and is not sent down a PRI,
and what overrides and combinations work?
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From Scott.Voll at wesd.org  Thu Dec 22 19:19:03 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 22 19:19:19 2005
Subject: [cisco-voip] Difference between National, CallManager,
	ISDN for Caller ID?
Message-ID: <D713462ED535184D830F6F6301753E7E01A12169@VISHNU.wesd.org>

If it's set to CM it will try to do it.  If you set it to national isdn
it sends that way.  Most times CM and figure it out, but some telcos
want it one way or the other and then you end up changing to what they
want.  

 

One of my PRIs is set to CM and the other is set to national isdn.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Thursday, December 22, 2005 3:19 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Difference between National, CallManager,ISDN for
Caller ID?

 

What is it?

Is this telco dependent, or do are they specific to customer needs?

Also, what is the difference of manipulations between:

Outbound Calls

Calling Line ID Presentation*

--- Not Selected --- Default Allowed Restricted 

 

Calling Party Selection*

--- Not Selected --- Originator First Redirect Number Last Redirect
Number First Redirect Number (External) Last Redirect Number (External) 

 

Called party IE number type unknown*

--- Not Selected --- Cisco CallManager Unknown National International
Subscriber 

 

Calling party IE number type unknown*

--- Not Selected --- Cisco CallManager Unknown National International
Subscriber 

 

Called Numbering Plan*

--- Not Selected --- Cisco CallManager ISDN National Standard Private
Unknown   

 

Calling Numbering Plan*

--- Not Selected --- Cisco CallManager ISDN National Standard Private
Unknown   

 

Number of digits to strip*

--- Not Selected --- 0 1 2 3 4 5 6   7 8 9 10 11 12 13  14 15 16 17 18
19 20 21   22 23 24 25 26 27 28  29 30 31 32 

 

Caller ID DN

 

 

 

SMDI Base Port*

 

 

 


In other words, how can you manipulate what is and is not sent down a
PRI, and what overrides and combinations work?

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From kthorngr at cisco.com  Thu Dec 22 19:42:03 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Dec 22 19:42:09 2005
Subject: [cisco-voip] IOS MGCP question
In-Reply-To: <94a1afde0512221241r264a1c0dw7d02354833ae71e9@mail.gmail.com>
References: <94a1afde0512220858w3b0a1925x1375ef9afd305925@mail.gmail.com>
	<f85c6ac8e36009d6b85a94a1688bdc81@cisco.com>
	<94a1afde0512221241r264a1c0dw7d02354833ae71e9@mail.gmail.com>
Message-ID: <0526c2d404e89100a8b28ec199e43e6e@cisco.com>

Hi Ed,

As long as you have enough main memory you can use the boot system 
command and boot the image from a TFTP server.  This would be fine for 
a lab environment but might not be ideal for production.  If a flash 
upgrade for the VG200 is still available you might be able to get away 
with the TFTP workaround for a short period.

Kevin
On Dec 22, 2005, at 3:41 PM, Ed Leatherman wrote:

> Kevin,
>
> Turns out none of the VG200's I have came with enough flash memory to
> upgrade to any versions supporting MGCP PRI backhaul.. they are all
> 8MB flash and the lowest version of 12.2T that hasnt been removed from
> CCO (that I could find) is just a bit bigger than 8MB. All the 8MB
> versions have been removed, too old I guess. I'll try to get H323
> working for this test instead.
>
> I hadn't used the feature navigator before, thanks for showing me.
>
> Ed
>
> On 12/22/05, Kevin Thorngren <kthorngr at cisco.com> wrote:
>> Hi Ed,
>>
>> According to the Feature Navigator 12.2(16a) on the VG200 does not
>> support MGCP PRI backhaul to CCM.  You will need to use a 12.2T 
>> version
>> or a 12.3 version on the VG200.
>>
>> You can search by feature from here:
>> http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
>>
>> Look for the feature "MGCP PRI backhaul and T1-CAS support for Call
>> Manager (IP-PBX)".  Then select the VG200.  This will give you all 
>> your
>> options and you can determine which IOS will best fit your needs.
>>
>> Also for a back to back config don't forget to but the "isdn
>> protocol-emulate network" command on the serial interface of one of 
>> the
>> routers to bring up layer 2.
>>
>> HTH,
>> Kevin
>>
>


From erickbe at yahoo.com  Thu Dec 22 23:54:04 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Dec 22 23:54:12 2005
Subject: [cisco-voip] Re: *answer* Hunt Pilot/List and Parked call reversion
In-Reply-To: <20051221173742.81581.qmail@web34313.mail.mud.yahoo.com>
Message-ID: <20051223045404.76819.qmail@web34309.mail.mud.yahoo.com>

Answer to this for others (hiding in a release notes
doc)... working as designed.

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/4_0/rel_note/401cmrn.htm#wp1717365

Configuration Tips

The following caveats exist for line groups, route
groups, route/hunt lists, and route patterns/hunt
pilots:


???When a parked call reverts to the original caller,
the reversion goes back to the hunt pilot and
therefore may not go back to the phone that parked the
call.

--- Erick Bergquist <erickbe at yahoo.com> wrote:

> Hi,
> 
> Does anyone know if calls to a hunt pilot, when a
> member of hunt list/linegroup answers call and parks
> call for the call to revert to the users phone who
> parked the call? 
> 
> right now members of the line group who park call
> made
> to the hunt pilot, when the parked call reverts it
> rings another member of the line group.
> 
> I've dug in service parameters, etc but not finding
> any knobs. Is this working as designed? 
> 
> 
> __________________________________________________
> 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
> 



	
		
__________________________________ 
Yahoo! for Good - Make a difference this year. 
http://brand.yahoo.com/cybergivingweek2005/

From CORTIZ at broward.org  Fri Dec 23 10:23:03 2005
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri Dec 23 10:23:11 2005
Subject: [cisco-voip] IOS MGCP question
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC2BC1@MREXVS01.bc.broward.cty>

I think you are better off with the VG224's.  They are IOS based and are
basically like configuring a router.  Unless the 248's have changed (I
haven't looked) they are not IOS based and I remember they were a little
more troublesome.  Maybe someone else out there can confirm this and you
will feel better about going this route.

Carlos

-----Original Message-----
From: Ed Leatherman [mailto:ealeatherman at gmail.com] 
Sent: Thursday, December 22, 2005 8:00 PM
To: Ortiz, Carlos
Subject: Re: [cisco-voip] IOS MGCP question

Hey Carlos good to hear from you. We're getting some VG224's in to use
for a few dormitories at a regional campus, hopefully they will work
out OK. I would have rather gone with some 248's and had fewer boxes
to manage, but the 'customer' wants dual connections to all boxes
where possible.

On 12/22/05, Ortiz, Carlos <CORTIZ at broward.org> wrote:
> Hey Ed, I remember you from CIPTUG! The newer VG224's work really well
> with MGCP.  The only issue I currently have is that my IOS version has
a
> AAA bug in it.  I need to upgrade my memory to fix this.  Really
hasn't
> impacted me though because the boxes run and I never touch them....
>
> Carlos
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
> Sent: Thursday, December 22, 2005 3:42 PM
> To: Kevin Thorngren
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] IOS MGCP question
>
> Kevin,
>
> Turns out none of the VG200's I have came with enough flash memory to
> upgrade to any versions supporting MGCP PRI backhaul.. they are all
> 8MB flash and the lowest version of 12.2T that hasnt been removed from
> CCO (that I could find) is just a bit bigger than 8MB. All the 8MB
> versions have been removed, too old I guess. I'll try to get H323
> working for this test instead.
>
> I hadn't used the feature navigator before, thanks for showing me.
>
> Ed
>
> On 12/22/05, Kevin Thorngren <kthorngr at cisco.com> wrote:
> > Hi Ed,
> >
> > According to the Feature Navigator 12.2(16a) on the VG200 does not
> > support MGCP PRI backhaul to CCM.  You will need to use a 12.2T
> version
> > or a 12.3 version on the VG200.
> >
> > You can search by feature from here:
> > http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
> >
> > Look for the feature "MGCP PRI backhaul and T1-CAS support for Call
> > Manager (IP-PBX)".  Then select the VG200.  This will give you all
> your
> > options and you can determine which IOS will best fit your needs.
> >
> > Also for a back to back config don't forget to but the "isdn
> > protocol-emulate network" command on the serial interface of one of
> the
> > routers to bring up layer 2.
> >
> > HTH,
> > Kevin
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From Aaron.Derooy at xerox.com  Fri Dec 23 10:29:09 2005
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Fri Dec 23 10:30:04 2005
Subject: [cisco-voip] Dial peer usage on H323 g/w
Message-ID: <E0EF33751DEA634DA817A51024A8E3F2056CAA0B@usa0300ms01.na.xerox.net>

Anybody know a quick way to monitor dial peers? I want to clean up my
h323 g/w and it would be nice to know which dial peers get used and
which never get used. There must me an easy way to figure this out
without tracking individual calls using debug and/or show commands.
Thanks,

Aaron 



From CMStaffin at ra.rockwell.com  Fri Dec 23 10:47:30 2005
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Fri Dec 23 10:47:11 2005
Subject: [cisco-voip] IOS MGCP question
Message-ID: <OF2A5E322F.B30D186A-ON862570E0.005683F3-862570E0.0056BEEA@ra.rockwell.com>

Ed/Carlos

You are correct the VG248 is not IOS based.  It is completely menu driven 
unfortionatly.  However it does talk Skinny back to call manager so the 
ports all come up and look like phones.  Very little configuration on the 
box itself.

Craig Staffin
____________________________
Rockwell Automation
Network Analyst
cmstaffin at ra.rockwell.com
414-382-2309






"Ortiz, Carlos" <CORTIZ at broward.org>
Sent by: cisco-voip-bounces at puck.nether.net
12/23/2005 09:23 AM

 
        To:     "Ed Leatherman" <ealeatherman at gmail.com>
        cc:     cisco-voip at puck.nether.net
        Subject:        RE: [cisco-voip] IOS MGCP question


I think you are better off with the VG224's.  They are IOS based and are
basically like configuring a router.  Unless the 248's have changed (I
haven't looked) they are not IOS based and I remember they were a little
more troublesome.  Maybe someone else out there can confirm this and you
will feel better about going this route.

Carlos

-----Original Message-----
From: Ed Leatherman [mailto:ealeatherman at gmail.com] 
Sent: Thursday, December 22, 2005 8:00 PM
To: Ortiz, Carlos
Subject: Re: [cisco-voip] IOS MGCP question

Hey Carlos good to hear from you. We're getting some VG224's in to use
for a few dormitories at a regional campus, hopefully they will work
out OK. I would have rather gone with some 248's and had fewer boxes
to manage, but the 'customer' wants dual connections to all boxes
where possible.

On 12/22/05, Ortiz, Carlos <CORTIZ at broward.org> wrote:
> Hey Ed, I remember you from CIPTUG! The newer VG224's work really well
> with MGCP.  The only issue I currently have is that my IOS version has
a
> AAA bug in it.  I need to upgrade my memory to fix this.  Really
hasn't
> impacted me though because the boxes run and I never touch them....
>
> Carlos
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
> Sent: Thursday, December 22, 2005 3:42 PM
> To: Kevin Thorngren
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] IOS MGCP question
>
> Kevin,
>
> Turns out none of the VG200's I have came with enough flash memory to
> upgrade to any versions supporting MGCP PRI backhaul.. they are all
> 8MB flash and the lowest version of 12.2T that hasnt been removed from
> CCO (that I could find) is just a bit bigger than 8MB. All the 8MB
> versions have been removed, too old I guess. I'll try to get H323
> working for this test instead.
>
> I hadn't used the feature navigator before, thanks for showing me.
>
> Ed
>
> On 12/22/05, Kevin Thorngren <kthorngr at cisco.com> wrote:
> > Hi Ed,
> >
> > According to the Feature Navigator 12.2(16a) on the VG200 does not
> > support MGCP PRI backhaul to CCM.  You will need to use a 12.2T
> version
> > or a 12.3 version on the VG200.
> >
> > You can search by feature from here:
> > http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
> >
> > Look for the feature "MGCP PRI backhaul and T1-CAS support for Call
> > Manager (IP-PBX)".  Then select the VG200.  This will give you all
> your
> > options and you can determine which IOS will best fit your needs.
> >
> > Also for a back to back config don't forget to but the "isdn
> > protocol-emulate network" command on the serial interface of one of
> the
> > routers to bring up layer 2.
> >
> > HTH,
> > Kevin
> >
>
> _______________________________________________
> 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  Fri Dec 23 10:57:59 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri Dec 23 10:58:07 2005
Subject: [cisco-voip] IOS MGCP question
References: <OF2A5E322F.B30D186A-ON862570E0.005683F3-862570E0.0056BEEA@ra.rockwell.com>
Message-ID: <001c01c607d9$a583e520$30196883@cfs.uoguelph.ca>

I'm starting a 'Save the VG248' campaign. Anyone want to join?

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter than an
IP phone on a long haul 10base2 connection"                                           LFJ
  ----- Original Message ----- 
  From: Craig M Staffin 
  To: Ortiz, Carlos 
  Cc: cisco-voip-bounces at puck.nether.net ; Ed Leatherman ; cisco-voip at puck.nether.net 
  Sent: Friday, December 23, 2005 10:47 AM
  Subject: RE: [cisco-voip] IOS MGCP question



  Ed/Carlos 

  You are correct the VG248 is not IOS based.  It is completely menu driven unfortionatly.  However it does talk Skinny back to call manager so the ports all come up and look like phones.  Very little configuration on the box itself. 

  Craig Staffin 
  ____________________________ 
  Rockwell Automation 
  Network Analyst 
  cmstaffin at ra.rockwell.com 
  414-382-2309 




       "Ortiz, Carlos" <CORTIZ at broward.org> 
        Sent by: cisco-voip-bounces at puck.nether.net 
        12/23/2005 09:23 AM 

               
                To:        "Ed Leatherman" <ealeatherman at gmail.com> 
                cc:        cisco-voip at puck.nether.net 
                Subject:        RE: [cisco-voip] IOS MGCP question 



  I think you are better off with the VG224's.  They are IOS based and are
  basically like configuring a router.  Unless the 248's have changed (I
  haven't looked) they are not IOS based and I remember they were a little
  more troublesome.  Maybe someone else out there can confirm this and you
  will feel better about going this route.

  Carlos

  -----Original Message-----
  From: Ed Leatherman [mailto:ealeatherman at gmail.com] 
  Sent: Thursday, December 22, 2005 8:00 PM
  To: Ortiz, Carlos
  Subject: Re: [cisco-voip] IOS MGCP question

  Hey Carlos good to hear from you. We're getting some VG224's in to use
  for a few dormitories at a regional campus, hopefully they will work
  out OK. I would have rather gone with some 248's and had fewer boxes
  to manage, but the 'customer' wants dual connections to all boxes
  where possible.

  On 12/22/05, Ortiz, Carlos <CORTIZ at broward.org> wrote:
  > Hey Ed, I remember you from CIPTUG! The newer VG224's work really well
  > with MGCP.  The only issue I currently have is that my IOS version has
  a
  > AAA bug in it.  I need to upgrade my memory to fix this.  Really
  hasn't
  > impacted me though because the boxes run and I never touch them....
  >
  > Carlos
  >
  > -----Original Message-----
  > From: cisco-voip-bounces at puck.nether.net
  > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
  > Sent: Thursday, December 22, 2005 3:42 PM
  > To: Kevin Thorngren
  > Cc: cisco-voip at puck.nether.net
  > Subject: Re: [cisco-voip] IOS MGCP question
  >
  > Kevin,
  >
  > Turns out none of the VG200's I have came with enough flash memory to
  > upgrade to any versions supporting MGCP PRI backhaul.. they are all
  > 8MB flash and the lowest version of 12.2T that hasnt been removed from
  > CCO (that I could find) is just a bit bigger than 8MB. All the 8MB
  > versions have been removed, too old I guess. I'll try to get H323
  > working for this test instead.
  >
  > I hadn't used the feature navigator before, thanks for showing me.
  >
  > Ed
  >
  > On 12/22/05, Kevin Thorngren <kthorngr at cisco.com> wrote:
  > > Hi Ed,
  > >
  > > According to the Feature Navigator 12.2(16a) on the VG200 does not
  > > support MGCP PRI backhaul to CCM.  You will need to use a 12.2T
  > version
  > > or a 12.3 version on the VG200.
  > >
  > > You can search by feature from here:
  > > http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
  > >
  > > Look for the feature "MGCP PRI backhaul and T1-CAS support for Call
  > > Manager (IP-PBX)".  Then select the VG200.  This will give you all
  > your
  > > options and you can determine which IOS will best fit your needs.
  > >
  > > Also for a back to back config don't forget to but the "isdn
  > > protocol-emulate network" command on the serial interface of one of
  > the
  > > routers to bring up layer 2.
  > >
  > > HTH,
  > > Kevin
  > >
  >
  > _______________________________________________
  > 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
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From CMStaffin at ra.rockwell.com  Fri Dec 23 11:17:14 2005
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Fri Dec 23 11:16:51 2005
Subject: [cisco-voip] IOS MGCP question
Message-ID: <OFFFCCD3CC.D31E53CE-ON862570E0.005922ED-862570E0.00597773@ra.rockwell.com>

Actually I would love to contribute right now that is our only way to 
integrate our Audix voicemail into the call manager cluster.  By using 
SMDI over the Async port of the VG248

Thanks,
__________________________
Craig Staffin
Network Analyst
Global Network and Telecomunnications
Rockwell Automation
cmstaffin at ra.rockwell.com
(414) 382-2309






"Lelio Fulgenzi" <lelio at uoguelph.ca>
12/23/2005 09:57 AM

 
        To:     "Ortiz, Carlos" <CORTIZ at broward.org>, "Craig M Staffin" 
<CMStaffin at ra.rockwell.com>
        cc:     <cisco-voip-bounces at puck.nether.net>, "Ed Leatherman" 
<ealeatherman at gmail.com>, <cisco-voip at puck.nether.net>
        Subject:        Re: [cisco-voip] IOS MGCP question


I'm starting a 'Save the VG248' campaign. Anyone want to join?
 
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I had a coffee and Coke at lunch today...and now, I've got more jitter 
than an
IP phone on a long haul 10base2 connection"            LFJ
----- Original Message ----- 
From: Craig M Staffin 
To: Ortiz, Carlos 
Cc: cisco-voip-bounces at puck.nether.net ; Ed Leatherman ; cisco-voip at puck.nether.net 
Sent: Friday, December 23, 2005 10:47 AM
Subject: RE: [cisco-voip] IOS MGCP question


Ed/Carlos 

You are correct the VG248 is not IOS based.  It is completely menu driven 
unfortionatly.  However it does talk Skinny back to call manager so the 
ports all come up and look like phones.  Very little configuration on the 
box itself. 

Craig Staffin 
____________________________ 
Rockwell Automation 
Network Analyst 
cmstaffin at ra.rockwell.com 
414-382-2309 





"Ortiz, Carlos" <CORTIZ at broward.org> 
Sent by: cisco-voip-bounces at puck.nether.net 
12/23/2005 09:23 AM 
        
        To:        "Ed Leatherman" <ealeatherman at gmail.com> 
        cc:        cisco-voip at puck.nether.net 
        Subject:        RE: [cisco-voip] IOS MGCP question



I think you are better off with the VG224's.  They are IOS based and are
basically like configuring a router.  Unless the 248's have changed (I
haven't looked) they are not IOS based and I remember they were a little
more troublesome.  Maybe someone else out there can confirm this and you
will feel better about going this route.

Carlos

-----Original Message-----
From: Ed Leatherman [mailto:ealeatherman at gmail.com] 
Sent: Thursday, December 22, 2005 8:00 PM
To: Ortiz, Carlos
Subject: Re: [cisco-voip] IOS MGCP question

Hey Carlos good to hear from you. We're getting some VG224's in to use
for a few dormitories at a regional campus, hopefully they will work
out OK. I would have rather gone with some 248's and had fewer boxes
to manage, but the 'customer' wants dual connections to all boxes
where possible.

On 12/22/05, Ortiz, Carlos <CORTIZ at broward.org> wrote:
> Hey Ed, I remember you from CIPTUG! The newer VG224's work really well
> with MGCP.  The only issue I currently have is that my IOS version has
a
> AAA bug in it.  I need to upgrade my memory to fix this.  Really
hasn't
> impacted me though because the boxes run and I never touch them....
>
> Carlos
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
> Sent: Thursday, December 22, 2005 3:42 PM
> To: Kevin Thorngren
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] IOS MGCP question
>
> Kevin,
>
> Turns out none of the VG200's I have came with enough flash memory to
> upgrade to any versions supporting MGCP PRI backhaul.. they are all
> 8MB flash and the lowest version of 12.2T that hasnt been removed from
> CCO (that I could find) is just a bit bigger than 8MB. All the 8MB
> versions have been removed, too old I guess. I'll try to get H323
> working for this test instead.
>
> I hadn't used the feature navigator before, thanks for showing me.
>
> Ed
>
> On 12/22/05, Kevin Thorngren <kthorngr at cisco.com> wrote:
> > Hi Ed,
> >
> > According to the Feature Navigator 12.2(16a) on the VG200 does not
> > support MGCP PRI backhaul to CCM.  You will need to use a 12.2T
> version
> > or a 12.3 version on the VG200.
> >
> > You can search by feature from here:
> > http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
> >
> > Look for the feature "MGCP PRI backhaul and T1-CAS support for Call
> > Manager (IP-PBX)".  Then select the VG200.  This will give you all
> your
> > options and you can determine which IOS will best fit your needs.
> >
> > Also for a back to back config don't forget to but the "isdn
> > protocol-emulate network" command on the serial interface of one of
> the
> > routers to bring up layer 2.
> >
> > HTH,
> > Kevin
> >
>
> _______________________________________________
> 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 PRuttman at foley.com  Fri Dec 23 11:25:20 2005
From: PRuttman at foley.com (Ruttman, Peter G.)
Date: Fri Dec 23 11:25:29 2005
Subject: [cisco-voip] Call rings phones once and then drops
Message-ID: <7E3064B7EFC16C41B185C41D141008B7C3D43F@foleyex97.foleylaw.com>

I had this problem when the PRIs got full and cisco had me change dsps
from flex to medium mode.  This fixed the problem for us.
 
Pete
 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Saturday, December 17, 2005 1:34 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call rings phones once and then drops



Calls ring in on local PRI trunks on 3845 router, makes its way to
callmanager, makes its way to phone and rings once on the phone.  But
that is it, calls instantly drop after first ring of the phone.  Does
not give person chance to answer just drops without any user
intervention. 
 
Using MGCP between 3845 and callmanager.
 
Any ideas?


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 Eric.A.Fretz at L-3com.com  Fri Dec 23 12:27:33 2005
From: Eric.A.Fretz at L-3com.com (Fretz, EA Eric @ IS)
Date: Fri Dec 23 12:27:52 2005
Subject: [cisco-voip] Dial peer usage on H323 g/w
Message-ID: <51A2C3ED40A3584281444F65BC9026270A884525@gvlexch02.gvl.is.l-3com.com>

You might look at some of the DIAL-CONTROL mibs available on your router.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of DeRooy, Aaron H
Sent: Friday, December 23, 2005 9:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Dial peer usage on H323 g/w


Anybody know a quick way to monitor dial peers? I want to clean up my h323
g/w and it would be nice to know which dial peers get used and which never
get used. There must me an easy way to figure this out without tracking
individual calls using debug and/or show commands. Thanks,

Aaron 


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

From jsteinberg at gmail.com  Fri Dec 23 12:52:44 2005
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Fri Dec 23 12:52:47 2005
Subject: [cisco-voip] IOS MGCP question
Message-ID: <f1b65f880512230952r2eae8701x@mail.gmail.com>

I agree, the VG248's do a fine job integrating the callmanager with
older voicemail systems that use SMDI integration.

However, you could also use the Cisco Message Interface service on the
CallManager for SMDI integration.  You would then run the SMDI link
from the CallManager's serial port to your voicemail system.  You
would still need a gateway of some sort but aleast you have options
other than VG248.

I have no problem with the VG248, it does a fine job - but it has
always seemed like the red headed stepchild.  The newest code 1.3.1
doesn't even exist on the bug toolkit.  I'm mentioned that to TAC
before - i think it is to encourage people to migrate to Unity. :)

Justin

> Date: Fri, 23 Dec 2005 10:17:14 -0600
> From: Craig M Staffin <CMStaffin at ra.rockwell.com>
> Subject: Re: [cisco-voip] IOS MGCP question
> To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Cc: cisco-voip-bounces at puck.nether.net, "Ortiz, Carlos"
>        <CORTIZ at broward.org>,   cisco-voip at puck.nether.net, Ed Leatherman
>        <ealeatherman at gmail.com>
> Message-ID:
>        <OFFFCCD3CC.D31E53CE-ON862570E0.005922ED-862570E0.00597773 at ra.rockwell.com>
>
> Content-Type: text/plain; charset="us-ascii"
>
> Actually I would love to contribute right now that is our only way to
> integrate our Audix voicemail into the call manager cluster.  By using
> SMDI over the Async port of the VG248
>
> Thanks,
> __________________________
> Craig Staffin
> Network Analyst
> Global Network and Telecomunnications
> Rockwell Automation
> cmstaffin at ra.rockwell.com
> (414) 382-2309
>
>


From CMStaffin at ra.rockwell.com  Fri Dec 23 13:02:58 2005
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Fri Dec 23 13:02:49 2005
Subject: [cisco-voip] IOS MGCP question
Message-ID: <OF9B672967.A770C2C2-ON862570E0.0062FC4A-862570E0.0063259D@ra.rockwell.com>

Correct we could use the port off of one of the call manager servers but 
we have a geography problem.  Our Call Manager server are located in 
Milwaukee and Cleaveland.  Where as our voicemail system is located in 
Milwaukee and also in Brazil.

So unfortionatly the only thing we can do is use the red headed stepchil 
and hope cisco does not end support for it or comes out with something to 
replace it wtih.

Thanks,
__________________________
Craig Staffin
Network Analyst
Global Network and Telecomunnications
Rockwell Automation
cmstaffin at ra.rockwell.com
(414) 382-2309






Justin Steinberg <jsteinberg at gmail.com>
Sent by: cisco-voip-bounces at puck.nether.net
12/23/2005 11:52 AM

 
        To:     CMStaffin at ra.rockwell.com, cisco-voip at puck.nether.net
        cc: 
        Subject:        Re: [cisco-voip] IOS MGCP question


I agree, the VG248's do a fine job integrating the callmanager with
older voicemail systems that use SMDI integration.

However, you could also use the Cisco Message Interface service on the
CallManager for SMDI integration.  You would then run the SMDI link
from the CallManager's serial port to your voicemail system.  You
would still need a gateway of some sort but aleast you have options
other than VG248.

I have no problem with the VG248, it does a fine job - but it has
always seemed like the red headed stepchild.  The newest code 1.3.1
doesn't even exist on the bug toolkit.  I'm mentioned that to TAC
before - i think it is to encourage people to migrate to Unity. :)

Justin

> Date: Fri, 23 Dec 2005 10:17:14 -0600
> From: Craig M Staffin <CMStaffin at ra.rockwell.com>
> Subject: Re: [cisco-voip] IOS MGCP question
> To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Cc: cisco-voip-bounces at puck.nether.net, "Ortiz, Carlos"
>        <CORTIZ at broward.org>,   cisco-voip at puck.nether.net, Ed Leatherman
>        <ealeatherman at gmail.com>
> Message-ID:
> 
<OFFFCCD3CC.D31E53CE-ON862570E0.005922ED-862570E0.00597773 at ra.rockwell.com>
>
> Content-Type: text/plain; charset="us-ascii"
>
> Actually I would love to contribute right now that is our only way to
> integrate our Audix voicemail into the call manager cluster.  By using
> SMDI over the Async port of the VG248
>
> Thanks,
> __________________________
> Craig Staffin
> Network Analyst
> Global Network and Telecomunnications
> Rockwell Automation
> cmstaffin at ra.rockwell.com
> (414) 382-2309
>
>

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


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From ciscozest at gmail.com  Fri Dec 23 13:07:40 2005
From: ciscozest at gmail.com (Ciscozest)
Date: Fri Dec 23 13:07:36 2005
Subject: [cisco-voip] Help: Problem  on AC Hunting System
References: <20051223045404.76819.qmail@web34309.mail.mud.yahoo.com>
Message-ID: <006a01c607eb$c65472d0$6301a8c0@windfall99>

I configured a Hunting group in CCM 3.3(4). the members are: DN 4701, DN 4702, DN 4703, and DN 8000 (voicemail). I use 
longest idle algorithm. Whenever I dial to the AC pilot, it went to either 4701 or 4702 or 4703 then to voicemail. What 
I want is to route the call to any of user DN then if it is not answered/busy, the call will be routed to another user 
DN and eventually to the voicemail. Can this be done? Also would like to know how to set ringing timeout for this 
hunting system? Thanks

regards,
Robin 


From candace_holman at harvard.edu  Fri Dec 23 13:22:47 2005
From: candace_holman at harvard.edu (Candace Holman)
Date: Fri Dec 23 13:22:56 2005
Subject: [cisco-voip] VM Pilot or FwdNoAns Issue
Message-ID: <43AC4077.90902@harvard.edu>

The readme says:*  "Caution*:  Do not apply this service release with OS 
version 2000.2.7, 2000.4.1 or previous OS release trains.  This service 
release is only compatible with the OS 2000.4.2 release train."

Did you install 2000.4.2 first?

Candace

    Have you gotten CCM and SDL traces of these calls?   An OS update  
    *should* have no impact on how CM call processing works.

    If it's getting widespread and you have data for it please open a TAC  
    case.

    -Ryan

    On Dec 22, 2005, at 2:45 PM, VoIP Forum wrote:

    Has anyone recently upgraded OS from the 2000.2.7 train to 2000.4.2?



    After an upgrade from 2000.2.7sr8 to 2000.4.2sr2 we have been getting  
    dispersed reports that calls are not rolling to VM after the NoAns  
    timeout. Does this sound familiar to anyone?

    Do we need an ES to resolve this?




From erickbe at yahoo.com  Sat Dec 24 00:20:51 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sat Dec 24 00:21:00 2005
Subject: [cisco-voip] Dial peer usage on H323 g/w
In-Reply-To: <51A2C3ED40A3584281444F65BC9026270A884525@gvlexch02.gvl.is.l-3com.com>
Message-ID: <20051224052051.21602.qmail@web34305.mail.mud.yahoo.com>

Hi,

You could also do a 'show dial-peer voice | inc
tag|Calls|Connect Time' and this will output the
dial-peer id (tag) and number of calls attempted,
successful, failed, etc and total time calls have been
connected using that dial-peer since counters were
cleared or router was rebooted.


--- "Fretz, EA Eric @ IS" <Eric.A.Fretz at L-3com.com>
wrote:

> You might look at some of the DIAL-CONTROL mibs
> available on your router.
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of DeRooy, Aaron H
> Sent: Friday, December 23, 2005 9:29 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Dial peer usage on H323 g/w
> 
> 
> Anybody know a quick way to monitor dial peers? I
> want to clean up my h323
> g/w and it would be nice to know which dial peers
> get used and which never
> get used. There must me an easy way to figure this
> out without tracking
> individual calls using debug and/or show commands.
> Thanks,
> 
> Aaron 
> 
> 
> _______________________________________________
> 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
> 



	
		
__________________________________ 
Yahoo! for Good - Make a difference this year. 
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From qpduong at yahoo.com  Sun Dec 25 23:26:16 2005
From: qpduong at yahoo.com (duongquang phon)
Date: Sun Dec 25 23:26:23 2005
Subject: [cisco-voip] no ringback tone with call from PSTN->CCM->ATA
Message-ID: <20051226042616.35746.qmail@web53715.mail.yahoo.com>

Hi all,

I have deployed a call scenario as follow:

PSTN---> AS5300---->CallManager----->ATA 186

and I have faced two problems as follow:
(i)first I see that CallManager converts all h323
messages with Faststart element that come from AS5300
or ATA into slow start mode.
(ii)Because of (i) all calls from PSTN to ATA will use
h323 slow start mode so it drives to another problem
as follow:
In slow start mode ATA 186 does not send h245 address
back to AS5300 until Connect message so no media
channel opened until Connect message so caller from
PSTN can not hear ringback tone from ATA. In oder to
solve this problem I have forced AS5300 to play
ringback tone locally by using command: tone ringback
alert-no-pi  under voip dial-peer but sometimes it can
play sometimes can not.

Any thoughts?

Thanks,
Phon



		
___________________________________________________________ 
Yahoo! Photos ? NEW, now offering a quality print service from just 8p a photo http://uk.photos.yahoo.com

From subs-cisco at cbvcollections.com  Mon Dec 26 02:40:36 2005
From: subs-cisco at cbvcollections.com (Cisco Subscriber)
Date: Mon Dec 26 02:41:03 2005
Subject: [cisco-voip] Difference between National, CallManager,
	ISDN for Caller ID?
Message-ID: <9AEF8665DE74FA43B3B987FB931E92AE0ADB14@CBVVANEM01.cbv.dns>

Had to set mine to unknown for call routing to happen correctly (see
previous posts regarding routing using 9.@/route filter).  Used to have
them set to Call Manager/National.  The only thing that allowed all
calls to route properly with my provider was to set it to unknown and
have them determine the numbering plan.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, December 22, 2005 4:19 PM
To: Jonathan Charles; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Difference between National, CallManager,ISDN
for Caller ID?



If it's set to CM it will try to do it.  If you set it to national isdn
it sends that way.  Most times CM and figure it out, but some telcos
want it one way or the other and then you end up changing to what they
want.  

 

One of my PRIs is set to CM and the other is set to national isdn.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Thursday, December 22, 2005 3:19 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Difference between National, CallManager,ISDN for
Caller ID?

 

What is it?

Is this telco dependent, or do are they specific to customer needs?

Also, what is the difference of manipulations between:

Outbound Calls

Calling Line ID Presentation*

--- Not Selected --- Default Allowed Restricted 	 

Calling Party Selection*

--- Not Selected --- Originator First Redirect Number Last Redirect
Number First Redirect Number (External) Last Redirect Number (External)


Called party IE number type unknown*

--- Not Selected --- Cisco CallManager Unknown National International
Subscriber 	 

Calling party IE number type unknown*

--- Not Selected --- Cisco CallManager Unknown National International
Subscriber 	 

Called Numbering Plan*

--- Not Selected --- Cisco CallManager ISDN National Standard Private
Unknown   

 

Calling Numbering Plan*

--- Not Selected --- Cisco CallManager ISDN National Standard Private
Unknown   

 

Number of digits to strip*

--- Not Selected --- 0 1 2 3 4 5 6   7 8 9 10 11 12 13  14 15 16 17 18
19 20 21   22 23 24 25 26 27 28  29 30 31 32 	 

Caller ID DN

 

 

 

SMDI Base Port*

 

 

 


In other words, how can you manipulate what is and is not sent down a
PRI, and what overrides and combinations work?


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From Antonino.Monteiro at oni.pt  Mon Dec 26 07:55:31 2005
From: Antonino.Monteiro at oni.pt (Antonino Mendes Monteiro)
Date: Mon Dec 26 07:55:39 2005
Subject: [cisco-voip] Call Manager Express and Ericsson MD110
Message-ID: <3D60AF1F07F3734BBCA585EECA7E52538782F9@excmat203.oni.pt>

Season Greetings,
 
We have a customer that pretends to connect a Ericsson MD110 PBX to a
Cisco Call Manager Express router through VoIP using H.323.
 
I took a look on Cisco web site, but only found references to Q.SIG and
ISDN implementations using PRI or BRI interfaces to MD110.
 
Does anybody has the experience of connecting Cisco products, namely
CCME to directly Ericsson MD110 using either H.323 or SIP?
 
Thanks in advanced,
Antonino
 
 
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From rratliff at cisco.com  Tue Dec 27 09:40:12 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue Dec 27 09:40:10 2005
Subject: [cisco-voip] no ringback tone with call from PSTN->CCM->ATA
In-Reply-To: <20051226042616.35746.qmail@web53715.mail.yahoo.com>
References: <20051226042616.35746.qmail@web53715.mail.yahoo.com>
Message-ID: <13A6FAFD-5D6E-4319-9F77-B1D3CFBDC659@cisco.com>

Phon,

Later versions of CM do support H.323 fast start both inbound and  
outbound.  Inbound faststart should be supported in 3.3 and later.

For your ringback issue check out http://www.cisco.com/en/US/tech/ 
tk1077/technologies_tech_note09186a0080094c33.shtml.

-Ryan

On Dec 25, 2005, at 11:26 PM, duongquang phon wrote:

Hi all,

I have deployed a call scenario as follow:

PSTN---> AS5300---->CallManager----->ATA 186

and I have faced two problems as follow:
(i)first I see that CallManager converts all h323
messages with Faststart element that come from AS5300
or ATA into slow start mode.
(ii)Because of (i) all calls from PSTN to ATA will use
h323 slow start mode so it drives to another problem
as follow:
In slow start mode ATA 186 does not send h245 address
back to AS5300 until Connect message so no media
channel opened until Connect message so caller from
PSTN can not hear ringback tone from ATA. In oder to
solve this problem I have forced AS5300 to play
ringback tone locally by using command: tone ringback
alert-no-pi  under voip dial-peer but sometimes it can
play sometimes can not.

Any thoughts?

Thanks,
Phon



		
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8p a photo http://uk.photos.yahoo.com
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From Ruben.Montes at eu.didata.com  Tue Dec 27 14:27:23 2005
From: Ruben.Montes at eu.didata.com (Ruben Montes)
Date: Tue Dec 27 14:27:27 2005
Subject: [cisco-voip] BRI connection goes up and down
Message-ID: <D305A0848689344BB98F14A3F13161B714ADC1@eubebrusvex.eu.didata.local>

Hello,

 

I have configured a BRI MGCP connection with a CCM 4.1(3) and a C3845
with IOS 12.3(14)T5.

 

I have a NM with 2 VIC with 2xBRI each one (4 BRI total). I have the
message B-channel out of service all the time in all the four BRIs. In
the router I can see line is going up and down all the time. I can see
that I receive a DISC from the telco (with a debug isdn q921) but I
don't know why. We have four BRIs so it is very strange to be a telco
problem...

If I do the config with H323 I haven't this error.

 

I'm pretty sure this a configuration issue related with ISDn
negotiation, but I don't know how to solve it (I've tried a bunch of
option in the CCM MGCP config page and also in the router...)

 

Regards,

 

Ruben

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From jgrace at digitelusa.net  Tue Dec 27 16:15:07 2005
From: jgrace at digitelusa.net (James Grace)
Date: Tue Dec 27 16:19:46 2005
Subject: [cisco-voip] bulk import is missing for 405
Message-ID: <200512271619203.SM01568@3729434770>

We are trying to use the bulk exchange import  and it missing.  Do and body
know where I can get it  we are using unity 4.0.5

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Scott.Voll at wesd.org  Tue Dec 27 16:25:18 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue Dec 27 16:25:29 2005
Subject: [cisco-voip] bulk import is missing for 405
Message-ID: <D713462ED535184D830F6F6301753E7E01A121B0@VISHNU.wesd.org>

Have you looked here? http://www.ciscounitytools.com/

 

Or is that one of the ones that's in the start --> programs --> unity?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, December 27, 2005 1:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] bulk import is missing for 405

 

We are trying to use the bulk exchange import  and it missing.  Do and
body know where I can get it  we are using unity 4.0.5

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From jgrace at digitelusa.net  Tue Dec 27 20:11:50 2005
From: jgrace at digitelusa.net (James)
Date: Tue Dec 27 20:11:53 2005
Subject: [cisco-voip] Configing
Message-ID: <1676083188-1135732311-cardhu_blackberry.rim.net-17719-@engine08-cell02>

Where to I cong the t1 seting for. A p2p on my gateway.  I'm looking for where do I put the bazs line coding and stuff.   The reason I'm asking is because. We don't get the controller on the routrer config.  Only serial
James D. Grace 
System Engineer
CCNP CCNA MCSE MCDBA 
Digitel Corp.

From wsisk at cisco.com  Tue Dec 27 20:14:14 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Tue Dec 27 20:14:24 2005
Subject: [cisco-voip] IOS MGCP question
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2BBC@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2BBC@MREXVS01.bc.broward.cty>
Message-ID: <E9F04921-D68A-4694-BA7D-D2F7D5904955@cisco.com>

And the vg224 lacks the advanced call control modes of the vg248  
(conference, park, transfer...)

But look for those coming out next year on the vg224.  Looks like the  
EFT is already in progress.

/Wes
On Dec 22, 2005, at 3:46 PM, Ortiz, Carlos wrote:

Hey Ed, I remember you from CIPTUG! The newer VG224's work really well
with MGCP.  The only issue I currently have is that my IOS version has a
AAA bug in it.  I need to upgrade my memory to fix this.  Really hasn't
impacted me though because the boxes run and I never touch them....

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Thursday, December 22, 2005 3:42 PM
To: Kevin Thorngren
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IOS MGCP question

Kevin,

Turns out none of the VG200's I have came with enough flash memory to
upgrade to any versions supporting MGCP PRI backhaul.. they are all
8MB flash and the lowest version of 12.2T that hasnt been removed from
CCO (that I could find) is just a bit bigger than 8MB. All the 8MB
versions have been removed, too old I guess. I'll try to get H323
working for this test instead.

I hadn't used the feature navigator before, thanks for showing me.

Ed

On 12/22/05, Kevin Thorngren <kthorngr at cisco.com> wrote:
> Hi Ed,
>
> According to the Feature Navigator 12.2(16a) on the VG200 does not
> support MGCP PRI backhaul to CCM.  You will need to use a 12.2T
version
> or a 12.3 version on the VG200.
>
> You can search by feature from here:
> http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
>
> Look for the feature "MGCP PRI backhaul and T1-CAS support for Call
> Manager (IP-PBX)".  Then select the VG200.  This will give you all
your
> options and you can determine which IOS will best fit your needs.
>
> Also for a back to back config don't forget to but the "isdn
> protocol-emulate network" command on the serial interface of one of
the
> routers to bring up layer 2.
>
> HTH,
> Kevin
>

_______________________________________________
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 wsisk at cisco.com  Tue Dec 27 20:15:41 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Tue Dec 27 20:15:59 2005
Subject: [cisco-voip] VM Pilot or FwdNoAns Issue
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2BB9@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2BB9@MREXVS01.bc.broward.cty>
Message-ID: <361D5B18-568D-48D9-A2BB-0777D6D4BA9C@cisco.com>

yes, migration to 2000.4.2 will be mandatory.  there will be no more  
releases for the 2000.4.1 or 2000.2.7 OS versions.  There are already  
some fixes in 2000.4.2 that are not in 2000.4.1 or 2000.2.7.

/Wes
On Dec 22, 2005, at 3:00 PM, Ortiz, Carlos wrote:

That?s not good news.  Is it mandatory to move to this train?



Carlos



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of VoIP Forum
Sent: Thursday, December 22, 2005 2:46 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VM Pilot or FwdNoAns Issue



Has anyone recently upgraded OS from the 2000.2.7 train to 2000.4.2?



After an upgrade from 2000.2.7sr8 to 2000.4.2sr2 we have been getting  
dispersed reports that calls are not rolling to VM after the NoAns  
timeout. Does this sound familiar to anyone?

Do we need an ES to resolve this?

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

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From wsisk at cisco.com  Tue Dec 27 20:18:54 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Tue Dec 27 20:19:04 2005
Subject: [cisco-voip] Configing
In-Reply-To: <1676083188-1135732311-cardhu_blackberry.rim.net-17719-@engine08-cell02>
References: <1676083188-1135732311-cardhu_blackberry.rim.net-17719-@engine08-cell02>
Message-ID: <F40BB762-DD4B-4874-AE0B-F13BBBEBC30D@cisco.com>

if it's the new hwic card that can be t1 or e1 you have to configure  
the card type first
http://www.ciscotaccc.com/core/showcase?case=K19239995

/Wes

On Dec 27, 2005, at 8:11 PM, James wrote:

Where to I cong the t1 seting for. A p2p on my gateway.  I'm looking  
for where do I put the bazs line coding and stuff.   The reason I'm  
asking is because. We don't get the controller on the routrer  
config.  Only serial
James D. Grace
System Engineer
CCNP CCNA MCSE MCDBA
Digitel Corp.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From kthorngr at cisco.com  Tue Dec 27 20:37:38 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Tue Dec 27 20:38:35 2005
Subject: [cisco-voip] Configing
In-Reply-To: <1676083188-1135732311-cardhu_blackberry.rim.net-17719-@engine08-cell02>
References: <1676083188-1135732311-cardhu_blackberry.rim.net-17719-@engine08-cell02>
Message-ID: <e5bea530e95a0e92e0c69e2b26af5510@cisco.com>

It would be good for you to provide more detail on what you are trying  
to do.

Are you trying to do voice or data?

Do you have WIC, VWIC or HWIC interfaces or something else?

You may need to add the "card type t1" command, for example:
http://www.cisco.com/en/US/products/hw/routers/ps221/ 
prod_configuration_guide09186a00800a4063.html

Kevin


On Dec 27, 2005, at 8:11 PM, James wrote:

> Where to I cong the t1 seting for. A p2p on my gateway.  I'm looking  
> for where do I put the bazs line coding and stuff.   The reason I'm  
> asking is because. We don't get the controller on the routrer config.   
> Only serial
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From techguy at gmail.com  Wed Dec 28 09:16:25 2005
From: techguy at gmail.com (TechGuy)
Date: Wed Dec 28 09:16:27 2005
Subject: [cisco-voip] Pulling voicemail off of unity to save?
Message-ID: <a1b07ea60512280616k7ccdf07ey600b703336b1864c@mail.gmail.com>

There are a couple of voicemails in a users account that I am hoping to
retrieve so that it can be stored as well as given to someone for legal
reasons.  I cannot seem to find an easy way to retrieve this voicemail.  I
know it is stored on an exchange message store of course but not sure of the
best way to access this.  Any recommendations?  I am sure I can do it in a
round about way, but thought I would ask to see if there was an easy way.

Regards.
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From Scott.Voll at wesd.org  Wed Dec 28 10:43:51 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 28 10:44:08 2005
Subject: [cisco-voip] Pulling voicemail off of unity to save?
Message-ID: <D713462ED535184D830F6F6301753E7E01A121D0@VISHNU.wesd.org>

UM or VM.  Does the user have access to the VMail?  Can you just reset
the password and go in and save the attachment?  Or you could give
yourself exchange admin rights and put yourself on there account and go
in that way.  Can you get to the web interface to download it?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Wednesday, December 28, 2005 6:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Pulling voicemail off of unity to save?

 

There are a couple of voicemails in a users account that I am hoping to
retrieve so that it can be stored as well as given to someone for legal
reasons.  I cannot seem to find an easy way to retrieve this voicemail.
I know it is stored on an exchange message store of course but not sure
of the best way to access this.  Any recommendations?  I am sure I can
do it in a round about way, but thought I would ask to see if there was
an easy way. 

 

Regards.

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From techguy at gmail.com  Wed Dec 28 11:06:21 2005
From: techguy at gmail.com (TechGuy)
Date: Wed Dec 28 11:06:26 2005
Subject: [cisco-voip] Pulling voicemail off of unity to save?
In-Reply-To: <E7FEA6FD4640244FBE5AA4DDB39897480141E842@CTG-MSNEXC01.staff.berbee.com>
References: <E7FEA6FD4640244FBE5AA4DDB39897480141E842@CTG-MSNEXC01.staff.berbee.com>
Message-ID: <a1b07ea60512280806h4e143d3dl5bb679d3cf76507f@mail.gmail.com>

Brad seems to be on to something here in regards to accessing the M: drive.
Within the M: drive there is the domain name and then the MBX folder.
Inside the MBX folder there are each individual mailbox folder, however no
matter what login I use I am unable to access any of these folders. Only
think I have not tried is the actual administrator account on the local box
itself, I have tried the highest level of domain access we have but even it
has no permissions to access these folders.



On 12/28/05, Jungemann, Brad <brad.jungemann at berbee.com> wrote:
>
>  What version of Unity? If it's Exchange 2000 you can go into the M: Drive
> and the Users folder to hunt down the message. Exchange 2003 it's not so
> easy. I'm taking that this is Voice Mail Only or is it Unified?
>
> -Brad
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *TechGuy
> *Sent:* Wednesday, December 28, 2005 8:16 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Pulling voicemail off of unity to save?
>
>
>
> There are a couple of voicemails in a users account that I am hoping to
> retrieve so that it can be stored as well as given to someone for legal
> reasons.  I cannot seem to find an easy way to retrieve this voicemail.  I
> know it is stored on an exchange message store of course but not sure of the
> best way to access this.  Any recommendations?  I am sure I can do it in a
> round about way, but thought I would ask to see if there was an easy way.
>
>
>
> Regards.
>
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From mike.jones at berbee.com  Wed Dec 28 11:44:52 2005
From: mike.jones at berbee.com (Jones, Mike)
Date: Wed Dec 28 11:45:02 2005
Subject: [cisco-voip] Pulling voicemail off of unity to save?
Message-ID: <0E6152300039CE48B75770462D6B49FF7AA61C@CTG-MSNEXC01.staff.berbee.com>

You want to be very careful when snooping in the Exchange 2000 M drive.
If you do not see it listed in the folder directory it may have been
hidden per the recommendations of Microsoft.

http://support.microsoft.com/kb/305145/en-us

 

Thanks,

Mike J

  

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Wednesday, December 28, 2005 10:06 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Pulling voicemail off of unity to save?

 

Brad seems to be on to something here in regards to accessing the M:
drive.  Within the M: drive there is the domain name and then the MBX
folder.  Inside the MBX folder there are each individual mailbox folder,
however no matter what login I use I am unable to access any of these
folders. Only think I have not tried is the actual administrator account
on the local box itself, I have tried the highest level of domain access
we have but even it has no permissions to access these folders. 



 

On 12/28/05, Jungemann, Brad <brad.jungemann at berbee.com> wrote: 

What version of Unity? If it's Exchange 2000 you can go into the M:
Drive and the Users folder to hunt down the message. Exchange 2003 it's
not so easy. I'm taking that this is Voice Mail Only or is it Unified? 

-Brad 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Wednesday, December 28, 2005 8:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Pulling voicemail off of unity to save?

 

There are a couple of voicemails in a users account that I am hoping to
retrieve so that it can be stored as well as given to someone for legal
reasons.  I cannot seem to find an easy way to retrieve this voicemail.
I know it is stored on an exchange message store of course but not sure
of the best way to access this.  Any recommendations?  I am sure I can
do it in a round about way, but thought I would ask to see if there was
an easy way. 

 

Regards.

 

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From Scott.Voll at wesd.org  Wed Dec 28 11:54:35 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 28 11:54:43 2005
Subject: [cisco-voip] Pulling voicemail off of unity to save?
Message-ID: <D713462ED535184D830F6F6301753E7E01A121D5@VISHNU.wesd.org>

I'm not sure with 2000 but with 2003.  Domain admin does not have
permission to get to email accounts. Explicitly denied.   This is by
design as I'm told.  You want exchange admin.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Wednesday, December 28, 2005 8:06 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Pulling voicemail off of unity to save?

 

Brad seems to be on to something here in regards to accessing the M:
drive.  Within the M: drive there is the domain name and then the MBX
folder.  Inside the MBX folder there are each individual mailbox folder,
however no matter what login I use I am unable to access any of these
folders. Only think I have not tried is the actual administrator account
on the local box itself, I have tried the highest level of domain access
we have but even it has no permissions to access these folders. 



 

On 12/28/05, Jungemann, Brad <brad.jungemann at berbee.com> wrote: 

What version of Unity? If it's Exchange 2000 you can go into the M:
Drive and the Users folder to hunt down the message. Exchange 2003 it's
not so easy. I'm taking that this is Voice Mail Only or is it Unified? 

-Brad 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Wednesday, December 28, 2005 8:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Pulling voicemail off of unity to save?

 

There are a couple of voicemails in a users account that I am hoping to
retrieve so that it can be stored as well as given to someone for legal
reasons.  I cannot seem to find an easy way to retrieve this voicemail.
I know it is stored on an exchange message store of course but not sure
of the best way to access this.  Any recommendations?  I am sure I can
do it in a round about way, but thought I would ask to see if there was
an easy way. 

 

Regards.

 

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From PRuttman at foley.com  Wed Dec 28 12:11:34 2005
From: PRuttman at foley.com (Ruttman, Peter G.)
Date: Wed Dec 28 12:11:39 2005
Subject: [cisco-voip] Upgrade to 4.1(3) SR2?
Message-ID: <7E3064B7EFC16C41B185C41D141008B7CC35B7@foleyex97.foleylaw.com>

We have a few issues that are addressed in SR2.  Has anyone on the
listed loaded SR2 in a production environment yet?  Does anyone have any
reason to tell us to wait?

Pete

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 pmowry at getgds.com  Wed Dec 28 12:18:42 2005
From: pmowry at getgds.com (Patrick Mowry)
Date: Wed Dec 28 12:18:46 2005
Subject: [cisco-voip] TCL script to parse *ANI*DNIS*
Message-ID: <4C792A099259634C83F97067671B37FE01A87AE8@GDSEXC.gds.pri>

Hello,
 I have a carrier providing a T1 E&M-II wink start trunk to a Fujitsu
F9600.  They provide caller-id information by formatting the digits send
as *ANI*DNIS.  If I configure the controller as:
ds0-group 1 timeslots 1-24 type e&m-wink-start dnis

I can place outbound calls, but for inbound calls I'll see a number like
*3375551212*3600* , and this of course would not match anything my
dail-peers or CallManager expects.  Can a TCL script be used to break
this number up and pass the caller-id and DID to CallManager?  The IOS
is 12.4T AdvancedIP Services, and CCM 4.1.3sr2

Thanks,

-Patrick


From Aaron.Derooy at xerox.com  Wed Dec 28 12:36:22 2005
From: Aaron.Derooy at xerox.com (DeRooy, Aaron H)
Date: Wed Dec 28 12:37:03 2005
Subject: [cisco-voip] Progress_ind command
Message-ID: <E0EF33751DEA634DA817A51024A8E3F2056CAE2C@usa0300ms01.na.xerox.net>

 

Does anybody know if ANY progress_ind command is needed to make sure 
DTMF tones get thru the IP phone network to/from a PSTN destination?
I've been experimenting and find that the "dtmf-relay h245-alphanumeric"
command breaks DTMF tones but the progess_ind commands have no effect.
>From what I've been able to find out, progress_ind command solve
"ringback" problems with the "alert" key word but not sure about "setup"
and "progress" key works.
Thanks,
 
Aaron


From zephy316 at gmail.com  Wed Dec 28 12:57:01 2005
From: zephy316 at gmail.com (Israel Lang)
Date: Wed Dec 28 12:57:05 2005
Subject: [cisco-voip] Unity Service Packs
Message-ID: <91e236d30512280957n1dd1bc56x32e0c10530b8a308@mail.gmail.com>

Anybody have a good document or can quickly tell me the compatibility matrix
between the various versions of Unity, Exchange and Windows 2003? In
particular I am interest in WIndows 2003 SP1, Exchange SP1 and SP2, and
Unity 4.0x and 4.1.

Thanks,

--
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol
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From Scott.Voll at wesd.org  Wed Dec 28 13:01:42 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 28 13:01:51 2005
Subject: [cisco-voip] Unity Service Packs
Message-ID: <D713462ED535184D830F6F6301753E7E01A121DB@VISHNU.wesd.org>

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2237/products_
device_support_table09186a0080531aab.html

 

The only thing I didn't like is that Windows 2003 was not on the list
yet it is approved.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Wednesday, December 28, 2005 9:57 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Service Packs

 

Anybody have a good document or can quickly tell me the compatibility
matrix between the various versions of Unity, Exchange and Windows 2003?
In particular I am interest in WIndows 2003 SP1, Exchange SP1 and SP2,
and Unity 4.0x and 4.1.

Thanks,

-- 
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol 

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From Scott.Voll at wesd.org  Wed Dec 28 13:04:35 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 28 13:04:44 2005
Subject: [cisco-voip] Unity Service Packs
Message-ID: <D713462ED535184D830F6F6301753E7E01A121DC@VISHNU.wesd.org>

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2237/products_
device_support_table09186a0080531aab.html#wp60952

 

Sorry-it's under the recommended SP area.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, December 28, 2005 10:02 AM
To: Israel Lang; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity Service Packs

 

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2237/products_
device_support_table09186a0080531aab.html

 

The only thing I didn't like is that Windows 2003 was not on the list
yet it is approved.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Wednesday, December 28, 2005 9:57 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Service Packs

 

Anybody have a good document or can quickly tell me the compatibility
matrix between the various versions of Unity, Exchange and Windows 2003?
In particular I am interest in WIndows 2003 SP1, Exchange SP1 and SP2,
and Unity 4.0x and 4.1.

Thanks,

-- 
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol 

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From SPackett at fenwick.com  Wed Dec 28 13:11:21 2005
From: SPackett at fenwick.com (Stu Packett)
Date: Wed Dec 28 13:11:31 2005
Subject: [cisco-voip] Pulling voicemail off of unity to save?
Message-ID: <965C0255FD5828438E34F9A68ECA76E740A6D7@svcexch02.firm.fenwick.llp>

If you're also the Exchange admin, I think the easiest way is to go into
their mailboxes.  If not, then the Exchange admin can easily retrieve
those voicemails.  I had to do the same a while back, but I'm also the
Exchange admin, so that made things easier.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Wednesday, December 28, 2005 6:16 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Pulling voicemail off of unity to save?


There are a couple of voicemails in a users account that I am hoping to
retrieve so that it can be stored as well as given to someone for legal
reasons.  I cannot seem to find an easy way to retrieve this voicemail.
I know it is stored on an exchange message store of course but not sure
of the best way to access this.  Any recommendations?  I am sure I can
do it in a round about way, but thought I would ask to see if there was
an easy way. 
 
Regards.
--------------------------------------------------------

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From rratliff at cisco.com  Wed Dec 28 13:33:43 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed Dec 28 13:33:44 2005
Subject: [cisco-voip] Unity Service Packs
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A121DB@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A121DB@VISHNU.wesd.org>
Message-ID: <FED75853-6F0F-42F8-8048-CDF8CC282879@cisco.com>

Yet another link...

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_unity/ 
cmptblty/msupdate.htm

-Ryan

On Dec 28, 2005, at 1:01 PM, Voll, Scott wrote:

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2237/ 
products_device_support_table09186a0080531aab.html



The only thing I didn?t like is that Windows 2003 was not on the list  
yet it is approved.



Scott



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Wednesday, December 28, 2005 9:57 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Service Packs



Anybody have a good document or can quickly tell me the compatibility  
matrix between the various versions of Unity, Exchange and Windows  
2003? In particular I am interest in WIndows 2003 SP1, Exchange SP1  
and SP2, and Unity 4.0x and 4.1.

Thanks,

-- 
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol

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



From jgrace at digitelusa.net  Wed Dec 28 13:38:54 2005
From: jgrace at digitelusa.net (James Grace)
Date: Wed Dec 28 13:43:36 2005
Subject: [cisco-voip] config p2p on my voice gw
Message-ID: <200512281342468.SM01568@3729434770>

On my gw im trying to config the line code and framing    with ser- mod
XXXX

But they wont show up under the serial int.  Im trying a add the cmd under
the serial interface.  Because this is not a VWIC card.  Can someone tell me
what am I missing 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Joe_Bell at adp.com  Wed Dec 28 13:51:27 2005
From: Joe_Bell at adp.com (Bell, Joe)
Date: Wed Dec 28 13:51:37 2005
Subject: [cisco-voip] Unity Service Packs
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3B34C880@EXCHANGEDS01.ds.ad.adp.com>

Unity installed on Windows 2003 is only supported in Unified messaging
mode, not voice mail.  I know that's weird, but it's true.  Also, the
Unity server cannot be a DC, it must be only a member server to a
preexisting domain.  Make sure you don't push the domain security policy
to that server!

 

Joe 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, December 28, 2005 10:05 AM
To: Israel Lang; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity Service Packs

 

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2237/products_
device_support_table09186a0080531aab.html#wp60952

 

Sorry-it's under the recommended SP area.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, December 28, 2005 10:02 AM
To: Israel Lang; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity Service Packs

 

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps2237/products_
device_support_table09186a0080531aab.html

 

The only thing I didn't like is that Windows 2003 was not on the list
yet it is approved.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Wednesday, December 28, 2005 9:57 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Service Packs

 

Anybody have a good document or can quickly tell me the compatibility
matrix between the various versions of Unity, Exchange and Windows 2003?
In particular I am interest in WIndows 2003 SP1, Exchange SP1 and SP2,
and Unity 4.0x and 4.1.

Thanks,

-- 
Have a great day!!!!!!!!!!
zephy316_gmail.com

Replace the _ with an @ symbol 



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 Scott.Voll at wesd.org  Wed Dec 28 14:01:39 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed Dec 28 14:01:48 2005
Subject: [cisco-voip] config p2p on my voice gw
Message-ID: <D713462ED535184D830F6F6301753E7E01A121E8@VISHNU.wesd.org>

Interface configuration

 

Service-module t1 <framing> or <linecode> 

 

In newer IOS is doesn't show up in the config.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Wednesday, December 28, 2005 10:39 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] config p2p on my voice gw

 

On my gw im trying to config the line code and framing    with ser- mod
XXXX

But they wont show up under the serial int.  Im trying a add the cmd
under the serial interface.  Because this is not a VWIC card.  Can
someone tell me what am I missing 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From kthorngr at cisco.com  Wed Dec 28 14:17:22 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed Dec 28 14:17:26 2005
Subject: [cisco-voip] config p2p on my voice gw
In-Reply-To: <200512281342468.SM01568@3729434770>
References: <200512281342468.SM01568@3729434770>
Message-ID: <9eb0ad16aacdeb19b702e1a8a122c4bb@cisco.com>

Still not sure what type of card you have.  Try using "show inventory"  
to get the part number.  If you have WIC-1DSU-T1 then take a look at  
this web page and use the service-module command:
http://www.cisco.com/en/US/products/hw/modules/ps3129/ 
products_tech_note09186a00800b1501.shtml

If you have something like the WIC-2T or WIC-1T then you will need an  
external CSU/DSU and a V.35 cable.  You will then configure the CSU/DSU  
with the Line Coding and Framing.
http://www.cisco.com/en/US/products/hw/routers/ps274/ 
products_tech_note09186a00800b0858.shtml

HTH,
Kevin

On Dec 28, 2005, at 1:38 PM, James Grace wrote:

> On my gw im trying to config the line code and framing??? with ser-  
> mod? XXXX
> But they wont show up under the serial int.? Im trying a add the cmd  
> under the serial interface.? Because this is not a VWIC card.? Can  
> someone tell me what am I missing
> ?
> James D. Grace
> CCNP CCNA MCSE MCDBA
> Sr. System Engineer / Professional Svc.
> Digitel Corporation
> ?
> _______________________________________________
> 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  Wed Dec 28 17:00:14 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Dec 28 17:00:23 2005
Subject: [cisco-voip] config p2p on my voice gw
In-Reply-To: <9eb0ad16aacdeb19b702e1a8a122c4bb@cisco.com>
Message-ID: <20051228220014.40494.qmail@web34309.mail.mud.yahoo.com>

The commands won't show up in the config if the
default options are used which is probably the case if
you're in the US and are using all 24 channels. 

A 'show service-module' will tell you details on how
it is currently running and give framing, lining,
clocking, clock speed, errors if any, alarms if any,
etc. 

--- Kevin Thorngren <kthorngr at cisco.com> wrote:

> Still not sure what type of card you have.  Try
> using "show inventory"  
> to get the part number.  If you have WIC-1DSU-T1
> then take a look at  
> this web page and use the service-module command:
>
http://www.cisco.com/en/US/products/hw/modules/ps3129/
> 
> products_tech_note09186a00800b1501.shtml
> 
> If you have something like the WIC-2T or WIC-1T then
> you will need an  
> external CSU/DSU and a V.35 cable.  You will then
> configure the CSU/DSU  
> with the Line Coding and Framing.
>
http://www.cisco.com/en/US/products/hw/routers/ps274/
> 
> products_tech_note09186a00800b0858.shtml
> 
> HTH,
> Kevin
> 
> On Dec 28, 2005, at 1:38 PM, James Grace wrote:
> 
> > On my gw im trying to config the line code and
> framing????????? with ser-  
> > mod??? XXXX
> > But they wont show up under the serial int.??? Im
> trying a add the cmd  
> > under the serial interface.??? Because this is not
a
> VWIC card.??? Can  
> > someone tell me what am I missing
> > ???
> > James D. Grace
> > CCNP CCNA MCSE MCDBA
> > Sr. System Engineer / Professional Svc.
> > Digitel Corporation
> > ???
> > _______________________________________________
> > 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
> 



		
__________________________________________ 
Yahoo! DSL ? Something to write home about. 
Just $16.99/mo. or less. 
dsl.yahoo.com 


From jonvoip at gmail.com  Wed Dec 28 17:08:33 2005
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed Dec 28 17:08:36 2005
Subject: [cisco-voip] No DT on one extension with CME
Message-ID: <5d093f9a0512281408o8703a40ud102b0e515c126a4@mail.gmail.com>

The cool thing is that the problem follows the DN, if you put it on another
phone, the other phone exhibits the same bizarre behavior.

You can pick up the phone, the first DN is selected, you hear no dialtone.
The display on the phone is as if you were making a normal call. You can
dial and it says ring out, but the other phone will not ring. Other DNs on
the same phone (in different spots) work fine.

Here is the config of the DN:

!
!
ephone-dn  66
 number 7918
 label 7918
 description Chris Welzenbach Line 1
 name Chris Welzenbach
 call-forward busy 17918
 call-forward noan 17918 timeout 15
 hold-alert 30 originator
!
!

Here is the config of the phone:

!
ephone  31
 mac-address 0013.80DF.81BE
 paging-dn 80
 type 7960
 button  1:66 2:67 3:40 4s41 5s8 6:21
 pin 7918
!

I have set the router to reboot at 2AM and that might clear it up, but I
would rather give them a real solution to the problem


Jonathan
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From Anthony.Mendoza at ca.com  Wed Dec 28 19:10:19 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Wed Dec 28 19:10:23 2005
Subject: [cisco-voip] Unity Not Recording :(
Message-ID: <D1F2DC1FDE5F8C4385B42578077F19576080A6@uswams21.ca.com>

All,  

We're running into a weird issue with a new Unity install.  Here's the
basics:
 
Office A:
Unity 4.X
CCM 4.0 Cluster
Exchange Server
15 channel PRI Gateway on 2651 Router
 
Office B:
Unity 4.X
Exchange Server
 
Office A and B connected via VPN Tunnel <100msec latency
 
We're currently 100% functioning fine in Office A, however, when I flip
VM profiles and point people to Office B's Unity server I am unable to
record VM messages from the PSTN.  

* If I dial an extension of a phone located in Office A from an internal
IP phone, it records fine to the Unity Server in Office B.  The message
then gets dropped in the Exchange Inbox as you would expect in Office B.


* If I call that same extension from PSTN it blows and I get nothing.
In fact Unity stops me 3-4 secs into it since it hears silence.
 
We already have a TAC issue open, but wanted to find out if anyone here
had any ideas?

How can I make sure that when the PSTN call gets routed to Unity it
shows up from #VOICE LAN IP# vs. #DATA LAN IP# below?

Here's a sanitized version of our 2651 config for reference:
!
! Last configuration change at 23:12:31 UTC Wed Dec 28 2005
! NVRAM config last updated at 23:12:47 UTC Wed Dec 28 2005
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0
voice-card 0
 dspfarm
 no local-bypass
!
isdn switch-type primary-net5
!
!
voice rtp send-recv

!
ccm-manager redundant-host #PUBLISHER IP#
ccm-manager mgcp
ccm-manager music-on-hold
!
!
controller E1 0/0
 pri-group timeslots 1-16 service mgcp
!
class-map match-all control
 match access-group 102
class-map match-all voice
 match access-group 101
!
!
policy-map QoS
 class voice
  set cos 5
 class control
  set cos 3
!
!
!
!
interface FastEthernet0/0
 description uksagr00-f0/0/0
 ip address #USED WITH VPN# 255.255.255.252
 speed 100
 full-duplex
!
interface Serial0/0:15
 no ip address
 no logging event link-status
 isdn switch-type primary-net5
 isdn incoming-voice voice
 isdn bind-l3 ccm-manager
 no cdp enable
!
interface FastEthernet0/1
 description TRUNK TO 3548 LAN
 no ip address
 duplex auto
 speed auto
!
interface FastEthernet0/1.2
 encapsulation dot1Q 2 native
 ip address #DATA LAN IP# 255.255.254.0
 ip access-group UKSLUNITY1 out
!
interface FastEthernet0/1.100
 encapsulation dot1Q 100
 ip address #VOICE LAN IP# 255.255.254.0
 ip policy route-map VOIP_MONITOR
!
ip classless
ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN GW#
ip http server
!
ip access-list extended VOIP_MONITOR
 permit ip #VOICE LAN# 0.0.1.255 #OFFICE C LAN# 0.0.0.255
 permit ip #VOICE LAN# 0.0.1.255 #OFFICE B LAN# 0.0.0.63
ip access-list extended UKSLUNITY1
 permit ip any host #UNITY SERVER IN OFC B# log
 permit ip any any
!
route-map VOIP_MONITOR permit 10
 match ip address VOIP_MONITOR
 set ip next-hop #TO VPN GW#
!
control-plane
!
!
!
voice-port 0/0:15
!
mgcp
mgcp call-agent #subscriber IP# service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp package-capability rtp-package
no mgcp package-capability res-package
mgcp package-capability sst-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
!
mgcp profile default
!

dial-peer voice 9 voip
 application mgcpapp
 destination-pattern 9T
!
----
All ideas and opinions expressed are mine and only mine not the company
I work for or that you see based on my email address.  If I say anything
rude or offensive blame me not my employer.


From erickbe at yahoo.com  Wed Dec 28 20:40:49 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed Dec 28 20:40:57 2005
Subject: [cisco-voip] Unity Not Recording :(
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F19576080A6@uswams21.ca.com>
Message-ID: <20051229014049.69414.qmail@web34304.mail.mud.yahoo.com>

Sounds like a routing issue of some sort, or port
blockage perhaps. When you dial in from PSTN and call
hits Unity in office B can you hear the greetings, etc
fine or is it just not recording or voice?

Any codec (region) differences between the 2 offices?
Is VPN doing any firewall'ish/acl'ish stuff that may
be effecting traffic? The second unity server have
correct router(s) being used to get back to PSTN
gateway?

When the call is connected, on gateway 'show call
active voice brief' should show the PSTN and IP leg of
call with IP address of remote host. Does this look
right? Try pinging the remote IP address from gateways
interface MGCP is sourced from. No reachability
issues? It may be useful also to watch the call in the
unity port status monitor and call viewer. 

With MGCP, try changing MGCP interface with bind
statement. I had a similar strange issue once and
bouncing MGCP didn't fix it but flip-flopping the MGCP
source interface (mgcp bind...) between LAN and WAN
interface and back again cleared it up (no mgcp/mgcp
didn't fix).

--- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:

> All,  
> 
> We're running into a weird issue with a new Unity
> install.  Here's the
> basics:
>  
> Office A:
> Unity 4.X
> CCM 4.0 Cluster
> Exchange Server
> 15 channel PRI Gateway on 2651 Router
>  
> Office B:
> Unity 4.X
> Exchange Server
>  
> Office A and B connected via VPN Tunnel <100msec
> latency
>  
> We're currently 100% functioning fine in Office A,
> however, when I flip
> VM profiles and point people to Office B's Unity
> server I am unable to
> record VM messages from the PSTN.  
> 
> * If I dial an extension of a phone located in
> Office A from an internal
> IP phone, it records fine to the Unity Server in
> Office B.  The message
> then gets dropped in the Exchange Inbox as you would
> expect in Office B.
> 
> 
> * If I call that same extension from PSTN it blows
> and I get nothing.
> In fact Unity stops me 3-4 secs into it since it
> hears silence.
>  
> We already have a TAC issue open, but wanted to find
> out if anyone here
> had any ideas?
> 
> How can I make sure that when the PSTN call gets
> routed to Unity it
> shows up from #VOICE LAN IP# vs. #DATA LAN IP#
> below?
> 
> Here's a sanitized version of our 2651 config for
> reference:
> !
> ! Last configuration change at 23:12:31 UTC Wed Dec
> 28 2005
> ! NVRAM config last updated at 23:12:47 UTC Wed Dec
> 28 2005
> !
> network-clock-participate wic 0
> network-clock-select 1 E1 0/0
> voice-card 0
>  dspfarm
>  no local-bypass
> !
> isdn switch-type primary-net5
> !
> !
> voice rtp send-recv
> 
> !
> ccm-manager redundant-host #PUBLISHER IP#
> ccm-manager mgcp
> ccm-manager music-on-hold
> !
> !
> controller E1 0/0
>  pri-group timeslots 1-16 service mgcp
> !
> class-map match-all control
>  match access-group 102
> class-map match-all voice
>  match access-group 101
> !
> !
> policy-map QoS
>  class voice
>   set cos 5
>  class control
>   set cos 3
> !
> !
> !
> !
> interface FastEthernet0/0
>  description uksagr00-f0/0/0
>  ip address #USED WITH VPN# 255.255.255.252
>  speed 100
>  full-duplex
> !
> interface Serial0/0:15
>  no ip address
>  no logging event link-status
>  isdn switch-type primary-net5
>  isdn incoming-voice voice
>  isdn bind-l3 ccm-manager
>  no cdp enable
> !
> interface FastEthernet0/1
>  description TRUNK TO 3548 LAN
>  no ip address
>  duplex auto
>  speed auto
> !
> interface FastEthernet0/1.2
>  encapsulation dot1Q 2 native
>  ip address #DATA LAN IP# 255.255.254.0
>  ip access-group UKSLUNITY1 out
> !
> interface FastEthernet0/1.100
>  encapsulation dot1Q 100
>  ip address #VOICE LAN IP# 255.255.254.0
>  ip policy route-map VOIP_MONITOR
> !
> ip classless
> ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN GW#
> ip http server
> !
> ip access-list extended VOIP_MONITOR
>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C LAN#
> 0.0.0.255
>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B LAN#
> 0.0.0.63
> ip access-list extended UKSLUNITY1
>  permit ip any host #UNITY SERVER IN OFC B# log
>  permit ip any any
> !
> route-map VOIP_MONITOR permit 10
>  match ip address VOIP_MONITOR
>  set ip next-hop #TO VPN GW#
> !
> control-plane
> !
> !
> !
> voice-port 0/0:15
> !
> mgcp
> mgcp call-agent #subscriber IP# service-type mgcp
> version 0.1
> mgcp dtmf-relay voip codec all mode out-of-band
> mgcp rtp unreachable timeout 1000 action notify
> mgcp package-capability rtp-package
> no mgcp package-capability res-package
> mgcp package-capability sst-package
> no mgcp package-capability fxr-package
> no mgcp timer receive-rtcp
> mgcp sdp simple
> mgcp fax t38 inhibit
> mgcp rtp payload-type g726r16 static
> !
> mgcp profile default
> !
> 
> dial-peer voice 9 voip
>  application mgcpapp
>  destination-pattern 9T
> !
> ----
> All ideas and opinions expressed are mine and only
> mine not the company
> I work for or that you see based on my email
> address.  If I say anything
> rude or offensive blame me not my employer.



		
__________________________________________ 
Yahoo! DSL ? Something to write home about. 
Just $16.99/mo. or less. 
dsl.yahoo.com 


From jburton at NETechCorp.com  Wed Dec 28 22:32:45 2005
From: jburton at NETechCorp.com (Jason Burton)
Date: Wed Dec 28 22:36:55 2005
Subject: [cisco-voip] No DT on one extension with CME
References: <5d093f9a0512281408o8703a40ud102b0e515c126a4@mail.gmail.com>
Message-ID: <FDC137145729AD438CDCA0F9FC6F723633958C@mail.netechcorp.com>

I had something similiar to the problem you are having.  What it ended up being was a call from that DN was stuck in Limbo connected to VM (CUE).  I was able to do a 'sh call voice active summary' to see the call and get the call id and then do a clear call voice.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jonathan Charles
Sent: Wed 12/28/2005 5:08 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No DT on one extension with CME


The cool thing is that the problem follows the DN, if you put it on another phone, the other phone exhibits the same bizarre behavior.

You can pick up the phone, the first DN is selected, you hear no dialtone. The display on the phone is as if you were making a normal call. You can dial and it says ring out, but the other phone will not ring. Other DNs on the same phone (in different spots) work fine.

Here is the config of the DN:

!
!
ephone-dn  66
 number 7918
 label 7918
 description Chris Welzenbach Line 1
 name Chris Welzenbach
 call-forward busy 17918
 call-forward noan 17918 timeout 15
 hold-alert 30 originator
!
!

Here is the config of the phone:

!
ephone  31
 mac-address 0013.80DF.81BE
 paging-dn 80
 type 7960
 button  1:66 2:67 3:40 4s41 5s8 6:21
 pin 7918
!

I have set the router to reboot at 2AM and that might clear it up, but I would rather give them a real solution to the problem


Jonathan



From it at cimgroup.com  Thu Dec 29 01:11:49 2005
From: it at cimgroup.com (IT)
Date: Thu Dec 29 01:07:53 2005
Subject: [cisco-voip] DeviceTransientConnection Error
Message-ID: <B487D4C1FE6D04439061018C6EBC109B02708DE4@exchange.cimgroup.com>

I just went through the event viewer on my CallManager 4.1 machine, not
because anythings wrong, but because its 10pm and im still at work.

And I found THOUSANDS of messages.  Here is the background,
cimgroup801grand is a 2801 connected with a p2p t1, and all the devices
connected to it work great. I have about 6x 7960s and they are all fine.
Its setup as a gateway, since it has a VIC2-2FXO in it for 911 calling.
The lines haven't been installed, so 911 calling is not turned on yet.
Is that why I am getting this error message every 30 seconds? Or is it
something else?

 

Event Type:       Error

Event Source:    Cisco CallManager

Event Category: None

Event ID:           3

Date:                12/28/2005

Time:                10:15:15 PM

User:                N/A

Computer:         CIMGROUPCCM01

Description:

Error: DeviceTransientConnection - Transient connection attempt.

  Connecting Port: 2427

  Device name [Optional].: *@cimgroup801grand.cimgroup.com

  Device IP address.: 10.60.1.129

  Device type. [Optional]: 254

  Reason Code [Optional].: 2

  App ID: Cisco CallManager

  Cluster ID: StandAloneCluster

  Node ID: 10.96.3.11

Explanation: A connection was established and immediately dropped before
completing registration. Incomplete registration may indicate a device
is rehoming in the middle of registration. The alarm could also indicate
a device misconfiguration, database error, or an illegal/unknown device
trying to attempt a connection.

Recommended Action: No action is required if this event was issued as a
result of a normal device rehome..

 


******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
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From erickbe at yahoo.com  Thu Dec 29 01:33:08 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Dec 29 01:33:17 2005
Subject: [cisco-voip] DeviceTransientConnection Error
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B02708DE4@exchange.cimgroup.com>
Message-ID: <20051229063308.28950.qmail@web34302.mail.mud.yahoo.com>

The reason code of 2 means no entry in database (port
not configured yet in ccm). Is this particular port
(endpoint) configured in call manager yet or
unconfigured? 

Usually from my experiences, if you see transient
connection on one server another server in the cluster
will have a device unregistered event log error with
more details as to port, etc trying to register. CCM
detailed traces will also have this info.

I'm betting that since you're still working on getting
the 911 lines installed these ports may not be
configured yet on the CCM gateway page and thus the
router is constantly trying to register them to CCM.
You can shutdown the voice ports on the router - or
simply configure the ports on the call manager side
but not assign a DN, etc to them. This way they show
registered on CCM side and later you can come back and
finish configuring them with specifics. This will also
save on resources because the router/gateway is no
longer constantly trying to register and the server
doesn't have to constantly try to process the
registration request and put entries in the event log.
It will also get rid of that error in event log ...
which will make troubleshooting real issues easier
since you won't have a bunch of noise in the logs to
sift through. 

--- IT <it at cimgroup.com> wrote:

> I just went through the event viewer on my
> CallManager 4.1 machine, not
> because anythings wrong, but because its 10pm and im
> still at work.
> 
> And I found THOUSANDS of messages.  Here is the
> background,
> cimgroup801grand is a 2801 connected with a p2p t1,
> and all the devices
> connected to it work great. I have about 6x 7960s
> and they are all fine.
> Its setup as a gateway, since it has a VIC2-2FXO in
> it for 911 calling.
> The lines haven't been installed, so 911 calling is
> not turned on yet.
> Is that why I am getting this error message every 30
> seconds? Or is it
> something else?
> 
>  
> 
> Event Type:       Error
> 
> Event Source:    Cisco CallManager
> 
> Event Category: None
> 
> Event ID:           3
> 
> Date:                12/28/2005
> 
> Time:                10:15:15 PM
> 
> User:                N/A
> 
> Computer:         CIMGROUPCCM01
> 
> Description:
> 
> Error: DeviceTransientConnection - Transient
> connection attempt.
> 
>   Connecting Port: 2427
> 
>   Device name [Optional].:
> *@cimgroup801grand.cimgroup.com
> 
>   Device IP address.: 10.60.1.129
> 
>   Device type. [Optional]: 254
> 
>   Reason Code [Optional].: 2
> 
>   App ID: Cisco CallManager
> 
>   Cluster ID: StandAloneCluster
> 
>   Node ID: 10.96.3.11
> 
> Explanation: A connection was established and
> immediately dropped before
> completing registration. Incomplete registration may
> indicate a device
> is rehoming in the middle of registration. The alarm
> could also indicate
> a device misconfiguration, database error, or an
> illegal/unknown device
> trying to attempt a connection.
> 
> Recommended Action: No action is required if this
> event was issued as a
> result of a normal device rehome..
> 
>  
> 
> 
>
******************************************************
> This email may contain trade secrets and/or other
> information that is 
> proprietary and/or confidential. Use of this by
> anyone other than the 
> intended recipient(s) is prohibited and may be
> unlawful. If you are not 
> the intended recipient, delete the e-mail and notify
> us immediately.
>
******************************************************
> 
> 
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



	
		
__________________________________ 
Yahoo! for Good - Make a difference this year. 
http://brand.yahoo.com/cybergivingweek2005/

From gary.sullivan at waters.nestle.com  Thu Dec 29 07:09:21 2005
From: gary.sullivan at waters.nestle.com (Sullivan, Gary, GREENWICH,
	Information Services)
Date: Thu Dec 29 07:27:02 2005
Subject: [cisco-voip] Upgrade to 4.1(3) SR2?
Message-ID: <E9B4C49E7DD9DA459F42493E5485677B038F476C@usgcte00.nestle.com>

We upgraded to this 2 weeks ago. the only issue I have is a PRI  to a Nortel M1 PBX. 
The PRI is up D-channel is established but can't make or receive calls.
I have a Cisco TAC case opened on it now.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net]On Behalf Of Ruttman, Peter G.
Sent: Wednesday, December 28, 2005 12:12 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Upgrade to 4.1(3) SR2?



We have a few issues that are addressed in SR2.  Has anyone on the listed loaded SR2 in a production environment yet?  Does anyone have any reason to tell us to wait?

Pete 


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. 


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From gary.sullivan at waters.nestle.com  Thu Dec 29 08:13:27 2005
From: gary.sullivan at waters.nestle.com (Sullivan, Gary, GREENWICH,
	Information Services)
Date: Thu Dec 29 08:15:48 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
Message-ID: <E9B4C49E7DD9DA459F42493E5485677B038F476D@usgcte00.nestle.com>

We have a PRI between are CCM and Nortel M1 pbx. Was working fine until we upgraded.
We upgraded are CCM from 3.3.4 to 4.1.3 . Now we can't make any calls. the D-channel is established.
but when I try to make a call from my IP phone it disconnects right away. when I call from Nortel to IP
it rings once then disconnects. I did a debug isdn q931 and see the Messages.
Any help is appreciated


ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP pd = 8  callref = 0x0001 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA98381 
                Exclusive, Channel 1 
        Display i = 0xB1, 'Gary Sullivan' 
        Calling Party Number i = 0x0081, '3102' 
                Plan:Unknown, Type:Unknown 
        Called Party Number i = 0x80, '0400' 
                Plan:Unknown, Type:Unknown

81 
        Cause i = 0x81B6 - Incoming calls are barred
GCTRT03#
GCTRT03#
GCTRT03#
GCTRT03#
GCTRT03#
Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8  callref = 0x17 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xE9808317 
                Exclusive, Interface 0, Channel 23 
        Display i = 0xB1, 'GARY SULLIVAN' 
        Calling Party Number i = 0x0980, '0543' 
                Plan:Private, Type:Unknown 
        Called Party Number i = 0x89, '3102' 
                Plan:Private, Type:Unknown


        Channel ID i = 0xA98397 
                Exclusive, Channel 23
Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8  callref = 0x97
Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX -> NOTIFY pd = 8  callref = 0x97 
        Connected Address i = 0x80, '3102' 
        Notification Ind i = 0xF1 
        Display i = 0xB2, 'Gary Sullivan'
Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x17 
        Cause i = 0x81E4 - Invalid information element contents
Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x17 
        Cause i = 0x81E0 - Mandatory information element missing


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From rratliff at cisco.com  Thu Dec 29 08:32:31 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu Dec 29 08:32:30 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
In-Reply-To: <E9B4C49E7DD9DA459F42493E5485677B038F476D@usgcte00.nestle.com>
References: <E9B4C49E7DD9DA459F42493E5485677B038F476D@usgcte00.nestle.com>
Message-ID: <B1DD47E1-DB39-4431-80EB-583BE662B878@cisco.com>

The first call appears to be rejected by your PBX but I think the  
debug is missing info.  The "Cause i = 0x81B6 - Incoming calls are  
barred" should probably appear in a Disconnect or Release message  
from the PBX.

The second one is also rejected by the PBX. I would guess because of  
the Display IE but the second Release message is a bit confusing.

Is this H.323 or MGCP?

-Ryan

On Dec 29, 2005, at 8:13 AM, Sullivan, Gary, GREENWICH, Information  
Services wrote:

We have a PRI between are CCM and Nortel M1 pbx. Was working fine  
until we upgraded.
We upgraded are CCM from 3.3.4 to 4.1.3 . Now we can't make any  
calls. the D-channel is established.
but when I try to make a call from my IP phone it disconnects right  
away. when I call from Nortel to IP
it rings once then disconnects. I did a debug isdn q931 and see the  
Messages.
Any help is appreciated


ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP pd = 8  callref =  
0x0001
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Display i = 0xB1, 'Gary Sullivan'
         Calling Party Number i = 0x0081, '3102'
                 Plan:Unknown, Type:Unknown
         Called Party Number i = 0x80, '0400'
                 Plan:Unknown, Type:Unknown

81
         Cause i = 0x81B6 - Incoming calls are barred
GCTRT03#
GCTRT03#
GCTRT03#
GCTRT03#
GCTRT03#
Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8  callref  
= 0x17
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xE9808317
                 Exclusive, Interface 0, Channel 23
         Display i = 0xB1, 'GARY SULLIVAN'
         Calling Party Number i = 0x0980, '0543'
                 Plan:Private, Type:Unknown
         Called Party Number i = 0x89, '3102'
                 Plan:Private, Type:Unknown


         Channel ID i = 0xA98397
                 Exclusive, Channel 23
Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8   
callref = 0x97
Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX -> NOTIFY pd = 8  callref  
= 0x97
         Connected Address i = 0x80, '3102'
         Notification Ind i = 0xF1
         Display i = 0xB2, 'Gary Sullivan'
Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8   
callref = 0x17
         Cause i = 0x81E4 - Invalid information element contents
Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8   
callref = 0x17
         Cause i = 0x81E0 - Mandatory information element missing


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


From kthorngr at cisco.com  Thu Dec 29 08:39:12 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Dec 29 08:39:16 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
In-Reply-To: <E9B4C49E7DD9DA459F42493E5485677B038F476D@usgcte00.nestle.com>
References: <E9B4C49E7DD9DA459F42493E5485677B038F476D@usgcte00.nestle.com>
Message-ID: <80b35633a933ed80060fe3ef9931f427@cisco.com>

Are you using DMS-100 Protocol?  If so, I believe you need to place a 
check mark in the "MCDN Channel Number Extension Bit Set to Zero" 
option in the CCM GW configuration.

Let us know if this helps.

Kevin
On Dec 29, 2005, at 8:13 AM, Sullivan, Gary, GREENWICH, Information 
Services wrote:

> We have a PRI between are CCM and Nortel M1 pbx. Was working fine 
> until we upgraded.
> We upgraded are CCM from 3.3.4 to 4.1.3 . Now we can't make any calls. 
> the D-channel is established.
> but when I try to make a call from my IP phone it disconnects right 
> away. when I call from Nortel to IP
> it rings once then disconnects. I did a debug isdn q931 and see the 
> Messages.
> Any help is appreciated
>
>
> ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP pd = 8? callref = 
> 0x0001
> ??????? Bearer Capability i = 0x8090A2
> ??????????????? Standard = CCITT
> ??????????????? Transer Capability = Speech?
> ??????????????? Transfer Mode = Circuit
> ??????????????? Transfer Rate = 64 kbit/s
> ??????? Channel ID i = 0xA98381
> ??????????????? Exclusive, Channel 1
> ??????? Display i = 0xB1, 'Gary Sullivan'
> ??????? Calling Party Number i = 0x0081, '3102'
> ??????????????? Plan:Unknown, Type:Unknown
> ??????? Called Party Number i = 0x80, '0400'
> ??????????????? Plan:Unknown, Type:Unknown
>
> 81
> ??????? Cause i = 0x81B6 - Incoming calls are barred
> GCTRT03#
> GCTRT03#
> GCTRT03#
> GCTRT03#
> GCTRT03#
> Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8? callref = 
> 0x17
> ??????? Bearer Capability i = 0x8090A2
> ??????????????? Standard = CCITT
> ??????????????? Transer Capability = Speech?
> ??????????????? Transfer Mode = Circuit
> ??????????????? Transfer Rate = 64 kbit/s
> ??????? Channel ID i = 0xE9808317
> ??????????????? Exclusive, Interface 0, Channel 23
> ??????? Display i = 0xB1, 'GARY SULLIVAN'
> ??????? Calling Party Number i = 0x0980, '0543'
> ??????????????? Plan:Private, Type:Unknown
> ??????? Called Party Number i = 0x89, '3102'
> ??????????????? Plan:Private, Type:Unknown
>
>
> ??????? Channel ID i = 0xA98397
> ??????????????? Exclusive, Channel 23
> Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8? 
> callref = 0x97
> Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX -> NOTIFY pd = 8? callref 
> = 0x97
> ??????? Connected Address i = 0x80, '3102'
> ??????? Notification Ind i = 0xF1
> ??????? Display i = 0xB2, 'Gary Sullivan'
> Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8? callref 
> = 0x17
> ??????? Cause i = 0x81E4 - Invalid information element contents
> Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8? callref 
> = 0x17
> ??????? Cause i = 0x81E0 - Mandatory information element missing
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From paland at gmail.com  Thu Dec 29 08:40:35 2005
From: paland at gmail.com (Patrick Aland)
Date: Thu Dec 29 08:40:37 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
In-Reply-To: <B1DD47E1-DB39-4431-80EB-583BE662B878@cisco.com>
References: <E9B4C49E7DD9DA459F42493E5485677B038F476D@usgcte00.nestle.com>
	<B1DD47E1-DB39-4431-80EB-583BE662B878@cisco.com>
Message-ID: <960c6a4d0512290540q1c7150c8v82cd9e4d178c74e6@mail.gmail.com>

try adding
 isdn channel-id invert extend-bit

Under your serial interface, i.e.

int ser0/0/0:23
 isdn channel-id invert extend-bit

I had the same problem with a Nortel 61C and the above fixed it.

On 12/29/05, Ryan Ratliff <rratliff at cisco.com> wrote:
>
> The first call appears to be rejected by your PBX but I think the
> debug is missing info.  The "Cause i = 0x81B6 - Incoming calls are
> barred" should probably appear in a Disconnect or Release message
> from the PBX.
>
> The second one is also rejected by the PBX. I would guess because of
> the Display IE but the second Release message is a bit confusing.
>
> Is this H.323 or MGCP?
>
> -Ryan
>
> On Dec 29, 2005, at 8:13 AM, Sullivan, Gary, GREENWICH, Information
> Services wrote:
>
> We have a PRI between are CCM and Nortel M1 pbx. Was working fine
> until we upgraded.
> We upgraded are CCM from 3.3.4 to 4.1.3 . Now we can't make any
> calls. the D-channel is established.
> but when I try to make a call from my IP phone it disconnects right
> away. when I call from Nortel to IP
> it rings once then disconnects. I did a debug isdn q931 and see the
> Messages.
> Any help is appreciated
>
>
> ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP pd = 8  callref =
> 0x0001
>          Bearer Capability i = 0x8090A2
>                  Standard = CCITT
>                  Transer Capability = Speech
>                  Transfer Mode = Circuit
>                  Transfer Rate = 64 kbit/s
>          Channel ID i = 0xA98381
>                  Exclusive, Channel 1
>          Display i = 0xB1, 'Gary Sullivan'
>          Calling Party Number i = 0x0081, '3102'
>                  Plan:Unknown, Type:Unknown
>          Called Party Number i = 0x80, '0400'
>                  Plan:Unknown, Type:Unknown
>
> 81
>          Cause i = 0x81B6 - Incoming calls are barred
> GCTRT03#
> GCTRT03#
> GCTRT03#
> GCTRT03#
> GCTRT03#
> Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8  callref
> = 0x17
>          Bearer Capability i = 0x8090A2
>                  Standard = CCITT
>                  Transer Capability = Speech
>                  Transfer Mode = Circuit
>                  Transfer Rate = 64 kbit/s
>          Channel ID i = 0xE9808317
>                  Exclusive, Interface 0, Channel 23
>          Display i = 0xB1, 'GARY SULLIVAN'
>          Calling Party Number i = 0x0980, '0543'
>                  Plan:Private, Type:Unknown
>          Called Party Number i = 0x89, '3102'
>                  Plan:Private, Type:Unknown
>
>
>          Channel ID i = 0xA98397
>                  Exclusive, Channel 23
> Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8
> callref = 0x97
> Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX -> NOTIFY pd = 8  callref
> = 0x97
>          Connected Address i = 0x80, '3102'
>          Notification Ind i = 0xF1
>          Display i = 0xB2, 'Gary Sullivan'
> Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8
> callref = 0x17
>          Cause i = 0x81E4 - Invalid information element contents
> Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8
> callref = 0x17
>          Cause i = 0x81E0 - Mandatory information element missing
>
>
> _______________________________________________
> 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
>



--
--Patrick
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From mfa at crec.ifas.ufl.edu  Thu Dec 29 08:48:25 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Thu Dec 29 08:48:33 2005
Subject: [cisco-voip] Upgrade to 4.1(3) SR2?
References: <200512281811.jBSIBY9J022951@puck.nether.net>
Message-ID: <012801c60c7e$8b8af470$0402a8c0@crec.ifas.ufl.edu>

We upgraded 2 weeks ago; no problems seen, but we're a small, basic 
environment (Pub/Sub, PRI to Verizon, no WAN, 250 stations, Unity/Exchange).

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

> Date: Wed, 28 Dec 2005 11:11:34 -0600
> From: "Ruttman, Peter G." <PRuttman at foley.com>
> Subject: [cisco-voip] Upgrade to 4.1(3) SR2?
> To: <cisco-voip at puck.nether.net>
> Message-ID:
> <7E3064B7EFC16C41B185C41D141008B7CC35B7 at foleyex97.foleylaw.com>
> Content-Type: text/plain; charset="us-ascii"
>
> We have a few issues that are addressed in SR2.  Has anyone on the
> listed loaded SR2 in a production environment yet?  Does anyone have any
> reason to tell us to wait?
>
> Pete
>


From gary.sullivan at waters.nestle.com  Thu Dec 29 08:54:51 2005
From: gary.sullivan at waters.nestle.com (Sullivan, Gary, GREENWICH,
	Information Services)
Date: Thu Dec 29 08:57:22 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
Message-ID: <E9B4C49E7DD9DA459F42493E5485677B038F476F@usgcte00.nestle.com>

MGCP

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Thursday, December 29, 2005 8:33 AM
To: Sullivan,Gary,GREENWICH,Information Services
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PRI between CCM (3725) and Nortel M1


The first call appears to be rejected by your PBX but I think the  
debug is missing info.  The "Cause i = 0x81B6 - Incoming calls are  
barred" should probably appear in a Disconnect or Release message  
from the PBX.

The second one is also rejected by the PBX. I would guess because of  
the Display IE but the second Release message is a bit confusing.

Is this H.323 or MGCP?

-Ryan

On Dec 29, 2005, at 8:13 AM, Sullivan, Gary, GREENWICH, Information  
Services wrote:

We have a PRI between are CCM and Nortel M1 pbx. Was working fine  
until we upgraded.
We upgraded are CCM from 3.3.4 to 4.1.3 . Now we can't make any  
calls. the D-channel is established.
but when I try to make a call from my IP phone it disconnects right  
away. when I call from Nortel to IP
it rings once then disconnects. I did a debug isdn q931 and see the  
Messages.
Any help is appreciated


ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP pd = 8  callref =  
0x0001
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Display i = 0xB1, 'Gary Sullivan'
         Calling Party Number i = 0x0081, '3102'
                 Plan:Unknown, Type:Unknown
         Called Party Number i = 0x80, '0400'
                 Plan:Unknown, Type:Unknown

81
         Cause i = 0x81B6 - Incoming calls are barred
GCTRT03#
GCTRT03#
GCTRT03#
GCTRT03#
GCTRT03#
Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8  callref  
= 0x17
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xE9808317
                 Exclusive, Interface 0, Channel 23
         Display i = 0xB1, 'GARY SULLIVAN'
         Calling Party Number i = 0x0980, '0543'
                 Plan:Private, Type:Unknown
         Called Party Number i = 0x89, '3102'
                 Plan:Private, Type:Unknown


         Channel ID i = 0xA98397
                 Exclusive, Channel 23
Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8   
callref = 0x97
Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX -> NOTIFY pd = 8  callref  
= 0x97
         Connected Address i = 0x80, '3102'
         Notification Ind i = 0xF1
         Display i = 0xB2, 'Gary Sullivan'
Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8   
callref = 0x17
         Cause i = 0x81E4 - Invalid information element contents
Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8   
callref = 0x17
         Cause i = 0x81E0 - Mandatory information element missing


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



From gary.sullivan at waters.nestle.com  Thu Dec 29 09:19:10 2005
From: gary.sullivan at waters.nestle.com (Sullivan, Gary, GREENWICH,
	Information Services)
Date: Thu Dec 29 09:21:33 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
Message-ID: <E9B4C49E7DD9DA459F42493E5485677B038F4770@usgcte00.nestle.com>

When I try to add that I get 
    GCTRT03(config-if)#isdn channel-id invert extend-bit
                                         ^
% Invalid input detected at '^' marker.

The router has is run image c3725-is-mz.122-15.T2.bin

-----Original Message-----
From: Patrick Aland [mailto:paland at gmail.com]
Sent: Thursday, December 29, 2005 8:41 AM
To: Ryan Ratliff
Cc: Sullivan,Gary,GREENWICH,Information Services; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PRI between CCM (3725) and Nortel M1


try adding 
 isdn channel-id invert extend-bit

Under your serial interface, i.e.

int ser0/0/0:23
 isdn channel-id invert extend-bit

I had the same problem with a Nortel 61C and the above fixed it. 


On 12/29/05, Ryan Ratliff < rratliff at cisco.com> wrote: 

The first call appears to be rejected by your PBX but I think the
debug is missing info.  The "Cause i = 0x81B6 - Incoming calls are
barred" should probably appear in a Disconnect or Release message
from the PBX. 

The second one is also rejected by the PBX. I would guess because of
the Display IE but the second Release message is a bit confusing.

Is this H.323 or MGCP?

-Ryan

On Dec 29, 2005, at 8:13 AM, Sullivan, Gary, GREENWICH, Information 
Services wrote:

We have a PRI between are CCM and Nortel M1 pbx. Was working fine
until we upgraded.
We upgraded are CCM from 3.3.4 to 4.1.3 . Now we can't make any
calls. the D-channel is established. 
but when I try to make a call from my IP phone it disconnects right
away. when I call from Nortel to IP
it rings once then disconnects. I did a debug isdn q931 and see the
Messages.
Any help is appreciated 


ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP pd = 8  callref =
0x0001
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transer Capability = Speech 
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Display i = 0xB1, 'Gary Sullivan'
         Calling Party Number i = 0x0081, '3102' 
                 Plan:Unknown, Type:Unknown
         Called Party Number i = 0x80, '0400'
                 Plan:Unknown, Type:Unknown

81
         Cause i = 0x81B6 - Incoming calls are barred
GCTRT03# 
GCTRT03#
GCTRT03#
GCTRT03#
GCTRT03#
Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8  callref
= 0x17
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transer Capability = Speech 
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xE9808317
                 Exclusive, Interface 0, Channel 23
         Display i = 0xB1, 'GARY SULLIVAN' 
         Calling Party Number i = 0x0980, '0543'
                 Plan:Private, Type:Unknown
         Called Party Number i = 0x89, '3102'
                 Plan:Private, Type:Unknown


         Channel ID i = 0xA98397 
                 Exclusive, Channel 23
Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8
callref = 0x97
Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX -> NOTIFY pd = 8  callref
= 0x97
         Connected Address i = 0x80, '3102' 
         Notification Ind i = 0xF1
         Display i = 0xB2, 'Gary Sullivan'
Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8
callref = 0x17
         Cause i = 0x81E4 - Invalid information element contents 
Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8
callref = 0x17
         Cause i = 0x81E0 - Mandatory information element missing


_______________________________________________
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  <https://puck.nether.net/mailman/listinfo/cisco-voip> 





-- 
--Patrick 

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From gary.sullivan at waters.nestle.com  Thu Dec 29 09:32:13 2005
From: gary.sullivan at waters.nestle.com (Sullivan, Gary, GREENWICH,
	Information Services)
Date: Thu Dec 29 09:32:55 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
Message-ID: <E9B4C49E7DD9DA459F42493E5485677B038F4771@usgcte00.nestle.com>

Yes using DMS-100 and have that check all ready.

-----Original Message-----
From: Kevin Thorngren [mailto:kthorngr at cisco.com]
Sent: Thursday, December 29, 2005 8:39 AM
To: Sullivan,Gary,GREENWICH,Information Services
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PRI between CCM (3725) and Nortel M1


Are you using DMS-100 Protocol? If so, I believe you need to place a check mark in the "MCDN Channel Number Extension Bit Set to Zero" option in the CCM GW configuration. 

Let us know if this helps. 

Kevin 
On Dec 29, 2005, at 8:13 AM, Sullivan, Gary, GREENWICH, Information Services wrote: 


We have a PRI between are CCM and Nortel M1 pbx. Was working fine until we upgraded. 
We upgraded are CCM from 3.3.4 to 4.1.3 . Now we can't make any calls. the D-channel is established. 
but when I try to make a call from my IP phone it disconnects right away. when I call from Nortel to IP 
it rings once then disconnects. I did a debug isdn q931 and see the Messages. 
Any help is appreciated 


ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP pd = 8  callref = 0x0001 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA98381 
                Exclusive, Channel 1 
        Display i = 0xB1, 'Gary Sullivan' 
        Calling Party Number i = 0x0081, '3102' 
                Plan:Unknown, Type:Unknown 
        Called Party Number i = 0x80, '0400' 
                Plan:Unknown, Type:Unknown 

81 
        Cause i = 0x81B6 - Incoming calls are barred 
GCTRT03# 
GCTRT03# 
GCTRT03# 
GCTRT03# 
GCTRT03# 
Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8  callref = 0x17 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xE9808317 
                Exclusive, Interface 0, Channel 23 
        Display i = 0xB1, 'GARY SULLIVAN' 
        Calling Party Number i = 0x0980, '0543' 
                Plan:Private, Type:Unknown 
        Called Party Number i = 0x89, '3102' 
                Plan:Private, Type:Unknown 


        Channel ID i = 0xA98397 
                Exclusive, Channel 23 
Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8  callref = 0x97 
Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX -> NOTIFY pd = 8  callref = 0x97 
        Connected Address i = 0x80, '3102' 
        Notification Ind i = 0xF1 
        Display i = 0xB2, 'Gary Sullivan' 
Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x17 
        Cause i = 0x81E4 - Invalid information element contents 
Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x17 
        Cause i = 0x81E0 - Mandatory information element missing 

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

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From gary.sullivan at waters.nestle.com  Thu Dec 29 09:33:47 2005
From: gary.sullivan at waters.nestle.com (Sullivan, Gary, GREENWICH,
	Information Services)
Date: Thu Dec 29 09:35:02 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
Message-ID: <E9B4C49E7DD9DA459F42493E5485677B038F4772@usgcte00.nestle.com>

I am also on a Nortel 61C rls 25.40

-----Original Message-----
From: Patrick Aland [mailto:paland at gmail.com]
Sent: Thursday, December 29, 2005 8:41 AM
To: Ryan Ratliff
Cc: Sullivan,Gary,GREENWICH,Information Services; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PRI between CCM (3725) and Nortel M1


try adding 
 isdn channel-id invert extend-bit

Under your serial interface, i.e.

int ser0/0/0:23
 isdn channel-id invert extend-bit

I had the same problem with a Nortel 61C and the above fixed it. 


On 12/29/05, Ryan Ratliff < rratliff at cisco.com> wrote: 

The first call appears to be rejected by your PBX but I think the
debug is missing info.  The "Cause i = 0x81B6 - Incoming calls are
barred" should probably appear in a Disconnect or Release message
from the PBX. 

The second one is also rejected by the PBX. I would guess because of
the Display IE but the second Release message is a bit confusing.

Is this H.323 or MGCP?

-Ryan

On Dec 29, 2005, at 8:13 AM, Sullivan, Gary, GREENWICH, Information 
Services wrote:

We have a PRI between are CCM and Nortel M1 pbx. Was working fine
until we upgraded.
We upgraded are CCM from 3.3.4 to 4.1.3 . Now we can't make any
calls. the D-channel is established. 
but when I try to make a call from my IP phone it disconnects right
away. when I call from Nortel to IP
it rings once then disconnects. I did a debug isdn q931 and see the
Messages.
Any help is appreciated 


ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP pd = 8  callref =
0x0001
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transer Capability = Speech 
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Display i = 0xB1, 'Gary Sullivan'
         Calling Party Number i = 0x0081, '3102' 
                 Plan:Unknown, Type:Unknown
         Called Party Number i = 0x80, '0400'
                 Plan:Unknown, Type:Unknown

81
         Cause i = 0x81B6 - Incoming calls are barred
GCTRT03# 
GCTRT03#
GCTRT03#
GCTRT03#
GCTRT03#
Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8  callref
= 0x17
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transer Capability = Speech 
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xE9808317
                 Exclusive, Interface 0, Channel 23
         Display i = 0xB1, 'GARY SULLIVAN' 
         Calling Party Number i = 0x0980, '0543'
                 Plan:Private, Type:Unknown
         Called Party Number i = 0x89, '3102'
                 Plan:Private, Type:Unknown


         Channel ID i = 0xA98397 
                 Exclusive, Channel 23
Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8
callref = 0x97
Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX -> NOTIFY pd = 8  callref
= 0x97
         Connected Address i = 0x80, '3102' 
         Notification Ind i = 0xF1
         Display i = 0xB2, 'Gary Sullivan'
Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8
callref = 0x17
         Cause i = 0x81E4 - Invalid information element contents 
Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8
callref = 0x17
         Cause i = 0x81E0 - Mandatory information element missing


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-- 
--Patrick 

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From jgrace at digitelusa.net  Thu Dec 29 10:10:10 2005
From: jgrace at digitelusa.net (James Grace)
Date: Thu Dec 29 10:14:54 2005
Subject: [cisco-voip] cm location 
Message-ID: <200512291014406.SM01568@3729434770>

We are having for some issue with not enough bandwidth.  I look at the cm
and we have 2 locataions configured.  But on our phone phones the locations
is set to none.  Now on those two loacation they were config as limit
bandwth on voice and video.   My question is none on the ip phone uses a
default of one the the locations settings.  Just wondering if  this is why
we are having out of bandwidth issues on the phones 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From kthorngr at cisco.com  Thu Dec 29 10:41:26 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Dec 29 10:41:36 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
In-Reply-To: <E9B4C49E7DD9DA459F42493E5485677B038F4771@usgcte00.nestle.com>
References: <E9B4C49E7DD9DA459F42493E5485677B038F4771@usgcte00.nestle.com>
Message-ID: <1da5c665edcfdfef8fdc83e8d9128283@cisco.com>

Typically this type of behavior is seen when the Extension Bit is set  
to 1 but should be set to 0 per DMS-100.

As Ryan stated earlier the Nortel is disconnecting the calls.  I'm not  
familiar with Nortel gear but are there debugs on the Nortel side you  
can look at to see why the calls are being disconnected?

Is it expecting the Extension Bit to be 1 or 0?

The " isdn channel-id invert extended" command was introduced in 12.2  
but it doesn't appear it is in the version you are running:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/ 
products_command_reference_chapter09186a008010b000.html#wp1138159

Kevin

On Dec 29, 2005, at 9:32 AM, Sullivan,Gary,GREENWICH,Information  
Services wrote:

> Yes using DMS-100 and have that check all ready.
>> -----Original Message-----
>> From: Kevin Thorngren [mailto:kthorngr at cisco.com]
>> Sent: Thursday, December 29, 2005 8:39 AM
>> To: Sullivan,Gary,GREENWICH,Information Services
>> Cc: cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] PRI between CCM (3725) and Nortel M1
>>
>> Are you using DMS-100 Protocol? If so, I believe you need to place a  
>> check mark in the "MCDN Channel Number Extension Bit Set to Zero"  
>> option in the CCM GW configuration.
>>
>> Let us know if this helps.
>>
>> Kevin
>> On Dec 29, 2005, at 8:13 AM, Sullivan, Gary, GREENWICH, Information  
>> Services wrote:
>>
>>> We have a PRI between are CCM and Nortel M1 pbx. Was working fine  
>>> until we upgraded.
>>> We upgraded are CCM from 3.3.4 to 4.1.3 . Now we can't make any  
>>> calls. the D-channel is established.
>>> but when I try to make a call from my IP phone it disconnects right  
>>> away. when I call from Nortel to IP
>>> it rings once then disconnects. I did a debug isdn q931 and see the  
>>> Messages.
>>> Any help is appreciated
>>>
>>>
>>> ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP pd = 8? callref  
>>> = 0x0001
>>> ??????? Bearer Capability i = 0x8090A2
>>> ??????????????? Standard = CCITT
>>> ??????????????? Transer Capability = Speech?
>>> ??????????????? Transfer Mode = Circuit
>>> ??????????????? Transfer Rate = 64 kbit/s
>>> ??????? Channel ID i = 0xA98381
>>> ??????????????? Exclusive, Channel 1
>>> ??????? Display i = 0xB1, 'Gary Sullivan'
>>> ??????? Calling Party Number i = 0x0081, '3102'
>>> ??????????????? Plan:Unknown, Type:Unknown
>>> ??????? Called Party Number i = 0x80, '0400'
>>> ??????????????? Plan:Unknown, Type:Unknown
>>>
>>> 81
>>> ??????? Cause i = 0x81B6 - Incoming calls are barred
>>> GCTRT03#
>>> GCTRT03#
>>> GCTRT03#
>>> GCTRT03#
>>> GCTRT03#
>>> Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP pd = 8? callref  
>>> = 0x17
>>> ??????? Bearer Capability i = 0x8090A2
>>> ??????????????? Standard = CCITT
>>> ??????????????? Transer Capability = Speech?
>>> ??????????????? Transfer Mode = Circuit
>>> ??????????????? Transfer Rate = 64 kbit/s
>>> ??????? Channel ID i = 0xE9808317
>>> ??????????????? Exclusive, Interface 0, Channel 23
>>> ??????? Display i = 0xB1, 'GARY SULLIVAN'
>>> ??????? Calling Party Number i = 0x0980, '0543'
>>> ??????????????? Plan:Private, Type:Unknown
>>> ??????? Called Party Number i = 0x89, '3102'
>>> ??????????????? Plan:Private, Type:Unknown
>>>
>>>
>>> ??????? Channel ID i = 0xA98397
>>> ??????????????? Exclusive, Channel 23
>>> Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX -> ALERTING pd = 8?  
>>> callref = 0x97
>>> Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX -> NOTIFY pd = 8?  
>>> callref = 0x97
>>> ??????? Connected Address i = 0x80, '3102'
>>> ??????? Notification Ind i = 0xF1
>>> ??????? Display i = 0xB2, 'Gary Sullivan'
>>> Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8?  
>>> callref = 0x17
>>> ??????? Cause i = 0x81E4 - Invalid information element contents
>>> Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <- RELEASE pd = 8?  
>>> callref = 0x17
>>> ??????? Cause i = 0x81E0 - Mandatory information element missing
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
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From kthorngr at cisco.com  Thu Dec 29 10:57:34 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu Dec 29 10:57:38 2005
Subject: [cisco-voip] cm location 
In-Reply-To: <200512291014406.SM01568@3729434770>
References: <200512291014406.SM01568@3729434770>
Message-ID: <0c4badb19b31688ddeeb50fca5d283da@cisco.com>

If you place a call between two phones that have None configured for 
the Location then you have unlimited bandwidth available.  The call 
won't use one of the configured Locations.  It sounds like another 
device, Transcoder maybe, is being invoked that is in one of the 
configured Locations.  CCM traces will show you what is happening.

Kevin
On Dec 29, 2005, at 10:10 AM, James Grace wrote:

> We are having for some issue with not enough bandwidth.? I look at the 
> cm and we have 2 locataions configured.? But on our phone phones the 
> locations is set to none.? Now on those two loacation they were config 
> as limit bandwth on voice and video.?? My question is none on the ip 
> phone uses a default of one the the locations settings.? Just 
> wondering if? this is why we are having out of bandwidth issues on the 
> phones
> ?
> James D. Grace
> CCNP CCNA MCSE MCDBA
> Sr. System Engineer / Professional Svc.
> Digitel Corporation
> ?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From wsisk at cisco.com  Thu Dec 29 13:14:42 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 13:14:53 2005
Subject: [cisco-voip] Progress_ind command
In-Reply-To: <E0EF33751DEA634DA817A51024A8E3F2056CAE2C@usa0300ms01.na.xerox.net>
References: <E0EF33751DEA634DA817A51024A8E3F2056CAE2C@usa0300ms01.na.xerox.net>
Message-ID: <113A03DE-2311-46E2-AB60-6866C6E557E1@cisco.com>

progress_ind really has very little to do with DTMF.

progress_ind determine *WHEN* audio is cut through.  Though that  
audio cut through (actually the associated h245 session) is a pre- 
requisite for DTMF exchange, the 2 are not directly linked as you  
seem to imply.

/Wes

On Dec 28, 2005, at 12:36 PM, DeRooy, Aaron H wrote:



Does anybody know if ANY progress_ind command is needed to make sure
DTMF tones get thru the IP phone network to/from a PSTN destination?
I've been experimenting and find that the "dtmf-relay h245-alphanumeric"
command breaks DTMF tones but the progess_ind commands have no effect.
> From what I've been able to find out, progress_ind command solve
"ringback" problems with the "alert" key word but not sure about "setup"
and "progress" key works.
Thanks,

Aaron

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From wsisk at cisco.com  Thu Dec 29 13:20:08 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 13:22:05 2005
Subject: [cisco-voip] Hunt Pilot/List and Parked call reversion
In-Reply-To: <20051221173742.81581.qmail@web34313.mail.mud.yahoo.com>
References: <20051221173742.81581.qmail@web34313.mail.mud.yahoo.com>
Message-ID: <F6B03BEB-712E-43A4-B8D1-74816D0721C6@cisco.com>

this is working as designed for calls presented through huntlists.

/Wes

On Dec 21, 2005, at 12:37 PM, Erick Bergquist wrote:

Hi,

Does anyone know if calls to a hunt pilot, when a
member of hunt list/linegroup answers call and parks
call for the call to revert to the users phone who
parked the call?

right now members of the line group who park call made
to the hunt pilot, when the parked call reverts it
rings another member of the line group.

I've dug in service parameters, etc but not finding
any knobs. Is this working as designed?


__________________________________________________
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Tired of spam?  Yahoo! Mail has the best spam protection around
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From wsisk at cisco.com  Thu Dec 29 13:21:53 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 13:22:12 2005
Subject: [cisco-voip] Directory Integration
In-Reply-To: <009401c60684$62db3260$30196883@cfs.uoguelph.ca>
References: <D713462ED535184D830F6F6301753E7E01A1213F@VISHNU.wesd.org>
	<009401c60684$62db3260$30196883@cfs.uoguelph.ca>
Message-ID: <006EC859-5346-4C0C-A3DF-59DA1DA04658@cisco.com>

you can have CM integrated with the default DCD and use the  
directorysearch from ipphoneServicesSDK to have a directory service  
on the phone that searches your enterprise directory.

/Wes
On Dec 21, 2005, at 6:15 PM, Lelio Fulgenzi wrote:

I really wish that there was more to offer with AD integration than  
just that. perhaps display name for a DN?

I also think AD integration gives you centralized PIN.

Chris Adams is on the list - Chris, can you remind me of that one  
problem you had where you had to populate a different field for  
something to work b/c the integration broke it?
------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: Voll, Scott
To: Corbett Enders ; cisco-voip at puck.nether.net
Sent: Wednesday, December 21, 2005 6:11 PM
Subject: RE: [cisco-voip] Directory Integration

Pros with AD integration.  One username and password.



Con?s: if your AD goes down, no one can log in.



You?re very reliant on a stable AD.  If you have lots of AD  
instability, don?t integrate.



Works fine for us.



Scott



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Corbett Enders
Sent: Tuesday, December 20, 2005 11:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Directory Integration



Is there a general best practice around integration with Windows  
Active Directory?  I'm thinking it is a good idea, but I'm wondering  
if anyone has precautions or reasons NOT to use this model?

Regards,
Corbett Enders.



_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip
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From wsisk at cisco.com  Thu Dec 29 13:43:11 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 13:43:26 2005
Subject: [cisco-voip] Directory Integration
In-Reply-To: <24DFD2417491E547AC80AE140706C4857F6D@EX001.homesbyavi.local>
References: <24DFD2417491E547AC80AE140706C4857F6D@EX001.homesbyavi.local>
Message-ID: <41AB5B28-1635-430A-9232-B701D20B3E4A@cisco.com>

take a look at CSCdz63349 in bug toolkit for instructions on how to  
hide users.

/Wes

On Dec 21, 2005, at 6:31 PM, Corbett Enders wrote:

I read something to that effect regarding hiding accounts from the  
Call Manager, by setting a custom field entry.

I have also noted an issue with regards to Exchange 2000, not sure  
yet if it applies to Exchange 2003 (iNetOrgPerson class).

Slight twist on my original question:  Have you performed the AD  
Integration and regretted it?  Did it break something you were trying  
to do elsewhere?  And on the flip side, did anyone not perform the AD  
Integration and now wish that you had?

Pros:  Single directory (the model everyone pitches, yet not often  
used), Simplified Management (AD Users & Computers?)
Cons: Increased dependency on AD domain controllers, possible latency  
for some of the call flow (lookups).


From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Wednesday, December 21, 2005 4:15 PM
To: Voll, Scott; Corbett Enders; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Directory Integration

I really wish that there was more to offer with AD integration than  
just that. perhaps display name for a DN?

I also think AD integration gives you centralized PIN.

Chris Adams is on the list - Chris, can you remind me of that one  
problem you had where you had to populate a different field for  
something to work b/c the integration broke it?
------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: Voll, Scott
To: Corbett Enders ; cisco-voip at puck.nether.net
Sent: Wednesday, December 21, 2005 6:11 PM
Subject: RE: [cisco-voip] Directory Integration

Pros with AD integration.  One username and password.



Con?s: if your AD goes down, no one can log in.



You?re very reliant on a stable AD.  If you have lots of AD  
instability, don?t integrate.



Works fine for us.



Scott



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Corbett Enders
Sent: Tuesday, December 20, 2005 11:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Directory Integration



Is there a general best practice around integration with Windows  
Active Directory?  I'm thinking it is a good idea, but I'm wondering  
if anyone has precautions or reasons NOT to use this model?

Regards,
Corbett Enders.



_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
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From wsisk at cisco.com  Thu Dec 29 13:45:09 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 13:45:36 2005
Subject: [cisco-voip] Saving IP phone directories
In-Reply-To: <015c01c6067a$66233d00$400cfb0a@crec.ifas.ufl.edu>
References: <200512212135.jBLLZcpd037476@puck.nether.net>
	<015c01c6067a$66233d00$400cfb0a@crec.ifas.ufl.edu>
Message-ID: <4AF855ED-180F-4475-B224-D5A7FE3C57EB@cisco.com>

In the IPPhoneServiceSDK there are cmds to retrieve the informatoin,  
but no way to automagically insert the data.

/Wes

On Dec 21, 2005, at 5:03 PM, Mike Armstrong wrote:

When replacing an IP phone, we usually just assign the old IP to the new
phone and make the substitution.  However, all the stored directories
(Missed Calls, etc.) are lost across a power reset.  Is there any way to
save those and reload them into the new phone, probably via some XML  
magic?

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

----- Original Message -----
From: <cisco-voip-request at puck.nether.net>
To: <cisco-voip at puck.nether.net>
Sent: Wednesday, December 21, 2005 4:35 PM
Subject: cisco-voip Digest, Vol 34, Issue 117


> 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. AC Pilot Point ringing busy (Erick Bergquist)
>   2. Re: AC Pilot Point ringing busy (Erick Bergquist)
>   3. Hunt Pilot/List and Parked call reversion (Erick Bergquist)
>   4. HELP---PASSWORD IS LOCKED  (Mike Carter)
>   5. Unity sending message (James)
>   6. - PGW220, SLT (2651) & AS5400 (wael ahmed el mezain)
>   7. RE: HELP---PASSWORD IS LOCKED  (Jim Koniecki)
>   8. New gw missing t1 controller (James)
>   9. RE: CDR Error (IT)
>  10. Unity Permissions Wizard - Not Taking Effect (Stu Packett)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 21 Dec 2005 09:00:24 -0800 (PST)
> From: Erick Bergquist <erickbe at yahoo.com>
> Subject: [cisco-voip] AC Pilot Point ringing busy
> To: cisco-voip at puck.nether.net
> Message-ID: <20051221170024.2060.qmail at web34311.mail.mud.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi, Have CCM 4.1(3)sr1 in lab and trying to get a
> attendant console pilot point going and have hunt
> group with 2-3 numbers in it (call directory numbers)
> and whenever I call pilot point # it is busy. It is
> not in a partition, etc and there are no matching
> numbers in dialplan. I reset the server many times,
> restarted TCD, etc, and the thing is still busy.
>
> DNA shows the number is blocked on it's analysis. In
> CCM Detailed traces it shows theres NoPotentialMatches
> when the number is called.
>
> Going through some TCD traces now to figure this out,
> but thought I'd see if anyone had seen this before.
> Not much to the setup and only a few phones on this
> lab setup. The ac user exists and has right password
> (12345) as I'm able to login with AC fine, etc.
>
> The problem I'm trying to work on is when calls come
> into a AC pilot for a client, when they park the call
> when it reverts off of park it rings back to the pilot
> # and not the users phone who parked the call and
> someone else in hunt group gets the call. I'm thinking
> it may be one of the service parameter settings but
> wanted to confirm it on my lab setup but having issues
> with AC.
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
>
> ------------------------------
>
> Message: 2
> Date: Wed, 21 Dec 2005 09:16:30 -0800 (PST)
> From: Erick Bergquist <erickbe at yahoo.com>
> Subject: Re: [cisco-voip] AC Pilot Point ringing busy
> To: erickbe at yahoo.com, cisco-voip at puck.nether.net
> Message-ID: <20051221171630.32993.qmail at web34308.mail.mud.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Nevermind.. got it working. Deleted the pilot point
> and added it back. Ugh.
>
> --- Erick Bergquist <erickbe at yahoo.com> wrote:
>
>> Hi, Have CCM 4.1(3)sr1 in lab and trying to get a
>> attendant console pilot point going and have hunt
>> group with 2-3 numbers in it (call directory
>> numbers)
>> and whenever I call pilot point # it is busy. It is
>> not in a partition, etc and there are no matching
>> numbers in dialplan. I reset the server many times,
>> restarted TCD, etc, and the thing is still busy.
>>
>> DNA shows the number is blocked on it's analysis. In
>> CCM Detailed traces it shows theres
>> NoPotentialMatches
>> when the number is called.
>>
>> Going through some TCD traces now to figure this
>> out,
>> but thought I'd see if anyone had seen this before.
>> Not much to the setup and only a few phones on this
>> lab setup. The ac user exists and has right password
>> (12345) as I'm able to login with AC fine, etc.
>>
>> The problem I'm trying to work on is when calls come
>> into a AC pilot for a client, when they park the
>> call
>> when it reverts off of park it rings back to the
>> pilot
>> # and not the users phone who parked the call and
>> someone else in hunt group gets the call. I'm
>> thinking
>> it may be one of the service parameter settings but
>> wanted to confirm it on my lab setup but having
>> issues
>> with AC.
>>
>>
>> __________________________________________________
>> 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
>>
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 21 Dec 2005 09:37:42 -0800 (PST)
> From: Erick Bergquist <erickbe at yahoo.com>
> Subject: [cisco-voip] Hunt Pilot/List and Parked call reversion
> To: cisco-voip at puck.nether.net
> Message-ID: <20051221173742.81581.qmail at web34313.mail.mud.yahoo.com>
> Content-Type: text/plain; charset=iso-8859-1
>
> Hi,
>
> Does anyone know if calls to a hunt pilot, when a
> member of hunt list/linegroup answers call and parks
> call for the call to revert to the users phone who
> parked the call?
>
> right now members of the line group who park call made
> to the hunt pilot, when the parked call reverts it
> rings another member of the line group.
>
> I've dug in service parameters, etc but not finding
> any knobs. Is this working as designed?
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
>
> ------------------------------
>
> Message: 4
> Date: Wed, 21 Dec 2005 12:29:19 -0600
> From: "Mike Carter" <mike.c at altatechnologies.com>
> Subject: [cisco-voip] HELP---PASSWORD IS LOCKED
> To: <cisco-voip at puck.nether.net>
> Message-ID: <BFEHKOIOENIIPHGJOCHDOEAPFDAA.mike.c at altatechnologies.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> HELP!!!!!!!
>
> not sure how but somehow one of our phones got locked----none of the
> default
> passwords work
>
> ideas????
>
> Michael Carter
> Account Manager
> Alta Technologies Inc
> 763-475-5327
> 800-546-2582 (327)
> AIM mikec at alta
> yahoo IM cartematalta
> mike.c at altatechnologies.com
> Sun IBM HP Cisco Nortel Juniper Adtran
> D-Link Emulex Qlogic Overland Storage
> We buy and sell all of the above...please email your offers or  
> requests.
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Wed, 21 Dec 2005 18:42:13 +0000 GMT
> From: "James" <jgrace at digitelusa.net>
> Subject: [cisco-voip] Unity sending message
> To: "CiscosupportUpuck" <cisco-voip at puck.nether.net>
> Message-ID:
> <364521873-1135190538-cardhu_blackberry.rim.net-23637- at engine18- 
> cell02>
>
> Content-Type: text/plain
>
> I have a question.  In unity when you crreate an vm and it give u the
> options like 4 to send message. And it give you the dial by name or ##
> for ext.   Is there a way to flip it around to ask for ext and then  
> name
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 21 Dec 2005 19:03:41 +0000
> From: "wael ahmed el mezain" <wael733 at hotmail.com>
> Subject: [cisco-voip] - PGW220, SLT (2651) & AS5400
> To: cisco-voip at puck.nether.net
> Message-ID: <BAY105-F244463F15E8BC7F498F361E7310 at phx.gbl>
> Content-Type: text/plain; format=flowed
>
> Hello Guys,
>
> Is there anyone install this system before to give me a hand  
> [PGW220, SLT
> (2651) & AS5400].
>
> I want to configure it to make interface between PSTN & Cisco VoIP  
> system
> (MGCP server, HSI server, BAMS server & 2651 as SLT and AS5400 as  
> Access
> Media) and how to config. the CCM to go to PGW gateway through the  
> 3745
> gatekeeper.
>
> You can add me on your MSN to make a chat (wael733 at hotmail.com).
>
> Thanks,
> Wael Ahmed
>
>
>
>
> ------------------------------
>
> Message: 7
> Date: Wed, 21 Dec 2005 14:34:51 -0500
> From: "Jim Koniecki" <jkoniecki at comcast.net>
> Subject: RE: [cisco-voip] HELP---PASSWORD IS LOCKED
> To: "'Mike Carter'" <mike.c at altatechnologies.com>,
> <cisco-voip at puck.nether.net>
> Message-ID: <000801c60665$9cfe1a80$c800a8c0 at Jalbert1>
> Content-Type: text/plain; charset="us-ascii"
>
> Mike,
>
> Please explain what you mean by locked? What are you trying to do or
> perform?
>
> What is the default password you are using ?
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Carter
> Sent: Wednesday, December 21, 2005 1:29 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] HELP---PASSWORD IS LOCKED
>
> HELP!!!!!!!
>
> not sure how but somehow one of our phones got locked----none of the
> default
> passwords work
>
> ideas????
>
> Michael Carter
> Account Manager
> Alta Technologies Inc
> 763-475-5327
> 800-546-2582 (327)
> AIM mikec at alta
> yahoo IM cartematalta
> mike.c at altatechnologies.com
> Sun IBM HP Cisco Nortel Juniper Adtran
> D-Link Emulex Qlogic Overland Storage
> We buy and sell all of the above...please email your offers or  
> requests.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> ------------------------------
>
> Message: 8
> Date: Wed, 21 Dec 2005 20:50:20 +0000 GMT
> From: "James" <jgrace at digitelusa.net>
> Subject: [cisco-voip] New gw missing t1 controller
> To: "CiscosupportUpuck" <cisco-voip at puck.nether.net>
> Message-ID:
> <1940869722-1135198224-cardhu_blackberry.rim.net-9259- at engine19- 
> cell02>
>
> Content-Type: text/plain
>
> We r installing a new gateway. And for somereason the serial show  
> up in
> the config. But the t1 interface do not. I ran into this some time  
> ago and
> I think we had to addACmd.
> James D. Grace
> System Engineer
> CCNP CCNA MCSE MCDBA
> Digitel Corp.
>
>
> ------------------------------
>
> Message: 9
> Date: Wed, 21 Dec 2005 13:04:17 -0800
> From: "IT" <it at cimgroup.com>
> Subject: RE: [cisco-voip] CDR Error
> To: "Ryan Ratliff" <rratliff at cisco.com>, "IT" <it at cimgroup.com>
> Cc: cisco-voip at puck.nether.net
> Message-ID:
> <B487D4C1FE6D04439061018C6EBC109B026B1C19 at exchange.cimgroup.com>
> Content-Type: text/plain; charset="us-ascii"
>
> It appears the problem is larger than that. I installed the ART on  
> 11/8,
> and the Max_date for any query has not moved from 11/8. I manually
> changed the max_date, but all the reports are just filled with 0's.  
> So I
> changed the max_date back.
> I have setup the Load Scheduler to run on default of midnight. When I
> check the Tbl_Event_Log in SQL, the CDRLoad Jobs are taking only a
> second or so (start->end time), and most of the utilization jobs are
> returning a status of -2. The Error_log table is filled with
> CtrPregenReports[generate] errors with Error_code 1020.
> Where else can I look?
>
> Thanks,
> Avidan
>
> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> Sent: Tuesday, December 20, 2005 9:23 PM
> To: IT
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CDR Error
>
> DirUser is the API ART uses for ldap access.
>
> Are you even able to login to the ART web page?
>
> Many times we see issues where the CDRs are being loaded just not
> associated the correct user.
> Are your reports user or department based?
>
> If so, try doing a CDR search on your extension to see if it finds
> calls.
>
> There is a known issue with ART where when it goes to retrieve the
> user a device is associated to if the directory returns multiple
> users ART will take the first one.  This can cause issues with IPCC
> agents, AC users, etc where you have lots of devices associated with
> application users.  The "owner userid" field on the phone device page
> was created to workaround this.
>
> Of course none of this may be relevant to your issue but it's a
> common problem.
>
> -Ryan
>
> On Dec 21, 2005, at 12:10 AM, IT wrote:
>
> It appears that none of my reports contain any data when I use CAR,
> even though I enabled the CDR loading.
>
> Here is the error I have in ARTSchError.log
>
> DirUser.OperationError
>
>             at DirUser.LoadComClass.retAupairClass(LoadComClass.java)
>
>             at DirUser.DirUser.<init>(DirUser.java)
>
>             at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java:
> 137)
>
>             at com.cisco.art.controller.CtrLoader.run(CtrLoader.java:
> 1356)
>
>             at com.cisco.art.scheduler.SchThread.run(SchThread.java:
> 550)
>
> DirUser.OperationError
>
>             at DirUser.LoadComClass.retAupairClass(LoadComClass.java)
>
>             at DirUser.DirUser.<init>(DirUser.java)
>
>             at com.cisco.art.controller.CtrLdap.<init>(CtrLdap.java:
> 137)
>
>             at com.cisco.art.controller.CtrLoader.run(CtrLoader.java:
> 1356)
>
>             at com.cisco.art.scheduler.SchThread.run(SchThread.java:
> 550)
>
>
>
> I have Active Directory integration installed with CM4.0. Has anyone
> seen this issue before?
>
>
>
> Thanks,
>
> Avidan
>
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are  
> not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> ------------------------------
>
> Message: 10
> Date: Wed, 21 Dec 2005 13:35:29 -0800
> From: "Stu Packett" <SPackett at fenwick.com>
> Subject: [cisco-voip] Unity Permissions Wizard - Not Taking Effect
> To: <cisco-voip at puck.nether.net>
> Message-ID:
> <965C0255FD5828438E34F9A68ECA76E740A686 at svcexch02.firm.fenwick.llp>
> Content-Type: text/plain; charset="us-ascii"
>
> I brought up a new Exchange server and a few Exchange stores on  
> existing
> servers, but after running the Unity Permissions Wizard, the  
> permissions
> are not taking effect.  When I looked at the Permissions Wizard  
> logs, it
> shows that UnityMsgSvc (Unity Message Store Service account) as
> 'Succeeded'.  I've waited over 15 minutes for the AD replication to
> occur and I've restarted Unity.  I even restarted the Exchange
> Information Store service on the new Exchange server, but I still  
> cannot
> get the permissions to carry over.  Because the permissions are not
> taking effect on Exchange, any user I enable on the new Exchange  
> server
> fails.
>
> Unity 4.1.(1) ES11 on Windows 2003
> Exchange 2003 SP1
>
> Thanks in advance.
> --------------------------------------------------------
>
> IRS Circular 230  Disclosure:  To ensure compliance with requirements
> imposed by the IRS, we inform you that any U.S. federal tax advice  
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> _______________________________________________
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> End of cisco-voip Digest, Vol 34, Issue 117
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From wsisk at cisco.com  Thu Dec 29 13:55:01 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 13:55:14 2005
Subject: [cisco-voip] Busy out channels on an mgcp gateway
In-Reply-To: <008f01c6063e$a56c1220$30196883@cfs.uoguelph.ca>
References: <7E3064B7EFC16C41B185C41D141008B7C3CE83@foleyex97.foleylaw.com>
	<008f01c6063e$a56c1220$30196883@cfs.uoguelph.ca>
Message-ID: <CF6ECC03-22CB-4450-859F-784DF250F690@cisco.com>

this does work gracefully if it is one of the supported protocols:
CSCed07090

/Wes

On Dec 21, 2005, at 9:55 AM, Lelio Fulgenzi wrote:

I've heard people use the following service parameter to do what you  
are asking. There are five of them available I think.


Change B-Channel Maintenance Status 1 :	This parameter allows Cisco  
CallManager to change individual B-Channel maintenance status for PRI  
interface in real time for troubleshooting. Input Format:
Device Name = B-channel Maintenance Status
Example:
S0/DS1-0 at SDA123456789ABC = 0000 1111 0000 0000 0000 0001(24 bits for  
T1. Channel number begins with 1 to 24 from left to right. The last  
one specifies the D-Channel, which does not get affected. In this  
example, 5th-8th B-Channels are out of service).
or,
S1/DS1-0 at DLS2-CM136-VG200-2.SELSIUSCMLAB.CISCO.COM =       0000 0000  
0000 0000 0000 0000 1111 1111(32 bits for E1. B-Channel number begins  
with 1 to 32 from left to right. The 16th channel specifies D- 
Channel. The last bit does not affect any channel. In this example,  
25th-31st channels are marked out of service).
Explanation:
The DEVICE NAME MUST exactly match the gateway name on the top of the  
gateway configuration web page. Copy and paste that field into this  
service parameter to avoid any manual inputting mistake. Ensure '='  
is mandatory and unique to distinguish the DEVICE NAME and BCHANNEL  
MAINTENANCE STATUS fields. BCHANNEL MAINTENANCE STATUS = xxxx xxxx  
xxxx xxxx xxxx xxxx xxxx xxxx
Where x can be:
0 -- In service
1 -- Graceful out of service (Change channel status until active call  
ends if an active call exists on that channel)
2 -- Forceful out of service (Tear down active call first; then,  
change channel status immediately if an active call exists on that  
channel).
The system treats any other value as invalid and the value does not  
take effect. Make sure that the total number of ?x? is either 24 for  
T1 or 32 for E1. Any other invalid length or mismatch (for example,  
24 for E1) gets treated as error, and no action gets taken for that  
device.
SPACE: Optional.
The spaces between device name and '=', between ?=? and BCHANNEL  
MAINTENANCE STATUS, and between x and x, are all optional. You cannot  
use any space inside the DEVICE NAME field.
To use this feature, the "Enable Status Poll" flag on the gateway  
configuration page must be checked and updated, then the gateway must  
be reset for the first time.After this, changing service parameter  
will take effect without resetting the gateway.
Maximum length: 255.
------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
----- Original Message -----
From: Ruttman, Peter G.
To: cisco-voip at puck.nether.net
Sent: Wednesday, December 21, 2005 9:43 AM
Subject: [cisco-voip] Busy out channels on an mgcp gateway

I need to reset an mgcp gateway PRI that has inbound DIDs associated  
with it.  Is it possible to busy out the channels on it so that new  
calls do not get pinned up on it?  Then I can safely reset the  
gateway during work hours.

Pete


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_______________________________________________
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From wsisk at cisco.com  Thu Dec 29 14:01:09 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 14:01:19 2005
Subject: [cisco-voip] TCL script to parse *ANI*DNIS*
In-Reply-To: <4C792A099259634C83F97067671B37FE01A87AE8@GDSEXC.gds.pri>
References: <4C792A099259634C83F97067671B37FE01A87AE8@GDSEXC.gds.pri>
Message-ID: <57CD2F6B-6558-464A-A161-789A4F3562E5@cisco.com>

htis is Feature group D signaling.  did youtr fgd ds0 group?

also, did you try eana for inbound and outbound?
http://www.cisco.com/en/US/partner/tech/tk652/tk653/ 
technologies_tech_note09186a00800e2560.shtml

otherwise:
http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps1834/ 
products_feature_guide09186a008007fe27.html

/Wes

On Dec 28, 2005, at 12:18 PM, Patrick Mowry wrote:

Hello,
  I have a carrier providing a T1 E&M-II wink start trunk to a Fujitsu
F9600.  They provide caller-id information by formatting the digits send
as *ANI*DNIS.  If I configure the controller as:
ds0-group 1 timeslots 1-24 type e&m-wink-start dnis

I can place outbound calls, but for inbound calls I'll see a number like
*3375551212*3600* , and this of course would not match anything my
dail-peers or CallManager expects.  Can a TCL script be used to break
this number up and pass the caller-id and DID to CallManager?  The IOS
is 12.4T AdvancedIP Services, and CCM 4.1.3sr2

Thanks,

-Patrick

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From nkhwaja at gmail.com  Thu Dec 29 14:41:49 2005
From: nkhwaja at gmail.com (Nasser Khwaja)
Date: Thu Dec 29 14:41:52 2005
Subject: [cisco-voip] IOS Command?
Message-ID: <23435d3d0512291141u4c5a58a4s8dbd689d7a9dff96@mail.gmail.com>

Hi,
My Router has an NM-HD-2V module with a VIC2-4FXO card. Which IOS command(s)
can exactly confirm this fact?
Thanks
-Nasser.
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From CMStaffin at ra.rockwell.com  Thu Dec 29 14:52:59 2005
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Thu Dec 29 14:52:32 2005
Subject: [cisco-voip] IOS Command?
Message-ID: <OFC713904D.01BE2776-ON862570E6.006D2FD0-862570E6.006D2C9F@ra.rockwell.com>

Show Inventory 

this command will tell you all modules installed.


Thanks,
__________________________
Craig Staffin
Network Analyst
Global Network and Telecomunnications
Rockwell Automation
cmstaffin at ra.rockwell.com
(414) 382-2309






Nasser Khwaja <nkhwaja at gmail.com>
Sent by: cisco-voip-bounces at puck.nether.net
12/29/2005 01:41 PM

 
        To:     cisco-voip at puck.nether.net
        cc: 
        Subject:        [cisco-voip] IOS Command?


Hi,
My Router has an NM-HD-2V module with a VIC2-4FXO card. Which IOS 
command(s) can exactly confirm this fact?
Thanks
-Nasser.
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From erickbe at yahoo.com  Thu Dec 29 15:54:00 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Dec 29 15:54:08 2005
Subject: [cisco-voip] PRI between CCM (3725) and Nortel M1
In-Reply-To: <E9B4C49E7DD9DA459F42493E5485677B038F4771@usgcte00.nestle.com>
Message-ID: <20051229205400.1514.qmail@web34312.mail.mud.yahoo.com>

Also try (if H323), or is it MGCP controlled?

isdn flip-chan-flag 

under the D-channel configuration. I had to do that
once when connecting to a Option 61C with cisco router
and DMS 100.

--- "Sullivan, Gary, GREENWICH, Information Services"
<gary.sullivan at waters.nestle.com> wrote:

> Yes using DMS-100 and have that check all ready.
> 
> -----Original Message-----
> From: Kevin Thorngren [mailto:kthorngr at cisco.com]
> Sent: Thursday, December 29, 2005 8:39 AM
> To: Sullivan,Gary,GREENWICH,Information Services
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] PRI between CCM (3725) and
> Nortel M1
> 
> 
> Are you using DMS-100 Protocol? If so, I believe you
> need to place a check mark in the "MCDN Channel
> Number Extension Bit Set to Zero" option in the CCM
> GW configuration. 
> 
> Let us know if this helps. 
> 
> Kevin 
> On Dec 29, 2005, at 8:13 AM, Sullivan, Gary,
> GREENWICH, Information Services wrote: 
> 
> 
> We have a PRI between are CCM and Nortel M1 pbx. Was
> working fine until we upgraded. 
> We upgraded are CCM from 3.3.4 to 4.1.3 . Now we
> can't make any calls. the D-channel is established. 
> but when I try to make a call from my IP phone it
> disconnects right away. when I call from Nortel to
> IP 
> it rings once then disconnects. I did a debug isdn
> q931 and see the Messages. 
> Any help is appreciated 
> 
> 
> ec 20 09:41:28.853: ISDN Se2/0:23 Q931: TX -> SETUP
> pd = 8  callref = 0x0001 
>         Bearer Capability i = 0x8090A2 
>                 Standard = CCITT 
>                 Transer Capability = Speech  
>                 Transfer Mode = Circuit 
>                 Transfer Rate = 64 kbit/s 
>         Channel ID i = 0xA98381 
>                 Exclusive, Channel 1 
>         Display i = 0xB1, 'Gary Sullivan' 
>         Calling Party Number i = 0x0081, '3102' 
>                 Plan:Unknown, Type:Unknown 
>         Called Party Number i = 0x80, '0400' 
>                 Plan:Unknown, Type:Unknown 
> 
> 81 
>         Cause i = 0x81B6 - Incoming calls are barred
> 
> GCTRT03# 
> GCTRT03# 
> GCTRT03# 
> GCTRT03# 
> GCTRT03# 
> Dec 20 09:41:38.894: ISDN Se2/0:23 Q931: RX <- SETUP
> pd = 8  callref = 0x17 
>         Bearer Capability i = 0x8090A2 
>                 Standard = CCITT 
>                 Transer Capability = Speech  
>                 Transfer Mode = Circuit 
>                 Transfer Rate = 64 kbit/s 
>         Channel ID i = 0xE9808317 
>                 Exclusive, Interface 0, Channel 23 
>         Display i = 0xB1, 'GARY SULLIVAN' 
>         Calling Party Number i = 0x0980, '0543' 
>                 Plan:Private, Type:Unknown 
>         Called Party Number i = 0x89, '3102' 
>                 Plan:Private, Type:Unknown 
> 
> 
>         Channel ID i = 0xA98397 
>                 Exclusive, Channel 23 
> Dec 20 09:41:39.206: ISDN Se2/0:23 Q931: TX ->
> ALERTING pd = 8  callref = 0x97 
> Dec 20 09:41:39.210: ISDN Se2/0:23 Q931: TX ->
> NOTIFY pd = 8  callref = 0x97 
>         Connected Address i = 0x80, '3102' 
>         Notification Ind i = 0xF1 
>         Display i = 0xB2, 'Gary Sullivan' 
> Dec 20 09:41:39.222: ISDN Se2/0:23 Q931: RX <-
> RELEASE pd = 8  callref = 0x17 
>         Cause i = 0x81E4 - Invalid information
> element contents 
> Dec 20 09:41:39.230: ISDN Se2/0:23 Q931: RX <-
> RELEASE pd = 8  callref = 0x17 
>         Cause i = 0x81E0 - Mandatory information
> element missing 
> 



	
		
__________________________________ 
Yahoo! for Good - Make a difference this year. 
http://brand.yahoo.com/cybergivingweek2005/

From wsisk at cisco.com  Thu Dec 29 15:54:34 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 15:54:44 2005
Subject: [cisco-voip] IOS Command?
In-Reply-To: <23435d3d0512291141u4c5a58a4s8dbd689d7a9dff96@mail.gmail.com>
References: <23435d3d0512291141u4c5a58a4s8dbd689d7a9dff96@mail.gmail.com>
Message-ID: <0E0D4632-8A84-41EA-A51E-73E120CFEB48@cisco.com>

'show diag'

/Wes

On Dec 29, 2005, at 2:41 PM, Nasser Khwaja wrote:

Hi,
My Router has an NM-HD-2V module with a VIC2-4FXO card. Which IOS  
command(s) can exactly confirm this fact?
Thanks
-Nasser.

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From wsisk at cisco.com  Thu Dec 29 16:00:14 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 16:00:26 2005
Subject: [cisco-voip] What's the difference
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A12122@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A12122@VISHNU.wesd.org>
Message-ID: <268C50B7-0E65-4A7B-832B-00C98699995F@cisco.com>

SEP = selsius ethernet phone
ADP = autogenerated device profile

ADP is the 'default profile' for an extension mobility phone that is  
logged out.

/Wes
On Dec 20, 2005, at 2:48 PM, Voll, Scott wrote:

What is the difference between a device with SEP and ADP prefixes?   
CM 4.1 but I think I remember it from 3.3.



TIA



Scott

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From wsisk at cisco.com  Thu Dec 29 16:03:12 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 16:03:49 2005
Subject: [cisco-voip] CM 4.1 to PBX GW troublshsooting
In-Reply-To: <20051220182216.YNBN29285.eastrmmtao03.cox.net@[172.18.52.8]>
References: <20051220182216.YNBN29285.eastrmmtao03.cox.net@[172.18.52.8]>
Message-ID: <BBD3664F-4265-4225-BDBB-4592780BD500@cisco.com>

start by observing  'sh isdn status' and 'sh isdn service' when calls  
are failing.  What is the status of all channels from the perspective  
of the router?  this assumes you are using PRI.  then proceed to 'deb  
isdn q931' to see why calls are failing.

If not PRI but CAS (ds0-group) how about 'sh voice port sum'.  What  
state are the voice ports in? matching debug in this case is 'debug  
vpm all'

/Wes


On Dec 20, 2005, at 1:21 PM, <bababooey at cox.net> <bababooey at cox.net>  
wrote:

I have a Call Manager 4.1.3 and a router interface to a pbx with a T1  
trunk.  It is configured for H.323.
I find that I must reset the T1 controller almost daily to restore  
the PBX access.  I believe the problem is on my side not the PBX,  
because PBX users can still ring ip phones on the Call Manager.

How do I begin to troubleshoot this problem?
Which debug would be helpful?

Thanks -
Bob


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From wsisk at cisco.com  Thu Dec 29 16:07:00 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 16:07:13 2005
Subject: [cisco-voip] Inbound Calling Name
In-Reply-To: <697161D0107BAD40AFE9A86B06509AC90643E17F@FCEXCLUSTER.FirstCharter.com>
References: <697161D0107BAD40AFE9A86B06509AC90643E17F@FCEXCLUSTER.FirstCharter.com>
Message-ID: <7DF1B720-4885-43EC-BABB-A2FE117A40DA@cisco.com>

4ess is the only protocol CM cname feature does not work for.
cname = name delivery via facility IE.

if your carrier/pbx can send it in a DisplayIE CM will display it.

/Wes
On Dec 19, 2005, at 4:39 PM, Jason Burwell wrote:

On CCM 3.3.4 can you get Inbound Calling Name when using 4ESS  
Protocol? Using 6608 as a Gateway.

Thanks!
Jason


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From wsisk at cisco.com  Thu Dec 29 16:09:48 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 16:10:15 2005
Subject: [cisco-voip] Export Fast Dials and Address Book
In-Reply-To: <f1b65f880512191355w4a50ec7cv@mail.gmail.com>
References: <f1b65f880512191355w4a50ec7cv@mail.gmail.com>
Message-ID: <A97C620A-46DE-4572-A6F1-C7563D61DFFA@cisco.com>

only way (albeit unsupported) is with LDIF commands to export/import  
the data.

There was a directory migration kit floating around that was  
available by special request.  Might check with your CiscoSE to see  
if it is still available.

/Wes
On Dec 19, 2005, at 4:55 PM, Justin Steinberg wrote:

I have two CallManager clusters that I'm combining into one cluster.
On both clusters I have users who aggresively use the fast dial and
address book IP phone services.  I'm looking for a way to export this
data so that I can take the contents of one cluster and add them to
the other cluster without overwriting what is currently there.

I can see the data when I browse via the DC Admin application but do
not see any options in the menu's to backup or export this data.

I also did not see this available in BAT.

Any help is greatly appreciated.

Thanks,

Justin

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From wsisk at cisco.com  Thu Dec 29 16:11:29 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 16:11:47 2005
Subject: [cisco-voip] Analog line extension
In-Reply-To: <52964DBFB59FA34F98E96921BE21C906B235B3@twt-exch.milano.twt>
References: <52964DBFB59FA34F98E96921BE21C906B235B3@twt-exch.milano.twt>
Message-ID: <057D8F32-57CC-482F-9C5B-FFE5B5845F4D@cisco.com>

HDA FXO module does the highest FXO port density I can think of.   
Other than the old 8/16 port fxo cards for the ICS chassis.

/Wes

On Dec 19, 2005, at 12:56 PM, Brian Turnbow wrote:

Hello all,
I have a need to extend 32 analog lines over an IP network with fxs  
on one side and fxo on the other side.
I need to set up hotline for each port so that when the phone goes  
off hook it connects automatically to the specified port on the other  
end.

I can't change to ISDN and need to maintain the actual analog  
configuration. (I've tried to convice them to change architecture but  
it's a no go)

That being said I thought of the cisco IAD, but it seems to only  
offer high port density with FXS lines not FXO?
Any suggestions on euipment ?

Thanks
Brian


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From wsisk at cisco.com  Thu Dec 29 16:17:47 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 16:18:02 2005
Subject: [cisco-voip] Newbie Question
In-Reply-To: <OF749F26ED.5C1F6E26-ON852570DC.00591208-852570DC.0059498E@bccnetworks.net>
References: <OF749F26ED.5C1F6E26-ON852570DC.00591208-852570DC.0059498E@bccnetworks.net>
Message-ID: <D4F530E0-2562-42C2-A975-71135E04FC2E@cisco.com>

aka "B-ACD" implemented in TCL:

http://www.cisco.com/en/US/customer/products/sw/voicesw/ps4625/ 
products_feature_guide_chapter09186a00804ad4c0.html

/Wes

On Dec 19, 2005, at 11:15 AM, PBrunnen at bccnetworks.net wrote:


Hello all!
    Sorry in advance for this question... but I was not sure where  
else to ask it.
I had heard that you can setup a router, such as a 2621, as a VoIP  
call center.
If anyone could provide any further details as to how this is  
accomplished, I would greatly appreciate it.
I have been searching around the net with no success except to run  
across this list.

Thank you again for any help you can provide.  -Cheers, Peter.


====================
In specification, Murphy's
Law supersedes Ohm's Law.
       --Unknown
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From wsisk at cisco.com  Thu Dec 29 16:19:18 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 16:19:35 2005
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 96
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B93@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC2B93@MREXVS01.bc.broward.cty>
Message-ID: <BA4C7E19-C5BD-44A3-ABE1-C4F0831E408C@cisco.com>

Carlos,

This config is convered in the Video with callmanager SRND:

http://www.cisco.com/go/srnd

/Wes
On Dec 19, 2005, at 10:29 AM, Ortiz, Carlos wrote:

Steve,



Would you mind pointing me out to the documentation you used for this  
setup?  Specifically the Gatekeeper and MCU setup.  We are not doing  
this yet, but I can see it on the roadmap.



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Steve G
Sent: Monday, December 19, 2005 10:10 AM
To: cisco-voip at puck.nether.net; it at cimgroup.com
Subject: [cisco-voip] Re: cisco-voip Digest, Vol 34, Issue 96



Avidan,

      I am planning a slightly larger setup using the 3544 chassis  
with the 2PRI gateway blade.  I have 12 videoconferencing systems  
spread out over our WAN.  90 percent of them are Polycom VSX7000  
models.  I have heard the Tandbergs will integrate pretty seemlessly  
with CallManager, but I do not have any experience using them.  I  
have setup a gatekeeper which allows the H.323 Video endpoints  
(Polycom stuff) to call Skinny endpoints (Cisco VTA, 7985...etc).   
It's a pretty slick setup and not too complicated to set-up, as long  
as you have control over the CCM and gatekeepers.  The most difficult  
part might be coordinating with other groups if that is required.



We are also looking at the MCU blade for the 3540 chassis.



Steve



---------- Forwarded message ----------
From: "IT" <it at cimgroup.com >
To: <cisco-voip at puck.nether.net>
Date: Sun, 18 Dec 2005 18:34:54 -0800
Subject: [cisco-voip] Video conf. equipment for IP/VC

I am planning to setup a 3511 IP/VC to handle 4 branch offices and  
their video conference needs. I currently have a Polycom Viewstation  
EX, but with the upcoming IP/VC installation, I didnt know if buying  
3 more of those was the right idea. What is everyone using out there  
for video conferencing with their IP/VC? Tandbergs? Polycoms?  Does  
getting MeetingPlace really help in small conferences?

thanks for the advice.
avidan

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From wsisk at cisco.com  Thu Dec 29 16:21:34 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 16:21:45 2005
Subject: [cisco-voip] Call rings phones once and then drops
In-Reply-To: <a1b07ea60512171219p288d1552qddda2d4820ebba47@mail.gmail.com>
References: <a1b07ea60512171219p288d1552qddda2d4820ebba47@mail.gmail.com>
Message-ID: <5E5B3920-5B7B-4355-B036-1DBAB7C30FFA@cisco.com>

CallManager Locations out of bandwidth causes this message, amoung  
other things.  Check the CCM traces... (or next hop address off the  
destination dial peer)

/Wes
On Dec 17, 2005, at 3:19 PM, TechGuy wrote:

Here is the debug:

emb3845#debug isdn q931
debug isdn q931 is              ON.
emb3845#
*Dec 17 20:07:23.806: ISDN Se0/1/1:23 Q931: RX <- SETUP pd = 8   
callref = 0x00D2
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98381
                 Exclusive, Channel 1
         Calling Party Number i = 0x2183, '3375015555'
                 Plan:ISDN, Type:National
         Called Party Number i = 0x80, '4080'
                 Plan:Unknown, Type:Unknown
*Dec 17 20:07:24.830: ISDN Se0/1/1:23 Q931: TX -> RELEASE_COMP pd =  
8  callref = 0x80D2
         Cause i = 0x80A9 - Temporary failure
emb3845#


More info on the problem.  This router has two local PRI trunks  
coming in to a 2MFT card.  It also has two CAS (E&M Wink) trunks  
coming into a 2MFT card for long distance and incoming toll free.

The two CAS trunks work fine.  We are able to get incoming calls via  
the CAS trunks mentioned above with no problems.  Also, we have  
another gateway configured on the same callmanager in a similiar  
fashion and all works fine.  This was all working earlier this  
morning.  No known changes occured between when it worked and stopped  
working.

Quite an interesting problem, we have TAC on the line now but thought  
I would see what others think.
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From wsisk at cisco.com  Thu Dec 29 16:26:59 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 16:27:30 2005
Subject: [cisco-voip] Sync address book plugin
In-Reply-To: <2130867875-1134783712-cardhu_blackberry.rim.net-11521-@engine21-cell02>
References: <2130867875-1134783712-cardhu_blackberry.rim.net-11521-@engine21-cell02>
Message-ID: <AD8AA9C1-F847-4E08-9CFE-ED3C5942E102@cisco.com>

you can export outlook or outlook express or most anything to a  
format that can be imported to the windows address book.  True, not a  
real time synch, but still better than manual.

/Wes
On Dec 16, 2005, at 8:42 PM, James wrote:

What would be the best way to sync cm with outlook address book.  I  
ran the install and did the sync. But it sync the window address  
book.  And this do not get populated unless the user were using  
outlook express. Just looking for some deployment ideals
James D. Grace
System Engineer
CCNP CCNA MCSE MCDBA
Digitel Corp.
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From mmilanov at sezampro.yu  Thu Dec 29 16:41:59 2005
From: mmilanov at sezampro.yu (Mladen Milanovic)
Date: Thu Dec 29 16:38:22 2005
Subject: [cisco-voip] showing more than 500 users in general search
Message-ID: <200512292138.jBTLcEW0037319@puck.nether.net>

Hi Group,
I have a system with more than 1,500 users. When my client try to list all
users through ccmadmin page: Users>Global Directory and try to search with
blank he receive a message:
 
"Warning! Search too general: results truncated. 
Greater than 500 records matched, but only the first 500 will be displayed. 
Please refine your search. 

500 matching record(s)" 

 
Is there any way to enable showing all users. Client would like to use this
approach to verifying that all users are entered in the system ?
 
 
Thanks.
Mladen
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From Anthony.Mendoza at ca.com  Thu Dec 29 17:46:28 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Thu Dec 29 17:46:38 2005
Subject: [cisco-voip] Unity Not Recording :(
Message-ID: <D1F2DC1FDE5F8C4385B42578077F19576080C9@uswams21.ca.com>

So we fixed a routing issue we did have, but no go.  We've also tried
the MGCP flip flop via:

conf t
! our data interface
mgcp bind media source-interface FastEthernet0/1.2
mgcp bind control source-interface FastEthernet0/1.2
! our voice interface
mgcp bind media source-interface FastEthernet0/1.100
mgcp bind control source-interface FastEthernet0/1.100
end

That didn't help either.

-Anthony
Important Phone Numbers
North America Service Desk: +1 631-342-3955
Euro-Service Desk: +44 (0) 1753 243456
AP-Service Desk: +61 2 9937 0510
Web: http://servicedesk.ca.com/
(remember to mention you are a Niku Acquisition employee)
 

> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> Sent: Wednesday, December 28, 2005 5:41 PM
> To: Mendoza, Anthony; Cisco VoIP
> Subject: Re: [cisco-voip] Unity Not Recording :(
> 
> Sounds like a routing issue of some sort, or port
> blockage perhaps. When you dial in from PSTN and call
> hits Unity in office B can you hear the greetings, etc
> fine or is it just not recording or voice?
> 
> Any codec (region) differences between the 2 offices?
> Is VPN doing any firewall'ish/acl'ish stuff that may
> be effecting traffic? The second unity server have
> correct router(s) being used to get back to PSTN
> gateway?
> 
> When the call is connected, on gateway 'show call
> active voice brief' should show the PSTN and IP leg of
> call with IP address of remote host. Does this look
> right? Try pinging the remote IP address from gateways
> interface MGCP is sourced from. No reachability
> issues? It may be useful also to watch the call in the
> unity port status monitor and call viewer. 
> 
> With MGCP, try changing MGCP interface with bind
> statement. I had a similar strange issue once and
> bouncing MGCP didn't fix it but flip-flopping the MGCP
> source interface (mgcp bind...) between LAN and WAN
> interface and back again cleared it up (no mgcp/mgcp
> didn't fix).
> 
> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:
> 
> > All,  
> > 
> > We're running into a weird issue with a new Unity
> > install.  Here's the
> > basics:
> >  
> > Office A:
> > Unity 4.X
> > CCM 4.0 Cluster
> > Exchange Server
> > 15 channel PRI Gateway on 2651 Router
> >  
> > Office B:
> > Unity 4.X
> > Exchange Server
> >  
> > Office A and B connected via VPN Tunnel <100msec
> > latency
> >  
> > We're currently 100% functioning fine in Office A,
> > however, when I flip
> > VM profiles and point people to Office B's Unity
> > server I am unable to
> > record VM messages from the PSTN.  
> > 
> > * If I dial an extension of a phone located in
> > Office A from an internal
> > IP phone, it records fine to the Unity Server in
> > Office B.  The message
> > then gets dropped in the Exchange Inbox as you would
> > expect in Office B.
> > 
> > 
> > * If I call that same extension from PSTN it blows
> > and I get nothing.
> > In fact Unity stops me 3-4 secs into it since it
> > hears silence.
> >  
> > We already have a TAC issue open, but wanted to find
> > out if anyone here
> > had any ideas?
> > 
> > How can I make sure that when the PSTN call gets
> > routed to Unity it
> > shows up from #VOICE LAN IP# vs. #DATA LAN IP#
> > below?
> > 
> > Here's a sanitized version of our 2651 config for
> > reference:
> > !
> > ! Last configuration change at 23:12:31 UTC Wed Dec
> > 28 2005
> > ! NVRAM config last updated at 23:12:47 UTC Wed Dec
> > 28 2005
> > !
> > network-clock-participate wic 0
> > network-clock-select 1 E1 0/0
> > voice-card 0
> >  dspfarm
> >  no local-bypass
> > !
> > isdn switch-type primary-net5
> > !
> > !
> > voice rtp send-recv
> > 
> > !
> > ccm-manager redundant-host #PUBLISHER IP#
> > ccm-manager mgcp
> > ccm-manager music-on-hold
> > !
> > !
> > controller E1 0/0
> >  pri-group timeslots 1-16 service mgcp
> > !
> > class-map match-all control
> >  match access-group 102
> > class-map match-all voice
> >  match access-group 101
> > !
> > !
> > policy-map QoS
> >  class voice
> >   set cos 5
> >  class control
> >   set cos 3
> > !
> > !
> > !
> > !
> > interface FastEthernet0/0
> >  description uksagr00-f0/0/0
> >  ip address #USED WITH VPN# 255.255.255.252
> >  speed 100
> >  full-duplex
> > !
> > interface Serial0/0:15
> >  no ip address
> >  no logging event link-status
> >  isdn switch-type primary-net5
> >  isdn incoming-voice voice
> >  isdn bind-l3 ccm-manager
> >  no cdp enable
> > !
> > interface FastEthernet0/1
> >  description TRUNK TO 3548 LAN
> >  no ip address
> >  duplex auto
> >  speed auto
> > !
> > interface FastEthernet0/1.2
> >  encapsulation dot1Q 2 native
> >  ip address #DATA LAN IP# 255.255.254.0
> >  ip access-group UKSLUNITY1 out
> > !
> > interface FastEthernet0/1.100
> >  encapsulation dot1Q 100
> >  ip address #VOICE LAN IP# 255.255.254.0
> >  ip policy route-map VOIP_MONITOR
> > !
> > ip classless
> > ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> > ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN GW#
> > ip http server
> > !
> > ip access-list extended VOIP_MONITOR
> >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C LAN#
> > 0.0.0.255
> >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B LAN#
> > 0.0.0.63
> > ip access-list extended UKSLUNITY1
> >  permit ip any host #UNITY SERVER IN OFC B# log
> >  permit ip any any
> > !
> > route-map VOIP_MONITOR permit 10
> >  match ip address VOIP_MONITOR
> >  set ip next-hop #TO VPN GW#
> > !
> > control-plane
> > !
> > !
> > !
> > voice-port 0/0:15
> > !
> > mgcp
> > mgcp call-agent #subscriber IP# service-type mgcp
> > version 0.1
> > mgcp dtmf-relay voip codec all mode out-of-band
> > mgcp rtp unreachable timeout 1000 action notify
> > mgcp package-capability rtp-package
> > no mgcp package-capability res-package
> > mgcp package-capability sst-package
> > no mgcp package-capability fxr-package
> > no mgcp timer receive-rtcp
> > mgcp sdp simple
> > mgcp fax t38 inhibit
> > mgcp rtp payload-type g726r16 static
> > !
> > mgcp profile default
> > !
> > 
> > dial-peer voice 9 voip
> >  application mgcpapp
> >  destination-pattern 9T
> > !
> > ----
> > All ideas and opinions expressed are mine and only
> > mine not the company
> > I work for or that you see based on my email
> > address.  If I say anything
> > rude or offensive blame me not my employer.
> 
> 
> 
> 		
> __________________________________________ 
> Yahoo! DSL - Something to write home about. 
> Just $16.99/mo. or less. 
> dsl.yahoo.com 
> 
> 
> 


From Scott.Voll at wesd.org  Thu Dec 29 17:50:57 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu Dec 29 17:51:13 2005
Subject: [cisco-voip] 7920's
Message-ID: <D713462ED535184D830F6F6301753E7E01A12251@VISHNU.wesd.org>

I have a couple 7920 and have used them for a while.  Currently they are
just using WEP.  Since I'm in the process of upgrading all our wireless
I wanted to know if there is a better option?  WPA, TKIP, something
better then WEP.  Any comments?

 

Scott

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From James.O'Farrell at valuelink.co.uk  Thu Dec 29 18:08:28 2005
From: James.O'Farrell at valuelink.co.uk (James O'Farrell)
Date: Thu Dec 29 18:08:33 2005
Subject: [cisco-voip] Outbound Calls
Message-ID: <3A04A78705126A4098F7BE9F4673ED9D0190CCED@cypher.visdomain.valuelink.co.uk>

Dose anyone have any idea why after upgrading to IOS 12.4(3b) and CME
3.3 I am unable to call out on my BRI channels.

 

Incoming calls appear to be unaffected.

 

In a debug isdn q931 I appear to be getting this;

 

Dec 29 21:19:47.694: ISDN BR1/1 Q931: RX <- SETUP_ACK pd = 8  callref =
0xC6 

        Channel ID i = 0x89

Dec 29 21:19:47.702: ISDN BR1/1 **ERROR**: host_information: Unexpected
INFO - c

all id 0x8046

Dec 29 21:19:51.773: ISDN BR1/1 Q931: RX <- CALL_PROC pd = 8  callref =
0xC6

Dec 29 21:19:51.801: ISDN BR1/1 Q931: RX <- ALERTING pd = 8  callref =
0xC6 

        Progress Ind i = 0x8488 - In-band info or appropriate now
available 

Dec 29 21:19:51.805: ISDN BR1/1 **ERROR**: host_alerting: Unexpected
Alert clear

ing the call

Dec 29 21:19:51.805: %ISDN-4-ISDN_UNEXPECTED_EVENT: BR1/1: Occurred at
../isdn/i

sdnif_rxstate.c:6371

Dec 29 21:19:51.813: ISDN BR1/1 Q931: TX -> DISCONNECT pd = 8  callref =
0x46 

        Cause i = 0x8090 - Normal call clearing

 

>From the callers point of view the call comes back as "unknown number"
at the other end the person being called the phone rings once then
disconnects.

 

Any help/suggestions would be much appreciated.

 

TIA

 

JOF

 


ValueLink is a specialist service provider of financial data to the major UK and International centres. Our clients require reliable, accurate data with maximum coverage on a fixed time delayed basis. We work closely with each of our clients to provide a service which meets their specific requirements and maximises efficiency of their process. 

For further information visit our website @ www.valuelink.co.uk

CONFIDENTIALITY: The information in this e-mail and any attachment is confidential. It is intended only for the named recipient(s). If you are not a named recipient, please notify the sender immediately and do not read, use, copy or disseminate this information.

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From SPackett at fenwick.com  Thu Dec 29 18:21:58 2005
From: SPackett at fenwick.com (Stu Packett)
Date: Thu Dec 29 18:22:12 2005
Subject: [cisco-voip] 7920's
Message-ID: <965C0255FD5828438E34F9A68ECA76E740A6EF@svcexch02.firm.fenwick.llp>

Scott:
If your access points are Cisco, try out LEAP.  I use LEAP with all my
1200 APs.  On my switch and AP, I have separate VLANs for wireless data
and wireless voice.  The switch is trunked to pass only those VLANS.
The AP is configured for MAC address authentication.  If you have RSA
ACS, you can take advantage of two factor authentication as well.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, December 29, 2005 2:51 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7920's



I have a couple 7920 and have used them for a while.  Currently they are
just using WEP.  Since I'm in the process of upgrading all our wireless
I wanted to know if there is a better option?  WPA, TKIP, something
better then WEP.  Any comments?

 

Scott
--------------------------------------------------------

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

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From wsisk at cisco.com  Thu Dec 29 18:53:51 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 18:54:17 2005
Subject: [cisco-voip] Progress_ind command
In-Reply-To: <E0EF33751DEA634DA817A51024A8E3F2056CB0B5@usa0300ms01.na.xerox.net>
References: <E0EF33751DEA634DA817A51024A8E3F2056CB0B5@usa0300ms01.na.xerox.net>
Message-ID: <A38611F6-AD07-4D09-A69A-1E23BFC8034B@cisco.com>

As you are using an h323 gateway, I suspect the problem is more to do  
with dial-peer matching.  We have seen far more problems caused by  
incorrect peer matching.

here is a decent discussion of progress_ind
http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps5013/ 
products_feature_guide09186a0080080e2d.html#wp1015327

For dial-peer matching, every call through the router must match an  
ingress peer and an egress peer.  The VOIP peer you match (whether  
ingress or egress) must have your desired h245 dtmf signal option  
configured.  Keep in mind it could match the default peer (aka dial- 
peer 0) which would not negotiate DTMF relay.

dial-peer matching and dial-peer 0:
http://www.cisco.com/en/US/customer/tech/tk652/tk90/ 
technologies_tech_note09186a008010fed1.shtml


/Wes


On Dec 29, 2005, at 3:23 PM, DeRooy, Aaron H wrote:

Thanks. If a user complained about tones not getting thru, progress_ind
commands would not help unless the reason for no tones was that no audio
connection was established in the first place, correct? In that case,
how would I know what progress_ind command (setup, alert, etc.)to use?
Can you send something that explains it besides the command ref.?
Obviously, I don't know much about the use of this command but want to.
Thanks

-----Original Message-----
From: Wes Sisk [mailto:wsisk at cisco.com]
Sent: Thursday, December 29, 2005 1:15 PM
To: DeRooy, Aaron H
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Progress_ind command

progress_ind really has very little to do with DTMF.

progress_ind determine *WHEN* audio is cut through.  Though that audio
cut through (actually the associated h245 session) is a pre- requisite
for DTMF exchange, the 2 are not directly linked as you seem to imply.

/Wes

On Dec 28, 2005, at 12:36 PM, DeRooy, Aaron H wrote:



Does anybody know if ANY progress_ind command is needed to make sure
DTMF tones get thru the IP phone network to/from a PSTN destination?
I've been experimenting and find that the "dtmf-relay h245-alphanumeric"
command breaks DTMF tones but the progess_ind commands have no effect.
> From what I've been able to find out, progress_ind command solve
"ringback" problems with the "alert" key word but not sure about "setup"
and "progress" key works.
Thanks,

Aaron

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

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From it at cimgroup.com  Thu Dec 29 19:47:08 2005
From: it at cimgroup.com (IT)
Date: Thu Dec 29 19:43:03 2005
Subject: [cisco-voip] show incoming calls on a PRI
Message-ID: <B487D4C1FE6D04439061018C6EBC109B027091CD@exchange.cimgroup.com>

My telco says I have new phone numbers on my PRI, but when I dial them I
get a busy tone. The new extensions are 9XXX, and the telco is sending 4
digits. If I have them send 10 digits, do I need to create a translation
pattern in callmanager?

 

I use 9 for an outside line. How can I tell the calls are coming
through? Is there a debug command I can issue on my 2801 with the serial
PRI card?

 


******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
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From it at cimgroup.com  Thu Dec 29 19:56:55 2005
From: it at cimgroup.com (IT)
Date: Thu Dec 29 19:52:49 2005
Subject: [cisco-voip] Using extensions beginngin with a 9
Message-ID: <B487D4C1FE6D04439061018C6EBC109B027091D9@exchange.cimgroup.com>

I have problem.
People dial 9 to get an outside line, so I have a bunch of translation
patterns beginning with a 9.

Now I have people joining the systems whose last 4 digits of their
numbers are 9XXX.  My telco is sending 4 digits with the call, and it is
ringing busy. However, when the telco switches to sending me 7 digits,
and I change the extension on the phone to be the full 7 digits, the
calls come through.

How can I use 4 digits with these extensions beginning with a 9?

 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
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From ray at oneunified.net  Thu Dec 29 20:04:51 2005
From: ray at oneunified.net (Ray Burkholder)
Date: Thu Dec 29 20:05:36 2005
Subject: [cisco-voip] show incoming calls on a PRI
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B027091CD@exchange.cimgroup.com>
Message-ID: <00c101c60cdd$0cce24e0$640a0a0a@oneunified.local>

debug isdn q931

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Thursday, December 29, 2005 20:47
Cc: Cisco-Voip
Subject: [cisco-voip] show incoming calls on a PRI



My telco says I have new phone numbers on my PRI, but when I dial them I get
a busy tone. The new extensions are 9XXX, and the telco is sending 4 digits.
If I have them send 10 digits, do I need to create a translation pattern in
callmanager?

 

I use 9 for an outside line. How can I tell the calls are coming through? Is
there a debug command I can issue on my 2801 with the serial PRI card?

 

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 


-- 
Scanned for viruses & dangerous content at One Unified
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From wsisk at cisco.com  Thu Dec 29 20:38:04 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 20:38:17 2005
Subject: [cisco-voip] show incoming calls on a PRI
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B027091CD@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B027091CD@exchange.cimgroup.com>
Message-ID: <F429FE27-E94C-487D-BBD2-048F3B9A9816@cisco.com>

If your 2801 is MGCP controlled, the call is presented over PRI  
backhaul directly to CM. You just need to handle the call inside CM  
with closest match routing.

If your 2801 uses h323 configuration with dial-peers, you will need  
to configure closing matching dial-peers or get 10 digits and use  
translation-rule config on the gateway.

/Wes
On Dec 29, 2005, at 7:47 PM, IT wrote:

My telco says I have new phone numbers on my PRI, but when I dial  
them I get a busy tone. The new extensions are 9XXX, and the telco is  
sending 4 digits. If I have them send 10 digits, do I need to create  
a translation pattern in callmanager?



I use 9 for an outside line. How can I tell the calls are coming  
through? Is there a debug command I can issue on my 2801 with the  
serial PRI card?



******************************************************
This email may contain trade secrets and/or other information that is
proprietary and/or confidential. Use of this by anyone other than the
intended recipient(s) is prohibited and may be unlawful. If you are not
the intended recipient, delete the e-mail and notify us immediately.
******************************************************

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

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From wsisk at cisco.com  Thu Dec 29 20:40:24 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 20:40:36 2005
Subject: [cisco-voip] 7920's
In-Reply-To: <D713462ED535184D830F6F6301753E7E01A12251@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01A12251@VISHNU.wesd.org>
Message-ID: <115AEA4F-5467-41DA-8AF1-B4169B271804@cisco.com>

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/ 
english/wip7920/relnotes/rn201.htm#wp107591

/Wes
On Dec 29, 2005, at 5:50 PM, Voll, Scott wrote:

I have a couple 7920 and have used them for a while.  Currently they  
are just using WEP.  Since I?m in the process of upgrading all our  
wireless I wanted to know if there is a better option?  WPA, TKIP,  
something better then WEP.  Any comments?



Scott

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

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From wsisk at cisco.com  Thu Dec 29 20:44:50 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 20:45:04 2005
Subject: [cisco-voip] Outbound Calls
In-Reply-To: <3A04A78705126A4098F7BE9F4673ED9D0190CCED@cypher.visdomain.valuelink.co.uk>
References: <3A04A78705126A4098F7BE9F4673ED9D0190CCED@cypher.visdomain.valuelink.co.uk>
Message-ID: <49540EFE-CCAE-4854-BC83-1CB1CA437ED6@cisco.com>

looks very similar to CSCei81193.  Looks like we aren't processing  
the setup ACK and it goes down hill from there.  looks like it is  
fixed in interim versions but nothing avail on cisco.com.  prob  
better to roll back for now.

/Wes

On Dec 29, 2005, at 6:08 PM, James O'Farrell wrote:

Dose anyone have any idea why after upgrading to IOS 12.4(3b) and CME  
3.3 I am unable to call out on my BRI channels.



Incoming calls appear to be unaffected.



In a debug isdn q931 I appear to be getting this;



Dec 29 21:19:47.694: ISDN BR1/1 Q931: RX <- SETUP_ACK pd = 8  callref  
= 0xC6

         Channel ID i = 0x89

Dec 29 21:19:47.702: ISDN BR1/1 **ERROR**: host_information:  
Unexpected INFO - c

all id 0x8046

Dec 29 21:19:51.773: ISDN BR1/1 Q931: RX <- CALL_PROC pd = 8  callref  
= 0xC6

Dec 29 21:19:51.801: ISDN BR1/1 Q931: RX <- ALERTING pd = 8  callref  
= 0xC6

         Progress Ind i = 0x8488 - In-band info or appropriate now  
available

Dec 29 21:19:51.805: ISDN BR1/1 **ERROR**: host_alerting: Unexpected  
Alert clear

ing the call

Dec 29 21:19:51.805: %ISDN-4-ISDN_UNEXPECTED_EVENT: BR1/1: Occurred  
at ../isdn/i

sdnif_rxstate.c:6371

Dec 29 21:19:51.813: ISDN BR1/1 Q931: TX -> DISCONNECT pd = 8   
callref = 0x46

         Cause i = 0x8090 - Normal call clearing



 From the callers point of view the call comes back as ?unknown  
number? at the other end the person being called the phone rings once  
then disconnects.



Any help/suggestions would be much appreciated.



TIA



JOF



ValueLink is a specialist service provider of financial data to the  
major UK and International centres. Our clients require reliable,  
accurate data with maximum coverage on a fixed time delayed basis. We  
work closely with each of our clients to provide a service which  
meets their specific requirements and maximises efficiency of their  
process.
For further information visit our website @ www.valuelink.co.uk
CONFIDENTIALITY: The information in this e-mail and any attachment is  
confidential. It is intended only for the named recipient(s). If you  
are not a named recipient, please notify the sender immediately and  
do not read, use, copy or disseminate this information.
ValueLink Information Services Ltd accepts no liability whatsoever  
for any direct or consequential loss arising from the use, or  
reliance on, this e-mail or it's contents.
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

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From erickbe at yahoo.com  Thu Dec 29 20:57:17 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Dec 29 20:57:25 2005
Subject: [cisco-voip] Using extensions beginngin with a 9
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B027091D9@exchange.cimgroup.com>
Message-ID: <20051230015717.30168.qmail@web34313.mail.mud.yahoo.com>

Put the translation patterns with 9 in a partition,
and configure proper CSSs for outbound calls from
phones and then a CSS for inbound calls from the
gateway. The CSS for inbound calls from gateways
shouldn't have the partition used for the translation
patterns. Assign this CSS to the gateway for incoming
calls. This way only those partitions in that CSS are
usable for inbound calls on the gateway. 

--- IT <it at cimgroup.com> wrote:

> I have problem.
> People dial 9 to get an outside line, so I have a
> bunch of translation
> patterns beginning with a 9.
> 
> Now I have people joining the systems whose last 4
> digits of their
> numbers are 9XXX.  My telco is sending 4 digits with
> the call, and it is
> ringing busy. However, when the telco switches to
> sending me 7 digits,
> and I change the extension on the phone to be the
> full 7 digits, the
> calls come through.
> 
> How can I use 4 digits with these extensions
> beginning with a 9?
> 
>  
> 
>
******************************************************
> This email may contain trade secrets and/or other
> information that is 
> proprietary and/or confidential. Use of this by
> anyone other than the 
> intended recipient(s) is prohibited and may be
> unlawful. If you are not 
> the intended recipient, delete the e-mail and notify
> us immediately.
>
******************************************************
> 
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



		
__________________________________________ 
Yahoo! DSL ? Something to write home about. 
Just $16.99/mo. or less. 
dsl.yahoo.com 


From wsisk at cisco.com  Thu Dec 29 21:28:49 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 21:29:04 2005
Subject: [cisco-voip] showing more than 500 users in general search
In-Reply-To: <200512292138.jBTLcEW0037319@puck.nether.net>
References: <200512292138.jBTLcEW0037319@puck.nether.net>
Message-ID: <52C21C17-C70F-45C8-BB37-9B2A7191C7EC@cisco.com>

no way that I can see, looks like a tag was added through AXL to  
specify a large max search results.

you cold try increasing the search limit for corp directory when  
searching from the phone and search there to see how many are  
returned.  this is configured under system->enterprise parameters
enable all user search ->False.

NOTE: this will cause CPU and Disk I/O spikes on the CM server, do  
not leave it False on a system this large.

If that does not allow you to verify, then try this:
open cmd prompt
enter:
dumpall out.txt

this dumps entire contents of dcdirectory to a flat text file called  
out.txt

take that file to your favorite *nix box, or windows box with Unix  
bin ports and run:

grep "^    InternetAddress" test | wc -l

/Wes
On Dec 29, 2005, at 4:41 PM, Mladen Milanovic wrote:

Hi Group,
I have a system with more than 1,500 users. When my client try to  
list all users through ccmadmin page: Users>Global Directory and try  
to search with blank he receive a message:

"Warning! Search too general: results truncated.
Greater than 500 records matched, but only the first 500 will be  
displayed.
Please refine your search.

500 matching record(s)"

Is there any way to enable showing all users. Client would like to  
use this approach to verifying that all users are entered in the  
system ?


Thanks.
Mladen
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From erickbe at yahoo.com  Thu Dec 29 21:29:46 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu Dec 29 21:29:53 2005
Subject: [cisco-voip] Unity Not Recording :(
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F19576080C9@uswams21.ca.com>
Message-ID: <20051230022946.47530.qmail@web34309.mail.mud.yahoo.com>

Ok. How about the other things noted in previous
reply? All look good? The 'show call act voice brief'
or 'show call act voice' (more details) will show the
remote IP address (should be unity server) during you
have a call up with problem. 

Have you figured out if it is one-way audi

Have you looked at mgcp debugs on router/gateway
(debug mgcp packet) to see if that reveals anything?
Maybe a sniffer trace is needed. 

--- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:

> So we fixed a routing issue we did have, but no go. 
> We've also tried
> the MGCP flip flop via:
> 
> conf t
> ! our data interface
> mgcp bind media source-interface FastEthernet0/1.2
> mgcp bind control source-interface FastEthernet0/1.2
> ! our voice interface
> mgcp bind media source-interface FastEthernet0/1.100
> mgcp bind control source-interface
> FastEthernet0/1.100
> end
> 
> That didn't help either.
> 
> -Anthony
> Important Phone Numbers
> North America Service Desk: +1 631-342-3955
> Euro-Service Desk: +44 (0) 1753 243456
> AP-Service Desk: +61 2 9937 0510
> Web: http://servicedesk.ca.com/
> (remember to mention you are a Niku Acquisition
> employee)
>  
> 
> > -----Original Message-----
> > From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> > Sent: Wednesday, December 28, 2005 5:41 PM
> > To: Mendoza, Anthony; Cisco VoIP
> > Subject: Re: [cisco-voip] Unity Not Recording :(
> > 
> > Sounds like a routing issue of some sort, or port
> > blockage perhaps. When you dial in from PSTN and
> call
> > hits Unity in office B can you hear the greetings,
> etc
> > fine or is it just not recording or voice?
> > 
> > Any codec (region) differences between the 2
> offices?
> > Is VPN doing any firewall'ish/acl'ish stuff that
> may
> > be effecting traffic? The second unity server have
> > correct router(s) being used to get back to PSTN
> > gateway?
> > 
> > When the call is connected, on gateway 'show call
> > active voice brief' should show the PSTN and IP
> leg of
> > call with IP address of remote host. Does this
> look
> > right? Try pinging the remote IP address from
> gateways
> > interface MGCP is sourced from. No reachability
> > issues? It may be useful also to watch the call in
> the
> > unity port status monitor and call viewer. 
> > 
> > With MGCP, try changing MGCP interface with bind
> > statement. I had a similar strange issue once and
> > bouncing MGCP didn't fix it but flip-flopping the
> MGCP
> > source interface (mgcp bind...) between LAN and
> WAN
> > interface and back again cleared it up (no
> mgcp/mgcp
> > didn't fix).
> > 
> > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> wrote:
> > 
> > > All,  
> > > 
> > > We're running into a weird issue with a new
> Unity
> > > install.  Here's the
> > > basics:
> > >  
> > > Office A:
> > > Unity 4.X
> > > CCM 4.0 Cluster
> > > Exchange Server
> > > 15 channel PRI Gateway on 2651 Router
> > >  
> > > Office B:
> > > Unity 4.X
> > > Exchange Server
> > >  
> > > Office A and B connected via VPN Tunnel <100msec
> > > latency
> > >  
> > > We're currently 100% functioning fine in Office
> A,
> > > however, when I flip
> > > VM profiles and point people to Office B's Unity
> > > server I am unable to
> > > record VM messages from the PSTN.  
> > > 
> > > * If I dial an extension of a phone located in
> > > Office A from an internal
> > > IP phone, it records fine to the Unity Server in
> > > Office B.  The message
> > > then gets dropped in the Exchange Inbox as you
> would
> > > expect in Office B.
> > > 
> > > 
> > > * If I call that same extension from PSTN it
> blows
> > > and I get nothing.
> > > In fact Unity stops me 3-4 secs into it since it
> > > hears silence.
> > >  
> > > We already have a TAC issue open, but wanted to
> find
> > > out if anyone here
> > > had any ideas?
> > > 
> > > How can I make sure that when the PSTN call gets
> > > routed to Unity it
> > > shows up from #VOICE LAN IP# vs. #DATA LAN IP#
> > > below?
> > > 
> > > Here's a sanitized version of our 2651 config
> for
> > > reference:
> > > !
> > > ! Last configuration change at 23:12:31 UTC Wed
> Dec
> > > 28 2005
> > > ! NVRAM config last updated at 23:12:47 UTC Wed
> Dec
> > > 28 2005
> > > !
> > > network-clock-participate wic 0
> > > network-clock-select 1 E1 0/0
> > > voice-card 0
> > >  dspfarm
> > >  no local-bypass
> > > !
> > > isdn switch-type primary-net5
> > > !
> > > !
> > > voice rtp send-recv
> > > 
> > > !
> > > ccm-manager redundant-host #PUBLISHER IP#
> > > ccm-manager mgcp
> > > ccm-manager music-on-hold
> > > !
> > > !
> > > controller E1 0/0
> > >  pri-group timeslots 1-16 service mgcp
> > > !
> > > class-map match-all control
> > >  match access-group 102
> > > class-map match-all voice
> > >  match access-group 101
> > > !
> > > !
> > > policy-map QoS
> > >  class voice
> > >   set cos 5
> > >  class control
> > >   set cos 3
> > > !
> > > !
> > > !
> > > !
> > > interface FastEthernet0/0
> > >  description uksagr00-f0/0/0
> > >  ip address #USED WITH VPN# 255.255.255.252
> > >  speed 100
> > >  full-duplex
> > > !
> > > interface Serial0/0:15
> > >  no ip address
> > >  no logging event link-status
> > >  isdn switch-type primary-net5
> > >  isdn incoming-voice voice
> > >  isdn bind-l3 ccm-manager
> > >  no cdp enable
> > > !
> > > interface FastEthernet0/1
> > >  description TRUNK TO 3548 LAN
> > >  no ip address
> > >  duplex auto
> > >  speed auto
> > > !
> > > interface FastEthernet0/1.2
> > >  encapsulation dot1Q 2 native
> > >  ip address #DATA LAN IP# 255.255.254.0
> > >  ip access-group UKSLUNITY1 out
> > > !
> > > interface FastEthernet0/1.100
> > >  encapsulation dot1Q 100
> > >  ip address #VOICE LAN IP# 255.255.254.0
> > >  ip policy route-map VOIP_MONITOR
> > > !
> > > ip classless
> > > ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> > > ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
> GW#
> > > ip http server
> > > !
> > > ip access-list extended VOIP_MONITOR
> > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C LAN#
> > > 0.0.0.255
> > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B LAN#
> > > 0.0.0.63
> > > ip access-list extended UKSLUNITY1
> > >  permit ip any host #UNITY SERVER IN OFC B# log
> > >  permit ip any any
> > > !
> > > route-map VOIP_MONITOR permit 10
> > >  match ip address VOIP_MONITOR
> > >  set ip next-hop #TO VPN GW#
> > > !
> > > control-plane
> > > !
> > > !
> > > !
> > > voice-port 0/0:15
> > > !
> > > mgcp
> > > mgcp call-agent #subscriber IP# service-type
> mgcp
> > > version 0.1
> > > mgcp dtmf-relay voip codec all mode out-of-band
> > > mgcp rtp unreachable timeout 1000 action notify
> > > mgcp package-capability rtp-package
> > > no mgcp package-capability res-package
> > > mgcp package-capability sst-package
> > > no mgcp package-capability fxr-package
> > > no mgcp timer receive-rtcp
> > > mgcp sdp simple
> > > mgcp fax t38 inhibit
> > > mgcp rtp payload-type g726r16 static
> > > !
> > > mgcp profile default
> > > !
> > > 
> > > dial-peer voice 9 voip
> > >  application mgcpapp
> > >  destination-pattern 9T
> > > !
> > > ----
> > > All ideas and opinions expressed are mine and
> only
> > > mine not the company
> > > I work for or that you see based on my email
> > > address.  If I say anything
> > > rude or offensive blame me not my employer.
> > 
> > 
> > 
> > 		
> > __________________________________________ 
> > Yahoo! DSL - Something to write home about. 
> > Just $16.99/mo. or less. 
> > dsl.yahoo.com 
> > 
> > 
> > 
> 



		
__________________________________________ 
Yahoo! DSL ? Something to write home about. 
Just $16.99/mo. or less. 
dsl.yahoo.com 


From wsisk at cisco.com  Thu Dec 29 21:33:06 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 21:33:18 2005
Subject: [cisco-voip] Caller Name ID possible via Facility IE with H.323?
In-Reply-To: <12544185.1134698829510.JavaMail.bfeeny@mac.com>
References: <472D0CBC7B31C1499B77C1DAA3308B3B2D76C2@EXCHANGEDS01.ds.ad.adp.com>
	<12544185.1134698829510.JavaMail.bfeeny@mac.com>
Message-ID: <423CF026-6DEB-4A80-8F00-267987558C1B@cisco.com>

it was implemented for SIP, but not for h323 because of some  
interesting challenges with the way h323 calls are set up.

/Wes
On Dec 15, 2005, at 9:07 PM, Brian Feeny wrote:

Thanks, I have a case open, but I am wanting a clear explanation of  
the "isdn supp-service name calling", and how it applies, if at all  
to H.323.

I am going to try to get Hellsouth to switch to Display IE, but I am  
not holding my breath.

Brian

On Thursday, December 15, 2005, at 07:22PM, Bell, Joe  
<Joe_Bell at adp.com> wrote:

> Brian,
>
> Unless someone knows any pfm, there is no way to accomplish caller- 
> id on
> a PRI with H.323 unless your provider switches the data into  
> Display IE
> (good luck with that).  Recently (using 12.4(4T), I opened a TAC case
> about this issue to see if there was any way to accomplish this or  
> if it
> had been addressed in the new 12.4 train.  TAC sent me a clear answer
> saying H.323 and caller-id are unsupported and in order to get it, I
> needed to go to MGCP.
>
> Joe Bell
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Feeny
> Sent: Thursday, December 15, 2005 11:23 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Caller Name ID possible via Facility IE with
> H.323?
>
>
> I have read the thread:
>
> https://puck.nether.net/pipermail/cisco-voip/2005-March/003509.html
>
> And was curious as to the outcome.  I have PRI's coming in from a  
> 5ESS,
> and the carrier is sending the ID in the facility IE (inbound calls to
> us), and of course CM appears to want it in the display IE.  I have  
> read
> of ways CM would take it via the Facility ID if using MGCP AND  
> NI2/3 or
> MGCP AND QSIG.  Does any solution at all exist for H.323?  I have  
> tried
> to set my switch type for NI and QSIG, both work as far as
> placing/receiving calls, but I still cannot get inbound name id.  I  
> have
> tried to set "isdn supp-service name calling" on the gateway as  
> well but
> it doesn't seem to fix it.  Gateways are running late 12.3T code,  
> CM is
> 4.1(3)SR2.
>
> Sure I can try to battle it out with Bellsouth and ask them to  
> change to
> display IE, but I am sure they will not do it for me, I'll try though.
> But I am hoping there is a solution for those of us with H.323  
> links.  I
> could possibly get them to switch the circuit to NI2 or QSIG.  Like I
> said switching the switch type on the gateway works, without having  
> the
> carrier do anything probably because its just compatible, but not sure
> if formally having them change the switch emulation would do anything.
> The q.931 output is definitely more "friendly" with QSIG/NI, as it
> decodes the facility IE.  With 5ESS it doesn't like to receive the  
> name
> id facility ie stuff, it errors on it (but of course call completes).
>
> Brian
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> 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.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From wsisk at cisco.com  Thu Dec 29 21:35:33 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 21:35:46 2005
Subject: [cisco-voip] Cisco 7935 firmware issues
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B0260D981@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B0260D981@exchange.cimgroup.com>
Message-ID: <BE67AEF3-B9E2-4549-B3C2-00699A7FCBFC@cisco.com>

your phone is a brick and can only be fixed with a special hardware  
box, sorry.  RMA it.

sorry.  the firmware is erased and you are in the polycom equivalent  
of rommon.

/Wes

On Dec 14, 2005, at 7:35 PM, IT wrote:

Hello,

I have inherited a Cisco 7935 and I am running CM 4.0
I am having trouble getting the 7935 to pick up new firmware. I have a
tftp server running on my CM Publisher, and the device defaults are set
correctly for the firmware version. However, the screen only displays:
Station IP: 10.96.3.58
Contacting: 10.96.3.11 (my tftp server)

When I go to http://10.96.3.58 I get:
Wind Web Server Error Report:
Server Error: 501 Not Implemented
No RPM for this combination of URL and method
/dynaform/index_Dyna.htm

I am able to telnet to 10.96.3.58, but I get a username & password
prompt, but I do not know the password, and cisco/cisco doesn't work.

Does anyone know how else I can force flash this device?

On another note, I had to explicitly set the VLAN on my switch, as the
device was picking up the data vlan when connecting, shouldn't this
device auto register on the voice vlan?

Thanks,
Avidan

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proprietary and/or confidential. Use of this by anyone other than the
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From wsisk at cisco.com  Thu Dec 29 21:36:56 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 21:37:09 2005
Subject: [cisco-voip] getting translation patterns in CDRs
In-Reply-To: <002701c60273$339efd70$ca01a8c0@deadfish>
References: <002701c60273$339efd70$ca01a8c0@deadfish>
Message-ID: <1E5CF149-C0F5-451E-94D3-25BAE1E578AB@cisco.com>

does not affect CDR, it was put in for troubleshooting and some  
aspects of DNA.

no way to get the xlation patterns in CDR currently :(  I think you  
guys harassed Wade about this in CM General Session at CIPTUG.

/Wes
On Dec 16, 2005, at 2:02 PM, Lelio Fulgenzi wrote:

I'd like to get translations in my CDRs somehow. Whether they are two  
entries or one entry or whatever. Can anyone tell me if the Digit  
Analysis Complexity service parameter affects CDRs at all? And  
whether it puts too much of a strain on CallManager?

Thanks, Lelio

Digit Analysis Complexity :	This parameter allows the user to toggle  
between two modes; that is, StandardAnalysis and  
TranslationAndAlternatePatternAnalysis. The  
TranslationAndAlternatePatternAnalysis mode gives information about  
the Translation pattern and the Alternate matches in the CCM Traces  
while finding a match for a pattern. The valid values for this field  
follow:
-- StandardAnalysis 
-- TranslationAndAlternatePatternAnalysis 

This is a required field.
Default: StandardAnalysis.

------------------------------------------------------------------------ 
--------
Lelio Fulgenzi, B.A.
Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"I had a coffee and Coke at lunch today...and now, I've got more  
jitter than an
IP phone on a long haul 10base2  
connection"                                           LFJ
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From wsisk at cisco.com  Thu Dec 29 21:42:28 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 21:42:40 2005
Subject: [cisco-voip] CRS Application Trigger Question
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BCF05EAD@CH-EXCH-02.intelltech.apptis.inc>
References: <2F09B2162CCDEA4EA05C089B24FBC2BCF05EAD@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <05292BA1-1FC2-4731-AC6A-DA7330FA7C47@cisco.com>

associate the jtapi trigger with a cti port group / media group that  
only has your desired number of sessions available.  I'm pretty sure  
in my previous mis configurations i've recevied reorder from the  
trigger when one of these was not set correctly:

jtapi trigger max sessions
cisco media group number of channels
cti port groups number of cti ports

/Wes

On Dec 15, 2005, at 4:24 PM, VoIP Forum wrote:

What happens to an incoming call when the Maximum Sessions limit is  
reached on a trigger? The help docs don?t help too much.



Max Sessions

Maximum number of simultaneous instances of the associated  
application that can run when triggered by this trigger



I need it to send a busy tone back to the caller, but currently it  
just rings and rings.



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From wsisk at cisco.com  Thu Dec 29 21:43:29 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 21:43:41 2005
Subject: [cisco-voip] Speed dial on softkey
In-Reply-To: <62570315-1134679182-cardhu_blackberry.rim.net-3710-@engine21-cell02>
References: <62570315-1134679182-cardhu_blackberry.rim.net-3710-@engine21-cell02>
Message-ID: <EE3574AD-8D9E-426D-839B-E6CEEA6AC445@cisco.com>

Nope, but you could have a URL or URI assigned to a line button in  
latest versions of CM.

/Wes
On Dec 15, 2005, at 3:39 PM, James wrote:

    I have a question.    Is it posible to create a speed dial on the  
softkey button
James D. Grace
System Engineer
CCNP CCNA MCSE MCDBA
Digitel Corp.
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From wsisk at cisco.com  Thu Dec 29 21:51:09 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Thu Dec 29 21:51:22 2005
Subject: [cisco-voip] deciphering 7960 Phone's internal webpages
In-Reply-To: <f1b65f880512151116n46d4b9a3g@mail.gmail.com>
References: <f1b65f880512151116n46d4b9a3g@mail.gmail.com>
Message-ID: <04B751B5-FA8B-49E5-A7D7-BCFEF0C472F7@cisco.com>

This is the date, time, reason for the last phone reset.

If the phone crashed due to watchdog timeout or other abnormal  
processor activity, you would get a bunch of hex in there that is  
basically a stack dump.

Right now there is no external decoder for these, and not even an  
automated internal decoder.  But we use the data in phone reset cases  
to see what function crashed the phone.

/Wes
On Dec 15, 2005, at 2:16 PM, Justin Steinberg wrote:

Does anyone have any detailed information on what's available on the
7960's internal webpages?

Some things are obvious, but then there are some sections such as
'debug display' and 'stack statistics' that can be a little more
cryptic.
Example:
2005-12-12 15:07:45.220 Code:8108, P:0, S:12300900

There is also the streaming statistics section - can anyone explain
those timestamps and the jitter values.  I'm thinking the jitter value
might be:
average,max

TIA

Justin

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From lelio at uoguelph.ca  Thu Dec 29 22:45:13 2005
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu Dec 29 22:47:44 2005
Subject: [cisco-voip] getting translation patterns in CDRs
References: <002701c60273$339efd70$ca01a8c0@deadfish>
	<1E5CF149-C0F5-451E-94D3-25BAE1E578AB@cisco.com>
Message-ID: <009301c60cf3$ccabb870$c901a8c0@EATIUSBIRDIUS>

Oh yeah, I remember that question..wasn't from me though. But now I understand how it could be useful.
  ----- Original Message ----- 
  From: Wes Sisk 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Thursday, December 29, 2005 9:36 PM
  Subject: Re: [cisco-voip] getting translation patterns in CDRs


  does not affect CDR, it was put in for troubleshooting and some aspects of DNA.


  no way to get the xlation patterns in CDR currently :(  I think you guys harassed Wade about this in CM General Session at CIPTUG.


  /Wes

  On Dec 16, 2005, at 2:02 PM, Lelio Fulgenzi wrote:


  I'd like to get translations in my CDRs somehow. Whether they are two entries or one entry or whatever. Can anyone tell me if the Digit Analysis Complexity service parameter affects CDRs at all? And whether it puts too much of a strain on CallManager?

  Thanks, Lelio


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

        Digit Analysis Complexity : This parameter allows the user to toggle between two modes; that is, StandardAnalysis and TranslationAndAlternatePatternAnalysis. The TranslationAndAlternatePatternAnalysis mode gives information about the Translation pattern and the Alternate matches in the CCM Traces while finding a match for a pattern. The valid values for this field follow: 
        -- StandardAnalysis 
        -- TranslationAndAlternatePatternAnalysis 

        This is a required field.
        Default: StandardAnalysis.
       


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


  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Network Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I had a coffee and Coke at lunch today...and now, I've got more jitter than an
  IP phone on a long haul 10base2 connection"                                           LFJ
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip

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From Anthony.Mendoza at ca.com  Fri Dec 30 00:02:51 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Fri Dec 30 00:03:02 2005
Subject: [cisco-voip] Unity Not Recording :(
Message-ID: <D1F2DC1FDE5F8C4385B42578077F1957FF93@uswams21.ca.com>

Yes, the show call act voice brief showed the IP of our Unity server.. I would say that it is 1 way audio right now.  We hear the Unity server greetings and such fine, but it doesn't hear us when we try to record a message.  I'll try the debug mgcp packet next..is that cpu intensive?  e.g., do I need to worry about the router crashing and burning when running that debug?  I'm in California and this particular 2 sites are in the UK..  so it's not easy to get over there to reset it :(

thanks for your help.  Hopefully TAC will have some news for us tomorrow as well.  I'll have to wait until I get a guy in the office next week to attempt a sniffer trace.

-----Original Message-----
From:	Erick Bergquist [mailto:erickbe at yahoo.com]
Sent:	Thu 12/29/2005 6:29 PM
To:	Mendoza, Anthony; Cisco VoIP
Cc:	
Subject:	RE: [cisco-voip] Unity Not Recording :(

Ok. How about the other things noted in previous
reply? All look good? The 'show call act voice brief'
or 'show call act voice' (more details) will show the
remote IP address (should be unity server) during you
have a call up with problem. 

Have you figured out if it is one-way audi

Have you looked at mgcp debugs on router/gateway
(debug mgcp packet) to see if that reveals anything?
Maybe a sniffer trace is needed. 

--- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:

> So we fixed a routing issue we did have, but no go. 
> We've also tried
> the MGCP flip flop via:
> 
> conf t
> ! our data interface
> mgcp bind media source-interface FastEthernet0/1.2
> mgcp bind control source-interface FastEthernet0/1.2
> ! our voice interface
> mgcp bind media source-interface FastEthernet0/1.100
> mgcp bind control source-interface
> FastEthernet0/1.100
> end
> 
> That didn't help either.
> 
> -Anthony
> Important Phone Numbers
> North America Service Desk: +1 631-342-3955
> Euro-Service Desk: +44 (0) 1753 243456
> AP-Service Desk: +61 2 9937 0510
> Web: http://servicedesk.ca.com/
> (remember to mention you are a Niku Acquisition
> employee)
>  
> 
> > -----Original Message-----
> > From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> > Sent: Wednesday, December 28, 2005 5:41 PM
> > To: Mendoza, Anthony; Cisco VoIP
> > Subject: Re: [cisco-voip] Unity Not Recording :(
> > 
> > Sounds like a routing issue of some sort, or port
> > blockage perhaps. When you dial in from PSTN and
> call
> > hits Unity in office B can you hear the greetings,
> etc
> > fine or is it just not recording or voice?
> > 
> > Any codec (region) differences between the 2
> offices?
> > Is VPN doing any firewall'ish/acl'ish stuff that
> may
> > be effecting traffic? The second unity server have
> > correct router(s) being used to get back to PSTN
> > gateway?
> > 
> > When the call is connected, on gateway 'show call
> > active voice brief' should show the PSTN and IP
> leg of
> > call with IP address of remote host. Does this
> look
> > right? Try pinging the remote IP address from
> gateways
> > interface MGCP is sourced from. No reachability
> > issues? It may be useful also to watch the call in
> the
> > unity port status monitor and call viewer. 
> > 
> > With MGCP, try changing MGCP interface with bind
> > statement. I had a similar strange issue once and
> > bouncing MGCP didn't fix it but flip-flopping the
> MGCP
> > source interface (mgcp bind...) between LAN and
> WAN
> > interface and back again cleared it up (no
> mgcp/mgcp
> > didn't fix).
> > 
> > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> wrote:
> > 
> > > All,  
> > > 
> > > We're running into a weird issue with a new
> Unity
> > > install.  Here's the
> > > basics:
> > >  
> > > Office A:
> > > Unity 4.X
> > > CCM 4.0 Cluster
> > > Exchange Server
> > > 15 channel PRI Gateway on 2651 Router
> > >  
> > > Office B:
> > > Unity 4.X
> > > Exchange Server
> > >  
> > > Office A and B connected via VPN Tunnel <100msec
> > > latency
> > >  
> > > We're currently 100% functioning fine in Office
> A,
> > > however, when I flip
> > > VM profiles and point people to Office B's Unity
> > > server I am unable to
> > > record VM messages from the PSTN.  
> > > 
> > > * If I dial an extension of a phone located in
> > > Office A from an internal
> > > IP phone, it records fine to the Unity Server in
> > > Office B.  The message
> > > then gets dropped in the Exchange Inbox as you
> would
> > > expect in Office B.
> > > 
> > > 
> > > * If I call that same extension from PSTN it
> blows
> > > and I get nothing.
> > > In fact Unity stops me 3-4 secs into it since it
> > > hears silence.
> > >  
> > > We already have a TAC issue open, but wanted to
> find
> > > out if anyone here
> > > had any ideas?
> > > 
> > > How can I make sure that when the PSTN call gets
> > > routed to Unity it
> > > shows up from #VOICE LAN IP# vs. #DATA LAN IP#
> > > below?
> > > 
> > > Here's a sanitized version of our 2651 config
> for
> > > reference:
> > > !
> > > ! Last configuration change at 23:12:31 UTC Wed
> Dec
> > > 28 2005
> > > ! NVRAM config last updated at 23:12:47 UTC Wed
> Dec
> > > 28 2005
> > > !
> > > network-clock-participate wic 0
> > > network-clock-select 1 E1 0/0
> > > voice-card 0
> > >  dspfarm
> > >  no local-bypass
> > > !
> > > isdn switch-type primary-net5
> > > !
> > > !
> > > voice rtp send-recv
> > > 
> > > !
> > > ccm-manager redundant-host #PUBLISHER IP#
> > > ccm-manager mgcp
> > > ccm-manager music-on-hold
> > > !
> > > !
> > > controller E1 0/0
> > >  pri-group timeslots 1-16 service mgcp
> > > !
> > > class-map match-all control
> > >  match access-group 102
> > > class-map match-all voice
> > >  match access-group 101
> > > !
> > > !
> > > policy-map QoS
> > >  class voice
> > >   set cos 5
> > >  class control
> > >   set cos 3
> > > !
> > > !
> > > !
> > > !
> > > interface FastEthernet0/0
> > >  description uksagr00-f0/0/0
> > >  ip address #USED WITH VPN# 255.255.255.252
> > >  speed 100
> > >  full-duplex
> > > !
> > > interface Serial0/0:15
> > >  no ip address
> > >  no logging event link-status
> > >  isdn switch-type primary-net5
> > >  isdn incoming-voice voice
> > >  isdn bind-l3 ccm-manager
> > >  no cdp enable
> > > !
> > > interface FastEthernet0/1
> > >  description TRUNK TO 3548 LAN
> > >  no ip address
> > >  duplex auto
> > >  speed auto
> > > !
> > > interface FastEthernet0/1.2
> > >  encapsulation dot1Q 2 native
> > >  ip address #DATA LAN IP# 255.255.254.0
> > >  ip access-group UKSLUNITY1 out
> > > !
> > > interface FastEthernet0/1.100
> > >  encapsulation dot1Q 100
> > >  ip address #VOICE LAN IP# 255.255.254.0
> > >  ip policy route-map VOIP_MONITOR
> > > !
> > > ip classless
> > > ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> > > ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
> GW#
> > > ip http server
> > > !
> > > ip access-list extended VOIP_MONITOR
> > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C LAN#
> > > 0.0.0.255
> > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B LAN#
> > > 0.0.0.63
> > > ip access-list extended UKSLUNITY1
> > >  permit ip any host #UNITY SERVER IN OFC B# log
> > >  permit ip any any
> > > !
> > > route-map VOIP_MONITOR permit 10
> > >  match ip address VOIP_MONITOR
> > >  set ip next-hop #TO VPN GW#
> > > !
> > > control-plane
> > > !
> > > !
> > > !
> > > voice-port 0/0:15
> > > !
> > > mgcp
> > > mgcp call-agent #subscriber IP# service-type
> mgcp
> > > version 0.1
> > > mgcp dtmf-relay voip codec all mode out-of-band
> > > mgcp rtp unreachable timeout 1000 action notify
> > > mgcp package-capability rtp-package
> > > no mgcp package-capability res-package
> > > mgcp package-capability sst-package
> > > no mgcp package-capability fxr-package
> > > no mgcp timer receive-rtcp
> > > mgcp sdp simple
> > > mgcp fax t38 inhibit
> > > mgcp rtp payload-type g726r16 static
> > > !
> > > mgcp profile default
> > > !
> > > 
> > > dial-peer voice 9 voip
> > >  application mgcpapp
> > >  destination-pattern 9T
> > > !
> > > ----
> > > All ideas and opinions expressed are mine and
> only
> > > mine not the company
> > > I work for or that you see based on my email
> > > address.  If I say anything
> > > rude or offensive blame me not my employer.
> > 
> > 
> > 
> > 		
> > __________________________________________ 
> > Yahoo! DSL - Something to write home about. 
> > Just $16.99/mo. or less. 
> > dsl.yahoo.com 
> > 
> > 
> > 
> 



		
__________________________________________ 
Yahoo! DSL ? Something to write home about. 
Just $16.99/mo. or less. 
dsl.yahoo.com 




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From erickbe at yahoo.com  Fri Dec 30 00:45:01 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Dec 30 00:45:08 2005
Subject: [cisco-voip] Unity Not Recording :(
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F1957FF93@uswams21.ca.com>
Message-ID: <20051230054501.79705.qmail@web34308.mail.mud.yahoo.com>

Ok. Hmm. Does unity respond to key pressses for
options on call handlers, dial-by-name, conversations,
etc fine? 

Do regular phone calls from this gateway to IP phones
at that site on subnet of unity server work fine? 

What version of Unity TSP?
One or Two NICs on Unity server?

That debug is not that intensive... unless theres a
lot of call setup or MGCP activity going on. You can
do it to the buffer or a syslog server to. Only need
to run it for period call comes in and connects to
unity then can stop debug. But now that you stated
it's one way audio and it shows the correct IP for
remote address, chances are the MGCP call
setup/audio-cutover is OK. 

As for sniffer traces, you may have ms network monitor
installed on the server you can use to get a trace
from unity server side of things. You can also pull up
the port status monitor tool under tools depot to see
if ports are seeing DTMF presses if you can't do
dial-by-name, etc above.


--- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:

> Yes, the show call act voice brief showed the IP of
> our Unity server.. I would say that it is 1 way
> audio right now.  We hear the Unity server greetings
> and such fine, but it doesn't hear us when we try to
> record a message.  I'll try the debug mgcp packet
> next..is that cpu intensive?  e.g., do I need to
> worry about the router crashing and burning when
> running that debug?  I'm in California and this
> particular 2 sites are in the UK..  so it's not easy
> to get over there to reset it :(
> 
> thanks for your help.  Hopefully TAC will have some
> news for us tomorrow as well.  I'll have to wait
> until I get a guy in the office next week to attempt
> a sniffer trace.
> 
> -----Original Message-----
> From:	Erick Bergquist [mailto:erickbe at yahoo.com]
> Sent:	Thu 12/29/2005 6:29 PM
> To:	Mendoza, Anthony; Cisco VoIP
> Cc:	
> Subject:	RE: [cisco-voip] Unity Not Recording :(
> 
> Ok. How about the other things noted in previous
> reply? All look good? The 'show call act voice
> brief'
> or 'show call act voice' (more details) will show
> the
> remote IP address (should be unity server) during
> you
> have a call up with problem. 
> 
> Have you figured out if it is one-way audi
> 
> Have you looked at mgcp debugs on router/gateway
> (debug mgcp packet) to see if that reveals anything?
> Maybe a sniffer trace is needed. 
> 
> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> wrote:
> 
> > So we fixed a routing issue we did have, but no
> go. 
> > We've also tried
> > the MGCP flip flop via:
> > 
> > conf t
> > ! our data interface
> > mgcp bind media source-interface FastEthernet0/1.2
> > mgcp bind control source-interface
> FastEthernet0/1.2
> > ! our voice interface
> > mgcp bind media source-interface
> FastEthernet0/1.100
> > mgcp bind control source-interface
> > FastEthernet0/1.100
> > end
> > 
> > That didn't help either.
> > 
> > -Anthony
> > Important Phone Numbers
> > North America Service Desk: +1 631-342-3955
> > Euro-Service Desk: +44 (0) 1753 243456
> > AP-Service Desk: +61 2 9937 0510
> > Web: http://servicedesk.ca.com/
> > (remember to mention you are a Niku Acquisition
> > employee)
> >  
> > 
> > > -----Original Message-----
> > > From: Erick Bergquist [mailto:erickbe at yahoo.com]
> 
> > > Sent: Wednesday, December 28, 2005 5:41 PM
> > > To: Mendoza, Anthony; Cisco VoIP
> > > Subject: Re: [cisco-voip] Unity Not Recording :(
> > > 
> > > Sounds like a routing issue of some sort, or
> port
> > > blockage perhaps. When you dial in from PSTN and
> > call
> > > hits Unity in office B can you hear the
> greetings,
> > etc
> > > fine or is it just not recording or voice?
> > > 
> > > Any codec (region) differences between the 2
> > offices?
> > > Is VPN doing any firewall'ish/acl'ish stuff that
> > may
> > > be effecting traffic? The second unity server
> have
> > > correct router(s) being used to get back to PSTN
> > > gateway?
> > > 
> > > When the call is connected, on gateway 'show
> call
> > > active voice brief' should show the PSTN and IP
> > leg of
> > > call with IP address of remote host. Does this
> > look
> > > right? Try pinging the remote IP address from
> > gateways
> > > interface MGCP is sourced from. No reachability
> > > issues? It may be useful also to watch the call
> in
> > the
> > > unity port status monitor and call viewer. 
> > > 
> > > With MGCP, try changing MGCP interface with bind
> > > statement. I had a similar strange issue once
> and
> > > bouncing MGCP didn't fix it but flip-flopping
> the
> > MGCP
> > > source interface (mgcp bind...) between LAN and
> > WAN
> > > interface and back again cleared it up (no
> > mgcp/mgcp
> > > didn't fix).
> > > 
> > > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> > wrote:
> > > 
> > > > All,  
> > > > 
> > > > We're running into a weird issue with a new
> > Unity
> > > > install.  Here's the
> > > > basics:
> > > >  
> > > > Office A:
> > > > Unity 4.X
> > > > CCM 4.0 Cluster
> > > > Exchange Server
> > > > 15 channel PRI Gateway on 2651 Router
> > > >  
> > > > Office B:
> > > > Unity 4.X
> > > > Exchange Server
> > > >  
> > > > Office A and B connected via VPN Tunnel
> <100msec
> > > > latency
> > > >  
> > > > We're currently 100% functioning fine in
> Office
> > A,
> > > > however, when I flip
> > > > VM profiles and point people to Office B's
> Unity
> > > > server I am unable to
> > > > record VM messages from the PSTN.  
> > > > 
> > > > * If I dial an extension of a phone located in
> > > > Office A from an internal
> > > > IP phone, it records fine to the Unity Server
> in
> > > > Office B.  The message
> > > > then gets dropped in the Exchange Inbox as you
> > would
> > > > expect in Office B.
> > > > 
> > > > 
> > > > * If I call that same extension from PSTN it
> > blows
> > > > and I get nothing.
> > > > In fact Unity stops me 3-4 secs into it since
> it
> > > > hears silence.
> > > >  
> > > > We already have a TAC issue open, but wanted
> to
> > find
> > > > out if anyone here
> > > > had any ideas?
> > > > 
> > > > How can I make sure that when the PSTN call
> gets
> > > > routed to Unity it
> > > > shows up from #VOICE LAN IP# vs. #DATA LAN IP#
> > > > below?
> > > > 
> > > > Here's a sanitized version of our 2651 config
> > for
> > > > reference:
> > > > !
> > > > ! Last configuration change at 23:12:31 UTC
> Wed
> > Dec
> > > > 28 2005
> > > > ! NVRAM config last updated at 23:12:47 UTC
> Wed
> > Dec
> > > > 28 2005
> > > > !
> > > > network-clock-participate wic 0
> > > > network-clock-select 1 E1 0/0
> > > > voice-card 0
> > > >  dspfarm
> > > >  no local-bypass
> > > > !
> > > > isdn switch-type primary-net5
> > > > !
> > > > !
> > > > voice rtp send-recv
> > > > 
> > > > !
> > > > ccm-manager redundant-host #PUBLISHER IP#
> > > > ccm-manager mgcp
> > > > ccm-manager music-on-hold
> > > > !
> > > > !
> > > > controller E1 0/0
> > > >  pri-group timeslots 1-16 service mgcp
> > > > !
> > > > class-map match-all control
> > > >  match access-group 102
> > > > class-map match-all voice
> > > >  match access-group 101
> > > > !
> > > > !
> > > > policy-map QoS
> > > >  class voice
> > > >   set cos 5
> > > >  class control
> > > >   set cos 3
> > > > !
> > > > !
> > > > !
> > > > !
> > > > interface FastEthernet0/0
> > > >  description uksagr00-f0/0/0
> > > >  ip address #USED WITH VPN# 255.255.255.252
> > > >  speed 100
> > > >  full-duplex
> > > > !
> > > > interface Serial0/0:15
> > > >  no ip address
> > > >  no logging event link-status
> > > >  isdn switch-type primary-net5
> > > >  isdn incoming-voice voice
> > > >  isdn bind-l3 ccm-manager
> > > >  no cdp enable
> > > > !
> > > > interface FastEthernet0/1
> > > >  description TRUNK TO 3548 LAN
> > > >  no ip address
> > > >  duplex auto
> > > >  speed auto
> > > > !
> > > > interface FastEthernet0/1.2
> > > >  encapsulation dot1Q 2 native
> > > >  ip address #DATA LAN IP# 255.255.254.0
> > > >  ip access-group UKSLUNITY1 out
> > > > !
> > > > interface FastEthernet0/1.100
> > > >  encapsulation dot1Q 100
> > > >  ip address #VOICE LAN IP# 255.255.254.0
> > > >  ip policy route-map VOIP_MONITOR
> > > > !
> > > > ip classless
> > > > ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> > > > ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
> > GW#
> > > > ip http server
> > > > !
> > > > ip access-list extended VOIP_MONITOR
> > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C
> LAN#
> > > > 0.0.0.255
> > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B
> LAN#
> > > > 0.0.0.63
> > > > ip access-list extended UKSLUNITY1
> > > >  permit ip any host #UNITY SERVER IN OFC B#
> log
> > > >  permit ip any any
> > > > !
> > > > route-map VOIP_MONITOR permit 10
> > > >  match ip address VOIP_MONITOR
> > > >  set ip next-hop #TO VPN GW#
> > > > !
> > > > control-plane
> > > > !
> > > > !
> > > > !
> > > > voice-port 0/0:15
> > > > !
> > > > mgcp
> > > > mgcp call-agent #subscriber IP# service-type
> > mgcp
> > > > version 0.1
> > > > mgcp dtmf-relay voip codec all mode
> out-of-band
> > > > mgcp rtp unreachable timeout 1000 action
> notify
> > > > mgcp package-capability rtp-package
> > > > no mgcp package-capability res-package
> > > > mgcp package-capability sst-package
> > > > no mgcp package-capability fxr-package
> > > > no mgcp timer receive-rtcp
> > > > mgcp sdp simple
> > > > mgcp fax t38 inhibit
> > > > mgcp rtp payload-type g726r16 static
> > > > !
> > > > mgcp profile default
> > > > !
> > > > 
> > > > dial-peer voice 9 voip
> > > >  application mgcpapp
> > > >  destination-pattern 9T
> > > > !
> > > > ----
> > > > All ideas and opinions expressed are mine and
> > only
> > > > mine not the company
> > > > I work for or that you see based on my email
> > > > address.  If I say anything
> > > > rude or offensive blame me not my employer.



		
__________________________________________ 
Yahoo! DSL ? Something to write home about. 
Just $16.99/mo. or less. 
dsl.yahoo.com 


From wsisk at cisco.com  Fri Dec 30 08:29:05 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Fri Dec 30 08:29:17 2005
Subject: [cisco-voip] showing more than 500 users in general search
In-Reply-To: <200512292138.jBTLcEW0037319@puck.nether.net>
References: <200512292138.jBTLcEW0037319@puck.nether.net>
Message-ID: <5BD7043B-4CB1-49B5-A853-CED508D268A0@cisco.com>

another unsupported but viable suggestion:  change the limit of 500:

ciscoCCNatMaxSearchResults attribute. The SystemProfile entry under  
the CiscoBase will contain this attribute.

You will have to go into DC Directory Manager to modify this attrib.

/Wes

On Dec 29, 2005, at 4:41 PM, Mladen Milanovic wrote:

Hi Group,
I have a system with more than 1,500 users. When my client try to  
list all users through ccmadmin page: Users>Global Directory and try  
to search with blank he receive a message:

"Warning! Search too general: results truncated.
Greater than 500 records matched, but only the first 500 will be  
displayed.
Please refine your search.

500 matching record(s)"

Is there any way to enable showing all users. Client would like to  
use this approach to verifying that all users are entered in the  
system ?


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

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From mfa at crec.ifas.ufl.edu  Fri Dec 30 10:58:30 2005
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Fri Dec 30 10:58:39 2005
Subject: [cisco-voip] Leap second unaccounted for 
References: <200512300503.jBU535kI097648@puck.nether.net>
Message-ID: <001f01c60d59$e217aa50$0402a8c0@crec.ifas.ufl.edu>

This is probably way down everyone's list of things to worry about, but 
December 31, 2005 is actually 86401 seconds long, since this year NIST 
sneaks in a leap second at midnight.  (The official clock goes from 23:59:59 
to 23:59:60 to 00:00:00.)  TOD will eventually self-correct, since most of 
us synchronize periodically with a NIST source, but anyone interested is 
precise interval measurement including this period should be careful of the 
time references used.  Excel's time functions, for example, do not appear to 
take this into account.  If anyone places a long-distance call to wish a 
Happy New Year in real time and CDR data is used for billing, the odds are 
we'll all give away that second without noticing.  Or caring, most likely...

Happy New Year,

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 


From Scott.Voll at wesd.org  Fri Dec 30 11:04:55 2005
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri Dec 30 11:05:06 2005
Subject: [cisco-voip] CRS Application Trigger Question
Message-ID: <D713462ED535184D830F6F6301753E7E01A12259@VISHNU.wesd.org>

How does it work if you have 7 different triggers / media groups / CTI
groups setup like this:

 

1

2

6

10

10

5

10

 

And you're licensed for 24 IVR ports.  If I were to have 24 calls over
the 7 groups it would busy out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Thursday, December 29, 2005 6:42 PM
To: VoIP Forum
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CRS Application Trigger Question

 

associate the jtapi trigger with a cti port group / media group that
only has your desired number of sessions available.  I'm pretty sure in
my previous mis configurations i've recevied reorder from the trigger
when one of these was not set correctly:

 

jtapi trigger max sessions

cisco media group number of channels

cti port groups number of cti ports

 

/Wes

 

On Dec 15, 2005, at 4:24 PM, VoIP Forum wrote:

 

What happens to an incoming call when the Maximum Sessions limit is
reached on a trigger? The help docs don't help too much.

 

Max Sessions 

Maximum number of simultaneous instances of the associated application
that can run when triggered by this trigger

 

I need it to send a busy tone back to the caller, but currently it just
rings and rings.

 

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

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

 

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From Anthony.Mendoza at ca.com  Fri Dec 30 11:08:43 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Fri Dec 30 11:08:48 2005
Subject: [cisco-voip] Unity Not Recording :(
Message-ID: <D1F2DC1FDE5F8C4385B42578077F1957FF94@uswams21.ca.com>

[Comments Below]

> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> Sent: Thursday, December 29, 2005 9:45 PM
> To: Mendoza, Anthony; Cisco VoIP
> Subject: RE: [cisco-voip] Unity Not Recording :(
> 
> Ok. Hmm. Does unity respond to key pressses for
> options on call handlers, dial-by-name, conversations,
> etc fine? 
Sure does, which is even more annoying.  If I'm logged into a mailbox, I
can choose to send a message, choosing the extension via dial by name,
but when I try to record a message, nothing...  same as when I try to
call an extension directly..

> Do regular phone calls from this gateway to IP phones
> at that site on subnet of unity server work fine? 
There are no phones in Office B on the Unity Subnet.

> 
> What version of Unity TSP?
Cisco Unity TSP version = 7.0(4)59)        

> One or Two NICs on Unity server?
2 NICs, only 1 enabled

> That debug is not that intensive... unless theres a
> lot of call setup or MGCP activity going on. You can
> do it to the buffer or a syslog server to. Only need
> to run it for period call comes in and connects to
> unity then can stop debug. But now that you stated
> it's one way audio and it shows the correct IP for
> remote address, chances are the MGCP call
> setup/audio-cutover is OK. 
Given your assumption above I didn't do the mgcp logging.

> 
> As for sniffer traces, you may have ms network monitor
> installed on the server you can use to get a trace
> from unity server side of things. You can also pull up
> the port status monitor tool under tools depot to see
> if ports are seeing DTMF presses if you can't do
> dial-by-name, etc above.
The port registers all the input I press...just not my voice...very
strange I say..  are the DTMF tones sent on a different channel or
protocol than voice?


> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:
> 
> > Yes, the show call act voice brief showed the IP of
> > our Unity server.. I would say that it is 1 way
> > audio right now.  We hear the Unity server greetings
> > and such fine, but it doesn't hear us when we try to
> > record a message.  I'll try the debug mgcp packet
> > next..is that cpu intensive?  e.g., do I need to
> > worry about the router crashing and burning when
> > running that debug?  I'm in California and this
> > particular 2 sites are in the UK..  so it's not easy
> > to get over there to reset it :(
> > 
> > thanks for your help.  Hopefully TAC will have some
> > news for us tomorrow as well.  I'll have to wait
> > until I get a guy in the office next week to attempt
> > a sniffer trace.
> > 
> > -----Original Message-----
> > From:	Erick Bergquist [mailto:erickbe at yahoo.com]
> > Sent:	Thu 12/29/2005 6:29 PM
> > To:	Mendoza, Anthony; Cisco VoIP
> > Cc:	
> > Subject:	RE: [cisco-voip] Unity Not Recording :(
> > 
> > Ok. How about the other things noted in previous
> > reply? All look good? The 'show call act voice
> > brief'
> > or 'show call act voice' (more details) will show
> > the
> > remote IP address (should be unity server) during
> > you
> > have a call up with problem. 
> > 
> > Have you figured out if it is one-way audi
> > 
> > Have you looked at mgcp debugs on router/gateway
> > (debug mgcp packet) to see if that reveals anything?
> > Maybe a sniffer trace is needed. 
> > 
> > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> > wrote:
> > 
> > > So we fixed a routing issue we did have, but no
> > go. 
> > > We've also tried
> > > the MGCP flip flop via:
> > > 
> > > conf t
> > > ! our data interface
> > > mgcp bind media source-interface FastEthernet0/1.2
> > > mgcp bind control source-interface
> > FastEthernet0/1.2
> > > ! our voice interface
> > > mgcp bind media source-interface
> > FastEthernet0/1.100
> > > mgcp bind control source-interface
> > > FastEthernet0/1.100
> > > end
> > > 
> > > That didn't help either.
> > > 
> > > -Anthony
> > >  
> > > 
> > > > -----Original Message-----
> > > > From: Erick Bergquist [mailto:erickbe at yahoo.com]
> > 
> > > > Sent: Wednesday, December 28, 2005 5:41 PM
> > > > To: Mendoza, Anthony; Cisco VoIP
> > > > Subject: Re: [cisco-voip] Unity Not Recording :(
> > > > 
> > > > Sounds like a routing issue of some sort, or
> > port
> > > > blockage perhaps. When you dial in from PSTN and
> > > call
> > > > hits Unity in office B can you hear the
> > greetings,
> > > etc
> > > > fine or is it just not recording or voice?
> > > > 
> > > > Any codec (region) differences between the 2
> > > offices?
> > > > Is VPN doing any firewall'ish/acl'ish stuff that
> > > may
> > > > be effecting traffic? The second unity server
> > have
> > > > correct router(s) being used to get back to PSTN
> > > > gateway?
> > > > 
> > > > When the call is connected, on gateway 'show
> > call
> > > > active voice brief' should show the PSTN and IP
> > > leg of
> > > > call with IP address of remote host. Does this
> > > look
> > > > right? Try pinging the remote IP address from
> > > gateways
> > > > interface MGCP is sourced from. No reachability
> > > > issues? It may be useful also to watch the call
> > in
> > > the
> > > > unity port status monitor and call viewer. 
> > > > 
> > > > With MGCP, try changing MGCP interface with bind
> > > > statement. I had a similar strange issue once
> > and
> > > > bouncing MGCP didn't fix it but flip-flopping
> > the
> > > MGCP
> > > > source interface (mgcp bind...) between LAN and
> > > WAN
> > > > interface and back again cleared it up (no
> > > mgcp/mgcp
> > > > didn't fix).
> > > > 
> > > > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> > > wrote:
> > > > 
> > > > > All,  
> > > > > 
> > > > > We're running into a weird issue with a new
> > > Unity
> > > > > install.  Here's the
> > > > > basics:
> > > > >  
> > > > > Office A:
> > > > > Unity 4.X
> > > > > CCM 4.0 Cluster
> > > > > Exchange Server
> > > > > 15 channel PRI Gateway on 2651 Router
> > > > >  
> > > > > Office B:
> > > > > Unity 4.X
> > > > > Exchange Server
> > > > >  
> > > > > Office A and B connected via VPN Tunnel
> > <100msec
> > > > > latency
> > > > >  
> > > > > We're currently 100% functioning fine in
> > Office
> > > A,
> > > > > however, when I flip
> > > > > VM profiles and point people to Office B's
> > Unity
> > > > > server I am unable to
> > > > > record VM messages from the PSTN.  
> > > > > 
> > > > > * If I dial an extension of a phone located in
> > > > > Office A from an internal
> > > > > IP phone, it records fine to the Unity Server
> > in
> > > > > Office B.  The message
> > > > > then gets dropped in the Exchange Inbox as you
> > > would
> > > > > expect in Office B.
> > > > > 
> > > > > 
> > > > > * If I call that same extension from PSTN it
> > > blows
> > > > > and I get nothing.
> > > > > In fact Unity stops me 3-4 secs into it since
> > it
> > > > > hears silence.
> > > > >  
> > > > > We already have a TAC issue open, but wanted
> > to
> > > find
> > > > > out if anyone here
> > > > > had any ideas?
> > > > > 
> > > > > How can I make sure that when the PSTN call
> > gets
> > > > > routed to Unity it
> > > > > shows up from #VOICE LAN IP# vs. #DATA LAN IP#
> > > > > below?
> > > > > 
> > > > > Here's a sanitized version of our 2651 config
> > > for
> > > > > reference:
> > > > > !
> > > > > ! Last configuration change at 23:12:31 UTC
> > Wed
> > > Dec
> > > > > 28 2005
> > > > > ! NVRAM config last updated at 23:12:47 UTC
> > Wed
> > > Dec
> > > > > 28 2005
> > > > > !
> > > > > network-clock-participate wic 0
> > > > > network-clock-select 1 E1 0/0
> > > > > voice-card 0
> > > > >  dspfarm
> > > > >  no local-bypass
> > > > > !
> > > > > isdn switch-type primary-net5
> > > > > !
> > > > > !
> > > > > voice rtp send-recv
> > > > > 
> > > > > !
> > > > > ccm-manager redundant-host #PUBLISHER IP#
> > > > > ccm-manager mgcp
> > > > > ccm-manager music-on-hold
> > > > > !
> > > > > !
> > > > > controller E1 0/0
> > > > >  pri-group timeslots 1-16 service mgcp
> > > > > !
> > > > > class-map match-all control
> > > > >  match access-group 102
> > > > > class-map match-all voice
> > > > >  match access-group 101
> > > > > !
> > > > > !
> > > > > policy-map QoS
> > > > >  class voice
> > > > >   set cos 5
> > > > >  class control
> > > > >   set cos 3
> > > > > !
> > > > > !
> > > > > !
> > > > > !
> > > > > interface FastEthernet0/0
> > > > >  description uksagr00-f0/0/0
> > > > >  ip address #USED WITH VPN# 255.255.255.252
> > > > >  speed 100
> > > > >  full-duplex
> > > > > !
> > > > > interface Serial0/0:15
> > > > >  no ip address
> > > > >  no logging event link-status
> > > > >  isdn switch-type primary-net5
> > > > >  isdn incoming-voice voice
> > > > >  isdn bind-l3 ccm-manager
> > > > >  no cdp enable
> > > > > !
> > > > > interface FastEthernet0/1
> > > > >  description TRUNK TO 3548 LAN
> > > > >  no ip address
> > > > >  duplex auto
> > > > >  speed auto
> > > > > !
> > > > > interface FastEthernet0/1.2
> > > > >  encapsulation dot1Q 2 native
> > > > >  ip address #DATA LAN IP# 255.255.254.0
> > > > >  ip access-group UKSLUNITY1 out
> > > > > !
> > > > > interface FastEthernet0/1.100
> > > > >  encapsulation dot1Q 100
> > > > >  ip address #VOICE LAN IP# 255.255.254.0
> > > > >  ip policy route-map VOIP_MONITOR
> > > > > !
> > > > > ip classless
> > > > > ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> > > > > ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
> > > GW#
> > > > > ip http server
> > > > > !
> > > > > ip access-list extended VOIP_MONITOR
> > > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C
> > LAN#
> > > > > 0.0.0.255
> > > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B
> > LAN#
> > > > > 0.0.0.63
> > > > > ip access-list extended UKSLUNITY1
> > > > >  permit ip any host #UNITY SERVER IN OFC B#
> > log
> > > > >  permit ip any any
> > > > > !
> > > > > route-map VOIP_MONITOR permit 10
> > > > >  match ip address VOIP_MONITOR
> > > > >  set ip next-hop #TO VPN GW#
> > > > > !
> > > > > control-plane
> > > > > !
> > > > > !
> > > > > !
> > > > > voice-port 0/0:15
> > > > > !
> > > > > mgcp
> > > > > mgcp call-agent #subscriber IP# service-type
> > > mgcp
> > > > > version 0.1
> > > > > mgcp dtmf-relay voip codec all mode
> > out-of-band
> > > > > mgcp rtp unreachable timeout 1000 action
> > notify
> > > > > mgcp package-capability rtp-package
> > > > > no mgcp package-capability res-package
> > > > > mgcp package-capability sst-package
> > > > > no mgcp package-capability fxr-package
> > > > > no mgcp timer receive-rtcp
> > > > > mgcp sdp simple
> > > > > mgcp fax t38 inhibit
> > > > > mgcp rtp payload-type g726r16 static
> > > > > !
> > > > > mgcp profile default
> > > > > !
> > > > > 
> > > > > dial-peer voice 9 voip
> > > > >  application mgcpapp
> > > > >  destination-pattern 9T
> > > > > !
> > > > > ----
> > > > > All ideas and opinions expressed are mine and
> > > only
> > > > > mine not the company
> > > > > I work for or that you see based on my email
> > > > > address.  If I say anything
> > > > > rude or offensive blame me not my employer.
> 
> 
> 
> 		
> __________________________________________ 
> Yahoo! DSL - Something to write home about. 
> Just $16.99/mo. or less. 
> dsl.yahoo.com 
> 
> 
> 


From Anthony.Mendoza at ca.com  Fri Dec 30 11:11:36 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Fri Dec 30 11:11:40 2005
Subject: [cisco-voip] Turning off Playback of "Undeliverable" message in
	Exchange
Message-ID: <D1F2DC1FDE5F8C4385B42578077F19576080D5@uswams21.ca.com>

All,
 
I've figured out how to turn off the feature which plays back read
receipts on the  handset when checking messages.  However, I have not
found the option to turn off "Undeliverable" messages.  For example.  If
I sent out an email to a Distribution List and some members have full
mailboxes or no longer exist, I get the standard Exchange bounce back.
This message (if marked as new) shows up as "new" message if I'm calling
in.  This is somewhat distracting for our end users since they only want
to hear about voicemail, not email when checking their Voicemail over
the phone.  Is there an additional option to turn this off?

We're on Unity 4.x on top of Exchange 2003.

Thanks,
-Anthony



From rratliff at cisco.com  Fri Dec 30 11:16:20 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Dec 30 11:16:18 2005
Subject: [cisco-voip] Unity Not Recording :(
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F1957FF94@uswams21.ca.com>
References: <D1F2DC1FDE5F8C4385B42578077F1957FF94@uswams21.ca.com>
Message-ID: <7EE3C818-9C95-40FB-A861-C244794B15A9@cisco.com>

DTMF is out of band, so the gateway tells CM you hit a digit, and CM  
relays that to Unity.

How bout a simple extended ping from whichever interface your media  
is bound to (or h.323 signalling if that's the case) to Unity?

-Ryan

On Dec 30, 2005, at 11:08 AM, Mendoza, Anthony wrote:

[Comments Below]

> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com]
> Sent: Thursday, December 29, 2005 9:45 PM
> To: Mendoza, Anthony; Cisco VoIP
> Subject: RE: [cisco-voip] Unity Not Recording :(
>
> Ok. Hmm. Does unity respond to key pressses for
> options on call handlers, dial-by-name, conversations,
> etc fine?
Sure does, which is even more annoying.  If I'm logged into a mailbox, I
can choose to send a message, choosing the extension via dial by name,
but when I try to record a message, nothing...  same as when I try to
call an extension directly..

> Do regular phone calls from this gateway to IP phones
> at that site on subnet of unity server work fine?
There are no phones in Office B on the Unity Subnet.

>
> What version of Unity TSP?
Cisco Unity TSP version = 7.0(4)59)

> One or Two NICs on Unity server?
2 NICs, only 1 enabled

> That debug is not that intensive... unless theres a
> lot of call setup or MGCP activity going on. You can
> do it to the buffer or a syslog server to. Only need
> to run it for period call comes in and connects to
> unity then can stop debug. But now that you stated
> it's one way audio and it shows the correct IP for
> remote address, chances are the MGCP call
> setup/audio-cutover is OK.
Given your assumption above I didn't do the mgcp logging.

>
> As for sniffer traces, you may have ms network monitor
> installed on the server you can use to get a trace
> from unity server side of things. You can also pull up
> the port status monitor tool under tools depot to see
> if ports are seeing DTMF presses if you can't do
> dial-by-name, etc above.
The port registers all the input I press...just not my voice...very
strange I say..  are the DTMF tones sent on a different channel or
protocol than voice?


> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:
>
>> Yes, the show call act voice brief showed the IP of
>> our Unity server.. I would say that it is 1 way
>> audio right now.  We hear the Unity server greetings
>> and such fine, but it doesn't hear us when we try to
>> record a message.  I'll try the debug mgcp packet
>> next..is that cpu intensive?  e.g., do I need to
>> worry about the router crashing and burning when
>> running that debug?  I'm in California and this
>> particular 2 sites are in the UK..  so it's not easy
>> to get over there to reset it :(
>>
>> thanks for your help.  Hopefully TAC will have some
>> news for us tomorrow as well.  I'll have to wait
>> until I get a guy in the office next week to attempt
>> a sniffer trace.
>>
>> -----Original Message-----
>> From:	Erick Bergquist [mailto:erickbe at yahoo.com]
>> Sent:	Thu 12/29/2005 6:29 PM
>> To:	Mendoza, Anthony; Cisco VoIP
>> Cc:	
>> Subject:	RE: [cisco-voip] Unity Not Recording :(
>>
>> Ok. How about the other things noted in previous
>> reply? All look good? The 'show call act voice
>> brief'
>> or 'show call act voice' (more details) will show
>> the
>> remote IP address (should be unity server) during
>> you
>> have a call up with problem.
>>
>> Have you figured out if it is one-way audi
>>
>> Have you looked at mgcp debugs on router/gateway
>> (debug mgcp packet) to see if that reveals anything?
>> Maybe a sniffer trace is needed.
>>
>> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
>> wrote:
>>
>>> So we fixed a routing issue we did have, but no
>> go.
>>> We've also tried
>>> the MGCP flip flop via:
>>>
>>> conf t
>>> ! our data interface
>>> mgcp bind media source-interface FastEthernet0/1.2
>>> mgcp bind control source-interface
>> FastEthernet0/1.2
>>> ! our voice interface
>>> mgcp bind media source-interface
>> FastEthernet0/1.100
>>> mgcp bind control source-interface
>>> FastEthernet0/1.100
>>> end
>>>
>>> That didn't help either.
>>>
>>> -Anthony
>>>
>>>
>>>> -----Original Message-----
>>>> From: Erick Bergquist [mailto:erickbe at yahoo.com]
>>
>>>> Sent: Wednesday, December 28, 2005 5:41 PM
>>>> To: Mendoza, Anthony; Cisco VoIP
>>>> Subject: Re: [cisco-voip] Unity Not Recording :(
>>>>
>>>> Sounds like a routing issue of some sort, or
>> port
>>>> blockage perhaps. When you dial in from PSTN and
>>> call
>>>> hits Unity in office B can you hear the
>> greetings,
>>> etc
>>>> fine or is it just not recording or voice?
>>>>
>>>> Any codec (region) differences between the 2
>>> offices?
>>>> Is VPN doing any firewall'ish/acl'ish stuff that
>>> may
>>>> be effecting traffic? The second unity server
>> have
>>>> correct router(s) being used to get back to PSTN
>>>> gateway?
>>>>
>>>> When the call is connected, on gateway 'show
>> call
>>>> active voice brief' should show the PSTN and IP
>>> leg of
>>>> call with IP address of remote host. Does this
>>> look
>>>> right? Try pinging the remote IP address from
>>> gateways
>>>> interface MGCP is sourced from. No reachability
>>>> issues? It may be useful also to watch the call
>> in
>>> the
>>>> unity port status monitor and call viewer.
>>>>
>>>> With MGCP, try changing MGCP interface with bind
>>>> statement. I had a similar strange issue once
>> and
>>>> bouncing MGCP didn't fix it but flip-flopping
>> the
>>> MGCP
>>>> source interface (mgcp bind...) between LAN and
>>> WAN
>>>> interface and back again cleared it up (no
>>> mgcp/mgcp
>>>> didn't fix).
>>>>
>>>> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
>>> wrote:
>>>>
>>>>> All,
>>>>>
>>>>> We're running into a weird issue with a new
>>> Unity
>>>>> install.  Here's the
>>>>> basics:
>>>>>
>>>>> Office A:
>>>>> Unity 4.X
>>>>> CCM 4.0 Cluster
>>>>> Exchange Server
>>>>> 15 channel PRI Gateway on 2651 Router
>>>>>
>>>>> Office B:
>>>>> Unity 4.X
>>>>> Exchange Server
>>>>>
>>>>> Office A and B connected via VPN Tunnel
>> <100msec
>>>>> latency
>>>>>
>>>>> We're currently 100% functioning fine in
>> Office
>>> A,
>>>>> however, when I flip
>>>>> VM profiles and point people to Office B's
>> Unity
>>>>> server I am unable to
>>>>> record VM messages from the PSTN.
>>>>>
>>>>> * If I dial an extension of a phone located in
>>>>> Office A from an internal
>>>>> IP phone, it records fine to the Unity Server
>> in
>>>>> Office B.  The message
>>>>> then gets dropped in the Exchange Inbox as you
>>> would
>>>>> expect in Office B.
>>>>>
>>>>>
>>>>> * If I call that same extension from PSTN it
>>> blows
>>>>> and I get nothing.
>>>>> In fact Unity stops me 3-4 secs into it since
>> it
>>>>> hears silence.
>>>>>
>>>>> We already have a TAC issue open, but wanted
>> to
>>> find
>>>>> out if anyone here
>>>>> had any ideas?
>>>>>
>>>>> How can I make sure that when the PSTN call
>> gets
>>>>> routed to Unity it
>>>>> shows up from #VOICE LAN IP# vs. #DATA LAN IP#
>>>>> below?
>>>>>
>>>>> Here's a sanitized version of our 2651 config
>>> for
>>>>> reference:
>>>>> !
>>>>> ! Last configuration change at 23:12:31 UTC
>> Wed
>>> Dec
>>>>> 28 2005
>>>>> ! NVRAM config last updated at 23:12:47 UTC
>> Wed
>>> Dec
>>>>> 28 2005
>>>>> !
>>>>> network-clock-participate wic 0
>>>>> network-clock-select 1 E1 0/0
>>>>> voice-card 0
>>>>>  dspfarm
>>>>>  no local-bypass
>>>>> !
>>>>> isdn switch-type primary-net5
>>>>> !
>>>>> !
>>>>> voice rtp send-recv
>>>>>
>>>>> !
>>>>> ccm-manager redundant-host #PUBLISHER IP#
>>>>> ccm-manager mgcp
>>>>> ccm-manager music-on-hold
>>>>> !
>>>>> !
>>>>> controller E1 0/0
>>>>>  pri-group timeslots 1-16 service mgcp
>>>>> !
>>>>> class-map match-all control
>>>>>  match access-group 102
>>>>> class-map match-all voice
>>>>>  match access-group 101
>>>>> !
>>>>> !
>>>>> policy-map QoS
>>>>>  class voice
>>>>>   set cos 5
>>>>>  class control
>>>>>   set cos 3
>>>>> !
>>>>> !
>>>>> !
>>>>> !
>>>>> interface FastEthernet0/0
>>>>>  description uksagr00-f0/0/0
>>>>>  ip address #USED WITH VPN# 255.255.255.252
>>>>>  speed 100
>>>>>  full-duplex
>>>>> !
>>>>> interface Serial0/0:15
>>>>>  no ip address
>>>>>  no logging event link-status
>>>>>  isdn switch-type primary-net5
>>>>>  isdn incoming-voice voice
>>>>>  isdn bind-l3 ccm-manager
>>>>>  no cdp enable
>>>>> !
>>>>> interface FastEthernet0/1
>>>>>  description TRUNK TO 3548 LAN
>>>>>  no ip address
>>>>>  duplex auto
>>>>>  speed auto
>>>>> !
>>>>> interface FastEthernet0/1.2
>>>>>  encapsulation dot1Q 2 native
>>>>>  ip address #DATA LAN IP# 255.255.254.0
>>>>>  ip access-group UKSLUNITY1 out
>>>>> !
>>>>> interface FastEthernet0/1.100
>>>>>  encapsulation dot1Q 100
>>>>>  ip address #VOICE LAN IP# 255.255.254.0
>>>>>  ip policy route-map VOIP_MONITOR
>>>>> !
>>>>> ip classless
>>>>> ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
>>>>> ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
>>> GW#
>>>>> ip http server
>>>>> !
>>>>> ip access-list extended VOIP_MONITOR
>>>>>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C
>> LAN#
>>>>> 0.0.0.255
>>>>>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B
>> LAN#
>>>>> 0.0.0.63
>>>>> ip access-list extended UKSLUNITY1
>>>>>  permit ip any host #UNITY SERVER IN OFC B#
>> log
>>>>>  permit ip any any
>>>>> !
>>>>> route-map VOIP_MONITOR permit 10
>>>>>  match ip address VOIP_MONITOR
>>>>>  set ip next-hop #TO VPN GW#
>>>>> !
>>>>> control-plane
>>>>> !
>>>>> !
>>>>> !
>>>>> voice-port 0/0:15
>>>>> !
>>>>> mgcp
>>>>> mgcp call-agent #subscriber IP# service-type
>>> mgcp
>>>>> version 0.1
>>>>> mgcp dtmf-relay voip codec all mode
>> out-of-band
>>>>> mgcp rtp unreachable timeout 1000 action
>> notify
>>>>> mgcp package-capability rtp-package
>>>>> no mgcp package-capability res-package
>>>>> mgcp package-capability sst-package
>>>>> no mgcp package-capability fxr-package
>>>>> no mgcp timer receive-rtcp
>>>>> mgcp sdp simple
>>>>> mgcp fax t38 inhibit
>>>>> mgcp rtp payload-type g726r16 static
>>>>> !
>>>>> mgcp profile default
>>>>> !
>>>>>
>>>>> dial-peer voice 9 voip
>>>>>  application mgcpapp
>>>>>  destination-pattern 9T
>>>>> !
>>>>> ----
>>>>> All ideas and opinions expressed are mine and
>>> only
>>>>> mine not the company
>>>>> I work for or that you see based on my email
>>>>> address.  If I say anything
>>>>> rude or offensive blame me not my employer.
>
>
>
> 		
> __________________________________________
> Yahoo! DSL - Something to write home about.
> Just $16.99/mo. or less.
> dsl.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  Fri Dec 30 11:18:29 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Dec 30 11:18:29 2005
Subject: [cisco-voip] Turning off Playback of "Undeliverable" message in
	Exchange
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F19576080D5@uswams21.ca.com>
References: <D1F2DC1FDE5F8C4385B42578077F19576080D5@uswams21.ca.com>
Message-ID: <2DF91955-F82C-4B86-A691-BB94E3006C01@cisco.com>

I was pretty sure Unity treats NDRs and receipts the same, but maybe  
that's just in Domino.

Have you checked the advanced settings tool?

-Ryan

On Dec 30, 2005, at 11:11 AM, Mendoza, Anthony wrote:

All,

I've figured out how to turn off the feature which plays back read
receipts on the  handset when checking messages.  However, I have not
found the option to turn off "Undeliverable" messages.  For example.  If
I sent out an email to a Distribution List and some members have full
mailboxes or no longer exist, I get the standard Exchange bounce back.
This message (if marked as new) shows up as "new" message if I'm calling
in.  This is somewhat distracting for our end users since they only want
to hear about voicemail, not email when checking their Voicemail over
the phone.  Is there an additional option to turn this off?

We're on Unity 4.x on top of Exchange 2003.

Thanks,
-Anthony


_______________________________________________
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From Anthony.Mendoza at ca.com  Fri Dec 30 11:19:41 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Fri Dec 30 11:19:46 2005
Subject: [cisco-voip] Turning off Playback of "Undeliverable" message in
	Exchange
Message-ID: <D1F2DC1FDE5F8C4385B42578077F19576080DA@uswams21.ca.com>

Yes, checked the advanced settings tool as that's where I turned off the
normal read receipts.  I thought the NDRs were treated the same as well,
but maybe not in Exchange?  Is this something I can submit as a Feature
Request/Bug if it isn't already fixed in a subsequent rev?

-Anthony

> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com] 
> Sent: Friday, December 30, 2005 8:18 AM
> To: Mendoza, Anthony
> Cc: Cisco VoIP
> Subject: Re: [cisco-voip] Turning off Playback of 
> "Undeliverable" message in Exchange
> 
> I was pretty sure Unity treats NDRs and receipts the same, but maybe  
> that's just in Domino.
> 
> Have you checked the advanced settings tool?
> 
> -Ryan
> 
> On Dec 30, 2005, at 11:11 AM, Mendoza, Anthony wrote:
> 
> All,
> 
> I've figured out how to turn off the feature which plays back read
> receipts on the  handset when checking messages.  However, I have not
> found the option to turn off "Undeliverable" messages.  For 
> example.  If
> I sent out an email to a Distribution List and some members have full
> mailboxes or no longer exist, I get the standard Exchange bounce back.
> This message (if marked as new) shows up as "new" message if 
> I'm calling
> in.  This is somewhat distracting for our end users since 
> they only want
> to hear about voicemail, not email when checking their Voicemail over
> the phone.  Is there an additional option to turn this off?
> 
> We're on Unity 4.x on top of Exchange 2003.
> 
> Thanks,
> -Anthony
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 


From Anthony.Mendoza at ca.com  Fri Dec 30 11:20:15 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Fri Dec 30 11:20:21 2005
Subject: [cisco-voip] Unity Not Recording :(
Message-ID: <D1F2DC1FDE5F8C4385B42578077F19576080DB@uswams21.ca.com>

Extended ping was originally the issue we thought, but once the option
to apply the policy route map was turned on for local usage, the
extended ping works now from the PRI GW in Office A to the Unity Server
in Office B.  Also, when you refer to the interface the media is bound
to that's through the command:

mgcp bind xxxx

correct?  If there's another bind command I may be missing let me know
and I'll double check that configuration and make sure there is an
explicit bind to the Voice GW interface.

Thanks,
-Anthony

> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com] 
> Sent: Friday, December 30, 2005 8:16 AM
> To: Mendoza, Anthony
> Cc: erickbe at yahoo.com; Cisco VoIP
> Subject: Re: [cisco-voip] Unity Not Recording :(
> 
> DTMF is out of band, so the gateway tells CM you hit a digit, and CM  
> relays that to Unity.
> 
> How bout a simple extended ping from whichever interface your media  
> is bound to (or h.323 signalling if that's the case) to Unity?
> 
> -Ryan
> 
> On Dec 30, 2005, at 11:08 AM, Mendoza, Anthony wrote:
> 
> [Comments Below]
> 
> > -----Original Message-----
> > From: Erick Bergquist [mailto:erickbe at yahoo.com]
> > Sent: Thursday, December 29, 2005 9:45 PM
> > To: Mendoza, Anthony; Cisco VoIP
> > Subject: RE: [cisco-voip] Unity Not Recording :(
> >
> > Ok. Hmm. Does unity respond to key pressses for
> > options on call handlers, dial-by-name, conversations,
> > etc fine?
> Sure does, which is even more annoying.  If I'm logged into a 
> mailbox, I
> can choose to send a message, choosing the extension via dial by name,
> but when I try to record a message, nothing...  same as when I try to
> call an extension directly..
> 
> > Do regular phone calls from this gateway to IP phones
> > at that site on subnet of unity server work fine?
> There are no phones in Office B on the Unity Subnet.
> 
> >
> > What version of Unity TSP?
> Cisco Unity TSP version = 7.0(4)59)
> 
> > One or Two NICs on Unity server?
> 2 NICs, only 1 enabled
> 
> > That debug is not that intensive... unless theres a
> > lot of call setup or MGCP activity going on. You can
> > do it to the buffer or a syslog server to. Only need
> > to run it for period call comes in and connects to
> > unity then can stop debug. But now that you stated
> > it's one way audio and it shows the correct IP for
> > remote address, chances are the MGCP call
> > setup/audio-cutover is OK.
> Given your assumption above I didn't do the mgcp logging.
> 
> >
> > As for sniffer traces, you may have ms network monitor
> > installed on the server you can use to get a trace
> > from unity server side of things. You can also pull up
> > the port status monitor tool under tools depot to see
> > if ports are seeing DTMF presses if you can't do
> > dial-by-name, etc above.
> The port registers all the input I press...just not my voice...very
> strange I say..  are the DTMF tones sent on a different channel or
> protocol than voice?
> 
> 
> > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:
> >
> >> Yes, the show call act voice brief showed the IP of
> >> our Unity server.. I would say that it is 1 way
> >> audio right now.  We hear the Unity server greetings
> >> and such fine, but it doesn't hear us when we try to
> >> record a message.  I'll try the debug mgcp packet
> >> next..is that cpu intensive?  e.g., do I need to
> >> worry about the router crashing and burning when
> >> running that debug?  I'm in California and this
> >> particular 2 sites are in the UK..  so it's not easy
> >> to get over there to reset it :(
> >>
> >> thanks for your help.  Hopefully TAC will have some
> >> news for us tomorrow as well.  I'll have to wait
> >> until I get a guy in the office next week to attempt
> >> a sniffer trace.
> >>
> >> -----Original Message-----
> >> From:	Erick Bergquist [mailto:erickbe at yahoo.com]
> >> Sent:	Thu 12/29/2005 6:29 PM
> >> To:	Mendoza, Anthony; Cisco VoIP
> >> Cc:	
> >> Subject:	RE: [cisco-voip] Unity Not Recording :(
> >>
> >> Ok. How about the other things noted in previous
> >> reply? All look good? The 'show call act voice
> >> brief'
> >> or 'show call act voice' (more details) will show
> >> the
> >> remote IP address (should be unity server) during
> >> you
> >> have a call up with problem.
> >>
> >> Have you figured out if it is one-way audi
> >>
> >> Have you looked at mgcp debugs on router/gateway
> >> (debug mgcp packet) to see if that reveals anything?
> >> Maybe a sniffer trace is needed.
> >>
> >> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> >> wrote:
> >>
> >>> So we fixed a routing issue we did have, but no
> >> go.
> >>> We've also tried
> >>> the MGCP flip flop via:
> >>>
> >>> conf t
> >>> ! our data interface
> >>> mgcp bind media source-interface FastEthernet0/1.2
> >>> mgcp bind control source-interface
> >> FastEthernet0/1.2
> >>> ! our voice interface
> >>> mgcp bind media source-interface
> >> FastEthernet0/1.100
> >>> mgcp bind control source-interface
> >>> FastEthernet0/1.100
> >>> end
> >>>
> >>> That didn't help either.
> >>>
> >>> -Anthony
> >>>
> >>>
> >>>> -----Original Message-----
> >>>> From: Erick Bergquist [mailto:erickbe at yahoo.com]
> >>
> >>>> Sent: Wednesday, December 28, 2005 5:41 PM
> >>>> To: Mendoza, Anthony; Cisco VoIP
> >>>> Subject: Re: [cisco-voip] Unity Not Recording :(
> >>>>
> >>>> Sounds like a routing issue of some sort, or
> >> port
> >>>> blockage perhaps. When you dial in from PSTN and
> >>> call
> >>>> hits Unity in office B can you hear the
> >> greetings,
> >>> etc
> >>>> fine or is it just not recording or voice?
> >>>>
> >>>> Any codec (region) differences between the 2
> >>> offices?
> >>>> Is VPN doing any firewall'ish/acl'ish stuff that
> >>> may
> >>>> be effecting traffic? The second unity server
> >> have
> >>>> correct router(s) being used to get back to PSTN
> >>>> gateway?
> >>>>
> >>>> When the call is connected, on gateway 'show
> >> call
> >>>> active voice brief' should show the PSTN and IP
> >>> leg of
> >>>> call with IP address of remote host. Does this
> >>> look
> >>>> right? Try pinging the remote IP address from
> >>> gateways
> >>>> interface MGCP is sourced from. No reachability
> >>>> issues? It may be useful also to watch the call
> >> in
> >>> the
> >>>> unity port status monitor and call viewer.
> >>>>
> >>>> With MGCP, try changing MGCP interface with bind
> >>>> statement. I had a similar strange issue once
> >> and
> >>>> bouncing MGCP didn't fix it but flip-flopping
> >> the
> >>> MGCP
> >>>> source interface (mgcp bind...) between LAN and
> >>> WAN
> >>>> interface and back again cleared it up (no
> >>> mgcp/mgcp
> >>>> didn't fix).
> >>>>
> >>>> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> >>> wrote:
> >>>>
> >>>>> All,
> >>>>>
> >>>>> We're running into a weird issue with a new
> >>> Unity
> >>>>> install.  Here's the
> >>>>> basics:
> >>>>>
> >>>>> Office A:
> >>>>> Unity 4.X
> >>>>> CCM 4.0 Cluster
> >>>>> Exchange Server
> >>>>> 15 channel PRI Gateway on 2651 Router
> >>>>>
> >>>>> Office B:
> >>>>> Unity 4.X
> >>>>> Exchange Server
> >>>>>
> >>>>> Office A and B connected via VPN Tunnel
> >> <100msec
> >>>>> latency
> >>>>>
> >>>>> We're currently 100% functioning fine in
> >> Office
> >>> A,
> >>>>> however, when I flip
> >>>>> VM profiles and point people to Office B's
> >> Unity
> >>>>> server I am unable to
> >>>>> record VM messages from the PSTN.
> >>>>>
> >>>>> * If I dial an extension of a phone located in
> >>>>> Office A from an internal
> >>>>> IP phone, it records fine to the Unity Server
> >> in
> >>>>> Office B.  The message
> >>>>> then gets dropped in the Exchange Inbox as you
> >>> would
> >>>>> expect in Office B.
> >>>>>
> >>>>>
> >>>>> * If I call that same extension from PSTN it
> >>> blows
> >>>>> and I get nothing.
> >>>>> In fact Unity stops me 3-4 secs into it since
> >> it
> >>>>> hears silence.
> >>>>>
> >>>>> We already have a TAC issue open, but wanted
> >> to
> >>> find
> >>>>> out if anyone here
> >>>>> had any ideas?
> >>>>>
> >>>>> How can I make sure that when the PSTN call
> >> gets
> >>>>> routed to Unity it
> >>>>> shows up from #VOICE LAN IP# vs. #DATA LAN IP#
> >>>>> below?
> >>>>>
> >>>>> Here's a sanitized version of our 2651 config
> >>> for
> >>>>> reference:
> >>>>> !
> >>>>> ! Last configuration change at 23:12:31 UTC
> >> Wed
> >>> Dec
> >>>>> 28 2005
> >>>>> ! NVRAM config last updated at 23:12:47 UTC
> >> Wed
> >>> Dec
> >>>>> 28 2005
> >>>>> !
> >>>>> network-clock-participate wic 0
> >>>>> network-clock-select 1 E1 0/0
> >>>>> voice-card 0
> >>>>>  dspfarm
> >>>>>  no local-bypass
> >>>>> !
> >>>>> isdn switch-type primary-net5
> >>>>> !
> >>>>> !
> >>>>> voice rtp send-recv
> >>>>>
> >>>>> !
> >>>>> ccm-manager redundant-host #PUBLISHER IP#
> >>>>> ccm-manager mgcp
> >>>>> ccm-manager music-on-hold
> >>>>> !
> >>>>> !
> >>>>> controller E1 0/0
> >>>>>  pri-group timeslots 1-16 service mgcp
> >>>>> !
> >>>>> class-map match-all control
> >>>>>  match access-group 102
> >>>>> class-map match-all voice
> >>>>>  match access-group 101
> >>>>> !
> >>>>> !
> >>>>> policy-map QoS
> >>>>>  class voice
> >>>>>   set cos 5
> >>>>>  class control
> >>>>>   set cos 3
> >>>>> !
> >>>>> !
> >>>>> !
> >>>>> !
> >>>>> interface FastEthernet0/0
> >>>>>  description uksagr00-f0/0/0
> >>>>>  ip address #USED WITH VPN# 255.255.255.252
> >>>>>  speed 100
> >>>>>  full-duplex
> >>>>> !
> >>>>> interface Serial0/0:15
> >>>>>  no ip address
> >>>>>  no logging event link-status
> >>>>>  isdn switch-type primary-net5
> >>>>>  isdn incoming-voice voice
> >>>>>  isdn bind-l3 ccm-manager
> >>>>>  no cdp enable
> >>>>> !
> >>>>> interface FastEthernet0/1
> >>>>>  description TRUNK TO 3548 LAN
> >>>>>  no ip address
> >>>>>  duplex auto
> >>>>>  speed auto
> >>>>> !
> >>>>> interface FastEthernet0/1.2
> >>>>>  encapsulation dot1Q 2 native
> >>>>>  ip address #DATA LAN IP# 255.255.254.0
> >>>>>  ip access-group UKSLUNITY1 out
> >>>>> !
> >>>>> interface FastEthernet0/1.100
> >>>>>  encapsulation dot1Q 100
> >>>>>  ip address #VOICE LAN IP# 255.255.254.0
> >>>>>  ip policy route-map VOIP_MONITOR
> >>>>> !
> >>>>> ip classless
> >>>>> ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> >>>>> ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
> >>> GW#
> >>>>> ip http server
> >>>>> !
> >>>>> ip access-list extended VOIP_MONITOR
> >>>>>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C
> >> LAN#
> >>>>> 0.0.0.255
> >>>>>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B
> >> LAN#
> >>>>> 0.0.0.63
> >>>>> ip access-list extended UKSLUNITY1
> >>>>>  permit ip any host #UNITY SERVER IN OFC B#
> >> log
> >>>>>  permit ip any any
> >>>>> !
> >>>>> route-map VOIP_MONITOR permit 10
> >>>>>  match ip address VOIP_MONITOR
> >>>>>  set ip next-hop #TO VPN GW#
> >>>>> !
> >>>>> control-plane
> >>>>> !
> >>>>> !
> >>>>> !
> >>>>> voice-port 0/0:15
> >>>>> !
> >>>>> mgcp
> >>>>> mgcp call-agent #subscriber IP# service-type
> >>> mgcp
> >>>>> version 0.1
> >>>>> mgcp dtmf-relay voip codec all mode
> >> out-of-band
> >>>>> mgcp rtp unreachable timeout 1000 action
> >> notify
> >>>>> mgcp package-capability rtp-package
> >>>>> no mgcp package-capability res-package
> >>>>> mgcp package-capability sst-package
> >>>>> no mgcp package-capability fxr-package
> >>>>> no mgcp timer receive-rtcp
> >>>>> mgcp sdp simple
> >>>>> mgcp fax t38 inhibit
> >>>>> mgcp rtp payload-type g726r16 static
> >>>>> !
> >>>>> mgcp profile default
> >>>>> !
> >>>>>
> >>>>> dial-peer voice 9 voip
> >>>>>  application mgcpapp
> >>>>>  destination-pattern 9T
> >>>>> !
> >>>>> ----
> >>>>> All ideas and opinions expressed are mine and
> >>> only
> >>>>> mine not the company
> >>>>> I work for or that you see based on my email
> >>>>> address.  If I say anything
> >>>>> rude or offensive blame me not my employer.
> >
> >
> >
> > 		
> > __________________________________________
> > Yahoo! DSL - Something to write home about.
> > Just $16.99/mo. or less.
> > dsl.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  Fri Dec 30 11:29:11 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Dec 30 11:29:12 2005
Subject: [cisco-voip] Unity Not Recording :(
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F19576080DB@uswams21.ca.com>
References: <D1F2DC1FDE5F8C4385B42578077F19576080DB@uswams21.ca.com>
Message-ID: <DB18F90F-4451-4C8B-8676-3EF5318E57E5@cisco.com>

You are correct there is one bind command for signaling and one for  
media.  Make sure you've got both!

-Ryan

On Dec 30, 2005, at 11:20 AM, Mendoza, Anthony wrote:

Extended ping was originally the issue we thought, but once the option
to apply the policy route map was turned on for local usage, the
extended ping works now from the PRI GW in Office A to the Unity Server
in Office B.  Also, when you refer to the interface the media is bound
to that's through the command:

mgcp bind xxxx

correct?  If there's another bind command I may be missing let me know
and I'll double check that configuration and make sure there is an
explicit bind to the Voice GW interface.

Thanks,
-Anthony

> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> Sent: Friday, December 30, 2005 8:16 AM
> To: Mendoza, Anthony
> Cc: erickbe at yahoo.com; Cisco VoIP
> Subject: Re: [cisco-voip] Unity Not Recording :(
>
> DTMF is out of band, so the gateway tells CM you hit a digit, and CM
> relays that to Unity.
>
> How bout a simple extended ping from whichever interface your media
> is bound to (or h.323 signalling if that's the case) to Unity?
>
> -Ryan
>
> On Dec 30, 2005, at 11:08 AM, Mendoza, Anthony wrote:
>
> [Comments Below]
>
>> -----Original Message-----
>> From: Erick Bergquist [mailto:erickbe at yahoo.com]
>> Sent: Thursday, December 29, 2005 9:45 PM
>> To: Mendoza, Anthony; Cisco VoIP
>> Subject: RE: [cisco-voip] Unity Not Recording :(
>>
>> Ok. Hmm. Does unity respond to key pressses for
>> options on call handlers, dial-by-name, conversations,
>> etc fine?
> Sure does, which is even more annoying.  If I'm logged into a
> mailbox, I
> can choose to send a message, choosing the extension via dial by name,
> but when I try to record a message, nothing...  same as when I try to
> call an extension directly..
>
>> Do regular phone calls from this gateway to IP phones
>> at that site on subnet of unity server work fine?
> There are no phones in Office B on the Unity Subnet.
>
>>
>> What version of Unity TSP?
> Cisco Unity TSP version = 7.0(4)59)
>
>> One or Two NICs on Unity server?
> 2 NICs, only 1 enabled
>
>> That debug is not that intensive... unless theres a
>> lot of call setup or MGCP activity going on. You can
>> do it to the buffer or a syslog server to. Only need
>> to run it for period call comes in and connects to
>> unity then can stop debug. But now that you stated
>> it's one way audio and it shows the correct IP for
>> remote address, chances are the MGCP call
>> setup/audio-cutover is OK.
> Given your assumption above I didn't do the mgcp logging.
>
>>
>> As for sniffer traces, you may have ms network monitor
>> installed on the server you can use to get a trace
>> from unity server side of things. You can also pull up
>> the port status monitor tool under tools depot to see
>> if ports are seeing DTMF presses if you can't do
>> dial-by-name, etc above.
> The port registers all the input I press...just not my voice...very
> strange I say..  are the DTMF tones sent on a different channel or
> protocol than voice?
>
>
>> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:
>>
>>> Yes, the show call act voice brief showed the IP of
>>> our Unity server.. I would say that it is 1 way
>>> audio right now.  We hear the Unity server greetings
>>> and such fine, but it doesn't hear us when we try to
>>> record a message.  I'll try the debug mgcp packet
>>> next..is that cpu intensive?  e.g., do I need to
>>> worry about the router crashing and burning when
>>> running that debug?  I'm in California and this
>>> particular 2 sites are in the UK..  so it's not easy
>>> to get over there to reset it :(
>>>
>>> thanks for your help.  Hopefully TAC will have some
>>> news for us tomorrow as well.  I'll have to wait
>>> until I get a guy in the office next week to attempt
>>> a sniffer trace.
>>>
>>> -----Original Message-----
>>> From:	Erick Bergquist [mailto:erickbe at yahoo.com]
>>> Sent:	Thu 12/29/2005 6:29 PM
>>> To:	Mendoza, Anthony; Cisco VoIP
>>> Cc:	
>>> Subject:	RE: [cisco-voip] Unity Not Recording :(
>>>
>>> Ok. How about the other things noted in previous
>>> reply? All look good? The 'show call act voice
>>> brief'
>>> or 'show call act voice' (more details) will show
>>> the
>>> remote IP address (should be unity server) during
>>> you
>>> have a call up with problem.
>>>
>>> Have you figured out if it is one-way audi
>>>
>>> Have you looked at mgcp debugs on router/gateway
>>> (debug mgcp packet) to see if that reveals anything?
>>> Maybe a sniffer trace is needed.
>>>
>>> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
>>> wrote:
>>>
>>>> So we fixed a routing issue we did have, but no
>>> go.
>>>> We've also tried
>>>> the MGCP flip flop via:
>>>>
>>>> conf t
>>>> ! our data interface
>>>> mgcp bind media source-interface FastEthernet0/1.2
>>>> mgcp bind control source-interface
>>> FastEthernet0/1.2
>>>> ! our voice interface
>>>> mgcp bind media source-interface
>>> FastEthernet0/1.100
>>>> mgcp bind control source-interface
>>>> FastEthernet0/1.100
>>>> end
>>>>
>>>> That didn't help either.
>>>>
>>>> -Anthony
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: Erick Bergquist [mailto:erickbe at yahoo.com]
>>>
>>>>> Sent: Wednesday, December 28, 2005 5:41 PM
>>>>> To: Mendoza, Anthony; Cisco VoIP
>>>>> Subject: Re: [cisco-voip] Unity Not Recording :(
>>>>>
>>>>> Sounds like a routing issue of some sort, or
>>> port
>>>>> blockage perhaps. When you dial in from PSTN and
>>>> call
>>>>> hits Unity in office B can you hear the
>>> greetings,
>>>> etc
>>>>> fine or is it just not recording or voice?
>>>>>
>>>>> Any codec (region) differences between the 2
>>>> offices?
>>>>> Is VPN doing any firewall'ish/acl'ish stuff that
>>>> may
>>>>> be effecting traffic? The second unity server
>>> have
>>>>> correct router(s) being used to get back to PSTN
>>>>> gateway?
>>>>>
>>>>> When the call is connected, on gateway 'show
>>> call
>>>>> active voice brief' should show the PSTN and IP
>>>> leg of
>>>>> call with IP address of remote host. Does this
>>>> look
>>>>> right? Try pinging the remote IP address from
>>>> gateways
>>>>> interface MGCP is sourced from. No reachability
>>>>> issues? It may be useful also to watch the call
>>> in
>>>> the
>>>>> unity port status monitor and call viewer.
>>>>>
>>>>> With MGCP, try changing MGCP interface with bind
>>>>> statement. I had a similar strange issue once
>>> and
>>>>> bouncing MGCP didn't fix it but flip-flopping
>>> the
>>>> MGCP
>>>>> source interface (mgcp bind...) between LAN and
>>>> WAN
>>>>> interface and back again cleared it up (no
>>>> mgcp/mgcp
>>>>> didn't fix).
>>>>>
>>>>> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
>>>> wrote:
>>>>>
>>>>>> All,
>>>>>>
>>>>>> We're running into a weird issue with a new
>>>> Unity
>>>>>> install.  Here's the
>>>>>> basics:
>>>>>>
>>>>>> Office A:
>>>>>> Unity 4.X
>>>>>> CCM 4.0 Cluster
>>>>>> Exchange Server
>>>>>> 15 channel PRI Gateway on 2651 Router
>>>>>>
>>>>>> Office B:
>>>>>> Unity 4.X
>>>>>> Exchange Server
>>>>>>
>>>>>> Office A and B connected via VPN Tunnel
>>> <100msec
>>>>>> latency
>>>>>>
>>>>>> We're currently 100% functioning fine in
>>> Office
>>>> A,
>>>>>> however, when I flip
>>>>>> VM profiles and point people to Office B's
>>> Unity
>>>>>> server I am unable to
>>>>>> record VM messages from the PSTN.
>>>>>>
>>>>>> * If I dial an extension of a phone located in
>>>>>> Office A from an internal
>>>>>> IP phone, it records fine to the Unity Server
>>> in
>>>>>> Office B.  The message
>>>>>> then gets dropped in the Exchange Inbox as you
>>>> would
>>>>>> expect in Office B.
>>>>>>
>>>>>>
>>>>>> * If I call that same extension from PSTN it
>>>> blows
>>>>>> and I get nothing.
>>>>>> In fact Unity stops me 3-4 secs into it since
>>> it
>>>>>> hears silence.
>>>>>>
>>>>>> We already have a TAC issue open, but wanted
>>> to
>>>> find
>>>>>> out if anyone here
>>>>>> had any ideas?
>>>>>>
>>>>>> How can I make sure that when the PSTN call
>>> gets
>>>>>> routed to Unity it
>>>>>> shows up from #VOICE LAN IP# vs. #DATA LAN IP#
>>>>>> below?
>>>>>>
>>>>>> Here's a sanitized version of our 2651 config
>>>> for
>>>>>> reference:
>>>>>> !
>>>>>> ! Last configuration change at 23:12:31 UTC
>>> Wed
>>>> Dec
>>>>>> 28 2005
>>>>>> ! NVRAM config last updated at 23:12:47 UTC
>>> Wed
>>>> Dec
>>>>>> 28 2005
>>>>>> !
>>>>>> network-clock-participate wic 0
>>>>>> network-clock-select 1 E1 0/0
>>>>>> voice-card 0
>>>>>>  dspfarm
>>>>>>  no local-bypass
>>>>>> !
>>>>>> isdn switch-type primary-net5
>>>>>> !
>>>>>> !
>>>>>> voice rtp send-recv
>>>>>>
>>>>>> !
>>>>>> ccm-manager redundant-host #PUBLISHER IP#
>>>>>> ccm-manager mgcp
>>>>>> ccm-manager music-on-hold
>>>>>> !
>>>>>> !
>>>>>> controller E1 0/0
>>>>>>  pri-group timeslots 1-16 service mgcp
>>>>>> !
>>>>>> class-map match-all control
>>>>>>  match access-group 102
>>>>>> class-map match-all voice
>>>>>>  match access-group 101
>>>>>> !
>>>>>> !
>>>>>> policy-map QoS
>>>>>>  class voice
>>>>>>   set cos 5
>>>>>>  class control
>>>>>>   set cos 3
>>>>>> !
>>>>>> !
>>>>>> !
>>>>>> !
>>>>>> interface FastEthernet0/0
>>>>>>  description uksagr00-f0/0/0
>>>>>>  ip address #USED WITH VPN# 255.255.255.252
>>>>>>  speed 100
>>>>>>  full-duplex
>>>>>> !
>>>>>> interface Serial0/0:15
>>>>>>  no ip address
>>>>>>  no logging event link-status
>>>>>>  isdn switch-type primary-net5
>>>>>>  isdn incoming-voice voice
>>>>>>  isdn bind-l3 ccm-manager
>>>>>>  no cdp enable
>>>>>> !
>>>>>> interface FastEthernet0/1
>>>>>>  description TRUNK TO 3548 LAN
>>>>>>  no ip address
>>>>>>  duplex auto
>>>>>>  speed auto
>>>>>> !
>>>>>> interface FastEthernet0/1.2
>>>>>>  encapsulation dot1Q 2 native
>>>>>>  ip address #DATA LAN IP# 255.255.254.0
>>>>>>  ip access-group UKSLUNITY1 out
>>>>>> !
>>>>>> interface FastEthernet0/1.100
>>>>>>  encapsulation dot1Q 100
>>>>>>  ip address #VOICE LAN IP# 255.255.254.0
>>>>>>  ip policy route-map VOIP_MONITOR
>>>>>> !
>>>>>> ip classless
>>>>>> ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
>>>>>> ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
>>>> GW#
>>>>>> ip http server
>>>>>> !
>>>>>> ip access-list extended VOIP_MONITOR
>>>>>>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C
>>> LAN#
>>>>>> 0.0.0.255
>>>>>>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B
>>> LAN#
>>>>>> 0.0.0.63
>>>>>> ip access-list extended UKSLUNITY1
>>>>>>  permit ip any host #UNITY SERVER IN OFC B#
>>> log
>>>>>>  permit ip any any
>>>>>> !
>>>>>> route-map VOIP_MONITOR permit 10
>>>>>>  match ip address VOIP_MONITOR
>>>>>>  set ip next-hop #TO VPN GW#
>>>>>> !
>>>>>> control-plane
>>>>>> !
>>>>>> !
>>>>>> !
>>>>>> voice-port 0/0:15
>>>>>> !
>>>>>> mgcp
>>>>>> mgcp call-agent #subscriber IP# service-type
>>>> mgcp
>>>>>> version 0.1
>>>>>> mgcp dtmf-relay voip codec all mode
>>> out-of-band
>>>>>> mgcp rtp unreachable timeout 1000 action
>>> notify
>>>>>> mgcp package-capability rtp-package
>>>>>> no mgcp package-capability res-package
>>>>>> mgcp package-capability sst-package
>>>>>> no mgcp package-capability fxr-package
>>>>>> no mgcp timer receive-rtcp
>>>>>> mgcp sdp simple
>>>>>> mgcp fax t38 inhibit
>>>>>> mgcp rtp payload-type g726r16 static
>>>>>> !
>>>>>> mgcp profile default
>>>>>> !
>>>>>>
>>>>>> dial-peer voice 9 voip
>>>>>>  application mgcpapp
>>>>>>  destination-pattern 9T
>>>>>> !
>>>>>> ----
>>>>>> All ideas and opinions expressed are mine and
>>>> only
>>>>>> mine not the company
>>>>>> I work for or that you see based on my email
>>>>>> address.  If I say anything
>>>>>> rude or offensive blame me not my employer.
>>
>>
>>
>> 		
>> __________________________________________
>> Yahoo! DSL - Something to write home about.
>> Just $16.99/mo. or less.
>> dsl.yahoo.com
>>
>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


From erickbe at yahoo.com  Fri Dec 30 11:29:28 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Dec 30 11:29:36 2005
Subject: [cisco-voip] Turning off Playback of "Undeliverable" message in
	Exchange
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F19576080DA@uswams21.ca.com>
Message-ID: <20051230162928.78295.qmail@web34311.mail.mud.yahoo.com>

There are 2 options in advanced settings tool for
receipts, but I think NDRs are still processed since
they are different. I don't know if that other option
will kill the NDRs also but there was much discussion
on this on the cisco unified messaging forum on the
cisco forums in past. You may want to try searching
there, or posting question with unity version info,
etc and see what they come back with now. The other
thing you can try doing is make rule on mailbox on
exchange to move those types of messages to another
folder where unity won't see them. 

--- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:

> Yes, checked the advanced settings tool as that's
> where I turned off the
> normal read receipts.  I thought the NDRs were
> treated the same as well,
> but maybe not in Exchange?  Is this something I can
> submit as a Feature
> Request/Bug if it isn't already fixed in a
> subsequent rev?
> 
> -Anthony
> 
> > -----Original Message-----
> > From: Ryan Ratliff [mailto:rratliff at cisco.com] 
> > Sent: Friday, December 30, 2005 8:18 AM
> > To: Mendoza, Anthony
> > Cc: Cisco VoIP
> > Subject: Re: [cisco-voip] Turning off Playback of 
> > "Undeliverable" message in Exchange
> > 
> > I was pretty sure Unity treats NDRs and receipts
> the same, but maybe  
> > that's just in Domino.
> > 
> > Have you checked the advanced settings tool?
> > 
> > -Ryan
> > 
> > On Dec 30, 2005, at 11:11 AM, Mendoza, Anthony
> wrote:
> > 
> > All,
> > 
> > I've figured out how to turn off the feature which
> plays back read
> > receipts on the  handset when checking messages. 
> However, I have not
> > found the option to turn off "Undeliverable"
> messages.  For 
> > example.  If
> > I sent out an email to a Distribution List and
> some members have full
> > mailboxes or no longer exist, I get the standard
> Exchange bounce back.
> > This message (if marked as new) shows up as "new"
> message if 
> > I'm calling
> > in.  This is somewhat distracting for our end
> users since 
> > they only want
> > to hear about voicemail, not email when checking
> their Voicemail over
> > the phone.  Is there an additional option to turn
> this off?
> > 
> > We're on Unity 4.x on top of Exchange 2003.
> > 
> > Thanks,
> > -Anthony
> > 
> > 
> > _______________________________________________
> > 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
> 



		
__________________________________________ 
Yahoo! DSL ? Something to write home about. 
Just $16.99/mo. or less. 
dsl.yahoo.com 


From Anthony.Mendoza at ca.com  Fri Dec 30 11:30:48 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Fri Dec 30 11:30:53 2005
Subject: [cisco-voip] Unity Not Recording :(
Message-ID: <D1F2DC1FDE5F8C4385B42578077F19576080E0@uswams21.ca.com>

Yeah, Erick gave me that suggestion the other day..  have that applied,
doesn't seem to help though:

mgcp bind media source-interface FastEthernet0/1.100
mgcp bind control source-interface FastEthernet0/1.100

:(

-Anthony
> -----Original Message-----
> From: Ryan Ratliff [mailto:rratliff at cisco.com] 
> Sent: Friday, December 30, 2005 8:29 AM
> To: Mendoza, Anthony
> Cc: erickbe at yahoo.com; Cisco VoIP
> Subject: Re: [cisco-voip] Unity Not Recording :(
> 
> You are correct there is one bind command for signaling and one for  
> media.  Make sure you've got both!
> 
> -Ryan
> 
> On Dec 30, 2005, at 11:20 AM, Mendoza, Anthony wrote:
> 
> Extended ping was originally the issue we thought, but once the option
> to apply the policy route map was turned on for local usage, the
> extended ping works now from the PRI GW in Office A to the 
> Unity Server
> in Office B.  Also, when you refer to the interface the media is bound
> to that's through the command:
> 
> mgcp bind xxxx
> 
> correct?  If there's another bind command I may be missing let me know
> and I'll double check that configuration and make sure there is an
> explicit bind to the Voice GW interface.
> 
> Thanks,
> -Anthony
> 
> > -----Original Message-----
> > From: Ryan Ratliff [mailto:rratliff at cisco.com]
> > Sent: Friday, December 30, 2005 8:16 AM
> > To: Mendoza, Anthony
> > Cc: erickbe at yahoo.com; Cisco VoIP
> > Subject: Re: [cisco-voip] Unity Not Recording :(
> >
> > DTMF is out of band, so the gateway tells CM you hit a digit, and CM
> > relays that to Unity.
> >
> > How bout a simple extended ping from whichever interface your media
> > is bound to (or h.323 signalling if that's the case) to Unity?
> >
> > -Ryan
> >
> > On Dec 30, 2005, at 11:08 AM, Mendoza, Anthony wrote:
> >
> > [Comments Below]
> >
> >> -----Original Message-----
> >> From: Erick Bergquist [mailto:erickbe at yahoo.com]
> >> Sent: Thursday, December 29, 2005 9:45 PM
> >> To: Mendoza, Anthony; Cisco VoIP
> >> Subject: RE: [cisco-voip] Unity Not Recording :(
> >>
> >> Ok. Hmm. Does unity respond to key pressses for
> >> options on call handlers, dial-by-name, conversations,
> >> etc fine?
> > Sure does, which is even more annoying.  If I'm logged into a
> > mailbox, I
> > can choose to send a message, choosing the extension via 
> dial by name,
> > but when I try to record a message, nothing...  same as 
> when I try to
> > call an extension directly..
> >
> >> Do regular phone calls from this gateway to IP phones
> >> at that site on subnet of unity server work fine?
> > There are no phones in Office B on the Unity Subnet.
> >
> >>
> >> What version of Unity TSP?
> > Cisco Unity TSP version = 7.0(4)59)
> >
> >> One or Two NICs on Unity server?
> > 2 NICs, only 1 enabled
> >
> >> That debug is not that intensive... unless theres a
> >> lot of call setup or MGCP activity going on. You can
> >> do it to the buffer or a syslog server to. Only need
> >> to run it for period call comes in and connects to
> >> unity then can stop debug. But now that you stated
> >> it's one way audio and it shows the correct IP for
> >> remote address, chances are the MGCP call
> >> setup/audio-cutover is OK.
> > Given your assumption above I didn't do the mgcp logging.
> >
> >>
> >> As for sniffer traces, you may have ms network monitor
> >> installed on the server you can use to get a trace
> >> from unity server side of things. You can also pull up
> >> the port status monitor tool under tools depot to see
> >> if ports are seeing DTMF presses if you can't do
> >> dial-by-name, etc above.
> > The port registers all the input I press...just not my voice...very
> > strange I say..  are the DTMF tones sent on a different channel or
> > protocol than voice?
> >
> >
> >> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:
> >>
> >>> Yes, the show call act voice brief showed the IP of
> >>> our Unity server.. I would say that it is 1 way
> >>> audio right now.  We hear the Unity server greetings
> >>> and such fine, but it doesn't hear us when we try to
> >>> record a message.  I'll try the debug mgcp packet
> >>> next..is that cpu intensive?  e.g., do I need to
> >>> worry about the router crashing and burning when
> >>> running that debug?  I'm in California and this
> >>> particular 2 sites are in the UK..  so it's not easy
> >>> to get over there to reset it :(
> >>>
> >>> thanks for your help.  Hopefully TAC will have some
> >>> news for us tomorrow as well.  I'll have to wait
> >>> until I get a guy in the office next week to attempt
> >>> a sniffer trace.
> >>>
> >>> -----Original Message-----
> >>> From:	Erick Bergquist [mailto:erickbe at yahoo.com]
> >>> Sent:	Thu 12/29/2005 6:29 PM
> >>> To:	Mendoza, Anthony; Cisco VoIP
> >>> Cc:	
> >>> Subject:	RE: [cisco-voip] Unity Not Recording :(
> >>>
> >>> Ok. How about the other things noted in previous
> >>> reply? All look good? The 'show call act voice
> >>> brief'
> >>> or 'show call act voice' (more details) will show
> >>> the
> >>> remote IP address (should be unity server) during
> >>> you
> >>> have a call up with problem.
> >>>
> >>> Have you figured out if it is one-way audi
> >>>
> >>> Have you looked at mgcp debugs on router/gateway
> >>> (debug mgcp packet) to see if that reveals anything?
> >>> Maybe a sniffer trace is needed.
> >>>
> >>> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> >>> wrote:
> >>>
> >>>> So we fixed a routing issue we did have, but no
> >>> go.
> >>>> We've also tried
> >>>> the MGCP flip flop via:
> >>>>
> >>>> conf t
> >>>> ! our data interface
> >>>> mgcp bind media source-interface FastEthernet0/1.2
> >>>> mgcp bind control source-interface
> >>> FastEthernet0/1.2
> >>>> ! our voice interface
> >>>> mgcp bind media source-interface
> >>> FastEthernet0/1.100
> >>>> mgcp bind control source-interface
> >>>> FastEthernet0/1.100
> >>>> end
> >>>>
> >>>> That didn't help either.
> >>>>
> >>>> -Anthony
> >>>>
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: Erick Bergquist [mailto:erickbe at yahoo.com]
> >>>
> >>>>> Sent: Wednesday, December 28, 2005 5:41 PM
> >>>>> To: Mendoza, Anthony; Cisco VoIP
> >>>>> Subject: Re: [cisco-voip] Unity Not Recording :(
> >>>>>
> >>>>> Sounds like a routing issue of some sort, or
> >>> port
> >>>>> blockage perhaps. When you dial in from PSTN and
> >>>> call
> >>>>> hits Unity in office B can you hear the
> >>> greetings,
> >>>> etc
> >>>>> fine or is it just not recording or voice?
> >>>>>
> >>>>> Any codec (region) differences between the 2
> >>>> offices?
> >>>>> Is VPN doing any firewall'ish/acl'ish stuff that
> >>>> may
> >>>>> be effecting traffic? The second unity server
> >>> have
> >>>>> correct router(s) being used to get back to PSTN
> >>>>> gateway?
> >>>>>
> >>>>> When the call is connected, on gateway 'show
> >>> call
> >>>>> active voice brief' should show the PSTN and IP
> >>>> leg of
> >>>>> call with IP address of remote host. Does this
> >>>> look
> >>>>> right? Try pinging the remote IP address from
> >>>> gateways
> >>>>> interface MGCP is sourced from. No reachability
> >>>>> issues? It may be useful also to watch the call
> >>> in
> >>>> the
> >>>>> unity port status monitor and call viewer.
> >>>>>
> >>>>> With MGCP, try changing MGCP interface with bind
> >>>>> statement. I had a similar strange issue once
> >>> and
> >>>>> bouncing MGCP didn't fix it but flip-flopping
> >>> the
> >>>> MGCP
> >>>>> source interface (mgcp bind...) between LAN and
> >>>> WAN
> >>>>> interface and back again cleared it up (no
> >>>> mgcp/mgcp
> >>>>> didn't fix).
> >>>>>
> >>>>> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> >>>> wrote:
> >>>>>
> >>>>>> All,
> >>>>>>
> >>>>>> We're running into a weird issue with a new
> >>>> Unity
> >>>>>> install.  Here's the
> >>>>>> basics:
> >>>>>>
> >>>>>> Office A:
> >>>>>> Unity 4.X
> >>>>>> CCM 4.0 Cluster
> >>>>>> Exchange Server
> >>>>>> 15 channel PRI Gateway on 2651 Router
> >>>>>>
> >>>>>> Office B:
> >>>>>> Unity 4.X
> >>>>>> Exchange Server
> >>>>>>
> >>>>>> Office A and B connected via VPN Tunnel
> >>> <100msec
> >>>>>> latency
> >>>>>>
> >>>>>> We're currently 100% functioning fine in
> >>> Office
> >>>> A,
> >>>>>> however, when I flip
> >>>>>> VM profiles and point people to Office B's
> >>> Unity
> >>>>>> server I am unable to
> >>>>>> record VM messages from the PSTN.
> >>>>>>
> >>>>>> * If I dial an extension of a phone located in
> >>>>>> Office A from an internal
> >>>>>> IP phone, it records fine to the Unity Server
> >>> in
> >>>>>> Office B.  The message
> >>>>>> then gets dropped in the Exchange Inbox as you
> >>>> would
> >>>>>> expect in Office B.
> >>>>>>
> >>>>>>
> >>>>>> * If I call that same extension from PSTN it
> >>>> blows
> >>>>>> and I get nothing.
> >>>>>> In fact Unity stops me 3-4 secs into it since
> >>> it
> >>>>>> hears silence.
> >>>>>>
> >>>>>> We already have a TAC issue open, but wanted
> >>> to
> >>>> find
> >>>>>> out if anyone here
> >>>>>> had any ideas?
> >>>>>>
> >>>>>> How can I make sure that when the PSTN call
> >>> gets
> >>>>>> routed to Unity it
> >>>>>> shows up from #VOICE LAN IP# vs. #DATA LAN IP#
> >>>>>> below?
> >>>>>>
> >>>>>> Here's a sanitized version of our 2651 config
> >>>> for
> >>>>>> reference:
> >>>>>> !
> >>>>>> ! Last configuration change at 23:12:31 UTC
> >>> Wed
> >>>> Dec
> >>>>>> 28 2005
> >>>>>> ! NVRAM config last updated at 23:12:47 UTC
> >>> Wed
> >>>> Dec
> >>>>>> 28 2005
> >>>>>> !
> >>>>>> network-clock-participate wic 0
> >>>>>> network-clock-select 1 E1 0/0
> >>>>>> voice-card 0
> >>>>>>  dspfarm
> >>>>>>  no local-bypass
> >>>>>> !
> >>>>>> isdn switch-type primary-net5
> >>>>>> !
> >>>>>> !
> >>>>>> voice rtp send-recv
> >>>>>>
> >>>>>> !
> >>>>>> ccm-manager redundant-host #PUBLISHER IP#
> >>>>>> ccm-manager mgcp
> >>>>>> ccm-manager music-on-hold
> >>>>>> !
> >>>>>> !
> >>>>>> controller E1 0/0
> >>>>>>  pri-group timeslots 1-16 service mgcp
> >>>>>> !
> >>>>>> class-map match-all control
> >>>>>>  match access-group 102
> >>>>>> class-map match-all voice
> >>>>>>  match access-group 101
> >>>>>> !
> >>>>>> !
> >>>>>> policy-map QoS
> >>>>>>  class voice
> >>>>>>   set cos 5
> >>>>>>  class control
> >>>>>>   set cos 3
> >>>>>> !
> >>>>>> !
> >>>>>> !
> >>>>>> !
> >>>>>> interface FastEthernet0/0
> >>>>>>  description uksagr00-f0/0/0
> >>>>>>  ip address #USED WITH VPN# 255.255.255.252
> >>>>>>  speed 100
> >>>>>>  full-duplex
> >>>>>> !
> >>>>>> interface Serial0/0:15
> >>>>>>  no ip address
> >>>>>>  no logging event link-status
> >>>>>>  isdn switch-type primary-net5
> >>>>>>  isdn incoming-voice voice
> >>>>>>  isdn bind-l3 ccm-manager
> >>>>>>  no cdp enable
> >>>>>> !
> >>>>>> interface FastEthernet0/1
> >>>>>>  description TRUNK TO 3548 LAN
> >>>>>>  no ip address
> >>>>>>  duplex auto
> >>>>>>  speed auto
> >>>>>> !
> >>>>>> interface FastEthernet0/1.2
> >>>>>>  encapsulation dot1Q 2 native
> >>>>>>  ip address #DATA LAN IP# 255.255.254.0
> >>>>>>  ip access-group UKSLUNITY1 out
> >>>>>> !
> >>>>>> interface FastEthernet0/1.100
> >>>>>>  encapsulation dot1Q 100
> >>>>>>  ip address #VOICE LAN IP# 255.255.254.0
> >>>>>>  ip policy route-map VOIP_MONITOR
> >>>>>> !
> >>>>>> ip classless
> >>>>>> ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> >>>>>> ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
> >>>> GW#
> >>>>>> ip http server
> >>>>>> !
> >>>>>> ip access-list extended VOIP_MONITOR
> >>>>>>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C
> >>> LAN#
> >>>>>> 0.0.0.255
> >>>>>>  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B
> >>> LAN#
> >>>>>> 0.0.0.63
> >>>>>> ip access-list extended UKSLUNITY1
> >>>>>>  permit ip any host #UNITY SERVER IN OFC B#
> >>> log
> >>>>>>  permit ip any any
> >>>>>> !
> >>>>>> route-map VOIP_MONITOR permit 10
> >>>>>>  match ip address VOIP_MONITOR
> >>>>>>  set ip next-hop #TO VPN GW#
> >>>>>> !
> >>>>>> control-plane
> >>>>>> !
> >>>>>> !
> >>>>>> !
> >>>>>> voice-port 0/0:15
> >>>>>> !
> >>>>>> mgcp
> >>>>>> mgcp call-agent #subscriber IP# service-type
> >>>> mgcp
> >>>>>> version 0.1
> >>>>>> mgcp dtmf-relay voip codec all mode
> >>> out-of-band
> >>>>>> mgcp rtp unreachable timeout 1000 action
> >>> notify
> >>>>>> mgcp package-capability rtp-package
> >>>>>> no mgcp package-capability res-package
> >>>>>> mgcp package-capability sst-package
> >>>>>> no mgcp package-capability fxr-package
> >>>>>> no mgcp timer receive-rtcp
> >>>>>> mgcp sdp simple
> >>>>>> mgcp fax t38 inhibit
> >>>>>> mgcp rtp payload-type g726r16 static
> >>>>>> !
> >>>>>> mgcp profile default
> >>>>>> !
> >>>>>>
> >>>>>> dial-peer voice 9 voip
> >>>>>>  application mgcpapp
> >>>>>>  destination-pattern 9T
> >>>>>> !
> >>>>>> ----
> >>>>>> All ideas and opinions expressed are mine and
> >>>> only
> >>>>>> mine not the company
> >>>>>> I work for or that you see based on my email
> >>>>>> address.  If I say anything
> >>>>>> rude or offensive blame me not my employer.
> >>
> >>
> >>
> >> 		
> >> __________________________________________
> >> Yahoo! DSL - Something to write home about.
> >> Just $16.99/mo. or less.
> >> dsl.yahoo.com
> >>
> >>
> >>
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> 
> 
> 
> 


From Anthony.Mendoza at ca.com  Fri Dec 30 11:31:52 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Fri Dec 30 11:31:56 2005
Subject: [cisco-voip] Turning off Playback of "Undeliverable" message in
	Exchange
Message-ID: <D1F2DC1FDE5F8C4385B42578077F19576080E1@uswams21.ca.com>

I'll take a look on the forums.  The only other setting I saw in
Advanced Settings was specific to Voicemail NDRs, not to email NDRs. 

Thanks again for the suggestion.  I'd like to avoid making rules if
possible, but that is a valid workaround.
-Anthony

> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> Sent: Friday, December 30, 2005 8:29 AM
> To: Mendoza, Anthony; Ryan Ratliff
> Cc: Cisco VoIP
> Subject: RE: [cisco-voip] Turning off Playback of 
> "Undeliverable" message in Exchange
> 
> There are 2 options in advanced settings tool for
> receipts, but I think NDRs are still processed since
> they are different. I don't know if that other option
> will kill the NDRs also but there was much discussion
> on this on the cisco unified messaging forum on the
> cisco forums in past. You may want to try searching
> there, or posting question with unity version info,
> etc and see what they come back with now. The other
> thing you can try doing is make rule on mailbox on
> exchange to move those types of messages to another
> folder where unity won't see them. 
> 
> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:
> 
> > Yes, checked the advanced settings tool as that's
> > where I turned off the
> > normal read receipts.  I thought the NDRs were
> > treated the same as well,
> > but maybe not in Exchange?  Is this something I can
> > submit as a Feature
> > Request/Bug if it isn't already fixed in a
> > subsequent rev?
> > 
> > -Anthony
> > 
> > > -----Original Message-----
> > > From: Ryan Ratliff [mailto:rratliff at cisco.com] 
> > > Sent: Friday, December 30, 2005 8:18 AM
> > > To: Mendoza, Anthony
> > > Cc: Cisco VoIP
> > > Subject: Re: [cisco-voip] Turning off Playback of 
> > > "Undeliverable" message in Exchange
> > > 
> > > I was pretty sure Unity treats NDRs and receipts
> > the same, but maybe  
> > > that's just in Domino.
> > > 
> > > Have you checked the advanced settings tool?
> > > 
> > > -Ryan
> > > 
> > > On Dec 30, 2005, at 11:11 AM, Mendoza, Anthony
> > wrote:
> > > 
> > > All,
> > > 
> > > I've figured out how to turn off the feature which
> > plays back read
> > > receipts on the  handset when checking messages. 
> > However, I have not
> > > found the option to turn off "Undeliverable"
> > messages.  For 
> > > example.  If
> > > I sent out an email to a Distribution List and
> > some members have full
> > > mailboxes or no longer exist, I get the standard
> > Exchange bounce back.
> > > This message (if marked as new) shows up as "new"
> > message if 
> > > I'm calling
> > > in.  This is somewhat distracting for our end
> > users since 
> > > they only want
> > > to hear about voicemail, not email when checking
> > their Voicemail over
> > > the phone.  Is there an additional option to turn
> > this off?
> > > 
> > > We're on Unity 4.x on top of Exchange 2003.
> > > 
> > > Thanks,
> > > -Anthony
> > > 
> > > 
> > > _______________________________________________
> > > 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
> > 
> 
> 
> 
> 		
> __________________________________________ 
> Yahoo! DSL - Something to write home about. 
> Just $16.99/mo. or less. 
> dsl.yahoo.com 
> 
> 
> 
> 


From wsisk at cisco.com  Fri Dec 30 11:39:36 2005
From: wsisk at cisco.com (Wes Sisk)
Date: Fri Dec 30 11:39:46 2005
Subject: [cisco-voip] Unity Not Recording :(
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F1957FF93@uswams21.ca.com>
References: <D1F2DC1FDE5F8C4385B42578077F1957FF93@uswams21.ca.com>
Message-ID: <B5DC7192-8748-4E4B-AF00-26D51B2B0F5F@cisco.com>

In that case sniffer trace would be most informative.   How many NICs  
in Unity server and how many are active?  Typical issue is packets  
delivered to wrong interface because Unity advertises wrong address  
in OpenReceiveChannelAck or in a teaming scenario one NIC responds to  
the ARP but Unity is not listening on that interface.

sniff at the gateway fast/enet interface.

/Wes


On Dec 30, 2005, at 12:02 AM, Mendoza, Anthony wrote:

Yes, the show call act voice brief showed the IP of our Unity  
server.. I would say that it is 1 way audio right now.  We hear the  
Unity server greetings and such fine, but it doesn't hear us when we  
try to record a message.  I'll try the debug mgcp packet next..is  
that cpu intensive?  e.g., do I need to worry about the router  
crashing and burning when running that debug?  I'm in California and  
this particular 2 sites are in the UK..  so it's not easy to get over  
there to reset it :(

thanks for your help.  Hopefully TAC will have some news for us  
tomorrow as well.  I'll have to wait until I get a guy in the office  
next week to attempt a sniffer trace.

-----Original Message-----
From:   Erick Bergquist [mailto:erickbe at yahoo.com]
Sent:   Thu 12/29/2005 6:29 PM
To:     Mendoza, Anthony; Cisco VoIP
Cc:
Subject:        RE: [cisco-voip] Unity Not Recording :(

Ok. How about the other things noted in previous
reply? All look good? The 'show call act voice brief'
or 'show call act voice' (more details) will show the
remote IP address (should be unity server) during you
have a call up with problem.

Have you figured out if it is one-way audi

Have you looked at mgcp debugs on router/gateway
(debug mgcp packet) to see if that reveals anything?
Maybe a sniffer trace is needed.

--- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:

 > So we fixed a routing issue we did have, but no go.
 > We've also tried
 > the MGCP flip flop via:
 >
 > conf t
 > ! our data interface
 > mgcp bind media source-interface FastEthernet0/1.2
 > mgcp bind control source-interface FastEthernet0/1.2
 > ! our voice interface
 > mgcp bind media source-interface FastEthernet0/1.100
 > mgcp bind control source-interface
 > FastEthernet0/1.100
 > end
 >
 > That didn't help either.
 >
 > -Anthony
 > Important Phone Numbers
 > North America Service Desk: +1 631-342-3955
 > Euro-Service Desk: +44 (0) 1753 243456
 > AP-Service Desk: +61 2 9937 0510
 > Web: http://servicedesk.ca.com/
 > (remember to mention you are a Niku Acquisition
 > employee)
 >
 >
 > > -----Original Message-----
 > > From: Erick Bergquist [mailto:erickbe at yahoo.com]
 > > Sent: Wednesday, December 28, 2005 5:41 PM
 > > To: Mendoza, Anthony; Cisco VoIP
 > > Subject: Re: [cisco-voip] Unity Not Recording :(
 > >
 > > Sounds like a routing issue of some sort, or port
 > > blockage perhaps. When you dial in from PSTN and
 > call
 > > hits Unity in office B can you hear the greetings,
 > etc
 > > fine or is it just not recording or voice?
 > >
 > > Any codec (region) differences between the 2
 > offices?
 > > Is VPN doing any firewall'ish/acl'ish stuff that
 > may
 > > be effecting traffic? The second unity server have
 > > correct router(s) being used to get back to PSTN
 > > gateway?
 > >
 > > When the call is connected, on gateway 'show call
 > > active voice brief' should show the PSTN and IP
 > leg of
 > > call with IP address of remote host. Does this
 > look
 > > right? Try pinging the remote IP address from
 > gateways
 > > interface MGCP is sourced from. No reachability
 > > issues? It may be useful also to watch the call in
 > the
 > > unity port status monitor and call viewer.
 > >
 > > With MGCP, try changing MGCP interface with bind
 > > statement. I had a similar strange issue once and
 > > bouncing MGCP didn't fix it but flip-flopping the
 > MGCP
 > > source interface (mgcp bind...) between LAN and
 > WAN
 > > interface and back again cleared it up (no
 > mgcp/mgcp
 > > didn't fix).
 > >
 > > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
 > wrote:
 > >
 > > > All,
 > > >
 > > > We're running into a weird issue with a new
 > Unity
 > > > install.  Here's the
 > > > basics:
 > > >
 > > > Office A:
 > > > Unity 4.X
 > > > CCM 4.0 Cluster
 > > > Exchange Server
 > > > 15 channel PRI Gateway on 2651 Router
 > > >
 > > > Office B:
 > > > Unity 4.X
 > > > Exchange Server
 > > >
 > > > Office A and B connected via VPN Tunnel <100msec
 > > > latency
 > > >
 > > > We're currently 100% functioning fine in Office
 > A,
 > > > however, when I flip
 > > > VM profiles and point people to Office B's Unity
 > > > server I am unable to
 > > > record VM messages from the PSTN.
 > > >
 > > > * If I dial an extension of a phone located in
 > > > Office A from an internal
 > > > IP phone, it records fine to the Unity Server in
 > > > Office B.  The message
 > > > then gets dropped in the Exchange Inbox as you
 > would
 > > > expect in Office B.
 > > >
 > > >
 > > > * If I call that same extension from PSTN it
 > blows
 > > > and I get nothing.
 > > > In fact Unity stops me 3-4 secs into it since it
 > > > hears silence.
 > > >
 > > > We already have a TAC issue open, but wanted to
 > find
 > > > out if anyone here
 > > > had any ideas?
 > > >
 > > > How can I make sure that when the PSTN call gets
 > > > routed to Unity it
 > > > shows up from #VOICE LAN IP# vs. #DATA LAN IP#
 > > > below?
 > > >
 > > > Here's a sanitized version of our 2651 config
 > for
 > > > reference:
 > > > !
 > > > ! Last configuration change at 23:12:31 UTC Wed
 > Dec
 > > > 28 2005
 > > > ! NVRAM config last updated at 23:12:47 UTC Wed
 > Dec
 > > > 28 2005
 > > > !
 > > > network-clock-participate wic 0
 > > > network-clock-select 1 E1 0/0
 > > > voice-card 0
 > > >  dspfarm
 > > >  no local-bypass
 > > > !
 > > > isdn switch-type primary-net5
 > > > !
 > > > !
 > > > voice rtp send-recv
 > > >
 > > > !
 > > > ccm-manager redundant-host #PUBLISHER IP#
 > > > ccm-manager mgcp
 > > > ccm-manager music-on-hold
 > > > !
 > > > !
 > > > controller E1 0/0
 > > >  pri-group timeslots 1-16 service mgcp
 > > > !
 > > > class-map match-all control
 > > >  match access-group 102
 > > > class-map match-all voice
 > > >  match access-group 101
 > > > !
 > > > !
 > > > policy-map QoS
 > > >  class voice
 > > >   set cos 5
 > > >  class control
 > > >   set cos 3
 > > > !
 > > > !
 > > > !
 > > > !
 > > > interface FastEthernet0/0
 > > >  description uksagr00-f0/0/0
 > > >  ip address #USED WITH VPN# 255.255.255.252
 > > >  speed 100
 > > >  full-duplex
 > > > !
 > > > interface Serial0/0:15
 > > >  no ip address
 > > >  no logging event link-status
 > > >  isdn switch-type primary-net5
 > > >  isdn incoming-voice voice
 > > >  isdn bind-l3 ccm-manager
 > > >  no cdp enable
 > > > !
 > > > interface FastEthernet0/1
 > > >  description TRUNK TO 3548 LAN
 > > >  no ip address
 > > >  duplex auto
 > > >  speed auto
 > > > !
 > > > interface FastEthernet0/1.2
 > > >  encapsulation dot1Q 2 native
 > > >  ip address #DATA LAN IP# 255.255.254.0
 > > >  ip access-group UKSLUNITY1 out
 > > > !
 > > > interface FastEthernet0/1.100
 > > >  encapsulation dot1Q 100
 > > >  ip address #VOICE LAN IP# 255.255.254.0
 > > >  ip policy route-map VOIP_MONITOR
 > > > !
 > > > ip classless
 > > > ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
 > > > ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
 > GW#
 > > > ip http server
 > > > !
 > > > ip access-list extended VOIP_MONITOR
 > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C LAN#
 > > > 0.0.0.255
 > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B LAN#
 > > > 0.0.0.63
 > > > ip access-list extended UKSLUNITY1
 > > >  permit ip any host #UNITY SERVER IN OFC B# log
 > > >  permit ip any any
 > > > !
 > > > route-map VOIP_MONITOR permit 10
 > > >  match ip address VOIP_MONITOR
 > > >  set ip next-hop #TO VPN GW#
 > > > !
 > > > control-plane
 > > > !
 > > > !
 > > > !
 > > > voice-port 0/0:15
 > > > !
 > > > mgcp
 > > > mgcp call-agent #subscriber IP# service-type
 > mgcp
 > > > version 0.1
 > > > mgcp dtmf-relay voip codec all mode out-of-band
 > > > mgcp rtp unreachable timeout 1000 action notify
 > > > mgcp package-capability rtp-package
 > > > no mgcp package-capability res-package
 > > > mgcp package-capability sst-package
 > > > no mgcp package-capability fxr-package
 > > > no mgcp timer receive-rtcp
 > > > mgcp sdp simple
 > > > mgcp fax t38 inhibit
 > > > mgcp rtp payload-type g726r16 static
 > > > !
 > > > mgcp profile default
 > > > !
 > > >
 > > > dial-peer voice 9 voip
 > > >  application mgcpapp
 > > >  destination-pattern 9T
 > > > !
 > > > ----
 > > > All ideas and opinions expressed are mine and
 > only
 > > > mine not the company
 > > > I work for or that you see based on my email
 > > > address.  If I say anything
 > > > rude or offensive blame me not my employer.
 > >
 > >
 > >
 > >
 > > __________________________________________
 > > Yahoo! DSL - Something to write home about.
 > > Just $16.99/mo. or less.
 > > dsl.yahoo.com
 > >
 > >
 > >
 >




__________________________________________
Yahoo! DSL ? Something to write home about.
Just $16.99/mo. or less.
dsl.yahoo.com




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

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From Anthony.Mendoza at ca.com  Fri Dec 30 11:41:00 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Fri Dec 30 11:41:06 2005
Subject: [cisco-voip] Unity Not Recording :(
Message-ID: <D1F2DC1FDE5F8C4385B42578077F19576080E2@uswams21.ca.com>

In regards to the Unity Server, there's 2 installed, but the second is
disabled via the OS.  I will run a trace on it and see what I get.

Thanks,
-Anthony 

> -----Original Message-----
> From: Wes Sisk [mailto:wsisk at cisco.com] 
> Sent: Friday, December 30, 2005 8:40 AM
> To: Mendoza, Anthony
> Cc: erickbe at yahoo.com; Cisco VoIP
> Subject: Re: [cisco-voip] Unity Not Recording :(
> 
> In that case sniffer trace would be most informative.   How 
> many NICs in Unity server and how many are active?  Typical 
> issue is packets delivered to wrong interface because Unity 
> advertises wrong address in OpenReceiveChannelAck or in a 
> teaming scenario one NIC responds to the ARP but Unity is not 
> listening on that interface.
> 
> sniff at the gateway fast/enet interface.
> 
> /Wes
> 
> 
> On Dec 30, 2005, at 12:02 AM, Mendoza, Anthony wrote:
> 
> 
> Yes, the show call act voice brief showed the IP of our Unity 
> server.. I would say that it is 1 way audio right now.  We 
> hear the Unity server greetings and such fine, but it doesn't 
> hear us when we try to record a message.  I'll try the debug 
> mgcp packet next..is that cpu intensive?  e.g., do I need to 
> worry about the router crashing and burning when running that 
> debug?  I'm in California and this particular 2 sites are in 
> the UK..  so it's not easy to get over there to reset it :(
> 
> thanks for your help.  Hopefully TAC will have some news for 
> us tomorrow as well.  I'll have to wait until I get a guy in 
> the office next week to attempt a sniffer trace.
> 
> -----Original Message-----
> From:   Erick Bergquist [mailto:erickbe at yahoo.com]
> Sent:   Thu 12/29/2005 6:29 PM
> To:     Mendoza, Anthony; Cisco VoIP
> Cc:    
> Subject:        RE: [cisco-voip] Unity Not Recording :(
> 
> Ok. How about the other things noted in previous
> reply? All look good? The 'show call act voice brief'
> or 'show call act voice' (more details) will show the
> remote IP address (should be unity server) during you
> have a call up with problem.
> 
> Have you figured out if it is one-way audi
> 
> Have you looked at mgcp debugs on router/gateway
> (debug mgcp packet) to see if that reveals anything?
> Maybe a sniffer trace is needed.
> 
> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:
> 
> > So we fixed a routing issue we did have, but no go.
> > We've also tried
> > the MGCP flip flop via:
> >
> > conf t
> > ! our data interface
> > mgcp bind media source-interface FastEthernet0/1.2
> > mgcp bind control source-interface FastEthernet0/1.2
> > ! our voice interface
> > mgcp bind media source-interface FastEthernet0/1.100
> > mgcp bind control source-interface
> > FastEthernet0/1.100
> > end
> >
> > That didn't help either.
> >
> > -Anthony
> > Important Phone Numbers
> > North America Service Desk: +1 631-342-3955
> > Euro-Service Desk: +44 (0) 1753 243456
> > AP-Service Desk: +61 2 9937 0510
> > Web: http://servicedesk.ca.com/
> > (remember to mention you are a Niku Acquisition
> > employee)
> > 
> >
> > > -----Original Message-----
> > > From: Erick Bergquist [mailto:erickbe at yahoo.com]
> > > Sent: Wednesday, December 28, 2005 5:41 PM
> > > To: Mendoza, Anthony; Cisco VoIP
> > > Subject: Re: [cisco-voip] Unity Not Recording :(
> > >
> > > Sounds like a routing issue of some sort, or port
> > > blockage perhaps. When you dial in from PSTN and
> > call
> > > hits Unity in office B can you hear the greetings,
> > etc
> > > fine or is it just not recording or voice?
> > >
> > > Any codec (region) differences between the 2
> > offices?
> > > Is VPN doing any firewall'ish/acl'ish stuff that
> > may
> > > be effecting traffic? The second unity server have
> > > correct router(s) being used to get back to PSTN
> > > gateway?
> > >
> > > When the call is connected, on gateway 'show call
> > > active voice brief' should show the PSTN and IP
> > leg of
> > > call with IP address of remote host. Does this
> > look
> > > right? Try pinging the remote IP address from
> > gateways
> > > interface MGCP is sourced from. No reachability
> > > issues? It may be useful also to watch the call in
> > the
> > > unity port status monitor and call viewer.
> > >
> > > With MGCP, try changing MGCP interface with bind
> > > statement. I had a similar strange issue once and
> > > bouncing MGCP didn't fix it but flip-flopping the
> > MGCP
> > > source interface (mgcp bind...) between LAN and
> > WAN
> > > interface and back again cleared it up (no
> > mgcp/mgcp
> > > didn't fix).
> > >
> > > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> > wrote:
> > >
> > > > All, 
> > > >
> > > > We're running into a weird issue with a new
> > Unity
> > > > install.  Here's the
> > > > basics:
> > > > 
> > > > Office A:
> > > > Unity 4.X
> > > > CCM 4.0 Cluster
> > > > Exchange Server
> > > > 15 channel PRI Gateway on 2651 Router
> > > > 
> > > > Office B:
> > > > Unity 4.X
> > > > Exchange Server
> > > > 
> > > > Office A and B connected via VPN Tunnel <100msec
> > > > latency
> > > > 
> > > > We're currently 100% functioning fine in Office
> > A,
> > > > however, when I flip
> > > > VM profiles and point people to Office B's Unity
> > > > server I am unable to
> > > > record VM messages from the PSTN. 
> > > >
> > > > * If I dial an extension of a phone located in
> > > > Office A from an internal
> > > > IP phone, it records fine to the Unity Server in
> > > > Office B.  The message
> > > > then gets dropped in the Exchange Inbox as you
> > would
> > > > expect in Office B.
> > > >
> > > >
> > > > * If I call that same extension from PSTN it
> > blows
> > > > and I get nothing.
> > > > In fact Unity stops me 3-4 secs into it since it
> > > > hears silence.
> > > > 
> > > > We already have a TAC issue open, but wanted to
> > find
> > > > out if anyone here
> > > > had any ideas?
> > > >
> > > > How can I make sure that when the PSTN call gets
> > > > routed to Unity it
> > > > shows up from #VOICE LAN IP# vs. #DATA LAN IP#
> > > > below?
> > > >
> > > > Here's a sanitized version of our 2651 config
> > for
> > > > reference:
> > > > !
> > > > ! Last configuration change at 23:12:31 UTC Wed
> > Dec
> > > > 28 2005
> > > > ! NVRAM config last updated at 23:12:47 UTC Wed
> > Dec
> > > > 28 2005
> > > > !
> > > > network-clock-participate wic 0
> > > > network-clock-select 1 E1 0/0
> > > > voice-card 0
> > > >  dspfarm
> > > >  no local-bypass
> > > > !
> > > > isdn switch-type primary-net5
> > > > !
> > > > !
> > > > voice rtp send-recv
> > > >
> > > > !
> > > > ccm-manager redundant-host #PUBLISHER IP#
> > > > ccm-manager mgcp
> > > > ccm-manager music-on-hold
> > > > !
> > > > !
> > > > controller E1 0/0
> > > >  pri-group timeslots 1-16 service mgcp
> > > > !
> > > > class-map match-all control
> > > >  match access-group 102
> > > > class-map match-all voice
> > > >  match access-group 101
> > > > !
> > > > !
> > > > policy-map QoS
> > > >  class voice
> > > >   set cos 5
> > > >  class control
> > > >   set cos 3
> > > > !
> > > > !
> > > > !
> > > > !
> > > > interface FastEthernet0/0
> > > >  description uksagr00-f0/0/0
> > > >  ip address #USED WITH VPN# 255.255.255.252
> > > >  speed 100
> > > >  full-duplex
> > > > !
> > > > interface Serial0/0:15
> > > >  no ip address
> > > >  no logging event link-status
> > > >  isdn switch-type primary-net5
> > > >  isdn incoming-voice voice
> > > >  isdn bind-l3 ccm-manager
> > > >  no cdp enable
> > > > !
> > > > interface FastEthernet0/1
> > > >  description TRUNK TO 3548 LAN
> > > >  no ip address
> > > >  duplex auto
> > > >  speed auto
> > > > !
> > > > interface FastEthernet0/1.2
> > > >  encapsulation dot1Q 2 native
> > > >  ip address #DATA LAN IP# 255.255.254.0
> > > >  ip access-group UKSLUNITY1 out
> > > > !
> > > > interface FastEthernet0/1.100
> > > >  encapsulation dot1Q 100
> > > >  ip address #VOICE LAN IP# 255.255.254.0
> > > >  ip policy route-map VOIP_MONITOR
> > > > !
> > > > ip classless
> > > > ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> > > > ip route #VOICE NETWORKS# 255.255.0.0 #TO VPN
> > GW#
> > > > ip http server
> > > > !
> > > > ip access-list extended VOIP_MONITOR
> > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C LAN#
> > > > 0.0.0.255
> > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B LAN#
> > > > 0.0.0.63
> > > > ip access-list extended UKSLUNITY1
> > > >  permit ip any host #UNITY SERVER IN OFC B# log
> > > >  permit ip any any
> > > > !
> > > > route-map VOIP_MONITOR permit 10
> > > >  match ip address VOIP_MONITOR
> > > >  set ip next-hop #TO VPN GW#
> > > > !
> > > > control-plane
> > > > !
> > > > !
> > > > !
> > > > voice-port 0/0:15
> > > > !
> > > > mgcp
> > > > mgcp call-agent #subscriber IP# service-type
> > mgcp
> > > > version 0.1
> > > > mgcp dtmf-relay voip codec all mode out-of-band
> > > > mgcp rtp unreachable timeout 1000 action notify
> > > > mgcp package-capability rtp-package
> > > > no mgcp package-capability res-package
> > > > mgcp package-capability sst-package
> > > > no mgcp package-capability fxr-package
> > > > no mgcp timer receive-rtcp
> > > > mgcp sdp simple
> > > > mgcp fax t38 inhibit
> > > > mgcp rtp payload-type g726r16 static
> > > > !
> > > > mgcp profile default
> > > > !
> > > >
> > > > dial-peer voice 9 voip
> > > >  application mgcpapp
> > > >  destination-pattern 9T
> > > > !
> > > > ----
> > > > All ideas and opinions expressed are mine and
> > only
> > > > mine not the company
> > > > I work for or that you see based on my email
> > > > address.  If I say anything
> > > > rude or offensive blame me not my employer.
> > >
> > >
> > >
> > >            
> > > __________________________________________
> > > Yahoo! DSL - Something to write home about.
> > > Just $16.99/mo. or less.
> > > dsl.yahoo.com
> > >
> > >
> > >
> >
> 
> 
> 
>                
> __________________________________________
> Yahoo! DSL - Something to write home about.
> Just $16.99/mo. or less.
> dsl.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  Fri Dec 30 11:46:55 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Dec 30 11:48:07 2005
Subject: [cisco-voip] Turning off Playback of "Undeliverable" message in
	Exchange
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F19576080E1@uswams21.ca.com>
References: <D1F2DC1FDE5F8C4385B42578077F19576080E1@uswams21.ca.com>
Message-ID: <7B9DE6B9-1D6A-4A66-926D-64F3B3A86AAD@cisco.com>

To Exchange a voicemail is simply an email with a wav attachment.  I  
can't imagine there is any difference between a VM NDR and an email NDR.

-Ryan

On Dec 30, 2005, at 11:31 AM, Mendoza, Anthony wrote:

I'll take a look on the forums.  The only other setting I saw in
Advanced Settings was specific to Voicemail NDRs, not to email NDRs.

Thanks again for the suggestion.  I'd like to avoid making rules if
possible, but that is a valid workaround.
-Anthony

> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com]
> Sent: Friday, December 30, 2005 8:29 AM
> To: Mendoza, Anthony; Ryan Ratliff
> Cc: Cisco VoIP
> Subject: RE: [cisco-voip] Turning off Playback of
> "Undeliverable" message in Exchange
>
> There are 2 options in advanced settings tool for
> receipts, but I think NDRs are still processed since
> they are different. I don't know if that other option
> will kill the NDRs also but there was much discussion
> on this on the cisco unified messaging forum on the
> cisco forums in past. You may want to try searching
> there, or posting question with unity version info,
> etc and see what they come back with now. The other
> thing you can try doing is make rule on mailbox on
> exchange to move those types of messages to another
> folder where unity won't see them.
>
> --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:
>
>> Yes, checked the advanced settings tool as that's
>> where I turned off the
>> normal read receipts.  I thought the NDRs were
>> treated the same as well,
>> but maybe not in Exchange?  Is this something I can
>> submit as a Feature
>> Request/Bug if it isn't already fixed in a
>> subsequent rev?
>>
>> -Anthony
>>
>>> -----Original Message-----
>>> From: Ryan Ratliff [mailto:rratliff at cisco.com]
>>> Sent: Friday, December 30, 2005 8:18 AM
>>> To: Mendoza, Anthony
>>> Cc: Cisco VoIP
>>> Subject: Re: [cisco-voip] Turning off Playback of
>>> "Undeliverable" message in Exchange
>>>
>>> I was pretty sure Unity treats NDRs and receipts
>> the same, but maybe
>>> that's just in Domino.
>>>
>>> Have you checked the advanced settings tool?
>>>
>>> -Ryan
>>>
>>> On Dec 30, 2005, at 11:11 AM, Mendoza, Anthony
>> wrote:
>>>
>>> All,
>>>
>>> I've figured out how to turn off the feature which
>> plays back read
>>> receipts on the  handset when checking messages.
>> However, I have not
>>> found the option to turn off "Undeliverable"
>> messages.  For
>>> example.  If
>>> I sent out an email to a Distribution List and
>> some members have full
>>> mailboxes or no longer exist, I get the standard
>> Exchange bounce back.
>>> This message (if marked as new) shows up as "new"
>> message if
>>> I'm calling
>>> in.  This is somewhat distracting for our end
>> users since
>>> they only want
>>> to hear about voicemail, not email when checking
>> their Voicemail over
>>> the phone.  Is there an additional option to turn
>> this off?
>>>
>>> We're on Unity 4.x on top of Exchange 2003.
>>>
>>> Thanks,
>>> -Anthony
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>
>
>
> 		
> __________________________________________
> Yahoo! DSL - Something to write home about.
> Just $16.99/mo. or less.
> dsl.yahoo.com
>
>
>
>


From mmilanov at sezampro.yu  Fri Dec 30 12:04:07 2005
From: mmilanov at sezampro.yu (Mladen Milanovic)
Date: Fri Dec 30 12:00:23 2005
Subject: [cisco-voip] showing more than 500 users in general search
In-Reply-To: <5BD7043B-4CB1-49B5-A853-CED508D268A0@cisco.com>
Message-ID: <200512301700.jBUH0Jko076808@puck.nether.net>

Wes,
thanks a lot on the reply. Defenetly I will try it.
 
Happy New Year :-)
 
Mladen

  _____  

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Friday, December 30, 2005 14:29
To: mmilanov at sezampro.yu
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] showing more than 500 users in general search


another unsupported but viable suggestion: change the limit of 500:

ciscoCCNatMaxSearchResults attribute. The SystemProfile entry under the
CiscoBase will contain this attribute.

You will have to go into DC Directory Manager to modify this attrib.


/Wes

On Dec 29, 2005, at 4:41 PM, Mladen Milanovic wrote:

Hi Group,
I have a system with more than 1,500 users. When my client try to list all
users through ccmadmin page: Users>Global Directory and try to search with
blank he receive a message:
"Warning! Search too general: results truncated. 
Greater than 500 records matched, but only the first 500 will be displayed. 
Please refine your search. 

500 matching record(s)" 

Is there any way to enable showing all users. Client would like to use this
approach to verifying that all users are entered in the system ?
Thanks.
Mladen
_______________________________________________
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  Fri Dec 30 12:25:51 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Dec 30 12:26:24 2005
Subject: [cisco-voip] Turning off Playback of "Undeliverable" message in
	Exchange
In-Reply-To: <7B9DE6B9-1D6A-4A66-926D-64F3B3A86AAD@cisco.com>
Message-ID: <20051230172551.91113.qmail@web34315.mail.mud.yahoo.com>

I searched the forums for those other discussions...

In Unity 4.0(5) and later there is an additional
option in the Advanced Settings Tool that lets you
exclude ALL receipts which includes NDRs - this option
is not available in 4.0(4). Only 4.0(5) and 4.1(1). 

--- Ryan Ratliff <rratliff at cisco.com> wrote:

> To Exchange a voicemail is simply an email with a
> wav attachment.  I  
> can't imagine there is any difference between a VM
> NDR and an email NDR.
> 
> -Ryan
> 
> On Dec 30, 2005, at 11:31 AM, Mendoza, Anthony
> wrote:
> 
> I'll take a look on the forums.  The only other
> setting I saw in
> Advanced Settings was specific to Voicemail NDRs,
> not to email NDRs.
> 
> Thanks again for the suggestion.  I'd like to avoid
> making rules if
> possible, but that is a valid workaround.
> -Anthony
> 
> > -----Original Message-----
> > From: Erick Bergquist [mailto:erickbe at yahoo.com]
> > Sent: Friday, December 30, 2005 8:29 AM
> > To: Mendoza, Anthony; Ryan Ratliff
> > Cc: Cisco VoIP
> > Subject: RE: [cisco-voip] Turning off Playback of
> > "Undeliverable" message in Exchange
> >
> > There are 2 options in advanced settings tool for
> > receipts, but I think NDRs are still processed
> since
> > they are different. I don't know if that other
> option
> > will kill the NDRs also but there was much
> discussion
> > on this on the cisco unified messaging forum on
> the
> > cisco forums in past. You may want to try
> searching
> > there, or posting question with unity version
> info,
> > etc and see what they come back with now. The
> other
> > thing you can try doing is make rule on mailbox on
> > exchange to move those types of messages to
> another
> > folder where unity won't see them.
> >
> > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> wrote:
> >
> >> Yes, checked the advanced settings tool as that's
> >> where I turned off the
> >> normal read receipts.  I thought the NDRs were
> >> treated the same as well,
> >> but maybe not in Exchange?  Is this something I
> can
> >> submit as a Feature
> >> Request/Bug if it isn't already fixed in a
> >> subsequent rev?
> >>
> >> -Anthony
> >>
> >>> -----Original Message-----
> >>> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> >>> Sent: Friday, December 30, 2005 8:18 AM
> >>> To: Mendoza, Anthony
> >>> Cc: Cisco VoIP
> >>> Subject: Re: [cisco-voip] Turning off Playback
> of
> >>> "Undeliverable" message in Exchange
> >>>
> >>> I was pretty sure Unity treats NDRs and receipts
> >> the same, but maybe
> >>> that's just in Domino.
> >>>
> >>> Have you checked the advanced settings tool?
> >>>
> >>> -Ryan
> >>>
> >>> On Dec 30, 2005, at 11:11 AM, Mendoza, Anthony
> >> wrote:
> >>>
> >>> All,
> >>>
> >>> I've figured out how to turn off the feature
> which
> >> plays back read
> >>> receipts on the  handset when checking messages.
> >> However, I have not
> >>> found the option to turn off "Undeliverable"
> >> messages.  For
> >>> example.  If
> >>> I sent out an email to a Distribution List and
> >> some members have full
> >>> mailboxes or no longer exist, I get the standard
> >> Exchange bounce back.
> >>> This message (if marked as new) shows up as
> "new"
> >> message if
> >>> I'm calling
> >>> in.  This is somewhat distracting for our end
> >> users since
> >>> they only want
> >>> to hear about voicemail, not email when checking
> >> their Voicemail over
> >>> the phone.  Is there an additional option to
> turn
> >> this off?
> >>>
> >>> We're on Unity 4.x on top of Exchange 2003.
> >>>
> >>> Thanks,
> >>> -Anthony
> >>>
> >>>
> >>> _______________________________________________
> >>> 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
> >>
> >
> >
> >
> > 		
> > __________________________________________
> > Yahoo! DSL - Something to write home about.
> > Just $16.99/mo. or less.
> > dsl.yahoo.com
> >
> >
> >
> >
> 
> 



		
__________________________________________ 
Yahoo! DSL ? Something to write home about. 
Just $16.99/mo. or less. 
dsl.yahoo.com 


From Anthony.Mendoza at ca.com  Fri Dec 30 12:27:26 2005
From: Anthony.Mendoza at ca.com (Mendoza, Anthony)
Date: Fri Dec 30 12:27:31 2005
Subject: [cisco-voip] Turning off Playback of "Undeliverable" message in
	Exchange
Message-ID: <D1F2DC1FDE5F8C4385B42578077F19576080E3@uswams21.ca.com>

Thanks for the tip.  I guess we'll upgrade to 4.0(5) after we get our
users migrated over.

-Anthony

> -----Original Message-----
> From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> Sent: Friday, December 30, 2005 9:26 AM
> To: Ryan Ratliff; Mendoza, Anthony
> Cc: erickbe at yahoo.com; Cisco VoIP
> Subject: Re: [cisco-voip] Turning off Playback of 
> "Undeliverable" message in Exchange
> 
> I searched the forums for those other discussions...
> 
> In Unity 4.0(5) and later there is an additional
> option in the Advanced Settings Tool that lets you
> exclude ALL receipts which includes NDRs - this option
> is not available in 4.0(4). Only 4.0(5) and 4.1(1). 
> 
> --- Ryan Ratliff <rratliff at cisco.com> wrote:
> 
> > To Exchange a voicemail is simply an email with a
> > wav attachment.  I  
> > can't imagine there is any difference between a VM
> > NDR and an email NDR.
> > 
> > -Ryan
> > 
> > On Dec 30, 2005, at 11:31 AM, Mendoza, Anthony
> > wrote:
> > 
> > I'll take a look on the forums.  The only other
> > setting I saw in
> > Advanced Settings was specific to Voicemail NDRs,
> > not to email NDRs.
> > 
> > Thanks again for the suggestion.  I'd like to avoid
> > making rules if
> > possible, but that is a valid workaround.
> > -Anthony
> > 
> > > -----Original Message-----
> > > From: Erick Bergquist [mailto:erickbe at yahoo.com]
> > > Sent: Friday, December 30, 2005 8:29 AM
> > > To: Mendoza, Anthony; Ryan Ratliff
> > > Cc: Cisco VoIP
> > > Subject: RE: [cisco-voip] Turning off Playback of
> > > "Undeliverable" message in Exchange
> > >
> > > There are 2 options in advanced settings tool for
> > > receipts, but I think NDRs are still processed
> > since
> > > they are different. I don't know if that other
> > option
> > > will kill the NDRs also but there was much
> > discussion
> > > on this on the cisco unified messaging forum on
> > the
> > > cisco forums in past. You may want to try
> > searching
> > > there, or posting question with unity version
> > info,
> > > etc and see what they come back with now. The
> > other
> > > thing you can try doing is make rule on mailbox on
> > > exchange to move those types of messages to
> > another
> > > folder where unity won't see them.
> > >
> > > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> > wrote:
> > >
> > >> Yes, checked the advanced settings tool as that's
> > >> where I turned off the
> > >> normal read receipts.  I thought the NDRs were
> > >> treated the same as well,
> > >> but maybe not in Exchange?  Is this something I
> > can
> > >> submit as a Feature
> > >> Request/Bug if it isn't already fixed in a
> > >> subsequent rev?
> > >>
> > >> -Anthony
> > >>
> > >>> -----Original Message-----
> > >>> From: Ryan Ratliff [mailto:rratliff at cisco.com]
> > >>> Sent: Friday, December 30, 2005 8:18 AM
> > >>> To: Mendoza, Anthony
> > >>> Cc: Cisco VoIP
> > >>> Subject: Re: [cisco-voip] Turning off Playback
> > of
> > >>> "Undeliverable" message in Exchange
> > >>>
> > >>> I was pretty sure Unity treats NDRs and receipts
> > >> the same, but maybe
> > >>> that's just in Domino.
> > >>>
> > >>> Have you checked the advanced settings tool?
> > >>>
> > >>> -Ryan
> > >>>
> > >>> On Dec 30, 2005, at 11:11 AM, Mendoza, Anthony
> > >> wrote:
> > >>>
> > >>> All,
> > >>>
> > >>> I've figured out how to turn off the feature
> > which
> > >> plays back read
> > >>> receipts on the  handset when checking messages.
> > >> However, I have not
> > >>> found the option to turn off "Undeliverable"
> > >> messages.  For
> > >>> example.  If
> > >>> I sent out an email to a Distribution List and
> > >> some members have full
> > >>> mailboxes or no longer exist, I get the standard
> > >> Exchange bounce back.
> > >>> This message (if marked as new) shows up as
> > "new"
> > >> message if
> > >>> I'm calling
> > >>> in.  This is somewhat distracting for our end
> > >> users since
> > >>> they only want
> > >>> to hear about voicemail, not email when checking
> > >> their Voicemail over
> > >>> the phone.  Is there an additional option to
> > turn
> > >> this off?
> > >>>
> > >>> We're on Unity 4.x on top of Exchange 2003.
> > >>>
> > >>> Thanks,
> > >>> -Anthony
> > >>>
> > >>>
> > >>> _______________________________________________
> > >>> 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
> > >>
> > >
> > >
> > >
> > > 		
> > > __________________________________________
> > > Yahoo! DSL - Something to write home about.
> > > Just $16.99/mo. or less.
> > > dsl.yahoo.com
> > >
> > >
> > >
> > >
> > 
> > 
> 
> 
> 
> 		
> __________________________________________ 
> Yahoo! DSL - Something to write home about. 
> Just $16.99/mo. or less. 
> dsl.yahoo.com 
> 
> 
> 
> 


From erickbe at yahoo.com  Fri Dec 30 13:12:58 2005
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri Dec 30 13:13:13 2005
Subject: [cisco-voip] Unity Not Recording :(
In-Reply-To: <D1F2DC1FDE5F8C4385B42578077F1957FF94@uswams21.ca.com>
Message-ID: <20051230181258.2862.qmail@web34303.mail.mud.yahoo.com>

Are there any phones at this office you can test PSTN
calls between this gateway and IP phone on to verify
two-way communication (RTP stream) is ok? It sounds
like RTP is not making it one direction and it sounds
like routing is good because DTMF is making it. So
either something is being blocked or not handled right
by the VPN rules or policy routing perhaps. Like Wes
said, a sniffer trace would reveal more of whats going
on. But testing with phones may be easier then unity.
If the phone calls work fine, and unity doesn't record
fine then that isolates issue further down also.

You can set up a access-list to permit rtp packets
from the gateways IP address and all other traffic
inbound on the remote sites router interface and look
at the counters on that ACL to see if the hits go up
on the RTP access list entry. This will tell you if
the traffic is getting that far or not. 

--- "Mendoza, Anthony" <Anthony.Mendoza at ca.com> wrote:

> [Comments Below]
> 
> > -----Original Message-----
> > From: Erick Bergquist [mailto:erickbe at yahoo.com] 
> > Sent: Thursday, December 29, 2005 9:45 PM
> > To: Mendoza, Anthony; Cisco VoIP
> > Subject: RE: [cisco-voip] Unity Not Recording :(
> > 
> > Ok. Hmm. Does unity respond to key pressses for
> > options on call handlers, dial-by-name,
> conversations,
> > etc fine? 
> Sure does, which is even more annoying.  If I'm
> logged into a mailbox, I
> can choose to send a message, choosing the extension
> via dial by name,
> but when I try to record a message, nothing...  same
> as when I try to
> call an extension directly..
> 
> > Do regular phone calls from this gateway to IP
> phones
> > at that site on subnet of unity server work fine? 
> There are no phones in Office B on the Unity Subnet.
> 
> > 
> > What version of Unity TSP?
> Cisco Unity TSP version = 7.0(4)59)        
> 
> > One or Two NICs on Unity server?
> 2 NICs, only 1 enabled
> 
> > That debug is not that intensive... unless theres
> a
> > lot of call setup or MGCP activity going on. You
> can
> > do it to the buffer or a syslog server to. Only
> need
> > to run it for period call comes in and connects to
> > unity then can stop debug. But now that you stated
> > it's one way audio and it shows the correct IP for
> > remote address, chances are the MGCP call
> > setup/audio-cutover is OK. 
> Given your assumption above I didn't do the mgcp
> logging.
> 
> > 
> > As for sniffer traces, you may have ms network
> monitor
> > installed on the server you can use to get a trace
> > from unity server side of things. You can also
> pull up
> > the port status monitor tool under tools depot to
> see
> > if ports are seeing DTMF presses if you can't do
> > dial-by-name, etc above.
> The port registers all the input I press...just not
> my voice...very
> strange I say..  are the DTMF tones sent on a
> different channel or
> protocol than voice?
> 
> 
> > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> wrote:
> > 
> > > Yes, the show call act voice brief showed the IP
> of
> > > our Unity server.. I would say that it is 1 way
> > > audio right now.  We hear the Unity server
> greetings
> > > and such fine, but it doesn't hear us when we
> try to
> > > record a message.  I'll try the debug mgcp
> packet
> > > next..is that cpu intensive?  e.g., do I need to
> > > worry about the router crashing and burning when
> > > running that debug?  I'm in California and this
> > > particular 2 sites are in the UK..  so it's not
> easy
> > > to get over there to reset it :(
> > > 
> > > thanks for your help.  Hopefully TAC will have
> some
> > > news for us tomorrow as well.  I'll have to wait
> > > until I get a guy in the office next week to
> attempt
> > > a sniffer trace.
> > > 
> > > -----Original Message-----
> > > From:	Erick Bergquist [mailto:erickbe at yahoo.com]
> > > Sent:	Thu 12/29/2005 6:29 PM
> > > To:	Mendoza, Anthony; Cisco VoIP
> > > Cc:	
> > > Subject:	RE: [cisco-voip] Unity Not Recording :(
> > > 
> > > Ok. How about the other things noted in previous
> > > reply? All look good? The 'show call act voice
> > > brief'
> > > or 'show call act voice' (more details) will
> show
> > > the
> > > remote IP address (should be unity server)
> during
> > > you
> > > have a call up with problem. 
> > > 
> > > Have you figured out if it is one-way audi
> > > 
> > > Have you looked at mgcp debugs on router/gateway
> > > (debug mgcp packet) to see if that reveals
> anything?
> > > Maybe a sniffer trace is needed. 
> > > 
> > > --- "Mendoza, Anthony" <Anthony.Mendoza at ca.com>
> > > wrote:
> > > 
> > > > So we fixed a routing issue we did have, but
> no
> > > go. 
> > > > We've also tried
> > > > the MGCP flip flop via:
> > > > 
> > > > conf t
> > > > ! our data interface
> > > > mgcp bind media source-interface
> FastEthernet0/1.2
> > > > mgcp bind control source-interface
> > > FastEthernet0/1.2
> > > > ! our voice interface
> > > > mgcp bind media source-interface
> > > FastEthernet0/1.100
> > > > mgcp bind control source-interface
> > > > FastEthernet0/1.100
> > > > end
> > > > 
> > > > That didn't help either.
> > > > 
> > > > -Anthony
> > > >  
> > > > 
> > > > > -----Original Message-----
> > > > > From: Erick Bergquist
> [mailto:erickbe at yahoo.com]
> > > 
> > > > > Sent: Wednesday, December 28, 2005 5:41 PM
> > > > > To: Mendoza, Anthony; Cisco VoIP
> > > > > Subject: Re: [cisco-voip] Unity Not
> Recording :(
> > > > > 
> > > > > Sounds like a routing issue of some sort, or
> > > port
> > > > > blockage perhaps. When you dial in from PSTN
> and
> > > > call
> > > > > hits Unity in office B can you hear the
> > > greetings,
> > > > etc
> > > > > fine or is it just not recording or voice?
> > > > > 
> > > > > Any codec (region) differences between the 2
> > > > offices?
> > > > > Is VPN doing any firewall'ish/acl'ish stuff
> that
> > > > may
> > > > > be effecting traffic? The second unity
> server
> > > have
> > > > > correct router(s) being used to get back to
> PSTN
> > > > > gateway?
> > > > > 
> > > > > When the call is connected, on gateway 'show
> > > call
> > > > > active voice brief' should show the PSTN and
> IP
> > > > leg of
> > > > > call with IP address of remote host. Does
> this
> > > > look
> > > > > right? Try pinging the remote IP address
> from
> > > > gateways
> > > > > interface MGCP is sourced from. No
> reachability
> > > > > issues? It may be useful also to watch the
> call
> > > in
> > > > the
> > > > > unity port status monitor and call viewer. 
> > > > > 
> > > > > With MGCP, try changing MGCP interface with
> bind
> > > > > statement. I had a similar strange issue
> once
> > > and
> > > > > bouncing MGCP didn't fix it but
> flip-flopping
> > > the
> > > > MGCP
> > > > > source interface (mgcp bind...) between LAN
> and
> > > > WAN
> > > > > interface and back again cleared it up (no
> > > > mgcp/mgcp
> > > > > didn't fix).
> > > > > 
> > > > > --- "Mendoza, Anthony"
> <Anthony.Mendoza at ca.com>
> > > > wrote:
> > > > > 
> > > > > > All,  
> > > > > > 
> > > > > > We're running into a weird issue with a
> new
> > > > Unity
> > > > > > install.  Here's the
> > > > > > basics:
> > > > > >  
> > > > > > Office A:
> > > > > > Unity 4.X
> > > > > > CCM 4.0 Cluster
> > > > > > Exchange Server
> > > > > > 15 channel PRI Gateway on 2651 Router
> > > > > >  
> > > > > > Office B:
> > > > > > Unity 4.X
> > > > > > Exchange Server
> > > > > >  
> > > > > > Office A and B connected via VPN Tunnel
> > > <100msec
> > > > > > latency
> > > > > >  
> > > > > > We're currently 100% functioning fine in
> > > Office
> > > > A,
> > > > > > however, when I flip
> > > > > > VM profiles and point people to Office B's
> > > Unity
> > > > > > server I am unable to
> > > > > > record VM messages from the PSTN.  
> > > > > > 
> > > > > > * If I dial an extension of a phone
> located in
> > > > > > Office A from an internal
> > > > > > IP phone, it records fine to the Unity
> Server
> > > in
> > > > > > Office B.  The message
> > > > > > then gets dropped in the Exchange Inbox as
> you
> > > > would
> > > > > > expect in Office B.
> > > > > > 
> > > > > > 
> > > > > > * If I call that same extension from PSTN
> it
> > > > blows
> > > > > > and I get nothing.
> > > > > > In fact Unity stops me 3-4 secs into it
> since
> > > it
> > > > > > hears silence.
> > > > > >  
> > > > > > We already have a TAC issue open, but
> wanted
> > > to
> > > > find
> > > > > > out if anyone here
> > > > > > had any ideas?
> > > > > > 
> > > > > > How can I make sure that when the PSTN
> call
> > > gets
> > > > > > routed to Unity it
> > > > > > shows up from #VOICE LAN IP# vs. #DATA LAN
> IP#
> > > > > > below?
> > > > > > 
> > > > > > Here's a sanitized version of our 2651
> config
> > > > for
> > > > > > reference:
> > > > > > !
> > > > > > ! Last configuration change at 23:12:31
> UTC
> > > Wed
> > > > Dec
> > > > > > 28 2005
> > > > > > ! NVRAM config last updated at 23:12:47
> UTC
> > > Wed
> > > > Dec
> > > > > > 28 2005
> > > > > > !
> > > > > > network-clock-participate wic 0
> > > > > > network-clock-select 1 E1 0/0
> > > > > > voice-card 0
> > > > > >  dspfarm
> > > > > >  no local-bypass
> > > > > > !
> > > > > > isdn switch-type primary-net5
> > > > > > !
> > > > > > !
> > > > > > voice rtp send-recv
> > > > > > 
> > > > > > !
> > > > > > ccm-manager redundant-host #PUBLISHER IP#
> > > > > > ccm-manager mgcp
> > > > > > ccm-manager music-on-hold
> > > > > > !
> > > > > > !
> > > > > > controller E1 0/0
> > > > > >  pri-group timeslots 1-16 service mgcp
> > > > > > !
> > > > > > class-map match-all control
> > > > > >  match access-group 102
> > > > > > class-map match-all voice
> > > > > >  match access-group 101
> > > > > > !
> > > > > > !
> > > > > > policy-map QoS
> > > > > >  class voice
> > > > > >   set cos 5
> > > > > >  class control
> > > > > >   set cos 3
> > > > > > !
> > > > > > !
> > > > > > !
> > > > > > !
> > > > > > interface FastEthernet0/0
> > > > > >  description uksagr00-f0/0/0
> > > > > >  ip address #USED WITH VPN#
> 255.255.255.252
> > > > > >  speed 100
> > > > > >  full-duplex
> > > > > > !
> > > > > > interface Serial0/0:15
> > > > > >  no ip address
> > > > > >  no logging event link-status
> > > > > >  isdn switch-type primary-net5
> > > > > >  isdn incoming-voice voice
> > > > > >  isdn bind-l3 ccm-manager
> > > > > >  no cdp enable
> > > > > > !
> > > > > > interface FastEthernet0/1
> > > > > >  description TRUNK TO 3548 LAN
> > > > > >  no ip address
> > > > > >  duplex auto
> > > > > >  speed auto
> > > > > > !
> > > > > > interface FastEthernet0/1.2
> > > > > >  encapsulation dot1Q 2 native
> > > > > >  ip address #DATA LAN IP# 255.255.254.0
> > > > > >  ip access-group UKSLUNITY1 out
> > > > > > !
> > > > > > interface FastEthernet0/1.100
> > > > > >  encapsulation dot1Q 100
> > > > > >  ip address #VOICE LAN IP# 255.255.254.0
> > > > > >  ip policy route-map VOIP_MONITOR
> > > > > > !
> > > > > > ip classless
> > > > > > ip route 0.0.0.0 0.0.0.0 #TO DATA LAN#
> > > > > > ip route #VOICE NETWORKS# 255.255.0.0 #TO
> VPN
> > > > GW#
> > > > > > ip http server
> > > > > > !
> > > > > > ip access-list extended VOIP_MONITOR
> > > > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE C
> > > LAN#
> > > > > > 0.0.0.255
> > > > > >  permit ip #VOICE LAN# 0.0.1.255 #OFFICE B
> > > LAN#
> > > > > > 0.0.0.63
> > > > > > ip access-list extended UKSLUNITY1
> > > > > >  permit ip any host #UNITY SERVER IN OFC
> B#
> > > log
> > > > > >  permit ip any any
> > > > > > !
> > > > > > route-map VOIP_MONITOR permit 10
> > > > > >  match ip address VOIP_MONITOR
> > > > > >  set ip next-hop #TO VPN GW#
> > > > > > !
> > > > > > control-plane
> > > > > > !
> > > > > > !
> > > > > > !
> > > > > > voice-port 0/0:15
> > > > > > !
> > > > > > mgcp
> > > > > > mgcp call-agent #subscriber IP#
> service-type
> > > > mgcp
> > > > > > version 0.1
> > > > > > mgcp dtmf-relay voip codec all mode
> > > out-of-band
> > > > > > mgcp rtp unreachable timeout 1000 action
> > > notify
> > > > > > mgcp package-capability rtp-package
> > > > > > no mgcp package-capability res-package
> > > > > > mgcp package-capability sst-package
> > > > > > no mgcp package-capability fxr-package
> > > > > > no mgcp timer receive-rtcp
> > > > > > mgcp sdp simple
> > > > > > mgcp fax t38 inhibit
> > > > > > mgcp rtp payload-type g726r16 static
> > > > > > !
> > > > > > mgcp profile default
> > > > > > !
> > > > > > 
> > > > > > dial-peer voice 9 voip
> > > > > >  application mgcpapp
> > > > > >  destination-pattern 9T
> > > > > > !
> > > > > > ----
> > > > > > All ideas and opinions expressed are mine
> and
> > > > only
> > > > > > mine not the company
> > > > > > I work for or that you see based on my
> email
> > > > > > address.  If I say anything
> > > > > > rude or offensive blame me not my
> employer.
> > 
> > 
> > 
> > 		
> > __________________________________________ 
> > Yahoo! DSL - Something to write home about. 
> > Just $16.99/mo. or less. 
> > dsl.yahoo.com 
> > 
> > 
> > 
> 



	
		
__________________________________ 
Yahoo! for Good - Make a difference this year. 
http://brand.yahoo.com/cybergivingweek2005/

From it at cimgroup.com  Fri Dec 30 14:24:23 2005
From: it at cimgroup.com (IT)
Date: Fri Dec 30 14:20:23 2005
Subject: [cisco-voip] TAC using a password decrypter
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0270935D@exchange.cimgroup.com>

I was on the phone yesterday with TAC troubleshooting an issue with my
IPMAUser login/password. We went into the registry on my CallManager,
and I read him the encrypted password key, and he said he used a
"password decrypter", and read back the password in plain text. He said
the utility was publicly accessible. What and where can I get this?

 

Also, how do I set up a line on CallManager so that 7960 users can see
"line status" of another user, without being able to answer the line or
call from it?

This is basically so co-workers can see if their neighbor is on the
phone, since they are too lazy to get up and check....

 

Thanks,

Avidan


******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
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From pwalenta at wi.rr.com  Fri Dec 30 14:44:08 2005
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Fri Dec 30 14:44:12 2005
Subject: [cisco-voip] TAC using a password decrypter
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B0270935D@exchange.cimgroup.com>
Message-ID: <200512301942.jBUJg4Z7028946@ms-smtp-01.rdc-kc.rr.com>

There is no way to set up a busy lamp field in the current versions of
CallManager.  However I believe that feature will be coming soon.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Friday, December 30, 2005 1:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] TAC using a password decrypter



I was on the phone yesterday with TAC troubleshooting an issue with my
IPMAUser login/password. We went into the registry on my CallManager, and I
read him the encrypted password key, and he said he used a "password
decrypter", and read back the password in plain text. He said the utility
was publicly accessible. What and where can I get this?

 

Also, how do I set up a line on CallManager so that 7960 users can see "line
status" of another user, without being able to answer the line or call from
it?

This is basically so co-workers can see if their neighbor is on the phone,
since they are too lazy to get up and check..

 

Thanks,

Avidan

******************************************************
This email may contain trade secrets and/or other information that is 
proprietary and/or confidential. Use of this by anyone other than the 
intended recipient(s) is prohibited and may be unlawful. If you are not 
the intended recipient, delete the e-mail and notify us immediately.
****************************************************** 

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From rratliff at cisco.com  Fri Dec 30 14:44:35 2005
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri Dec 30 14:44:35 2005
Subject: [cisco-voip] TAC using a password decrypter
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B0270935D@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B0270935D@exchange.cimgroup.com>
Message-ID: <6AE07A62-0019-48C5-BC0A-056272A4C4E0@cisco.com>

The utility is not publicly available AFAIK.   The correct way to  
"decrypt" these passwords is to use the passwordutils utility to  
encrypt your known passwords and check against what is in your registry.

-Ryan

On Dec 30, 2005, at 2:24 PM, IT wrote:

I was on the phone yesterday with TAC troubleshooting an issue with  
my IPMAUser login/password. We went into the registry on my  
CallManager, and I read him the encrypted password key, and he said  
he used a ?password decrypter?, and read back the password in plain  
text. He said the utility was publicly accessible. What and where can  
I get this?



Also, how do I set up a line on CallManager so that 7960 users can  
see ?line status? of another user, without being able to answer the  
line or call from it?

This is basically so co-workers can see if their neighbor is on the  
phone, since they are too lazy to get up and check?.



Thanks,

Avidan

******************************************************
This email may contain trade secrets and/or other information that is
proprietary and/or confidential. Use of this by anyone other than the
intended recipient(s) is prohibited and may be unlawful. If you are not
the intended recipient, delete the e-mail and notify us immediately.
******************************************************

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



From kthorngr at cisco.com  Fri Dec 30 15:26:32 2005
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri Dec 30 15:26:38 2005
Subject: [cisco-voip] TAC using a password decrypter
In-Reply-To: <6AE07A62-0019-48C5-BC0A-056272A4C4E0@cisco.com>
References: <B487D4C1FE6D04439061018C6EBC109B0270935D@exchange.cimgroup.com>
	<6AE07A62-0019-48C5-BC0A-056272A4C4E0@cisco.com>
Message-ID: <9f48ac4e437be6238c6ae4763f859998@cisco.com>

Ryan is correct.  We don't have a publicly available tool to decrypt 
passwords.  There are very few instances where it would be needed and 
an administrator should be able to reset the service passwords through 
the CCM tools provided.

Please unicast the case number to me so I can make sure the TAC CE 
understands this.

Kevin
On Dec 30, 2005, at 2:44 PM, Ryan Ratliff wrote:

> The utility is not publicly available AFAIK.   The correct way to
> "decrypt" these passwords is to use the passwordutils utility to
> encrypt your known passwords and check against what is in your 
> registry.
>
> -Ryan
>
> On Dec 30, 2005, at 2:24 PM, IT wrote:
>
> I was on the phone yesterday with TAC troubleshooting an issue with
> my IPMAUser login/password. We went into the registry on my
> CallManager, and I read him the encrypted password key, and he said
> he used a ?password decrypter?, and read back the password in plain
> text. He said the utility was publicly accessible. What and where can
> I get this?
>
>
>
> Also, how do I set up a line on CallManager so that 7960 users can
> see ?line status? of another user, without being able to answer the
> line or call from it?
>
> This is basically so co-workers can see if their neighbor is on the
> phone, since they are too lazy to get up and check?.
>
>
>
> Thanks,
>
> Avidan
>
> ******************************************************
> This email may contain trade secrets and/or other information that is
> proprietary and/or confidential. Use of this by anyone other than the
> intended recipient(s) is prohibited and may be unlawful. If you are not
> the intended recipient, delete the e-mail and notify us immediately.
> ******************************************************
>
> _______________________________________________
> 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 pb2 at bigpond.net.au  Sat Dec 31 23:05:56 2005
From: pb2 at bigpond.net.au (Paul Barnes)
Date: Sat Dec 31 23:06:13 2005
Subject: [cisco-voip] 3810 router as sip ua - calling line ID
Message-ID: <007501c60e88$aacbc560$be00a8c0@spineynorman2>

Hi 
I have an mc3810 router working as a sip ua to connect a voip network to an Ericsson MD110 pabx via the 3810's E1 voice interface. I'm using qsig as the E1 sig protocol
It works pretty well, voice, ringing etc all work ok between the two systems, however can only get Calling Line ID (I think ANI in the US) to work one way - ie the pabx extension number is displayed on the sip handset on an incoming call, however the voip extension number is not displayed on the pabx handset when calling the other way.
Debug q931 and voip ccapi shows the ANI and dnis numbers in the signalling exchange so I'm inclined to think it could be a pabx setting?
I can't find any cisco specific interface commands related to the passing of CLI information across an E1. Would appreciate it if anybody has any ideas?
thanks
paul 
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