From parnold at 454.com  Fri Sep  1 02:22:01 2006
From: parnold at 454.com (Arnold, Paul)
Date: Fri, 1 Sep 2006 02:22:01 -0400
Subject: [cisco-voip] Voice activation feature on Cisco IP Phone
In-Reply-To: <20060830132555.39488.qmail@web52004.mail.yahoo.com>
Message-ID: <AD922568C8E1C546B98A94DB60C5403A034D18BC@NELSON.labrat.com>

There are ~$19.99 analog answering machines that read out caller id
number verbally. (No name, just number.)
I use one at home.
Direct her extension to an analog port with the answering machine
attached?

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gu Sang Jung
Sent: Wednesday, August 30, 2006 9:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Voice activation feature on Cisco IP Phone


 
The problem is like this.
One of my customer can't see the screen on her phone and therefore does
not know who's trying to call even though Voice T1-line provide calling
number information. She was wondering if it would be possible to set her
current phone for voice acitivation... So that when someone calls it
will tell her who's calling/what phone # they are calling from.
Is there any possible solution with Cisco CCM ?
 
thank in advance
 
gusang jung
 
CCIE VOIEC &RS (12972),MCSE
Network Enginner

________________________________

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From netfortius at gmail.com  Fri Sep  1 08:24:10 2006
From: netfortius at gmail.com (Netfortius)
Date: Fri, 1 Sep 2006 07:24:10 -0500
Subject: [cisco-voip] [Semi-OT] E911 and legal issues
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F003063747@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F003063747@USNAEXCH.na.didata.local>
Message-ID: <200609010724.10787.netfortius@gmail.com>

This makes perfect sense - thank you!

Stefan

On Thursday 31 August 2006 22:28, you wrote:
> I've been down this path several times.
>
> What you need to do is call the local non-emergency number for the
> area(s) that you need this deployed.  You will then need to work with
> them to identify the actual pop that services your location.  They
> usually have someone who can answer these questions and outline the
> local requirements and laws.  I always make it a point to request on
> upcoming changes to the laws as well, but they are not usually excited
> about change.
>
> The next step to take before looking into something like CER is to call
> the local carrier.  It's embarrassing to sell a customer CER and then
> find out the local carrier cannot accept dynamic ELI updates.  Usually
> the carrier can give you the comparison of what is required by law and
> what they can provide from the Telco's perspective.

From CORTIZ at broward.org  Fri Sep  1 08:59:19 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 1 Sep 2006 08:59:19 -0400
Subject: [cisco-voip] Save Ringer Volume settings
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A4747E@MREXVS01.bc.broward.cty>

There was a post here yesterday about saving ring volume settings.  Not
sure if this applies but read below and see if this is the case with
your Call Mgr setup.  

 

 

Settings Access :

Indicates whether the Settings button on the phone is functional. When
Settings Access is enabled, you can change the phone network
configuration, ring type, and volume on the phone. When Settings Access
is disabled, the Settings button is completely disabled; no options
appear when you press the button. Also, you cannot adjust the ringer
volume or save any volume settings. When Settings Access is restricted,
you can only access User Preferences and volume settings. By default,
Settings Access is enabled. 
This is a required field.
Default: Enabled.

 

Carlos

 

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From rratliff at cisco.com  Fri Sep  1 08:59:21 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 1 Sep 2006 08:59:21 -0400
Subject: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?
In-Reply-To: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD7533@SVCEXCH01.firm.fenwick.llp>
References: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD7533@SVCEXCH01.firm.fenwick.llp>
Message-ID: <7824C76F-2540-4C15-BFA3-81AFBC6CB903@cisco.com>

Since there was no autoupdate path to CIPC 2.0 there was no package  
released to update the device defaults.  I would expect that when the  
next version is released they will include the files necessary to  
update device defaults.

-Ryan

On Aug 31, 2006, at 2:17 PM, Stu Packett wrote:

My concern is, how does the CIPC client know to get an update from  
the TFTP server when my device defaults still shows 1.1(3)?  For  
example, if Cisco releases 2.0(2), the client has no idea that 2.0(2)  
is out because the TFTP server is still set to 1.1(3).  I know in  
previous versions of CIPC, they had you run a SCCP install, but if  
the new CIPC 2.0 uses the same load as the 797x, then I guess I  
shouldn't worry much.  Thanks.

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Thursday, August 31, 2006 11:12 AM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?

You said it works fine so I think you answered your own question.  I  
think since it uses SCCP it should use the same load that any 797X  
should use.  I have 2.0(1a) running and have not had to install any  
additional phone loads?..



Carlos



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, August 31, 2006 1:36 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?



I just downloaded IP Communicator 2.0(1a) and installed it on my  
laptop and it works just fine, but don't I need to install the IP  
Communicator phone load on the TFTP server?  My 'Device Defaults'  
currently shows CIPC-1-1-3-0 under 'Cisco IP Communicator' and I  
would assume it should show CIPC-2-0-1a under it.  I remember when I  
first installed version 1.1(3), I had to run some device package  
installation so I created the files under the TFTP directory.  Is  
this not the case for version 2.0?  I can't seem to find (if it  
exists) the phone loads for CIPC 2.0.  Is anyone experiencing the  
same thing I am?  Thanks in advance.



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



From CORTIZ at broward.org  Fri Sep  1 09:11:32 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 1 Sep 2006 09:11:32 -0400
Subject: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A4747F@MREXVS01.bc.broward.cty>

Now I see what you are talking about.  Scratch what I said
yesterday.....

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Friday, September 01, 2006 8:59 AM
To: Stu Packett
Cc: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?

Since there was no autoupdate path to CIPC 2.0 there was no package  
released to update the device defaults.  I would expect that when the  
next version is released they will include the files necessary to  
update device defaults.

-Ryan

On Aug 31, 2006, at 2:17 PM, Stu Packett wrote:

My concern is, how does the CIPC client know to get an update from  
the TFTP server when my device defaults still shows 1.1(3)?  For  
example, if Cisco releases 2.0(2), the client has no idea that 2.0(2)  
is out because the TFTP server is still set to 1.1(3).  I know in  
previous versions of CIPC, they had you run a SCCP install, but if  
the new CIPC 2.0 uses the same load as the 797x, then I guess I  
shouldn't worry much.  Thanks.

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Thursday, August 31, 2006 11:12 AM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?

You said it works fine so I think you answered your own question.  I  
think since it uses SCCP it should use the same load that any 797X  
should use.  I have 2.0(1a) running and have not had to install any  
additional phone loads.....



Carlos



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, August 31, 2006 1:36 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?



I just downloaded IP Communicator 2.0(1a) and installed it on my  
laptop and it works just fine, but don't I need to install the IP  
Communicator phone load on the TFTP server?  My 'Device Defaults'  
currently shows CIPC-1-1-3-0 under 'Cisco IP Communicator' and I  
would assume it should show CIPC-2-0-1a under it.  I remember when I  
first installed version 1.1(3), I had to run some device package  
installation so I created the files under the TFTP directory.  Is  
this not the case for version 2.0?  I can't seem to find (if it  
exists) the phone loads for CIPC 2.0.  Is anyone experiencing the  
same thing I am?  Thanks in advance.



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


From lelio at uoguelph.ca  Fri Sep  1 09:12:13 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 1 Sep 2006 09:12:13 -0400
Subject: [cisco-voip] Save Ringer Volume settings
References: <26B357EF9FB7D04A989C9E82A9E5436E03A4747E@MREXVS01.bc.broward.cty>
Message-ID: <002501c6cdc8$3dca76f0$c801a8c0@EATIUSBIRDIUS>

The "save" softkey is the one that appears immediately after you hit the "settings" button, while you are still in the menu. 


  ----- Original Message ----- 
  From: Ortiz, Carlos 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, September 01, 2006 8:59 AM
  Subject: [cisco-voip] Save Ringer Volume settings


  There was a post here yesterday about saving ring volume settings.  Not sure if this applies but read below and see if this is the case with your Call Mgr setup.  

   

   

        Settings Access :
       Indicates whether the Settings button on the phone is functional. When Settings Access is enabled, you can change the phone network configuration, ring type, and volume on the phone. When Settings Access is disabled, the Settings button is completely disabled; no options appear when you press the button. Also, you cannot adjust the ringer volume or save any volume settings. When Settings Access is restricted, you can only access User Preferences and volume settings. By default, Settings Access is enabled. 
        This is a required field.
        Default: Enabled.

         

        Carlos
       

   



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


  _______________________________________________
  cisco-voip mailing list
  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 Sep  1 10:07:10 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Fri, 1 Sep 2006 09:07:10 -0500
Subject: [cisco-voip] [Semi-OT] E911 and legal issues
In-Reply-To: <200608312047.32748.netfortius@gmail.com>
Message-ID: <OFDB4C3B50.0D36D9F9-ON862571DC.004D0464-862571DC.004D7BC6@ra.rockwell.com>

Check with the local PSAP.  As far as I know Chicago is the only city that 
has and actually enforces an E911 law.

Also from my understanding you can make 911 ring to an internal team first 
as long as it is posted so that the employees understand that that is in 
fact what is happening.

Craig





Netfortius <netfortius at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
08/31/2006 08:47 PM
Please respond to
netfortius at gmail.com


To
cisco-voip at puck.nether.net
cc

Subject
[cisco-voip] [Semi-OT] E911 and legal issues






Sorry for the off-topic, but google-ing around didn't reveal much to me, 
so 
here it goes: has any one of the members of this list implemented an E911 
solution? I have a client who needs this installed, but we do not know if 
the 
configuration has to be related to legal aspects in the state/county/town 
the 
company resides in. Here is what I am referring to: 

- under the old, analog system, the methodology of calling 911 had the 
number 
actually routed to an emergency crew, properly trained in CPR and other 
first 
emergency responder methods, who would then decide if the call had to 
really 
go to 911. One of the big advantages of this method was also the fact that 

the internal crew had full knowledge of the location of each person, to 
the 
cube and office level

- someone I was discussing this issue with said to me that the above was 
only 
possible under very old laws, allowing such, and that the new laws require 

911 calls to be sent directly to the 911 services, and that the location 
will 
have to be passed on based on the information from a database my customer 
will have to deliver (and maintain?!?) in coordination with the 911 folks, 

such that each phone will be identified in a specifc area of the building, 

based on participation in a switch-port connection. To me this sounds 
extreme, because one of the advnatages VoIP has been bringin to the table 
was ... well ... mobility of phones (no matter where I plug them, they 
register) - so how the heck are we going to be able to continuously update 

the switch-port info?!?

Any pointers will be highly appreciated - to narrow the scope I would be 
mostly interested in USA - Illinois, but of course other states' 
deployment 
experience will also be valuable.

If you feel the above is totally off-topic, please accept my apologies, 
and - 
if anythiung useful to be added - please email me directly.

Thank you,
Stefan
_______________________________________________
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cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From selt9 at yahoo.com  Fri Sep  1 12:15:03 2006
From: selt9 at yahoo.com (Selt9)
Date: Fri, 1 Sep 2006 09:15:03 -0700 (PDT)
Subject: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E03A4747F@MREXVS01.bc.broward.cty>
Message-ID: <20060901161503.18306.qmail@web52308.mail.yahoo.com>

no, such files will not be included because upgrade from 1.1.x to 2.x is not supported and will not be...
  so, if they include such files there will be a condition to raise all present installation of CIPC to 2.0 version before making next version available on tftp server...
   
  

"Ortiz, Carlos" <CORTIZ at broward.org> wrote:
  Now I see what you are talking about. Scratch what I said
yesterday.....

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Friday, September 01, 2006 8:59 AM
To: Stu Packett
Cc: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?

Since there was no autoupdate path to CIPC 2.0 there was no package 
released to update the device defaults. I would expect that when the 
next version is released they will include the files necessary to 
update device defaults.

-Ryan

On Aug 31, 2006, at 2:17 PM, Stu Packett wrote:

My concern is, how does the CIPC client know to get an update from 
the TFTP server when my device defaults still shows 1.1(3)? For 
example, if Cisco releases 2.0(2), the client has no idea that 2.0(2) 
is out because the TFTP server is still set to 1.1(3). I know in 
previous versions of CIPC, they had you run a SCCP install, but if 
the new CIPC 2.0 uses the same load as the 797x, then I guess I 
shouldn't worry much. Thanks.

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Thursday, August 31, 2006 11:12 AM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?

You said it works fine so I think you answered your own question. I 
think since it uses SCCP it should use the same load that any 797X 
should use. I have 2.0(1a) running and have not had to install any 
additional phone loads.....



Carlos



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, August 31, 2006 1:36 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CIPC-2-0-1a.loads - Does it exist?



I just downloaded IP Communicator 2.0(1a) and installed it on my 
laptop and it works just fine, but don't I need to install the IP 
Communicator phone load on the TFTP server? My 'Device Defaults' 
currently shows CIPC-1-1-3-0 under 'Cisco IP Communicator' and I 
would assume it should show CIPC-2-0-1a under it. I remember when I 
first installed version 1.1(3), I had to run some device package 
installation so I created the files under the TFTP directory. Is 
this not the case for version 2.0? I can't seem to find (if it 
exists) the phone loads for CIPC 2.0. Is anyone experiencing the 
same thing I am? 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 jason.aarons at us.didata.com  Fri Sep  1 15:19:48 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 1 Sep 2006 15:19:48 -0400
Subject: [cisco-voip] What is command to disable (busy out) b-channel on T1
	PRI?
Message-ID: <C1FE15183DA37645BC0633BC604E44F003064109@USNAEXCH.na.didata.local>

What is command to disable (busy out) a single b-channel on PRI? Show
isdn service shows all in-service, I need to take a channel out of
service -jason




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From jneiberger at gmail.com  Fri Sep  1 15:29:50 2006
From: jneiberger at gmail.com (John Neiberger)
Date: Fri, 1 Sep 2006 13:29:50 -0600
Subject: [cisco-voip] What is command to disable (busy out) b-channel on
	T1 PRI?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F003064109@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F003064109@USNAEXCH.na.didata.local>
Message-ID: <547ad0fe0609011229x69b4cdfwcf5d054d158e8d2f@mail.gmail.com>

Is this on an MGCP or H.323 gateway?

On 9/1/06, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
>
>
>
> What is command to disable (busy out) a single b-channel on PRI? Show isdn
> service shows all in-service, I need to take a channel out of service -jason
>
>
> ________________________________
>
>
>
>
>
>
>  Disclaimer:
>
>  This e-mail communication and any attachments may contain
>  confidential and privileged information and is for use by the
>  designated addressee(s) named above only. If you are not the
>  intended addressee, you are hereby notified that you have received
>  this communication in error and that any use or reproduction of
>  this email or its contents is strictly prohibited and may be
>  unlawful. If you have received this communication in error, please
>  notify us immediately by replying to this message and deleting it
>  from your computer. Thank you.
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> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From DylanV at semaphore.com  Fri Sep  1 15:31:51 2006
From: DylanV at semaphore.com (Dylan Vanderhoof)
Date: Fri, 1 Sep 2006 12:31:51 -0700
Subject: [cisco-voip] SNMP monitoring MGCP gateway
Message-ID: <F6F72F553C7E754A945CB88CD8804B42417F1E@exchange01.semaphore.lan>

Is there any useful information that can be polled via SNMP on an MGCP
voice gateway?  I'm looking at my VG200 and not finding much (I assume
due to the backhaul), but I figured I'd ask.

Obviously call stats are available via SNMP on the callmanager.

Thanks,
Dylam


From voipermike at gmail.com  Fri Sep  1 15:33:01 2006
From: voipermike at gmail.com (Mike Mike)
Date: Fri, 1 Sep 2006 14:33:01 -0500
Subject: [cisco-voip] What is command to disable (busy out) b-channel on
	T1 PRI?
In-Reply-To: <547ad0fe0609011229x69b4cdfwcf5d054d158e8d2f@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F003064109@USNAEXCH.na.didata.local>
	<547ad0fe0609011229x69b4cdfwcf5d054d158e8d2f@mail.gmail.com>
Message-ID: <8bbfc1f40609011233tff5284cy77bd13b49d606a33@mail.gmail.com>

I believe you need to go into the controller config and busyout the ds0

On 9/1/06, John Neiberger <jneiberger at gmail.com> wrote:
>
> Is this on an MGCP or H.323 gateway?
>
> On 9/1/06, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
> >
> >
> >
> >
> > What is command to disable (busy out) a single b-channel on PRI? Show
> isdn
> > service shows all in-service, I need to take a channel out of service
> -jason
> >
> >
> > ________________________________
> >
> >
> >
> >
> >
> >
> >  Disclaimer:
> >
> >  This e-mail communication and any attachments may contain
> >  confidential and privileged information and is for use by the
> >  designated addressee(s) named above only. If you are not the
> >  intended addressee, you are hereby notified that you have received
> >  this communication in error and that any use or reproduction of
> >  this email or its contents is strictly prohibited and may be
> >  unlawful. If you have received this communication in error, please
> >  notify us immediately by replying to this message and deleting it
> >  from your computer. Thank you.
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
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From lelio at uoguelph.ca  Fri Sep  1 15:46:23 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 1 Sep 2006 15:46:23 -0400
Subject: [cisco-voip] What is command to disable (busy out) b-channel on
	T1PRI?
References: <C1FE15183DA37645BC0633BC604E44F003064109@USNAEXCH.na.didata.local>
Message-ID: <005101c6cdff$4e4a13f0$c801a8c0@EATIUSBIRDIUS>

There is a CallManager service parameter that you can use. It accepts a string of 0s and 1s as the argument. I believe 0 is enabled, and 1 is disabled. So, if you put 00000000000000000000001 the last channel is disabled...or something like that. I think it's called maintenance status. I don't think it's a supported way of implementing partial T1 service but others on the list have done it like this.

  ----- Original Message ----- 
  From: Jason Aarons (US) 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, September 01, 2006 3:19 PM
  Subject: [cisco-voip] What is command to disable (busy out) b-channel on T1PRI?


  What is command to disable (busy out) a single b-channel on PRI? Show isdn service shows all in-service, I need to take a channel out of service -jason



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From voipermike at gmail.com  Fri Sep  1 15:49:35 2006
From: voipermike at gmail.com (Mike Mike)
Date: Fri, 1 Sep 2006 14:49:35 -0500
Subject: [cisco-voip] What is command to disable (busy out) b-channel on
	T1PRI?
In-Reply-To: <005101c6cdff$4e4a13f0$c801a8c0@EATIUSBIRDIUS>
References: <C1FE15183DA37645BC0633BC604E44F003064109@USNAEXCH.na.didata.local>
	<005101c6cdff$4e4a13f0$c801a8c0@EATIUSBIRDIUS>
Message-ID: <8bbfc1f40609011249u74b37ad9pb5a7fd52df7b5b3d@mail.gmail.com>

Here is a link for busying out the ds0:

http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/products_feature_guide09186a0080080091.html

On 9/1/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  There is a CallManager service parameter that you can use. It accepts a
> string of 0s and 1s as the argument. I believe 0 is enabled, and 1 is
> disabled. So, if you put 00000000000000000000001 the last channel is
> disabled...or something like that. I think it's called maintenance status. I
> don't think it's a supported way of implementing partial T1 service but
> others on the list have done it like this.
>
>
> ----- Original Message -----
> *From:* Jason Aarons (US) <jason.aarons at us.didata.com>
> *To:* cisco-voip at puck.nether.net
> *Sent:* Friday, September 01, 2006 3:19 PM
> *Subject:* [cisco-voip] What is command to disable (busy out) b-channel on
> T1PRI?
>
>  What is command to disable (busy out) a single b-channel on PRI? Show
> isdn service shows all in-service, I need to take a channel out of service
> -jason
>
> ------------------------------
>
>
> *
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful. If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
> *
>
> ------------------------------
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> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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>
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From duncanw at otc.edu  Fri Sep  1 16:38:52 2006
From: duncanw at otc.edu (DUNCAN, WILLIAM B.)
Date: Fri, 1 Sep 2006 15:38:52 -0500
Subject: [cisco-voip] Unity Digital Domain
Message-ID: <06D1B6D4926222458F803D0D3EDCCB7E84E07C@EXM1.otc.edu>

I have been attempting to get digital networking setup between two unity
servers on our network.  They connect to the same CCM cluster but are on
different vlans.  If I open global subscribers I see both of these
servers, but when I add a dialing domain on one server, it never appears
in the drop down list on the other.  I can manually add the dialing
domain on the second, but you can only send messages one way.  Any
ideas?

 

Thanks everyone,

Brian Duncan

Coordinator of Telecommunications

Ozarks Technical Community College

 

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From tednugent69 at yahoo.com  Fri Sep  1 17:16:30 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 1 Sep 2006 14:16:30 -0700 (PDT)
Subject: [cisco-voip] What is command to disable (busy out) b-channel on
	T1 PRI?
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F003064109@USNAEXCH.na.didata.local>
Message-ID: <20060901211630.40868.qmail@web36107.mail.mud.yahoo.com>

It's an advanced CM Service Param. It's only needed if
the telco is sending top-down (ascending-starting at 1
going up) which would require you to send bottom-up.
(decending-starting at 23 going down)

Change B-Channel Maintenance Status 1 : This parameter
allows Cisco CallManager to change individual
B-Channel maintenance status for PRI and CAS
interfaces 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. In the case of PRI, the last
bit specifies the D-Channel, which does not get
affected. In this example, the 5th through 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 the D-Channel. The
last bit does not affect any channel. In this example,
the 25th through 31st channels are marked out of
service.)
Explanation:
The Device Name as specified in the Gateway
Configuration page in Cisco CallManager Administration
MUST exactly match the gateway name that this field
specifies. To avoid any manual inputting mistake, copy
the Device Name from Cisco CallManager Administration
and paste it into this service parameter. "=" is
mandatory and unique to distinguish the Device Name
and B-Channel 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.)
Because values other than 0,1, or 2 are invalid, the
system ignores them. 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. 
PRI Specific:
To use this feature, the Enable status poll check box
must be checked on the PRI Gateway Configuration
window. If it is not, you need to check the check box,
click Update, and reset the gateway for the service
parameter change to take effect.

Maximum length: 255.
 

--- "Jason Aarons (US)" <jason.aarons at us.didata.com>
wrote:

> What is command to disable (busy out) a single
> b-channel on PRI? Show
> isdn service shows all in-service, I need to take a
> channel out of
> service -jason
> 
> 
> 
> 
> -----------------------------------------
> Disclaimer:
> 
> This e-mail communication and any attachments may
> contain
> confidential and privileged information and is for
> use by the
> designated addressee(s) named above only.  If you
> are not the
> intended addressee, you are hereby notified that you
> have received
> this communication in error and that any use or
> reproduction of
> this email or its contents is strictly prohibited
> and may be
> unlawful.  If you have received this communication
> in error, please
> notify us immediately by replying to this message
> and deleting it
> from your computer. Thank you.
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> 


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From cisco.ipt at gmail.com  Fri Sep  1 17:35:55 2006
From: cisco.ipt at gmail.com (cisco.ipt)
Date: Fri, 1 Sep 2006 15:35:55 -0600
Subject: [cisco-voip] VoIP Monitoring Solutions
Message-ID: <8ebc12a90609011435taadd131r3f68083e4694f7dc@mail.gmail.com>

I am looking into best products to use for Cisco Voice Network Monitoring, I
am curious what others thoughts are on the below listed Applications and
requirements.

Requirements:

   - e911 Testing - is there anything out there that supports this?
   - Circuit Utilization - what application?
   - CDR Records - Best application for this?
   - Multicast validation or testing?
   - System Support - troubleshooting VoIP



Applications for NMS:
Cisco NMS
- Cisco Unified Operations Manager 1.1
- CiscoWorks IP Communications Operations Manager 1.0

Prognosis
- PROGNOSIS IP Telephony Manager
- PROGNOSIS IP Telephony Express
- PROGNOSIS IP Telephony Assessor
- PROGNOSIS IPCC Management Reporting
- PROGNOSIS for SRST

NetIQ
- AppManager for VoIP
- NetIQ Vivinet Diagnostics
- Vivinet Assessor

Qovia
- Qovia Service Manager monitoring and management software
- Qovia Sensors

Clarus Systems
- ClarusIPC - Testing Solution

Anything else anyone using?



- - - - -
thanks,
cisco.ipt at gmail.com
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From lelio at uoguelph.ca  Fri Sep  1 17:49:00 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 1 Sep 2006 17:49:00 -0400
Subject: [cisco-voip] VoIP Monitoring Solutions
References: <8ebc12a90609011435taadd131r3f68083e4694f7dc@mail.gmail.com>
Message-ID: <004601c6ce10$6f5dba90$c801a8c0@EATIUSBIRDIUS>

I'm not sure about others on the list and I don't mean to start a flame war on a friday, but I'd really like to see names associated with posts. I can understand why people use alternate email addresses, i.e. for filtering, etc. But I'd like to be able to put a name to a message. 

cisco-ipt at gmail.com who are you?!?!?

lelio "hoping it's not my boss" fulgenzi
  ----- Original Message ----- 
  From: cisco.ipt 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, September 01, 2006 5:35 PM
  Subject: [cisco-voip] VoIP Monitoring Solutions


  I am looking into best products to use for Cisco Voice Network Monitoring, I am curious what others thoughts are on the below listed Applications and requirements.

  Requirements:

    a.. e911 Testing - is there anything out there that supports this?
    b.. Circuit Utilization - what application?
    c.. CDR Records - Best application for this?
    d.. Multicast validation or testing? 
    e.. System Support - troubleshooting VoIP


  Applications for NMS: 
  Cisco NMS
  - Cisco Unified Operations Manager 1.1
  - CiscoWorks IP Communications Operations Manager 1.0 

  Prognosis
  - PROGNOSIS IP Telephony Manager
  - PROGNOSIS IP Telephony Express
  - PROGNOSIS IP Telephony Assessor
  - PROGNOSIS IPCC Management Reporting
  - PROGNOSIS for SRST 

  NetIQ 
  - AppManager for VoIP
  - NetIQ Vivinet Diagnostics
  - Vivinet Assessor

  Qovia
  - Qovia Service Manager monitoring and management software
  - Qovia Sensors

  Clarus Systems
  - ClarusIPC - Testing Solution 

  Anything else anyone using?



  - - - - - 
  thanks,
  cisco.ipt at gmail.com



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  https://puck.nether.net/mailman/listinfo/cisco-voip
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From DylanV at semaphore.com  Fri Sep  1 19:28:00 2006
From: DylanV at semaphore.com (Dylan Vanderhoof)
Date: Fri, 1 Sep 2006 16:28:00 -0700
Subject: [cisco-voip] SNMP monitoring MGCP gateway
Message-ID: <F6F72F553C7E754A945CB88CD8804B42417F21@exchange01.semaphore.lan>

Replying to myself, but I need to restate the question, since some
snmpwalking of a few callmanagers of different versions showed nearly
nothing use.

Can I get usage stats on trunks/gateways via snmp at all with an MGCP
backhaul in place?  Am I missing some obvious OID on my VG200 gateway?

I know AXL is available, but its not really an option at this point for
what I'm looking at.

Thanks,
Dylan

> -----Original Message-----
> From: Dylan Vanderhoof 
> Sent: Friday, September 01, 2006 12:32 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] SNMP monitoring MGCP gateway
> 
> 
> Is there any useful information that can be polled via SNMP on an MGCP
> voice gateway?  I'm looking at my VG200 and not finding much (I assume
> due to the backhaul), but I figured I'd ask.
> 
> Obviously call stats are available via SNMP on the callmanager.
> 
> Thanks,
> Dylan
> 


From bababooey at cox.net  Sat Sep  2 00:07:57 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Sat, 2 Sep 2006 00:07:57 -0400
Subject: [cisco-voip] Cisco voice gw - caller id
References: <mailman.2039.1157147342.1681.cisco-voip@puck.nether.net>
Message-ID: <000301c6ce45$60023320$0f0110ac@corporation2>

I have in my current lab setup the following:
CCM 4.1 ->3825 -- t1 --3825 <-- CCM 4.1

I am using the 1MFT1 module between routers back to back t1 cable.
I have configured the H323 Gw on both CCM's and the Cisco IOS Voice GW 
routers.
I can call between the CCM's but get "Unknown number" on the receving IP 
Phone.

What do I need to do to pass the caller id info?
Can I pass the caller-id info with H323 and T1 or do I need MGCP/PRI?

The caller-id command does not seem to be available under the voice-port 
config.
I am using 12.4 ip adv service IOS on the Voice GW routers.

Thanks




From marcus.lundbom at invid.se  Sat Sep  2 07:33:34 2006
From: marcus.lundbom at invid.se (Marcus Lundbom)
Date: Sat, 2 Sep 2006 13:33:34 +0200
Subject: [cisco-voip] CME calling name
Message-ID: <30A8B4535CC19243A7C505E5156642D9277C09@gbgsrv02.gbg.invid.local>

Hi list,

Would it be possible to configure a CME to actually pass calling name to
an outgoing PRI? If so, can someone please enlighten me on how? :)

Best regards,


/M


From lelio at uoguelph.ca  Sat Sep  2 08:25:10 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat, 2 Sep 2006 08:25:10 -0400
Subject: [cisco-voip] CME calling name
References: <30A8B4535CC19243A7C505E5156642D9277C09@gbgsrv02.gbg.invid.local>
Message-ID: <001d01c6ce8a$d5930a70$c801a8c0@EATIUSBIRDIUS>

I spent some time investigating this. I'm sure I'll be corrected if I've got anything slighty ajar. 

Some background. In the U.S. the calling name is presented to the called party by the called party's service provider via SS7 database lookups. Meanwhile, in Canada, the calling name is presented to the called party by the calling party's service provider. 

While CallManager can be configured to 'allow calling party name' out PRIs, the value used is the same calling party name used for internal calls. There is no option in CallManager to override the internal calling party name with an enterprise calling party name - it is expecting the PSTN to do that for you. Unfortunately, in Canada, the PRI service model does not allow for that. Supposedly, it does, but the service providers have not paid the licencing fees or something like that.

Anyways, I'm not sure how much of this translates to CME, but it might be a start.

If you're looking for the internal calling name to be passed, check the system parameters (or equiv) on CME. If you're looking for an enterprise parameter, ask your SP if they can override it.

Good Luck


  ----- Original Message ----- 
  From: Marcus Lundbom 
  To: cisco-voip at puck.nether.net 
  Sent: Saturday, September 02, 2006 7:33 AM
  Subject: [cisco-voip] CME calling name


  Hi list,

  Would it be possible to configure a CME to actually pass calling name to
  an outgoing PRI? If so, can someone please enlighten me on how? :)

  Best regards,


  /M

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From SCASPER at mandtbank.com  Sat Sep  2 11:17:49 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Sat, 02 Sep 2006 11:17:49 -0400
Subject: [cisco-voip] SRST transfer user 4 digits to an 11 digit extension
Message-ID: <s4f9687e.046@comrcm01intgwia.mandtbank.com>


 We use 11 digit extensions at our site and would like to be able to
transfer using 4 digits when in SRST mode just as can be done when not
in SRST. Transfer works fine to an the actual 11 digit extension but we
have not been able to figure out how to allow a 4 digit transfer. Here
is the SRST config:

call-manager-fallback
 max-conferences 4 gain -6
 transfer-system full-blind
 ip source-address 10.120.224.1 port 2000 strict-match
 max-ephones 40
 max-dn 80 dual-line
 transfer-pattern 1716849....
 transfer-pattern 2...
 voicemail 914103476301
 translation-profile incoming 2
 translation-profile outgoing 2
 call-forward busy 17168492502
 call-forward noan 17168492501 timeout 16
 time-zone 12                     

and the associated translation

voice translation-rule 2
 rule 1 /^\(2...\)$/ /1716849\1/  

voice translation-profile 2
 translate calling 2
 translate called 2    

The translation rule is good because inbound 4 digit DID works when in
SRST, any advice would be appreciated.

         



Thanks!

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026


From rratliff at cisco.com  Sat Sep  2 11:30:49 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Sat, 2 Sep 2006 11:30:49 -0400
Subject: [cisco-voip] Cisco voice gw - caller id
In-Reply-To: <000301c6ce45$60023320$0f0110ac@corporation2>
References: <mailman.2039.1157147342.1681.cisco-voip@puck.nether.net>
	<000301c6ce45$60023320$0f0110ac@corporation2>
Message-ID: <E51E3E83-21C9-41EF-9A00-5CABA30690F7@cisco.com>

Is your T1 a CAS trunk or PRI?


-Ryan

On Sep 2, 2006, at 12:07 AM, Bob A. Bowie wrote:

I have in my current lab setup the following:
CCM 4.1 ->3825 -- t1 --3825 <-- CCM 4.1

I am using the 1MFT1 module between routers back to back t1 cable.
I have configured the H323 Gw on both CCM's and the Cisco IOS Voice GW
routers.
I can call between the CCM's but get "Unknown number" on the receving IP
Phone.

What do I need to do to pass the caller id info?
Can I pass the caller-id info with H323 and T1 or do I need MGCP/PRI?

The caller-id command does not seem to be available under the voice-port
config.
I am using 12.4 ip adv service IOS on the Voice GW routers.

Thanks



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


From rratliff at cisco.com  Sat Sep  2 11:34:06 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Sat, 2 Sep 2006 11:34:06 -0400
Subject: [cisco-voip] CME calling name
In-Reply-To: <001d01c6ce8a$d5930a70$c801a8c0@EATIUSBIRDIUS>
References: <30A8B4535CC19243A7C505E5156642D9277C09@gbgsrv02.gbg.invid.local>
	<001d01c6ce8a$d5930a70$c801a8c0@EATIUSBIRDIUS>
Message-ID: <EBE5C82E-6D65-46C8-862D-7C42908C13DA@cisco.com>

Configure a name under the ephone-dn that is making the call.

Now if you are using a PRI under the serial x/y:23 interface  
configure 'isdn outgoing display-ie' and CME will provide calling  
name to the PSTN.
You can verify this via 'debug isdn q931'.

As Lelio mentioned depending on where you are located the remote  
telco may or may not pay any attention whatsoever to the name you  
provide.

-Ryan

On Sep 2, 2006, at 8:25 AM, Lelio Fulgenzi wrote:

I spent some time investigating this. I'm sure I'll be corrected if  
I've got anything slighty ajar.

Some background. In the U.S. the calling name is presented to the  
called party by the called party's service provider via SS7 database  
lookups. Meanwhile, in Canada, the calling name is presented to the  
called party by the calling party's service provider.

While CallManager can be configured to 'allow calling party name' out  
PRIs, the value used is the same calling party name used for internal  
calls. There is no option in CallManager to override the internal  
calling party name with an enterprise calling party name - it is  
expecting the PSTN to do that for you. Unfortunately, in Canada, the  
PRI service model does not allow for that. Supposedly, it does, but  
the service providers have not paid the licencing fees or something  
like that.

Anyways, I'm not sure how much of this translates to CME, but it  
might be a start.

If you're looking for the internal calling name to be passed, check  
the system parameters (or equiv) on CME. If you're looking for an  
enterprise parameter, ask your SP if they can override it.

Good Luck


----- Original Message -----
From: Marcus Lundbom
To: cisco-voip at puck.nether.net
Sent: Saturday, September 02, 2006 7:33 AM
Subject: [cisco-voip] CME calling name

Hi list,

Would it be possible to configure a CME to actually pass calling name to
an outgoing PRI? If so, can someone please enlighten me on how? :)

Best regards,


/M

_______________________________________________
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  Sat Sep  2 11:41:05 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Sat, 2 Sep 2006 11:41:05 -0400
Subject: [cisco-voip] SRST transfer user 4 digits to an 11 digit
	extension
In-Reply-To: <s4f9687e.046@comrcm01intgwia.mandtbank.com>
References: <s4f9687e.046@comrcm01intgwia.mandtbank.com>
Message-ID: <297B60A7-CBDB-4479-9DE4-5C3D3121875E@cisco.com>

I haven't played with the transfer-pattern config too much so I'm not  
sure what affect that would have on this behavior.

Have you tried applying the translation-profile to a pots dial-peer  
with a destination-pattern of 2...$?

How about running a 'debug voice translation' to see if it's getting  
applied?

-Ryan

On Sep 2, 2006, at 11:17 AM, STEVEN CASPER wrote:


  We use 11 digit extensions at our site and would like to be able to
transfer using 4 digits when in SRST mode just as can be done when not
in SRST. Transfer works fine to an the actual 11 digit extension but we
have not been able to figure out how to allow a 4 digit transfer. Here
is the SRST config:

call-manager-fallback
  max-conferences 4 gain -6
  transfer-system full-blind
  ip source-address 10.120.224.1 port 2000 strict-match
  max-ephones 40
  max-dn 80 dual-line
  transfer-pattern 1716849....
  transfer-pattern 2...
  voicemail 914103476301
  translation-profile incoming 2
  translation-profile outgoing 2
  call-forward busy 17168492502
  call-forward noan 17168492501 timeout 16
  time-zone 12

and the associated translation

voice translation-rule 2
  rule 1 /^\(2...\)$/ /1716849\1/

voice translation-profile 2
  translate calling 2
  translate called 2

The translation rule is good because inbound 4 digit DID works when in
SRST, any advice would be appreciated.





Thanks!

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

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


From SCASPER at mandtbank.com  Sat Sep  2 12:16:10 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Sat, 02 Sep 2006 12:16:10 -0400
Subject: [cisco-voip] SRST transfer user 4 digits to an 11 digitextension
Message-ID: <s4f97623.065@comrcm01intgwia.mandtbank.com>

Thanks! Does a POTS dial peer need a destination like a Voice port or a
IP address or can it just be used to route calls "internal" to the
router?




Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

>>> "Ryan Ratliff" <rratliff at cisco.com> 09/02/06 11:41 AM >>>
I haven't played with the transfer-pattern config too much so I'm not 

sure what affect that would have on this behavior.

Have you tried applying the translation-profile to a pots dial-peer  
with a destination-pattern of 2...$?

How about running a 'debug voice translation' to see if it's getting  
applied?

-Ryan

On Sep 2, 2006, at 11:17 AM, STEVEN CASPER wrote:


  We use 11 digit extensions at our site and would like to be able to
transfer using 4 digits when in SRST mode just as can be done when not
in SRST. Transfer works fine to an the actual 11 digit extension but
we
have not been able to figure out how to allow a 4 digit transfer. Here
is the SRST config:

call-manager-fallback
  max-conferences 4 gain -6
  transfer-system full-blind
  ip source-address 10.120.224.1 port 2000 strict-match
  max-ephones 40
  max-dn 80 dual-line
  transfer-pattern 1716849....
  transfer-pattern 2...
  voicemail 914103476301
  translation-profile incoming 2
  translation-profile outgoing 2
  call-forward busy 17168492502
  call-forward noan 17168492501 timeout 16
  time-zone 12

and the associated translation

voice translation-rule 2
  rule 1 /^\(2...\)$/ /1716849\1/

voice translation-profile 2
  translate calling 2
  translate called 2

The translation rule is good because inbound 4 digit DID works when in
SRST, any advice would be appreciated.





Thanks!

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

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




From rratliff at cisco.com  Sat Sep  2 12:35:08 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Sat, 2 Sep 2006 12:35:08 -0400
Subject: [cisco-voip] SRST transfer user 4 digits to an 11 digitextension
In-Reply-To: <s4f97623.064@comrcm01intgwia.mandtbank.com>
References: <s4f97623.064@comrcm01intgwia.mandtbank.com>
Message-ID: <DCDE1280-C0E9-4D58-B2B0-2ECC4627B50C@cisco.com>

The POTS dial-peer would need a destination if you want to use it as  
an outbound dial-peer (which you do).  Your goal here it to allow  
people in SRST mode to dial 4 digits and route via the PSTN to other  
sites correct?

-Ryan

On Sep 2, 2006, at 12:16 PM, STEVEN CASPER wrote:

Thanks! Does a POTS dial peer need a destination like a Voice port or a
IP address or can it just be used to route calls "internal" to the
router?




Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

>>> "Ryan Ratliff" <rratliff at cisco.com> 09/02/06 11:41 AM >>>
I haven't played with the transfer-pattern config too much so I'm not

sure what affect that would have on this behavior.

Have you tried applying the translation-profile to a pots dial-peer
with a destination-pattern of 2...$?

How about running a 'debug voice translation' to see if it's getting
applied?

-Ryan

On Sep 2, 2006, at 11:17 AM, STEVEN CASPER wrote:


   We use 11 digit extensions at our site and would like to be able to
transfer using 4 digits when in SRST mode just as can be done when not
in SRST. Transfer works fine to an the actual 11 digit extension but
we
have not been able to figure out how to allow a 4 digit transfer. Here
is the SRST config:

call-manager-fallback
   max-conferences 4 gain -6
   transfer-system full-blind
   ip source-address 10.120.224.1 port 2000 strict-match
   max-ephones 40
   max-dn 80 dual-line
   transfer-pattern 1716849....
   transfer-pattern 2...
   voicemail 914103476301
   translation-profile incoming 2
   translation-profile outgoing 2
   call-forward busy 17168492502
   call-forward noan 17168492501 timeout 16
   time-zone 12

and the associated translation

voice translation-rule 2
   rule 1 /^\(2...\)$/ /1716849\1/

voice translation-profile 2
   translate calling 2
   translate called 2

The translation rule is good because inbound 4 digit DID works when in
SRST, any advice would be appreciated.





Thanks!

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

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


From SCASPER at mandtbank.com  Sat Sep  2 13:52:43 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Sat, 02 Sep 2006 13:52:43 -0400
Subject: [cisco-voip] SRST transfer user 4 digits to an 11digitextension
Message-ID: <s4f98cd7.008@comrcm01intgwia.mandtbank.com>

No, I would like to be able to do a 4 digit transfer to another phone at
the same site. We use 11 digit extensions but mask them. Under normal
conditions the users can dial/transfer using 4 digits to other phones at
their site. I am hoping to duplicate this functionality under SRST.

Steve

>>> "Ryan Ratliff" <rratliff at cisco.com> 09/02/06 12:35 PM >>>
The POTS dial-peer would need a destination if you want to use it as  
an outbound dial-peer (which you do).  Your goal here it to allow  
people in SRST mode to dial 4 digits and route via the PSTN to other  
sites correct?

-Ryan

On Sep 2, 2006, at 12:16 PM, STEVEN CASPER wrote:

Thanks! Does a POTS dial peer need a destination like a Voice port or
a
IP address or can it just be used to route calls "internal" to the
router?




Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

>>> "Ryan Ratliff" <rratliff at cisco.com> 09/02/06 11:41 AM >>>
I haven't played with the transfer-pattern config too much so I'm not

sure what affect that would have on this behavior.

Have you tried applying the translation-profile to a pots dial-peer
with a destination-pattern of 2...$?

How about running a 'debug voice translation' to see if it's getting
applied?

-Ryan

On Sep 2, 2006, at 11:17 AM, STEVEN CASPER wrote:


   We use 11 digit extensions at our site and would like to be able to
transfer using 4 digits when in SRST mode just as can be done when not
in SRST. Transfer works fine to an the actual 11 digit extension but
we
have not been able to figure out how to allow a 4 digit transfer. Here
is the SRST config:

call-manager-fallback
   max-conferences 4 gain -6
   transfer-system full-blind
   ip source-address 10.120.224.1 port 2000 strict-match
   max-ephones 40
   max-dn 80 dual-line
   transfer-pattern 1716849....
   transfer-pattern 2...
   voicemail 914103476301
   translation-profile incoming 2
   translation-profile outgoing 2
   call-forward busy 17168492502
   call-forward noan 17168492501 timeout 16
   time-zone 12

and the associated translation

voice translation-rule 2
   rule 1 /^\(2...\)$/ /1716849\1/

voice translation-profile 2
   translate calling 2
   translate called 2

The translation rule is good because inbound 4 digit DID works when in
SRST, any advice would be appreciated.





Thanks!

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

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




From wsisk at cisco.com  Sat Sep  2 21:53:48 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Sat, 2 Sep 2006 21:53:48 -0400
Subject: [cisco-voip] SNMP monitoring MGCP gateway
In-Reply-To: <F6F72F553C7E754A945CB88CD8804B42417F21@exchange01.semaphore.lan>
References: <F6F72F553C7E754A945CB88CD8804B42417F21@exchange01.semaphore.lan>
Message-ID: <FDFFA176-0CF1-4A7F-B99A-B07107A6A7CC@cisco.com>

Dylan,

Is CSCsa67949 what you are looking for:
<B>Symptom:</B>
No way to determine from CCM when all outbound PRI trunks are busy.

<B>Conditions:</B>
CallManager MGCP PRI endpoints.

<B>Workaround:</B>
If an IOS gateway, use SNMP to poll
1.3.6.1.4.1.9.10.19.1.1.9.1
Description  	An entry in the DS0 Usage table.

Use cpmDS1UsageSlotIndex and cpmDS1UsagePortIndex to specify which  
ds1,then use  cpmDS1ActiveDS0s to find the number of active calls per  
ds1.

If 6608 or dt-24+ there is no workaround other than possibly ART/CAR  
Gateway utilization reports



/Wes

On Sep 1, 2006, at 7:28 PM, Dylan Vanderhoof wrote:

Replying to myself, but I need to restate the question, since some
snmpwalking of a few callmanagers of different versions showed nearly
nothing use.

Can I get usage stats on trunks/gateways via snmp at all with an MGCP
backhaul in place?  Am I missing some obvious OID on my VG200 gateway?

I know AXL is available, but its not really an option at this point for
what I'm looking at.

Thanks,
Dylan

> -----Original Message-----
> From: Dylan Vanderhoof
> Sent: Friday, September 01, 2006 12:32 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] SNMP monitoring MGCP gateway
>
>
> Is there any useful information that can be polled via SNMP on an MGCP
> voice gateway?  I'm looking at my VG200 and not finding much (I assume
> due to the backhaul), but I figured I'd ask.
>
> Obviously call stats are available via SNMP on the callmanager.
>
> Thanks,
> Dylan
>

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


From erickbe at yahoo.com  Sun Sep  3 00:56:04 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sat, 2 Sep 2006 21:56:04 -0700 (PDT)
Subject: [cisco-voip] SRST transfer user 4 digits to an 11digitextension
In-Reply-To: <s4f98cd7.008@comrcm01intgwia.mandtbank.com>
Message-ID: <20060903045604.23960.qmail@web37208.mail.mud.yahoo.com>

Have you tried using translate called, or alias under call-manager-fallback?

Have not personally tried calling a alias # from a IP phone in fallback, so not sure if that will work for calls from IP Phones offhand. It is meant for incoming calls. 


----- Original Message ----
From: STEVEN CASPER <SCASPER at mandtbank.com>
To: rratliff at cisco.com
Cc: cisco-voip at puck.nether.net
Sent: Saturday, September 2, 2006 12:52:43 PM
Subject: Re: [cisco-voip] SRST transfer user 4 digits to an 11digitextension

No, I would like to be able to do a 4 digit transfer to another phone at
the same site. We use 11 digit extensions but mask them. Under normal
conditions the users can dial/transfer using 4 digits to other phones at
their site. I am hoping to duplicate this functionality under SRST.

Steve

>>> "Ryan Ratliff" <rratliff at cisco.com> 09/02/06 12:35 PM >>>
The POTS dial-peer would need a destination if you want to use it as  
an outbound dial-peer (which you do).  Your goal here it to allow  
people in SRST mode to dial 4 digits and route via the PSTN to other  
sites correct?

-Ryan

On Sep 2, 2006, at 12:16 PM, STEVEN CASPER wrote:

Thanks! Does a POTS dial peer need a destination like a Voice port or
a
IP address or can it just be used to route calls "internal" to the
router?




Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

>>> "Ryan Ratliff" <rratliff at cisco.com> 09/02/06 11:41 AM >>>
I haven't played with the transfer-pattern config too much so I'm not

sure what affect that would have on this behavior.

Have you tried applying the translation-profile to a pots dial-peer
with a destination-pattern of 2...$?

How about running a 'debug voice translation' to see if it's getting
applied?

-Ryan

On Sep 2, 2006, at 11:17 AM, STEVEN CASPER wrote:


   We use 11 digit extensions at our site and would like to be able to
transfer using 4 digits when in SRST mode just as can be done when not
in SRST. Transfer works fine to an the actual 11 digit extension but
we
have not been able to figure out how to allow a 4 digit transfer. Here
is the SRST config:

call-manager-fallback
   max-conferences 4 gain -6
   transfer-system full-blind
   ip source-address 10.120.224.1 port 2000 strict-match
   max-ephones 40
   max-dn 80 dual-line
   transfer-pattern 1716849....
   transfer-pattern 2...
   voicemail 914103476301
   translation-profile incoming 2
   translation-profile outgoing 2
   call-forward busy 17168492502
   call-forward noan 17168492501 timeout 16
   time-zone 12

and the associated translation

voice translation-rule 2
   rule 1 /^\(2...\)$/ /1716849\1/

voice translation-profile 2
   translate calling 2
   translate called 2

The translation rule is good because inbound 4 digit DID works when in
SRST, any advice would be appreciated.





Thanks!

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

_______________________________________________
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 millerman1 at cox.net  Sun Sep  3 08:44:37 2006
From: millerman1 at cox.net (Steve Miller)
Date: Sun, 3 Sep 2006 08:44:37 -0400
Subject: [cisco-voip] Allowing Limited Admin Rights for Unity Passcode
	Changes?
References: <20060903045604.23960.qmail@web37208.mail.mud.yahoo.com>
Message-ID: <000701c6cf56$b78df9f0$6401a8c0@millersw>

      Allowing Limited Admin Rights for Unity Passcode Changes

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


As instructed by "Configuring Call Manager and Unity" (book), I have created
a new class of service and template which is supposed to allow subscribers
with this class of service to change Unity passcodes and re-enable
mailboxes, but it either doesn't work or I don't understand how it is
supposed to work. Is this ability to be accessed from the mailbox itself
(TUI)? I changed my mailbox to the new template/COS, but there is no change
that I can see. What am I missing? This must be an easy question, but no one 
has chimed in.
Please help! Thanks.

Steve



From PEKOU at gr.ibm.com  Mon Sep  4 05:45:59 2006
From: PEKOU at gr.ibm.com (Evi Pekou)
Date: Mon, 4 Sep 2006 12:45:59 +0300
Subject: [cisco-voip] Advanced transfer feature
Message-ID: <OFD18B6859.8B47DC96-ONC22571DF.0032AD82-C22571DF.003545D8@gr.ibm.com>

Does the Callmanager (4.1.3) support the following feature:
an incoming call answered by the operator and transfered to an internal IP 
Phone should  return to the operator after the internal phone hangs up, 
so that the operator can transfer it to another IP Phone if requested by 
the caller?

Can anyone think of a way to implement this?

Thanks,
Evi
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From PEKOU at gr.ibm.com  Mon Sep  4 06:30:51 2006
From: PEKOU at gr.ibm.com (Evi Pekou)
Date: Mon, 4 Sep 2006 13:30:51 +0300
Subject: [cisco-voip] Barge in
Message-ID: <OF92193E03.C3CB2E81-ONC22571DF.0038E63F-C22571DF.00396192@gr.ibm.com>

Is it possible, when using the barge in feature, for the other party in 
the existing conversation (not the one that is barged in) 
not to hear the person that initiated the barge in speaking?

Thanks,
Evi
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From nmarus at gmail.com  Mon Sep  4 10:51:31 2006
From: nmarus at gmail.com (Nick Marus)
Date: Mon, 4 Sep 2006 10:51:31 -0400
Subject: [cisco-voip] Barge in
In-Reply-To: <OF92193E03.C3CB2E81-ONC22571DF.0038E63F-C22571DF.00396192@gr.ibm.com>
References: <OF92193E03.C3CB2E81-ONC22571DF.0038E63F-C22571DF.00396192@gr.ibm.com>
Message-ID: <1a9f6d3f0609040751v462b2643ud00d1c3a1c9b4389@mail.gmail.com>

The feature you are talking about is called "whisper page" and as far as I
know cant be done on cisco without the use of a 3rd party product. Look at
ipcelerate as they have soemthing that can do this.

On 9/4/06, Evi Pekou <PEKOU at gr.ibm.com> wrote:
>
>
> Is it possible, when using the barge in feature, for the other party in
> the existing conversation (not the one that is barged in)
> not to hear the person that initiated the barge in speaking?
>
> Thanks,
>
> Evi
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Nick Marus
nmarus at gmail.com
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From jgrace at digitel.net  Mon Sep  4 14:17:02 2006
From: jgrace at digitel.net (James Grace)
Date: Mon, 4 Sep 2006 14:17:02 -0400
Subject: [cisco-voip] unity log on accts
Message-ID: <200609041416640.SM01472@jgm>

We have a site with unified messaging,  and we want to find a way to
delegate the admin responsibilities.  We have the admin acct that was
created during the install and it works find.  But is there a way to create
groups in AD and assign the users in that group correct rights to administer
unity

 

James D. Grace 

CCNP CCNA MCSE MCSA 

Digitel Corporation

System Engineer / Professional Svc.

MSN messenger ID:  jgrace29 at hotmail.com

Mac IChat ID:  jgrace29 at mac.com

 

 

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From pwalenta at wi.rr.com  Mon Sep  4 15:05:39 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Mon, 4 Sep 2006 14:05:39 -0500
Subject: [cisco-voip] unity log on accts
In-Reply-To: <200609041416640.SM01472@jgm>
Message-ID: <000901c6d055$1cea42f0$0232a8c0@pwnhome.com>

There is a way in Unity to create a CoS - class of service - that determines
what people can and cannot do in the Unity web admin interface.  I'd start
there.

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Monday, September 04, 2006 1:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] unity log on accts



We have a site with unified messaging,  and we want to find a way to
delegate the admin responsibilities.  We have the admin acct that was
created during the install and it works find.  But is there a way to create
groups in AD and assign the users in that group correct rights to administer
unity

 

James D. Grace 

CCNP CCNA MCSE MCSA 

Digitel Corporation

System Engineer / Professional Svc.

MSN messenger ID:  jgrace29 at hotmail.com

Mac IChat ID:  jgrace29 at mac.com

 

 

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From voip at pentenrieder.de  Mon Sep  4 16:54:27 2006
From: voip at pentenrieder.de (Andreas Pentenrieder)
Date: Mon, 4 Sep 2006 22:54:27 +0200
Subject: [cisco-voip] Unity
Message-ID: <001a01c6d064$5040aa40$6721a8c0@avodaq.com>

HI There,


I have got questions regarding Unity/ Unity Connection. May be there is somebody who can help me:


1. User PIN
Is it possible to disable the PIN verification for subscribers? As soon as the subscriber dials the voicemail number he is authenticated (so no PIN verification).

2. Scenario with Unity Connection

- Two IP Phones
- First IP Phone DN: 1000
- Second IP Phone DN: 2000
- One Voicemail box for DN 1000, with an alternate extension 2000
- Several new voice messages for DN 1000

First IP Phone calls Unity Connection and presses 1 to listen to new voice messages. During first IP Phone is listening to new voice message, the second IP Phone calls the Unity Connection, too. Presses 1 to listen to new voice messages.
What will happen? Will the second IP Phone here the next voice message (the message after the message currently playing on the first IP Phone), the same voice message as the first IP Phone, etc?


Thanks for your help


Andreas Pentenrieder

CCIE Voice, CCVP, CCNP, CCNA, CCSI
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From luis.simonet at gmail.com  Tue Sep  5 01:16:57 2006
From: luis.simonet at gmail.com (Luis Simonet)
Date: Tue, 5 Sep 2006 01:16:57 -0400
Subject: [cisco-voip] Cisco Unity Connection 1.2.1
Message-ID: <e268831b0609042216g3c0469wf9ffe4cb32b963a0@mail.gmail.com>

I need to install Unity Connection on an existing Windows server 2003, I
will used it for training. I have the license from Cisco. But when I try to
run the installation it wants me to install 2003 from the DVD that came with
Unity Connection. Is there a way to bypass it and use the version of 2003
that is already install.

Thanks
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From jgrace at digitel.net  Tue Sep  5 09:04:54 2006
From: jgrace at digitel.net (James Grace)
Date: Tue, 5 Sep 2006 09:04:54 -0400
Subject: [cisco-voip] MOH drop folder issue in callmanager
Message-ID: <200609050904906.SM01472@jgm>

For some reason im having issue with drop mp3 or wav files into the
dropaudiosoucehere dir on callmanager.  I may be doing something wrong here.
First thig is when I drop on the pub nothing happens.  Secondly I have 2
wav. Files I drop 1 in the subscribers folder and it works but when I drop
the other file it does nothing.  The 2 wav file are the same size, format
etc.  They are 2 different wav files.  Ive rebooted both server and check
the translater service on both pub and sub

 

James D. Grace 

CCNP CCNA MCSE MCSA 

Digitel Corporation

System Engineer / Professional Svc.

MSN messenger ID:  jgrace29 at hotmail.com

Mac IChat ID:  jgrace29 at mac.com

 

 

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From rratliff at cisco.com  Tue Sep  5 09:12:30 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 5 Sep 2006 09:12:30 -0400
Subject: [cisco-voip] SRST transfer user 4 digits to an 11digitextension
In-Reply-To: <20060903045604.23960.qmail@web37208.mail.mud.yahoo.com>
References: <20060903045604.23960.qmail@web37208.mail.mud.yahoo.com>
Message-ID: <6DDD29EE-1777-46A6-A20D-00F7DF204DF8@cisco.com>

I messed with it a bit in CME but haven't had much luck getting  
translation-profiles applied at the ephone-dn.  I assume the same  
issues would be seen setting them globally in the call-manager- 
fallback config.

The definition of the transfer-pattern command implies that it only  
controls the digits you are allowed to transfer to.  By default the  
system only allows you to transfer to other IP phones.  In SRST I  
assume this means that 11 digit transfers would work by default.   
This would lead me to believe your current config of the 4-digit  
transfer-pattern and the profile applied globally would work.

-Ryan

On Sep 3, 2006, at 12:56 AM, Erick Bergquist wrote:

Have you tried using translate called, or alias under call-manager- 
fallback?

Have not personally tried calling a alias # from a IP phone in  
fallback, so not sure if that will work for calls from IP Phones  
offhand. It is meant for incoming calls.


----- Original Message ----
From: STEVEN CASPER <SCASPER at mandtbank.com>
To: rratliff at cisco.com
Cc: cisco-voip at puck.nether.net
Sent: Saturday, September 2, 2006 12:52:43 PM
Subject: Re: [cisco-voip] SRST transfer user 4 digits to an  
11digitextension

No, I would like to be able to do a 4 digit transfer to another phone at
the same site. We use 11 digit extensions but mask them. Under normal
conditions the users can dial/transfer using 4 digits to other phones at
their site. I am hoping to duplicate this functionality under SRST.

Steve

>>> "Ryan Ratliff" <rratliff at cisco.com> 09/02/06 12:35 PM >>>
The POTS dial-peer would need a destination if you want to use it as
an outbound dial-peer (which you do).  Your goal here it to allow
people in SRST mode to dial 4 digits and route via the PSTN to other
sites correct?

-Ryan

On Sep 2, 2006, at 12:16 PM, STEVEN CASPER wrote:

Thanks! Does a POTS dial peer need a destination like a Voice port or
a
IP address or can it just be used to route calls "internal" to the
router?




Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

>>> "Ryan Ratliff" <rratliff at cisco.com> 09/02/06 11:41 AM >>>
I haven't played with the transfer-pattern config too much so I'm not

sure what affect that would have on this behavior.

Have you tried applying the translation-profile to a pots dial-peer
with a destination-pattern of 2...$?

How about running a 'debug voice translation' to see if it's getting
applied?

-Ryan

On Sep 2, 2006, at 11:17 AM, STEVEN CASPER wrote:


    We use 11 digit extensions at our site and would like to be able to
transfer using 4 digits when in SRST mode just as can be done when not
in SRST. Transfer works fine to an the actual 11 digit extension but
we
have not been able to figure out how to allow a 4 digit transfer. Here
is the SRST config:

call-manager-fallback
    max-conferences 4 gain -6
    transfer-system full-blind
    ip source-address 10.120.224.1 port 2000 strict-match
    max-ephones 40
    max-dn 80 dual-line
    transfer-pattern 1716849....
    transfer-pattern 2...
    voicemail 914103476301
    translation-profile incoming 2
    translation-profile outgoing 2
    call-forward busy 17168492502
    call-forward noan 17168492501 timeout 16
    time-zone 12

and the associated translation

voice translation-rule 2
    rule 1 /^\(2...\)$/ /1716849\1/

voice translation-profile 2
    translate calling 2
    translate called 2

The translation rule is good because inbound 4 digit DID works when in
SRST, any advice would be appreciated.





Thanks!

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

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From voipermike at gmail.com  Tue Sep  5 09:36:04 2006
From: voipermike at gmail.com (Mike Mike)
Date: Tue, 5 Sep 2006 08:36:04 -0500
Subject: [cisco-voip] MOH drop folder issue in callmanager
In-Reply-To: <200609050904906.SM01472@jgm>
References: <200609050904906.SM01472@jgm>
Message-ID: <8bbfc1f40609050636t6d1edf9dq43f427512675a467@mail.gmail.com>

A former peer of mine discovered a similar if not duplicate issue. Here is a
copy of the info he provided:

Scenario:



User drops a wav file or mp3 into the DropMoHFilesHere directory and it just
sits there, it will not convert?



Problem:



File is not inheriting permissions from parent directory and you get an
'file access denied' in the traces? Even though 'Allow inheritable
permissions from parent to propagate to this object' is checked on the
parent directory.



Solution:



Right click on the file in question, go to properties, click the security
tab, click add, sort by name of account and add these accounts to the file
permissions: CCMCDR, CCMService, CCMServiceRW and CCMUser. When you hit
apply, the file will disappear? If not, give each of these accounts full
control.


Hope this helps!

Mike

On 9/5/06, James Grace <jgrace at digitel.net> wrote:
>
>  For some reason im having issue with drop mp3 or wav files into the
> dropaudiosoucehere dir on callmanager.  I may be doing something wrong
> here.  First thig is when I drop on the pub nothing happens.  Secondly I
> have 2 wav. Files I drop 1 in the subscribers folder and it works but when I
> drop the other file it does nothing.  The 2 wav file are the same size,
> format etc.  They are 2 different wav files.  Ive rebooted both server and
> check the translater service on both pub and sub
>
>
>
> James D. Grace
>
> CCNP CCNA MCSE MCSA
>
> Digitel Corporation
>
> System Engineer / Professional Svc.
>
> MSN messenger ID:  jgrace29 at hotmail.com
>
> Mac IChat ID:  jgrace29 at mac.com
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From CORTIZ at broward.org  Tue Sep  5 09:36:39 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue, 5 Sep 2006 09:36:39 -0400
Subject: [cisco-voip] Allowing Limited Admin Rights for Unity
	PasscodeChanges?
In-Reply-To: <000701c6cf56$b78df9f0$6401a8c0@millersw>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A47489@MREXVS01.bc.broward.cty>

Did you go in to the Class Of service >> System Access Area and change
to the desired access for this COS?  I use this same method to allow the
helpdesk to reset passwords and such.

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Steve Miller
Sent: Sunday, September 03, 2006 8:45 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Allowing Limited Admin Rights for Unity
PasscodeChanges?

      Allowing Limited Admin Rights for Unity Passcode Changes

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


As instructed by "Configuring Call Manager and Unity" (book), I have
created
a new class of service and template which is supposed to allow
subscribers
with this class of service to change Unity passcodes and re-enable
mailboxes, but it either doesn't work or I don't understand how it is
supposed to work. Is this ability to be accessed from the mailbox itself
(TUI)? I changed my mailbox to the new template/COS, but there is no
change
that I can see. What am I missing? This must be an easy question, but no
one 
has chimed in.
Please help! Thanks.

Steve


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From MLinsemier at apcapital.com  Tue Sep  5 09:45:07 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Tue, 5 Sep 2006 09:45:07 -0400
Subject: [cisco-voip] Which Cisco VoIP Products properly mark DSCP
Message-ID: <AB0A35F2486F81468971AA68558BE71904AD22C2@smieln01winm01.micoanet.com>

All,

 

I am curious if anyone knows of a matrix or table that states what Cisco
VoIP products and versions properly set DSCP values once they pass
information onto the network.  I am specifically looking for the
following products.

 

*	CallManager
*	Unity
*	IPCC Express
*	MeetingPlace Express

 

It would be nice to know what products I can simply trust DSCP values
and which other ones will have to have custom QoS that properly marks
voice.

 

Sincerely,

 

Matthew M. Linsemier, CCSP/CCNP/CCDA
Senior Network Engineer
American Physicians
Voice: 517.324.6695
Fax: 517.333.2812
E-Mail: mlinsemier at apcapital.com

 


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From gpate at kenttech.com  Tue Sep  5 10:26:17 2006
From: gpate at kenttech.com (Gary L. Pate)
Date: Tue, 5 Sep 2006 10:26:17 -0400
Subject: [cisco-voip] Cisco's Emergency Responder
Message-ID: <002e01c6d0f7$400de5f0$0d1e14ac@gpatelt>

Can anyone recommend any training resources on Cisco's Emergency Responder?
I've used Global Knowledge in the past for other courses, but they do not
have an ER course. I have no experience with ER, but would like to learn for
future implementations. Any suggested resource would be great!

 

Gary Pate

Kent Technologies Inc.

 

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From lohnert at gmail.com  Tue Sep  5 11:02:17 2006
From: lohnert at gmail.com (Martin Lohnert)
Date: Tue, 5 Sep 2006 17:02:17 +0200
Subject: [cisco-voip] facility ie
Message-ID: <1bfd8b6f0609050802k243153afm9e7915937b795e22@mail.gmail.com>

Hello,
Is there any way to find out what an unsupported facility ie actually is? I
have a q.sig E1 to a legacy pbx, and in debug q.931 we see the following:
----------------------
Aug 24 10:52:06.868: ISDN Se0/2/1:15 Q931: RX <- FACILITY pd = 8  callref =
0x274B
        Facility i =
0x9FAA06800100820100A11002021B6D06072B0C028852B66D160131
                Protocol Profile =  Networking Extensions
                0x0100
                0x820100
                0xA11002021B6D06072B0C028852B66D160131
                Component = Invoke component, Unsupported operation
Aug 24 10:52:06.872: ISDN  **ERROR**: isdnAsn1Component: error: unknown type
10
Aug 24 10:52:06.872: ISDN  **ERROR**: isdn_decode_facility_ie:
isdnAsn1DecComponent failed
-----------------------

Cheers, and TIA!
ML
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From jneiberger at gmail.com  Tue Sep  5 11:05:16 2006
From: jneiberger at gmail.com (John Neiberger)
Date: Tue, 5 Sep 2006 09:05:16 -0600
Subject: [cisco-voip] facility ie
In-Reply-To: <1bfd8b6f0609050802k243153afm9e7915937b795e22@mail.gmail.com>
References: <1bfd8b6f0609050802k243153afm9e7915937b795e22@mail.gmail.com>
Message-ID: <547ad0fe0609050805l5db4207fj2369cb010fd57632@mail.gmail.com>

For similar reasons, I would *love* to find some sort of facility
message decoder. Let us know if you find one anywhere.

Thanks,
John

On 9/5/06, Martin Lohnert <lohnert at gmail.com> wrote:
>
> Hello,
> Is there any way to find out what an unsupported facility ie actually is? I
> have a q.sig E1 to a legacy pbx, and in debug q.931 we see the following:
> ----------------------
> Aug 24 10:52:06.868: ISDN Se0/2/1:15 Q931: RX <- FACILITY pd = 8  callref =
> 0x274B
>         Facility i =
> 0x9FAA06800100820100A11002021B6D06072B0C028852B66D160131
>                 Protocol Profile =  Networking Extensions
>                 0x0100
>                 0x820100
>                 0xA11002021B6D06072B0C028852B66D160131
>                 Component = Invoke component, Unsupported operation
> Aug 24 10:52:06.872: ISDN  **ERROR**: isdnAsn1Component: error: unknown type
> 10
> Aug 24 10:52:06.872: ISDN  **ERROR**: isdn_decode_facility_ie:
> isdnAsn1DecComponent failed
> -----------------------
>
> Cheers, and TIA!
>
> ML
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From rratliff at cisco.com  Tue Sep  5 11:14:10 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 5 Sep 2006 11:14:10 -0400
Subject: [cisco-voip] facility ie
In-Reply-To: <1bfd8b6f0609050802k243153afm9e7915937b795e22@mail.gmail.com>
References: <1bfd8b6f0609050802k243153afm9e7915937b795e22@mail.gmail.com>
Message-ID: <092C8FC5-75BD-473F-8C70-D711BAF6E38F@cisco.com>

If the call is going to CallManager we do a decent job of decoding  
them in the ccm traces.

-Ryan

On Sep 5, 2006, at 11:02 AM, Martin Lohnert wrote:

Hello,
Is there any way to find out what an unsupported facility ie actually  
is? I have a q.sig E1 to a legacy pbx, and in debug q.931 we see the  
following:
----------------------
Aug 24 10:52:06.868: ISDN Se0/2/1:15 Q931: RX <- FACILITY pd = 8   
callref = 0x274B
         Facility i =  
0x9FAA06800100820100A11002021B6D06072B0C028852B66D160131
                 Protocol Profile =  Networking Extensions
                 0x0100
                 0x820100
                 0xA11002021B6D06072B0C028852B66D160131
                 Component = Invoke component, Unsupported operation
Aug 24 10:52:06.872: ISDN  **ERROR**: isdnAsn1Component: error:  
unknown type 10
Aug 24 10:52:06.872: ISDN  **ERROR**: isdn_decode_facility_ie:  
isdnAsn1DecComponent failed
-----------------------

Cheers, and TIA!
ML
_______________________________________________
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 Sep  5 11:24:42 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 5 Sep 2006 08:24:42 -0700
Subject: [cisco-voip] Unity
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C7973@Apollo.wesd.org>

I don't know about question 2 but there is a option to allow the user
calling from the correct extension not to enter a pin.  I don't remember
where, but it's in the newer versions of Unity.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Andreas
Pentenrieder
Sent: Monday, September 04, 2006 1:54 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity

 

HI There,

 

 

I have got questions regarding Unity/ Unity Connection. May be there is
somebody who can help me:

 

 

1. User PIN

Is it possible to disable the PIN verification for subscribers? As soon
as the subscriber dials the voicemail number he is authenticated (so no
PIN verification).

 

2. Scenario with Unity Connection

 

- Two IP Phones

- First IP Phone DN: 1000

- Second IP Phone DN: 2000

- One Voicemail box for DN 1000, with an alternate extension 2000

- Several new voice messages for DN 1000

 

First IP Phone calls Unity Connection and presses 1 to listen to new
voice messages. During first IP Phone is listening to new voice message,
the second IP Phone calls the Unity Connection, too. Presses 1 to listen
to new voice messages.

What will happen? Will the second IP Phone here the next voice message
(the message after the message currently playing on the first IP Phone),
the same voice message as the first IP Phone, etc?

 

 

Thanks for your help

 

 

Andreas Pentenrieder

 

CCIE Voice, CCVP, CCNP, CCNA, CCSI

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From wsisk at cisco.com  Tue Sep  5 11:28:29 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 05 Sep 2006 11:28:29 -0400
Subject: [cisco-voip] facility ie
In-Reply-To: <1bfd8b6f0609050802k243153afm9e7915937b795e22@mail.gmail.com>
References: <1bfd8b6f0609050802k243153afm9e7915937b795e22@mail.gmail.com>
Message-ID: <44FD979D.4090805@cisco.com>

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From gusangjung at yahoo.com  Tue Sep  5 11:30:22 2006
From: gusangjung at yahoo.com (Gu Sang Jung)
Date: Tue, 5 Sep 2006 08:30:22 -0700 (PDT)
Subject: [cisco-voip] Fax Over IP service with Catalysist 6509 T1-blade
Message-ID: <20060905153022.67533.qmail@web52008.mail.yahoo.com>

Hi There
   
  One of my customer want to integrate with Exchange mail with Fax service. Unfortuately, I can't use Cisco Fax IP service because customer use 6608-T1 Voice T1-card on Catalyst 6500 switch. 
  Until now, I can't find a solution to provide this situation. Any advise and help will be very appreciated.
   
  gusang jung
  CCIE VOICE&RS(12972),MCSE
  WI,USA

 		
---------------------------------
 All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.
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From Scott.Voll at wesd.org  Tue Sep  5 11:37:16 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 5 Sep 2006 08:37:16 -0700
Subject: [cisco-voip] Advanced transfer feature
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C797E@Apollo.wesd.org>

Make a translation of 0 for the operator  the have the user Transfer to
0.  Best I can think of.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Evi Pekou
Sent: Monday, September 04, 2006 2:46 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Advanced transfer feature

 


Does the Callmanager (4.1.3) support the following feature: 
an incoming call answered by the operator and transfered to an internal
IP Phone should  return to the operator after the internal phone hangs
up, 
so that the operator can transfer it to another IP Phone if requested by
the caller? 

Can anyone think of a way to implement this? 

Thanks, 
Evi

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From jason.aarons at us.didata.com  Tue Sep  5 12:56:10 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 5 Sep 2006 12:56:10 -0400
Subject: [cisco-voip] Cisco's Emergency Responder
Message-ID: <C1FE15183DA37645BC0633BC604E44F0030A1255@USNAEXCH.na.didata.local>

I'm guessing the issue is Cisco doesn't produce CER official curriculum.
The instructor would have to write their own manuals/labs.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gary L. Pate
Sent: Tuesday, September 05, 2006 10:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Cisco's Emergency Responder

 

Can anyone recommend any training resources on Cisco's Emergency
Responder? I've used Global Knowledge in the past for other courses, but
they do not have an ER course. I have no experience with ER, but would
like to learn for future implementations. Any suggested resource would
be great!

 

Gary Pate

Kent Technologies Inc.

 




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From ppauly at gmail.com  Tue Sep  5 16:01:49 2006
From: ppauly at gmail.com (Peter Pauly)
Date: Tue, 05 Sep 2006 16:01:49 -0400
Subject: [cisco-voip] Who called 911?
Message-ID: <44FDD7AD.6020404@gmail.com>

Every time our security people are informed that someone made a 911
call, they call me to find out who it was (extension, name). I run a
query in query analyzer against the CDR database.

Are there any applications available that the security guys could run
from their desktop that would query callmanager over the network and
display 911 calls? What you guys doing about this problem?

On a related question, our CallManager sends caller id information just
fine. When I call my cellphone, I can see the extension of where the
call came from. But when I make a test call to 911, the PSAP is only
displaying the common billing phone number, not the caller-id
information. Is my telco doing something wrong?

From lelio at uoguelph.ca  Tue Sep  5 16:10:23 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 5 Sep 2006 16:10:23 -0400
Subject: [cisco-voip] Who called 911?
References: <44FDD7AD.6020404@gmail.com>
Message-ID: <014c01c6d127$521cd9b0$47196883@cfs.uoguelph.ca>

We have similar requests, although not that often. Something I have thought about would be the following:
  a.. purchase Crystal Reports (the edition with web forms)
  b.. create a web form that queries the database 
  c.. present information to user
While you could go about this a different way, i.e. creating website with SQL calls to the database directly, the idea of Crystal Reports available for making multiple reports without rewriting new web pages appeals to me.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Peter Pauly 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, September 05, 2006 4:01 PM
  Subject: [cisco-voip] Who called 911?


  Every time our security people are informed that someone made a 911
  call, they call me to find out who it was (extension, name). I run a
  query in query analyzer against the CDR database.

  Are there any applications available that the security guys could run
  from their desktop that would query callmanager over the network and
  display 911 calls? What you guys doing about this problem?

  On a related question, our CallManager sends caller id information just
  fine. When I call my cellphone, I can see the extension of where the
  call came from. But when I make a test call to 911, the PSAP is only
  displaying the common billing phone number, not the caller-id
  information. Is my telco doing something wrong?
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From sil at infiltrated.net  Tue Sep  5 16:22:09 2006
From: sil at infiltrated.net (J. Oquendo)
Date: Tue, 05 Sep 2006 16:22:09 -0400
Subject: [cisco-voip] Who called 911?
In-Reply-To: <44FDD7AD.6020404@gmail.com>
References: <44FDD7AD.6020404@gmail.com>
Message-ID: <44FDDC71.80702@infiltrated.net>

Peter Pauly wrote:
> Every time our security people are informed that someone made a 911
> call, they call me to find out who it was (extension, name). I run a
> query in query analyzer against the CDR database.
>
> Are there any applications available that the security guys could run
> from their desktop that would query callmanager over the network and
> display 911 calls? What you guys doing about this problem?
>
> On a related question, our CallManager sends caller id information just
> fine. When I call my cellphone, I can see the extension of where the
> call came from. But when I make a test call to 911, the PSAP is only
> displaying the common billing phone number, not the caller-id
> information. Is my telco doing something wrong?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

Your internal ANI mapping/translation might is likely wrong on your end. 
(ELIN is your friend!)
http://www.cisco.com/univercd/cc/td/doc/product/voice/respond/res12/admin12/e911trbl.htm#wp1022781

Anyhow, why not have emergency calls go through your security desk first?

-- 
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 

The happiness of society is the end of government.
John Adams


From su1droot at gmail.com  Tue Sep  5 16:53:06 2006
From: su1droot at gmail.com (su1droot)
Date: Tue, 5 Sep 2006 16:53:06 -0400
Subject: [cisco-voip] Who called 911?
In-Reply-To: <44FDD7AD.6020404@gmail.com>
References: <44FDD7AD.6020404@gmail.com>
Message-ID: <f83dffc80609051353m11086291k4cd6b9b37d4beef9@mail.gmail.com>

If the PSAP is only seeing your BTN you may want to verify with your Telco
you have per station E911 service.  With one of the carriers for a customer
of ours that is a extra feature that costs more money.

On 9/5/06, Peter Pauly <ppauly at gmail.com> wrote:
>
> Every time our security people are informed that someone made a 911
> call, they call me to find out who it was (extension, name). I run a
> query in query analyzer against the CDR database.
>
> Are there any applications available that the security guys could run
> from their desktop that would query callmanager over the network and
> display 911 calls? What you guys doing about this problem?
>
> On a related question, our CallManager sends caller id information just
> fine. When I call my cellphone, I can see the extension of where the
> call came from. But when I make a test call to 911, the PSAP is only
> displaying the common billing phone number, not the caller-id
> information. Is my telco doing something wrong?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
--
ben
email: locutus at the-collective.net
web: http://www.the-collective.net/~locutus
irc: su1droot; aim: su1droot; icq: 11818040; y!: su1droot
GPG key: 0x87F108E2
"We exist only in each others memories, as long as you
remember everyone you have met they still exist in you"
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From milay at cisco.com  Tue Sep  5 16:56:06 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Tue, 5 Sep 2006 13:56:06 -0700
Subject: [cisco-voip] Who called 911?
Message-ID: <413FABF5DCE22849AF86F442E472B78BB62DA2@xmb-sjc-219.amer.cisco.com>

Enable debugging on the GW & verify if the calling party info is being
presented to the PRI. if so you need to do as stated below & add a
feature.
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of su1droot
Sent: Tuesday, September 05, 2006 3:53 PM
To: Peter Pauly
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Who called 911?


If the PSAP is only seeing your BTN you may want to verify with your
Telco you have per station E911 service.  With one of the carriers for a
customer of ours that is a extra feature that costs more money.


On 9/5/06, Peter Pauly <ppauly at gmail.com> wrote: 

	Every time our security people are informed that someone made a
911
	call, they call me to find out who it was (extension, name). I
run a
	query in query analyzer against the CDR database.
	
	Are there any applications available that the security guys
could run 
	from their desktop that would query callmanager over the network
and
	display 911 calls? What you guys doing about this problem?
	
	On a related question, our CallManager sends caller id
information just
	fine. When I call my cellphone, I can see the extension of where
the 
	call came from. But when I make a test call to 911, the PSAP is
only
	displaying the common billing phone number, not the caller-id
	information. Is my telco doing something wrong?
	_______________________________________________ 
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip 
	




-- 
--
ben
email: locutus at the-collective.net
web: http://www.the-collective.net/~locutus 
irc: su1droot; aim: su1droot; icq: 11818040; y!: su1droot
GPG key: 0x87F108E2
"We exist only in each others memories, as long as you
remember everyone you have met they still exist in you" 
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From jgrace at digitel.net  Tue Sep  5 18:32:09 2006
From: jgrace at digitel.net (James Grace)
Date: Tue, 5 Sep 2006 18:32:09 -0400
Subject: [cisco-voip] background logos on 7941 -61
Message-ID: <20060905183162.SM01472@jgm>

Do any one have doc on how to put background logos on the 7941 61 ipphones.
Ive worked out the idle screensaver but, ive seen some phones out there with
the background changed

 

James D. Grace 

CCNP CCNA MCSE MCSA 

Digitel Corporation

System Engineer / Professional Svc.

MSN messenger ID:  jgrace29 at hotmail.com

Mac IChat ID:  jgrace29 at mac.com

 

 

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From SPackett at fenwick.com  Tue Sep  5 18:40:02 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Tue, 5 Sep 2006 15:40:02 -0700
Subject: [cisco-voip] Who called 911?
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD7579@SVCEXCH01.firm.fenwick.llp>

An application we use for non-technical folks to view who called 911 is
from Variphy.

http://www.variphy.com/ 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Peter Pauly
Sent: Tuesday, September 05, 2006 1:02 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Who called 911?

Every time our security people are informed that someone made a 911
call, they call me to find out who it was (extension, name). I run a
query in query analyzer against the CDR database.

Are there any applications available that the security guys could run
from their desktop that would query callmanager over the network and
display 911 calls? What you guys doing about this problem?

On a related question, our CallManager sends caller id information just
fine. When I call my cellphone, I can see the extension of where the
call came from. But when I make a test call to 911, the PSAP is only
displaying the common billing phone number, not the caller-id
information. Is my telco doing something wrong?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From kthorngr at cisco.com  Tue Sep  5 20:25:24 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Tue, 5 Sep 2006 20:25:24 -0400
Subject: [cisco-voip] background logos on 7941 -61
In-Reply-To: <20060905183162.SM01472@jgm>
References: <20060905183162.SM01472@jgm>
Message-ID: <87444b7c8177c986e14cc7c03f9e7923@cisco.com>

How about this from the 7941/61 docs:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ 
ipp7961/admin/4_2/7961cus.htm#wp1030672

Kevin
On Sep 5, 2006, at 6:32 PM, James Grace wrote:

> Do any one have doc on how to put background logos on the 7941 61  
> ipphones.? Ive worked out the idle screensaver but, ive seen some  
> phones out there with the background changed
> ?
> James D. Grace
> CCNP CCNA MCSE MCSA
> Digitel Corporation
> System Engineer / Professional Svc.
> MSN messenger ID:? jgrace29 at hotmail.com
> Mac IChat ID:? jgrace29 at mac.com
> ?
> ?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From ealeatherman at gmail.com  Wed Sep  6 08:47:16 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 6 Sep 2006 08:47:16 -0400
Subject: [cisco-voip] Who called 911?
In-Reply-To: <44FDD7AD.6020404@gmail.com>
References: <44FDD7AD.6020404@gmail.com>
Message-ID: <94a1afde0609060547y8079d99o63fe645acfefd921@mail.gmail.com>

Cisco's emergency responder product provides a HTTP interface for
security/management to see 911 call information, including whatever location
information you have programmed in for it. Also lets them add notes to calls
if they want to record what they found out for each call. Not worth buying
CER just for that, but if you need the other stuff it provides, its a nice
additional feature.

I believe Ben is right, there is probably something extra you need to pay
your telco for to get the digits to the 911 center... another charge for the
phone company, who'da thought.

On 9/5/06, Peter Pauly <ppauly at gmail.com> wrote:
>
> Every time our security people are informed that someone made a 911
> call, they call me to find out who it was (extension, name). I run a
> query in query analyzer against the CDR database.
>
> Are there any applications available that the security guys could run
> from their desktop that would query callmanager over the network and
> display 911 calls? What you guys doing about this problem?
>
> On a related question, our CallManager sends caller id information just
> fine. When I call my cellphone, I can see the extension of where the
> call came from. But when I make a test call to 911, the PSAP is only
> displaying the common billing phone number, not the caller-id
> information. Is my telco doing something wrong?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From Matt.Slaga at us.didata.com  Wed Sep  6 08:56:40 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 6 Sep 2006 08:56:40 -0400
Subject: [cisco-voip] Who called 911?
Message-ID: <C1FE15183DA37645BC0633BC604E44F0030FF5A8@USNAEXCH.na.didata.local>

You can also use IPCelerate's product for this.  In fact, when someone
dials 911 you have the most un-technical way to notify your security
guards.  When a user calls 911, a pre-recorded message is played over
your security guard's IP phone telling them that someone dialed 911,
where that person is and what phone it was called from all in real time.
Before the person has even hung up the phone in most cases, security can
already be on the way.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Wednesday, September 06, 2006 8:47 AM
To: Peter Pauly
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Who called 911?

 

Cisco's emergency responder product provides a HTTP interface for
security/management to see 911 call information, including whatever
location information you have programmed in for it. Also lets them add
notes to calls if they want to record what they found out for each call.
Not worth buying CER just for that, but if you need the other stuff it
provides, its a nice additional feature. 

I believe Ben is right, there is probably something extra you need to
pay your telco for to get the digits to the 911 center... another charge
for the phone company, who'da thought.

On 9/5/06, Peter Pauly <ppauly at gmail.com> wrote:

Every time our security people are informed that someone made a 911
call, they call me to find out who it was (extension, name). I run a
query in query analyzer against the CDR database.

Are there any applications available that the security guys could run 
from their desktop that would query callmanager over the network and
display 911 calls? What you guys doing about this problem?

On a related question, our CallManager sends caller id information just
fine. When I call my cellphone, I can see the extension of where the 
call came from. But when I make a test call to 911, the PSAP is only
displaying the common billing phone number, not the caller-id
information. Is my telco doing something wrong?
_______________________________________________ 
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 




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




-----------------------------------------
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This e-mail communication and any attachments may contain
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From lelio at uoguelph.ca  Wed Sep  6 09:13:31 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 6 Sep 2006 09:13:31 -0400
Subject: [cisco-voip] password/PIN restrictions on CallManager
Message-ID: <005401c6d1b6$3fd53bf0$47196883@cfs.uoguelph.ca>

Are there any password/PIN restrictions on CallManager that are easily accessible, describable, etc? All I see on the login page selections are the number of digits/characters.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
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From CMStaffin at ra.rockwell.com  Wed Sep  6 12:02:04 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Wed, 6 Sep 2006 11:02:04 -0500
Subject: [cisco-voip] background logos on 7941 -61
In-Reply-To: <87444b7c8177c986e14cc7c03f9e7923@cisco.com>
Message-ID: <OF7D403F4C.58E096F4-ON862571E1.00574DCB-862571E1.005800AB@ra.rockwell.com>

Kevin,

I am just curious this process is very manual.  In other words you need to 
set it for each phone.  Pretty labor intensive for 20,000 phones.  Is 
there a way in CCM that you can force the phones to get lets say a company 
logo as the background?

Craig





Kevin Thorngren <kthorngr at cisco.com> 
Sent by: cisco-voip-bounces at puck.nether.net
09/05/2006 07:25 PM

To
"James Grace" <jgrace at digitel.net>
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] background logos on 7941 -61






How about this from the 7941/61 docs:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ 
ipp7961/admin/4_2/7961cus.htm#wp1030672

Kevin
On Sep 5, 2006, at 6:32 PM, James Grace wrote:

> Do any one have doc on how to put background logos on the 7941 61 
> ipphones.  Ive worked out the idle screensaver but, ive seen some 
> phones out there with the background changed
>  
> James D. Grace
> CCNP CCNA MCSE MCSA
> Digitel Corporation
> System Engineer / Professional Svc.
> MSN messenger ID:  jgrace29 at hotmail.com
> Mac IChat ID:  jgrace29 at mac.com
>  
>  
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From SPackett at fenwick.com  Wed Sep  6 12:49:44 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Wed, 6 Sep 2006 09:49:44 -0700
Subject: [cisco-voip] background logos on 7941 -61
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD757E@SVCEXCH01.firm.fenwick.llp>

Craig,
I'm glad you brought that up as I'm about to deploy several 7971 phones
and would like to know if there is a bulk way of pushing the logos out.
I didn't seem to mind when I did it for 10 or so phones, but I will be
getting several more in the next few weeks.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Craig M Staffin
Sent: Wednesday, September 06, 2006 9:02 AM
To: Kevin Thorngren
Cc: cisco-voip-bounces at puck.nether.net; James Grace;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61



Kevin, 

I am just curious this process is very manual.  In other words you need
to set it for each phone.  Pretty labor intensive for 20,000 phones.  Is
there a way in CCM that you can force the phones to get lets say a
company logo as the background? 

Craig





Kevin Thorngren <kthorngr at cisco.com> 
Sent by: cisco-voip-bounces at puck.nether.net 

09/05/2006 07:25 PM 

To
"James Grace" <jgrace at digitel.net> 
cc
cisco-voip at puck.nether.net 
Subject
Re: [cisco-voip] background logos on 7941 -61

	




How about this from the 7941/61 docs:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ 
ipp7961/admin/4_2/7961cus.htm#wp1030672

Kevin
On Sep 5, 2006, at 6:32 PM, James Grace wrote:

> Do any one have doc on how to put background logos on the 7941 61  
> ipphones.  Ive worked out the idle screensaver but, ive seen some  
> phones out there with the background changed
>  
> James D. Grace
> CCNP CCNA MCSE MCSA
> Digitel Corporation
> System Engineer / Professional Svc.
> MSN messenger ID:  jgrace29 at hotmail.com
> Mac IChat ID:  jgrace29 at mac.com
>  
>  
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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From jonvoip at gmail.com  Wed Sep  6 12:57:02 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 6 Sep 2006 11:57:02 -0500
Subject: [cisco-voip] background logos on 7941 -61
In-Reply-To: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD757E@SVCEXCH01.firm.fenwick.llp>
References: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD757E@SVCEXCH01.firm.fenwick.llp>
Message-ID: <5d093f9a0609060957p103c482fg589ef6e5d8b2088d@mail.gmail.com>

On the 7971, it is easy, just load the logo ont he TFTP server... For the
others, change Enterprise Parameters for the Idle screen... that will do it
globally.

However, each phone is going to need to select that image from the settings
on the 7971s...




Jonathan

On 9/6/06, Stu Packett <SPackett at fenwick.com> wrote:
>
>  Craig,
> I'm glad you brought that up as I'm about to deploy several 7971 phones
> and would like to know if there is a bulk way of pushing the logos out.  I
> didn't seem to mind when I did it for 10 or so phones, but I will be getting
> several more in the next few weeks.
>
>  ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Craig M Staffin
> *Sent:* Wednesday, September 06, 2006 9:02 AM
> *To:* Kevin Thorngren
> *Cc:* cisco-voip-bounces at puck.nether.net; James Grace;
> cisco-voip at puck.nether.net
>
> *Subject:* Re: [cisco-voip] background logos on 7941 -61
>
>
> Kevin,
>
> I am just curious this process is very manual.  In other words you need to
> set it for each phone.  Pretty labor intensive for 20,000 phones.  Is there
> a way in CCM that you can force the phones to get lets say a company logo as
> the background?
>
> Craig
>
>
>
>
>   *Kevin Thorngren <kthorngr at cisco.com>*
> Sent by: cisco-voip-bounces at puck.nether.net
>
> 09/05/2006 07:25 PM
>    To
> "James Grace" <jgrace at digitel.net>  cc
> cisco-voip at puck.nether.net  Subject
> Re: [cisco-voip] background logos on 7941 -61
>
>
>
>
>
>
> How about this from the 7941/61 docs:
> http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/
> ipp7961/admin/4_2/7961cus.htm#wp1030672
>
> Kevin
> On Sep 5, 2006, at 6:32 PM, James Grace wrote:
>
> > Do any one have doc on how to put background logos on the 7941 61
> > ipphones.  Ive worked out the idle screensaver but, ive seen some
> > phones out there with the background changed
> >
> > James D. Grace
> > CCNP CCNA MCSE MCSA
> > Digitel Corporation
> > System Engineer / Professional Svc.
> > MSN messenger ID:  jgrace29 at hotmail.com
> > Mac IChat ID:  jgrace29 at mac.com
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From CORTIZ at broward.org  Wed Sep  6 14:02:42 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed, 6 Sep 2006 14:02:42 -0400
Subject: [cisco-voip] Setting up local DSP resources on 2821
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F117E31@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A474AE@MREXVS01.bc.broward.cty>

Were the questions regarding DSPFARM configs in the email below ever
answered? 

I have configured my DSPFARM on a 2811 but how do you actually make
these resources the first choice for the users at that remote site?  I
assume some configuration in the conf bridge resources and MRGL but I
can't find a doc specifically for the 2800 routers.  Anyone have a link?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, July 18, 2006 3:20 PM
To: Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821


Hi Ted

Sounds like you have done it a couple of times.

Q1. I notice in the config examples they have an example that has

voice-card 1
no dspfarm
dsp services dspfarm

bit confusing - why no dsprarm and right below it dsp services dspfarm
???

Q2. They also show
associate profile 10 register CFB123456789966
associate profile 6 register MTP123456789988

where does the values after register come from - are they arbitrary
things one choose and need to be unique per site and per profile I
register etc.???


oip


From wsisk at cisco.com  Wed Sep  6 14:08:28 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 06 Sep 2006 14:08:28 -0400
Subject: [cisco-voip] background logos on 7941 -61
In-Reply-To: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD757E@SVCEXCH01.firm.fenwick.llp>
References: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD757E@SVCEXCH01.firm.fenwick.llp>
Message-ID: <44FF0E9C.5090507@cisco.com>

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From jason.aarons at us.didata.com  Wed Sep  6 14:25:14 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 6 Sep 2006 14:25:14 -0400
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe at 100%
	cpu
Message-ID: <C1FE15183DA37645BC0633BC604E44F0030FFD1E@USNAEXCH.na.didata.local>

Any ideas why winmgmt.exe might be at 100% on both boxes in my CRS
4.0(4) cluster ? -jason

 

OS 2000.2.7sr8

 

Can I disable the Windows Instrument Management service and the Windows
Instrument Management Extensions service?




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From sales at doretel.com  Wed Sep  6 14:27:07 2006
From: sales at doretel.com (Sales@Doretel)
Date: Wed, 6 Sep 2006 14:27:07 -0400
Subject: [cisco-voip] Cisco Brand Protection Contacts
Message-ID: <0c5101c6d1e2$0fc32860$6501a8c0@ShaneCompaq>

I know there is some Cisco guys on here. I have a customer that bought some WIC cards that are fake from a grey market vendor. The contact I had at Brand Protection is no longer in that position anymore. Can someone please give me a call/email/im if you can help me point this poor customer in the right direction.

Thanks in advance...

Shane Breen
Doretel Communications, Inc.
Office: 404.755.5721 
sbreen at doretel.com
AIM: shanebreen2003
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From rratliff at cisco.com  Wed Sep  6 14:25:23 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 6 Sep 2006 14:25:23 -0400
Subject: [cisco-voip] Setting up local DSP resources on 2821
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E03A474AE@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E03A474AE@MREXVS01.bc.broward.cty>
Message-ID: <BE227AFD-5D69-4EF8-98DF-1A99E46D36BB@cisco.com>

It's CallManager that decides what resources a device uses. The  
configuration on the router is simply to get it registered.

You put the device in a MRG, and then the order of MRGs in the MRGL  
decides preference.

-Ryan

On Sep 6, 2006, at 2:02 PM, Ortiz, Carlos wrote:

Were the questions regarding DSPFARM configs in the email below ever
answered?

I have configured my DSPFARM on a 2811 but how do you actually make
these resources the first choice for the users at that remote site?  I
assume some configuration in the conf bridge resources and MRGL but I
can't find a doc specifically for the 2800 routers.  Anyone have a link?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, July 18, 2006 3:20 PM
To: Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821


Hi Ted

Sounds like you have done it a couple of times.

Q1. I notice in the config examples they have an example that has

voice-card 1
no dspfarm
dsp services dspfarm

bit confusing - why no dsprarm and right below it dsp services dspfarm
???

Q2. They also show
associate profile 10 register CFB123456789966
associate profile 6 register MTP123456789988

where does the values after register come from - are they arbitrary
things one choose and need to be unique per site and per profile I
register etc.???


oip

_______________________________________________
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  Wed Sep  6 14:26:57 2006
From: bills at tns.its.psu.edu (Simon, Bill)
Date: Wed, 06 Sep 2006 14:26:57 -0400
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe at
 100%	cpu
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0030FFD1E@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0030FFD1E@USNAEXCH.na.didata.local>
Message-ID: <44FF12F1.9070306@tns.its.psu.edu>

Consider this a "me too" post - we've been seeing some CPU-high behavior 
from those services lately as well.  Restarting the services seems to 
bring it back down to normal.

Jason Aarons (US) wrote:
> Any ideas why winmgmt.exe might be at 100% on both boxes in my CRS 
> 4.0(4) cluster ? ?jason
> 
>  
> 
> OS 2000.2.7sr8
> 
>  
> 
> Can I disable the Windows Instrument Management service and the Windows 
> Instrument Management Extensions service?

From baryonyx5 at gmail.com  Wed Sep  6 14:27:24 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Wed, 6 Sep 2006 13:27:24 -0500
Subject: [cisco-voip] FXO calls disconnect when answered
In-Reply-To: <a749d0620608301358r21c32348u283a1f512ba290e5@mail.gmail.com>
References: <a749d0620608251127w53bb9ea2t469ed382be972f3a@mail.gmail.com>
	<36E9A2CB-ACE5-43E8-BC92-50FD05CC9CE0@cisco.com>
	<a749d0620608251504s1c978385g444f21b5ad0080f@mail.gmail.com>
	<8E3927B4-E4E2-43B1-AA60-EDEA575047E7@cisco.com>
	<a749d0620608301358r21c32348u283a1f512ba290e5@mail.gmail.com>
Message-ID: <a749d0620609061127g31adb5adk1b762808d57844f6@mail.gmail.com>

Just an update on this issue...

It looks like it was a hardware problem.  I shipped up a separate 1760
running the same code and had no problems.

I'm not sure yet what component is to blame.  I assume its either the voice
card or possibly the PVDM.

Thanks for all the help; I've learned a lot from your suggestions.

On 8/30/06, Kris Seraphine <baryonyx5 at gmail.com> wrote:
>
> Hi
>
> Here are debugs with the timer sup-disconnect set to 1500.  The results
> were the same. I'm at about  the end of  my rope on this one.
>
> I should also add the following info for completeness:
> These lines work without problem in the old key system
> when I try to place an outbound call from an IP phone at the site, the
> call disconnects immediately (no ringback)
> I get the same results with the test ground start line as I got with the
> production loop start lines.
>
> Arcadia-1760(config-voiceport)#do sh run | beg voice-port
> voice-port 2/0
>  signal groundStart
>  no vad
>  no comfort-noise
>  timing sup-disconnect 1500
>  connection plar opx xxxxxxxxxx
>
>
> Arcadia-1760(config-voiceport)#do sh debug
> VTSP:
>   debug voip vtsp event is ON (filter is OFF)
>   debug voip vtsp session is ON (filter is OFF)
>   debug voip vtsp error software is ON
>   debug voip vtsp error call is ON (filter is OFF)
>   debug voip vtsp inout is ON (filter is OFF)
> Voice Port Module signaling debugging is on
>
>
> Arcadia-1760(config-voiceport)#
> 032054: Aug 30 15:43:10.286: htsp_process_event: [2/0, FXOGS_ONHOOK,
> E_DSP_SIG_0000]fxogs_onhook_ringing
> 032055: Aug 30 15:43: 10.286: htsp_timer_stop
> 032056: Aug 30 15:43:10.286: htsp_timer - 125 msec
> 032057: Aug 30 15:43:10.414: htsp_process_event: [2/0,
> FXOGS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxogs_wait_ring_min_timer
> 032058: Aug 30 15:43: 10.414: htsp_timer - 10000 msec
> Arcadia-1760(config-voiceport)#
> 032059: Aug 30 15:43:12.086: htsp_process_event: [2/0, FXOGS_RINGING,
> E_DSP_SIG_0100]
> 032060: Aug 30 15:43:12.086: fxogs_ringing_not:
> 032061: Aug 30 15:43: 12.086: htsp_timer_stop
> 032062: Aug 30 15:43:12.086: htsp_timer_stop2
> 032063: Aug 30 15:43:12.086: htsp_timer_stop3 htsp_setup_ind
> 032064: Aug 30 15:43:12.086: [2/0] get_local_station_id calling num=
> calling name= calling time=08/30 15:43  orig called=
> 032065: Aug 30 15:43:12.090:
> //-1/xxxxxxxxxxxx/VTSP:(2/0):-1:-1:-1/vtsp_allocate_cdb:
>    CDB=0x83B5FEF0
> 032066: Aug 30 15:43:12.090:
> //-1/xxxxxxxxxxxx/VTSP:(2/0):-1:-1:-1/vtsp_do_call_setup_ind:
>    Event=E_TSP_SETUP_IND
>    Progress Indication=3, CarrierIDCode=, Info Trans Capability=144,
> Source Carrier ID=, tg_label_flag=0
> 032067: Aug 30 15:43:12.094:
> //-1/FCA9AEE082BC/VTSP:(2/0):-1:-1:-1/vtsp_timer:
>    Timer Start Time=292230322, Timer Value=180000(ms)
> 032068: Aug 30 15:43:12.094:
> //-1/FCA9AEE082BC/VTSP:(2/0):-1:-1:-1/vtsp_do_normal_call_setup_ind:
>    IC State Machine (ICSM) Added
> 032069: Aug 30 15:43:12.094:
> //-1/FCA9AEE082BC/VTSP:(2/0):-1:-1:-1/vtsp_insert_cdb:
> 032070: Aug 30 15:43:12.098:
> //-1/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_SETUP_IND_PEND, event:E_VTSP_DSM_OPEN_SUCCESS]
> 032071: Aug 30 15:43:12.098:
> //-1/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_setup_ind_pend_success:
> 032072: Aug 30 15:43:12.098:
> //-1/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_realloc_cdb:
>    CDB=0x83B5FEF0
> 032073: Aug 30 15:43:12.098:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_insert_cdb:
> 032074: Aug 30 15:43: 12.102:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_timer:
>    Timer Start Time=292230322, Timer Value=180000(ms)
> 032075: Aug 30 15:43:12.102: htsp_process_event: [2/0,
> FXOGS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
> 032076: Aug 30 15:43: 12.102: fxogs_wait_setup_ack:
> 032077: Aug 30 15:43:12.110:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_spi_event_handler:
>    Event Category=1, Event=137
> 032078: Aug 30 15:43:12.110:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_SETUP_INDICATED, event:E_CC_PROCEEDING]
> 032079: Aug 30 15:43:12.114:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_proceeding:
>    Progress Indication=0
> 032080: Aug 30 15:43:12.114:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292230324
> 032081: Aug 30 15:43:12.114: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_HTSP_PROCEEDING]
> 032082: Aug 30 15:43:12.218:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_PROCEEDING, event:E_CC_ALERT]
> 032083: Aug 30 15:43:12.218:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_alert:
>    Progress Indication=0, Signal Indication=1, Setup Progress Indication=3
> 032084: Aug 30 15:43:12.218:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292230334htsp_alert_notify
> 032085: Aug 30 15:43:12.222:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_alert:
>    Progress Indication=0, Tone=Ring Back
> 032086: Aug 30 15:43:12.226:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_ALERTING, event:E_CC_SERVICE_MSG]
> 032087: Aug 30 15:43:12.226:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_service_msg_down:
> Arcadia-1760(config-voiceport)#
> 032088: Aug 30 15:43:12.226:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292230335htsp_call_service_msghtsp_call_service_msg not
> EFXS (2)
> Arcadia-1760(config-voiceport)#
> 032089: Aug 30 15:43:16.046: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_DSP_SIG_0000]fxogs_opx_offhook_ringing
> 032090: Aug 30 15:43:16.046: htsp_timer_stop2
> 032091: Aug 30 15:43:16.050: htsp_timer_stop
> Arcadia-1760(config-voiceport)#
> 032092: Aug 30 15:43:18.086: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_DSP_SIG_0100]
> 032093: Aug 30 15:43:18.086: fxogs_stop_disc_timer
> 032094: Aug 30 15:43:18.086: htsp_timer_stop2
> Arcadia-1760(config-voiceport)#
> 032095: Aug 30 15:43:20.093:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_ALERTING, event:E_CC_SERVICE_MSG]
> 032096: Aug 30 15:43:20.093:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_service_msg_down:
> 032097: Aug 30 15:43:20.093:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292231123htsp_call_service_msghtsp_call_service_msg not
> EFXS (2)
> 032098: Aug 30 15:43:20.269: htsp_call_bridged invoked
> 032099: Aug 30 15:43:20.273:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_dsm_save_fax_config:
>    Fax Relay=DISABLED - 'fax rate disabled' set (dial-peer)
>    Primary Fax Protocol=IGNORE_FAX_RELAY, Fallback Fax
> Protocol=IGNORE_FAX_RELAY
>    Fax Parameters Set By=Dialpeer, Peer=101
> 032100: Aug 30 15:43:20.277 :
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_ALERTING, event:E_CC_DO_CAPS_IND]
> 032101: Aug 30 15:43:20.277:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_dsm_peer_event_cb:
>    Event=E_DSM_CC_CAPS_IND
> 032102: Aug 30 15:43:20.281:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_ALERTING, event:E_CC_CAPS_IND]
> 032103: Aug 30 15:43:20.281:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_dsm_peer_event_cb:
>    Event=E_DSM_CC_CAPS_ACK
> 032104: Aug 30 15:43:20.285:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_ALERTING, event:E_CC_CONNECT]
> 032105: Aug 30 15:43:20.285:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_alert_connect:
>    Progress Indication=0
> 032106: Aug 30 15:43:20.285:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_ring_noan_timer_stop:
>    Timer Stop Time=292231142
> 032107: Aug 30 15:43:20.285: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_HTSP_CONNECT]fxogs_proc_voice
> 032108: Aug 30 15:43:20.289: htsp_timer_stop
> 032109: Aug 30 15:43:20.289: [2/0] set signal state = 0xC timestamp = 0
> Arcadia-1760(config-voiceport)#
> 032110: Aug 30 15:43:20.293:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_dsm_feature_notify_cb:
>    Feature ID=0, Feature Status=1
> 032111: Aug 30 15:43:20.293:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_dsm_peer_event_cb:
>    Event=E_DSM_CC_MODIFY_MEDIA_IND
> 032112: Aug 30 15:43:20.293: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_HTSP_VOICE_CUT_THROUGH]
> 032113: Aug 30 15:43:20.549: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_DSP_SIG_1100]
> 032114: Aug 30 15:43:20.549: fxogs_no_tipgnd
> 032115: Aug 30 15:43:20.549: htsp_timer_stop
> 032116: Aug 30 15:43:20.549: htsp_timer_stop2
> 032117: Aug 30 15:43:20.549: htsp_timer2 - 350 msec
> 032118: Aug 30 15:43:20.901: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_HTSP_EVENT_TIMER2]fxogs_offhook_disc
> 032119: Aug 30 15:43:20.901: htsp_timer_stop
> 032120: Aug 30 15:43: 20.901: [2/0] set signal state = 0x4 timestamp = 0
> 032121: Aug 30 15:43:20.901: htsp_timer - 2000 msec
> 032122: Aug 30 15:43:20.905:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_CONNECT, event:E_TSP_DISCONNECT_IND]
> 032123: Aug 30 15:43:20.905:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_generate_disc:
>    Cause Value=16
> 032124: Aug 30 15:43:20.905:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292231204
> 032125: Aug 30 15:43:20.905:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_ring_noan_timer_stop:
>    Timer Stop Time=292231204
> 032126: Aug 30 15:43:20.905:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_set_release_source:
>    Release Direction=PSTN, Release Source=Calling Party-PSTN
> 032127: Aug 30 15:43:20.909:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_cc_call_disconnected:
>    Cause Value=16
> 032128: Aug 30 15:43:20.909:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_timer:
>    Timer Start Time=292231205, Timer Value=15000(ms)
> 032129: Aug 30 15:43:20.909:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_generate_disc:
>    Return Code=0
> 032130: Aug 30 15:43:20.917:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_CONNECT, event:E_CC_DISCONNECT]
> 032131: Aug 30 15:43:20.921:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_disconnect:
>    Cause Value=16, Previous Cause Value=16
> 032132: Aug 30 15:43:20.921:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_ring_noan_timer_stop:
>    Timer Stop Time=292231206
> 032133: Aug 30 15:43:20.921:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_dsm_save_fax_config:
>    Fax Relay=DISABLED - 'fax rate disabled' set (dial-peer)
>    Primary Fax Protocol=IGNORE_FAX_RELAY, Fallback Fax
> Protocol=IGNORE_FAX_RELAY
>    Fax Parameters Set By=Dialpeer, Peer=101
> 032134: Aug 30 15:43:20.921: htsp_timer_stop3
> 032135: Aug 30 15:43:20.929:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_WAIT_STATS, event:E_VTSP_DSM_STATS_COMPLETE]
> 032136: Aug 30 15:43:20.929:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_dsm_dsp_stats_complete:
> 032137: Aug 30 15:43:20.929:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_timer:
>    Timer Start Time=292231207, Timer Value=60000(ms)
> 032138: Aug 30 15:43:20.929: htsp_process_event: [2/0, FXOGS_GUARD_OUT,
> E_HTSP_RELEASE_REQ]fxogs_guard_out_release
> 032139: Aug 30 15:43:20.933:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_WAIT_RELEASE, event:E_TSP_DISCONNECT_CONF]
> 032140: Aug 30 15:43:20.933:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/act_wrelease_release:
> 032141: Aug 30 15:43:20.933:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_play_busy_timer_stop:
>    Timer Stop Time=292231207
> 032142: Aug 30 15:43:20.933:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292231207
> 032143: Aug 30 15:43:20.933:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:0:1/vtsp_do_call_history:
>    Coder Rate=16
> 032144: Aug 30 15:43:20.941:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:-1:-1/vtsp_process_event:
>    [state:S_CLOSE_DSPRM, event:E_VTSP_DSM_CLOSE_COMPLETE]
> 032145: Aug 30 15:43:20.941:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:-1:-1/act_terminate:
> Arcadia-1760(config-voiceport)#
> 032146: Aug 30 15:43:20.941:
> //642/FCA9AEE082BC/VTSP:(2/0):-1:-1:-1/vtsp_timer_stop:
>    Timer Stop Time=292231208
> 032147: Aug 30 15:43:20.945:
> //-1/FCA9AEE082BC/VTSP:(2/0):-1:-1:-1/vtsp_free_cdb:
>    CDB=0x83B5FEF0
> 032148: Aug 30 15:43:21.509: htsp_process_event: [2/0, FXOGS_GUARD_OUT,
> E_DSP_SIG_0100]
> 032149: Aug 30 15:43:21.509:
> fxogs_guardout_tip_groundfxogs_onhook_tip_ground
> 032150: Aug 30 15:43:21.509 : htsp_timer - 7000 msec
> Arcadia-1760(config-voiceport)#
> 032151: Aug 30 15:43:22.049: htsp_process_event: [2/0, FXOGS_TIP_GROUND,
> E_DSP_SIG_0000]fxogs_tipground_ringing
> 032152: Aug 30 15:43:22.049: htsp_timer_stop
> Arcadia-1760(config-voiceport)#
> 032153: Aug 30 15:43:24.057: htsp_process_event: [2/0, FXOGS_RINGING,
> E_DSP_SIG_0100]
> 032154: Aug 30 15:43:24.061: fxogs_ringing_not:
> 032155: Aug 30 15:43:24.061: htsp_timer_stop
> 032156: Aug 30 15:43:24.061: htsp_timer_stop2
> 032157: Aug 30 15:43:24.061: htsp_timer_stop3 htsp_setup_ind
> 032158: Aug 30 15:43:24.061: [2/0] get_local_station_id calling num=
> calling name= calling time=08/30 15:43  orig called=
> 032159: Aug 30 15:43:24.061:
> //-1/xxxxxxxxxxxx/VTSP:(2/0):-1:-1:-1/vtsp_allocate_cdb:
>    CDB=0x83B5FC48
> 032160: Aug 30 15:43:24.065:
> //-1/xxxxxxxxxxxx/VTSP:(2/0):-1:-1:-1/vtsp_do_call_setup_ind:
>    Event=E_TSP_SETUP_IND
>    Progress Indication=3, CarrierIDCode=, Info Trans Capability=144,
> Source Carrier ID=, tg_label_flag=0
> 032161: Aug 30 15:43:24.065:
> //-1/03CD0D6282BF/VTSP:(2/0):-1:-1:-1/vtsp_timer:
>    Timer Start Time=292231521, Timer Value=180000(ms)
> 032162: Aug 30 15:43:24.065:
> //-1/03CD0D6282BF/VTSP:(2/0):-1:-1:-1/vtsp_do_normal_call_setup_ind:
>    IC State Machine (ICSM) Added
> 032163: Aug 30 15:43:24.069:
> //-1/03CD0D6282BF/VTSP:(2/0):-1:-1:-1/vtsp_insert_cdb:
> 032164: Aug 30 15:43:24.069:
> //-1/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_SETUP_IND_PEND, event:E_VTSP_DSM_OPEN_SUCCESS]
> 032165: Aug 30 15:43:24.073:
> //-1/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_setup_ind_pend_success:
> 032166: Aug 30 15:43:24.073:
> //-1/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_realloc_cdb:
>    CDB=0x83B5FC48
> 032167: Aug 30 15:43:24.073:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_insert_cdb:
> 032168: Aug 30 15:43: 24.077:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_timer:
>    Timer Start Time=292231522, Timer Value=180000(ms)
> 032169: Aug 30 15:43:24.077: htsp_process_event: [2/0,
> FXOGS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
> 032170: Aug 30 15:43: 24.077: fxogs_wait_setup_ack:
> 032171: Aug 30 15:43:24.085:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_spi_event_handler:
>    Event Category=1, Event=137
> 032172: Aug 30 15:43:24.085:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_SETUP_INDICATED, event:E_CC_PROCEEDING]
> 032173: Aug 30 15:43:24.085:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_proceeding:
>    Progress Indication=0
> 032174: Aug 30 15:43:24.089:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292231523
> 032175: Aug 30 15:43:24.093: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_HTSP_PROCEEDING]
> 032176: Aug 30 15:43:24.197:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_PROCEEDING, event:E_CC_ALERT]
> 032177: Aug 30 15:43:24.197:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_alert:
>    Progress Indication=0, Signal Indication=1, Setup Progress Indication=3
> 032178: Aug 30 15:43:24.197:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292231534htsp_alert_notify
> 032179: Aug 30 15:43:24.201:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_alert:
>    Progress Indication=0, Tone=Ring Back
> 032180: Aug 30 15:43:24.205:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_ALERTING, event:E_CC_SERVICE_MSG]
> 032181: Aug 30 15:43:24.205:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_service_msg_down:
> Arcadia-1760(config-voiceport)#
> 032182: Aug 30 15:43:24.205:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292231535htsp_call_service_msghtsp_call_service_msg not
> EFXS (2)
> Arcadia-1760(config-voiceport)#
> 032183: Aug 30 15:43:25.681: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_DSP_SIG_1100]
> 032184: Aug 30 15:43: 25.681: fxogs_no_tipgnd
> 032185: Aug 30 15:43:25.681: htsp_timer_stop
> 032186: Aug 30 15:43:25.681: htsp_timer_stop2
> 032187: Aug 30 15:43:25.681: htsp_timer2 - 350 msec
> 032188: Aug 30 15:43:26.033: htsp_process_event: [2/0, FXOGS_OFFHOOK,
> E_HTSP_EVENT_TIMER2]fxogs_offhook_disc
> 032189: Aug 30 15:43:26.033: htsp_timer_stop
> 032190: Aug 30 15:43:26.033: [2/0] set signal state = 0x4 timestamp = 0
> 032191: Aug 30 15:43:26.033: htsp_timer - 2000 msec
> 032192: Aug 30 15:43:26.033:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_ALERTING, event:E_TSP_DISCONNECT_IND]
> 032193: Aug 30 15:43:26.033:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_generate_disc:
>    Cause Value=16
> 032194: Aug 30 15:43:26.037:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292231718
> 032195: Aug 30 15:43:26.037:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_ring_noan_timer_stop:
>    Timer Stop Time=292231718
> 032196: Aug 30 15:43:26.037:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_set_release_source:
>    Release Direction=PSTN, Release Source=Calling Party-PSTN
> 032197: Aug 30 15:43:26.037:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_cc_call_disconnected:
>    Cause Value=16
> 032198: Aug 30 15:43:26.037:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_timer:
>    Timer Start Time=292231718, Timer Value=15000(ms)
> 032199: Aug 30 15:43:26.037:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_generate_disc:
>    Return Code=0
> 032200: Aug 30 15:43:26.045:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_WAIT_HOST_DISC, event:E_CC_DISCONNECT]
> 032201: Aug 30 15:43:26.045:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_host_disc_cleanup:
>    dsm_indicate_error=0, short_cleardown_timeout=1, Mode=0
> 032202: Aug 30 15:43: 26.045:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_dsm_save_fax_config:
>    Call Entry=NULL, Call Id=-1
> 032203: Aug 30 15:43:26.045:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_host_disc_cleanup:
>    Exit State=16
> 032204: Aug 30 15:43:26.049:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_WAIT_STATS, event:E_VTSP_DSM_STATS_COMPLETE]
> 032205: Aug 30 15:43:26.049:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_dsm_dsp_stats_complete:
> 032206: Aug 30 15:43:26.049:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_timer:
>    Timer Start Time=292231719, Timer Value=60000(ms)
> 032207: Aug 30 15:43:26.053: htsp_process_event: [2/0, FXOGS_GUARD_OUT,
> E_HTSP_RELEASE_REQ]fxogs_guard_out_release
> 032208: Aug 30 15:43:26.053:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_process_event:
>    [state:S_WAIT_RELEASE, event:E_TSP_DISCONNECT_CONF]
> 032209: Aug 30 15:43:26.057:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/act_wrelease_release:
> 032210: Aug 30 15:43:26.057:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_play_busy_timer_stop:
>    Timer Stop Time=292231720
> 032211: Aug 30 15:43:26.057:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_timer_stop:
>    Timer Stop Time=292231720
> 032212: Aug 30 15:43:26.057:
> //644/03CD0D6282BF/VTSP:(2/0):-1:0:1/vtsp_do_call_history:
>    Coder Rate=-1
> 032213: Aug 30 15:43:26.065:
> //644/03CD0D6282BF/VTSP:(2/0):-1:-1:-1/vtsp_process_event:
>    [state:S_CLOSE_DSPRM, event:E_VTSP_DSM_CLOSE_COMPLETE]
>
>
>
> --
> kris seraphine
>



-- 
kris seraphine
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From netfortius at gmail.com  Wed Sep  6 14:43:49 2006
From: netfortius at gmail.com (Netfortius)
Date: Wed, 6 Sep 2006 13:43:49 -0500
Subject: [cisco-voip] Unity (integrated) questions vis-a-vis Outlook,
	VMO and a separate Exchange server
Message-ID: <200609061343.49819.netfortius@gmail.com>

Could someone point me to a very thorough description of the architecture of 
Unity - the integrated version (the one with "built-in" exchange data 
store) - and its relationship with that Exchange data store, Outlook 
capabilities as MAPI client for an existing Exchange email server, VMO usage 
in such an environment, etc.? Here are a couple of things I am trying to 
answer, associated with a deployment of such:

- if Outlook is a MAPI client - could I "go" after the messages in an 
integrated unity environment (its exchange data store) as if it was an 
exchange server?
- if the above is true (which I think it is, simply because the OWA is 
supposed to do the same thing), and knowing that Outlook can only accept an 
Exchange serve account at a time - is there any relationship I could 
establish between the data store associated with the unity and an Exchange 
server already used as email server, such that I can get the voice messages 
from unity/exch data store in the same time with emails (trust + 
<something>)?
- what is the VMO role in such an environment? If it is capable of creating 
voice messages to be sent to other mailboxes, isn't that equivalent to being 
able to be a "MAPI" client of the unity data store, or use Outlook for such - 
and - under this scenario - wouldn't this allow "pulling" into some Outlook 
local folder voice messages, from the unity data store, while Outlook is also 
an email client of the mail Exchange server?

I am sorry if the above questions do not make much sense, but I am trying to 
gain a better understanding of this architecture, and I am not sure how 
things actually work (some people even telling me that VMO is not merely an 
Outlook form, but also having some DLLs associated with it, upon 
installation, which allow it to communicate directly with the Unity 
server?!?)

Oh - we'll be using Unity ver 4.2.

Pointers to RTFMs, books, links, etc will be appreciated

TIA,
Stefan

From CORTIZ at broward.org  Wed Sep  6 14:56:55 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed, 6 Sep 2006 14:56:55 -0400
Subject: [cisco-voip] Setting up local DSP resources on 2821
In-Reply-To: <BE227AFD-5D69-4EF8-98DF-1A99E46D36BB@cisco.com>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A474B5@MREXVS01.bc.broward.cty>

I ran a show media resource status on the router and see the following:

Lowrise#sh media resource status 
 Resource Providers:

 Resource  Provider ID :: FLEX_DSPRM Status :: REGISTERED
 Service Profiles
 MTP :: 
 TRANSCODING :: 
 CONFERENCING :: 1 
 Applications : 
 Application ID : SCCP, Status : REGISTERED

I guess that looks as it should.....

What appears incorrect is 

Profile ID = 1, Service = CONFERENCING, Resource ID = 1  
 Profile Description :  
 Profile Admin State : UP 
 Profile Operation State : ACTIVE IN PROGRESS 
 Application : SCCP   Status : ASSOCIATION IN PROGRESS 
 Resource Provider : FLEX_DSPRM   Status : UP 
 Number of Resource Configured : 2 
 Number of Resource Available : 2

There is a required parameter on the Call Manager:

Conference Bridge Name  Enter the same name that exists in the gateway
Command Line Interface (CLI). 


What name are they referring too?

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Wednesday, September 06, 2006 2:25 PM
To: Ortiz, Carlos
Cc: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821

It's CallManager that decides what resources a device uses. The  
configuration on the router is simply to get it registered.

You put the device in a MRG, and then the order of MRGs in the MRGL  
decides preference.

-Ryan

On Sep 6, 2006, at 2:02 PM, Ortiz, Carlos wrote:

Were the questions regarding DSPFARM configs in the email below ever
answered?

I have configured my DSPFARM on a 2811 but how do you actually make
these resources the first choice for the users at that remote site?  I
assume some configuration in the conf bridge resources and MRGL but I
can't find a doc specifically for the 2800 routers.  Anyone have a link?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, July 18, 2006 3:20 PM
To: Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821


Hi Ted

Sounds like you have done it a couple of times.

Q1. I notice in the config examples they have an example that has

voice-card 1
no dspfarm
dsp services dspfarm

bit confusing - why no dsprarm and right below it dsp services dspfarm
???

Q2. They also show
associate profile 10 register CFB123456789966
associate profile 6 register MTP123456789988

where does the values after register come from - are they arbitrary
things one choose and need to be unique per site and per profile I
register etc.???


oip

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


From ErasmuE4 at telkom.co.za  Wed Sep  6 14:57:12 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 6 Sep 2006 20:57:12 +0200
Subject: [cisco-voip] Setting up local DSP resources on 2821
References: <26B357EF9FB7D04A989C9E82A9E5436E03A474AE@MREXVS01.bc.broward.cty>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D5D5@CNTRRA20-XCS01.telkom.co.za>


Most of them

I also have some sample configs on my laptop at work which I will make
available tomorrow. I managed to get hardware assisted (enhanced IOS)
conferencing on a router going nicely after reading the docs and
getting help from the group. And it is fairly simple to get it
configured in CCM as a media resource.

I also managed to find out the following useful info. cisco often
refers to DSPs and how many conferences etc one can do per DSP. I
figured out now that as far as PVDM2s go a PVDM2-16 counts as one DSP,
a 2-32 counts as 2 and a 2-48 as three and 2-64 as 4. Plus that
generaly for G711 a single DSP (that is one chip on the PVDM can do 8
x 8 meaning 8 conferences with 8 participants. This usually drops to
something like 2 x 8 if you also allow G729 to conference in.

Also - avoid a PVDM2-8 as far as conferencing go because it is limited
because it counts as a half DSP. Using a different chip to the rest.
It can handle a very small conference apparently but I have not tried
it. Also keep in mind that a single chip can only handle conferencing
and no other function at the same time -- that is share for other
services like transcoding and MTP. However - a single PVDM can do
multiple things like transcoding, conferencing etc as long as
conferencing has at least a dedicated dsp on the PVDM.

A lot of people are also worried about transcoding - normally when
they have remote branches with G729 across a wan for onnet call to a
central site and both the HQ and reote site use G711 in the site. This
is not nromally true. The cisco IP phones supports G729 natively and
if you have your regions (for codec selection) and locations correctly
configured the call manager signalling will tell a remote phone to use
the G729 codec when talking onnet to an HQ phone across the IP wan.

still learning so I hop I have it correct here.

erik erasmus
Telkom SA


________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Wed 2006-09-06 20:02
To: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up local DSP resources on 2821



Were the questions regarding DSPFARM configs in the email below ever
answered?

I have configured my DSPFARM on a 2811 but how do you actually make
these resources the first choice for the users at that remote site?  I
assume some configuration in the conf bridge resources and MRGL but I
can't find a doc specifically for the 2800 routers.  Anyone have a
link?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, July 18, 2006 3:20 PM
To: Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821


Hi Ted

Sounds like you have done it a couple of times.

Q1. I notice in the config examples they have an example that has

voice-card 1
no dspfarm
dsp services dspfarm

bit confusing - why no dsprarm and right below it dsp services dspfarm
???

Q2. They also show
associate profile 10 register CFB123456789966
associate profile 6 register MTP123456789988

where does the values after register come from - are they arbitrary
things one choose and need to be unique per site and per profile I
register etc.???


oip




~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From kthorngr at cisco.com  Wed Sep  6 14:59:42 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed, 6 Sep 2006 14:59:42 -0400
Subject: [cisco-voip] Cisco Brand Protection Contacts
In-Reply-To: <0c5101c6d1e2$0fc32860$6501a8c0@ShaneCompaq>
References: <0c5101c6d1e2$0fc32860$6501a8c0@ShaneCompaq>
Message-ID: <7dba46f847578de0e891559610797659@cisco.com>

See if you can get to this web site:
http://www.cisco.com/web/partners/brand-protection/index.html

There  is a link for "Reporting Breaches of Intellectual Property" 
which I believe is what you are looking for.

HTH,
Kevin
On Sep 6, 2006, at 2:27 PM, Sales at Doretel wrote:

> I know there is some Cisco guys on here. I have a customer that bought 
> some WIC cards that are fake from a grey market vendor. The contact I 
> had at Brand Protection is no longer in that position anymore. Can 
> someone please give me a call/email/im if you can help me point this 
> poor customer in the right direction.
> ?
> Thanks in advance...
> ?
> Shane Breen
> Doretel Communications, Inc.
> Office: 404.755.5721?
> sbreen at doretel.com
> AIM: shanebreen2003
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From CORTIZ at broward.org  Wed Sep  6 15:03:32 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed, 6 Sep 2006 15:03:32 -0400
Subject: [cisco-voip] Setting up local DSP resources on 2821
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F84D5D5@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A474B6@MREXVS01.bc.broward.cty>

I just sent an email regarding the config on the Call Mgr side.  If you
have a chance please let me know what Conf Bridge name they are
referring to in the CM pages.  I read it needs to match the name on the
gateway....which one?

 

Carlos

 

________________________________

From: Erik Erasmus (E) [mailto:ErasmuE4 at telkom.co.za] 
Sent: Wednesday, September 06, 2006 2:57 PM
To: Ortiz, Carlos; Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up local DSP resources on 2821

 

Most of them

 

I also have some sample configs on my laptop at work which I will make
available tomorrow. I managed to get hardware assisted (enhanced IOS)
conferencing on a router going nicely after reading the docs and getting
help from the group. And it is fairly simple to get it configured in CCM
as a media resource.

 

I also managed to find out the following useful info. cisco often refers
to DSPs and how many conferences etc one can do per DSP. I figured out
now that as far as PVDM2s go a PVDM2-16 counts as one DSP, a 2-32 counts
as 2 and a 2-48 as three and 2-64 as 4. Plus that generaly for G711 a
single DSP (that is one chip on the PVDM can do 8 x 8 meaning 8
conferences with 8 participants. This usually drops to something like 2
x 8 if you also allow G729 to conference in.

 

Also - avoid a PVDM2-8 as far as conferencing go because it is limited
because it counts as a half DSP. Using a different chip to the rest. It
can handle a very small conference apparently but I have not tried it.
Also keep in mind that a single chip can only handle conferencing and no
other function at the same time -- that is share for other services like
transcoding and MTP. However - a single PVDM can do multiple things like
transcoding, conferencing etc as long as conferencing has at least a
dedicated dsp on the PVDM.

 

A lot of people are also worried about transcoding - normally when they
have remote branches with G729 across a wan for onnet call to a central
site and both the HQ and reote site use G711 in the site. This is not
nromally true. The cisco IP phones supports G729 natively and if you
have your regions (for codec selection) and locations correctly
configured the call manager signalling will tell a remote phone to use
the G729 codec when talking onnet to an HQ phone across the IP wan.

 

still learning so I hop I have it correct here.

 

erik erasmus

Telkom SA

 

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Wed 2006-09-06 20:02
To: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up local DSP resources on 2821

Were the questions regarding DSPFARM configs in the email below ever
answered?

I have configured my DSPFARM on a 2811 but how do you actually make
these resources the first choice for the users at that remote site?  I
assume some configuration in the conf bridge resources and MRGL but I
can't find a doc specifically for the 2800 routers.  Anyone have a link?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, July 18, 2006 3:20 PM
To: Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821


Hi Ted

Sounds like you have done it a couple of times.

Q1. I notice in the config examples they have an example that has

voice-card 1
no dspfarm
dsp services dspfarm

bit confusing - why no dsprarm and right below it dsp services dspfarm
???

Q2. They also show
associate profile 10 register CFB123456789966
associate profile 6 register MTP123456789988

where does the values after register come from - are they arbitrary
things one choose and need to be unique per site and per profile I
register etc.???


oip

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at 
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
	
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From rratliff at cisco.com  Wed Sep  6 15:03:16 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 6 Sep 2006 15:03:16 -0400
Subject: [cisco-voip] Setting up local DSP resources on 2821
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E03A474B5@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E03A474B5@MREXVS01.bc.broward.cty>
Message-ID: <AD0E6F72-6009-439C-A1C6-2BAEEA9464F0@cisco.com>

The name in CCMAdmin is either a prefix followed by the mac-address  
of the interface specified by the 'sccp local' command OR a name  
specified by you in the dspfarm profile 'associate profile x register  
y' where x is the number of the dspfarm profile and y is any name you  
want.

If you are using the older PDVMs that don't support dspfarm profiles  
the name is not configurable in IOS and will always be the prefix  
(CFB, MTP) followed by the mac-address of the interface.

-Ryan

On Sep 6, 2006, at 2:56 PM, Ortiz, Carlos wrote:

I ran a show media resource status on the router and see the following:

Lowrise#sh media resource status
  Resource Providers:

  Resource  Provider ID :: FLEX_DSPRM Status :: REGISTERED
  Service Profiles
  MTP ::
  TRANSCODING ::
  CONFERENCING :: 1
  Applications :
  Application ID : SCCP, Status : REGISTERED

I guess that looks as it should.....

What appears incorrect is

Profile ID = 1, Service = CONFERENCING, Resource ID = 1
  Profile Description :
  Profile Admin State : UP
  Profile Operation State : ACTIVE IN PROGRESS
  Application : SCCP   Status : ASSOCIATION IN PROGRESS
  Resource Provider : FLEX_DSPRM   Status : UP
  Number of Resource Configured : 2
  Number of Resource Available : 2

There is a required parameter on the Call Manager:

Conference Bridge Name  Enter the same name that exists in the gateway
Command Line Interface (CLI).


What name are they referring too?

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Wednesday, September 06, 2006 2:25 PM
To: Ortiz, Carlos
Cc: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821

It's CallManager that decides what resources a device uses. The
configuration on the router is simply to get it registered.

You put the device in a MRG, and then the order of MRGs in the MRGL
decides preference.

-Ryan

On Sep 6, 2006, at 2:02 PM, Ortiz, Carlos wrote:

Were the questions regarding DSPFARM configs in the email below ever
answered?

I have configured my DSPFARM on a 2811 but how do you actually make
these resources the first choice for the users at that remote site?  I
assume some configuration in the conf bridge resources and MRGL but I
can't find a doc specifically for the 2800 routers.  Anyone have a link?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, July 18, 2006 3:20 PM
To: Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821


Hi Ted

Sounds like you have done it a couple of times.

Q1. I notice in the config examples they have an example that has

voice-card 1
no dspfarm
dsp services dspfarm

bit confusing - why no dsprarm and right below it dsp services dspfarm
???

Q2. They also show
associate profile 10 register CFB123456789966
associate profile 6 register MTP123456789988

where does the values after register come from - are they arbitrary
things one choose and need to be unique per site and per profile I
register etc.???


oip

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


From ErasmuE4 at telkom.co.za  Wed Sep  6 15:07:36 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 6 Sep 2006 21:07:36 +0200
Subject: [cisco-voip] Setting up local DSP resources on 2821
References: <26B357EF9FB7D04A989C9E82A9E5436E03A474B6@MREXVS01.bc.broward.cty>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D5D7@CNTRRA20-XCS01.telkom.co.za>


yes - can tomorow - send more help

it is simple -- the name you apply in the IOS router -- if it is the
newer DSPs is aname you select and does not have to be a mac address
or anything specific - pick a nice name like branch1-conb1 etc and
then in ccm use the same name when adding the resource - skinny
messages will then make sure that resource registers correctly


cisco has a doc -- something like ccm - ios integration etc. can not
remember but it explains nicely.


erik


________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Wed 2006-09-06 21:03
To: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up local DSP resources on 2821



I just sent an email regarding the config on the Call Mgr side.  If
you have a chance please let me know what Conf Bridge name they are
referring to in the CM pages.  I read it needs to match the name on
the gateway....which one?



Carlos



________________________________

From: Erik Erasmus (E) [mailto:ErasmuE4 at telkom.co.za]
Sent: Wednesday, September 06, 2006 2:57 PM
To: Ortiz, Carlos; Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up local DSP resources on 2821



Most of them



I also have some sample configs on my laptop at work which I will make
available tomorrow. I managed to get hardware assisted (enhanced IOS)
conferencing on a router going nicely after reading the docs and
getting help from the group. And it is fairly simple to get it
configured in CCM as a media resource.



I also managed to find out the following useful info. cisco often
refers to DSPs and how many conferences etc one can do per DSP. I
figured out now that as far as PVDM2s go a PVDM2-16 counts as one DSP,
a 2-32 counts as 2 and a 2-48 as three and 2-64 as 4. Plus that
generaly for G711 a single DSP (that is one chip on the PVDM can do 8
x 8 meaning 8 conferences with 8 participants. This usually drops to
something like 2 x 8 if you also allow G729 to conference in.



Also - avoid a PVDM2-8 as far as conferencing go because it is limited
because it counts as a half DSP. Using a different chip to the rest.
It can handle a very small conference apparently but I have not tried
it. Also keep in mind that a single chip can only handle conferencing
and no other function at the same time -- that is share for other
services like transcoding and MTP. However - a single PVDM can do
multiple things like transcoding, conferencing etc as long as
conferencing has at least a dedicated dsp on the PVDM.



A lot of people are also worried about transcoding - normally when
they have remote branches with G729 across a wan for onnet call to a
central site and both the HQ and reote site use G711 in the site. This
is not nromally true. The cisco IP phones supports G729 natively and
if you have your regions (for codec selection) and locations correctly
configured the call manager signalling will tell a remote phone to use
the G729 codec when talking onnet to an HQ phone across the IP wan.



still learning so I hop I have it correct here.



erik erasmus

Telkom SA





________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Wed 2006-09-06 20:02
To: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up local DSP resources on 2821

Were the questions regarding DSPFARM configs in the email below ever
answered?

I have configured my DSPFARM on a 2811 but how do you actually make
these resources the first choice for the users at that remote site?  I
assume some configuration in the conf bridge resources and MRGL but I
can't find a doc specifically for the 2800 routers.  Anyone have a
link?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, July 18, 2006 3:20 PM
To: Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821


Hi Ted

Sounds like you have done it a couple of times.

Q1. I notice in the config examples they have an example that has

voice-card 1
no dspfarm
dsp services dspfarm

bit confusing - why no dsprarm and right below it dsp services dspfarm
???

Q2. They also show
associate profile 10 register CFB123456789966
associate profile 6 register MTP123456789988

where does the values after register come from - are they arbitrary
things one choose and need to be unique per site and per profile I
register etc.???


oip

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From mark.turpin at bryancave.com  Wed Sep  6 15:13:34 2006
From: mark.turpin at bryancave.com (Turpin, Mark)
Date: Wed, 6 Sep 2006 15:13:34 -0400
Subject: [cisco-voip] Unity (integrated) questions vis-a-vis Outlook,
	VMO and a separate Exchange server
In-Reply-To: <200609061343.49819.netfortius@gmail.com>
Message-ID: <2496838411CFA74BAFD10B4C3757616708FA95@FRMEXBE02.bc.firm.local>

A google search for 'unity architecture' yielded some decent results.

I'd recommend http://www.ciscounitytools.com/Documents.htm as a good
starting point.

-Mark

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Netfortius
Sent: Wednesday, September 06, 2006 1:44 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity (integrated) questions vis-a-vis Outlook,VMO
and a separate Exchange server

Could someone point me to a very thorough description of the
architecture of 
Unity - the integrated version (the one with "built-in" exchange data 
store) - and its relationship with that Exchange data store, Outlook 
capabilities as MAPI client for an existing Exchange email server, VMO
usage 
in such an environment, etc.? Here are a couple of things I am trying to

answer, associated with a deployment of such:

- if Outlook is a MAPI client - could I "go" after the messages in an 
integrated unity environment (its exchange data store) as if it was an 
exchange server?
- if the above is true (which I think it is, simply because the OWA is 
supposed to do the same thing), and knowing that Outlook can only accept
an 
Exchange serve account at a time - is there any relationship I could 
establish between the data store associated with the unity and an
Exchange 
server already used as email server, such that I can get the voice
messages 
from unity/exch data store in the same time with emails (trust + 
<something>)?
- what is the VMO role in such an environment? If it is capable of
creating 
voice messages to be sent to other mailboxes, isn't that equivalent to
being 
able to be a "MAPI" client of the unity data store, or use Outlook for
such - 
and - under this scenario - wouldn't this allow "pulling" into some
Outlook 
local folder voice messages, from the unity data store, while Outlook is
also 
an email client of the mail Exchange server?

I am sorry if the above questions do not make much sense, but I am
trying to 
gain a better understanding of this architecture, and I am not sure how 
things actually work (some people even telling me that VMO is not merely
an 
Outlook form, but also having some DLLs associated with it, upon 
installation, which allow it to communicate directly with the Unity 
server?!?)

Oh - we'll be using Unity ver 4.2.

Pointers to RTFMs, books, links, etc will be appreciated

TIA,
Stefan
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip 
 
This electronic message is from a law firm. It may contain confidential or privileged information. If you received this transmission in error, please reply to the sender to advise of the error and delete this transmission and any attachments.
 
IRS Circular 230 Disclosure: To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice contained in this communication (including any attachments) is not intended or written 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.


From ErasmuE4 at telkom.co.za  Wed Sep  6 15:22:27 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 6 Sep 2006 21:22:27 +0200
Subject: [cisco-voip] Setting up local DSP resources on 2821
References: <26B357EF9FB7D04A989C9E82A9E5436E03A474B6@MREXVS01.bc.broward.cty>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D5D9@CNTRRA20-XCS01.telkom.co.za>


Carlos

search for this doc

Cisco CallManager and Cisco IOS Interoperability Guide
 

I found the chapter with the example on conferencing with the PVDM2
example very helpful



erik

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Wed 2006-09-06 21:03
To: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up local DSP resources on 2821



I just sent an email regarding the config on the Call Mgr side.  If
you have a chance please let me know what Conf Bridge name they are
referring to in the CM pages.  I read it needs to match the name on
the gateway....which one?



Carlos



________________________________

From: Erik Erasmus (E) [mailto:ErasmuE4 at telkom.co.za]
Sent: Wednesday, September 06, 2006 2:57 PM
To: Ortiz, Carlos; Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up local DSP resources on 2821



Most of them



I also have some sample configs on my laptop at work which I will make
available tomorrow. I managed to get hardware assisted (enhanced IOS)
conferencing on a router going nicely after reading the docs and
getting help from the group. And it is fairly simple to get it
configured in CCM as a media resource.



I also managed to find out the following useful info. cisco often
refers to DSPs and how many conferences etc one can do per DSP. I
figured out now that as far as PVDM2s go a PVDM2-16 counts as one DSP,
a 2-32 counts as 2 and a 2-48 as three and 2-64 as 4. Plus that
generaly for G711 a single DSP (that is one chip on the PVDM can do 8
x 8 meaning 8 conferences with 8 participants. This usually drops to
something like 2 x 8 if you also allow G729 to conference in.



Also - avoid a PVDM2-8 as far as conferencing go because it is limited
because it counts as a half DSP. Using a different chip to the rest.
It can handle a very small conference apparently but I have not tried
it. Also keep in mind that a single chip can only handle conferencing
and no other function at the same time -- that is share for other
services like transcoding and MTP. However - a single PVDM can do
multiple things like transcoding, conferencing etc as long as
conferencing has at least a dedicated dsp on the PVDM.



A lot of people are also worried about transcoding - normally when
they have remote branches with G729 across a wan for onnet call to a
central site and both the HQ and reote site use G711 in the site. This
is not nromally true. The cisco IP phones supports G729 natively and
if you have your regions (for codec selection) and locations correctly
configured the call manager signalling will tell a remote phone to use
the G729 codec when talking onnet to an HQ phone across the IP wan.



still learning so I hop I have it correct here.



erik erasmus

Telkom SA





________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Wed 2006-09-06 20:02
To: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Setting up local DSP resources on 2821

Were the questions regarding DSPFARM configs in the email below ever
answered?

I have configured my DSPFARM on a 2811 but how do you actually make
these resources the first choice for the users at that remote site?  I
assume some configuration in the conf bridge resources and MRGL but I
can't find a doc specifically for the 2800 routers.  Anyone have a
link?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, July 18, 2006 3:20 PM
To: Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821


Hi Ted

Sounds like you have done it a couple of times.

Q1. I notice in the config examples they have an example that has

voice-card 1
no dspfarm
dsp services dspfarm

bit confusing - why no dsprarm and right below it dsp services dspfarm
???

Q2. They also show
associate profile 10 register CFB123456789966
associate profile 6 register MTP123456789988

where does the values after register come from - are they arbitrary
things one choose and need to be unique per site and per profile I
register etc.???


oip

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
-------------- next part --------------
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From kthorngr at cisco.com  Wed Sep  6 15:25:03 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed, 6 Sep 2006 15:25:03 -0400
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe at
	100%	cpu
In-Reply-To: <44FF12F1.9070306@tns.its.psu.edu>
References: <C1FE15183DA37645BC0633BC604E44F0030FFD1E@USNAEXCH.na.didata.local>
	<44FF12F1.9070306@tns.its.psu.edu>
Message-ID: <62c4045931cd101a29197edbc3ce1bf6@cisco.com>

Well, I have not done much with IPCC X but I found a case where MS 
found that overly large application, system and security event logs 
caused high CPU utilization.  Might be worth checking out.

Kevin
On Sep 6, 2006, at 2:26 PM, Simon, Bill wrote:

> Consider this a "me too" post - we've been seeing some CPU-high 
> behavior
> from those services lately as well.  Restarting the services seems to
> bring it back down to normal.
>
> Jason Aarons (US) wrote:
>> Any ideas why winmgmt.exe might be at 100% on both boxes in my CRS
>> 4.0(4) cluster ? ?jason
>>
>>
>>
>> OS 2000.2.7sr8
>>
>>
>>
>> Can I disable the Windows Instrument Management service and the 
>> Windows
>> Instrument Management Extensions service?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



From CORTIZ at broward.org  Wed Sep  6 15:38:19 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Wed, 6 Sep 2006 15:38:19 -0400
Subject: [cisco-voip] Setting up local DSP resources on 2821
In-Reply-To: <AD0E6F72-6009-439C-A1C6-2BAEEA9464F0@cisco.com>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A474B7@MREXVS01.bc.broward.cty>

I guess my original question wasn't clear, but that's what I needed to
know.  Now that I have made the name in dspfarm profile config on the
gateway and the CM CFB configuration page the same I see the following
from show dspfarm profile 1:

Profile ID = 1, Service = CONFERENCING, Resource ID = 1  
 Profile Description :  
 Profile Admin State : UP 
 Profile Operation State : ACTIVE 
 Application : SCCP   Status : ASSOCIATED 
 Resource Provider : FLEX_DSPRM   Status : UP 
 Number of Resource Configured : 2 
 Number of Resource Available : 2

Thanks for the help!

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Wednesday, September 06, 2006 3:03 PM
To: Ortiz, Carlos
Cc: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821

The name in CCMAdmin is either a prefix followed by the mac-address  
of the interface specified by the 'sccp local' command OR a name  
specified by you in the dspfarm profile 'associate profile x register  
y' where x is the number of the dspfarm profile and y is any name you  
want.

If you are using the older PDVMs that don't support dspfarm profiles  
the name is not configurable in IOS and will always be the prefix  
(CFB, MTP) followed by the mac-address of the interface.

-Ryan

On Sep 6, 2006, at 2:56 PM, Ortiz, Carlos wrote:

I ran a show media resource status on the router and see the following:

Lowrise#sh media resource status
  Resource Providers:

  Resource  Provider ID :: FLEX_DSPRM Status :: REGISTERED
  Service Profiles
  MTP ::
  TRANSCODING ::
  CONFERENCING :: 1
  Applications :
  Application ID : SCCP, Status : REGISTERED

I guess that looks as it should.....

What appears incorrect is

Profile ID = 1, Service = CONFERENCING, Resource ID = 1
  Profile Description :
  Profile Admin State : UP
  Profile Operation State : ACTIVE IN PROGRESS
  Application : SCCP   Status : ASSOCIATION IN PROGRESS
  Resource Provider : FLEX_DSPRM   Status : UP
  Number of Resource Configured : 2
  Number of Resource Available : 2

There is a required parameter on the Call Manager:

Conference Bridge Name  Enter the same name that exists in the gateway
Command Line Interface (CLI).


What name are they referring too?

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Wednesday, September 06, 2006 2:25 PM
To: Ortiz, Carlos
Cc: Erik Erasmus (E); Ted Nugent; Wydra, Jason;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821

It's CallManager that decides what resources a device uses. The
configuration on the router is simply to get it registered.

You put the device in a MRG, and then the order of MRGs in the MRGL
decides preference.

-Ryan

On Sep 6, 2006, at 2:02 PM, Ortiz, Carlos wrote:

Were the questions regarding DSPFARM configs in the email below ever
answered?

I have configured my DSPFARM on a 2811 but how do you actually make
these resources the first choice for the users at that remote site?  I
assume some configuration in the conf bridge resources and MRGL but I
can't find a doc specifically for the 2800 routers.  Anyone have a link?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Tuesday, July 18, 2006 3:20 PM
To: Ted Nugent; Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Setting up local DSP resources on 2821


Hi Ted

Sounds like you have done it a couple of times.

Q1. I notice in the config examples they have an example that has

voice-card 1
no dspfarm
dsp services dspfarm

bit confusing - why no dsprarm and right below it dsp services dspfarm
???

Q2. They also show
associate profile 10 register CFB123456789966
associate profile 6 register MTP123456789988

where does the values after register come from - are they arbitrary
things one choose and need to be unique per site and per profile I
register etc.???


oip

_______________________________________________
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  Wed Sep  6 16:03:27 2006
From: bills at tns.its.psu.edu (Simon, Bill)
Date: Wed, 06 Sep 2006 16:03:27 -0400
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe at
 100%	cpu
In-Reply-To: <62c4045931cd101a29197edbc3ce1bf6@cisco.com>
References: <C1FE15183DA37645BC0633BC604E44F0030FFD1E@USNAEXCH.na.didata.local>
	<44FF12F1.9070306@tns.its.psu.edu>
	<62c4045931cd101a29197edbc3ce1bf6@cisco.com>
Message-ID: <44FF298F.1030401@tns.its.psu.edu>

How would you define "overly large"?



Kevin Thorngren wrote:
> Well, I have not done much with IPCC X but I found a case where MS found 
> that overly large application, system and security event logs caused 
> high CPU utilization.  Might be worth checking out.
> 
> Kevin
> On Sep 6, 2006, at 2:26 PM, Simon, Bill wrote:
> 
>> Consider this a "me too" post - we've been seeing some CPU-high behavior
>> from those services lately as well.  Restarting the services seems to
>> bring it back down to normal.
>>
>> Jason Aarons (US) wrote:
>>> Any ideas why winmgmt.exe might be at 100% on both boxes in my CRS
>>> 4.0(4) cluster ? ?jason
>>>
>>>
>>>
>>> OS 2000.2.7sr8
>>>
>>>
>>>
>>> Can I disable the Windows Instrument Management service and the Windows
>>> Instrument Management Extensions service?

From jason.aarons at us.didata.com  Wed Sep  6 16:04:49 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 6 Sep 2006 16:04:49 -0400
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe at
	100%	cpu
Message-ID: <C1FE15183DA37645BC0633BC604E44F0030FFF5A@USNAEXCH.na.didata.local>

Each of the 4 logs in Event viewer had be increased to 99,999 in size by
the customer, I put them back to 8,000 and cleared them.

Do you know of a MS KB article that matches this?

-----Original Message-----
From: Kevin Thorngren [mailto:kthorngr at cisco.com] 
Sent: Wednesday, September 06, 2006 3:25 PM
To: Simon, Bill; Jason Aarons (US)
Cc: ciscovoip Voip
Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
at 100% cpu

Well, I have not done much with IPCC X but I found a case where MS 
found that overly large application, system and security event logs 
caused high CPU utilization.  Might be worth checking out.

Kevin
On Sep 6, 2006, at 2:26 PM, Simon, Bill wrote:

> Consider this a "me too" post - we've been seeing some CPU-high 
> behavior
> from those services lately as well.  Restarting the services seems to
> bring it back down to normal.
>
> Jason Aarons (US) wrote:
>> Any ideas why winmgmt.exe might be at 100% on both boxes in my CRS
>> 4.0(4) cluster ? -jason
>>
>>
>>
>> OS 2000.2.7sr8
>>
>>
>>
>> Can I disable the Windows Instrument Management service and the 
>> Windows
>> Instrument Management Extensions service?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


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

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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notify us immediately by replying to this message and deleting it
from your computer. Thank you.



From kthorngr at cisco.com  Wed Sep  6 16:25:52 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed, 6 Sep 2006 16:25:52 -0400
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe at
	100%	cpu
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0030FFF5A@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0030FFF5A@USNAEXCH.na.didata.local>
Message-ID: <b5d3d01249fb5c187d7616c927e4d7bb@cisco.com>

I did not see one referenced in the case.  It would be interesting to 
find out if this does resolve the CPU spike.

Kevin
On Sep 6, 2006, at 4:04 PM, Jason Aarons ((US)) wrote:

> Each of the 4 logs in Event viewer had be increased to 99,999 in size 
> by
> the customer, I put them back to 8,000 and cleared them.
>
> Do you know of a MS KB article that matches this?
>
> -----Original Message-----
> From: Kevin Thorngren [mailto:kthorngr at cisco.com]
> Sent: Wednesday, September 06, 2006 3:25 PM
> To: Simon, Bill; Jason Aarons (US)
> Cc: ciscovoip Voip
> Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
> at 100% cpu
>
> Well, I have not done much with IPCC X but I found a case where MS
> found that overly large application, system and security event logs
> caused high CPU utilization.  Might be worth checking out.
>
> Kevin
> On Sep 6, 2006, at 2:26 PM, Simon, Bill wrote:
>
>> Consider this a "me too" post - we've been seeing some CPU-high
>> behavior
>> from those services lately as well.  Restarting the services seems to
>> bring it back down to normal.
>>
>> Jason Aarons (US) wrote:
>>> Any ideas why winmgmt.exe might be at 100% on both boxes in my CRS
>>> 4.0(4) cluster ? -jason
>>>
>>>
>>>
>>> OS 2000.2.7sr8
>>>
>>>
>>>
>>> Can I disable the Windows Instrument Management service and the
>>> Windows
>>> Instrument Management Extensions service?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From kthorngr at cisco.com  Wed Sep  6 16:29:28 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed, 6 Sep 2006 16:29:28 -0400
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe at
	100%	cpu
In-Reply-To: <44FF298F.1030401@tns.its.psu.edu>
References: <C1FE15183DA37645BC0633BC604E44F0030FFD1E@USNAEXCH.na.didata.local>
	<44FF12F1.9070306@tns.its.psu.edu>
	<62c4045931cd101a29197edbc3ce1bf6@cisco.com>
	<44FF298F.1030401@tns.its.psu.edu>
Message-ID: <97a0928d73d349a9a04813e31100e847@cisco.com>

In the case I referenced they found that one of the logs was 260MB with 
over 1 million events.  They exported the log and set the max size to 
100MB (I am not sure if these were recommendations from MS or just a 
size they chose to capture a certain number of days of activity).  
Although the specific process was never tracked down the conclusion 
form MS was that a particular process was accessing the event logs 
causing winmgmt to spike the CPU.

I don't have a definition for "overly large" though ;-)

Be careful when exporting the logs as I have seen this cause CPU spikes 
as well.

Kevin
On Sep 6, 2006, at 4:03 PM, Simon, Bill wrote:

> How would you define "overly large"?
>
>
>
> Kevin Thorngren wrote:
>> Well, I have not done much with IPCC X but I found a case where MS 
>> found that overly large application, system and security event logs 
>> caused high CPU utilization.  Might be worth checking out.
>> Kevin
>> On Sep 6, 2006, at 2:26 PM, Simon, Bill wrote:
>>> Consider this a "me too" post - we've been seeing some CPU-high 
>>> behavior
>>> from those services lately as well.  Restarting the services seems to
>>> bring it back down to normal.
>>>
>>> Jason Aarons (US) wrote:
>>>> Any ideas why winmgmt.exe might be at 100% on both boxes in my CRS
>>>> 4.0(4) cluster ? ?jason
>>>>
>>>>
>>>>
>>>> OS 2000.2.7sr8
>>>>
>>>>
>>>>
>>>> Can I disable the Windows Instrument Management service and the 
>>>> Windows
>>>> Instrument Management Extensions service?
>



From jgrace at digitel.net  Wed Sep  6 16:50:45 2006
From: jgrace at digitel.net (James Grace)
Date: Wed, 6 Sep 2006 16:50:45 -0400
Subject: [cisco-voip] can we use bat to create cti port or cti point
Message-ID: <200609061650484.SM01472@jgm>

We have a lot of user to add that only will have voice mail and no ip
phones.  I going to create cti port or points to forward incoming calls to
unity.  Can this be done in bulk

 

James D. Grace 

CCNP CCNA MCSE MCSA 

Digitel Corporation

System Engineer / Professional Svc.

MSN messenger ID:  jgrace29 at hotmail.com

Mac IChat ID:  jgrace29 at mac.com

 

 

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From CMStaffin at ra.rockwell.com  Wed Sep  6 17:33:55 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Wed, 6 Sep 2006 16:33:55 -0500
Subject: [cisco-voip] can we use bat to create cti port or cti point
In-Reply-To: <200609061650484.SM01472@jgm>
Message-ID: <OF6AA84A18.F6057496-ON862571E1.007570F8-862571E1.00765EED@ra.rockwell.com>

James,

What we have actually done is use extension mobility.  That way if the 
user ever does come into the office they can log into a phone and you can 
import them with BAT :)

Craig




"James Grace" <jgrace at digitel.net> 
Sent by: cisco-voip-bounces at puck.nether.net
09/06/2006 03:50 PM

To
<cisco-voip at puck.nether.net>
cc

Subject
[cisco-voip] can we use bat to create cti port or cti point






We have a lot of user to add that only will have voice mail and no ip 
phones.  I going to create cti port or points to forward incoming calls to 
unity.  Can this be done in bulk
 
James D. Grace 
CCNP CCNA MCSE MCSA 
Digitel Corporation
System Engineer / Professional Svc.
MSN messenger ID:  jgrace29 at hotmail.com
Mac IChat ID:  jgrace29 at mac.com
 
 _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From kds850 at hotmail.com  Thu Sep  7 02:24:59 2006
From: kds850 at hotmail.com (kevin k)
Date: Thu, 07 Sep 2006 02:24:59 -0400
Subject: [cisco-voip] ccm 4.0 on laptop - help
Message-ID: <BAY105-F30F82567331811D340DD2C8E360@phx.gbl>

hi all

trying to learn ccm.  i got the steps on how to install on pc went through.
i did vmware + cisco os sr
all were good.
when i try to install  i get the following error message in the log.
"	Did not find file:	"C:\Program Files\Common 
Files\Cisco\Logs\CCMServices.ctl"
09/07/2006 02:01:39.109 CCMSetup.cpp fnrsInitialize	FindFirstFile: Error: 2: 
The system cannot find the file specified.
09/07/2006 02:01:39.109 CCMSetup.cpp fnrsInitialize	iuFindFile: Function 
ended.
"
i can not go on without making it work.
any idea?

thanks

_________________________________________________________________
All-in-one security and maintenance for your PC.  Get a free 90-day trial!   
http://www.windowsonecare.com/trial.aspx?sc_cid=msn_hotmail


From dandy at volja.net  Thu Sep  7 06:19:23 2006
From: dandy at volja.net (Aleksandar Ristoski)
Date: Thu, 7 Sep 2006 12:19:23 +0200
Subject: [cisco-voip] translation of empty calling number
Message-ID: <200609071219.23979.dandy@volja.net>

Hi!

As the subject says, is there a possibility to "translate"or attach some 
string if the calling number is empty? I'll be gratefull to all if You can 
share a hint.

Best regards!

From acm at netuse.de  Thu Sep  7 07:04:43 2006
From: acm at netuse.de (Ahmad Cheikh Moussa)
Date: Thu, 07 Sep 2006 13:04:43 +0200
Subject: [cisco-voip] Callmanager 4.1(3) and SRST
Message-ID: <44FFFCCB.7070102@netuse.de>

Hi!

Quick question. Can I define, which IP Phone
has the ability to use SRST and which not ?
Is there a way to do so via device pool ?
I have approximately 300 Phones, but a gateway which
only support SRST for 48 Phones. I thought, that I define
the 48 Phones, which are allowed to use the SRST.

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 KWhitson at cmbtec.com  Thu Sep  7 07:12:16 2006
From: KWhitson at cmbtec.com (Karl Whitson)
Date: Thu, 7 Sep 2006 07:12:16 -0400
Subject: [cisco-voip] Callmanager 4.1(3) and SRST
In-Reply-To: <44FFFCCB.7070102@netuse.de>
Message-ID: <8A4D67E299AA254CAB2B5985EEDD865804BA97@cmb-bm-sbs.CMB-BM.Local>

Ahmad,

First configure your SRST router in Call Manager under Menu option System, SRST.  Then create two Device Pools one with the SRST router you configured in step 1.  The other Device Pool with the SRST Reference set as Disable.  Then assign the Device Pools to the appropriate devices.  Only those devices with the SRST reference will register with the gateway during a WAN outage.

Karl Whitson
CMB Technologies
40 Cape Rd.
Milford, MA  01757
DID:  (508) 422-8413
Cell:  (603) 930-4189
Fax:  (508) 634-8678
Karl.Whitson at cmbtec.com
www.cmbtec.com
TO OPEN A SUPPORT CASE EMAIL SUPPORT at CMBTEC.COM OR CALL (508) 422-8429
 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ahmad Cheikh Moussa
Sent: Thursday, September 07, 2006 7:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Callmanager 4.1(3) and SRST

Hi!

Quick question. Can I define, which IP Phone
has the ability to use SRST and which not ?
Is there a way to do so via device pool ?
I have approximately 300 Phones, but a gateway which
only support SRST for 48 Phones. I thought, that I define
the 48 Phones, which are allowed to use the SRST.

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 acm at netuse.de  Thu Sep  7 07:19:15 2006
From: acm at netuse.de (Ahmad Cheikh Moussa)
Date: Thu, 07 Sep 2006 13:19:15 +0200
Subject: [cisco-voip] Callmanager 4.1(3) and SRST
In-Reply-To: <8A4D67E299AA254CAB2B5985EEDD865804BA97@cmb-bm-sbs.CMB-BM.Local>
References: <8A4D67E299AA254CAB2B5985EEDD865804BA97@cmb-bm-sbs.CMB-BM.Local>
Message-ID: <45000033.6080708@netuse.de>

Hi!

Karl Whitson wrote:
> Ahmad,
> 
> First configure your SRST router in Call Manager under Menu option System, SRST.  Then create two Device Pools one with the SRST router you configured in step 1.  The other Device Pool with the SRST Reference set as Disable.  Then assign the Device Pools to the appropriate devices.  Only those devices with the SRST reference will register with the gateway during a WAN outage.
> 
Thanks for your reply. I only searched under the callmanager group and
not in the device pool itself.

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 KWhitson at cmbtec.com  Thu Sep  7 07:31:57 2006
From: KWhitson at cmbtec.com (Karl Whitson)
Date: Thu, 7 Sep 2006 07:31:57 -0400
Subject: [cisco-voip] translation of empty calling number
In-Reply-To: <200609071219.23979.dandy@volja.net>
Message-ID: <8A4D67E299AA254CAB2B5985EEDD865804BA98@cmb-bm-sbs.CMB-BM.Local>

Aleksandar,

In Call Manager configuration go to Service, Service Parameters and
select your Publisher server and Cisco CallManager as the service.
Press the Advanced button.  Scroll down through the list until you get
to the "Clusterwide Parameters (Device - General)" section and go to
"Unknown Caller ID" option.  This will allow you to enter a Number for a
call with no Caller ID.  The "Unknown Caller ID Text" option will allow
you to enter the text to be displayed on the phone. 

Karl Whitson
CMB Technologies
40 Cape Rd.
Milford, MA  01757
DID:  (508) 422-8413
Cell:  (603) 930-4189
Fax:  (508) 634-8678
Karl.Whitson at cmbtec.com
www.cmbtec.com
TO OPEN A SUPPORT CASE EMAIL SUPPORT at CMBTEC.COM OR CALL (508) 422-8429
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Aleksandar
Ristoski
Sent: Thursday, September 07, 2006 6:19 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] translation of empty calling number

Hi!

As the subject says, is there a possibility to "translate"or attach some

string if the calling number is empty? I'll be gratefull to all if You
can 
share a hint.

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


From Matt.Slaga at us.didata.com  Thu Sep  7 08:22:43 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 7 Sep 2006 08:22:43 -0400
Subject: [cisco-voip] can we use bat to create cti port or cti point
Message-ID: <C1FE15183DA37645BC0633BC604E44F0031003CB@USNAEXCH.na.didata.local>

You can use BAT to import CTI Ports but not route points.  When you
create your device template, select cti port as the device type.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Wednesday, September 06, 2006 4:51 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] can we use bat to create cti port or cti point

 

We have a lot of user to add that only will have voice mail and no ip
phones.  I going to create cti port or points to forward incoming calls
to unity.  Can this be done in bulk

 

James D. Grace 

CCNP CCNA MCSE MCSA 

Digitel Corporation

System Engineer / Professional Svc.

MSN messenger ID:  jgrace29 at hotmail.com

Mac IChat ID:  jgrace29 at mac.com

 

 




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

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From ealeatherman at gmail.com  Thu Sep  7 08:50:19 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 7 Sep 2006 08:50:19 -0400
Subject: [cisco-voip] Cisco Unity Connection 1.2.1
In-Reply-To: <e268831b0609042216g3c0469wf9ffe4cb32b963a0@mail.gmail.com>
References: <e268831b0609042216g3c0469wf9ffe4cb32b963a0@mail.gmail.com>
Message-ID: <94a1afde0609070550u3a953882h2a6c05fd152ad115@mail.gmail.com>

Luis, there is a command-line switch you can use to install connections on a
regular win2k3 server instead of using a platform configuration disc. I
believe it is /nopcd but its been a few months since I had to do it.

On 9/5/06, Luis Simonet <luis.simonet at gmail.com> wrote:
>
> I need to install Unity Connection on an existing Windows server 2003, I
> will used it for training. I have the license from Cisco. But when I try to
> run the installation it wants me to install 2003 from the DVD that came with
> Unity Connection. Is there a way to bypass it and use the version of 2003
> that is already install.
>
> Thanks
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Thu Sep  7 08:55:29 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 7 Sep 2006 08:55:29 -0400
Subject: [cisco-voip] Unity
In-Reply-To: <001a01c6d064$5040aa40$6721a8c0@avodaq.com>
References: <001a01c6d064$5040aa40$6721a8c0@avodaq.com>
Message-ID: <94a1afde0609070555x45e8a06ew8014a59133d7836b@mail.gmail.com>

I would kinda expect it to just play the next message in the queue. As soon
as the first person starts listening to the first message, that message
would get marked read in the message store (assuming it works like
Unity+exchange), so when the next caller tried to listen to new message I
would think it would start with the next message marked as unread, which
would be the second message. Never tried it before though :)

2. Scenario with Unity Connection
>
> - Two IP Phones
> - First IP Phone DN: 1000
> - Second IP Phone DN: 2000
> - One Voicemail box for DN 1000, with an alternate extension 2000
> - Several new voice messages for DN 1000
>
> First IP Phone calls Unity Connection and presses 1 to listen to new voice
> messages. During first IP Phone is listening to new voice message, the
> second IP Phone calls the Unity Connection, too. Presses 1 to listen to new
> voice messages.
> What will happen? Will the second IP Phone here the next voice message
> (the message after the message currently playing on the first IP Phone), the
> same voice message as the first IP Phone, etc?
>
>

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Thu Sep  7 08:58:40 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 7 Sep 2006 08:58:40 -0400
Subject: [cisco-voip] Which Cisco VoIP Products properly mark DSCP
In-Reply-To: <AB0A35F2486F81468971AA68558BE71904AD22C2@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE71904AD22C2@smieln01winm01.micoanet.com>
Message-ID: <94a1afde0609070558i139012c5m6d7693f129276edf@mail.gmail.com>

As far as I know, callmanager sets its its qos markings properly and you can
trust it. We are running Unity version 4.0.4 and it does not mark its
traffic properly, so we have to manually do that ourselves. I do not know
about the other 2 products, but I would be interested in finding out if IPCC
does, if I get time i'll pop a sniffer on one and find out.

On 9/5/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>  All,
>
>
>
> I am curious if anyone knows of a matrix or table that states what Cisco
> VoIP products and versions properly set DSCP values once they pass
> information onto the network.  I am specifically looking for the following
> products.
>
>
>
>    - CallManager
>    - Unity
>    - IPCC Express
>    - MeetingPlace Express
>
>
>
> It would be nice to know what products I can simply trust DSCP values and
> which other ones will have to have custom QoS that properly marks voice.
>
>
>
> Sincerely,
>
>
>
> Matthew M. Linsemier, CCSP/CCNP/CCDA
> Senior Network Engineer
> American Physicians
> Voice: 517.324.6695
> Fax: 517.333.2812
> E-Mail: mlinsemier at apcapital.com
>
>
>
> ------------------------------
>
> *CONFIDENTIALITY STATEMENT*
>
> This communication and any attachments are *CONFIDENTIAL* and may be
> protected by one or more legal privileges. It is intended solely for the use
> of the addressee identified above. If you are not the intended recipient,
> any use, disclosure, copying or distribution of this communication is *
> UNAUTHORIZED*. Neither this information block, the typed name of the
> sender, nor anything else in this message is intended to constitute an
> electronic signature unless a specific statement to the contrary is included
> in this message. If you have received this communication in error, please
> immediately contact me and delete this communication from your computer.
> Thank you.
>
> ------------------------------
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From zephy316 at gmail.com  Thu Sep  7 09:18:18 2006
From: zephy316 at gmail.com (Israel Lang)
Date: Thu, 7 Sep 2006 09:18:18 -0400
Subject: [cisco-voip] Cisco Unity Connection 1.2.1
In-Reply-To: <94a1afde0609070550u3a953882h2a6c05fd152ad115@mail.gmail.com>
References: <e268831b0609042216g3c0469wf9ffe4cb32b963a0@mail.gmail.com>
	<94a1afde0609070550u3a953882h2a6c05fd152ad115@mail.gmail.com>
Message-ID: <91e236d30609070618l2758ba98k81db9b728bf0d768@mail.gmail.com>

I will warn you though that TAC will basically tell you they can't support
you if you do this. In fact in one instance they thought I had used the
switch and basically told me to wipe and start over using the PCD.

I have also had some weird things happen on my test box, that were only
corrected when I installed with the PCD.

On 9/7/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> Luis, there is a command-line switch you can use to install connections on
> a regular win2k3 server instead of using a platform configuration disc. I
> believe it is /nopcd but its been a few months since I had to do it.
>
> On 9/5/06, Luis Simonet <luis.simonet at gmail.com> wrote:
>
> > I need to install Unity Connection on an existing Windows server 2003, I
> will used it for training. I have the license from Cisco. But when I try to
> run the installation it wants me to install 2003 from the DVD that came with
> Unity Connection. Is there a way to bypass it and use the version of 2003
> that is already install.
>
> Thanks
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


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

Replace the _ with an @ symbol
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From Nick.Kassel at Charles-Stanley.co.uk  Thu Sep  7 09:46:36 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 7 Sep 2006 14:46:36 +0100
Subject: [cisco-voip] IPT Paging Issues
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC014519AD@CSEXCH01.charles-stanley.co.uk>

We currently have Berbee Informacast for paging through Call Manager and
our IP phones and are having trouble with sending broadcasts to remote
sites, the multicasting seems to work intermittently, we may receive
broadcasts fine one time but others there may be up to a 20 second
delay. We are using 1760 routers with 3560 switches, our WAN is MPLS
using GRE tunnels. 

The IOS we are testing with is c1700-sv8y7-mz.123-4.t and
c1700-sv8y7-mz.123-11.T3 the latter being worse with hardly any
broadcasts coming through at all.

 

The following is the config on our branch routers, can any one see If
I'm missing something or no of a better IOS version to use.

 

 

!

ip cef

ip multicast-routing

!

interface Tunnel0

 bandwidth 2000

 ip unnumbered Loopback0

 ip pim sparse-mode

 ip ospf network point-to-point

 ip ospf cost 25

 ip ospf mtu-ignore

 keepalive 10 2

 tunnel source XXXXXXXXXXX

 tunnel destination XXXXXXXXXXX

!

interface Tunnel1

 bandwidth 2000

 ip unnumbered Loopback1

 ip ospf network point-to-point

 ip ospf mtu-ignore

 keepalive 10 2

 tunnel source XXXXXXXXXXX

 tunnel destination XXXXXXXXXXX

!

interface Loopback0

 ip address XXXXXXXXXXXXXXXXXXX

!

interface Loopback1

 ip address XXXXXXXXXXXXXXXXXXX

!

interface FastEthernet0/0

 no ip address

 ip pim sparse-mode

 speed auto

 full-duplex

 no cdp enable

 hold-queue 100 out

!

interface FastEthernet0/0.10

 encapsulation dot1Q 1 native

 ip address XXXXXXXXXXXXXX

 ip pim sparse-mode

 no cdp enable

!

interface FastEthernet0/0.20

 encapsulation dot1Q 2

 ip address XXXXXXXXXXXXXX

 ip pim sparse-mode

 ip igmp static-group 239.0.1.2

 no cdp enable

 h323-gateway voip bind srcaddr XXXXXXXX

!

interface Serial1/0

 bandwidth 2000

 no ip address

 encapsulation frame-relay IETF

 ip ospf network broadcast

 fair-queue

 frame-relay lmi-type ansi

!

interface Serial1/0.10 point-to-point

 bandwidth 2000

 ip unnumbered Loopback0

 no cdp enable

 frame-relay interface-dlci 101 IETF

!

interface Serial1/0.20 point-to-point

 bandwidth 2000

 ip unnumbered Loopback1

 no cdp enable

!

no ip http server

ip pim bidir-enable

ip pim rp-address 129.XXX.X.XX

ip mroute 239.0.1.2 255.255.255.255 Tunnel0

ip mroute 172.XX.0.0 255.255.240.0 Tunnel0

ip mroute 129.XXX.0.0 255.255.0.0 Tunnel0

!

!

!

 

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From tuskerkeg at gmail.com  Thu Sep  7 10:38:54 2006
From: tuskerkeg at gmail.com (tusker keg)
Date: Thu, 7 Sep 2006 17:38:54 +0300
Subject: [cisco-voip] Cabling Standards for IPT
Message-ID: <ef905eb80609070738x82e3e66t6481ce104c9b9492@mail.gmail.com>

Howdy'

Was wondering if there are any standards /BCP references out there for
a successful IPT deployment as far LAN cabling is concerned


Regards

./tusker

From db7td at gmx.de  Thu Sep  7 12:02:42 2006
From: db7td at gmx.de (Dietmar)
Date: Thu, 7 Sep 2006 18:02:42 +0200
Subject: [cisco-voip] 7941 volume too low
Message-ID: <200609071802.42970.db7td@gmx.de>

Hi all,

we recently started to use 7941 instead of 7940. The 7941 is very nice but 
there is one big problem: Compared to the 7940s, the microphone of the 
telephone receiver seems to be not so sensitive as the one of the 7940. Far 
end users are complaining about the low volume. For external calls, this can 
be corrected with the "output attenuation" parameter on the GW, but for 
IP-to-IP calls, I don't know where to adjust this. Firmware upgrade to the 
latest version did not solve this issue.

Is there anyone out there who knows how to solve this?


Thanks,
  Dietmar

From Scott.Voll at wesd.org  Thu Sep  7 12:59:30 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu, 7 Sep 2006 09:59:30 -0700
Subject: [cisco-voip] CM 5.x or 6.x
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C7DB4@Apollo.wesd.org>

Does anyone know what the interface to CM 5.x and 6.x uses?  Is it JAVA?
What version?  Is there a IPCCx version compatible with CM 5.x?

 

Thanks

 

Scott

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From milay at cisco.com  Thu Sep  7 13:21:53 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Thu, 7 Sep 2006 10:21:53 -0700
Subject: [cisco-voip] CM 5.x or 6.x
Message-ID: <413FABF5DCE22849AF86F442E472B78BB63523@xmb-sjc-219.amer.cisco.com>

JTAPI is still used to for communication with CCM in 5.0.x. You did not
mention if this was for IPCC Enterprise or Express. Regardless IPCC
enterprise 7.1.1 is supported with 5.0.2 & IPCC Express 4.5.1 & 4.5.2
are also supported with 5.0.x, see links below for detail.
 
http://www.cisco.com/application/pdf/en/us/guest/products/ps1844/c1609/c
cmigration_09186a008031a0a7.pdf
 
http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, September 07, 2006 12:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CM 5.x or 6.x



Does anyone know what the interface to CM 5.x and 6.x uses?  Is it JAVA?
What version?  Is there a IPCCx version compatible with CM 5.x?

 

Thanks

 

Scott

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From rratliff at cisco.com  Thu Sep  7 13:30:32 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 7 Sep 2006 13:30:32 -0400
Subject: [cisco-voip] CM 5.x or 6.x
In-Reply-To: <407055A92CECCB499C922A2D35FC19A6066C7DB4@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A6066C7DB4@Apollo.wesd.org>
Message-ID: <F3B56D5E-7F2D-4936-AF6C-EBA8211C738D@cisco.com>

I have no idea what CM 6.x uses (will use?).  5.x uses a web browser  
for the CCMAdmin and ssh for the console.

According to http://www.cisco.com/univercd/cc/td/doc/systems/unified/ 
unified1/relnotes/rnc50ipt.htm#wp1326480 IPCC Express 4.5(2) will  
work with CM 5.0(2) and later.

-Ryan

On Sep 7, 2006, at 12:59 PM, Voll, Scott wrote:

Does anyone know what the interface to CM 5.x and 6.x uses?  Is it  
JAVA?  What version?  Is there a IPCCx version compatible with CM 5.x?



Thanks



Scott

_______________________________________________
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 Sep  7 13:32:02 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu, 7 Sep 2006 10:32:02 -0700
Subject: [cisco-voip] CM 5.x or 6.x
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C7DCE@Apollo.wesd.org>

I'm talking about the Front end Java not the backend.  At least I think
of JTAPI as a backend.

 

Scott

 

________________________________

From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Thursday, September 07, 2006 10:22 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CM 5.x or 6.x

 

JTAPI is still used to for communication with CCM in 5.0.x. You did not
mention if this was for IPCC Enterprise or Express. Regardless IPCC
enterprise 7.1.1 is supported with 5.0.2 & IPCC Express 4.5.1 & 4.5.2
are also supported with 5.0.x, see links below for detail.

 

http://www.cisco.com/application/pdf/en/us/guest/products/ps1844/c1609/c
cmigration_09186a008031a0a7.pdf

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf

 

Mike

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, September 07, 2006 12:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CM 5.x or 6.x

Does anyone know what the interface to CM 5.x and 6.x uses?  Is it JAVA?
What version?  Is there a IPCCx version compatible with CM 5.x?

 

Thanks

 

Scott

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From Matt.Slaga at us.didata.com  Thu Sep  7 13:44:39 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 7 Sep 2006 13:44:39 -0400
Subject: [cisco-voip] CM 5.x or 6.x
Message-ID: <C1FE15183DA37645BC0633BC604E44F003135B35@USNAEXCH.na.didata.local>

CCM 6.0 will continue to be java based, however with 6.0 they will be
re-offering a windows 2003 platform.  Cisco is working on creating a
platform independent java version that can be used for linux or windows.

 

  

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, September 07, 2006 1:32 PM
To: Mike Lay (milay); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 5.x or 6.x

 

I'm talking about the Front end Java not the backend.  At least I think
of JTAPI as a backend.

 

Scott

 

________________________________

From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Thursday, September 07, 2006 10:22 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CM 5.x or 6.x

 

JTAPI is still used to for communication with CCM in 5.0.x. You did not
mention if this was for IPCC Enterprise or Express. Regardless IPCC
enterprise 7.1.1 is supported with 5.0.2 & IPCC Express 4.5.1 & 4.5.2
are also supported with 5.0.x, see links below for detail.

 

http://www.cisco.com/application/pdf/en/us/guest/products/ps1844/c1609/c
cmigration_09186a008031a0a7.pdf

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf

 

Mike

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, September 07, 2006 12:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CM 5.x or 6.x

Does anyone know what the interface to CM 5.x and 6.x uses?  Is it JAVA?
What version?  Is there a IPCCx version compatible with CM 5.x?

 

Thanks

 

Scott




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From wsisk at cisco.com  Thu Sep  7 14:13:22 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 07 Sep 2006 14:13:22 -0400
Subject: [cisco-voip] Cabling Standards for IPT
In-Reply-To: <ef905eb80609070738x82e3e66t6481ce104c9b9492@mail.gmail.com>
References: <ef905eb80609070738x82e3e66t6481ce104c9b9492@mail.gmail.com>
Message-ID: <45006142.1080107@cisco.com>

I believe cat5e is the minimum requirement.

/Wes

tusker keg wrote:
> Howdy'
> 
> Was wondering if there are any standards /BCP references out there for
> a successful IPT deployment as far LAN cabling is concerned
> 
> 
> Regards
> 
> ./tusker
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From wsisk at cisco.com  Thu Sep  7 14:16:49 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 07 Sep 2006 14:16:49 -0400
Subject: [cisco-voip] CM 5.x or 6.x
In-Reply-To: <407055A92CECCB499C922A2D35FC19A6066C7DB4@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A6066C7DB4@Apollo.wesd.org>
Message-ID: <45006211.30002@cisco.com>

Scott,

Can you elaborate a bit more on what you are looking for?

CM5.x uses a web browser for administration. You must run the browser on 
your PC, there is is no GUI access on the server such as XWindows, 
windows, or even a valid configuration intervace via CLI or ascii.

the console is ascii only and only used for troubleshooting.

For telephony control jtapi is still used, although updated with new 
features.

IPCCX version 4.5 is certified with CM5.0

/Wes

Voll, Scott wrote:
> Does anyone know what the interface to CM 5.x and 6.x uses?  Is it 
> JAVA?  What version?  Is there a IPCCx version compatible with CM 5.x?
> 
>  
> 
> Thanks
> 
>  
> 
> Scott
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> 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 Sep  7 14:25:49 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu, 7 Sep 2006 11:25:49 -0700
Subject: [cisco-voip] CM 5.x or 6.x
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C7DED@Apollo.wesd.org>

I'm referring to what Version of JAVA do you have to run on your PC
Browser in order to do administration. (IE> MS VM, Java 1.4 or 1.5,
etc.)

Scott

-----Original Message-----
From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Thursday, September 07, 2006 11:17 AM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CM 5.x or 6.x

Scott,

Can you elaborate a bit more on what you are looking for?

CM5.x uses a web browser for administration. You must run the browser on

your PC, there is is no GUI access on the server such as XWindows, 
windows, or even a valid configuration intervace via CLI or ascii.

the console is ascii only and only used for troubleshooting.

For telephony control jtapi is still used, although updated with new 
features.

IPCCX version 4.5 is certified with CM5.0

/Wes

Voll, Scott wrote:
> Does anyone know what the interface to CM 5.x and 6.x uses?  Is it 
> JAVA?  What version?  Is there a IPCCx version compatible with CM 5.x?
> 
>  
> 
> Thanks
> 
>  
> 
> Scott
> 
> 
>
------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From jason.aarons at us.didata.com  Thu Sep  7 17:03:24 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 7 Sep 2006 17:03:24 -0400
Subject: [cisco-voip] Cabling Standards for IPT
Message-ID: <C1FE15183DA37645BC0633BC604E44F003135FE7@USNAEXCH.na.didata.local>

I have a couple phones in warehouses running on Category3 at 10/Half, it
works, powered locally, it was too expensive to run a Cat5 run. Not a
good idea in my view, but hey it works.

The specs show Cat3 in Table A-1 Cisco IP Phone Specifications  
http://www.cisco.com/en/US/products/hw/phones/ps379/products_administrat
ion_guide_chapter09186a00800b607e.html#wp1003719



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Thursday, September 07, 2006 2:13 PM
To: tusker keg
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Cabling Standards for IPT

I believe cat5e is the minimum requirement.

/Wes

tusker keg wrote:
> Howdy'
> 
> Was wondering if there are any standards /BCP references out there for
> a successful IPT deployment as far LAN cabling is concerned
> 
> 
> Regards
> 
> ./tusker
> _______________________________________________
> 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  Thu Sep  7 17:32:40 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 7 Sep 2006 17:32:40 -0400
Subject: [cisco-voip] simple ldap query tools (command line preferred)
Message-ID: <000c01c6d2c5$258ebea0$47196883@cfs.uoguelph.ca>

Anyone have any simple LDAP query tools they use on Windows? I want to write a quick batch file that will allow admin staff to search for userIDs based on a controlled device, i.e. MAC address.

I can do what I need to do with DC Directory, but it doesn't meet our needs.

Lelio

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
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From rduric at uoguelph.ca  Thu Sep  7 21:18:36 2006
From: rduric at uoguelph.ca (Rasim Duric)
Date: Thu, 7 Sep 2006 21:18:36 -0400
Subject: [cisco-voip] JTAPI subsystem in partial_service
Message-ID: <BD89624EE83BD349BD31A08B453DE244BB142A@uogexb201.cfs.uoguelph.ca>

Hi all,

JTAPI subsystem is flapping between in_service and partial_service on
one of our IPCCX applications. I've stopped and restarted the CRS engine
but it would run for a while and back in partial_service again. I can
see in the MIVR trace file some CTI ports going OOS but they always stay
registered with CCM and are associated with the proper JTAPI user. The
calls to these OOS CTI ports would ring the hunt pilot since they would
hit it as the CTI Route Point forwarding target (FW on Failure Ext/Int,
FW Busy Int/Ext, FW No Answer Int/Ext). No FW targets defined for the
CTI ports though. 

What could be causing this issue? My impression is that this problem has
started since we upgraded CCM 3.3 to 4.1(3)sr3a and IPCCX 3.1 to 3.5(3)
but not sure.

Is anybody experiencing such a behaviour and would an upgrade to CCM
4.1(3)sr3c fix the problem as TAC suggested? TIA.



Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca



From milay at cisco.com  Thu Sep  7 21:26:26 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Thu, 7 Sep 2006 18:26:26 -0700
Subject: [cisco-voip] JTAPI subsystem in partial_service
Message-ID: <413FABF5DCE22849AF86F442E472B78BBC68BD@xmb-sjc-219.amer.cisco.com>

Just a SWAG but did you update the JTAPIClient after the upgrade? Also
forward on failure is not the same as forward on not registered,
upcoming feature.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rasim Duric
Sent: Thursday, September 07, 2006 8:19 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] JTAPI subsystem in partial_service

Hi all,

JTAPI subsystem is flapping between in_service and partial_service on
one of our IPCCX applications. I've stopped and restarted the CRS engine
but it would run for a while and back in partial_service again. I can
see in the MIVR trace file some CTI ports going OOS but they always stay
registered with CCM and are associated with the proper JTAPI user. The
calls to these OOS CTI ports would ring the hunt pilot since they would
hit it as the CTI Route Point forwarding target (FW on Failure Ext/Int,
FW Busy Int/Ext, FW No Answer Int/Ext). No FW targets defined for the
CTI ports though. 

What could be causing this issue? My impression is that this problem has
started since we upgraded CCM 3.3 to 4.1(3)sr3a and IPCCX 3.1 to 3.5(3)
but not sure.

Is anybody experiencing such a behaviour and would an upgrade to CCM
4.1(3)sr3c fix the problem as TAC suggested? TIA.



Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca


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


From rduric at uoguelph.ca  Thu Sep  7 21:30:18 2006
From: rduric at uoguelph.ca (Rasim Duric)
Date: Thu, 7 Sep 2006 21:30:18 -0400
Subject: [cisco-voip] JTAPI subsystem in partial_service
In-Reply-To: <413FABF5DCE22849AF86F442E472B78BBC68BD@xmb-sjc-219.amer.cisco.com>
Message-ID: <BD89624EE83BD349BD31A08B453DE244BB142B@uogexb201.cfs.uoguelph.ca>

Yes, I did update the JTAPI client after the upgrade.

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Thursday, September 07, 2006 9:26 PM
To: Rasim Duric; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] JTAPI subsystem in partial_service

Just a SWAG but did you update the JTAPIClient after the upgrade? Also
forward on failure is not the same as forward on not registered,
upcoming feature.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rasim Duric
Sent: Thursday, September 07, 2006 8:19 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] JTAPI subsystem in partial_service

Hi all,

JTAPI subsystem is flapping between in_service and partial_service on
one of our IPCCX applications. I've stopped and restarted the CRS engine
but it would run for a while and back in partial_service again. I can
see in the MIVR trace file some CTI ports going OOS but they always stay
registered with CCM and are associated with the proper JTAPI user. The
calls to these OOS CTI ports would ring the hunt pilot since they would
hit it as the CTI Route Point forwarding target (FW on Failure Ext/Int,
FW Busy Int/Ext, FW No Answer Int/Ext). No FW targets defined for the
CTI ports though. 

What could be causing this issue? My impression is that this problem has
started since we upgraded CCM 3.3 to 4.1(3)sr3a and IPCCX 3.1 to 3.5(3)
but not sure.

Is anybody experiencing such a behaviour and would an upgrade to CCM
4.1(3)sr3c fix the problem as TAC suggested? TIA.



Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca


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


From jgrace at digitel.net  Thu Sep  7 23:46:57 2006
From: jgrace at digitel.net (James Grace)
Date: Thu, 7 Sep 2006 23:46:57 -0400
Subject: [cisco-voip] background logos on 7941 -61
In-Reply-To: <44FF0E9C.5090507@cisco.com>
Message-ID: <200609072346859.SM01472@jgm>

Hey Wes

I use the doc to create and place the background on my CM ver 4.1.3.  this
my lab.but when I tried to add that same logo to my production CM ver 4.2.1
it does not show up.  I created the dir structure Desktops\320x95x4\  and
have the correct files in their.  But no show  when I goto the phone
settings

 

  _____  

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, September 06, 2006 2:08 PM
To: Stu Packett
Cc: Craig M Staffin; Kevin Thorngren; cisco-voip-bounces at puck.nether.net;
James Grace; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61

 

For all ambitious enough, this is possible through the phone SDK and pushing
instructions to each phone. I did a proof of concept for a group of phones.
my bat script looks like this:

wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute

wget is pretty straight forward, as are user and password.   the user and
password are for a user configured in callmanager's directory that is
assocaited with the phone.  If you are using Berbee or some of the other
phone control apps, they repoint the Authentication URL enterprise parameter
so the phone always gets a successful auth regardless of the user/pass.
the --post-file is a text file that contains the XML encoding for the
desired instruction, CiscoIpPhoneExecute in all cases looks something like
this:
kp1.txt sends the instruction to press keypad button 1:
XML=%3CCiscoIPPhoneExecute%3E%3CExecuteItem+Priority%3D%220%22+URL%3D%22Key:
KeyPad1%22%2F%3E%3C%2FCiscoIPPhoneExecute%3E

it's heavy on file i/o but it's flexible enough for me to recreate the
*interesting* phone use scenarios i need to recreate and test en-mass.

/Wes


Stu Packett wrote: 

Craig,

I'm glad you brought that up as I'm about to deploy several 7971 phones and
would like to know if there is a bulk way of pushing the logos out.  I
didn't seem to mind when I did it for 10 or so phones, but I will be getting
several more in the next few weeks.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Craig M Staffin
Sent: Wednesday, September 06, 2006 9:02 AM
To: Kevin Thorngren
Cc: cisco-voip-bounces at puck.nether.net; James Grace;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61


Kevin, 

I am just curious this process is very manual.  In other words you need to
set it for each phone.  Pretty labor intensive for 20,000 phones.  Is there
a way in CCM that you can force the phones to get lets say a company logo as
the background? 

Craig






Kevin Thorngren  <mailto:kthorngr at cisco.com> <kthorngr at cisco.com> 
Sent by: cisco-voip-bounces at puck.nether.net 

09/05/2006 07:25 PM 


To

"James Grace"  <mailto:jgrace at digitel.net> <jgrace at digitel.net> 


cc

cisco-voip at puck.nether.net 


Subject

Re: [cisco-voip] background logos on 7941 -61

 


 

 




How about this from the 7941/61 docs:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ 
ipp7961/admin/4_2/7961cus.htm#wp1030672

Kevin
On Sep 5, 2006, at 6:32 PM, James Grace wrote:

> Do any one have doc on how to put background logos on the 7941 61  
> ipphones.  Ive worked out the idle screensaver but, ive seen some  
> phones out there with the background changed
>  
> James D. Grace
> CCNP CCNA MCSE MCSA
> Digitel Corporation
> System Engineer / Professional Svc.
> MSN messenger ID:  jgrace29 at hotmail.com
> Mac IChat ID:  jgrace29 at mac.com
>  
>  
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




 





  _____  



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
  
---
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From SPackett at fenwick.com  Fri Sep  8 00:01:51 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu, 7 Sep 2006 21:01:51 -0700
Subject: [cisco-voip] background logos on 7941 -61
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD75AD@SVCEXCH01.firm.fenwick.llp>

James:
Maybe you had a typo, buy my directory is under C:\Program
Files\Cisco\TFTPPath\Desktops\320x212x12.  Also, I needed to restart the
TFTP service when I made the changes.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, September 07, 2006 8:47 PM
To: 'Wes Sisk'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61



Hey Wes

I use the doc to create and place the background on my CM ver 4.1.3.
this my lab...but when I tried to add that same logo to my production CM
ver 4.2.1  it does not show up.  I created the dir structure
Desktops\320x95x4\  and have the correct files in their.  But no show
when I goto the phone settings

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, September 06, 2006 2:08 PM
To: Stu Packett
Cc: Craig M Staffin; Kevin Thorngren;
cisco-voip-bounces at puck.nether.net; James Grace;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61

 

For all ambitious enough, this is possible through the phone SDK and
pushing instructions to each phone. I did a proof of concept for a group
of phones.  my bat script looks like this:

wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute

wget is pretty straight forward, as are user and password.   the user
and password are for a user configured in callmanager's directory that
is assocaited with the phone.  If you are using Berbee or some of the
other phone control apps, they repoint the Authentication URL enterprise
parameter so the phone always gets a successful auth regardless of the
user/pass.
the --post-file is a text file that contains the XML encoding for the
desired instruction, CiscoIpPhoneExecute in all cases looks something
like this:
kp1.txt sends the instruction to press keypad button 1:
XML=%3CCiscoIPPhoneExecute%3E%3CExecuteItem+Priority%3D%220%22+URL%3D%22
Key:KeyPad1%22%2F%3E%3C%2FCiscoIPPhoneExecute%3E

it's heavy on file i/o but it's flexible enough for me to recreate the
*interesting* phone use scenarios i need to recreate and test en-mass.

/Wes


Stu Packett wrote: 

Craig,

I'm glad you brought that up as I'm about to deploy several 7971 phones
and would like to know if there is a bulk way of pushing the logos out.
I didn't seem to mind when I did it for 10 or so phones, but I will be
getting several more in the next few weeks.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Craig M Staffin
Sent: Wednesday, September 06, 2006 9:02 AM
To: Kevin Thorngren
Cc: cisco-voip-bounces at puck.nether.net; James Grace;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61


Kevin, 

I am just curious this process is very manual.  In other words you need
to set it for each phone.  Pretty labor intensive for 20,000 phones.  Is
there a way in CCM that you can force the phones to get lets say a
company logo as the background? 

Craig





Kevin Thorngren <kthorngr at cisco.com> <mailto:kthorngr at cisco.com>  
Sent by: cisco-voip-bounces at puck.nether.net 

09/05/2006 07:25 PM 

To

"James Grace" <jgrace at digitel.net> <mailto:jgrace at digitel.net>  

cc

cisco-voip at puck.nether.net 

Subject

Re: [cisco-voip] background logos on 7941 -61

 

 

 




How about this from the 7941/61 docs:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ 
ipp7961/admin/4_2/7961cus.htm#wp1030672

Kevin
On Sep 5, 2006, at 6:32 PM, James Grace wrote:

> Do any one have doc on how to put background logos on the 7941 61  
> ipphones.  Ive worked out the idle screensaver but, ive seen some  
> phones out there with the background changed
>  
> James D. Grace
> CCNP CCNA MCSE MCSA
> Digitel Corporation
> System Engineer / Professional Svc.
> MSN messenger ID:  jgrace29 at hotmail.com
> Mac IChat ID:  jgrace29 at mac.com
>  
>  
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




 



________________________________



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
  
---
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From manoj.kalpage at gmail.com  Fri Sep  8 07:18:14 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Fri, 8 Sep 2006 20:18:14 +0900
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall
Message-ID: <bbe09c1d0609080418j42130bamfb7fc50832cc5ce6@mail.gmail.com>

Hi All,
Does any one has configured PIX firewall to connect internet IP phones to
Call Manager. I have configure firewall to open all the port which CCM need
but still no luck. Bellow is the config of my PIX. Am i missing anything?

Here is the link I refered to open the TCP and UDP Ports

http://www.cisco.com/application/pdf/en/us/guest/products/ps5820/c1693/ccmigration_09186a0080536eae.pdf

Thank you in advance.
Manoj

:
PIX Version 6.3(5)
interface ethernet0 auto
interface ethernet1 auto
nameif ethernet0 outside security0
nameif ethernet1 inside security100
enable password u2zabJUOK.TTL3K1 encrypted
passwd 1P5CrRl.dL8Oe4k2 encrypted
hostname PBXLPIX01
domain-name pbxl.jp
clock timezone JST 9
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 pptp 1723
fixup protocol rsh 514
fixup protocol rtsp 554
fixup protocol sip 5060
fixup protocol sip udp 5060
fixup protocol skinny 2000
fixup protocol smtp 25
fixup protocol snmp 161
fixup protocol sqlnet 1521
fixup protocol tftp 69
names
object-group service outbound-tcp tcp
  port-object eq www
  port-object eq https
  port-object eq smtp
  port-object eq ftp
  port-object eq pop3
  port-object eq imap4
  port-object eq domain
  port-object eq 123
  port-object eq ssh
  port-object eq citrix-ica
object-group service outbound-udp udp
  port-object eq domain
  port-object eq ntp
object-group service mail-inbound tcp
  port-object eq www
  port-object eq https
  port-object eq smtp
object-group service VoIP-udp udp
  port-object range 16384 32768
  port-object eq tftp
object-group service VoIP-tcp tcp
  port-object eq 3804
  port-object eq 2443
  port-object eq 2000
  port-object eq www
  port-object eq 69
  port-object eq https
access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group VoIP-tcp
access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group VoIP-udp
access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group
outbound-tcp
access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group
outbound-udp
access-list 101 permit tcp any host 210.81.12.195 object-group mail-inbound
access-list 101 permit tcp any host 210.81.12.196 object-group VoIP-tcp
access-list 101 permit udp any host 210.81.12.196 object-group VoIP-udp
access-list 101 permit tcp any host 210.81.12.197 object-group VoIP-tcp
access-list 101 permit udp any host 210.81.12.197 object-group VoIP-udp
pager lines 24
logging on
logging trap informational
logging host inside 172.16.0.26
logging host inside 172.16.0.12
icmp permit any unreachable outside
icmp permit any outside
mtu outside 1500
mtu inside 1500
ip address outside xxx.xxx.xxx.xxx 255.255.255.240
ip address inside 172.16.0.2 255.255.0.0
ip audit info action alarm
ip audit attack action alarm
ip local pool pbxlpool 10.1.0.100-10.1.0.200
pdm locationxxx.xxx.xxx.xxx 255.255.255.255 outside
pdm history enable
arp timeout 14400
global (outside) 1 interface
nat (inside) 0 access-list VPNREMOTE
nat (inside) 1 172.16.0.0 255.255.0.0 0 0
static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
255.255.255.255 0 1000
static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
255.255.255.255 0 1000
static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
255.255.255.255 0 1000
access-group 101 in interface outside
access-group 102 in interface inside
route outside 0.0.0.0 0.0.0.0 210.81.12.193 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 sip-disconnect 0:02:00 sip-invite 0:03:00
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

aaa authentication ssh console LOCAL

http 172.16.0.12 255.255.255.255 inside
snmp-server host inside 172.16.0.12
snmp-server location pbxl-pix-datacentre

snmp-server community pbxl
snmp-server enable traps
floodguard enable

telnet 172.16.0.0 255.255.0.0 inside
telnet 192.168.0.0 255.255.255.0 inside
telnet timeout 60
ssh 210.101.94.211 255.255.255.255 outside
ssh 0.0.0.0 0.0.0.0 outside
ssh 172.16.0.12 255.255.255.255 inside
ssh 172.16.0.0 255.255.0.0 inside
ssh 192.168.1.0 255.255.255.0 inside
ssh timeout 60
console timeout 0
PBXLPIX01(config)#
PBXLPIX01(config)#
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From jgrace at digitel.net  Fri Sep  8 09:26:36 2006
From: jgrace at digitel.net (James Grace)
Date: Fri, 8 Sep 2006 09:26:36 -0400
Subject: [cisco-voip] background logos on 7941 -61
In-Reply-To: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD75AD@SVCEXCH01.firm.fenwick.llp>
Message-ID: <200609080925546.SM01472@jgm>

Thank you.

I checked for typos and all is the same.  Now question.  What do the dir
names "320x212x12"  because my is 320x96x4 ..  I was thinking that this is
just a description of the logo

 

\320x212x12

 

  _____  

From: Stu Packett [mailto:SPackett at fenwick.com] 
Sent: Friday, September 08, 2006 12:02 AM
To: James Grace; Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] background logos on 7941 -61

 

James:
Maybe you had a typo, buy my directory is under C:\Program
Files\Cisco\TFTPPath\Desktops\320x212x12.  Also, I needed to restart the
TFTP service when I made the changes.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, September 07, 2006 8:47 PM
To: 'Wes Sisk'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61

Hey Wes

I use the doc to create and place the background on my CM ver 4.1.3.  this
my lab.but when I tried to add that same logo to my production CM ver 4.2.1
it does not show up.  I created the dir structure Desktops\320x95x4\  and
have the correct files in their.  But no show  when I goto the phone
settings

 

  _____  

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, September 06, 2006 2:08 PM
To: Stu Packett
Cc: Craig M Staffin; Kevin Thorngren; cisco-voip-bounces at puck.nether.net;
James Grace; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61

 

For all ambitious enough, this is possible through the phone SDK and pushing
instructions to each phone. I did a proof of concept for a group of phones.
my bat script looks like this:

wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute

wget is pretty straight forward, as are user and password.   the user and
password are for a user configured in callmanager's directory that is
assocaited with the phone.  If you are using Berbee or some of the other
phone control apps, they repoint the Authentication URL enterprise parameter
so the phone always gets a successful auth regardless of the user/pass.
the --post-file is a text file that contains the XML encoding for the
desired instruction, CiscoIpPhoneExecute in all cases looks something like
this:
kp1.txt sends the instruction to press keypad button 1:
XML=%3CCiscoIPPhoneExecute%3E%3CExecuteItem+Priority%3D%220%22+URL%3D%22Key:
KeyPad1%22%2F%3E%3C%2FCiscoIPPhoneExecute%3E

it's heavy on file i/o but it's flexible enough for me to recreate the
*interesting* phone use scenarios i need to recreate and test en-mass.

/Wes


Stu Packett wrote: 

Craig,

I'm glad you brought that up as I'm about to deploy several 7971 phones and
would like to know if there is a bulk way of pushing the logos out.  I
didn't seem to mind when I did it for 10 or so phones, but I will be getting
several more in the next few weeks.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Craig M Staffin
Sent: Wednesday, September 06, 2006 9:02 AM
To: Kevin Thorngren
Cc: cisco-voip-bounces at puck.nether.net; James Grace;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61


Kevin, 

I am just curious this process is very manual.  In other words you need to
set it for each phone.  Pretty labor intensive for 20,000 phones.  Is there
a way in CCM that you can force the phones to get lets say a company logo as
the background? 

Craig





Kevin Thorngren  <mailto:kthorngr at cisco.com> <kthorngr at cisco.com> 
Sent by: cisco-voip-bounces at puck.nether.net 

09/05/2006 07:25 PM 


To

"James Grace"  <mailto:jgrace at digitel.net> <jgrace at digitel.net> 


cc

cisco-voip at puck.nether.net 


Subject

Re: [cisco-voip] background logos on 7941 -61

 


 

 




How about this from the 7941/61 docs:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ 
ipp7961/admin/4_2/7961cus.htm#wp1030672

Kevin
On Sep 5, 2006, at 6:32 PM, James Grace wrote:

> Do any one have doc on how to put background logos on the 7941 61  
> ipphones.  Ive worked out the idle screensaver but, ive seen some  
> phones out there with the background changed
>  
> James D. Grace
> CCNP CCNA MCSE MCSA
> Digitel Corporation
> System Engineer / Professional Svc.
> MSN messenger ID:  jgrace29 at hotmail.com
> Mac IChat ID:  jgrace29 at mac.com
>  
>  
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



 
 
 





  _____  



 
 
 
_______________________________________________
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 Sep  8 12:01:46 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 8 Sep 2006 09:01:46 -0700
Subject: [cisco-voip] OT Express 500
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C7F47@Apollo.wesd.org>

I know there has been a lot of talk about the express 500 series
switches and how much they are hated.  I wanted to see for me self and
so far I'm not impressed.

 

Can the E500's do dot1q trunking of more then a voice and a access vlan?
If so, can someone share with me the way to set one up as a Dot1q trunk
to a 3560.  

 

Thanks

 

Scott

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From SPackett at fenwick.com  Fri Sep  8 12:37:32 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Fri, 8 Sep 2006 09:37:32 -0700
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD75B4@SVCEXCH01.firm.fenwick.llp>

Manoj:
Is your PIX giving out DHCP addresses?  On my PIX 501, I have it setup
as a DHCP server and these are my DHCP commands:
 
dhcpd address xxx.xxx.xxx.xxx
dhcpd dns xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
dhcpd wins xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
dhcpd lease 36000
dhcpd ping_timeout 750
dhcpd domain internaldomain.com
dhcpd option 150 ip xxx.xxx.xxx.xxx <--TFTP address
dhcpd enable inside

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Manoj Kalpage
Sent: Friday, September 08, 2006 4:18 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall


Hi All,
Does any one has configured PIX firewall to connect internet IP phones
to Call Manager. I have configure firewall to open all the port which
CCM need but still no luck. Bellow is the config of my PIX. Am i missing
anything? 

Here is the link I refered to open the TCP and UDP Ports

http://www.cisco.com/application/pdf/en/us/guest/products/ps5820/c1693/c
cmigration_09186a0080536eae.pdf

Thank you in advance.
Manoj

:
PIX Version 6.3(5)
interface ethernet0 auto
interface ethernet1 auto 
nameif ethernet0 outside security0
nameif ethernet1 inside security100
enable password u2zabJUOK.TTL3K1 encrypted
passwd 1P5CrRl.dL8Oe4k2 encrypted
hostname PBXLPIX01
domain-name pbxl.jp
clock timezone JST 9
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 pptp 1723 
fixup protocol rsh 514
fixup protocol rtsp 554
fixup protocol sip 5060
fixup protocol sip udp 5060
fixup protocol skinny 2000
fixup protocol smtp 25
fixup protocol snmp 161
fixup protocol sqlnet 1521 
fixup protocol tftp 69
names
object-group service outbound-tcp tcp
  port-object eq www
  port-object eq https
  port-object eq smtp
  port-object eq ftp
  port-object eq pop3
  port-object eq imap4 
  port-object eq domain
  port-object eq 123
  port-object eq ssh
  port-object eq citrix-ica
object-group service outbound-udp udp
  port-object eq domain
  port-object eq ntp
object-group service mail-inbound tcp 
  port-object eq www
  port-object eq https
  port-object eq smtp
object-group service VoIP-udp udp
  port-object range 16384 32768
  port-object eq tftp
object-group service VoIP-tcp tcp
  port-object eq 3804 
  port-object eq 2443
  port-object eq 2000
  port-object eq www
  port-object eq 69
  port-object eq https
access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group
VoIP-tcp
access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group
VoIP-udp
access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group
outbound-tcp
access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group
outbound-udp 
access-list 101 permit tcp any host 210.81.12.195 object-group
mail-inbound
access-list 101 permit tcp any host 210.81.12.196 object-group VoIP-tcp 
access-list 101 permit udp any host 210.81.12.196 object-group VoIP-udp
access-list 101 permit tcp any host 210.81.12.197 object-group VoIP-tcp
access-list 101 permit udp any host 210.81.12.197 object-group VoIP-udp
pager lines 24
logging on
logging trap informational
logging host inside 172.16.0.26
logging host inside 172.16.0.12
icmp permit any unreachable outside
icmp permit any outside
mtu outside 1500
mtu inside 1500
ip address outside xxx.xxx.xxx.xxx 255.255.255.240
ip address inside 172.16.0.2 255.255.0.0
ip audit info action alarm
ip audit attack action alarm 
ip local pool pbxlpool 10.1.0.100-10.1.0.200
pdm locationxxx.xxx.xxx.xxx 255.255.255.255 outside
pdm history enable
arp timeout 14400
global (outside) 1 interface
nat (inside) 0 access-list VPNREMOTE 
nat (inside) 1 172.16.0.0 255.255.0.0 0 0
static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
255.255.255.255 0 1000
static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
255.255.255.255 0 1000
static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
255.255.255.255 0 1000
access-group 101 in interface outside
access-group 102 in interface inside
route outside 0.0.0.0 0.0.0.0 210.81.12.193 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 sip-disconnect 0:02:00 sip-invite 0:03:00 
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

aaa authentication ssh console LOCAL

http 172.16.0.12 255.255.255.255 inside
snmp-server host inside 172.16.0.12
snmp-server location pbxl-pix-datacentre

snmp-server community pbxl
snmp-server enable traps
floodguard enable

telnet 172.16.0.0 255.255.0.0 inside
telnet 192.168.0.0 255.255.255.0 inside
telnet timeout 60 
ssh 210.101.94.211 255.255.255.255 outside
ssh 0.0.0.0 0.0.0.0 outside
ssh 172.16.0.12 255.255.255.255 inside
ssh 172.16.0.0 255.255.0.0 inside
ssh 192.168.1.0 255.255.255.0 inside
ssh timeout 60
console timeout 0
PBXLPIX01(config)#
PBXLPIX01(config)#



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From cbaird at ipcelerate.com  Fri Sep  8 13:11:27 2006
From: cbaird at ipcelerate.com (Chris Baird)
Date: Fri, 8 Sep 2006 12:11:27 -0500
Subject: [cisco-voip]  Grouping Service URLs
In-Reply-To: <BD89624EE83BD349BD31A08B453DE244BB142A@uogexb201.cfs.uoguelph.ca>
Message-ID: <027201c6d369$d7ccae20$026310ac@VAIOBAIRD>

Hello,

Is there a way I can group several IP phone service URLs?  For example, I
have a group of users (Group A) that should have ext mobility, fast dials,
etc but another group (Group B) should only have ext mobility and stock
quotes.  Can I create a group to subscribe to group A and a separate group
for B?  

Thanks,
Chris 




From pmowry at getgds.com  Fri Sep  8 14:41:05 2006
From: pmowry at getgds.com (Patrick Mowry)
Date: Fri, 8 Sep 2006 13:41:05 -0500
Subject: [cisco-voip] Is there any such thing as a null destination for an
	IOS dial-peer?
In-Reply-To: <mailman.5.1157731206.24968.cisco-voip@puck.nether.net>
Message-ID: <4C792A099259634C83F97067671B37FE0263B586@GDSEXC.gds.pri>

Hello,

  We have a site using CallManager Express where some users are
restricted from calling certain area codes and directory assistance, but
not others.  So an example dial-peer I have to make it work is:

dial-peer voice 31 pots
 corlist outgoing Block-NC
 description ** NC Blocked dial-peer **
 destination-pattern 91252.......
 progress_ind setup enable 3
 progress_ind progress enable 8
 port 0/0/0:1
 forward-digits 0
 prefix ###

So a dn who's class of restriction contains Block-NC, matches this
dial-peer, we send only ### to the carrier so the users hears a reorder
tone (fast busy) instead of the call going through.  And this is what I
want to happen, except I would prefer not to take up a channel on the T1
circuit just to return a fast busy tone.  I thought of routing the call
to a CUE auto-attendant number that would only say you are not allowed
to call this area code then hang up.  But they did not want to use up a
Unity session for this either.

Since I only want to block these calls for some users and not all, does
anyone have a suggestion on a better way to do this?

Thanks,

-Patrick


From rratliff at cisco.com  Fri Sep  8 14:45:22 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 8 Sep 2006 14:45:22 -0400
Subject: [cisco-voip] Is there any such thing as a null destination for
	an IOS dial-peer?
In-Reply-To: <4C792A099259634C83F97067671B37FE0263B586@GDSEXC.gds.pri>
References: <4C792A099259634C83F97067671B37FE0263B586@GDSEXC.gds.pri>
Message-ID: <8C3E4304-CEC0-4888-A100-E05B303DD05C@cisco.com>

Rather than having a special COR to block calls why not just have COR  
setup so they don't match any dial-peers?  This will give them a fast  
busy and not tie up a channel.

-Ryan

On Sep 8, 2006, at 2:41 PM, Patrick Mowry wrote:

Hello,

   We have a site using CallManager Express where some users are
restricted from calling certain area codes and directory assistance, but
not others.  So an example dial-peer I have to make it work is:

dial-peer voice 31 pots
  corlist outgoing Block-NC
  description ** NC Blocked dial-peer **
  destination-pattern 91252.......
  progress_ind setup enable 3
  progress_ind progress enable 8
  port 0/0/0:1
  forward-digits 0
  prefix ###

So a dn who's class of restriction contains Block-NC, matches this
dial-peer, we send only ### to the carrier so the users hears a reorder
tone (fast busy) instead of the call going through.  And this is what I
want to happen, except I would prefer not to take up a channel on the T1
circuit just to return a fast busy tone.  I thought of routing the call
to a CUE auto-attendant number that would only say you are not allowed
to call this area code then hang up.  But they did not want to use up a
Unity session for this either.

Since I only want to block these calls for some users and not all, does
anyone have a suggestion on a better way to do this?

Thanks,

-Patrick

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


From jgrace at digitel.net  Fri Sep  8 15:55:13 2006
From: jgrace at digitel.net (James Grace)
Date: Fri, 8 Sep 2006 15:55:13 -0400
Subject: [cisco-voip] background logos on 7941 -61
In-Reply-To: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD75AD@SVCEXCH01.firm.fenwick.llp>
Message-ID: <200609081554187.SM01472@jgm>

I just change my dir to  \320x212x12  and the logo comes up on my ip
communicator.  But not on any of the 7941 61 

 

  _____  

From: Stu Packett [mailto:SPackett at fenwick.com] 
Sent: Friday, September 08, 2006 12:02 AM
To: James Grace; Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] background logos on 7941 -61

 

James:
Maybe you had a typo, buy my directory is under C:\Program
Files\Cisco\TFTPPath\Desktops\320x212x12.  Also, I needed to restart the
TFTP service when I made the changes.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, September 07, 2006 8:47 PM
To: 'Wes Sisk'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61

Hey Wes

I use the doc to create and place the background on my CM ver 4.1.3.  this
my lab.but when I tried to add that same logo to my production CM ver 4.2.1
it does not show up.  I created the dir structure Desktops\320x95x4\  and
have the correct files in their.  But no show  when I goto the phone
settings

 

  _____  

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, September 06, 2006 2:08 PM
To: Stu Packett
Cc: Craig M Staffin; Kevin Thorngren; cisco-voip-bounces at puck.nether.net;
James Grace; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61

 

For all ambitious enough, this is possible through the phone SDK and pushing
instructions to each phone. I did a proof of concept for a group of phones.
my bat script looks like this:

wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute

wget is pretty straight forward, as are user and password.   the user and
password are for a user configured in callmanager's directory that is
assocaited with the phone.  If you are using Berbee or some of the other
phone control apps, they repoint the Authentication URL enterprise parameter
so the phone always gets a successful auth regardless of the user/pass.
the --post-file is a text file that contains the XML encoding for the
desired instruction, CiscoIpPhoneExecute in all cases looks something like
this:
kp1.txt sends the instruction to press keypad button 1:
XML=%3CCiscoIPPhoneExecute%3E%3CExecuteItem+Priority%3D%220%22+URL%3D%22Key:
KeyPad1%22%2F%3E%3C%2FCiscoIPPhoneExecute%3E

it's heavy on file i/o but it's flexible enough for me to recreate the
*interesting* phone use scenarios i need to recreate and test en-mass.

/Wes


Stu Packett wrote: 

Craig,

I'm glad you brought that up as I'm about to deploy several 7971 phones and
would like to know if there is a bulk way of pushing the logos out.  I
didn't seem to mind when I did it for 10 or so phones, but I will be getting
several more in the next few weeks.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Craig M Staffin
Sent: Wednesday, September 06, 2006 9:02 AM
To: Kevin Thorngren
Cc: cisco-voip-bounces at puck.nether.net; James Grace;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61


Kevin, 

I am just curious this process is very manual.  In other words you need to
set it for each phone.  Pretty labor intensive for 20,000 phones.  Is there
a way in CCM that you can force the phones to get lets say a company logo as
the background? 

Craig





Kevin Thorngren  <mailto:kthorngr at cisco.com> <kthorngr at cisco.com> 
Sent by: cisco-voip-bounces at puck.nether.net 

09/05/2006 07:25 PM 


To

"James Grace"  <mailto:jgrace at digitel.net> <jgrace at digitel.net> 


cc

cisco-voip at puck.nether.net 


Subject

Re: [cisco-voip] background logos on 7941 -61

 


 

 




How about this from the 7941/61 docs:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ 
ipp7961/admin/4_2/7961cus.htm#wp1030672

Kevin
On Sep 5, 2006, at 6:32 PM, James Grace wrote:

> Do any one have doc on how to put background logos on the 7941 61  
> ipphones.  Ive worked out the idle screensaver but, ive seen some  
> phones out there with the background changed
>  
> James D. Grace
> CCNP CCNA MCSE MCSA
> Digitel Corporation
> System Engineer / Professional Svc.
> MSN messenger ID:  jgrace29 at hotmail.com
> Mac IChat ID:  jgrace29 at mac.com
>  
>  
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



 
 
 





  _____  



 
 
 
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From milay at cisco.com  Fri Sep  8 16:20:15 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Fri, 8 Sep 2006 13:20:15 -0700
Subject: [cisco-voip] background logos on 7941 -61
Message-ID: <413FABF5DCE22849AF86F442E472B78BBC6BA4@xmb-sjc-219.amer.cisco.com>

The 7941/61 phones support images with the resolution of 320x196x4 so if
you are trying to use a 320x212x12 image on a 7941/61, it will not work.
Also IPC emulates a 7970 thus it works with the 320x212x12 image. 
 
The link that Wes posted gives the correct instructions for the 7941/61
phones. Make sure you make the resolution is correct, the xml file is
correct, the directory is correct & restart TFTP. Other wise you will
need to look at detailed TFTP &/or sniffer from the phone to see how far
the phone is getting when loading the image.:
 
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i
pp7961/admin/4_2/7961cus.htm#wp1030672
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Friday, September 08, 2006 2:55 PM
To: 'Stu Packett'; Wes Sisk (wsisk)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61



I just change my dir to  \320x212x12  and the logo comes up on my ip
communicator.  But not on any of the 7941 61 

 

________________________________

From: Stu Packett [mailto:SPackett at fenwick.com] 
Sent: Friday, September 08, 2006 12:02 AM
To: James Grace; Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] background logos on 7941 -61

 

James:
Maybe you had a typo, buy my directory is under C:\Program
Files\Cisco\TFTPPath\Desktops\320x212x12.  Also, I needed to restart the
TFTP service when I made the changes.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Thursday, September 07, 2006 8:47 PM
To: 'Wes Sisk'
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61

Hey Wes

I use the doc to create and place the background on my CM ver 4.1.3.
this my lab...but when I tried to add that same logo to my production CM
ver 4.2.1  it does not show up.  I created the dir structure
Desktops\320x95x4\  and have the correct files in their.  But no show
when I goto the phone settings

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, September 06, 2006 2:08 PM
To: Stu Packett
Cc: Craig M Staffin; Kevin Thorngren;
cisco-voip-bounces at puck.nether.net; James Grace;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61

 

For all ambitious enough, this is possible through the phone SDK and
pushing instructions to each phone. I did a proof of concept for a group
of phones.  my bat script looks like this:

wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=kp2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk1.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=sk2.txt
http://172.18.104.45/CGI/Execute
sleep 2
wgetgnu --http-user=myuser --http-passwd=pass --post-file=settings.txt
http://172.18.104.45/CGI/Execute

wget is pretty straight forward, as are user and password.   the user
and password are for a user configured in callmanager's directory that
is assocaited with the phone.  If you are using Berbee or some of the
other phone control apps, they repoint the Authentication URL enterprise
parameter so the phone always gets a successful auth regardless of the
user/pass.
the --post-file is a text file that contains the XML encoding for the
desired instruction, CiscoIpPhoneExecute in all cases looks something
like this:
kp1.txt sends the instruction to press keypad button 1:
XML=%3CCiscoIPPhoneExecute%3E%3CExecuteItem+Priority%3D%220%22+URL%3D%22
Key:KeyPad1%22%2F%3E%3C%2FCiscoIPPhoneExecute%3E

it's heavy on file i/o but it's flexible enough for me to recreate the
*interesting* phone use scenarios i need to recreate and test en-mass.

/Wes


Stu Packett wrote: 

Craig,

I'm glad you brought that up as I'm about to deploy several 7971 phones
and would like to know if there is a bulk way of pushing the logos out.
I didn't seem to mind when I did it for 10 or so phones, but I will be
getting several more in the next few weeks.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Craig M Staffin
Sent: Wednesday, September 06, 2006 9:02 AM
To: Kevin Thorngren
Cc: cisco-voip-bounces at puck.nether.net; James Grace;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] background logos on 7941 -61


Kevin, 

I am just curious this process is very manual.  In other words you need
to set it for each phone.  Pretty labor intensive for 20,000 phones.  Is
there a way in CCM that you can force the phones to get lets say a
company logo as the background? 

Craig




Kevin Thorngren <kthorngr at cisco.com> <mailto:kthorngr at cisco.com>  
Sent by: cisco-voip-bounces at puck.nether.net 

09/05/2006 07:25 PM 

To

"James Grace" <jgrace at digitel.net> <mailto:jgrace at digitel.net>  

cc

cisco-voip at puck.nether.net 

Subject

Re: [cisco-voip] background logos on 7941 -61

 

 

 




How about this from the 7941/61 docs:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ 
ipp7961/admin/4_2/7961cus.htm#wp1030672

Kevin
On Sep 5, 2006, at 6:32 PM, James Grace wrote:

> Do any one have doc on how to put background logos on the 7941 61  
> ipphones.  Ive worked out the idle screensaver but, ive seen some  
> phones out there with the background changed
>  
> James D. Grace
> CCNP CCNA MCSE MCSA
> Digitel Corporation
> System Engineer / Professional Svc.
> MSN messenger ID:  jgrace29 at hotmail.com
> Mac IChat ID:  jgrace29 at mac.com
>  
>  
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



 
 
 



________________________________



 
 
 
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From pmowry at getgds.com  Fri Sep  8 17:02:33 2006
From: pmowry at getgds.com (Patrick Mowry)
Date: Fri, 8 Sep 2006 16:02:33 -0500
Subject: [cisco-voip] Is there any such thing as a null destination for
	an IOS dial-peer?
In-Reply-To: <8C3E4304-CEC0-4888-A100-E05B303DD05C@cisco.com>
Message-ID: <4C792A099259634C83F97067671B37FE0263B633@GDSEXC.gds.pri>

Because I want them to be able to call every US domestic area code
except for 7.  So I assume it is easier to have a 91.......... pattern
accessable to everyone and then 7 more specific peers to block those
that are not allowed to be called than to build peers that match
everything else except the 7 not allowed.  And it is not a system wide
restriction.  I only need to block them for 6 people out of 50.  Their
job function is legally not allowed to call the area codes I need to
block.

Am I wrong in that dial-peer assessment?

Thanks again,

-Patrick

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Friday, September 08, 2006 1:45 PM
To: Patrick Mowry
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Is there any such thing as a null destination
for an IOS dial-peer?

Rather than having a special COR to block calls why not just have COR  
setup so they don't match any dial-peers?  This will give them a fast  
busy and not tie up a channel.

-Ryan

On Sep 8, 2006, at 2:41 PM, Patrick Mowry wrote:

Hello,

   We have a site using CallManager Express where some users are
restricted from calling certain area codes and directory assistance, but
not others.  So an example dial-peer I have to make it work is:

dial-peer voice 31 pots
  corlist outgoing Block-NC
  description ** NC Blocked dial-peer **
  destination-pattern 91252.......
  progress_ind setup enable 3
  progress_ind progress enable 8
  port 0/0/0:1
  forward-digits 0
  prefix ###

So a dn who's class of restriction contains Block-NC, matches this
dial-peer, we send only ### to the carrier so the users hears a reorder
tone (fast busy) instead of the call going through.  And this is what I
want to happen, except I would prefer not to take up a channel on the T1
circuit just to return a fast busy tone.  I thought of routing the call
to a CUE auto-attendant number that would only say you are not allowed
to call this area code then hang up.  But they did not want to use up a
Unity session for this either.

Since I only want to block these calls for some users and not all, does
anyone have a suggestion on a better way to do this?

Thanks,

-Patrick

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



From bloemhard_johan at silverfalls.k12.or.us  Fri Sep  8 18:34:34 2006
From: bloemhard_johan at silverfalls.k12.or.us (Johan Bloemhard)
Date: Fri, 8 Sep 2006 15:34:34 -0700
Subject: [cisco-voip] Automatic Email/SMS Notification for Emergency Calls
	Solution
Message-ID: <ED9CD6C06120064590FC3BBA964D32A7FCF7F0@msexch1.silverfalls.k12.or.us>

Since so many of you have assisted me, I would like to share a little
script that I made that checks for someone dialing an emergency number
(eg 911) and then have the system email a distribution list (emails/sms)
of managers which should be notified for emergency purposes.

 

The script itself isn't a great mystery but hopefully will save some
time for you.

 

You can set it up to poll on the interval you wish.  Once it finds a
call record, it adds the pkid to a log database so it doesn't poll it
again in the future.  Because it's not intensive on the server/DB, I set
it up to poll every minute.  

 

You can pull information such as the calling party id, line label,
internal calling id, CSS, and more. 

 

This does require the basic knowledge of web servers, PHP, SQL and
requires users to be setup on the corresponding databases.  For myself,
I use CDRReporter and changed CDRReporter to db_owner on CCM0300 so that
it has select privileges.  The PHP server needs to be compiled for mssql
extension support.

 

This is a sample email that I have sent to an administrator distribution
list:

Extension 6343 with label (Mr. Smith x6343) and internal caller id (John
Smith) at the High School called 911 at 08/22/06 05:28:00 PM for 0.3
min. This extension is registered to smith_john.

 

Administrators love it and hopefully some of you will find it helpful as
well.

 

Feel free to contact me if you have any questions.

 

 

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

 


______________________
Disclaimer: This message may be legally privileged and/or is intended only for the use of the addressee(s).  The content and views expressed in this email may represent the views of the sender and not those of Silver Falls School District. 

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From bloemhard_johan at silverfalls.k12.or.us  Sat Sep  9 00:48:36 2006
From: bloemhard_johan at silverfalls.k12.or.us (Johan Bloemhard)
Date: Fri, 8 Sep 2006 21:48:36 -0700
Subject: [cisco-voip] Who called 911?
Message-ID: <000001c6d3cb$34ecc933$e4046b0a@silverfalls.k12.or.us>

I was just catching up on my emails and noticed your question, sent a couple of days back. I just finished a tiny app that I posted earlier which you could use for your situation. It's quite flexible. If you would like to see additional fields which can be added, change debug to 1. Just make sure to turn off emailing or have limited emailing so that everyone isn't bombarded with emails from 911 calls.

It's more proactive and emails after a 911 call is placed. With just some minor changes to your environment, you could have a free solution!

Let me know if you have any questions.

Johan Bloemhard

- Sent from my handheld



-----Original Message-----
From: "Peter Pauly"<ppauly at gmail.com>
Sent: 9/5/06 1:02:38 PM
To: "cisco-voip at puck.nether.net"<cisco-voip at puck.nether.net>
Subject: [cisco-voip] Who called 911?

Every time our security people are informed that someone made a 911
call, they call me to find out who it was (extension, name). I run a
query in query analyzer against the CDR database.

Are there any applications available that the security guys could run
from their desktop that would query callmanager over the network and
display 911 calls? What you guys doing about this problem?

On a related question, our CallManager sends caller id information just
fine. When I call my cellphone, I can see the extension of where the
call came from. But when I make a test call to 911, the PSAP is only
displaying the common billing phone number, not the caller-id
information. Is my telco doing something wrong?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

______________________
Disclaimer: This message may be legally privileged and/or is intended only for the use of the addressee(s).  The content and views expressed in this email may represent the views of the sender and not those of Silver Falls School District. 



From manoj.kalpage at gmail.com  Sat Sep  9 03:19:58 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Sat, 9 Sep 2006 16:19:58 +0900
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall
In-Reply-To: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD75B4@SVCEXCH01.firm.fenwick.llp>
References: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD75B4@SVCEXCH01.firm.fenwick.llp>
Message-ID: <bbe09c1d0609090019j6cab2b0dibeb19ec52f5e5ef8@mail.gmail.com>

Stu,
Thank you for the reply, I use windows 2003 DHCP server for my phones in LAN
but I can get my outside phone connect to CCM through internet. Do you have
IP phones connect to your CCM through internet without using VPN?

Thanks,
Manoj



On 9/9/06, Stu Packett <SPackett at fenwick.com> wrote:
>
>  Manoj:
> Is your PIX giving out DHCP addresses?  On my PIX 501, I have it setup as
> a DHCP server and these are my DHCP commands:
>
> dhcpd address xxx.xxx.xxx.xxx
> dhcpd dns xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
> dhcpd wins xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
> dhcpd lease 36000
> dhcpd ping_timeout 750
> dhcpd domain internaldomain.com
> dhcpd option 150 ip xxx.xxx.xxx.xxx <--TFTP address
> dhcpd enable inside
>
>  ------------------------------
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Manoj Kalpage
> *Sent:* Friday, September 08, 2006 4:18 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Internet IP phone connect through PIX Firewall
>
>
>  Hi All,
> Does any one has configured PIX firewall to connect internet IP phones to
> Call Manager. I have configure firewall to open all the port which CCM need
> but still no luck. Bellow is the config of my PIX. Am i missing anything?
>
> Here is the link I refered to open the TCP and UDP Ports
>
>
> http://www.cisco.com/application/pdf/en/us/guest/products/ps5820/c1693/ccmigration_09186a0080536eae.pdf
>
> Thank you in advance.
> Manoj
>
> :
> PIX Version 6.3(5)
> interface ethernet0 auto
> interface ethernet1 auto
> nameif ethernet0 outside security0
> nameif ethernet1 inside security100
> enable password u2zabJUOK.TTL3K1 encrypted
> passwd 1P5CrRl.dL8Oe4k2 encrypted
> hostname PBXLPIX01
> domain-name pbxl.jp
> clock timezone JST 9
> 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 pptp 1723
> fixup protocol rsh 514
> fixup protocol rtsp 554
> fixup protocol sip 5060
> fixup protocol sip udp 5060
> fixup protocol skinny 2000
> fixup protocol smtp 25
> fixup protocol snmp 161
> fixup protocol sqlnet 1521
> fixup protocol tftp 69
> names
> object-group service outbound-tcp tcp
>   port-object eq www
>   port-object eq https
>   port-object eq smtp
>   port-object eq ftp
>   port-object eq pop3
>   port-object eq imap4
>   port-object eq domain
>   port-object eq 123
>   port-object eq ssh
>   port-object eq citrix-ica
> object-group service outbound-udp udp
>   port-object eq domain
>   port-object eq ntp
> object-group service mail-inbound tcp
>   port-object eq www
>   port-object eq https
>   port-object eq smtp
> object-group service VoIP-udp udp
>   port-object range 16384 32768
>   port-object eq tftp
> object-group service VoIP-tcp tcp
>   port-object eq 3804
>   port-object eq 2443
>   port-object eq 2000
>   port-object eq www
>   port-object eq 69
>   port-object eq https
> access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group
> VoIP-tcp
> access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group
> VoIP-udp
> access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group
> outbound-tcp
> access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group
> outbound-udp
> access-list 101 permit tcp any host 210.81.12.195 object-group
> mail-inbound
> access-list 101 permit tcp any host 210.81.12.196 object-group VoIP-tcp
> access-list 101 permit udp any host 210.81.12.196 object-group VoIP-udp
> access-list 101 permit tcp any host 210.81.12.197 object-group VoIP-tcp
> access-list 101 permit udp any host 210.81.12.197 object-group VoIP-udp
>
> pager lines 24
> logging on
> logging trap informational
> logging host inside 172.16.0.26
> logging host inside 172.16.0.12
> icmp permit any unreachable outside
> icmp permit any outside
> mtu outside 1500
> mtu inside 1500
> ip address outside xxx.xxx.xxx.xxx 255.255.255.240
> ip address inside 172.16.0.2 255.255.0.0
>
> ip audit info action alarm
> ip audit attack action alarm
> ip local pool pbxlpool 10.1.0.100-10.1.0.200
> pdm locationxxx.xxx.xxx.xxx 255.255.255.255 outside
>
> pdm history enable
> arp timeout 14400
> global (outside) 1 interface
> nat (inside) 0 access-list VPNREMOTE
> nat (inside) 1 172.16.0.0 255.255.0.0 0 0
> static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask 255.255.255.255
> 0 1000
> static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
> 255.255.255.255 0 1000
> static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
> 255.255.255.255 0 1000
>
> access-group 101 in interface outside
> access-group 102 in interface inside
> route outside 0.0.0.0 0.0.0.0 210.81.12.193 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 sip-disconnect 0:02:00 sip-invite 0:03:00
> 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
>
> aaa authentication ssh console LOCAL
>
> http 172.16.0.12 255.255.255.255 inside
>
> snmp-server host inside 172.16.0.12
> snmp-server location pbxl-pix-datacentre
>
> snmp-server community pbxl
> snmp-server enable traps
> floodguard enable
>
> telnet 172.16.0.0 255.255.0.0 inside
> telnet 192.168.0.0 255.255.255.0 inside
> telnet timeout 60
> ssh 210.101.94.211 255.255.255.255 outside
> ssh 0.0.0.0 0.0.0.0 outside
> ssh 172.16.0.12 255.255.255.255 inside
> ssh 172.16.0.0 255.255.0.0 inside
> ssh 192.168.1.0 255.255.255.0 inside
>
> ssh timeout 60
> console timeout 0
> PBXLPIX01(config)#
> PBXLPIX01(config)#
>
>
>
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From tennille.spence at gsa.gov  Sat Sep  9 13:04:18 2006
From: tennille.spence at gsa.gov (tennille.spence at gsa.gov)
Date: Sat, 9 Sep 2006 12:04:18 -0500
Subject: [cisco-voip] Assistance with restoring a unity 3.1 voicemail account
Message-ID: <OF70E0E243.570531AC-ON862571E4.005D576F-862571E4.005DC735@gsa.gov>


Does anyone have any guidance on restoring messages for a single unity
voicemail account fro unity 3.1? The account was removed from the system
accidentally. I do have a system backup using DIRT but I have never
restored from it before and wasn't sure if I could simply select a single
voicemail to restore or if it would restore everyone's account and
messages.

Tennille Spence
IT Business Liaison
PBS Technology Branch
816-823-1357 Office
Tennille Spence
www.gsa.gov/r6


"To be without some of the things you want is an indispensable part of
happiness." - Bertrand Russell



From SPackett at fenwick.com  Sat Sep  9 15:41:45 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Sat, 9 Sep 2006 12:41:45 -0700
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC56F55E@SVCEXCH01.firm.fenwick.llp>

Sorry, I have never tried without the VPN.  I thought best practice was
to use the VPN because it was not advised to put the CCM on the public
internet.  If you do get your config working, I'd like to get a copy of
your config just for reference.  Thanks.

________________________________

From: Manoj Kalpage [mailto:manoj.kalpage at gmail.com] 
Sent: Saturday, September 09, 2006 12:20 AM
To: Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Internet IP phone connect through PIX Firewall


Stu, 
Thank you for the reply, I use windows 2003 DHCP server for my phones in
LAN but I can get my outside phone connect to CCM through internet. Do
you have IP phones connect to your CCM through internet without using
VPN?  
 
Thanks,
Manoj


 
On 9/9/06, Stu Packett <SPackett at fenwick.com> wrote: 

	Manoj:
	Is your PIX giving out DHCP addresses?  On my PIX 501, I have it
setup as a DHCP server and these are my DHCP commands: 
	 
	dhcpd address xxx.xxx.xxx.xxx
	dhcpd dns xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
	dhcpd wins xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
	dhcpd lease 36000 
	dhcpd ping_timeout 750
	dhcpd domain internaldomain.com <http://internaldomain.com/> 
	dhcpd option 150 ip xxx.xxx.xxx.xxx <--TFTP address 
	dhcpd enable inside

________________________________

	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 Manoj Kalpage
	Sent: Friday, September 08, 2006 4:18 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Internet IP phone connect through PIX
Firewall
	
	 
	
	Hi All,
	Does any one has configured PIX firewall to connect internet IP
phones to Call Manager. I have configure firewall to open all the port
which CCM need but still no luck. Bellow is the config of my PIX. Am i
missing anything? 
	
	Here is the link I refered to open the TCP and UDP Ports
	
	
http://www.cisco.com/application/pdf/en/us/guest/products/ps5820/c1693/c
cmigration_09186a0080536eae.pdf
	
	Thank you in advance.
	Manoj
	
	:
	PIX Version 6.3(5)
	interface ethernet0 auto
	interface ethernet1 auto 
	nameif ethernet0 outside security0
	nameif ethernet1 inside security100
	enable password u2zabJUOK.TTL3K1 encrypted
	passwd 1P5CrRl.dL8Oe4k2 encrypted
	hostname PBXLPIX01
	domain-name pbxl.jp <http://pbxl.jp/> 
	
	clock timezone JST 9
	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 pptp 1723 
	fixup protocol rsh 514
	fixup protocol rtsp 554
	fixup protocol sip 5060
	fixup protocol sip udp 5060
	fixup protocol skinny 2000
	fixup protocol smtp 25
	fixup protocol snmp 161 
	fixup protocol sqlnet 1521 
	fixup protocol tftp 69
	names
	object-group service outbound-tcp tcp
	  port-object eq www
	  port-object eq https
	  port-object eq smtp
	  port-object eq ftp
	  port-object eq pop3 
	  port-object eq imap4 
	  port-object eq domain
	  port-object eq 123
	  port-object eq ssh
	  port-object eq citrix-ica
	object-group service outbound-udp udp
	  port-object eq domain
	  port-object eq ntp 
	object-group service mail-inbound tcp 
	  port-object eq www
	  port-object eq https
	  port-object eq smtp
	object-group service VoIP-udp udp
	  port-object range 16384 32768
	  port-object eq tftp
	object-group service VoIP-tcp tcp 
	  port-object eq 3804 
	  port-object eq 2443
	  port-object eq 2000
	  port-object eq www
	  port-object eq 69
	  port-object eq https
	
	access-list 102 permit tcp 172.16.0.0 <http://172.16.0.0/>
255.255.0.0 <http://255.255.0.0/>  any object-group VoIP-tcp
	access-list 102 permit udp 172.16.0.0 <http://172.16.0.0/>
255.255.0.0 <http://255.255.0.0/>  any object-group VoIP-udp
	access-list 102 permit tcp 172.16.0.0 <http://172.16.0.0/>
255.255.0.0 <http://255.255.0.0/>  any object-group outbound-tcp
	access-list 102 permit udp 172.16.0.0 <http://172.16.0.0/>
255.255.0.0 <http://255.255.0.0/>  any object-group outbound-udp 
	access-list 101 permit tcp any host 210.81.12.195
<http://210.81.12.195/>  object-group mail-inbound 
	access-list 101 permit tcp any host 210.81.12.196
<http://210.81.12.196/>  object-group VoIP-tcp 
	access-list 101 permit udp any host 210.81.12.196
<http://210.81.12.196/>  object-group VoIP-udp
	access-list 101 permit tcp any host 210.81.12.197
<http://210.81.12.197/>  object-group VoIP-tcp
	access-list 101 permit udp any host 210.81.12.197
<http://210.81.12.197/>  object-group VoIP-udp 
	
	pager lines 24
	logging on
	logging trap informational
	logging host inside 172.16.0.26 <http://172.16.0.26/>  
	
	logging host inside 172.16.0.12 <http://172.16.0.12/> 
	
	icmp permit any unreachable outside
	icmp permit any outside
	mtu outside 1500
	mtu inside 1500
	ip address outside xxx.xxx.xxx.xxx 255.255.255.240
<http://255.255.255.240/> 
	
	ip address inside 172.16.0.2 <http://172.16.0.2/>  255.255.0.0
<http://255.255.0.0/> 
	
	ip audit info action alarm
	ip audit attack action alarm 
	ip local pool pbxlpool 10.1.0.100-10.1.0.200
	
	pdm locationxxx.xxx.xxx.xxx 255.255.255.255
<http://255.255.255.255/>  outside
	
	pdm history enable
	arp timeout 14400
	global (outside) 1 interface
	nat (inside) 0 access-list VPNREMOTE 
	
	nat (inside) 1 172.16.0.0 <http://172.16.0.0/>  255.255.0.0
<http://255.255.0.0/>  0 0
	static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
255.255.255.255 <http://255.255.255.255/> 0 1000
	static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
255.255.255.255 <http://255.255.255.255/>  0 1000
	static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
255.255.255.255 <http://255.255.255.255/>  0 1000
	
	access-group 101 in interface outside
	access-group 102 in interface inside
	
	route outside 0.0.0.0 <http://0.0.0.0/>  0.0.0.0
<http://0.0.0.0/>  210.81.12.193 <http://210.81.12.193/>  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 sip-disconnect 0:02:00 sip-invite 0:03:00 
	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
	
	aaa authentication ssh console LOCAL
	
	
	http 172.16.0.12 <http://172.16.0.12/>  255.255.255.255
<http://255.255.255.255/>  inside
	
	snmp-server host inside 172.16.0.12 <http://172.16.0.12/> 
	
	snmp-server location pbxl-pix-datacentre
	
	snmp-server community pbxl
	snmp-server enable traps
	floodguard enable
	
	
	telnet 172.16.0.0 <http://172.16.0.0/>  255.255.0.0
<http://255.255.0.0/>  inside
	telnet 192.168.0.0 <http://192.168.0.0/>  255.255.255.0
<http://255.255.255.0/>  inside
	telnet timeout 60 
	ssh 210.101.94.211 <http://210.101.94.211/>  255.255.255.255
<http://255.255.255.255/>  outside
	ssh 0.0.0.0 <http://0.0.0.0/>  0.0.0.0 <http://0.0.0.0/>
outside
	ssh 172.16.0.12 <http://172.16.0.12/>  255.255.255.255
<http://255.255.255.255/>  inside
	ssh 172.16.0.0 <http://172.16.0.0/>  255.255.0.0
<http://255.255.0.0/>  inside
	ssh 192.168.1.0 <http://192.168.1.0/> 255.255.255.0
<http://255.255.255.0/>  inside
	
	ssh timeout 60
	console timeout 0
	PBXLPIX01(config)#
	PBXLPIX01(config)#
	
	
	


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From Brian.Henry at apptis.com  Sat Sep  9 16:56:18 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Sat, 9 Sep 2006 16:56:18 -0400
Subject: [cisco-voip] Assistance with restoring a unity 3.1 voicemail
	account
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC028382EB@CH-EXCH-02.intelltech.apptis.inc>

By using DiRT it will restore the entire Unity configuration. If you are also using the backup utility for backing up the voicemail mailbox's then it will restore them also. You cannot restore an individual person by using the utilities.
 
However if you are using the backup utility then it will create a .PST file for every mailbox.  Create the person back in Unity as if you were creating a new user.
Then use Outlook on another computer to open the individuals email/mailbox and then import the PST for that user.
 
Just a thought.  Of course the settings for that person as far as conversation, notification, etc.. will have to be put back manually.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of tennille.spence at gsa.gov
Sent: Sat 9/9/2006 1:04 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Assistance with restoring a unity 3.1 voicemail account




Does anyone have any guidance on restoring messages for a single unity
voicemail account fro unity 3.1? The account was removed from the system
accidentally. I do have a system backup using DIRT but I have never
restored from it before and wasn't sure if I could simply select a single
voicemail to restore or if it would restore everyone's account and
messages.

Tennille Spence
IT Business Liaison
PBS Technology Branch
816-823-1357 Office
Tennille Spence
www.gsa.gov/r6


"To be without some of the things you want is an indispensable part of
happiness." - Bertrand Russell


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


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From erickbe at yahoo.com  Sat Sep  9 22:16:57 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sat, 9 Sep 2006 19:16:57 -0700 (PDT)
Subject: [cisco-voip] Assistance with restoring a unity 3.1 voicemail
	account
In-Reply-To: <OF70E0E243.570531AC-ON862571E4.005D576F-862571E4.005DC735@gsa.gov>
Message-ID: <20060910021657.34923.qmail@web37212.mail.mud.yahoo.com>

Is the mail account still on the mailstore, or was that also removed?

----- Original Message ----
From: tennille.spence at gsa.gov
To: cisco-voip at puck.nether.net
Sent: Saturday, September 9, 2006 12:04:18 PM
Subject: [cisco-voip] Assistance with restoring a unity 3.1 voicemail account


Does anyone have any guidance on restoring messages for a single unity
voicemail account fro unity 3.1? The account was removed from the system
accidentally. I do have a system backup using DIRT but I have never
restored from it before and wasn't sure if I could simply select a single
voicemail to restore or if it would restore everyone's account and
messages.

Tennille Spence
IT Business Liaison
PBS Technology Branch
816-823-1357 Office
Tennille Spence
www.gsa.gov/r6


"To be without some of the things you want is an indispensable part of
happiness." - Bertrand Russell


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





From aman.chugh at gmail.com  Sun Sep 10 00:42:54 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Sun, 10 Sep 2006 10:12:54 +0530
Subject: [cisco-voip] Qos for V3PN
Message-ID: <d4c697ad0609092142n4fc57598q76a7cce3d1534c45@mail.gmail.com>

Hi All,

 I working on a configuration for Qos for a site to site ipsec vpn tunnel on
which we run voice,The link is 512 kbps and also supports nat traffic to the
internet.I have gone thorugh the QOS srnd for V3PN  and have put together a
config,please let me know if there is any gotcha to be aware of.This remote
site currently uses around 6 ipphones registered to a central site
callmanager and there are no gateways at this site.I am doing this Qos
configuration as Occassionly I see email traffic overwhelming my voice
traffic.Please share your experience with the same scenario.

1. Classify traffic using ACL

ip access-list extended Voice-bearer
remark Match the UDP ports that VoIP Uses for Bearer Traffic
permit udp any any range 16384 32767

ip access-list extended Voice-signalling
remark SCCP
permit tcp any any range 2000 2002

2.Make class map

class-map match-all Voice-rtp
description  Bearer Traffic
match access-group name Voice-bearer

class-map match-all Voice-control
description  Control Traffic (SCCP)
match access-group name Voice-signalling
3.policy-map encr-traffic
class Voice-rtp
priority 256
class Voice-control
bandwidth 128

4.Apply the policy on serial interface connected to the internet
service-policy output encr-traffic
5. Also add the qos pre-classify on serial interface connected to the
internet


Aman
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From ml at iridescent.org  Sun Sep 10 00:38:40 2006
From: ml at iridescent.org (Lawrence E. Bakst)
Date: Sat, 9 Sep 2006 21:38:40 -0700
Subject: [cisco-voip] Cabling Standards for IPT
In-Reply-To: <45006142.1080107@cisco.com>
References: <ef905eb80609070738x82e3e66t6481ce104c9b9492@mail.gmail.com>
	<45006142.1080107@cisco.com>
Message-ID: <p06230985c12945ecba11@[192.168.0.7]>

With GbE now a commodity technology I can't imagine running anything other than high quality CAT-6e cable, unless cost is major concern. If you look carefully you can even find cable that will probably work with 10 GbE over copper. Most of the new Cisco phones run GbE. There is a significant difference in cable and jacks that meet cat5e vs cat6 when it comes to specifications. If cost is a concern I'd suggest running cat6 cable and get cat5e jacks as you can upgrade those latter with relative ease.

Best,

leb


At 2:13 PM -0400 9/7/06, Wes Sisk wrote:
>I believe cat5e is the minimum requirement.
>
>/Wes
>
>tusker keg wrote:
>> Howdy'
>>
>> Was wondering if there are any standards /BCP references out there for
>> a successful IPT deployment as far LAN cabling is concerned
>>
>>
>> Regards
>>
>> ./tusker
>> _______________________________________________
>> 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  Sun Sep 10 09:28:38 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Sun, 10 Sep 2006 09:28:38 -0400
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall
In-Reply-To: <01C8F21F30DE9740B53FD2EF05C3A8FC56F55E@SVCEXCH01.firm.fenwick.llp>
References: <01C8F21F30DE9740B53FD2EF05C3A8FC56F55E@SVCEXCH01.firm.fenwick.llp>
Message-ID: <33AFDCDE-4C00-43F3-82E3-47D6423A4B2F@cisco.com>

Manoj,

Are you doing NAT on your PIX?  If so, you will need special CM+PIX  
config.

phones download SEP<mac>.cnf.xml from TFTP server.  Inside this XML  
file is a listing of which CM servers the phone should register to  
using SCCP.  This file also tells the phone what TCP port to use for  
the SCCP communication.  The CM servers are listed in this by name or  
IP address based on how your CM is configured under system->server.

For NAT traversal, your CM will have to be configured using host  
names.  Your PIX will have to do DNS fixup.  Your phone will receive  
the name cm1.manoj.com.  the phone must do DNS lookup on this and  
receive the external address of your CM server.

Otherwise, you must use a valid internet address for the IP of your  
CM server.

/Wes
On Sep 9, 2006, at 3:41 PM, Stu Packett wrote:

Sorry, I have never tried without the VPN.  I thought best practice  
was to use the VPN because it was not advised to put the CCM on the  
public internet.  If you do get your config working, I'd like to get  
a copy of your config just for reference.  Thanks.

From: Manoj Kalpage [mailto:manoj.kalpage at gmail.com]
Sent: Saturday, September 09, 2006 12:20 AM
To: Stu Packett
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Internet IP phone connect through PIX Firewall

Stu,
Thank you for the reply, I use windows 2003 DHCP server for my phones  
in LAN but I can get my outside phone connect to CCM through  
internet. Do you have IP phones connect to your CCM through internet  
without using VPN?

Thanks,
Manoj



On 9/9/06, Stu Packett <SPackett at fenwick.com> wrote:
Manoj:
Is your PIX giving out DHCP addresses?  On my PIX 501, I have it  
setup as a DHCP server and these are my DHCP commands:

dhcpd address xxx.xxx.xxx.xxx
dhcpd dns xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
dhcpd wins xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
dhcpd lease 36000
dhcpd ping_timeout 750
dhcpd domain internaldomain.com
dhcpd option 150 ip xxx.xxx.xxx.xxx <--TFTP address
dhcpd enable inside

From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip- 
bounces at puck.nether.net] On Behalf Of Manoj Kalpage
Sent: Friday, September 08, 2006 4:18 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall


Hi All,
Does any one has configured PIX firewall to connect internet IP  
phones to Call Manager. I have configure firewall to open all the  
port which CCM need but still no luck. Bellow is the config of my  
PIX. Am i missing anything?

Here is the link I refered to open the TCP and UDP Ports

http://www.cisco.com/application/pdf/en/us/guest/products/ps5820/ 
c1693/ccmigration_09186a0080536eae.pdf

Thank you in advance.
Manoj

:
PIX Version 6.3(5)
interface ethernet0 auto
interface ethernet1 auto
nameif ethernet0 outside security0
nameif ethernet1 inside security100
enable password u2zabJUOK.TTL3K1 encrypted
passwd 1P5CrRl.dL8Oe4k2 encrypted
hostname PBXLPIX01
domain-name pbxl.jp
clock timezone JST 9
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 pptp 1723
fixup protocol rsh 514
fixup protocol rtsp 554
fixup protocol sip 5060
fixup protocol sip udp 5060
fixup protocol skinny 2000
fixup protocol smtp 25
fixup protocol snmp 161
fixup protocol sqlnet 1521
fixup protocol tftp 69
names
object-group service outbound-tcp tcp
   port-object eq www
   port-object eq https
   port-object eq smtp
   port-object eq ftp
   port-object eq pop3
   port-object eq imap4
   port-object eq domain
   port-object eq 123
   port-object eq ssh
   port-object eq citrix-ica
object-group service outbound-udp udp
   port-object eq domain
   port-object eq ntp
object-group service mail-inbound tcp
   port-object eq www
   port-object eq https
   port-object eq smtp
object-group service VoIP-udp udp
   port-object range 16384 32768
   port-object eq tftp
object-group service VoIP-tcp tcp
   port-object eq 3804
   port-object eq 2443
   port-object eq 2000
   port-object eq www
   port-object eq 69
   port-object eq https
access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group  
VoIP-tcp
access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group  
VoIP-udp
access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group  
outbound-tcp
access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group  
outbound-udp
access-list 101 permit tcp any host 210.81.12.195 object-group mail- 
inbound
access-list 101 permit tcp any host 210.81.12.196 object-group VoIP-tcp
access-list 101 permit udp any host 210.81.12.196 object-group VoIP-udp
access-list 101 permit tcp any host 210.81.12.197 object-group VoIP-tcp
access-list 101 permit udp any host 210.81.12.197 object-group VoIP-udp

pager lines 24
logging on
logging trap informational
logging host inside 172.16.0.26
logging host inside 172.16.0.12
icmp permit any unreachable outside
icmp permit any outside
mtu outside 1500
mtu inside 1500
ip address outside xxx.xxx.xxx.xxx 255.255.255.240
ip address inside 172.16.0.2 255.255.0.0

ip audit info action alarm
ip audit attack action alarm
ip local pool pbxlpool 10.1.0.100-10.1.0.200
pdm locationxxx.xxx.xxx.xxx 255.255.255.255 outside

pdm history enable
arp timeout 14400
global (outside) 1 interface
nat (inside) 0 access-list VPNREMOTE
nat (inside) 1 172.16.0.0 255.255.0.0 0 0
static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask  
255.255.255.255 0 1000
static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask  
255.255.255.255 0 1000
static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask  
255.255.255.255 0 1000

access-group 101 in interface outside
access-group 102 in interface inside
route outside 0.0.0.0 0.0.0.0 210.81.12.193 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 sip-disconnect 0:02:00 sip-invite 0:03:00
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

aaa authentication ssh console LOCAL

http 172.16.0.12 255.255.255.255 inside

snmp-server host inside 172.16.0.12
snmp-server location pbxl-pix-datacentre

snmp-server community pbxl
snmp-server enable traps
floodguard enable

telnet 172.16.0.0 255.255.0.0 inside
telnet 192.168.0.0 255.255.255.0 inside
telnet timeout 60
ssh 210.101.94.211 255.255.255.255 outside
ssh 0.0.0.0 0.0.0.0 outside
ssh 172.16.0.12 255.255.255.255 inside
ssh 172.16.0.0 255.255.0.0 inside
ssh 192.168.1.0 255.255.255.0 inside

ssh timeout 60
console timeout 0
PBXLPIX01(config)#
PBXLPIX01(config)#



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

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From lee.pedder at gmail.com  Sun Sep 10 12:10:32 2006
From: lee.pedder at gmail.com (Lee Pedder)
Date: Sun, 10 Sep 2006 17:10:32 +0100
Subject: [cisco-voip] Qos for V3PN
In-Reply-To: <d4c697ad0609092142n4fc57598q76a7cce3d1534c45@mail.gmail.com>
References: <d4c697ad0609092142n4fc57598q76a7cce3d1534c45@mail.gmail.com>
Message-ID: <81eaebb50609100910r3df74579td926a2b5de6bbd64@mail.gmail.com>

Consider using IP DSCP values as they tend to make your QoS policy
easier to manage. Cisco phones will set some DSCP values by default.
E.g.

class-map match-all VOICE
  description IP Phones mark Voice to EF by default
 match ip dscp ef
class-map match-all CALL-SIGNALING
  description Call Signalling Marked Packets from Phones
 match ip dscp cs3

Also, if you are using GRE over IPsec be aware of the packetization
overhead this can add. For a G.729 call using 20ms packetization
intervals, this results in a 56kbps stream on the WAN when adding UDP,
IP, GRE, ESP and IP headers. You can reduce this slightly by changing
codec settings to 30ms but this increases call latency. I have not
tried this personally.

Also, ensure you set up the phones in a location with the appropriate
bandwidth setting in CCM. This should allow the exact number of calls
your QoS policy can handle, and present users with a "not enough
bandwidth" message on subsequent calls. Because of overhead, this will
need some tweaking as the bandwidth you tell callmanager will not take
this into account.

From aman.chugh at gmail.com  Sun Sep 10 16:21:49 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Mon, 11 Sep 2006 01:51:49 +0530
Subject: [cisco-voip] Qos for V3PN
In-Reply-To: <81eaebb50609100910r3df74579td926a2b5de6bbd64@mail.gmail.com>
References: <d4c697ad0609092142n4fc57598q76a7cce3d1534c45@mail.gmail.com>
	<81eaebb50609100910r3df74579td926a2b5de6bbd64@mail.gmail.com>
Message-ID: <d4c697ad0609101321k260e848fsdaac9c789e13e76e@mail.gmail.com>

Thanks for your Input,I am not using GRE with IPSEC ,it's only IPSEC.I did
apply the inital QOS configuration and will do some testing this week,will
keep you updated about the results.

Aman


On 9/10/06, Lee Pedder <lee.pedder at gmail.com> wrote:
>
> Consider using IP DSCP values as they tend to make your QoS policy
> easier to manage. Cisco phones will set some DSCP values by default.
> E.g.
>
> class-map match-all VOICE
> description IP Phones mark Voice to EF by default
> match ip dscp ef
> class-map match-all CALL-SIGNALING
> description Call Signalling Marked Packets from Phones
> match ip dscp cs3
>
> Also, if you are using GRE over IPsec be aware of the packetization
> overhead this can add. For a G.729 call using 20ms packetization
> intervals, this results in a 56kbps stream on the WAN when adding UDP,
> IP, GRE, ESP and IP headers. You can reduce this slightly by changing
> codec settings to 30ms but this increases call latency. I have not
> tried this personally.
>
> Also, ensure you set up the phones in a location with the appropriate
> bandwidth setting in CCM. This should allow the exact number of calls
> your QoS policy can handle, and present users with a "not enough
> bandwidth" message on subsequent calls. Because of overhead, this will
> need some tweaking as the bandwidth you tell callmanager will not take
> this into account.
>
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From smgustafson at gmail.com  Sun Sep 10 22:57:56 2006
From: smgustafson at gmail.com (Steve G)
Date: Sun, 10 Sep 2006 19:57:56 -0700
Subject: [cisco-voip] Unity Call Transfer "wait while I transfer your call"
Message-ID: <8a1f34b50609101957g552dabe7s6b0fd424deb00e7f@mail.gmail.com>

Hello,
Is it possible to disable the voice of Unity saying "wait while I transfer
your call" once the caller selects a transfer option?

If you can let me know the process to disable this, I would appreciate it.

Thank you,
Steve
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From erickbe at yahoo.com  Mon Sep 11 00:11:55 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon, 11 Sep 2006 12:11:55 +0800 (CST)
Subject: [cisco-voip] Unity Call Transfer "wait while I transfer your
	call"
In-Reply-To: <8a1f34b50609101957g552dabe7s6b0fd424deb00e7f@mail.gmail.com>
Message-ID: <20060911041155.88019.qmail@web37203.mail.mud.yahoo.com>

Check out this cisco forum post.

http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=IP%20Communications%20and%20Video&topic=Unified%20Communications&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40%40.1ddb8400/0#selected_message

Posted by lindborg   -  		Jun 22, 2006, 8:06am PST

" This isn't TAC supported, of course, and upgrades will replace the prompt with the original system one, but you can edit the "Please wait while I transfer your call prompt" by rerecording this prompt: 
 
\Commserver\Localize\Prompts\ENU\AvPHGreet\AvPhGreetENU005.WAV 
 
No reboot or anything like that should be required. I'd make a backup of the original prompt just in case you want to go back or this isn't the guy you really wanted to change. You can also pull it from the original install CD if you get in a jam. 
 
As a side note - you can find all the prompt wording for all prompts in the PROMPT.INI files in the \localize\prompts\ENU\G711\xxx directories - just search on the string you're looking for across all .INI files in that folder tree and it should come up with the prompt name you're looking for. "


----- Original Message ----
From: Steve G <smgustafson at gmail.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Sunday, September 10, 2006 9:57:56 PM
Subject: [cisco-voip] Unity Call Transfer "wait while I transfer your call"

Hello,
 Is it possible to disable the voice of Unity saying "wait while I transfer your call" once the caller selects a transfer option?
  
 If you can let me know the process to disable this, I would appreciate it.
  
 Thank you,
 Steve
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




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From ml at iridescent.org  Mon Sep 11 02:03:25 2006
From: ml at iridescent.org (Lawrence E. Bakst)
Date: Sun, 10 Sep 2006 23:03:25 -0700
Subject: [cisco-voip] SCCP ver 5 load for 7960 compatible with CME 3.3
Message-ID: <p06230990c12a940ed9d3@[192.168.0.7]>

Hi All,

I am current using
load 7960-7940 P00305000600
with CME 3.3.

I have a mix of 7960, 7970, and 7920 phones.

Everything is SCCP ver 5 except the 7060's which are SCCP ver 4 and I have problems with them where after a conversation has been going for some time (minutes) the person at the other end sometimes hears a garbled conversation. Hanging up and re-dialing fixes it until it happens again. This is going out through an analog FXO port.

So I was thinking of advancing the load for the 7960's.

I am stuck with CME 3.3 for now. Any advice?

To answer my own question, I tried 2 loads:

8.0.4
P00308000400
which is SCCP ver 9

6.0.5
P00306000500
which is SCCP ver 5

I decided to stick with 6.0.5.

Anyone going up to 6.x should know that once you do that you are stuck with signed loads.

Anyone (from Cisco) know what versions of SCCP are in CME 3.3, 3.4 and 4.0? 

It would nice to have that info so you can at least sync versions.

Anyway to query the SCCP version from the cli or via MIB?

Best,

leb


From cj.alan at gmail.com  Mon Sep 11 03:43:58 2006
From: cj.alan at gmail.com (Alan Su)
Date: Mon, 11 Sep 2006 15:43:58 +0800
Subject: [cisco-voip]  long duration active call but no channel occupied
Message-ID: <8f9f19f00609110043n3ecb98e9hb0a5384eae856465@mail.gmail.com>

Dear All,

I have a very werid situation in my customer's voip environment:

Last nite, the network administrator report the abnormal traffic, and in
voice gateway, we found out that there is a call last for "5 weeks" long and
still active. But when we try to show voice port summary, there is no any
channel been used !!

Attached is the "sh call active voice brief" message, any comment would be
very grateful.


Regards,
Alan
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02H#sho call act voi brie

<ID>: <CallID> <start>hs.<index> +<connect> pid:<peer_id> <dir> <addr> <state>

  dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes>

 IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>

  delay:<last>/<min>/<max>ms <codec>

  media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>

  MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>

 

   last <buf event time>s dur:<Min>/<Max>s

 FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

  <codec> (payload size)

 ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>

  <codec> (payload size)

 Tele <int> (callID) [channel_id] tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l

> i/o:<l>/<l> dBm

  MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<

drops>

         speeds(bps): local <rx>/<tx> remote <rx>/<tx>

 Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<m

anf>

 bw: <req>/<act> codec: <audio>/<video>

  tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 by

tes>

 rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 byt

es>

 

 

Telephony call-legs: 2

SIP call-legs: 3

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 5

1524 : 3258 810008234ms.1 +830 pid:999 Originate 852221 active

 dur 5w6d tx:22518/439573 rx:31552/614042

 IP 172.28.222.254:18192 SRTP: off rtt:0ms pl:585290/900ms lost:2575/1152/52 del

ay:60/60/250ms g729r8

 media inactive detected:n media contrl rcvd:n/a timestamp:n/a

33E1 : 16411 136729798ms.1 +2460 pid:0 Answer 1 connected

 dur 00:01:35 tx:2716/434560 rx:4848/775680

 IP 172.16.168.234:32548 SRTP: off rtt:0ms pl:94645/0ms lost:0/1/0 delay:60/60/6

5ms g711alaw

 media inactive detected:n media contrl rcvd:n/a timestamp:n/a

33E1 : 16412 136730188ms.1 +2070 pid:100 Originate 8675521 active

 dur 00:01:35 tx:4849/814632 rx:2716/434560

 Tele 0/3/0:15 (16412) [0/3/0.3] tx:97315/54320/0ms g711alaw noise:-57 acom:45

i/0:-31/-54 dBm

 

33E3 : 16413 136800218ms.1 +2910 pid:0 Answer 1 connected

 dur 00:00:24 tx:694/111040 rx:1321/211360

 IP 172.16.168.234:32672 SRTP: off rtt:0ms pl:24055/85ms lost:0/2/1 delay:105/65

/125ms g711alaw

 media inactive detected:n media contrl rcvd:n/a timestamp:n/a

33E3 : 16414 136800598ms.1 +2530 pid:100 Originate 8675521 active

 dur 00:00:24 tx:1321/221928 rx:694/111040

 Tele 0/3/0:15 (16414) [0/3/0.1] tx:26895/13880/0ms g711alaw noise:-67 acom:34

i/0:-68/-33 dBm

 

Telephony call-legs: 2

SIP call-legs: 3

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 5


From pmarques74 at gmail.com  Mon Sep 11 07:06:52 2006
From: pmarques74 at gmail.com (pedro marques)
Date: Mon, 11 Sep 2006 12:06:52 +0100
Subject: [cisco-voip] CCM 4.1(3) with 2N Voice Blue Enterprise GSM VoIP
	Gateway
Message-ID: <d2216e4e0609110406w58f74644t8ad05156a3fe2356@mail.gmail.com>

Hello all,

The scenario:

A CCM running 4.1(3)sr2 connected to a 2N VBE GSM VoIP Gateway through H.323
.

The problem:

In a redundant scenario, where if there are no channels available in the GSM
gateway the CallManager should redirect the calls to an alternate route (via
ISDN), CallManager is not acting as expected because it will never switch to
the defined alternate route.

What I did to try to solve the problem:

In CCMs' Service Parameters I manipulated the "Stop Routing on Unallocated
Number Flag" and "Stop Routing on User Busy Flag" parameters and changed
their values from True to False. After that I tried several of the "Reject
call code" and "No route code" options present in the config page of the
H.323 line, namelly the following ones:

"Local endpoint application cleared call", "Local endpoint did not accept
call", "Local endpoint declined to answer call", "Local endpoint busy",
"Local endpoint congested" and "Call cleared due to no circuit channel
available".

Unfortunatelly none of this tests solved the problem.

So I end up asking the group if anyone already ran into this or a similar
problem that can provide some hints on how to solve this situation.

Best regards,

-- 
Pedro Miguel Marques
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From KWhitson at cmbtec.com  Mon Sep 11 07:39:09 2006
From: KWhitson at cmbtec.com (Karl Whitson)
Date: Mon, 11 Sep 2006 07:39:09 -0400
Subject: [cisco-voip] Unity Call Transfer "wait while I transfer your
	call"
In-Reply-To: <8a1f34b50609101957g552dabe7s6b0fd424deb00e7f@mail.gmail.com>
Message-ID: <8A4D67E299AA254CAB2B5985EEDD865804BAD1@cmb-bm-sbs.CMB-BM.Local>

Steve,

 

With Unity version 4.1 and later, this option may be turned off/on per
subscriber.

On the Subscriber account go to Call Transfer.

Go to While Transferring, Notify Caller?

Click the checkbox labeled "Do not play the "Wait while I transfer your
call prompt"."

If you want to change this setting for all existing subscribers at the
same time you can use the Bulk Edit Utility in the Unity Tools Depot.

 

Karl Whitson

CMB Technologies

40 Cape Rd.

Milford, MA  01757

DID:  (508) 422-8413

Cell:  (603) 930-4189

Fax:  (508) 634-8678

Karl.Whitson at cmbtec.com

www.cmbtec.com

TO OPEN A SUPPORT CASE EMAIL SUPPORT at CMBTEC.COM
<mailto:SUPPORT at CMBTEC.COM>  OR CALL (508) 422-8429

  

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Steve G
Sent: Sunday, September 10, 2006 10:58 PM
To: ciscovoip
Subject: [cisco-voip] Unity Call Transfer "wait while I transfer your
call"

 

Hello,

Is it possible to disable the voice of Unity saying "wait while I
transfer your call" once the caller selects a transfer option?

 

If you can let me know the process to disable this, I would appreciate
it.

 

Thank you,

Steve

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From dwinter at ctiusa.com  Mon Sep 11 07:59:40 2006
From: dwinter at ctiusa.com (David Winter)
Date: Mon, 11 Sep 2006 07:59:40 -0400
Subject: [cisco-voip] Unity Call Transfer "wait while I transfer
	yourcall"
In-Reply-To: <8A4D67E299AA254CAB2B5985EEDD865804BAD1@cmb-bm-sbs.CMB-BM.Local>
References: <8A4D67E299AA254CAB2B5985EEDD865804BAD1@cmb-bm-sbs.CMB-BM.Local>
Message-ID: <45054FAC.2040306@ctiusa.com>

"OK... if you promise to be extra careful, understand that this isn't a 
suported operation and that future upgrades may (will) over write these 
changes and you'll have to do this again, here's the deal.

You need to replace the prompt in question with a silent prompt 
(basically blank but not quite empty) so that Unity's conversation code 
wont get cranky when it goes to look for a prompt it's expecting and 
it's not there.

So, step one is to find a blank prompt. There are several. I suggest using:

\Commserver\Localize\Prompts\ENU\AvAddrSearch\AvAddrSearchENU003.WAV.

You want to copy this blank prompt over the prompt you don't want to 
hear. That guy is:

\Commserver\Localize\Prompts\ENU\AvPHGreet\AvPhGreetENU005.WAV

This is the "Please wait while I transfer your call". This should do 
what you want. No reboot or anything like that should be required. I'd 
make a backup of the original prompt just in case you want to go back or 
this isn't the guy you really wanted to remove. You can also pull it 
from the original install CD if you get in a jam.


Karl Whitson wrote:
>
> Steve,
>
>  
>
> With Unity version 4.1 and later, this option may be turned off/on per 
> subscriber.
>
> On the Subscriber account go to Call Transfer.
>
> Go to While Transferring, Notify Caller?
>
> Click the checkbox labeled "Do not play the "Wait while I transfer 
> your call prompt"."
>
> If you want to change this setting for all existing subscribers at the 
> same time you can use the Bulk Edit Utility in the Unity Tools Depot.
>
>  
>
> Karl Whitson
>
> CMB Technologies
>
> 40 Cape Rd.
>
> Milford, MA  01757
>
> DID:  (508) 422-8413
>
> Cell:  (603) 930-4189
>
> Fax:  (508) 634-8678
>
> Karl.Whitson at cmbtec.com <mailto:Karl.Whitson at cmbtec.com>
>
> www.cmbtec.com <http://www.cmbtec.com>
>
> //TO OPEN A SUPPORT CASE EMAIL SUPPORT at CMBTEC.CO 
> <mailto:SUPPORT at CMBTEC.COM>M <mailto:SUPPORT at CMBTEC.COM>//// OR CALL// 
> (508) 422-8429
>
>  
>
> ------------------------------------------------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Steve G
> *Sent:* Sunday, September 10, 2006 10:58 PM
> *To:* ciscovoip
> *Subject:* [cisco-voip] Unity Call Transfer "wait while I transfer 
> your call"
>
>  
>
> Hello,
>
> Is it possible to disable the voice of Unity saying "wait while I 
> transfer your call" once the caller selects a transfer option?
>
>  
>
> If you can let me know the process to disable this, I would appreciate it.
>
>  
>
> Thank you,
>
> Steve
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

-- 
David Winter
Coleman Technologies
WestFL Network Engineer
777 South Harbour Island Boulevard
Suite 240
Tampa, Florida  33602
Phone: 813-233-2023 x333
Cell: 813-417-1830
FAX: 407-284-6670
AIM: mobofool
Y!: vt_fool
ICQ: 3563403
MSN :dwinter at vt.edu

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From tennille.spence at gsa.gov  Mon Sep 11 10:52:06 2006
From: tennille.spence at gsa.gov (tennille.spence at gsa.gov)
Date: Mon, 11 Sep 2006 09:52:06 -0500
Subject: [cisco-voip] Assistance with restoring a unity 3.1 voicemail
 account
In-Reply-To: <20060910021657.34923.qmail@web37212.mail.mud.yahoo.com>
Message-ID: <OF65E4D154.185AE8E7-ON862571E6.00518190-862571E6.0051ACA6@gsa.gov>


Unfortunately the unity account and the exchange account were removed. All
that remains is the recorded name wav file and an active directory account.
I have the pst file from a previous backup and was hoping I could somehow
recreate the account from that.


Tennille Spence
IT Business Liaison
PBS Technology Branch
816-823-1357 Office
Tennille Spence
www.gsa.gov/r6


"To be without some of the things you want is an indispensable part of
happiness." - Bertrand Russell


                                                                           
                                                                           
             "Erick Bergquist"                                             
             <erickbe at yahoo.co                                          To 
             m>                        tennille.spence at gsa.gov,            
                                       cisco-voip at puck.nether.net          
             09/09/2006 09:16                                           cc 
             PM                                                            
                                                                   Subject 
                                       Re: [cisco-voip] Assistance with    
             Please respond to         restoring a unity 3.1 voicemail     
             "Erick Bergquist"         account                             
             <erickbe at yahoo.co                                             
                    m>                                                     
                                                                           
                                                                           
                                                                           
                                                                           




Is the mail account still on the mailstore, or was that also removed?

----- Original Message ----
From: tennille.spence at gsa.gov
To: cisco-voip at puck.nether.net
Sent: Saturday, September 9, 2006 12:04:18 PM
Subject: [cisco-voip] Assistance with restoring a unity 3.1 voicemail
account


Does anyone have any guidance on restoring messages for a single unity
voicemail account fro unity 3.1? The account was removed from the system
accidentally. I do have a system backup using DIRT but I have never
restored from it before and wasn't sure if I could simply select a single
voicemail to restore or if it would restore everyone's account and
messages.

Tennille Spence
IT Business Liaison
PBS Technology Branch
816-823-1357 Office
Tennille Spence
www.gsa.gov/r6


"To be without some of the things you want is an indispensable part of
happiness." - Bertrand Russell


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









From lelio at uoguelph.ca  Mon Sep 11 10:56:10 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 11 Sep 2006 10:56:10 -0400
Subject: [cisco-voip] requirements for WAN link with centralized deployment
Message-ID: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>

Just wondering if there are posted requirements for WAN links with centralized deployments. I know that I've read somewhere about "if you want to have a offsite subscriber you have to have Xms with X delay, etc. etc. etc." just wondering if there is something similar when you are hanging a phone on the other end and it is talking to other phones and subscribers. 

I recall a small app you could run on a router that would do some tests.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
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From jneiberger at gmail.com  Mon Sep 11 11:06:49 2006
From: jneiberger at gmail.com (John Neiberger)
Date: Mon, 11 Sep 2006 09:06:49 -0600
Subject: [cisco-voip] requirements for WAN link with centralized
	deployment
In-Reply-To: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>
References: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>
Message-ID: <547ad0fe0609110806n5a0f6ee4i1d54a780737a2645@mail.gmail.com>

> Just wondering if there are posted requirements for WAN links with
> centralized deployments. I know that I've read somewhere about "if you want
> to have a offsite subscriber you have to have Xms with X delay, etc. etc.
> etc." just wondering if there is something similar when you are hanging a
> phone on the other end and it is talking to other phones and subscribers.
>
> I recall a small app you could run on a router that would do some tests.

The phones aren't a big deal. As long as your one-way delay isn't over
150 ms or so, you should be fine. As I recall, the maximum
*round-trip* delay over a WAN for a subscriber is 40 ms, which is
pretty stringent.

John

From lelio at uoguelph.ca  Mon Sep 11 11:12:20 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 11 Sep 2006 11:12:20 -0400
Subject: [cisco-voip] requirements for WAN link with centralized
	deployment
References: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>
	<547ad0fe0609110806n5a0f6ee4i1d54a780737a2645@mail.gmail.com>
Message-ID: <010701c6d5b4$ad355160$47196883@cfs.uoguelph.ca>

Do you know if this is written down anywhere? We are writing up specs for our remote sites and I'd like to include a reference where possible.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: John Neiberger 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Monday, September 11, 2006 11:06 AM
  Subject: Re: [cisco-voip] requirements for WAN link with centralized deployment


  > Just wondering if there are posted requirements for WAN links with
  > centralized deployments. I know that I've read somewhere about "if you want
  > to have a offsite subscriber you have to have Xms with X delay, etc. etc.
  > etc." just wondering if there is something similar when you are hanging a
  > phone on the other end and it is talking to other phones and subscribers.
  >
  > I recall a small app you could run on a router that would do some tests.

  The phones aren't a big deal. As long as your one-way delay isn't over
  150 ms or so, you should be fine. As I recall, the maximum
  *round-trip* delay over a WAN for a subscriber is 40 ms, which is
  pretty stringent.

  John
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From jneiberger at gmail.com  Mon Sep 11 11:23:30 2006
From: jneiberger at gmail.com (John Neiberger)
Date: Mon, 11 Sep 2006 09:23:30 -0600
Subject: [cisco-voip] requirements for WAN link with centralized
	deployment
In-Reply-To: <010701c6d5b4$ad355160$47196883@cfs.uoguelph.ca>
References: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>
	<547ad0fe0609110806n5a0f6ee4i1d54a780737a2645@mail.gmail.com>
	<010701c6d5b4$ad355160$47196883@cfs.uoguelph.ca>
Message-ID: <547ad0fe0609110823o5be1b2edy5a8b32f6756f15d8@mail.gmail.com>

Look in the Cisco IP Telephony Solution Reference Network Design
(SRND). It's in the section called "Clustering over the WAN". That's
probably a very good document for you to read through prior to making
any decisions.

On 9/11/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>
> Do you know if this is written down anywhere? We are writing up specs for
> our remote sites and I'd like to include a reference where possible.
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."
>
> ----- Original Message -----
> From: John Neiberger
> To: Lelio Fulgenzi
> Cc: cisco-voip at puck.nether.net
> Sent: Monday, September 11, 2006 11:06 AM
> Subject: Re: [cisco-voip] requirements for WAN link with centralized
> deployment
>
> > Just wondering if there are posted requirements for WAN links with
> > centralized deployments. I know that I've read somewhere about "if you
> want
> > to have a offsite subscriber you have to have Xms with X delay, etc. etc.
> > etc." just wondering if there is something similar when you are hanging a
> > phone on the other end and it is talking to other phones and subscribers.
> >
> > I recall a small app you could run on a router that would do some tests.
>
> The phones aren't a big deal. As long as your one-way delay isn't over
> 150 ms or so, you should be fine. As I recall, the maximum
> *round-trip* delay over a WAN for a subscriber is 40 ms, which is
> pretty stringent.
>
> John
>

From bob at smalltime.com  Mon Sep 11 11:26:02 2006
From: bob at smalltime.com (Robert Kulagowski)
Date: Mon, 11 Sep 2006 10:26:02 -0500
Subject: [cisco-voip] requirements for WAN link with
	centralized	deployment
In-Reply-To: <010701c6d5b4$ad355160$47196883@cfs.uoguelph.ca>
References: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>	<547ad0fe0609110806n5a0f6ee4i1d54a780737a2645@mail.gmail.com>
	<010701c6d5b4$ad355160$47196883@cfs.uoguelph.ca>
Message-ID: <4505800A.2070107@smalltime.com>

Lelio Fulgenzi wrote:
> Do you know if this is written down anywhere? We are writing up specs 
> for our remote sites and I'd like to include a reference where possible.

Page 2-17 of the SRND is your friend.  Heck, the entire SRND is your friend!


From lelio at uoguelph.ca  Mon Sep 11 11:27:06 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 11 Sep 2006 11:27:06 -0400
Subject: [cisco-voip] requirements for WAN link with centralized
	deployment
References: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>
	<547ad0fe0609110806n5a0f6ee4i1d54a780737a2645@mail.gmail.com>
	<010701c6d5b4$ad355160$47196883@cfs.uoguelph.ca>
	<547ad0fe0609110823o5be1b2edy5a8b32f6756f15d8@mail.gmail.com>
Message-ID: <012c01c6d5b6$bd4ef3b0$47196883@cfs.uoguelph.ca>

Thanks. I should have said that I am reading through that as we speak and was looking for anything that might not be included there. 

I could have sworn there was a command line utility that you can run on the router that sends data back and forth to a stub on another router and it tells you if it passes voice quality link testing or something like that. I've never seen that in the SRND.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: John Neiberger 
  To: Lelio Fulgenzi 
  Cc: cisco-voip at puck.nether.net 
  Sent: Monday, September 11, 2006 11:23 AM
  Subject: Re: [cisco-voip] requirements for WAN link with centralized deployment


  Look in the Cisco IP Telephony Solution Reference Network Design
  (SRND). It's in the section called "Clustering over the WAN". That's
  probably a very good document for you to read through prior to making
  any decisions.

  On 9/11/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  >
  >
  > Do you know if this is written down anywhere? We are writing up specs for
  > our remote sites and I'd like to include a reference where possible.
  >
  >
  > --------------------------------------------------------------------------------
  > Lelio Fulgenzi, B.A.
  > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  > "I can eat fifty eggs." "Nobody can eat fifty eggs."
  >
  > ----- Original Message -----
  > From: John Neiberger
  > To: Lelio Fulgenzi
  > Cc: cisco-voip at puck.nether.net
  > Sent: Monday, September 11, 2006 11:06 AM
  > Subject: Re: [cisco-voip] requirements for WAN link with centralized
  > deployment
  >
  > > Just wondering if there are posted requirements for WAN links with
  > > centralized deployments. I know that I've read somewhere about "if you
  > want
  > > to have a offsite subscriber you have to have Xms with X delay, etc. etc.
  > > etc." just wondering if there is something similar when you are hanging a
  > > phone on the other end and it is talking to other phones and subscribers.
  > >
  > > I recall a small app you could run on a router that would do some tests.
  >
  > The phones aren't a big deal. As long as your one-way delay isn't over
  > 150 ms or so, you should be fine. As I recall, the maximum
  > *round-trip* delay over a WAN for a subscriber is 40 ms, which is
  > pretty stringent.
  >
  > John
  >
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From AFrutin at apcapital.com  Mon Sep 11 11:42:28 2006
From: AFrutin at apcapital.com (Frutin, Amy)
Date: Mon, 11 Sep 2006 11:42:28 -0400
Subject: [cisco-voip] simple LDAP query tools
Message-ID: <AB0A35F2486F81468971AA68558BE71904BE4907@smieln01winm01.micoanet.com>

I installed this application
http://www.softerra.com/portfolio_ldap-browser.htm on my desktop awhile
ago for LDAP searches; didn't use it very extensively, but if I remember
correctly it worked great (and it's free).

 

Link to the d/l: http://download.softerra.com/files/ldapbrowser26.msi.


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

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From manoj.kalpage at gmail.com  Mon Sep 11 11:44:44 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Tue, 12 Sep 2006 00:44:44 +0900
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall
References: <01C8F21F30DE9740B53FD2EF05C3A8FC56F55E@SVCEXCH01.firm.fenwick.llp>
	<33AFDCDE-4C00-43F3-82E3-47D6423A4B2F@cisco.com>
Message-ID: <012301c6d5b9$37acdee0$6500a8c0@computec.lk>

Wes,
Thank You for the reply. Yes I have NAT on my PIX. Regarding DNS fixup, do i need additional settings on my PIX except the fixup protocol dns maximum-length 512 ? 

Best Regards,
Manoj



  ----- Original Message ----- 
  From: Wes Sisk 
  To: Manoj Kalpage 
  Cc: ciscovoip Voip ; Stu Packett 
  Sent: Sunday, September 10, 2006 10:28 PM
  Subject: Re: [cisco-voip] Internet IP phone connect through PIX Firewall


  Manoj,


  Are you doing NAT on your PIX?  If so, you will need special CM+PIX config.


  phones download SEP<mac>.cnf.xml from TFTP server.  Inside this XML file is a listing of which CM servers the phone should register to using SCCP.  This file also tells the phone what TCP port to use for the SCCP communication.  The CM servers are listed in this by name or IP address based on how your CM is configured under system->server.


  For NAT traversal, your CM will have to be configured using host names.  Your PIX will have to do DNS fixup.  Your phone will receive the name cm1.manoj.com.  the phone must do DNS lookup on this and receive the external address of your CM server.


  Otherwise, you must use a valid internet address for the IP of your CM server.


  /Wes

  On Sep 9, 2006, at 3:41 PM, Stu Packett wrote:


  Sorry, I have never tried without the VPN.  I thought best practice was to use the VPN because it was not advised to put the CCM on the public internet.  If you do get your config working, I'd like to get a copy of your config just for reference.  Thanks.



------------------------------------------------------------------------------
  From: Manoj Kalpage [mailto:manoj.kalpage at gmail.com] 
  Sent: Saturday, September 09, 2006 12:20 AM
  To: Stu Packett
  Cc: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Internet IP phone connect through PIX Firewall


  Stu, 
  Thank you for the reply, I use windows 2003 DHCP server for my phones in LAN but I can get my outside phone connect to CCM through internet. Do you have IP phones connect to your CCM through internet without using VPN?  

  Thanks,
  Manoj


   
  On 9/9/06, Stu Packett <SPackett at fenwick.com> wrote: 
    Manoj:
    Is your PIX giving out DHCP addresses?  On my PIX 501, I have it setup as a DHCP server and these are my DHCP commands: 

    dhcpd address xxx.xxx.xxx.xxx
    dhcpd dns xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
    dhcpd wins xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
    dhcpd lease 36000 
    dhcpd ping_timeout 750
    dhcpd domain internaldomain.com
    dhcpd option 150 ip xxx.xxx.xxx.xxx <--TFTP address 
    dhcpd enable inside



----------------------------------------------------------------------------
    From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip-bounces at puck.nether.net] On Behalf Of Manoj Kalpage
    Sent: Friday, September 08, 2006 4:18 AM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] Internet IP phone connect through PIX Firewall

     
    Hi All,
    Does any one has configured PIX firewall to connect internet IP phones to Call Manager. I have configure firewall to open all the port which CCM need but still no luck. Bellow is the config of my PIX. Am i missing anything? 

    Here is the link I refered to open the TCP and UDP Ports

    http://www.cisco.com/application/pdf/en/us/guest/products/ps5820/c1693/ccmigration_09186a0080536eae.pdf

    Thank you in advance.
    Manoj

    :
    PIX Version 6.3(5)
    interface ethernet0 auto
    interface ethernet1 auto 
    nameif ethernet0 outside security0
    nameif ethernet1 inside security100
    enable password u2zabJUOK.TTL3K1 encrypted
    passwd 1P5CrRl.dL8Oe4k2 encrypted
    hostname PBXLPIX01
    domain-name pbxl.jp

    clock timezone JST 9
    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 pptp 1723 
    fixup protocol rsh 514
    fixup protocol rtsp 554
    fixup protocol sip 5060
    fixup protocol sip udp 5060
    fixup protocol skinny 2000
    fixup protocol smtp 25
    fixup protocol snmp 161 
    fixup protocol sqlnet 1521 
    fixup protocol tftp 69
    names
    object-group service outbound-tcp tcp
      port-object eq www
      port-object eq https
      port-object eq smtp
      port-object eq ftp
      port-object eq pop3 
      port-object eq imap4 
      port-object eq domain
      port-object eq 123
      port-object eq ssh
      port-object eq citrix-ica
    object-group service outbound-udp udp
      port-object eq domain
      port-object eq ntp 
    object-group service mail-inbound tcp 
      port-object eq www
      port-object eq https
      port-object eq smtp
    object-group service VoIP-udp udp
      port-object range 16384 32768
      port-object eq tftp
    object-group service VoIP-tcp tcp 
      port-object eq 3804 
      port-object eq 2443
      port-object eq 2000
      port-object eq www
      port-object eq 69
      port-object eq https

    access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group VoIP-tcp
    access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group VoIP-udp
    access-list 102 permit tcp 172.16.0.0 255.255.0.0 any object-group outbound-tcp
    access-list 102 permit udp 172.16.0.0 255.255.0.0 any object-group outbound-udp 
    access-list 101 permit tcp any host 210.81.12.195 object-group mail-inbound 
    access-list 101 permit tcp any host 210.81.12.196 object-group VoIP-tcp 
    access-list 101 permit udp any host 210.81.12.196 object-group VoIP-udp
    access-list 101 permit tcp any host 210.81.12.197 object-group VoIP-tcp
    access-list 101 permit udp any host 210.81.12.197 object-group VoIP-udp 

    pager lines 24
    logging on
    logging trap informational
    logging host inside 172.16.0.26 

    logging host inside 172.16.0.12

    icmp permit any unreachable outside
    icmp permit any outside
    mtu outside 1500
    mtu inside 1500
    ip address outside xxx.xxx.xxx.xxx 255.255.255.240

    ip address inside 172.16.0.2 255.255.0.0

    ip audit info action alarm
    ip audit attack action alarm 
    ip local pool pbxlpool 10.1.0.100-10.1.0.200

    pdm locationxxx.xxx.xxx.xxx 255.255.255.255 outside

    pdm history enable
    arp timeout 14400
    global (outside) 1 interface
    nat (inside) 0 access-list VPNREMOTE 

    nat (inside) 1 172.16.0.0 255.255.0.0 0 0
    static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask 255.255.255.255 0 1000
    static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask 255.255.255.255 0 1000
    static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask 255.255.255.255 0 1000

    access-group 101 in interface outside
    access-group 102 in interface inside

    route outside 0.0.0.0 0.0.0.0 210.81.12.193 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 sip-disconnect 0:02:00 sip-invite 0:03:00 
    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

    aaa authentication ssh console LOCAL


    http 172.16.0.12 255.255.255.255 inside

    snmp-server host inside 172.16.0.12

    snmp-server location pbxl-pix-datacentre

    snmp-server community pbxl
    snmp-server enable traps
    floodguard enable


    telnet 172.16.0.0 255.255.0.0 inside
    telnet 192.168.0.0 255.255.255.0 inside
    telnet timeout 60 
    ssh 210.101.94.211 255.255.255.255 outside
    ssh 0.0.0.0 0.0.0.0 outside
    ssh 172.16.0.12 255.255.255.255 inside
    ssh 172.16.0.0 255.255.0.0 inside
    ssh 192.168.1.0 255.255.255.0 inside

    ssh timeout 60
    console timeout 0
    PBXLPIX01(config)#
    PBXLPIX01(config)#





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

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From BRiley at jackhenry.com  Mon Sep 11 12:19:38 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Mon, 11 Sep 2006 11:19:38 -0500
Subject: [cisco-voip] IPCC BARS compatibility
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002FCA74D@smoxchg.jhacorp.com>

 

Can someone send me the link on what version of BARS is compatible with
IPCC 4.0.4? I have BARS 4.10 and when I put it on the IPCC server none
of the CRS services would start. After I uninstalled BARS the services
started back up. I was going to try BARS 4.11 but I want to make sure
this does not happen again.  Thanks.

 

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 Scott.Voll at wesd.org  Mon Sep 11 12:39:24 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 11 Sep 2006 09:39:24 -0700
Subject: [cisco-voip] IPCC BARS compatibility
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C80CA@Apollo.wesd.org>

I just looked at this last weekend and according to the metric only
4.0.10 is supported on IPCCx.

 

Scott

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley III
Sent: Monday, September 11, 2006 9:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC BARS compatibility

 

 

Can someone send me the link on what version of BARS is compatible with
IPCC 4.0.4? I have BARS 4.10 and when I put it on the IPCC server none
of the CRS services would start. After I uninstalled BARS the services
started back up. I was going to try BARS 4.11 but I want to make sure
this does not happen again.  Thanks.

 

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
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From BRiley at jackhenry.com  Mon Sep 11 13:53:44 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Mon, 11 Sep 2006 12:53:44 -0500
Subject: [cisco-voip] IPCC BARS compatibility
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002FCA82A@smoxchg.jhacorp.com>

Well that's too bad considering 4.10 is what caused the services to
stop. I will give it another try with 4.10 and see what happens. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 11, 2006 11:39 AM
To: Bill Riley III; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC BARS compatibility

 

I just looked at this last weekend and according to the metric only
4.0.10 is supported on IPCCx.

 

Scott

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley III
Sent: Monday, September 11, 2006 9:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC BARS compatibility

 

 

Can someone send me the link on what version of BARS is compatible with
IPCC 4.0.4? I have BARS 4.10 and when I put it on the IPCC server none
of the CRS services would start. After I uninstalled BARS the services
started back up. I was going to try BARS 4.11 but I want to make sure
this does not happen again.  Thanks.

 

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.
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
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From wsisk at cisco.com  Mon Sep 11 14:04:10 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 11 Sep 2006 14:04:10 -0400
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall
In-Reply-To: <012301c6d5b9$37acdee0$6500a8c0@computec.lk>
References: <01C8F21F30DE9740B53FD2EF05C3A8FC56F55E@SVCEXCH01.firm.fenwick.llp>
	<33AFDCDE-4C00-43F3-82E3-47D6423A4B2F@cisco.com>
	<012301c6d5b9$37acdee0$6500a8c0@computec.lk>
Message-ID: <4505A51A.10706@cisco.com>

I believe this is what you need:
http://www.cisco.com/en/US/partner/products/sw/secursw/ps2120/products_configuration_guide_chapter09186a008017278b.html#wp1063720

and this, but with inside/outsider reversed:
http://www.ciscotaccc.com/kaidara-advisor/security/showcase?case=K72511039

/Wes

Manoj Kalpage wrote:
> Wes,
> Thank You for the reply. Yes I have NAT on my PIX. Regarding DNS fixup, 
> do i need additional settings on my PIX except the fixup protocol dns 
> maximum-length 512 ?
>  
> Best Regards,
> Manoj
>  
> 
>     ----- Original Message -----
>     *From:* Wes Sisk <mailto:wsisk at cisco.com>
>     *To:* Manoj Kalpage <mailto:manoj.kalpage at gmail.com>
>     *Cc:* ciscovoip Voip <mailto:cisco-voip at puck.nether.net> ; Stu
>     Packett <mailto:SPackett at fenwick.com>
>     *Sent:* Sunday, September 10, 2006 10:28 PM
>     *Subject:* Re: [cisco-voip] Internet IP phone connect through PIX
>     Firewall
> 
>     Manoj,
> 
>     Are you doing NAT on your PIX?  If so, you will need special CM+PIX
>     config.
> 
>     phones download SEP<mac>.cnf.xml from TFTP server.  Inside this XML
>     file is a listing of which CM servers the phone should register to
>     using SCCP.  This file also tells the phone what TCP port to use for
>     the SCCP communication.  The CM servers are listed in this by name
>     or IP address based on how your CM is configured under system->server.
> 
>     For NAT traversal, your CM will have to be configured using host
>     names.  Your PIX will have to do DNS fixup.  Your phone will receive
>     the name cm1.manoj.com.  the phone must do DNS lookup on this and
>     receive the external address of your CM server.
> 
>     Otherwise, you must use a valid internet address for the IP of your
>     CM server.
> 
>     /Wes
>     On Sep 9, 2006, at 3:41 PM, Stu Packett wrote:
> 
>     Sorry, I have never tried without the VPN.  I thought best practice
>     was to use the VPN because it was not advised to put the CCM on the
>     public internet.  If you do get your config working, I'd like to get
>     a copy of your config just for reference.  Thanks.
> 
>     ------------------------------------------------------------------------
>     *From:* Manoj Kalpage [mailto:manoj.kalpage at gmail.com]
>     *Sent:* Saturday, September 09, 2006 12:20 AM
>     *To:* Stu Packett
>     *Cc:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     *Subject:* Re: [cisco-voip] Internet IP phone connect through PIX
>     Firewall
> 
>     Stu,
>     Thank you for the reply, I use windows 2003 DHCP server for my
>     phones in LAN but I can get my outside phone connect to CCM through
>     internet. Do you have IP phones connect to your CCM through internet
>     without using VPN? 
>      
>     Thanks,
>     Manoj
> 
> 
>      
>     On 9/9/06, *Stu Packett* <SPackett at fenwick.com
>     <mailto:SPackett at fenwick.com>> wrote:
> 
>         Manoj:
>         Is your PIX giving out DHCP addresses?  On my PIX 501, I have it
>         setup as a DHCP server and these are my DHCP commands:
>          
>         dhcpd address xxx.xxx.xxx.xxx
>         dhcpd dns xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
>         dhcpd wins xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
>         dhcpd lease 36000
>         dhcpd ping_timeout 750
>         dhcpd domain internaldomain.com <http://internaldomain.com/>
>         dhcpd option 150 ip xxx.xxx.xxx.xxx <--TFTP address
>         dhcpd enable inside
> 
>         ------------------------------------------------------------------------
>         *From:* cisco-voip-bounces at puck.nether.net
>         <mailto:cisco-voip-bounces at puck.nether.net> [mailto:
>         cisco-voip-bounces at puck.nether.net
>         <mailto:cisco-voip-bounces at puck.nether.net>] *On Behalf Of
>         *Manoj Kalpage
>         *Sent:* Friday, September 08, 2006 4:18 AM
>         *To:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>         *Subject:* [cisco-voip] Internet IP phone connect through PIX
>         Firewall
> 
>          
>         Hi All,
>         Does any one has configured PIX firewall to connect internet IP
>         phones to Call Manager. I have configure firewall to open all
>         the port which CCM need but still no luck. Bellow is the config
>         of my PIX. Am i missing anything?
> 
>         Here is the link I refered to open the TCP and UDP Ports
> 
>         http://www.cisco.com/application/pdf/en/us/guest/products/ps5820/c1693/ccmigration_09186a0080536eae.pdf
> 
>         Thank you in advance.
>         Manoj
> 
>         :
>         PIX Version 6.3(5)
>         interface ethernet0 auto
>         interface ethernet1 auto
>         nameif ethernet0 outside security0
>         nameif ethernet1 inside security100
>         enable password u2zabJUOK.TTL3K1 encrypted
>         passwd 1P5CrRl.dL8Oe4k2 encrypted
>         hostname PBXLPIX01
>         domain-name pbxl.jp <http://pbxl.jp/>
>         clock timezone JST 9
>         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 pptp 1723
>         fixup protocol rsh 514
>         fixup protocol rtsp 554
>         fixup protocol sip 5060
>         fixup protocol sip udp 5060
>         fixup protocol skinny 2000
>         fixup protocol smtp 25
>         fixup protocol snmp 161
>         fixup protocol sqlnet 1521
>         fixup protocol tftp 69
>         names
>         object-group service outbound-tcp tcp
>           port-object eq www
>           port-object eq https
>           port-object eq smtp
>           port-object eq ftp
>           port-object eq pop3
>           port-object eq imap4
>           port-object eq domain
>           port-object eq 123
>           port-object eq ssh
>           port-object eq citrix-ica
>         object-group service outbound-udp udp
>           port-object eq domain
>           port-object eq ntp
>         object-group service mail-inbound tcp
>           port-object eq www
>           port-object eq https
>           port-object eq smtp
>         object-group service VoIP-udp udp
>           port-object range 16384 32768
>           port-object eq tftp
>         object-group service VoIP-tcp tcp
>           port-object eq 3804
>           port-object eq 2443
>           port-object eq 2000
>           port-object eq www
>           port-object eq 69
>           port-object eq https
>         access-list 102 permit tcp 172.16.0.0 <http://172.16.0.0/>
>         255.255.0.0 <http://255.255.0.0/> any object-group VoIP-tcp
>         access-list 102 permit udp 172.16.0.0 <http://172.16.0.0/>
>         255.255.0.0 <http://255.255.0.0/> any object-group VoIP-udp
>         access-list 102 permit tcp 172.16.0.0 <http://172.16.0.0/>
>         255.255.0.0 <http://255.255.0.0/> any object-group outbound-tcp
>         access-list 102 permit udp 172.16.0.0 <http://172.16.0.0/>
>         255.255.0.0 <http://255.255.0.0/> any object-group outbound-udp
>         access-list 101 permit tcp any host 210.81.12.195
>         <http://210.81.12.195/> object-group mail-inbound
>         access-list 101 permit tcp any host 210.81.12.196
>         <http://210.81.12.196/> object-group VoIP-tcp
>         access-list 101 permit udp any host 210.81.12.196
>         <http://210.81.12.196/> object-group VoIP-udp
>         access-list 101 permit tcp any host 210.81.12.197
>         <http://210.81.12.197/> object-group VoIP-tcp
>         access-list 101 permit udp any host 210.81.12.197
>         <http://210.81.12.197/> object-group VoIP-udp
> 
>         pager lines 24
>         logging on
>         logging trap informational
>         logging host inside 172.16.0.26 <http://172.16.0.26/>
>         logging host inside 172.16.0.12 <http://172.16.0.12/>
>         icmp permit any unreachable outside
>         icmp permit any outside
>         mtu outside 1500
>         mtu inside 1500
>         ip address outside xxx.xxx.xxx.xxx 255.255.255.240
>         <http://255.255.255.240/>
>         ip address inside 172.16.0.2 <http://172.16.0.2/> 255.255.0.0
>         <http://255.255.0.0/>
> 
>         ip audit info action alarm
>         ip audit attack action alarm
>         ip local pool pbxlpool 10.1.0.100-10.1.0.200
>         pdm locationxxx.xxx.xxx.xxx 255.255.255.255
>         <http://255.255.255.255/> outside
> 
>         pdm history enable
>         arp timeout 14400
>         global (outside) 1 interface
>         nat (inside) 0 access-list VPNREMOTE
>         nat (inside) 1 172.16.0.0 <http://172.16.0.0/> 255.255.0.0
>         <http://255.255.0.0/> 0 0
>         static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
>         255.255.255.255 <http://255.255.255.255/>0 1000
>         static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
>         255.255.255.255 <http://255.255.255.255/> 0 1000
>         static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
>         255.255.255.255 <http://255.255.255.255/> 0 1000
> 
>         access-group 101 in interface outside
>         access-group 102 in interface inside
>         route outside 0.0.0.0 <http://0.0.0.0/> 0.0.0.0
>         <http://0.0.0.0/> 210.81.12.193 <http://210.81.12.193/> 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 sip-disconnect 0:02:00 sip-invite 0:03:00
>         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
> 
>         aaa authentication ssh console LOCAL
> 
>         http 172.16.0.12 <http://172.16.0.12/> 255.255.255.255
>         <http://255.255.255.255/> inside
> 
>         snmp-server host inside 172.16.0.12 <http://172.16.0.12/>
>         snmp-server location pbxl-pix-datacentre
> 
>         snmp-server community pbxl
>         snmp-server enable traps
>         floodguard enable
> 
>         telnet 172.16.0.0 <http://172.16.0.0/> 255.255.0.0
>         <http://255.255.0.0/> inside
>         telnet 192.168.0.0 <http://192.168.0.0/> 255.255.255.0
>         <http://255.255.255.0/> inside
>         telnet timeout 60
>         ssh 210.101.94.211 <http://210.101.94.211/> 255.255.255.255
>         <http://255.255.255.255/> outside
>         ssh 0.0.0.0 <http://0.0.0.0/> 0.0.0.0 <http://0.0.0.0/> outside
>         ssh 172.16.0.12 <http://172.16.0.12/> 255.255.255.255
>         <http://255.255.255.255/> inside
>         ssh 172.16.0.0 <http://172.16.0.0/> 255.255.0.0
>         <http://255.255.0.0/> inside
>         ssh 192.168.1.0 <http://192.168.1.0/>255.255.255.0
>         <http://255.255.255.0/> inside
> 
>         ssh timeout 60
>         console timeout 0
>         PBXLPIX01(config)#
>         PBXLPIX01(config)#
> 
> 
> 
>     _______________________________________________
>     cisco-voip mailing list
>     cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>     https://puck.nether.net/mailman/listinfo/cisco-voip
> 

From Scott.Voll at wesd.org  Mon Sep 11 14:04:19 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 11 Sep 2006 11:04:19 -0700
Subject: [cisco-voip] IPCC BARS compatibility
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C8115@Apollo.wesd.org>

Are you sure it's not a CSA issue?  I had a problem with the CSA client
and upgraded it to solve the problem.

 

Scott

 

PS.  I have BARS 4.0.10 on my IPCCx 4.0.1sr1 build 055 working fine.

 

________________________________

From: Bill Riley III [mailto:BRiley at jackhenry.com] 
Sent: Monday, September 11, 2006 10:54 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC BARS compatibility

 

Well that's too bad considering 4.10 is what caused the services to
stop. I will give it another try with 4.10 and see what happens. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 11, 2006 11:39 AM
To: Bill Riley III; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC BARS compatibility

 

I just looked at this last weekend and according to the metric only
4.0.10 is supported on IPCCx.

 

Scott

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley III
Sent: Monday, September 11, 2006 9:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC BARS compatibility

 

 

Can someone send me the link on what version of BARS is compatible with
IPCC 4.0.4? I have BARS 4.10 and when I put it on the IPCC server none
of the CRS services would start. After I uninstalled BARS the services
started back up. I was going to try BARS 4.11 but I want to make sure
this does not happen again.  Thanks.

 

NOTICE: This electronic mail message and any files transmitted with it
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or distribution is strictly prohibited. If you have received this
message in error, please immediately
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are intended exclusively
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From wsisk at cisco.com  Mon Sep 11 14:06:59 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 11 Sep 2006 14:06:59 -0400
Subject: [cisco-voip] requirements for WAN link with centralized
	deployment
In-Reply-To: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>
References: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>
Message-ID: <4505A5C3.5050109@cisco.com>

http://www.cisco.com/go/srnd is your friend

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guide_chapter09186a00806e8c27.html#wp1045846

/Wes

Lelio Fulgenzi wrote:
> Just wondering if there are posted requirements for WAN links with 
> centralized deployments. I know that I've read somewhere about "if you 
> want to have a offsite subscriber you have to have Xms with X delay, 
> etc. etc. etc." just wondering if there is something similar when you 
> are hanging a phone on the other end and it is talking to other phones 
> and subscribers.
>  
> I recall a small app you could run on a router that would do some tests.
>  
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From BRiley at jackhenry.com  Mon Sep 11 14:09:34 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Mon, 11 Sep 2006 13:09:34 -0500
Subject: [cisco-voip] IPCC BARS compatibility
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002FCA85A@smoxchg.jhacorp.com>

I ruled out CSA already. I also have BARS 4.0.10 on a IPXXx 4.0.2 and it
is running just fine as well. I will give it another try and see what
happens. 

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, September 11, 2006 1:04 PM
To: Bill Riley III; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC BARS compatibility

 

Are you sure it's not a CSA issue?  I had a problem with the CSA client
and upgraded it to solve the problem.

 

Scott

 

PS.  I have BARS 4.0.10 on my IPCCx 4.0.1sr1 build 055 working fine.

 

________________________________

From: Bill Riley III [mailto:BRiley at jackhenry.com] 
Sent: Monday, September 11, 2006 10:54 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC BARS compatibility

 

Well that's too bad considering 4.10 is what caused the services to
stop. I will give it another try with 4.10 and see what happens. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 11, 2006 11:39 AM
To: Bill Riley III; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC BARS compatibility

 

I just looked at this last weekend and according to the metric only
4.0.10 is supported on IPCCx.

 

Scott

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley III
Sent: Monday, September 11, 2006 9:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC BARS compatibility

 

 

Can someone send me the link on what version of BARS is compatible with
IPCC 4.0.4? I have BARS 4.10 and when I put it on the IPCC server none
of the CRS services would start. After I uninstalled BARS the services
started back up. I was going to try BARS 4.11 but I want to make sure
this does not happen again.  Thanks.

 

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.
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.
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 ealeatherman at gmail.com  Mon Sep 11 15:19:06 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 11 Sep 2006 15:19:06 -0400
Subject: [cisco-voip] requirements for WAN link with centralized
	deployment
In-Reply-To: <012c01c6d5b6$bd4ef3b0$47196883@cfs.uoguelph.ca>
References: <00ed01c6d5b2$6b5b2c30$47196883@cfs.uoguelph.ca>
	<547ad0fe0609110806n5a0f6ee4i1d54a780737a2645@mail.gmail.com>
	<010701c6d5b4$ad355160$47196883@cfs.uoguelph.ca>
	<547ad0fe0609110823o5be1b2edy5a8b32f6756f15d8@mail.gmail.com>
	<012c01c6d5b6$bd4ef3b0$47196883@cfs.uoguelph.ca>
Message-ID: <94a1afde0609111219v23667622ja399fa00581c231f@mail.gmail.com>

Lelio, I think you are looking for IP SLAs feature on IOS, its a particular
feature set that can run tests for delay, packet loss, jitter, etc. I've
never used it personally but I remember hearing about it a few times. If you
use it let me know what you think :)

see if this helps: http://www.cisco.com/go/ipsla

Ed

On 9/11/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  Thanks. I should have said that I am reading through that as we speak and
> was looking for anything that might not be included there.
>
> I could have sworn there was a command line utility that you can run on
> the router that sends data back and forth to a stub on another router and it
> tells you if it passes voice quality link testing or something like that.
> I've never seen that in the SRND.
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."
>
> ----- Original Message -----
> *From:* John Neiberger <jneiberger at gmail.com>
> *To:* Lelio Fulgenzi <lelio at uoguelph.ca>
> *Cc:* cisco-voip at puck.nether.net
> *Sent:* Monday, September 11, 2006 11:23 AM
> *Subject:* Re: [cisco-voip] requirements for WAN link with centralized
> deployment
>
> Look in the Cisco IP Telephony Solution Reference Network Design
> (SRND). It's in the section called "Clustering over the WAN". That's
> probably a very good document for you to read through prior to making
> any decisions.
>
> On 9/11/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> >
> >
> > Do you know if this is written down anywhere? We are writing up specs
> for
> > our remote sites and I'd like to include a reference where possible.
> >
> >
> >
> --------------------------------------------------------------------------------
> > Lelio Fulgenzi, B.A.
> > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> > "I can eat fifty eggs." "Nobody can eat fifty eggs."
> >
> > ----- Original Message -----
> > From: John Neiberger
> > To: Lelio Fulgenzi
> > Cc: cisco-voip at puck.nether.net
> > Sent: Monday, September 11, 2006 11:06 AM
> > Subject: Re: [cisco-voip] requirements for WAN link with centralized
> > deployment
> >
> > > Just wondering if there are posted requirements for WAN links with
> > > centralized deployments. I know that I've read somewhere about "if you
> > want
> > > to have a offsite subscriber you have to have Xms with X delay, etc.
> etc.
> > > etc." just wondering if there is something similar when you are
> hanging a
> > > phone on the other end and it is talking to other phones and
> subscribers.
> > >
> > > I recall a small app you could run on a router that would do some
> tests.
> >
> > The phones aren't a big deal. As long as your one-way delay isn't over
> > 150 ms or so, you should be fine. As I recall, the maximum
> > *round-trip* delay over a WAN for a subscriber is 40 ms, which is
> > pretty stringent.
> >
> > John
> >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From Matt.Slaga at us.didata.com  Mon Sep 11 15:19:09 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 11 Sep 2006 15:19:09 -0400
Subject: [cisco-voip] IPCC BARS compatibility
Message-ID: <C1FE15183DA37645BC0633BC604E44F0031D171F@USNAEXCH.na.didata.local>

I've only seen this happen when CSA or AV was not shut down before the
BARS installation.

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley III
Sent: Monday, September 11, 2006 2:10 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC BARS compatibility

 

I ruled out CSA already. I also have BARS 4.0.10 on a IPXXx 4.0.2 and it
is running just fine as well. I will give it another try and see what
happens. 

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, September 11, 2006 1:04 PM
To: Bill Riley III; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC BARS compatibility

 

Are you sure it's not a CSA issue?  I had a problem with the CSA client
and upgraded it to solve the problem.

 

Scott

 

PS.  I have BARS 4.0.10 on my IPCCx 4.0.1sr1 build 055 working fine.

 

________________________________

From: Bill Riley III [mailto:BRiley at jackhenry.com] 
Sent: Monday, September 11, 2006 10:54 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC BARS compatibility

 

Well that's too bad considering 4.10 is what caused the services to
stop. I will give it another try with 4.10 and see what happens. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 11, 2006 11:39 AM
To: Bill Riley III; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC BARS compatibility

 

I just looked at this last weekend and according to the metric only
4.0.10 is supported on IPCCx.

 

Scott

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley III
Sent: Monday, September 11, 2006 9:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC BARS compatibility

 

 

Can someone send me the link on what version of BARS is compatible with
IPCC 4.0.4? I have BARS 4.10 and when I put it on the IPCC server none
of the CRS services would start. After I uninstalled BARS the services
started back up. I was going to try BARS 4.11 but I want to make sure
this does not happen again.  Thanks.

 

NOTICE: This electronic mail message and any files transmitted with it
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message in error, please immediately
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-----------------------------------------
Disclaimer:

This e-mail communication and any attachments may contain
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From baryonyx5 at gmail.com  Mon Sep 11 17:56:58 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Mon, 11 Sep 2006 16:56:58 -0500
Subject: [cisco-voip] call quality issues between gateway and Unity
Message-ID: <a749d0620609111456g272d78ebne39720e0a8a35ddb@mail.gmail.com>

Hi

I have a customer with a multi site centralized CCM 4.13 cluster.  The
remote sites are getting complaints of poor voice quality from customers
when being transferred to Unity (which is located in the central site) .
The callers cannot understand the greetings and describe them as choppy.
There are no problems with external calls transferred from the remote sites
to IP phones at the central site or for intersite calls.  It seems to be
limited to calls where the RTP stream is between a remote voice gateway and
Unity.  These calls are G.729 so Unity has to transcode the greetings on the
fly but if that were the problem I'd expect remote users to have the same
problem when calling into check their messages or leave messages for other
users.  Also, the Unity server is not very heavily used; it generally has no
more than 9 active calls at the most.

I suspect the gateway might be the problem.  It's a 2851 running MGCP.  I
guess I'd like to know if anyone agrees or disagrees with this assumption.
Also, I've had a lot of problems with MGCP code on the ISRs.  Can someone
recommend an IOS version that is stable for and MGCP gateway and doesn't
suffer from DSP firmware issues?

I should mention that Unity is version 4.04 sr1 and the gateway is running
12.4(5).

Thanks

-- 
kris seraphine
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From wsisk at cisco.com  Mon Sep 11 19:00:36 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 11 Sep 2006 19:00:36 -0400
Subject: [cisco-voip] call quality issues between gateway and Unity
In-Reply-To: <a749d0620609111456g272d78ebne39720e0a8a35ddb@mail.gmail.com>
References: <a749d0620609111456g272d78ebne39720e0a8a35ddb@mail.gmail.com>
Message-ID: <60299CCB-4272-4FB7-80D2-F6FA1871EF57@cisco.com>

Kris,

This sounds very similar to something we fixed long ago.  Take a look  
at 'sh voice call' to see if it's jitter:
http://www.cisco.com/en/US/tech/tk652/tk698/ 
technologies_tech_note09186a00800945df.shtml

CSCea35850 - Unity RTP stream has inherent jitter


If you are using non-default packetization periods watch out for
CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and  
G711/30ms

/Wes

On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:

Hi

I have a customer with a multi site centralized CCM 4.13 cluster.   
The remote sites are getting complaints of poor voice quality from  
customers when being transferred to Unity (which is located in the  
central site) .  The callers cannot understand the greetings and  
describe them as choppy.  There are no problems with external calls  
transferred from the remote sites to IP phones at the central site or  
for intersite calls.  It seems to be limited to calls where the RTP  
stream is between a remote voice gateway and Unity.  These calls are  
G.729 so Unity has to transcode the greetings on the fly but if that  
were the problem I'd expect remote users to have the same problem  
when calling into check their messages or leave messages for other  
users.  Also, the Unity server is not very heavily used; it generally  
has no more than 9 active calls at the most.

I suspect the gateway might be the problem.  It's a 2851 running  
MGCP.  I guess I'd like to know if anyone agrees or disagrees with  
this assumption.  Also, I've had a lot of problems with MGCP code on  
the ISRs.  Can someone recommend an IOS version that is stable for  
and MGCP gateway and doesn't suffer from DSP firmware issues?

I should mention that Unity is version 4.04 sr1 and the gateway is  
running 12.4(5).

Thanks

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


From erickbe at yahoo.com  Mon Sep 11 22:19:09 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon, 11 Sep 2006 19:19:09 -0700 (PDT)
Subject: [cisco-voip] Delayed audio-cutover with H.323 gateway
Message-ID: <20060912021909.82356.qmail@web37214.mail.mud.yahoo.com>

Hi,

I'm having a issue where an inbound call comes in FXO port and connection plar's to a shared DN across 5 IP Phones. The phone call gets answered fine, but the user on IP phone saids "Welcome to BlahBlah" and the outside caller doesn't hear the first time they say "Welcome To BlahBlah" and the IP Phone user repeats their greeting and they hear it the second time fine.

There are no errors on any interfaces that I can see, and the gateway has voice rtp send-recv enabled. Issue does not happen on calls over PRI, just the analog FXOs. I have disabled caller-id on the voice-port and also tried connection plar opx with no difference. Sometimes it works fine but rarely. 

CCM is version 4.1(3)sr3c and the gateway is a 1760 running 12.4(8a). 

A similar problem can happen when the call is parked or put on hold also, now and then.



From baryonyx5 at gmail.com  Mon Sep 11 22:40:19 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Mon, 11 Sep 2006 21:40:19 -0500
Subject: [cisco-voip] call quality issues between gateway and Unity
In-Reply-To: <60299CCB-4272-4FB7-80D2-F6FA1871EF57@cisco.com>
References: <a749d0620609111456g272d78ebne39720e0a8a35ddb@mail.gmail.com>
	<60299CCB-4272-4FB7-80D2-F6FA1871EF57@cisco.com>
Message-ID: <a749d0620609111940v646f9f65hbca957c451f34324@mail.gmail.com>

Thanks for the quick response.

That does sound like my problem although the solutions don't apply to me.

I haven't been able to catch an active call that has these symptoms but I
took a look at the last 100 calls via the show call hist voice command and
the values don't look to me to show much jitter.

There were no lost packets
Only a few calls had 2 or 3 late packets; one call had 7 late packets.
Only a few calls had non-zero values for the GapFilledWith... lines.

I might be interpreting the output wrong though.



On 9/11/06, Wes Sisk <wsisk at cisco.com> wrote:
>
> Kris,
>
> This sounds very similar to something we fixed long ago.  Take a look
> at 'sh voice call' to see if it's jitter:
> http://www.cisco.com/en/US/tech/tk652/tk698/
> technologies_tech_note09186a00800945df.shtml
>
> CSCea35850 - Unity RTP stream has inherent jitter
>
>
> If you are using non-default packetization periods watch out for
> CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and
> G711/30ms
>
> /Wes
>
> On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:
>
> Hi
>
> I have a customer with a multi site centralized CCM 4.13 cluster.
> The remote sites are getting complaints of poor voice quality from
> customers when being transferred to Unity (which is located in the
> central site) .  The callers cannot understand the greetings and
> describe them as choppy.  There are no problems with external calls
> transferred from the remote sites to IP phones at the central site or
> for intersite calls.  It seems to be limited to calls where the RTP
> stream is between a remote voice gateway and Unity.  These calls are
> G.729 so Unity has to transcode the greetings on the fly but if that
> were the problem I'd expect remote users to have the same problem
> when calling into check their messages or leave messages for other
> users.  Also, the Unity server is not very heavily used; it generally
> has no more than 9 active calls at the most.
>
> I suspect the gateway might be the problem.  It's a 2851 running
> MGCP.  I guess I'd like to know if anyone agrees or disagrees with
> this assumption.  Also, I've had a lot of problems with MGCP code on
> the ISRs.  Can someone recommend an IOS version that is stable for
> and MGCP gateway and doesn't suffer from DSP firmware issues?
>
> I should mention that Unity is version 4.04 sr1 and the gateway is
> running 12.4(5).
>
> Thanks
>
> --
> kris seraphine
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
kris seraphine
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From hansel.gunawan at multipolar.co.id  Mon Sep 11 23:09:47 2006
From: hansel.gunawan at multipolar.co.id (Hansel Gideon Gunawan)
Date: Tue, 12 Sep 2006 10:09:47 +0700
Subject: [cisco-voip] Intercluster trunk (gatekeeper controlled) behind
	ASA/NAT
Message-ID: <FCF1EA6D9A9506418E4DC24F01E0D14A05C65C92@mpcserver.multipolar.com>

Guys,

 

I'm trying to make an inter-cluster trunk using gatekeeper (tried h225
and intercluster trunk). Topology attached (I hope attaching pic is
permitted):



Gatekeeper registration and signaling seems to be ok, since the CCMs can
be seen on "show gatekeeper endpoint" and extension dialing from one
site to another resulted in the phone ringing. The thing is, the audio
won't got through. It looked like the h245 did not work properly.

 

The CCM is statically NAT'ed, and I already configured the ASAs to do
h323 inspection and permit any ip and icmp packets.

 

Anybody knows what's wrong? Tried this with intercluster trunk
non-gatekeeper controlled and worked all right.

 

Thanks in advance.

 

Regards,

 

Hansel Gideon Gunawan

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From zhaoye at 188.com  Tue Sep 12 04:51:20 2006
From: zhaoye at 188.com (=?gb2312?B?1dTsxw==?=)
Date: Tue, 12 Sep 2006 16:51:20 +0800 (CST)
Subject: [cisco-voip]  Monitor Historical PRI Usage
Message-ID: <45067508.00000B.21612@bj188app3.188.com>

I find some IOS version of router can use  OID 1.3.6.1.4.1.9.10.19.1.1.4.0  by mrtg ,but others not.
how can i do it ? it is impossible to change the IOS version of router in production, can you give some suggestion for me ,thanks
 
successfully:

Cisco 3660 c3660-js-mz.122-19b.bin
Cisco AS5300 c5300-js-mz.122-1a.bin
Cisco 3640 c3640-jsx-mz.122-27.bin
Cisco 3845 c3845-ipvoicek9-mz.124-1a.bin
Cisco 3640 c3640-is-mz.122-3.bin
Cisco 3745 c3745-is-mz.123-6.bin
cisco 3660 c3660-js-mz.122-19b.bin
cisco 3660 c3660-js-mz.122-19b.bin
Cisco 3745 c3745-js-mz.123-13a.bin

 
fail:
 
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 3845 c3845-ipvoice-mz.124-2.T.bin
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 2620 c2600-jk8s-mz.122-1a.bin
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 2821 c2800nm-spservicesk9-mz.123-11.T7.bin
 
best regards
zhaoye 2006-9-12
 
 
 
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From zhaoye at 188.com  Tue Sep 12 04:59:21 2006
From: zhaoye at 188.com (=?gb2312?B?1dTsxw==?=)
Date: Tue, 12 Sep 2006 16:59:21 +0800 (CST)
Subject: [cisco-voip]  VoIP Monitoring Solutions
Message-ID: <450676E9.00002B.17868@bj188app3.188.com>

   - Circuit Utilization - what application?      
   you can use solarwinds to do it .
                              
   - CDR Records - Best application for this?
   you can use Cisco ACS and SQl, I had done it successfully.

   - 

 
 
 
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From BRiley at jackhenry.com  Tue Sep 12 09:25:06 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Tue, 12 Sep 2006 08:25:06 -0500
Subject: [cisco-voip] IPCC BARS compatibility
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002FCAB49@smoxchg.jhacorp.com>

I reinstalled last night and everything is working. Thanks for
everyone's help.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Matt Slaga (US)
Sent: Monday, September 11, 2006 2:19 PM
To: Bill Riley III; Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC BARS compatibility

 

I've only seen this happen when CSA or AV was not shut down before the
BARS installation.

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley III
Sent: Monday, September 11, 2006 2:10 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC BARS compatibility

 

I ruled out CSA already. I also have BARS 4.0.10 on a IPXXx 4.0.2 and it
is running just fine as well. I will give it another try and see what
happens. 

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, September 11, 2006 1:04 PM
To: Bill Riley III; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC BARS compatibility

 

Are you sure it's not a CSA issue?  I had a problem with the CSA client
and upgraded it to solve the problem.

 

Scott

 

PS.  I have BARS 4.0.10 on my IPCCx 4.0.1sr1 build 055 working fine.

 

________________________________

From: Bill Riley III [mailto:BRiley at jackhenry.com] 
Sent: Monday, September 11, 2006 10:54 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC BARS compatibility

 

Well that's too bad considering 4.10 is what caused the services to
stop. I will give it another try with 4.10 and see what happens. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 11, 2006 11:39 AM
To: Bill Riley III; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC BARS compatibility

 

I just looked at this last weekend and according to the metric only
4.0.10 is supported on IPCCx.

 

Scott

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.
pdf

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Riley III
Sent: Monday, September 11, 2006 9:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC BARS compatibility

 

 

Can someone send me the link on what version of BARS is compatible with
IPCC 4.0.4? I have BARS 4.10 and when I put it on the IPCC server none
of the CRS services would start. After I uninstalled BARS the services
started back up. I was going to try BARS 4.11 but I want to make sure
this does not happen again.  Thanks.

 

NOTICE: This electronic mail message and any files transmitted with it
are intended exclusively
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together with any attachment, may contain confidential and/or privileged
information. Any unauthorized review, use, printing, saving, copying,
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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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information. Any unauthorized review, use, printing, saving, copying,
disclosure 
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message in error, please immediately
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________________________________



Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the
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this communication in error and that any use or reproduction of
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NOTICE: This electronic mail message and any files transmitted with it are intended exclusively
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From wsisk at cisco.com  Tue Sep 12 09:31:19 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 12 Sep 2006 09:31:19 -0400
Subject: [cisco-voip] Delayed audio-cutover with H.323 gateway
In-Reply-To: <20060912021909.82356.qmail@web37214.mail.mud.yahoo.com>
References: <20060912021909.82356.qmail@web37214.mail.mud.yahoo.com>
Message-ID: <523879D3-62CE-4E34-8E45-B40B202E235B@cisco.com>

What is round trip time from gateway to CM?  Are you using fast start  
on CM and the gateway?

/Wes

On Sep 11, 2006, at 10:19 PM, Erick Bergquist wrote:

Hi,

I'm having a issue where an inbound call comes in FXO port and  
connection plar's to a shared DN across 5 IP Phones. The phone call  
gets answered fine, but the user on IP phone saids "Welcome to  
BlahBlah" and the outside caller doesn't hear the first time they say  
"Welcome To BlahBlah" and the IP Phone user repeats their greeting  
and they hear it the second time fine.

There are no errors on any interfaces that I can see, and the gateway  
has voice rtp send-recv enabled. Issue does not happen on calls over  
PRI, just the analog FXOs. I have disabled caller-id on the voice- 
port and also tried connection plar opx with no difference. Sometimes  
it works fine but rarely.

CCM is version 4.1(3)sr3c and the gateway is a 1760 running 12.4(8a).

A similar problem can happen when the call is parked or put on hold  
also, now and then.


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


From erickbe at yahoo.com  Tue Sep 12 10:06:38 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue, 12 Sep 2006 07:06:38 -0700 (PDT)
Subject: [cisco-voip] Delayed audio-cutover with H.323 gateway
In-Reply-To: <523879D3-62CE-4E34-8E45-B40B202E235B@cisco.com>
Message-ID: <20060912140638.74372.qmail@web37209.mail.mud.yahoo.com>

4 to 8ms on average. Fast start not enabled on GW - let me look into that.

----- Original Message ----
From: Wes Sisk <wsisk at cisco.com>
To: Erick Bergquist <erickbe at yahoo.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Sent: Tuesday, September 12, 2006 8:31:19 AM
Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway

What is round trip time from gateway to CM?  Are you using fast start  
on CM and the gateway?

/Wes

On Sep 11, 2006, at 10:19 PM, Erick Bergquist wrote:

Hi,

I'm having a issue where an inbound call comes in FXO port and  
connection plar's to a shared DN across 5 IP Phones. The phone call  
gets answered fine, but the user on IP phone saids "Welcome to  
BlahBlah" and the outside caller doesn't hear the first time they say  
"Welcome To BlahBlah" and the IP Phone user repeats their greeting  
and they hear it the second time fine.

There are no errors on any interfaces that I can see, and the gateway  
has voice rtp send-recv enabled. Issue does not happen on calls over  
PRI, just the analog FXOs. I have disabled caller-id on the voice- 
port and also tried connection plar opx with no difference. Sometimes  
it works fine but rarely.

CCM is version 4.1(3)sr3c and the gateway is a 1760 running 12.4(8a).

A similar problem can happen when the call is parked or put on hold  
also, now and then.


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cisco-voip at puck.nether.net
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From BRiley at jackhenry.com  Tue Sep 12 10:10:11 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Tue, 12 Sep 2006 09:10:11 -0500
Subject: [cisco-voip] SRST inbound call
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002FCAC00@smoxchg.jhacorp.com>

I have several remote branches where I have broadcast hunt groups setup
in call manager to ring multiple phones when a call comes in on the FXO
ports. The FXO ports have a PLAR to the hunt pilot which rings most of
the phones at that location. I have a translation rule and dial-peers
setup so that when the GW cannot contact the call manager it will
translate the DN of the hunt pilot to a DN that is local to that branch.
All of that works correctly when the phone is switched to SRST but it
only rings one phone. I know SRST does not have any sort of broadcast
hunt groups in its functionality but I am trying to find a way to make
the second inbound call ring to a different phone if the first phone is
busy. 

 

 

Thanks,

 

 

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
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From bills at tns.its.psu.edu  Tue Sep 12 10:18:04 2006
From: bills at tns.its.psu.edu (Simon, Bill)
Date: Tue, 12 Sep 2006 10:18:04 -0400
Subject: [cisco-voip] VM ports all busy - give callers a friendly message
Message-ID: <4506C19C.2070402@tns.its.psu.edu>

When the voicemail ports are all busy, new callers getting forwarded to 
voicemail (or subscribers attempting to login) just get a reorder tone.

Is there any way to set up the annunciator or some other function to 
play back a friendlier "All lines are busy" sort of message?



From CORTIZ at broward.org  Tue Sep 12 10:22:43 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue, 12 Sep 2006 10:22:43 -0400
Subject: [cisco-voip] Unity Message limit warnings
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A47503@MREXVS01.bc.broward.cty>

Is there any way to turn off the message limit warnings in Unity when a
user is nearing their limit and logs on to their phone to hear messages?
I would rather leave this task to Exchange which already does a good job
of informing the users.

 

Carlos

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From rsingleton at novateck.com  Tue Sep 12 11:01:27 2006
From: rsingleton at novateck.com (Robert Singleton)
Date: Tue, 12 Sep 2006 10:01:27 -0500
Subject: [cisco-voip] SRST inbound call
In-Reply-To: <13591CB4B2359F4D96E2606DD6B840A002FCAC00@smoxchg.jhacorp.com>
References: <13591CB4B2359F4D96E2606DD6B840A002FCAC00@smoxchg.jhacorp.com>
Message-ID: <200609121002609.SM01032@mail.leasenet.net>

On Tue, 2006-09-12 at 09:10 -0500, Bill Riley III wrote:
> I know SRST does not have any sort of broadcast hunt groups in its
> functionality but I am trying to find a way to make the second inbound
> call ring to a different phone if the first phone is busy. 

I have a similar setup and will be very interested in any advice anyone
can offer!

Robert

> 


From helder.guarda at gs.com  Tue Sep 12 11:07:12 2006
From: helder.guarda at gs.com (Guarda, Helder)
Date: Tue, 12 Sep 2006 11:07:12 -0400
Subject: [cisco-voip] SRST inbound call
In-Reply-To: <1158073287.2035.12.camel@robert.morsco.com>
Message-ID: <F7DB0A6356C76D41B18EBC2CB175DE7603D11EC1@gsmbnmp04es.firmwide.corp.gs.com>

Try this:

 SUMMARY STEPS
1. call-manager-fallback
Enters call-manager fallback configuration mode
2. alias tag number-pattern to alternate-number [preference
preference-value] [cfw number timeout
timeout-value] [huntstop]
Creates a set of rules for rerouting calls to phones that are
unavailable during SRST
3. max-dn max-directory-numbers preference preference-order]
Sets maximum number of directory numbers that can be supported during
SRST
4. end
Return to privileged EXEC mode
5. show dial-peer voice summary
Displays information for the voice dial peers
Example:

The following example sets the preference keyword in the alias command
to a lower preference than the preference value created by the max-dn
command.  Setting this value lower allows for the cfw command to take
effect.  The incoming call to extension 36565 hunts to alias because it
has a lower preference, and no-answer/busy calls to 36565 are forwarded
to 36575.  All incoming calls to other extensions in SRST mode are
forwarded to 36599 after 10 seconds.

call-manager-fallback
     alias 1 36565 to 36570 preference 1 cfw 36575 timeout 10
     max-dn 10 preference 2 
     call-forward busy 36599
     call-forward noan 36599 timeout 10 	

I have this working at various locations and it works fine.

helder

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Singleton
Sent: Tuesday, September 12, 2006 11:01 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SRST inbound call

On Tue, 2006-09-12 at 09:10 -0500, Bill Riley III wrote:
> I know SRST does not have any sort of broadcast hunt groups in its 
> functionality but I am trying to find a way to make the second inbound

> call ring to a different phone if the first phone is busy.

I have a similar setup and will be very interested in any advice anyone
can offer!

Robert

> 

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


From BRiley at jackhenry.com  Tue Sep 12 11:34:53 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Tue, 12 Sep 2006 10:34:53 -0500
Subject: [cisco-voip] SRST inbound call
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002FCAD2D@smoxchg.jhacorp.com>

I think that is going to be it. Thanks for your help.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Guarda, Helder
Sent: Tuesday, September 12, 2006 10:07 AM
To: Robert Singleton; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SRST inbound call

Try this:

 SUMMARY STEPS
1. call-manager-fallback
Enters call-manager fallback configuration mode
2. alias tag number-pattern to alternate-number [preference
preference-value] [cfw number timeout
timeout-value] [huntstop]
Creates a set of rules for rerouting calls to phones that are
unavailable during SRST
3. max-dn max-directory-numbers preference preference-order]
Sets maximum number of directory numbers that can be supported during
SRST
4. end
Return to privileged EXEC mode
5. show dial-peer voice summary
Displays information for the voice dial peers
Example:

The following example sets the preference keyword in the alias command
to a lower preference than the preference value created by the max-dn
command.  Setting this value lower allows for the cfw command to take
effect.  The incoming call to extension 36565 hunts to alias because it
has a lower preference, and no-answer/busy calls to 36565 are forwarded
to 36575.  All incoming calls to other extensions in SRST mode are
forwarded to 36599 after 10 seconds.

call-manager-fallback
     alias 1 36565 to 36570 preference 1 cfw 36575 timeout 10
     max-dn 10 preference 2 
     call-forward busy 36599
     call-forward noan 36599 timeout 10 	

I have this working at various locations and it works fine.

helder

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert
Singleton
Sent: Tuesday, September 12, 2006 11:01 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] SRST inbound call

On Tue, 2006-09-12 at 09:10 -0500, Bill Riley III wrote:
> I know SRST does not have any sort of broadcast hunt groups in its 
> functionality but I am trying to find a way to make the second inbound

> call ring to a different phone if the first phone is busy.

I have a similar setup and will be very interested in any advice anyone
can offer!

Robert

> 

_______________________________________________
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
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.


From jgrace at digitel.net  Tue Sep 12 12:07:47 2006
From: jgrace at digitel.net (James Grace)
Date: Tue, 12 Sep 2006 12:07:47 -0400
Subject: [cisco-voip] unity checking VM externally
Message-ID: <200609121207421.SM01472@jgm>

Is there a way to config  unity to allow external user checking there vm to
just have to enter there password.  Instead of having to enter ( * ) and
then the ext 

 

James D. Grace 

CCNP CCNA MCSE MCSA 

Digitel Corporation

System Engineer / Professional Svc.

Email:  <mailto:jgrace at digitel.net> jgrace at digitel.net

Email:  <mailto:grace.jd at gmail.com> grace.jd at gmail.com

MSN messenger ID:  jgrace29 at hotmail.com

Mac IChat ID:  jgrace29 at mac.com

 

 

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From lelio at uoguelph.ca  Tue Sep 12 12:13:03 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 12 Sep 2006 12:13:03 -0400
Subject: [cisco-voip] unity checking VM externally
References: <200609121207421.SM01472@jgm>
Message-ID: <024501c6d686$530758f0$47196883@cfs.uoguelph.ca>

I'm pretty sure you can create a DID that goes to 'attempt sign in' and add the external DN as an alternate extension.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: James Grace 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, September 12, 2006 12:07 PM
  Subject: [cisco-voip] unity checking VM externally


  Is there a way to config  unity to allow external user checking there vm to just have to enter there password.  Instead of having to enter ( * ) and then the ext 

   

  James D. Grace 

  CCNP CCNA MCSE MCSA 

  Digitel Corporation

  System Engineer / Professional Svc.

  Email: jgrace at digitel.net

  Email: grace.jd at gmail.com

  MSN messenger ID:  jgrace29 at hotmail.com

  Mac IChat ID:  jgrace29 at mac.com

   

   



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


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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 tim.reimers at asheville.k12.nc.us  Tue Sep 12 12:17:12 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue, 12 Sep 2006 12:17:12 -0400
Subject: [cisco-voip] ip service...
Message-ID: <050944EE48FF384F9C663F7048C47DA704B418A3@webmail.asheville.k12.nc.us>

 

Hi folks---

 

Isn't there a service that can be configured on an IP phone that allows
a user to immediately report a call in which they had trouble with echo
or jitter?

 

I know the 'info' button shows the codec, etc-but it doesn't log that to
a technician for later analysis

(and have you ever tried to have a user read the codec info to you?
That's -really- fun to listen to...not! )

 

Thanks Tim

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From lelio at uoguelph.ca  Tue Sep 12 12:18:40 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 12 Sep 2006 12:18:40 -0400
Subject: [cisco-voip] ip service...
References: <050944EE48FF384F9C663F7048C47DA704B418A3@webmail.asheville.k12.nc.us>
Message-ID: <027c01c6d687$1cffc110$47196883@cfs.uoguelph.ca>

QRT

it's a softkey that you can show.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803f52dd.html

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Tim Reimers 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, September 12, 2006 12:17 PM
  Subject: [cisco-voip] ip service...


   

  Hi folks---

   

  Isn't there a service that can be configured on an IP phone that allows a user to immediately report a call in which they had trouble with echo or jitter?

   

  I know the 'info' button shows the codec, etc-but it doesn't log that to a technician for later analysis

  (and have you ever tried to have a user read the codec info to you? That's -really- fun to listen to.not! )

   

  Thanks Tim



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


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  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From tim.reimers at asheville.k12.nc.us  Tue Sep 12 12:20:30 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue, 12 Sep 2006 12:20:30 -0400
Subject: [cisco-voip] ip service...
In-Reply-To: <027c01c6d687$1cffc110$47196883@cfs.uoguelph.ca>
Message-ID: <050944EE48FF384F9C663F7048C47DA704B418A6@webmail.asheville.k12.nc.us>

Thanks Lelio and Scott...

 

 

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Tuesday, September 12, 2006 12:19 PM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ip service...

 

QRT

 

it's a softkey that you can show.

 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administra
tion_guide_chapter09186a00803f52dd.html

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."

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

	From: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  

	To: cisco-voip at puck.nether.net 

	Sent: Tuesday, September 12, 2006 12:17 PM

	Subject: [cisco-voip] ip service...

	 

	 

	Hi folks---

	 

	Isn't there a service that can be configured on an IP phone that
allows a user to immediately report a call in which they had trouble
with echo or jitter?

	 

	I know the 'info' button shows the codec, etc-but it doesn't log
that to a technician for later analysis

	(and have you ever tried to have a user read the codec info to
you? That's -really- fun to listen to...not! )

	 

	Thanks Tim

	
________________________________


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

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From Nick.Kassel at Charles-Stanley.co.uk  Tue Sep 12 12:26:09 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 12 Sep 2006 17:26:09 +0100
Subject: [cisco-voip] unity checking VM externally
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC014519CA@CSEXCH01.charles-stanley.co.uk>

Create a translation pattern from a DID and put the Unity voicemail
pilot in the called party transform mask.

 

You then need to add the number they are calling from into the Alternate
extensions which I believe you can have 10 different ones of, then when
you call from that number, you just enter your pin. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, September 12, 2006 5:13 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] unity checking VM externally

 

I'm pretty sure you can create a DID that goes to 'attempt sign in' and
add the external DN as an alternate extension.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."

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

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

	To: cisco-voip at puck.nether.net 

	Sent: Tuesday, September 12, 2006 12:07 PM

	Subject: [cisco-voip] unity checking VM externally

	 

	Is there a way to config  unity to allow external user checking
there vm to just have to enter there password.  Instead of having to
enter ( * ) and then the ext 

	 

	James D. Grace 

	CCNP CCNA MCSE MCSA 

	Digitel Corporation

	System Engineer / Professional Svc.

	Email: jgrace at digitel.net <mailto:jgrace at digitel.net> 

	Email: grace.jd at gmail.com <mailto:grace.jd at gmail.com> 

	MSN messenger ID:  jgrace29 at hotmail.com

	Mac IChat ID:  jgrace29 at mac.com

	 

	 

	
________________________________


	_______________________________________________
	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 Sep 12 12:28:33 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 12 Sep 2006 12:28:33 -0400
Subject: [cisco-voip] unity checking VM externally
References: <253FAE7A06B11B49A3A29770B2B8BAFC014519CA@CSEXCH01.charles-stanley.co.uk>
Message-ID: <02cd01c6d688$7ee8a530$47196883@cfs.uoguelph.ca>

yup, that's the ticket.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Nick Kassel 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Tuesday, September 12, 2006 12:26 PM
  Subject: RE: [cisco-voip] unity checking VM externally


  Create a translation pattern from a DID and put the Unity voicemail pilot in the called party transform mask.



  You then need to add the number they are calling from into the Alternate extensions which I believe you can have 10 different ones of, then when you call from that number, you just enter your pin. 




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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Tuesday, September 12, 2006 5:13 PM
  To: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] unity checking VM externally



  I'm pretty sure you can create a DID that goes to 'attempt sign in' and add the external DN as an alternate extension.



  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I can eat fifty eggs." "Nobody can eat fifty eggs."

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

    From: James Grace 

    To: cisco-voip at puck.nether.net 

    Sent: Tuesday, September 12, 2006 12:07 PM

    Subject: [cisco-voip] unity checking VM externally



    Is there a way to config  unity to allow external user checking there vm to just have to enter there password.  Instead of having to enter ( * ) and then the ext 



    James D. Grace 

    CCNP CCNA MCSE MCSA 

    Digitel Corporation

    System Engineer / Professional Svc.

    Email: jgrace at digitel.net

    Email: grace.jd at gmail.com

    MSN messenger ID:  jgrace29 at hotmail.com

    Mac IChat ID:  jgrace29 at mac.com






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

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From chris.ellington at nsi1.com  Tue Sep 12 12:35:26 2006
From: chris.ellington at nsi1.com (Chris Ellington)
Date: Tue, 12 Sep 2006 12:35:26 -0400
Subject: [cisco-voip] Delayed audio-cutover with H.323 gateway
In-Reply-To: <mailman.7.1158076805.69520.cisco-voip@puck.nether.net>
Message-ID: <C12C5A0E.10343%chris.ellington@nsi1.com>

There is absolutely something wrong with 12.4T code - I had a similar issue
with delayed audio path that my TAC guy could never figure out; eventually
the gateway quit passing any traffic and a reload solved everything.

This is a CMM running 12.4(6)T3 code (which admittedly is not the latest
code for the CMM).

I had my TAC case open for 3 weeks with no resolution but am looking to roll
back to the latest 12.3T code for the CMM.

Chris


On 9/12/06 12:00 PM, "cisco-voip-request at puck.nether.net"
<cisco-voip-request at puck.nether.net> wrote:

> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Tue, 12 Sep 2006 09:31:19 -0400
> From: Wes Sisk <wsisk at cisco.com>
> Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway
> To: Erick Bergquist <erickbe at yahoo.com>
> Cc: ciscovoip <cisco-voip at puck.nether.net>
> Message-ID: <523879D3-62CE-4E34-8E45-B40B202E235B at cisco.com>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
> 
> What is round trip time from gateway to CM?  Are you using fast start
> on CM and the gateway?
> 
> /Wes
> 
> On Sep 11, 2006, at 10:19 PM, Erick Bergquist wrote:
> 
> Hi,
> 
> I'm having a issue where an inbound call comes in FXO port and
> connection plar's to a shared DN across 5 IP Phones. The phone call
> gets answered fine, but the user on IP phone saids "Welcome to
> BlahBlah" and the outside caller doesn't hear the first time they say
> "Welcome To BlahBlah" and the IP Phone user repeats their greeting
> and they hear it the second time fine.
> 
> There are no errors on any interfaces that I can see, and the gateway
> has voice rtp send-recv enabled. Issue does not happen on calls over
> PRI, just the analog FXOs. I have disabled caller-id on the voice-
> port and also tried connection plar opx with no difference. Sometimes
> it works fine but rarely.
> 
> CCM is version 4.1(3)sr3c and the gateway is a 1760 running 12.4(8a).
> 
> A similar problem can happen when the call is parked or put on hold
> also, now and then.
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com



From Scott.Voll at wesd.org  Tue Sep 12 12:38:04 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 12 Sep 2006 09:38:04 -0700
Subject: [cisco-voip] unity checking VM externally
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C828F@Apollo.wesd.org>

If they don't want to ever leave themselves a message you can just put
the external number in as an alternate extension and it will always ask
them for their password.

 

Scott

 

PS.  I don't like this option personally as sometimes I like to leave
myself messages like Lelio.  You know the message like, "you are a great
guy, and people really like you!" :-)

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, September 12, 2006 9:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] unity checking VM externally

 

Is there a way to config  unity to allow external user checking there vm
to just have to enter there password.  Instead of having to enter ( * )
and then the ext 

 

James D. Grace 

CCNP CCNA MCSE MCSA 

Digitel Corporation

System Engineer / Professional Svc.

Email: jgrace at digitel.net <mailto:jgrace at digitel.net> 

Email: grace.jd at gmail.com <mailto:grace.jd at gmail.com> 

MSN messenger ID:  jgrace29 at hotmail.com

Mac IChat ID:  jgrace29 at mac.com

 

 

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From Scott.Voll at wesd.org  Tue Sep 12 12:40:51 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 12 Sep 2006 09:40:51 -0700
Subject: [cisco-voip] Delayed audio-cutover with H.323 gateway
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C8294@Apollo.wesd.org>

Why would you run T code on a CMM?  I'm running 12.4(7a) without issues.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Chris Ellington
Sent: Tuesday, September 12, 2006 9:35 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway

There is absolutely something wrong with 12.4T code - I had a similar
issue
with delayed audio path that my TAC guy could never figure out;
eventually
the gateway quit passing any traffic and a reload solved everything.

This is a CMM running 12.4(6)T3 code (which admittedly is not the latest
code for the CMM).

I had my TAC case open for 3 weeks with no resolution but am looking to
roll
back to the latest 12.3T code for the CMM.

Chris


On 9/12/06 12:00 PM, "cisco-voip-request at puck.nether.net"
<cisco-voip-request at puck.nether.net> wrote:

> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Tue, 12 Sep 2006 09:31:19 -0400
> From: Wes Sisk <wsisk at cisco.com>
> Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway
> To: Erick Bergquist <erickbe at yahoo.com>
> Cc: ciscovoip <cisco-voip at puck.nether.net>
> Message-ID: <523879D3-62CE-4E34-8E45-B40B202E235B at cisco.com>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
> 
> What is round trip time from gateway to CM?  Are you using fast start
> on CM and the gateway?
> 
> /Wes
> 
> On Sep 11, 2006, at 10:19 PM, Erick Bergquist wrote:
> 
> Hi,
> 
> I'm having a issue where an inbound call comes in FXO port and
> connection plar's to a shared DN across 5 IP Phones. The phone call
> gets answered fine, but the user on IP phone saids "Welcome to
> BlahBlah" and the outside caller doesn't hear the first time they say
> "Welcome To BlahBlah" and the IP Phone user repeats their greeting
> and they hear it the second time fine.
> 
> There are no errors on any interfaces that I can see, and the gateway
> has voice rtp send-recv enabled. Issue does not happen on calls over
> PRI, just the analog FXOs. I have disabled caller-id on the voice-
> port and also tried connection plar opx with no difference. Sometimes
> it works fine but rarely.
> 
> CCM is version 4.1(3)sr3c and the gateway is a 1760 running 12.4(8a).
> 
> A similar problem can happen when the call is parked or put on hold
> also, now and then.
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com


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


From zephy316 at gmail.com  Tue Sep 12 13:18:00 2006
From: zephy316 at gmail.com (Israel Lang)
Date: Tue, 12 Sep 2006 13:18:00 -0400
Subject: [cisco-voip] unity checking VM externally
In-Reply-To: <407055A92CECCB499C922A2D35FC19A6066C828F@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A6066C828F@Apollo.wesd.org>
Message-ID: <91e236d30609121018y6a40e938y214e921fb67b6c4@mail.gmail.com>

alternative extensions.

On 9/12/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>  If they don't want to ever leave themselves a message you can just put
>  the external number in as an alternate extension and it will always ask
> them for their password.
>
>
>
> Scott
>
>
>
> PS.  I don't like this option personally as sometimes I like to leave
> myself messages like Lelio.  You know the message like, "you are a great
> guy, and people really like you!" J
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *James Grace
> *Sent:* Tuesday, September 12, 2006 9:08 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] unity checking VM externally
>
>
>
> Is there a way to config  unity to allow external user checking there vm
> to just have to enter there password.  Instead of having to enter ( * ) and
> then the ext
>
>
>
> James D. Grace
>
> CCNP CCNA MCSE MCSA
>
> Digitel Corporation
>
> System Engineer / Professional Svc.
>
> Email: jgrace at digitel.net
>
> *Email: **grace.jd at gmail.com* <grace.jd at gmail.com>
>
> MSN messenger ID:  jgrace29 at hotmail.com
>
> Mac IChat ID:  jgrace29 at mac.com
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


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

Replace the _ with an @ symbol
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From wsisk at cisco.com  Tue Sep 12 14:57:17 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 12 Sep 2006 14:57:17 -0400
Subject: [cisco-voip] Delayed audio-cutover with H.323 gateway
In-Reply-To: <20060912140638.74372.qmail@web37209.mail.mud.yahoo.com>
References: <20060912140638.74372.qmail@web37209.mail.mud.yahoo.com>
Message-ID: <4507030D.1040700@cisco.com>

Fast start should not apply here.  make sure you are not dropping 
packets or retransmitting due to network loss between the 2 but fast 
start should only be necessary when you're looking at 40msec+ RTT gw to 
CM.  You have problems either on the GW or on the CM with getting those 
channels opened fast enough.

off the wall idea - enable 'file close thread flag' to make sure you 
aren't hitting that issue.  then get CCM and SDL traces from an 
occurrence.  look in the traces to obeserve the time from user offhook 
to startmediatransmittion to phone and open receive channel ack from the 
phone.  If that is small, then it is a problem with audio cut through on 
the gw itself.

/Wes

Erick Bergquist wrote:
> 4 to 8ms on average. Fast start not enabled on GW - let me look into that.
> 
> ----- Original Message ----
> From: Wes Sisk <wsisk at cisco.com>
> To: Erick Bergquist <erickbe at yahoo.com>
> Cc: ciscovoip <cisco-voip at puck.nether.net>
> Sent: Tuesday, September 12, 2006 8:31:19 AM
> Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway
> 
> What is round trip time from gateway to CM?  Are you using fast start  
> on CM and the gateway?
> 
> /Wes
> 
> On Sep 11, 2006, at 10:19 PM, Erick Bergquist wrote:
> 
> Hi,
> 
> I'm having a issue where an inbound call comes in FXO port and  
> connection plar's to a shared DN across 5 IP Phones. The phone call  
> gets answered fine, but the user on IP phone saids "Welcome to  
> BlahBlah" and the outside caller doesn't hear the first time they say  
> "Welcome To BlahBlah" and the IP Phone user repeats their greeting  
> and they hear it the second time fine.
> 
> There are no errors on any interfaces that I can see, and the gateway  
> has voice rtp send-recv enabled. Issue does not happen on calls over  
> PRI, just the analog FXOs. I have disabled caller-id on the voice- 
> port and also tried connection plar opx with no difference. Sometimes  
> it works fine but rarely.
> 
> CCM is version 4.1(3)sr3c and the gateway is a 1760 running 12.4(8a).
> 
> A similar problem can happen when the call is parked or put on hold  
> also, now and then.
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 

From LSOCARRAS at broward.org  Tue Sep 12 15:25:08 2006
From: LSOCARRAS at broward.org (Socarras, Luis)
Date: Tue, 12 Sep 2006 15:25:08 -0400
Subject: [cisco-voip] Meet-Me conference call
Message-ID: <5BFD4B266FCBED44938AE3BB35ACD0AA014FF291@UVEXVS01.bc.broward.cty>

Is there a way to initiate a Meetme conference call from an external
phone?  

 

I know it can be done using IPcommunicator, or an internal phone with
the Meetme softkey.

 

Thanks,

Luis

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From Scott.Voll at wesd.org  Tue Sep 12 15:26:30 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 12 Sep 2006 12:26:30 -0700
Subject: [cisco-voip] Meet-Me conference call
Message-ID: <407055A92CECCB499C922A2D35FC19A605D63206@Apollo.wesd.org>

Not with out it being started from an ip phone

Scott

--------------------------
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: Tue Sep 12 12:25:08 2006
Subject: [cisco-voip] Meet-Me conference call

Is there a way to initiate a Meetme conference call from an external phone?  

 

I know it can be done using IPcommunicator, or an internal phone with the Meetme softkey.

 

Thanks,

Luis

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From wsisk at cisco.com  Tue Sep 12 15:50:02 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 12 Sep 2006 15:50:02 -0400
Subject: [cisco-voip] VM ports all busy - give callers a friendly message
In-Reply-To: <4506C19C.2070402@tns.its.psu.edu>
References: <4506C19C.2070402@tns.its.psu.edu>
Message-ID: <45070F6A.1060003@cisco.com>

configure the hunt pilot to hunt busy to a dummy dn that is CFA to an 
IVR script.

Simon, Bill wrote:
> When the voicemail ports are all busy, new callers getting forwarded to 
> voicemail (or subscribers attempting to login) just get a reorder tone.
> 
> Is there any way to set up the annunciator or some other function to 
> play back a friendlier "All lines are busy" sort of message?
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

From jgrace at digitelusa.net  Tue Sep 12 15:51:27 2006
From: jgrace at digitelusa.net (James Grace)
Date: 12 Sep 2006 19:51:27 GMT
Subject: [cisco-voip] Error in AC "unable to make additional call
Message-ID: <SUWCL19IP018vjqueol00001fa9@smtpoutcl19.wirelesssyncvzw.net>

Our ac is working but we keep getting this error after we handle a call   
Error in AC "unable to make additional call
Noow we have a pilot and hunt for the ac user

James D Grace
CCNP CCNA MCSE MCDBA
Sr. System Engineer
MSN Messenger ID: jgrace29 at hotmail.com
MAC IChat ID: 
jgrace29 at mac.com

From bills at tns.its.psu.edu  Tue Sep 12 15:51:57 2006
From: bills at tns.its.psu.edu (Simon, Bill)
Date: Tue, 12 Sep 2006 15:51:57 -0400
Subject: [cisco-voip] VM ports all busy - give callers a friendly message
In-Reply-To: <45070F6A.1060003@cisco.com>
References: <4506C19C.2070402@tns.its.psu.edu> <45070F6A.1060003@cisco.com>
Message-ID: <45070FDD.8010303@tns.its.psu.edu>

That's what I thought someone might say.  Is there any way to do it with 
just CallManager?

Wes Sisk wrote:
> configure the hunt pilot to hunt busy to a dummy dn that is CFA to an 
> IVR script.
> 
> Simon, Bill wrote:
>> When the voicemail ports are all busy, new callers getting forwarded 
>> to voicemail (or subscribers attempting to login) just get a reorder 
>> tone.
>>
>> Is there any way to set up the annunciator or some other function to 
>> play back a friendlier "All lines are busy" sort of message?


From lelio at uoguelph.ca  Tue Sep 12 15:54:46 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 12 Sep 2006 15:54:46 -0400
Subject: [cisco-voip] VM ports all busy - give callers a friendly message
References: <4506C19C.2070402@tns.its.psu.edu> <45070F6A.1060003@cisco.com>
	<45070FDD.8010303@tns.its.psu.edu>
Message-ID: <044701c6d6a5$4c3f69d0$47196883@cfs.uoguelph.ca>

Is it a costing issue? CallManager comes with a free X port autoattendant (IPCC eXpress I think). Use that and not only will you get a nice message, but if you use Unity as your AA, you get a back autoattendant.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Simon, Bill 
  To: Wes Sisk 
  Cc: Cisco Voip 
  Sent: Tuesday, September 12, 2006 3:51 PM
  Subject: Re: [cisco-voip] VM ports all busy - give callers a friendly message


  That's what I thought someone might say.  Is there any way to do it with 
  just CallManager?

  Wes Sisk wrote:
  > configure the hunt pilot to hunt busy to a dummy dn that is CFA to an 
  > IVR script.
  > 
  > Simon, Bill wrote:
  >> When the voicemail ports are all busy, new callers getting forwarded 
  >> to voicemail (or subscribers attempting to login) just get a reorder 
  >> tone.
  >>
  >> Is there any way to set up the annunciator or some other function to 
  >> play back a friendlier "All lines are busy" sort of message?

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From tim.reimers at asheville.k12.nc.us  Tue Sep 12 16:02:52 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue, 12 Sep 2006 16:02:52 -0400
Subject: [cisco-voip] ip service...
In-Reply-To: <027c01c6d687$1cffc110$47196883@cfs.uoguelph.ca>
Message-ID: <050944EE48FF384F9C663F7048C47DA704B4199B@webmail.asheville.k12.nc.us>

 
 
I get the message "Document empty or invalid" when I try to run a
report..
 
I see that it's part of Extended Services, and puts a QRT button on the
phone.
 
I cannot find any directions for 4.01 on how to configure the CCM so
that the button becomes available to users...
 
Anyone have a link on how to enable the QRT function on the phones?
 
Tim
 
________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Tuesday, September 12, 2006 12:19 PM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ip service...


QRT
 
it's a softkey that you can show.
 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administra
tion_guide_chapter09186a00803f52dd.html
 
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."

	----- Original Message ----- 
	From: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  
	To: cisco-voip at puck.nether.net 
	Sent: Tuesday, September 12, 2006 12:17 PM
	Subject: [cisco-voip] ip service...


	 

	Hi folks---

	 

	Isn't there a service that can be configured on an IP phone that
allows a user to immediately report a call in which they had trouble
with echo or jitter?

	 

	I know the 'info' button shows the codec, etc-but it doesn't log
that to a technician for later analysis

	(and have you ever tried to have a user read the codec info to
you? That's -really- fun to listen to...not! )

	 

	Thanks Tim

	
________________________________


	

	_______________________________________________
	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  Tue Sep 12 16:05:34 2006
From: bills at tns.its.psu.edu (Simon, Bill)
Date: Tue, 12 Sep 2006 16:05:34 -0400
Subject: [cisco-voip] VM ports all busy - give callers a friendly message
In-Reply-To: <044701c6d6a5$4c3f69d0$47196883@cfs.uoguelph.ca>
References: <4506C19C.2070402@tns.its.psu.edu> <45070F6A.1060003@cisco.com>
	<45070FDD.8010303@tns.its.psu.edu>
	<044701c6d6a5$4c3f69d0$47196883@cfs.uoguelph.ca>
Message-ID: <4507130E.8000102@tns.its.psu.edu>

It's more an issue of complexity.  We have IPCC enterprise.  If I had to 
choose the most reliable part of our IPT setup I'd choose CallManager 
over Unity or IPCC.  Therefore when those systems are in the process of 
sucking mud I would like CallManager to be able to respond gracefully.

Another thing I'm trying to get a sense of is the usefulness of the 
Annunciator.  Seems to me that it doesn't annunciate much of anything 
except to tell the user he dialed some incorrect digits.  I'd like just 
a few more intercept messages to deliver to a caller to explain a situation.


Lelio Fulgenzi wrote:
> Is it a costing issue? CallManager comes with a free X port 
> autoattendant (IPCC eXpress I think). Use that and not only will you get 
> a nice message, but if you use Unity as your AA, you get a back 
> autoattendant.
>  
>  
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."
> 
>     ----- Original Message -----
>     *From:* Simon, Bill <mailto:bills at tns.its.psu.edu>
>     *To:* Wes Sisk <mailto:wsisk at cisco.com>
>     *Cc:* Cisco Voip <mailto:cisco-voip at puck.nether.net>
>     *Sent:* Tuesday, September 12, 2006 3:51 PM
>     *Subject:* Re: [cisco-voip] VM ports all busy - give callers a
>     friendly message
> 
>     That's what I thought someone might say.  Is there any way to do it
>     with
>     just CallManager?
> 
>     Wes Sisk wrote:
>      > configure the hunt pilot to hunt busy to a dummy dn that is CFA
>     to an
>      > IVR script.
>      >
>      > Simon, Bill wrote:
>      >> When the voicemail ports are all busy, new callers getting
>     forwarded
>      >> to voicemail (or subscribers attempting to login) just get a
>     reorder
>      >> tone.
>      >>
>      >> Is there any way to set up the annunciator or some other
>     function to
>      >> play back a friendlier "All lines are busy" sort of message?

From tim.reimers at asheville.k12.nc.us  Tue Sep 12 16:07:53 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue, 12 Sep 2006 16:07:53 -0400
Subject: [cisco-voip] ip service...
In-Reply-To: <027c01c6d687$1cffc110$47196883@cfs.uoguelph.ca>
Message-ID: <050944EE48FF384F9C663F7048C47DA704B419A0@webmail.asheville.k12.nc.us>

btw- the "extended functions" service is indeed running---
I did think that perhaps it was just set to 'off'

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Tuesday, September 12, 2006 12:19 PM
To: Tim Reimers; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ip service...


QRT
 
it's a softkey that you can show.
 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administra
tion_guide_chapter09186a00803f52dd.html
 
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."

	----- Original Message ----- 
	From: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  
	To: cisco-voip at puck.nether.net 
	Sent: Tuesday, September 12, 2006 12:17 PM
	Subject: [cisco-voip] ip service...


	 

	Hi folks---

	 

	Isn't there a service that can be configured on an IP phone that
allows a user to immediately report a call in which they had trouble
with echo or jitter?

	 

	I know the 'info' button shows the codec, etc-but it doesn't log
that to a technician for later analysis

	(and have you ever tried to have a user read the codec info to
you? That's -really- fun to listen to...not! )

	 

	Thanks Tim

	
________________________________


	

	_______________________________________________
	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 Sep 12 16:09:01 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 12 Sep 2006 16:09:01 -0400
Subject: [cisco-voip] ip service...
References: <050944EE48FF384F9C663F7048C47DA704B4199B@webmail.asheville.k12.nc.us>
Message-ID: <046001c6d6a7$4a0916f0$47196883@cfs.uoguelph.ca>

It's a softkey that you have to enable via the softkey template configuration screen.

device > device settings > softkey template

select your template, then:

highlite callmanager > configure softkey layout 

quality report tool (QRT) appears as valid options for the following states:

Connected
Connected Conference
Connected Transfer
On Hook

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Tim Reimers 
  To: Lelio Fulgenzi 
  Cc: Cisco Voip 
  Sent: Tuesday, September 12, 2006 4:02 PM
  Subject: RE: [cisco-voip] ip service...




  I get the message "Document empty or invalid" when I try to run a report..

  I see that it's part of Extended Services, and puts a QRT button on the phone.

  I cannot find any directions for 4.01 on how to configure the CCM so that the button becomes available to users...

  Anyone have a link on how to enable the QRT function on the phones?

  Tim


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

  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
  Sent: Tuesday, September 12, 2006 12:19 PM
  To: Tim Reimers; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] ip service...


  QRT

  it's a softkey that you can show.

  http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803f52dd.html

  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I can eat fifty eggs." "Nobody can eat fifty eggs."
    ----- Original Message ----- 
    From: Tim Reimers 
    To: cisco-voip at puck.nether.net 
    Sent: Tuesday, September 12, 2006 12:17 PM
    Subject: [cisco-voip] ip service...


     

    Hi folks---

     

    Isn't there a service that can be configured on an IP phone that allows a user to immediately report a call in which they had trouble with echo or jitter?

     

    I know the 'info' button shows the codec, etc-but it doesn't log that to a technician for later analysis

    (and have you ever tried to have a user read the codec info to you? That's -really- fun to listen to.not! )

     

    Thanks Tim



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


    _______________________________________________
    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  Tue Sep 12 16:13:03 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 12 Sep 2006 16:13:03 -0400
Subject: [cisco-voip] VM ports all busy - give callers a friendly message
In-Reply-To: <4507130E.8000102@tns.its.psu.edu>
References: <4506C19C.2070402@tns.its.psu.edu> <45070F6A.1060003@cisco.com>
	<45070FDD.8010303@tns.its.psu.edu>
	<044701c6d6a5$4c3f69d0$47196883@cfs.uoguelph.ca>
	<4507130E.8000102@tns.its.psu.edu>
Message-ID: <450714CF.7080400@cisco.com>

Unfortunately Annunciator is currently pretty limited in functionality. 
  We can't really define disconnect codes to invoke ANN or record 
alternate messages.  I've heard it's coming, but have not heard a date 
put to that.

/Wes

Simon, Bill wrote:
> It's more an issue of complexity.  We have IPCC enterprise.  If I had to 
> choose the most reliable part of our IPT setup I'd choose CallManager 
> over Unity or IPCC.  Therefore when those systems are in the process of 
> sucking mud I would like CallManager to be able to respond gracefully.
> 
> Another thing I'm trying to get a sense of is the usefulness of the 
> Annunciator.  Seems to me that it doesn't annunciate much of anything 
> except to tell the user he dialed some incorrect digits.  I'd like just 
> a few more intercept messages to deliver to a caller to explain a 
> situation.
> 
> 
> Lelio Fulgenzi wrote:
>> Is it a costing issue? CallManager comes with a free X port 
>> autoattendant (IPCC eXpress I think). Use that and not only will you 
>> get a nice message, but if you use Unity as your AA, you get a back 
>> autoattendant.
>>  
>>  
>> -------------------------------------------------------------------------------- 
>>
>> Lelio Fulgenzi, B.A.
>> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
>> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>> "I can eat fifty eggs." "Nobody can eat fifty eggs."
>>
>>     ----- Original Message -----
>>     *From:* Simon, Bill <mailto:bills at tns.its.psu.edu>
>>     *To:* Wes Sisk <mailto:wsisk at cisco.com>
>>     *Cc:* Cisco Voip <mailto:cisco-voip at puck.nether.net>
>>     *Sent:* Tuesday, September 12, 2006 3:51 PM
>>     *Subject:* Re: [cisco-voip] VM ports all busy - give callers a
>>     friendly message
>>
>>     That's what I thought someone might say.  Is there any way to do it
>>     with
>>     just CallManager?
>>
>>     Wes Sisk wrote:
>>      > configure the hunt pilot to hunt busy to a dummy dn that is CFA
>>     to an
>>      > IVR script.
>>      >
>>      > Simon, Bill wrote:
>>      >> When the voicemail ports are all busy, new callers getting
>>     forwarded
>>      >> to voicemail (or subscribers attempting to login) just get a
>>     reorder
>>      >> tone.
>>      >>
>>      >> Is there any way to set up the annunciator or some other
>>     function to
>>      >> play back a friendlier "All lines are busy" sort of message?

From jgrace at digitel.net  Tue Sep 12 16:29:54 2006
From: jgrace at digitel.net (James Grace)
Date: Tue, 12 Sep 2006 16:29:54 -0400
Subject: [cisco-voip] having a unity timezone error
Message-ID: <20060912162978.SM01472@jgm>

We have a unity unity unified message w/exchange.  Running into an issue
where our server is configured with a central timezone.  Now we have a
remote site and those subscribers are configure with est.  for some reason
there VM time stamp is 2 hour ahead verse 1 hour.  When we change the
timezone for those subscriber to default,  the stamp is correct 1 hrs ahead.
Anybody seen this issue.

 

James D. Grace 

CCNP CCNA MCSE MCSA 

Digitel Corporation

System Engineer / Professional Svc.

Email:  <mailto:jgrace at digitel.net> jgrace at digitel.net

Email:  <mailto:grace.jd at gmail.com> grace.jd at gmail.com

MSN messenger ID:  jgrace29 at hotmail.com

Mac IChat ID:  jgrace29 at mac.com

 

 

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From jason.aarons at us.didata.com  Tue Sep 12 16:42:58 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 12 Sep 2006 16:42:58 -0400
Subject: [cisco-voip] Speed Dial 9+456+variable
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032146E2@USNAEXCH.na.didata.local>

Customer's Avaya system allowed them to dial 9 plus 456 plus any
variable length of 4 digits for a speed dial button.

 

Any thoughts how to mimic this in CallManager?

 

Eg Press Speedial then 6756 would dial 94566756




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From techguy at gmail.com  Tue Sep 12 17:14:13 2006
From: techguy at gmail.com (TechGuy)
Date: Tue, 12 Sep 2006 16:14:13 -0500
Subject: [cisco-voip] Managing DID's ?
Message-ID: <a1b07ea60609121414l7fda34aas1222a5c9dccfea58@mail.gmail.com>

I have one major issue/complaint if you will with callmanager, and
that is in regards to managing DID's or extension ranges or whatever
you want to call it.

I have no experience with any other phone systems or PBX's so nothing
to base this on but my own personal experience of having to manage
callmanager over the years.

Just seems there should be a way to manage them both DID's /
extensions.  I am refering to which are used or not.  I guess I think
of it in terms of DHCP kinda.  You put in the given range your working
with, and you can easily add and delete from within that range and see
statistics based on that range.  Such as how many are used, or
available.  When you delete a user/phone/extension maybe it goes back
into this available pool or something.

Right now all I can do is search say extension beginning with 145 and
see if anything is availble between 1450 - 1459 and so on.  I suspect
everyone has to deal with this same mess and I have to wonder if there
is a better way that I am simply not aware of.

Real nightmare for me right now due to acquisitions of others and
using their DID ranges which now seem to be all over the place for me.
 I have ranges using every number except 0 and 8, 8 is out outside
access number.  So I am running into issues now trying to make it all
work with new offices and what not.  I blame the person who setup four
digit extensions.

Thoughts and suggestions?

From Scott.Voll at wesd.org  Tue Sep 12 17:18:56 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 12 Sep 2006 14:18:56 -0700
Subject: [cisco-voip] Managing DID's ?
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C838A@Apollo.wesd.org>

First off use the route plan report to get what is available.

You might want to go to CIPTUG conference next month.  There were many
workshops on this subject.

What about changing your dialing to 5 digits?  Seems like a lot of work,
but might save you headaches in the future.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of TechGuy
Sent: Tuesday, September 12, 2006 2:14 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Managing DID's ?

I have one major issue/complaint if you will with callmanager, and
that is in regards to managing DID's or extension ranges or whatever
you want to call it.

I have no experience with any other phone systems or PBX's so nothing
to base this on but my own personal experience of having to manage
callmanager over the years.

Just seems there should be a way to manage them both DID's /
extensions.  I am refering to which are used or not.  I guess I think
of it in terms of DHCP kinda.  You put in the given range your working
with, and you can easily add and delete from within that range and see
statistics based on that range.  Such as how many are used, or
available.  When you delete a user/phone/extension maybe it goes back
into this available pool or something.

Right now all I can do is search say extension beginning with 145 and
see if anything is availble between 1450 - 1459 and so on.  I suspect
everyone has to deal with this same mess and I have to wonder if there
is a better way that I am simply not aware of.

Real nightmare for me right now due to acquisitions of others and
using their DID ranges which now seem to be all over the place for me.
 I have ranges using every number except 0 and 8, 8 is out outside
access number.  So I am running into issues now trying to make it all
work with new offices and what not.  I blame the person who setup four
digit extensions.

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


From voipermike at gmail.com  Tue Sep 12 17:21:41 2006
From: voipermike at gmail.com (Mike Mike)
Date: Tue, 12 Sep 2006 16:21:41 -0500
Subject: [cisco-voip] Managing DID's ?
In-Reply-To: <a1b07ea60609121414l7fda34aas1222a5c9dccfea58@mail.gmail.com>
References: <a1b07ea60609121414l7fda34aas1222a5c9dccfea58@mail.gmail.com>
Message-ID: <8bbfc1f40609121421l142cd0dex4401eb6803ccff94@mail.gmail.com>

How about the route plan report. Save it as a .csv and sort as necessary.
Click Route Plan > Route Plan Report > View IN File (upper right hand
corner) and then save as .csv

On 9/12/06, TechGuy <techguy at gmail.com> wrote:
>
> I have one major issue/complaint if you will with callmanager, and
> that is in regards to managing DID's or extension ranges or whatever
> you want to call it.
>
> I have no experience with any other phone systems or PBX's so nothing
> to base this on but my own personal experience of having to manage
> callmanager over the years.
>
> Just seems there should be a way to manage them both DID's /
> extensions.  I am refering to which are used or not.  I guess I think
> of it in terms of DHCP kinda.  You put in the given range your working
> with, and you can easily add and delete from within that range and see
> statistics based on that range.  Such as how many are used, or
> available.  When you delete a user/phone/extension maybe it goes back
> into this available pool or something.
>
> Right now all I can do is search say extension beginning with 145 and
> see if anything is availble between 1450 - 1459 and so on.  I suspect
> everyone has to deal with this same mess and I have to wonder if there
> is a better way that I am simply not aware of.
>
> Real nightmare for me right now due to acquisitions of others and
> using their DID ranges which now seem to be all over the place for me.
> I have ranges using every number except 0 and 8, 8 is out outside
> access number.  So I am running into issues now trying to make it all
> work with new offices and what not.  I blame the person who setup four
> digit extensions.
>
> Thoughts and suggestions?
> _______________________________________________
> 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  Tue Sep 12 17:59:10 2006
From: it at cimgroup.com (IT)
Date: Tue, 12 Sep 2006 14:59:10 -0700
Subject: [cisco-voip] vulnerable gateway?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A6066C838A@Apollo.wesd.org>
Message-ID: <B487D4C1FE6D04439061018C6EBC109B03D5EF84@exchange.cimgroup.com>

I just got a call from my long distance provider that someone has been
using my PRI for many international calls. I check my CDR database
tables, and it appears that calls have been coming from one of my branch
office 2801's. But, in the CDR table, the origDeviceName alternates
between the name of the gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those
ports open, because when I tried to do a "access-list 100 deny udp any
any eq 1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up
these gateways in the first place locked them down, but hindsight is
20/20.

Thanks,
Avidan


From puckcisco at cumhur.com  Tue Sep 12 18:19:01 2006
From: puckcisco at cumhur.com (puckcisco at cumhur.com)
Date: Wed, 13 Sep 2006 01:19:01 +0300
Subject: [cisco-voip] vulnerable gateway?
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B03D5EF84@exchange.cimgroup.com>
Message-ID: <08bc01c6d6b9$73582c50$c8a8a8c0@izmir>

Dear Avidan,

H323 use TCP 1720 (not udp)  port for call initiation also don't forget to
block SIP ports TCP/UDP 5060.

Regards.
Cumhur

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, September 13, 2006 12:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] vulnerable gateway?

I just got a call from my long distance provider that someone has been
using my PRI for many international calls. I check my CDR database
tables, and it appears that calls have been coming from one of my branch
office 2801's. But, in the CDR table, the origDeviceName alternates
between the name of the gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those
ports open, because when I tried to do a "access-list 100 deny udp any
any eq 1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up
these gateways in the first place locked them down, but hindsight is
20/20.

Thanks,
Avidan

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



-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date: 12/09/2006



From it at cimgroup.com  Tue Sep 12 18:28:43 2006
From: it at cimgroup.com (IT)
Date: Tue, 12 Sep 2006 15:28:43 -0700
Subject: [cisco-voip] vulnerable gateway?
In-Reply-To: <08bc01c6d6b9$73582c50$c8a8a8c0@izmir>
Message-ID: <B487D4C1FE6D04439061018C6EBC109B03D5EFD6@exchange.cimgroup.com>

Actually, I tried both UDP and TCP.
Would it still show up under a portscan? TAC seems to think they came in
through voicemail...

-----Original Message-----
From: cumbur [mailto:zeus at cumhur.com] On Behalf Of puckcisco at cumhur.com
Sent: Tuesday, September 12, 2006 3:19 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

Dear Avidan,

H323 use TCP 1720 (not udp)  port for call initiation also don't forget
to
block SIP ports TCP/UDP 5060.

Regards.
Cumhur

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, September 13, 2006 12:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] vulnerable gateway?

I just got a call from my long distance provider that someone has been
using my PRI for many international calls. I check my CDR database
tables, and it appears that calls have been coming from one of my branch
office 2801's. But, in the CDR table, the origDeviceName alternates
between the name of the gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those
ports open, because when I tried to do a "access-list 100 deny udp any
any eq 1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up
these gateways in the first place locked them down, but hindsight is
20/20.

Thanks,
Avidan

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



-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date:
12/09/2006




From Scott.Voll at wesd.org  Tue Sep 12 18:36:53 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 12 Sep 2006 15:36:53 -0700
Subject: [cisco-voip] vulnerable gateway?
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C83DE@Apollo.wesd.org>

I would agree with TAC per your CDR of CiscoUM-VI1.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Tuesday, September 12, 2006 3:29 PM
To: puckcisco at cumhur.com; IT; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

Actually, I tried both UDP and TCP.
Would it still show up under a portscan? TAC seems to think they came in
through voicemail...

-----Original Message-----
From: cumbur [mailto:zeus at cumhur.com] On Behalf Of puckcisco at cumhur.com
Sent: Tuesday, September 12, 2006 3:19 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

Dear Avidan,

H323 use TCP 1720 (not udp)  port for call initiation also don't forget
to
block SIP ports TCP/UDP 5060.

Regards.
Cumhur

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, September 13, 2006 12:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] vulnerable gateway?

I just got a call from my long distance provider that someone has been
using my PRI for many international calls. I check my CDR database
tables, and it appears that calls have been coming from one of my branch
office 2801's. But, in the CDR table, the origDeviceName alternates
between the name of the gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those
ports open, because when I tried to do a "access-list 100 deny udp any
any eq 1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up
these gateways in the first place locked them down, but hindsight is
20/20.

Thanks,
Avidan

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



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Checked by AVG Free Edition.
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12/09/2006



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


From it at cimgroup.com  Tue Sep 12 18:42:04 2006
From: it at cimgroup.com (IT)
Date: Tue, 12 Sep 2006 15:42:04 -0700
Subject: [cisco-voip] vulnerable gateway?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A6066C83DE@Apollo.wesd.org>
Message-ID: <B487D4C1FE6D04439061018C6EBC109B03D5F003@exchange.cimgroup.com>

But where in unity is someone able to route their call to any arbitrary
phone number?

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, September 12, 2006 3:37 PM
To: IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

I would agree with TAC per your CDR of CiscoUM-VI1.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Tuesday, September 12, 2006 3:29 PM
To: puckcisco at cumhur.com; IT; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

Actually, I tried both UDP and TCP.
Would it still show up under a portscan? TAC seems to think they came in
through voicemail...

-----Original Message-----
From: cumbur [mailto:zeus at cumhur.com] On Behalf Of puckcisco at cumhur.com
Sent: Tuesday, September 12, 2006 3:19 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

Dear Avidan,

H323 use TCP 1720 (not udp)  port for call initiation also don't forget
to
block SIP ports TCP/UDP 5060.

Regards.
Cumhur

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, September 13, 2006 12:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] vulnerable gateway?

I just got a call from my long distance provider that someone has been
using my PRI for many international calls. I check my CDR database
tables, and it appears that calls have been coming from one of my branch
office 2801's. But, in the CDR table, the origDeviceName alternates
between the name of the gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those
ports open, because when I tried to do a "access-list 100 deny udp any
any eq 1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up
these gateways in the first place locked them down, but hindsight is
20/20.

Thanks,
Avidan

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



-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date:
12/09/2006



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


From puckcisco at cumhur.com  Tue Sep 12 18:52:21 2006
From: puckcisco at cumhur.com (puckcisco at cumhur.com)
Date: Wed, 13 Sep 2006 01:52:21 +0300
Subject: [cisco-voip] vulnerable gateway?
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B03D5F003@exchange.cimgroup.com>
Message-ID: <08c601c6d6be$1b9d1ed0$c8a8a8c0@izmir>

It was old issue but check it.

http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=IP%20Commun
ications%20and%20Video&topic=Unified%20Communications&CommCmd=MB%3Fcmd%3Dpas
s_through%26location%3Doutline%40%5E1%40%40.1dd617c0/0#selected_message


-----Original Message-----
From: IT [mailto:it at cimgroup.com] 
Sent: Wednesday, September 13, 2006 1:42 AM
To: Voll, Scott; IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

But where in unity is someone able to route their call to any arbitrary
phone number?

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org]
Sent: Tuesday, September 12, 2006 3:37 PM
To: IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

I would agree with TAC per your CDR of CiscoUM-VI1.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Tuesday, September 12, 2006 3:29 PM
To: puckcisco at cumhur.com; IT; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

Actually, I tried both UDP and TCP.
Would it still show up under a portscan? TAC seems to think they came in
through voicemail...

-----Original Message-----
From: cumbur [mailto:zeus at cumhur.com] On Behalf Of puckcisco at cumhur.com
Sent: Tuesday, September 12, 2006 3:19 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

Dear Avidan,

H323 use TCP 1720 (not udp)  port for call initiation also don't forget to
block SIP ports TCP/UDP 5060.

Regards.
Cumhur

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, September 13, 2006 12:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] vulnerable gateway?

I just got a call from my long distance provider that someone has been using
my PRI for many international calls. I check my CDR database tables, and it
appears that calls have been coming from one of my branch office 2801's.
But, in the CDR table, the origDeviceName alternates between the name of the
gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those ports
open, because when I tried to do a "access-list 100 deny udp any any eq
1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up these
gateways in the first place locked them down, but hindsight is 20/20.

Thanks,
Avidan

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



--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date:
12/09/2006



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



-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date: 12/09/2006



From puckcisco at cumhur.com  Tue Sep 12 18:59:15 2006
From: puckcisco at cumhur.com (puckcisco at cumhur.com)
Date: Wed, 13 Sep 2006 01:59:15 +0300
Subject: [cisco-voip] vulnerable gateway?
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B03D5F003@exchange.cimgroup.com>
Message-ID: <08c701c6d6bf$1366a730$c8a8a8c0@izmir>

Also read this article, check passwords and other details.

http://neworder.box.sk/newsread.php?newsid=13082
 
Try to trace your CDR maybe you will find entry point.

-----Original Message-----
From: IT [mailto:it at cimgroup.com] 
Sent: Wednesday, September 13, 2006 1:42 AM
To: Voll, Scott; IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

But where in unity is someone able to route their call to any arbitrary
phone number?

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org]
Sent: Tuesday, September 12, 2006 3:37 PM
To: IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

I would agree with TAC per your CDR of CiscoUM-VI1.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Tuesday, September 12, 2006 3:29 PM
To: puckcisco at cumhur.com; IT; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

Actually, I tried both UDP and TCP.
Would it still show up under a portscan? TAC seems to think they came in
through voicemail...

-----Original Message-----
From: cumbur [mailto:zeus at cumhur.com] On Behalf Of puckcisco at cumhur.com
Sent: Tuesday, September 12, 2006 3:19 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

Dear Avidan,

H323 use TCP 1720 (not udp)  port for call initiation also don't forget to
block SIP ports TCP/UDP 5060.

Regards.
Cumhur

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, September 13, 2006 12:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] vulnerable gateway?

I just got a call from my long distance provider that someone has been using
my PRI for many international calls. I check my CDR database tables, and it
appears that calls have been coming from one of my branch office 2801's.
But, in the CDR table, the origDeviceName alternates between the name of the
gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those ports
open, because when I tried to do a "access-list 100 deny udp any any eq
1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up these
gateways in the first place locked them down, but hindsight is 20/20.

Thanks,
Avidan

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



--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date:
12/09/2006



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



-- 
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date: 12/09/2006



From rvillagran at coasin.cl  Tue Sep 12 19:34:14 2006
From: rvillagran at coasin.cl (Rodrigo Villagran)
Date: Tue, 12 Sep 2006 19:34:14 -0400
Subject: [cisco-voip] Cabling Standards for IPT
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F003135FE7@USNAEXCH.na.didata.local>
Message-ID: <200609122332.k8CNW8J5004362@mail.coasin.cl>

Hi,
I've do some installations with Cat3, but as a recommendation don't use it
with a PC behind the phone, it cause some issues........bad voice quality.

Regards
Rodrigo
-----Mensaje original-----
De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de Jason Aarons (US)
Enviado el: jueves, 07 de septiembre de 2006 17:03
Para: ciscovoip Voip
Asunto: Re: [cisco-voip] Cabling Standards for IPT

I have a couple phones in warehouses running on Category3 at 10/Half, it
works, powered locally, it was too expensive to run a Cat5 run. Not a
good idea in my view, but hey it works.

The specs show Cat3 in Table A-1 Cisco IP Phone Specifications  
http://www.cisco.com/en/US/products/hw/phones/ps379/products_administrat
ion_guide_chapter09186a00800b607e.html#wp1003719



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Thursday, September 07, 2006 2:13 PM
To: tusker keg
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Cabling Standards for IPT

I believe cat5e is the minimum requirement.

/Wes

tusker keg wrote:
> Howdy'
> 
> Was wondering if there are any standards /BCP references out there for
> a successful IPT deployment as far LAN cabling is concerned
> 
> 
> Regards
> 
> ./tusker
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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

This e-mail communication and any attachments may contain
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intended addressee, you are hereby notified that you have received
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this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
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_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip


From erickbe at yahoo.com  Tue Sep 12 21:22:28 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue, 12 Sep 2006 18:22:28 -0700 (PDT)
Subject: [cisco-voip] Delayed audio-cutover with H.323 gateway
In-Reply-To: <4507030D.1040700@cisco.com>
Message-ID: <20060913012228.72422.qmail@web37206.mail.mud.yahoo.com>

Ok. I have a couple traces I am going through (CCM and debugs from router). 

The interesting part is the user is hearing a clicking sound when the delay happens (like someone is hanging up a phone) when the audio cuts in and becomes hearable. Sometimes it is very loud. Sometimes not as loud. 

When they transfer a call to another IP Phone, usually when the other person picks up the caller on PSTN side doesn't hear them for 4 to 8 seconds. Sometimes it works fine with no delay. I am leaning more toward a GW or network issue somewhere at this point. Calls on PRI's do not have this problem - just analog ports.

----- Original Message ----
From: Wes Sisk <wsisk at cisco.com>
To: Erick Bergquist <erickbe at yahoo.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Sent: Tuesday, September 12, 2006 1:57:17 PM
Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway

Fast start should not apply here.  make sure you are not dropping 
packets or retransmitting due to network loss between the 2 but fast 
start should only be necessary when you're looking at 40msec+ RTT gw to 
CM.  You have problems either on the GW or on the CM with getting those 
channels opened fast enough.

off the wall idea - enable 'file close thread flag' to make sure you 
aren't hitting that issue.  then get CCM and SDL traces from an 
occurrence.  look in the traces to obeserve the time from user offhook 
to startmediatransmittion to phone and open receive channel ack from the 
phone.  If that is small, then it is a problem with audio cut through on 
the gw itself.

/Wes

Erick Bergquist wrote:
> 4 to 8ms on average. Fast start not enabled on GW - let me look into that.
> 
> ----- Original Message ----
> From: Wes Sisk <wsisk at cisco.com>
> To: Erick Bergquist <erickbe at yahoo.com>
> Cc: ciscovoip <cisco-voip at puck.nether.net>
> Sent: Tuesday, September 12, 2006 8:31:19 AM
> Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway
> 
> What is round trip time from gateway to CM?  Are you using fast start  
> on CM and the gateway?
> 
> /Wes
> 
> On Sep 11, 2006, at 10:19 PM, Erick Bergquist wrote:
> 
> Hi,
> 
> I'm having a issue where an inbound call comes in FXO port and  
> connection plar's to a shared DN across 5 IP Phones. The phone call  
> gets answered fine, but the user on IP phone saids "Welcome to  
> BlahBlah" and the outside caller doesn't hear the first time they say  
> "Welcome To BlahBlah" and the IP Phone user repeats their greeting  
> and they hear it the second time fine.
> 
> There are no errors on any interfaces that I can see, and the gateway  
> has voice rtp send-recv enabled. Issue does not happen on calls over  
> PRI, just the analog FXOs. I have disabled caller-id on the voice- 
> port and also tried connection plar opx with no difference. Sometimes  
> it works fine but rarely.
> 
> CCM is version 4.1(3)sr3c and the gateway is a 1760 running 12.4(8a).
> 
> A similar problem can happen when the call is parked or put on hold  
> also, now and then.
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 





From erickbe at yahoo.com  Tue Sep 12 21:37:45 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue, 12 Sep 2006 18:37:45 -0700 (PDT)
Subject: [cisco-voip] Phone settings page,
	Restricted option for settings drop down gone
Message-ID: <20060913013745.63102.qmail@web37204.mail.mud.yahoo.com>

Under the phone configuration, for settings access the drop down only has Enabled/Disabled now and the Restricted setting is no longer listed.  Is there a global setting somewhere or something else to disable this from showing up , or something else going on?  I have looked around and tried to find a doc on cisco but so far have pulled up nothing. This is on CCM 4.0(2a). 

Thanks, Erick


From zthoran at yahoo.com  Tue Sep 12 22:00:18 2006
From: zthoran at yahoo.com (Zevier Thoran)
Date: Tue, 12 Sep 2006 19:00:18 -0700 (PDT)
Subject: [cisco-voip] Mysterious Inbound calls
Message-ID: <20060913020018.94070.qmail@web37107.mail.mud.yahoo.com>

Hi,
Every evening at 5pm, My Callmanager system gets hit with about 20 anonymous calls on the operator number. On answering a call the operator just gets dead air.
Any suggestions how to solve this problem?
Regards - Zevier.

 		
---------------------------------
Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2?/min or less.
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From lelio at uoguelph.ca  Tue Sep 12 22:27:54 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 12 Sep 2006 22:27:54 -0400
Subject: [cisco-voip] Managing DID's ?
References: <a1b07ea60609121414l7fda34aas1222a5c9dccfea58@mail.gmail.com>
Message-ID: <001d01c6d6dc$390313d0$ca01a8c0@deadfish>

I hear ya on this one. Others have mentioned the route plan report, which is very helpful, but it's not a true DN management system. Something that could easily be built in though is an 'assign next available number' from a range you have programmed. The problem comes, with us anyways, where you are managing multiple systems.

I think a true number management system is too complex to be included in CallManager. Too many variables. 


  ----- Original Message ----- 
  From: TechGuy 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, September 12, 2006 5:14 PM
  Subject: [cisco-voip] Managing DID's ?


  I have one major issue/complaint if you will with callmanager, and
  that is in regards to managing DID's or extension ranges or whatever
  you want to call it.

  I have no experience with any other phone systems or PBX's so nothing
  to base this on but my own personal experience of having to manage
  callmanager over the years.

  Just seems there should be a way to manage them both DID's /
  extensions.  I am refering to which are used or not.  I guess I think
  of it in terms of DHCP kinda.  You put in the given range your working
  with, and you can easily add and delete from within that range and see
  statistics based on that range.  Such as how many are used, or
  available.  When you delete a user/phone/extension maybe it goes back
  into this available pool or something.

  Right now all I can do is search say extension beginning with 145 and
  see if anything is availble between 1450 - 1459 and so on.  I suspect
  everyone has to deal with this same mess and I have to wonder if there
  is a better way that I am simply not aware of.

  Real nightmare for me right now due to acquisitions of others and
  using their DID ranges which now seem to be all over the place for me.
   I have ranges using every number except 0 and 8, 8 is out outside
  access number.  So I am running into issues now trying to make it all
  work with new offices and what not.  I blame the person who setup four
  digit extensions.

  Thoughts and suggestions?
  _______________________________________________
  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 Sep 12 22:39:24 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 12 Sep 2006 22:39:24 -0400
Subject: [cisco-voip] Managing DID's ?
References: <a1b07ea60609121414l7fda34aas1222a5c9dccfea58@mail.gmail.com>
	<001d01c6d6dc$390313d0$ca01a8c0@deadfish>
Message-ID: <003001c6d6dd$d360c9d0$ca01a8c0@deadfish>

on that note, does anyone out there have any number mangement system they use?
  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: TechGuy ; cisco-voip at puck.nether.net 
  Sent: Tuesday, September 12, 2006 10:27 PM
  Subject: Re: [cisco-voip] Managing DID's ?


  I hear ya on this one. Others have mentioned the route plan report, which is very helpful, but it's not a true DN management system. Something that could easily be built in though is an 'assign next available number' from a range you have programmed. The problem comes, with us anyways, where you are managing multiple systems.

  I think a true number management system is too complex to be included in CallManager. Too many variables. 


    ----- Original Message ----- 
    From: TechGuy 
    To: cisco-voip at puck.nether.net 
    Sent: Tuesday, September 12, 2006 5:14 PM
    Subject: [cisco-voip] Managing DID's ?


    I have one major issue/complaint if you will with callmanager, and
    that is in regards to managing DID's or extension ranges or whatever
    you want to call it.

    I have no experience with any other phone systems or PBX's so nothing
    to base this on but my own personal experience of having to manage
    callmanager over the years.

    Just seems there should be a way to manage them both DID's /
    extensions.  I am refering to which are used or not.  I guess I think
    of it in terms of DHCP kinda.  You put in the given range your working
    with, and you can easily add and delete from within that range and see
    statistics based on that range.  Such as how many are used, or
    available.  When you delete a user/phone/extension maybe it goes back
    into this available pool or something.

    Right now all I can do is search say extension beginning with 145 and
    see if anything is availble between 1450 - 1459 and so on.  I suspect
    everyone has to deal with this same mess and I have to wonder if there
    is a better way that I am simply not aware of.

    Real nightmare for me right now due to acquisitions of others and
    using their DID ranges which now seem to be all over the place for me.
     I have ranges using every number except 0 and 8, 8 is out outside
    access number.  So I am running into issues now trying to make it all
    work with new offices and what not.  I blame the person who setup four
    digit extensions.

    Thoughts and suggestions?
    _______________________________________________
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    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip



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


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From wsisk at cisco.com  Tue Sep 12 22:41:29 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 12 Sep 2006 22:41:29 -0400
Subject: [cisco-voip] Speed Dial 9+456+variable
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0032146E2@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0032146E2@USNAEXCH.na.didata.local>
Message-ID: <189BF100-559E-4E50-BD7E-328DC54E814F@cisco.com>

sounds like a translation pattern?  *.XXXX with predot discard and  
prefix 9456
then have users dial * followed by 4 digit number

/Wes

On Sep 12, 2006, at 4:42 PM, Jason Aarons ((US)) wrote:

Customer?s Avaya system allowed them to dial 9 plus 456 plus any  
variable length of 4 digits for a speed dial button.



Any thoughts how to mimic this in CallManager?



Eg Press Speedial then 6756 would dial 94566756





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From wsisk at cisco.com  Tue Sep 12 22:44:21 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 12 Sep 2006 22:44:21 -0400
Subject: [cisco-voip] Mysterious Inbound calls
In-Reply-To: <20060913020018.94070.qmail@web37107.mail.mud.yahoo.com>
References: <20060913020018.94070.qmail@web37107.mail.mud.yahoo.com>
Message-ID: <C159D7F3-FDBA-4819-B28A-C50815DA044B@cisco.com>

Do you use Unity voicemail?  Is your the operator's specific  
extension (any hunt pilot that routes to that phone) configured under  
the operator CH in Unity?

/Wes

On Sep 12, 2006, at 10:00 PM, Zevier Thoran wrote:

Hi,
Every evening at 5pm, My Callmanager system gets hit with about 20  
anonymous calls on the operator number. On answering a call the  
operator just gets dead air.
Any suggestions how to solve this problem?
Regards - Zevier.

Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30 
+ countries) for 2?/min or less.
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From wsisk at cisco.com  Tue Sep 12 22:47:24 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 12 Sep 2006 22:47:24 -0400
Subject: [cisco-voip] vulnerable gateway?
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B03D5EF84@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B03D5EF84@exchange.cimgroup.com>
Message-ID: <AAA48F2B-73C8-45F7-B48D-1B12E2D6E596@cisco.com>

the calls from CiscoUM-VI1 are most likely someone hitting your Unity  
Voicemail Auto Attendant and transferring out to an international  
number.

/Wes
On Sep 12, 2006, at 5:59 PM, IT wrote:

I just got a call from my long distance provider that someone has been
using my PRI for many international calls. I check my CDR database
tables, and it appears that calls have been coming from one of my branch
office 2801's. But, in the CDR table, the origDeviceName alternates
between the name of the gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those
ports open, because when I tried to do a "access-list 100 deny udp any
any eq 1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up
these gateways in the first place locked them down, but hindsight is
20/20.

Thanks,
Avidan

_______________________________________________
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From wsisk at cisco.com  Tue Sep 12 22:51:15 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 12 Sep 2006 22:51:15 -0400
Subject: [cisco-voip] Managing DID's ?
In-Reply-To: <a1b07ea60609121414l7fda34aas1222a5c9dccfea58@mail.gmail.com>
References: <a1b07ea60609121414l7fda34aas1222a5c9dccfea58@mail.gmail.com>
Message-ID: <01B17239-CEFC-4320-991D-3CD90C02DCB7@cisco.com>

Unfortunately not as straight forward as it sounds because you can do  
trunks off to voice gateways or other PBX's and use virtual  
addressing on those.

Route Plan report is about the best approach going.  Use a  
translation pattern like XXXX to intercept the unassigned numbers and  
route them to your preferred call treatment.

/Wes
On Sep 12, 2006, at 5:14 PM, TechGuy wrote:

I have one major issue/complaint if you will with callmanager, and
that is in regards to managing DID's or extension ranges or whatever
you want to call it.

I have no experience with any other phone systems or PBX's so nothing
to base this on but my own personal experience of having to manage
callmanager over the years.

Just seems there should be a way to manage them both DID's /
extensions.  I am refering to which are used or not.  I guess I think
of it in terms of DHCP kinda.  You put in the given range your working
with, and you can easily add and delete from within that range and see
statistics based on that range.  Such as how many are used, or
available.  When you delete a user/phone/extension maybe it goes back
into this available pool or something.

Right now all I can do is search say extension beginning with 145 and
see if anything is availble between 1450 - 1459 and so on.  I suspect
everyone has to deal with this same mess and I have to wonder if there
is a better way that I am simply not aware of.

Real nightmare for me right now due to acquisitions of others and
using their DID ranges which now seem to be all over the place for me.
  I have ranges using every number except 0 and 8, 8 is out outside
access number.  So I am running into issues now trying to make it all
work with new offices and what not.  I blame the person who setup four
digit extensions.

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


From erickbe at yahoo.com  Tue Sep 12 23:02:35 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue, 12 Sep 2006 20:02:35 -0700 (PDT)
Subject: [cisco-voip] vulnerable gateway?
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B03D5F003@exchange.cimgroup.com>
Message-ID: <20060913030235.43498.qmail@web37215.mail.mud.yahoo.com>

You can limit this with the restriction tables in unity and ultimately with the CSS set on the Call Manager VM Port configuration. 

----- Original Message ----
From: IT <it at cimgroup.com>
To: "Voll, Scott" <Scott.Voll at wesd.org>; IT <it at cimgroup.com>; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Sent: Tuesday, September 12, 2006 5:42:04 PM
Subject: Re: [cisco-voip] vulnerable gateway?

But where in unity is someone able to route their call to any arbitrary
phone number?

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, September 12, 2006 3:37 PM
To: IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

I would agree with TAC per your CDR of CiscoUM-VI1.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Tuesday, September 12, 2006 3:29 PM
To: puckcisco at cumhur.com; IT; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

Actually, I tried both UDP and TCP.
Would it still show up under a portscan? TAC seems to think they came in
through voicemail...

-----Original Message-----
From: cumbur [mailto:zeus at cumhur.com] On Behalf Of puckcisco at cumhur.com
Sent: Tuesday, September 12, 2006 3:19 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

Dear Avidan,

H323 use TCP 1720 (not udp)  port for call initiation also don't forget
to
block SIP ports TCP/UDP 5060.

Regards.
Cumhur

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, September 13, 2006 12:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] vulnerable gateway?

I just got a call from my long distance provider that someone has been
using my PRI for many international calls. I check my CDR database
tables, and it appears that calls have been coming from one of my branch
office 2801's. But, in the CDR table, the origDeviceName alternates
between the name of the gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those
ports open, because when I tried to do a "access-list 100 deny udp any
any eq 1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up
these gateways in the first place locked them down, but hindsight is
20/20.

Thanks,
Avidan

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



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Checked by AVG Free Edition.
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12/09/2006



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From zthoran at yahoo.com  Tue Sep 12 23:32:41 2006
From: zthoran at yahoo.com (Zevier Thoran)
Date: Tue, 12 Sep 2006 20:32:41 -0700 (PDT)
Subject: [cisco-voip] Mysterious Inbound calls
In-Reply-To: <C159D7F3-FDBA-4819-B28A-C50815DA044B@cisco.com>
Message-ID: <20060913033241.49264.qmail@web37101.mail.mud.yahoo.com>

Hi Wes,
I am just taking over this system and not fully familiar...but was told that all these calls do come in via a press press '0' for Operator Unity Auto Attendant Call Handler. But why should these calls only come in at 5pm? Is their a fix?
Many thanks in advance......./Zevier

Wes Sisk <wsisk at cisco.com> wrote: Do you use Unity voicemail?  Is your the operator's specific extension (any hunt pilot that routes to that phone) configured under the operator CH in Unity?

/Wes

On Sep 12, 2006, at 10:00 PM, Zevier Thoran wrote:

Hi,
Every evening at 5pm, My Callmanager system gets hit with about 20 anonymous calls on the operator number. On answering a call the operator just gets dead air.
Any suggestions how to solve this problem?
Regards - Zevier.
   


---------------------------------
Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2??/min or less._______________________________________________
cisco-voip mailing list
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From manoj.kalpage at gmail.com  Wed Sep 13 05:19:45 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Wed, 13 Sep 2006 18:19:45 +0900
Subject: [cisco-voip] ISP and VPN Failover for Call Manager based VOIP
	network
In-Reply-To: <bbe09c1d0609120004m2451c068ma29bf5a6c1f36364@mail.gmail.com>
References: <bbe09c1d0609120004m2451c068ma29bf5a6c1f36364@mail.gmail.com>
Message-ID: <bbe09c1d0609130219k29481ba1p427a9d81bb5b9ffe@mail.gmail.com>

 Dear All,
I am looking for ISP fail over for VoIP network. We have small
enterprise VoIP network. If I explain our network bit, Basically we
have call manager and unity server in main office with PIX515. All the
branch offices has PIX 501. With attached fail over solution I am going to
create two tunnels from each branch office and have them connected to each
firewall in main office. I think this way if one PIX515 fail at main office,
still branch office can be connected through second PIX515. Bellow is the
router configuration for routing between two PIX 515. This configuration
itself doesn't mean anything without looking at a diagram.I need to test
this but I don't have enough gears with me right now and also I don't have
100% confidence on this. So, I would like to share with you folks. Any
comments and ideas would be greatly appreciated.

Please find the diagram bellow link (Sorry it's han written one )
http://proxy.f2.ymdb.yahoofs.jp/bc/857e55a/bc/bd7f/failover.jpg?bcQM9BFBNirrJIWq

best regards,
Manoj


ip cef

!####Establish sla monitors for use in tracking objects####!

ip sla monitor 1
type echo protocol ipIcmpEcho 174.16.0.1
threshold 3
frequency 5
ip sla monitor schedule 1 life forever start-time now

ip sla monitor 2
type echo protocol ipIcmpEcho 173.16.0.1
threshold 3
frequency 5
ip sla monitor schedule 2 life forever start-time now
!

!####Configure Tracking objects (referencing IP SLA monitor's above)####!

track 101 rtr 1 reachability
!
track 102 rtr 2 reachability
!
!
!
!
!####Configure Interfaces with NAT####!

interface FastEthernet 0/1
ip address 172.16.0.1 255.255.0.0
ip nat inside

!
interface Fastethernet 0/0
ip address 173.16.0.2 255.255.255.0
ip nat outside

!
interface Fastethernet 0/2
ip address 174.16.0.2 255.255.255.0
ip nat outside

!
ip classless
!####Configure gateway of last resort with tracking objects####!
ip route 0.0.0.0 0.0.0.0 173.16.0.1 track 101
ip route 0.0.0.0 0.0.0.0 174.16.0.1 track 102

!####Configure NAT statements for most outbound traffic####!
ip nat inside source route-map ISP1 interface FastEthernet 0/0 overload
ip nat inside source route-map ISP2 interface FastEthernet 0/2 overload

!
access-list 10 permit 172.16.0.0 0.0.0.255
access-list 101 permit icmp any host 173.16.0.1 echo
access-list 102 permit icmp any host 174.16.0.1 echo

!
!####Configure route maps for reference in NAT statements####!
route-map ISP2 permit 10
match ip address 10
match interface Fastethernet 0/1
!
route-map ISP1 permit 10
match ip address 10
match interface Fastethernet 0/0
!
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From manoj.kalpage at gmail.com  Wed Sep 13 05:21:53 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Wed, 13 Sep 2006 18:21:53 +0900
Subject: [cisco-voip] Internet IP phone connect through PIX Firewall
In-Reply-To: <4505A51A.10706@cisco.com>
References: <01C8F21F30DE9740B53FD2EF05C3A8FC56F55E@SVCEXCH01.firm.fenwick.llp>
	<33AFDCDE-4C00-43F3-82E3-47D6423A4B2F@cisco.com>
	<012301c6d5b9$37acdee0$6500a8c0@computec.lk>
	<4505A51A.10706@cisco.com>
Message-ID: <bbe09c1d0609130221u303cc36flc827b7ed5095b029@mail.gmail.com>

Thanks Wes, I will try this and post the outcome.



On 9/12/06, Wes Sisk <wsisk at cisco.com> wrote:
>
> I believe this is what you need:
>
> http://www.cisco.com/en/US/partner/products/sw/secursw/ps2120/products_configuration_guide_chapter09186a008017278b.html#wp1063720
>
> and this, but with inside/outsider reversed:
> http://www.ciscotaccc.com/kaidara-advisor/security/showcase?case=K72511039
>
> /Wes
>
> Manoj Kalpage wrote:
> > Wes,
> > Thank You for the reply. Yes I have NAT on my PIX. Regarding DNS fixup,
> > do i need additional settings on my PIX except the fixup protocol dns
> > maximum-length 512 ?
> >
> > Best Regards,
> > Manoj
> >
> >
> >     ----- Original Message -----
> >     *From:* Wes Sisk <mailto:wsisk at cisco.com>
> >     *To:* Manoj Kalpage <mailto:manoj.kalpage at gmail.com>
> >     *Cc:* ciscovoip Voip <mailto:cisco-voip at puck.nether.net> ; Stu
> >     Packett <mailto:SPackett at fenwick.com>
> >     *Sent:* Sunday, September 10, 2006 10:28 PM
> >     *Subject:* Re: [cisco-voip] Internet IP phone connect through PIX
> >     Firewall
> >
> >     Manoj,
> >
> >     Are you doing NAT on your PIX?  If so, you will need special CM+PIX
> >     config.
> >
> >     phones download SEP<mac>.cnf.xml from TFTP server.  Inside this XML
> >     file is a listing of which CM servers the phone should register to
> >     using SCCP.  This file also tells the phone what TCP port to use for
> >     the SCCP communication.  The CM servers are listed in this by name
> >     or IP address based on how your CM is configured under
> system->server.
> >
> >     For NAT traversal, your CM will have to be configured using host
> >     names.  Your PIX will have to do DNS fixup.  Your phone will receive
> >     the name cm1.manoj.com.  the phone must do DNS lookup on this and
> >     receive the external address of your CM server.
> >
> >     Otherwise, you must use a valid internet address for the IP of your
> >     CM server.
> >
> >     /Wes
> >     On Sep 9, 2006, at 3:41 PM, Stu Packett wrote:
> >
> >     Sorry, I have never tried without the VPN.  I thought best practice
> >     was to use the VPN because it was not advised to put the CCM on the
> >     public internet.  If you do get your config working, I'd like to get
> >     a copy of your config just for reference.  Thanks.
> >
> >
> ------------------------------------------------------------------------
> >     *From:* Manoj Kalpage [mailto:manoj.kalpage at gmail.com]
> >     *Sent:* Saturday, September 09, 2006 12:20 AM
> >     *To:* Stu Packett
> >     *Cc:* cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> >     *Subject:* Re: [cisco-voip] Internet IP phone connect through PIX
> >     Firewall
> >
> >     Stu,
> >     Thank you for the reply, I use windows 2003 DHCP server for my
> >     phones in LAN but I can get my outside phone connect to CCM through
> >     internet. Do you have IP phones connect to your CCM through internet
> >     without using VPN?
> >
> >     Thanks,
> >     Manoj
> >
> >
> >
> >     On 9/9/06, *Stu Packett* <SPackett at fenwick.com
> >     <mailto:SPackett at fenwick.com>> wrote:
> >
> >         Manoj:
> >         Is your PIX giving out DHCP addresses?  On my PIX 501, I have it
> >         setup as a DHCP server and these are my DHCP commands:
> >
> >         dhcpd address xxx.xxx.xxx.xxx
> >         dhcpd dns xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
> >         dhcpd wins xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
> >         dhcpd lease 36000
> >         dhcpd ping_timeout 750
> >         dhcpd domain internaldomain.com <http://internaldomain.com/>
> >         dhcpd option 150 ip xxx.xxx.xxx.xxx <--TFTP address
> >         dhcpd enable inside
> >
> >
> ------------------------------------------------------------------------
> >         *From:* cisco-voip-bounces at puck.nether.net
> >         <mailto:cisco-voip-bounces at puck.nether.net> [mailto:
> >         cisco-voip-bounces at puck.nether.net
> >         <mailto:cisco-voip-bounces at puck.nether.net>] *On Behalf Of
> >         *Manoj Kalpage
> >         *Sent:* Friday, September 08, 2006 4:18 AM
> >         *To:* cisco-voip at puck.nether.net <mailto:
> cisco-voip at puck.nether.net>
> >         *Subject:* [cisco-voip] Internet IP phone connect through PIX
> >         Firewall
> >
> >
> >         Hi All,
> >         Does any one has configured PIX firewall to connect internet IP
> >         phones to Call Manager. I have configure firewall to open all
> >         the port which CCM need but still no luck. Bellow is the config
> >         of my PIX. Am i missing anything?
> >
> >         Here is the link I refered to open the TCP and UDP Ports
> >
> >
> http://www.cisco.com/application/pdf/en/us/guest/products/ps5820/c1693/ccmigration_09186a0080536eae.pdf
> >
> >         Thank you in advance.
> >         Manoj
> >
> >         :
> >         PIX Version 6.3(5)
> >         interface ethernet0 auto
> >         interface ethernet1 auto
> >         nameif ethernet0 outside security0
> >         nameif ethernet1 inside security100
> >         enable password u2zabJUOK.TTL3K1 encrypted
> >         passwd 1P5CrRl.dL8Oe4k2 encrypted
> >         hostname PBXLPIX01
> >         domain-name pbxl.jp <http://pbxl.jp/>
> >         clock timezone JST 9
> >         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 pptp 1723
> >         fixup protocol rsh 514
> >         fixup protocol rtsp 554
> >         fixup protocol sip 5060
> >         fixup protocol sip udp 5060
> >         fixup protocol skinny 2000
> >         fixup protocol smtp 25
> >         fixup protocol snmp 161
> >         fixup protocol sqlnet 1521
> >         fixup protocol tftp 69
> >         names
> >         object-group service outbound-tcp tcp
> >           port-object eq www
> >           port-object eq https
> >           port-object eq smtp
> >           port-object eq ftp
> >           port-object eq pop3
> >           port-object eq imap4
> >           port-object eq domain
> >           port-object eq 123
> >           port-object eq ssh
> >           port-object eq citrix-ica
> >         object-group service outbound-udp udp
> >           port-object eq domain
> >           port-object eq ntp
> >         object-group service mail-inbound tcp
> >           port-object eq www
> >           port-object eq https
> >           port-object eq smtp
> >         object-group service VoIP-udp udp
> >           port-object range 16384 32768
> >           port-object eq tftp
> >         object-group service VoIP-tcp tcp
> >           port-object eq 3804
> >           port-object eq 2443
> >           port-object eq 2000
> >           port-object eq www
> >           port-object eq 69
> >           port-object eq https
> >         access-list 102 permit tcp 172.16.0.0 <http://172.16.0.0/>
> >         255.255.0.0 <http://255.255.0.0/> any object-group VoIP-tcp
> >         access-list 102 permit udp 172.16.0.0 <http://172.16.0.0/>
> >         255.255.0.0 <http://255.255.0.0/> any object-group VoIP-udp
> >         access-list 102 permit tcp 172.16.0.0 <http://172.16.0.0/>
> >         255.255.0.0 <http://255.255.0.0/> any object-group outbound-tcp
> >         access-list 102 permit udp 172.16.0.0 <http://172.16.0.0/>
> >         255.255.0.0 <http://255.255.0.0/> any object-group outbound-udp
> >         access-list 101 permit tcp any host 210.81.12.195
> >         <http://210.81.12.195/> object-group mail-inbound
> >         access-list 101 permit tcp any host 210.81.12.196
> >         <http://210.81.12.196/> object-group VoIP-tcp
> >         access-list 101 permit udp any host 210.81.12.196
> >         <http://210.81.12.196/> object-group VoIP-udp
> >         access-list 101 permit tcp any host 210.81.12.197
> >         <http://210.81.12.197/> object-group VoIP-tcp
> >         access-list 101 permit udp any host 210.81.12.197
> >         <http://210.81.12.197/> object-group VoIP-udp
> >
> >         pager lines 24
> >         logging on
> >         logging trap informational
> >         logging host inside 172.16.0.26 <http://172.16.0.26/>
> >         logging host inside 172.16.0.12 <http://172.16.0.12/>
> >         icmp permit any unreachable outside
> >         icmp permit any outside
> >         mtu outside 1500
> >         mtu inside 1500
> >         ip address outside xxx.xxx.xxx.xxx 255.255.255.240
> >         <http://255.255.255.240/>
> >         ip address inside 172.16.0.2 <http://172.16.0.2/> 255.255.0.0
> >         <http://255.255.0.0/>
> >
> >         ip audit info action alarm
> >         ip audit attack action alarm
> >         ip local pool pbxlpool 10.1.0.100-10.1.0.200
> >         pdm locationxxx.xxx.xxx.xxx 255.255.255.255
> >         <http://255.255.255.255/> outside
> >
> >         pdm history enable
> >         arp timeout 14400
> >         global (outside) 1 interface
> >         nat (inside) 0 access-list VPNREMOTE
> >         nat (inside) 1 172.16.0.0 <http://172.16.0.0/> 255.255.0.0
> >         <http://255.255.0.0/> 0 0
> >         static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
> >         255.255.255.255 <http://255.255.255.255/>0 1000
> >         static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
> >         255.255.255.255 <http://255.255.255.255/> 0 1000
> >         static (inside,outside) xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx netmask
> >         255.255.255.255 <http://255.255.255.255/> 0 1000
> >
> >         access-group 101 in interface outside
> >         access-group 102 in interface inside
> >         route outside 0.0.0.0 <http://0.0.0.0/> 0.0.0.0
> >         <http://0.0.0.0/> 210.81.12.193 <http://210.81.12.193/> 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 sip-disconnect 0:02:00 sip-invite 0:03:00
> >         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
> >
> >         aaa authentication ssh console LOCAL
> >
> >         http 172.16.0.12 <http://172.16.0.12/> 255.255.255.255
> >         <http://255.255.255.255/> inside
> >
> >         snmp-server host inside 172.16.0.12 <http://172.16.0.12/>
> >         snmp-server location pbxl-pix-datacentre
> >
> >         snmp-server community pbxl
> >         snmp-server enable traps
> >         floodguard enable
> >
> >         telnet 172.16.0.0 <http://172.16.0.0/> 255.255.0.0
> >         <http://255.255.0.0/> inside
> >         telnet 192.168.0.0 <http://192.168.0.0/> 255.255.255.0
> >         <http://255.255.255.0/> inside
> >         telnet timeout 60
> >         ssh 210.101.94.211 <http://210.101.94.211/> 255.255.255.255
> >         <http://255.255.255.255/> outside
> >         ssh 0.0.0.0 <http://0.0.0.0/> 0.0.0.0 <http://0.0.0.0/> outside
> >         ssh 172.16.0.12 <http://172.16.0.12/> 255.255.255.255
> >         <http://255.255.255.255/> inside
> >         ssh 172.16.0.0 <http://172.16.0.0/> 255.255.0.0
> >         <http://255.255.0.0/> inside
> >         ssh 192.168.1.0 <http://192.168.1.0/>255.255.255.0
> >         <http://255.255.255.0/> inside
> >
> >         ssh timeout 60
> >         console timeout 0
> >         PBXLPIX01(config)#
> >         PBXLPIX01(config)#
> >
> >
> >
> >     _______________________________________________
> >     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
> >
>
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From manoj.kalpage at gmail.com  Wed Sep 13 05:27:40 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Wed, 13 Sep 2006 18:27:40 +0900
Subject: [cisco-voip] ip service...
In-Reply-To: <046001c6d6a7$4a0916f0$47196883@cfs.uoguelph.ca>
References: <050944EE48FF384F9C663F7048C47DA704B4199B@webmail.asheville.k12.nc.us>
	<046001c6d6a7$4a0916f0$47196883@cfs.uoguelph.ca>
Message-ID: <bbe09c1d0609130227i62f7a1cfg649f027d18cf0630@mail.gmail.com>

Sorry to interrupt the post but I am having a problem accessing QRT report
but I got QRT soft Key configured on phone and I can leave the feed back.
Bellow is the error


Microsoft JScript runtime error '800a004c'

Path not found

/CCMService/QRTReportResult.asp, line 109
Best regards,
Manoj



On 9/13/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  It's a softkey that you have to enable via the softkey template
> configuration screen.
>
> device > device settings > softkey template
>
> select your template, then:
>
> highlite callmanager > configure softkey layout
>
> quality report tool (QRT) appears as valid options for the following
> states:
>
> Connected
> Connected Conference
> Connected Transfer
> On Hook
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."
>
>  ----- Original Message -----
> *From:* Tim Reimers <tim.reimers at asheville.k12.nc.us>
>  *To:* Lelio Fulgenzi <lelio at uoguelph.ca>
> *Cc:* Cisco Voip <cisco-voip at puck.nether.net>
> *Sent:* Tuesday, September 12, 2006 4:02 PM
> *Subject:* RE: [cisco-voip] ip service...
>
>
>
>
> I get the message "Document empty or invalid" when I try to run a report..
>
> I see that it's part of Extended Services, and puts a QRT button on the
> phone.
>
> I cannot find any directions for 4.01 on how to configure the CCM so that
> the button becomes available to users...
>
> Anyone have a link on how to enable the QRT function on the phones?
>
> Tim
>
>  ------------------------------
>  *From:* Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> *Sent:* Tuesday, September 12, 2006 12:19 PM
> *To:* Tim Reimers; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] ip service...
>
>
>  QRT
>
> it's a softkey that you can show.
>
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803f52dd.html
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."
>
> ----- Original Message -----
> *From:* Tim Reimers <tim.reimers at asheville.k12.nc.us>
> *To:* cisco-voip at puck.nether.net
> *Sent:* Tuesday, September 12, 2006 12:17 PM
> *Subject:* [cisco-voip] ip service...
>
>
>
>
>
> Hi folks---
>
>
>
> Isn't there a service that can be configured on an IP phone that allows a
> user to immediately report a call in which they had trouble with echo or
> jitter?
>
>
>
> I know the 'info' button shows the codec, etc?but it doesn't log that to a
> technician for later analysis
>
> (and have you ever tried to have a user read the codec info to you? That's
> ?really- fun to listen to?not! )
>
>
>
> Thanks Tim
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From akoul at dataways.gr  Wed Sep 13 07:04:28 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Wed, 13 Sep 2006 14:04:28 +0300
Subject: [cisco-voip] sip phone cannot reset to factory defaults
Message-ID: <4507E5BC.4080504@dataways.gr>

Hi,
cisco depot has send me some 7960 phones to do some lab tests.
The phone has sip image (POS3-06-2-00) and **# does not unlock the 
network config menu to change the static settings that it has.
Also, i tried pressing # as it boots but nothing happens!! It boots 
correctly!!!
Any ideas?

From MLinsemier at apcapital.com  Wed Sep 13 09:04:42 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Wed, 13 Sep 2006 09:04:42 -0400
Subject: [cisco-voip] ISP and VPN Failover for Call Manager based
	VOIPnetwork
In-Reply-To: <bbe09c1d0609130219k29481ba1p427a9d81bb5b9ffe@mail.gmail.com>
Message-ID: <AB0A35F2486F81468971AA68558BE71904CEF16A@smieln01winm01.micoanet.com>

Manoj,

 

Do you currently have private lines or some other circuits
interconnecting your offices or are you planning to use VPN exclusively
for voice and data?  My major concern when using a Cisco PIX for voice
would be Quality of Service.  While the PIX can preserve DSCP values as
they are passed across the tunnels, unless anything has changed in 7.x,
it doesn't have the ability to perform marking, LLQ prioritization, and
traffic shaping.  This means that before any traffic is passed to the
PIX, the device behind it (a switch or router) will have to perform some
of these functions (say marking or traffic shaping).  In regards to LLQ
you are out of luck.  

 

For our Teleworker VPN network we utilize a 2851 at the head-end and
failover site and 871/877 routers at our remotes.  This gives us the
capability to mark, LLQ, and shape traffic at the edge, before it is
passed on to the ISP.  Additionally we utilize DMVPN and GRE to maintain
routing information (EIGRP) and to dynamically handle routing changes
when we loose a VPN link (say to our head-end).  I think you can do some
least cost routing type things on the PIX to achieve the same effect,
but it's much easier in IOS.

 

Your ideas are sound in my opinion.  I'm sure that there are some people
that are handling voice fine using Cisco PIX's however we had mixed
results when we were using them.  Once we moved to the IOS VPN several
of our QoS issues were resolved.  Regardless, you always have to
remember that it still is the Internet and not a private network
connection, so you get what you get.

 

Hope this helps,

 

-Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Manoj Kalpage
Sent: Wednesday, September 13, 2006 5:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ISP and VPN Failover for Call Manager based
VOIPnetwork

 

Dear All,

I am looking for ISP fail over for VoIP network. We have small
enterprise VoIP network. If I explain our network bit, Basically we have
call manager and unity server in main office with PIX515. All the branch
offices has PIX 501. With attached fail over solution I am going to
create two tunnels from each branch office and have them connected to
each firewall in main office. I think this way if one PIX515 fail at
main office, still branch office can be connected through second PIX515.
Bellow is the router configuration for routing between two PIX 515. This
configuration itself doesn't mean anything without looking at a
diagram.I need to test this but I don't have enough gears with me right
now and also I don't have 100% confidence on this. So, I would like to
share with you folks. Any comments and ideas would be greatly
appreciated.

 

Please find the diagram bellow link (Sorry it's han written one )

http://proxy.f2.ymdb.yahoofs.jp/bc/857e55a/bc/bd7f/failover.jpg?bcQM9BFB
NirrJIWq

 

best regards,

Manoj

 


ip cef

!####Establish sla monitors for use in tracking objects####!

ip sla monitor 1
type echo protocol ipIcmpEcho 174.16.0.1 <http://174.16.0.1/> 
threshold 3
frequency 5
ip sla monitor schedule 1 life forever start-time now 

ip sla monitor 2
type echo protocol ipIcmpEcho 173.16.0.1 <http://173.16.0.1/> 
threshold 3
frequency 5
ip sla monitor schedule 2 life forever start-time now 
!

!####Configure Tracking objects (referencing IP SLA monitor's
above)####!

track 101 rtr 1 reachability
!
track 102 rtr 2 reachability
!
!
!
!
!####Configure Interfaces with NAT####!

interface FastEthernet 0/1
ip address 172.16.0.1 <http://172.16.0.1/>  255.255.0.0
<http://255.255.0.0/> 
ip nat inside

!
interface Fastethernet 0/0
ip address 173.16.0.2 <http://173.16.0.2/>  255.255.255.0
<http://255.255.255.0/> 
ip nat outside

!
interface Fastethernet 0/2
ip address 174.16.0.2 <http://174.16.0.2/>  255.255.255.0
<http://255.255.255.0/> 
ip nat outside

!
ip classless
!####Configure gateway of last resort with tracking objects####!
ip route 0.0.0.0 <http://0.0.0.0/>  0.0.0.0 <http://0.0.0.0/>
173.16.0.1 <http://173.16.0.1/>  track 101 
ip route 0.0.0.0 <http://0.0.0.0/>  0.0.0.0 <http://0.0.0.0/>
174.16.0.1 <http://174.16.0.1/>  track 102

!####Configure NAT statements for most outbound traffic####!
ip nat inside source route-map ISP1 interface FastEthernet 0/0 overload
ip nat inside source route-map ISP2 interface FastEthernet 0/2 overload

!
access-list 10 permit 172.16.0.0 <http://172.16.0.0/>  0.0.0.255
<http://0.0.0.255/> 
access-list 101 permit icmp any host 173.16.0.1 <http://173.16.0.1/>
echo
access-list 102 permit icmp any host 174.16.0.1 <http://174.16.0.1/>
echo

!
!####Configure route maps for reference in NAT statements####!
route-map ISP2 permit 10
match ip address 10
match interface Fastethernet 0/1
!
route-map ISP1 permit 10
match ip address 10
match interface Fastethernet 0/0 
! 

 





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

CONFIDENTIALITY STATEMENT

This communication and any attachments are CONFIDENTIAL and may be 
protected by one or more legal privileges. It is intended solely
for the use of the addressee identified above. If you are not 
the intended recipient, any use, disclosure, copying or 
distribution of this communication is UNAUTHORIZED. Neither this 
information block, the typed name of the sender, nor anything 
else in this message is intended to constitute an electronic 
signature unless a specific statement to the contrary is 
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From wsisk at cisco.com  Wed Sep 13 09:47:26 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 13 Sep 2006 09:47:26 -0400
Subject: [cisco-voip] Mysterious Inbound calls
In-Reply-To: <20060913033241.49264.qmail@web37101.mail.mud.yahoo.com>
References: <20060913033241.49264.qmail@web37101.mail.mud.yahoo.com>
Message-ID: <45080BEE.8050407@cisco.com>

Zevier,

There is an issue where:
user calls unity to check voicemail or calls into unity auto attendant
user then receives a inbound call. user places unity on hold to answer 
incoming call.  The 'after greeting' action on the all subscribers 
defaults to send call to opening greeting.  If the opening greeting call 
handler is configured to transfer to operator then the operator will 
receive and no one will be on the other end.

However, this does not explain 20+calls all happening at 5PM.  Please 
open a TAC case and provide traces, details of directory numbers, and times.

/WEs

Zevier Thoran wrote:
> Hi Wes,
> I am just taking over this system and not fully familiar...but was told 
> that all these calls do come in via a press press '0' for Operator Unity 
> Auto Attendant Call Handler. But why should these calls only come in at 
> 5pm? Is their a fix?
> Many thanks in advance......./Zevier
> 
> */Wes Sisk <wsisk at cisco.com>/* wrote:
> 
>     Do you use Unity voicemail?  Is your the operator's specific
>     extension (any hunt pilot that routes to that phone) configured
>     under the operator CH in Unity?
> 
>     /Wes
> 
>     On Sep 12, 2006, at 10:00 PM, Zevier Thoran wrote:
> 
>     Hi,
>     Every evening at 5pm, My Callmanager system gets hit with about 20
>     anonymous calls on the operator number. On answering a call the
>     operator just gets dead air.
>     Any suggestions how to solve this problem?
>     Regards - Zevier.
> 
>     ------------------------------------------------------------------------
>     Yahoo! Messenger with Voice. Make PC-to-Phone Calls
>     <http://us.rd.yahoo.com/mail_us/taglines/postman1/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com>
>     to the US (and 30+ countries) for 2??/min or less.
>     _______________________________________________
>     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
> 
> 
> ------------------------------------------------------------------------
> Stay in the know. Pulse on the new Yahoo.com. Check it out. 
> <http://us.rd.yahoo.com/evt=42974/*http://www.yahoo.com/preview>

From bloemhard_johan at silverfalls.k12.or.us  Wed Sep 13 10:56:16 2006
From: bloemhard_johan at silverfalls.k12.or.us (Johan Bloemhard)
Date: Wed, 13 Sep 2006 07:56:16 -0700
Subject: [cisco-voip] ip service...
Message-ID: <C763DC29A34B6B418AFBA81F75B2633A4DC8@msexch2.silverfalls.k12.or.us>

If you look in the QRTReportResult.asp file, which should be located in
"C:\CiscoWebs\Service", what does it say?  By default, it should say
"folder = fso.GetFolder("c:\\Program Files\\Cisco\\QRT");"

 

Verify that "c:\Program Files\Cisco\QRT" exists.  If it doesn't, then
QRT may not be setup correctly.  Verify your log file settings under
Service Parameters->Cisco Extended Functions, Log File settings.

 

If your log file settings are different, you may have to change either
the log file settings to match the line in the ASP file or vice versa.

 

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

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Manoj Kalpage
Sent: Wednesday, September 13, 2006 2:28 AM
To: Lelio Fulgenzi
Cc: Cisco Voip
Subject: Re: [cisco-voip] ip service...

 

Sorry to interrupt the post but I am having a problem accessing QRT
report but I got QRT soft Key configured on phone and I can leave the
feed back. Bellow is the error

  

Microsoft JScript runtime error '800a004c' 

Path not found 

/CCMService/QRTReportResult.asp, line 109 

Best regards,

Manoj



 

On 9/13/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote: 

It's a softkey that you have to enable via the softkey template
configuration screen.

 

device > device settings > softkey template

 

select your template, then:

 

highlite callmanager > configure softkey layout 

 

quality report tool (QRT) appears as valid options for the following
states:

 

Connected

Connected Conference

Connected Transfer

On Hook

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."

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

From: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  

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

Cc: Cisco Voip <mailto:cisco-voip at puck.nether.net>  

Sent: Tuesday, September 12, 2006 4:02 PM

Subject: RE: [cisco-voip] ip service...


 

 

 

I get the message "Document empty or invalid" when I try to run a
report..

 

I see that it's part of Extended Services, and puts a QRT button on the
phone.

 

I cannot find any directions for 4.01 on how to configure the CCM so
that the button becomes available to users...

 

Anyone have a link on how to enable the QRT function on the phones?

 

Tim

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca ] 
Sent: Tuesday, September 12, 2006 12:19 PM
To: Tim Reimers; cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] ip service...

 

QRT

 

it's a softkey that you can show.

 

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administra
tion_guide_chapter09186a00803f52dd.html 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN) 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."

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

	From: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  

	To: cisco-voip at puck.nether.net 

	Sent: Tuesday, September 12, 2006 12:17 PM

	Subject: [cisco-voip] ip service...

	
	 

	 

	Hi folks---

	 

	Isn't there a service that can be configured on an IP phone that
allows a user to immediately report a call in which they had trouble
with echo or jitter? 

	 

	I know the 'info' button shows the codec, etc-but it doesn't log
that to a technician for later analysis

	(and have you ever tried to have a user read the codec info to
you? That's -really- fun to listen to...not! )

	 

	Thanks Tim

	
________________________________


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


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______________________
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From cbaird at ipcelerate.com  Wed Sep 13 10:59:47 2006
From: cbaird at ipcelerate.com (Chris Baird)
Date: Wed, 13 Sep 2006 09:59:47 -0500
Subject: [cisco-voip] 7970s not registering after CCM upgrade to 4.1(2)
Message-ID: <00f001c6d745$421670c0$333210ac@VAIOBAIRD>

We upgraded our CCM last night from 4.0(2)asr2c to 4.1(2).  Now only 4 of 20
7970s will register, the other 16 show unregistered but they are getting a
DCHP address from the CCM.  The 7970s were on 8.02 and we upgraded that to
8.04sr1 but are seeing the same results.  Our 7960s are not seeing any
issues.

 

Any ideas?

 

Thanks,

Chris

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From SCASPER at mandtbank.com  Wed Sep 13 11:05:34 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Wed, 13 Sep 2006 11:05:34 -0400
Subject: [cisco-voip] call quality issues between gateway and Unity
Message-ID: <s507e61a.028@comrcm01intgwia.mandtbank.com>

Wes,
I see that Playout-Delay is not supported on MGCP platforms so how is
jitter handled in a MGCP/Call Manager enviroment?

Thanks!
Steve

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
Kris,

This sounds very similar to something we fixed long ago.  Take a look 

at 'sh voice call' to see if it's jitter:
http://www.cisco.com/en/US/tech/tk652/tk698/ 
technologies_tech_note09186a00800945df.shtml

CSCea35850 - Unity RTP stream has inherent jitter


If you are using non-default packetization periods watch out for
CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and  
G711/30ms

/Wes

On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:

Hi

I have a customer with a multi site centralized CCM 4.13 cluster.   
The remote sites are getting complaints of poor voice quality from  
customers when being transferred to Unity (which is located in the  
central site) .  The callers cannot understand the greetings and  
describe them as choppy.  There are no problems with external calls  
transferred from the remote sites to IP phones at the central site or 

for intersite calls.  It seems to be limited to calls where the RTP  
stream is between a remote voice gateway and Unity.  These calls are  
G.729 so Unity has to transcode the greetings on the fly but if that  
were the problem I'd expect remote users to have the same problem  
when calling into check their messages or leave messages for other  
users.  Also, the Unity server is not very heavily used; it generally 

has no more than 9 active calls at the most.

I suspect the gateway might be the problem.  It's a 2851 running  
MGCP.  I guess I'd like to know if anyone agrees or disagrees with  
this assumption.  Also, I've had a lot of problems with MGCP code on  
the ISRs.  Can someone recommend an IOS version that is stable for  
and MGCP gateway and doesn't suffer from DSP firmware issues?

I should mention that Unity is version 4.04 sr1 and the gateway is  
running 12.4(5).

Thanks

-- 
kris seraphine
_______________________________________________
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cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 

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



From aman.chugh at gmail.com  Wed Sep 13 11:17:49 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Wed, 13 Sep 2006 20:47:49 +0530
Subject: [cisco-voip] ISP and VPN Failover for Call Manager based
	VOIPnetwork
In-Reply-To: <AB0A35F2486F81468971AA68558BE71904CEF16A@smieln01winm01.micoanet.com>
References: <bbe09c1d0609130219k29481ba1p427a9d81bb5b9ffe@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE71904CEF16A@smieln01winm01.micoanet.com>
Message-ID: <d4c697ad0609130817s2f5b8ee5h8d97dbcd4c35bc11@mail.gmail.com>

I kinda agree with Matthew,I have been having mix results for Voice Quality
with a Site to Site VPN  with a 2801 and PIX on the other end,I have
configured QOS and now when ever I use g729 codec ,I get terrible voice
quality and if i change it back to g711 it's is very good,but cant use g711
as it consumes quite a lot of bandwitdh On 512 kbps link with the VPN
header.I have a TAC case openned and Tac has recommened to try somethings
out.I will keep you guys posted.

Aman


On 9/13/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>   Manoj,
>
>
>
> Do you currently have private lines or some other circuits interconnecting
> your offices or are you planning to use VPN exclusively for voice and data?
> My major concern when using a Cisco PIX for voice would be Quality of
> Service.  While the PIX can preserve DSCP values as they are passed across
> the tunnels, unless anything has changed in 7.x, it doesn't have the
> ability to perform marking, LLQ prioritization, and traffic shaping.  This
> means that before any traffic is passed to the PIX, the device behind it (a
> switch or router) will have to perform some of these functions (say marking
> or traffic shaping).  In regards to LLQ you are out of luck.
>
>
>
> For our Teleworker VPN network we utilize a 2851 at the head-end and
> failover site and 871/877 routers at our remotes.  This gives us the
> capability to mark, LLQ, and shape traffic at the edge, before it is passed
> on to the ISP.  Additionally we utilize DMVPN and GRE to maintain routing
> information (EIGRP) and to dynamically handle routing changes when we loose
> a VPN link (say to our head-end).  I think you can do some least cost
> routing type things on the PIX to achieve the same effect, but it's much
> easier in IOS.
>
>
>
> Your ideas are sound in my opinion.  I'm sure that there are some people
> that are handling voice fine using Cisco PIX's however we had mixed results
> when we were using them.  Once we moved to the IOS VPN several of our QoS
> issues were resolved.  Regardless, you always have to remember that it still
> is the Internet and not a private network connection, so you get what you
> get.
>
>
>
> Hope this helps,
>
>
>
> -Matt
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Manoj Kalpage
> *Sent:* Wednesday, September 13, 2006 5:20 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] ISP and VPN Failover for Call Manager based
> VOIPnetwork
>
>
>
> Dear All,
>
> I am looking for ISP fail over for VoIP network. We have small
> enterprise VoIP network. If I explain our network bit, Basically we
> have call manager and unity server in main office with PIX515. All the
> branch offices has PIX 501. With attached fail over solution I am going to
> create two tunnels from each branch office and have them connected to each
> firewall in main office. I think this way if one PIX515 fail at main office,
> still branch office can be connected through second PIX515. Bellow is the
> router configuration for routing between two PIX 515. This configuration
> itself doesn't mean anything without looking at a diagram.I need to test
> this but I don't have enough gears with me right now and also I don't have
> 100% confidence on this. So, I would like to share with you folks. Any
> comments and ideas would be greatly appreciated.
>
>
>
> Please find the diagram bellow link (Sorry it's han written one )
>
>
> http://proxy.f2.ymdb.yahoofs.jp/bc/857e55a/bc/bd7f/failover.jpg?bcQM9BFBNirrJIWq
>
>
>
> best regards,
>
> Manoj
>
>
>
>
> ip cef
>
> !####Establish sla monitors for use in tracking objects####!
>
> ip sla monitor 1
> type echo protocol ipIcmpEcho 174.16.0.1
> threshold 3
> frequency 5
> ip sla monitor schedule 1 life forever start-time now
>
> ip sla monitor 2
> type echo protocol ipIcmpEcho 173.16.0.1
> threshold 3
> frequency 5
> ip sla monitor schedule 2 life forever start-time now
> !
>
> !####Configure Tracking objects (referencing IP SLA monitor's above)####!
>
> track 101 rtr 1 reachability
> !
> track 102 rtr 2 reachability
> !
> !
> !
> !
> !####Configure Interfaces with NAT####!
>
> interface FastEthernet 0/1
> ip address 172.16.0.1 255.255.0.0
> ip nat inside
>
> !
> interface Fastethernet 0/0
> ip address 173.16.0.2 255.255.255.0
> ip nat outside
>
> !
> interface Fastethernet 0/2
> ip address 174.16.0.2 255.255.255.0
> ip nat outside
>
> !
> ip classless
> !####Configure gateway of last resort with tracking objects####!
> ip route 0.0.0.0 0.0.0.0 173.16.0.1 track 101
> ip route 0.0.0.0 0.0.0.0 174.16.0.1 track 102
>
> !####Configure NAT statements for most outbound traffic####!
> ip nat inside source route-map ISP1 interface FastEthernet 0/0 overload
> ip nat inside source route-map ISP2 interface FastEthernet 0/2 overload
>
> !
> access-list 10 permit 172.16.0.0 0.0.0.255
> access-list 101 permit icmp any host 173.16.0.1 echo
> access-list 102 permit icmp any host 174.16.0.1 echo
>
> !
> !####Configure route maps for reference in NAT statements####!
> route-map ISP2 permit 10
> match ip address 10
> match interface Fastethernet 0/1
> !
> route-map ISP1 permit 10
> match ip address 10
> match interface Fastethernet 0/0
> !
>
>
>
>
>  ------------------------------
>
> *CONFIDENTIALITY STATEMENT*
>
> This communication and any attachments are *CONFIDENTIAL* and may be
> protected by one or more legal privileges. It is intended solely for the use
> of the addressee identified above. If you are not the intended recipient,
> any use, disclosure, copying or distribution of this communication is *
> UNAUTHORIZED*. Neither this information block, the typed name of the
> sender, nor anything else in this message is intended to constitute an
> electronic signature unless a specific statement to the contrary is included
> in this message. If you have received this communication in error, please
> immediately contact me and delete this communication from your computer.
> Thank you.
>
> ------------------------------
>
>
> _______________________________________________
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From MLinsemier at apcapital.com  Wed Sep 13 11:27:06 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Wed, 13 Sep 2006 11:27:06 -0400
Subject: [cisco-voip] call quality issues between gateway and Unity
In-Reply-To: <s507e61a.028@comrcm01intgwia.mandtbank.com>
Message-ID: <AB0A35F2486F81468971AA68558BE71904CEF24B@smieln01winm01.micoanet.com>

At one point in time Playout-Delay was supported on MGCP because we used
to do this back in the day on our VG200 gateways to resolve some of the
jitter we were getting from a tie-line we were using to connect Nortel
users to CallManager and Unity Voicemail.  This was three years ago so
maybe things have changed since then.

Jitter and Unity seem to be a weird thing.  While we don't experience
day to day issues with unity, occasionally we will still get some jitter
when you first hear the employees greeting or "enter your password".  

-Matt

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of STEVEN CASPER
Sent: Wednesday, September 13, 2006 11:06 AM
To: wsisk at cisco.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] call quality issues between gateway and Unity

Wes,
I see that Playout-Delay is not supported on MGCP platforms so how is
jitter handled in a MGCP/Call Manager enviroment?

Thanks!
Steve

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
Kris,

This sounds very similar to something we fixed long ago.  Take a look 

at 'sh voice call' to see if it's jitter:
http://www.cisco.com/en/US/tech/tk652/tk698/ 
technologies_tech_note09186a00800945df.shtml

CSCea35850 - Unity RTP stream has inherent jitter


If you are using non-default packetization periods watch out for
CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and  
G711/30ms

/Wes

On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:

Hi

I have a customer with a multi site centralized CCM 4.13 cluster.   
The remote sites are getting complaints of poor voice quality from  
customers when being transferred to Unity (which is located in the  
central site) .  The callers cannot understand the greetings and  
describe them as choppy.  There are no problems with external calls  
transferred from the remote sites to IP phones at the central site or 

for intersite calls.  It seems to be limited to calls where the RTP  
stream is between a remote voice gateway and Unity.  These calls are  
G.729 so Unity has to transcode the greetings on the fly but if that  
were the problem I'd expect remote users to have the same problem  
when calling into check their messages or leave messages for other  
users.  Also, the Unity server is not very heavily used; it generally 

has no more than 9 active calls at the most.

I suspect the gateway might be the problem.  It's a 2851 running  
MGCP.  I guess I'd like to know if anyone agrees or disagrees with  
this assumption.  Also, I've had a lot of problems with MGCP code on  
the ISRs.  Can someone recommend an IOS version that is stable for  
and MGCP gateway and doesn't suffer from DSP firmware issues?

I should mention that Unity is version 4.04 sr1 and the gateway is  
running 12.4(5).

Thanks

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

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

CONFIDENTIALITY STATEMENT

This communication and any attachments are CONFIDENTIAL and may be 
protected by one or more legal privileges. It is intended solely
for the use of the addressee identified above. If you are not 
the intended recipient, any use, disclosure, copying or 
distribution of this communication is UNAUTHORIZED. Neither this 
information block, the typed name of the sender, nor anything 
else in this message is intended to constitute an electronic 
signature unless a specific statement to the contrary is 
included in this message. If you have received this communication in 
error, please immediately contact me and delete this communication from
your computer. Thank you.

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



From garethsc at gmail.com  Wed Sep 13 12:31:03 2006
From: garethsc at gmail.com (Gareth Cafirma)
Date: Wed, 13 Sep 2006 12:31:03 -0400
Subject: [cisco-voip] Phone Ring Setting not working
Message-ID: <8dc3ea820609130931s3924d84ay1953f970b334288f@mail.gmail.com>

Hi guys,

What could be the problem when a DN's Phone ring setting (Idle), (active) is
set to disable but it still rings when you call it? This DN is shared among
couple of phones with different settings, main phone is set to ring for
both, secondary phone set to ring when idle and ring once when active.
Appeciate any help.

On 9/13/06, 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. Re: Mysterious Inbound calls (Wes Sisk)
>   2. Re: ip service... (Johan Bloemhard)
>
>
>
> ---------- Forwarded message ----------
> From: Wes Sisk <wsisk at cisco.com>
> To: Zevier Thoran <zthoran at yahoo.com>
> Date: Wed, 13 Sep 2006 09:47:26 -0400
> Subject: Re: [cisco-voip] Mysterious Inbound calls
> Zevier,
>
> There is an issue where:
> user calls unity to check voicemail or calls into unity auto attendant
> user then receives a inbound call. user places unity on hold to answer
> incoming call.  The 'after greeting' action on the all subscribers
> defaults to send call to opening greeting.  If the opening greeting call
> handler is configured to transfer to operator then the operator will
> receive and no one will be on the other end.
>
> However, this does not explain 20+calls all happening at 5PM.  Please
> open a TAC case and provide traces, details of directory numbers, and
> times.
>
> /WEs
>
> Zevier Thoran wrote:
> > Hi Wes,
> > I am just taking over this system and not fully familiar...but was told
> > that all these calls do come in via a press press '0' for Operator Unity
> > Auto Attendant Call Handler. But why should these calls only come in at
> > 5pm? Is their a fix?
> > Many thanks in advance......./Zevier
> >
> > */Wes Sisk <wsisk at cisco.com>/* wrote:
> >
> >     Do you use Unity voicemail?  Is your the operator's specific
> >     extension (any hunt pilot that routes to that phone) configured
> >     under the operator CH in Unity?
> >
> >     /Wes
> >
> >     On Sep 12, 2006, at 10:00 PM, Zevier Thoran wrote:
> >
> >     Hi,
> >     Every evening at 5pm, My Callmanager system gets hit with about 20
> >     anonymous calls on the operator number. On answering a call the
> >     operator just gets dead air.
> >     Any suggestions how to solve this problem?
> >     Regards - Zevier.
> >
> >
> ------------------------------------------------------------------------
> >     Yahoo! Messenger with Voice. Make PC-to-Phone Calls
> >     <
> http://us.rd.yahoo.com/mail_us/taglines/postman1/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com
> >
> >     to the US (and 30+ countries) for 2?/min or less.
> >     _______________________________________________
> >     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
> >
> >
> > ------------------------------------------------------------------------
> > Stay in the know. Pulse on the new Yahoo.com <http://yahoo.com/>. Check
> it out.
> > <http://us.rd.yahoo.com/evt=42974/*http://www.yahoo.com/preview>
>
>
>
>
> ---------- Forwarded message ----------
> From: "Johan Bloemhard" <bloemhard_johan at silverfalls.k12.or.us>
> To: "Manoj Kalpage" <manoj.kalpage at gmail.com>
> Date: Wed, 13 Sep 2006 07:56:16 -0700
> Subject: Re: [cisco-voip] ip service...
>
> If you look in the QRTReportResult.asp file, which should be located in
> "C:\CiscoWebs\Service", what does it say?  By default, it should say "folder
> = fso.GetFolder("c:\\Program Files\\Cisco\\QRT");"
>
>
>
> Verify that "c:\Program Files\Cisco\QRT" exists.  If it doesn't, then QRT
> may not be setup correctly.  Verify your log file settings under Service
> Parameters->Cisco Extended Functions, Log File settings.
>
>
>
> If your log file settings are different, you may have to change either the
> log file settings to match the line in the ASP file or vice versa.
>
>
>
> Johan Bloemhard
> Senior Information Technology Specialist
> ___________________________________
> *Silver** Falls School District** 4J*
> 802 Schlador Street
> Silverton, OR 97381
> w: 503.873.6331
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Manoj Kalpage
> *Sent:* Wednesday, September 13, 2006 2:28 AM
> *To:* Lelio Fulgenzi
> *Cc:* Cisco Voip
> *Subject:* Re: [cisco-voip] ip service...
>
>
>
> Sorry to interrupt the post but I am having a problem accessing QRT report
> but I got QRT soft Key configured on phone and I can leave the feed back.
> Bellow is the error
>
>
>
> Microsoft JScript runtime error '800a004c'
>
> Path not found
>
> /CCMService/QRTReportResult.asp, line 109
>
> Best regards,
>
> Manoj
>
>
>
>
>
> On 9/13/06, *Lelio Fulgenzi* <lelio at uoguelph.ca> wrote:
>
> It's a softkey that you have to enable via the softkey template
> configuration screen.
>
>
>
> device > device settings > softkey template
>
>
>
> select your template, then:
>
>
>
> highlite callmanager > configure softkey layout
>
>
>
> quality report tool (QRT) appears as valid options for the following
> states:
>
>
>
> Connected
>
> Connected Conference
>
> Connected Transfer
>
> On Hook
>
>
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."
>
> ----- Original Message -----
>
> *From:* Tim Reimers <tim.reimers at asheville.k12.nc.us>
>
> *To:* Lelio Fulgenzi <lelio at uoguelph.ca>
>
> *Cc:* Cisco Voip <cisco-voip at puck.nether.net>
>
> *Sent:* Tuesday, September 12, 2006 4:02 PM
>
> *Subject:* RE: [cisco-voip] ip service...
>
>
>
>
>
>
>
>
> I get the message "Document empty or invalid" when I try to run a report..
>
>
>
> I see that it's part of Extended Services, and puts a QRT button on the
> phone.
>
>
>
> I cannot find any directions for 4.01 on how to configure the CCM so that
> the button becomes available to users...
>
>
>
> Anyone have a link on how to enable the QRT function on the phones?
>
>
>
> Tim
>
>
>  ------------------------------
>
> *From:* Lelio Fulgenzi [mailto:lelio at uoguelph.ca ]
> *Sent:* Tuesday, September 12, 2006 12:19 PM
> *To:* Tim Reimers; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] ip service...
>
>
>
> QRT
>
>
>
> it's a softkey that you can show.
>
>
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803f52dd.html
>
>
>
>
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."
>
>  ----- Original Message -----
>
> *From:* Tim Reimers <tim.reimers at asheville.k12.nc.us>
>
> *To:* cisco-voip at puck.nether.net
>
> *Sent:* Tuesday, September 12, 2006 12:17 PM
>
> *Subject:* [cisco-voip] ip service...
>
>
>
>
>
>
> Hi folks---
>
>
>
> Isn't there a service that can be configured on an IP phone that allows a
> user to immediately report a call in which they had trouble with echo or
> jitter?
>
>
>
> I know the 'info' button shows the codec, etc?but it doesn't log that to a
> technician for later analysis
>
> (and have you ever tried to have a user read the codec info to you? That's
> ?really- fun to listen to?not! )
>
>
>
> Thanks Tim
>  ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> ------------------------------
> *Disclaimer*: This message may be legally privileged and/or is intended
> only for the use of the addressee(s). The content and views expressed in
> this email may represent the views of the sender and not those of Silver
> Falls School District.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From SPackett at fenwick.com  Wed Sep 13 13:45:37 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Wed, 13 Sep 2006 10:45:37 -0700
Subject: [cisco-voip] 7970s not registering after CCM upgrade to 4.1(2)
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD760C@SVCEXCH01.firm.fenwick.llp>

I had the same experience when I upgrade from 4.0(2a) sr? to 4.1(3) and
my 7971s did not get the latest firmware.  I had to manually go to each
phone and unplug them and plug them back for them to download the latest
firmware.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Chris Baird
Sent: Wednesday, September 13, 2006 8:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7970s not registering after CCM upgrade to 4.1(2)



We upgraded our CCM last night from 4.0(2)asr2c to 4.1(2).  Now only 4
of 20 7970s will register, the other 16 show unregistered but they are
getting a DCHP address from the CCM.  The 7970s were on 8.02 and we
upgraded that to 8.04sr1 but are seeing the same results.  Our 7960s are
not seeing any issues.

 

Any ideas?

 

Thanks,

Chris

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From baryonyx5 at gmail.com  Wed Sep 13 14:31:43 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Wed, 13 Sep 2006 13:31:43 -0500
Subject: [cisco-voip] IP phone users hearing other users' calls
Message-ID: <a749d0620609131131l790c7aect64b2bc3e526dff8c@mail.gmail.com>

I was contacted by a business that has an ICS7750 running CCM 3.3.2.  They
are having a strange problem.  A user will be on a external call and begin
hearing audio from another separate call between a different IP phone and
external caller.  According to the customer, only the IP phone users can
hear each others conversations, the external parties do not know its
happening.  The problem is random but has been happening for quite a while.
I'm told that when it happens, the other call is not muffled or softer but
sounds loud and clear.

My first thought is that they have a wiring issue with their analog lines
but I would think the PSTN end of the call would notice the problem as well
if this were the problem.

I'm not sure how this could happen on the IP side of the call but since its
a 7750 running old code I thought I'd ask.

If anyone has any thoughts or has experienced a similar problem, please let
me know.

The MRP is running 12.2(4r)XM, I haven't been able to get into the SSP yet.

thanks

-- 
kris seraphine
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From tim.reimers at asheville.k12.nc.us  Wed Sep 13 15:27:55 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 13 Sep 2006 15:27:55 -0400
Subject: [cisco-voip] BAT question--  CCM 4.01
Message-ID: <050944EE48FF384F9C663F7048C47DA704B41B8F@webmail.asheville.k12.nc.us>

Is it possible to update the softkey template globally using BAT?

I've designed a new softkey template that applies QRT and MCIS
but I cannot figure out how to get that softkey template to be in use on
the phones globally..
(or even a test subset of phones)

Softkey template doesn't appear in the BAT menus for phone or for line.

I could change it on every phone, but that would NOT be workable with
500 phones...

Does this mean that softkey settings are globally configured somewhere
else?

I see device profiles as a possibility, but my vendor didn't configre
any device profile
apparently. When I do a search for "device profile" and leave the box
empty, 
there's nothing reported in the search

ideas??

thanks, Tim
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From gpate at kenttech.com  Wed Sep 13 15:42:19 2006
From: gpate at kenttech.com (Gary L. Pate)
Date: Wed, 13 Sep 2006 15:42:19 -0400
Subject: [cisco-voip] FXS - Fax Issue
In-Reply-To: <a749d0620609131131l790c7aect64b2bc3e526dff8c@mail.gmail.com>
Message-ID: <001401c6d76c$b9a2eca0$7000a8c0@gpatelt>

I'm having an issue with faxes that I'm hoping someone could help me with.
Here is the scenario:

 

The fax numbers comes in via a PRI at a location where the callmanager is
located. Then there is an ATM circuit (1.5) to the customer location where a
28xx series router is located. This router has FXS ports for the fax
machines. The router is running 12.4 (7b) and is using MGCP. (Config
attached). I keep having issues with sending/receiving faxes, especially
faxes with multiple pages. For example, a 7 page fax may only come across
with 3 pages, and then I lose it.

Any ideas?

 

 

 

 

Current configuration : 7359 bytes

!

! Last configuration change at 14:29:05 EDT Tue Aug 29 2006

! NVRAM config last updated at 14:19:21 EDT Tue Aug 29 2006

!

version 12.4

service timestamps debug datetime localtime show-timezone

service timestamps log datetime localtime show-timezone

service password-encryption

!

hostname FLT3FBCRT01

!

boot-start-marker

boot-end-marker

!

logging buffered 8192 informational

enable secret 5 $1$s39s$1p25IOzozk5eoZofbQdvy0

enable password 7 075E754D08084D54

!

no aaa new-model

!

resource policy

!

clock timezone EST -5

clock summer-time EDT recurring

ip cef

!

!

no ip dhcp use vrf connected

no ip dhcp conflict logging

!

ip dhcp pool CBS

   network 10.254.252.0 255.255.255.128

   default-router 10.254.252.1

   dns-server 10.255.255.200 65.115.171.194

   option 150 ip 10.255.255.210

   lease 2

!

!

no ip domain lookup

!

voice-card 0

!

!

!

!

!

!

!

!

!

!

!

!

!

crypto pki trustpoint TP-self-signed-4206964125

 enrollment selfsigned

 subject-name cn=IOS-Self-Signed-Certificate-4206964125

 revocation-check none

 rsakeypair TP-self-signed-4206964125

!

!

crypto pki certificate chain TP-self-signed-4206964125

 certificate self-signed 01

  30820244 308201AD A0030201 02020101 300D0609 2A864886 F70D0101 04050030

  31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274

  69666963 6174652D 34323036 39363431 3235301E 170D3036 30373136 30343033

  32355A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649

  4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D34 32303639

  36343132 3530819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281

  8100AFAC DD4F1533 DEC505AD 48E8BF81 52758C16 3DA39E8A 98731CEB 8F9B1827

  49C72EE6 158CCD93 37B63BA9 1237325C 72080995 2068D6F8 DB296B45 643F5778

  5DD67CFA 97F0A7E2 77F6BBF6 E43658B2 C85B95D4 7EB0502C A4BDD590 963B31C8

  A877004F 13DD3FF5 F9DCD3D2 1B1341F1 77EBBFAD FE86DF65 81CD232D 299C1571

  56E30203 010001A3 6C306A30 0F060355 1D130101 FF040530 030101FF 30170603

  551D1104 10300E82 0C464C54 33464243 52543031 2E301F06 03551D23 04183016

  80141EDE C76712C5 EFA40117 6612C951 E7080EDA 1F41301D 0603551D 0E041604

  141EDEC7 6712C5EF A4011766 12C951E7 080EDA1F 41300D06 092A8648 86F70D01

  01040500 03818100 65A6D08B E26C80B3 6E4B3725 5A398282 D28048D1 D0FB2281

  FD8719A4 9632EB07 C898B58A 044D99F4 D7ECF2B0 FB30BA75 46FB964A 88913FA5

  DF79D189 932C2648 B781955E 7A909D72 EAE5E18D E67FFCBA 828F4A8F F99511DF

  58CD947E 340C14EE 0D3013FB A41AC1B5 92C86A52 A4181945 C8CE8AAB D556FF98

  D436AD08 CD3F7CF6

  quit

!

class-map match-any call-sig

 match ip dscp cs3

 match ip dscp af31

class-map match-any protocols

 match ip dscp cs6

 match ip dscp cs7

class-map match-all voice

 match ip dscp ef

!

!

policy-map Voice_QoS

 class voice

  priority percent 50

 class call-sig

  bandwidth percent 10

 class protocols

  bandwidth percent 5

 class class-default

   police 1000000 conform-action transmit  exceed-action drop

!

!

!

!

interface FastEthernet0/0

 bandwidth 1500

 ip address 10.254.252.129 255.255.255.128

 load-interval 30

 speed 100

 full-duplex

 no mop enabled

 service-policy output Voice_QoS

!

interface FastEthernet0/1

 no ip address

 shutdown

 duplex auto

 speed auto

!

interface FastEthernet0/1/0

!

interface FastEthernet0/1/1

!

interface FastEthernet0/1/2

!

interface FastEthernet0/1/3

!

interface FastEthernet0/1/4

!

interface FastEthernet0/1/5

!

interface FastEthernet0/1/6

!

interface FastEthernet0/1/7

!

interface FastEthernet0/1/8

!

interface Vlan1

 ip address 10.254.252.1 255.255.255.128

!

ip route 0.0.0.0 0.0.0.0 10.254.252.254

ip route 10.255.255.0 255.255.255.0 10.254.252.254

!

!

ip http server

ip http authentication local

ip http secure-server

!

logging 10.255.255.215

!

!

control-plane

!

!

!

voice-port 0/2/0

 cptone C1

 description Fax Machine

!

voice-port 0/2/1

 cptone C1

 description Petra Fax Machine

!

voice-port 0/2/2

 cptone C1

 description Credit card machine

!

voice-port 0/2/3

 cptone C1

!

voice-port 0/3/0

!

voice-port 0/3/1

!

ccm-manager fallback-mgcp

ccm-manager redundant-host 10.255.255.211

ccm-manager mgcp

ccm-manager music-on-hold

ccm-manager config server 10.255.255.210

ccm-manager config

ccm-manager download-tones

!

mgcp

mgcp call-agent 10.255.255.210 2427 service-type mgcp version 0.1

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

mgcp package-capability pre-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface FastEthernet0/0

mgcp bind media source-interface FastEthernet0/0

!

mgcp profile default

!

!

!

dial-peer voice 999020 pots

 service mgcpapp

 port 0/2/0

!

dial-peer voice 999021 pots

 service mgcpapp

 port 0/2/1

!

dial-peer voice 999022 pots

 service mgcpapp

 port 0/2/2

!

dial-peer voice 999023 pots

 service mgcpapp

 port 0/2/3

!

!

line con 0

 login local

line aux 0

line vty 0 4

 password 7 03550F0A400E751D

 login

 transport input telnet ssh

line vty 5 15

 password 7 03550F0A400E751D

 login

 transport input telnet ssh

!

scheduler allocate 20000 1000

ntp clock-period 17179981

ntp update-calendar

ntp server 128.118.25.3 prefer

end

 

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From Scott.Voll at wesd.org  Wed Sep 13 16:09:25 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 13 Sep 2006 13:09:25 -0700
Subject: [cisco-voip] FXS - Fax Issue
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C8570@Apollo.wesd.org>

Fax related things to check:

 

Fax can NOT be running super G3 33.4k and can NOT be running ECM (error
correction mode)

 

Call needs to be G711 all the way through.

 

No Fax relay.  It needs to be Fax passthrough.  Check on MGCP gateway as
my CMM PRI's defaulted to Fax relay.

 

Hope that gives you a starting point.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gary L. Pate
Sent: Wednesday, September 13, 2006 12:42 PM
To: 'cisco-voip'
Subject: [cisco-voip] FXS - Fax Issue

 

I'm having an issue with faxes that I'm hoping someone could help me
with. Here is the scenario:

 

The fax numbers comes in via a PRI at a location where the callmanager
is located. Then there is an ATM circuit (1.5) to the customer location
where a 28xx series router is located. This router has FXS ports for the
fax machines. The router is running 12.4 (7b) and is using MGCP. (Config
attached). I keep having issues with sending/receiving faxes, especially
faxes with multiple pages. For example, a 7 page fax may only come
across with 3 pages, and then I lose it.

Any ideas?

 

 

 

 

Current configuration : 7359 bytes

!

! Last configuration change at 14:29:05 EDT Tue Aug 29 2006

! NVRAM config last updated at 14:19:21 EDT Tue Aug 29 2006

!

version 12.4

service timestamps debug datetime localtime show-timezone

service timestamps log datetime localtime show-timezone

service password-encryption

!

hostname FLT3FBCRT01

!

boot-start-marker

boot-end-marker

!

logging buffered 8192 informational

enable secret 5 $1$s39s$1p25IOzozk5eoZofbQdvy0

enable password 7 075E754D08084D54

!

no aaa new-model

!

resource policy

!

clock timezone EST -5

clock summer-time EDT recurring

ip cef

!

!

no ip dhcp use vrf connected

no ip dhcp conflict logging

!

ip dhcp pool CBS

   network 10.254.252.0 255.255.255.128

   default-router 10.254.252.1

   dns-server 10.255.255.200 65.115.171.194

   option 150 ip 10.255.255.210

   lease 2

!

!

no ip domain lookup

!

voice-card 0

!

!

!

!

!

!

!

!

!

!

!

!

!

crypto pki trustpoint TP-self-signed-4206964125

 enrollment selfsigned

 subject-name cn=IOS-Self-Signed-Certificate-4206964125

 revocation-check none

 rsakeypair TP-self-signed-4206964125

!

!

crypto pki certificate chain TP-self-signed-4206964125

 certificate self-signed 01

  30820244 308201AD A0030201 02020101 300D0609 2A864886 F70D0101
04050030

  31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D
43657274

  69666963 6174652D 34323036 39363431 3235301E 170D3036 30373136
30343033

  32355A17 0D323030 31303130 30303030 305A3031 312F302D 06035504
03132649

  4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D34
32303639

  36343132 3530819F 300D0609 2A864886 F70D0101 01050003 818D0030
81890281

  8100AFAC DD4F1533 DEC505AD 48E8BF81 52758C16 3DA39E8A 98731CEB
8F9B1827

  49C72EE6 158CCD93 37B63BA9 1237325C 72080995 2068D6F8 DB296B45
643F5778

  5DD67CFA 97F0A7E2 77F6BBF6 E43658B2 C85B95D4 7EB0502C A4BDD590
963B31C8

  A877004F 13DD3FF5 F9DCD3D2 1B1341F1 77EBBFAD FE86DF65 81CD232D
299C1571

  56E30203 010001A3 6C306A30 0F060355 1D130101 FF040530 030101FF
30170603

  551D1104 10300E82 0C464C54 33464243 52543031 2E301F06 03551D23
04183016

  80141EDE C76712C5 EFA40117 6612C951 E7080EDA 1F41301D 0603551D
0E041604

  141EDEC7 6712C5EF A4011766 12C951E7 080EDA1F 41300D06 092A8648
86F70D01

  01040500 03818100 65A6D08B E26C80B3 6E4B3725 5A398282 D28048D1
D0FB2281

  FD8719A4 9632EB07 C898B58A 044D99F4 D7ECF2B0 FB30BA75 46FB964A
88913FA5

  DF79D189 932C2648 B781955E 7A909D72 EAE5E18D E67FFCBA 828F4A8F
F99511DF

  58CD947E 340C14EE 0D3013FB A41AC1B5 92C86A52 A4181945 C8CE8AAB
D556FF98

  D436AD08 CD3F7CF6

  quit

!

class-map match-any call-sig

 match ip dscp cs3

 match ip dscp af31

class-map match-any protocols

 match ip dscp cs6

 match ip dscp cs7

class-map match-all voice

 match ip dscp ef

!

!

policy-map Voice_QoS

 class voice

  priority percent 50

 class call-sig

  bandwidth percent 10

 class protocols

  bandwidth percent 5

 class class-default

   police 1000000 conform-action transmit  exceed-action drop

!

!

!

!

interface FastEthernet0/0

 bandwidth 1500

 ip address 10.254.252.129 255.255.255.128

 load-interval 30

 speed 100

 full-duplex

 no mop enabled

 service-policy output Voice_QoS

!

interface FastEthernet0/1

 no ip address

 shutdown

 duplex auto

 speed auto

!

interface FastEthernet0/1/0

!

interface FastEthernet0/1/1

!

interface FastEthernet0/1/2

!

interface FastEthernet0/1/3

!

interface FastEthernet0/1/4

!

interface FastEthernet0/1/5

!

interface FastEthernet0/1/6

!

interface FastEthernet0/1/7

!

interface FastEthernet0/1/8

!

interface Vlan1

 ip address 10.254.252.1 255.255.255.128

!

ip route 0.0.0.0 0.0.0.0 10.254.252.254

ip route 10.255.255.0 255.255.255.0 10.254.252.254

!

!

ip http server

ip http authentication local

ip http secure-server

!

logging 10.255.255.215

!

!

control-plane

!

!

!

voice-port 0/2/0

 cptone C1

 description Fax Machine

!

voice-port 0/2/1

 cptone C1

 description Petra Fax Machine

!

voice-port 0/2/2

 cptone C1

 description Credit card machine

!

voice-port 0/2/3

 cptone C1

!

voice-port 0/3/0

!

voice-port 0/3/1

!

ccm-manager fallback-mgcp

ccm-manager redundant-host 10.255.255.211

ccm-manager mgcp

ccm-manager music-on-hold

ccm-manager config server 10.255.255.210

ccm-manager config

ccm-manager download-tones

!

mgcp

mgcp call-agent 10.255.255.210 2427 service-type mgcp version 0.1

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

mgcp package-capability pre-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface FastEthernet0/0

mgcp bind media source-interface FastEthernet0/0

!

mgcp profile default

!

!

!

dial-peer voice 999020 pots

 service mgcpapp

 port 0/2/0

!

dial-peer voice 999021 pots

 service mgcpapp

 port 0/2/1

!

dial-peer voice 999022 pots

 service mgcpapp

 port 0/2/2

!

dial-peer voice 999023 pots

 service mgcpapp

 port 0/2/3

!

!

line con 0

 login local

line aux 0

line vty 0 4

 password 7 03550F0A400E751D

 login

 transport input telnet ssh

line vty 5 15

 password 7 03550F0A400E751D

 login

 transport input telnet ssh

!

scheduler allocate 20000 1000

ntp clock-period 17179981

ntp update-calendar

ntp server 128.118.25.3 prefer

end

 

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From fday at wilson-co.com  Wed Sep 13 16:16:08 2006
From: fday at wilson-co.com (Faye Day)
Date: Wed, 13 Sep 2006 16:16:08 -0400
Subject: [cisco-voip] Unity and BackupExec
Message-ID: <EE2B04B25891B149A42E963DA4F41437011F0806@WCOEXCHANGE.wilson-co.com>

Has anybody out there ever figured out a way to get BackupExec 9.0 to
run a Unity backup to a network drive folder instead of to an onboard
tape? 

 

Running Unity 4.0(4) sr1 and BU exec 9.0 Rev 4454 on a Windows 2000 sp4
server. I'm trying to backup to a folder on another Win2k sp4 server. 

 

Thanks,

 

Faye Day

 

Wilson County Technology Services

Assistant Director

(252) 206-1024

 

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From wcdixo at aurora.lib.il.us  Wed Sep 13 16:18:51 2006
From: wcdixo at aurora.lib.il.us (Dixon, Wayne)
Date: Wed, 13 Sep 2006 15:18:51 -0500
Subject: [cisco-voip] Unity and BackupExec
In-Reply-To: <EE2B04B25891B149A42E963DA4F41437011F0806@WCOEXCHANGE.wilson-co.com>
Message-ID: <6D8BEE91433474478D294A8867FFDC6EE3F8C3@WMAIN3.aurorapubliclibrary.org>

Faye,
 
Do you have the proper license key for that, it's not a standard option
available on the Backup Exec software.  That could be the issue.  You
can use the Backup-to-disk feature for a local drive (that comes with
Backup Exec, but not to a network drive.
 
Wayne Dixon 
Network Assistant
Aurora Public Library
1 E. Benton Street
Aurora, IL 60505
Phone: 630-264-4257
Fax: 630-896-3209
~~~~~~~~~~~~~~~~~~~~~~
 

	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Faye Day
	Sent: Wednesday, September 13, 2006 3:16 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Unity and BackupExec
	
	

	Has anybody out there ever figured out a way to get BackupExec
9.0 to run a Unity backup to a network drive folder instead of to an
onboard tape? 

	 

	Running Unity 4.0(4) sr1 and BU exec 9.0 Rev 4454 on a Windows
2000 sp4 server. I'm trying to backup to a folder on another Win2k sp4
server. 

	 

	Thanks,

	 

	Faye Day

	 

	Wilson County Technology Services

	Assistant Director

	(252) 206-1024

	 

-------------- next part --------------
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From bills at tns.its.psu.edu  Wed Sep 13 16:20:53 2006
From: bills at tns.its.psu.edu (Simon, Bill)
Date: Wed, 13 Sep 2006 16:20:53 -0400
Subject: [cisco-voip] Unity and BackupExec
In-Reply-To: <EE2B04B25891B149A42E963DA4F41437011F0806@WCOEXCHANGE.wilson-co.com>
References: <EE2B04B25891B149A42E963DA4F41437011F0806@WCOEXCHANGE.wilson-co.com>
Message-ID: <45086825.2030700@tns.its.psu.edu>

You need Backup Exec for Windows Servers - and you'll install the Backup 
Exec software on the server where you want to store the backups.  Then 
load the backup agent on the Unity server.  Essentially the Backup Exec 
server then performs the backup on a remote system and stores the files 
locally.

Faye Day wrote:
> Has anybody out there ever figured out a way to get BackupExec 9.0 to 
> run a Unity backup to a network drive folder instead of to an onboard tape?
> 
>  
> 
> Running Unity 4.0(4) sr1 and BU exec 9.0 Rev 4454 on a Windows 2000 sp4 
> server. I?m trying to backup to a folder on another Win2k sp4 server.

From gpate at kenttech.com  Wed Sep 13 16:49:53 2006
From: gpate at kenttech.com (Gary L. Pate)
Date: Wed, 13 Sep 2006 16:49:53 -0400
Subject: [cisco-voip] FXS - Fax Issue
In-Reply-To: <407055A92CECCB499C922A2D35FC19A6066C8570@Apollo.wesd.org>
Message-ID: <003e01c6d776$29df2340$7000a8c0@gpatelt>

This is my MGCP config on the router. I assume that is the gateway config
your referring to (on the IOS) versus the MGCP config on the Callmanager. (I
don' t see any options for pass through in the MGCP config on CCM ver. 4.2).

 

mgcp

mgcp call-agent 10.255.255.210 2427 service-type mgcp version 0.1

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

mgcp package-capability pre-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface FastEthernet0/0

mgcp bind media source-interface FastEthernet0/0

 

 

After reviewing a doc on pass through, this is what it looks like I need:


SUMMARY STEPS 


1. mgcp package-capability rtp-package 

2. mgcp modem passthrough voip mode nse 

3. mgcp modem passthrough voip codec {g711ulaw | g711alaw} 

4. mgcp modem passthrough voip redundancy [sample-duration [10 | 20]]
[maximum-sessions sessions] 

5. mgcp timer nse-response t38 time 

6. mgcp fax t.38 inhibit 

Here was an example though:

 

mgcp
mgcp call-agent 10.3.222.1 service-type mgcp version 0.1
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp fax t38 inhibit

 

The example is not using all the options from above, plus there is another
command in the example:

ccm-manager mgcp
no ccm-manager fax protocol cisco

 

Is this needed.?

 

 

  _____  

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Wednesday, September 13, 2006 4:09 PM
To: GPate at kenttech.com; cisco-voip
Subject: RE: [cisco-voip] FXS - Fax Issue

 

Fax related things to check:

 

Fax can NOT be running super G3 33.4k and can NOT be running ECM (error
correction mode)

 

Call needs to be G711 all the way through.

 

No Fax relay.  It needs to be Fax passthrough.  Check on MGCP gateway as my
CMM PRI's defaulted to Fax relay.

 

Hope that gives you a starting point.

 

Scott

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gary L. Pate
Sent: Wednesday, September 13, 2006 12:42 PM
To: 'cisco-voip'
Subject: [cisco-voip] FXS - Fax Issue

 

I'm having an issue with faxes that I'm hoping someone could help me with.
Here is the scenario:

 

The fax numbers comes in via a PRI at a location where the callmanager is
located. Then there is an ATM circuit (1.5) to the customer location where a
28xx series router is located. This router has FXS ports for the fax
machines. The router is running 12.4 (7b) and is using MGCP. (Config
attached). I keep having issues with sending/receiving faxes, especially
faxes with multiple pages. For example, a 7 page fax may only come across
with 3 pages, and then I lose it.

Any ideas?

 

 

 

 

Current configuration : 7359 bytes

!

! Last configuration change at 14:29:05 EDT Tue Aug 29 2006

! NVRAM config last updated at 14:19:21 EDT Tue Aug 29 2006

!

version 12.4

service timestamps debug datetime localtime show-timezone

service timestamps log datetime localtime show-timezone

service password-encryption

!

hostname FLT3FBCRT01

!

boot-start-marker

boot-end-marker

!

logging buffered 8192 informational

enable secret 5 $1$s39s$1p25IOzozk5eoZofbQdvy0

enable password 7 075E754D08084D54

!

no aaa new-model

!

resource policy

!

clock timezone EST -5

clock summer-time EDT recurring

ip cef

!

!

no ip dhcp use vrf connected

no ip dhcp conflict logging

!

ip dhcp pool CBS

   network 10.254.252.0 255.255.255.128

   default-router 10.254.252.1

   dns-server 10.255.255.200 65.115.171.194

   option 150 ip 10.255.255.210

   lease 2

!

!

no ip domain lookup

!

voice-card 0

!

!

!

!

!

!

!

!

!

!

!

!

!

crypto pki trustpoint TP-self-signed-4206964125

 enrollment selfsigned

 subject-name cn=IOS-Self-Signed-Certificate-4206964125

 revocation-check none

 rsakeypair TP-self-signed-4206964125

!

!

crypto pki certificate chain TP-self-signed-4206964125

 certificate self-signed 01

  30820244 308201AD A0030201 02020101 300D0609 2A864886 F70D0101 04050030

  31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274

  69666963 6174652D 34323036 39363431 3235301E 170D3036 30373136 30343033

  32355A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649

  4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D34 32303639

  36343132 3530819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281

  8100AFAC DD4F1533 DEC505AD 48E8BF81 52758C16 3DA39E8A 98731CEB 8F9B1827

  49C72EE6 158CCD93 37B63BA9 1237325C 72080995 2068D6F8 DB296B45 643F5778

  5DD67CFA 97F0A7E2 77F6BBF6 E43658B2 C85B95D4 7EB0502C A4BDD590 963B31C8

  A877004F 13DD3FF5 F9DCD3D2 1B1341F1 77EBBFAD FE86DF65 81CD232D 299C1571

  56E30203 010001A3 6C306A30 0F060355 1D130101 FF040530 030101FF 30170603

  551D1104 10300E82 0C464C54 33464243 52543031 2E301F06 03551D23 04183016

  80141EDE C76712C5 EFA40117 6612C951 E7080EDA 1F41301D 0603551D 0E041604

  141EDEC7 6712C5EF A4011766 12C951E7 080EDA1F 41300D06 092A8648 86F70D01

  01040500 03818100 65A6D08B E26C80B3 6E4B3725 5A398282 D28048D1 D0FB2281

  FD8719A4 9632EB07 C898B58A 044D99F4 D7ECF2B0 FB30BA75 46FB964A 88913FA5

  DF79D189 932C2648 B781955E 7A909D72 EAE5E18D E67FFCBA 828F4A8F F99511DF

  58CD947E 340C14EE 0D3013FB A41AC1B5 92C86A52 A4181945 C8CE8AAB D556FF98

  D436AD08 CD3F7CF6

  quit

!

class-map match-any call-sig

 match ip dscp cs3

 match ip dscp af31

class-map match-any protocols

 match ip dscp cs6

 match ip dscp cs7

class-map match-all voice

 match ip dscp ef

!

!

policy-map Voice_QoS

 class voice

  priority percent 50

 class call-sig

  bandwidth percent 10

 class protocols

  bandwidth percent 5

 class class-default

   police 1000000 conform-action transmit  exceed-action drop

!

!

!

!

interface FastEthernet0/0

 bandwidth 1500

 ip address 10.254.252.129 255.255.255.128

 load-interval 30

 speed 100

 full-duplex

 no mop enabled

 service-policy output Voice_QoS

!

interface FastEthernet0/1

 no ip address

 shutdown

 duplex auto

 speed auto

!

interface FastEthernet0/1/0

!

interface FastEthernet0/1/1

!

interface FastEthernet0/1/2

!

interface FastEthernet0/1/3

!

interface FastEthernet0/1/4

!

interface FastEthernet0/1/5

!

interface FastEthernet0/1/6

!

interface FastEthernet0/1/7

!

interface FastEthernet0/1/8

!

interface Vlan1

 ip address 10.254.252.1 255.255.255.128

!

ip route 0.0.0.0 0.0.0.0 10.254.252.254

ip route 10.255.255.0 255.255.255.0 10.254.252.254

!

!

ip http server

ip http authentication local

ip http secure-server

!

logging 10.255.255.215

!

!

control-plane

!

!

!

voice-port 0/2/0

 cptone C1

 description Fax Machine

!

voice-port 0/2/1

 cptone C1

 description Petra Fax Machine

!

voice-port 0/2/2

 cptone C1

 description Credit card machine

!

voice-port 0/2/3

 cptone C1

!

voice-port 0/3/0

!

voice-port 0/3/1

!

ccm-manager fallback-mgcp

ccm-manager redundant-host 10.255.255.211

ccm-manager mgcp

ccm-manager music-on-hold

ccm-manager config server 10.255.255.210

ccm-manager config

ccm-manager download-tones

!

mgcp

mgcp call-agent 10.255.255.210 2427 service-type mgcp version 0.1

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

mgcp package-capability pre-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface FastEthernet0/0

mgcp bind media source-interface FastEthernet0/0

!

mgcp profile default

!

!

!

dial-peer voice 999020 pots

 service mgcpapp

 port 0/2/0

!

dial-peer voice 999021 pots

 service mgcpapp

 port 0/2/1

!

dial-peer voice 999022 pots

 service mgcpapp

 port 0/2/2

!

dial-peer voice 999023 pots

 service mgcpapp

 port 0/2/3

!

!

line con 0

 login local

line aux 0

line vty 0 4

 password 7 03550F0A400E751D

 login

 transport input telnet ssh

line vty 5 15

 password 7 03550F0A400E751D

 login

 transport input telnet ssh

!

scheduler allocate 20000 1000

ntp clock-period 17179981

ntp update-calendar

ntp server 128.118.25.3 prefer

end

 

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From MLinsemier at apcapital.com  Wed Sep 13 17:03:18 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Wed, 13 Sep 2006 17:03:18 -0400
Subject: [cisco-voip] FXS - Fax Issue
In-Reply-To: <003e01c6d776$29df2340$7000a8c0@gpatelt>
Message-ID: <AB0A35F2486F81468971AA68558BE71904CEF439@smieln01winm01.micoanet.com>

This is what we have configured to support ATA186's (similar to FXS) on
our CMM MGCP Voice Gateways:

 

mgcp

mgcp call-agent x.x.x.x 2427 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

mgcp rtp unreachable timeout 1000 action notify

mgcp modem passthrough voip mode nse

mgcp package-capability rtp-package

mgcp package-capability sst-package

no mgcp timer receive-rtcp

mgcp fax t38 inhibit

mgcp bind control source-interface GigabitEthernet1/0

mgcp bind media source-interface GigabitEthernet1/0

 

The biggest issues we have ever had with Faxing and MGCP has been
SuperG3 fax modes and ECM.  Make sure you turn these off.

 

On a side note, there was some work being done on native T.38 faxing
(full speed).  Anyone know where this is at?

 

-Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gary L. Pate
Sent: Wednesday, September 13, 2006 4:50 PM
To: 'Voll, Scott'; 'cisco-voip'
Subject: Re: [cisco-voip] FXS - Fax Issue

 

This is my MGCP config on the router. I assume that is the gateway
config your referring to (on the IOS) versus the MGCP config on the
Callmanager. (I don' t see any options for pass through in the MGCP
config on CCM ver. 4.2).

 

mgcp

mgcp call-agent 10.255.255.210 2427 service-type mgcp version 0.1

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

mgcp package-capability pre-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface FastEthernet0/0

mgcp bind media source-interface FastEthernet0/0

 

 

After reviewing a doc on pass through, this is what it looks like I
need:


SUMMARY STEPS 


1.  mgcp package-capability rtp-package 

2. mgcp modem passthrough voip mode nse 

3. mgcp modem passthrough voip codec {g711ulaw | g711alaw} 

4. mgcp modem passthrough voip redundancy [sample-duration [10 | 20]]
[maximum-sessions sessions] 

5. mgcp timer nse-response t38 time 

6. mgcp fax t.38 inhibit 

Here was an example though:

 

mgcp
mgcp call-agent 10.3.222.1 service-type mgcp version 0.1
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp fax t38 inhibit

 

The example is not using all the options from above, plus there is
another command in the example:

ccm-manager mgcp
no ccm-manager fax protocol cisco

 

Is this needed.?

 

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Wednesday, September 13, 2006 4:09 PM
To: GPate at kenttech.com; cisco-voip
Subject: RE: [cisco-voip] FXS - Fax Issue

 

Fax related things to check:

 

Fax can NOT be running super G3 33.4k and can NOT be running ECM (error
correction mode)

 

Call needs to be G711 all the way through.

 

No Fax relay.  It needs to be Fax passthrough.  Check on MGCP gateway as
my CMM PRI's defaulted to Fax relay.

 

Hope that gives you a starting point.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gary L. Pate
Sent: Wednesday, September 13, 2006 12:42 PM
To: 'cisco-voip'
Subject: [cisco-voip] FXS - Fax Issue

 

I'm having an issue with faxes that I'm hoping someone could help me
with. Here is the scenario:

 

The fax numbers comes in via a PRI at a location where the callmanager
is located. Then there is an ATM circuit (1.5) to the customer location
where a 28xx series router is located. This router has FXS ports for the
fax machines. The router is running 12.4 (7b) and is using MGCP. (Config
attached). I keep having issues with sending/receiving faxes, especially
faxes with multiple pages. For example, a 7 page fax may only come
across with 3 pages, and then I lose it.

Any ideas?

 

 

 

 

Current configuration : 7359 bytes

!

! Last configuration change at 14:29:05 EDT Tue Aug 29 2006

! NVRAM config last updated at 14:19:21 EDT Tue Aug 29 2006

!

version 12.4

service timestamps debug datetime localtime show-timezone

service timestamps log datetime localtime show-timezone

service password-encryption

!

hostname FLT3FBCRT01

!

boot-start-marker

boot-end-marker

!

logging buffered 8192 informational

enable secret 5 $1$s39s$1p25IOzozk5eoZofbQdvy0

enable password 7 075E754D08084D54

!

no aaa new-model

!

resource policy

!

clock timezone EST -5

clock summer-time EDT recurring

ip cef

!

!

no ip dhcp use vrf connected

no ip dhcp conflict logging

!

ip dhcp pool CBS

   network 10.254.252.0 255.255.255.128

   default-router 10.254.252.1

   dns-server 10.255.255.200 65.115.171.194

   option 150 ip 10.255.255.210

   lease 2

!

!

no ip domain lookup

!

voice-card 0

!

!

!

!

!

!

!

!

!

!

!

!

!

crypto pki trustpoint TP-self-signed-4206964125

 enrollment selfsigned

 subject-name cn=IOS-Self-Signed-Certificate-4206964125

 revocation-check none

 rsakeypair TP-self-signed-4206964125

!

!

crypto pki certificate chain TP-self-signed-4206964125

 certificate self-signed 01

  30820244 308201AD A0030201 02020101 300D0609 2A864886 F70D0101
04050030

  31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D
43657274

  69666963 6174652D 34323036 39363431 3235301E 170D3036 30373136
30343033

  32355A17 0D323030 31303130 30303030 305A3031 312F302D 06035504
03132649

  4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D34
32303639

  36343132 3530819F 300D0609 2A864886 F70D0101 01050003 818D0030
81890281

  8100AFAC DD4F1533 DEC505AD 48E8BF81 52758C16 3DA39E8A 98731CEB
8F9B1827

  49C72EE6 158CCD93 37B63BA9 1237325C 72080995 2068D6F8 DB296B45
643F5778

  5DD67CFA 97F0A7E2 77F6BBF6 E43658B2 C85B95D4 7EB0502C A4BDD590
963B31C8

  A877004F 13DD3FF5 F9DCD3D2 1B1341F1 77EBBFAD FE86DF65 81CD232D
299C1571

  56E30203 010001A3 6C306A30 0F060355 1D130101 FF040530 030101FF
30170603

  551D1104 10300E82 0C464C54 33464243 52543031 2E301F06 03551D23
04183016

  80141EDE C76712C5 EFA40117 6612C951 E7080EDA 1F41301D 0603551D
0E041604

  141EDEC7 6712C5EF A4011766 12C951E7 080EDA1F 41300D06 092A8648
86F70D01

  01040500 03818100 65A6D08B E26C80B3 6E4B3725 5A398282 D28048D1
D0FB2281

  FD8719A4 9632EB07 C898B58A 044D99F4 D7ECF2B0 FB30BA75 46FB964A
88913FA5

  DF79D189 932C2648 B781955E 7A909D72 EAE5E18D E67FFCBA 828F4A8F
F99511DF

  58CD947E 340C14EE 0D3013FB A41AC1B5 92C86A52 A4181945 C8CE8AAB
D556FF98

  D436AD08 CD3F7CF6

  quit

!

class-map match-any call-sig

 match ip dscp cs3

 match ip dscp af31

class-map match-any protocols

 match ip dscp cs6

 match ip dscp cs7

class-map match-all voice

 match ip dscp ef

!

!

policy-map Voice_QoS

 class voice

  priority percent 50

 class call-sig

  bandwidth percent 10

 class protocols

  bandwidth percent 5

 class class-default

   police 1000000 conform-action transmit  exceed-action drop

!

!

!

!

interface FastEthernet0/0

 bandwidth 1500

 ip address 10.254.252.129 255.255.255.128

 load-interval 30

 speed 100

 full-duplex

 no mop enabled

 service-policy output Voice_QoS

!

interface FastEthernet0/1

 no ip address

 shutdown

 duplex auto

 speed auto

!

interface FastEthernet0/1/0

!

interface FastEthernet0/1/1

!

interface FastEthernet0/1/2

!

interface FastEthernet0/1/3

!

interface FastEthernet0/1/4

!

interface FastEthernet0/1/5

!

interface FastEthernet0/1/6

!

interface FastEthernet0/1/7

!

interface FastEthernet0/1/8

!

interface Vlan1

 ip address 10.254.252.1 255.255.255.128

!

ip route 0.0.0.0 0.0.0.0 10.254.252.254

ip route 10.255.255.0 255.255.255.0 10.254.252.254

!

!

ip http server

ip http authentication local

ip http secure-server

!

logging 10.255.255.215

!

!

control-plane

!

!

!

voice-port 0/2/0

 cptone C1

 description Fax Machine

!

voice-port 0/2/1

 cptone C1

 description Petra Fax Machine

!

voice-port 0/2/2

 cptone C1

 description Credit card machine

!

voice-port 0/2/3

 cptone C1

!

voice-port 0/3/0

!

voice-port 0/3/1

!

ccm-manager fallback-mgcp

ccm-manager redundant-host 10.255.255.211

ccm-manager mgcp

ccm-manager music-on-hold

ccm-manager config server 10.255.255.210

ccm-manager config

ccm-manager download-tones

!

mgcp

mgcp call-agent 10.255.255.210 2427 service-type mgcp version 0.1

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

mgcp package-capability pre-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

mgcp bind control source-interface FastEthernet0/0

mgcp bind media source-interface FastEthernet0/0

!

mgcp profile default

!

!

!

dial-peer voice 999020 pots

 service mgcpapp

 port 0/2/0

!

dial-peer voice 999021 pots

 service mgcpapp

 port 0/2/1

!

dial-peer voice 999022 pots

 service mgcpapp

 port 0/2/2

!

dial-peer voice 999023 pots

 service mgcpapp

 port 0/2/3

!

!

line con 0

 login local

line aux 0

line vty 0 4

 password 7 03550F0A400E751D

 login

 transport input telnet ssh

line vty 5 15

 password 7 03550F0A400E751D

 login

 transport input telnet ssh

!

scheduler allocate 20000 1000

ntp clock-period 17179981

ntp update-calendar

ntp server 128.118.25.3 prefer

end

 


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From wsisk at cisco.com  Wed Sep 13 18:30:38 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 13 Sep 2006 18:30:38 -0400
Subject: [cisco-voip] call quality issues between gateway and Unity
In-Reply-To: <s507e61a.027@comrcm01intgwia.mandtbank.com>
References: <s507e61a.027@comrcm01intgwia.mandtbank.com>
Message-ID: <4508868E.1070502@cisco.com>

Steven,

Where do you get that playout-delay is not supported?

/Wes

STEVEN CASPER wrote:
> Wes,
> I see that Playout-Delay is not supported on MGCP platforms so how is
> jitter handled in a MGCP/Call Manager enviroment?
> 
> Thanks!
> Steve
> 
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026
> 
>>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
> Kris,
> 
> This sounds very similar to something we fixed long ago.  Take a look 
> 
> at 'sh voice call' to see if it's jitter:
> http://www.cisco.com/en/US/tech/tk652/tk698/ 
> technologies_tech_note09186a00800945df.shtml
> 
> CSCea35850 - Unity RTP stream has inherent jitter
> 
> 
> If you are using non-default packetization periods watch out for
> CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and  
> G711/30ms
> 
> /Wes
> 
> On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:
> 
> Hi
> 
> I have a customer with a multi site centralized CCM 4.13 cluster.   
> The remote sites are getting complaints of poor voice quality from  
> customers when being transferred to Unity (which is located in the  
> central site) .  The callers cannot understand the greetings and  
> describe them as choppy.  There are no problems with external calls  
> transferred from the remote sites to IP phones at the central site or 
> 
> for intersite calls.  It seems to be limited to calls where the RTP  
> stream is between a remote voice gateway and Unity.  These calls are  
> G.729 so Unity has to transcode the greetings on the fly but if that  
> were the problem I'd expect remote users to have the same problem  
> when calling into check their messages or leave messages for other  
> users.  Also, the Unity server is not very heavily used; it generally 
> 
> has no more than 9 active calls at the most.
> 
> I suspect the gateway might be the problem.  It's a 2851 running  
> MGCP.  I guess I'd like to know if anyone agrees or disagrees with  
> this assumption.  Also, I've had a lot of problems with MGCP code on  
> the ISRs.  Can someone recommend an IOS version that is stable for  
> and MGCP gateway and doesn't suffer from DSP firmware issues?
> 
> I should mention that Unity is version 4.04 sr1 and the gateway is  
> running 12.4(5).
> 
> Thanks
> 

From yezhi98 at 126.com  Wed Sep 13 20:28:45 2006
From: yezhi98 at 126.com (yezhi98)
Date: Thu, 14 Sep 2006 08:28:45 +0800 (CST)
Subject: [cisco-voip] Why some IOS in cisco router can use OID get E1/T1
 usage status and others got fai l
Message-ID: <4508A23D.00010B.11146@bj126app7.126.com>



I find some IOS version of router can use  OID 1.3.6.1.4.1.9.10.19.1.1.4.0  by mrtg to get E1/T1 channel usage ,but others not. how can i do it ? it is impossible to change the IOS version of router in production, can you give some suggestion for me ,thanks
 
successfully:

Cisco 3660 c3660-js-mz.122-19b.bin
Cisco AS5300 c5300-js-mz.122-1a.bin
Cisco 3640 c3640-jsx-mz.122-27.bin
Cisco 3845 c3845-ipvoicek9-mz.124-1a.bin
Cisco 3640 c3640-is-mz.122-3.bin
Cisco 3745 c3745-is-mz.123-6.bin
cisco 3660 c3660-js-mz.122-19b.bin
cisco 3660 c3660-js-mz.122-19b.bin
Cisco 3745 c3745-js-mz.123-13a.bin

 
fail:
 
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 3845 c3845-ipvoice-mz.124-2.T.bin
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 2620 c2600-jk8s-mz.122-1a.bin
Cisco 3845 c3845-adventerprisek9-mz.123-11.T7.bin
Cisco 2821 c2800nm-spservicesk9-mz.123-11.T7.bin
 
best regards
zhaoye 2006-9-14
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From erickbe at yahoo.com  Wed Sep 13 23:17:59 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 13 Sep 2006 20:17:59 -0700 (PDT)
Subject: [cisco-voip] Delayed audio-cutover with H.323 gateway
In-Reply-To: <20060913012228.72422.qmail@web37206.mail.mud.yahoo.com>
Message-ID: <20060914031759.61841.qmail@web37202.mail.mud.yahoo.com>

FYI,

Got this fixed up by adjusting QoS policies on routers.  The core issue was the DNs involved were shared across a number of phones and the amount of control/signalling traffic exceeded the 64 packet queue-size (default). Increasing the queue-limit of the control class fixed the problem. 

----- Original Message ----
From: Erick Bergquist <erickbe at yahoo.com>
To: Wes Sisk <wsisk at cisco.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Sent: Tuesday, September 12, 2006 8:22:28 PM
Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway

Ok. I have a couple traces I am going through (CCM and debugs from router). 

The interesting part is the user is hearing a clicking sound when the delay happens (like someone is hanging up a phone) when the audio cuts in and becomes hearable. Sometimes it is very loud. Sometimes not as loud. 

When they transfer a call to another IP Phone, usually when the other person picks up the caller on PSTN side doesn't hear them for 4 to 8 seconds. Sometimes it works fine with no delay. I am leaning more toward a GW or network issue somewhere at this point. Calls on PRI's do not have this problem - just analog ports.

----- Original Message ----
From: Wes Sisk <wsisk at cisco.com>
To: Erick Bergquist <erickbe at yahoo.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Sent: Tuesday, September 12, 2006 1:57:17 PM
Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway

Fast start should not apply here.  make sure you are not dropping 
packets or retransmitting due to network loss between the 2 but fast 
start should only be necessary when you're looking at 40msec+ RTT gw to 
CM.  You have problems either on the GW or on the CM with getting those 
channels opened fast enough.

off the wall idea - enable 'file close thread flag' to make sure you 
aren't hitting that issue.  then get CCM and SDL traces from an 
occurrence.  look in the traces to obeserve the time from user offhook 
to startmediatransmittion to phone and open receive channel ack from the 
phone.  If that is small, then it is a problem with audio cut through on 
the gw itself.

/Wes

Erick Bergquist wrote:
> 4 to 8ms on average. Fast start not enabled on GW - let me look into that.
> 
> ----- Original Message ----
> From: Wes Sisk <wsisk at cisco.com>
> To: Erick Bergquist <erickbe at yahoo.com>
> Cc: ciscovoip <cisco-voip at puck.nether.net>
> Sent: Tuesday, September 12, 2006 8:31:19 AM
> Subject: Re: [cisco-voip] Delayed audio-cutover with H.323 gateway
> 
> What is round trip time from gateway to CM?  Are you using fast start  
> on CM and the gateway?
> 
> /Wes
> 
> On Sep 11, 2006, at 10:19 PM, Erick Bergquist wrote:
> 
> Hi,
> 
> I'm having a issue where an inbound call comes in FXO port and  
> connection plar's to a shared DN across 5 IP Phones. The phone call  
> gets answered fine, but the user on IP phone saids "Welcome to  
> BlahBlah" and the outside caller doesn't hear the first time they say  
> "Welcome To BlahBlah" and the IP Phone user repeats their greeting  
> and they hear it the second time fine.
> 
> There are no errors on any interfaces that I can see, and the gateway  
> has voice rtp send-recv enabled. Issue does not happen on calls over  
> PRI, just the analog FXOs. I have disabled caller-id on the voice- 
> port and also tried connection plar opx with no difference. Sometimes  
> it works fine but rarely.
> 
> CCM is version 4.1(3)sr3c and the gateway is a 1760 running 12.4(8a).
> 
> A similar problem can happen when the call is parked or put on hold  
> also, now and then.
> 
> 
> _______________________________________________
> 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 Nick.Kassel at Charles-Stanley.co.uk  Thu Sep 14 04:29:52 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 14 Sep 2006 09:29:52 +0100
Subject: [cisco-voip] vulnerable gateway?
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC014519CF@CSEXCH01.charles-stanley.co.uk>

What partitions does the Voicemail CSS need to access, we appear to have
spurious partitions added to this CSS and I'm not sure they need to be
there?

-----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, September 13, 2006 4:03 AM
To: IT; Voll, Scott; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

You can limit this with the restriction tables in unity and ultimately
with the CSS set on the Call Manager VM Port configuration. 

----- Original Message ----
From: IT <it at cimgroup.com>
To: "Voll, Scott" <Scott.Voll at wesd.org>; IT <it at cimgroup.com>;
puckcisco at cumhur.com; cisco-voip at puck.nether.net
Sent: Tuesday, September 12, 2006 5:42:04 PM
Subject: Re: [cisco-voip] vulnerable gateway?

But where in unity is someone able to route their call to any arbitrary
phone number?

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, September 12, 2006 3:37 PM
To: IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

I would agree with TAC per your CDR of CiscoUM-VI1.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Tuesday, September 12, 2006 3:29 PM
To: puckcisco at cumhur.com; IT; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

Actually, I tried both UDP and TCP.
Would it still show up under a portscan? TAC seems to think they came in
through voicemail...

-----Original Message-----
From: cumbur [mailto:zeus at cumhur.com] On Behalf Of puckcisco at cumhur.com
Sent: Tuesday, September 12, 2006 3:19 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

Dear Avidan,

H323 use TCP 1720 (not udp)  port for call initiation also don't forget
to
block SIP ports TCP/UDP 5060.

Regards.
Cumhur

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, September 13, 2006 12:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] vulnerable gateway?

I just got a call from my long distance provider that someone has been
using my PRI for many international calls. I check my CDR database
tables, and it appears that calls have been coming from one of my branch
office 2801's. But, in the CDR table, the origDeviceName alternates
between the name of the gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those
ports open, because when I tried to do a "access-list 100 deny udp any
any eq 1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up
these gateways in the first place locked them down, but hindsight is
20/20.

Thanks,
Avidan

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



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12/09/2006



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

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From manoj.kalpage at gmail.com  Thu Sep 14 05:44:33 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Thu, 14 Sep 2006 18:44:33 +0900
Subject: [cisco-voip] ISP and VPN Failover for Call Manager based
	VOIPnetwork
In-Reply-To: <AB0A35F2486F81468971AA68558BE71904CEF16A@smieln01winm01.micoanet.com>
References: <bbe09c1d0609130219k29481ba1p427a9d81bb5b9ffe@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE71904CEF16A@smieln01winm01.micoanet.com>
Message-ID: <bbe09c1d0609140244l249cff8aw8f05f7c277d5b53f@mail.gmail.com>

Hi Matthew,
What a wonderful reply. Thank you very much for your reply. I was thinking
in wrong way. We have just 1Mbps full duplex Internet connection from
Verizon and we are experiencing lot of voice quality issues recently. I know
now I should move to router based VPN. Can I do EIGRP between different
provide without having service agreement? What I heard I have to pay extra
money for EIGRP. After read your reply I did some research on the web and
found bellow link from Cisco. Do you think this is enough information for me
to implement VoIP environment you have suggested?
http://www.cisco.com/warp/public/471/dcmvpn.html
By any chance, do you have a sample configuration of your network which I
can refer?

Best Regards,
Manoj



On 9/13/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>   Manoj,
>
>
>
> Do you currently have private lines or some other circuits interconnecting
> your offices or are you planning to use VPN exclusively for voice and data?
> My major concern when using a Cisco PIX for voice would be Quality of
> Service.  While the PIX can preserve DSCP values as they are passed across
> the tunnels, unless anything has changed in 7.x, it doesn't have the
> ability to perform marking, LLQ prioritization, and traffic shaping.  This
> means that before any traffic is passed to the PIX, the device behind it (a
> switch or router) will have to perform some of these functions (say marking
> or traffic shaping).  In regards to LLQ you are out of luck.
>
>
>
> For our Teleworker VPN network we utilize a 2851 at the head-end and
> failover site and 871/877 routers at our remotes.  This gives us the
> capability to mark, LLQ, and shape traffic at the edge, before it is passed
> on to the ISP.  Additionally we utilize DMVPN and GRE to maintain routing
> information (EIGRP) and to dynamically handle routing changes when we loose
> a VPN link (say to our head-end).  I think you can do some least cost
> routing type things on the PIX to achieve the same effect, but it's much
> easier in IOS.
>
>
>
> Your ideas are sound in my opinion.  I'm sure that there are some people
> that are handling voice fine using Cisco PIX's however we had mixed results
> when we were using them.  Once we moved to the IOS VPN several of our QoS
> issues were resolved.  Regardless, you always have to remember that it still
> is the Internet and not a private network connection, so you get what you
> get.
>
>
>
> Hope this helps,
>
>
>
> -Matt
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Manoj Kalpage
> *Sent:* Wednesday, September 13, 2006 5:20 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] ISP and VPN Failover for Call Manager based
> VOIPnetwork
>
>
>
> Dear All,
>
> I am looking for ISP fail over for VoIP network. We have small
> enterprise VoIP network. If I explain our network bit, Basically we
> have call manager and unity server in main office with PIX515. All the
> branch offices has PIX 501. With attached fail over solution I am going to
> create two tunnels from each branch office and have them connected to each
> firewall in main office. I think this way if one PIX515 fail at main office,
> still branch office can be connected through second PIX515. Bellow is the
> router configuration for routing between two PIX 515. This configuration
> itself doesn't mean anything without looking at a diagram.I need to test
> this but I don't have enough gears with me right now and also I don't have
> 100% confidence on this. So, I would like to share with you folks. Any
> comments and ideas would be greatly appreciated.
>
>
>
> Please find the diagram bellow link (Sorry it's han written one )
>
>
> http://proxy.f2.ymdb.yahoofs.jp/bc/857e55a/bc/bd7f/failover.jpg?bcQM9BFBNirrJIWq
>
>
>
> best regards,
>
> Manoj
>
>
>
>
> ip cef
>
> !####Establish sla monitors for use in tracking objects####!
>
> ip sla monitor 1
> type echo protocol ipIcmpEcho 174.16.0.1
> threshold 3
> frequency 5
> ip sla monitor schedule 1 life forever start-time now
>
> ip sla monitor 2
> type echo protocol ipIcmpEcho 173.16.0.1
> threshold 3
> frequency 5
> ip sla monitor schedule 2 life forever start-time now
> !
>
> !####Configure Tracking objects (referencing IP SLA monitor's above)####!
>
> track 101 rtr 1 reachability
> !
> track 102 rtr 2 reachability
> !
> !
> !
> !
> !####Configure Interfaces with NAT####!
>
> interface FastEthernet 0/1
> ip address 172.16.0.1 255.255.0.0
> ip nat inside
>
> !
> interface Fastethernet 0/0
> ip address 173.16.0.2 255.255.255.0
> ip nat outside
>
> !
> interface Fastethernet 0/2
> ip address 174.16.0.2 255.255.255.0
> ip nat outside
>
> !
> ip classless
> !####Configure gateway of last resort with tracking objects####!
> ip route 0.0.0.0 0.0.0.0 173.16.0.1 track 101
> ip route 0.0.0.0 0.0.0.0 174.16.0.1 track 102
>
> !####Configure NAT statements for most outbound traffic####!
> ip nat inside source route-map ISP1 interface FastEthernet 0/0 overload
> ip nat inside source route-map ISP2 interface FastEthernet 0/2 overload
>
> !
> access-list 10 permit 172.16.0.0 0.0.0.255
> access-list 101 permit icmp any host 173.16.0.1 echo
> access-list 102 permit icmp any host 174.16.0.1 echo
>
> !
> !####Configure route maps for reference in NAT statements####!
> route-map ISP2 permit 10
> match ip address 10
> match interface Fastethernet 0/1
> !
> route-map ISP1 permit 10
> match ip address 10
> match interface Fastethernet 0/0
> !
>
>
>
>
>  ------------------------------
>
> *CONFIDENTIALITY STATEMENT*
>
> This communication and any attachments are *CONFIDENTIAL* and may be
> protected by one or more legal privileges. It is intended solely for the use
> of the addressee identified above. If you are not the intended recipient,
> any use, disclosure, copying or distribution of this communication is *
> UNAUTHORIZED*. Neither this information block, the typed name of the
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> ------------------------------
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>
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From aman.chugh at gmail.com  Thu Sep 14 05:53:40 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Thu, 14 Sep 2006 15:23:40 +0530
Subject: [cisco-voip] ISP and VPN Failover for Call Manager based
	VOIPnetwork
In-Reply-To: <bbe09c1d0609140244l249cff8aw8f05f7c277d5b53f@mail.gmail.com>
References: <bbe09c1d0609130219k29481ba1p427a9d81bb5b9ffe@mail.gmail.com>
	<AB0A35F2486F81468971AA68558BE71904CEF16A@smieln01winm01.micoanet.com>
	<bbe09c1d0609140244l249cff8aw8f05f7c277d5b53f@mail.gmail.com>
Message-ID: <d4c697ad0609140253q21760380t663b5f67dc2b75a6@mail.gmail.com>

Manoj,

reffer to the V3PN SRND ,it has the configurations and examples and even
test results of what you are tying to do.

Thanks
Aman


On 9/14/06, Manoj Kalpage <manoj.kalpage at gmail.com> wrote:
>
>  Hi Matthew,
> What a wonderful reply. Thank you very much for your reply. I was thinking
> in wrong way. We have just 1Mbps full duplex Internet connection from
> Verizon and we are experiencing lot of voice quality issues recently. I know
> now I should move to router based VPN. Can I do EIGRP between different
> provide without having service agreement? What I heard I have to pay extra
> money for EIGRP. After read your reply I did some research on the web and
> found bellow link from Cisco. Do you think this is enough information for me
> to implement VoIP environment you have suggested?
> http://www.cisco.com/warp/public/471/dcmvpn.html
> By any chance, do you have a sample configuration of your network which I
> can refer?
>
> Best Regards,
>  Manoj
>
>
>
>  On 9/13/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
> >   Manoj,
> >
> >
> >
> > Do you currently have private lines or some other circuits
> > interconnecting your offices or are you planning to use VPN exclusively for
> > voice and data?  My major concern when using a Cisco PIX for voice would be
> > Quality of Service.  While the PIX can preserve DSCP values as they are
> > passed across the tunnels, unless anything has changed in 7.x, it
> > doesn't have the ability to perform marking, LLQ prioritization, and traffic
> > shaping.  This means that before any traffic is passed to the PIX, the
> > device behind it (a switch or router) will have to perform some of these
> > functions (say marking or traffic shaping).  In regards to LLQ you are out
> > of luck.
> >
> >
> >
> > For our Teleworker VPN network we utilize a 2851 at the head-end and
> > failover site and 871/877 routers at our remotes.  This gives us the
> > capability to mark, LLQ, and shape traffic at the edge, before it is passed
> > on to the ISP.  Additionally we utilize DMVPN and GRE to maintain routing
> > information (EIGRP) and to dynamically handle routing changes when we loose
> > a VPN link (say to our head-end).  I think you can do some least cost
> > routing type things on the PIX to achieve the same effect, but it's much
> > easier in IOS.
> >
> >
> >
> > Your ideas are sound in my opinion.  I'm sure that there are some people
> > that are handling voice fine using Cisco PIX's however we had mixed results
> > when we were using them.  Once we moved to the IOS VPN several of our QoS
> > issues were resolved.  Regardless, you always have to remember that it still
> > is the Internet and not a private network connection, so you get what you
> > get.
> >
> >
> >
> > Hope this helps,
> >
> >
> >
> > -Matt
> >
> >
> >  ------------------------------
> >
> > *From:* cisco-voip-bounces at puck.nether.net [mailto:
> > cisco-voip-bounces at puck.nether.net] *On Behalf Of *Manoj Kalpage
> > *Sent:* Wednesday, September 13, 2006 5:20 AM
> > *To:* cisco-voip at puck.nether.net
> > *Subject:* [cisco-voip] ISP and VPN Failover for Call Manager based
> > VOIPnetwork
> >
> >
> >
> > Dear All,
> >
> > I am looking for ISP fail over for VoIP network. We have small
> > enterprise VoIP network. If I explain our network bit, Basically we
> > have call manager and unity server in main office with PIX515. All the
> > branch offices has PIX 501. With attached fail over solution I am going to
> > create two tunnels from each branch office and have them connected to each
> > firewall in main office. I think this way if one PIX515 fail at main office,
> > still branch office can be connected through second PIX515. Bellow is the
> > router configuration for routing between two PIX 515. This configuration
> > itself doesn't mean anything without looking at a diagram.I need to test
> > this but I don't have enough gears with me right now and also I don't have
> > 100% confidence on this. So, I would like to share with you folks. Any
> > comments and ideas would be greatly appreciated.
> >
> >
> >
> > Please find the diagram bellow link (Sorry it's han written one )
> >
> >
> > http://proxy.f2.ymdb.yahoofs.jp/bc/857e55a/bc/bd7f/failover.jpg?bcQM9BFBNirrJIWq
> >
> >
> >
> > best regards,
> >
> > Manoj
> >
> >
> >
> >
> > ip cef
> >
> > !####Establish sla monitors for use in tracking objects####!
> >
> > ip sla monitor 1
> > type echo protocol ipIcmpEcho 174.16.0.1
> > threshold 3
> > frequency 5
> > ip sla monitor schedule 1 life forever start-time now
> >
> > ip sla monitor 2
> > type echo protocol ipIcmpEcho 173.16.0.1
> > threshold 3
> > frequency 5
> > ip sla monitor schedule 2 life forever start-time now
> > !
> >
> > !####Configure Tracking objects (referencing IP SLA monitor's
> > above)####!
> >
> > track 101 rtr 1 reachability
> > !
> > track 102 rtr 2 reachability
> > !
> > !
> > !
> > !
> > !####Configure Interfaces with NAT####!
> >
> > interface FastEthernet 0/1
> > ip address 172.16.0.1 255.255.0.0
> > ip nat inside
> >
> > !
> > interface Fastethernet 0/0
> > ip address 173.16.0.2 255.255.255.0
> > ip nat outside
> >
> > !
> > interface Fastethernet 0/2
> > ip address 174.16.0.2 255.255.255.0
> > ip nat outside
> >
> > !
> > ip classless
> > !####Configure gateway of last resort with tracking objects####!
> > ip route 0.0.0.0 0.0.0.0 173.16.0.1 track 101
> > ip route 0.0.0.0 0.0.0.0 174.16.0.1 track 102
> >
> > !####Configure NAT statements for most outbound traffic####!
> > ip nat inside source route-map ISP1 interface FastEthernet 0/0 overload
> > ip nat inside source route-map ISP2 interface FastEthernet 0/2 overload
> >
> > !
> > access-list 10 permit 172.16.0.0 0.0.0.255
> > access-list 101 permit icmp any host 173.16.0.1 echo
> > access-list 102 permit icmp any host 174.16.0.1 echo
> >
> > !
> > !####Configure route maps for reference in NAT statements####!
> > route-map ISP2 permit 10
> > match ip address 10
> > match interface Fastethernet 0/1
> > !
> > route-map ISP1 permit 10
> > match ip address 10
> > match interface Fastethernet 0/0
> > !
> >
> >
> >
> >
> >  ------------------------------
> >
> > *CONFIDENTIALITY STATEMENT*
> >
> > This communication and any attachments are *CONFIDENTIAL* and may be
> > protected by one or more legal privileges. It is intended solely for the use
> > of the addressee identified above. If you are not the intended recipient,
> > any use, disclosure, copying or distribution of this communication is *
> > UNAUTHORIZED*. Neither this information block, the typed name of the
> > sender, nor anything else in this message is intended to constitute an
> > electronic signature unless a specific statement to the contrary is included
> > in this message. If you have received this communication in error, please
> > immediately contact me and delete this communication from your computer.
> > Thank you.
> >
> > ------------------------------
> >
> >
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From SCASPER at mandtbank.com  Thu Sep 14 08:14:17 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Thu, 14 Sep 2006 08:14:17 -0400
Subject: [cisco-voip] call quality issues between gateway and Unity
Message-ID: <s5090f6b.066@comrcm01intgwia.mandtbank.com>

It is in the document Playout Delay Enhancements:

http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/products_feature_guide09186a008008033c.html


in the link on Understanding Jitter that you posted.

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800945df.shtml


Steve

>>> "Wes Sisk" <wsisk at cisco.com> 09/13/06 6:30 PM >>>
Steven,

Where do you get that playout-delay is not supported?

/Wes

STEVEN CASPER wrote:
> Wes,
> I see that Playout-Delay is not supported on MGCP platforms so how
is
> jitter handled in a MGCP/Call Manager enviroment?
> 
> Thanks!
> Steve
> 
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026
> 
>>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
> Kris,
> 
> This sounds very similar to something we fixed long ago.  Take a look

> 
> at 'sh voice call' to see if it's jitter:
> http://www.cisco.com/en/US/tech/tk652/tk698/ 
> technologies_tech_note09186a00800945df.shtml
> 
> CSCea35850 - Unity RTP stream has inherent jitter
> 
> 
> If you are using non-default packetization periods watch out for
> CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and 

> G711/30ms
> 
> /Wes
> 
> On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:
> 
> Hi
> 
> I have a customer with a multi site centralized CCM 4.13 cluster.   
> The remote sites are getting complaints of poor voice quality from  
> customers when being transferred to Unity (which is located in the  
> central site) .  The callers cannot understand the greetings and  
> describe them as choppy.  There are no problems with external calls 

> transferred from the remote sites to IP phones at the central site or

> 
> for intersite calls.  It seems to be limited to calls where the RTP 

> stream is between a remote voice gateway and Unity.  These calls are 

> G.729 so Unity has to transcode the greetings on the fly but if that 

> were the problem I'd expect remote users to have the same problem  
> when calling into check their messages or leave messages for other  
> users.  Also, the Unity server is not very heavily used; it generally

> 
> has no more than 9 active calls at the most.
> 
> I suspect the gateway might be the problem.  It's a 2851 running  
> MGCP.  I guess I'd like to know if anyone agrees or disagrees with  
> this assumption.  Also, I've had a lot of problems with MGCP code on 

> the ISRs.  Can someone recommend an IOS version that is stable for  
> and MGCP gateway and doesn't suffer from DSP firmware issues?
> 
> I should mention that Unity is version 4.04 sr1 and the gateway is  
> running 12.4(5).
> 
> Thanks
> 



From lelio at uoguelph.ca  Thu Sep 14 08:40:03 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 14 Sep 2006 08:40:03 -0400
Subject: [cisco-voip] vulnerable gateway?
References: <253FAE7A06B11B49A3A29770B2B8BAFC014519CF@CSEXCH01.charles-stanley.co.uk>
Message-ID: <002e01c6d7fa$f052ec80$47196883@cfs.uoguelph.ca>

I believe all you really need are all the partitions the phones are in and if you are sending any calls to IPCC, you need the partitions the CTI route points and CTI ports are in. In addition, if you are using CTI route points as phantom numbers for forwarding, etc. And any partition that contains translations. 

The best would be to review each of your partitions and decide if Unity has to call it. I don't think there are any hard and fast rules of what it needs, etc. Previous versions of callmanager (3.x and below I believe) that used call forwarding would need access to the partition the voicemail ports are in as well. That was definately a hard and fast rule.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Nick Kassel 
  To: Erick Bergquist ; IT ; Voll, Scott ; puckcisco at cumhur.com ; cisco-voip at puck.nether.net 
  Sent: Thursday, September 14, 2006 4:29 AM
  Subject: Re: [cisco-voip] vulnerable gateway?


  What partitions does the Voicemail CSS need to access, we appear to have
  spurious partitions added to this CSS and I'm not sure they need to be
  there?

  -----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, September 13, 2006 4:03 AM
  To: IT; Voll, Scott; puckcisco at cumhur.com; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] vulnerable gateway?

  You can limit this with the restriction tables in unity and ultimately
  with the CSS set on the Call Manager VM Port configuration. 

  ----- Original Message ----
  From: IT <it at cimgroup.com>
  To: "Voll, Scott" <Scott.Voll at wesd.org>; IT <it at cimgroup.com>;
  puckcisco at cumhur.com; cisco-voip at puck.nether.net
  Sent: Tuesday, September 12, 2006 5:42:04 PM
  Subject: Re: [cisco-voip] vulnerable gateway?

  But where in unity is someone able to route their call to any arbitrary
  phone number?

  -----Original Message-----
  From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
  Sent: Tuesday, September 12, 2006 3:37 PM
  To: IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
  Subject: RE: [cisco-voip] vulnerable gateway?

  I would agree with TAC per your CDR of CiscoUM-VI1.

  Scott

  -----Original Message-----
  From: cisco-voip-bounces at puck.nether.net
  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
  Sent: Tuesday, September 12, 2006 3:29 PM
  To: puckcisco at cumhur.com; IT; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] vulnerable gateway?

  Actually, I tried both UDP and TCP.
  Would it still show up under a portscan? TAC seems to think they came in
  through voicemail...

  -----Original Message-----
  From: cumbur [mailto:zeus at cumhur.com] On Behalf Of puckcisco at cumhur.com
  Sent: Tuesday, September 12, 2006 3:19 PM
  To: IT; cisco-voip at puck.nether.net
  Subject: RE: [cisco-voip] vulnerable gateway?

  Dear Avidan,

  H323 use TCP 1720 (not udp)  port for call initiation also don't forget
  to
  block SIP ports TCP/UDP 5060.

  Regards.
  Cumhur

  -----Original Message-----
  From: cisco-voip-bounces at puck.nether.net
  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
  Sent: Wednesday, September 13, 2006 12:59 AM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] vulnerable gateway?

  I just got a call from my long distance provider that someone has been
  using my PRI for many international calls. I check my CDR database
  tables, and it appears that calls have been coming from one of my branch
  office 2801's. But, in the CDR table, the origDeviceName alternates
  between the name of the gateway and CiscoUM-VI1.
  I ran a port scan against the router, and found that h.323 and callbook
  ports were open to the public. I shutdown the interface that had those
  ports open, because when I tried to do a "access-list 100 deny udp any
  any eq 1720" it still shows as open on the portscan.

  How can I secure/lock H.323 on these branch devices?
  How did someone utilize my gateway to make these calls?
  How can I avoid this in the future?

  I guess I should have made sure that the consulting group that set up
  these gateways in the first place locked them down, but hindsight is
  20/20.

  Thanks,
  Avidan

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



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  12/09/2006



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

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




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From wsisk at cisco.com  Thu Sep 14 09:01:09 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 14 Sep 2006 09:01:09 -0400
Subject: [cisco-voip] call quality issues between gateway and Unity
In-Reply-To: <s5090f6b.065@comrcm01intgwia.mandtbank.com>
References: <s5090f6b.065@comrcm01intgwia.mandtbank.com>
Message-ID: <F5D84B82-E7C4-4613-95AB-03E2262C9D2D@cisco.com>

Steven,

Thank you for the clarification. Take a look at the Release-note of  
the cited bug in bug toolkit.

There is a different command to use with MGCP:
mgcp playout adaptive 60 40 200

Same underlying behavior, just modified for the call signaling protocol.

/Wes

On Sep 14, 2006, at 8:14 AM, STEVEN CASPER wrote:

It is in the document Playout Delay Enhancements:

http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/ 
products_feature_guide09186a008008033c.html


in the link on Understanding Jitter that you posted.

http://www.cisco.com/en/US/tech/tk652/tk698/ 
technologies_tech_note09186a00800945df.shtml


Steve

>>> "Wes Sisk" <wsisk at cisco.com> 09/13/06 6:30 PM >>>
Steven,

Where do you get that playout-delay is not supported?

/Wes

STEVEN CASPER wrote:
> Wes,
> I see that Playout-Delay is not supported on MGCP platforms so how
is
> jitter handled in a MGCP/Call Manager enviroment?
>
> Thanks!
> Steve
>
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026
>
>>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
> Kris,
>
> This sounds very similar to something we fixed long ago.  Take a look

>
> at 'sh voice call' to see if it's jitter:
> http://www.cisco.com/en/US/tech/tk652/tk698/
> technologies_tech_note09186a00800945df.shtml
>
> CSCea35850 - Unity RTP stream has inherent jitter
>
>
> If you are using non-default packetization periods watch out for
> CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and

> G711/30ms
>
> /Wes
>
> On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:
>
> Hi
>
> I have a customer with a multi site centralized CCM 4.13 cluster.
> The remote sites are getting complaints of poor voice quality from
> customers when being transferred to Unity (which is located in the
> central site) .  The callers cannot understand the greetings and
> describe them as choppy.  There are no problems with external calls

> transferred from the remote sites to IP phones at the central site or

>
> for intersite calls.  It seems to be limited to calls where the RTP

> stream is between a remote voice gateway and Unity.  These calls are

> G.729 so Unity has to transcode the greetings on the fly but if that

> were the problem I'd expect remote users to have the same problem
> when calling into check their messages or leave messages for other
> users.  Also, the Unity server is not very heavily used; it generally

>
> has no more than 9 active calls at the most.
>
> I suspect the gateway might be the problem.  It's a 2851 running
> MGCP.  I guess I'd like to know if anyone agrees or disagrees with
> this assumption.  Also, I've had a lot of problems with MGCP code on

> the ISRs.  Can someone recommend an IOS version that is stable for
> and MGCP gateway and doesn't suffer from DSP firmware issues?
>
> I should mention that Unity is version 4.04 sr1 and the gateway is
> running 12.4(5).
>
> Thanks
>


From Nick.Kassel at Charles-Stanley.co.uk  Thu Sep 14 09:12:40 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 14 Sep 2006 14:12:40 +0100
Subject: [cisco-voip] vulnerable gateway?
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC014519D6@CSEXCH01.charles-stanley.co.uk>

Thanks Lelio

 

I thought that was the case but didn't want to remove them too soon just
in case.

 

 

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Thursday, September 14, 2006 1:40 PM
To: Nick Kassel; Erick Bergquist; IT; Voll, Scott; puckcisco at cumhur.com;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

 

I believe all you really need are all the partitions the phones are in
and if you are sending any calls to IPCC, you need the partitions the
CTI route points and CTI ports are in. In addition, if you are using CTI
route points as phantom numbers for forwarding, etc. And any partition
that contains translations. 

 

The best would be to review each of your partitions and decide if Unity
has to call it. I don't think there are any hard and fast rules of what
it needs, etc. Previous versions of callmanager (3.x and below I
believe) that used call forwarding would need access to the partition
the voicemail ports are in as well. That was definately a hard and fast
rule.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."

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

	From: Nick Kassel <mailto:Nick.Kassel at charles-stanley.co.uk>  

	To: Erick Bergquist <mailto:erickbe at yahoo.com>  ; IT
<mailto:it at cimgroup.com>  ; Voll, Scott <mailto:Scott.Voll at wesd.org>  ;
puckcisco at cumhur.com ; cisco-voip at puck.nether.net 

	Sent: Thursday, September 14, 2006 4:29 AM

	Subject: Re: [cisco-voip] vulnerable gateway?

	 

	What partitions does the Voicemail CSS need to access, we appear
to have
	spurious partitions added to this CSS and I'm not sure they need
to be
	there?
	
	-----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, September 13, 2006 4:03 AM
	To: IT; Voll, Scott; puckcisco at cumhur.com;
cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] vulnerable gateway?
	
	You can limit this with the restriction tables in unity and
ultimately
	with the CSS set on the Call Manager VM Port configuration. 
	
	----- Original Message ----
	From: IT <it at cimgroup.com>
	To: "Voll, Scott" <Scott.Voll at wesd.org>; IT <it at cimgroup.com>;
	puckcisco at cumhur.com; cisco-voip at puck.nether.net
	Sent: Tuesday, September 12, 2006 5:42:04 PM
	Subject: Re: [cisco-voip] vulnerable gateway?
	
	But where in unity is someone able to route their call to any
arbitrary
	phone number?
	
	-----Original Message-----
	From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
	Sent: Tuesday, September 12, 2006 3:37 PM
	To: IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] vulnerable gateway?
	
	I would agree with TAC per your CDR of CiscoUM-VI1.
	
	Scott
	
	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
	[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
	Sent: Tuesday, September 12, 2006 3:29 PM
	To: puckcisco at cumhur.com; IT; cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] vulnerable gateway?
	
	Actually, I tried both UDP and TCP.
	Would it still show up under a portscan? TAC seems to think they
came in
	through voicemail...
	
	-----Original Message-----
	From: cumbur [mailto:zeus at cumhur.com] On Behalf Of
puckcisco at cumhur.com
	Sent: Tuesday, September 12, 2006 3:19 PM
	To: IT; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] vulnerable gateway?
	
	Dear Avidan,
	
	H323 use TCP 1720 (not udp)  port for call initiation also don't
forget
	to
	block SIP ports TCP/UDP 5060.
	
	Regards.
	Cumhur
	
	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
	[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
	Sent: Wednesday, September 13, 2006 12:59 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] vulnerable gateway?
	
	I just got a call from my long distance provider that someone
has been
	using my PRI for many international calls. I check my CDR
database
	tables, and it appears that calls have been coming from one of
my branch
	office 2801's. But, in the CDR table, the origDeviceName
alternates
	between the name of the gateway and CiscoUM-VI1.
	I ran a port scan against the router, and found that h.323 and
callbook
	ports were open to the public. I shutdown the interface that had
those
	ports open, because when I tried to do a "access-list 100 deny
udp any
	any eq 1720" it still shows as open on the portscan.
	
	How can I secure/lock H.323 on these branch devices?
	How did someone utilize my gateway to make these calls?
	How can I avoid this in the future?
	
	I guess I should have made sure that the consulting group that
set up
	these gateways in the first place locked them down, but
hindsight is
	20/20.
	
	Thanks,
	Avidan
	
	_______________________________________________
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	https://puck.nether.net/mailman/listinfo/cisco-voip
	
	
	
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From CMStaffin at ra.rockwell.com  Thu Sep 14 09:14:34 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Thu, 14 Sep 2006 08:14:34 -0500
Subject: [cisco-voip] ISP and VPN Failover for Call Manager
	based	VOIPnetwork
In-Reply-To: <bbe09c1d0609140244l249cff8aw8f05f7c277d5b53f@mail.gmail.com>
Message-ID: <OF3BC4355C.B4F7B2CC-ON862571E9.00488E28-862571E9.0048BBDC@ra.rockwell.com>

Manoj,

Your best solution is to do IPSEC tunnell within a GRE tunnell.  This will 
allow you to do EIGRP as well as CDP works now through GRE.  This can and 
does work across as many vendors as you want.  We are currently useing 
this exact setup across atleast 10 different vendors.  The only thing to 
be careful of is be very picky about your vendors and make sure that the 
peering between your two vendors is very good and reliable.

Craig





"Manoj Kalpage" <manoj.kalpage at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
09/14/2006 04:44 AM

To
"Linsemier, Matthew" <MLinsemier at apcapital.com>
cc
cisco-voip at puck.nether.net
Subject
Re: [cisco-voip] ISP and VPN Failover for Call Manager based VOIPnetwork






Hi Matthew,
What a wonderful reply. Thank you very much for your reply. I was thinking 
in wrong way. We have just 1Mbps full duplex Internet connection from 
Verizon and we are experiencing lot of voice quality issues recently. I 
know now I should move to router based VPN. Can I do EIGRP between 
different provide without having service agreement? What I heard I have to 
pay extra money for EIGRP. After read your reply I did some research on 
the web and found bellow link from Cisco. Do you think this is enough 
information for me to implement VoIP environment you have suggested? 
http://www.cisco.com/warp/public/471/dcmvpn.html
By any chance, do you have a sample configuration of your network which I 
can refer?
 
Best Regards,
Manoj
 

 
On 9/13/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote: 
Manoj,
 
Do you currently have private lines or some other circuits interconnecting 
your offices or are you planning to use VPN exclusively for voice and 
data?  My major concern when using a Cisco PIX for voice would be Quality 
of Service.  While the PIX can preserve DSCP values as they are passed 
across the tunnels, unless anything has changed in 7.x, it doesn't have 
the ability to perform marking, LLQ prioritization, and traffic shaping. 
This means that before any traffic is passed to the PIX, the device behind 
it (a switch or router) will have to perform some of these functions (say 
marking or traffic shaping).  In regards to LLQ you are out of luck. 
 
For our Teleworker VPN network we utilize a 2851 at the head-end and 
failover site and 871/877 routers at our remotes.  This gives us the 
capability to mark, LLQ, and shape traffic at the edge, before it is 
passed on to the ISP.  Additionally we utilize DMVPN and GRE to maintain 
routing information (EIGRP) and to dynamically handle routing changes when 
we loose a VPN link (say to our head-end).  I think you can do some least 
cost routing type things on the PIX to achieve the same effect, but it's 
much easier in IOS. 
 
Your ideas are sound in my opinion.  I'm sure that there are some people 
that are handling voice fine using Cisco PIX's however we had mixed 
results when we were using them.  Once we moved to the IOS VPN several of 
our QoS issues were resolved.  Regardless, you always have to remember 
that it still is the Internet and not a private network connection, so you 
get what you get. 
 
Hope this helps,
 
-Matt
 

From: cisco-voip-bounces at puck.nether.net [mailto:
cisco-voip-bounces at puck.nether.net] On Behalf Of Manoj Kalpage
Sent: Wednesday, September 13, 2006 5:20 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ISP and VPN Failover for Call Manager based 
VOIPnetwork
 
Dear All,
I am looking for ISP fail over for VoIP network. We have small enterprise 
VoIP network. If I explain our network bit, Basically we have call manager 
and unity server in main office with PIX515. All the branch offices has 
PIX 501. With attached fail over solution I am going to create two tunnels 
from each branch office and have them connected to each firewall in main 
office. I think this way if one PIX515 fail at main office, still branch 
office can be connected through second PIX515. Bellow is the router 
configuration for routing between two PIX 515. This configuration itself 
doesn't mean anything without looking at a diagram.I need to test this but 
I don't have enough gears with me right now and also I don't have 100% 
confidence on this. So, I would like to share with you folks. Any comments 
and ideas would be greatly appreciated. 
 
Please find the diagram bellow link (Sorry it's han written one )
http://proxy.f2.ymdb.yahoofs.jp/bc/857e55a/bc/bd7f/failover.jpg?bcQM9BFBNirrJIWq
 
best regards,
Manoj
 

ip cef
!####Establish sla monitors for use in tracking objects####!
ip sla monitor 1
type echo protocol ipIcmpEcho 174.16.0.1
threshold 3
frequency 5
ip sla monitor schedule 1 life forever start-time now 
ip sla monitor 2
type echo protocol ipIcmpEcho 173.16.0.1
threshold 3
frequency 5
ip sla monitor schedule 2 life forever start-time now 
!
!####Configure Tracking objects (referencing IP SLA monitor's above)####!
track 101 rtr 1 reachability
!
track 102 rtr 2 reachability
!
!
!
!
!####Configure Interfaces with NAT####!
interface FastEthernet 0/1
ip address 172.16.0.1 255.255.0.0
ip nat inside
!
interface Fastethernet 0/0
ip address 173.16.0.2 255.255.255.0
ip nat outside
!
interface Fastethernet 0/2
ip address 174.16.0.2 255.255.255.0
ip nat outside
!
ip classless
!####Configure gateway of last resort with tracking objects####!
ip route 0.0.0.0 0.0.0.0 173.16.0.1 track 101 
ip route 0.0.0.0 0.0.0.0 174.16.0.1 track 102
!####Configure NAT statements for most outbound traffic####!
ip nat inside source route-map ISP1 interface FastEthernet 0/0 overload
ip nat inside source route-map ISP2 interface FastEthernet 0/2 overload 
!
access-list 10 permit 172.16.0.0 0.0.0.255
access-list 101 permit icmp any host 173.16.0.1 echo
access-list 102 permit icmp any host 174.16.0.1 echo
!
!####Configure route maps for reference in NAT statements####!
route-map ISP2 permit 10
match ip address 10
match interface Fastethernet 0/1 
!
route-map ISP1 permit 10
match ip address 10
match interface Fastethernet 0/0 
! 
 


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From Matt.Slaga at us.didata.com  Thu Sep 14 09:17:31 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 14 Sep 2006 09:17:31 -0400
Subject: [cisco-voip] BAT question--  CCM 4.01
Message-ID: <C1FE15183DA37645BC0633BC604E44F00324910D@USNAEXCH.na.didata.local>

Unless you already have your softkey templates assigned per device (not
set to default), the easiest way to change it is through the device
pools.  It is generally a good idea to leave all DP settings as default
on the phones so you don't have to change potentially hundreds or
thousands of individual devices but just a handful of device pools.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, September 13, 2006 3:28 PM
To: Cisco Voip
Subject: [cisco-voip] BAT question-- CCM 4.01

 

Is it possible to update the softkey template globally using BAT? 

I've designed a new softkey template that applies QRT and MCIS 
but I cannot figure out how to get that softkey template to be in use on
the phones globally.. 
(or even a test subset of phones) 

Softkey template doesn't appear in the BAT menus for phone or for line. 

I could change it on every phone, but that would NOT be workable with
500 phones... 

Does this mean that softkey settings are globally configured somewhere
else? 

I see device profiles as a possibility, but my vendor didn't configre
any device profile 
apparently. When I do a search for "device profile" and leave the box
empty, 
there's nothing reported in the search 

ideas?? 

thanks, Tim 




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From SCASPER at mandtbank.com  Thu Sep 14 09:31:58 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Thu, 14 Sep 2006 09:31:58 -0400
Subject: [cisco-voip] call quality issues between gateway and Unity
Message-ID: <s509219f.076@comrcm01intgwia.mandtbank.com>

Thanks Wes,
This is very interesting. So is the recommendation to add "mgcp playout
adaptive 60 40 200" to all voice gateways that allow non IP phone users
to access Unity? I have a lot of gateways that connect Nortel users to
Unity via QSIG gateways. We occasionally get complaints of canned or
robotic voice quality. We are running CCM 4.1.3sr1 with Unity 4.05.

Steve

>>> "Wes Sisk" <wsisk at cisco.com> 09/14/06 9:01 AM >>>
Steven,

Thank you for the clarification. Take a look at the Release-note of  
the cited bug in bug toolkit.

There is a different command to use with MGCP:
mgcp playout adaptive 60 40 200

Same underlying behavior, just modified for the call signaling
protocol.

/Wes

On Sep 14, 2006, at 8:14 AM, STEVEN CASPER wrote:

It is in the document Playout Delay Enhancements:

http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/ 
products_feature_guide09186a008008033c.html


in the link on Understanding Jitter that you posted.

http://www.cisco.com/en/US/tech/tk652/tk698/ 
technologies_tech_note09186a00800945df.shtml


Steve

>>> "Wes Sisk" <wsisk at cisco.com> 09/13/06 6:30 PM >>>
Steven,

Where do you get that playout-delay is not supported?

/Wes

STEVEN CASPER wrote:
> Wes,
> I see that Playout-Delay is not supported on MGCP platforms so how
is
> jitter handled in a MGCP/Call Manager enviroment?
>
> Thanks!
> Steve
>
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026
>
>>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
> Kris,
>
> This sounds very similar to something we fixed long ago.  Take a
look

>
> at 'sh voice call' to see if it's jitter:
> http://www.cisco.com/en/US/tech/tk652/tk698/ 
> technologies_tech_note09186a00800945df.shtml
>
> CSCea35850 - Unity RTP stream has inherent jitter
>
>
> If you are using non-default packetization periods watch out for
> CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and

> G711/30ms
>
> /Wes
>
> On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:
>
> Hi
>
> I have a customer with a multi site centralized CCM 4.13 cluster.
> The remote sites are getting complaints of poor voice quality from
> customers when being transferred to Unity (which is located in the
> central site) .  The callers cannot understand the greetings and
> describe them as choppy.  There are no problems with external calls

> transferred from the remote sites to IP phones at the central site
or

>
> for intersite calls.  It seems to be limited to calls where the RTP

> stream is between a remote voice gateway and Unity.  These calls are

> G.729 so Unity has to transcode the greetings on the fly but if that

> were the problem I'd expect remote users to have the same problem
> when calling into check their messages or leave messages for other
> users.  Also, the Unity server is not very heavily used; it
generally

>
> has no more than 9 active calls at the most.
>
> I suspect the gateway might be the problem.  It's a 2851 running
> MGCP.  I guess I'd like to know if anyone agrees or disagrees with
> this assumption.  Also, I've had a lot of problems with MGCP code on

> the ISRs.  Can someone recommend an IOS version that is stable for
> and MGCP gateway and doesn't suffer from DSP firmware issues?
>
> I should mention that Unity is version 4.04 sr1 and the gateway is
> running 12.4(5).
>
> Thanks
>




From wsisk at cisco.com  Thu Sep 14 10:42:28 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 14 Sep 2006 10:42:28 -0400
Subject: [cisco-voip] call quality issues between gateway and Unity
In-Reply-To: <s509219f.075@comrcm01intgwia.mandtbank.com>
References: <s509219f.075@comrcm01intgwia.mandtbank.com>
Message-ID: <45096A54.3070501@cisco.com>

Steven,

No, this is not a blanket best practice recommendation.  It was a 
workaround for the issue cited in that cdets.   If you have the time, 
the best approach is to identify the cause of the robotic voice and 
modify the smallest possible set of options to address that issue.

/Wes

STEVEN CASPER wrote:
> Thanks Wes,
> This is very interesting. So is the recommendation to add "mgcp playout
> adaptive 60 40 200" to all voice gateways that allow non IP phone users
> to access Unity? I have a lot of gateways that connect Nortel users to
> Unity via QSIG gateways. We occasionally get complaints of canned or
> robotic voice quality. We are running CCM 4.1.3sr1 with Unity 4.05.
> 
> Steve
> 
>>>> "Wes Sisk" <wsisk at cisco.com> 09/14/06 9:01 AM >>>
> Steven,
> 
> Thank you for the clarification. Take a look at the Release-note of  
> the cited bug in bug toolkit.
> 
> There is a different command to use with MGCP:
> mgcp playout adaptive 60 40 200
> 
> Same underlying behavior, just modified for the call signaling
> protocol.
> 
> /Wes
> 
> On Sep 14, 2006, at 8:14 AM, STEVEN CASPER wrote:
> 
> It is in the document Playout Delay Enhancements:
> 
> http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/ 
> products_feature_guide09186a008008033c.html
> 
> 
> in the link on Understanding Jitter that you posted.
> 
> http://www.cisco.com/en/US/tech/tk652/tk698/ 
> technologies_tech_note09186a00800945df.shtml
> 
> 
> Steve
> 
>>>> "Wes Sisk" <wsisk at cisco.com> 09/13/06 6:30 PM >>>
> Steven,
> 
> Where do you get that playout-delay is not supported?
> 
> /Wes
> 
> STEVEN CASPER wrote:
>> Wes,
>> I see that Playout-Delay is not supported on MGCP platforms so how
> is
>> jitter handled in a MGCP/Call Manager enviroment?
>>
>> Thanks!
>> Steve
>>
>> Steve Casper
>> Voice Technologies
>> M&T Bank
>> (410) 347-6026
>>
>>>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
>> Kris,
>>
>> This sounds very similar to something we fixed long ago.  Take a
> look
> 
>> at 'sh voice call' to see if it's jitter:
>> http://www.cisco.com/en/US/tech/tk652/tk698/ 
>> technologies_tech_note09186a00800945df.shtml
>>
>> CSCea35850 - Unity RTP stream has inherent jitter
>>
>>
>> If you are using non-default packetization periods watch out for
>> CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and
> 
>> G711/30ms
>>
>> /Wes
>>
>> On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:
>>
>> Hi
>>
>> I have a customer with a multi site centralized CCM 4.13 cluster.
>> The remote sites are getting complaints of poor voice quality from
>> customers when being transferred to Unity (which is located in the
>> central site) .  The callers cannot understand the greetings and
>> describe them as choppy.  There are no problems with external calls
> 
>> transferred from the remote sites to IP phones at the central site
> or
> 
>> for intersite calls.  It seems to be limited to calls where the RTP
> 
>> stream is between a remote voice gateway and Unity.  These calls are
> 
>> G.729 so Unity has to transcode the greetings on the fly but if that
> 
>> were the problem I'd expect remote users to have the same problem
>> when calling into check their messages or leave messages for other
>> users.  Also, the Unity server is not very heavily used; it
> generally
> 
>> has no more than 9 active calls at the most.
>>
>> I suspect the gateway might be the problem.  It's a 2851 running
>> MGCP.  I guess I'd like to know if anyone agrees or disagrees with
>> this assumption.  Also, I've had a lot of problems with MGCP code on
> 
>> the ISRs.  Can someone recommend an IOS version that is stable for
>> and MGCP gateway and doesn't suffer from DSP firmware issues?
>>
>> I should mention that Unity is version 4.04 sr1 and the gateway is
>> running 12.4(5).
>>
>> Thanks
>>

From baryonyx5 at gmail.com  Thu Sep 14 10:54:40 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Thu, 14 Sep 2006 09:54:40 -0500
Subject: [cisco-voip] IP phone users hearing other users' calls
In-Reply-To: <005301c6d765$6393b0d0$47196883@cfs.uoguelph.ca>
References: <a749d0620609131131l790c7aect64b2bc3e526dff8c@mail.gmail.com>
	<005301c6d765$6393b0d0$47196883@cfs.uoguelph.ca>
Message-ID: <a749d0620609140754k2ea6aadep1d26f058401a5474@mail.gmail.com>

No, an IP address conflict wouldn't manifest itself in this way.

Thanks for the reply though.

On 9/13/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>  could it be an IP address conflict?
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."
>
> ----- Original Message -----
> *From:* Kris Seraphine <baryonyx5 at gmail.com>
> *To:* cisco-voip <cisco-voip at puck.nether.net>
> *Sent:* Wednesday, September 13, 2006 2:31 PM
> *Subject:* [cisco-voip] IP phone users hearing other users' calls
>
> I was contacted by a business that has an ICS7750 running CCM 3.3.2.  They
> are having a strange problem.  A user will be on a external call and begin
> hearing audio from another separate call between a different IP phone and
> external caller.  According to the customer, only the IP phone users can
> hear each others conversations, the external parties do not know its
> happening.  The problem is random but has been happening for quite a while.
> I'm told that when it happens, the other call is not muffled or softer but
> sounds loud and clear.
>
> My first thought is that they have a wiring issue with their analog lines
> but I would think the PSTN end of the call would notice the problem as well
> if this were the problem.
>
> I'm not sure how this could happen on the IP side of the call but since
> its a 7750 running old code I thought I'd ask.
>
> If anyone has any thoughts or has experienced a similar problem, please
> let me know.
>
> The MRP is running 12.2(4r)XM, I haven't been able to get into the SSP
> yet.
>
> thanks
>
> --
> kris seraphine
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
kris seraphine
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From Richard.vanBerkel at telindus.nl  Thu Sep 14 11:16:55 2006
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Thu, 14 Sep 2006 17:16:55 +0200
Subject: [cisco-voip] Unity (Unified Messaging) Importing users through
	Active Directory
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D886E9A452@tnl1mx1.telindus.intra>

 

Hi,

 

I have a problem importing users through active directory. The
permissions seem to be ok.

 

Exchange 2003 delegated control:

 

UnityInstall      Exchange  Full Administrator

UnityDirSvc       Exchange  Full Administrator

UnityMsgStoreSvc  Exchange  Full Administrator

 

Active directory Unity Accounts have the following rights:

*         Installation Account:   UnityInstall      Domain Admin      +
Local Administrator

*         Administration:         UnityAdmin        Domain User       +
Local Administrator 

*         Directory Services:     UnityDirSvc       Domain Admin      +
Local Administrator

*         Unity Message Store:    UnityMsgStoreSvc  Domain Admin      +
Local Administrator

 

Problem is that whenever I import a user,I get an error message.

The user is in the subscriber list of the unity server, but there is no
way of selecting this user.

I can find the user by doing a search on extension.

When it finds the users there is no Surname or Firstname.

 

The only thing I can do is deleting it with Bulk delete.

 

Any ideas?

 

 

Richard van Berkel

Senior Customer Service

Profile: MCSE-2003, Microsoft Exchange 2003, Cisco Certified Voip
Specialist

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

Telindus B.V.

Savannahweg 19 

3542 AW Utrecht

Netherlands

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

mailto: Richard.vanBerkel at telindus.nl
<mailto:mailto:%20Richard.vanBerkel at telindus.nl> 

tel:         +31 302 477 515 

fax:       +31 302 477 711 

mobile: +31 628 903 277

 

Telindus. Change things your way.

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

	
	 	 
 

 






33rd International Telindus Symposium
Thursday, 26 October 2006, Tour & Taxis, Brussels, Belgium 
Change your vision & see how ICT changes your business.
The one-day ICT event for business professionals!

Programme and free registration > http://www.telindussymposium.com


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From Sechaba.Mokoena at btgroup.co.za  Thu Sep 14 11:22:19 2006
From: Sechaba.Mokoena at btgroup.co.za (Sechaba Mokoena)
Date: Thu, 14 Sep 2006 17:22:19 +0200
Subject: [cisco-voip]  Device Profiles and Phones in BAT
Message-ID: <55C485040101F640952DA48A70AD738501F3F783@midms01.btg>

Hi,

 

I have phones and devices profiles that share the same DN. Using BAT, I
wanted to change the line Calling Search Space on the phone, but this
also changes the line Calling Search Space on the device profile. Is
there a specific query that I can run in BAT that will only change the
line Calling Search Space on the phone and not the line Calling Search
Space on the Device Profile?

 

Regards,

 

Sechaba Mokoena

Systems Engineer

Bytes Technology Group - Systems Integration


Tel : (+27) (11) 205-7000 |  Fax: (+27) (11) 205-7584  |  Cell : (+27)
84 798-6803
Web  : www.btgroup.co.za  |  Press Office : www.itweb.co.za/office/bytes
<blocked::http://www.itweb.co.za/office/bytes>    |   Licensing :
www.purelicensing.co.za <blocked::http://www.purelicensing.co.za/>  

 

Bytes Systems Integration (Pty) Ltd : Registration No: 1995/012031/07
A subsidiary of Bytes Technology Group SA (Pty) Ltd,   In association
with KAGISO
P O Box 5905, Halfway House, 1685, Midrand;  Bytes Business Park, Block
B, 241 Third Road, Halfway Gardens, Midrand, South Africa 

E-Mail Disclaimer: http://www.altron.co.za/email.asp
<blocked::blocked::http://www.altron.co.za/email.asp>  or phone: (+27)
(11) 205-7000

 


The information contained in this message and or attachments 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
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From CORTIZ at broward.org  Thu Sep 14 11:34:47 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Thu, 14 Sep 2006 11:34:47 -0400
Subject: [cisco-voip] Unity (Unified Messaging) Importing users
	throughActive Directory
In-Reply-To: <B42A9A0E2A14B943A3B7EA2FA7A0D886E9A452@tnl1mx1.telindus.intra>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A4753F@MREXVS01.bc.broward.cty>

What version of Unity are you running?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Richard van
Berkel
Sent: Thursday, September 14, 2006 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity (Unified Messaging) Importing users
throughActive Directory

 

 

Hi,

 

I have a problem importing users through active directory. The
permissions seem to be ok.

 

Exchange 2003 delegated control:

 

UnityInstall      Exchange  Full Administrator

UnityDirSvc       Exchange  Full Administrator

UnityMsgStoreSvc  Exchange  Full Administrator

 

Active directory Unity Accounts have the following rights:

*         Installation Account:   UnityInstall      Domain Admin      +
Local Administrator

*         Administration:         UnityAdmin        Domain User       +
Local Administrator 

*         Directory Services:     UnityDirSvc       Domain Admin      +
Local Administrator

*         Unity Message Store:    UnityMsgStoreSvc  Domain Admin      +
Local Administrator

 

Problem is that whenever I import a user,I get an error message.

The user is in the subscriber list of the unity server, but there is no
way of selecting this user.

I can find the user by doing a search on extension.

When it finds the users there is no Surname or Firstname.

 

The only thing I can do is deleting it with Bulk delete.

 

Any ideas?

 

 

Richard van Berkel

Senior Customer Service

Profile: MCSE-2003, Microsoft Exchange 2003, Cisco Certified Voip
Specialist

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

Telindus B.V.

Savannahweg 19 

3542 AW Utrecht

Netherlands

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

mailto: Richard.vanBerkel at telindus.nl
<mailto:mailto:%20Richard.vanBerkel at telindus.nl> 

tel:         +31 302 477 515 

fax:       +31 302 477 711 

mobile: +31 628 903 277

 

Telindus. Change things your way.

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

	
	 	 
 

 

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

33rd International Telindus Symposium

Thursday, 26 October 2006, Tour & Taxis, Brussels, Belgium 

Change your vision & see how ICT changes your business.

The one-day ICT event for business professionals!

 

Programme and free registration > http://www.telindussymposium.com

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From Richard.vanBerkel at telindus.nl  Thu Sep 14 11:47:13 2006
From: Richard.vanBerkel at telindus.nl (Richard van Berkel)
Date: Thu, 14 Sep 2006 17:47:13 +0200
Subject: [cisco-voip] Unity (Unified Messaging) Importing users
	throughActive Directory
Message-ID: <B42A9A0E2A14B943A3B7EA2FA7A0D886E9A45B@tnl1mx1.telindus.intra>

Version 4.0.5 

 

Groeten Richard,

Mailto:Richard.vanBerkel at Telindus.nl
<mailto:Richard.vanBerkel at Telindus.nl> 

  

<skype:richardvanberkel?call> 

<http://www.skype.com/go/download>  

 

 

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: donderdag 14 september 2006 17:35
To: Richard van Berkel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity (Unified Messaging) Importing users
throughActive Directory

 

What version of Unity are you running?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Richard van
Berkel
Sent: Thursday, September 14, 2006 11:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity (Unified Messaging) Importing users
throughActive Directory

 

 

Hi,

 

I have a problem importing users through active directory. The
permissions seem to be ok.

 

Exchange 2003 delegated control:

 

UnityInstall      Exchange  Full Administrator

UnityDirSvc       Exchange  Full Administrator

UnityMsgStoreSvc  Exchange  Full Administrator

 

Active directory Unity Accounts have the following rights:

*         Installation Account:   UnityInstall      Domain Admin      +
Local Administrator

*         Administration:         UnityAdmin        Domain User       +
Local Administrator 

*         Directory Services:     UnityDirSvc       Domain Admin      +
Local Administrator

*         Unity Message Store:    UnityMsgStoreSvc  Domain Admin      +
Local Administrator

 

Problem is that whenever I import a user,I get an error message.

The user is in the subscriber list of the unity server, but there is no
way of selecting this user.

I can find the user by doing a search on extension.

When it finds the users there is no Surname or Firstname.

 

The only thing I can do is deleting it with Bulk delete.

 

Any ideas?

 

 

Richard van Berkel

Senior Customer Service

Profile: MCSE-2003, Microsoft Exchange 2003, Cisco Certified Voip
Specialist

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

Telindus B.V.

Savannahweg 19 

3542 AW Utrecht

Netherlands

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

mailto: Richard.vanBerkel at telindus.nl
<mailto:mailto:%20Richard.vanBerkel at telindus.nl> 

tel:         +31 302 477 515 

fax:       +31 302 477 711 

mobile: +31 628 903 277

 

Telindus. Change things your way.

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

	
	 	 
 

 

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

33rd International Telindus Symposium

Thursday, 26 October 2006, Tour & Taxis, Brussels, Belgium 

Change your vision & see how ICT changes your business.

The one-day ICT event for business professionals!

 

Programme and free registration > http://www.telindussymposium.com

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From tim.reimers at asheville.k12.nc.us  Thu Sep 14 13:31:34 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Thu, 14 Sep 2006 13:31:34 -0400
Subject: [cisco-voip] QRT Viewer problem...
Message-ID: <050944EE48FF384F9C663F7048C47DA704CD32CA@webmail.asheville.k12.nc.us>

Hi everyone-

I've configured the QRT button on a few phones-- have done some test
'pushes' of that button.

I've restarted the "Cisco Extended Functions" service on both CCMs

When I go into the QRT Viewer I get a popup screen that says:
"Document empty or invalid" "Could not do Problem Reporting" 

It's been a day now since I did my first QRT button push, so I'm sure
there's some sort of data in there (or I assume there is)

All other CDR seems to work fine, so I know the system is recording data
about calls from phones correctly already.

(I successfully enabled Malicious Caller functionality in Device Pools,
and that worked fine)

So I don't think the problem is at the Extended Functions layer, nor is
it perhaps the phones-
It's something about the logfile not getting created when QRT events
arrive, and then there's no data for the report viewer to display

Thanks, Tim


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From SPackett at fenwick.com  Thu Sep 14 13:50:16 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu, 14 Sep 2006 10:50:16 -0700
Subject: [cisco-voip] IP Communicator - AutoUpdate Error
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD7628@SVCEXCH01.firm.fenwick.llp>

We did our first install of IP Communicator v2.0.1.1, but whenever we
open the application, we get a pop-up message that says "The following
AutoUpdate path is not supported:  2.0.1.1 -> 1.1.3.0"  You then click
OK and you're able to work in IP Communicator just fine.

I asked a similar question a few weeks ago as my device defaults shows
"CIPC-1-1-3-0", because Cisco does not have the sccp file that installs
on the TFTP server.  Has anyone run into this?  Does anyone have the
sccp file that you install on the CallManager?

Thanks in advance,
Stu "see you at CIPTUG 2006" Packett
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From BDKusa at sgh.com  Thu Sep 14 14:03:47 2006
From: BDKusa at sgh.com (Ben D. Kusa)
Date: Thu, 14 Sep 2006 14:03:47 -0400
Subject: [cisco-voip] IP Communicator - AutoUpdate Error
In-Reply-To: <01C8F21F30DE9740B53FD2EF05C3A8FC01CD7628@SVCEXCH01.firm.fenwick.llp>
Message-ID: <8A9E98B05F39B14E82E546ECF9A8196901422089@exchange.sgh.com>

I got around that by blanking out the device default for IP
Communicator, that way it does not try an auto-update.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, September 14, 2006 1:50 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IP Communicator - AutoUpdate Error

 

We did our first install of IP Communicator v2.0.1.1, but whenever we
open the application, we get a pop-up message that says "The following
AutoUpdate path is not supported:  2.0.1.1 -> 1.1.3.0"  You then click
OK and you're able to work in IP Communicator just fine.

I asked a similar question a few weeks ago as my device defaults shows
"CIPC-1-1-3-0", because Cisco does not have the sccp file that installs
on the TFTP server.  Has anyone run into this?  Does anyone have the
sccp file that you install on the CallManager?

Thanks in advance, 
Stu "see you at CIPTUG 2006" Packett 

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From it at cimgroup.com  Thu Sep 14 15:08:23 2006
From: it at cimgroup.com (IT)
Date: Thu, 14 Sep 2006 12:08:23 -0700
Subject: [cisco-voip] vulnerable gateway?
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC014519CF@CSEXCH01.charles-stanley.co.uk>
Message-ID: <B487D4C1FE6D04439061018C6EBC109B03DC4573@exchange.cimgroup.com>

This was the problem. Kind of silly, and TAC had no idea.  Glad it was
quick and relatively painless. Now we are more secure...

http://www.cisco.com/en/US/products/products_security_advisory09186a0080
37cd59.shtml


-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Thursday, September 14, 2006 1:30 AM
To: Erick Bergquist; IT; Voll, Scott; puckcisco at cumhur.com;
cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

What partitions does the Voicemail CSS need to access, we appear to have
spurious partitions added to this CSS and I'm not sure they need to be
there?

-----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, September 13, 2006 4:03 AM
To: IT; Voll, Scott; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

You can limit this with the restriction tables in unity and ultimately
with the CSS set on the Call Manager VM Port configuration. 

----- Original Message ----
From: IT <it at cimgroup.com>
To: "Voll, Scott" <Scott.Voll at wesd.org>; IT <it at cimgroup.com>;
puckcisco at cumhur.com; cisco-voip at puck.nether.net
Sent: Tuesday, September 12, 2006 5:42:04 PM
Subject: Re: [cisco-voip] vulnerable gateway?

But where in unity is someone able to route their call to any arbitrary
phone number?

-----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, September 12, 2006 3:37 PM
To: IT; puckcisco at cumhur.com; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

I would agree with TAC per your CDR of CiscoUM-VI1.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Tuesday, September 12, 2006 3:29 PM
To: puckcisco at cumhur.com; IT; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] vulnerable gateway?

Actually, I tried both UDP and TCP.
Would it still show up under a portscan? TAC seems to think they came in
through voicemail...

-----Original Message-----
From: cumbur [mailto:zeus at cumhur.com] On Behalf Of puckcisco at cumhur.com
Sent: Tuesday, September 12, 2006 3:19 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] vulnerable gateway?

Dear Avidan,

H323 use TCP 1720 (not udp)  port for call initiation also don't forget
to
block SIP ports TCP/UDP 5060.

Regards.
Cumhur

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, September 13, 2006 12:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] vulnerable gateway?

I just got a call from my long distance provider that someone has been
using my PRI for many international calls. I check my CDR database
tables, and it appears that calls have been coming from one of my branch
office 2801's. But, in the CDR table, the origDeviceName alternates
between the name of the gateway and CiscoUM-VI1.
I ran a port scan against the router, and found that h.323 and callbook
ports were open to the public. I shutdown the interface that had those
ports open, because when I tried to do a "access-list 100 deny udp any
any eq 1720" it still shows as open on the portscan.

How can I secure/lock H.323 on these branch devices?
How did someone utilize my gateway to make these calls?
How can I avoid this in the future?

I guess I should have made sure that the consulting group that set up
these gateways in the first place locked them down, but hindsight is
20/20.

Thanks,
Avidan

_______________________________________________
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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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 lelio at uoguelph.ca  Thu Sep 14 15:23:21 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 14 Sep 2006 15:23:21 -0400
Subject: [cisco-voip] SRST and rebooted router and phones
Message-ID: <00c601c6d833$3da14a30$47196883@cfs.uoguelph.ca>

Can someone remind me what happens when SRST is enabled and the phones reboot? I'm pretty sure if the router reboots SRST breaks, but I can't remember if it breaks when the phone reboots/looses power.

I'm trying to spec out uptime requirements and I feel comfortable asking them source a UPS that will keep the router up for hours (if not a day), but would rather leave the UPS runtime phone up to them and their budget.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
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From omar.parihuana at gmail.com  Thu Sep 14 17:18:43 2006
From: omar.parihuana at gmail.com (omar parihuana)
Date: Thu, 14 Sep 2006 16:18:43 -0500
Subject: [cisco-voip] 2801 for Two E1 and full voice channels
Message-ID: <834c50110609141418g744ccfd4ydcb981d98940b905@mail.gmail.com>

Hi Folks,

We're going to purchase a Cisco 28xx, we need 2E1 with 60 voice
channels enable. I think that an Cisco 2801 with VWIC-2MFT-E1 and
PVDM2-64 will be enough. I don't sure if the VWIC will work on HWIC
slot in 2801 or I need purchase an NM-HDV-2E1-60 (and one 2811- the
price difference is high) or maybe 2801 isn't enough. Any suggest will
be appreciated.

Thanks!

-- 
Omar E.P.T
-----------------
Certified Networking Professionals make better Connections!

From bob at smalltime.com  Thu Sep 14 17:18:35 2006
From: bob at smalltime.com (Robert Kulagowski)
Date: Thu, 14 Sep 2006 16:18:35 -0500
Subject: [cisco-voip] 7936 randomly not generating certain portions of the
	HTML admin page
Message-ID: <4509C72B.8030506@smalltime.com>

When I use the admin web page of the 7936, I get all sorts of anomalies, 
like occasionally the top page of the page will say "The page cannot be 
displayed" rather than the listing of the firmware version, etc.

The left-side icons also usually sometimes come up, sometimes not.  If I 
do a shift-refresh in IE, I'll usually get some of the icons to fill in, 
sometimes not.

Is this a known issue?  Firmware 3.3.11 and .12.

From David.Fielder at us.didata.com  Thu Sep 14 17:52:10 2006
From: David.Fielder at us.didata.com (David Fielder (US))
Date: Thu, 14 Sep 2006 17:52:10 -0400
Subject: [cisco-voip] Converting from H.323 to MGCP
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032804AB@USNAEXCH.na.didata.local>

I know this is a long message but any help would be appreciated.

 

I am experiencing a strange problem with a gateway conversion I am
doing. Attached are the debugs and configs. My client is running H.323
voice gateways and we are trying to switch them over to MGCP. So far I
have been only partially successfully at getting there Main gateway over
to MGCP. This is a 3745 with two NM-HDV2 2T1 so there are four PRI's
connected to this router with configurations as follows:

 

1/0 - configured but not functional

 

1/1 - PRI to Nortel PBX 

            Switch Type=Primary-ni

2/0 - Second PRI to Nortel PBX

            Switch Type=Primary-ni

2/1 - PRI to the PSTN (Bellsouth)

            Switch Type=Primary-qsig

 

The existing gateway configuration on the CallManager is configured to
use the following Plan and Types for H.323

 

Called Party Number Type = National

Called Numbering Plan = National Standard

Calling Party Number Type = National

Calling Numbering Plan = ISDN  

 

I created a new Gateway in CallManager for MGCP which mirrored the H.323
for CSS, DP, and MRGL...etc. It also had the same numbering Plan and
Type for each port configured. I created three new Route Groups one for
connection to the PSTN and one each for the connections to the PBX. I
created two new Route Lists one for the port to the PSTN and one for the
ports to the PBX. 

 

Below is the configuration I used on the IOS gateway:

 

ccm-manager fallback-mgcp

ccm-manager redundant-host 172.17.1.20

ccm-manager mgcp

!

controller T1 1/1

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

controller T1 2/0

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

controller T1 2/1

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

interface Serial1/1:23

 no ip address

 no logging event link-status

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

interface Serial2/0:23

 no ip address

 no logging event link-status

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

interface Serial2/1:23

 no ip address

 no logging event link-status

 isdn switch-type primary-qsig

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

mgcp

mgcp call-agent 172.17.1.21 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

!

mgcp profile default

!

dial-peer voice 1 pots

 application mgcpapp

 port 2/1:23

!

dial-peer voice 2 pots

 application mgcpapp

 port 1/1:23

!

dial-peer voice 3 pots

 application mgcpapp

 port 2/0:23

 

Once I completed the changes and pointed my Route Patterns to the
appropriate Route List. I was unable to make any four digit or external
calls out or in. Every time I would try to make a call out to the PSTN.
I would get the following message in the Q931 debugs:

 

  Cause i = 0x829C - Invalid number format (incomplete number)

Sep  7 02:21:06.914: ISDN Se2/1:23 Q931: TX -> SETUP pd = 8  callref =
0x0006

        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 = 0x2881, '4879'

                Plan:National, Type:National

        Called Party Number i = 0xA1, '3154850'

                Plan:ISDN, Type:National

Sep  7 02:21:06.982: ISDN Se2/1:23 Q931: RX <- STATUS pd = 8  callref =
0x8006

        Cause i = 0x82E46C - Invalid information element contents

        Call State i = 0x01

Sep  7 02:21:07.030: ISDN Se2/1:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x

8006

 

I also captured a debug of successful calls when using H.323 and this is
the result of that debug:

 

Sep  7 02:01:25.344: ISDN Se1/0:23 Q931: pak_private_number: Invalid
type/plan 0

x2 0x8 may be overriden; sw-type 22

Sep  7 02:01:25.348: ISDN Se1/0:23 **ERROR**: CC_CHAN_GetIdleDMSChan:
Interface

de-Activated

Sep  7 02:01:25.348: ISDN Se1/0:23 **ERROR**: CCPMSG_OutCall: fails with
cause 0

x22

Sep  7 02:01:25.348: ISDN Se2/1:23 Q931: pak_private_number: Invalid
type/plan 0

x2 0x8 may be overriden; sw-type 22

Sep  7 02:01:25.352: ISDN Se2/1:23 Q931: Applying typeplan for sw-type
0x16 is 0

x0 0x0, Called num 6695129

Sep  7 02:01:25.352: ISDN Se2/1:23 Q931: TX -> SETUP pd = 8  callref =
0x02A5

        Bearer Capability i = 0x8090A2

                Standard = CCITT

                Transer Capability = Speech

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0xA98397

                Exclusive, Channel 23

        Calling Party Number i = 0x2181, '4879'

                Plan:ISDN, Type:National

        Called Party Number i = 0x80, '6695129'

                Plan:Unknown, Type:Unknown

Sep  7 02:01:25.780: ISDN Se2/1:23 Q931: RX <- CALL_PROC pd = 8  callref
= 0x82A

5

        Channel ID i = 0xA98397

                Exclusive, Channel 23

Sep  7 02:01:26.632: ISDN Se2/1:23 Q931: RX <- ALERTING pd = 8  callref
= 0x82A5

 

        Progress Ind i = 0x8288 - In-band info or appropriate now
available

Sep  7 02:01:33.878: ISDN Se2/1:23 Q931: RX <- CONNECT pd = 8  callref =
0x82A5

Sep  7 02:01:33.882: ISDN Se2/1:23 Q931: TX -> CONNECT_ACK pd = 8
callref = 0x0

2A5

Sep  7 02:01:36.782: ISDN Se2/1:23 Q931: RX <- DISCONNECT pd = 8
callref = 0x82

A5

        Cause i = 0x8290 - Normal call clearing

 

I noticed that even though the CallManger config for the gateway had the
called party plan and type as National and ISDN, they were showing in
the debug as Unknown. So I changed the Plan and Type for the MGCP
Gateway port 2/1 to match what I was seeing in the debugs for H.323 and
I was able to make calls out to the PSTN. That was working great.
However, I was still unable to make any calls to the PBX on the other
two ports. I changed the plan and type and I was then able to make calls
using my IP Communicator on my PC because I am making the changes
remotely but the people in the office were still unable to make any 4
digit calls. They were able to make calls to the PSTN. 4digit calls
coming in would make it to the destination but the person receiving the
calls would be unable to answer them. They were able to do a call
forward all to a cell phone and get the calls. I captured another Q931
debug for the four digit calls and was getting this information:

 

Sep 14 12:05:16.228: ISDN Se1/1:23 Q931: TX -> SETUP pd = 8  callref =
0x0013

        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 = '4624'

        Calling Party Number i = 0x2181, '4624'

                Plan:ISDN, Type:National

        Called Party Number i = 0x80, '3411'

                Plan:Unknown, Type:Unknown

Sep 14 12:05:16.276: ISDN Se1/1:23 Q931: RX <- STATUS pd = 8  callref =
0x8013

        Cause i = 0x81E3 - Information element not implemented

        Call State i = 0x06

Sep 14 12:05:16.280: ISDN Se1/1:23 Q931: TX -> DISCONNECT pd = 8
callref = 0x00

13

        Cause i = 0x80E2 - Message not compatible with call state or not
implement

 

Sep 14 12:05:16.324: ISDN Se1/1:23 Q931: RX <- RELEASE pd = 8  callref =
0x8013

        Cause i = 0x81E4 - Invalid information element contents

Sep 14 12:05:16.332: ISDN Se1/1:23 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x

0013

 

I have rolled them back to H.323 until I can resolve the issue. There
are six other gateways to convert and I don't want to run into the
problem there because they are not close by if I needed to go onsite for
any reason. I would appreciate any insight you guys might have on this
or any other troubleshooting steps you may recommend. I have verified
that the endpoints are up and the gateways are registered. I have
attached the full configs, debugs and sh vers for the gateway. I have
also opened a TAC case and they have been unhelpful so far. They did say
that the version of code was unsupported. I plan to upgrade the code
this weekend. 

 

The version of code is 12.3(8)T3 and the CallManager is 4.1(3)sr3

 

 

 

David Fielder

Senior Engineer

Dimension Data

Office: (336) 510-9431

Mobile: (336) 669-5129

david.fielder at us.didata.com

 




-----------------------------------------
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From Scott.Voll at wesd.org  Thu Sep 14 17:55:02 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu, 14 Sep 2006 14:55:02 -0700
Subject: [cisco-voip] Converting from H.323 to MGCP
Message-ID: <407055A92CECCB499C922A2D35FC19A6066C87A1@Apollo.wesd.org>

Are you sure Bellsouth is qsig?  I thought qsig was for PBX type
connection and ni was more of a telco one?  Am I out to lunch?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of David Fielder
(US)
Sent: Thursday, September 14, 2006 2:52 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Converting from H.323 to MGCP

 

I know this is a long message but any help would be appreciated.

 

I am experiencing a strange problem with a gateway conversion I am
doing. Attached are the debugs and configs. My client is running H.323
voice gateways and we are trying to switch them over to MGCP. So far I
have been only partially successfully at getting there Main gateway over
to MGCP. This is a 3745 with two NM-HDV2 2T1 so there are four PRI's
connected to this router with configurations as follows:

 

1/0 - configured but not functional

 

1/1 - PRI to Nortel PBX 

            Switch Type=Primary-ni

2/0 - Second PRI to Nortel PBX

            Switch Type=Primary-ni

2/1 - PRI to the PSTN (Bellsouth)

            Switch Type=Primary-qsig

 

The existing gateway configuration on the CallManager is configured to
use the following Plan and Types for H.323

 

Called Party Number Type = National

Called Numbering Plan = National Standard

Calling Party Number Type = National

Calling Numbering Plan = ISDN  

 

I created a new Gateway in CallManager for MGCP which mirrored the H.323
for CSS, DP, and MRGL...etc. It also had the same numbering Plan and
Type for each port configured. I created three new Route Groups one for
connection to the PSTN and one each for the connections to the PBX. I
created two new Route Lists one for the port to the PSTN and one for the
ports to the PBX. 

 

Below is the configuration I used on the IOS gateway:

 

ccm-manager fallback-mgcp

ccm-manager redundant-host 172.17.1.20

ccm-manager mgcp

!

controller T1 1/1

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

controller T1 2/0

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

controller T1 2/1

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

interface Serial1/1:23

 no ip address

 no logging event link-status

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

interface Serial2/0:23

 no ip address

 no logging event link-status

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

interface Serial2/1:23

 no ip address

 no logging event link-status

 isdn switch-type primary-qsig

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

mgcp

mgcp call-agent 172.17.1.21 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

!

mgcp profile default

!

dial-peer voice 1 pots

 application mgcpapp

 port 2/1:23

!

dial-peer voice 2 pots

 application mgcpapp

 port 1/1:23

!

dial-peer voice 3 pots

 application mgcpapp

 port 2/0:23

 

Once I completed the changes and pointed my Route Patterns to the
appropriate Route List. I was unable to make any four digit or external
calls out or in. Every time I would try to make a call out to the PSTN.
I would get the following message in the Q931 debugs:

 

  Cause i = 0x829C - Invalid number format (incomplete number)

Sep  7 02:21:06.914: ISDN Se2/1:23 Q931: TX -> SETUP pd = 8  callref =
0x0006

        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 = 0x2881, '4879'

                Plan:National, Type:National

        Called Party Number i = 0xA1, '3154850'

                Plan:ISDN, Type:National

Sep  7 02:21:06.982: ISDN Se2/1:23 Q931: RX <- STATUS pd = 8  callref =
0x8006

        Cause i = 0x82E46C - Invalid information element contents

        Call State i = 0x01

Sep  7 02:21:07.030: ISDN Se2/1:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x

8006

 

I also captured a debug of successful calls when using H.323 and this is
the result of that debug:

 

Sep  7 02:01:25.344: ISDN Se1/0:23 Q931: pak_private_number: Invalid
type/plan 0

x2 0x8 may be overriden; sw-type 22

Sep  7 02:01:25.348: ISDN Se1/0:23 **ERROR**: CC_CHAN_GetIdleDMSChan:
Interface

de-Activated

Sep  7 02:01:25.348: ISDN Se1/0:23 **ERROR**: CCPMSG_OutCall: fails with
cause 0

x22

Sep  7 02:01:25.348: ISDN Se2/1:23 Q931: pak_private_number: Invalid
type/plan 0

x2 0x8 may be overriden; sw-type 22

Sep  7 02:01:25.352: ISDN Se2/1:23 Q931: Applying typeplan for sw-type
0x16 is 0

x0 0x0, Called num 6695129

Sep  7 02:01:25.352: ISDN Se2/1:23 Q931: TX -> SETUP pd = 8  callref =
0x02A5

        Bearer Capability i = 0x8090A2

                Standard = CCITT

                Transer Capability = Speech

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0xA98397

                Exclusive, Channel 23

        Calling Party Number i = 0x2181, '4879'

                Plan:ISDN, Type:National

        Called Party Number i = 0x80, '6695129'

                Plan:Unknown, Type:Unknown

Sep  7 02:01:25.780: ISDN Se2/1:23 Q931: RX <- CALL_PROC pd = 8  callref
= 0x82A

5

        Channel ID i = 0xA98397

                Exclusive, Channel 23

Sep  7 02:01:26.632: ISDN Se2/1:23 Q931: RX <- ALERTING pd = 8  callref
= 0x82A5

 

        Progress Ind i = 0x8288 - In-band info or appropriate now
available

Sep  7 02:01:33.878: ISDN Se2/1:23 Q931: RX <- CONNECT pd = 8  callref =
0x82A5

Sep  7 02:01:33.882: ISDN Se2/1:23 Q931: TX -> CONNECT_ACK pd = 8
callref = 0x0

2A5

Sep  7 02:01:36.782: ISDN Se2/1:23 Q931: RX <- DISCONNECT pd = 8
callref = 0x82

A5

        Cause i = 0x8290 - Normal call clearing

 

I noticed that even though the CallManger config for the gateway had the
called party plan and type as National and ISDN, they were showing in
the debug as Unknown. So I changed the Plan and Type for the MGCP
Gateway port 2/1 to match what I was seeing in the debugs for H.323 and
I was able to make calls out to the PSTN. That was working great.
However, I was still unable to make any calls to the PBX on the other
two ports. I changed the plan and type and I was then able to make calls
using my IP Communicator on my PC because I am making the changes
remotely but the people in the office were still unable to make any 4
digit calls. They were able to make calls to the PSTN. 4digit calls
coming in would make it to the destination but the person receiving the
calls would be unable to answer them. They were able to do a call
forward all to a cell phone and get the calls. I captured another Q931
debug for the four digit calls and was getting this information:

 

Sep 14 12:05:16.228: ISDN Se1/1:23 Q931: TX -> SETUP pd = 8  callref =
0x0013

        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 = '4624'

        Calling Party Number i = 0x2181, '4624'

                Plan:ISDN, Type:National

        Called Party Number i = 0x80, '3411'

                Plan:Unknown, Type:Unknown

Sep 14 12:05:16.276: ISDN Se1/1:23 Q931: RX <- STATUS pd = 8  callref =
0x8013

        Cause i = 0x81E3 - Information element not implemented

        Call State i = 0x06

Sep 14 12:05:16.280: ISDN Se1/1:23 Q931: TX -> DISCONNECT pd = 8
callref = 0x00

13

        Cause i = 0x80E2 - Message not compatible with call state or not
implement

 

Sep 14 12:05:16.324: ISDN Se1/1:23 Q931: RX <- RELEASE pd = 8  callref =
0x8013

        Cause i = 0x81E4 - Invalid information element contents

Sep 14 12:05:16.332: ISDN Se1/1:23 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x

0013

 

I have rolled them back to H.323 until I can resolve the issue. There
are six other gateways to convert and I don't want to run into the
problem there because they are not close by if I needed to go onsite for
any reason. I would appreciate any insight you guys might have on this
or any other troubleshooting steps you may recommend. I have verified
that the endpoints are up and the gateways are registered. I have
attached the full configs, debugs and sh vers for the gateway. I have
also opened a TAC case and they have been unhelpful so far. They did say
that the version of code was unsupported. I plan to upgrade the code
this weekend. 

 

The version of code is 12.3(8)T3 and the CallManager is 4.1(3)sr3

 

 

 

David Fielder

Senior Engineer

Dimension Data

Office: (336) 510-9431

Mobile: (336) 669-5129

david.fielder at us.didata.com

 

________________________________



Disclaimer:

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


-------------- next part --------------
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From jason.aarons at us.didata.com  Thu Sep 14 18:13:43 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 14 Sep 2006 18:13:43 -0400
Subject: [cisco-voip] Converting from H.323 to MGCP
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032804E5@USNAEXCH.na.didata.local>

Lots of Nortel's can't do QSIG due to licensing/software limitation,
thus NI or Primary 5ess is used.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Thursday, September 14, 2006 5:55 PM
To: David Fielder (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Converting from H.323 to MGCP

 

Are you sure Bellsouth is qsig?  I thought qsig was for PBX type
connection and ni was more of a telco one?  Am I out to lunch?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of David Fielder
(US)
Sent: Thursday, September 14, 2006 2:52 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Converting from H.323 to MGCP

 

I know this is a long message but any help would be appreciated.

 

I am experiencing a strange problem with a gateway conversion I am
doing. Attached are the debugs and configs. My client is running H.323
voice gateways and we are trying to switch them over to MGCP. So far I
have been only partially successfully at getting there Main gateway over
to MGCP. This is a 3745 with two NM-HDV2 2T1 so there are four PRI's
connected to this router with configurations as follows:

 

1/0 - configured but not functional

 

1/1 - PRI to Nortel PBX 

            Switch Type=Primary-ni

2/0 - Second PRI to Nortel PBX

            Switch Type=Primary-ni

2/1 - PRI to the PSTN (Bellsouth)

            Switch Type=Primary-qsig

 

The existing gateway configuration on the CallManager is configured to
use the following Plan and Types for H.323

 

Called Party Number Type = National

Called Numbering Plan = National Standard

Calling Party Number Type = National

Calling Numbering Plan = ISDN  

 

I created a new Gateway in CallManager for MGCP which mirrored the H.323
for CSS, DP, and MRGL...etc. It also had the same numbering Plan and
Type for each port configured. I created three new Route Groups one for
connection to the PSTN and one each for the connections to the PBX. I
created two new Route Lists one for the port to the PSTN and one for the
ports to the PBX. 

 

Below is the configuration I used on the IOS gateway:

 

ccm-manager fallback-mgcp

ccm-manager redundant-host 172.17.1.20

ccm-manager mgcp

!

controller T1 1/1

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

controller T1 2/0

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

controller T1 2/1

 framing esf

 linecode b8zs

 pri-group timeslots 1-24 service mgcp

!

interface Serial1/1:23

 no ip address

 no logging event link-status

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

interface Serial2/0:23

 no ip address

 no logging event link-status

 isdn switch-type primary-ni

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

interface Serial2/1:23

 no ip address

 no logging event link-status

 isdn switch-type primary-qsig

 isdn incoming-voice voice

 isdn bind-l3 ccm-manager

 no cdp enable

!

mgcp

mgcp call-agent 172.17.1.21 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

!

mgcp profile default

!

dial-peer voice 1 pots

 application mgcpapp

 port 2/1:23

!

dial-peer voice 2 pots

 application mgcpapp

 port 1/1:23

!

dial-peer voice 3 pots

 application mgcpapp

 port 2/0:23

 

Once I completed the changes and pointed my Route Patterns to the
appropriate Route List. I was unable to make any four digit or external
calls out or in. Every time I would try to make a call out to the PSTN.
I would get the following message in the Q931 debugs:

 

  Cause i = 0x829C - Invalid number format (incomplete number)

Sep  7 02:21:06.914: ISDN Se2/1:23 Q931: TX -> SETUP pd = 8  callref =
0x0006

        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 = 0x2881, '4879'

                Plan:National, Type:National

        Called Party Number i = 0xA1, '3154850'

                Plan:ISDN, Type:National

Sep  7 02:21:06.982: ISDN Se2/1:23 Q931: RX <- STATUS pd = 8  callref =
0x8006

        Cause i = 0x82E46C - Invalid information element contents

        Call State i = 0x01

Sep  7 02:21:07.030: ISDN Se2/1:23 Q931: RX <- RELEASE_COMP pd = 8
callref = 0x

8006

 

I also captured a debug of successful calls when using H.323 and this is
the result of that debug:

 

Sep  7 02:01:25.344: ISDN Se1/0:23 Q931: pak_private_number: Invalid
type/plan 0

x2 0x8 may be overriden; sw-type 22

Sep  7 02:01:25.348: ISDN Se1/0:23 **ERROR**: CC_CHAN_GetIdleDMSChan:
Interface

de-Activated

Sep  7 02:01:25.348: ISDN Se1/0:23 **ERROR**: CCPMSG_OutCall: fails with
cause 0

x22

Sep  7 02:01:25.348: ISDN Se2/1:23 Q931: pak_private_number: Invalid
type/plan 0

x2 0x8 may be overriden; sw-type 22

Sep  7 02:01:25.352: ISDN Se2/1:23 Q931: Applying typeplan for sw-type
0x16 is 0

x0 0x0, Called num 6695129

Sep  7 02:01:25.352: ISDN Se2/1:23 Q931: TX -> SETUP pd = 8  callref =
0x02A5

        Bearer Capability i = 0x8090A2

                Standard = CCITT

                Transer Capability = Speech

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0xA98397

                Exclusive, Channel 23

        Calling Party Number i = 0x2181, '4879'

                Plan:ISDN, Type:National

        Called Party Number i = 0x80, '6695129'

                Plan:Unknown, Type:Unknown

Sep  7 02:01:25.780: ISDN Se2/1:23 Q931: RX <- CALL_PROC pd = 8  callref
= 0x82A

5

        Channel ID i = 0xA98397

                Exclusive, Channel 23

Sep  7 02:01:26.632: ISDN Se2/1:23 Q931: RX <- ALERTING pd = 8  callref
= 0x82A5

 

        Progress Ind i = 0x8288 - In-band info or appropriate now
available

Sep  7 02:01:33.878: ISDN Se2/1:23 Q931: RX <- CONNECT pd = 8  callref =
0x82A5

Sep  7 02:01:33.882: ISDN Se2/1:23 Q931: TX -> CONNECT_ACK pd = 8
callref = 0x0

2A5

Sep  7 02:01:36.782: ISDN Se2/1:23 Q931: RX <- DISCONNECT pd = 8
callref = 0x82

A5

        Cause i = 0x8290 - Normal call clearing

 

I noticed that even though the CallManger config for the gateway had the
called party plan and type as National and ISDN, they were showing in
the debug as Unknown. So I changed the Plan and Type for the MGCP
Gateway port 2/1 to match what I was seeing in the debugs for H.323 and
I was able to make calls out to the PSTN. That was working great.
However, I was still unable to make any calls to the PBX on the other
two ports. I changed the plan and type and I was then able to make calls
using my IP Communicator on my PC because I am making the changes
remotely but the people in the office were still unable to make any 4
digit calls. They were able to make calls to the PSTN. 4digit calls
coming in would make it to the destination but the person receiving the
calls would be unable to answer them. They were able to do a call
forward all to a cell phone and get the calls. I captured another Q931
debug for the four digit calls and was getting this information:

 

Sep 14 12:05:16.228: ISDN Se1/1:23 Q931: TX -> SETUP pd = 8  callref =
0x0013

        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 = '4624'

        Calling Party Number i = 0x2181, '4624'

                Plan:ISDN, Type:National

        Called Party Number i = 0x80, '3411'

                Plan:Unknown, Type:Unknown

Sep 14 12:05:16.276: ISDN Se1/1:23 Q931: RX <- STATUS pd = 8  callref =
0x8013

        Cause i = 0x81E3 - Information element not implemented

        Call State i = 0x06

Sep 14 12:05:16.280: ISDN Se1/1:23 Q931: TX -> DISCONNECT pd = 8
callref = 0x00

13

        Cause i = 0x80E2 - Message not compatible with call state or not
implement

 

Sep 14 12:05:16.324: ISDN Se1/1:23 Q931: RX <- RELEASE pd = 8  callref =
0x8013

        Cause i = 0x81E4 - Invalid information element contents

Sep 14 12:05:16.332: ISDN Se1/1:23 Q931: TX -> RELEASE_COMP pd = 8
callref = 0x

0013

 

I have rolled them back to H.323 until I can resolve the issue. There
are six other gateways to convert and I don't want to run into the
problem there because they are not close by if I needed to go onsite for
any reason. I would appreciate any insight you guys might have on this
or any other troubleshooting steps you may recommend. I have verified
that the endpoints are up and the gateways are registered. I have
attached the full configs, debugs and sh vers for the gateway. I have
also opened a TAC case and they have been unhelpful so far. They did say
that the version of code was unsupported. I plan to upgrade the code
this weekend. 

 

The version of code is 12.3(8)T3 and the CallManager is 4.1(3)sr3

 

 

 

David Fielder

Senior Engineer

Dimension Data

Office: (336) 510-9431

Mobile: (336) 669-5129

david.fielder at us.didata.com

 

________________________________



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From swood.listserv at gmail.com  Thu Sep 14 19:04:00 2006
From: swood.listserv at gmail.com (Spencer Wood)
Date: Thu, 14 Sep 2006 19:04:00 -0400
Subject: [cisco-voip] Moving VM Accounts Between Unity Servers
Message-ID: <531f35b30609141604u52e7e0b8v72889a8fdf31455b@mail.gmail.com>

We have a couple of Unity Servers (All in the same AD Domain), and we are
trying to move mailboxes/subscribers from one Unity Server to Another.

I tried using the GSM tool (Which is pretty nice), but I'm not having any
luck with the moving part.  When I select the mailbox to move, the next
screen ask which server to move the mailbox to, and all I get is a blank
list.

I made sure that both servers have the "Stream Files" directory shared out,
and that the user I'm running the GSM tool from are admin's on both boxes..

Any thoughts???

Spencer
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From jryan.fields at gmail.com  Thu Sep 14 22:10:49 2006
From: jryan.fields at gmail.com (Ryan Fields)
Date: Fri, 15 Sep 2006 10:10:49 +0800
Subject: [cisco-voip] IPCC Express supervisor features
Message-ID: <c1fadd3a0609141910o6997a478g907770f6a7ae48d1@mail.gmail.com>

Hi All,

Is it possible to use supervisor features like agent monitoring, silent
monitoring, barge-in, intercept and coaching
from a remote location? The scenario would be having the whole IPCC,
Callmanager, IVR system and agents located
in Asia while the supervisor will monitor or perform call-control functions
in California.

Thanks,
Ryan
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From gpate at kenttech.com  Thu Sep 14 22:58:25 2006
From: gpate at kenttech.com (Gary L. Pate)
Date: Thu, 14 Sep 2006 22:58:25 -0400
Subject: [cisco-voip] IPCC silent monitoring
In-Reply-To: <6D8BEE91433474478D294A8867FFDC6EE3F8C3@WMAIN3.aurorapubliclibrary.org>
Message-ID: <000b01c6d872$d07a38a0$1ffb14ac@gpatelt>

I have a client using IPCC 4.0 enhanced version, and they are able to use
all the enhanced features except the monitoring function. As I understand
it, this version does not require you to span the ports, but their ports are
spanned anyway because they are using a third party recording device to
record all conversations. Anyway, barge in, intercept, etc. work, and when
they do the monitoring, the applications acts like its working, but they do
not hear anything coming out of the PC speakers even though its been
confirmed that the speakers do work. Is there anything special that the
workstation needs to monitor?

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From netfortius at gmail.com  Fri Sep 15 07:23:47 2006
From: netfortius at gmail.com (Netfortius)
Date: Fri, 15 Sep 2006 06:23:47 -0500
Subject: [cisco-voip] [On a tangent - Nortel & Unity] {Was: call quality
	issues between gateway and Unity}
In-Reply-To: <s509219f.076@comrcm01intgwia.mandtbank.com>
References: <s509219f.076@comrcm01intgwia.mandtbank.com>
Message-ID: <200609150623.47794.netfortius@gmail.com>

Steve,

I have a question related to this last paragraph (sorry for hi-jacking part of 
this thread): I am in the process of migrating - one location at a time - a 
Nortel-based (Meridian) community, to Cisco-based VoIP w/same infrastructure 
as the one mentioned by you (CM and Unity). One problem I am having is that I 
need to run TWO voice mail systems in parallel (Call Pilot on Nortel, and 
Unity on Cisco), because I could not find a way to signal the existance of a 
message in Unity, to a Nortel-side phone, by turning on the message lights on 
the latter. How do you handle this?

Thanks,
Stefan

On Thursday 14 September 2006 08:31, STEVEN CASPER wrote:
> Thanks Wes,
> This is very interesting. So is the recommendation to add "mgcp playout
> adaptive 60 40 200" to all voice gateways that allow non IP phone users
> to access Unity? I have a lot of gateways that connect Nortel users to
> Unity via QSIG gateways. We occasionally get complaints of canned or
> robotic voice quality. We are running CCM 4.1.3sr1 with Unity 4.05.
>
> Steve
<snip>

From bob at smalltime.com  Fri Sep 15 07:22:28 2006
From: bob at smalltime.com (Robert Kulagowski)
Date: Fri, 15 Sep 2006 06:22:28 -0500
Subject: [cisco-voip] Managing DID's ?
In-Reply-To: <a1b07ea60609121414l7fda34aas1222a5c9dccfea58@mail.gmail.com>
References: <a1b07ea60609121414l7fda34aas1222a5c9dccfea58@mail.gmail.com>
Message-ID: <450A8CF4.3080307@smalltime.com>

> Right now all I can do is search say extension beginning with 145 and
> see if anything is availble between 1450 - 1459 and so on.  I suspect
> everyone has to deal with this same mess and I have to wonder if there
> is a better way that I am simply not aware of.

What I did was to use the BAT and insert dummy phones and encode the 
status in the description:  "Miami Available-xxx xxx xxxx", etc.  That 
way when I need a DID I can just do a search for phone where description 
contains available and see what I've got.  When a DID becomes free I 
change the description to put it back in the available pool.

From Matt.Slaga at us.didata.com  Fri Sep 15 08:28:38 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 15 Sep 2006 08:28:38 -0400
Subject: [cisco-voip] IPCC silent monitoring
Message-ID: <C1FE15183DA37645BC0633BC604E44F00328070A@USNAEXCH.na.didata.local>

Make sure the network card is capable of 802.1q packet tagging.  Also,
make sure the phone is configured (in CCM) for PC Voice VLAN Access.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gary L. Pate
Sent: Thursday, September 14, 2006 10:58 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC silent monitoring

 

I have a client using IPCC 4.0 enhanced version, and they are able to
use all the enhanced features except the monitoring function. As I
understand it, this version does not require you to span the ports, but
their ports are spanned anyway because they are using a third party
recording device to record all conversations. Anyway, barge in,
intercept, etc. work, and when they do the monitoring, the applications
acts like its working, but they do not hear anything coming out of the
PC speakers even though its been confirmed that the speakers do work. Is
there anything special that the workstation needs to monitor?




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

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
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this communication in error and that any use or reproduction of
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From ealeatherman at gmail.com  Fri Sep 15 08:34:55 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 15 Sep 2006 08:34:55 -0400
Subject: [cisco-voip] IPCC silent monitoring
In-Reply-To: <000b01c6d872$d07a38a0$1ffb14ac@gpatelt>
References: <6D8BEE91433474478D294A8867FFDC6EE3F8C3@WMAIN3.aurorapubliclibrary.org>
	<000b01c6d872$d07a38a0$1ffb14ac@gpatelt>
Message-ID: <94a1afde0609150534h10e58aa3r75bbda462d20b408@mail.gmail.com>

Are your agent IP Phones setup to span the voice VLAN to the pc port? the
agent desktop app is what sends the audio that you are monitoring off to
either your desktop or the IPCC server (cant remember which), and it has to
have the audio traffic from the phone to be able to do it. sounds like
everything is working for you but the RTP stream is getting stopped
somewhere, like a one-way audio problem with normal ip phone service.

On 9/14/06, Gary L. Pate <gpate at kenttech.com> wrote:
>
>  I have a client using IPCC 4.0 enhanced version, and they are able to use
> all the enhanced features except the monitoring function. As I understand
> it, this version does not require you to span the ports, but their ports are
> spanned anyway because they are using a third party recording device to
> record all conversations. Anyway, barge in, intercept, etc. work, and when
> they do the monitoring, the applications acts like its working, but they do
> not hear anything coming out of the PC speakers even though its been
> confirmed that the speakers do work. Is there anything special that the
> workstation needs to monitor?
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From wsisk at cisco.com  Fri Sep 15 09:20:28 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 15 Sep 2006 09:20:28 -0400
Subject: [cisco-voip] IPCC silent monitoring
In-Reply-To: <000b01c6d872$d07a38a0$1ffb14ac@gpatelt>
References: <000b01c6d872$d07a38a0$1ffb14ac@gpatelt>
Message-ID: <450AA89C.4060204@cisco.com>

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From MLinsemier at apcapital.com  Fri Sep 15 09:28:26 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Fri, 15 Sep 2006 09:28:26 -0400
Subject: [cisco-voip] IPCC silent monitoring
In-Reply-To: <450AA89C.4060204@cisco.com>
Message-ID: <AB0A35F2486F81468971AA68558BE71904CEF8A4@smieln01winm01.micoanet.com>

Wes,

 

I thought with IPCC Express 4.0 there was no need for SPAN or Remote
SPAN ports anymore for monitoring?  Has something changed?

 

-Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Friday, September 15, 2006 9:20 AM
To: GPate at kenttech.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC silent monitoring

 

Gary,

Make sure you span the voice vlan of the agent phones to the designated
NIC of the monitoring server.  take a packet capture at that interface
to make sure you are seeing all the vlan traffic.  Then make sure your
server NIC correctly passes dot1q tagged packets up to the OS.  You can
verify both in 1 step by attempting a packet capture with network
monitor or ethereal on the Monitor server.

A good starting point:
http://www.cisco.com/en/US/products/sw/custcosw/ps1001/products_tech_not
e09186a008010e6ba.shtml

/Wes

Gary L. Pate wrote: 

I have a client using IPCC 4.0 enhanced version, and they are able to
use all the enhanced features except the monitoring function. As I
understand it, this version does not require you to span the ports, but
their ports are spanned anyway because they are using a third party
recording device to record all conversations. Anyway, barge in,
intercept, etc. work, and when they do the monitoring, the applications
acts like its working, but they do not hear anything coming out of the
PC speakers even though its been confirmed that the speakers do work. Is
there anything special that the workstation needs to monitor?

 



________________________________



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

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

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the intended recipient, any use, disclosure, copying or 
distribution of this communication is UNAUTHORIZED. Neither this 
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From comvox at optonline.net  Fri Sep 15 09:45:26 2006
From: comvox at optonline.net (comvox at optonline.net)
Date: Fri, 15 Sep 2006 13:45:26 +0000 (GMT)
Subject: [cisco-voip] Compare and Contrast
Message-ID: <e366864a59be.450aae76@optonline.net>

All,

I have a sales office that's going to be relocating.  At the moment, they're on an Option 11, 3.0.  There are other sales offices... mostly Nortel 3.0, where we're using VoIP trunking interoffice, and some softphones for remote users.
 
Management is considering installing a Cisco Call manager in the new facilities where the sales office will move to.  I'm not familiar with Cisco voice products. I'm the primary voice administrator across the offices, with local site admins. I'm not an installer. I can install/setup/etc line cards, etc. I do know BARS, Meridian Mail, and am pretty creative with interoffice routing, failover from VoIP to PSTN, etc..  Politics and economics are playing a large part in this consideration, and for the seemingly small say I have in the matter, I'd like at least to be able to intelligently put together a "pros and cons" of switching the system, since I'll inevitably be the guy that's called when something's not working properly.  Besides the obvious things like having to train the users on a new system, can anyone provide me with some technical comparisons between a Cisco Call Manager's capabilities and the capabilities of a similarly spec'ed Nortel system (BCMXXX)? 
 
Since the Option 11 is getting old, we're leaning towards a Nortel replacement should we decide against Cisco. 
 
Here are a couple of concerns:
1. How easily will the Call Manager network with the Meridians for VoIP trunking?  We're 3.0 on the Nortel side, so SIP is not in the picture.
2. How do the 2 manufacturers' multimedia/collaboration tools/products compare? 
3. How do the call center apps compare?  
4. How smart is it to put IP phones on all desktops when Disaster Recovery and Redundancy planning are just about the biggest issues right now in my department? 
 
At the moment, none of our offices is using Unified Messaging solutions. One office suddenly having this capability shouldn't be the reason it becomes necessary for all offices to have it. It's obvious that a business case should be made first for the solution, and then considerations for installations and rollouts come next,  but you know how it goes... 
 
Thanks in advance for all of your advice..
 
-- Chris 
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From Matt.Slaga at us.didata.com  Fri Sep 15 09:52:19 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 15 Sep 2006 09:52:19 -0400
Subject: [cisco-voip] IPCC silent monitoring
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032808D3@USNAEXCH.na.didata.local>

It now gives you two options, one for agent desktop unicast and the
other for the old fashioned span ports.  If you are using the IP phone
agent, you only have the SPAN option.  If you are using the agent
desktop, you have both options.

 

One of the requirements for the agent desktop is an 802.1q compliant
network card.  The phone is sending tagged packets to the PC and the PC
has to know what to do with them.  Almost all NICs in the past few years
have this functionality but I've run into a few cases where it doesn't.
In these cases, you have to use SPAN.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Linsemier,
Matthew
Sent: Friday, September 15, 2006 9:28 AM
To: Wes Sisk; GPate at kenttech.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC silent monitoring

 

Wes,

 

I thought with IPCC Express 4.0 there was no need for SPAN or Remote
SPAN ports anymore for monitoring?  Has something changed?

 

-Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Friday, September 15, 2006 9:20 AM
To: GPate at kenttech.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC silent monitoring

 

Gary,

Make sure you span the voice vlan of the agent phones to the designated
NIC of the monitoring server.  take a packet capture at that interface
to make sure you are seeing all the vlan traffic.  Then make sure your
server NIC correctly passes dot1q tagged packets up to the OS.  You can
verify both in 1 step by attempting a packet capture with network
monitor or ethereal on the Monitor server.

A good starting point:
http://www.cisco.com/en/US/products/sw/custcosw/ps1001/products_tech_not
e09186a008010e6ba.shtml

/Wes

Gary L. Pate wrote: 

I have a client using IPCC 4.0 enhanced version, and they are able to
use all the enhanced features except the monitoring function. As I
understand it, this version does not require you to span the ports, but
their ports are spanned anyway because they are using a third party
recording device to record all conversations. Anyway, barge in,
intercept, etc. work, and when they do the monitoring, the applications
acts like its working, but they do not hear anything coming out of the
PC speakers even though its been confirmed that the speakers do work. Is
there anything special that the workstation needs to monitor?

 
 
 



________________________________



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


CONFIDENTIALITY STATEMENT

This communication and any attachments are CONFIDENTIAL and may be
protected by one or more legal privileges. It is intended solely for the
use of the addressee identified above. If you are not the intended
recipient, any use, disclosure, copying or distribution of this
communication is UNAUTHORIZED. Neither this information block, the typed
name of the sender, nor anything else in this message is intended to
constitute an electronic signature unless a specific statement to the
contrary is included in this message. If you have received this
communication in error, please immediately contact me and delete this
communication from your computer. Thank you.

________________________________




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

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
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From akoul at dataways.gr  Fri Sep 15 09:56:30 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Fri, 15 Sep 2006 16:56:30 +0300
Subject: [cisco-voip] Can IPMA service work with no Partitions/CSS cinfigured
Message-ID: <450AB10E.1020009@dataways.gr>

Imagine a big installation where it was never desired to have partitions 
and CSS.
Can we now implement the IPMA omiting the part of the configuration of 
partitions/CSS. To leave these fields blank for all,
lines, managers, assistants, everyone and route points?

From jason.aarons at us.didata.com  Fri Sep 15 09:57:22 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 15 Sep 2006 09:57:22 -0400
Subject: [cisco-voip] IPCC silent monitoring
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032808E5@USNAEXCH.na.didata.local>

In IPCC Express 4.X there are two types of monitoring;

 

A.	SPAN Based (old school)
B.	Agent Monitoring (CAD acts as sniffer on desktops, new to 4.x)

 

If option B is being used, is the NIC a Intel 10/100 Pro or a 3COM 905?
I've had issues with these NICs not being able to sniff packets tagged.
There are CCO.

 

TAC has a extensive troubleshooting process for monitoring I haven't
seen on CCO anywhere.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Gary L. Pate
Sent: Thursday, September 14, 2006 10:58 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC silent monitoring

 

I have a client using IPCC 4.0 enhanced version, and they are able to
use all the enhanced features except the monitoring function. As I
understand it, this version does not require you to span the ports, but
their ports are spanned anyway because they are using a third party
recording device to record all conversations. Anyway, barge in,
intercept, etc. work, and when they do the monitoring, the applications
acts like its working, but they do not hear anything coming out of the
PC speakers even though its been confirmed that the speakers do work. Is
there anything special that the workstation needs to monitor?




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

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
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From Christian.Rupp at etavis.ch  Fri Sep 15 09:59:32 2006
From: Christian.Rupp at etavis.ch (Rupp Christian)
Date: Fri, 15 Sep 2006 15:59:32 +0200
Subject: [cisco-voip] to class internal number as external number
Message-ID: <30C88EC7EED69C4992E4CBCA1B7ADCF4010D6029@BASMXCL01.etavis.local>

Hello 
 
do anyone know how I can class an internal call as an external call?
the problem is, that I want to use the call forward busy external rule
for one special internal dn.
 
Thanks a lot for the ideas. 
 
Regards,
Christian
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From gpate at kenttech.com  Fri Sep 15 10:02:44 2006
From: gpate at kenttech.com (Gary L. Pate)
Date: Fri, 15 Sep 2006 10:02:44 -0400
Subject: [cisco-voip] IPCC silent monitoring
In-Reply-To: <94a1afde0609150534h10e58aa3r75bbda462d20b408@mail.gmail.com>
Message-ID: <002b01c6d8cf$9f3eca00$20fb14ac@gpatelt>

All of the ports are spanned, but as I understand it, version 4.0 does not
require span port when using CAD. Others have also responded with a
suggestion about the NIC card, so I'm going to look at that. I'll let
everyone know if that is the problem.

 

 

 

  _____  

From: Ed Leatherman [mailto:ealeatherman at gmail.com] 
Sent: Friday, September 15, 2006 8:35 AM
To: GPate at kenttech.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC silent monitoring

 

Are your agent IP Phones setup to span the voice VLAN to the pc port? the
agent desktop app is what sends the audio that you are monitoring off to
either your desktop or the IPCC server (cant remember which), and it has to
have the audio traffic from the phone to be able to do it. sounds like
everything is working for you but the RTP stream is getting stopped
somewhere, like a one-way audio problem with normal ip phone service. 

On 9/14/06, Gary L. Pate <gpate at kenttech.com> wrote:

I have a client using IPCC 4.0 enhanced version, and they are able to use
all the enhanced features except the monitoring function. As I understand
it, this version does not require you to span the ports, but their ports are
spanned anyway because they are using a third party recording device to
record all conversations. Anyway, barge in, intercept, etc. work, and when
they do the monitoring, the applications acts like its working, but they do
not hear anything coming out of the PC speakers even though its been
confirmed that the speakers do work. Is there anything special that the
workstation needs to monitor?


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






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

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From wsisk at cisco.com  Fri Sep 15 10:06:05 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 15 Sep 2006 10:06:05 -0400
Subject: [cisco-voip] IPCC silent monitoring
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0032808E5@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0032808E5@USNAEXCH.na.didata.local>
Message-ID: <450AB34D.4090100@cisco.com>

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From wsisk at cisco.com  Fri Sep 15 10:10:03 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 15 Sep 2006 10:10:03 -0400
Subject: [cisco-voip] to class internal number as external number
In-Reply-To: <30C88EC7EED69C4992E4CBCA1B7ADCF4010D6029@BASMXCL01.etavis.local>
References: <30C88EC7EED69C4992E4CBCA1B7ADCF4010D6029@BASMXCL01.etavis.local>
Message-ID: <450AB43B.4040004@cisco.com>

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From wsisk at cisco.com  Fri Sep 15 10:15:01 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 15 Sep 2006 10:15:01 -0400
Subject: [cisco-voip] IPCC silent monitoring
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0032808E5@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0032808E5@USNAEXCH.na.didata.local>
Message-ID: <450AB565.8070803@cisco.com>

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From nathan.a.reeves at gmail.com  Fri Sep 15 11:01:27 2006
From: nathan.a.reeves at gmail.com (Nathan Reeves)
Date: Fri, 15 Sep 2006 23:01:27 +0800
Subject: [cisco-voip] Any issues in utilising the 'mobile' field in the
	DCDirectory?
Message-ID: <62833d8a0609150801k5a22fe7fv2e0400eeae877828@mail.gmail.com>

Anyone know if cisco have issues in using the mobile phone number
field in the DCDirectory in CCM?  I'd be looking to run a sync of this
field from our AD domain to the DCDirectory and then allowing users to
search the mobile number field using a modified asp page.

Is anyone else using any of the non CCM fields in production?

Cheers

Nathan

From baryonyx5 at gmail.com  Fri Sep 15 12:04:03 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Fri, 15 Sep 2006 11:04:03 -0500
Subject: [cisco-voip] Increase volume on IP side of FXO call
Message-ID: <a749d0620609150904n5eed276bh9a4a639248114317@mail.gmail.com>

Hi

Is it possible to increase the call volume for the IP phone user on calls
through an FXO port?  If I'm not mistaken, the input gain and output
attenuation only affect the analog side of the call.

I have a customer with a  remote location where the volume on external calls
is very low and apparantly using the IP phone volume control doesn't help
enough.

Thanks

-- 
kris seraphine
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From tim.reimers at asheville.k12.nc.us  Fri Sep 15 12:54:32 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Fri, 15 Sep 2006 12:54:32 -0400
Subject: [cisco-voip] Question regarding AD integration and corporate
	directories...
Message-ID: <050944EE48FF384F9C663F7048C47DA704CD34D5@webmail.asheville.k12.nc.us>




Regarding the plugin for AD for the CCMs- mentioned here:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a0080094493.shtml

What does this accomplish for CCM users?

Is it a 'free' option for a 4.01 non-Unifed Messaging environment?

I'm trying to figure out if I should ask my vendor to install this as an
OEM standard option,
and what effect it may have on my existing users on CCM and users on
Unity and the Corporate Directory that now exists.

I don't quite understand the difference between having this plugin
installed and the AD changes made,
versus using the internal CCM users?

I can only find Cisco's brief discussions, that don't really give a
'features' list of what extra capabilities you have with this installed,
that you don't have without it...

Tim

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From Scott.Voll at wesd.org  Fri Sep 15 13:10:32 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 15 Sep 2006 10:10:32 -0700
Subject: [cisco-voip] Question regarding AD integration and
	corporatedirectories...
Message-ID: <407055A92CECCB499C922A2D35FC19A606933C3A@Apollo.wesd.org>

Really it's just so you don't have to use DC directory.  We have the
plugin because it's on less username / password our users have to know.

 

IE> if you give them access to https://CM/ccmuser they can use there
domain username and password rather then DCD username/password.

 

Or 

 

IPCCx agents can also use there Domain username/password.

 

Really, that's the only advantage I know of.

 

Scott

 

PS. No affect on Unity

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Friday, September 15, 2006 9:55 AM
To: Cisco Voip
Subject: [cisco-voip] Question regarding AD integration and
corporatedirectories...

 





Regarding the plugin for AD for the CCMs- mentioned here: 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a0080094493.shtml
<http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note
09186a0080094493.shtml>  

What does this accomplish for CCM users? 

Is it a 'free' option for a 4.01 non-Unifed Messaging environment? 

I'm trying to figure out if I should ask my vendor to install this as an
OEM standard option, 
and what effect it may have on my existing users on CCM and users on
Unity and the Corporate Directory that now exists. 

I don't quite understand the difference between having this plugin
installed and the AD changes made, 
versus using the internal CCM users? 

I can only find Cisco's brief discussions, that don't really give a
'features' list of what extra capabilities you have with this installed,
that you don't have without it...

Tim 

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From milay at cisco.com  Fri Sep 15 13:17:48 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Fri, 15 Sep 2006 10:17:48 -0700
Subject: [cisco-voip] Question regarding AD integration and
	corporatedirectories...
Message-ID: <413FABF5DCE22849AF86F442E472B78BC3B898@xmb-sjc-219.amer.cisco.com>

The AD plug-in allows you to not need to manage an additional account
for users. In addition you have the ability to search your corporate
directory from your IP phone with no additional xml scripting.
 
If you do decide to integrate with AD you will need to add any special
accounts utilized by CTI applications into AD, AC, CRS etc... Also you
will have to reassociate any devices to accounts, example would be if
you have IP Phones associated to user accounts.
 
Mike 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Friday, September 15, 2006 11:55 AM
To: Cisco Voip
Subject: [cisco-voip] Question regarding AD integration and
corporatedirectories...






Regarding the plugin for AD for the CCMs- mentioned here: 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a0080094493.shtml
<http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note
09186a0080094493.shtml>  

What does this accomplish for CCM users? 

Is it a 'free' option for a 4.01 non-Unifed Messaging environment? 

I'm trying to figure out if I should ask my vendor to install this as an
OEM standard option, 
and what effect it may have on my existing users on CCM and users on
Unity and the Corporate Directory that now exists. 

I don't quite understand the difference between having this plugin
installed and the AD changes made, 
versus using the internal CCM users? 

I can only find Cisco's brief discussions, that don't really give a
'features' list of what extra capabilities you have with this installed,
that you don't have without it...

Tim 

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From milay at cisco.com  Fri Sep 15 13:22:50 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Fri, 15 Sep 2006 10:22:50 -0700
Subject: [cisco-voip] Question regarding AD integration
	andcorporatedirectories...
Message-ID: <413FABF5DCE22849AF86F442E472B78BC3B89D@xmb-sjc-219.amer.cisco.com>

Almost forgot! AD has a limitation of the number of devices that can be
associated to a single user, ~750. So if you have a situation where you
need to associate multiple phones, or route points to a single user,
beware of this. This is a Microsoft limitation not a Cisco limitation.
 
Mike 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Lay
(milay)
Sent: Friday, September 15, 2006 12:18 PM
To: Tim Reimers; Cisco Voip
Subject: Re: [cisco-voip] Question regarding AD integration
andcorporatedirectories...


The AD plug-in allows you to not need to manage an additional account
for users. In addition you have the ability to search your corporate
directory from your IP phone with no additional xml scripting.
 
If you do decide to integrate with AD you will need to add any special
accounts utilized by CTI applications into AD, AC, CRS etc... Also you
will have to reassociate any devices to accounts, example would be if
you have IP Phones associated to user accounts.
 
Mike 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Friday, September 15, 2006 11:55 AM
To: Cisco Voip
Subject: [cisco-voip] Question regarding AD integration and
corporatedirectories...






Regarding the plugin for AD for the CCMs- mentioned here: 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a0080094493.shtml
<http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note
09186a0080094493.shtml>  

What does this accomplish for CCM users? 

Is it a 'free' option for a 4.01 non-Unifed Messaging environment? 

I'm trying to figure out if I should ask my vendor to install this as an
OEM standard option, 
and what effect it may have on my existing users on CCM and users on
Unity and the Corporate Directory that now exists. 

I don't quite understand the difference between having this plugin
installed and the AD changes made, 
versus using the internal CCM users? 

I can only find Cisco's brief discussions, that don't really give a
'features' list of what extra capabilities you have with this installed,
that you don't have without it...

Tim 

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From robert.bell1 at comcast.net  Fri Sep 15 14:08:30 2006
From: robert.bell1 at comcast.net (robert.bell1 at comcast.net)
Date: Fri, 15 Sep 2006 18:08:30 +0000
Subject: [cisco-voip] Avaya VoIP solution and Cisco autoqos (QOS) options
Message-ID: <091520061808.14958.450AEC1D000DBF8C00003A6E2207003201CE04040A0DD29B9D0A0D019D@comcast.net>

We have an Avaya VoIP solution, which I'm still getting familar with. I am much more comfortable with Cisco's VoIP solutions, but my question is simple...I think.  I want to make sure QOS is setup properly.  I am wondering if I ran autoqos on all switch ports and router ports from end - to - end  would the Avaya equipment be properly covered. I did some research and so far all I've come up with is to do this:

class-map match-any VoIP
 match ip dscp 46

policy-map voipQoS
 class VoIP
  priority 816
  class class-default
  fair-queue
  random-detect dscp-based

interface Serial0
 description T1
 ip address 172.16.0.1
 service-policy output voipQoS


Now I know I would have to adjust it for my environment, but I'm not familiar with dscp 46 and what that does or how that sets the priority on the Avaya VoIP equipment.  I would really like to see some sample configs of cisco switch ports and router interfaces along with a quick explaination of why it this works or how maybe another way is better.  
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From pmowry at getgds.com  Fri Sep 15 15:14:05 2006
From: pmowry at getgds.com (Patrick Mowry)
Date: Fri, 15 Sep 2006 14:14:05 -0500
Subject: [cisco-voip] Question regarding AD integration and corporate
	directories...
In-Reply-To: <mailman.9798.1158340979.1681.cisco-voip@puck.nether.net>
Message-ID: <4C792A099259634C83F97067671B37FE026853F4@GDSEXC.gds.pri>

Hello,
  To be blunt, I hate AD integration with a passion.  It makes every AD
user show up in the standard "Corporate Directory".  Unless you change
the description of the user to CiscoPrivateUser, but some sites actually
use the description field for something useful.  It expects the phone
attribute to be the users extension when most sites I've been to has the
full e164 number in this field to sync to their PDA.  When AD has a
hiccup your call center users can't login.  IF you talk about AD
integration, some admins expect this to mean administering CallManager
users and phones from within Microsofts administration tools.

  I much prefer to keep DCDirectory in place and have a script update
whatever field in AD I want with the phone number assignments as so on.
 
I just feel it is more trouble than it's worth for just remembering 1
password. Except for call centers, how many users login to callmanager
anyway.

-Patrick
________________________________________________________________________
____
Date: Fri, 15 Sep 2006 12:54:32 -0400
From: "Tim Reimers" <tim.reimers at asheville.k12.nc.us>
Subject: [cisco-voip] Question regarding AD integration and corporate
	directories...
To: "Cisco Voip" <cisco-voip at puck.nether.net>
Message-ID:
	
<050944EE48FF384F9C663F7048C47DA704CD34D5 at webmail.asheville.k12.nc.us>
Content-Type: text/plain; charset="us-ascii"




Regarding the plugin for AD for the CCMs- mentioned here:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a0080094493.shtml

What does this accomplish for CCM users?

Is it a 'free' option for a 4.01 non-Unifed Messaging environment?

I'm trying to figure out if I should ask my vendor to install this as an
OEM standard option,
and what effect it may have on my existing users on CCM and users on
Unity and the Corporate Directory that now exists.

I don't quite understand the difference between having this plugin
installed and the AD changes made,
versus using the internal CCM users?

I can only find Cisco's brief discussions, that don't really give a
'features' list of what extra capabilities you have with this installed,
that you don't have without it...

Tim



From milay at cisco.com  Fri Sep 15 15:23:52 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Fri, 15 Sep 2006 12:23:52 -0700
Subject: [cisco-voip] Question regarding AD integration and
	corporatedirectories...
Message-ID: <413FABF5DCE22849AF86F442E472B78BC3B966@xmb-sjc-219.amer.cisco.com>

"It makes every AD user show up in the standard "Corporate Directory".
Unless you change the description of the user to CiscoPrivateUser, but
some sites actually use the description field for something useful."

This can be managed by where your search base is pointed. To be more
specific point the search base to an OU that has only the users you want
to see. If you have all of your users in a single OU then there is not
much you can do other than change the description as you have mentioned.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Patrick Mowry
Sent: Friday, September 15, 2006 2:14 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Question regarding AD integration and
corporatedirectories...

Hello,
  To be blunt, I hate AD integration with a passion.  It makes every AD
user show up in the standard "Corporate Directory".  Unless you change
the description of the user to CiscoPrivateUser, but some sites actually
use the description field for something useful.  It expects the phone
attribute to be the users extension when most sites I've been to has the
full e164 number in this field to sync to their PDA.  When AD has a
hiccup your call center users can't login.  IF you talk about AD
integration, some admins expect this to mean administering CallManager
users and phones from within Microsofts administration tools.

  I much prefer to keep DCDirectory in place and have a script update
whatever field in AD I want with the phone number assignments as so on.
 
I just feel it is more trouble than it's worth for just remembering 1
password. Except for call centers, how many users login to callmanager
anyway.

-Patrick
________________________________________________________________________
____
Date: Fri, 15 Sep 2006 12:54:32 -0400
From: "Tim Reimers" <tim.reimers at asheville.k12.nc.us>
Subject: [cisco-voip] Question regarding AD integration and corporate
	directories...
To: "Cisco Voip" <cisco-voip at puck.nether.net>
Message-ID:
	
<050944EE48FF384F9C663F7048C47DA704CD34D5 at webmail.asheville.k12.nc.us>
Content-Type: text/plain; charset="us-ascii"




Regarding the plugin for AD for the CCMs- mentioned here:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note0
9186a0080094493.shtml

What does this accomplish for CCM users?

Is it a 'free' option for a 4.01 non-Unifed Messaging environment?

I'm trying to figure out if I should ask my vendor to install this as an
OEM standard option, and what effect it may have on my existing users on
CCM and users on Unity and the Corporate Directory that now exists.

I don't quite understand the difference between having this plugin
installed and the AD changes made, versus using the internal CCM users?

I can only find Cisco's brief discussions, that don't really give a
'features' list of what extra capabilities you have with this installed,
that you don't have without it...

Tim


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


From jason.aarons at us.didata.com  Fri Sep 15 17:23:17 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 15 Sep 2006 17:23:17 -0400
Subject: [cisco-voip] FW: Cannot leave voice mail from internal phone
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032810B5@USNAEXCH.na.didata.local>

We cannot leave voicemails for end users from certain phones. Any ideas?
Unity 4.0(4).

 
Event Type: Error
Event Source: CiscoUnity_ConvMsg
Event Category: Network 
Event ID: 10015
Date:  9/15/2006
Time:  1:15:46 PM
User:  N/A
Computer: ITIT-UNITY01
Description:
IAvDohMessage::Submit returned [0x800404dc] on line 683 of file
e:\views\cs_ue4.0.3.159\un_Core2\ConversationEng\AvStateSvr\AvSMsgSend.c
pp 
For more information, click: http://www.CiscoUnitySupport.com/find.php  


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

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



From erickbe at yahoo.com  Fri Sep 15 18:52:36 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri, 15 Sep 2006 15:52:36 -0700 (PDT)
Subject: [cisco-voip] Version Stability Opinions
In-Reply-To: <44F7A2A9.90104@cisco.com>
Message-ID: <20060915225236.88464.qmail@web37210.mail.mud.yahoo.com>

Any other workarounds to this CMR bug below, besides stopping CDR Insert until a upgrade to 4.2(3) can be done?  Seeing this issue where the regsvc.exe service spikes CPU while CDRInsert is trying to insert CMR file which is causing call processing delays, etc.  Tried the manual workaround below but the field already exists in the database and table and field properties are correct. 

Event Type: Error
Event  Source: Cisco Anonymous Alarms
Event Category: None
Event  ID: 3
Date:  9/11/2006
Time:  8:49:34  AM
User:  N/A
Computer: CMPUB
Description:
Error:  UNKNOWN_ALARM:kErrorInsertingCDR
  Error: Error executing "insert into  CallDetailRecordDiagnostic  (cdrRecordType,globalCallID_callManagerId,globalCallID_callId,nodeId,dire
   File: \\CMPUB\CDR\CMR\cmr_CMPUB-Cluster01091120060848
   ErrorCode: 207
  State: 42S22
. 


----- Original Message ----
From: Wes Sisk <wsisk at cisco.com>
To: Robin Inderberg <robin.inderberg at candidator.se>
Cc: cisco-voip at puck.nether.net
Sent: Thursday, August 31, 2006 10:02:01 PM
Subject: Re: [cisco-voip] Version Stability Opinions

      For anyone upgrading from 4.1 to 4.2(1) please be aware of 
 CSCse68968 CMR files not inserted into DB on upgrade from 4.1 to 4.2
 
 Calldetailrecorddiagnostics did not get updated with a field it requires during upgrade.  Workaround:
 ?    Stop insertCDR Service 
 ?    Through Enterprise manager or Query analyzer, execute on CDR database 
 ALTER TABLE CallDetailRecordDiagnostic ADD varVQMetrics varchar(600) DEFAULT NULL NULL 
 ?    In CDR file location, move CMR_xxxxxxx files in BAD folder to CMR folder 
 ?    Start insertCDR Service 
 
 This is only resolved via this manual workaround or with CM 4.2(3) which just posted to Cisco.com tonight.
 
 /Wes
 
 Robin Inderberg wrote:                                      Hi Duncan,
      
    We are running CCM 4.2(1)sr1b and Cisco Unity 4.0(5), and we are not expericen any problems at all.
    The only problem we found now were on the upgrade to 4.2(1)sr1b, and Cisco Unity did Crash? So a restart after this one, everything worked well again.
      
    And some more problem is the bug in Cisco 7920 with 3.0 and the HLOG function, it doesn?t work at all. The softkey is removed.
    And in 2.01 it shows wrong when you have loged in and loged out from a hlog? 
      
    I don?t know if it is CCM 4.2, or if it is my setup, but the SIP trunk, after installing CCM 4.2 don?t work very well.
      
    The MRM seems to have some problems, atleast is it what I have localised the problem too. 
      
    Is this a problem anyone else has?
      
    If anyone is interrested in the debugs from it, I have them right here. 
      
    Hope you got any information that could be valuable for you Duncan .   
      
       Med v?nliga h?lsningar/Best Regards
      
       Robin Inderberg
    
         
   From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of DUNCAN, WILLIAM B.
   Sent: den 31 augusti 2006 18:58
   To: cisco-voip at puck.nether.net
   Subject: [cisco-voip] Version Stability Opinions
    
     
    I was interested in what you all thought were currently the best versions of CCM and Unity from a stability standpoint.  We are running CCM 4.13 and Unity 4.04 BTW.  
      
    Thanks,
    Brian Duncan
    Coordinator of Telecommunications
    Ozarks Technical    Community College
      
    
    _______________________________________________ cisco-voip mailing list cisco-voip at puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip       _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




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From peder at networkoblivion.com  Fri Sep 15 21:36:38 2006
From: peder at networkoblivion.com (Peder @ NetworkOblivion)
Date: Fri, 15 Sep 2006 20:36:38 -0500
Subject: [cisco-voip] PRI, SIP & CallerID Name
Message-ID: <450B5526.9070701@networkoblivion.com>

Does anybody know how to get a Cisco gateway with PRI to pass CallerID 
name over SIP to a gateway?  I'm pretty sure it's supported, but I've 
googled everything I can and I can't find the right combination of key 
words.  I have this vague memory of it being a new feature in 12.3T, but 
I can't recall where I saw/heard that.

PA




From tim.reimers at asheville.k12.nc.us  Sat Sep 16 12:08:38 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Sat, 16 Sep 2006 12:08:38 -0400
Subject: [cisco-voip] Reset Unity mailbox password..
Message-ID: <050944EE48FF384F9C663F7048C47DA7030EC52E@webmail.asheville.k12.nc.us>

 
hello everyone--
 
I'm looking for a way to allow a subscribers Unity mailbox password to be reset in a way OTHER than through the Unity webpages (and TermServ to Unity)
 
I really really need to limit the access of people to full Unity to just a few..
.. but at the same time, those few of us with full access CANNOT keep up with the requests for resetting of passwords. (I know, you'd think that's a user education thing..)
 
So -- I'm hoping for something like Greetings Administrator to have rights to reset passwords on any mailbox -- similar to how the person using GA can select any call handler and work with it..
I'm hoping there's a level of access I can grant to a  additional staff members at remote sites just to be able to clear and reset a password using their phone and something like Greetings Administrator?
 
If that's not clear, just email me and I'll try to confuse you some more... ;-)
 
thanks, Tim
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From jonvoip at gmail.com  Sat Sep 16 14:02:27 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 16 Sep 2006 13:02:27 -0500
Subject: [cisco-voip] FW: Cannot leave voice mail from internal phone
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0032810B5@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0032810B5@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0609161102g67960779u122db484d69179e8@mail.gmail.com>

Calling Search space for those phones does not include the vmail ports...



Jonathan

On 9/15/06, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
> We cannot leave voicemails for end users from certain phones. Any ideas?
> Unity 4.0(4).
>
>
> Event Type: Error
> Event Source: CiscoUnity_ConvMsg
> Event Category: Network
> Event ID: 10015
> Date:  9/15/2006
> Time:  1:15:46 PM
> User:  N/A
> Computer: ITIT-UNITY01
> Description:
> IAvDohMessage::Submit returned [0x800404dc] on line 683 of file
> e:\views\cs_ue4.0.3.159\un_Core2\ConversationEng\AvStateSvr\AvSMsgSend.c
> pp
> For more information, click: http://www.CiscoUnitySupport.com/find.php
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From aman.chugh at gmail.com  Sat Sep 16 15:21:42 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Sun, 17 Sep 2006 00:51:42 +0530
Subject: [cisco-voip] FW: Cannot leave voice mail from internal phone
In-Reply-To: <5d093f9a0609161102g67960779u122db484d69179e8@mail.gmail.com>
References: <C1FE15183DA37645BC0633BC604E44F0032810B5@USNAEXCH.na.didata.local>
	<5d093f9a0609161102g67960779u122db484d69179e8@mail.gmail.com>
Message-ID: <d4c697ad0609161221m30bd9c1bn4d8c91e31131ee8a@mail.gmail.com>

Not possible in this case as the call is reaching unity and then failing as
event viewer indicates,I would think it is that we cant leave messages for
some subscribers in Unity, i would try to recreate them or check the
permission on the message store where there mailboxes are stored.

Aman


On 9/16/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Calling Search space for those phones does not include the vmail ports...
>
>
>
> Jonathan
>
> On 9/15/06, Jason Aarons (US) < jason.aarons at us.didata.com> wrote:
> >
> > We cannot leave voicemails for end users from certain phones. Any ideas?
> >
> > Unity 4.0(4).
> >
> >
> > Event Type: Error
> > Event Source: CiscoUnity_ConvMsg
> > Event Category: Network
> > Event ID: 10015
> > Date:  9/15/2006
> > Time:  1:15:46 PM
> > User:  N/A
> > Computer: ITIT-UNITY01
> > Description:
> > IAvDohMessage::Submit returned [0x800404dc] on line 683 of file
> > e:\views\cs_ue4.0.3.159\un_Core2\ConversationEng\AvStateSvr\AvSMsgSend.c
> > pp
> > For more information, click: http://www.CiscoUnitySupport.com/find.php<http://www.ciscounitysupport.com/find.php>
> >
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain
> > confidential and privileged information and is for use by the
> > designated addressee(s) named above only.  If you are not the
> > intended addressee, you are hereby notified that you have received
> > this communication in error and that any use or reproduction of
> > this email or its contents is strictly prohibited and may be
> > unlawful.  If you have received this communication in error, please
> > notify us immediately by replying to this message and deleting it
> > from your computer. Thank you.
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From jgrace at digitelusa.net  Sat Sep 16 15:26:35 2006
From: jgrace at digitelusa.net (James Grace)
Date: 16 Sep 2006 19:26:35 GMT
Subject: [cisco-voip] fwd Unity messaging will not play
Message-ID: <SUWCL19IP01pfoyrBgb0000562c@smtpoutcl19.wirelesssyncvzw.net>

We are fwd unity unified messages to my smart phone from our exchange server.  I get the message as a wave file .  But will not play.  Do I need a certain codec on this phone to play this wave file.  Samsung i730 windows mobile 2003.  I can play every thing else on this phone

James D Grace
CCNP CCNA MCSE MCDBA
Sr. System Engineer
MSN Messenger ID: jgrace29 at hotmail.com
MAC IChat ID: 
jgrace29 at mac.com

From adman at adman.net  Sat Sep 16 16:15:31 2006
From: adman at adman.net (Adam Blomfield)
Date: Sat, 16 Sep 2006 15:15:31 -0500
Subject: [cisco-voip] fwd Unity messaging will not play
In-Reply-To: <SUWCL19IP01pfoyrBgb0000562c@smtpoutcl19.wirelesssyncvzw.net>
References: <SUWCL19IP01pfoyrBgb0000562c@smtpoutcl19.wirelesssyncvzw.net>
Message-ID: <c6f32ce40609161315u2ed3eec2y4ecf960fd457a34b@mail.gmail.com>

On 16 Sep 2006 19:26:35 GMT, James Grace <jgrace at digitelusa.net> wrote:
> We are fwd unity unified messages to my smart phone from our exchange server.  I get the message as a wave file .  But will not play.  Do I need a certain codec on this phone to play this wave file.  Samsung i730 windows mobile 2003.  I can play every thing else on this phone
>
> James D Grace
> CCNP CCNA MCSE MCDBA
> Sr. System Engineer
> MSN Messenger ID: jgrace29 at hotmail.com
> MAC IChat ID:
> jgrace29 at mac.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
Try downloading TCPMP from http://tcpmp.corecodec.org/download.

From SPackett at fenwick.com  Sat Sep 16 18:49:04 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Sat, 16 Sep 2006 15:49:04 -0700
Subject: [cisco-voip] fwd Unity messaging will not play
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC56F575@SVCEXCH01.firm.fenwick.llp>

We install an application called mVoicemail from MotionApps on all our
smart phones.  It plays the .wav files just fine as the application can
handle the codecs.  I highly suggest you give it a try.  A good amount
of our users are just playing the .wav files instead of calling into the
voicemail system to check voicemails. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Saturday, September 16, 2006 12:27 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] fwd Unity messaging will not play

We are fwd unity unified messages to my smart phone from our exchange
server.  I get the message as a wave file .  But will not play.  Do I
need a certain codec on this phone to play this wave file.  Samsung i730
windows mobile 2003.  I can play every thing else on this phone

James D Grace
CCNP CCNA MCSE MCDBA
Sr. System Engineer
MSN Messenger ID: jgrace29 at hotmail.com
MAC IChat ID: 
jgrace29 at mac.com
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From wsisk at cisco.com  Sat Sep 16 19:36:11 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Sat, 16 Sep 2006 19:36:11 -0400
Subject: [cisco-voip] Version Stability Opinions
In-Reply-To: <20060915225236.88464.qmail@web37210.mail.mud.yahoo.com>
References: <20060915225236.88464.qmail@web37210.mail.mud.yahoo.com>
Message-ID: <450C8A6B.6060403@cisco.com>

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From lelio at uoguelph.ca  Sat Sep 16 20:18:14 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat, 16 Sep 2006 20:18:14 -0400
Subject: [cisco-voip] Reset Unity mailbox password..
References: <050944EE48FF384F9C663F7048C47DA7030EC52E@webmail.asheville.k12.nc.us>
Message-ID: <003e01c6d9ee$c52025a0$ca01a8c0@deadfish>

I have not seen anything like this. Although it would make for a great utility, say send their reset password to their email address. I cannot find any reference to an API like SOAP for unity, that would have helped immensely. You can try posting in the Cisco forums, I'm pretty sure Jeff Lindborgh reads that. Jeff was the primary developer for Unity and I believe he still is.

If your users have the domain password available to them, check out the PCA login page. It's a little different than the web/sa pages and might fit your needs.

  ----- Original Message ----- 
  From: Tim Reimers 
  To: cisco-voip at puck.nether.net 
  Cc: Alan Surrette 
  Sent: Saturday, September 16, 2006 12:08 PM
  Subject: [cisco-voip] Reset Unity mailbox password..



  hello everyone--

  I'm looking for a way to allow a subscribers Unity mailbox password to be reset in a way OTHER than through the Unity webpages (and TermServ to Unity)

  I really really need to limit the access of people to full Unity to just a few..
  .. but at the same time, those few of us with full access CANNOT keep up with the requests for resetting of passwords. (I know, you'd think that's a user education thing..)

  So -- I'm hoping for something like Greetings Administrator to have rights to reset passwords on any mailbox -- similar to how the person using GA can select any call handler and work with it..
  I'm hoping there's a level of access I can grant to a  additional staff members at remote sites just to be able to clear and reset a password using their phone and something like Greetings Administrator?

  If that's not clear, just email me and I'll try to confuse you some more... ;-)

  thanks, Tim


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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From tim.reimers at asheville.k12.nc.us  Sat Sep 16 21:18:47 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Sat, 16 Sep 2006 21:18:47 -0400
Subject: [cisco-voip] Reset Unity mailbox password..
References: <050944EE48FF384F9C663F7048C47DA7030EC52E@webmail.asheville.k12.nc.us>
	<003e01c6d9ee$c52025a0$ca01a8c0@deadfish>
Message-ID: <050944EE48FF384F9C663F7048C47DA7030EC52F@webmail.asheville.k12.nc.us>

You know, now that  I think about it--- hmm...
 
I'll have to check around in AD on Unity and see where they've embedded the password in some AD field-- I'm betting that the mailbox password is in there somewhere..
 
We have some internal experience in scripting AD for our main domain --- if it's there in AD in some schema extended field... hmmm..
 
time to break out that LDAP browser that someone posted a link to - I did manage to get that to connect to Unity's AD, though not our data AD or the CCM DCDirectory..
 
 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Sat 9/16/2006 8:18 PM
To: Tim Reimers; cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: Re: [cisco-voip] Reset Unity mailbox password..


I have not seen anything like this. Although it would make for a great utility, say send their reset password to their email address. I cannot find any reference to an API like SOAP for unity, that would have helped immensely. You can try posting in the Cisco forums, I'm pretty sure Jeff Lindborgh reads that. Jeff was the primary developer for Unity and I believe he still is.
 
If your users have the domain password available to them, check out the PCA login page. It's a little different than the web/sa pages and might fit your needs.
 

	----- Original Message ----- 
	From: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  
	To: cisco-voip at puck.nether.net 
	Cc: Alan Surrette <mailto:alan.surrette at asheville.k12.nc.us>  
	Sent: Saturday, September 16, 2006 12:08 PM
	Subject: [cisco-voip] Reset Unity mailbox password..

	 
	hello everyone--
	 
	I'm looking for a way to allow a subscribers Unity mailbox password to be reset in a way OTHER than through the Unity webpages (and TermServ to Unity)
	 
	I really really need to limit the access of people to full Unity to just a few..
	.. but at the same time, those few of us with full access CANNOT keep up with the requests for resetting of passwords. (I know, you'd think that's a user education thing..)
	 
	So -- I'm hoping for something like Greetings Administrator to have rights to reset passwords on any mailbox -- similar to how the person using GA can select any call handler and work with it..
	I'm hoping there's a level of access I can grant to a  additional staff members at remote sites just to be able to clear and reset a password using their phone and something like Greetings Administrator?
	 
	If that's not clear, just email me and I'll try to confuse you some more... ;-)
	 
	thanks, Tim

	
________________________________


	

	_______________________________________________
	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 Sep 16 21:27:01 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat, 16 Sep 2006 21:27:01 -0400
Subject: [cisco-voip] Reset Unity mailbox password..
References: <050944EE48FF384F9C663F7048C47DA7030EC52E@webmail.asheville.k12.nc.us>
	<003e01c6d9ee$c52025a0$ca01a8c0@deadfish>
	<050944EE48FF384F9C663F7048C47DA7030EC52F@webmail.asheville.k12.nc.us>
Message-ID: <001901c6d9f8$6057ee50$ca01a8c0@deadfish>

I didn't look in AD, but when I looked in the Unity SQL enterprise manager, there was a phone password field in the subscribers table. If it's there, I don't think it will be in AD. Or will it?
  ----- Original Message ----- 
  From: Tim Reimers 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Cc: Alan Surrette 
  Sent: Saturday, September 16, 2006 9:18 PM
  Subject: RE: [cisco-voip] Reset Unity mailbox password..


  You know, now that  I think about it--- hmm...

  I'll have to check around in AD on Unity and see where they've embedded the password in some AD field-- I'm betting that the mailbox password is in there somewhere..

  We have some internal experience in scripting AD for our main domain --- if it's there in AD in some schema extended field... hmmm..

  time to break out that LDAP browser that someone posted a link to - I did manage to get that to connect to Unity's AD, though not our data AD or the CCM DCDirectory..




------------------------------------------------------------------------------
  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
  Sent: Sat 9/16/2006 8:18 PM
  To: Tim Reimers; cisco-voip at puck.nether.net
  Cc: Alan Surrette
  Subject: Re: [cisco-voip] Reset Unity mailbox password..


  I have not seen anything like this. Although it would make for a great utility, say send their reset password to their email address. I cannot find any reference to an API like SOAP for unity, that would have helped immensely. You can try posting in the Cisco forums, I'm pretty sure Jeff Lindborgh reads that. Jeff was the primary developer for Unity and I believe he still is.

  If your users have the domain password available to them, check out the PCA login page. It's a little different than the web/sa pages and might fit your needs.

    ----- Original Message ----- 
    From: Tim Reimers 
    To: cisco-voip at puck.nether.net 
    Cc: Alan Surrette 
    Sent: Saturday, September 16, 2006 12:08 PM
    Subject: [cisco-voip] Reset Unity mailbox password..



    hello everyone--

    I'm looking for a way to allow a subscribers Unity mailbox password to be reset in a way OTHER than through the Unity webpages (and TermServ to Unity)

    I really really need to limit the access of people to full Unity to just a few..
    .. but at the same time, those few of us with full access CANNOT keep up with the requests for resetting of passwords. (I know, you'd think that's a user education thing..)

    So -- I'm hoping for something like Greetings Administrator to have rights to reset passwords on any mailbox -- similar to how the person using GA can select any call handler and work with it..
    I'm hoping there's a level of access I can grant to a  additional staff members at remote sites just to be able to clear and reset a password using their phone and something like Greetings Administrator?

    If that's not clear, just email me and I'll try to confuse you some more... ;-)

    thanks, Tim


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


    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip
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From jim at tgasolutions.com  Sat Sep 16 22:42:26 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Sat, 16 Sep 2006 22:42:26 -0400
Subject: [cisco-voip] Reset Unity mailbox password..
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC1E40D2@exchange.TGASolutions.local>

Take a look over at www.ciscounitytools.com
 
There are some architecture notes there...
 
J

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Saturday, September 16, 2006 9:19 PM
To: Lelio Fulgenzi; cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: Re: [cisco-voip] Reset Unity mailbox password..


You know, now that  I think about it--- hmm...
 
I'll have to check around in AD on Unity and see where they've embedded
the password in some AD field-- I'm betting that the mailbox password is
in there somewhere..
 
We have some internal experience in scripting AD for our main domain ---
if it's there in AD in some schema extended field... hmmm..
 
time to break out that LDAP browser that someone posted a link to - I
did manage to get that to connect to Unity's AD, though not our data AD
or the CCM DCDirectory..
 
 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Sat 9/16/2006 8:18 PM
To: Tim Reimers; cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: Re: [cisco-voip] Reset Unity mailbox password..


I have not seen anything like this. Although it would make for a great
utility, say send their reset password to their email address. I cannot
find any reference to an API like SOAP for unity, that would have helped
immensely. You can try posting in the Cisco forums, I'm pretty sure Jeff
Lindborgh reads that. Jeff was the primary developer for Unity and I
believe he still is.
 
If your users have the domain password available to them, check out the
PCA login page. It's a little different than the web/sa pages and might
fit your needs.
 

	----- Original Message ----- 
	From: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  
	To: cisco-voip at puck.nether.net 
	Cc: Alan Surrette <mailto:alan.surrette at asheville.k12.nc.us>  
	Sent: Saturday, September 16, 2006 12:08 PM
	Subject: [cisco-voip] Reset Unity mailbox password..

	 
	hello everyone--
	 
	I'm looking for a way to allow a subscribers Unity mailbox
password to be reset in a way OTHER than through the Unity webpages (and
TermServ to Unity)
	 
	I really really need to limit the access of people to full Unity
to just a few..
	.. but at the same time, those few of us with full access CANNOT
keep up with the requests for resetting of passwords. (I know, you'd
think that's a user education thing..)
	 
	So -- I'm hoping for something like Greetings Administrator to
have rights to reset passwords on any mailbox -- similar to how the
person using GA can select any call handler and work with it..
	I'm hoping there's a level of access I can grant to a
additional staff members at remote sites just to be able to clear and
reset a password using their phone and something like Greetings
Administrator?
	 
	If that's not clear, just email me and I'll try to confuse you
some more... ;-)
	 
	thanks, Tim

	
________________________________


	

	_______________________________________________
	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 Sep 16 23:31:04 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat, 16 Sep 2006 23:31:04 -0400
Subject: [cisco-voip] Reset Unity mailbox password..
References: <D2BBAE709B4D874DAC556C7EB9D938FC1E40D2@exchange.TGASolutions.local>
Message-ID: <004101c6da09$b5352e40$ca01a8c0@deadfish>

Thanks Jim. Looks like there might be something there that Tim can use: http://www.ciscounitytools.com/Documents/programaticAdmin.doc


Update subscriber's password
Setting the password is easy going through the stored procedure since it will take care of hashing your raw phone password into an MD5 string for you on the back end.  MD5 strings should always be exactly 32 characters here.  Since phone passwords can only be 20 digits in length, the stored procedure assumes that if the string is 32 characters long it's already been hashed into an MD5 string and will pass it through directly.  Otherwise it'll hash it for you on the back end so you can simply pass through the raw digits for the phone password you want to set and it'll take care of it for you.  If you're paranoid you can also hash the phone password into it's MD5 string and then pass that into the stored procedure instead.  It'll handle it either way.

 

When setting the password for a subscriber it's up to you to be sure it complies with the password policy for the site.  If you pass in a blank string for the password, for instance, the phone password will be cleared.  You can check the phone password policies in the PwPolicy table in UnityDB.  The stored procedure does not do any enforcement of the site's policy, so make sure you're careful here.

 

    oCommand.CommandText = "sp_ModifySubscriber"

 

    oCommand.Parameters.Item("@SubscriberObjectID") = oCommand.CreateParameter("SubscriberObjectID", adGUID, adParamInput, , strSubscriberObjectID)

 

    'The phone password is passed in as clear text in this example - it gets crunched and encrypted by the SP during the subscriber update process.  You can pass in a "pre hashed" MD5 string yourself - if the string is exactly 32 characters long the stored proc will pass it through "as is".

    oCommand.Parameters.Item("@PWDTMF") = oCommand.CreateParameter("PWDTMF", adVarChar, adParamInput, , txtPhonePassword.Text)

    

'Passwords are not written through to the directory, no need to synch.

        oCommand.Parameters.Item("@DirectorySync") = oCommand.CreateParameter("DirectorySync", adInteger, adParamInput, , 0)

    

    'Let the stored procedure call rip.

    oCommand.Execute

 

Since phone passwords are not pushed into the directory there's no need to issue a synchronization request for this.  You're done.

  ----- Original Message ----- 
  From: Jim McBurnett 
  To: Tim Reimers ; Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Cc: Alan Surrette 
  Sent: Saturday, September 16, 2006 10:42 PM
  Subject: RE: [cisco-voip] Reset Unity mailbox password..


  Take a look over at www.ciscounitytools.com

  There are some architecture notes there...

  J



------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
  Sent: Saturday, September 16, 2006 9:19 PM
  To: Lelio Fulgenzi; cisco-voip at puck.nether.net
  Cc: Alan Surrette
  Subject: Re: [cisco-voip] Reset Unity mailbox password..


  You know, now that  I think about it--- hmm...

  I'll have to check around in AD on Unity and see where they've embedded the password in some AD field-- I'm betting that the mailbox password is in there somewhere..

  We have some internal experience in scripting AD for our main domain --- if it's there in AD in some schema extended field... hmmm..

  time to break out that LDAP browser that someone posted a link to - I did manage to get that to connect to Unity's AD, though not our data AD or the CCM DCDirectory..




------------------------------------------------------------------------------
  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
  Sent: Sat 9/16/2006 8:18 PM
  To: Tim Reimers; cisco-voip at puck.nether.net
  Cc: Alan Surrette
  Subject: Re: [cisco-voip] Reset Unity mailbox password..


  I have not seen anything like this. Although it would make for a great utility, say send their reset password to their email address. I cannot find any reference to an API like SOAP for unity, that would have helped immensely. You can try posting in the Cisco forums, I'm pretty sure Jeff Lindborgh reads that. Jeff was the primary developer for Unity and I believe he still is.

  If your users have the domain password available to them, check out the PCA login page. It's a little different than the web/sa pages and might fit your needs.

    ----- Original Message ----- 
    From: Tim Reimers 
    To: cisco-voip at puck.nether.net 
    Cc: Alan Surrette 
    Sent: Saturday, September 16, 2006 12:08 PM
    Subject: [cisco-voip] Reset Unity mailbox password..



    hello everyone--

    I'm looking for a way to allow a subscribers Unity mailbox password to be reset in a way OTHER than through the Unity webpages (and TermServ to Unity)

    I really really need to limit the access of people to full Unity to just a few..
    .. but at the same time, those few of us with full access CANNOT keep up with the requests for resetting of passwords. (I know, you'd think that's a user education thing..)

    So -- I'm hoping for something like Greetings Administrator to have rights to reset passwords on any mailbox -- similar to how the person using GA can select any call handler and work with it..
    I'm hoping there's a level of access I can grant to a  additional staff members at remote sites just to be able to clear and reset a password using their phone and something like Greetings Administrator?

    If that's not clear, just email me and I'll try to confuse you some more... ;-)

    thanks, Tim


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


    _______________________________________________
    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 Sep 16 23:31:12 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat, 16 Sep 2006 23:31:12 -0400
Subject: [cisco-voip] Reset Unity mailbox password..
References: <D2BBAE709B4D874DAC556C7EB9D938FC1E40D2@exchange.TGASolutions.local>
Message-ID: <004501c6da09$b9fd3da0$ca01a8c0@deadfish>

Thanks Jim. Looks like there might be something there that Tim can use: http://www.ciscounitytools.com/Documents/programaticAdmin.doc


Update subscriber's password
Setting the password is easy going through the stored procedure since it will take care of hashing your raw phone password into an MD5 string for you on the back end.  MD5 strings should always be exactly 32 characters here.  Since phone passwords can only be 20 digits in length, the stored procedure assumes that if the string is 32 characters long it's already been hashed into an MD5 string and will pass it through directly.  Otherwise it'll hash it for you on the back end so you can simply pass through the raw digits for the phone password you want to set and it'll take care of it for you.  If you're paranoid you can also hash the phone password into it's MD5 string and then pass that into the stored procedure instead.  It'll handle it either way.

 

When setting the password for a subscriber it's up to you to be sure it complies with the password policy for the site.  If you pass in a blank string for the password, for instance, the phone password will be cleared.  You can check the phone password policies in the PwPolicy table in UnityDB.  The stored procedure does not do any enforcement of the site's policy, so make sure you're careful here.

 

    oCommand.CommandText = "sp_ModifySubscriber"

 

    oCommand.Parameters.Item("@SubscriberObjectID") = oCommand.CreateParameter("SubscriberObjectID", adGUID, adParamInput, , strSubscriberObjectID)

 

    'The phone password is passed in as clear text in this example - it gets crunched and encrypted by the SP during the subscriber update process.  You can pass in a "pre hashed" MD5 string yourself - if the string is exactly 32 characters long the stored proc will pass it through "as is".

    oCommand.Parameters.Item("@PWDTMF") = oCommand.CreateParameter("PWDTMF", adVarChar, adParamInput, , txtPhonePassword.Text)

    

'Passwords are not written through to the directory, no need to synch.

        oCommand.Parameters.Item("@DirectorySync") = oCommand.CreateParameter("DirectorySync", adInteger, adParamInput, , 0)

    

    'Let the stored procedure call rip.

    oCommand.Execute

 

Since phone passwords are not pushed into the directory there's no need to issue a synchronization request for this.  You're done.

  ----- Original Message ----- 
  From: Jim McBurnett 
  To: Tim Reimers ; Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Cc: Alan Surrette 
  Sent: Saturday, September 16, 2006 10:42 PM
  Subject: RE: [cisco-voip] Reset Unity mailbox password..


  Take a look over at www.ciscounitytools.com

  There are some architecture notes there...

  J



------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
  Sent: Saturday, September 16, 2006 9:19 PM
  To: Lelio Fulgenzi; cisco-voip at puck.nether.net
  Cc: Alan Surrette
  Subject: Re: [cisco-voip] Reset Unity mailbox password..


  You know, now that  I think about it--- hmm...

  I'll have to check around in AD on Unity and see where they've embedded the password in some AD field-- I'm betting that the mailbox password is in there somewhere..

  We have some internal experience in scripting AD for our main domain --- if it's there in AD in some schema extended field... hmmm..

  time to break out that LDAP browser that someone posted a link to - I did manage to get that to connect to Unity's AD, though not our data AD or the CCM DCDirectory..




------------------------------------------------------------------------------
  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
  Sent: Sat 9/16/2006 8:18 PM
  To: Tim Reimers; cisco-voip at puck.nether.net
  Cc: Alan Surrette
  Subject: Re: [cisco-voip] Reset Unity mailbox password..


  I have not seen anything like this. Although it would make for a great utility, say send their reset password to their email address. I cannot find any reference to an API like SOAP for unity, that would have helped immensely. You can try posting in the Cisco forums, I'm pretty sure Jeff Lindborgh reads that. Jeff was the primary developer for Unity and I believe he still is.

  If your users have the domain password available to them, check out the PCA login page. It's a little different than the web/sa pages and might fit your needs.

    ----- Original Message ----- 
    From: Tim Reimers 
    To: cisco-voip at puck.nether.net 
    Cc: Alan Surrette 
    Sent: Saturday, September 16, 2006 12:08 PM
    Subject: [cisco-voip] Reset Unity mailbox password..



    hello everyone--

    I'm looking for a way to allow a subscribers Unity mailbox password to be reset in a way OTHER than through the Unity webpages (and TermServ to Unity)

    I really really need to limit the access of people to full Unity to just a few..
    .. but at the same time, those few of us with full access CANNOT keep up with the requests for resetting of passwords. (I know, you'd think that's a user education thing..)

    So -- I'm hoping for something like Greetings Administrator to have rights to reset passwords on any mailbox -- similar to how the person using GA can select any call handler and work with it..
    I'm hoping there's a level of access I can grant to a  additional staff members at remote sites just to be able to clear and reset a password using their phone and something like Greetings Administrator?

    If that's not clear, just email me and I'll try to confuse you some more... ;-)

    thanks, Tim


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


    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip
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From cshaffer at gmail.com  Sun Sep 17 00:20:28 2006
From: cshaffer at gmail.com (Curt Shaffer)
Date: Sun, 17 Sep 2006 00:20:28 -0400
Subject: [cisco-voip] Fax over IP
Message-ID: <005901c6da10$9ca0beb0$181b10ac@betsy>

I am currently running an Asterisk VoIP server at home. I need to fax
reliably and this does not seem to work for me with this solution. Outgoing
seems to work great but incoming is about 25%. Anyway I am running a 1760
router here and I was thinking about configuring an FXS port on it for the
fax machine and still doing SIP to the Asterisk box for registration. I know
this is not an open source list but my question is have any of you out there
had any reliability with this type of setup, not meaning Asterisk per say,
but with SIP on the front end to a router with an FXS card with a Fax
machine connected to it. If so are they any comments or suggestions you can
make for me to help make this at least a little more reliable? If you need
any other information, please ask and I can provide you with any details you
need.

 

Thanks!

 

Curt

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From tdobrev at telelink.bg  Sun Sep 17 10:18:31 2006
From: tdobrev at telelink.bg (Teodor Dobrev)
Date: Sun, 17 Sep 2006 17:18:31 +0300
Subject: [cisco-voip] CRS4.5 agent can't login
Message-ID: <89560EBC93004045847D8683424E0E97015A590E@exch1.telelink.bg>

Hi Guys,

I have just installed CRS 4.5 with CCM 5.0  but I am facing this issue:

I can't login with the CAD ( neither with ip phone agent )  - it says 

" The ID you entered was not found "

I searched a few troubleshooting docs and i noticed that I don't have
the C:\Program Files\Cisco\Desktop_config folder.

Have you faced similar issue before I will be glad to receive a tip!

 

 

BR,

Teo

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From jgrace at digitel.net  Sun Sep 17 19:34:28 2006
From: jgrace at digitel.net (James Grace)
Date: Sun, 17 Sep 2006 19:34:28 -0400
Subject: [cisco-voip] way to change the message notification schedule
Message-ID: <200609171933984.SM01512@jgm>

We have about 150 users all with message notification.  The default schedule
is 8 to 5 mon to fri.  Is there some type of way to change this in bulk?  I
change the subscriber's vm schedule but this does not change the message
notification schedule 

 

James D Grace 

CCNP CCNA MCSE MCDBA

System Engineer / Professional Svc.

Digitel Corp.

Email:  <mailto:grace.jg at gmail> grace.jg at gmail 

email: jgrace at digitel.net 

MSN id: grace.jd at gmail.com

 

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From jgrace at digitel.net  Sun Sep 17 20:35:46 2006
From: jgrace at digitel.net (James Grace)
Date: Sun, 17 Sep 2006 20:35:46 -0400
Subject: [cisco-voip] trying to find the cisco doc on how to prefix the
	redial in cm
Message-ID: <200609172034437.SM01512@jgm>

trying to find the cisco doc on how to prefix the redial in cm.  I seen a
doc that gave you the steps.  But can not find it anymore.  

 

James D Grace 

CCNP CCNA MCSE MCDBA

System Engineer / Professional Svc.

Digitel Corp.

Email:  <mailto:grace.jg at gmail> grace.jg at gmail 

email: jgrace at digitel.net 

MSN id: grace.jd at gmail.com

 

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From erickbe at yahoo.com  Sun Sep 17 21:07:45 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sun, 17 Sep 2006 18:07:45 -0700 (PDT)
Subject: [cisco-voip] trying to find the cisco doc on how to prefix the
	redial in cm
In-Reply-To: <200609172034437.SM01512@jgm>
Message-ID: <20060918010745.60537.qmail@web37205.mail.mud.yahoo.com>

Here is the doc with 2 ways to do it in Call Manager.  If you have a H.323 gateway you can do it with translation profiles, and I believe there is a different doc that covers that but don't have it offhand.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00805b6f33.shtml

----- Original Message ----
From: James Grace <jgrace at digitel.net>
To: cisco-voip at puck.nether.net
Sent: Sunday, September 17, 2006 7:35:46 PM
Subject: [cisco-voip] trying to find the cisco doc on how to prefix the redial in cm

             trying to find the cisco doc on how to prefix the redial in cm.  I seen a doc that gave you the steps.  But can not find it anymore.  
     
   James D Grace 
   CCNP CCNA MCSE MCDBA
   System Engineer / Professional Svc.
   Digitel Corp.
   Email: grace.jg at gmail 
   email: jgrace at digitel.net 
   MSN id: grace.jd at gmail.com
     
   
     _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




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From gpate at kenttech.com  Sun Sep 17 21:36:10 2006
From: gpate at kenttech.com (Gary L. Pate)
Date: Sun, 17 Sep 2006 21:36:10 -0400
Subject: [cisco-voip] Best Practices
Message-ID: <00a501c6dac2$d2052160$25fb14ac@gpatelt>

I'm trying to determine a "best practices" when setting up inbound DID's,
extensions, etc. Let me explain:

 

In most situations I've been involved in, I've used the last 4 digits of a
DID as someone's extension. I make sure the LEC sends 10 digit's, and I do a
significant digits of 4 on the gateway. This has always worked well, and is
easy to configure. However, in a recent install, this customer was using 3
digit extensions, so I again had the LEC send 10 digits, and I made the
gateway significant digits of 3. Again, this worked, but the problem was
that I started running into to many situations where I had duplicate 3
digits. For example, a DID might be 593-3183, so the phone extension was
183. But I also had a existing fax line number that was 444-4183. As you can
see, the last 3 digits are the same. I could not make a translation pattern
of 444183 because the gateway was only sending 3 digits. I ended up having
the LEC forward the 4444183 to one of the DID ranges, and then gave the fax
the number from that range, but that is not what I want to do in the future.


I guess what I want to know is what is the "best practice" going forward? Is
it a good idea to have the LEC send 10 digits, and if so, what should I set
on the gateway for significant digit's, and how should I handle my internal
phones that are using the last four digits of the DID as their extension?
I've read a document on using translation patterns to prefix digits, etc,
but I'm not sure if that is the directions I should go. 

Can someone offer advice on this, and perhaps provide an example of how to
do this with both a 4 digit example, and a 3 digit example? I really want to
have more control over the lines, and not have to worry much about duplicate
numbers or having the LEC getting involved.

 

Gary

 

 

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From tim.reimers at asheville.k12.nc.us  Sun Sep 17 21:39:53 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Sun, 17 Sep 2006 21:39:53 -0400
Subject: [cisco-voip] Reset Unity mailbox password..
References: <D2BBAE709B4D874DAC556C7EB9D938FC1E40D2@exchange.TGASolutions.local>
	<004501c6da09$b9fd3da0$ca01a8c0@deadfish>
Message-ID: <050944EE48FF384F9C663F7048C47DA7030EC531@webmail.asheville.k12.nc.us>

Cool! 
 
Thanks...
 
that's the biggest request we get -- other things, I think we can continue to handle internally..
 
It may take the skill of our app developer to do this, not me.. .but that does look as though it's do-able...
 
thanks guys..
Tim

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Sat 9/16/2006 11:31 PM
To: Jim McBurnett; Tim Reimers; cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: Re: [cisco-voip] Reset Unity mailbox password..


Thanks Jim. Looks like there might be something there that Tim can use: http://www.ciscounitytools.com/Documents/programaticAdmin.doc
 
 

Update subscriber's password


Setting the password is easy going through the stored procedure since it will take care of hashing your raw phone password into an MD5 string for you on the back end.  MD5 strings should always be exactly 32 characters here.  Since phone passwords can only be 20 digits in length, the stored procedure assumes that if the string is 32 characters long it's already been hashed into an MD5 string and will pass it through directly.  Otherwise it'll hash it for you on the back end so you can simply pass through the raw digits for the phone password you want to set and it'll take care of it for you.  If you're paranoid you can also hash the phone password into it's MD5 string and then pass that into the stored procedure instead.  It'll handle it either way.

 

When setting the password for a subscriber it's up to you to be sure it complies with the password policy for the site.  If you pass in a blank string for the password, for instance, the phone password will be cleared.  You can check the phone password policies in the PwPolicy table in UnityDB.  The stored procedure does not do any enforcement of the site's policy, so make sure you're careful here.

 

    oCommand.CommandText = "sp_ModifySubscriber"

 

    oCommand.Parameters.Item("@SubscriberObjectID") = oCommand.CreateParameter("SubscriberObjectID", adGUID, adParamInput, , strSubscriberObjectID)

 

    'The phone password is passed in as clear text in this example - it gets crunched and encrypted by the SP during the subscriber update process.  You can pass in a "pre hashed" MD5 string yourself - if the string is exactly 32 characters long the stored proc will pass it through "as is".

    oCommand.Parameters.Item("@PWDTMF") = oCommand.CreateParameter("PWDTMF", adVarChar, adParamInput, , txtPhonePassword.Text)

    

'Passwords are not written through to the directory, no need to synch.

        oCommand.Parameters.Item("@DirectorySync") = oCommand.CreateParameter("DirectorySync", adInteger, adParamInput, , 0)

    

    'Let the stored procedure call rip.

    oCommand.Execute

 

Since phone passwords are not pushed into the directory there's no need to issue a synchronization request for this.  You're done.

	----- Original Message ----- 
	From: Jim McBurnett <mailto:jim at tgasolutions.com>  
	To: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  ; Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  ; cisco-voip at puck.nether.net 
	Cc: Alan Surrette <mailto:alan.surrette at asheville.k12.nc.us>  
	Sent: Saturday, September 16, 2006 10:42 PM
	Subject: RE: [cisco-voip] Reset Unity mailbox password..

	Take a look over at www.ciscounitytools.com
	 
	There are some architecture notes there...
	 
	J

________________________________

	From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
	Sent: Saturday, September 16, 2006 9:19 PM
	To: Lelio Fulgenzi; cisco-voip at puck.nether.net
	Cc: Alan Surrette
	Subject: Re: [cisco-voip] Reset Unity mailbox password..
	
	
	You know, now that  I think about it--- hmm...
	 
	I'll have to check around in AD on Unity and see where they've embedded the password in some AD field-- I'm betting that the mailbox password is in there somewhere..
	 
	We have some internal experience in scripting AD for our main domain --- if it's there in AD in some schema extended field... hmmm..
	 
	time to break out that LDAP browser that someone posted a link to - I did manage to get that to connect to Unity's AD, though not our data AD or the CCM DCDirectory..
	 
	 

________________________________

	From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
	Sent: Sat 9/16/2006 8:18 PM
	To: Tim Reimers; cisco-voip at puck.nether.net
	Cc: Alan Surrette
	Subject: Re: [cisco-voip] Reset Unity mailbox password..
	
	
	I have not seen anything like this. Although it would make for a great utility, say send their reset password to their email address. I cannot find any reference to an API like SOAP for unity, that would have helped immensely. You can try posting in the Cisco forums, I'm pretty sure Jeff Lindborgh reads that. Jeff was the primary developer for Unity and I believe he still is.
	 
	If your users have the domain password available to them, check out the PCA login page. It's a little different than the web/sa pages and might fit your needs.
	 

		----- Original Message ----- 
		From: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  
		To: cisco-voip at puck.nether.net 
		Cc: Alan Surrette <mailto:alan.surrette at asheville.k12.nc.us>  
		Sent: Saturday, September 16, 2006 12:08 PM
		Subject: [cisco-voip] Reset Unity mailbox password..

		 
		hello everyone--
		 
		I'm looking for a way to allow a subscribers Unity mailbox password to be reset in a way OTHER than through the Unity webpages (and TermServ to Unity)
		 
		I really really need to limit the access of people to full Unity to just a few..
		.. but at the same time, those few of us with full access CANNOT keep up with the requests for resetting of passwords. (I know, you'd think that's a user education thing..)
		 
		So -- I'm hoping for something like Greetings Administrator to have rights to reset passwords on any mailbox -- similar to how the person using GA can select any call handler and work with it..
		I'm hoping there's a level of access I can grant to a  additional staff members at remote sites just to be able to clear and reset a password using their phone and something like Greetings Administrator?
		 
		If that's not clear, just email me and I'll try to confuse you some more... ;-)
		 
		thanks, Tim

		
________________________________


		

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

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From jgrace at digitel.net  Sun Sep 17 22:23:14 2006
From: jgrace at digitel.net (James Grace)
Date: Sun, 17 Sep 2006 22:23:14 -0400
Subject: [cisco-voip] add the prefix on missed and incoming call on CM
Message-ID: <200609172222859.SM01512@jgm>

I trying to configure cm to add prefix  to the incoming and missed calls in
our directory on cm..   I went thu the doc on how to configure it.  But do
any one know how to add one for incoming local and LD.  It seems that I can
only create a translation pattern for one or the other.   

 

James D Grace 

CCNP CCNA MCSE MCDBA

System Engineer / Professional Svc.

Digitel Corp.

Email:  <mailto:grace.jg at gmail> grace.jg at gmail 

email: jgrace at digitel.net 

MSN id: grace.jd at gmail.com

 

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From erickbe at yahoo.com  Sun Sep 17 22:52:52 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sun, 17 Sep 2006 19:52:52 -0700 (PDT)
Subject: [cisco-voip] add the prefix on missed and incoming call on CM
In-Reply-To: <200609172222859.SM01512@jgm>
Message-ID: <20060918025252.50222.qmail@web37210.mail.mud.yahoo.com>

Well, in CCM it is more of all or nothing thing since there isn't really a good way to filter and decide what to add what prefixes to. If you have a MGCP Gateway and Q.931 (PRI) interface you can set the service parameters and what prefix you want for national/unknown/subscriber and international number types. But then this also depends solely on the numbers coming in with that number plan so it can match right, and still no way to do any filter or match logic in call manager at moment so there will be hits and misses still. 

To get full control of this, a H.323 gateway and translation rules/profiles would work better. 
There may be another cisco application that may work better or a third party app. 


----- Original Message ----
From: James Grace <jgrace at digitel.net>
To: cisco-voip at puck.nether.net
Sent: Sunday, September 17, 2006 9:23:14 PM
Subject: [cisco-voip] add the prefix on missed and incoming call on CM

             I trying to configure cm to add prefix  to the incoming and missed calls in our directory on cm..   I went thu the doc on how to configure it.  But do any one know how to add one for incoming local and LD.  It seems that I can only create a translation pattern for one or the other.   
     
   James D Grace 
   CCNP CCNA MCSE MCDBA
   System Engineer / Professional Svc.
   Digitel Corp.
   Email: grace.jg at gmail 
   email: jgrace at digitel.net 
   MSN id: grace.jd at gmail.com
     
   
     _______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




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From Matt.Slaga at us.didata.com  Mon Sep 18 07:23:04 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 18 Sep 2006 07:23:04 -0400
Subject: [cisco-voip] Reset Unity mailbox password..
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032C9DAF@USNAEXCH.na.didata.local>

For our customers, we create a "Help Desk" COS in Unity and assign very
granular access to things like changing passwords.  When they log into
the Unity SA, they only see the options specified.

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Sunday, September 17, 2006 9:40 PM
To: Lelio Fulgenzi; Jim McBurnett; cisco-voip at puck.nether.net
Cc: Kelly Looney; Alan Surrette
Subject: Re: [cisco-voip] Reset Unity mailbox password..

 

Cool! 

 

Thanks...

 

that's the biggest request we get -- other things, I think we can
continue to handle internally..

 

It may take the skill of our app developer to do this, not me.. .but
that does look as though it's do-able...

 

thanks guys..

Tim

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
Sent: Sat 9/16/2006 11:31 PM
To: Jim McBurnett; Tim Reimers; cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: Re: [cisco-voip] Reset Unity mailbox password..

Thanks Jim. Looks like there might be something there that Tim can use:
http://www.ciscounitytools.com/Documents/programaticAdmin.doc

 

 


Update subscriber's password


Setting the password is easy going through the stored procedure since it
will take care of hashing your raw phone password into an MD5 string for
you on the back end.  MD5 strings should always be exactly 32 characters
here.  Since phone passwords can only be 20 digits in length, the stored
procedure assumes that if the string is 32 characters long it's already
been hashed into an MD5 string and will pass it through directly.
Otherwise it'll hash it for you on the back end so you can simply pass
through the raw digits for the phone password you want to set and it'll
take care of it for you.  If you're paranoid you can also hash the phone
password into it's MD5 string and then pass that into the stored
procedure instead.  It'll handle it either way.

 

When setting the password for a subscriber it's up to you to be sure it
complies with the password policy for the site.  If you pass in a blank
string for the password, for instance, the phone password will be
cleared.  You can check the phone password policies in the PwPolicy
table in UnityDB.  The stored procedure does not do any enforcement of
the site's policy, so make sure you're careful here.

 

    oCommand.CommandText = "sp_ModifySubscriber"

 

    oCommand.Parameters.Item("@SubscriberObjectID") =
oCommand.CreateParameter("SubscriberObjectID", adGUID, adParamInput, ,
strSubscriberObjectID)

 

    'The phone password is passed in as clear text in this example - it
gets crunched and encrypted by the SP during the subscriber update
process.  You can pass in a "pre hashed" MD5 string yourself - if the
string is exactly 32 characters long the stored proc will pass it
through "as is".

    oCommand.Parameters.Item("@PWDTMF") =
oCommand.CreateParameter("PWDTMF", adVarChar, adParamInput, ,
txtPhonePassword.Text)

    

'Passwords are not written through to the directory, no need to synch.

        oCommand.Parameters.Item("@DirectorySync") =
oCommand.CreateParameter("DirectorySync", adInteger, adParamInput, , 0)

    

    'Let the stored procedure call rip.

    oCommand.Execute

 

Since phone passwords are not pushed into the directory there's no need
to issue a synchronization request for this.  You're done.

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

	From: Jim McBurnett <mailto:jim at tgasolutions.com>  

	To: Tim Reimers <mailto:tim.reimers at asheville.k12.nc.us>  ;
Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  ; cisco-voip at puck.nether.net 

	Cc: Alan Surrette <mailto:alan.surrette at asheville.k12.nc.us>  

	Sent: Saturday, September 16, 2006 10:42 PM

	Subject: RE: [cisco-voip] Reset Unity mailbox password..

	 

	Take a look over at www.ciscounitytools.com

	 

	There are some architecture notes there...

	 

	J

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
	Sent: Saturday, September 16, 2006 9:19 PM
	To: Lelio Fulgenzi; cisco-voip at puck.nether.net
	Cc: Alan Surrette
	Subject: Re: [cisco-voip] Reset Unity mailbox password..

	You know, now that  I think about it--- hmm...

	 

	I'll have to check around in AD on Unity and see where they've
embedded the password in some AD field-- I'm betting that the mailbox
password is in there somewhere..

	 

	We have some internal experience in scripting AD for our main
domain --- if it's there in AD in some schema extended field... hmmm..

	 

	time to break out that LDAP browser that someone posted a link
to - I did manage to get that to connect to Unity's AD, though not our
data AD or the CCM DCDirectory..

	 

	 

	 

	
________________________________


	From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
	Sent: Sat 9/16/2006 8:18 PM
	To: Tim Reimers; cisco-voip at puck.nether.net
	Cc: Alan Surrette
	Subject: Re: [cisco-voip] Reset Unity mailbox password..

	I have not seen anything like this. Although it would make for a
great utility, say send their reset password to their email address. I
cannot find any reference to an API like SOAP for unity, that would have
helped immensely. You can try posting in the Cisco forums, I'm pretty
sure Jeff Lindborgh reads that. Jeff was the primary developer for Unity
and I believe he still is.

	 

	If your users have the domain password available to them, check
out the PCA login page. It's a little different than the web/sa pages
and might fit your needs.

	 

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

		From: Tim Reimers
<mailto:tim.reimers at asheville.k12.nc.us>  

		To: cisco-voip at puck.nether.net 

		Cc: Alan Surrette
<mailto:alan.surrette at asheville.k12.nc.us>  

		Sent: Saturday, September 16, 2006 12:08 PM

		Subject: [cisco-voip] Reset Unity mailbox password..

		 

		 

		hello everyone--

		 

		I'm looking for a way to allow a subscribers Unity
mailbox password to be reset in a way OTHER than through the Unity
webpages (and TermServ to Unity)

		 

		I really really need to limit the access of people to
full Unity to just a few..

		.. but at the same time, those few of us with full
access CANNOT keep up with the requests for resetting of passwords. (I
know, you'd think that's a user education thing..)

		 

		So -- I'm hoping for something like Greetings
Administrator to have rights to reset passwords on any mailbox --
similar to how the person using GA can select any call handler and work
with it..

		I'm hoping there's a level of access I can grant to a
additional staff members at remote sites just to be able to clear and
reset a password using their phone and something like Greetings
Administrator?

		 

		If that's not clear, just email me and I'll try to
confuse you some more... ;-)

		 

		thanks, Tim

		
________________________________


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




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From ErasmuE4 at telkom.co.za  Mon Sep 18 07:45:28 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Mon, 18 Sep 2006 13:45:28 +0200
Subject: [cisco-voip] auto QoS for Ip telephony solution
Message-ID: <472E195E692EE74BAE4E7B2880A78C1FAC17ED@CNTRRA20-XCS01.telkom.co.za>


Hi



Does any one have good experience on the use of auto QoS in an IP
Telephony solution. The scenario is as follows. Centralised call
manager and remote branches with IP phones. Locations, regions etc.
All working fine. At remote branches we have a voice gateway with a
trunk to the local lan switch (two subinterfaces - one for data vlan
and one for voice vlan)



Currently at branches we use auto-QoS VoIP cisco-phone on all the 3560
series switchports connection to cisco phones and the same at HQ on
the 4500 series switches.



1.	I want to know what is recommended for the trunk interface on
the switch between the branch gateway and the lan switch on the 802.1Q
trunk and also on the router main / subinterfaces.
2.	Also what is recommended on switch ports connecting to the
Cisco Call Manager at HQ. Is it auto QoS VoIP trust ???



The auto QoS on the phone ports are working fine and I would prefer to
use auto-QoS for the additional links too where possible, excluding
the WAN links on the MPLS cloud where we have configured QoS manually.
Mainly because it is easier and consistent.







Erik Erasmus

Telkom SA

': +27 12 680 8320

*: +27 12 680 7388

*: +27 82 4106420

*:  erasmue4 at telkom.co.za



      





~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From ealeatherman at gmail.com  Mon Sep 18 08:45:31 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 18 Sep 2006 08:45:31 -0400
Subject: [cisco-voip] Best Practices
In-Reply-To: <00a501c6dac2$d2052160$25fb14ac@gpatelt>
References: <00a501c6dac2$d2052160$25fb14ac@gpatelt>
Message-ID: <94a1afde0609180545x2f81569se8dfde70802b6425@mail.gmail.com>

Hi Gary,

I think its a good idea to get 10 digits from the LEC, then you alway have
more numbers to work with. At that point you could take more than 3 digits
as significant and process them through a translation pattern to get them to
the extension you want.

For example, if the called number from LEC is XXX-593-3183, take 4 digits as
significant and have a 3XXX translation pattern intercept the calls from
your gateway and pass them to the correct partition/phone. You could have a
4XXX pattern catch calls to 593-4183 etc. If you still run into problems you
could take more sig digits and make the pattern more specific to the
telephon exchange or something.

Is there a requirement that you can only have significant digits set to 3?

Hope that helps/makes sense on monday morning.

On 9/17/06, Gary L. Pate <gpate at kenttech.com> wrote:
>
>  I'm trying to determine a "best practices" when setting up inbound DID's,
> extensions, etc. Let me explain:
>
>
>
> In most situations I've been involved in, I've used the last 4 digits of a
> DID as someone's extension. I make sure the LEC sends 10 digit's, and I do a
> significant digits of 4 on the gateway. This has always worked well, and is
> easy to configure. However, in a recent install, this customer was using 3
> digit extensions, so I again had the LEC send 10 digits, and I made the
> gateway significant digits of 3. Again, this worked, but the problem was
> that I started running into to many situations where I had duplicate 3
> digits. For example, a DID might be 593-3183, so the phone extension was
> 183. But I also had a existing fax line number that was 444-4183. As you can
> see, the last 3 digits are the same. I could not make a translation pattern
> of 444183 because the gateway was only sending 3 digits. I ended up having
> the LEC forward the 4444183 to one of the DID ranges, and then gave the fax
> the number from that range, but that is not what I want to do in the future.
>
>
> I guess what I want to know is what is the "best practice" going forward?
> Is it a good idea to have the LEC send 10 digits, and if so, what should I
> set on the gateway for significant digit's, and how should I handle my
> internal phones that are using the last four digits of the DID as their
> extension? I've read a document on using translation patterns to prefix
> digits, etc, but I'm not sure if that is the directions I should go.
>
> Can someone offer advice on this, and perhaps provide an example of how to
> do this with both a 4 digit example, and a 3 digit example? I really want to
> have more control over the lines, and not have to worry much about duplicate
> numbers or having the LEC getting involved.
>
>
>
> Gary
>
-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Mon Sep 18 08:52:30 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 18 Sep 2006 08:52:30 -0400
Subject: [cisco-voip] Best Practices
References: <00a501c6dac2$d2052160$25fb14ac@gpatelt>
	<94a1afde0609180545x2f81569se8dfde70802b6425@mail.gmail.com>
Message-ID: <003a01c6db21$4d8ba940$47196883@cfs.uoguelph.ca>

I would have to agree. We request 7 digits from the carrier now, but will likely need to change this to 10 as we have recently had a new area code introduced. As you know, any change on the carrier side is big $$, so by sending you all the digits, you can do just about anything you want.

Thanks goodness they don't charge you per digit sent. ;)

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Ed Leatherman 
  To: GPate at kenttech.com 
  Cc: cisco-voip at puck.nether.net 
  Sent: Monday, September 18, 2006 8:45 AM
  Subject: Re: [cisco-voip] Best Practices


  Hi Gary,

  I think its a good idea to get 10 digits from the LEC, then you alway have more numbers to work with. At that point you could take more than 3 digits as significant and process them through a translation pattern to get them to the extension you want. 

  For example, if the called number from LEC is XXX-593-3183, take 4 digits as significant and have a 3XXX translation pattern intercept the calls from your gateway and pass them to the correct partition/phone. You could have a 4XXX pattern catch calls to 593-4183 etc. If you still run into problems you could take more sig digits and make the pattern more specific to the telephon exchange or something. 

  Is there a requirement that you can only have significant digits set to 3?

  Hope that helps/makes sense on monday morning.


  On 9/17/06, Gary L. Pate <gpate at kenttech.com> wrote:
    I'm trying to determine a "best practices" when setting up inbound DID's, extensions, etc. Let me explain:



    In most situations I've been involved in, I've used the last 4 digits of a DID as someone's extension. I make sure the LEC sends 10 digit's, and I do a significant digits of 4 on the gateway. This has always worked well, and is easy to configure. However, in a recent install, this customer was using 3 digit extensions, so I again had the LEC send 10 digits, and I made the gateway significant digits of 3. Again, this worked, but the problem was that I started running into to many situations where I had duplicate 3 digits. For example, a DID might be 593-3183, so the phone extension was 183. But I also had a existing fax line number that was 444-4183. As you can see, the last 3 digits are the same. I could not make a translation pattern of 444183 because the gateway was only sending 3 digits. I ended up having the LEC forward the 4444183 to one of the DID ranges, and then gave the fax the number from that range, but that is not what I want to do in the future. 

    I guess what I want to know is what is the "best practice" going forward? Is it a good idea to have the LEC send 10 digits, and if so, what should I set on the gateway for significant digit's, and how should I handle my internal phones that are using the last four digits of the DID as their extension? I've read a document on using translation patterns to prefix digits, etc, but I'm not sure if that is the directions I should go. 

    Can someone offer advice on this, and perhaps provide an example of how to do this with both a 4 digit example, and a 3 digit example? I really want to have more control over the lines, and not have to worry much about duplicate numbers or having the LEC getting involved.



    Gary

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


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


  _______________________________________________
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From ealeatherman at gmail.com  Mon Sep 18 08:55:36 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 18 Sep 2006 08:55:36 -0400
Subject: [cisco-voip] FW: Cannot leave voice mail from internal phone
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0032810B5@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0032810B5@USNAEXCH.na.didata.local>
Message-ID: <94a1afde0609180555h6ead898eob8c6d29878f1c50f@mail.gmail.com>

Hi Jason,

We had this sort of problem directly after we installed MS Exchange 2k3
hotfix 903158. When we uninstalled that hotfix the users who could not get
voicemail worked fine, much to their disappointment. Not sure if that helps
but I figured i'd toss it out there.

Ed

On 9/15/06, Jason Aarons (US) <jason.aarons at us.didata.com> wrote:
>
> We cannot leave voicemails for end users from certain phones. Any ideas?
> Unity 4.0(4).
>
>
> Event Type: Error
> Event Source: CiscoUnity_ConvMsg
> Event Category: Network
> Event ID: 10015
> Date:  9/15/2006
> Time:  1:15:46 PM
> User:  N/A
> Computer: ITIT-UNITY01
> Description:
> IAvDohMessage::Submit returned [0x800404dc] on line 683 of file
> e:\views\cs_ue4.0.3.159\un_Core2\ConversationEng\AvStateSvr\AvSMsgSend.c
> pp
> For more information, click: http://www.CiscoUnitySupport.com/find.php
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
>
> _______________________________________________
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> cisco-voip at puck.nether.net
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-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From bloemhard_johan at silverfalls.k12.or.us  Mon Sep 18 10:05:37 2006
From: bloemhard_johan at silverfalls.k12.or.us (Johan Bloemhard)
Date: Mon, 18 Sep 2006 07:05:37 -0700
Subject: [cisco-voip] H.323 Gateway Security - Prevent Unauthorized Usage
Message-ID: <C763DC29A34B6B418AFBA81F75B2633A027272@msexch2.silverfalls.k12.or.us>

We have a 2801 with a H.323 gateway (PSTN).  I just came across a
student that was using IP Blue (VTGO) on their handheld and had
configured it using our H.323 gateway.  I imagine that he got the IP of
the H.323 gateway off of a phone.  Obviously he didn't have internal
dialing but could dial out with a problem.

 

How can we prevent this?  ACLs on the 2801?  How are other organizations
tackling this problem?

 

TIA

 

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

 


______________________
Disclaimer: This message may be legally privileged and/or is intended only for the use of the addressee(s).  The content and views expressed in this email may represent the views of the sender and not those of Silver Falls School District. 

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From ealeatherman at gmail.com  Mon Sep 18 10:29:41 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 18 Sep 2006 10:29:41 -0400
Subject: [cisco-voip] auto QoS for Ip telephony solution
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1FAC17ED@CNTRRA20-XCS01.telkom.co.za>
References: <472E195E692EE74BAE4E7B2880A78C1FAC17ED@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <94a1afde0609180729h653ce374k88a32401a5c8834d@mail.gmail.com>

Hi Erik,

I'm no qos expert by any stretch, this is just what we do and it seems to
work out for us.. traffic markings come out end-to-end how we want them to
at least. If someone else sees something wrong with it i'd love to hear it.

I've been using "auto qos voip trust" on both those instances, i've been
adding to that "mls qos trust dscp" to the callmanager ports and layer 3
links, because the auto trust statement seems to put mls qos trust cos
instead. To my understanding the callmanager servers mark their traffic with
the dscp bits.

Ed

On 9/18/06, Erik Erasmus (E) <ErasmuE4 at telkom.co.za> wrote:
>
>
> Currently at branches we use auto-QoS VoIP cisco-phone on all the 3560
> series switchports connection to cisco phones and the same at HQ on the 4500
> series switches.
>
>
>
>    1. I want to know what is recommended for the trunk interface on the
>    switch between the branch gateway and the lan switch on the 802.1Qtrunk and also on the router main / subinterfaces.
>    2. Also what is recommended on switch ports connecting to the Cisco
>    Call Manager at HQ. Is it auto QoS VoIP trust ???
>
>
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From akoul at dataways.gr  Mon Sep 18 10:31:52 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Mon, 18 Sep 2006 17:31:52 +0300
Subject: [cisco-voip] ***SPAM***Re: Can IPMA service work with no
 Partitions/CSS cinfigured
In-Reply-To: <20060915144634.32775.qmail@web53611.mail.yahoo.com>
References: <20060915144634.32775.qmail@web53611.mail.yahoo.com>
Message-ID: <450EADD8.1090809@dataways.gr>

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From rratliff at cisco.com  Mon Sep 18 10:33:12 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 18 Sep 2006 10:33:12 -0400
Subject: [cisco-voip] Can IPMA service work with no Partitions/CSS
	cinfigured
In-Reply-To: <450AB10E.1020009@dataways.gr>
References: <450AB10E.1020009@dataways.gr>
Message-ID: <60FF5428-50FC-4795-B216-9285DFD0276A@cisco.com>

Not if you want the IPMA route point to intercept calls for the Manager.

You can however put your IPMA RP into the None partition and have the  
CSS applied only to it.

-Ryan

On Sep 15, 2006, at 9:56 AM, Anthony Kouloglou wrote:

Imagine a big installation where it was never desired to have partitions
and CSS.
Can we now implement the IPMA omiting the part of the configuration of
partitions/CSS. To leave these fields blank for all,
lines, managers, assistants, everyone and route points?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From MLinsemier at apcapital.com  Mon Sep 18 10:36:53 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Mon, 18 Sep 2006 10:36:53 -0400
Subject: [cisco-voip] auto QoS for Ip telephony solution
In-Reply-To: <94a1afde0609180729h653ce374k88a32401a5c8834d@mail.gmail.com>
Message-ID: <AB0A35F2486F81468971AA68558BE71904CEFC2F@smieln01winm01.micoanet.com>

This is a good topic for discussion.  I posted to this list a while back
about which Cisco software packages, specifically their VoIP software
line, does and does not mark traffic before sending it on to the switch.
Because I have never received a firm answer, I always thought it to be a
best practice to mark / re-mark this traffic to be sure.  If there was a
definitive with version numbers list I may default to trusting DSCP on
the switches using Auto-QoS.  Until then, as long as re-marking the
traffic doesn't have significant overhead, I don't see the harm unless
your QoS remarking was misconfigured.

 

Any of you Cisco guys want to shed some light on this?

 

-Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Monday, September 18, 2006 10:30 AM
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] auto QoS for Ip telephony solution

 

Hi Erik,

I'm no qos expert by any stretch, this is just what we do and it seems
to work out for us.. traffic markings come out end-to-end how we want
them to at least. If someone else sees something wrong with it i'd love
to hear it. 

I've been using "auto qos voip trust" on both those instances, i've been
adding to that "mls qos trust dscp" to the callmanager ports and layer 3
links, because the auto trust statement seems to put mls qos trust cos
instead. To my understanding the callmanager servers mark their traffic
with the dscp bits. 

Ed

On 9/18/06, Erik Erasmus (E) <ErasmuE4 at telkom.co.za> wrote:

 

Currently at branches we use auto-QoS VoIP cisco-phone on all the 3560
series switchports connection to cisco phones and the same at HQ on the
4500 series switches. 

 

1.	I want to know what is recommended for the trunk interface on
the switch between the branch gateway and the lan switch on the 802.1Q
trunk and also on the router main / subinterfaces. 
2.	Also what is recommended on switch ports connecting to the Cisco
Call Manager at HQ. Is it auto QoS VoIP trust ???

 




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


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

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From Matt.Slaga at us.didata.com  Mon Sep 18 10:43:59 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 18 Sep 2006 10:43:59 -0400
Subject: [cisco-voip] H.323 Gateway Security - Prevent Unauthorized Usage
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032CA1E8@USNAEXCH.na.didata.local>

ACLs on the router specifying H323 control traffic to/from the
callmanager servers (and gatekeepers if applicable) would be all that is
needed to prevent this.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Johan Bloemhard
Sent: Monday, September 18, 2006 10:06 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] H.323 Gateway Security - Prevent Unauthorized
Usage

 

We have a 2801 with a H.323 gateway (PSTN).  I just came across a
student that was using IP Blue (VTGO) on their handheld and had
configured it using our H.323 gateway.  I imagine that he got the IP of
the H.323 gateway off of a phone.  Obviously he didn't have internal
dialing but could dial out with a problem.

 

How can we prevent this?  ACLs on the 2801?  How are other organizations
tackling this problem?

 

TIA

 

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

 

________________________________

Disclaimer: This message may be legally privileged and/or is intended
only for the use of the addressee(s). The content and views expressed in
this email may represent the views of the sender and not those of Silver
Falls School District. 




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From rlatorre at unislumin.com  Mon Sep 18 11:00:26 2006
From: rlatorre at unislumin.com (Ryan LaTorre)
Date: Mon, 18 Sep 2006 11:00:26 -0400
Subject: [cisco-voip] The CMM and SPAN
Message-ID: <18082B35CF0EBD4C8D265874508A615C03A8DAC7@hermes.unislumin.com>


The other day I encountered what appears to be a bug, but I could not
locate anything in the bug navigator.

Trying to SPAN the backplane port of a CMM module in a Cat 6500 with
Sup720, for the purposes of IPCCx recording/monitoring (in 3.x).  The
SPAN configuration is very straight forward (I believe) but I found that
when I SPAN both directions I receive only the transmit packets,
resulting in only recording one side of the conversation.  I confirmed
this by changing the SPAN source to receive only and NO packets were
mirrored whatsoever.  From what I can tell, it seems I can't SPAN
packets coming from the CMM.

Are there any SPAN/CMM caveats I should be aware of, or does this really
sound like a defect? I didn't see anything in release notes or guides.

Thanks,
Ryan

Attention:

Privileged/Confidential Information may be contained in this message.  Disclosure to any person other than the named recipient is unauthorized.  If you are not the intended recipient, please delete all copies of this information and kindly notify the sender by reply email.  Opinions, conclusions and other information in this message that do not relate to the official business of UNIS LUMIN Inc. shall be understood as neither given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries reserve the right to monitor all e-mail communications through its networks.  Thank you.
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From Scott.Voll at wesd.org  Mon Sep 18 11:17:30 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 18 Sep 2006 08:17:30 -0700
Subject: [cisco-voip] The CMM and SPAN
Message-ID: <407055A92CECCB499C922A2D35FC19A606933DA6@Apollo.wesd.org>

Is the layer three vlan interface on the Cat 6509?  If so, try to SPAN
the layer 3 interface rather then the CMM port.

 

What version of IOS are you using on the 65xx and the CMM?

 

I'm using 12.2(18)SXF4 and 12.4(7a) without problem

 

SPAN Config:

 

Vlan 102 is my Core Voice VLAN and 202 is my RSPAN

 

monitor session 2 source vlan 102

monitor session 2 destination remote vlan 202

monitor session 4 destination interface Fa3/44 <-- IPCCx Monitor server

monitor session 4 source remote vlan 202

 

 

Sh mon sess all

 

Session 2

---------

Type                   : Remote Source Session

Source VLANs           :

    Both               : 102

Destination RSPAN VLAN : 202

 

Session 4

---------

Type                   : Remote Destination Session

Source RSPAN VLAN      : 202

Destination Ports      : Fa3/44

 

Scott

 

PS.  All your agents are on the same vlan as your IPCCx server, right?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan LaTorre
Sent: Monday, September 18, 2006 8:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] The CMM and SPAN

 

 

The other day I encountered what appears to be a bug, but I could not
locate anything in the bug navigator. 

Trying to SPAN the backplane port of a CMM module in a Cat 6500 with
Sup720, for the purposes of IPCCx recording/monitoring (in 3.x).  The
SPAN configuration is very straight forward (I believe) but I found that
when I SPAN both directions I receive only the transmit packets,
resulting in only recording one side of the conversation.  I confirmed
this by changing the SPAN source to receive only and NO packets were
mirrored whatsoever.  From what I can tell, it seems I can't SPAN
packets coming from the CMM.

Are there any SPAN/CMM caveats I should be aware of, or does this really
sound like a defect? I didn't see anything in release notes or guides.

Thanks, 
Ryan 

________________________________

Privileged/Confidential Information may be contained in this message.
Disclosure to any person other than the named recipient is unauthorized.
If you are not the intended recipient, please delete all copies of this
information and kindly notify the sender by reply email.  Opinions,
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the official business of UNIS LUMIN Inc. shall be understood as neither
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From MLinsemier at apcapital.com  Mon Sep 18 11:17:44 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Mon, 18 Sep 2006 11:17:44 -0400
Subject: [cisco-voip] The CMM and SPAN
In-Reply-To: <18082B35CF0EBD4C8D265874508A615C03A8DAC7@hermes.unislumin.com>
Message-ID: <AB0A35F2486F81468971AA68558BE71904CEFC77@smieln01winm01.micoanet.com>

Ryan,

 

As a alternative, have you tried spanning one of the VLAN ports that
your phones are on (another method that can be used for
recording/monitoring)?  We utilize CMM's as well and if I remember I had
a similar issue back in the day when I initially tried configuring
spanning of these ports.  I ended up segmenting all of the phones
utilized for our CAC (where we needed recording/montiroing) into their
own VLAN and then used RSPAN to span both directions back to the IPCC
Express server. You could do this as a test to make sure that it is
definitely a CMM issue and not something else

 

That being said, if you have SMARTNet contract on IPCC Express, 4.x
doesn't require the spanning of any ports if you are using Agent
Desktop.

 

-Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan LaTorre
Sent: Monday, September 18, 2006 11:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] The CMM and SPAN

 

 

The other day I encountered what appears to be a bug, but I could not
locate anything in the bug navigator. 

Trying to SPAN the backplane port of a CMM module in a Cat 6500 with
Sup720, for the purposes of IPCCx recording/monitoring (in 3.x).  The
SPAN configuration is very straight forward (I believe) but I found that
when I SPAN both directions I receive only the transmit packets,
resulting in only recording one side of the conversation.  I confirmed
this by changing the SPAN source to receive only and NO packets were
mirrored whatsoever.  From what I can tell, it seems I can't SPAN
packets coming from the CMM.

Are there any SPAN/CMM caveats I should be aware of, or does this really
sound like a defect? I didn't see anything in release notes or guides.

Thanks, 
Ryan 

________________________________

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


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

CONFIDENTIALITY STATEMENT

This communication and any attachments are CONFIDENTIAL and may be 
protected by one or more legal privileges. It is intended solely
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the intended recipient, any use, disclosure, copying or 
distribution of this communication is UNAUTHORIZED. Neither this 
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From aman.chugh at gmail.com  Mon Sep 18 11:21:31 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Mon, 18 Sep 2006 20:51:31 +0530
Subject: [cisco-voip] H.323 Gateway Security - Prevent Unauthorized Usage
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0032CA1E8@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0032CA1E8@USNAEXCH.na.didata.local>
Message-ID: <d4c697ad0609180821t7ed5d679g3512c75915012bda@mail.gmail.com>

You could also try to block this particular subnet/IP address (IP BLUE) to
communicate with Callmanager on TCP port 1720,1719 using a extended ACL.

Aman


On 9/18/06, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
>
>   ACLs on the router specifying H323 control traffic to/from the
> callmanager servers (and gatekeepers if applicable) would be all that is
> needed to prevent this.
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Johan Bloemhard
> *Sent:* Monday, September 18, 2006 10:06 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] H.323 Gateway Security - Prevent Unauthorized
> Usage
>
>
>
> We have a 2801 with a H.323 gateway (PSTN).  I just came across a student
> that was using IP Blue (VTGO) on their handheld and had configured it using
> our H.323 gateway.  I imagine that he got the IP of the H.323 gateway off
> of a phone.  Obviously he didn't have internal dialing but could dial out
> with a problem.
>
>
>
> How can we prevent this?  ACLs on the 2801?  How are other organizations
> tackling this problem?
>
>
>
> TIA
>
>
>
> Johan Bloemhard
> Senior Information Technology Specialist
> ___________________________________
> *Silver** Falls School District** 4J*
> 802 Schlador Street
> Silverton, OR 97381
> w: 503.873.6331
>
>
>
> ------------------------------
> *Disclaimer*: This message may be legally privileged and/or is intended
> only for the use of the addressee(s). The content and views expressed in
> this email may represent the views of the sender and not those of Silver
> Falls School District.
>
>   ------------------------------
>
>
> *
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful. If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
> *
>
> _______________________________________________
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>
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>
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From rlatorre at unislumin.com  Mon Sep 18 11:35:10 2006
From: rlatorre at unislumin.com (Ryan LaTorre)
Date: Mon, 18 Sep 2006 11:35:10 -0400
Subject: [cisco-voip] The CMM and SPAN
Message-ID: <18082B35CF0EBD4C8D265874508A615C03A8DAEB@hermes.unislumin.com>

Thanks for the ideas guys, I should've mentioned that I did implement a
successful workaround, so I do have recording operational right now. I
just don't like leaving things like this unresolved! Also, IPCCx 4.x
upgrade is in the planning stages, but I'm just trying to figure this
out because it seems like the switch is not operating as it should. I
admit it could be my config, but it's only two lines!
 
My config (that doesn't quite work):
monitor session 2 source interface Gi4/1 both (the CMM backplane port)
monitor session 2 destination interface Gig8/1 (the IPCCx monitor port)
 
The reason we're not spanning the VLAN is because when we do that I get
two copies of each frame on the TX side, since currently the CMM and the
IP phones are in the same VLAN.  This is using 12.2(18)SXF3 on the s720
and 12.4(7a) on the CMM. Pretty close versions, that's for sure.  Maybe
it's not a bug...
 


________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, September 18, 2006 11:18 AM
To: Ryan LaTorre; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] The CMM and SPAN



Is the layer three vlan interface on the Cat 6509?  If so, try to SPAN
the layer 3 interface rather then the CMM port.

 

What version of IOS are you using on the 65xx and the CMM?

 

I'm using 12.2(18)SXF4 and 12.4(7a) without problem

 

SPAN Config:

 

Vlan 102 is my Core Voice VLAN and 202 is my RSPAN

 

monitor session 2 source vlan 102

monitor session 2 destination remote vlan 202

monitor session 4 destination interface Fa3/44 <-- IPCCx Monitor server

monitor session 4 source remote vlan 202

 

 

Sh mon sess all

 

Session 2

---------

Type                   : Remote Source Session

Source VLANs           :

    Both               : 102

Destination RSPAN VLAN : 202

 

Session 4

---------

Type                   : Remote Destination Session

Source RSPAN VLAN      : 202

Destination Ports      : Fa3/44

 

Scott

 

PS.  All your agents are on the same vlan as your IPCCx server, right?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan LaTorre
Sent: Monday, September 18, 2006 8:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] The CMM and SPAN

 

 

The other day I encountered what appears to be a bug, but I could not
locate anything in the bug navigator. 

Trying to SPAN the backplane port of a CMM module in a Cat 6500 with
Sup720, for the purposes of IPCCx recording/monitoring (in 3.x).  The
SPAN configuration is very straight forward (I believe) but I found that
when I SPAN both directions I receive only the transmit packets,
resulting in only recording one side of the conversation.  I confirmed
this by changing the SPAN source to receive only and NO packets were
mirrored whatsoever.  From what I can tell, it seems I can't SPAN
packets coming from the CMM.

Are there any SPAN/CMM caveats I should be aware of, or does this really
sound like a defect? I didn't see anything in release notes or guides.

Thanks, 
Ryan 

________________________________

Privileged/Confidential Information may be contained in this message.
Disclosure to any person other than the named recipient is unauthorized.
If you are not the intended recipient, please delete all copies of this
information and kindly notify the sender by reply email.  Opinions,
conclusions and other information in this message that do not relate to
the official business of UNIS LUMIN Inc. shall be understood as neither
given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries
reserve the right to monitor all e-mail communications through its
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From David.Fielder at us.didata.com  Mon Sep 18 12:25:45 2006
From: David.Fielder at us.didata.com (David Fielder (US))
Date: Mon, 18 Sep 2006 12:25:45 -0400
Subject: [cisco-voip] MGCP and FXS ports
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032CA426@USNAEXCH.na.didata.local>

I have a fax machine connected to an FXS port off my gateway. I have the
Gateway configured for MGCP. The port is configured in CallManager and
the DN for the fax machine has been added. When I call the fax either
from and external line or 4 digits I get fax tone. When they attempt to
make an outbound call from the fax machine, they get dial tone and it
will attempt to dial the digits and then immediately go to a fast busy.
If I do a show voice port summary, I see the port go off hook. I have
looked through the CallManager's logs and I don't think it ever reaches
the CM. I have configured an IP phone to emulate the port and DN
settings and I am able to make call internal and external. So it appears
that the call routing in CM is fine. All the fax machines on the FXS
ports are doing this. Has anyone ever seen this before?

 

David Fielder

Senior Engineer

Dimension Data

Office: (336) 510-9431

Mobile: (336) 669-5129

david.fielder at us.didata.com

 

 

 




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From Matt.Slaga at us.didata.com  Mon Sep 18 12:40:24 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 18 Sep 2006 12:40:24 -0400
Subject: [cisco-voip] H.323 Gateway Security - Prevent Unauthorized Usage
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032CA483@USNAEXCH.na.didata.local>

Problem is that IP blue is not making any connection or call dialog with
CallManager itself, it is the gateway that is being requested directly.
CallManager never knows that the call is being made.

 

 

 

________________________________

From: Aman Chugh [mailto:aman.chugh at gmail.com] 
Sent: Monday, September 18, 2006 11:22 AM
To: Matt Slaga (US)
Cc: Johan Bloemhard; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H.323 Gateway Security - Prevent Unauthorized
Usage

 

You could also try to block this particular subnet/IP address (IP BLUE)
to communicate with Callmanager on TCP port 1720,1719 using a extended
ACL.

 

Aman

 

On 9/18/06, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote: 

ACLs on the router specifying H323 control traffic to/from the
callmanager servers (and gatekeepers if applicable) would be all that is
needed to prevent this. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Johan Bloemhard
Sent: Monday, September 18, 2006 10:06 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] H.323 Gateway Security - Prevent Unauthorized
Usage

 

We have a 2801 with a H.323 gateway (PSTN).  I just came across a
student that was using IP Blue (VTGO) on their handheld and had
configured it using our H.323 gateway.  I imagine that he got the IP of
the H.323 gateway off of a phone.  Obviously he didn't have internal
dialing but could dial out with a problem.

 

How can we prevent this?  ACLs on the 2801?  How are other organizations
tackling this problem?

 

TIA

 

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

 

________________________________

Disclaimer: This message may be legally privileged and/or is intended
only for the use of the addressee(s). The content and views expressed in
this email may represent the views of the sender and not those of Silver
Falls School District. 

________________________________

 


Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the 
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notify us immediately by replying to this message and deleting it
from your computer. Thank you.


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



 




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From ErasmuE4 at telkom.co.za  Mon Sep 18 12:41:07 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Mon, 18 Sep 2006 18:41:07 +0200
Subject: [cisco-voip] auto QoS for Ip telephony solution
References: <472E195E692EE74BAE4E7B2880A78C1FAC17ED@CNTRRA20-XCS01.telkom.co.za>
	<94a1afde0609180729h653ce374k88a32401a5c8834d@mail.gmail.com>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D5DB@CNTRRA20-XCS01.telkom.co.za>


Thanks Ed

I think it sound like good advice - I was basically planning to try it
like you say.
If one does not have extensive resources it is a lot easier and
possibly good enough in most cases to use auto QoS.

thanks for your advice - will let the group know if during my
implementation I see something interseting.

________________________________

From: Ed Leatherman [mailto:ealeatherman at gmail.com]
Sent: Mon 2006-09-18 16:29
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] auto QoS for Ip telephony solution


Hi Erik,

I'm no qos expert by any stretch, this is just what we do and it seems
to work out for us.. traffic markings come out end-to-end how we want
them to at least. If someone else sees something wrong with it i'd
love to hear it.

I've been using "auto qos voip trust" on both those instances, i've
been adding to that "mls qos trust dscp" to the callmanager ports and
layer 3 links, because the auto trust statement seems to put mls qos
trust cos instead. To my understanding the callmanager servers mark
their traffic with the dscp bits.

Ed


On 9/18/06, Erik Erasmus (E) <ErasmuE4 at telkom.co.za> wrote:


	Currently at branches we use auto-QoS VoIP cisco-phone on all
the 3560 series switchports connection to cisco phones and the same at
HQ on the 4500 series switches.

	

	1.	I want to know what is recommended for the trunk
interface on the switch between the branch gateway and the lan switch
on the 802.1Q trunk and also on the router main / subinterfaces.
	2.	Also what is recommended on switch ports connecting to
the Cisco Call Manager at HQ. Is it auto QoS VoIP trust ???

	




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


~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From rlatorre at unislumin.com  Mon Sep 18 12:43:25 2006
From: rlatorre at unislumin.com (Ryan LaTorre)
Date: Mon, 18 Sep 2006 12:43:25 -0400
Subject: [cisco-voip] MGCP and FXS ports
Message-ID: <18082B35CF0EBD4C8D265874508A615C03A8DB1E@hermes.unislumin.com>

 
The first thing that comes to mind is the possibility that the fax
machines are not programmed to dial a 9 or other external access code
before the number.  This is a common issue when moving fax machines from
POTS lines to an inside system.  If this is it, you can either reprogram
the individual fax machines, or you could create a CSS/route pattern
that allows external dialing without an access code.
 
-Ryan
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of David Fielder
(US)
Sent: Monday, September 18, 2006 12:26 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP and FXS ports



I have a fax machine connected to an FXS port off my gateway. I have the
Gateway configured for MGCP. The port is configured in CallManager and
the DN for the fax machine has been added. When I call the fax either
from and external line or 4 digits I get fax tone. When they attempt to
make an outbound call from the fax machine, they get dial tone and it
will attempt to dial the digits and then immediately go to a fast busy.
If I do a show voice port summary, I see the port go off hook. I have
looked through the CallManager's logs and I don't think it ever reaches
the CM. I have configured an IP phone to emulate the port and DN
settings and I am able to make call internal and external. So it appears
that the call routing in CM is fine. All the fax machines on the FXS
ports are doing this. Has anyone ever seen this before?

 

David Fielder

Senior Engineer

Dimension Data

Office: (336) 510-9431

Mobile: (336) 669-5129

david.fielder at us.didata.com

 

 

 

________________________________



Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the
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this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
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Attention:

Privileged/Confidential Information may be contained in this message.  Disclosure to any person other than the named recipient is unauthorized.  If you are not the intended recipient, please delete all copies of this information and kindly notify the sender by reply email.  Opinions, conclusions and other information in this message that do not relate to the official business of UNIS LUMIN Inc. shall be understood as neither given nor endorsed by it.  UNIS LUMIN Inc. and any of its subsidiaries reserve the right to monitor all e-mail communications through its networks.  Thank you.
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From GRhodes at rbh.com  Mon Sep 18 13:51:08 2006
From: GRhodes at rbh.com (Rhodes, Geoff)
Date: Mon, 18 Sep 2006 13:51:08 -0400
Subject: [cisco-voip] Ok to run HP Openview client on CCM 4.x servers?
Message-ID: <6350516B6159444E9D646523EB4526F4037B193A@apollo.rbh.local>

I assume it's ok, but wanted to see if anyone is doing it and any
problems they have seen.  We want to make sure any critical server we
have is watched by Openview.
 
Thanks!
 
Geoff Rhodes
Director, Information Technology
Robinson, Bradshaw & Hinson, P.A.
704 / 377-8188
grhodes at rbh.com
 
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From milay at cisco.com  Mon Sep 18 14:01:48 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Mon, 18 Sep 2006 11:01:48 -0700
Subject: [cisco-voip] Ok to run HP Openview client on CCM 4.x servers?
Message-ID: <413FABF5DCE22849AF86F442E472B78BC3BC99@xmb-sjc-219.amer.cisco.com>

Below are the supported HP Openview components that are supported with
CCM, we only list support for 4.0 & 4.1:
 
IP Communications - Platform/Remote Agents 5.2
http://www.cisco.com/pcgi-bin/ecoa/displayProfile.pl?PARTNER_ID=41474
 
IP Communications - Platform/Remote Agents 8.10 & 7.27
http://www.cisco.com/pcgi-bin/ecoa/displayProfile.pl?PARTNER_ID=41471
 
IP Communications - Platform/Remote Agents 3.65
http://www.cisco.com/pcgi-bin/ecoa/displayProfile.pl?PARTNER_ID=41473
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rhodes, Geoff
Sent: Monday, September 18, 2006 12:51 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Ok to run HP Openview client on CCM 4.x servers?


I assume it's ok, but wanted to see if anyone is doing it and any
problems they have seen.  We want to make sure any critical server we
have is watched by Openview.
 
Thanks!
 
Geoff Rhodes
Director, Information Technology
Robinson, Bradshaw & Hinson, P.A.
704 / 377-8188
grhodes at rbh.com
 
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From jcampbell at ci.davidson.nc.us  Mon Sep 18 14:24:43 2006
From: jcampbell at ci.davidson.nc.us (Campbell, Jeremy P.)
Date: Mon, 18 Sep 2006 14:24:43 -0400
Subject: [cisco-voip] mcs7815i rails
Message-ID: <F81B53847D3DC945A55B555B5555CC79D1F81B@davnc03.ci.davidson.nc.us>

What IBM model does the MCS-7815i correspond to?  It seems crazy to
spend $600+ on Cisco branded rails for these units.  I'm assuming the
IBM rails work.  Have others done this?

 

Thanks,

Jeremy

 

_____________________________
Jeremy P. Campbell
Information Systems Director
Town of Davidson, NC
704-940-9632 - voice, direct
704-892-7591 ext. 248 - voice, main
704-892-3971 - fax
JCampbell at ci.davidson.nc.us

 

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From ppauly at gmail.com  Mon Sep 11 13:24:35 2006
From: ppauly at gmail.com (Peter Pauly)
Date: Mon, 11 Sep 2006 13:24:35 -0400
Subject: [cisco-voip] Is it safe to use SQL Server Authentication instead of
	Windows Authentication?
Message-ID: <45059BD3.2060908@gmail.com>

I want to access the CDR database from outside CallManager. Is is safe
to change the database properties from "Use Windows Authentication" to
"Use SQL Server Authentication"?

Would it be better to leave it as is and create a local windows account?


From milay at cisco.com  Mon Sep 18 14:36:00 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Mon, 18 Sep 2006 11:36:00 -0700
Subject: [cisco-voip] Is it safe to use SQL Server Authentication
	instead ofWindows Authentication?
Message-ID: <413FABF5DCE22849AF86F442E472B78BC3BCE7@xmb-sjc-219.amer.cisco.com>

You can change this with no negative impact on the server performance.
However this was changed as part of making the server more secure.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Peter Pauly
Sent: Monday, September 11, 2006 12:25 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Is it safe to use SQL Server Authentication
instead ofWindows Authentication?

I want to access the CDR database from outside CallManager. Is is safe
to change the database properties from "Use Windows Authentication" to
"Use SQL Server Authentication"?

Would it be better to leave it as is and create a local windows account?

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


From aman.chugh at gmail.com  Mon Sep 18 15:06:59 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Tue, 19 Sep 2006 00:36:59 +0530
Subject: [cisco-voip] Is it safe to use SQL Server Authentication
	instead ofWindows Authentication?
In-Reply-To: <413FABF5DCE22849AF86F442E472B78BC3BCE7@xmb-sjc-219.amer.cisco.com>
References: <413FABF5DCE22849AF86F442E472B78BC3BCE7@xmb-sjc-219.amer.cisco.com>
Message-ID: <d4c697ad0609181206s332e84f6i8a028f5720d5b548@mail.gmail.com>

I would suggest not to change it as if you apply any callmanager upgrades or
service releases those check the database authentication setting and might
cause it to fail,Also I am sure this not TAC supported.Creating a windows
user with rights on that database would be a easier way out

Aman


On 9/19/06, Mike Lay (milay) <milay at cisco.com> wrote:
>
> You can change this with no negative impact on the server performance.
> However this was changed as part of making the server more secure.
>
> Mike
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Peter Pauly
> Sent: Monday, September 11, 2006 12:25 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Is it safe to use SQL Server Authentication
> instead ofWindows Authentication?
>
> I want to access the CDR database from outside CallManager. Is is safe
> to change the database properties from "Use Windows Authentication" to
> "Use SQL Server Authentication"?
>
> Would it be better to leave it as is and create a local windows account?
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From Matt.Slaga at us.didata.com  Mon Sep 18 15:42:56 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 18 Sep 2006 15:42:56 -0400
Subject: [cisco-voip] Is it safe to use SQL Server Authenticationinstead
	ofWindows Authentication?
Message-ID: <C1FE15183DA37645BC0633BC604E44F0032CA8D8@USNAEXCH.na.didata.local>

This is TAC supported and the main configuration for third party CDR
server applications.  We have a 3rd party CDR reporting product that
access the CDR database using SQL Server Authentication and we got the
okay from the Cisco CallManager developers and confirmation that the
servers will continue to be a TAC supported system.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Aman Chugh
Sent: Monday, September 18, 2006 3:07 PM
To: Mike Lay (milay)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Is it safe to use SQL Server
Authenticationinstead ofWindows Authentication?

 

I would suggest not to change it as if you apply any callmanager
upgrades or service releases those check the database authentication
setting and might cause it to fail,Also I am sure this not TAC
supported.Creating a windows user with rights on that database would be
a easier way out 

 

Aman

 

On 9/19/06, Mike Lay (milay) <milay at cisco.com> wrote: 

You can change this with no negative impact on the server performance.
However this was changed as part of making the server more secure. 

Mike

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net ] On Behalf Of Peter Pauly
Sent: Monday, September 11, 2006 12:25 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Is it safe to use SQL Server Authentication 
instead ofWindows Authentication?

I want to access the CDR database from outside CallManager. Is is safe
to change the database properties from "Use Windows Authentication" to
"Use SQL Server Authentication"? 

Would it be better to leave it as is and create a local windows account?

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

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

 




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

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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From rsingleton at novateck.com  Mon Sep 18 17:11:16 2006
From: rsingleton at novateck.com (Robert Singleton)
Date: Mon, 18 Sep 2006 16:11:16 -0500
Subject: [cisco-voip] Best Practices
In-Reply-To: <00a501c6dac2$d2052160$25fb14ac@gpatelt>
References: <00a501c6dac2$d2052160$25fb14ac@gpatelt>
Message-ID: <200609181612921.SM00844@mail.leasenet.net>


On Sun, 2006-09-17 at 21:36 -0400, Gary L. Pate wrote: 
> I?m trying to determine a ?best practices? when setting up inbound
> DID?s, extensions, etc. Let me explain:
> 
>  
> 
> In most situations I?ve been involved in, I?ve used the last 4 digits
> of a DID as someone?s extension. I make sure the LEC sends 10 digit?s,
> and I do a significant digits of 4 on the gateway.

I leave significant digits on the gateway at 10, and make all my number
manipulation in CallManager. In my case, however, my extension DNs are
not at all related to the assigned DIDs and only about 10 percent of all
telephones even have a DID. By having a separate translation pattern for
each assigned DID, I have very granular control over DID assignments.

For example, in the DFW metroplex, I have two blocks of 50 numbers, one
in Dallas, one in Fort Worth. I have 197 telephones in 10 offices for
which those would be local calls and 43 more phones in 6 more area for
which those would be interlata LD calls, however only 62 of those 240
phones have (or even need) DIDs assigned.

Robert





From baryonyx5 at gmail.com  Mon Sep 18 17:38:18 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Mon, 18 Sep 2006 16:38:18 -0500
Subject: [cisco-voip] Problems with 7941 hookswitch
Message-ID: <a749d0620609181438s75b68085maeaae1d01a9fe54b@mail.gmail.com>

Hi

Is anyone else seeing a lot of 7941's with bad hookswitches?

We've had to RMA about 10 of them in the last month.  The switch pops up
normally when the phone is taken off hook but the user doesn't get dial
tone.  If you press the hookswitch 5 or 6 times eventually you get a tone
but there's definately something not working right.

Just wondering.

-- 
kris seraphine
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From lelio at uoguelph.ca  Mon Sep 18 17:50:23 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 18 Sep 2006 17:50:23 -0400
Subject: [cisco-voip] Problems with 7941 hookswitch
References: <a749d0620609181438s75b68085maeaae1d01a9fe54b@mail.gmail.com>
Message-ID: <005401c6db6c$7246eec0$ca01a8c0@deadfish>

there is a Cisco document which explains that the hook switch has a self cleaning mechanism and by pressing it in quick succession x number of times, it cleans it and will continue to work normally again.

in actuality, i think it is a self-destructing mechanism. we have had to replace a large number of phones even after we took them apart and cleaned the electronics.

i did some calculations on the cost of maintenance per phone, and the payback was at about 160 phones (at list price). so, if you are replacing more than 160 phones per year, it's better to buy maintenance. if you get discounts, then the payback is quicker (less phones needing to be replaced).


  ----- Original Message ----- 
  From: Kris Seraphine 
  To: cisco-voip 
  Sent: Monday, September 18, 2006 5:38 PM
  Subject: [cisco-voip] Problems with 7941 hookswitch


  Hi

  Is anyone else seeing a lot of 7941's with bad hookswitches? 

  We've had to RMA about 10 of them in the last month.  The switch pops up normally when the phone is taken off hook but the user doesn't get dial tone.  If you press the hookswitch 5 or 6 times eventually you get a tone but there's definately something not working right. 

  Just wondering. 

  -- 
  kris seraphine 


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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From baryonyx5 at gmail.com  Mon Sep 18 17:53:54 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Mon, 18 Sep 2006 16:53:54 -0500
Subject: [cisco-voip] Recommendations for inexpensive monitoring tool
Message-ID: <a749d0620609181453l1a72d242t63edddcb1042a46d@mail.gmail.com>

Hi

I have a small customer with a centralized CCM cluster and 13 remote sites
with between 4 and 25 phones at each site.  They currently have no network
monitoring solution implemented and they are not willing to spend much money
on one.  I'm looking for recommendations for  inexpensive WAN link
monitoring applications.  I'd like to see utilization on different classes
of traffic; either  based on port or better yet on dscp values.  I'm
thinking about using PRTG but thought I'd see if anyone else has a better
suggestion.

thanks

-- 
kris seraphine
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From ray at oneunified.net  Mon Sep 18 19:55:14 2006
From: ray at oneunified.net (Ray Burkholder)
Date: Mon, 18 Sep 2006 20:55:14 -0300
Subject: [cisco-voip] The CMM and SPAN
In-Reply-To: <18082B35CF0EBD4C8D265874508A615C03A8DAEB@hermes.unislumin.com>
Message-ID: <003a01c6db7d$e4e02e90$640a0a0a@oneunified.local>

The trick of doing this across switches and preventing replicated packets is
to:
 
a) when creating a vlan, use the extra command to turn it into a special
spanning/remote vlan
b) when using the monitor session command, use it in 'remote' mode, such as
what was provided in the example 
 
These two details will get you a single stream of packets to the
destinination monitoring port when running across multiple switches.

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan LaTorre
Sent: Monday, September 18, 2006 12:35
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] The CMM and SPAN


Thanks for the ideas guys, I should've mentioned that I did implement a
successful workaround, so I do have recording operational right now. I just
don't like leaving things like this unresolved! Also, IPCCx 4.x upgrade is
in the planning stages, but I'm just trying to figure this out because it
seems like the switch is not operating as it should. I admit it could be my
config, but it's only two lines!
 
My config (that doesn't quite work):
monitor session 2 source interface Gi4/1 both (the CMM backplane port)
monitor session 2 destination interface Gig8/1 (the IPCCx monitor port)
 
The reason we're not spanning the VLAN is because when we do that I get two
copies of each frame on the TX side, since currently the CMM and the IP
phones are in the same VLAN.  This is using 12.2(18)SXF3 on the s720 and
12.4(7a) on the CMM. Pretty close versions, that's for sure.  Maybe it's not
a bug...
 


  _____  

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, September 18, 2006 11:18 AM
To: Ryan LaTorre; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] The CMM and SPAN



Is the layer three vlan interface on the Cat 6509?  If so, try to SPAN the
layer 3 interface rather then the CMM port.

 

What version of IOS are you using on the 65xx and the CMM?

 

I'm using 12.2(18)SXF4 and 12.4(7a) without problem

 

SPAN Config:

 

Vlan 102 is my Core Voice VLAN and 202 is my RSPAN

 

monitor session 2 source vlan 102

monitor session 2 destination remote vlan 202

monitor session 4 destination interface Fa3/44 <-- IPCCx Monitor server

monitor session 4 source remote vlan 202

 

 

Sh mon sess all

 

Session 2

---------

Type                   : Remote Source Session

Source VLANs           :

    Both               : 102

Destination RSPAN VLAN : 202

 

Session 4

---------

Type                   : Remote Destination Session

Source RSPAN VLAN      : 202

Destination Ports      : Fa3/44

 

Scott

 

PS.  All your agents are on the same vlan as your IPCCx server, right?

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan LaTorre
Sent: Monday, September 18, 2006 8:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] The CMM and SPAN

 

 

The other day I encountered what appears to be a bug, but I could not locate
anything in the bug navigator. 

Trying to SPAN the backplane port of a CMM module in a Cat 6500 with Sup720,
for the purposes of IPCCx recording/monitoring (in 3.x).  The SPAN
configuration is very straight forward (I believe) but I found that when I
SPAN both directions I receive only the transmit packets, resulting in only
recording one side of the conversation.  I confirmed this by changing the
SPAN source to receive only and NO packets were mirrored whatsoever.  From
what I can tell, it seems I can't SPAN packets coming from the CMM.

Are there any SPAN/CMM caveats I should be aware of, or does this really
sound like a defect? I didn't see anything in release notes or guides.

Thanks, 
Ryan 

  _____  

Privileged/Confidential Information may be contained in this message.
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-- 
Scanned for viruses & dangerous content at One Unified
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From ray at oneunified.net  Mon Sep 18 19:59:30 2006
From: ray at oneunified.net (Ray Burkholder)
Date: Mon, 18 Sep 2006 20:59:30 -0300
Subject: [cisco-voip] Recommendations for inexpensive monitoring tool
In-Reply-To: <a749d0620609181453l1a72d242t63edddcb1042a46d@mail.gmail.com>
Message-ID: <004801c6db7e$7c304be0$640a0a0a@oneunified.local>

If you were to run a Debian based server, you can get Cricket as a package.
Then use the tools from Acktomic's site to automatically scan interfaces and
create Cricket configuration files.  This will include QoS configurations.
You'll then be able to monitor statistics on a per class/interface basis.

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kris Seraphine
Sent: Monday, September 18, 2006 18:54
To: cisco-voip
Subject: [cisco-voip] Recommendations for inexpensive monitoring tool


Hi

I have a small customer with a centralized CCM cluster and 13 remote sites
with between 4 and 25 phones at each site.  They currently have no network
monitoring solution implemented and they are not willing to spend much money
on one.  I'm looking for recommendations for  inexpensive WAN link
monitoring applications.  I'd like to see utilization on different classes
of traffic; either  based on port or better yet on dscp values.  I'm
thinking about using PRTG but thought I'd see if anyone else has a better
suggestion. 

thanks

-- 
kris seraphine 
-- 
Scanned for viruses & dangerous content at One Unified
<http://www.oneunified.net>  and is believed to be clean. 

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From CMStaffin at ra.rockwell.com  Mon Sep 18 21:30:47 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Mon, 18 Sep 2006 20:30:47 -0500
Subject: [cisco-voip] Problems with 7941 hookswitch
In-Reply-To: <a749d0620609181438s75b68085maeaae1d01a9fe54b@mail.gmail.com>
Message-ID: <OF08538C7A.7A2D58E7-ON862571EE.00083E46-862571EE.00085076@ra.rockwell.com>

YES!!

we have RMA'd about 20 of them already  except ours are 7961 not 41

Craig





"Kris Seraphine" <baryonyx5 at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
09/18/2006 04:38 PM

To
cisco-voip <cisco-voip at puck.nether.net>
cc

Subject
[cisco-voip] Problems with 7941 hookswitch






Hi

Is anyone else seeing a lot of 7941's with bad hookswitches? 

We've had to RMA about 10 of them in the last month.  The switch pops up 
normally when the phone is taken off hook but the user doesn't get dial 
tone.  If you press the hookswitch 5 or 6 times eventually you get a tone 
but there's definately something not working right. 

Just wondering. 

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

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From netfortius at gmail.com  Mon Sep 18 22:19:02 2006
From: netfortius at gmail.com (Netfortius)
Date: Mon, 18 Sep 2006 21:19:02 -0500
Subject: [cisco-voip] Problems with 7941 hookswitch
In-Reply-To: <a749d0620609181438s75b68085maeaae1d01a9fe54b@mail.gmail.com>
References: <a749d0620609181438s75b68085maeaae1d01a9fe54b@mail.gmail.com>
Message-ID: <200609182119.02349.netfortius@gmail.com>

Just finalizing a site install, and already started seeing some 7941s behaving 
like this. I was just wondering if we were plain unlucky with a bad batch, 
when in fact looks like Cisco has screwed up royally :(

Stefan

On Monday 18 September 2006 16:38, Kris Seraphine wrote:
> Hi
>
> Is anyone else seeing a lot of 7941's with bad hookswitches?
>
> We've had to RMA about 10 of them in the last month.  The switch pops up
> normally when the phone is taken off hook but the user doesn't get dial
> tone.  If you press the hookswitch 5 or 6 times eventually you get a tone
> but there's definately something not working right.
>
> Just wondering.

From baryonyx5 at gmail.com  Tue Sep 19 00:37:10 2006
From: baryonyx5 at gmail.com (Kris Seraphine)
Date: Mon, 18 Sep 2006 23:37:10 -0500
Subject: [cisco-voip] call quality issues between gateway and Unity
In-Reply-To: <45096A54.3070501@cisco.com>
References: <s509219f.075@comrcm01intgwia.mandtbank.com>
	<45096A54.3070501@cisco.com>
Message-ID: <a749d0620609182137m3fedefb5ha3ccfbaa629183f2@mail.gmail.com>

Just to update this issue, I found out that callers hear the system prompts
clear as a bell.  It's only the user greetings that are a problem.  The
users are recording their greetings over the WAN using G.729 so it seems the
gateway is having problems playing these out to the PSTN user.

I also noticed that the gateway seems to be hitting CSCsc12570 and calls
between it and Unity are not honoring the region settings.  I'm not sure how
all this fits together but hopefully an IOS upgrade will help.

On 9/14/06, Wes Sisk <wsisk at cisco.com> wrote:
>
> Steven,
>
> No, this is not a blanket best practice recommendation.  It was a
> workaround for the issue cited in that cdets.   If you have the time,
> the best approach is to identify the cause of the robotic voice and
> modify the smallest possible set of options to address that issue.
>
> /Wes
>
> STEVEN CASPER wrote:
> > Thanks Wes,
> > This is very interesting. So is the recommendation to add "mgcp playout
> > adaptive 60 40 200" to all voice gateways that allow non IP phone users
> > to access Unity? I have a lot of gateways that connect Nortel users to
> > Unity via QSIG gateways. We occasionally get complaints of canned or
> > robotic voice quality. We are running CCM 4.1.3sr1 with Unity 4.05.
> >
> > Steve
> >
> >>>> "Wes Sisk" <wsisk at cisco.com> 09/14/06 9:01 AM >>>
> > Steven,
> >
> > Thank you for the clarification. Take a look at the Release-note of
> > the cited bug in bug toolkit.
> >
> > There is a different command to use with MGCP:
> > mgcp playout adaptive 60 40 200
> >
> > Same underlying behavior, just modified for the call signaling
> > protocol.
> >
> > /Wes
> >
> > On Sep 14, 2006, at 8:14 AM, STEVEN CASPER wrote:
> >
> > It is in the document Playout Delay Enhancements:
> >
> > http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/
> > products_feature_guide09186a008008033c.html
> >
> >
> > in the link on Understanding Jitter that you posted.
> >
> > http://www.cisco.com/en/US/tech/tk652/tk698/
> > technologies_tech_note09186a00800945df.shtml
> >
> >
> > Steve
> >
> >>>> "Wes Sisk" <wsisk at cisco.com> 09/13/06 6:30 PM >>>
> > Steven,
> >
> > Where do you get that playout-delay is not supported?
> >
> > /Wes
> >
> > STEVEN CASPER wrote:
> >> Wes,
> >> I see that Playout-Delay is not supported on MGCP platforms so how
> > is
> >> jitter handled in a MGCP/Call Manager enviroment?
> >>
> >> Thanks!
> >> Steve
> >>
> >> Steve Casper
> >> Voice Technologies
> >> M&T Bank
> >> (410) 347-6026
> >>
> >>>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
> >> Kris,
> >>
> >> This sounds very similar to something we fixed long ago.  Take a
> > look
> >
> >> at 'sh voice call' to see if it's jitter:
> >> http://www.cisco.com/en/US/tech/tk652/tk698/
> >> technologies_tech_note09186a00800945df.shtml
> >>
> >> CSCea35850 - Unity RTP stream has inherent jitter
> >>
> >>
> >> If you are using non-default packetization periods watch out for
> >> CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and
> >
> >> G711/30ms
> >>
> >> /Wes
> >>
> >> On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:
> >>
> >> Hi
> >>
> >> I have a customer with a multi site centralized CCM 4.13 cluster.
> >> The remote sites are getting complaints of poor voice quality from
> >> customers when being transferred to Unity (which is located in the
> >> central site) .  The callers cannot understand the greetings and
> >> describe them as choppy.  There are no problems with external calls
> >
> >> transferred from the remote sites to IP phones at the central site
> > or
> >
> >> for intersite calls.  It seems to be limited to calls where the RTP
> >
> >> stream is between a remote voice gateway and Unity.  These calls are
> >
> >> G.729 so Unity has to transcode the greetings on the fly but if that
> >
> >> were the problem I'd expect remote users to have the same problem
> >> when calling into check their messages or leave messages for other
> >> users.  Also, the Unity server is not very heavily used; it
> > generally
> >
> >> has no more than 9 active calls at the most.
> >>
> >> I suspect the gateway might be the problem.  It's a 2851 running
> >> MGCP.  I guess I'd like to know if anyone agrees or disagrees with
> >> this assumption.  Also, I've had a lot of problems with MGCP code on
> >
> >> the ISRs.  Can someone recommend an IOS version that is stable for
> >> and MGCP gateway and doesn't suffer from DSP firmware issues?
> >>
> >> I should mention that Unity is version 4.04 sr1 and the gateway is
> >> running 12.4(5).
> >>
> >> Thanks
> >>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
kris seraphine
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From balbler at nts.at  Tue Sep 19 02:24:17 2006
From: balbler at nts.at (Bernhard Albler)
Date: Tue, 19 Sep 2006 08:24:17 +0200
Subject: [cisco-voip] Problems with 7941 hookswitch
In-Reply-To: <200609182119.02349.netfortius@gmail.com>
Message-ID: <5A8C69748176434F92F1EECD7D21F72C78B56F@srvgraz03.nts.at>

Hi
> -----Urspr?ngliche Nachricht-----
> Von: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] Im Auftrag von Netfortius
> Gesendet: Dienstag, 19. September 2006 04:19
> An: cisco-voip at puck.nether.net
> Betreff: Re: [cisco-voip] Problems with 7941 hookswitch
> 
> Just finalizing a site install, and already started seeing some 7941s
> behaving
> like this. I was just wondering if we were plain unlucky with a bad batch,
> when in fact looks like Cisco has screwed up royally :(

We are seeing the same issue right now. Defect rates are around 5% of the phones right now. Can any of the cisco guys comment on this? Is this internally known already?

Regards
bernhard
------------------------------------------
DI (FH) Bernhard Albler, NTS AG 
Tel.: +43 (316) 405 455 -0, email: balbler at nts.at 
 


From manoj.kalpage at gmail.com  Tue Sep 19 03:44:49 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Tue, 19 Sep 2006 16:44:49 +0900
Subject: [cisco-voip] Cisco Router and VPN Module
Message-ID: <bbe09c1d0609190044g218f3d5bq6bb6c88d4e43b50a@mail.gmail.com>

My understanding is with Advanced IP service feature set any descent router
can create VPN (and DMVPN using GRE over IPSEC) tunnels between routers and
having VPN module speed up the process. Am I right?

Please someone correct me.

Thanks,
Manoj
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From CORTIZ at broward.org  Tue Sep 19 09:09:00 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue, 19 Sep 2006 09:09:00 -0400
Subject: [cisco-voip] call quality issues between gateway and Unity
In-Reply-To: <a749d0620609182137m3fedefb5ha3ccfbaa629183f2@mail.gmail.com>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A47581@MREXVS01.bc.broward.cty>

This may seem silly but make sure that the actual files were not
recorded when the WAN was congested.  We had complaints one time from a
location of garbled greetings but it turned out that the recordings were
done during WAN congestion and just needed to be recorded again....

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kris Seraphine
Sent: Tuesday, September 19, 2006 12:37 AM
To: Wes Sisk
Cc: cisco-voip at puck.nether.net; STEVEN CASPER
Subject: Re: [cisco-voip] call quality issues between gateway and Unity

 

Just to update this issue, I found out that callers hear the system
prompts clear as a bell.  It's only the user greetings that are a
problem.  The users are recording their greetings over the WAN using
G.729 so it seems the gateway is having problems playing these out to
the PSTN user.  

I also noticed that the gateway seems to be hitting CSCsc12570 and calls
between it and Unity are not honoring the region settings.  I'm not sure
how all this fits together but hopefully an IOS upgrade will help. 

On 9/14/06, Wes Sisk <wsisk at cisco.com> wrote:

Steven,

No, this is not a blanket best practice recommendation.  It was a
workaround for the issue cited in that cdets.   If you have the time,
the best approach is to identify the cause of the robotic voice and 
modify the smallest possible set of options to address that issue.

/Wes

STEVEN CASPER wrote:
> Thanks Wes,
> This is very interesting. So is the recommendation to add "mgcp
playout
> adaptive 60 40 200" to all voice gateways that allow non IP phone
users 
> to access Unity? I have a lot of gateways that connect Nortel users to
> Unity via QSIG gateways. We occasionally get complaints of canned or
> robotic voice quality. We are running CCM 4.1.3sr1 with Unity 4.05.
>
> Steve
>
>>>> "Wes Sisk" <wsisk at cisco.com> 09/14/06 9:01 AM >>>
> Steven,
>
> Thank you for the clarification. Take a look at the Release-note of 
> the cited bug in bug toolkit.
>
> There is a different command to use with MGCP:
> mgcp playout adaptive 60 40 200
>
> Same underlying behavior, just modified for the call signaling
> protocol.
>
> /Wes
>
> On Sep 14, 2006, at 8:14 AM, STEVEN CASPER wrote:
>
> It is in the document Playout Delay Enhancements:
>
> http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/
> products_feature_guide09186a008008033c.html
>
>
> in the link on Understanding Jitter that you posted.
>
> http://www.cisco.com/en/US/tech/tk652/tk698/
> technologies_tech_note09186a00800945df.shtml
>
>
> Steve
>
>>>> "Wes Sisk" <wsisk at cisco.com > 09/13/06 6:30 PM >>>
> Steven,
>
> Where do you get that playout-delay is not supported?
>
> /Wes
>
> STEVEN CASPER wrote:
>> Wes,
>> I see that Playout-Delay is not supported on MGCP platforms so how 
> is
>> jitter handled in a MGCP/Call Manager enviroment?
>>
>> Thanks!
>> Steve
>>
>> Steve Casper
>> Voice Technologies
>> M&T Bank
>> (410) 347-6026 
>>
>>>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
>> Kris,
>>
>> This sounds very similar to something we fixed long ago.  Take a 
> look
>
>> at 'sh voice call' to see if it's jitter:
>> http://www.cisco.com/en/US/tech/tk652/tk698/
>> technologies_tech_note09186a00800945df.shtml 
>>
>> CSCea35850 - Unity RTP stream has inherent jitter
>>
>>
>> If you are using non-default packetization periods watch out for
>> CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and
>
>> G711/30ms
>>
>> /Wes
>>
>> On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:
>>
>> Hi
>>
>> I have a customer with a multi site centralized CCM 4.13 cluster.
>> The remote sites are getting complaints of poor voice quality from
>> customers when being transferred to Unity (which is located in the
>> central site) .  The callers cannot understand the greetings and 
>> describe them as choppy.  There are no problems with external calls
>
>> transferred from the remote sites to IP phones at the central site
> or
>
>> for intersite calls.  It seems to be limited to calls where the RTP 
>
>> stream is between a remote voice gateway and Unity.  These calls are
>
>> G.729 so Unity has to transcode the greetings on the fly but if that
>
>> were the problem I'd expect remote users to have the same problem 
>> when calling into check their messages or leave messages for other
>> users.  Also, the Unity server is not very heavily used; it
> generally
>
>> has no more than 9 active calls at the most. 
>>
>> I suspect the gateway might be the problem.  It's a 2851 running
>> MGCP.  I guess I'd like to know if anyone agrees or disagrees with
>> this assumption.  Also, I've had a lot of problems with MGCP code on 
>
>> the ISRs.  Can someone recommend an IOS version that is stable for
>> and MGCP gateway and doesn't suffer from DSP firmware issues?
>>
>> I should mention that Unity is version 4.04 sr1 and the gateway is
>> running 12.4(5).
>>
>> Thanks
>>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
kris seraphine 

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From Court.Schuett at us.millwardbrown.com  Tue Sep 19 09:47:59 2006
From: Court.Schuett at us.millwardbrown.com (Schuett, Court (MBNAP it))
Date: Tue, 19 Sep 2006 08:47:59 -0500
Subject: [cisco-voip] Problems with 7941 hookswitch
Message-ID: <98AA0AC6C161614396D4976A32D21D0302E8C5D1@usnapwxc001.am.group.local>

We're experiencing the same problems.  We just deployed 350 phones to
our corporate headquarters and have had about 15 failures so far, so 5%
sounds about right.  We didn't get maintenance on them either so we're
happy to have them fail now rather than later, but it is really
irritating.  It just creates a lot of extra work for us that we really
didn't need.  Our were scattered between 41s and 61s as well.  
 
We've also noticed that they seem to disconnect from the Ethernet jacks
more often.  If people are moving their phones around, they have a
tendency to jiggle loose and then reboot.  Again, never had that problem
with the 40s or 60s but it seems pretty chronic with the 41s and 61s.
 
 
 
Court Schuett
Network Systems Engineer
Millward Brown, Naperville
w: 630-955-8983
f: 630-245-5755
e: 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 Craig M Staffin
Sent: Monday, September 18, 2006 8:31 PM
To: Kris Seraphine
Cc: cisco-voip-bounces at puck.nether.net; cisco-voip
Subject: Re: [cisco-voip] Problems with 7941 hookswitch



YES!! 

we have RMA'd about 20 of them already  except ours are 7961 not 41 

Craig





"Kris Seraphine" <baryonyx5 at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net 

09/18/2006 04:38 PM 

To
cisco-voip <cisco-voip at puck.nether.net> 
cc
Subject
[cisco-voip] Problems with 7941 hookswitch

	




Hi

Is anyone else seeing a lot of 7941's with bad hookswitches? 

We've had to RMA about 10 of them in the last month.  The switch pops up
normally when the phone is taken off hook but the user doesn't get dial
tone.  If you press the hookswitch 5 or 6 times eventually you get a
tone but there's definately something not working right. 

Just wondering. 

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



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From CORTIZ at broward.org  Tue Sep 19 09:59:25 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue, 19 Sep 2006 09:59:25 -0400
Subject: [cisco-voip] Problems with 7941 hookswitch
In-Reply-To: <98AA0AC6C161614396D4976A32D21D0302E8C5D1@usnapwxc001.am.group.local>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A47588@MREXVS01.bc.broward.cty>

We just started ordering these so that's not good to hear.  Only have
about 15 7941 & 7961 deployed but no problems yet.  We do have 50+ on
order for our next deployment though?  Is this being addressed formally
by Cisco?????

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Schuett, Court
(MBNAP it)
Sent: Tuesday, September 19, 2006 9:48 AM
To: cisco-voip-bounces at puck.nether.net; cisco-voip
Subject: Re: [cisco-voip] Problems with 7941 hookswitch

 

We're experiencing the same problems.  We just deployed 350 phones to
our corporate headquarters and have had about 15 failures so far, so 5%
sounds about right.  We didn't get maintenance on them either so we're
happy to have them fail now rather than later, but it is really
irritating.  It just creates a lot of extra work for us that we really
didn't need.  Our were scattered between 41s and 61s as well.  

 

We've also noticed that they seem to disconnect from the Ethernet jacks
more often.  If people are moving their phones around, they have a
tendency to jiggle loose and then reboot.  Again, never had that problem
with the 40s or 60s but it seems pretty chronic with the 41s and 61s.

 

 

 

Court Schuett

Network Systems Engineer

Millward Brown, Naperville

w: 630-955-8983

f: 630-245-5755

e: 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 Craig M Staffin
Sent: Monday, September 18, 2006 8:31 PM
To: Kris Seraphine
Cc: cisco-voip-bounces at puck.nether.net; cisco-voip
Subject: Re: [cisco-voip] Problems with 7941 hookswitch


YES!! 

we have RMA'd about 20 of them already  except ours are 7961 not 41 

Craig





"Kris Seraphine" <baryonyx5 at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net 

09/18/2006 04:38 PM 

To

cisco-voip <cisco-voip at puck.nether.net> 

cc

 

Subject

[cisco-voip] Problems with 7941 hookswitch

 

 

 




Hi

Is anyone else seeing a lot of 7941's with bad hookswitches? 

We've had to RMA about 10 of them in the last month.  The switch pops up
normally when the phone is taken off hook but the user doesn't get dial
tone.  If you press the hookswitch 5 or 6 times eventually you get a
tone but there's definately something not working right. 

Just wondering. 

-- 
kris seraphine _______________________________________________
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. 
==================================================== 

 

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From lelio at uoguelph.ca  Tue Sep 19 10:03:13 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 19 Sep 2006 10:03:13 -0400
Subject: [cisco-voip] Problems with 7941 hookswitch
References: <26B357EF9FB7D04A989C9E82A9E5436E03A47588@MREXVS01.bc.broward.cty>
Message-ID: <007501c6dbf4$5a16e1b0$47196883@cfs.uoguelph.ca>

click here for more information about the self-destructing, ahem, I mean self-cleaning hookswitches ;)

i think that is their official answer:

to quote:
  A thorough analysis of a large number of Cisco IP Phones that have been reported as hardware failures has been completed and our research reveals that the great majority of these phones do not have any determinable failure. The standard by which the phone industry measures phone failures is based on Bell Corp/Telcordia standards and the standard acceptable failure rate is set at 4 percent. The Cisco IP Phone's overall failure rate is well under the acceptable industry average. Many of the problems that have been reported as hardware failures are really either operational or cleanup issues. This document describes some common troubleshoot steps to be taken before you attempt to replace the Cisco IP Phone.

we have over 7000 phones, so 4% is about 280 phones. not very pretty.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Ortiz, Carlos 
  To: Schuett, Court (MBNAP it) ; cisco-voip-bounces at puck.nether.net ; cisco-voip 
  Sent: Tuesday, September 19, 2006 9:59 AM
  Subject: Re: [cisco-voip] Problems with 7941 hookswitch


  We just started ordering these so that's not good to hear.  Only have about 15 7941 & 7961 deployed but no problems yet.  We do have 50+ on order for our next deployment though?  Is this being addressed formally by Cisco?????

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Schuett, Court (MBNAP it)
  Sent: Tuesday, September 19, 2006 9:48 AM
  To: cisco-voip-bounces at puck.nether.net; cisco-voip
  Subject: Re: [cisco-voip] Problems with 7941 hookswitch

   

  We're experiencing the same problems.  We just deployed 350 phones to our corporate headquarters and have had about 15 failures so far, so 5% sounds about right.  We didn't get maintenance on them either so we're happy to have them fail now rather than later, but it is really irritating.  It just creates a lot of extra work for us that we really didn't need.  Our were scattered between 41s and 61s as well.  

   

  We've also noticed that they seem to disconnect from the Ethernet jacks more often.  If people are moving their phones around, they have a tendency to jiggle loose and then reboot.  Again, never had that problem with the 40s or 60s but it seems pretty chronic with the 41s and 61s.

   

   

   

  Court Schuett

  Network Systems Engineer

  Millward Brown, Naperville

  w: 630-955-8983

  f: 630-245-5755

  e: 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 Craig M Staffin
  Sent: Monday, September 18, 2006 8:31 PM
  To: Kris Seraphine
  Cc: cisco-voip-bounces at puck.nether.net; cisco-voip
  Subject: Re: [cisco-voip] Problems with 7941 hookswitch


  YES!! 

  we have RMA'd about 20 of them already  except ours are 7961 not 41 

  Craig





        "Kris Seraphine" <baryonyx5 at gmail.com> 
        Sent by: cisco-voip-bounces at puck.nether.net 

        09/18/2006 04:38 PM 
       To
             cisco-voip <cisco-voip at puck.nether.net> 
             
              cc
              
             
              Subject
             [cisco-voip] Problems with 7941 hookswitch
             

         

               
              
             

       




  Hi

  Is anyone else seeing a lot of 7941's with bad hookswitches? 

  We've had to RMA about 10 of them in the last month.  The switch pops up normally when the phone is taken off hook but the user doesn't get dial tone.  If you press the hookswitch 5 or 6 times eventually you get a tone but there's definately something not working right. 

  Just wondering. 

  -- 
  kris seraphine _______________________________________________
  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
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From mfa at crec.ifas.ufl.edu  Tue Sep 19 10:14:10 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Tue, 19 Sep 2006 10:14:10 -0400
Subject: [cisco-voip] Problems with 7941 hookswitch
References: <mailman.11208.1158623738.1681.cisco-voip@puck.nether.net>
Message-ID: <02e101c6dbf5$e0c44f30$400cfb0a@crec.ifas.ufl.edu>

>From a mechanical viewpoint, the 79xx phones are junk, IMHO.  We probably 
RMA-replace 10% of our phone population every year,  almost all because of 
switchhook and deskstand problems.  I've asked several times if we could buy 
replacement parts and fix the common problems ourselves, but never received 
any response; I suspect a local repair might complicate future RMAs.  A 
while back I had to escalate an SR because the first engineer claimed we 
were abusing our phones, but after I offered his supervisor a chance to 
discuss the abuse with the 64-year-old lady who was the alleged abuser, he 
declined and we've had no problems since.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

> Date: Mon, 18 Sep 2006 16:38:18 -0500
> From: "Kris Seraphine" <baryonyx5 at gmail.com>
> Subject: [cisco-voip] Problems with 7941 hookswitch
> To: cisco-voip <cisco-voip at puck.nether.net>
> Message-ID:
> <a749d0620609181438s75b68085maeaae1d01a9fe54b at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi
>
> Is anyone else seeing a lot of 7941's with bad hookswitches?
>
> We've had to RMA about 10 of them in the last month.  The switch pops up
> normally when the phone is taken off hook but the user doesn't get dial
> tone.  If you press the hookswitch 5 or 6 times eventually you get a tone
> but there's definately something not working right.
>
> Just wondering.
>
> -- 
> kris seraphine


From lelio at uoguelph.ca  Tue Sep 19 10:19:51 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 19 Sep 2006 10:19:51 -0400
Subject: [cisco-voip] Problems with 7941 hookswitch
References: <mailman.11208.1158623738.1681.cisco-voip@puck.nether.net>
	<02e101c6dbf5$e0c44f30$400cfb0a@crec.ifas.ufl.edu>
Message-ID: <008801c6dbf6$abbc9b20$47196883@cfs.uoguelph.ca>

we have a 64 year old lady in the office who could take me in a bar brawl any day.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Mike Armstrong 
  To: cisco-voip at puck.nether.net ; baryonyx5 at gmail.com 
  Sent: Tuesday, September 19, 2006 10:14 AM
  Subject: Re: [cisco-voip] Problems with 7941 hookswitch


  >From a mechanical viewpoint, the 79xx phones are junk, IMHO.  We probably 
  RMA-replace 10% of our phone population every year,  almost all because of 
  switchhook and deskstand problems.  I've asked several times if we could buy 
  replacement parts and fix the common problems ourselves, but never received 
  any response; I suspect a local repair might complicate future RMAs.  A 
  while back I had to escalate an SR because the first engineer claimed we 
  were abusing our phones, but after I offered his supervisor a chance to 
  discuss the abuse with the 64-year-old lady who was the alleged abuser, he 
  declined and we've had no problems since.

  Mike Armstrong
  UF/IFAS CREC
  Lake Alfred, FL

  > Date: Mon, 18 Sep 2006 16:38:18 -0500
  > From: "Kris Seraphine" <baryonyx5 at gmail.com>
  > Subject: [cisco-voip] Problems with 7941 hookswitch
  > To: cisco-voip <cisco-voip at puck.nether.net>
  > Message-ID:
  > <a749d0620609181438s75b68085maeaae1d01a9fe54b at mail.gmail.com>
  > Content-Type: text/plain; charset="iso-8859-1"
  >
  > Hi
  >
  > Is anyone else seeing a lot of 7941's with bad hookswitches?
  >
  > We've had to RMA about 10 of them in the last month.  The switch pops up
  > normally when the phone is taken off hook but the user doesn't get dial
  > tone.  If you press the hookswitch 5 or 6 times eventually you get a tone
  > but there's definately something not working right.
  >
  > Just wondering.
  >
  > -- 
  > kris seraphine

  _______________________________________________
  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  Tue Sep 19 11:05:42 2006
From: balbler at nts.at (Bernhard Albler)
Date: Tue, 19 Sep 2006 17:05:42 +0200
Subject: [cisco-voip] Problems with 7941 hookswitch
In-Reply-To: <008801c6dbf6$abbc9b20$47196883@cfs.uoguelph.ca>
Message-ID: <5A8C69748176434F92F1EECD7D21F72C78B5E4@srvgraz03.nts.at>

Hi,

 

I can only underscore your comment. The Hookswitch has always been a
weakness of the cisco phones. However, I have never seen the failure
rate with the older 7940 / 7960 phones that we are now seeing with the
79X1 phones. While we ever only had 7940/7960ies which went bad after
some time, with 79X1s which are not working out of the box, dead on
arrival. So I do believe that this is an issue related to the new
models.

I still wonder if you guys are talking about the same lady. Maybe the
engineer backed down because of fear ;-).

regards

bernhard

 

 

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

DI (FH) Bernhard Albler, NTS AG 

Tel.: +43 (316) 405 455 -0, email: balbler at nts.at 

 

________________________________

Von: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] Im Auftrag von Lelio
Fulgenzi
Gesendet: Dienstag, 19. September 2006 16:20
An: Mike Armstrong; cisco-voip at puck.nether.net; baryonyx5 at gmail.com
Betreff: Re: [cisco-voip] Problems with 7941 hookswitch

 

we have a 64 year old lady in the office who could take me in a bar
brawl any day.

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."

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

	From: Mike Armstrong <mailto:mfa at crec.ifas.ufl.edu>  

	To: cisco-voip at puck.nether.net ; baryonyx5 at gmail.com 

	Sent: Tuesday, September 19, 2006 10:14 AM

	Subject: Re: [cisco-voip] Problems with 7941 hookswitch

	 

	>From a mechanical viewpoint, the 79xx phones are junk, IMHO.
We probably 
	RMA-replace 10% of our phone population every year,  almost all
because of 
	switchhook and deskstand problems.  I've asked several times if
we could buy 
	replacement parts and fix the common problems ourselves, but
never received 
	any response; I suspect a local repair might complicate future
RMAs.  A 
	while back I had to escalate an SR because the first engineer
claimed we 
	were abusing our phones, but after I offered his supervisor a
chance to 
	discuss the abuse with the 64-year-old lady who was the alleged
abuser, he 
	declined and we've had no problems since.
	
	Mike Armstrong
	UF/IFAS CREC
	Lake Alfred, FL
	
	> Date: Mon, 18 Sep 2006 16:38:18 -0500
	> From: "Kris Seraphine" <baryonyx5 at gmail.com>
	> Subject: [cisco-voip] Problems with 7941 hookswitch
	> To: cisco-voip <cisco-voip at puck.nether.net>
	> Message-ID:
	> <a749d0620609181438s75b68085maeaae1d01a9fe54b at mail.gmail.com>
	> Content-Type: text/plain; charset="iso-8859-1"
	>
	> Hi
	>
	> Is anyone else seeing a lot of 7941's with bad hookswitches?
	>
	> We've had to RMA about 10 of them in the last month.  The
switch pops up
	> normally when the phone is taken off hook but the user doesn't
get dial
	> tone.  If you press the hookswitch 5 or 6 times eventually you
get a tone
	> but there's definately something not working right.
	>
	> Just wondering.
	>
	> -- 
	> kris seraphine
	
	_______________________________________________
	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  Tue Sep 19 11:09:24 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Tue, 19 Sep 2006 11:09:24 -0400
Subject: [cisco-voip] Problems with 7941 hookswitch
References: <mailman.11535.1158674600.1681.cisco-voip@puck.nether.net>
Message-ID: <043301c6dbfd$9808cc00$400cfb0a@crec.ifas.ufl.edu>

4% sounds awfully high to me; I've been in the telephony game since 1960; 
have only seen one Western Electric 500 handset fail, and that was because I 
had just beat the daylights out of it.  It's a long story, but no jury in 
the world would convict me.

Mike

> Date: Tue, 19 Sep 2006 10:03:13 -0400
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> Subject: Re: [cisco-voip] Problems with 7941 hookswitch
> To: "Ortiz, Carlos" <CORTIZ at broward.org>, "Schuett, Court \(MBNAP
> it\)" <Court.Schuett at us.millwardbrown.com>,
> <cisco-voip-bounces at puck.nether.net>, "cisco-voip"
> <cisco-voip at puck.nether.net>
> Message-ID: <007501c6dbf4$5a16e1b0$47196883 at cfs.uoguelph.ca>
> Content-Type: text/plain; charset="iso-8859-1"
>
> click here for more information about the self-destructing, ahem, I mean 
> self-cleaning hookswitches ;)
>
> i think that is their official answer:
>
> to quote:
>  A thorough analysis of a large number of Cisco IP Phones that have been 
> reported as hardware failures has been completed and our research reveals 
> that the great majority of these phones do not have any determinable 
> failure. The standard by which the phone industry measures phone failures 
> is based on Bell Corp/Telcordia standards and the standard acceptable 
> failure rate is set at 4 percent. The Cisco IP Phone's overall failure 
> rate is well under the acceptable industry average. Many of the problems 
> that have been reported as hardware failures are really either operational 
> or cleanup issues. This document describes some common troubleshoot steps 
> to be taken before you attempt to replace the Cisco IP Phone.
>
> we have over 7000 phones, so 4% is about 280 phones. not very pretty.
>
> --------------------------------------------------------------------------------
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."


From pwalenta at wi.rr.com  Tue Sep 19 11:43:24 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Tue, 19 Sep 2006 10:43:24 -0500
Subject: [cisco-voip] call quality issues between gateway and Unity
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E03A47581@MREXVS01.bc.broward.cty>
Message-ID: <000001c6dc02$588f27e0$0232a8c0@pwnhome.com>

If WAN congestion is causing audio issues with Unity, the it would sound
like QoS settings might be off.  QoS if done right, should protect voice, no
matter how much traffic is present (in my experience, you can keep voice
running smoothly, even if you have a severely overloaded WAN connection).

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: Tuesday, September 19, 2006 8:09 AM
To: Kris Seraphine; Wes Sisk
Cc: cisco-voip at puck.nether.net; STEVEN CASPER
Subject: Re: [cisco-voip] call quality issues between gateway and Unity



This may seem silly but make sure that the actual files were not recorded
when the WAN was congested.  We had complaints one time from a location of
garbled greetings but it turned out that the recordings were done during WAN
congestion and just needed to be recorded again..

 

Carlos

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kris Seraphine
Sent: Tuesday, September 19, 2006 12:37 AM
To: Wes Sisk
Cc: cisco-voip at puck.nether.net; STEVEN CASPER
Subject: Re: [cisco-voip] call quality issues between gateway and Unity

 

Just to update this issue, I found out that callers hear the system prompts
clear as a bell.  It's only the user greetings that are a problem.  The
users are recording their greetings over the WAN using G.729 so it seems the
gateway is having problems playing these out to the PSTN user.  

I also noticed that the gateway seems to be hitting CSCsc12570 and calls
between it and Unity are not honoring the region settings.  I'm not sure how
all this fits together but hopefully an IOS upgrade will help. 

On 9/14/06, Wes Sisk <wsisk at cisco.com> wrote:

Steven,

No, this is not a blanket best practice recommendation.  It was a
workaround for the issue cited in that cdets.   If you have the time,
the best approach is to identify the cause of the robotic voice and 
modify the smallest possible set of options to address that issue.

/Wes

STEVEN CASPER wrote:
> Thanks Wes,
> This is very interesting. So is the recommendation to add "mgcp playout
> adaptive 60 40 200" to all voice gateways that allow non IP phone users 
> to access Unity? I have a lot of gateways that connect Nortel users to
> Unity via QSIG gateways. We occasionally get complaints of canned or
> robotic voice quality. We are running CCM 4.1.3sr1 with Unity 4.05.
>
> Steve
>
>>>> "Wes Sisk" <wsisk at cisco.com> 09/14/06 9:01 AM >>>
> Steven,
>
> Thank you for the clarification. Take a look at the Release-note of 
> the cited bug in bug toolkit.
>
> There is a different command to use with MGCP:
> mgcp playout adaptive 60 40 200
>
> Same underlying behavior, just modified for the call signaling
> protocol.
>
> /Wes
>
> On Sep 14, 2006, at 8:14 AM, STEVEN CASPER wrote:
>
> It is in the document Playout Delay Enhancements:
>
> http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/
> products_feature_guide09186a008008033c.html
>
>
> in the link on Understanding Jitter that you posted.
>
> http://www.cisco.com/en/US/tech/tk652/tk698/
> technologies_tech_note09186a00800945df.shtml
>
>
> Steve
>
>>>> "Wes Sisk" <wsisk at cisco.com  <mailto:wsisk at cisco.com> > 09/13/06 6:30
PM >>>
> Steven,
>
> Where do you get that playout-delay is not supported?
>
> /Wes
>
> STEVEN CASPER wrote:
>> Wes,
>> I see that Playout-Delay is not supported on MGCP platforms so how 
> is
>> jitter handled in a MGCP/Call Manager enviroment?
>>
>> Thanks!
>> Steve
>>
>> Steve Casper
>> Voice Technologies
>> M&T Bank
>> (410) 347-6026 
>>
>>>>> "Wes Sisk" <wsisk at cisco.com> 09/11/06 7:00 PM >>>
>> Kris,
>>
>> This sounds very similar to something we fixed long ago.  Take a 
> look
>
>> at 'sh voice call' to see if it's jitter:
>> http://www.cisco.com/en/US/tech/tk652/tk698/
>> technologies_tech_note09186a00800945df.shtml 
>>
>> CSCea35850 - Unity RTP stream has inherent jitter
>>
>>
>> If you are using non-default packetization periods watch out for
>> CSCed52913 Unity RTP stream has jitter with G.729 at 30 & 60 ms and
>
>> G711/30ms
>>
>> /Wes
>>
>> On Sep 11, 2006, at 5:56 PM, Kris Seraphine wrote:
>>
>> Hi
>>
>> I have a customer with a multi site centralized CCM 4.13 cluster.
>> The remote sites are getting complaints of poor voice quality from
>> customers when being transferred to Unity (which is located in the
>> central site) .  The callers cannot understand the greetings and 
>> describe them as choppy.  There are no problems with external calls
>
>> transferred from the remote sites to IP phones at the central site
> or
>
>> for intersite calls.  It seems to be limited to calls where the RTP 
>
>> stream is between a remote voice gateway and Unity.  These calls are
>
>> G.729 so Unity has to transcode the greetings on the fly but if that
>
>> were the problem I'd expect remote users to have the same problem 
>> when calling into check their messages or leave messages for other
>> users.  Also, the Unity server is not very heavily used; it
> generally
>
>> has no more than 9 active calls at the most. 
>>
>> I suspect the gateway might be the problem.  It's a 2851 running
>> MGCP.  I guess I'd like to know if anyone agrees or disagrees with
>> this assumption.  Also, I've had a lot of problems with MGCP code on 
>
>> the ISRs.  Can someone recommend an IOS version that is stable for
>> and MGCP gateway and doesn't suffer from DSP firmware issues?
>>
>> I should mention that Unity is version 4.04 sr1 and the gateway is
>> running 12.4(5).
>>
>> Thanks
>>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
kris seraphine 

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From keith.klevenski at rig.net  Tue Sep 19 13:05:48 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Tue, 19 Sep 2006 12:05:48 -0500
Subject: [cisco-voip] 4.1.3 Upgrade Assistant?
Message-ID: <7F0CE0A1A0961E498A41329535342266D30177@HOUEXH02.rignet.org>

Hi all,

 

I'm going to be upgrading our 3.3.5 cluster to 4.1.3 this week and I am
not able to locate the 4.1.3 upgrade assistant on CCO however I do see
4.1.2.  Anyone know where it is or am I blind?

 

Thx.

 

Keith  

 

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From lelio at uoguelph.ca  Tue Sep 19 13:10:22 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 19 Sep 2006 13:10:22 -0400
Subject: [cisco-voip] 4.1.3 Upgrade Assistant?
References: <7F0CE0A1A0961E498A41329535342266D30177@HOUEXH02.rignet.org>
Message-ID: <012801c6dc0e$7ddb3a50$47196883@cfs.uoguelph.ca>

Upgrade assistants are now cumulative, so that means, always use the latest one. It's also in the utilities section, the latest one should be 4.2 I think.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Keith Klevenski 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, September 19, 2006 1:05 PM
  Subject: [cisco-voip] 4.1.3 Upgrade Assistant?


  Hi all,

   

  I'm going to be upgrading our 3.3.5 cluster to 4.1.3 this week and I am not able to locate the 4.1.3 upgrade assistant on CCO however I do see 4.1.2.  Anyone know where it is or am I blind?

   

  Thx.

   

  Keith  

   



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


  _______________________________________________
  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 digitel.net  Tue Sep 19 13:15:07 2006
From: jgrace at digitel.net (James Grace)
Date: Tue, 19 Sep 2006 13:15:07 -0400
Subject: [cisco-voip] message notification schedule change in bulk
Message-ID: <20060919131493.SM01512@jgm>

We have about 150 users all with message notification.  The default schedule
is 8 to 5 mon to fri.  Is there some type of way to change this in bulk?  I
change the subscriber's vm schedule but this does not change the message
notification schedule 

 

 

 

James D Grace 

CCNP CCNA MCSE MCDBA

System Engineer / Professional Svc.

Digitel Corp.

Email:  <mailto:grace.jd at gmail> grace.jd at gmail <mailto:grace.jd at gmail.com>
.com

email: jgrace at digitel.net 

MSN id: grace.jd at gmail.com

 

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From Matt.Slaga at us.didata.com  Tue Sep 19 13:25:08 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 19 Sep 2006 13:25:08 -0400
Subject: [cisco-voip] message notification schedule change in bulk
Message-ID: <C1FE15183DA37645BC0633BC604E44F00330DA7B@USNAEXCH.na.didata.local>

There is no way to do this that I know of.  Well, not without changing
subscriber templates, deleting and re-adding your users.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, September 19, 2006 1:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] message notification schedule change in bulk

 

We have about 150 users all with message notification.  The default
schedule is 8 to 5 mon to fri.  Is there some type of way to change this
in bulk?  I change the subscriber's vm schedule but this does not change
the message notification schedule 

 

 

 

James D Grace 

CCNP CCNA MCSE MCDBA

System Engineer / Professional Svc.

Digitel Corp.

Email:  <mailto:grace.jd at gmail> grace.jd at gmail
<mailto:grace.jd at gmail.com> .com

email: jgrace at digitel.net 

MSN id: grace.jd at gmail.com

 




-----------------------------------------
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From jgrace at digitel.net  Tue Sep 19 15:28:06 2006
From: jgrace at digitel.net (James Grace)
Date: Tue, 19 Sep 2006 15:28:06 -0400
Subject: [cisco-voip] message notification schedule change in bulk
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00330DA7B@USNAEXCH.na.didata.local>
Message-ID: <20060919152700.SM01512@jgm>

Well the thing is that the subscriber template do not change the message
notification,  only the VM active under profiles

 

  _____  

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Tuesday, September 19, 2006 1:25 PM
To: James Grace; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] message notification schedule change in bulk

 

There is no way to do this that I know of.  Well, not without changing
subscriber templates, deleting and re-adding your users.

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, September 19, 2006 1:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] message notification schedule change in bulk

 

We have about 150 users all with message notification.  The default schedule
is 8 to 5 mon to fri.  Is there some type of way to change this in bulk?  I
change the subscriber's vm schedule but this does not change the message
notification schedule 

 

 

 

James D Grace 

CCNP CCNA MCSE MCDBA

System Engineer / Professional Svc.

Digitel Corp.

Email:  <mailto:grace.jd at gmail> grace.jd at gmail <mailto:grace.jd at gmail.com>
.com

email: jgrace at digitel.net 

MSN id: grace.jd at gmail.com

 


  _____  



Disclaimer:

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only. If you are not the
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this communication in error and that any use or reproduction of
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From reuben-cisco-voip at reub.net  Tue Sep 19 18:25:11 2006
From: reuben-cisco-voip at reub.net (Reuben Farrelly)
Date: Wed, 20 Sep 2006 08:25:11 +1000
Subject: [cisco-voip] Problems with 7941 hookswitch
In-Reply-To: <02e101c6dbf5$e0c44f30$400cfb0a@crec.ifas.ufl.edu>
References: <mailman.11208.1158623738.1681.cisco-voip@puck.nether.net>
	<02e101c6dbf5$e0c44f30$400cfb0a@crec.ifas.ufl.edu>
Message-ID: <45106E47.1020606@reub.net>

+1 here with the hookswitch problem.  On my particular phone it has only started
happening in the last week or so.

I've also my own 7941 phone completely die twice due to catastrophic failure of
the flash, the phone tries to start up and gets stuck 'Waiting for flash...'
about four lines into the bootup sequence.  You'll only see this message if you
hook up your RJ12 console into the back of the phone and watch it boot, but it
has been the same problem both times.

Last time I did the RMA on it I asked the engineer if cisco were interested in
doing a failure analysis on it as it seemed odd that two phones could die in two
months with the identical same fault (which clearly looked like a component
failure) but no, sorry, not interested......

I wonder how long it will take before we start seeing field notices about these
devices.

Reuben


On 6:59 a.m., Mike Armstrong wrote:
>>From a mechanical viewpoint, the 79xx phones are junk, IMHO.  We probably 
> RMA-replace 10% of our phone population every year,  almost all because of 
> switchhook and deskstand problems.  I've asked several times if we could buy 
> replacement parts and fix the common problems ourselves, but never received 
> any response; I suspect a local repair might complicate future RMAs.  A 
> while back I had to escalate an SR because the first engineer claimed we 
> were abusing our phones, but after I offered his supervisor a chance to 
> discuss the abuse with the 64-year-old lady who was the alleged abuser, he 
> declined and we've had no problems since.
> 
> Mike Armstrong
> UF/IFAS CREC
> Lake Alfred, FL
> 
>> Date: Mon, 18 Sep 2006 16:38:18 -0500
>> From: "Kris Seraphine" <baryonyx5 at gmail.com>
>> Subject: [cisco-voip] Problems with 7941 hookswitch
>> To: cisco-voip <cisco-voip at puck.nether.net>
>> Message-ID:
>> <a749d0620609181438s75b68085maeaae1d01a9fe54b at mail.gmail.com>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Hi
>>
>> Is anyone else seeing a lot of 7941's with bad hookswitches?
>>
>> We've had to RMA about 10 of them in the last month.  The switch pops up
>> normally when the phone is taken off hook but the user doesn't get dial
>> tone.  If you press the hookswitch 5 or 6 times eventually you get a tone
>> but there's definately something not working right.
>>
>> Just wondering.
>>
>> -- 
>> kris seraphine
> 
> 

From lohnert at gmail.com  Wed Sep 20 06:01:42 2006
From: lohnert at gmail.com (Martin Lohnert)
Date: Wed, 20 Sep 2006 12:01:42 +0200
Subject: [cisco-voip] VG224 cables
Message-ID: <1bfd8b6f0609200301v2c142dc9ka22d2062c4a37fb7@mail.gmail.com>

Hi,
where do you guys get the RJ21-to-RJ45-patch-panel cables for VG224's ? So
far we've made them ourselves, but it just takes too much time and effort
and lacks reliability. I've searched the web for a good source, but most
places seem to have patch panels with 24 Rj45 ports and 2 RJ21's while VG224
has only one RJ21... Two of these can also be used for a VG248. Any tips who
manufactures/sells these...?

Cheers,
Martin
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From jason.aarons at us.didata.com  Wed Sep 20 07:54:00 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 20 Sep 2006 07:54:00 -0400
Subject: [cisco-voip] VG224 cables
Message-ID: <C1FE15183DA37645BC0633BC604E44F00333FB40@USNAEXCH.na.didata.local>

AbleComm sold me a "tailblock" that we then cross-connected the
modems/faxes.

 

http://www.phonegeeks.com/tailwitatcab.html

 

For next project I need to go to from VG224 to RJ-45 block

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Martin Lohnert
Sent: Wednesday, September 20, 2006 6:02 AM
To: Cisco Voip
Subject: [cisco-voip] VG224 cables

 

Hi,

where do you guys get the RJ21-to-RJ45-patch-panel cables for VG224's ?
So far we've made them ourselves, but it just takes too much time and
effort and lacks reliability. I've searched the web for a good source,
but most places seem to have patch panels with 24 Rj45 ports and 2
RJ21's while VG224 has only one RJ21... Two of these can also be used
for a VG248. Any tips who manufactures/sells these...? 

 

Cheers,

Martin

 




-----------------------------------------
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From luis.simonet at gmail.com  Wed Sep 20 09:09:08 2006
From: luis.simonet at gmail.com (Luis Simonet)
Date: Wed, 20 Sep 2006 09:09:08 -0400
Subject: [cisco-voip] Protocol mismatch
Message-ID: <e268831b0609200609u278e57e1m85d13c5d4328452a@mail.gmail.com>

I have a few 7940 that will not come up and are displaying protocol
mismatch. Any ideas why?

Luis
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From PRuttman at foley.com  Wed Sep 20 10:07:58 2006
From: PRuttman at foley.com (Ruttman, Peter G.)
Date: Wed, 20 Sep 2006 09:07:58 -0500
Subject: [cisco-voip] ccmadmin lock-ups
Message-ID: <7E3064B7EFC16C41B185C41D141008B7023AC292@FOLEYEX97.foleylaw.com>

I received a new laptop with SP2 and I am having IE lock-ups and having
IE close while in ccmadmin.  Has anyone else experienced this and found
a solution?  I have tried a couple versions of java as well with
1.5.0_06-b05 on there currently.  I downgraded to this from a newer one
because other non-SP2 users are using this without issue.  Any help?

Pete Ruttman

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 dekeel at pvamu.edu  Wed Sep 20 10:14:57 2006
From: dekeel at pvamu.edu (Keel,Donald)
Date: Wed, 20 Sep 2006 09:14:57 -0500
Subject: [cisco-voip] ccmadmin lock-ups
Message-ID: <857527AC06FE4F4C81FF40C0CAECBB0288ABC5@PVEXCH01.PV.PVPANTHER.NET>

I have the same issue two month back. But for me it was the Virus Scan
software on the server I change the software not to scan the web file on
the "C:\CiscoWebs" and fix my issue. 

 

Donald Keel * Network Administrator 2, Managed Services * Sungard *
Higher Education * Prairie View A&M University * P.O. Box 4019, Prairie
View, TX 77446 * Tel 936-261-9319 * Fax 936-261-2244 * dekeel at pvamu.edu
<mailto:dekeel at pvamu.edu>  * Donald.Keel at sungardhe.com
<mailto:Donald.Keel at sungardhe.com>  * www.sungardhe.com
<http://www.sungardhe.com/> 

CONFIDENTIALITY: This e-mail (including any attachments) may contain
confidential, proprietary and privileged information, and unauthorized
disclosure or use is prohibited. If you received this e-mail in error,
please notify the sender and delete this e-mail from your system.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter
G.
Sent: Wednesday, September 20, 2006 9:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ccmadmin lock-ups

 

I received a new laptop with SP2 and I am having IE lock-ups and having
IE close while in ccmadmin.  Has anyone else experienced this and found
a solution?  I have tried a couple versions of java as well with
1.5.0_06-b05 on there currently.  I downgraded to this from a newer one
because other non-SP2 users are using this without issue.  Any help?

Pete Ruttman 


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
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From keith.klevenski at rig.net  Wed Sep 20 10:26:00 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Wed, 20 Sep 2006 09:26:00 -0500
Subject: [cisco-voip] ccmadmin lock-ups
Message-ID: <7F0CE0A1A0961E498A41329535342266D3021C@HOUEXH02.rignet.org>

I have regularly had IE lockups and freezes at two different jobs on many different ccm servers connecting from many different clients for the last 2 ? years.  I've just accepted that it is just part of the game.  Sometimes I will search for a phone or route pattern and as soon as I click Find the entire browser will just disappear.  I thought it may have to do with using the Maxthon browser (tabbed front end for IE), but the same thing happens in IE, Maxthon, and other tabbed IE front ends.  I've tried IETab with Firefox as well, but it makes no difference since IE is still being used.  It just brings FF down with it.

 

I will certainly try configure the virus scanner not to scan CiscoWebs and see if that makes a difference.  Although it doesn't happen every day it happens often enough to be considered very annoying.  I'd say about once or twice a week the browser will lock up on me on average.

 

Keith

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keel,Donald
Sent: Wednesday, September 20, 2006 9:15 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have the same issue two month back. But for me it was the Virus Scan software on the server I change the software not to scan the web file on the "C:\CiscoWebs" and fix my issue. 

 

Donald Keel * Network Administrator 2, Managed Services * Sungard * Higher Education * Prairie View A&M University * P.O. Box 4019, Prairie View, TX 77446 * Tel 936-261-9319 * Fax 936-261-2244 * dekeel at pvamu.edu <mailto:dekeel at pvamu.edu>  * Donald.Keel at sungardhe.com <mailto:Donald.Keel at sungardhe.com>  * www.sungardhe.com <http://www.sungardhe.com/> 

CONFIDENTIALITY: This e-mail (including any attachments) may contain confidential, proprietary and privileged information, and unauthorized disclosure or use is prohibited. If you received this e-mail in error, please notify the sender and delete this e-mail from your system.

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter G.
Sent: Wednesday, September 20, 2006 9:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ccmadmin lock-ups

 

I received a new laptop with SP2 and I am having IE lock-ups and having IE close while in ccmadmin.  Has anyone else experienced this and found a solution?  I have tried a couple versions of java as well with 1.5.0_06-b05 on there currently.  I downgraded to this from a newer one because other non-SP2 users are using this without issue.  Any help?

Pete Ruttman 


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 wsisk at cisco.com  Wed Sep 20 10:27:09 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 20 Sep 2006 10:27:09 -0400
Subject: [cisco-voip] Protocol mismatch
In-Reply-To: <e268831b0609200609u278e57e1m85d13c5d4328452a@mail.gmail.com>
References: <e268831b0609200609u278e57e1m85d13c5d4328452a@mail.gmail.com>
Message-ID: <45114FBD.4020400@cisco.com>

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From jason.aarons at us.didata.com  Wed Sep 20 10:30:24 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 20 Sep 2006 10:30:24 -0400
Subject: [cisco-voip] ccmadmin lock-ups
Message-ID: <C1FE15183DA37645BC0633BC604E44F00333FE93@USNAEXCH.na.didata.local>

Does Sun Microsystems offer any kind of support? I think the issue is around the Java Plug In and/or code. I've seen lockups more in CCMAdmin 4.x than 3.x.

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Wednesday, September 20, 2006 10:26 AM
To: Keel,Donald; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have regularly had IE lockups and freezes at two different jobs on many different ccm servers connecting from many different clients for the last 2 ? years.  I've just accepted that it is just part of the game.  Sometimes I will search for a phone or route pattern and as soon as I click Find the entire browser will just disappear.  I thought it may have to do with using the Maxthon browser (tabbed front end for IE), but the same thing happens in IE, Maxthon, and other tabbed IE front ends.  I've tried IETab with Firefox as well, but it makes no difference since IE is still being used.  It just brings FF down with it.

 

I will certainly try configure the virus scanner not to scan CiscoWebs and see if that makes a difference.  Although it doesn't happen every day it happens often enough to be considered very annoying.  I'd say about once or twice a week the browser will lock up on me on average.

 

Keith

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keel,Donald
Sent: Wednesday, September 20, 2006 9:15 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have the same issue two month back. But for me it was the Virus Scan software on the server I change the software not to scan the web file on the "C:\CiscoWebs" and fix my issue. 

 

Donald Keel * Network Administrator 2, Managed Services * Sungard * Higher Education * Prairie View A&M University * P.O. Box 4019, Prairie View, TX 77446 * Tel 936-261-9319 * Fax 936-261-2244 * dekeel at pvamu.edu <mailto:dekeel at pvamu.edu>  * Donald.Keel at sungardhe.com <mailto:Donald.Keel at sungardhe.com>  * www.sungardhe.com <http://www.sungardhe.com/> 

CONFIDENTIALITY: This e-mail (including any attachments) may contain confidential, proprietary and privileged information, and unauthorized disclosure or use is prohibited. If you received this e-mail in error, please notify the sender and delete this e-mail from your system.

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter G.
Sent: Wednesday, September 20, 2006 9:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ccmadmin lock-ups

 

I received a new laptop with SP2 and I am having IE lock-ups and having IE close while in ccmadmin.  Has anyone else experienced this and found a solution?  I have tried a couple versions of java as well with 1.5.0_06-b05 on there currently.  I downgraded to this from a newer one because other non-SP2 users are using this without issue.  Any help?

Pete Ruttman 


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 mark.turpin at bryancave.com  Wed Sep 20 11:17:19 2006
From: mark.turpin at bryancave.com (Turpin, Mark)
Date: Wed, 20 Sep 2006 11:17:19 -0400
Subject: [cisco-voip] ccmadmin lock-ups
In-Reply-To: <7F0CE0A1A0961E498A41329535342266D3021C@HOUEXH02.rignet.org>
Message-ID: <2496838411CFA74BAFD10B4C3757616708FC43@FRMEXBE02.bc.firm.local>

I have to chime in on this thread.  Everyone I work with, some with different builds, some with different JRE's, doesn't matter... we all lock up regularly.  It drives all of us nuts!  I agree it seems JRE related, but as much as the problem annoys me I haven't had the time to research it on Sun's site.  I saw it in 3.x and agree again that it seems more prevalent in 4.x (running 4.2 now).

 

For me, I see it regularly on the phone or gateway pages.  Sometimes on route patterns, but since most of my work revolves around phones & gateways day in and day out I probably see it there more due to my usage patterns.

 

I really wish there was something that could be done on Cisco's side to address the issue that we've all come to just accept as an annoying hazard of the job....

 

-Mark

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Wednesday, September 20, 2006 9:26 AM
To: Keel,Donald; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have regularly had IE lockups and freezes at two different jobs on many different ccm servers connecting from many different clients for the last 2 ? years.  I've just accepted that it is just part of the game.  Sometimes I will search for a phone or route pattern and as soon as I click Find the entire browser will just disappear.  I thought it may have to do with using the Maxthon browser (tabbed front end for IE), but the same thing happens in IE, Maxthon, and other tabbed IE front ends.  I've tried IETab with Firefox as well, but it makes no difference since IE is still being used.  It just brings FF down with it.

 

I will certainly try configure the virus scanner not to scan CiscoWebs and see if that makes a difference.  Although it doesn't happen every day it happens often enough to be considered very annoying.  I'd say about once or twice a week the browser will lock up on me on average.

 

Keith

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keel,Donald
Sent: Wednesday, September 20, 2006 9:15 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have the same issue two month back. But for me it was the Virus Scan software on the server I change the software not to scan the web file on the "C:\CiscoWebs" and fix my issue. 

 

Donald Keel * Network Administrator 2, Managed Services * Sungard * Higher Education * Prairie View A&M University * P.O. Box 4019, Prairie View, TX 77446 * Tel 936-261-9319 * Fax 936-261-2244 * dekeel at pvamu.edu <mailto:dekeel at pvamu.edu>  * Donald.Keel at sungardhe.com <mailto:Donald.Keel at sungardhe.com>  * www.sungardhe.com <http://www.sungardhe.com/> 

CONFIDENTIALITY: This e-mail (including any attachments) may contain confidential, proprietary and privileged information, and unauthorized disclosure or use is prohibited. If you received this e-mail in error, please notify the sender and delete this e-mail from your system.

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter G.
Sent: Wednesday, September 20, 2006 9:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ccmadmin lock-ups

 

I received a new laptop with SP2 and I am having IE lock-ups and having IE close while in ccmadmin.  Has anyone else experienced this and found a solution?  I have tried a couple versions of java as well with 1.5.0_06-b05 on there currently.  I downgraded to this from a newer one because other non-SP2 users are using this without issue.  Any help?

Pete Ruttman 


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. 
 
This electronic message is from a law firm. It may contain confidential or privileged information. If you received this transmission in error, please reply to the sender to advise of the error and delete this transmission and any attachments.
 
IRS Circular 230 Disclosure: To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice contained in this communication (including any attachments) is not intended or written 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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From tim.reimers at asheville.k12.nc.us  Wed Sep 20 11:26:27 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 20 Sep 2006 11:26:27 -0400
Subject: [cisco-voip] Time on CCMs and phones..
Message-ID: <050944EE48FF384F9C663F7048C47DA704CD3AA3@webmail.asheville.k12.nc.us>

hi everyone-

I'm having trouble with drifting time on phones-

My setup -
acsccm1 and acsccm2 (publisher/subscriber on 4.01)
acsunity (Unity 4.x)

Supposedly, acsccm1 and acsccm2 get their time from acsunity
acsunity gets its time from ntp.bellsouth.net
That's the same timeserver my main data network uses.

What's weird is that all three 'sets' of servers vary in their time -
ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
But Unity this AM was more than 10 minutes different from the two CCMs
And all the VOIP servers were at least 12 minutes different from my data
timeservers.

No firewalls are blocking timeserver ports- a freeware 'get the time'
utility can use ntp.bellsouth.net and successfully get the time and
display it - on all three servers.
No timezone differences (that would result in whole hours being
different anyhow, not just minute differences)


Why are these servers not keeping time with each other, when both are
looking to ACSUNITY, at the very least, and 
Unity is looking at the same public timeserver that my main DCs are for
my data network?


Here's some w32TM output from the various servers.
>From ACSCCM1  - you can see that it's NTP server is acsunity
C:\>w32tm -source -v
W32Time: BEGIN:InitAdjIncr
W32Time:    Adj 156254 , Incr 156250  fAdjust 0
W32Time:    156254 Adj!=Incr 156250
W32Time: END:Line 2503
W32Time: BEGIN:TsUpTheThread
W32Time: END  Line 1407
W32Time: TimeMMInit()
W32Time: Kernel timer : using default maximum resolution
W32Time:                MaximumTime = 156250
W32Time:                CurrentTime = 9766
W32Time: Timer calibrated, looped 1 times
W32Time: BEGIN:InitTmCfg
W32Time: END:Line 807
W32Time: BEGIN:InitTmCli
W32Time: END:Line 2596
W32Time: BEGIN:InitTmData
W32Time: END:Line 2618
W32Time: AvoidTimeSyncOnWan 0
W32Time: ntpserver - acsunity
W32Time: BEGIN:CMOSSynchSet
W32Time:    Setting adjustment 156254 - Bool  0
W32Time:    BEGIN:SetTSTimeRes
W32Time:    END:Line 1295
W32Time: END:Line 864
W32Time: BEGIN:InitializeDC
W32Time:    BEGIN:GetRole
W32Time:       Role is 'workstation'
W32Time:    END  Line 672
W32Time: END:Line 704





w32TM output from ACSUNITY - shows ntp.bellsouth.net 
C:\Documents and Settings\Administrator>w32tm -v -source
W32Time: BEGIN:InitAdjIncr
W32Time:    Adj 156250 , Incr 156250  fAdjust 0
W32Time: END:Line 2503
W32Time: BEGIN:TsUpTheThread
W32Time: END  Line 1407
W32Time: TimeMMInit()
W32Time: Kernel timer : using default maximum resolution
W32Time:                MaximumTime = 156250
W32Time:                CurrentTime = 9766
W32Time: Timer calibrated, looped 1 times
W32Time: BEGIN:InitTmCfg
W32Time: END:Line 807
W32Time: BEGIN:InitTmCli
W32Time: END:Line 2596
W32Time: BEGIN:InitTmData
W32Time: END:Line 2618
W32Time: AvoidTimeSyncOnWan 0
W32Time: ntpserver - ntp.bellsouth.net
W32Time: BEGIN:CMOSSynchSet
W32Time:    Setting adjustment 156250 - Bool  0
W32Time:    BEGIN:SetTSTimeRes
W32Time:    END:Line 1295
W32Time: END:Line 864
W32Time: BEGIN:InitializeDC
W32Time:    BEGIN:GetRole
W32Time:       Role is 'PDC'
W32Time:    END  Line 672
W32Time:    BEGIN:FetchParentDomainName
W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
ptr 0
W32Time:    END:Line 782
W32Time: END:Line 704
W32Time: Server: Binding to 1 NIC.
W32Time: bind failed: 0x80072740
W32Time: Logging event 0xC0000031. 15 min until this event is allowed
again.
W32Time: 0xC0000031 reported to System Log in Event Viewer
W32Time: BEGIN:FinishCleanup
W32Time:    BEGIN:TsUpTheThread
W32Time:    END  Line 1407
W32Time:    BEGIN:UnInitializeDC
W32Time:       Ptrs 0 - 0
W32Time:    END:Line 727
W32Time:    Time service stopped.
W32Time: END:Line 407




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From keith.klevenski at rig.net  Wed Sep 20 11:31:33 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Wed, 20 Sep 2006 10:31:33 -0500
Subject: [cisco-voip] ccmadmin lock-ups
Message-ID: <7F0CE0A1A0961E498A41329535342266D3022C@HOUEXH02.rignet.org>

I'm glad to hear I'm not the only one experiencing this!

 

________________________________

From: Turpin, Mark [mailto:mark.turpin at bryancave.com] 
Sent: Wednesday, September 20, 2006 10:17 AM
To: Keith Klevenski; Keel,Donald; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ccmadmin lock-ups

 

I have to chime in on this thread.  Everyone I work with, some with different builds, some with different JRE's, doesn't matter... we all lock up regularly.  It drives all of us nuts!  I agree it seems JRE related, but as much as the problem annoys me I haven't had the time to research it on Sun's site.  I saw it in 3.x and agree again that it seems more prevalent in 4.x (running 4.2 now).

 

For me, I see it regularly on the phone or gateway pages.  Sometimes on route patterns, but since most of my work revolves around phones & gateways day in and day out I probably see it there more due to my usage patterns.

 

I really wish there was something that could be done on Cisco's side to address the issue that we've all come to just accept as an annoying hazard of the job....

 

-Mark

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Wednesday, September 20, 2006 9:26 AM
To: Keel,Donald; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have regularly had IE lockups and freezes at two different jobs on many different ccm servers connecting from many different clients for the last 2 ? years.  I've just accepted that it is just part of the game.  Sometimes I will search for a phone or route pattern and as soon as I click Find the entire browser will just disappear.  I thought it may have to do with using the Maxthon browser (tabbed front end for IE), but the same thing happens in IE, Maxthon, and other tabbed IE front ends.  I've tried IETab with Firefox as well, but it makes no difference since IE is still being used.  It just brings FF down with it.

 

I will certainly try configure the virus scanner not to scan CiscoWebs and see if that makes a difference.  Although it doesn't happen every day it happens often enough to be considered very annoying.  I'd say about once or twice a week the browser will lock up on me on average.

 

Keith

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keel,Donald
Sent: Wednesday, September 20, 2006 9:15 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have the same issue two month back. But for me it was the Virus Scan software on the server I change the software not to scan the web file on the "C:\CiscoWebs" and fix my issue. 

 

Donald Keel * Network Administrator 2, Managed Services * Sungard * Higher Education * Prairie View A&M University * P.O. Box 4019, Prairie View, TX 77446 * Tel 936-261-9319 * Fax 936-261-2244 * dekeel at pvamu.edu <mailto:dekeel at pvamu.edu>  * Donald.Keel at sungardhe.com <mailto:Donald.Keel at sungardhe.com>  * www.sungardhe.com <http://www.sungardhe.com/> 

CONFIDENTIALITY: This e-mail (including any attachments) may contain confidential, proprietary and privileged information, and unauthorized disclosure or use is prohibited. If you received this e-mail in error, please notify the sender and delete this e-mail from your system.

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter G.
Sent: Wednesday, September 20, 2006 9:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ccmadmin lock-ups

 

I received a new laptop with SP2 and I am having IE lock-ups and having IE close while in ccmadmin.  Has anyone else experienced this and found a solution?  I have tried a couple versions of java as well with 1.5.0_06-b05 on there currently.  I downgraded to this from a newer one because other non-SP2 users are using this without issue.  Any help?

Pete Ruttman 


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.

 

This electronic message is from a law firm. It may contain confidential or privileged information. If you received this transmission in error, please reply to the sender to advise of the error and delete this transmission and any attachments.

 

IRS Circular 230 Disclosure: To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice contained in this communication (including any attachments) is not intended or written 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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From Timothy_Frazee at adp.com  Wed Sep 20 11:45:04 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Wed, 20 Sep 2006 10:45:04 -0500
Subject: [cisco-voip] ccmadmin lock-ups
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3B01260A23@EXCHANGEDS01.ds.ad.adp.com>

What about using the JRE that's in the c:\utils dir. I've always had good luck with that one.

 

Timothy Frazee 

Network Solutions

314.249.8697 cellular phone

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Wednesday, September 20, 2006 10:32 AM
To: Turpin, Mark; Keel,Donald; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I'm glad to hear I'm not the only one experiencing this!

 

________________________________

From: Turpin, Mark [mailto:mark.turpin at bryancave.com] 
Sent: Wednesday, September 20, 2006 10:17 AM
To: Keith Klevenski; Keel,Donald; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ccmadmin lock-ups

 

I have to chime in on this thread.  Everyone I work with, some with different builds, some with different JRE's, doesn't matter... we all lock up regularly.  It drives all of us nuts!  I agree it seems JRE related, but as much as the problem annoys me I haven't had the time to research it on Sun's site.  I saw it in 3.x and agree again that it seems more prevalent in 4.x (running 4.2 now).

 

For me, I see it regularly on the phone or gateway pages.  Sometimes on route patterns, but since most of my work revolves around phones & gateways day in and day out I probably see it there more due to my usage patterns.

 

I really wish there was something that could be done on Cisco's side to address the issue that we've all come to just accept as an annoying hazard of the job....

 

-Mark

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Wednesday, September 20, 2006 9:26 AM
To: Keel,Donald; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have regularly had IE lockups and freezes at two different jobs on many different ccm servers connecting from many different clients for the last 2 ? years.  I've just accepted that it is just part of the game.  Sometimes I will search for a phone or route pattern and as soon as I click Find the entire browser will just disappear.  I thought it may have to do with using the Maxthon browser (tabbed front end for IE), but the same thing happens in IE, Maxthon, and other tabbed IE front ends.  I've tried IETab with Firefox as well, but it makes no difference since IE is still being used.  It just brings FF down with it.

 

I will certainly try configure the virus scanner not to scan CiscoWebs and see if that makes a difference.  Although it doesn't happen every day it happens often enough to be considered very annoying.  I'd say about once or twice a week the browser will lock up on me on average.

 

Keith

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keel,Donald
Sent: Wednesday, September 20, 2006 9:15 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have the same issue two month back. But for me it was the Virus Scan software on the server I change the software not to scan the web file on the "C:\CiscoWebs" and fix my issue. 

 

Donald Keel * Network Administrator 2, Managed Services * Sungard * Higher Education * Prairie View A&M University * P.O. Box 4019, Prairie View, TX 77446 * Tel 936-261-9319 * Fax 936-261-2244 * dekeel at pvamu.edu <mailto:dekeel at pvamu.edu>  * Donald.Keel at sungardhe.com <mailto:Donald.Keel at sungardhe.com>  * www.sungardhe.com <http://www.sungardhe.com/> 

CONFIDENTIALITY: This e-mail (including any attachments) may contain confidential, proprietary and privileged information, and unauthorized disclosure or use is prohibited. If you received this e-mail in error, please notify the sender and delete this e-mail from your system.

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter G.
Sent: Wednesday, September 20, 2006 9:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ccmadmin lock-ups

 

I received a new laptop with SP2 and I am having IE lock-ups and having IE close while in ccmadmin.  Has anyone else experienced this and found a solution?  I have tried a couple versions of java as well with 1.5.0_06-b05 on there currently.  I downgraded to this from a newer one because other non-SP2 users are using this without issue.  Any help?

Pete Ruttman 


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.

 

This electronic message is from a law firm. It may contain confidential or privileged information. If you received this transmission in error, please reply to the sender to advise of the error and delete this transmission and any attachments.

 

IRS Circular 230 Disclosure: To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice contained in this communication (including any attachments) is not intended or written 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.



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 BRogers at pasadenaisd.org  Wed Sep 20 11:47:05 2006
From: BRogers at pasadenaisd.org (Brad Rogers)
Date: Wed, 20 Sep 2006 10:47:05 -0500
Subject: [cisco-voip] ccmadmin lock-ups
In-Reply-To: <7F0CE0A1A0961E498A41329535342266D3022C@HOUEXH02.rignet.org>
Message-ID: <CB3DDB018C56BB42A32D5C63006B76ED0100414F@adminexchg03.pasadenaisd.org>

We have the same problem.  It's terribly annoying.

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Wednesday, September 20, 2006 10:32 AM
To: Turpin, Mark; Keel,Donald; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I'm glad to hear I'm not the only one experiencing this!

 

________________________________

From: Turpin, Mark [mailto:mark.turpin at bryancave.com] 
Sent: Wednesday, September 20, 2006 10:17 AM
To: Keith Klevenski; Keel,Donald; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] ccmadmin lock-ups

 

I have to chime in on this thread.  Everyone I work with, some with different builds, some with different JRE's, doesn't matter... we all lock up regularly.  It drives all of us nuts!  I agree it seems JRE related, but as much as the problem annoys me I haven't had the time to research it on Sun's site.  I saw it in 3.x and agree again that it seems more prevalent in 4.x (running 4.2 now).

 

For me, I see it regularly on the phone or gateway pages.  Sometimes on route patterns, but since most of my work revolves around phones & gateways day in and day out I probably see it there more due to my usage patterns.

 

I really wish there was something that could be done on Cisco's side to address the issue that we've all come to just accept as an annoying hazard of the job....

 

-Mark

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Wednesday, September 20, 2006 9:26 AM
To: Keel,Donald; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have regularly had IE lockups and freezes at two different jobs on many different ccm servers connecting from many different clients for the last 2 ? years.  I've just accepted that it is just part of the game.  Sometimes I will search for a phone or route pattern and as soon as I click Find the entire browser will just disappear.  I thought it may have to do with using the Maxthon browser (tabbed front end for IE), but the same thing happens in IE, Maxthon, and other tabbed IE front ends.  I've tried IETab with Firefox as well, but it makes no difference since IE is still being used.  It just brings FF down with it.

 

I will certainly try configure the virus scanner not to scan CiscoWebs and see if that makes a difference.  Although it doesn't happen every day it happens often enough to be considered very annoying.  I'd say about once or twice a week the browser will lock up on me on average.

 

Keith

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keel,Donald
Sent: Wednesday, September 20, 2006 9:15 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups

 

I have the same issue two month back. But for me it was the Virus Scan software on the server I change the software not to scan the web file on the "C:\CiscoWebs" and fix my issue. 

 

Donald Keel * Network Administrator 2, Managed Services * Sungard * Higher Education * Prairie View A&M University * P.O. Box 4019, Prairie View, TX 77446 * Tel 936-261-9319 * Fax 936-261-2244 * dekeel at pvamu.edu <mailto:dekeel at pvamu.edu>  * Donald.Keel at sungardhe.com <mailto:Donald.Keel at sungardhe.com>  * www.sungardhe.com <http://www.sungardhe.com/> 

CONFIDENTIALITY: This e-mail (including any attachments) may contain confidential, proprietary and privileged information, and unauthorized disclosure or use is prohibited. If you received this e-mail in error, please notify the sender and delete this e-mail from your system.

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter G.
Sent: Wednesday, September 20, 2006 9:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ccmadmin lock-ups

 

I received a new laptop with SP2 and I am having IE lock-ups and having IE close while in ccmadmin.  Has anyone else experienced this and found a solution?  I have tried a couple versions of java as well with 1.5.0_06-b05 on there currently.  I downgraded to this from a newer one because other non-SP2 users are using this without issue.  Any help?

Pete Ruttman 


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.

 

This electronic message is from a law firm. It may contain confidential or privileged information. If you received this transmission in error, please reply to the sender to advise of the error and delete this transmission and any attachments.

 

IRS Circular 230 Disclosure: To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice contained in this communication (including any attachments) is not intended or written 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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From bills at tns.its.psu.edu  Wed Sep 20 11:47:19 2006
From: bills at tns.its.psu.edu (Simon, Bill)
Date: Wed, 20 Sep 2006 11:47:19 -0400
Subject: [cisco-voip] Time on CCMs and phones..
In-Reply-To: <050944EE48FF384F9C663F7048C47DA704CD3AA3@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA704CD3AA3@webmail.asheville.k12.nc.us>
Message-ID: <45116287.9020901@tns.its.psu.edu>

We've found that the NTPD service (Cisco NTP) that comes with 
CallManager is right-on and very reliable.  We use that instead of the 
Windows time service.  That would be my recommendation.


Tim Reimers wrote:
> hi everyone-
> 
> I'm having trouble with drifting time on phones-
> 
> My setup -
> acsccm1 and acsccm2 (publisher/subscriber on 4.01)
> acsunity (Unity 4.x)
> 
> Supposedly, acsccm1 and acsccm2 get their time from acsunity
> acsunity gets its time from ntp.bellsouth.net
> That?s the same timeserver my main data network uses.
> 
> What's weird is that all three 'sets' of servers vary in their time -
> ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
> But Unity this AM was more than 10 minutes different from the two CCMs
> And all the VOIP servers were at least 12 minutes different from my data 
> timeservers.
> 
> No firewalls are blocking timeserver ports- a freeware 'get the time' 
> utility can use ntp.bellsouth.net and successfully get the time and 
> display it - on all three servers.
> 
> No timezone differences (that would result in whole hours being 
> different anyhow, not just minute differences)
> 
> 
> Why are these servers not keeping time with each other, when both are 
> looking to ACSUNITY, at the very least, and
> Unity is looking at the same public timeserver that my main DCs are for 
> my data network?
> 
> 
> Here's some w32TM output from the various servers.
> _*From ACSCCM1  - you can see that it's NTP server is acsunity*_**
> C:\>w32tm -source -v
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156254 , Incr 156250  fAdjust 0
> W32Time:    156254 Adj!=Incr 156250
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - acsunity
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156254 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'workstation'
> W32Time:    END  Line 672
> W32Time: END:Line 704
> 
> 
> 
> 
> 
> _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_**
> C:\Documents and Settings\Administrator>w32tm -v -source
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156250 , Incr 156250  fAdjust 0
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - ntp.bellsouth.net
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156250 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'PDC'
> W32Time:    END  Line 672
> W32Time:    BEGIN:FetchParentDomainName
> W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with ptr 0
> W32Time:    END:Line 782
> W32Time: END:Line 704
> W32Time: Server: Binding to 1 NIC.
> W32Time: bind failed: 0x80072740
> W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
> again.
> W32Time: 0xC0000031 reported to System Log in Event Viewer
> W32Time: BEGIN:FinishCleanup
> W32Time:    BEGIN:TsUpTheThread
> W32Time:    END  Line 1407
> W32Time:    BEGIN:UnInitializeDC
> W32Time:       Ptrs 0 - 0
> W32Time:    END:Line 727
> W32Time:    Time service stopped.
> W32Time: END:Line 407
> 
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

-- 
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/

From milay at cisco.com  Wed Sep 20 11:55:08 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Wed, 20 Sep 2006 08:55:08 -0700
Subject: [cisco-voip] Time on CCMs and phones..
Message-ID: <413FABF5DCE22849AF86F442E472B78BCB5B2A@xmb-sjc-219.amer.cisco.com>

You may have XNTP already install on your servers, installed as part of
any 3.3.3 or greater CCM install. Please ensure that you do not have
more than on time client running on any given server at one time. At
Cisco we prefer XNTP for a CCM implementation, by default the Pub is the
master & the Subs point to the Pub for time.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Simon, Bill
Sent: Wednesday, September 20, 2006 10:47 AM
To: Tim Reimers
Cc: Cisco Voip
Subject: Re: [cisco-voip] Time on CCMs and phones..

We've found that the NTPD service (Cisco NTP) that comes with
CallManager is right-on and very reliable.  We use that instead of the
Windows time service.  That would be my recommendation.


Tim Reimers wrote:
> hi everyone-
> 
> I'm having trouble with drifting time on phones-
> 
> My setup -
> acsccm1 and acsccm2 (publisher/subscriber on 4.01) acsunity (Unity 
> 4.x)
> 
> Supposedly, acsccm1 and acsccm2 get their time from acsunity acsunity 
> gets its time from ntp.bellsouth.net That's the same timeserver my 
> main data network uses.
> 
> What's weird is that all three 'sets' of servers vary in their time -
> ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
> But Unity this AM was more than 10 minutes different from the two CCMs

> And all the VOIP servers were at least 12 minutes different from my 
> data timeservers.
> 
> No firewalls are blocking timeserver ports- a freeware 'get the time' 
> utility can use ntp.bellsouth.net and successfully get the time and 
> display it - on all three servers.
> 
> No timezone differences (that would result in whole hours being 
> different anyhow, not just minute differences)
> 
> 
> Why are these servers not keeping time with each other, when both are 
> looking to ACSUNITY, at the very least, and Unity is looking at the 
> same public timeserver that my main DCs are for my data network?
> 
> 
> Here's some w32TM output from the various servers.
> _*From ACSCCM1  - you can see that it's NTP server is acsunity*_** 
> C:\>w32tm -source -v
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156254 , Incr 156250  fAdjust 0
> W32Time:    156254 Adj!=Incr 156250
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - acsunity
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156254 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'workstation'
> W32Time:    END  Line 672
> W32Time: END:Line 704
> 
> 
> 
> 
> 
> _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_** 
> C:\Documents and Settings\Administrator>w32tm -v -source
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156250 , Incr 156250  fAdjust 0
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - ntp.bellsouth.net
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156250 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'PDC'
> W32Time:    END  Line 672
> W32Time:    BEGIN:FetchParentDomainName
> W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
ptr 0
> W32Time:    END:Line 782
> W32Time: END:Line 704
> W32Time: Server: Binding to 1 NIC.
> W32Time: bind failed: 0x80072740
> W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
> again.
> W32Time: 0xC0000031 reported to System Log in Event Viewer
> W32Time: BEGIN:FinishCleanup
> W32Time:    BEGIN:TsUpTheThread
> W32Time:    END  Line 1407
> W32Time:    BEGIN:UnInitializeDC
> W32Time:       Ptrs 0 - 0
> W32Time:    END:Line 727
> W32Time:    Time service stopped.
> W32Time: END:Line 407
> 
> 
> 
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

--
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


From mfong at pacific.edu  Wed Sep 20 12:25:53 2006
From: mfong at pacific.edu (Melvin Fong)
Date: Wed, 20 Sep 2006 09:25:53 -0700
Subject: [cisco-voip] busy out PRI
Message-ID: <451109210200004E00012ECD@gwsmtp.pacific.edu>

Does anyone know of a way to busy out a PRI using call manager?
I don't see anything on the gateway config.  The only way I know of is
to busy the port that is connected to the PRI.  The question is whether
CM will recognize the circuit as not usable. If the port is disabled
then the D link should go down and hopefully CM will send the calls to
the 2nd PRI on the route list.  I haven't tested this yet and hoped
someone else has already.  Thanks
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From janderson at enventis.com  Wed Sep 20 12:50:20 2006
From: janderson at enventis.com (Jeff Anderson)
Date: Wed, 20 Sep 2006 11:50:20 -0500
Subject: [cisco-voip] Problems with 7941 hookswitch
In-Reply-To: <45106E47.1020606@reub.net>
Message-ID: <20060920115020825.00000002296@janderson-3948>

We have been experiencing this problem as well. I believe it has happened at every one of our deployments so far, but only with 41/61's phones. I think the problem is bug 

http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCsc99161&Submit=Search

I just noticed right now that they updated the bug. Before it only said the 7970's were affected, but my TAC guy told me several cases with 7941/61 have been reported as well. Now it includes those phone models.

I just tested my first 8.0(4)SR1 phone-load this week. I put it on a couple phones that were experiencing the problem the most. The jury is still out. If you are experiencing this problem a reset of the phone should correct the issue for a while.
 

Jeff Anderson



From rwolfe at Burwood.com  Wed Sep 20 13:04:20 2006
From: rwolfe at Burwood.com (Wolfe, Robert)
Date: Wed, 20 Sep 2006 12:04:20 -0500
Subject: [cisco-voip] Re-Routing Unallocated Calls
Message-ID: <97F472A9F637E44B954D7FBABC800430F5DA98@bur-msg-02.Burwood.com>

If a call is determined "Unallocated" by the gateway/callmanager, is
there a way to automatically send these calls to an auto attendant (or
even get them answered by callmanager and some kind of treatement
performed)?

 

Thanks in advance,

 

-rob

 

  <http://www.burwood.com> 

Robert Wolfe
Network Consultant
Burwood Group, Inc.

IP:  312.327.4634
Mobile:  312.593.0605
E-Mail:  rwolfe <mailto:rwolfe at burwood.com> @burwood.com
<mailto:rwolfe at burwood.com> 

 

  <http://www.burwood.com> 

 

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From rratliff at cisco.com  Wed Sep 20 13:03:34 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 20 Sep 2006 13:03:34 -0400
Subject: [cisco-voip] busy out PRI
In-Reply-To: <451109210200004E00012ECD@gwsmtp.pacific.edu>
References: <451109210200004E00012ECD@gwsmtp.pacific.edu>
Message-ID: <2BB03FB6-85DA-47F6-9EB8-3BAF1264EE5B@cisco.com>

Take a look at the description for the advanced CM service parameter  
"Change B-Channel Maintenance Status".
This is for MGCP PRIs only of course.  For H.323 it's all done in IOS.

-Ryan

On Sep 20, 2006, at 12:25 PM, Melvin Fong wrote:

Does anyone know of a way to busy out a PRI using call manager?
I don't see anything on the gateway config.  The only way I know of  
is to busy the port that is connected to the PRI.  The question is  
whether CM will recognize the circuit as not usable. If the port is  
disabled then the D link should go down and hopefully CM will send  
the calls to the 2nd PRI on the route list.  I haven't tested this  
yet and hoped someone else has already.  Thanks
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From mark.turpin at bryancave.com  Wed Sep 20 13:25:35 2006
From: mark.turpin at bryancave.com (Turpin, Mark)
Date: Wed, 20 Sep 2006 13:25:35 -0400
Subject: [cisco-voip] Re-Routing Unallocated Calls
In-Reply-To: <97F472A9F637E44B954D7FBABC800430F5DA98@bur-msg-02.Burwood.com>
Message-ID: <2496838411CFA74BAFD10B4C3757616708FC51@FRMEXBE02.bc.firm.local>

Sure, just setup a X mask route and point it at a Unity call handler or
your operator's DN.

 

For instance, if you're a 4-digit dial plan, suck in XXXX by putting it
in the partition where your line 1 dn's are and route it to 0 for the
operator (if so configured) or say 1000 if that were the DN for your
call handler.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wolfe, Robert
Sent: Wednesday, September 20, 2006 12:04 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Re-Routing Unallocated Calls

 

If a call is determined "Unallocated" by the gateway/callmanager, is
there a way to automatically send these calls to an auto attendant (or
even get them answered by callmanager and some kind of treatement
performed)?

 

Thanks in advance,

 

-rob

 

  <http://www.burwood.com> 

Robert Wolfe
Network Consultant
Burwood Group, Inc.

IP:  312.327.4634
Mobile:  312.593.0605
E-Mail:  rwolfe <mailto:rwolfe at burwood.com> @burwood.com
<mailto:rwolfe at burwood.com> 

 

  <http://www.burwood.com> 
 
This electronic message is from a law firm. It may contain confidential or privileged information. If you received this transmission in error, please reply to the sender to advise of the error and delete this transmission and any attachments.
 
IRS Circular 230 Disclosure: To ensure compliance with requirements imposed by the IRS, we inform you that any U.S. federal tax advice contained in this communication (including any attachments) is not intended or written 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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From ErasmuE4 at telkom.co.za  Wed Sep 20 13:36:09 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 20 Sep 2006 19:36:09 +0200
Subject: [cisco-voip] ccmadmin lock-ups
Message-ID: <472E195E692EE74BAE4E7B2880A78C1FAC187D@CNTRRA20-XCS01.telkom.co.za>


I also had some scary experiences where a call manager CPU goes to
100% - sometimes it comes down again but had cases where I hover my
mouse over the little java icon in the right hand corner and every
time I do it the CPU shoots up again.



Erik Erasmus

Telkom SA

': +27 12 680 8320

*: +27 12 680 7388

*: +27 82 4106420

*:  erasmue4 at telkom.co.za



      



________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith
Klevenski
Sent: 20 September 2006 04:26 PM
To: Keel,Donald; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups



I have regularly had IE lockups and freezes at two different jobs on
many different ccm servers connecting from many different clients for
the last 2 ? years.  I've just accepted that it is just part of the
game.  Sometimes I will search for a phone or route pattern and as
soon as I click Find the entire browser will just disappear.  I
thought it may have to do with using the Maxthon browser (tabbed front
end for IE), but the same thing happens in IE, Maxthon, and other
tabbed IE front ends.  I've tried IETab with Firefox as well, but it
makes no difference since IE is still being used.  It just brings FF
down with it.



I will certainly try configure the virus scanner not to scan CiscoWebs
and see if that makes a difference.  Although it doesn't happen every
day it happens often enough to be considered very annoying.  I'd say
about once or twice a week the browser will lock up on me on average.



Keith



________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keel,Donald
Sent: Wednesday, September 20, 2006 9:15 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccmadmin lock-ups



I have the same issue two month back. But for me it was the Virus Scan
software on the server I change the software not to scan the web file
on the "C:\CiscoWebs" and fix my issue.



Donald Keel * Network Administrator 2, Managed Services * Sungard *
Higher Education * Prairie View A&M University * P.O. Box 4019,
Prairie View, TX 77446 * Tel 936-261-9319 * Fax 936-261-2244 *
dekeel at pvamu.edu <mailto:dekeel at pvamu.edu>  *
Donald.Keel at sungardhe.com <mailto:Donald.Keel at sungardhe.com>  *
www.sungardhe.com <http://www.sungardhe.com/>

CONFIDENTIALITY: This e-mail (including any attachments) may contain
confidential, proprietary and privileged information, and unauthorized
disclosure or use is prohibited. If you received this e-mail in error,
please notify the sender and delete this e-mail from your system.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman,
Peter G.
Sent: Wednesday, September 20, 2006 9:08 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ccmadmin lock-ups



I received a new laptop with SP2 and I am having IE lock-ups and
having IE close while in ccmadmin.  Has anyone else experienced this
and found a solution?  I have tried a couple versions of java as well
with 1.5.0_06-b05 on there currently.  I downgraded to this from a
newer one because other non-SP2 users are using this without issue.
Any help?

Pete Ruttman


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From tim.reimers at asheville.k12.nc.us  Wed Sep 20 13:44:20 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 20 Sep 2006 13:44:20 -0400
Subject: [cisco-voip] Time on CCMs and phones..
Message-ID: <050944EE48FF384F9C663F7048C47DA704CD3B34@webmail.asheville.k12.nc.us>

hmmm..

I don't see that on my system- either in Services in serviceability
or in NT services in Administrative Tools...

wonder where it gets installed from in 4.01 ?? 

-----Original Message-----
From: Simon, Bill [mailto:bills at tns.its.psu.edu] 
Sent: Wednesday, September 20, 2006 11:47 AM
To: Tim Reimers
Cc: Cisco Voip
Subject: Re: [cisco-voip] Time on CCMs and phones..

We've found that the NTPD service (Cisco NTP) that comes with
CallManager is right-on and very reliable.  We use that instead of the
Windows time service.  That would be my recommendation.


Tim Reimers wrote:
> hi everyone-
> 
> I'm having trouble with drifting time on phones-
> 
> My setup -
> acsccm1 and acsccm2 (publisher/subscriber on 4.01) acsunity (Unity 
> 4.x)
> 
> Supposedly, acsccm1 and acsccm2 get their time from acsunity acsunity 
> gets its time from ntp.bellsouth.net That's the same timeserver my 
> main data network uses.
> 
> What's weird is that all three 'sets' of servers vary in their time -
> ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
> But Unity this AM was more than 10 minutes different from the two CCMs

> And all the VOIP servers were at least 12 minutes different from my 
> data timeservers.
> 
> No firewalls are blocking timeserver ports- a freeware 'get the time' 
> utility can use ntp.bellsouth.net and successfully get the time and 
> display it - on all three servers.
> 
> No timezone differences (that would result in whole hours being 
> different anyhow, not just minute differences)
> 
> 
> Why are these servers not keeping time with each other, when both are 
> looking to ACSUNITY, at the very least, and Unity is looking at the 
> same public timeserver that my main DCs are for my data network?
> 
> 
> Here's some w32TM output from the various servers.
> _*From ACSCCM1  - you can see that it's NTP server is acsunity*_** 
> C:\>w32tm -source -v
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156254 , Incr 156250  fAdjust 0
> W32Time:    156254 Adj!=Incr 156250
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - acsunity
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156254 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'workstation'
> W32Time:    END  Line 672
> W32Time: END:Line 704
> 
> 
> 
> 
> 
> _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_** 
> C:\Documents and Settings\Administrator>w32tm -v -source
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156250 , Incr 156250  fAdjust 0
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - ntp.bellsouth.net
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156250 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'PDC'
> W32Time:    END  Line 672
> W32Time:    BEGIN:FetchParentDomainName
> W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
ptr 0
> W32Time:    END:Line 782
> W32Time: END:Line 704
> W32Time: Server: Binding to 1 NIC.
> W32Time: bind failed: 0x80072740
> W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
> again.
> W32Time: 0xC0000031 reported to System Log in Event Viewer
> W32Time: BEGIN:FinishCleanup
> W32Time:    BEGIN:TsUpTheThread
> W32Time:    END  Line 1407
> W32Time:    BEGIN:UnInitializeDC
> W32Time:       Ptrs 0 - 0
> W32Time:    END:Line 727
> W32Time:    Time service stopped.
> W32Time: END:Line 407
> 
> 
> 
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

--
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/


From tim.reimers at asheville.k12.nc.us  Wed Sep 20 14:00:57 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 20 Sep 2006 14:00:57 -0400
Subject: [cisco-voip] Time on CCMs and phones..
Message-ID: <050944EE48FF384F9C663F7048C47DA704CD3B47@webmail.asheville.k12.nc.us>


In 
c:\program files\Cisco\Xntpd\
I found install.bat

Reading that, it made mention of the usual ntp.conf (located in Windows
dir)

ok - so I found ntp.conf and it contains this:
server 127.127.1.1		# Set Local Clock to Authoritive Time
Source
fudge 127.127.1.1 stratum 5	# Resets Stratum from default 3 to 5
driftfile %windir%\ntp.drift	# path for drift file

Looks rather as if my vendor never configured this at all.

I changed it to this:
server ntp.bellsouth.net		# Set Local Clock to Authoritive
Time Source
fudge ntp.bellsouth.net stratum 5	# Resets Stratum from default 3
to 5
driftfile %windir%\ntp.drift	# path for drift file 

Is that all I'd need to do?

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Wednesday, September 20, 2006 11:55 AM
To: Simon, Bill; Tim Reimers
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

You may have XNTP already install on your servers, installed as part of
any 3.3.3 or greater CCM install. Please ensure that you do not have
more than on time client running on any given server at one time. At
Cisco we prefer XNTP for a CCM implementation, by default the Pub is the
master & the Subs point to the Pub for time.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Simon, Bill
Sent: Wednesday, September 20, 2006 10:47 AM
To: Tim Reimers
Cc: Cisco Voip
Subject: Re: [cisco-voip] Time on CCMs and phones..

We've found that the NTPD service (Cisco NTP) that comes with
CallManager is right-on and very reliable.  We use that instead of the
Windows time service.  That would be my recommendation.


Tim Reimers wrote:
> hi everyone-
> 
> I'm having trouble with drifting time on phones-
> 
> My setup -
> acsccm1 and acsccm2 (publisher/subscriber on 4.01) acsunity (Unity
> 4.x)
> 
> Supposedly, acsccm1 and acsccm2 get their time from acsunity acsunity 
> gets its time from ntp.bellsouth.net That's the same timeserver my 
> main data network uses.
> 
> What's weird is that all three 'sets' of servers vary in their time -
> ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
> But Unity this AM was more than 10 minutes different from the two CCMs

> And all the VOIP servers were at least 12 minutes different from my 
> data timeservers.
> 
> No firewalls are blocking timeserver ports- a freeware 'get the time' 
> utility can use ntp.bellsouth.net and successfully get the time and 
> display it - on all three servers.
> 
> No timezone differences (that would result in whole hours being 
> different anyhow, not just minute differences)
> 
> 
> Why are these servers not keeping time with each other, when both are 
> looking to ACSUNITY, at the very least, and Unity is looking at the 
> same public timeserver that my main DCs are for my data network?
> 
> 
> Here's some w32TM output from the various servers.
> _*From ACSCCM1  - you can see that it's NTP server is acsunity*_** 
> C:\>w32tm -source -v
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156254 , Incr 156250  fAdjust 0
> W32Time:    156254 Adj!=Incr 156250
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - acsunity
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156254 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'workstation'
> W32Time:    END  Line 672
> W32Time: END:Line 704
> 
> 
> 
> 
> 
> _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_** 
> C:\Documents and Settings\Administrator>w32tm -v -source
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156250 , Incr 156250  fAdjust 0
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - ntp.bellsouth.net
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156250 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'PDC'
> W32Time:    END  Line 672
> W32Time:    BEGIN:FetchParentDomainName
> W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
ptr 0
> W32Time:    END:Line 782
> W32Time: END:Line 704
> W32Time: Server: Binding to 1 NIC.
> W32Time: bind failed: 0x80072740
> W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
> again.
> W32Time: 0xC0000031 reported to System Log in Event Viewer
> W32Time: BEGIN:FinishCleanup
> W32Time:    BEGIN:TsUpTheThread
> W32Time:    END  Line 1407
> W32Time:    BEGIN:UnInitializeDC
> W32Time:       Ptrs 0 - 0
> W32Time:    END:Line 727
> W32Time:    Time service stopped.
> W32Time: END:Line 407
> 
> 
> 
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

--
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


From mfong at pacific.edu  Wed Sep 20 14:28:32 2006
From: mfong at pacific.edu (Melvin Fong)
Date: Wed, 20 Sep 2006 11:28:32 -0700
Subject: [cisco-voip] busy out PRI
In-Reply-To: <962D315C-0340-4F55-A682-82B79AF2AFF0@cisco.com>
References: <451109210200004E00012ECD@gwsmtp.pacific.edu>
	<2BB03FB6-85DA-47F6-9EB8-3BAF1264EE5B@cisco.com>
	<45111FE50200004E00012EF2@gwsmtp.pacific.edu>
	<962D315C-0340-4F55-A682-82B79AF2AFF0@cisco.com>
Message-ID: <451125E00200004E00012F04@gwsmtp.pacific.edu>

Thanks, this looks like it will help

>>> Ryan Ratliff <rratliff at cisco.com> 9/20/06 11:04:53 >>>
Click on the name of the parameter to see the description for it.   
This has all the info you need.

-Ryan

On Sep 20, 2006, at 2:03 PM, Melvin Fong wrote:

Ryan,
    I found the "Change B-Channel Maintnance Status" but I am having  
trouble
finding documentation as to what I enter into the field and as to  
what status 1-5 are.  Can you point me in a direction?  I would  
appreciate it. Thanks
Mel

>>> Ryan Ratliff <rratliff at cisco.com> 9/20/06 10:03:34 >>>
Take a look at the description for the advanced CM service parameter
"Change B-Channel Maintenance Status".
This is for MGCP PRIs only of course.  For H.323 it's all done in IOS.

-Ryan

On Sep 20, 2006, at 12:25 PM, Melvin Fong wrote:

Does anyone know of a way to busy out a PRI using call manager?
I don't see anything on the gateway config.  The only way I know of
is to busy the port that is connected to the PRI.  The question is
whether CM will recognize the circuit as not usable. If the port is
disabled then the D link should go down and hopefully CM will send
the calls to the 2nd PRI on the route list.  I haven't tested this
yet and hoped someone else has already.  Thanks
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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From tim.reimers at asheville.k12.nc.us  Wed Sep 20 14:46:14 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 20 Sep 2006 14:46:14 -0400
Subject: [cisco-voip] Time on CCMs and phones..
Message-ID: <050944EE48FF384F9C663F7048C47DA704CD3B7D@webmail.asheville.k12.nc.us>

also, is that the same for Unity?
I don't find Xntp on my Unity install--

so if it is Cisco's opinion that Xntp be used on Unity as well, how do I
find it to install?
 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Wednesday, September 20, 2006 11:55 AM
To: Simon, Bill; Tim Reimers
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

You may have XNTP already install on your servers, installed as part of
any 3.3.3 or greater CCM install. Please ensure that you do not have
more than on time client running on any given server at one time. At
Cisco we prefer XNTP for a CCM implementation, by default the Pub is the
master & the Subs point to the Pub for time.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Simon, Bill
Sent: Wednesday, September 20, 2006 10:47 AM
To: Tim Reimers
Cc: Cisco Voip
Subject: Re: [cisco-voip] Time on CCMs and phones..

We've found that the NTPD service (Cisco NTP) that comes with
CallManager is right-on and very reliable.  We use that instead of the
Windows time service.  That would be my recommendation.


Tim Reimers wrote:
> hi everyone-
> 
> I'm having trouble with drifting time on phones-
> 
> My setup -
> acsccm1 and acsccm2 (publisher/subscriber on 4.01) acsunity (Unity
> 4.x)
> 
> Supposedly, acsccm1 and acsccm2 get their time from acsunity acsunity 
> gets its time from ntp.bellsouth.net That's the same timeserver my 
> main data network uses.
> 
> What's weird is that all three 'sets' of servers vary in their time -
> ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
> But Unity this AM was more than 10 minutes different from the two CCMs

> And all the VOIP servers were at least 12 minutes different from my 
> data timeservers.
> 
> No firewalls are blocking timeserver ports- a freeware 'get the time' 
> utility can use ntp.bellsouth.net and successfully get the time and 
> display it - on all three servers.
> 
> No timezone differences (that would result in whole hours being 
> different anyhow, not just minute differences)
> 
> 
> Why are these servers not keeping time with each other, when both are 
> looking to ACSUNITY, at the very least, and Unity is looking at the 
> same public timeserver that my main DCs are for my data network?
> 
> 
> Here's some w32TM output from the various servers.
> _*From ACSCCM1  - you can see that it's NTP server is acsunity*_** 
> C:\>w32tm -source -v
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156254 , Incr 156250  fAdjust 0
> W32Time:    156254 Adj!=Incr 156250
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - acsunity
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156254 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'workstation'
> W32Time:    END  Line 672
> W32Time: END:Line 704
> 
> 
> 
> 
> 
> _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_** 
> C:\Documents and Settings\Administrator>w32tm -v -source
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156250 , Incr 156250  fAdjust 0
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - ntp.bellsouth.net
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156250 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'PDC'
> W32Time:    END  Line 672
> W32Time:    BEGIN:FetchParentDomainName
> W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
ptr 0
> W32Time:    END:Line 782
> W32Time: END:Line 704
> W32Time: Server: Binding to 1 NIC.
> W32Time: bind failed: 0x80072740
> W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
> again.
> W32Time: 0xC0000031 reported to System Log in Event Viewer
> W32Time: BEGIN:FinishCleanup
> W32Time:    BEGIN:TsUpTheThread
> W32Time:    END  Line 1407
> W32Time:    BEGIN:UnInitializeDC
> W32Time:       Ptrs 0 - 0
> W32Time:    END:Line 727
> W32Time:    Time service stopped.
> W32Time: END:Line 407
> 
> 
> 
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

--
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


From wsisk at cisco.com  Wed Sep 20 14:59:45 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 20 Sep 2006 14:59:45 -0400
Subject: [cisco-voip] Time on CCMs and phones..
In-Reply-To: <050944EE48FF384F9C663F7048C47DA704CD3B34@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA704CD3B34@webmail.asheville.k12.nc.us>
Message-ID: <45118FA1.2020605@cisco.com>

An HTML attachment was scrubbed...
URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20060920/a8aa2a8d/attachment.html 

From bababooey at cox.net  Wed Sep 20 15:01:13 2006
From: bababooey at cox.net (bababooey at cox.net)
Date: Wed, 20 Sep 2006 15:01:13 -0400
Subject: [cisco-voip] Cisco 2800/3800 Wic cards
Message-ID: <5722994.1158778873778.JavaMail.root@eastrmwml03.mgt.cox.net>

Can someone help me out, can I put a VWIC-2MFT1 in the HWIC slot of the 28/3800 series routers?  Or do I need to add onboard DSP modules to support them.
Is there a good link to describe this?

I need a 2800 with on T1 interface to a PBX.  What is the best config, 
router-vwic+dsp's  or
router-nm-hd-ve-vwic (dsp onboard nm card)


Thanks for your help


From milay at cisco.com  Wed Sep 20 15:10:38 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Wed, 20 Sep 2006 12:10:38 -0700
Subject: [cisco-voip] Time on CCMs and phones..
Message-ID: <413FABF5DCE22849AF86F442E472B78BCB5C76@xmb-sjc-219.amer.cisco.com>

XNTP will only be on CallManager. Also there was a cutoff & it may not
be in earlier versions of 4.0. Here is a URL for configuring time sync
for CCM. Please note this doc needs to be updates, we do not like to see
Microsoft Time used for CCM time sync. This document covers the install
as well.:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configurat
ion_example09186a008009470f.shtml

Mike 

-----Original Message-----
From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
Sent: Wednesday, September 20, 2006 1:46 PM
To: Mike Lay (milay)
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

also, is that the same for Unity?
I don't find Xntp on my Unity install--

so if it is Cisco's opinion that Xntp be used on Unity as well, how do I
find it to install?
 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com]
Sent: Wednesday, September 20, 2006 11:55 AM
To: Simon, Bill; Tim Reimers
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

You may have XNTP already install on your servers, installed as part of
any 3.3.3 or greater CCM install. Please ensure that you do not have
more than on time client running on any given server at one time. At
Cisco we prefer XNTP for a CCM implementation, by default the Pub is the
master & the Subs point to the Pub for time.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Simon, Bill
Sent: Wednesday, September 20, 2006 10:47 AM
To: Tim Reimers
Cc: Cisco Voip
Subject: Re: [cisco-voip] Time on CCMs and phones..

We've found that the NTPD service (Cisco NTP) that comes with
CallManager is right-on and very reliable.  We use that instead of the
Windows time service.  That would be my recommendation.


Tim Reimers wrote:
> hi everyone-
> 
> I'm having trouble with drifting time on phones-
> 
> My setup -
> acsccm1 and acsccm2 (publisher/subscriber on 4.01) acsunity (Unity
> 4.x)
> 
> Supposedly, acsccm1 and acsccm2 get their time from acsunity acsunity 
> gets its time from ntp.bellsouth.net That's the same timeserver my 
> main data network uses.
> 
> What's weird is that all three 'sets' of servers vary in their time -
> ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
> But Unity this AM was more than 10 minutes different from the two CCMs

> And all the VOIP servers were at least 12 minutes different from my 
> data timeservers.
> 
> No firewalls are blocking timeserver ports- a freeware 'get the time' 
> utility can use ntp.bellsouth.net and successfully get the time and 
> display it - on all three servers.
> 
> No timezone differences (that would result in whole hours being 
> different anyhow, not just minute differences)
> 
> 
> Why are these servers not keeping time with each other, when both are 
> looking to ACSUNITY, at the very least, and Unity is looking at the 
> same public timeserver that my main DCs are for my data network?
> 
> 
> Here's some w32TM output from the various servers.
> _*From ACSCCM1  - you can see that it's NTP server is acsunity*_** 
> C:\>w32tm -source -v
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156254 , Incr 156250  fAdjust 0
> W32Time:    156254 Adj!=Incr 156250
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - acsunity
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156254 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'workstation'
> W32Time:    END  Line 672
> W32Time: END:Line 704
> 
> 
> 
> 
> 
> _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_** 
> C:\Documents and Settings\Administrator>w32tm -v -source
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156250 , Incr 156250  fAdjust 0
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - ntp.bellsouth.net
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156250 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'PDC'
> W32Time:    END  Line 672
> W32Time:    BEGIN:FetchParentDomainName
> W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
ptr 0
> W32Time:    END:Line 782
> W32Time: END:Line 704
> W32Time: Server: Binding to 1 NIC.
> W32Time: bind failed: 0x80072740
> W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
> again.
> W32Time: 0xC0000031 reported to System Log in Event Viewer
> W32Time: BEGIN:FinishCleanup
> W32Time:    BEGIN:TsUpTheThread
> W32Time:    END  Line 1407
> W32Time:    BEGIN:UnInitializeDC
> W32Time:       Ptrs 0 - 0
> W32Time:    END:Line 727
> W32Time:    Time service stopped.
> W32Time: END:Line 407
> 
> 
> 
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

--
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


From tim.reimers at asheville.k12.nc.us  Wed Sep 20 15:43:27 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 20 Sep 2006 15:43:27 -0400
Subject: [cisco-voip] Time on CCMs and phones..
Message-ID: <050944EE48FF384F9C663F7048C47DA704CD3BF4@webmail.asheville.k12.nc.us>

so should I install this on Unity as well?

Both CCMs seem to have it-- but Unity does not-- and since it's a DC, I
thought maybe Cisco was treating that differently than the CCMs... 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Wednesday, September 20, 2006 3:11 PM
To: Tim Reimers
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

XNTP will only be on CallManager. Also there was a cutoff & it may not
be in earlier versions of 4.0. Here is a URL for configuring time sync
for CCM. Please note this doc needs to be updates, we do not like to see
Microsoft Time used for CCM time sync. This document covers the install
as well.:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configurat
ion_example09186a008009470f.shtml

Mike 

-----Original Message-----
From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us]
Sent: Wednesday, September 20, 2006 1:46 PM
To: Mike Lay (milay)
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

also, is that the same for Unity?
I don't find Xntp on my Unity install--

so if it is Cisco's opinion that Xntp be used on Unity as well, how do I
find it to install?
 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com]
Sent: Wednesday, September 20, 2006 11:55 AM
To: Simon, Bill; Tim Reimers
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

You may have XNTP already install on your servers, installed as part of
any 3.3.3 or greater CCM install. Please ensure that you do not have
more than on time client running on any given server at one time. At
Cisco we prefer XNTP for a CCM implementation, by default the Pub is the
master & the Subs point to the Pub for time.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Simon, Bill
Sent: Wednesday, September 20, 2006 10:47 AM
To: Tim Reimers
Cc: Cisco Voip
Subject: Re: [cisco-voip] Time on CCMs and phones..

We've found that the NTPD service (Cisco NTP) that comes with
CallManager is right-on and very reliable.  We use that instead of the
Windows time service.  That would be my recommendation.


Tim Reimers wrote:
> hi everyone-
> 
> I'm having trouble with drifting time on phones-
> 
> My setup -
> acsccm1 and acsccm2 (publisher/subscriber on 4.01) acsunity (Unity
> 4.x)
> 
> Supposedly, acsccm1 and acsccm2 get their time from acsunity acsunity 
> gets its time from ntp.bellsouth.net That's the same timeserver my 
> main data network uses.
> 
> What's weird is that all three 'sets' of servers vary in their time -
> ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
> But Unity this AM was more than 10 minutes different from the two CCMs

> And all the VOIP servers were at least 12 minutes different from my 
> data timeservers.
> 
> No firewalls are blocking timeserver ports- a freeware 'get the time' 
> utility can use ntp.bellsouth.net and successfully get the time and 
> display it - on all three servers.
> 
> No timezone differences (that would result in whole hours being 
> different anyhow, not just minute differences)
> 
> 
> Why are these servers not keeping time with each other, when both are 
> looking to ACSUNITY, at the very least, and Unity is looking at the 
> same public timeserver that my main DCs are for my data network?
> 
> 
> Here's some w32TM output from the various servers.
> _*From ACSCCM1  - you can see that it's NTP server is acsunity*_** 
> C:\>w32tm -source -v
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156254 , Incr 156250  fAdjust 0
> W32Time:    156254 Adj!=Incr 156250
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - acsunity
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156254 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'workstation'
> W32Time:    END  Line 672
> W32Time: END:Line 704
> 
> 
> 
> 
> 
> _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_** 
> C:\Documents and Settings\Administrator>w32tm -v -source
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156250 , Incr 156250  fAdjust 0
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - ntp.bellsouth.net
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156250 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'PDC'
> W32Time:    END  Line 672
> W32Time:    BEGIN:FetchParentDomainName
> W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
ptr 0
> W32Time:    END:Line 782
> W32Time: END:Line 704
> W32Time: Server: Binding to 1 NIC.
> W32Time: bind failed: 0x80072740
> W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
> again.
> W32Time: 0xC0000031 reported to System Log in Event Viewer
> W32Time: BEGIN:FinishCleanup
> W32Time:    BEGIN:TsUpTheThread
> W32Time:    END  Line 1407
> W32Time:    BEGIN:UnInitializeDC
> W32Time:       Ptrs 0 - 0
> W32Time:    END:Line 727
> W32Time:    Time service stopped.
> W32Time: END:Line 407
> 
> 
> 
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

--
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


From milay at cisco.com  Wed Sep 20 16:00:48 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Wed, 20 Sep 2006 13:00:48 -0700
Subject: [cisco-voip] Time on CCMs and phones..
Message-ID: <413FABF5DCE22849AF86F442E472B78BCB5CAA@xmb-sjc-219.amer.cisco.com>

Sorry I can not speak to Unity.

Mike 

-----Original Message-----
From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
Sent: Wednesday, September 20, 2006 2:43 PM
To: Mike Lay (milay)
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

so should I install this on Unity as well?

Both CCMs seem to have it-- but Unity does not-- and since it's a DC, I
thought maybe Cisco was treating that differently than the CCMs... 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com]
Sent: Wednesday, September 20, 2006 3:11 PM
To: Tim Reimers
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

XNTP will only be on CallManager. Also there was a cutoff & it may not
be in earlier versions of 4.0. Here is a URL for configuring time sync
for CCM. Please note this doc needs to be updates, we do not like to see
Microsoft Time used for CCM time sync. This document covers the install
as well.:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configurat
ion_example09186a008009470f.shtml

Mike 

-----Original Message-----
From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us]
Sent: Wednesday, September 20, 2006 1:46 PM
To: Mike Lay (milay)
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

also, is that the same for Unity?
I don't find Xntp on my Unity install--

so if it is Cisco's opinion that Xntp be used on Unity as well, how do I
find it to install?
 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com]
Sent: Wednesday, September 20, 2006 11:55 AM
To: Simon, Bill; Tim Reimers
Cc: Cisco Voip
Subject: RE: [cisco-voip] Time on CCMs and phones..

You may have XNTP already install on your servers, installed as part of
any 3.3.3 or greater CCM install. Please ensure that you do not have
more than on time client running on any given server at one time. At
Cisco we prefer XNTP for a CCM implementation, by default the Pub is the
master & the Subs point to the Pub for time.

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Simon, Bill
Sent: Wednesday, September 20, 2006 10:47 AM
To: Tim Reimers
Cc: Cisco Voip
Subject: Re: [cisco-voip] Time on CCMs and phones..

We've found that the NTPD service (Cisco NTP) that comes with
CallManager is right-on and very reliable.  We use that instead of the
Windows time service.  That would be my recommendation.


Tim Reimers wrote:
> hi everyone-
> 
> I'm having trouble with drifting time on phones-
> 
> My setup -
> acsccm1 and acsccm2 (publisher/subscriber on 4.01) acsunity (Unity
> 4.x)
> 
> Supposedly, acsccm1 and acsccm2 get their time from acsunity acsunity 
> gets its time from ntp.bellsouth.net That's the same timeserver my 
> main data network uses.
> 
> What's weird is that all three 'sets' of servers vary in their time -
> ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
> But Unity this AM was more than 10 minutes different from the two CCMs

> And all the VOIP servers were at least 12 minutes different from my 
> data timeservers.
> 
> No firewalls are blocking timeserver ports- a freeware 'get the time' 
> utility can use ntp.bellsouth.net and successfully get the time and 
> display it - on all three servers.
> 
> No timezone differences (that would result in whole hours being 
> different anyhow, not just minute differences)
> 
> 
> Why are these servers not keeping time with each other, when both are 
> looking to ACSUNITY, at the very least, and Unity is looking at the 
> same public timeserver that my main DCs are for my data network?
> 
> 
> Here's some w32TM output from the various servers.
> _*From ACSCCM1  - you can see that it's NTP server is acsunity*_** 
> C:\>w32tm -source -v
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156254 , Incr 156250  fAdjust 0
> W32Time:    156254 Adj!=Incr 156250
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - acsunity
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156254 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'workstation'
> W32Time:    END  Line 672
> W32Time: END:Line 704
> 
> 
> 
> 
> 
> _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_** 
> C:\Documents and Settings\Administrator>w32tm -v -source
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156250 , Incr 156250  fAdjust 0
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - ntp.bellsouth.net
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156250 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'PDC'
> W32Time:    END  Line 672
> W32Time:    BEGIN:FetchParentDomainName
> W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
ptr 0
> W32Time:    END:Line 782
> W32Time: END:Line 704
> W32Time: Server: Binding to 1 NIC.
> W32Time: bind failed: 0x80072740
> W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
> again.
> W32Time: 0xC0000031 reported to System Log in Event Viewer
> W32Time: BEGIN:FinishCleanup
> W32Time:    BEGIN:TsUpTheThread
> W32Time:    END  Line 1407
> W32Time:    BEGIN:UnInitializeDC
> W32Time:       Ptrs 0 - 0
> W32Time:    END:Line 727
> W32Time:    Time service stopped.
> W32Time: END:Line 407
> 
> 
> 
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

--
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


From mfa at crec.ifas.ufl.edu  Wed Sep 20 16:32:29 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Wed, 20 Sep 2006 16:32:29 -0400
Subject: [cisco-voip] How to find a phone by serial number?
References: <mailman.429.1158773165.15977.cisco-voip@puck.nether.net>
Message-ID: <023c01c6dcf3$e49017d0$400cfb0a@crec.ifas.ufl.edu>

TAC tracks all products by serial number, but CCM tracks phones by MAC (in 
the form SEP<MAC> device name), and never, it seems, the twain shall meet. 
Does anyone have a way of locating a phone by serial number, other than 
manually maintaining a SN/MAC translation table somewhere?  Is the SN stored 
in the Publisher SQL database somewhere?

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 


From pwalenta at wi.rr.com  Wed Sep 20 16:47:23 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Wed, 20 Sep 2006 15:47:23 -0500
Subject: [cisco-voip] Time on CCMs and phones..
In-Reply-To: <050944EE48FF384F9C663F7048C47DA704CD3B34@webmail.asheville.k12.nc.us>
Message-ID: <00de01c6dcf5$fae333d0$0232a8c0@pwnhome.com>

Look for a directory c:\program files\cisco\xntp (going on memory here).
You should also see "Network Time Protocol" listed as a service, and it
should be started.  The main configuration file I believe is
C:\WINNT\ntp.conf (a directory listing should find it if that's not the
exact name).  Once edited, restart the time service, and that should correct
the CCM time (if it's 10 minutes off, it will correct it slowly). 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, September 20, 2006 12:44 PM
To: Simon, Bill
Cc: Cisco Voip
Subject: Re: [cisco-voip] Time on CCMs and phones..

hmmm..

I don't see that on my system- either in Services in serviceability or in NT
services in Administrative Tools...

wonder where it gets installed from in 4.01 ?? 

-----Original Message-----
From: Simon, Bill [mailto:bills at tns.its.psu.edu]
Sent: Wednesday, September 20, 2006 11:47 AM
To: Tim Reimers
Cc: Cisco Voip
Subject: Re: [cisco-voip] Time on CCMs and phones..

We've found that the NTPD service (Cisco NTP) that comes with CallManager is
right-on and very reliable.  We use that instead of the Windows time
service.  That would be my recommendation.


Tim Reimers wrote:
> hi everyone-
> 
> I'm having trouble with drifting time on phones-
> 
> My setup -
> acsccm1 and acsccm2 (publisher/subscriber on 4.01) acsunity (Unity
> 4.x)
> 
> Supposedly, acsccm1 and acsccm2 get their time from acsunity acsunity 
> gets its time from ntp.bellsouth.net That's the same timeserver my 
> main data network uses.
> 
> What's weird is that all three 'sets' of servers vary in their time -
> ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
> But Unity this AM was more than 10 minutes different from the two CCMs

> And all the VOIP servers were at least 12 minutes different from my 
> data timeservers.
> 
> No firewalls are blocking timeserver ports- a freeware 'get the time' 
> utility can use ntp.bellsouth.net and successfully get the time and 
> display it - on all three servers.
> 
> No timezone differences (that would result in whole hours being 
> different anyhow, not just minute differences)
> 
> 
> Why are these servers not keeping time with each other, when both are 
> looking to ACSUNITY, at the very least, and Unity is looking at the 
> same public timeserver that my main DCs are for my data network?
> 
> 
> Here's some w32TM output from the various servers.
> _*From ACSCCM1  - you can see that it's NTP server is acsunity*_** 
> C:\>w32tm -source -v
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156254 , Incr 156250  fAdjust 0
> W32Time:    156254 Adj!=Incr 156250
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - acsunity
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156254 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'workstation'
> W32Time:    END  Line 672
> W32Time: END:Line 704
> 
> 
> 
> 
> 
> _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_** 
> C:\Documents and Settings\Administrator>w32tm -v -source
> W32Time: BEGIN:InitAdjIncr
> W32Time:    Adj 156250 , Incr 156250  fAdjust 0
> W32Time: END:Line 2503
> W32Time: BEGIN:TsUpTheThread
> W32Time: END  Line 1407
> W32Time: TimeMMInit()
> W32Time: Kernel timer : using default maximum resolution
> W32Time:                MaximumTime = 156250
> W32Time:                CurrentTime = 9766
> W32Time: Timer calibrated, looped 1 times
> W32Time: BEGIN:InitTmCfg
> W32Time: END:Line 807
> W32Time: BEGIN:InitTmCli
> W32Time: END:Line 2596
> W32Time: BEGIN:InitTmData
> W32Time: END:Line 2618
> W32Time: AvoidTimeSyncOnWan 0
> W32Time: ntpserver - ntp.bellsouth.net
> W32Time: BEGIN:CMOSSynchSet
> W32Time:    Setting adjustment 156250 - Bool  0
> W32Time:    BEGIN:SetTSTimeRes
> W32Time:    END:Line 1295
> W32Time: END:Line 864
> W32Time: BEGIN:InitializeDC
> W32Time:    BEGIN:GetRole
> W32Time:       Role is 'PDC'
> W32Time:    END  Line 672
> W32Time:    BEGIN:FetchParentDomainName
> W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
ptr 0
> W32Time:    END:Line 782
> W32Time: END:Line 704
> W32Time: Server: Binding to 1 NIC.
> W32Time: bind failed: 0x80072740
> W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
> again.
> W32Time: 0xC0000031 reported to System Log in Event Viewer
> W32Time: BEGIN:FinishCleanup
> W32Time:    BEGIN:TsUpTheThread
> W32Time:    END  Line 1407
> W32Time:    BEGIN:UnInitializeDC
> W32Time:       Ptrs 0 - 0
> W32Time:    END:Line 727
> W32Time:    Time service stopped.
> W32Time: END:Line 407
> 
> 
> 
> 
> 
> ----------------------------------------------------------------------
> --
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip

--
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


From lelio at uoguelph.ca  Wed Sep 20 18:02:49 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 20 Sep 2006 18:02:49 -0400
Subject: [cisco-voip] Time on CCMs and phones..
References: <413FABF5DCE22849AF86F442E472B78BCB5CAA@xmb-sjc-219.amer.cisco.com>
Message-ID: <018701c6dd00$83478a00$47196883@cfs.uoguelph.ca>

Unity should get it's time from the domain controller. The domain controller should be NTP synced...

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Mike Lay (milay) 
  To: Tim Reimers 
  Cc: Cisco Voip 
  Sent: Wednesday, September 20, 2006 4:00 PM
  Subject: Re: [cisco-voip] Time on CCMs and phones..


  Sorry I can not speak to Unity.

  Mike 

  -----Original Message-----
  From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
  Sent: Wednesday, September 20, 2006 2:43 PM
  To: Mike Lay (milay)
  Cc: Cisco Voip
  Subject: RE: [cisco-voip] Time on CCMs and phones..

  so should I install this on Unity as well?

  Both CCMs seem to have it-- but Unity does not-- and since it's a DC, I
  thought maybe Cisco was treating that differently than the CCMs... 

  -----Original Message-----
  From: Mike Lay (milay) [mailto:milay at cisco.com]
  Sent: Wednesday, September 20, 2006 3:11 PM
  To: Tim Reimers
  Cc: Cisco Voip
  Subject: RE: [cisco-voip] Time on CCMs and phones..

  XNTP will only be on CallManager. Also there was a cutoff & it may not
  be in earlier versions of 4.0. Here is a URL for configuring time sync
  for CCM. Please note this doc needs to be updates, we do not like to see
  Microsoft Time used for CCM time sync. This document covers the install
  as well.:

  http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configurat
  ion_example09186a008009470f.shtml

  Mike 

  -----Original Message-----
  From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us]
  Sent: Wednesday, September 20, 2006 1:46 PM
  To: Mike Lay (milay)
  Cc: Cisco Voip
  Subject: RE: [cisco-voip] Time on CCMs and phones..

  also, is that the same for Unity?
  I don't find Xntp on my Unity install--

  so if it is Cisco's opinion that Xntp be used on Unity as well, how do I
  find it to install?
   

  -----Original Message-----
  From: Mike Lay (milay) [mailto:milay at cisco.com]
  Sent: Wednesday, September 20, 2006 11:55 AM
  To: Simon, Bill; Tim Reimers
  Cc: Cisco Voip
  Subject: RE: [cisco-voip] Time on CCMs and phones..

  You may have XNTP already install on your servers, installed as part of
  any 3.3.3 or greater CCM install. Please ensure that you do not have
  more than on time client running on any given server at one time. At
  Cisco we prefer XNTP for a CCM implementation, by default the Pub is the
  master & the Subs point to the Pub for time.

  Mike 

  -----Original Message-----
  From: cisco-voip-bounces at puck.nether.net
  [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Simon, Bill
  Sent: Wednesday, September 20, 2006 10:47 AM
  To: Tim Reimers
  Cc: Cisco Voip
  Subject: Re: [cisco-voip] Time on CCMs and phones..

  We've found that the NTPD service (Cisco NTP) that comes with
  CallManager is right-on and very reliable.  We use that instead of the
  Windows time service.  That would be my recommendation.


  Tim Reimers wrote:
  > hi everyone-
  > 
  > I'm having trouble with drifting time on phones-
  > 
  > My setup -
  > acsccm1 and acsccm2 (publisher/subscriber on 4.01) acsunity (Unity
  > 4.x)
  > 
  > Supposedly, acsccm1 and acsccm2 get their time from acsunity acsunity 
  > gets its time from ntp.bellsouth.net That's the same timeserver my 
  > main data network uses.
  > 
  > What's weird is that all three 'sets' of servers vary in their time -
  > ACSCCM1 and CCM2 usually are not far apart- 15 seconds at most.
  > But Unity this AM was more than 10 minutes different from the two CCMs

  > And all the VOIP servers were at least 12 minutes different from my 
  > data timeservers.
  > 
  > No firewalls are blocking timeserver ports- a freeware 'get the time' 
  > utility can use ntp.bellsouth.net and successfully get the time and 
  > display it - on all three servers.
  > 
  > No timezone differences (that would result in whole hours being 
  > different anyhow, not just minute differences)
  > 
  > 
  > Why are these servers not keeping time with each other, when both are 
  > looking to ACSUNITY, at the very least, and Unity is looking at the 
  > same public timeserver that my main DCs are for my data network?
  > 
  > 
  > Here's some w32TM output from the various servers.
  > _*From ACSCCM1  - you can see that it's NTP server is acsunity*_** 
  > C:\>w32tm -source -v
  > W32Time: BEGIN:InitAdjIncr
  > W32Time:    Adj 156254 , Incr 156250  fAdjust 0
  > W32Time:    156254 Adj!=Incr 156250
  > W32Time: END:Line 2503
  > W32Time: BEGIN:TsUpTheThread
  > W32Time: END  Line 1407
  > W32Time: TimeMMInit()
  > W32Time: Kernel timer : using default maximum resolution
  > W32Time:                MaximumTime = 156250
  > W32Time:                CurrentTime = 9766
  > W32Time: Timer calibrated, looped 1 times
  > W32Time: BEGIN:InitTmCfg
  > W32Time: END:Line 807
  > W32Time: BEGIN:InitTmCli
  > W32Time: END:Line 2596
  > W32Time: BEGIN:InitTmData
  > W32Time: END:Line 2618
  > W32Time: AvoidTimeSyncOnWan 0
  > W32Time: ntpserver - acsunity
  > W32Time: BEGIN:CMOSSynchSet
  > W32Time:    Setting adjustment 156254 - Bool  0
  > W32Time:    BEGIN:SetTSTimeRes
  > W32Time:    END:Line 1295
  > W32Time: END:Line 864
  > W32Time: BEGIN:InitializeDC
  > W32Time:    BEGIN:GetRole
  > W32Time:       Role is 'workstation'
  > W32Time:    END  Line 672
  > W32Time: END:Line 704
  > 
  > 
  > 
  > 
  > 
  > _*w32TM output from ACSUNITY - shows ntp.bellsouth.net*_** 
  > C:\Documents and Settings\Administrator>w32tm -v -source
  > W32Time: BEGIN:InitAdjIncr
  > W32Time:    Adj 156250 , Incr 156250  fAdjust 0
  > W32Time: END:Line 2503
  > W32Time: BEGIN:TsUpTheThread
  > W32Time: END  Line 1407
  > W32Time: TimeMMInit()
  > W32Time: Kernel timer : using default maximum resolution
  > W32Time:                MaximumTime = 156250
  > W32Time:                CurrentTime = 9766
  > W32Time: Timer calibrated, looped 1 times
  > W32Time: BEGIN:InitTmCfg
  > W32Time: END:Line 807
  > W32Time: BEGIN:InitTmCli
  > W32Time: END:Line 2596
  > W32Time: BEGIN:InitTmData
  > W32Time: END:Line 2618
  > W32Time: AvoidTimeSyncOnWan 0
  > W32Time: ntpserver - ntp.bellsouth.net
  > W32Time: BEGIN:CMOSSynchSet
  > W32Time:    Setting adjustment 156250 - Bool  0
  > W32Time:    BEGIN:SetTSTimeRes
  > W32Time:    END:Line 1295
  > W32Time: END:Line 864
  > W32Time: BEGIN:InitializeDC
  > W32Time:    BEGIN:GetRole
  > W32Time:       Role is 'PDC'
  > W32Time:    END  Line 672
  > W32Time:    BEGIN:FetchParentDomainName
  > W32Time:       NetLogonGetTimeServiceParentDomain() returned 54b with
  ptr 0
  > W32Time:    END:Line 782
  > W32Time: END:Line 704
  > W32Time: Server: Binding to 1 NIC.
  > W32Time: bind failed: 0x80072740
  > W32Time: Logging event 0xC0000031. 15 min until this event is allowed 
  > again.
  > W32Time: 0xC0000031 reported to System Log in Event Viewer
  > W32Time: BEGIN:FinishCleanup
  > W32Time:    BEGIN:TsUpTheThread
  > W32Time:    END  Line 1407
  > W32Time:    BEGIN:UnInitializeDC
  > W32Time:       Ptrs 0 - 0
  > W32Time:    END:Line 727
  > W32Time:    Time service stopped.
  > W32Time: END:Line 407
  > 
  > 
  > 
  > 
  > 
  > ----------------------------------------------------------------------
  > --
  > 
  > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip

  --
  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

  _______________________________________________
  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  Wed Sep 20 19:23:46 2006
From: techguy at gmail.com (TechGuy)
Date: Wed, 20 Sep 2006 18:23:46 -0500
Subject: [cisco-voip] Upgrading CM 3.3(5) to 4.1(3) ?
Message-ID: <a1b07ea60609201623l776f6339paecc4df49f9d1003@mail.gmail.com>

I will be performing an upgrade on CM in a couple of weeks.  Going to
go from 3.3(5) to 4.1(3).  I have done this upgrade before at a
previous employer but much more nervous about this one due to the
shear size of the deployment.  I am nervous about the end user affect
/ fallout to be honest.

Here they are setup with each line in a partition, such as Line 1 in
"PAR_Line_1" and Line 2 "PAR_Line_2" they do this to forward line 1 to
line 2 if line 1 is busy or something.  Now in 4.1(3) each line can
support multiple calls so it changes things a bit.

Anyone have any docs on what is changed and how it affects the end
users?  How did you manage alot of this?  Just curious.

Thanks!

From SPackett at fenwick.com  Wed Sep 20 19:51:16 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Wed, 20 Sep 2006 16:51:16 -0700
Subject: [cisco-voip] How to find a phone by serial number?
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC56F588@SVCEXCH01.firm.fenwick.llp>

Mike:
We purchased "Phone Scan Tool" from CMReports.com and used it once and
it worked just fine.  This maybe something you probably want to purchase
for $49.00 US. 

-----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, September 20, 2006 1:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How to find a phone by serial number?

TAC tracks all products by serial number, but CCM tracks phones by MAC
(in the form SEP<MAC> device name), and never, it seems, the twain shall
meet. 
Does anyone have a way of locating a phone by serial number, other than
manually maintaining a SN/MAC translation table somewhere?  Is the SN
stored in the Publisher SQL database somewhere?

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 

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


From Brian.Henry at apptis.com  Wed Sep 20 20:13:09 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Wed, 20 Sep 2006 20:13:09 -0400
Subject: [cisco-voip] How to find a phone by serial number?
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC028382FC@CH-EXCH-02.intelltech.apptis.inc>

The SN come from the phone itself and not CCM. Here is a link to a vendor that has a tool for what you are trying to accomplish.
http://www.syn-apps.net/About.aspx
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Mike Armstrong
Sent: Wed 9/20/2006 4:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] How to find a phone by serial number?



TAC tracks all products by serial number, but CCM tracks phones by MAC (in
the form SEP<MAC> device name), and never, it seems, the twain shall meet.
Does anyone have a way of locating a phone by serial number, other than
manually maintaining a SN/MAC translation table somewhere?  Is the SN stored
in the Publisher SQL database somewhere?

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

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


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From lelio at uoguelph.ca  Wed Sep 20 23:49:12 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 20 Sep 2006 23:49:12 -0400
Subject: [cisco-voip] Upgrading CM 3.3(5) to 4.1(3) ?
References: <a1b07ea60609201623l776f6339paecc4df49f9d1003@mail.gmail.com>
Message-ID: <001101c6dd30$e727c5f0$cb01a8c0@deadfish>

documentation? you got to be kidding, right?

i wish i was joking, but the only way i felt confident was reviewing the configuration options and comparing them screen by screen. printing the enterprise parameters as well as all the system parameters. only then will you catch everything. 

it would be nice if there was documentation that stated all the changes and differences, but it's just not there (or at least not accurate).

voicemail ports is another thing you will want to worry about. forwarding from one VM port to the other is handled much differently.

depending on what phone load you are running, you might see differences in phone behaviour. that's what was important for us, documenting that. a beta group helped us iron all that out.

what do you mean by shear size of the deployment? we have over 7200 phones and 8 servers in total. aside from the amount of time it took, everything went pretty well. aside from the fact that our phones hit the TFTP server pretty hard and some of them didn't come back with the ringlists, etc. be prepared to reset your phones a few times after the upgrades or at least try to upgrade them before the upgrade.

and plan for your O/S updates in a seperate window. 


  ----- Original Message ----- 
  From: TechGuy 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, September 20, 2006 7:23 PM
  Subject: [cisco-voip] Upgrading CM 3.3(5) to 4.1(3) ?


  I will be performing an upgrade on CM in a couple of weeks.  Going to
  go from 3.3(5) to 4.1(3).  I have done this upgrade before at a
  previous employer but much more nervous about this one due to the
  shear size of the deployment.  I am nervous about the end user affect
  / fallout to be honest.

  Here they are setup with each line in a partition, such as Line 1 in
  "PAR_Line_1" and Line 2 "PAR_Line_2" they do this to forward line 1 to
  line 2 if line 1 is busy or something.  Now in 4.1(3) each line can
  support multiple calls so it changes things a bit.

  Anyone have any docs on what is changed and how it affects the end
  users?  How did you manage alot of this?  Just curious.

  Thanks!
  _______________________________________________
  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  Thu Sep 21 03:41:50 2006
From: Ruben.Montes at eu.didata.com (Ruben Montes (EU))
Date: Thu, 21 Sep 2006 09:41:50 +0200
Subject: [cisco-voip] Unity integration Lotus Domino: cannot complete call
Message-ID: <FA233791F398B848B978464507739E8F38117E@EUBEBRUSVEX1.eu.didata.local>

Hello,

When a subscriber tries to connect to the Unity Voice Mail System, after
the password introduction the unity play a message saying that the
system is temporally down and cannot complete the call. 



Error in windows event viewer: 

Source: CiscoUnity_ConvSubscrib 

Event ID: 119 

Description: 

An unexpected error has occured while a subscriber is on the phone with
Unity. This can potentially result in the subscriber hearing the
failsafe conversation. Technical details are -
IAvDohMailUser::Get(AVP_MAIL_SERVER) returned [0x8004010f] on line 3696
of file e:\Views\cs_CU4.0.4.221\un_Conv1\Scripted\ConvSub\SubSignIn.cpp.

I'm also having in the Eventviewer an error every 5 minutes:
CiscoUnity_MALLn.

I have read somewhere that is related with a problem to access the mail
store located in the Domino.

Have anyone faced this problem?

Thanks in advance,

Ruben

 






From kfedor at amt.ru  Thu Sep 21 04:22:57 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Thu, 21 Sep 2006 12:22:57 +0400
Subject: [cisco-voip]  Upgrading Unity 3.1(5) to 4.1(2) ?
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC70648ED58@amthqex01.amt.ru>

Hi all,

I am a little panic with Cisco upgrade plan :)

Couldn't understand what I must purchase to upgrade Unity 3.1(5)UM
with 100 users and 10 ports to Unity 4.2(1) ??


-----------------------
Sincerely Yours,
Konstantin Fedorov



From rwolfe at Burwood.com  Thu Sep 21 05:35:06 2006
From: rwolfe at Burwood.com (Wolfe, Robert)
Date: Thu, 21 Sep 2006 04:35:06 -0500
Subject: [cisco-voip] Re-Routing Unallocated Calls
Message-ID: <97F472A9F637E44B954D7FBABC800430F93DB8@bur-msg-02.Burwood.com>

Works great.  Put a translation for 44XXXX to route to Unity (we are on
a 6 digit plan) in a separate Translation-Inbound partition seen by the
campus CSS and it routes right through.

 

Cheers!

 

-rob

 

Rob Wolfe  |  Burwood Group  |  Chicago |  rwolfe at burwood.com

Cell: 312.593.0605  |  Fax: 312.327.4601  |  IP: 312.327.4634

________________________________

From: Turpin, Mark [mailto:mark.turpin at bryancave.com] 
Sent: Wednesday, September 20, 2006 6:26 PM
To: Wolfe, Robert; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Re-Routing Unallocated Calls

 

Sure, just setup a X mask route and point it at a Unity call handler or
your operator's DN.

 

For instance, if you're a 4-digit dial plan, suck in XXXX by putting it
in the partition where your line 1 dn's are and route it to 0 for the
operator (if so configured) or say 1000 if that were the DN for your
call handler.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wolfe, Robert
Sent: Wednesday, September 20, 2006 12:04 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Re-Routing Unallocated Calls

 

If a call is determined "Unallocated" by the gateway/callmanager, is
there a way to automatically send these calls to an auto attendant (or
even get them answered by callmanager and some kind of treatement
performed)?

 

Thanks in advance,

 

-rob

 

  <http://www.burwood.com> 

Robert Wolfe
Network Consultant
Burwood Group, Inc.

IP:  312.327.4634
Mobile:  312.593.0605
E-Mail:  rwolfe <mailto:rwolfe at burwood.com> @burwood.com
<mailto:rwolfe at burwood.com> 

 

  <http://www.burwood.com> 

 

 

This electronic message is from a law firm. It may contain confidential
or privileged information. If you received this transmission in error,
please reply to the sender to advise of the error and delete this
transmission and any attachments.

 

IRS Circular 230 Disclosure: To ensure compliance with requirements
imposed by the IRS, we inform you that any U.S. federal tax advice
contained in this communication (including any attachments) is not
intended or written 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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From jrybak at annese.com  Thu Sep 21 06:38:58 2006
From: jrybak at annese.com (Jake Rybak)
Date: Thu, 21 Sep 2006 06:38:58 -0400
Subject: [cisco-voip] Unity integration Lotus Domino: cannot complete
	call
In-Reply-To: <FA233791F398B848B978464507739E8F38117E@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <005a01c6dd6a$374e2d60$380114ac@none>


The CiscoUnity_MALLn error every five minutes is the broadcast manager
service (CsBMsgConnector) and you stop the errors by disabling the service
CsBMsgConnector. If you need broadcast messages leave the service running
and the errors do not cause problems.

The errors should not be related to the failsafe.

jake 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruben Montes (EU)
Sent: Thursday, September 21, 2006 3:42 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity integration Lotus Domino: cannot complete call

Hello,

When a subscriber tries to connect to the Unity Voice Mail System, after
the password introduction the unity play a message saying that the
system is temporally down and cannot complete the call. 



Error in windows event viewer: 

Source: CiscoUnity_ConvSubscrib 

Event ID: 119 

Description: 

An unexpected error has occured while a subscriber is on the phone with
Unity. This can potentially result in the subscriber hearing the
failsafe conversation. Technical details are -
IAvDohMailUser::Get(AVP_MAIL_SERVER) returned [0x8004010f] on line 3696
of file e:\Views\cs_CU4.0.4.221\un_Conv1\Scripted\ConvSub\SubSignIn.cpp.

I'm also having in the Eventviewer an error every 5 minutes:
CiscoUnity_MALLn.

I have read somewhere that is related with a problem to access the mail
store located in the Domino.

Have anyone faced this problem?

Thanks in advance,

Ruben

 





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


From kfedor at amt.ru  Thu Sep 21 07:00:14 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Thu, 21 Sep 2006 15:00:14 +0400
Subject: [cisco-voip]  Upgrading Unity 3.1(5) to 4.1(2)
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC707BE4885@amthqex01.amt.ru>

Hi all,

I am a little panic with Cisco upgrade plan :)

Couldn't understand what I must purchase to upgrade Unity 3.1(5)UM
with 100 users and 10 ports to Unity 4.2(1) ??


-----------------------
Sincerely Yours,
Konstantin Fedorov



From ealeatherman at gmail.com  Thu Sep 21 09:00:56 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 21 Sep 2006 09:00:56 -0400
Subject: [cisco-voip] Cisco 2800/3800 Wic cards
In-Reply-To: <5722994.1158778873778.JavaMail.root@eastrmwml03.mgt.cox.net>
References: <5722994.1158778873778.JavaMail.root@eastrmwml03.mgt.cox.net>
Message-ID: <94a1afde0609210600o1557ffc9ge4701e38ee0bbd44@mail.gmail.com>

I just set up a 2811 with a VWIC-2MFT-T1, you will need some onboard DSPs if
you want to use the HWIC slots.

If you go to CCO and search for the DSP Calculator it will tell you what
dsps you will need for your application.

On 9/20/06, bababooey at cox.net <bababooey at cox.net> wrote:
>
> Can someone help me out, can I put a VWIC-2MFT1 in the HWIC slot of the
> 28/3800 series routers?  Or do I need to add onboard DSP modules to support
> them.
> Is there a good link to describe this?
>
> I need a 2800 with on T1 interface to a PBX.  What is the best config,
> router-vwic+dsp's  or
> router-nm-hd-ve-vwic (dsp onboard nm card)
>
>
> Thanks for your help
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From wsisk at cisco.com  Thu Sep 21 11:53:45 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 21 Sep 2006 11:53:45 -0400
Subject: [cisco-voip] How to find a phone by serial number?
In-Reply-To: <023c01c6dcf3$e49017d0$400cfb0a@crec.ifas.ufl.edu>
References: <mailman.429.1158773165.15977.cisco-voip@puck.nether.net>
	<023c01c6dcf3$e49017d0$400cfb0a@crec.ifas.ufl.edu>
Message-ID: <4512B589.9030406@cisco.com>

the phone s/n is available on the web page hosted out of the phone.  you 
can get script happy, or use the perl mod candace sent out to this list 
earlier, or pickup one of the commercial apps.

/Wes

Mike Armstrong wrote:
> TAC tracks all products by serial number, but CCM tracks phones by MAC (in 
> the form SEP<MAC> device name), and never, it seems, the twain shall meet. 
> Does anyone have a way of locating a phone by serial number, other than 
> manually maintaining a SN/MAC translation table somewhere?  Is the SN stored 
> in the Publisher SQL database somewhere?
>
> Mike Armstrong
> UF/IFAS CREC
> Lake Alfred, FL 
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From jgrace at digitelusa.net  Thu Sep 21 12:50:00 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu, 21 Sep 2006 12:50:00 -0400
Subject: [cisco-voip] cm phone image size when remote site download
Message-ID: <200609211250.AA124780574@mail.digitelusa.net>

Im doing a site where the cm is at the central location.  We are already having some bandwith issues prior to the cut.  My question is what the size of the phone loads from CM when we try to bring the remote phones up.  I know the bin file is one size but is that the sames size that will come across the wan for the phone to download and reg to CM

 


---



From SCASPER at mandtbank.com  Thu Sep 21 13:20:41 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Thu, 21 Sep 2006 13:20:41 -0400
Subject: [cisco-voip] [On a tangent - Nortel & Unity] {Was: call
 qualityissues between gateway and Unity}
Message-ID: <s51291c7.022@comrcm01intgwia.mandtbank.com>

Hi Stefen,

Sorry for the late reply, I just got back to looking at the group. We
use QSIG PRIs between our Call Manager gateways and our Nortel PBX
system to enable MWI from Unity to the Nortel phones.

Steve

Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

>>> "Netfortius" <netfortius at gmail.com> 09/15/06 7:23 AM >>>
Steve,

I have a question related to this last paragraph (sorry for hi-jacking
part of 
this thread): I am in the process of migrating - one location at a time
- a 
Nortel-based (Meridian) community, to Cisco-based VoIP w/same
infrastructure 
as the one mentioned by you (CM and Unity). One problem I am having is
that I 
need to run TWO voice mail systems in parallel (Call Pilot on Nortel,
and 
Unity on Cisco), because I could not find a way to signal the existance
of a 
message in Unity, to a Nortel-side phone, by turning on the message
lights on 
the latter. How do you handle this?

Thanks,
Stefan

On Thursday 14 September 2006 08:31, STEVEN CASPER wrote:
> Thanks Wes,
> This is very interesting. So is the recommendation to add "mgcp
playout
> adaptive 60 40 200" to all voice gateways that allow non IP phone
users
> to access Unity? I have a lot of gateways that connect Nortel users
to
> Unity via QSIG gateways. We occasionally get complaints of canned or
> robotic voice quality. We are running CCM 4.1.3sr1 with Unity 4.05.
>
> Steve
<snip>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 



From barry.oliver at provista-uk.com  Thu Sep 21 13:24:22 2006
From: barry.oliver at provista-uk.com (barry.oliver at provista-uk.com)
Date: Thu, 21 Sep 2006 18:24:22 +0100
Subject: [cisco-voip] Automated Reply Re: "cisco-voip Digest, Vol 43,
	Issue 109"
In-Reply-To: <mailman.323.1158831312.70603.cisco-voip@puck.nether.net>
Message-ID: <E1GQSHe-0006FC-RC@hermes.hosts.co.uk>

Dear cisco-voip-request at puck.nether.net

Thank you for your email. I am on annual leave at present, returning on the 25th of September. 

For any urgent enquiries can you please contact Steve McKee who will assist in my absence.  

Steve McKee
steve.mckee at provista-uk.com
07814 921 132.

Or to raise a call with our service desk please email servicedesk at provista-uk.com

Many thanks,

Barry

provista ip communications
 
t.0141 354 8883
f.0141 354 8884
 
www.provista-uk.com


From pyago at adomo.com  Thu Sep 21 13:50:17 2006
From: pyago at adomo.com (Paul Yago)
Date: Thu, 21 Sep 2006 10:50:17 -0700
Subject: [cisco-voip] CTI Port is Busy
Message-ID: <170BC64B51964949A28FE8A48EF1D4EB01820D9F@corpexchsrv01.adomo.com>

Hello,

I'm getting a fast busy when dialing into a registered CTI Port and the
CTI manager logs are giving me the following entry. CCM 4.1(3).

"deviceStateChangedNotify-Alerting State received but filter is off"

Any insight is appreciated.

Thanks,
Paul


From pyago at adomo.com  Thu Sep 21 14:40:12 2006
From: pyago at adomo.com (Paul Yago)
Date: Thu, 21 Sep 2006 11:40:12 -0700
Subject: [cisco-voip]  Cisco's G729
Message-ID: <170BC64B51964949A28FE8A48EF1D4EB01820DDA@corpexchsrv01.adomo.com>

Does anyone know what version of G729 that Cisco is using?

I'm registering a terminal with CiscoG729MediaCapability, but I'm not
sure if they're using G729AB or G729A. 

If they are using G729AB and my terminal is using G729A, will the other
Cisco terminal downgrade to G729A?

I'm also curious about G729CISCO7960 if anyone has some insight?

Thanks,
Paul 


From BRogers at pasadenaisd.org  Thu Sep 21 15:34:52 2006
From: BRogers at pasadenaisd.org (Brad Rogers)
Date: Thu, 21 Sep 2006 14:34:52 -0500
Subject: [cisco-voip] MCID
Message-ID: <CB3DDB018C56BB42A32D5C63006B76ED01004184@adminexchg03.pasadenaisd.org>

I know when you push the MCID button it flags the call as malicious on
CallManager, but does anybody know if it flags it on the carrier's side
as well?

 

 

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From roharvey at flemingc.on.ca  Thu Sep 21 15:38:29 2006
From: roharvey at flemingc.on.ca (Robert Harvey)
Date: Thu, 21 Sep 2006 15:38:29 -0400
Subject: [cisco-voip] Problems with 7941 hookswitch
Message-ID: <s512b1ff.052@freedom7.flemingc.on.ca>

Oh My,

We've had the hookswitch and phone stand issue here for years now! 
Cisco TAC states that the hookswitch mechanism is self-cleaning and
requires one to merely toggle the hookswitch 20-30 times in rapid
succession to rectify the issue.  Not so in my experience.  What I've
been doing is opening the phone and manually cleaning the rubber contact
peice and circuit board portions of the hookswitch to fix the issue. 
What I also do is straighten the metal spring mechanism to cause more
pressure on the hookswitch contact.  It seems to work great; however, it
is a manual process that takes ~5 minutes per phone.  We have no easy
fix for the broken phone stand, but in our student residences I use 3/8
" self-taping screws to permanently fix it in the full upright position
once they fail.  My question is why the newer 7941 and 7961 share the
same design flaws as the 7940 and 7960?  Cisco definitely is aware of
these issues from all the complaints I read from System Admins in the
forums and on this list...  This is very frustrating and causes me many
headaches.  At which point does a full recall take place when a product
demonstrates such absolute quality issues such as this?  I'll sign any
petition!!!

----------------------------------------------------------------
Robert Harvey
Network Support Analyst
Fleming College
Peterborough, ON, Canada
Office (705) 749-5513
Fax (705) 749-5540
experience.success
--------------------------------------------------------------



>From a mechanical viewpoint, the 79xx phones are junk, IMHO.  We
probably 
RMA-replace 10% of our phone population every year,  almost all because
of 
switchhook and deskstand problems.  I've asked several times if we
could buy 
replacement parts and fix the common problems ourselves, but never
received 
any response; I suspect a local repair might complicate future RMAs.  A

while back I had to escalate an SR because the first engineer claimed
we 
were abusing our phones, but after I offered his supervisor a chance to

discuss the abuse with the 64-year-old lady who was the alleged abuser,
he 
declined and we've had no problems since.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL

From roharvey at flemingc.on.ca  Thu Sep 21 15:46:02 2006
From: roharvey at flemingc.on.ca (Robert Harvey)
Date: Thu, 21 Sep 2006 15:46:02 -0400
Subject: [cisco-voip]  MGCP and FXS ports
Message-ID: <s512b3c7.097@freedom7.flemingc.on.ca>

Apologies if someone has already said this.  Make sure ip routing is
enabled.  I had a similar issue with a bunch of vg224s and the cause was
that I had neglected to enable ip routing within IOS.

I hope that helps!

----------------------------------------------------------------
Robert Harvey
Network Support Analyst
Fleming College
Peterborough, ON, Canada
Office (705) 749-5513
Fax (705) 749-5540
experience.success
----------------------------------------------------------------





I have a fax machine connected to an FXS port off my gateway. I have
the
Gateway configured for MGCP. The port is configured in CallManager and
the DN for the fax machine has been added. When I call the fax either
from and external line or 4 digits I get fax tone. When they attempt
to
make an outbound call from the fax machine, they get dial tone and it
will attempt to dial the digits and then immediately go to a fast
busy.
If I do a show voice port summary, I see the port go off hook. I have
looked through the CallManager's logs and I don't think it ever
reaches
the CM. I have configured an IP phone to emulate the port and DN
settings and I am able to make call internal and external. So it
appears
that the call routing in CM is fine. All the fax machines on the FXS
ports are doing this. Has anyone ever seen this before?

 

David Fielder

Senior Engineer

Dimension Data

Office: (336) 510-9431

Mobile: (336) 669-5129

david.fielder at us.didata.com 


From pyago at adomo.com  Thu Sep 21 15:56:35 2006
From: pyago at adomo.com (Paul Yago)
Date: Thu, 21 Sep 2006 12:56:35 -0700
Subject: [cisco-voip] CTI Port is Busy
Message-ID: <170BC64B51964949A28FE8A48EF1D4EB01820DF6@corpexchsrv01.adomo.com>

Never mind... I found the problem.
I was attempting to call into a G711 only enabled terminal with a G729
only phone, thus the fast busy.

Thanks

-----Original Message-----
From: Paul Yago 
Sent: Thursday, September 21, 2006 10:50 AM
To: cisco-voip at puck.nether.net
Cc: Paul Yago
Subject: RE: [cisco-voip] CTI Port is Busy

Hello,

I'm getting a fast busy when dialing into a registered CTI Port and the
CTI manager logs are giving me the following entry. CCM 4.1(3).

"deviceStateChangedNotify-Alerting State received but filter is off"

Any insight is appreciated.

Thanks,
Paul


From Matt.Slaga at us.didata.com  Thu Sep 21 16:42:37 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 21 Sep 2006 16:42:37 -0400
Subject: [cisco-voip] Upgrading Unity 3.1(5) to 4.1(2)
Message-ID: <C1FE15183DA37645BC0633BC604E44F00337C384@USNAEXCH.na.didata.local>

Well, it depends.  Did you purchase SAU or SASU with your original Unity
purchase?  If so, you just need to use the Product Upgrade Tool to order
the necessary software at no cost.  Otherwise, you will need to get with
your Cisco reseller to calculate it all out.



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Fedorov,
Konstantin
Sent: Thursday, September 21, 2006 7:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Upgrading Unity 3.1(5) to 4.1(2)

Hi all,

I am a little panic with Cisco upgrade plan :)

Couldn't understand what I must purchase to upgrade Unity 3.1(5)UM
with 100 users and 10 ports to Unity 4.2(1) ??


-----------------------
Sincerely Yours,
Konstantin Fedorov


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

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This e-mail communication and any attachments may contain
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notify us immediately by replying to this message and deleting it
from your computer. Thank you.



From erickbe at yahoo.com  Thu Sep 21 21:30:18 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu, 21 Sep 2006 18:30:18 -0700 (PDT)
Subject: [cisco-voip] One users calls not showing up in ART Department report
Message-ID: <20060922013018.54575.qmail@web37204.mail.mud.yahoo.com>

Have a CDR/ART report where we do a user department report and select this one users extension and it reports 0 calls made.  There are other users with same setup that show calls for this report.  I looked at the ART database and the billing_data table and did a query and see plenty of entries for this extension/user making calls. 

Checked the global directory and compared this user against others who show data for same report, and they are set for same department, etc. 

Anyone have any ideas? 


From ealeatherman at gmail.com  Thu Sep 21 21:47:34 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 21 Sep 2006 21:47:34 -0400
Subject: [cisco-voip] Cisco's G729
In-Reply-To: <170BC64B51964949A28FE8A48EF1D4EB01820DDA@corpexchsrv01.adomo.com>
References: <170BC64B51964949A28FE8A48EF1D4EB01820DDA@corpexchsrv01.adomo.com>
Message-ID: <94a1afde0609211847g3d654acbu2b7e943d976ecbc1@mail.gmail.com>

I thought they were using g729a

On 9/21/06, Paul Yago <pyago at adomo.com> wrote:
>
> Does anyone know what version of G729 that Cisco is using?
>
> I'm registering a terminal with CiscoG729MediaCapability, but I'm not
> sure if they're using G729AB or G729A.
>
> If they are using G729AB and my terminal is using G729A, will the other
> Cisco terminal downgrade to G729A?
>
> I'm also curious about G729CISCO7960 if anyone has some insight?
>
> Thanks,
> Paul
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From erickbe at yahoo.com  Thu Sep 21 22:39:10 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu, 21 Sep 2006 19:39:10 -0700 (PDT)
Subject: [cisco-voip] One users calls not showing up in ART Department
	report
In-Reply-To: <20060922013018.54575.qmail@web37204.mail.mud.yahoo.com>
Message-ID: <20060922023910.22026.qmail@web37208.mail.mud.yahoo.com>


Nevermind ( I think )... this is due to the users phone being associated to both their user id, and the rmjtapi user for IPCC stuff and I recall ART did not do well sometimes if a device was associated to multiple users.  Is setting the controlled device or owner fields the way to fix this or ?  They need the phone associated to the rmjtapi for ipcc. 


----- Original Message ----
From: Erick Bergquist <erickbe at yahoo.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Thursday, September 21, 2006 8:30:18 PM
Subject: [cisco-voip] One users calls not showing up in ART Department report

Have a CDR/ART report where we do a user department report and select this one users extension and it reports 0 calls made.  There are other users with same setup that show calls for this report.  I looked at the ART database and the billing_data table and did a query and see plenty of entries for this extension/user making calls. 

Checked the global directory and compared this user against others who show data for same report, and they are set for same department, etc. 

Anyone have any ideas? 

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





From bob at smalltime.com  Fri Sep 22 06:50:36 2006
From: bob at smalltime.com (Robert Kulagowski)
Date: Fri, 22 Sep 2006 05:50:36 -0500
Subject: [cisco-voip] cm phone image size when remote site download
In-Reply-To: <200609211250.AA124780574@mail.digitelusa.net>
References: <200609211250.AA124780574@mail.digitelusa.net>
Message-ID: <4513BFFC.5060905@smalltime.com>

James Grace wrote:
> Im doing a site where the cm is at the central location.  We are already having some bandwith issues prior to the cut.  My question is what the size of the phone loads from CM when we try to bring the remote phones up.  I know the bin file is one size but is that the sames size that will come across the wan for the phone to download and reg to CM

This was discussed about one month ago (check the archives.)  Yes, with 
current versions of the phone firmware, each one is going to 
individually pull it from the TFTP server.  I had a similar situation 
where I was upgrading a number of phones and didn't want to do it over 
the WAN, so I configured a TFTP server local at the site (you can use 
the router itself as a TFTP server, or setup something like the 
SolarWinds TFTP server on a Windows box 
(http://www.solarwinds.net/Tools/Free_tools/TFTP_Server/) and then 
configure the "Alternate TFTP" setting in Call Manager under Device > Phone.

Cisco packages each phone firmware load as a .exe and a .zip; just grab 
the .zip and unzip it into your TFTP server directory at the site if 
you're going to do it the "manual" way.  Worked well for me.

Someone from Cisco said that future versions of phone firmware may use 
some sort of peer-to-peer scheme.

From milay at cisco.com  Fri Sep 22 08:41:37 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Fri, 22 Sep 2006 05:41:37 -0700
Subject: [cisco-voip] One users calls not showing up in ART Department
	report
Message-ID: <413FABF5DCE22849AF86F442E472B78BCB62BB@xmb-sjc-219.amer.cisco.com>

A couple of things come to mind:

1) The user does not have a DN assigned in Global Directory or "User
Owner ID" not assigned on the device.
2) The DN is a shared line 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erick Bergquist
Sent: Thursday, September 21, 2006 8:30 PM
To: ciscovoip
Subject: [cisco-voip] One users calls not showing up in ART Department
report

Have a CDR/ART report where we do a user department report and select
this one users extension and it reports 0 calls made.  There are other
users with same setup that show calls for this report.  I looked at the
ART database and the billing_data table and did a query and see plenty
of entries for this extension/user making calls. 

Checked the global directory and compared this user against others who
show data for same report, and they are set for same department, etc. 

Anyone have any ideas? 

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


From rratliff at cisco.com  Fri Sep 22 09:33:29 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 22 Sep 2006 09:33:29 -0400
Subject: [cisco-voip] One users calls not showing up in ART Department
	report
In-Reply-To: <20060922023910.22026.qmail@web37208.mail.mud.yahoo.com>
References: <20060922023910.22026.qmail@web37208.mail.mud.yahoo.com>
Message-ID: <BBE2AE10-6668-4FCB-996B-5EE2347EB3DF@cisco.com>

Set owner userid in the Device page in CCMAdmin.   Note that this  
setting is not compatible with EM.

Also note that since this sets a field in the CDR record itself it  
will not change the records already in the ART database.  To do that  
you need to wipe the ART db, then remove all device associations  
except for the user you want, let ART re-import the entire CDR db,  
then re-associate the device to any application users necessary.    
Remember this will affect your CTI apps while the device is  
unassociated.

-Ryan

On Sep 21, 2006, at 10:39 PM, Erick Bergquist wrote:


Nevermind ( I think )... this is due to the users phone being  
associated to both their user id, and the rmjtapi user for IPCC stuff  
and I recall ART did not do well sometimes if a device was associated  
to multiple users.  Is setting the controlled device or owner fields  
the way to fix this or ?  They need the phone associated to the  
rmjtapi for ipcc.


----- Original Message ----
From: Erick Bergquist <erickbe at yahoo.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Thursday, September 21, 2006 8:30:18 PM
Subject: [cisco-voip] One users calls not showing up in ART  
Department report

Have a CDR/ART report where we do a user department report and select  
this one users extension and it reports 0 calls made.  There are  
other users with same setup that show calls for this report.  I  
looked at the ART database and the billing_data table and did a query  
and see plenty of entries for this extension/user making calls.

Checked the global directory and compared this user against others  
who show data for same report, and they are set for same department,  
etc.

Anyone have any ideas?

_______________________________________________
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  Fri Sep 22 09:44:33 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 22 Sep 2006 09:44:33 -0400
Subject: [cisco-voip] Cisco's G729
In-Reply-To: <170BC64B51964949A28FE8A48EF1D4EB01820DDA@corpexchsrv01.adomo.com>
References: <170BC64B51964949A28FE8A48EF1D4EB01820DDA@corpexchsrv01.adomo.com>
Message-ID: <4513E8C1.8060100@cisco.com>

We do multiple, it really depends on the endpoint.  Some of your 
verbiage looks CTI related, so I'm going to assume that is the interface 
you are trying to use.

Most of the ipphones register capabilities that include all variants of 
g.729.  Example of a 7960 registration:
09/22/2006 09:35:55.594 CCM|StationInit: (0000189) CapabilitiesRes 
capCount=7 caps= 25(120) 4(40) 2(40) 15(60) 16(60) 11(60) 12(60) 
|<CLID::496_Evoice-R5-CM1><NID::172.18.110.96><CT::1,100,119,1.1257817><IP::14.48.24.12><DEV::SEP000A41F97CC4>

This decodes to:
25(120) wideband
4(40) 711ulaw
2(40) 711alaw
15(60) 729b
16(60) 729ab
11(60) 729
12(60) 729a

from:
typedef enum {
    Media_Payload_NonStandard = 1,
    Media_Payload_G711Alaw64k = 2,
    Media_Payload_G711Alaw56k = 3,      // "restricted"
    Media_Payload_G711Ulaw64k = 4,
    Media_Payload_G711Ulaw56k = 5,      // "restricted"
    Media_Payload_G722_64k = 6,
    Media_Payload_G722_56k = 7,
    Media_Payload_G722_48k = 8,
    Media_Payload_G7231 = 9,
    Media_Payload_G728 = 10,
    Media_Payload_G729 = 11,
    Media_Payload_G729AnnexA = 12,
    Media_Payload_Is11172AudioCap = 13,
    Media_Payload_Is13818AudioCap = 14,
    Media_Payload_G729AnnexB = 15,
    Media_Payload_G729AnnexAwAnnexB = 16,
    Media_Payload_GSM_Full_Rate = 18,
    Media_Payload_GSM_Half_Rate = 19,
    Media_Payload_GSM_Enhanced_Full_Rate = 20,
   Media_Payload_Wide_Band_256k = 25,
    Media_Payload_Data64 = 32,
    Media_Payload_Data56 = 33,
    Media_Payload_GSM = 80,
    Media_Payload_ActiveVoice = 81,
    Media_Payload_G726_32K = 82,
    Media_Payload_G726_24K = 83,
    Media_Payload_G726_16K = 84,

Next is what ccm.exe considers as a 'match' between endpoints.  CCM 
matches based on the enum values above.  If it cannot find an enum 
match, then it generates a 'codec mismatch'.  If ccm finds a codec 
mismatch and cannot find a transcoder the default behavior is to connect 
the 2 endpoints, telling each to use their preferred codec which has 
bandwidth utilization less than the limit configured in Regions for 
callmanager.  There are a few interesting scenarios for h.323 - see the 
CM service parameter "Stirp G.729 Annex B (Silence Suppression) from 
Capabilities".  Also see CSCse92656 for one known sub-optimal endpoint.

/Wes

Paul Yago wrote:
> Does anyone know what version of G729 that Cisco is using?
>
> I'm registering a terminal with CiscoG729MediaCapability, but I'm not
> sure if they're using G729AB or G729A. 
>
> If they are using G729AB and my terminal is using G729A, will the other
> Cisco terminal downgrade to G729A?
>
> I'm also curious about G729CISCO7960 if anyone has some insight?
>
> Thanks,
> Paul 
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From erickbe at yahoo.com  Fri Sep 22 09:46:51 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri, 22 Sep 2006 06:46:51 -0700 (PDT)
Subject: [cisco-voip] One users calls not showing up in ART Department
	report
In-Reply-To: <BBE2AE10-6668-4FCB-996B-5EE2347EB3DF@cisco.com>
Message-ID: <20060922134651.35812.qmail@web37211.mail.mud.yahoo.com>

Right. Problem is this is a 3.3 system so Owner User ID not available until a upgrade is done.

So if we remove all the device associations except for one, and wipe ART that will corect the data for a bit. But is there any way to maybe put the device associations back in a certain order to get it to stick with the user id of the user and not rmjtapi or NULL?  Some of the records have a NULL user id now but theres devices associated but it's probably choosing NULL due to seeing multples and going with first thing it see's.

----- Original Message ----
From: Ryan Ratliff <rratliff at cisco.com>
To: Erick Bergquist <erickbe at yahoo.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Sent: Friday, September 22, 2006 8:33:29 AM
Subject: Re: [cisco-voip] One users calls not showing up in ART Department report

Set owner userid in the Device page in CCMAdmin.   Note that this  
setting is not compatible with EM.

Also note that since this sets a field in the CDR record itself it  
will not change the records already in the ART database.  To do that  
you need to wipe the ART db, then remove all device associations  
except for the user you want, let ART re-import the entire CDR db,  
then re-associate the device to any application users necessary.    
Remember this will affect your CTI apps while the device is  
unassociated.

-Ryan

On Sep 21, 2006, at 10:39 PM, Erick Bergquist wrote:


Nevermind ( I think )... this is due to the users phone being  
associated to both their user id, and the rmjtapi user for IPCC stuff  
and I recall ART did not do well sometimes if a device was associated  
to multiple users.  Is setting the controlled device or owner fields  
the way to fix this or ?  They need the phone associated to the  
rmjtapi for ipcc.


----- Original Message ----
From: Erick Bergquist <erickbe at yahoo.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Thursday, September 21, 2006 8:30:18 PM
Subject: [cisco-voip] One users calls not showing up in ART  
Department report

Have a CDR/ART report where we do a user department report and select  
this one users extension and it reports 0 calls made.  There are  
other users with same setup that show calls for this report.  I  
looked at the ART database and the billing_data table and did a query  
and see plenty of entries for this extension/user making calls.

Checked the global directory and compared this user against others  
who show data for same report, and they are set for same department,  
etc.

Anyone have any ideas?

_______________________________________________
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 jcampbell at ci.davidson.nc.us  Fri Sep 22 09:51:26 2006
From: jcampbell at ci.davidson.nc.us (Campbell, Jeremy P.)
Date: Fri, 22 Sep 2006 09:51:26 -0400
Subject: [cisco-voip] mcs-7815i rails
Message-ID: <F81B53847D3DC945A55B555B5555CC79D1FA61@davnc03.ci.davidson.nc.us>

Apologies for the repost, but someone on this list knows the answer to
my question.  

 

What IBM model does the MCS-7815i correspond to?  It seems crazy to
spend $600+ on Cisco branded rails for these units.  I'm assuming the
IBM rails work.  Have others done this?

 

Thanks,

Jeremy

 

_____________________________
Jeremy P. Campbell
Information Systems Director
Town of Davidson, NC
704-940-9632 - voice, direct
704-892-7591 ext. 248 - voice, main
704-892-3971 - fax
JCampbell at ci.davidson.nc.us

 

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From rratliff at cisco.com  Fri Sep 22 09:59:55 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Fri, 22 Sep 2006 09:59:55 -0400
Subject: [cisco-voip] One users calls not showing up in ART Department
	report
In-Reply-To: <20060922134651.35812.qmail@web37211.mail.mud.yahoo.com>
References: <20060922134651.35812.qmail@web37211.mail.mud.yahoo.com>
Message-ID: <7BAC7118-369B-4BBC-A9E4-63F599389F68@cisco.com>

I can't tell you for sure as I've never seen a good explanation of  
what order the directory returns the list of users associated to a  
given device and whether ART uses the first, last, or just picks at  
random.

If the user in question has a userid higher in the alphabet than  
rmjtapi then I think it's safe to assume it's not alphabetical.  In  
that case try removing all device associations, then adding the user  
followed by rmjtapi.  Maybe this will get it so the right user is  
picked.

-Ryan

On Sep 22, 2006, at 9:46 AM, Erick Bergquist wrote:

Right. Problem is this is a 3.3 system so Owner User ID not available  
until a upgrade is done.

So if we remove all the device associations except for one, and wipe  
ART that will corect the data for a bit. But is there any way to  
maybe put the device associations back in a certain order to get it  
to stick with the user id of the user and not rmjtapi or NULL?  Some  
of the records have a NULL user id now but theres devices associated  
but it's probably choosing NULL due to seeing multples and going with  
first thing it see's.

----- Original Message ----
From: Ryan Ratliff <rratliff at cisco.com>
To: Erick Bergquist <erickbe at yahoo.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Sent: Friday, September 22, 2006 8:33:29 AM
Subject: Re: [cisco-voip] One users calls not showing up in ART  
Department report

Set owner userid in the Device page in CCMAdmin.   Note that this
setting is not compatible with EM.

Also note that since this sets a field in the CDR record itself it
will not change the records already in the ART database.  To do that
you need to wipe the ART db, then remove all device associations
except for the user you want, let ART re-import the entire CDR db,
then re-associate the device to any application users necessary.
Remember this will affect your CTI apps while the device is
unassociated.

-Ryan

On Sep 21, 2006, at 10:39 PM, Erick Bergquist wrote:


Nevermind ( I think )... this is due to the users phone being
associated to both their user id, and the rmjtapi user for IPCC stuff
and I recall ART did not do well sometimes if a device was associated
to multiple users.  Is setting the controlled device or owner fields
the way to fix this or ?  They need the phone associated to the
rmjtapi for ipcc.


----- Original Message ----
From: Erick Bergquist <erickbe at yahoo.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Thursday, September 21, 2006 8:30:18 PM
Subject: [cisco-voip] One users calls not showing up in ART
Department report

Have a CDR/ART report where we do a user department report and select
this one users extension and it reports 0 calls made.  There are
other users with same setup that show calls for this report.  I
looked at the ART database and the billing_data table and did a query
and see plenty of entries for this extension/user making calls.

Checked the global directory and compared this user against others
who show data for same report, and they are set for same department,
etc.

Anyone have any ideas?

_______________________________________________
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 gpate at kenttech.com  Fri Sep 22 10:12:36 2006
From: gpate at kenttech.com (Gary L. Pate)
Date: Fri, 22 Sep 2006 10:12:36 -0400
Subject: [cisco-voip] Unity problem
Message-ID: <000901c6de51$2822e2d0$111e14ac@gpatelt>

Recently, I added my ID to the "System Event" and "Unaddressed Messages"
public distribution list. As I'm sure you know, only the Example
Administrator is a member of this by default. I added my ID to know of
things that are happening that could indicate a problem. After adding my ID,
I started receiving voice mails that were destined for subscribers. Funny
thing, the voice mails all sound normal as if they received the subscribers
greeting, unlike the normal type of things the Example Admin would get. I
have received no complaint that users aren't receiving their voice mails,
but recently have started getting sporadic mention of delayed voice mails.
This Unity server handles voice mail for multiple customers, and is not
really functioning in a "Unified Messaging" capacity. The different
customers all have different emails not associated with the Unity server.
Under Active Directory under the Exchange Features tab of a user, we have
the delivery options set to "Deliver messages to both forwarding address and
mailbox", and we have their normal SMTP address setup. With another
customer, we have the delivery options setup only to forward.

Here is what I believe is happening:

 

For the users that have the "Deliver to both" setup, I believe they are
getting their voice mails via the phone, and probably not getting the email
wave file, and the users that are set to only "forward" are not getting
their voice mails at all.

 

The problem with this though is that I don't think its happening to every
voice mail. I think some of them are actually being delivered.

 

When I checked the Example administrators account, there were over 5000
voice mail in there, and when I look at a mailbox of a "forward only"
account, I see the voice mails sitting there as if they were never
forwarded.

 

I hope this makes sense as it's difficult to describe exactly with the
different scenarios.

 

Any thoughts on why these do not seem to be getting forwarded to the user's
mailbox? It's not just one user, but multiple.

 

Gary

 

 

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From Scott.Voll at wesd.org  Fri Sep 22 11:12:40 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 22 Sep 2006 08:12:40 -0700
Subject: [cisco-voip] MCID
Message-ID: <407055A92CECCB499C922A2D35FC19A606934616@Apollo.wesd.org>

My understanding is NO.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Rogers
Sent: Thursday, September 21, 2006 12:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MCID

 

I know when you push the MCID button it flags the call as malicious on
CallManager, but does anybody know if it flags it on the carrier's side
as well?

 

 

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From MillerS at DicksteinShapiro.COM  Fri Sep 22 11:24:20 2006
From: MillerS at DicksteinShapiro.COM (Miller, Steve)
Date: Fri, 22 Sep 2006 11:24:20 -0400
Subject: [cisco-voip] MCID
Message-ID: <B9B5B75DF9145F44A21111DD92CC49000453D39E@DCEX2.DSMO.COM>

How do you create this button?
 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202) 420-2201
millers at dicksteinshapiro.com 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, September 22, 2006 11:13 AM
To: Brad Rogers; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MCID



My understanding is NO.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Rogers
Sent: Thursday, September 21, 2006 12:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MCID

 

I know when you push the MCID button it flags the call as malicious on
CallManager, but does anybody know if it flags it on the carrier's side
as well?

 

 


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From Scott.Voll at wesd.org  Fri Sep 22 11:34:46 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 22 Sep 2006 08:34:46 -0700
Subject: [cisco-voip] MCID
Message-ID: <407055A92CECCB499C922A2D35FC19A606934631@Apollo.wesd.org>

You have to add it to a softkey template.

 

Scott

 

________________________________

From: Miller, Steve [mailto:MillerS at DicksteinShapiro.COM] 
Sent: Friday, September 22, 2006 8:24 AM
To: Voll, Scott; Brad Rogers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MCID

 

How do you create this button?

 

Steve Miller
Telecom Engineer
Dickstein Shapiro LLP
1825 Eye Street NW | Washington, DC 20006
Tel (202) 420-3370 Fax (202) 420-2201
millers at dicksteinshapiro.com 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, September 22, 2006 11:13 AM
To: Brad Rogers; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MCID

My understanding is NO.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Rogers
Sent: Thursday, September 21, 2006 12:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MCID

 

I know when you push the MCID button it flags the call as malicious on
CallManager, but does anybody know if it flags it on the carrier's side
as well?

 

 

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From wsisk at cisco.com  Fri Sep 22 11:37:14 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 22 Sep 2006 11:37:14 -0400
Subject: [cisco-voip] MCID
In-Reply-To: <407055A92CECCB499C922A2D35FC19A606934616@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A606934616@Apollo.wesd.org>
Message-ID: <4514032A.4040207@cisco.com>

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From erickbe at yahoo.com  Fri Sep 22 12:28:43 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri, 22 Sep 2006 09:28:43 -0700 (PDT)
Subject: [cisco-voip] One users calls not showing up in ART Department
	report
Message-ID: <20060922162843.87722.qmail@web37215.mail.mud.yahoo.com>

Thanks for the help. Thats what I figured as well, that it would be hit and miss if we did something like below. 

----- Original Message ----
From: Ryan Ratliff <rratliff at cisco.com>
To: Erick Bergquist <erickbe at yahoo.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Sent: Friday, September 22, 2006 8:59:55 AM
Subject: Re: [cisco-voip] One users calls not showing up in ART Department report

I can't tell you for sure as I've never seen a good explanation of  
what order the directory returns the list of users associated to a  
given device and whether ART uses the first, last, or just picks at  
random.

If the user in question has a userid higher in the alphabet than  
rmjtapi then I think it's safe to assume it's not alphabetical.  In  
that case try removing all device associations, then adding the user  
followed by rmjtapi.  Maybe this will get it so the right user is  
picked.

-Ryan

On Sep 22, 2006, at 9:46 AM, Erick Bergquist wrote:

Right. Problem is this is a 3.3 system so Owner User ID not available  
until a upgrade is done.

So if we remove all the device associations except for one, and wipe  
ART that will corect the data for a bit. But is there any way to  
maybe put the device associations back in a certain order to get it  
to stick with the user id of the user and not rmjtapi or NULL?  Some  
of the records have a NULL user id now but theres devices associated  
but it's probably choosing NULL due to seeing multples and going with  
first thing it see's.

----- Original Message ----
From: Ryan Ratliff <rratliff at cisco.com>
To: Erick Bergquist <erickbe at yahoo.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Sent: Friday, September 22, 2006 8:33:29 AM
Subject: Re: [cisco-voip] One users calls not showing up in ART  
Department report

Set owner userid in the Device page in CCMAdmin.   Note that this
setting is not compatible with EM.

Also note that since this sets a field in the CDR record itself it
will not change the records already in the ART database.  To do that
you need to wipe the ART db, then remove all device associations
except for the user you want, let ART re-import the entire CDR db,
then re-associate the device to any application users necessary.
Remember this will affect your CTI apps while the device is
unassociated.

-Ryan

On Sep 21, 2006, at 10:39 PM, Erick Bergquist wrote:


Nevermind ( I think )... this is due to the users phone being
associated to both their user id, and the rmjtapi user for IPCC stuff
and I recall ART did not do well sometimes if a device was associated
to multiple users.  Is setting the controlled device or owner fields
the way to fix this or ?  They need the phone associated to the
rmjtapi for ipcc.


----- Original Message ----
From: Erick Bergquist <erickbe at yahoo.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Thursday, September 21, 2006 8:30:18 PM
Subject: [cisco-voip] One users calls not showing up in ART
Department report

Have a CDR/ART report where we do a user department report and select
this one users extension and it reports 0 calls made.  There are
other users with same setup that show calls for this report.  I
looked at the ART database and the billing_data table and did a query
and see plenty of entries for this extension/user making calls.

Checked the global directory and compared this user against others
who show data for same report, and they are set for same department,
etc.

Anyone have any ideas?

_______________________________________________
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 milay at cisco.com  Fri Sep 22 14:35:12 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Fri, 22 Sep 2006 11:35:12 -0700
Subject: [cisco-voip] mcs-7815i rails
Message-ID: <413FABF5DCE22849AF86F442E472B78BCB6425@xmb-sjc-219.amer.cisco.com>

The 7815I is based on an IBM x206 & x206m platforms.
 
ftp://ftp.software.ibm.com/common/ssi/rep_sp/n/XSD00310USEN/XSD00310USEN
.PDF
ftp://ftp.software.ibm.com/common/ssi/rep_sp/n/XSD01232USEN/XSD01232USEN
.PDF
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Campbell,
Jeremy P.
Sent: Friday, September 22, 2006 8:51 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] mcs-7815i rails



Apologies for the repost, but someone on this list knows the answer to
my question.  

 

What IBM model does the MCS-7815i correspond to?  It seems crazy to
spend $600+ on Cisco branded rails for these units.  I'm assuming the
IBM rails work.  Have others done this?

 

Thanks,

Jeremy

 

_____________________________
Jeremy P. Campbell
Information Systems Director
Town of Davidson, NC
704-940-9632 - voice, direct
704-892-7591 ext. 248 - voice, main
704-892-3971 - fax
JCampbell at ci.davidson.nc.us

 

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From KNoel at lincare.com  Fri Sep 22 15:22:57 2006
From: KNoel at lincare.com (Noel, Keith)
Date: Fri, 22 Sep 2006 15:22:57 -0400
Subject: [cisco-voip] CISCO IPCC/Unified Contact Center Enterprise
	AS400/iSeries
Message-ID: <EFAF5FF5FC1B35438B01CD486745BAF703F7EE5D@HQEXCH.Lincare.com>

Has anyone interfaced a IPCC (now Unified Contact Center Enterprise)
with an AS400/iSeries? Thanks.

Keith Noel
Phone: 1.727.431.8231
Fax: 1.727.531.7873

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From wsisk at cisco.com  Fri Sep 22 16:25:49 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 22 Sep 2006 16:25:49 -0400
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe at
 100%	cpu
In-Reply-To: <44FF12F1.9070306@tns.its.psu.edu>
References: <C1FE15183DA37645BC0633BC604E44F0030FFD1E@USNAEXCH.na.didata.local>
	<44FF12F1.9070306@tns.its.psu.edu>
Message-ID: <451446CD.4070007@cisco.com>

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From jason.aarons at us.didata.com  Fri Sep 22 16:56:30 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 22 Sep 2006 16:56:30 -0400
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe at
	100%	cpu
Message-ID: <C1FE15183DA37645BC0633BC604E44F0033B9C48@USNAEXCH.na.didata.local>

Is there a ES special or hotfix? or still pending resolution?  

 

I'm only seeing it on IPCC Cluster, does notes show if CallManager or
other platforms are seeing this?

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Friday, September 22, 2006 4:26 PM
To: Simon, Bill
Cc: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
at 100% cpu

 

I like this one:
CSCsf17895 winmgmt.exe consumes a large amount of CPU over time until
100%

/Wes

Simon, Bill wrote: 

Consider this a "me too" post - we've been seeing some CPU-high behavior

from those services lately as well.  Restarting the services seems to 
bring it back down to normal.
 
Jason Aarons (US) wrote:
  

	Any ideas why winmgmt.exe might be at 100% on both boxes in my
CRS 
	4.0(4) cluster ? -jason
	 
	 
	 
	OS 2000.2.7sr8
	 
	 
	 
	Can I disable the Windows Instrument Management service and the
Windows 
	Instrument Management Extensions service?
	    

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



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From barry.oliver at provista-uk.com  Fri Sep 22 16:59:01 2006
From: barry.oliver at provista-uk.com (barry.oliver at provista-uk.com)
Date: Fri, 22 Sep 2006 21:59:01 +0100
Subject: [cisco-voip] Automated Reply Re: "cisco-voip Digest, Vol 43,
	Issue 115"
In-Reply-To: <mailman.595.1158958620.76842.cisco-voip@puck.nether.net>
Message-ID: <E1GQs6v-0007Xi-HZ@hermes.hosts.co.uk>

Dear cisco-voip-request at puck.nether.net

Thank you for your email. I am on annual leave at present, returning on the 25th of September. 

For any urgent enquiries can you please contact Steve McKee who will assist in my absence.  

Steve McKee
steve.mckee at provista-uk.com
07814 921 132.

Or to raise a call with our service desk please email servicedesk at provista-uk.com

Many thanks,

Barry

provista ip communications
 
t.0141 354 8883
f.0141 354 8884
 
www.provista-uk.com


From jason.aarons at us.didata.com  Fri Sep 22 17:10:12 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 22 Sep 2006 17:10:12 -0400
Subject: [cisco-voip] winmgmt.exe at 100% CPU
Message-ID: <C1FE15183DA37645BC0633BC604E44F0033B9C80@USNAEXCH.na.didata.local>

CSCsf17895 

Resolution: Reboot your server

Severity 3 - Moderate

Things fail under unusual circumstances, or minor features do not work
at all, or things fail but there is a low-impact workaround

 

I hope they raise the severity!!! I know the one customer didn't think
it was a Severity 3!  Not good when your publisher or CallCenter is at
100% cpu.

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Friday, September 22, 2006 4:26 PM
To: Simon, Bill
Cc: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
at 100% cpu

 

I like this one:
CSCsf17895 winmgmt.exe consumes a large amount of CPU over time until
100%

/Wes

Simon, Bill wrote: 

Consider this a "me too" post - we've been seeing some CPU-high behavior

from those services lately as well.  Restarting the services seems to 
bring it back down to normal.
 
Jason Aarons (US) wrote:
  

	Any ideas why winmgmt.exe might be at 100% on both boxes in my
CRS 
	4.0(4) cluster ? -jason
	 
	 
	 
	OS 2000.2.7sr8
	 
	 
	 
	Can I disable the Windows Instrument Management service and the
Windows 
	Instrument Management Extensions service?
	    

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



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From milay at cisco.com  Fri Sep 22 17:25:33 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Fri, 22 Sep 2006 14:25:33 -0700
Subject: [cisco-voip] winmgmt.exe at 100% CPU
Message-ID: <413FABF5DCE22849AF86F442E472B78BCB64D6@xmb-sjc-219.amer.cisco.com>

The workaround needs to be updated. You can work around this by turning
off winmgmt logging.
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Friday, September 22, 2006 4:10 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] winmgmt.exe at 100% CPU



CSCsf17895 

Resolution: Reboot your server

Severity 3 - Moderate

Things fail under unusual circumstances, or minor features do not work
at all, or things fail but there is a low-impact workaround

 

I hope they raise the severity!!! I know the one customer didn't think
it was a Severity 3!  Not good when your publisher or CallCenter is at
100% cpu.

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Friday, September 22, 2006 4:26 PM
To: Simon, Bill
Cc: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
at 100% cpu

 

I like this one:
CSCsf17895 winmgmt.exe consumes a large amount of CPU over time until
100%

/Wes

Simon, Bill wrote: 

Consider this a "me too" post - we've been seeing some CPU-high behavior

from those services lately as well.  Restarting the services seems to 
bring it back down to normal.
 
Jason Aarons (US) wrote:
  

	Any ideas why winmgmt.exe might be at 100% on both boxes in my
CRS 
	4.0(4) cluster ? -jason
	 
	 
	 
	OS 2000.2.7sr8
	 
	 
	 
	Can I disable the Windows Instrument Management service and the
Windows 
	Instrument Management Extensions service?
	    

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

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
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From jason.aarons at us.didata.com  Fri Sep 22 17:48:54 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Fri, 22 Sep 2006 17:48:54 -0400
Subject: [cisco-voip] winmgmt.exe at 100% CPU
Message-ID: <C1FE15183DA37645BC0633BC604E44F0033B9CFE@USNAEXCH.na.didata.local>

Can you elaborate on steps to doing this?  Do you mean setting Event
Viewer log file(s) size to zero?

 

________________________________

From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Friday, September 22, 2006 5:26 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] winmgmt.exe at 100% CPU

 

The workaround needs to be updated. You can work around this by turning
off winmgmt logging.

 

Mike

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Friday, September 22, 2006 4:10 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] winmgmt.exe at 100% CPU

CSCsf17895 

Resolution: Reboot your server

Severity 3 - Moderate

Things fail under unusual circumstances, or minor features do not work
at all, or things fail but there is a low-impact workaround

 

I hope they raise the severity!!! I know the one customer didn't think
it was a Severity 3!  Not good when your publisher or CallCenter is at
100% cpu.

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Friday, September 22, 2006 4:26 PM
To: Simon, Bill
Cc: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
at 100% cpu

 

I like this one:
CSCsf17895 winmgmt.exe consumes a large amount of CPU over time until
100%

/Wes

Simon, Bill wrote: 

Consider this a "me too" post - we've been seeing some CPU-high behavior

from those services lately as well.  Restarting the services seems to 
bring it back down to normal.
 
Jason Aarons (US) wrote:
  

	Any ideas why winmgmt.exe might be at 100% on both boxes in my
CRS 
	4.0(4) cluster ? -jason
	 
	 
	 
	OS 2000.2.7sr8
	 
	 
	 
	Can I disable the Windows Instrument Management service and the
Windows 
	Instrument Management Extensions service?
	    

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


Disclaimer: This e-mail communication and any attachments may contain
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error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
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From ealeatherman at gmail.com  Sun Sep 24 10:41:02 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Sun, 24 Sep 2006 10:41:02 -0400
Subject: [cisco-voip] Passing calls from CCM to Unity CallHandler
Message-ID: <94a1afde0609240741y19b4dafawac52d99a39a0164a@mail.gmail.com>

Hi folks,

I'm curious how other people setup call handlers in CCM and Unity..
specifically how you route a call to an extension directly to a Unity Call
Handler. Is there a "Cisco Recommended" way to do this?

Right now i'm making hunt pilots and pointing them at the Unity lines and
then creating a call routing rule. In the past though I've used dumby phones
or CTI Route ports to forward to Unity. That seemed kinda clunky so I
started using the hunt pilots once I realized I could.

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Sun Sep 24 10:46:30 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun, 24 Sep 2006 10:46:30 -0400
Subject: [cisco-voip] Passing calls from CCM to Unity CallHandler
References: <94a1afde0609240741y19b4dafawac52d99a39a0164a@mail.gmail.com>
Message-ID: <002b01c6dfe8$39597ad0$c801a8c0@EATIUSBIRDIUS>

We still use CTI route points. The reason we like it, is because they offer us a few things:
  a.. multiple DNs per CTI route point allow us to create a CTI route point per group (department) and keep all the DNs in there.
  b.. forwarding options allow us to redirect calls when we have to
We just upgraded to 4.1(3) so we really haven't explored using hunt groups. Problem is, we'd likely never get the resources to move them all over and rather than having two ways to do it, we'll just stick with one.
  ----- Original Message ----- 
  From: Ed Leatherman 
  To: cisco-voip at puck.nether.net 
  Sent: Sunday, September 24, 2006 10:41 AM
  Subject: [cisco-voip] Passing calls from CCM to Unity CallHandler


  Hi folks, 

  I'm curious how other people setup call handlers in CCM and Unity.. specifically how you route a call to an extension directly to a Unity Call Handler. Is there a "Cisco Recommended" way to do this? 

  Right now i'm making hunt pilots and pointing them at the Unity lines and then creating a call routing rule. In the past though I've used dumby phones or CTI Route ports to forward to Unity. That seemed kinda clunky so I started using the hunt pilots once I realized I could. 

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


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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From aman.chugh at gmail.com  Sun Sep 24 10:47:10 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Sun, 24 Sep 2006 20:17:10 +0530
Subject: [cisco-voip] Passing calls from CCM to Unity CallHandler
In-Reply-To: <94a1afde0609240741y19b4dafawac52d99a39a0164a@mail.gmail.com>
References: <94a1afde0609240741y19b4dafawac52d99a39a0164a@mail.gmail.com>
Message-ID: <d4c697ad0609240747m3c7091c8qf558304bdcb40d5e@mail.gmail.com>

Translation patterns is what I would recommend ,if you want a call to land
directly in Unity.

Aman


On 9/24/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> Hi folks,
>
> I'm curious how other people setup call handlers in CCM and Unity..
> specifically how you route a call to an extension directly to a Unity Call
> Handler. Is there a "Cisco Recommended" way to do this?
>
> Right now i'm making hunt pilots and pointing them at the Unity lines and
> then creating a call routing rule. In the past though I've used dumby phones
> or CTI Route ports to forward to Unity. That seemed kinda clunky so I
> started using the hunt pilots once I realized I could.
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From lelio at uoguelph.ca  Sun Sep 24 10:59:22 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sun, 24 Sep 2006 10:59:22 -0400
Subject: [cisco-voip] Passing calls from CCM to Unity CallHandler
References: <94a1afde0609240741y19b4dafawac52d99a39a0164a@mail.gmail.com>
	<d4c697ad0609240747m3c7091c8qf558304bdcb40d5e@mail.gmail.com>
Message-ID: <002101c6dfea$0505df10$c801a8c0@EATIUSBIRDIUS>

With translations, you have to send them directly to a Unity port in order for them to work because it's as if the user dialed the number directly. 

With CTI route points, because they are forwarded, Unity/CCM will remember the first redirected number and slap them into the appropriate voicemailbox/call handler.

Again, there's many ways to skin this cat.....
  ----- Original Message ----- 
  From: Aman Chugh 
  To: Ed Leatherman 
  Cc: cisco-voip at puck.nether.net 
  Sent: Sunday, September 24, 2006 10:47 AM
  Subject: Re: [cisco-voip] Passing calls from CCM to Unity CallHandler


  Translation patterns is what I would recommend ,if you want a call to land directly in Unity.

  Aman

   
  On 9/24/06, Ed Leatherman <ealeatherman at gmail.com> wrote: 
    Hi folks, 

    I'm curious how other people setup call handlers in CCM and Unity.. specifically how you route a call to an extension directly to a Unity Call Handler. Is there a "Cisco Recommended" way to do this? 

    Right now i'm making hunt pilots and pointing them at the Unity lines and then creating a call routing rule. In the past though I've used dumby phones or CTI Route ports to forward to Unity. That seemed kinda clunky so I started using the hunt pilots once I realized I could. 

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







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


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From milay at cisco.com  Sun Sep 24 11:40:26 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Sun, 24 Sep 2006 08:40:26 -0700
Subject: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
	at100%	cpu
Message-ID: <413FABF5DCE22849AF86F442E472B78BCB6642@xmb-sjc-219.amer.cisco.com>

There is not ES for this at this time. CCM has a similar issue, the bug
Wes noted. The workaround is to disable winmgmt trace. I do not know
that the IPCC Express issue is the same, however you can try the
workaround with no negative affect.:
 
http://support.microsoft.com/default.aspx?scid=kb;en-us;830075
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Friday, September 22, 2006 3:57 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
at100% cpu



Is there a ES special or hotfix? or still pending resolution?  

 

I'm only seeing it on IPCC Cluster, does notes show if CallManager or
other platforms are seeing this?

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Friday, September 22, 2006 4:26 PM
To: Simon, Bill
Cc: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
at 100% cpu

 

I like this one:
CSCsf17895 winmgmt.exe consumes a large amount of CPU over time until
100%

/Wes

Simon, Bill wrote: 

Consider this a "me too" post - we've been seeing some CPU-high behavior

from those services lately as well.  Restarting the services seems to 
bring it back down to normal.
 
Jason Aarons (US) wrote:
  

	Any ideas why winmgmt.exe might be at 100% on both boxes in my
CRS 
	4.0(4) cluster ? -jason
	 
	 
	 
	OS 2000.2.7sr8
	 
	 
	 
	Can I disable the Windows Instrument Management service and the
Windows 
	Instrument Management Extensions service?
	    

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

________________________________

Disclaimer: This e-mail communication and any attachments may contain
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From milay at cisco.com  Sun Sep 24 11:41:03 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Sun, 24 Sep 2006 08:41:03 -0700
Subject: [cisco-voip] winmgmt.exe at 100% CPU
Message-ID: <413FABF5DCE22849AF86F442E472B78BCB6643@xmb-sjc-219.amer.cisco.com>

Not event log, winmgmt trace.:
http://support.microsoft.com/default.aspx?scid=kb;en-us;830075
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Friday, September 22, 2006 4:49 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] winmgmt.exe at 100% CPU



Can you elaborate on steps to doing this?  Do you mean setting Event
Viewer log file(s) size to zero?

 

________________________________

From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Friday, September 22, 2006 5:26 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] winmgmt.exe at 100% CPU

 

The workaround needs to be updated. You can work around this by turning
off winmgmt logging.

 

Mike

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Friday, September 22, 2006 4:10 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] winmgmt.exe at 100% CPU

CSCsf17895 

Resolution: Reboot your server

Severity 3 - Moderate

Things fail under unusual circumstances, or minor features do not work
at all, or things fail but there is a low-impact workaround

 

I hope they raise the severity!!! I know the one customer didn't think
it was a Severity 3!  Not good when your publisher or CallCenter is at
100% cpu.

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Friday, September 22, 2006 4:26 PM
To: Simon, Bill
Cc: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
at 100% cpu

 

I like this one:
CSCsf17895 winmgmt.exe consumes a large amount of CPU over time until
100%

/Wes

Simon, Bill wrote: 

Consider this a "me too" post - we've been seeing some CPU-high behavior

from those services lately as well.  Restarting the services seems to 
bring it back down to normal.
 
Jason Aarons (US) wrote:
  

	Any ideas why winmgmt.exe might be at 100% on both boxes in my
CRS 
	4.0(4) cluster ? -jason
	 
	 
	 
	OS 2000.2.7sr8
	 
	 
	 
	Can I disable the Windows Instrument Management service and the
Windows 
	Instrument Management Extensions service?
	    

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


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

________________________________

Disclaimer: This e-mail communication and any attachments may contain
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From barry.oliver at provista-uk.com  Sun Sep 24 11:44:16 2006
From: barry.oliver at provista-uk.com (barry.oliver at provista-uk.com)
Date: Sun, 24 Sep 2006 16:44:16 +0100
Subject: [cisco-voip] Automated Reply Re: "cisco-voip Digest, Vol 43,
	Issue 118"
In-Reply-To: <mailman.1194.1159112434.76842.cisco-voip@puck.nether.net>
Message-ID: <E1GRW9Q-0000fx-2A@athena.hosts.co.uk>

Dear cisco-voip-request at puck.nether.net

Thank you for your email. I am on annual leave at present, returning on the 25th of September. 

For any urgent enquiries can you please contact Steve McKee who will assist in my absence.  

Steve McKee
steve.mckee at provista-uk.com
07814 921 132.

Or to raise a call with our service desk please email servicedesk at provista-uk.com

Many thanks,

Barry

provista ip communications
 
t.0141 354 8883
f.0141 354 8884
 
www.provista-uk.com


From rsingleton at novateck.com  Sun Sep 24 17:12:03 2006
From: rsingleton at novateck.com (Robert Singleton)
Date: Sun, 24 Sep 2006 16:12:03 -0500
Subject: [cisco-voip] outbound only FXO
In-Reply-To: <170BC64B51964949A28FE8A48EF1D4EB01820DDA@corpexchsrv01.adomo.com>
References: <170BC64B51964949A28FE8A48EF1D4EB01820DDA@corpexchsrv01.adomo.com>
Message-ID: <1159132323.19690.10.camel@robert.morsco.com>

Most of my branch offices are pure VoIP, with no local PSTN gateway at
all. Some of the more critical branches would like to have a couple of
POTS lines connected to FXO ports and SRST configured, which is in
itself fairly easy. The trouble is that we'd like to use existing lines
that are otherwise only used for the burglar alarm or credit card
machine. As such, I do not want the gateway to answer these lines except
in fallback mode. I though setting the ports as "Outbound" in Gateway
Configuration in CCM would do this. Unfortunately, the gateway still
picks up the line, but now it gives the caller one ring then reorder
because it (obviously) doesn't know where to send a caller on an
"Outbound" line.

Any ideas?

I am using 2811's configured with VIC2-4FXO cards, running IOS
12.3(14)T7, with loop start CO lines.

Thanks!

Robert


From bababooey at cox.net  Mon Sep 25 00:18:13 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Mon, 25 Sep 2006 00:18:13 -0400
Subject: [cisco-voip] Caller-id and name debugging
References: <mailman.3.1159113604.73312.cisco-voip@puck.nether.net>
Message-ID: <000f01c6e059$9e56d600$0f0110ac@corporation2>

Hello, I have a CCM 4.1.3 with an mgcp gw running a PRI on 2811 isr.
I am trying to pass caller-id name and number, i am getting the calling 
number but not the name.

Which debug do I need to run on the voice gw to see if the info is passing?
Which CCM trace would display if the call manager is sending and receiving 
the info?

Thanks. 



From aman.chugh at gmail.com  Mon Sep 25 00:26:41 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Mon, 25 Sep 2006 09:56:41 +0530
Subject: [cisco-voip] Caller-id and name debugging
In-Reply-To: <000f01c6e059$9e56d600$0f0110ac@corporation2>
References: <mailman.3.1159113604.73312.cisco-voip@puck.nether.net>
	<000f01c6e059$9e56d600$0f0110ac@corporation2>
Message-ID: <d4c697ad0609242126s17909522i4cfe20cb4512a804@mail.gmail.com>

debug isdn q931.

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080237271.shtml

Aman


On 9/25/06, Bob A. Bowie <bababooey at cox.net> wrote:
>
> Hello, I have a CCM 4.1.3 with an mgcp gw running a PRI on 2811 isr.
> I am trying to pass caller-id name and number, i am getting the calling
> number but not the name.
>
> Which debug do I need to run on the voice gw to see if the info is
> passing?
> Which CCM trace would display if the call manager is sending and receiving
> the info?
>
> Thanks.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From sil at infiltrated.net  Mon Sep 25 00:28:18 2006
From: sil at infiltrated.net (J. Oquendo)
Date: Sun, 24 Sep 2006 23:28:18 -0500
Subject: [cisco-voip] Caller-id and name debugging
In-Reply-To: <000f01c6e059$9e56d600$0f0110ac@corporation2>
References: <mailman.3.1159113604.73312.cisco-voip@puck.nether.net>
	<000f01c6e059$9e56d600$0f0110ac@corporation2>
Message-ID: <20060925042818.GA35714@infiltrated.net>

On Mon, 25 Sep 2006, Bob A. Bowie wrote:

> Hello, I have a CCM 4.1.3 with an mgcp gw running a PRI on 2811 isr.
> I am trying to pass caller-id name and number, i am getting the calling 
> number but not the name.
> 
> Which debug do I need to run on the voice gw to see if the info is passing?
> Which CCM trace would display if the call manager is sending and receiving 
> the info?
> 
> Thanks. 


debug isdn q931 

-- 
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil infiltrated . net http://www.infiltrated.net

"How a man plays the game shows something of his
character - how he loses shows all" - Mr. Luckey 

From Nick.Kassel at Charles-Stanley.co.uk  Mon Sep 25 07:48:04 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 25 Sep 2006 12:48:04 +0100
Subject: [cisco-voip] Unassigned Directory Numbers
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A07@CSEXCH01.charles-stanley.co.uk>

We seem to have a large amount of Unassigned DN on our call manager, is
it best practice to delete these from the system or will they not cause
us any issues.

 

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

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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You accept that any instructions are deemed to have been given at the time the
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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 Charles Stanley Securities 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 milay at cisco.com  Mon Sep 25 08:01:44 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Mon, 25 Sep 2006 05:01:44 -0700
Subject: [cisco-voip] Unassigned Directory Numbers
Message-ID: <413FABF5DCE22849AF86F442E472B78BD1FE9E@xmb-sjc-219.amer.cisco.com>

This feature was added in 4.0, a few customers did not want the DN's to
be deleted from the system after they were removed from the phones. If
there is no use for these DN's please feel free to delete them.
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, September 25, 2006 6:48 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unassigned Directory Numbers



We seem to have a large amount of Unassigned DN on our call manager, is
it best practice to delete these from the system or will they not cause
us any issues.

 




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

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
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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 Charles Stanley Securities 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
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VirusScan and SurfControl Email Filter software.
	
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From rratliff at cisco.com  Mon Sep 25 08:51:19 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 25 Sep 2006 08:51:19 -0400
Subject: [cisco-voip] outbound only FXO
In-Reply-To: <1159132323.19690.10.camel@robert.morsco.com>
Message-ID: <009801c6e0a1$4cf88630$05000100@amer.cisco.com>

Why add them to CM at all if they are only to be used in SRST mode? 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert Singleton
Sent: Sunday, September 24, 2006 5:12 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] outbound only FXO

Most of my branch offices are pure VoIP, with no local PSTN gateway at all.
Some of the more critical branches would like to have a couple of POTS lines
connected to FXO ports and SRST configured, which is in itself fairly easy.
The trouble is that we'd like to use existing lines that are otherwise only
used for the burglar alarm or credit card machine. As such, I do not want
the gateway to answer these lines except in fallback mode. I though setting
the ports as "Outbound" in Gateway Configuration in CCM would do this.
Unfortunately, the gateway still picks up the line, but now it gives the
caller one ring then reorder because it (obviously) doesn't know where to
send a caller on an "Outbound" line.

Any ideas?

I am using 2811's configured with VIC2-4FXO cards, running IOS 12.3(14)T7,
with loop start CO lines.

Thanks!

Robert

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

From wsisk at cisco.com  Mon Sep 25 09:19:10 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 25 Sep 2006 09:19:10 -0400
Subject: [cisco-voip] outbound only FXO
In-Reply-To: <1159132323.19690.10.camel@robert.morsco.com>
References: <170BC64B51964949A28FE8A48EF1D4EB01820DDA@corpexchsrv01.adomo.com>
	<1159132323.19690.10.camel@robert.morsco.com>
Message-ID: <4517D74E.8020805@cisco.com>

Robert,

If you don't want to allow inbound calls simply do not configure a plar 
destination.  Since these are FXO there is no method of receiving called 
party number so the call will not route to any destination.   There is a 
possible toll fraud issue there if your config allows an external user 
to draw dialtone and then enter any digits to dial a destination.

Another option would be to PLAR the ports to an invalid number. 

/Wes

Robert Singleton wrote:
> Most of my branch offices are pure VoIP, with no local PSTN gateway at
> all. Some of the more critical branches would like to have a couple of
> POTS lines connected to FXO ports and SRST configured, which is in
> itself fairly easy. The trouble is that we'd like to use existing lines
> that are otherwise only used for the burglar alarm or credit card
> machine. As such, I do not want the gateway to answer these lines except
> in fallback mode. I though setting the ports as "Outbound" in Gateway
> Configuration in CCM would do this. Unfortunately, the gateway still
> picks up the line, but now it gives the caller one ring then reorder
> because it (obviously) doesn't know where to send a caller on an
> "Outbound" line.
>
> Any ideas?
>
> I am using 2811's configured with VIC2-4FXO cards, running IOS
> 12.3(14)T7, with loop start CO lines.
>
> Thanks!
>
> Robert
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From PEKOU at gr.ibm.com  Mon Sep 25 09:36:31 2006
From: PEKOU at gr.ibm.com (Evi Pekou)
Date: Mon, 25 Sep 2006 16:36:31 +0300
Subject: [cisco-voip] CallManager & IVR TCL script
Message-ID: <OFA557BD33.DB49392F-ONC22571F4.00485AD1-C22571F4.004A95AE@gr.ibm.com>

Hi, 

Is it possible to integrate IOS TCL scripts with CallManager?
I have a CCM 4.1 installation and we need to add some welcome messages 
depending on the number dialed by the customer.
Can we implement this using TCL scripts or do I need the IVR application?

Thanks,
Evi.
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From rratliff at cisco.com  Mon Sep 25 09:46:08 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 25 Sep 2006 09:46:08 -0400
Subject: [cisco-voip] CallManager & IVR TCL script
In-Reply-To: <OFA557BD33.DB49392F-ONC22571F4.00485AD1-C22571F4.004A95AE@gr.ibm.com>
Message-ID: <010b01c6e0a8$f437c120$05000100@amer.cisco.com>

There are several different ways to do this.

You can use IPCC Express, Unity, or even a TCL script on an H.323 router.
For the router if you want to intercept calls before they ever reach CM you
will have to use H.323.

It really depends on your comfort level and what products are available to
you. 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Evi Pekou
Sent: Monday, September 25, 2006 9:37 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager & IVR TCL script


Hi, 

Is it possible to integrate IOS TCL scripts with CallManager? 
I have a CCM 4.1 installation and we need to add some welcome messages
depending on the number dialed by the customer. 
Can we implement this using TCL scripts or do I need the IVR application? 

Thanks,
Evi.

From Nick.Kassel at Charles-Stanley.co.uk  Mon Sep 25 10:10:27 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 25 Sep 2006 15:10:27 +0100
Subject: [cisco-voip] Unassigned Directory Numbers
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A08@CSEXCH01.charles-stanley.co.uk>

Thanks Mike

 

What about numbers that show up in the route plan report for Unassigned
DN but are actually in use and assigned to a different partition?

 

 

 

________________________________

From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Monday, September 25, 2006 1:02 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unassigned Directory Numbers

 

This feature was added in 4.0, a few customers did not want the DN's to
be deleted from the system after they were removed from the phones. If
there is no use for these DN's please feel free to delete them.

 

Mike

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, September 25, 2006 6:48 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unassigned Directory Numbers

We seem to have a large amount of Unassigned DN on our call manager, is
it best practice to delete these from the system or will they not cause
us any issues.

 

 


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or all
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Access to this e-mail by any other person is prohibited. If you are not
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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 Charles Stanley Securities 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,
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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 Charles Stanley Securities 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 milay at cisco.com  Mon Sep 25 10:11:20 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Mon, 25 Sep 2006 07:11:20 -0700
Subject: [cisco-voip] Unassigned Directory Numbers
Message-ID: <413FABF5DCE22849AF86F442E472B78BD1FEBE@xmb-sjc-219.amer.cisco.com>

Those are considered shared lines & can be deleted also.
 
Mike

________________________________

From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Monday, September 25, 2006 9:10 AM
To: Mike Lay (milay)
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unassigned Directory Numbers



Thanks Mike

 

What about numbers that show up in the route plan report for Unassigned
DN but are actually in use and assigned to a different partition?

 

 

 

________________________________

From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Monday, September 25, 2006 1:02 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unassigned Directory Numbers

 

This feature was added in 4.0, a few customers did not want the DN's to
be deleted from the system after they were removed from the phones. If
there is no use for these DN's please feel free to delete them.

 

Mike

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, September 25, 2006 6:48 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unassigned Directory Numbers

We seem to have a large amount of Unassigned DN on our call manager, is
it best practice to delete these from the system or will they not cause
us any issues.

 

 


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

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 Charles Stanley Securities 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 Charles Stanley Securities 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 wsisk at cisco.com  Mon Sep 25 10:16:00 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 25 Sep 2006 10:16:00 -0400
Subject: [cisco-voip] Unassigned Directory Numbers
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451A08@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451A08@CSEXCH01.charles-stanley.co.uk>
Message-ID: <4517E4A0.3050502@cisco.com>

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From rratliff at cisco.com  Mon Sep 25 10:17:17 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 25 Sep 2006 10:17:17 -0400
Subject: [cisco-voip] Unassigned Directory Numbers
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451A08@CSEXCH01.charles-stanley.co.uk>
Message-ID: <013401c6e0ad$4ea22110$05000100@amer.cisco.com>

A directory number is a combination of the number and partition.  If you
have a number in partition x assigned to phone A and the same number in
partition y showing up as an unassigned DN then deleting the unassigned DN
will have no impact on phone A whatsoever. 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, September 25, 2006 10:10 AM
To: Mike Lay (milay)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unassigned Directory Numbers

Thanks Mike

 

What about numbers that show up in the route plan report for Unassigned DN
but are actually in use and assigned to a different partition?

 

 

 

________________________________

From: Mike Lay (milay) [mailto:milay at cisco.com]
Sent: Monday, September 25, 2006 1:02 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unassigned Directory Numbers

 

This feature was added in 4.0, a few customers did not want the DN's to be
deleted from the system after they were removed from the phones. If there is
no use for these DN's please feel free to delete them.

 

Mike

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, September 25, 2006 6:48 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unassigned Directory Numbers

We seem to have a large amount of Unassigned DN on our call manager, is it
best practice to delete these from the system or will they not cause us any
issues.

 

 


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

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 Charles Stanley Securities 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 Charles Stanley Securities 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 Nick.Kassel at Charles-Stanley.co.uk  Mon Sep 25 10:35:26 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 25 Sep 2006 15:35:26 +0100
Subject: [cisco-voip] Unassigned Directory Numbers
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A09@CSEXCH01.charles-stanley.co.uk>

Many thanks, I shall Delete all Found Items then.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Monday, September 25, 2006 3:17 PM
To: Nick Kassel; 'Mike Lay (milay)'
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unassigned Directory Numbers

A directory number is a combination of the number and partition.  If you
have a number in partition x assigned to phone A and the same number in
partition y showing up as an unassigned DN then deleting the unassigned
DN
will have no impact on phone A whatsoever. 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, September 25, 2006 10:10 AM
To: Mike Lay (milay)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Unassigned Directory Numbers

Thanks Mike

 

What about numbers that show up in the route plan report for Unassigned
DN
but are actually in use and assigned to a different partition?

 

 

 

________________________________

From: Mike Lay (milay) [mailto:milay at cisco.com]
Sent: Monday, September 25, 2006 1:02 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unassigned Directory Numbers

 

This feature was added in 4.0, a few customers did not want the DN's to
be
deleted from the system after they were removed from the phones. If
there is
no use for these DN's please feel free to delete them.

 

Mike

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, September 25, 2006 6:48 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unassigned Directory Numbers

We seem to have a large amount of Unassigned DN on our call manager, is
it
best practice to delete these from the system or will they not cause us
any
issues.

 

 


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

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 Charles Stanley Securities 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 Charles Stanley Securities 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 Charles Stanley Securities 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 BRiley at jackhenry.com  Mon Sep 25 10:39:43 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Mon, 25 Sep 2006 09:39:43 -0500
Subject: [cisco-voip] Ephone CH1 Down Call Manager Express
Message-ID: <2B3745BC9D31514381A6FCC5E72D937310A213@SMOMAIL.jhacorp.com>

 

I have an issue where an inbound call translates to a DN but the caller
is getting sent directly to VM instead of ringing the phone. I have
attached some debug voip ccapi inout and the ephone status. It shows the
call being redirect to their second DN 2135 which it should if it is
busy and then to VM 8000. If I reset the phone it will start working but
this has been happening to this site sporadically and I do not want to
constantly reset phones. Any suggestion on why CH1 and CH2 show down on
the phone? 

 

 

ephone-24 Mac:0015.2BD4.17A2 TCP socket:[18] activeLine:0 REGISTERED in
SCCP ver 6 and Server in ver 5

mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:1 paging 0 debug:0
caps:7

IP:10.10.200.14 51160 Telecaster 7940  keepalive 9445 max_line 2

button 1: dn 24 number 1185 CH1   DOWN         CH2   DOWN

button 2: dn 54 number 1100 CH1   IDLE         CH2   BUSY
feature-ring shared

 

 

 

 

Sep 25 09:25:51: //53986/32946F87945C/CCAPI/cc_process_call_setup_ind:

   >>>>CCAPI handed cid 53986 with tag 1 to app
"_ManagedAppProcess_Default"

Sep 25 09:25:51: //53986/32946F87945C/CCAPI/ccCallProceeding:

   Progress Indication=NULL(0)

Sep 25 09:25:51: //-1/xxxxxxxxxxxx/CCAPI/ccUpdateRedirectingNumber:

   type=6 redirecting_number=1185

Sep 25 09:25:51: //-1/xxxxxxxxxxxx/CCAPI/ccUpdateRedirectNumber:

   type=6  Original Called Number=1185, Calling Number=XXXXXXXXXX,
Calling DN=-1,

   Redirect Number=2185, Redirect Reason=1

Sep 25 09:25:51: //-1/xxxxxxxxxxxx/CCAPI/ccUpdateRedirectingNumber:

   type=6 redirecting_number=1185

Sep 25 09:25:51: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

 

 

 

Sep 25 09:25:51:  vsacount in free is 10

Sep 25 09:25:51: //53986/32946F87945C/CCAPI/ccGetTBCTCap:

   TGRM TBCT Enabled; TBCT Capability=0

Sep 25 09:25:51: //53986/32946F87945C/CCAPI/ccGetTBCTCap:

   TGRM TBCT Enabled; TBCT Capability=0

Sep 25 09:25:51: //-1/xxxxxxxxxxxx/CCAPI/ccUpdateRedirectNumber:

   type=6  Original Called Number=2185, Calling Number=XXXXXXXXXX,
Calling DN=-1,

   Redirect Number=8000, Redirect Reason=1

Sep 25 09:25:51: //-1/xxxxxxxxxxxx/CCAPI/ccUpdateRedirectingNumber:

   type=6 redirecting_number=2185

Sep 25 09:25:51: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

 

Thanks,

Bill Riley

 

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 manoj.kalpage at gmail.com  Mon Sep 25 11:19:15 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Tue, 26 Sep 2006 00:19:15 +0900
Subject: [cisco-voip] Head-end VPN Failover for VoIP
Message-ID: <bbe09c1d0609250819n543440d9l1b37a6bcb1c28b23@mail.gmail.com>

Dear All,
Does anyone know whether I can find step by step sample configuration for
Head-end VPN Failover for VoIP?

I found two methods for Cisco IOS VPN as bellow. Does both supports for
Head-end VPN Failover?
DMVPN using GRE over IPSEC
GRE over IPSec with EIGRP

If So, can I use either of combination bellow
DMVPN using GRE over IPSEC + Head-end VPN Failover
GRE over IPSec with EIGRP + Head-end VPN Failover

I have been researching this for couple of week but still I did not find the
much information. I would greatly appreciate  your  kind reply.

Best Regards,
Manoj
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From sil at infiltrated.net  Mon Sep 25 11:47:35 2006
From: sil at infiltrated.net (J. Oquendo)
Date: Mon, 25 Sep 2006 11:47:35 -0400
Subject: [cisco-voip] Head-end VPN Failover for VoIP
In-Reply-To: <bbe09c1d0609250819n543440d9l1b37a6bcb1c28b23@mail.gmail.com>
References: <bbe09c1d0609250819n543440d9l1b37a6bcb1c28b23@mail.gmail.com>
Message-ID: <4517FA17.1050906@infiltrated.net>

Manoj Kalpage wrote:
> Dear All,
> Does anyone know whether I can find step by step sample configuration 
> for Head-end VPN Failover for VoIP?
>
> I found two methods for Cisco IOS VPN as bellow. Does both supports 
> for Head-end VPN Failover?
> DMVPN using GRE over IPSEC
> GRE over IPSec with EIGRP
>
> If So, can I use either of combination bellow
> DMVPN using GRE over IPSEC + Head-end VPN Failover
> GRE over IPSec with EIGRP + Head-end VPN Failover
>
> I have been researching this for couple of week but still I did not 
> find the much information. I would greatly appreciate  your  kind reply.
>
> Best Regards,
> Manoj
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   
http://www.cisco.com/en/US/products/ps6660/products_white_paper0900aecd8051bf3b.shtml


-- 
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 

The happiness of society is the end of government.
John Adams


From bababooey at cox.net  Mon Sep 25 12:29:19 2006
From: bababooey at cox.net (bababooey at cox.net)
Date: Mon, 25 Sep 2006 12:29:19 -0400
Subject: [cisco-voip] Caller-id and name debugging
Message-ID: <5503797.1159201759047.JavaMail.root@eastrmwml03.mgt.cox.net>

>From the Voice-GW, debug q931. This is a call from CCm to PBX.
Where is the calling name supposed to be?

2811#
*Sep 25 16:32:16.651: ISDN Se1/0/0:23 Q931: TX -> SETUP pd = 8  callref = 0x0003 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transfer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA98397 
                Exclusive, Channel 23 
        Net Specific Fac i = 0x00E6 
        Calling Party Number i = 0x0081, '5670800' 
                Plan:Unknown, Type:Unknown 
        Called Party Number i = 0x80, '2757001' 
                Plan:Unknown, Type:Unknown
*Sep 25 16:32:16.671: ISDN Se1/0/0:23 Q931: RX <- CALL_PROC pd = 8  callref = 0x8003 
        Channel ID i = 0xA98397 
                Exclusive, Channel 23
*Sep 25 16:32:16.675: ISDN Se1/0/0:23 Q931: RX <- ALERTING pd = 8  callref = 0x8003 
        Progress Ind i = 0x8081 - Call not end-to-end ISDN, may have in-band info 
*Sep 25 16:32:36.235: ISDN Se1/0/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0x0003 
        Cause i = 0x8090 - Normal call clearing
*Sep 25 16:32:36.247: ISDN Se1/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x8003
*Sep 25 16:32:36.267: ISDN Se1/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x0003
2811#


---- Aman Chugh <aman.chugh at gmail.com> wrote: 
> debug isdn q931.
> 
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080237271.shtml
> 
> Aman
> 
> 
> On 9/25/06, Bob A. Bowie <bababooey at cox.net> wrote:
> >
> > Hello, I have a CCM 4.1.3 with an mgcp gw running a PRI on 2811 isr.
> > I am trying to pass caller-id name and number, i am getting the calling
> > number but not the name.
> >
> > Which debug do I need to run on the voice gw to see if the info is
> > passing?
> > Which CCM trace would display if the call manager is sending and receiving
> > the info?
> >
> > Thanks.
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >


From rratliff at cisco.com  Mon Sep 25 12:39:30 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 25 Sep 2006 12:39:30 -0400
Subject: [cisco-voip] Caller-id and name debugging
In-Reply-To: <5503797.1159201759047.JavaMail.root@eastrmwml03.mgt.cox.net>
Message-ID: <01a801c6e0c1$2c780190$05000100@amer.cisco.com>

It can be in a few places actually.

In the CCMAdmin page for this endpoint do you have "Display IE Outbound"
checked?  This is required for CM to present the Display IE in the Q931
Setup message. 

It will require a reset of the gateway to take affect (will drop calls).  If
you have 'ccm-manager config' enabled simply reset it via CCMAdmin.  If you
are not using that feature you will need to do a 'no mgcp/mgcp' on the
router.

-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of bababooey at cox.net
Sent: Monday, September 25, 2006 12:29 PM
To: Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Caller-id and name debugging

>From the Voice-GW, debug q931. This is a call from CCm to PBX.
Where is the calling name supposed to be?

2811#
*Sep 25 16:32:16.651: ISDN Se1/0/0:23 Q931: TX -> SETUP pd = 8  callref =
0x0003 
        Bearer Capability i = 0x8090A2 
                Standard = CCITT 
                Transfer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA98397 
                Exclusive, Channel 23 
        Net Specific Fac i = 0x00E6 
        Calling Party Number i = 0x0081, '5670800' 
                Plan:Unknown, Type:Unknown 
        Called Party Number i = 0x80, '2757001' 
                Plan:Unknown, Type:Unknown *Sep 25 16:32:16.671: ISDN
Se1/0/0:23 Q931: RX <- CALL_PROC pd = 8  callref = 0x8003 
        Channel ID i = 0xA98397 
                Exclusive, Channel 23
*Sep 25 16:32:16.675: ISDN Se1/0/0:23 Q931: RX <- ALERTING pd = 8  callref =
0x8003 
        Progress Ind i = 0x8081 - Call not end-to-end ISDN, may have in-band
info *Sep 25 16:32:36.235: ISDN Se1/0/0:23 Q931: TX -> DISCONNECT pd = 8
callref = 0x0003 
        Cause i = 0x8090 - Normal call clearing *Sep 25 16:32:36.247: ISDN
Se1/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 0x8003 *Sep 25
16:32:36.267: ISDN Se1/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref =
0x0003 2811#


---- Aman Chugh <aman.chugh at gmail.com> wrote: 
> debug isdn q931.
> 
>
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a0080237271.shtml
> 
> Aman
> 
> 
> On 9/25/06, Bob A. Bowie <bababooey at cox.net> wrote:
> >
> > Hello, I have a CCM 4.1.3 with an mgcp gw running a PRI on 2811 isr.
> > I am trying to pass caller-id name and number, i am getting the calling
> > number but not the name.
> >
> > Which debug do I need to run on the voice gw to see if the info is
> > passing?
> > Which CCM trace would display if the call manager is sending and
receiving
> > the info?
> >
> > Thanks.
> >
> >
> > _______________________________________________
> > 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 milay at cisco.com  Mon Sep 25 13:31:25 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Mon, 25 Sep 2006 10:31:25 -0700
Subject: [cisco-voip] winmgmt.exe at 100% CPU
Message-ID: <413FABF5DCE22849AF86F442E472B78BD1FFBC@xmb-sjc-219.amer.cisco.com>

Sorry but the logging should be disabled rather than error as stated in
the MS kbase link below.:
 
Workaround:

1. Click Start, click Run, type wmimgmt.msc, and then click OK. 
2. Right-click WMI Control (Local), and then click Properties. 
3. Click the Logging tab, and then click "Disabled" under Logging level.

4. Click OK. 

Thanks,
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Mike Lay
(milay)
Sent: Sunday, September 24, 2006 10:41 AM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] winmgmt.exe at 100% CPU


Not event log, winmgmt trace.:
http://support.microsoft.com/default.aspx?scid=kb;en-us;830075
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Friday, September 22, 2006 4:49 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] winmgmt.exe at 100% CPU



Can you elaborate on steps to doing this?  Do you mean setting Event
Viewer log file(s) size to zero?

 

________________________________

From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Friday, September 22, 2006 5:26 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] winmgmt.exe at 100% CPU

 

The workaround needs to be updated. You can work around this by turning
off winmgmt logging.

 

Mike

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Friday, September 22, 2006 4:10 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] winmgmt.exe at 100% CPU

CSCsf17895 

Resolution: Reboot your server

Severity 3 - Moderate

Things fail under unusual circumstances, or minor features do not work
at all, or things fail but there is a low-impact workaround

 

I hope they raise the severity!!! I know the one customer didn't think
it was a Severity 3!  Not good when your publisher or CallCenter is at
100% cpu.

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Friday, September 22, 2006 4:26 PM
To: Simon, Bill
Cc: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Two IPCC X Cluster 4.x boxes with winmgmt.exe
at 100% cpu

 

I like this one:
CSCsf17895 winmgmt.exe consumes a large amount of CPU over time until
100%

/Wes

Simon, Bill wrote: 

Consider this a "me too" post - we've been seeing some CPU-high behavior

from those services lately as well.  Restarting the services seems to 
bring it back down to normal.
 
Jason Aarons (US) wrote:
  

	Any ideas why winmgmt.exe might be at 100% on both boxes in my
CRS 
	4.0(4) cluster ? -jason
	 
	 
	 
	OS 2000.2.7sr8
	 
	 
	 
	Can I disable the Windows Instrument Management service and the
Windows 
	Instrument Management Extensions service?
	    

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


Disclaimer: This e-mail communication and any attachments may contain
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________________________________

Disclaimer: This e-mail communication and any attachments may contain
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From tim.reimers at asheville.k12.nc.us  Mon Sep 25 14:15:37 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Mon, 25 Sep 2006 14:15:37 -0400
Subject: [cisco-voip] MCID
In-Reply-To: <4514032A.4040207@cisco.com>
Message-ID: <050944EE48FF384F9C663F7048C47DA704E94739@webmail.asheville.k12.nc.us>

Hey Wes---
 
How do you know if your PRI supports it?
 
Ours is currently running H.323 -- going to convert to MGCP shortly.
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Friday, September 22, 2006 11:37 AM
To: Voll, Scott
Cc: Brad Rogers; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MCID


CM supports MCID-O, basically user invokation sent to network.  It will
work with ISDN PRI specs that support the feature.  It does not work
with QSIG, CAS, analog, etc.  The message sent to a carrier is a
FacilityMessage with FacilityIE.  Traces should look something like the
attached.
 
/Wes

Voll, Scott wrote: 

	My understanding is NO.

	

	Scott

	

	
________________________________


	From: cisco-voip-bounces at puck.nether.net [
mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Rogers
	Sent: Thursday, September 21, 2006 12:35 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] MCID

	

	I know when you push the MCID button it flags the call as
malicious on CallManager, but does anybody know if it flags it on the
carrier's side as well?

	

	

	
________________________________


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

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From tim.reimers at asheville.k12.nc.us  Mon Sep 25 14:26:26 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Mon, 25 Sep 2006 14:26:26 -0400
Subject: [cisco-voip] any way to schedule reboots of phones??
Message-ID: <050944EE48FF384F9C663F7048C47DA704E94748@webmail.asheville.k12.nc.us>

hi everyone-

I just made a global change to all phones on the system--

I can't restart all phones during the day---

Is there some sort of scheduleable reboot of all phones? (one I could
configure and not have to remember to reboot all phones tonight)

thanks, Tim
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From wsisk at cisco.com  Mon Sep 25 14:29:29 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 25 Sep 2006 14:29:29 -0400
Subject: [cisco-voip] any way to schedule reboots of phones??
In-Reply-To: <050944EE48FF384F9C663F7048C47DA704E94748@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA704E94748@webmail.asheville.k12.nc.us>
Message-ID: <45182009.60907@cisco.com>

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From wsisk at cisco.com  Mon Sep 25 14:31:32 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 25 Sep 2006 14:31:32 -0400
Subject: [cisco-voip] MCID
In-Reply-To: <050944EE48FF384F9C663F7048C47DA704E94739@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA704E94739@webmail.asheville.k12.nc.us>
Message-ID: <45182084.1020806@cisco.com>

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From cwhitten at nexband.com  Mon Sep 25 14:35:21 2006
From: cwhitten at nexband.com (Chad Whitten)
Date: Mon, 25 Sep 2006 13:35:21 -0500
Subject: [cisco-voip] weird music on hold problem
Message-ID: <200609251335.21319.cwhitten@nexband.com>

I have a problem with music on hold.  I have removed the music on hold from 
all the phones, all the lines, the gateway and the device pool configuration.  
have restarted all phones, have restarted the gateway, have restarted the moh 
server, and have restarted the call managers but ive still got the default 
"sample audio source" that sounds like funeral home music.

does the callmanager play the default music if nothing is selected?  if not, 
how do i make the music go away?
-- 
Chad Whitten
Chief Operations Officer
neXband Communications
cwhitten at nexband.com - Email
chadwick at jabber.nexband.com - Jabber
601-988-0101 - Phone
601-988-0016 - Fax

From rratliff at cisco.com  Mon Sep 25 14:36:49 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 25 Sep 2006 14:36:49 -0400
Subject: [cisco-voip] weird music on hold problem
In-Reply-To: <200609251335.21319.cwhitten@nexband.com>
Message-ID: <001101c6e0d1$9046bee0$05000100@amer.cisco.com>

Shut down the MOH server?

I'm pretty sure there is a default MOH source service parameter.

-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Chad Whitten
Sent: Monday, September 25, 2006 2:35 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] weird music on hold problem

I have a problem with music on hold.  I have removed the music on hold from
all the phones, all the lines, the gateway and the device pool
configuration.  
have restarted all phones, have restarted the gateway, have restarted the
moh server, and have restarted the call managers but ive still got the
default "sample audio source" that sounds like funeral home music.

does the callmanager play the default music if nothing is selected?  if not,
how do i make the music go away?
--
Chad Whitten
Chief Operations Officer
neXband Communications
cwhitten at nexband.com - Email
chadwick at jabber.nexband.com - Jabber
601-988-0101 - Phone
601-988-0016 - Fax
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

From bob at smalltime.com  Mon Sep 25 14:38:52 2006
From: bob at smalltime.com (Robert Kulagowski)
Date: Mon, 25 Sep 2006 13:38:52 -0500
Subject: [cisco-voip] Caller-id and name debugging
In-Reply-To: <5503797.1159201759047.JavaMail.root@eastrmwml03.mgt.cox.net>
References: <5503797.1159201759047.JavaMail.root@eastrmwml03.mgt.cox.net>
Message-ID: <4518223C.1090400@smalltime.com>

bababooey at cox.net wrote:
>>From the Voice-GW, debug q931. This is a call from CCm to PBX.
> Where is the calling name supposed to be?

RX <- SETUP pd = 8  callref = 0x1E2C
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transfer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98382
                 Exclusive, Channel 2
         Facility i = 0x9F8B0100A10F02017606072A8648CE1500040A0100
                 Protocol Profile =  Networking Extensions
                 0xA10F02017606072A8648CE1500040A0100
                 Component = Invoke component
                         Invoke Id = 118
                         Operation = InformationFollowing (calling_name)
                                 Name information in subsequent FACILITY 
message
         Calling Party Number i = 0x2181, '<snip>'
                 Plan:ISDN, Type:National
         Called Party Number i = 0x80, '<snip>'
                 Plan:Unknown, Type:Unknown
Sep 25 13:35:48.820: ISDN Se0/2/0:23 Q931: RX <- FACILITY pd = 8 
callref = 0x1E2C
         Facility i = 
0x9F8B0100A117020177020100800F544453202020202020202020202020
                 Protocol Profile =  Networking Extensions
                 0xA117020177020100800F544453202020202020202020202020
                 Component = Invoke component
                         Invoke Id = 119
                         Operation = CallingName
                                 Name presentation allowed
                                 Name = TDS
Sep 25 13:35:48.856: ISDN Se0/2/0:23 Q931: TX -> CALL_PROC pd = 8 
callref = 0x9E2C
         Channel ID i = 0xA98382
                 Exclusive, Channel 2
Sep 25 13:35:48.856: ISDN Se0/2/0:23 Q931: TX -> ALERTING pd = 8 
callref = 0x9E2C
         Progress Ind i = 0x8088 - In-band info or appropriate now available

=======

 From your example it doesn't look like calling name is being delivered, 
and in mine the calling name was "TDS".  But watch your debug for a 
little while and make sure that the one example you sent wasn't a fluke. 
  If you're consistently not getting it, tell your provider.


From Scott.Voll at wesd.org  Mon Sep 25 14:43:20 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 25 Sep 2006 11:43:20 -0700
Subject: [cisco-voip] any way to schedule reboots of phones??
Message-ID: <407055A92CECCB499C922A2D35FC19A60693497C@Apollo.wesd.org>

Can't you schedule the switches to reboot (reload at HH:MM)....... That
would take care of the issue.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Monday, September 25, 2006 11:29 AM
To: Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

 

Nothing directly.

If they are EM phones, you could schedule a forced EM logout.

schedule a .bat job to:
net stop "cisco callmanager"
net start "cisco callmanager"

schedule a windows reboot

/Wes

Tim Reimers wrote: 

hi everyone- 

I just made a global change to all phones on the system-- 

I can't restart all phones during the day--- 

Is there some sort of scheduleable reboot of all phones? (one I could
configure and not have to remember to reboot all phones tonight)

thanks, Tim 

 



________________________________



 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
  
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From sil at infiltrated.net  Mon Sep 25 14:41:57 2006
From: sil at infiltrated.net (J. Oquendo)
Date: Mon, 25 Sep 2006 14:41:57 -0400
Subject: [cisco-voip] any way to schedule reboots of phones??
In-Reply-To: <407055A92CECCB499C922A2D35FC19A60693497C@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A60693497C@Apollo.wesd.org>
Message-ID: <451822F5.7080803@infiltrated.net>

Voll, Scott wrote:
>
> Can?t you schedule the switches to reboot (reload at HH:MM)??. That 
> would take care of the issue.
>
> Scott
>
> ------------------------------------------------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Wes Sisk
> *Sent:* Monday, September 25, 2006 11:29 AM
> *To:* Tim Reimers
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] any way to schedule reboots of phones??
>
> Nothing directly.
>
> If they are EM phones, you could schedule a forced EM logout.
>
> schedule a .bat job to:
> net stop "cisco callmanager"
> net start "cisco callmanager"
>
> schedule a windows reboot
>
> /Wes
>
> Tim Reimers wrote:
>
> hi everyone-
>
> I just made a global change to all phones on the system--
>
> I can't restart all phones during the day---
>
> Is there some sort of scheduleable reboot of all phones? (one I could 
> configure and not have to remember to reboot all phones tonight)
>
> thanks, Tim
>

telephony-service
reset sequence-all

I could be mistaken, 99.99999% of the times I am


-- 
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 

The happiness of society is the end of government.
John Adams


From Scott.Voll at wesd.org  Mon Sep 25 14:48:41 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 25 Sep 2006 11:48:41 -0700
Subject: [cisco-voip] any way to schedule reboots of phones??
Message-ID: <407055A92CECCB499C922A2D35FC19A606934985@Apollo.wesd.org>

Where do you enter that command?

Scott


-----Original Message-----
From: J. Oquendo [mailto:sil at infiltrated.net] 
Sent: Monday, September 25, 2006 11:42 AM
To: Voll, Scott
Cc: Wes Sisk; Tim Reimers; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

Voll, Scott wrote:
>
> Can't you schedule the switches to reboot (reload at HH:MM).......
That 
> would take care of the issue.
>
> Scott
>
>
------------------------------------------------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Wes Sisk
> *Sent:* Monday, September 25, 2006 11:29 AM
> *To:* Tim Reimers
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] any way to schedule reboots of phones??
>
> Nothing directly.
>
> If they are EM phones, you could schedule a forced EM logout.
>
> schedule a .bat job to:
> net stop "cisco callmanager"
> net start "cisco callmanager"
>
> schedule a windows reboot
>
> /Wes
>
> Tim Reimers wrote:
>
> hi everyone-
>
> I just made a global change to all phones on the system--
>
> I can't restart all phones during the day---
>
> Is there some sort of scheduleable reboot of all phones? (one I could 
> configure and not have to remember to reboot all phones tonight)
>
> thanks, Tim
>

telephony-service
reset sequence-all

I could be mistaken, 99.99999% of the times I am


-- 
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 

The happiness of society is the end of government.
John Adams



From rratliff at cisco.com  Mon Sep 25 14:52:21 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 25 Sep 2006 14:52:21 -0400
Subject: [cisco-voip] Caller-id and name debugging
In-Reply-To: <4518223C.1090400@smalltime.com>
Message-ID: <002501c6e0d3$bb89a840$05000100@amer.cisco.com>

His was outbound so it would have been included in a Display IE if CM were
configured to send it.

-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert Kulagowski
Sent: Monday, September 25, 2006 2:39 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Caller-id and name debugging

bababooey at cox.net wrote:
>>From the Voice-GW, debug q931. This is a call from CCm to PBX.
> Where is the calling name supposed to be?

RX <- SETUP pd = 8  callref = 0x1E2C
         Bearer Capability i = 0x8090A2
                 Standard = CCITT
                 Transfer Capability = Speech
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98382
                 Exclusive, Channel 2
         Facility i = 0x9F8B0100A10F02017606072A8648CE1500040A0100
                 Protocol Profile =  Networking Extensions
                 0xA10F02017606072A8648CE1500040A0100
                 Component = Invoke component
                         Invoke Id = 118
                         Operation = InformationFollowing (calling_name)
                                 Name information in subsequent FACILITY
message
         Calling Party Number i = 0x2181, '<snip>'
                 Plan:ISDN, Type:National
         Called Party Number i = 0x80, '<snip>'
                 Plan:Unknown, Type:Unknown Sep 25 13:35:48.820: ISDN
Se0/2/0:23 Q931: RX <- FACILITY pd = 8 callref = 0x1E2C
         Facility i =
0x9F8B0100A117020177020100800F544453202020202020202020202020
                 Protocol Profile =  Networking Extensions
                 0xA117020177020100800F544453202020202020202020202020
                 Component = Invoke component
                         Invoke Id = 119
                         Operation = CallingName
                                 Name presentation allowed
                                 Name = TDS Sep 25 13:35:48.856: ISDN
Se0/2/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x9E2C
         Channel ID i = 0xA98382
                 Exclusive, Channel 2
Sep 25 13:35:48.856: ISDN Se0/2/0:23 Q931: TX -> ALERTING pd = 8 callref =
0x9E2C
         Progress Ind i = 0x8088 - In-band info or appropriate now available

=======

 From your example it doesn't look like calling name is being delivered, and
in mine the calling name was "TDS".  But watch your debug for a little while
and make sure that the one example you sent wasn't a fluke. 
  If you're consistently not getting it, tell your provider.

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

From BRiley at jackhenry.com  Mon Sep 25 14:57:29 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Mon, 25 Sep 2006 13:57:29 -0500
Subject: [cisco-voip] any way to schedule reboots of phones??
Message-ID: <2B3745BC9D31514381A6FCC5E72D937310A4FC@SMOMAIL.jhacorp.com>

I think that's in express.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 25, 2006 1:49 PM
To: J. Oquendo
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

Where do you enter that command?

Scott


-----Original Message-----
From: J. Oquendo [mailto:sil at infiltrated.net] 
Sent: Monday, September 25, 2006 11:42 AM
To: Voll, Scott
Cc: Wes Sisk; Tim Reimers; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

Voll, Scott wrote:
>
> Can't you schedule the switches to reboot (reload at HH:MM).......
That 
> would take care of the issue.
>
> Scott
>
>
------------------------------------------------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Wes Sisk
> *Sent:* Monday, September 25, 2006 11:29 AM
> *To:* Tim Reimers
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] any way to schedule reboots of phones??
>
> Nothing directly.
>
> If they are EM phones, you could schedule a forced EM logout.
>
> schedule a .bat job to:
> net stop "cisco callmanager"
> net start "cisco callmanager"
>
> schedule a windows reboot
>
> /Wes
>
> Tim Reimers wrote:
>
> hi everyone-
>
> I just made a global change to all phones on the system--
>
> I can't restart all phones during the day---
>
> Is there some sort of scheduleable reboot of all phones? (one I could 
> configure and not have to remember to reboot all phones tonight)
>
> thanks, Tim
>

telephony-service
reset sequence-all

I could be mistaken, 99.99999% of the times I am


-- 
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 

The happiness of society is the end of government.
John Adams


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


From tim.reimers at asheville.k12.nc.us  Mon Sep 25 15:01:47 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Mon, 25 Sep 2006 15:01:47 -0400
Subject: [cisco-voip] MCID
In-Reply-To: <45182084.1020806@cisco.com>
Message-ID: <050944EE48FF384F9C663F7048C47DA704E94779@webmail.asheville.k12.nc.us>

So I'm guessing that going to MGCP would allow the CM to "know" the ISDN
protocol ??

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Monday, September 25, 2006 2:32 PM
To: Tim Reimers
Cc: Voll, Scott; Brad Rogers; cisco-voip at puck.nether.net; Alan Surrette;
Scruggs, Maury G II [LTD]
Subject: Re: [cisco-voip] MCID


Tim,

Since it's h323 CM does not know the ISDN protocol.  I am not aware of
any limitations in the ISDN stack on the routers.

/Wes

Tim Reimers wrote: 

	Hey Wes---
	 
	How do you know if your PRI supports it?
	 
	Ours is currently running H.323 -- going to convert to MGCP
shortly.
	 

________________________________

	From: cisco-voip-bounces at puck.nether.net [
mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
	Sent: Friday, September 22, 2006 11:37 AM
	To: Voll, Scott
	Cc: Brad Rogers; cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] MCID
	
	
	CM supports MCID-O, basically user invokation sent to network.
It will work with ISDN PRI specs that support the feature.  It does not
work with QSIG, CAS, analog, etc.  The message sent to a carrier is a
FacilityMessage with FacilityIE.  Traces should look something like the
attached.
	 
	/Wes
	
	Voll, Scott wrote: 

		My understanding is NO.

		

		Scott

		

		
________________________________


		From: cisco-voip-bounces at puck.nether.net [
mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Rogers
		Sent: Thursday, September 21, 2006 12:35 PM
		To: cisco-voip at puck.nether.net
		Subject: [cisco-voip] MCID

		

		I know when you push the MCID button it flags the call
as malicious on CallManager, but does anybody know if it flags it on the
carrier's side as well?

		

		

		
________________________________


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

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From Matt.Slaga at us.didata.com  Mon Sep 25 15:01:56 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 25 Sep 2006 15:01:56 -0400
Subject: [cisco-voip] any way to schedule reboots of phones??
Message-ID: <C1FE15183DA37645BC0633BC604E44F0034017CC@USNAEXCH.na.didata.local>

That's a pretty hefty push for just a reset of phones.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 25, 2006 2:43 PM
To: Wes Sisk; Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

 

Can't you schedule the switches to reboot (reload at HH:MM)....... That
would take care of the issue.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Monday, September 25, 2006 11:29 AM
To: Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

 

Nothing directly.

If they are EM phones, you could schedule a forced EM logout.

schedule a .bat job to:
net stop "cisco callmanager"
net start "cisco callmanager"

schedule a windows reboot

/Wes

Tim Reimers wrote: 

hi everyone- 

I just made a global change to all phones on the system-- 

I can't restart all phones during the day--- 

Is there some sort of scheduleable reboot of all phones? (one I could
configure and not have to remember to reboot all phones tonight)

thanks, Tim 

 
 
 



________________________________



 
 
 
_______________________________________________
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 wsisk at cisco.com  Mon Sep 25 15:16:08 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 25 Sep 2006 15:16:08 -0400
Subject: [cisco-voip] MCID
In-Reply-To: <050944EE48FF384F9C663F7048C47DA704E94779@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA704E94779@webmail.asheville.k12.nc.us>
Message-ID: <45182AF8.8010108@cisco.com>

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From Scott.Voll at wesd.org  Mon Sep 25 15:59:51 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 25 Sep 2006 12:59:51 -0700
Subject: [cisco-voip] any way to schedule reboots of phones??
Message-ID: <407055A92CECCB499C922A2D35FC19A6069349D7@Apollo.wesd.org>

Well, the easiest would be VPN in from home.... Select all phones and
restart.  But if you don't want user intervention I was just trying to
come up with a useable option.

 

Scott

 

 

 

________________________________

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Monday, September 25, 2006 12:02 PM
To: Voll, Scott; Wes Sisk; Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] any way to schedule reboots of phones??

 

That's a pretty hefty push for just a reset of phones.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 25, 2006 2:43 PM
To: Wes Sisk; Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

 

Can't you schedule the switches to reboot (reload at HH:MM)....... That
would take care of the issue.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Monday, September 25, 2006 11:29 AM
To: Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

 

Nothing directly.

If they are EM phones, you could schedule a forced EM logout.

schedule a .bat job to:
net stop "cisco callmanager"
net start "cisco callmanager"

schedule a windows reboot

/Wes

Tim Reimers wrote: 

hi everyone- 

I just made a global change to all phones on the system-- 

I can't restart all phones during the day--- 

Is there some sort of scheduleable reboot of all phones? (one I could
configure and not have to remember to reboot all phones tonight)

thanks, Tim 

 
 
 
 
 



________________________________



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

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
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From tim.reimers at asheville.k12.nc.us  Mon Sep 25 16:17:06 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Mon, 25 Sep 2006 16:17:06 -0400
Subject: [cisco-voip] any way to schedule reboots of phones??
In-Reply-To: <407055A92CECCB499C922A2D35FC19A6069349D7@Apollo.wesd.org>
Message-ID: <050944EE48FF384F9C663F7048C47DA704E947E9@webmail.asheville.k12.nc.us>

That's probably what I'll do...
 
Right as yet, the new softkey template isn't working for the main
building.. not sure why...
 
Wes -- what's an "EM" phone?
 
That bat file would be easy to implement...

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, September 25, 2006 4:00 PM
To: Matt Slaga (US); Wes Sisk; Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] any way to schedule reboots of phones??



Well, the easiest would be VPN in from home.... Select all phones and
restart.  But if you don't want user intervention I was just trying to
come up with a useable option.

 

Scott

 

 

 

________________________________

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Monday, September 25, 2006 12:02 PM
To: Voll, Scott; Wes Sisk; Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] any way to schedule reboots of phones??

 

That's a pretty hefty push for just a reset of phones.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 25, 2006 2:43 PM
To: Wes Sisk; Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

 

Can't you schedule the switches to reboot (reload at HH:MM)....... That
would take care of the issue.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Monday, September 25, 2006 11:29 AM
To: Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

 

Nothing directly.

If they are EM phones, you could schedule a forced EM logout.

schedule a .bat job to:
net stop "cisco callmanager"
net start "cisco callmanager"

schedule a windows reboot

/Wes

Tim Reimers wrote: 

hi everyone- 

I just made a global change to all phones on the system-- 

I can't restart all phones during the day--- 

Is there some sort of scheduleable reboot of all phones? (one I could
configure and not have to remember to reboot all phones tonight)

thanks, Tim 

 
 
 
 
 



________________________________



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

________________________________

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

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From Scott.Voll at wesd.org  Mon Sep 25 16:22:50 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 25 Sep 2006 13:22:50 -0700
Subject: [cisco-voip] any way to schedule reboots of phones??
Message-ID: <407055A92CECCB499C922A2D35FC19A6069349EA@Apollo.wesd.org>

Extension Mobility

 

________________________________

From: Tim Reimers [mailto:tim.reimers at asheville.k12.nc.us] 
Sent: Monday, September 25, 2006 1:17 PM
To: Voll, Scott; Matt Slaga (US); Wes Sisk
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] any way to schedule reboots of phones??

 

That's probably what I'll do...

 

Right as yet, the new softkey template isn't working for the main
building.. not sure why...

 

Wes -- what's an "EM" phone?

 

That bat file would be easy to implement...

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, September 25, 2006 4:00 PM
To: Matt Slaga (US); Wes Sisk; Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] any way to schedule reboots of phones??

Well, the easiest would be VPN in from home.... Select all phones and
restart.  But if you don't want user intervention I was just trying to
come up with a useable option.

 

Scott

 

 

 

________________________________

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Monday, September 25, 2006 12:02 PM
To: Voll, Scott; Wes Sisk; Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] any way to schedule reboots of phones??

 

That's a pretty hefty push for just a reset of phones.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 25, 2006 2:43 PM
To: Wes Sisk; Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

 

Can't you schedule the switches to reboot (reload at HH:MM)....... That
would take care of the issue.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Monday, September 25, 2006 11:29 AM
To: Tim Reimers
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

 

Nothing directly.

If they are EM phones, you could schedule a forced EM logout.

schedule a .bat job to:
net stop "cisco callmanager"
net start "cisco callmanager"

schedule a windows reboot

/Wes

Tim Reimers wrote: 

hi everyone- 

I just made a global change to all phones on the system-- 

I can't restart all phones during the day--- 

Is there some sort of scheduleable reboot of all phones? (one I could
configure and not have to remember to reboot all phones tonight)

thanks, Tim 

 
 
 
 
 
 
 



________________________________



 
 
 
 
 
 
 
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From wsisk at cisco.com  Mon Sep 25 17:32:54 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Mon, 25 Sep 2006 17:32:54 -0400
Subject: [cisco-voip] any way to schedule reboots of phones??
In-Reply-To: <050944EE48FF384F9C663F7048C47DA704E947E9@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA704E947E9@webmail.asheville.k12.nc.us>
Message-ID: <45184B06.9040709@cisco.com>

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From erickbe at yahoo.com  Mon Sep 25 21:09:33 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon, 25 Sep 2006 20:09:33 -0500 (CDT)
Subject: [cisco-voip] any way to schedule reboots of phones??
In-Reply-To: <2B3745BC9D31514381A6FCC5E72D937310A4FC@SMOMAIL.jhacorp.com>
Message-ID: <20060926010933.6179.qmail@web37205.mail.mud.yahoo.com>

How about a time-based access list to block port 2000 at night for a bit? :) 

I had a client once where there phones were rebooting at random times, which turned out to be after a certain period but event logs showed a normal reason code and it turned out to be EM auto-logout was resetting the phone. I wish the event log showed a different reason code for that type of reset situation.

----- Original Message ----
From: Bill Riley III <BRiley at jackhenry.com>
To: "Voll, Scott" <Scott.Voll at wesd.org>; J. Oquendo <sil at infiltrated.net>
Cc: cisco-voip at puck.nether.net
Sent: Monday, September 25, 2006 1:57:29 PM
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

I think that's in express.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, September 25, 2006 1:49 PM
To: J. Oquendo
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

Where do you enter that command?

Scott


-----Original Message-----
From: J. Oquendo [mailto:sil at infiltrated.net] 
Sent: Monday, September 25, 2006 11:42 AM
To: Voll, Scott
Cc: Wes Sisk; Tim Reimers; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] any way to schedule reboots of phones??

Voll, Scott wrote:
>
> Can't you schedule the switches to reboot (reload at HH:MM).......
That 
> would take care of the issue.
>
> Scott
>
>
------------------------------------------------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] *On Behalf Of *Wes Sisk
> *Sent:* Monday, September 25, 2006 11:29 AM
> *To:* Tim Reimers
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] any way to schedule reboots of phones??
>
> Nothing directly.
>
> If they are EM phones, you could schedule a forced EM logout.
>
> schedule a .bat job to:
> net stop "cisco callmanager"
> net start "cisco callmanager"
>
> schedule a windows reboot
>
> /Wes
>
> Tim Reimers wrote:
>
> hi everyone-
>
> I just made a global change to all phones on the system--
>
> I can't restart all phones during the day---
>
> Is there some sort of scheduleable reboot of all phones? (one I could 
> configure and not have to remember to reboot all phones tonight)
>
> thanks, Tim
>

telephony-service
reset sequence-all

I could be mistaken, 99.99999% of the times I am


-- 
====================================================
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 

The happiness of society is the end of government.
John Adams


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From Nick.Kassel at Charles-Stanley.co.uk  Tue Sep 26 07:09:15 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 26 Sep 2006 12:09:15 +0100
Subject: [cisco-voip] No of calls made
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A0E@CSEXCH01.charles-stanley.co.uk>

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

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From Joe.Pollere at us.didata.com  Tue Sep 26 08:26:31 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Tue, 26 Sep 2006 08:26:31 -0400
Subject: [cisco-voip] No of calls made
Message-ID: <C65033A817B5734392E55CB9DF44610406D90A36@USNAEXCH.na.didata.local>

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_note09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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

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
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You accept that any instructions are deemed to have been given at the
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You acknowledge that e-mails are not secure and you accept the risk of
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************************************************************************
************************


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

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

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

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, International Securities Markets Association, and
The London International Financial Futures &
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This footnote also confirms that this email message has been swept by
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From MLinsemier at apcapital.com  Tue Sep 26 09:54:59 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Tue, 26 Sep 2006 09:54:59 -0400
Subject: [cisco-voip] 7935 / 7936 - That key is not active here
Message-ID: <AB0A35F2486F81468971AA68558BE71904F038BB@smieln01winm01.micoanet.com>

All,

 

Apparently (I was told today) that a few weeks ago several of the 7935
and 7936 conference phones started acting flakey when our employees
attempted to conference in additional users.  They would have one person
on the phone and attempt to conference in a second and when they pushed
the conference softkey (after having them on the line) they would
receive a error saying that "That key is not active here".  They powered
cycled the phone and still got this error.  They then changed the order
in which the called people and the problem seemed to resolve itself.
Has anyone else ran into this issue?  Will they get this as a warning if
there is not enough conferencing resources available in the DSP farm?

 

-Matt


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From Nick.Kassel at Charles-Stanley.co.uk  Tue Sep 26 10:18:53 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 26 Sep 2006 15:18:53 +0100
Subject: [cisco-voip] No of calls made
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A12@CSEXCH01.charles-stanley.co.uk>

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com] 
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_note09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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

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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You accept that any instructions are deemed to have been given at the
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You acknowledge that e-mails are not secure and you accept the risk of
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************************************************************************
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, 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.

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
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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 Charles Stanley Securities are divisions of Charles Stanley & Co. Ltd

Authorised and Regulated by the Financial Services Authority, Member of the
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From akoul at dataways.gr  Tue Sep 26 10:27:36 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Tue, 26 Sep 2006 17:27:36 +0300
Subject: [cisco-voip] What dows max-conn exactly do under a pots dial-peer?
Message-ID: <451938D8.1020300@dataways.gr>

Hi all,
i have found the syntax for this command but i cannot find exactly the 
maximum number of that command is?
So, suppose you have a E1 controller with 30 voice channels available.
if you use the max-conn 20 under the dial-peer pots means that no more 
than 20 calls can be achieved by that router?

Thanks

From Hans-Peter.Walter at tds.de  Tue Sep 26 10:32:47 2006
From: Hans-Peter.Walter at tds.de (Hans-Peter Walter)
Date: Tue, 26 Sep 2006 16:32:47 +0200
Subject: [cisco-voip] Antwort: Re:  No of calls made
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451A12@CSEXCH01.charles-stanley.co.uk>
Message-ID: <OF998B9176.13A5460E-ONC12571F5.004FD9AE-C12571F5.004FF4B4@de.tds-global.com>

I use for a quick check:

http://www.onlineconversion.com/unix_time.htm
http://www.patrickkempf.de/time.php





"Nick Kassel" <Nick.Kassel at charles-stanley.co.uk> 
Gesendet von: cisco-voip-bounces at puck.nether.net
26.09.2006 16:18

An
"Joe Pollere (US)" <Joe.Pollere at us.didata.com>, 
<cisco-voip at puck.nether.net>
Kopie

Thema
Re: [cisco-voip] No of calls made






Thanks for that Joe, I am a little confused with the conversion of date 
and time though as I have no sql experience, have you done this? 
 

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com] 
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made
 
You could use a sql query to search the CDR database:
 
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_tech_note09186a00801d71c5.shtml
 
 
 
 

From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made
 
I have been asked to try and find out how many calls the Call Managers 
have made over the last 12 months, can anyone help with how to extract 
this sort of information?
 
 
 


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

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

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

Authorised and Regulated by the Financial Services Authority, Member of 
the
London Stock Exchange, 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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The information contained in this e-mail is strictly confidential, some or 
all
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Access to this e-mail by any other person is prohibited. If you are not 
the
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You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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

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

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

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

This footnote also confirms that this email message has been swept by 
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that any use or reproduction of this email or its contents is strictly 
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From Joe.Pollere at us.didata.com  Tue Sep 26 10:46:49 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Tue, 26 Sep 2006 10:46:49 -0400
Subject: [cisco-voip] Antwort: Re:  No of calls made
Message-ID: <C65033A817B5734392E55CB9DF44610406D90DBE@USNAEXCH.na.didata.local>

Nick its pretty simple. I use the attached spreadsheet to do the
conversion. 

 

________________________________

From: Hans-Peter Walter [mailto:Hans-Peter.Walter at tds.de] 
Sent: Tuesday, September 26, 2006 10:33 AM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net; cisco-voip-bounces at puck.nether.net; Joe
Pollere (US)
Subject: Antwort: Re: [cisco-voip] No of calls made

 


I use for a quick check: 

http://www.onlineconversion.com/unix_time.htm 
http://www.patrickkempf.de/time.php 





"Nick Kassel" <Nick.Kassel at charles-stanley.co.uk> 
Gesendet von: cisco-voip-bounces at puck.nether.net 

26.09.2006 16:18 

An

"Joe Pollere (US)" <Joe.Pollere at us.didata.com>,
<cisco-voip at puck.nether.net> 

Kopie

 

Thema

Re: [cisco-voip] No of calls made

 

 

 




Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 
  

 

________________________________


From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com] 
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made 
  
You could use a sql query to search the CDR database: 
  
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_note09186a00801d71c5.shtml
<http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_t
ech_note09186a00801d71c5.shtml>  
  
  
  
  

 

________________________________


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made 
  
I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information? 
  
  
  


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Charles Stanley & Co. Ltd
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Registered in England No. 1903304

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

Authorised and Regulated by the Financial Services Authority, Member of
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The London International Financial Futures &
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This footnote also confirms that this email message has been swept by
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The information contained in this e-mail is strictly confidential, some
or all
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Access to this e-mail by any other person is prohibited. If you are not
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

Authorised and Regulated by the Financial Services Authority, Member of
the
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From Nick.Kassel at Charles-Stanley.co.uk  Tue Sep 26 10:49:02 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 26 Sep 2006 15:49:02 +0100
Subject: [cisco-voip] No of calls made
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A13@CSEXCH01.charles-stanley.co.uk>

I presume I have to have the Cisco CDR Insert service activated for this
to work? I do not have it activated on both of my subscribers, if this
is the case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com] 
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_note09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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

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 Charles Stanley Securities 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
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You accept that any instructions are deemed to have been given at the
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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 Charles Stanley Securities 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.

 

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
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The information contained in this e-mail is strictly confidential, some or all
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Access to this e-mail by any other person is prohibited. If you are not the
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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 Charles Stanley Securities 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 rratliff at cisco.com  Tue Sep 26 11:16:04 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 26 Sep 2006 11:16:04 -0400
Subject: [cisco-voip] What dows max-conn exactly do under a pots
	dial-peer?
In-Reply-To: <451938D8.1020300@dataways.gr>
Message-ID: <011b01c6e17e$af7531e0$05000100@amer.cisco.com>

It specifies the max number of calls that can be made using that dial-peer
at a given time. 

I'm not sure if it counts inbound and outbound legs or treats them
separately.  I would not use it on an inbound dial-peer.
 
-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony Kouloglou
Sent: Tuesday, September 26, 2006 10:28 AM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] What dows max-conn exactly do under a pots dial-peer?

Hi all,
i have found the syntax for this command but i cannot find exactly the
maximum number of that command is?
So, suppose you have a E1 controller with 30 voice channels available.
if you use the max-conn 20 under the dial-peer pots means that no more than
20 calls can be achieved by that router?

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

From rratliff at cisco.com  Tue Sep 26 11:17:35 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 26 Sep 2006 11:17:35 -0400
Subject: [cisco-voip] No of calls made
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451A13@CSEXCH01.charles-stanley.co.uk>
Message-ID: <012201c6e17e$e59da2c0$05000100@amer.cisco.com>

CDR Insert only runs on the publisher and is responsible for inserting the
CDR records generated by the CM process on each call processing node into
the database.  Since the CDR database only runs on the publisher it wastes
resources if you enable it on a subscriber (and it will throw errors to the
app log stating so if you do run it on a sub). 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this to
work? I do not have it activated on both of my subscribers, if this is the
case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date and
time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_tech_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers have
made over the last 12 months, can anyone help with how to extract this sort
of information?

 

 

 


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

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
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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 Charles Stanley Securities 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
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You accept that any instructions are deemed to have been given at the time
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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 Charles Stanley Securities 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 Charles Stanley Securities 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.
	

________________________________

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


From Nick.Kassel at Charles-Stanley.co.uk  Tue Sep 26 11:20:41 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 26 Sep 2006 16:20:41 +0100
Subject: [cisco-voip] No of calls made
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A14@CSEXCH01.charles-stanley.co.uk>

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the
CDR records generated by the CM process on each call processing node
into
the database.  Since the CDR database only runs on the publisher it
wastes
resources if you enable it on a subscriber (and it will throw errors to
the
app log stating so if you do run it on a sub). 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to
work? I do not have it activated on both of my subscribers, if this is
the
case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and
time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have
made over the last 12 months, can anyone help with how to extract this
sort
of information?

 

 

 


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

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 Charles Stanley Securities 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
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You acknowledge that e-mails are not secure and you accept the risk of
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
of
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Authorised and Regulated by the Financial Services Authority, Member of
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The
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This footnote also confirms that this email message has been swept by
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The information contained in this e-mail is strictly confidential, some
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You acknowledge that e-mails are not secure and you accept the risk of
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********************


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

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

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

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

This footnote also confirms that this email message has been swept by
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________________________________

Disclaimer: This e-mail communication and any attachments may contain
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The information contained in this e-mail is strictly confidential, some or all
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, 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 Nick.Kassel at Charles-Stanley.co.uk  Tue Sep 26 11:33:45 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 26 Sep 2006 16:33:45 +0100
Subject: [cisco-voip] No of calls made
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A15@CSEXCH01.charles-stanley.co.uk>

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the
CDR records generated by the CM process on each call processing node
into
the database.  Since the CDR database only runs on the publisher it
wastes
resources if you enable it on a subscriber (and it will throw errors to
the
app log stating so if you do run it on a sub). 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to
work? I do not have it activated on both of my subscribers, if this is
the
case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and
time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have
made over the last 12 months, can anyone help with how to extract this
sort
of information?

 

 

 


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Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

This footnote also confirms that this email message has been swept by
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The information contained in this e-mail is strictly confidential, some
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You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
****
********************


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

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

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

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

This footnote also confirms that this email message has been swept by
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The information contained in this e-mail is strictly confidential, some
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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.
************************************************************************
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********************


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

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

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

Authorised and Regulated by the Financial Services Authority, Member of
the
London Stock Exchange, 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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________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you
are hereby notified that you have received this communication in error
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that any use or reproduction of this email or its contents is strictly
prohibited and may be unlawful. If you have received this communication
in
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deleting
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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
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Access to this e-mail by any other person is prohibited. If you are not
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You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


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

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

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

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

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cisco-voip at puck.nether.net
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The information contained in this e-mail is strictly confidential, some or all
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Access to this e-mail by any other person is prohibited. If you are not the
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You acknowledge that e-mails are not secure and you accept the risk of
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************************************************************************************************


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

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

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

Authorised and Regulated by the Financial Services Authority, Member of the
London Stock Exchange, 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 Joe.Pollere at us.didata.com  Tue Sep 26 11:45:26 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Tue, 26 Sep 2006 11:45:26 -0400
Subject: [cisco-voip] No of calls made
Message-ID: <C65033A817B5734392E55CB9DF44610406D90F85@USNAEXCH.na.didata.local>

That query is correct. I just ran it on a live publisher. Is CDR enabled
under service parameters for all servers?



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 11:34 AM
To: Nick Kassel; Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the
CDR records generated by the CM process on each call processing node
into
the database.  Since the CDR database only runs on the publisher it
wastes
resources if you enable it on a subscriber (and it will throw errors to
the
app log stating so if you do run it on a sub). 


-Ryan 
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to
work? I do not have it activated on both of my subscribers, if this is
the
case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and
time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have
made over the last 12 months, can anyone help with how to extract this
sort
of information?

 

 

 


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

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 Charles Stanley Securities 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 Charles Stanley Securities 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 Charles Stanley Securities 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.
	

________________________________

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

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

The information 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 Charles Stanley Securities 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
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From milay at cisco.com  Tue Sep 26 11:54:11 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Tue, 26 Sep 2006 08:54:11 -0700
Subject: [cisco-voip] No of calls made
Message-ID: <413FABF5DCE22849AF86F442E472B78BD20354@xmb-sjc-219.amer.cisco.com>

Are records being inserted into the DB? Are there files backing up in
the \CDR, CMR, & Bad directories? Like stated by someone else, do you
have CDR's & CMR's enabled on all nodes? I assume that the records are
being classified properly....

Mike

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:34 AM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the CDR records generated by the CM process on each call processing node
into the database.  Since the CDR database only runs on the publisher it
wastes resources if you enable it on a subscriber (and it will throw
errors to the app log stating so if you do run it on a sub). 


-Ryan
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to work? I do not have it activated on both of my subscribers, if this
is the case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
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Authorised and Regulated by the Financial Services Authority, Member of
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
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

This footnote also confirms that this email message has been swept by
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The information contained in this e-mail is strictly confidential, some
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You acknowledge that e-mails are not secure and you accept the risk of
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

This footnote also confirms that this email message has been swept by
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________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
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error and that any use or reproduction of this email or its contents is
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************************


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

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

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

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

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Registered Office: 25 Luke Street London EC2A 4AR

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

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

Authorised and Regulated by the Financial Services Authority, Member of
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From Nick.Kassel at Charles-Stanley.co.uk  Tue Sep 26 12:03:03 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 26 Sep 2006 17:03:03 +0100
Subject: [cisco-voip] No of calls made
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A16@CSEXCH01.charles-stanley.co.uk>

Thanks guys, it appears that the CDR Enabled Flag is set to False and I
have no records in any of the directories. Is this the only way I would
have been able to get this information? 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Tuesday, September 26, 2006 4:54 PM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Are records being inserted into the DB? Are there files backing up in
the \CDR, CMR, & Bad directories? Like stated by someone else, do you
have CDR's & CMR's enabled on all nodes? I assume that the records are
being classified properly....

Mike

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:34 AM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the CDR records generated by the CM process on each call processing node
into the database.  Since the CDR database only runs on the publisher it
wastes resources if you enable it on a subscriber (and it will throw
errors to the app log stating so if you do run it on a sub). 


-Ryan
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to work? I do not have it activated on both of my subscribers, if this
is the case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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

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 Charles Stanley Securities 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,
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You accept that any instructions are deemed to have been given at the
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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 Charles Stanley Securities 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
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You accept that any instructions are deemed to have been given at the
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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 Charles Stanley Securities 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.
	

________________________________

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

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

The information 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
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You accept that any instructions are deemed to have been given at the
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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 Charles Stanley Securities 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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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
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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 Charles Stanley Securities 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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https://puck.nether.net/mailman/listinfo/cisco-voip
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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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From Joe.Pollere at us.didata.com  Tue Sep 26 12:06:32 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Tue, 26 Sep 2006 12:06:32 -0400
Subject: [cisco-voip] No of calls made
Message-ID: <C65033A817B5734392E55CB9DF44610406D9103F@USNAEXCH.na.didata.local>

You might be able to look at the calls completed counter under Call
Manager performance counters. I'm not sure if or when they get reset
though. I assume its after a reboot. 

Joe

-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, September 26, 2006 12:03 PM
To: Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Cc: Joe Pollere (US)
Subject: RE: [cisco-voip] No of calls made

Thanks guys, it appears that the CDR Enabled Flag is set to False and I
have no records in any of the directories. Is this the only way I would
have been able to get this information? 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Tuesday, September 26, 2006 4:54 PM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Are records being inserted into the DB? Are there files backing up in
the \CDR, CMR, & Bad directories? Like stated by someone else, do you
have CDR's & CMR's enabled on all nodes? I assume that the records are
being classified properly....

Mike

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:34 AM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the CDR records generated by the CM process on each call processing node
into the database.  Since the CDR database only runs on the publisher it
wastes resources if you enable it on a subscriber (and it will throw
errors to the app log stating so if you do run it on a sub). 


-Ryan
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to work? I do not have it activated on both of my subscribers, if this
is the case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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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 Charles Stanley Securities are divisions
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Charles Stanley & Co. Ltd
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Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
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Authorised and Regulated by the Financial Services Authority, Member of
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Disclaimer: This e-mail communication and any attachments may contain
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You acknowledge that e-mails are not secure and you accept the risk of
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

Authorised and Regulated by the Financial Services Authority, Member of
the London Stock Exchange, International Securities Markets Association,
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Registered Office: 25 Luke Street London EC2A 4AR

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

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

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Registered in England No. 1903304

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the
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From gpate at kenttech.com  Tue Sep 26 12:11:52 2006
From: gpate at kenttech.com (Gary L. Pate)
Date: Tue, 26 Sep 2006 12:11:52 -0400
Subject: [cisco-voip] No of calls made
In-Reply-To: <C65033A817B5734392E55CB9DF44610406D9103F@USNAEXCH.na.didata.local>
Message-ID: <000f01c6e186$7d2e6e60$1201280a@gpatelt>

Yea, I'm not sure when those get reset either. In case your not sure where
to look Nick, you can bring up the Real Time Monitoring Tool (assuming its
been installed), and then go to devices and then pull up the gateway, and
make sure you looking at the PRI. There is a column at the bottom that shows
calls per gateway and the total calls.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Pollere (US)
Sent: Tuesday, September 26, 2006 12:07 PM
To: Nick Kassel; Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

You might be able to look at the calls completed counter under Call
Manager performance counters. I'm not sure if or when they get reset
though. I assume its after a reboot. 

Joe

-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, September 26, 2006 12:03 PM
To: Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Cc: Joe Pollere (US)
Subject: RE: [cisco-voip] No of calls made

Thanks guys, it appears that the CDR Enabled Flag is set to False and I
have no records in any of the directories. Is this the only way I would
have been able to get this information? 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Tuesday, September 26, 2006 4:54 PM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Are records being inserted into the DB? Are there files backing up in
the \CDR, CMR, & Bad directories? Like stated by someone else, do you
have CDR's & CMR's enabled on all nodes? I assume that the records are
being classified properly....

Mike

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:34 AM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the CDR records generated by the CM process on each call processing node
into the database.  Since the CDR database only runs on the publisher it
wastes resources if you enable it on a subscriber (and it will throw
errors to the app log stating so if you do run it on a sub). 


-Ryan
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to work? I do not have it activated on both of my subscribers, if this
is the case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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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 Charles Stanley Securities 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
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The information contained in this e-mail is strictly confidential, some
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You acknowledge that e-mails are not secure and you accept the risk of
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

This footnote also confirms that this email message has been swept by
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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
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Access to this e-mail by any other person is prohibited. If you are not
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You acknowledge that e-mails are not secure and you accept the risk of
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

This footnote also confirms that this email message has been swept by
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Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
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you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
strictly prohibited and may be unlawful. If you have received this
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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
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From Nick.Kassel at Charles-Stanley.co.uk  Tue Sep 26 12:19:45 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Tue, 26 Sep 2006 17:19:45 +0100
Subject: [cisco-voip] No of calls made
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A18@CSEXCH01.charles-stanley.co.uk>

The trouble is I am trying to find out the total amount of calls made on
the call managers within the last 12 months, we have over 20 branch
offices each with their own local voice gateway and all of these will
have been reset within this period so the RTMT will not have kept this
information after a reset.

Thanks for all your help anyway.


-----Original Message-----
From: Gary L. Pate [mailto:gpate at kenttech.com] 
Sent: Tuesday, September 26, 2006 5:12 PM
To: 'Joe Pollere (US)'; Nick Kassel; 'Mike Lay (milay)'; 'Ryan Ratliff
(rratliff)'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Yea, I'm not sure when those get reset either. In case your not sure
where
to look Nick, you can bring up the Real Time Monitoring Tool (assuming
its
been installed), and then go to devices and then pull up the gateway,
and
make sure you looking at the PRI. There is a column at the bottom that
shows
calls per gateway and the total calls.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Pollere
(US)
Sent: Tuesday, September 26, 2006 12:07 PM
To: Nick Kassel; Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

You might be able to look at the calls completed counter under Call
Manager performance counters. I'm not sure if or when they get reset
though. I assume its after a reboot. 

Joe

-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, September 26, 2006 12:03 PM
To: Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Cc: Joe Pollere (US)
Subject: RE: [cisco-voip] No of calls made

Thanks guys, it appears that the CDR Enabled Flag is set to False and I
have no records in any of the directories. Is this the only way I would
have been able to get this information? 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Tuesday, September 26, 2006 4:54 PM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Are records being inserted into the DB? Are there files backing up in
the \CDR, CMR, & Bad directories? Like stated by someone else, do you
have CDR's & CMR's enabled on all nodes? I assume that the records are
being classified properly....

Mike

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:34 AM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the CDR records generated by the CM process on each call processing node
into the database.  Since the CDR database only runs on the publisher it
wastes resources if you enable it on a subscriber (and it will throw
errors to the app log stating so if you do run it on a sub). 


-Ryan
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to work? I do not have it activated on both of my subscribers, if this
is the case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

This footnote also confirms that this email message has been swept by
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Registered Office: 25 Luke Street London EC2A 4AR

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

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

Authorised and Regulated by the Financial Services Authority, Member of
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Disclaimer: This e-mail communication and any attachments may contain
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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 Charles Stanley Securities 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,
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Access to this e-mail by any other person is prohibited. If you are not
the intended recipient, any use, disclosure, copying, printing,
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You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


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

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

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

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

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only.
Access to this e-mail by any other person is prohibited. If you are not
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of,
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You acknowledge that e-mails are not secure and you accept the risk of
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

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Disclaimer:

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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The information contained in this e-mail is strictly confidential, some or all
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Charles Stanley & Co. Ltd
Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

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From Joe.Pollere at us.didata.com  Tue Sep 26 12:30:26 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Tue, 26 Sep 2006 12:30:26 -0400
Subject: [cisco-voip] No of calls made
Message-ID: <C65033A817B5734392E55CB9DF44610406D910D6@USNAEXCH.na.didata.local>

Actually I was referring to the Calls Completed Call Manager performance
counters under administrative tools. This would give you counts for all
calls completed by a node. It's a moot point though since we don't know
when the last reset occurred.

Joe

-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, September 26, 2006 12:20 PM
To: GPate at kenttech.com; Joe Pollere (US); Mike Lay (milay); Ryan Ratliff
(rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

The trouble is I am trying to find out the total amount of calls made on
the call managers within the last 12 months, we have over 20 branch
offices each with their own local voice gateway and all of these will
have been reset within this period so the RTMT will not have kept this
information after a reset.

Thanks for all your help anyway.


-----Original Message-----
From: Gary L. Pate [mailto:gpate at kenttech.com] 
Sent: Tuesday, September 26, 2006 5:12 PM
To: 'Joe Pollere (US)'; Nick Kassel; 'Mike Lay (milay)'; 'Ryan Ratliff
(rratliff)'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Yea, I'm not sure when those get reset either. In case your not sure
where
to look Nick, you can bring up the Real Time Monitoring Tool (assuming
its
been installed), and then go to devices and then pull up the gateway,
and
make sure you looking at the PRI. There is a column at the bottom that
shows
calls per gateway and the total calls.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Pollere
(US)
Sent: Tuesday, September 26, 2006 12:07 PM
To: Nick Kassel; Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

You might be able to look at the calls completed counter under Call
Manager performance counters. I'm not sure if or when they get reset
though. I assume its after a reboot. 

Joe

-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, September 26, 2006 12:03 PM
To: Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Cc: Joe Pollere (US)
Subject: RE: [cisco-voip] No of calls made

Thanks guys, it appears that the CDR Enabled Flag is set to False and I
have no records in any of the directories. Is this the only way I would
have been able to get this information? 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Tuesday, September 26, 2006 4:54 PM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Are records being inserted into the DB? Are there files backing up in
the \CDR, CMR, & Bad directories? Like stated by someone else, do you
have CDR's & CMR's enabled on all nodes? I assume that the records are
being classified properly....

Mike

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:34 AM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the CDR records generated by the CM process on each call processing node
into the database.  Since the CDR database only runs on the publisher it
wastes resources if you enable it on a subscriber (and it will throw
errors to the app log stating so if you do run it on a sub). 


-Ryan
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to work? I do not have it activated on both of my subscribers, if this
is the case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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From Matt.Slaga at us.didata.com  Tue Sep 26 14:39:22 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 26 Sep 2006 14:39:22 -0400
Subject: [cisco-voip] No of calls made
Message-ID: <C1FE15183DA37645BC0633BC604E44F0034343C6@USNAEXCH.na.didata.local>

A potential option is to determine the time of the last system reset (in
event view), take the total count of calls in RTMT and multiply it by
the length of time since the last reset divided by 12.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Pollere
(US)
Sent: Tuesday, September 26, 2006 12:30 PM
To: Nick Kassel; GPate at kenttech.com; Mike Lay (milay); Ryan Ratliff
(rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Actually I was referring to the Calls Completed Call Manager performance
counters under administrative tools. This would give you counts for all
calls completed by a node. It's a moot point though since we don't know
when the last reset occurred.

Joe

-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, September 26, 2006 12:20 PM
To: GPate at kenttech.com; Joe Pollere (US); Mike Lay (milay); Ryan Ratliff
(rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

The trouble is I am trying to find out the total amount of calls made on
the call managers within the last 12 months, we have over 20 branch
offices each with their own local voice gateway and all of these will
have been reset within this period so the RTMT will not have kept this
information after a reset.

Thanks for all your help anyway.


-----Original Message-----
From: Gary L. Pate [mailto:gpate at kenttech.com] 
Sent: Tuesday, September 26, 2006 5:12 PM
To: 'Joe Pollere (US)'; Nick Kassel; 'Mike Lay (milay)'; 'Ryan Ratliff
(rratliff)'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Yea, I'm not sure when those get reset either. In case your not sure
where
to look Nick, you can bring up the Real Time Monitoring Tool (assuming
its
been installed), and then go to devices and then pull up the gateway,
and
make sure you looking at the PRI. There is a column at the bottom that
shows
calls per gateway and the total calls.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Pollere
(US)
Sent: Tuesday, September 26, 2006 12:07 PM
To: Nick Kassel; Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

You might be able to look at the calls completed counter under Call
Manager performance counters. I'm not sure if or when they get reset
though. I assume its after a reboot. 

Joe

-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, September 26, 2006 12:03 PM
To: Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Cc: Joe Pollere (US)
Subject: RE: [cisco-voip] No of calls made

Thanks guys, it appears that the CDR Enabled Flag is set to False and I
have no records in any of the directories. Is this the only way I would
have been able to get this information? 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Tuesday, September 26, 2006 4:54 PM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Are records being inserted into the DB? Are there files backing up in
the \CDR, CMR, & Bad directories? Like stated by someone else, do you
have CDR's & CMR's enabled on all nodes? I assume that the records are
being classified properly....

Mike

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:34 AM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the CDR records generated by the CM process on each call processing node
into the database.  Since the CDR database only runs on the publisher it
wastes resources if you enable it on a subscriber (and it will throw
errors to the app log stating so if you do run it on a sub). 


-Ryan
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to work? I do not have it activated on both of my subscribers, if this
is the case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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You acknowledge that e-mails are not secure and you accept the risk of
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Tel: 0207 739 8200 Fax: 0207 739 7798
Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
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Authorised and Regulated by the Financial Services Authority, Member of
the London Stock Exchange, International Securities Markets Association,
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Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

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Registered Office: 25 Luke Street London EC2A 4AR

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

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

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

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From ealeatherman at gmail.com  Tue Sep 26 16:21:21 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Tue, 26 Sep 2006 16:21:21 -0400
Subject: [cisco-voip] FastDials in CM5
Message-ID: <94a1afde0609261321x7cdc0f3ek1b1d79914363594a@mail.gmail.com>

Anyone familiar with how to configure a fastdials service URL button in
Callmanager 5? I found where the URL and parameters changed, but I can't
find what the fast dials URL would be. Users arent going to want to dial in
their userID, PIN, hit some more buttons, etc, to get to "Fast" dials :).

-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From jgrace at digitelusa.net  Tue Sep 26 16:59:46 2006
From: jgrace at digitelusa.net (James Grace)
Date: Tue, 26 Sep 2006 16:59:46 -0400
Subject: [cisco-voip] a router for transcoding
Message-ID: <200609261659.AA22806738@mail.digitelusa.net>

we have a central site and 4 remote sites.  we only have cm servers and unity server at the central site.  we put a router at the central site for transding overflow if needed.  my question is why would we need a router for transcoding overflow.  Someone told me that it would be for conference calls on the system.  can someone help me understand this.  our unity server is set for g729 
---



From bob at smalltime.com  Tue Sep 26 17:07:05 2006
From: bob at smalltime.com (Robert Kulagowski)
Date: Tue, 26 Sep 2006 16:07:05 -0500
Subject: [cisco-voip] Configuring pickup groups
Message-ID: <45199679.40600@smalltime.com>

We have a site that has a PagePac unit connected to a FXS port.  CM is 
4.1.3sr2.

When the receptionist steps away / end of day, they forward the main 
number to the overhead night bell.

The Night Bell is in a pickup group, as are all the phones, so that 
after hours anyone can pickup incoming calls.

The problem is that because all phones are in the same pickup group, any 
user can pickup anyone else's ringing phone, which they don't want.  The 
other issue is that once a user hears the overhead, they press Pickup 
and the incoming call starts ringing at that phone, they get CallerID, 
and then can't hit iDivert to send it back to the main number's VM box 
if they don't want the call.  Instead, they have to wait for RNA in 
order for the call to end up in the main number VM.

I investigated the other pickup options:
Group Pickup might work, but means that it's now a two-step process to 
pickup a call, but it will allow me to put the phones in one pickup 
group and have the overhead in another pickup group.  I would remove 
"Pickup" from the softkey template for the phones.

I tried to configure OPickup, thinking that I could put the phones in a 
pickup group that had a partition that wasn't in their calling search 
space, but have the overhead in an associated pickup group that _was_ in 
their CSS.  Doesn't work; CallManager won't let you configure a DN with 
a pickup group whose partition isn't in your CSS.

Any other ideas on how to implement this functionality?


From Scott.Voll at wesd.org  Tue Sep 26 17:33:59 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 26 Sep 2006 14:33:59 -0700
Subject: [cisco-voip] a router for transcoding
Message-ID: <407055A92CECCB499C922A2D35FC19A606B43CF7@Apollo.wesd.org>

Transcoding would be used if you were using IPCCx for example since it
only does G711 and your Remote sites are doing G729 to the central site.
This will allow the Transcoding from remote site to IPCCx agent.

You can also use DSP resources on a Router for Conference calling, but
they need to be configured to do that.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, September 26, 2006 2:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] a router for transcoding

we have a central site and 4 remote sites.  we only have cm servers and
unity server at the central site.  we put a router at the central site
for transding overflow if needed.  my question is why would we need a
router for transcoding overflow.  Someone told me that it would be for
conference calls on the system.  can someone help me understand this.
our unity server is set for g729 
---


_______________________________________________
cisco-voip mailing list
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From lelio at uoguelph.ca  Tue Sep 26 17:45:21 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 26 Sep 2006 17:45:21 -0400
Subject: [cisco-voip] forwarding information presented to cell phones
Message-ID: <02d401c6e1b5$10f13050$47196883@cfs.uoguelph.ca>

This has been raised before, but I'm hoping someone who has this working can work out a few details with me. I'd like to be able to display the original calling number to the end device, but right now I am seeing my outbound mask. I've got route patterns set up so that they don't use the calling party external phone number mask and I input all 10 digits of the Calling Party Transform Mask that I want. This is what I see on the cell phone.

I recall reading somewhere that if I were to change this route pattern to use the calling party external phone number mask, the correct forwarded number would be presented on the Cell phone. 

Can anyone confirm this? I have setup my gateways so that the "originator" is selected as calling party presentation (or whatever that field was).

Thanks, Lelio

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
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From Scott.Voll at wesd.org  Tue Sep 26 17:56:00 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 26 Sep 2006 14:56:00 -0700
Subject: [cisco-voip] multi calls getting messed up
Message-ID: <407055A92CECCB499C922A2D35FC19A606B43D0C@Apollo.wesd.org>

I'm running CM 4.1.3sr2 with a 7960g running Firmware 7.02.300 and my
3640 VGW has a PRI.

 

With that all being said, has anyone seen issues where if 4 or 5 calls
all come in at the same time and the receptionist starts putting
everyone on hold she can't get back to the calls to do what ever she may
need to do.

 

I allow 16 calls with a 8 call busy trigger.  I thought it might be a
Bandwidth issue so I have increased the bandwidth to that region.  Is
this a Firmware issue?

 

One happens with multiple calls all coming in at the same time.  She
doesn't have the issue when she has multiple calls on hold if they are
all spaced out.

 

TIA

 

Scott

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From lelio at uoguelph.ca  Tue Sep 26 18:09:40 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 26 Sep 2006 18:09:40 -0400
Subject: [cisco-voip] multi calls getting messed up
References: <407055A92CECCB499C922A2D35FC19A606B43D0C@Apollo.wesd.org>
Message-ID: <035601c6e1b8$76a78540$47196883@cfs.uoguelph.ca>

I think I ran into that with the early versions of 7. Try FW 8 or higher and then try to replicate the problem.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Voll, Scott 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, September 26, 2006 5:56 PM
  Subject: [cisco-voip] multi calls getting messed up


  I'm running CM 4.1.3sr2 with a 7960g running Firmware 7.02.300 and my 3640 VGW has a PRI.

   

  With that all being said, has anyone seen issues where if 4 or 5 calls all come in at the same time and the receptionist starts putting everyone on hold she can't get back to the calls to do what ever she may need to do.

   

  I allow 16 calls with a 8 call busy trigger.  I thought it might be a Bandwidth issue so I have increased the bandwidth to that region.  Is this a Firmware issue?

   

  One happens with multiple calls all coming in at the same time.  She doesn't have the issue when she has multiple calls on hold if they are all spaced out.

   

  TIA

   

  Scott



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


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From erickbe at yahoo.com  Tue Sep 26 20:18:12 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue, 26 Sep 2006 17:18:12 -0700 (PDT)
Subject: [cisco-voip] No of calls made
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0034343C6@USNAEXCH.na.didata.local>
Message-ID: <20060927001812.77983.qmail@web37212.mail.mud.yahoo.com>

The Calls completed performance counter counts every type of call also (I believe) so probably is not an accurrate measurement since it includes transfers, MWI calls, etc. It gives a general idea but you need to  factor in other activity that increments this counter. I wish they would add more counters for MWIcalls, calltransfers, etc. 

----- Original Message ----
From: Matt Slaga (US) <Matt.Slaga at us.didata.com>
To: Joe Pollere (US) <Joe.Pollere at us.didata.com>; Nick Kassel <Nick.Kassel at charles-stanley.co.uk>; GPate at kenttech.com; Mike Lay (milay) <milay at cisco.com>; Ryan Ratliff (rratliff) <rratliff at cisco.com>; cisco-voip at puck.nether.net
Sent: Tuesday, September 26, 2006 1:39:22 PM
Subject: Re: [cisco-voip] No of calls made

A potential option is to determine the time of the last system reset (in
event view), take the total count of calls in RTMT and multiply it by
the length of time since the last reset divided by 12.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Pollere
(US)
Sent: Tuesday, September 26, 2006 12:30 PM
To: Nick Kassel; GPate at kenttech.com; Mike Lay (milay); Ryan Ratliff
(rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Actually I was referring to the Calls Completed Call Manager performance
counters under administrative tools. This would give you counts for all
calls completed by a node. It's a moot point though since we don't know
when the last reset occurred.

Joe

-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, September 26, 2006 12:20 PM
To: GPate at kenttech.com; Joe Pollere (US); Mike Lay (milay); Ryan Ratliff
(rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

The trouble is I am trying to find out the total amount of calls made on
the call managers within the last 12 months, we have over 20 branch
offices each with their own local voice gateway and all of these will
have been reset within this period so the RTMT will not have kept this
information after a reset.

Thanks for all your help anyway.


-----Original Message-----
From: Gary L. Pate [mailto:gpate at kenttech.com] 
Sent: Tuesday, September 26, 2006 5:12 PM
To: 'Joe Pollere (US)'; Nick Kassel; 'Mike Lay (milay)'; 'Ryan Ratliff
(rratliff)'; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Yea, I'm not sure when those get reset either. In case your not sure
where
to look Nick, you can bring up the Real Time Monitoring Tool (assuming
its
been installed), and then go to devices and then pull up the gateway,
and
make sure you looking at the PRI. There is a column at the bottom that
shows
calls per gateway and the total calls.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joe Pollere
(US)
Sent: Tuesday, September 26, 2006 12:07 PM
To: Nick Kassel; Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

You might be able to look at the calls completed counter under Call
Manager performance counters. I'm not sure if or when they get reset
though. I assume its after a reboot. 

Joe

-----Original Message-----
From: Nick Kassel [mailto:Nick.Kassel at Charles-Stanley.co.uk] 
Sent: Tuesday, September 26, 2006 12:03 PM
To: Mike Lay (milay); Ryan Ratliff (rratliff);
cisco-voip at puck.nether.net
Cc: Joe Pollere (US)
Subject: RE: [cisco-voip] No of calls made

Thanks guys, it appears that the CDR Enabled Flag is set to False and I
have no records in any of the directories. Is this the only way I would
have been able to get this information? 

-----Original Message-----
From: Mike Lay (milay) [mailto:milay at cisco.com] 
Sent: Tuesday, September 26, 2006 4:54 PM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

Are records being inserted into the DB? Are there files backing up in
the \CDR, CMR, & Bad directories? Like stated by someone else, do you
have CDR's & CMR's enabled on all nodes? I assume that the records are
being classified properly....

Mike

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:34 AM
To: Nick Kassel; Ryan Ratliff (rratliff); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I have entered the formula 

select * from CallDetailRecord where datetimeconnect > '1127606400'

and when I click on the check mark it says The command(s) completed
successfully but I don not get any results when I execute the query just
(0 row(s) affected)

Any idea's? 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 4:21 PM
To: Ryan Ratliff; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

Ok, I do have the CDR Insert activated on the Publisher, I must be doing
something worn in the query analyser.

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, September 26, 2006 4:18 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

CDR Insert only runs on the publisher and is responsible for inserting
the CDR records generated by the CM process on each call processing node
into the database.  Since the CDR database only runs on the publisher it
wastes resources if you enable it on a subscriber (and it will throw
errors to the app log stating so if you do run it on a sub). 


-Ryan
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 10:49 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

I presume I have to have the Cisco CDR Insert service activated for this
to work? I do not have it activated on both of my subscribers, if this
is the case is there no way I can find this information out?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 3:19 PM
To: Joe Pollere (US); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] No of calls made

 

Thanks for that Joe, I am a little confused with the conversion of date
and time though as I have no sql experience, have you done this? 

 

________________________________

From: Joe Pollere (US) [mailto:Joe.Pollere at us.didata.com]
Sent: Tuesday, September 26, 2006 1:27 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] No of calls made

 

You could use a sql query to search the CDR database:

 

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_n
ote09186a00801d71c5.shtml

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Tuesday, September 26, 2006 7:09 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No of calls made

 

I have been asked to try and find out how many calls the Call Managers
have made over the last 12 months, can anyone help with how to extract
this sort of information?

 

 

 


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

The information contained in this e-mail is strictly confidential, some
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Access to this e-mail by any other person is prohibited. If you are not
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You accept that any instructions are deemed to have been given at the
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Registered in England No. 1903304

Charles Stanley Sutherlands and Charles Stanley Securities are divisions
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From jsteinberg at gmail.com  Tue Sep 26 21:31:13 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 26 Sep 2006 21:31:13 -0400
Subject: [cisco-voip] OPickup with Analog phones on VG248
Message-ID: <f1b65f880609261831y35e4ddefr3eeab2078b847408@mail.gmail.com>

Is this possible?  I am running 1.3.1 code on a VG248 and have the following
options in the feature codes:

Call Pickup **3
Group Pickup **4

I do not see OPickup as a feature code within the VG248 running 1.3.1.  I
also don't see anything mentioned in release notes for 1.3.1es8.2.  I'm
wondering if there is new VG248 code that has the OPickup feature option
that is in CCM 4.1.3.

Any ideas?

justin
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From bcarter at sentinel.com  Wed Sep 27 01:07:53 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Wed, 27 Sep 2006 00:07:53 -0500
Subject: [cisco-voip] forwarding information presented to cell phones
Message-ID: <6A0744639B86DD459433108DBF101D2601189260@MAIL01.sentinel.com>

Cisco Mobility Manager

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, September 26, 2006 4:45 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] forwarding information presented to cell phones


This has been raised before, but I'm hoping someone who has this working
can work out a few details with me. I'd like to be able to display the
original calling number to the end device, but right now I am seeing my
outbound mask. I've got route patterns set up so that they don't use the
calling party external phone number mask and I input all 10 digits of
the Calling Party Transform Mask that I want. This is what I see on the
cell phone.
 
I recall reading somewhere that if I were to change this route pattern
to use the calling party external phone number mask, the correct
forwarded number would be presented on the Cell phone. 
 
Can anyone confirm this? I have setup my gateways so that the
"originator" is selected as calling party presentation (or whatever that
field was).
 
Thanks, Lelio
 
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."


This email may contain proprietary and confidential information for the sole use of the intended recipient. 
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From lelio at uoguelph.ca  Wed Sep 27 01:11:45 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 27 Sep 2006 01:11:45 -0400
Subject: [cisco-voip] forwarding information presented to cell phones
References: <6A0744639B86DD459433108DBF101D2601189260@MAIL01.sentinel.com>
Message-ID: <004401c6e1f3$6daffb30$ca01a8c0@deadfish>

I don't need the complexity, I just want to forward a phone to a cell phone, none of that fancy dual ringing stuff.
  ----- Original Message ----- 
  From: Carter, Bill 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Wednesday, September 27, 2006 1:07 AM
  Subject: RE: [cisco-voip] forwarding information presented to cell phones


  Cisco Mobility Manager



------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Tuesday, September 26, 2006 4:45 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] forwarding information presented to cell phones


  This has been raised before, but I'm hoping someone who has this working can work out a few details with me. I'd like to be able to display the original calling number to the end device, but right now I am seeing my outbound mask. I've got route patterns set up so that they don't use the calling party external phone number mask and I input all 10 digits of the Calling Party Transform Mask that I want. This is what I see on the cell phone.

  I recall reading somewhere that if I were to change this route pattern to use the calling party external phone number mask, the correct forwarded number would be presented on the Cell phone. 

  Can anyone confirm this? I have setup my gateways so that the "originator" is selected as calling party presentation (or whatever that field was).

  Thanks, Lelio

  --------------------------------------------------------------------------------
  Lelio Fulgenzi, B.A.
  Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
  "I can eat fifty eggs." "Nobody can eat fifty eggs."

This email may contain proprietary and confidential information for the sole use of the intended recipient. 
Any review, retransmission, dissemination, or other use of this information by persons or entities other than 
the intended recipient is prohibited. If you are not the intended recipient, please contact the sender and 
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From ErasmuE4 at telkom.co.za  Wed Sep 27 02:32:51 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 27 Sep 2006 08:32:51 +0200
Subject: [cisco-voip] a router for transcoding
In-Reply-To: <407055A92CECCB499C922A2D35FC19A606B43CF7@Apollo.wesd.org>
Message-ID: <472E195E692EE74BAE4E7B2880A78C1FAC194B@CNTRRA20-XCS01.telkom.co.za>


Samples of trans-coding needed:


1. you have branches connecting to central site at G729 and want to
conference from branch to HQ -- onnet across the IP wan -- you need
trans-coding because the software conference bridge with call manager
does not support G729 -- That is if you don't configure some DSP
resources at HQ for hardware conferencing that does support G711 and
g729, in which case you won't need trans-coding.


2. Another example is when you have a unity express voicemail solution
in a branch. Supports G711 only so PSTN calls to voicemail will work
fine but as soon as you have an onnet call at G729 to the person in a
branch and it rolls to voicemail you will need trans-coding in a
branch



Erik Erasmus
Telkom SA
': +27 12 680 8320
?: +27 12 680 7388
?: +27 82 4106420
*:  erasmue4 at telkom.co.za

       

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: 26 September 2006 11:34 PM
To: jgrace at digitelusa.net; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] a router for transcoding

Transcoding would be used if you were using IPCCx for example since it
only does G711 and your Remote sites are doing G729 to the central
site.
This will allow the Transcoding from remote site to IPCCx agent.

You can also use DSP resources on a Router for Conference calling, but
they need to be configured to do that.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, September 26, 2006 2:00 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] a router for transcoding

we have a central site and 4 remote sites.  we only have cm servers
and
unity server at the central site.  we put a router at the central site
for transding overflow if needed.  my question is why would we need a
router for transcoding overflow.  Someone told me that it would be for
conference calls on the system.  can someone help me understand this.
our unity server is set for g729
---


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This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
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From loris.vicinanza at shearman.com  Wed Sep 27 04:06:29 2006
From: loris.vicinanza at shearman.com (Loris Vicinanza)
Date: Wed, 27 Sep 2006 10:06:29 +0200
Subject: [cisco-voip] IP Communicator
Message-ID: <OFE5FB751C.C75C547D-ONC12571F6.002BCD7A-C12571F6.002C8BCB@shearman.com>

Hi, just wondering if somebody of you ever experienced a dump every time
you try to dial a number off an IP Communicator (ver 2.0.1.0)

Here's the error it generates in the event viewer:

 The bugcheck was: 0x0000008e (0xc0000005, 0xf71dff07, 0xeb1be4d4,
0x00000000

Thanks,

Loris



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From balbler at nts.at  Wed Sep 27 05:30:57 2006
From: balbler at nts.at (Bernhard Albler)
Date: Wed, 27 Sep 2006 11:30:57 +0200
Subject: [cisco-voip] Strange Messages in Phone Debug Log
Message-ID: <5A8C69748176434F92F1EECD7D21F72C78B8D2@srvgraz03.nts.at>

Hi,

A customer reported some strange issues with his phone (does not always
dial etc.). I looked at the phone (7960, FW 7.2(4)) and I saw the
following messages:
Datei nicht gefunden: Austria/7960-tones.xml  
  2006-06-18 09:30:58.038 Code:8106, P:0, S:12300800 
  Datei nicht gefunden: Austria/7960-tones.xml  
  2006-06-18 09:51:53.668 Code:8106, P:0, S:12300800 
  Datei nicht gefunden: Austria/7960-tones.xml  
  2006-06-18 10:10:51.578 Code:8106, P:0, S:12300800 
  2006-06-18 10:10:59.928 Code:30f, P:0, S:12300800 
  Config Task Call Stack: 
  0009082e 
  00056044 
  00055ae8 
  000565a2 
  00053d24 
  0008146c 
  0008146c 
  00081566 
  000815a6 
  0005283c 
  0008161a 
  00084482 
  0006e5e6 
  00065f7c 
  000533a2 
  3187252172 Code:0, P:0, S:12300000 
  NET Task Call Stack: 
  00416332 
  004163d2 
  00412d16 
  0040358e 
  00403a8a 
  00411496 
  Datei nicht gefunden: Austria/7960-tones.xml  
  2006-08-19 18:45:26.974 Code:8103, P:0, S:12300800 
  2006-08-19 18:50:42.264 Code:8103, P:0, S:12300806


Has anyone an idea, what the message (Code 8103 etc.) mean? Is this
something one should be concerened about or are these only debug
messages without relevance?

Many thanks
bernhard


From ealeatherman at gmail.com  Wed Sep 27 08:38:36 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 27 Sep 2006 08:38:36 -0400
Subject: [cisco-voip] FastDials in CM5
In-Reply-To: <94a1afde0609261321x7cdc0f3ek1b1d79914363594a@mail.gmail.com>
References: <94a1afde0609261321x7cdc0f3ek1b1d79914363594a@mail.gmail.com>
Message-ID: <94a1afde0609270538q1f350ff6ja266eb5078331888@mail.gmail.com>

Oops, i should clarify, i found the URL for personal directory/fast dials
menu - but not the URL that goes directly to Fast Dials.. if I get time i'll
hook up a sniffer and see if I can find it that way.

On 9/26/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> Anyone familiar with how to configure a fastdials service URL button in
> Callmanager 5? I found where the URL and parameters changed, but I can't
> find what the fast dials URL would be. Users arent going to want to dial in
> their userID, PIN, hit some more buttons, etc, to get to "Fast" dials :).
>
> --
> Ed Leatherman
> Senior Voice Engineer
> West Virginia University
> Telecommunications and Network Operations




-- 
Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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From acm at netuse.de  Wed Sep 27 09:05:09 2006
From: acm at netuse.de (Ahmad Cheikh Moussa)
Date: Wed, 27 Sep 2006 15:05:09 +0200
Subject: [cisco-voip] MGCP
Message-ID: <451A7705.6010300@netuse.de>

Hi!

CAN I use MGCP with a VIC2-2BRI-NT/TE instead of VWIC-2MFT-E1 ?
I need this, because I want to test a configuration in my lab .
Somneone told me that this module is not supported with VIC2-2BRI-NT/TE.

Is that correct ?

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 PEKOU at gr.ibm.com  Wed Sep 27 09:15:23 2006
From: PEKOU at gr.ibm.com (Evi Pekou)
Date: Wed, 27 Sep 2006 16:15:23 +0300
Subject: [cisco-voip] Start charging after IVR message
Message-ID: <OFC0D13F9A.13C5AF1F-ONC22571F6.00474253-C22571F6.0048A55D@gr.ibm.com>

Hi,

We have an implementation of CM 4.1 with co-resident IPCC Express.
There are no agents, the IPCC is only used for playing a welcome message 
depending on the number called by the customer. 
The requirment is for the customer not to be charged for the welcome 
message.  A message should be sent to the provider immediately after IVR 
message for the charging to start.
Any ideas how to implement this?

Regards,
Evi.
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From Nick.Kassel at Charles-Stanley.co.uk  Wed Sep 27 09:48:39 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed, 27 Sep 2006 14:48:39 +0100
Subject: [cisco-voip] CDR Analysis & Reporting
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A1B@CSEXCH01.charles-stanley.co.uk>

I have been testing the CDR Analysis & Reporting in our lab, is it
possible to create custom reports on items such as device pool or groups
of users and then combine this with reports such as the gateway reports.
I did notice on one report it had a department field but I was unable to
choose a department.

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

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From jsteinberg at gmail.com  Wed Sep 27 10:34:45 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 27 Sep 2006 10:34:45 -0400
Subject: [cisco-voip] Configuring pickup groups
In-Reply-To: <45199679.40600@smalltime.com>
References: <45199679.40600@smalltime.com>
Message-ID: <f1b65f880609270734i6c1d5d4dj22e6ed61bb59c634@mail.gmail.com>

You can do this with oPickup.

Configure an pickup group "Everyone", add all phones to this group.
Configure an pickup group "Overhead night bell", add the DN of the night
bell to this group.

Then add the park number for the "Overhead night bell" pickup group as a
current pickup group member of the "everyone" pickup group.

Fwd the receptionist phone to a DN that is a member of "overhead night bell"
pickup group.

Any phone in the "everyone" pickup group can dial opickup and answer the
night bell.

You could put all phones, night bell dn, and pickup groups in the same
partition and that would be fine.

Justin

On 9/26/06, Robert Kulagowski <bob at smalltime.com> wrote:
>
> We have a site that has a PagePac unit connected to a FXS port.  CM is
> 4.1.3sr2.
>
> When the receptionist steps away / end of day, they forward the main
> number to the overhead night bell.
>
> The Night Bell is in a pickup group, as are all the phones, so that
> after hours anyone can pickup incoming calls.
>
> The problem is that because all phones are in the same pickup group, any
> user can pickup anyone else's ringing phone, which they don't want.  The
> other issue is that once a user hears the overhead, they press Pickup
> and the incoming call starts ringing at that phone, they get CallerID,
> and then can't hit iDivert to send it back to the main number's VM box
> if they don't want the call.  Instead, they have to wait for RNA in
> order for the call to end up in the main number VM.
>
> I investigated the other pickup options:
> Group Pickup might work, but means that it's now a two-step process to
> pickup a call, but it will allow me to put the phones in one pickup
> group and have the overhead in another pickup group.  I would remove
> "Pickup" from the softkey template for the phones.
>
> I tried to configure OPickup, thinking that I could put the phones in a
> pickup group that had a partition that wasn't in their calling search
> space, but have the overhead in an associated pickup group that _was_ in
> their CSS.  Doesn't work; CallManager won't let you configure a DN with
> a pickup group whose partition isn't in your CSS.
>
> Any other ideas on how to implement this functionality?
>
> _______________________________________________
> 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  Wed Sep 27 10:39:28 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 27 Sep 2006 10:39:28 -0400
Subject: [cisco-voip] CDR Analysis & Reporting
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC01451A1B@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC01451A1B@CSEXCH01.charles-stanley.co.uk>
Message-ID: <451A8D20.3070204@cisco.com>

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From wsisk at cisco.com  Wed Sep 27 10:44:00 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 27 Sep 2006 10:44:00 -0400
Subject: [cisco-voip] Start charging after IVR message
In-Reply-To: <OFC0D13F9A.13C5AF1F-ONC22571F6.00474253-C22571F6.0048A55D@gr.ibm.com>
References: <OFC0D13F9A.13C5AF1F-ONC22571F6.00474253-C22571F6.0048A55D@gr.ibm.com>
Message-ID: <451A8E30.8070600@cisco.com>

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From wsisk at cisco.com  Wed Sep 27 10:46:52 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 27 Sep 2006 10:46:52 -0400
Subject: [cisco-voip] Strange Messages in Phone Debug Log
In-Reply-To: <5A8C69748176434F92F1EECD7D21F72C78B8D2@srvgraz03.nts.at>
References: <5A8C69748176434F92F1EECD7D21F72C78B8D2@srvgraz03.nts.at>
Message-ID: <451A8EDC.9010502@cisco.com>

I believe 8103 and 8106 are normal phone resets and no so concerning.  
The 30f with corresponding Config Task Call Stack represents a crash in 
the phone software and that is the stack trace.  This would cause the 
phone to reboot to recover.

/Wes

Bernhard Albler wrote:
> Hi,
>
> A customer reported some strange issues with his phone (does not always
> dial etc.). I looked at the phone (7960, FW 7.2(4)) and I saw the
> following messages:
> Datei nicht gefunden: Austria/7960-tones.xml  
>   2006-06-18 09:30:58.038 Code:8106, P:0, S:12300800 
>   Datei nicht gefunden: Austria/7960-tones.xml  
>   2006-06-18 09:51:53.668 Code:8106, P:0, S:12300800 
>   Datei nicht gefunden: Austria/7960-tones.xml  
>   2006-06-18 10:10:51.578 Code:8106, P:0, S:12300800 
>   2006-06-18 10:10:59.928 Code:30f, P:0, S:12300800 
>   Config Task Call Stack: 
>   0009082e 
>   00056044 
>   00055ae8 
>   000565a2 
>   00053d24 
>   0008146c 
>   0008146c 
>   00081566 
>   000815a6 
>   0005283c 
>   0008161a 
>   00084482 
>   0006e5e6 
>   00065f7c 
>   000533a2 
>   3187252172 Code:0, P:0, S:12300000 
>   NET Task Call Stack: 
>   00416332 
>   004163d2 
>   00412d16 
>   0040358e 
>   00403a8a 
>   00411496 
>   Datei nicht gefunden: Austria/7960-tones.xml  
>   2006-08-19 18:45:26.974 Code:8103, P:0, S:12300800 
>   2006-08-19 18:50:42.264 Code:8103, P:0, S:12300806
>
>
> Has anyone an idea, what the message (Code 8103 etc.) mean? Is this
> something one should be concerened about or are these only debug
> messages without relevance?
>
> Many thanks
> bernhard
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   

From wsisk at cisco.com  Wed Sep 27 10:48:17 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 27 Sep 2006 10:48:17 -0400
Subject: [cisco-voip] OPickup with Analog phones on VG248
In-Reply-To: <f1b65f880609261831y35e4ddefr3eeab2078b847408@mail.gmail.com>
References: <f1b65f880609261831y35e4ddefr3eeab2078b847408@mail.gmail.com>
Message-ID: <451A8F31.7080503@cisco.com>

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From Nick.Kassel at Charles-Stanley.co.uk  Wed Sep 27 11:00:24 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Wed, 27 Sep 2006 16:00:24 +0100
Subject: [cisco-voip] CDR Analysis & Reporting
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A1C@CSEXCH01.charles-stanley.co.uk>

Wes

 

How do I do the Department reports, when I search users it only gives me
First and Last name. The only place I can see Department listed is in
under User Reports for Bills but there is no option to select any
departments.

 

Nick

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, September 27, 2006 3:39 PM
To: Nick Kassel
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CDR Analysis & Reporting

 

Nick,

populate the department information for all your users in the global
directory and you can use department reports.  There is no correlation
to device pools and no real customization of reports.

/Wes

Nick Kassel wrote: 

I have been testing the CDR Analysis & Reporting in our lab, is it
possible to create custom reports on items such as device pool or groups
of users and then combine this with reports such as the gateway reports.
I did notice on one report it had a department field but I was unable to
choose a department.

 


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


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

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

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

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

This footnote also confirms that this email message has been swept by
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________________________________



 
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip
  
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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.
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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You accept that any instructions are deemed to have been given at the time the
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************************************************************************************************


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

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

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

Authorised and Regulated by the Financial Services Authority, Member of the
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From tomlemay at adelphia.net  Wed Sep 27 11:06:49 2006
From: tomlemay at adelphia.net (tomlemay at adelphia.net)
Date: Wed, 27 Sep 2006 11:06:49 -0400
Subject: [cisco-voip] VTA and MS Office  Communicator
Message-ID: <5366152.1159369609595.JavaMail.root@web32>

Has anyone gotten VTA web Camera to work with MS Office Communicator IM?  I think that VTA steps on MS Office Communicator since VTA runs in the system tray.  Do we need to uninstall VTA for MS Office Communicator to work?

Thanks, Tom

From acm at netuse.de  Wed Sep 27 11:08:57 2006
From: acm at netuse.de (Ahmad Cheikh-Moussa)
Date: Wed, 27 Sep 2006 17:08:57 +0200
Subject: [cisco-voip] MGCP
In-Reply-To: <451A7705.6010300@netuse.de>
References: <451A7705.6010300@netuse.de>
Message-ID: <20060927150857.GA13399@raysrv3.netuse.de>

Hi!

On Sep 27, 06, Ahmad Cheikh Moussa wrote:
> Hi!
> 
> CAN I use MGCP with a VIC2-2BRI-NT/TE instead of VWIC-2MFT-E1 ?
> I need this, because I want to test a configuration in my lab .
> Somneone told me that this module is not supported with VIC2-2BRI-NT/TE.
Here is a write error. I mean .. not supported with  MGCP.
Can someone confirm this ?

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 GRhodes at rbh.com  Wed Sep 27 11:28:58 2006
From: GRhodes at rbh.com (Rhodes, Geoff)
Date: Wed, 27 Sep 2006 11:28:58 -0400
Subject: [cisco-voip] Windows updates on Unity server
Message-ID: <6350516B6159444E9D646523EB4526F4037B19ED@apollo.rbh.local>

I know this has been gone over before, but I forgot to keep those
emails.  Can we apply Microsoft security patches to our Unity server as
we do other "normal" servers?  I know with our Call Manager servers we
wait until Cisco releases their version of the patches, but I want to
make sure Unity is cool with the normal MS patches.  We use WSUS to
deploy the MS patches.

Thanks in advance.

Geoff Rhodes
Director, Information Technology
Robinson, Bradshaw & Hinson, P.A.
704 / 377-8188
grhodes at rbh.com
 


From Scott.Voll at wesd.org  Wed Sep 27 11:39:19 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 27 Sep 2006 08:39:19 -0700
Subject: [cisco-voip] Windows updates on Unity server
Message-ID: <407055A92CECCB499C922A2D35FC19A606B43DC7@Apollo.wesd.org>

MS updates are supported on Unity servers, but Cisco likes to test
Service Packs before installing.  Usually within 60 days.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Rhodes, Geoff
Sent: Wednesday, September 27, 2006 8:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Windows updates on Unity server

I know this has been gone over before, but I forgot to keep those
emails.  Can we apply Microsoft security patches to our Unity server as
we do other "normal" servers?  I know with our Call Manager servers we
wait until Cisco releases their version of the patches, but I want to
make sure Unity is cool with the normal MS patches.  We use WSUS to
deploy the MS patches.

Thanks in advance.

Geoff Rhodes
Director, Information Technology
Robinson, Bradshaw & Hinson, P.A.
704 / 377-8188
grhodes at rbh.com
 

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


From bob at smalltime.com  Wed Sep 27 11:45:33 2006
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 27 Sep 2006 10:45:33 -0500
Subject: [cisco-voip] Configuring pickup groups
In-Reply-To: <f1b65f880609270734i6c1d5d4dj22e6ed61bb59c634@mail.gmail.com>
References: <45199679.40600@smalltime.com>
	<f1b65f880609270734i6c1d5d4dj22e6ed61bb59c634@mail.gmail.com>
Message-ID: <451A9C9D.1010301@smalltime.com>

Justin Steinberg wrote:
> You can do this with oPickup.
> 
> Configure an pickup group "Everyone", add all phones to this group.
> Configure an pickup group "Overhead night bell", add the DN of the night 
> bell to this group.
> 
> Then add the park number for the "Overhead night bell" pickup group as a 
> current pickup group member of the "everyone" pickup group.
> 
> Fwd the receptionist phone to a DN that is a member of "overhead night 
> bell" pickup group.
> 
> Any phone in the "everyone" pickup group can dial opickup and answer the 
> night bell.
> 
> You could put all phones, night bell dn, and pickup groups in the same 
> partition and that would be fine.

Right; I tried this, but the issue is that any phone can still pickup 
any other ringing phone (because they're all in the same pickup group), 
which is not what I want.

From MLinsemier at apcapital.com  Wed Sep 27 12:03:01 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Wed, 27 Sep 2006 12:03:01 -0400
Subject: [cisco-voip] 1 RxLost (7960) or 1 Rcvr Discarded (7970) when
	placing calls
Message-ID: <AB0A35F2486F81468971AA68558BE71904F03D07@smieln01winm01.micoanet.com>

All,

 

A while back I had noticed that any time I place a phone call out on a
7960 or 7970 there seems to always be 1 RxLost packet or Rcvr Discarded
that seems to be lost.  It happens immediately when you lift the handset
to place a call and never happens again.  We are running phone load
8.0(4) but I know it has been around for a since at least 8.0(1) (when
we noticed it).  Does anyone know if this is a actual problem or a
cosmetic issue? 

 

Matt


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From techguy at gmail.com  Wed Sep 27 12:03:12 2006
From: techguy at gmail.com (Tech Guy)
Date: Wed, 27 Sep 2006 11:03:12 -0500
Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?
Message-ID: <def9a76c0609270903q56fb4cc4s2cf48f9c5b5cfdcf@mail.gmail.com>

Currently running 3.3(5) on an MCS7835H1 and MCS7825H1, each have 1GB of
RAM.

Upgrading to 4.1(3) and the min. requirements are 1GB of RAM.  Just
wondering if there are any known issues with running on 1GB, if I decide to
upgrade the RAM should we really see any benefit?  Just wondering what is
the most affected on CallManagers with regards to RAM?

Trivial I know, but curious/concerned.
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From lelio at uoguelph.ca  Wed Sep 27 12:15:24 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 27 Sep 2006 12:15:24 -0400
Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?
References: <def9a76c0609270903q56fb4cc4s2cf48f9c5b5cfdcf@mail.gmail.com>
Message-ID: <028e01c6e250$235e4700$47196883@cfs.uoguelph.ca>

I think you have to watch out for the TFTP server since it keeps all the configs in memory. If you have lots of phones you should up it to 2GB. I can't remember what the threshold is, but we have 7500 and were told to move up to 2gb.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Tech Guy 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, September 27, 2006 12:03 PM
  Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?


  Currently running 3.3(5) on an MCS7835H1 and MCS7825H1, each have 1GB of RAM.

  Upgrading to 4.1(3) and the min. requirements are 1GB of RAM.  Just wondering if there are any known issues with running on 1GB, if I decide to upgrade the RAM should we really see any benefit?  Just wondering what is the most affected on CallManagers with regards to RAM? 

  Trivial I know, but curious/concerned.


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


  _______________________________________________
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  cisco-voip at puck.nether.net
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From Scott.Voll at wesd.org  Wed Sep 27 12:16:24 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 27 Sep 2006 09:16:24 -0700
Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?
Message-ID: <407055A92CECCB499C922A2D35FC19A606B43DF0@Apollo.wesd.org>

My understanding is that Cisco changed the requirement for 4.1 to 2
gig's of ram.

 

We upgraded from 3.3 to 4.1 before cisco required 2 gig's of ram and we
only have 30mb available on the Pub and 60mb available on the Sub(small
deploy ~300 phones).

 

You will definitely want 1.5gig's min and 2gig's would be best.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tech Guy
Sent: Wednesday, September 27, 2006 9:03 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?

 

Currently running 3.3(5) on an MCS7835H1 and MCS7825H1, each have 1GB of
RAM.

 

Upgrading to 4.1(3) and the min. requirements are 1GB of RAM.  Just
wondering if there are any known issues with running on 1GB, if I decide
to upgrade the RAM should we really see any benefit?  Just wondering
what is the most affected on CallManagers with regards to RAM? 

 

Trivial I know, but curious/concerned.

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From Matt.Slaga at us.didata.com  Wed Sep 27 12:16:55 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 27 Sep 2006 12:16:55 -0400
Subject: [cisco-voip] VTA and MS Office  Communicator
Message-ID: <C1FE15183DA37645BC0633BC604E44F00346D285@USNAEXCH.na.didata.local>

Are you trying to get the VTA camera to work with Office Communicator
web cast?  If so, this will not work.  The VTA camera is dedicated to
VTA and not usable by other applications. 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
tomlemay at adelphia.net
Sent: Wednesday, September 27, 2006 11:07 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VTA and MS Office Communicator

Has anyone gotten VTA web Camera to work with MS Office Communicator IM?
I think that VTA steps on MS Office Communicator since VTA runs in the
system tray.  Do we need to uninstall VTA for MS Office Communicator to
work?

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

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

This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
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unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.



From milay at cisco.com  Wed Sep 27 12:18:58 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Wed, 27 Sep 2006 09:18:58 -0700
Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?
Message-ID: <413FABF5DCE22849AF86F442E472B78BD207BF@xmb-sjc-219.amer.cisco.com>

1g would be the minimum requirement for CallManager. The overall
performance of the server will suffer if all RAM is consumed. All
services on the server will be affect, from dial tone to writing trace.
 
This states 4.0 but really should be 4.x:
 
http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_bulletin0900ae
cd80284099.html
 
Mike

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tech Guy
Sent: Wednesday, September 27, 2006 11:03 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?


Currently running 3.3(5) on an MCS7835H1 and MCS7825H1, each have 1GB of
RAM.
 
Upgrading to 4.1(3) and the min. requirements are 1GB of RAM.  Just
wondering if there are any known issues with running on 1GB, if I decide
to upgrade the RAM should we really see any benefit?  Just wondering
what is the most affected on CallManagers with regards to RAM? 
 
Trivial I know, but curious/concerned.
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From JosephAN.CTR at mcbbutler.usmc.mil  Wed Sep 27 12:34:55 2006
From: JosephAN.CTR at mcbbutler.usmc.mil (Joseph CTR Alan N)
Date: Thu, 28 Sep 2006 01:34:55 +0900
Subject: [cisco-voip] Alternate site to download Win OS Upgrades? CCO
	Acce	ss Molasses?
Message-ID: <E50B4D2263D1BA42BCC28FFC48D460C7037C5789@mcbb02e.mcbbutler.usmc.mil>


Maybe it's just me, though it seems like access to CCO and the Cisco website
in general has come to a crawl in recent days.

Does anyone have / know of an alternate site to pull down the OS Upgrade
files? Specifically win-OS-Upgrade-K9.2000-4-3a.exe or
win-OS-Upgrade-K9.2000-2-7.exe?

I'm trying to pull down win-OS-Upgrade-K9.2000-2-7.exe from CCO and am only
getting 8K/sec on my 30Mb cable connection. All the speedtests from
California say I'm getting well over a T1 worth of throughput.

Wait ... I'm up to a whopping 17K/sec now. Only 6 1/2 hours to go.

Thanks,

Joe

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From db7td at gmx.de  Wed Sep 27 12:35:24 2006
From: db7td at gmx.de (Dietmar)
Date: Wed, 27 Sep 2006 18:35:24 +0200
Subject: [cisco-voip] forwarding information presented to cell phones
In-Reply-To: <02d401c6e1b5$10f13050$47196883@cfs.uoguelph.ca>
References: <02d401c6e1b5$10f13050$47196883@cfs.uoguelph.ca>
Message-ID: <200609271835.24711.db7td@gmx.de>

Configure the external phone number mask (e.g. XXX) for every phone separately 
in the phone's DN configuration. Remove all transformations/maskings from 
your route patterns and check "use external phone number mask".


Dietmar


> This has been raised before, but I'm hoping someone who has this working
> can work out a few details with me. I'd like to be able to display the
> original calling number to the end device, but right now I am seeing my
> outbound mask. I've got route patterns set up so that they don't use the
> calling party external phone number mask and I input all 10 digits of the
> Calling Party Transform Mask that I want. This is what I see on the cell
> phone.
>
> I recall reading somewhere that if I were to change this route pattern to
> use the calling party external phone number mask, the correct forwarded
> number would be presented on the Cell phone.
>
> Can anyone confirm this? I have setup my gateways so that the "originator"
> is selected as calling party presentation (or whatever that field was).
>
> Thanks, Lelio
>
> ---------------------------------------------------------------------------
>----- Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "I can eat fifty eggs." "Nobody can eat fifty eggs."

-- 
E-Mail: db7td at gmx.de

From crl at rti.org  Wed Sep 27 12:39:06 2006
From: crl at rti.org (Lindahl, Carl)
Date: Wed, 27 Sep 2006 12:39:06 -0400
Subject: [cisco-voip] Alternate site to download Win OS Upgrades?
	CCOAcce	ss Molasses?
Message-ID: <F8DE0271F906D547B77294843E22B559A46C4B@rtpwexc10.RCC_NT.RTI.ORG>

Your not the only one. I have noticed a tremendous slow down and inability to access certain IOS for routers.


 -----Original Message-----
From: 	Joseph CTR Alan N [mailto:JosephAN.CTR at mcbbutler.usmc.mil]
Sent:	Wed Sep 27 12:35:18 2006
To:	'cisco-voip at puck.nether.net '
Subject:	[cisco-voip] Alternate site to download Win OS Upgrades? CCOAcce	ss Molasses?


Maybe it's just me, though it seems like access to CCO and the Cisco website
in general has come to a crawl in recent days.

Does anyone have / know of an alternate site to pull down the OS Upgrade
files? Specifically win-OS-Upgrade-K9.2000-4-3a.exe or
win-OS-Upgrade-K9.2000-2-7.exe?

I'm trying to pull down win-OS-Upgrade-K9.2000-2-7.exe from CCO and am only
getting 8K/sec on my 30Mb cable connection. All the speedtests from
California say I'm getting well over a T1 worth of throughput.

Wait ... I'm up to a whopping 17K/sec now. Only 6 1/2 hours to go.

Thanks,

Joe



From Scott.Voll at wesd.org  Wed Sep 27 12:53:49 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 27 Sep 2006 09:53:49 -0700
Subject: [cisco-voip] Alternate site to download Win OS Upgrades?
	CCOAcce	ss Molasses?
Message-ID: <407055A92CECCB499C922A2D35FC19A606B43E24@Apollo.wesd.org>

Sorry to here you're having those problems.  I can download the 2.7.exe
at 2.2mb / sec.

I'm not sure it's Cisco that's the problem.

Scott


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joseph CTR Alan
N
Sent: Wednesday, September 27, 2006 9:35 AM
To: 'cisco-voip at puck.nether.net '
Subject: [cisco-voip] Alternate site to download Win OS Upgrades?
CCOAcce ss Molasses?


Maybe it's just me, though it seems like access to CCO and the Cisco
website
in general has come to a crawl in recent days.

Does anyone have / know of an alternate site to pull down the OS Upgrade
files? Specifically win-OS-Upgrade-K9.2000-4-3a.exe or
win-OS-Upgrade-K9.2000-2-7.exe?

I'm trying to pull down win-OS-Upgrade-K9.2000-2-7.exe from CCO and am
only
getting 8K/sec on my 30Mb cable connection. All the speedtests from
California say I'm getting well over a T1 worth of throughput.

Wait ... I'm up to a whopping 17K/sec now. Only 6 1/2 hours to go.

Thanks,

Joe



From wsisk at cisco.com  Wed Sep 27 13:14:34 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 27 Sep 2006 13:14:34 -0400
Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?
In-Reply-To: <def9a76c0609270903q56fb4cc4s2cf48f9c5b5cfdcf@mail.gmail.com>
References: <def9a76c0609270903q56fb4cc4s2cf48f9c5b5cfdcf@mail.gmail.com>
Message-ID: <451AB17A.8050701@cisco.com>

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From jburton at NETechCorp.com  Wed Sep 27 13:26:14 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Wed, 27 Sep 2006 13:26:14 -0400
Subject: [cisco-voip] PoE Question
Message-ID: <961200C9-6E8E-4127-9138-9E60EA619898@mimectl>

Does anyone have a link on Cisco's web site that will explain the power budgeting for PoE on their switches.  Basically I'm looking for proof that it's either possible or not to have phones plugged into every port on a switch and not cause any problems with power.  I've heard both that you can't use all the ports and that you can, so I need the definitive answer.


thanks again!
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From bcarter at sentinel.com  Wed Sep 27 13:31:27 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Wed, 27 Sep 2006 12:31:27 -0500
Subject: [cisco-voip] Less Route Patterns for Toll Free
Message-ID: <6A0744639B86DD459433108DBF101D26011892AD@MAIL01.sentinel.com>

 
I am looking for a way in CallManager to reduce the 5 route patterns for
toll free.  Right no I use:
9.1[800]XXXXXXX
9.1[855]XXXXXXX
9.1[866]XXXXXXX
9.1[877]XXXXXXX
9.1[888]XXXXXXX

Any suggestions?


This email may contain proprietary and confidential information for the sole use of the intended recipient. 
Any review, retransmission, dissemination, or other use of this information by persons or entities other than 
the intended recipient is prohibited. If you are not the intended recipient, please contact the sender and 
delete all copies. To the extent that opinions are expressed in this message, they are not necessarily the 
opinions of Sentinel Technologies or any of its affiliates, employees, directors, officers or shareholders.


From jburton at NETechCorp.com  Wed Sep 27 13:37:19 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Wed, 27 Sep 2006 13:37:19 -0400
Subject: [cisco-voip] Less Route Patterns for Toll Free
In-Reply-To: <6A0744639B86DD459433108DBF101D26011892AD@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26011892AD@MAIL01.sentinel.com>
Message-ID: <6AD0D9D5-E3FC-4638-B867-8219DFD2D8D9@mimectl>

You can use a route filter that looks for area codes 800,855,866,877,888 and then have a route pattern 8.@ that references that route filter



From: Carter, Bill
Sent: Wed 9/27/2006 1:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Less Route Patterns for Toll Free


 
I am looking for a way in CallManager to reduce the 5 route patterns for
toll free.  Right no I use:
9.1[800]XXXXXXX
9.1[855]XXXXXXX
9.1[866]XXXXXXX
9.1[877]XXXXXXX
9.1[888]XXXXXXX

Any suggestions?


This email may contain proprietary and confidential information for the sole use of the intended recipient. 
Any review, retransmission, dissemination, or other use of this information by persons or entities other than 
the intended recipient is prohibited. If you are not the intended recipient, please contact the sender and 
delete all copies. To the extent that opinions are expressed in this message, they are not necessarily the 
opinions of Sentinel Technologies or any of its affiliates, employees, directors, officers or shareholders.

_______________________________________________
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 Sep 27 13:38:13 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 27 Sep 2006 10:38:13 -0700
Subject: [cisco-voip] PoE Question
Message-ID: <407055A92CECCB499C922A2D35FC19A606B43E5B@Apollo.wesd.org>

Well it depends.

 

My understanding is you can fully populate a 24 port PoE switch with
anything.

 

Or a 48 port with the older phones using the 6.3 watts.

 

But if you are using the newer phones which use twice the watts you can
only plug in about half.

 

Doesn't make much sense to me that cisco would do that..... but that's
what my Partner / reseller is telling me.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Wednesday, September 27, 2006 10:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PoE Question

 

Does anyone have a link on Cisco's web site that will explain the power
budgeting for PoE on their switches.  Basically I'm looking for proof
that it's either possible or not to have phones plugged into every port
on a switch and not cause any problems with power.  I've heard both that
you can't use all the ports and that you can, so I need the definitive
answer.

 

 

thanks again!

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From lelio at uoguelph.ca  Wed Sep 27 13:41:34 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 27 Sep 2006 13:41:34 -0400
Subject: [cisco-voip] PoE Question
References: <961200C9-6E8E-4127-9138-9E60EA619898@mimectl>
Message-ID: <02e801c6e25c$2d4282c0$47196883@cfs.uoguelph.ca>

I'm pretty sure this is specific to the switch you are using. For example, I think the one-piece switches have enough juice to power a phone on each port using PoE (cisco inline power) but the chasis switches like the 4000 require external power supplies if you want to use a lot of phones. And boy do they generate heat!

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Jason Burton 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, September 27, 2006 1:26 PM
  Subject: [cisco-voip] PoE Question


  Does anyone have a link on Cisco's web site that will explain the power budgeting for PoE on their switches.  Basically I'm looking for proof that it's either possible or not to have phones plugged into every port on a switch and not cause any problems with power.  I've heard both that you can't use all the ports and that you can, so I need the definitive answer.


  thanks again!


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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Wed Sep 27 13:42:29 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 27 Sep 2006 13:42:29 -0400
Subject: [cisco-voip] Less Route Patterns for Toll Free
References: <6A0744639B86DD459433108DBF101D26011892AD@MAIL01.sentinel.com>
Message-ID: <02fe01c6e25c$585f0a50$47196883@cfs.uoguelph.ca>

I use 9.@ and a route filter. ;)

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Carter, Bill 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, September 27, 2006 1:31 PM
  Subject: [cisco-voip] Less Route Patterns for Toll Free



  I am looking for a way in CallManager to reduce the 5 route patterns for
  toll free.  Right no I use:
  9.1[800]XXXXXXX
  9.1[855]XXXXXXX
  9.1[866]XXXXXXX
  9.1[877]XXXXXXX
  9.1[888]XXXXXXX

  Any suggestions?


  This email may contain proprietary and confidential information for the sole use of the intended recipient. 
  Any review, retransmission, dissemination, or other use of this information by persons or entities other than 
  the intended recipient is prohibited. If you are not the intended recipient, please contact the sender and 
  delete all copies. To the extent that opinions are expressed in this message, they are not necessarily the 
  opinions of Sentinel Technologies or any of its affiliates, employees, directors, officers or shareholders.

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Matt.Slaga at us.didata.com  Wed Sep 27 13:49:19 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 27 Sep 2006 13:49:19 -0400
Subject: [cisco-voip] PoE Question
Message-ID: <C1FE15183DA37645BC0633BC604E44F00346D49F@USNAEXCH.na.didata.local>

It is only the 7970s ad 71s that are the power hogs.  The 41/61s with
gigabit Ethernet that use slightly more power, but not the 15.4 watts
that the executive handsets use.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, September 27, 2006 1:38 PM
To: Jason Burton; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PoE Question

 

Well it depends.

 

My understanding is you can fully populate a 24 port PoE switch with
anything.

 

Or a 48 port with the older phones using the 6.3 watts.

 

But if you are using the newer phones which use twice the watts you can
only plug in about half.

 

Doesn't make much sense to me that cisco would do that..... but that's
what my Partner / reseller is telling me.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Wednesday, September 27, 2006 10:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PoE Question

 

Does anyone have a link on Cisco's web site that will explain the power
budgeting for PoE on their switches.  Basically I'm looking for proof
that it's either possible or not to have phones plugged into every port
on a switch and not cause any problems with power.  I've heard both that
you can't use all the ports and that you can, so I need the definitive
answer.

 

 

thanks again!




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

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
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From Scott.Voll at wesd.org  Wed Sep 27 13:52:59 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 27 Sep 2006 10:52:59 -0700
Subject: [cisco-voip] PoE Question
Message-ID: <407055A92CECCB499C922A2D35FC19A606B43E71@Apollo.wesd.org>

I thought the 7911's were power hungry too.

 

Scott

 

________________________________

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Wednesday, September 27, 2006 10:49 AM
To: Voll, Scott; Jason Burton; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] PoE Question

 

It is only the 7970s ad 71s that are the power hogs.  The 41/61s with
gigabit Ethernet that use slightly more power, but not the 15.4 watts
that the executive handsets use.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, September 27, 2006 1:38 PM
To: Jason Burton; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PoE Question

 

Well it depends.

 

My understanding is you can fully populate a 24 port PoE switch with
anything.

 

Or a 48 port with the older phones using the 6.3 watts.

 

But if you are using the newer phones which use twice the watts you can
only plug in about half.

 

Doesn't make much sense to me that cisco would do that..... but that's
what my Partner / reseller is telling me.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Wednesday, September 27, 2006 10:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PoE Question

 

Does anyone have a link on Cisco's web site that will explain the power
budgeting for PoE on their switches.  Basically I'm looking for proof
that it's either possible or not to have phones plugged into every port
on a switch and not cause any problems with power.  I've heard both that
you can't use all the ports and that you can, so I need the definitive
answer.

 

 

thanks again!

________________________________

Disclaimer: This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the designated
addressee(s) named above only. If you are not the intended addressee,
you are hereby notified that you have received this communication in
error and that any use or reproduction of this email or its contents is
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From wsisk at cisco.com  Wed Sep 27 14:08:13 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 27 Sep 2006 14:08:13 -0400
Subject: [cisco-voip] PoE Question
In-Reply-To: <961200C9-6E8E-4127-9138-9E60EA619898@mimectl>
References: <961200C9-6E8E-4127-9138-9E60EA619898@mimectl>
Message-ID: <451ABE0D.7000202@cisco.com>

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From jburton at NETechCorp.com  Wed Sep 27 14:10:37 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Wed, 27 Sep 2006 14:10:37 -0400
Subject: [cisco-voip] PoE Question
In-Reply-To: <451ABE0D.7000202@cisco.com>
References: <961200C9-6E8E-4127-9138-9E60EA619898@mimectl>,
	<451ABE0D.7000202@cisco.com>
Message-ID: <2F8F38A3-35A8-456D-9503-64F83C770D04@mimectl>

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From pwalenta at wi.rr.com  Wed Sep 27 14:13:45 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Wed, 27 Sep 2006 13:13:45 -0500
Subject: [cisco-voip] Alternate site to download Win OS
	Upgrades?	CCOAccess Molasses?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A606B43E24@Apollo.wesd.org>
Message-ID: <000801c6e260$ac9f59e0$0232a8c0@pwnhome.com>

There were some peering problems in San Jose this past weekend and I'm sure
some issues still linger (several Tier 1/Tier 2 providers decided to
restructure some networks).  So depending on how you're getting there, you
could still see slow access problems. 

-----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, September 27, 2006 11:54 AM
To: Joseph CTR Alan N; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Alternate site to download Win OS Upgrades?
CCOAccess Molasses?

Sorry to here you're having those problems.  I can download the 2.7.exe at
2.2mb / sec.

I'm not sure it's Cisco that's the problem.

Scott


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joseph CTR Alan N
Sent: Wednesday, September 27, 2006 9:35 AM
To: 'cisco-voip at puck.nether.net '
Subject: [cisco-voip] Alternate site to download Win OS Upgrades?
CCOAcce ss Molasses?


Maybe it's just me, though it seems like access to CCO and the Cisco website
in general has come to a crawl in recent days.

Does anyone have / know of an alternate site to pull down the OS Upgrade
files? Specifically win-OS-Upgrade-K9.2000-4-3a.exe or
win-OS-Upgrade-K9.2000-2-7.exe?

I'm trying to pull down win-OS-Upgrade-K9.2000-2-7.exe from CCO and am only
getting 8K/sec on my 30Mb cable connection. All the speedtests from
California say I'm getting well over a T1 worth of throughput.

Wait ... I'm up to a whopping 17K/sec now. Only 6 1/2 hours to go.

Thanks,

Joe


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


From bob at smalltime.com  Wed Sep 27 14:13:48 2006
From: bob at smalltime.com (Robert Kulagowski)
Date: Wed, 27 Sep 2006 13:13:48 -0500
Subject: [cisco-voip] PoE Question
In-Reply-To: <961200C9-6E8E-4127-9138-9E60EA619898@mimectl>
References: <961200C9-6E8E-4127-9138-9E60EA619898@mimectl>
Message-ID: <451ABF5C.1070604@smalltime.com>

Jason Burton wrote:
> Does anyone have a link on Cisco's web site that will explain the power 
> budgeting for PoE on their switches.  Basically I'm looking for proof 
> that it's either possible or not to have phones plugged into every port 
> on a switch and not cause any problems with power.  I've heard both that 
> you can't use all the ports and that you can, so I need the definitive 
> answer.

You can use all the ports on a 48 port PoE switch (we have 3750's) with 
7970's if you limit the power.

interface FastEthernet1/0/1
  power inline consumption 6700

Sets it to a max of 6.7watts, which on the 7970 means that the display 
is half brightness.  Our users never saw full brightness on their 
phones, and we limit the power on 24 port switches as well (so that 
there aren't complaints of "his phone is brighter than mine!").

We're running;
# sh ver
Cisco IOS Software, C3750 Software (C3750-IPBASEK9-M), Version 
12.2(25)SED, RELEASE SOFTWARE (fc1)
Copyright (c) 1986-2005 by Cisco Systems, Inc.
Compiled Sat 22-Oct-05 12:43 by yenanh
Image text-base: 0x00003000, data-base: 0x00F43C70

I think this was the first release that supported the power inline 
consumption command.

From mihi at ludens.elte.hu  Wed Sep 27 14:54:35 2006
From: mihi at ludens.elte.hu (MIHI)
Date: Wed, 27 Sep 2006 20:54:35 +0200
Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?
In-Reply-To: <def9a76c0609270903q56fb4cc4s2cf48f9c5b5cfdcf@mail.gmail.com>
Message-ID: <0J69001KNLQN7A60@invitel.hu>

Also note that the latest MCS OS also requires 2 GB memory.

 

"This Cisco Unified Communications operating system requires a minimum of 2
GB of memory on the server. If the installation process detects less than 2
GB memory on the server, the installation aborts."

http://cco.cisco.com/univercd/cc/td/doc/product/voice/iptel_os/install/os200
043.htm#wp491062

 

Marton

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tech Guy
Sent: Wednesday, September 27, 2006 6:03 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager 4.1(3) Memory/RAM ?

 

Currently running 3.3(5) on an MCS7835H1 and MCS7825H1, each have 1GB of
RAM.

 

Upgrading to 4.1(3) and the min. requirements are 1GB of RAM.  Just
wondering if there are any known issues with running on 1GB, if I decide to
upgrade the RAM should we really see any benefit?  Just wondering what is
the most affected on CallManagers with regards to RAM? 

 

Trivial I know, but curious/concerned.

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From jozjan at cnl.tuke.sk  Wed Sep 27 16:05:23 2006
From: jozjan at cnl.tuke.sk (Jozef Janitor)
Date: Wed, 27 Sep 2006 22:05:23 +0200
Subject: [cisco-voip] Cisco IP Communicator and Realtek High Definition Audio
Message-ID: <006301c6e270$450667a0$cf1336e0$@tuke.sk>

Hi,

 

Has anyone of you tried the Cisco IP Communicator setup with the Realtek
High Definition Audio?

I've got a weird result: the IP Communicator didn't recognize my sound card
L

 



 

Any ideas how to solve this problem? I am using the 2.0.1.01 version of IP
Communicator on the new Lenovo 3000 V100 notebook.

 

Thank you.

 

Best regards,

   Jozef Janitor

   Computer Networks Laboratory

   www.cnl.tuke.sk

 

 

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From tim.reimers at asheville.k12.nc.us  Wed Sep 27 16:54:13 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 27 Sep 2006 16:54:13 -0400
Subject: [cisco-voip] question regarding 'prime line'
Message-ID: <050944EE48FF384F9C663F7048C47DA704E94CDD@webmail.asheville.k12.nc.us>


Question on answering lines with wireless headsets:

Here's my configuration:

We have a number of secretaries with their own personal line as #1
They also have the main school number as their second/third button.
They want to use wireless headsets and be able to be away from their
desk and answer the main school line.
My understanding is that they can only answer line#1 with most
headsets-- since the headset does not have a way to select a line
remotely to answer.

Therefore--
We've considered reprogramming their phones to  have the main # as the
first button, and have their personal line as second or third.
That way, the wireless headset will answer the main line correctly:

Upon inquiring of our vendor what the ramifications would be of making
the main# the first button, the answer was:
"When they hit Messages, they'll get the Vmbox for the system, not their
own mailbox"
because we've set the option below to "true"

"Always Use Prime Line for Voice Message : This parameter specifies
valid value as True or False. If the flag is set to True, when the
message button is pressed, primary line gets chosen and becomes the
active line. If the flag is set to False, phone automatically select a
line with voice mail as the active line based on the line status. 
This is a required field.
Default: false."  (Ours is "true")

If I change that,  and someone has two lines on their phone with two
different Vmboxes (ie, the school main# with a CallHandler set up for
RNA on that line, their own personal line, and then shared lines for the
principal, the AP, etc etc)
How will the Messages button function?
What will happen when there are -no- active lines? Would that button
take you to the CH for the main school line on button #1 ??
Would it take you to the next available 'on hook' line and the related
mailbox?
or would it be random -- such that they might find that the Messages
button one time gets their VM, another time gets them the Principals vm
on line #3, or AP on #4, etc?

Mainly, I guess I'm looking to find out if the approach of 'main line as
button#1' to fix the headset issue is the right way to go, and if that
IS the right direction, 
what does "based on the line status" mean with respect to the option
above?

Does this make sense to folks? I can re-explain if it's not clear...

thanks, Tim



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From KWhitson at cmbtec.com  Wed Sep 27 17:16:21 2006
From: KWhitson at cmbtec.com (Karl Whitson)
Date: Wed, 27 Sep 2006 17:16:21 -0400
Subject: [cisco-voip] question regarding 'prime line'
In-Reply-To: <050944EE48FF384F9C663F7048C47DA704E94CDD@webmail.asheville.k12.nc.us>
Message-ID: <8A4D67E299AA254CAB2B5985EEDD86581577CA@cmb-bm-sbs.CMB-BM.Local>

Tim,

 

You may be over engineering the solution.  I use a wireless headset (GN
9120) with a handset lifter and do remote answer all day long on all
three lines.  The key is to connect the wireless headset up to the
handset port (inline with the handset) and not use the headset port on
the phone.  Then when any of the lines are ringing, when you press the
remote answer button it will lift the receiver and they can begin to
talk, no matter which line is ringing.  They will even get a beep in the
ear with the GN 9120 and 9350 (not sure of other models).  Then you will
not have to adjust this setting and risk having a negative effect on
other users.  I find that having the personal line as line one is a good
rule of thumb and a good design. 

 

Karl Whitson

CMB Technologies

40 Cape Rd.

Milford, MA  01757

DID:  (508) 422-8413

Cell:  (603) 930-4189

Fax:  (508) 634-8678

Karl.Whitson at cmbtec.com

www.cmbtec.com

TO OPEN A SUPPORT CASE EMAIL SUPPORT at CMBTEC.COM
<mailto:SUPPORT at CMBTEC.COM>  OR CALL (508) 422-8429

  

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, September 27, 2006 4:54 PM
To: cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: [cisco-voip] question regarding 'prime line'

 

 

Question on answering lines with wireless headsets: 

Here's my configuration: 

We have a number of secretaries with their own personal line as #1 
They also have the main school number as their second/third button. 
They want to use wireless headsets and be able to be away from their
desk and answer the main school line. 
My understanding is that they can only answer line#1 with most
headsets-- since the headset does not have a way to select a line
remotely to answer.

Therefore-- 
We've considered reprogramming their phones to  have the main # as the
first button, and have their personal line as second or third.

That way, the wireless headset will answer the main line correctly: 

Upon inquiring of our vendor what the ramifications would be of making
the main# the first button, the answer was: 
"When they hit Messages, they'll get the Vmbox for the system, not their
own mailbox" 
because we've set the option below to "true" 

"Always Use Prime Line for Voice Message : This parameter specifies
valid value as True or False. If the flag is set to True, when the
message button is pressed, primary line gets chosen and becomes the
active line. If the flag is set to False, phone automatically select a
line with voice mail as the active line based on the line status.
This is a required field.
Default: false."  (Ours is "true") 

If I change that,  and someone has two lines on their phone with two
different Vmboxes (ie, the school main# with a CallHandler set up for
RNA on that line, their own personal line, and then shared lines for the
principal, the AP, etc etc)

How will the Messages button function? 
What will happen when there are -no- active lines? Would that button
take you to the CH for the main school line on button #1 ??

Would it take you to the next available 'on hook' line and the related
mailbox? 
or would it be random -- such that they might find that the Messages
button one time gets their VM, another time gets them the Principals vm
on line #3, or AP on #4, etc?

Mainly, I guess I'm looking to find out if the approach of 'main line as
button#1' to fix the headset issue is the right way to go, and if that
IS the right direction, 

what does "based on the line status" mean with respect to the option
above? 

Does this make sense to folks? I can re-explain if it's not clear... 

thanks, Tim 

 

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From keith.klevenski at rig.net  Wed Sep 27 17:27:35 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Wed, 27 Sep 2006 16:27:35 -0500
Subject: [cisco-voip] question regarding 'prime line'
Message-ID: <7F0CE0A1A0961E498A4132953534226662E1DE@HOUEXH02.rignet.org>

Tim,

 

My impression of setting Always Use Prime Line for Voice Message to true
means that whenever the messages button is pressed it will check the vm
for line 1 no matter if there are other lines on the phone with other vm
boxes configured.  It will simply always check the vm for line 1 when
pressed no matter what.  To check vm for other lines you'd have to go
off hook on the other line and then press the messages or dial the vm
pilot number of course.  I am pretty confident about this as we have
this in place now in one of our clusters.

 

We had this issue recently where the person wanted to press the messages
button and have it go to line 1's vm as this was on a 7960 with a 7914
expansion module.  Every time they pressed the messages button it would
go to what they thought was a seemingly random line which was actually a
line that had a new vm waiting.

 

If this setting is left at the default of False then if the line that
has new vm waiting is in use then I would assume it would go to the next
line that is not in use that has a new vm.  I would also assume if there
were no lines with new vm then line 1 would be used.  This would be what
is meant by 'based on line status' is whether the line is in use or not.
If in use it will just go to the next line with new vm that is not in
use.

 

Again, these are assumptions that seem to make sense as I haven't sat
down and tested all of this.  Hope this helps.

 

Keith 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, September 27, 2006 3:54 PM
To: cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: [cisco-voip] question regarding 'prime line'

 

 

Question on answering lines with wireless headsets: 

Here's my configuration: 

We have a number of secretaries with their own personal line as #1 
They also have the main school number as their second/third button. 
They want to use wireless headsets and be able to be away from their
desk and answer the main school line. 
My understanding is that they can only answer line#1 with most
headsets-- since the headset does not have a way to select a line
remotely to answer.

Therefore-- 
We've considered reprogramming their phones to  have the main # as the
first button, and have their personal line as second or third.

That way, the wireless headset will answer the main line correctly: 

Upon inquiring of our vendor what the ramifications would be of making
the main# the first button, the answer was: 
"When they hit Messages, they'll get the Vmbox for the system, not their
own mailbox" 
because we've set the option below to "true" 

"Always Use Prime Line for Voice Message : This parameter specifies
valid value as True or False. If the flag is set to True, when the
message button is pressed, primary line gets chosen and becomes the
active line. If the flag is set to False, phone automatically select a
line with voice mail as the active line based on the line status.
This is a required field.
Default: false."  (Ours is "true") 

If I change that,  and someone has two lines on their phone with two
different Vmboxes (ie, the school main# with a CallHandler set up for
RNA on that line, their own personal line, and then shared lines for the
principal, the AP, etc etc)

How will the Messages button function? 
What will happen when there are -no- active lines? Would that button
take you to the CH for the main school line on button #1 ??

Would it take you to the next available 'on hook' line and the related
mailbox? 
or would it be random -- such that they might find that the Messages
button one time gets their VM, another time gets them the Principals vm
on line #3, or AP on #4, etc?

Mainly, I guess I'm looking to find out if the approach of 'main line as
button#1' to fix the headset issue is the right way to go, and if that
IS the right direction, 

what does "based on the line status" mean with respect to the option
above? 

Does this make sense to folks? I can re-explain if it's not clear... 

thanks, Tim 

 

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From SCASPER at mandtbank.com  Wed Sep 27 19:55:31 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Wed, 27 Sep 2006 19:55:31 -0400
Subject: [cisco-voip]  CDR IP Address Conversion
Message-ID: <s51ad740.029@comrcm01intgwia.mandtbank.com>


Does anyone have a tool or formula they could share that can be used to
convert the signed decimal value in the CDR database to an IP address?

Thanks!
Steve




Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026


From lelio at uoguelph.ca  Wed Sep 27 20:00:02 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 27 Sep 2006 20:00:02 -0400
Subject: [cisco-voip] CDR IP Address Conversion
References: <s51ad740.029@comrcm01intgwia.mandtbank.com>
Message-ID: <003d01c6e291$0e4c8930$ca01a8c0@deadfish>

found this using google...

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_troubleshooting_guide_chapter09186a0080382d82.html#wp1003527


  ----- Original Message ----- 
  From: STEVEN CASPER 
  To: cisco-voip at puck.nether.net 
  Sent: Wednesday, September 27, 2006 7:55 PM
  Subject: [cisco-voip] CDR IP Address Conversion



  Does anyone have a tool or formula they could share that can be used to
  convert the signed decimal value in the CDR database to an IP address?

  Thanks!
  Steve




  Steve Casper
  Voice Technologies
  M&T Bank
  (410) 347-6026

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From cisco at adman.net  Wed Sep 27 20:32:49 2006
From: cisco at adman.net (Adam)
Date: Wed, 27 Sep 2006 19:32:49 -0500
Subject: [cisco-voip] Licensing Question
Message-ID: <c6f32ce40609271732g2ed3280em9eb176118f020239@mail.gmail.com>

I have a couple of dozen phones on a CCME system that I would like to
migrate to a full CallManager system. I know that my CCME feature
license on the router entitles me to a SRST license of the same user
count, but I can't find anywhere if the CCME ULs for the phone
automatically translate to a CCM UL. They are the same price so it
would make sense, but I can't find it in writing anywhere. Anyone know
if this is the case and where it is documented?

Thanks,
Adam

From tim.reimers at asheville.k12.nc.us  Wed Sep 27 21:05:26 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 27 Sep 2006 21:05:26 -0400
Subject: [cisco-voip] question regarding 'prime line'
References: <8A4D67E299AA254CAB2B5985EEDD86581577CA@cmb-bm-sbs.CMB-BM.Local>
Message-ID: <050944EE48FF384F9C663F7048C47DA7030EC557@webmail.asheville.k12.nc.us>

we'll try that--
 
thanks!

________________________________

From: Karl Whitson [mailto:KWhitson at cmbtec.com]
Sent: Wed 9/27/2006 5:16 PM
To: Tim Reimers; cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: RE: [cisco-voip] question regarding 'prime line'



Tim,

 

You may be over engineering the solution.  I use a wireless headset (GN 9120) with a handset lifter and do remote answer all day long on all three lines.  The key is to connect the wireless headset up to the handset port (inline with the handset) and not use the headset port on the phone.  Then when any of the lines are ringing, when you press the remote answer button it will lift the receiver and they can begin to talk, no matter which line is ringing.  They will even get a beep in the ear with the GN 9120 and 9350 (not sure of other models).  Then you will not have to adjust this setting and risk having a negative effect on other users.  I find that having the personal line as line one is a good rule of thumb and a good design. 

 

Karl Whitson

CMB Technologies

40 Cape Rd.

Milford, MA  01757

DID:  (508) 422-8413

Cell:  (603) 930-4189

Fax:  (508) 634-8678

Karl.Whitson at cmbtec.com

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

TO OPEN A SUPPORT CASE EMAIL SUPPORT at CMBTEC.COM <mailto:SUPPORT at CMBTEC.COM>  OR CALL (508) 422-8429

  <https://webmail2.asheville.k12.nc.us/exchange/tim.reimers/Drafts/RE:%20%5Bcisco-voip%5D%20question%20regarding%20%27prime%20line%27.EML/1_multipart/image001.jpg> 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, September 27, 2006 4:54 PM
To: cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: [cisco-voip] question regarding 'prime line'

 

 

Question on answering lines with wireless headsets: 

Here's my configuration: 

We have a number of secretaries with their own personal line as #1 
They also have the main school number as their second/third button. 
They want to use wireless headsets and be able to be away from their desk and answer the main school line. 
My understanding is that they can only answer line#1 with most headsets-- since the headset does not have a way to select a line remotely to answer.

Therefore-- 
We've considered reprogramming their phones to  have the main # as the first button, and have their personal line as second or third.

That way, the wireless headset will answer the main line correctly: 

Upon inquiring of our vendor what the ramifications would be of making the main# the first button, the answer was: 
"When they hit Messages, they'll get the Vmbox for the system, not their own mailbox" 
because we've set the option below to "true" 

"Always Use Prime Line for Voice Message : This parameter specifies valid value as True or False. If the flag is set to True, when the message button is pressed, primary line gets chosen and becomes the active line. If the flag is set to False, phone automatically select a line with voice mail as the active line based on the line status.
This is a required field.
Default: false."  (Ours is "true") 

If I change that,  and someone has two lines on their phone with two different Vmboxes (ie, the school main# with a CallHandler set up for RNA on that line, their own personal line, and then shared lines for the principal, the AP, etc etc)

How will the Messages button function? 
What will happen when there are -no- active lines? Would that button take you to the CH for the main school line on button #1 ??

Would it take you to the next available 'on hook' line and the related mailbox? 
or would it be random -- such that they might find that the Messages button one time gets their VM, another time gets them the Principals vm on line #3, or AP on #4, etc?

Mainly, I guess I'm looking to find out if the approach of 'main line as button#1' to fix the headset issue is the right way to go, and if that IS the right direction, 

what does "based on the line status" mean with respect to the option above? 

Does this make sense to folks? I can re-explain if it's not clear... 

thanks, Tim 

 

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From tim.reimers at asheville.k12.nc.us  Wed Sep 27 22:01:00 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 27 Sep 2006 22:01:00 -0400
Subject: [cisco-voip] question regarding 'prime line'
References: <7F0CE0A1A0961E498A4132953534226662E1DE@HOUEXH02.rignet.org>
Message-ID: <050944EE48FF384F9C663F7048C47DA7030EC558@webmail.asheville.k12.nc.us>

Thanks!
 
I think I'll probably follow the idea mentioned before, if the type of headset that's been ordered will do it-
but this does give a good understanding of what 'status' might be..
Didn't think of "with a waiting VM"...
 
T

________________________________

From: Keith Klevenski [mailto:keith.klevenski at rig.net]
Sent: Wed 9/27/2006 5:27 PM
To: Tim Reimers; cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: RE: [cisco-voip] question regarding 'prime line'



Tim,

 

My impression of setting Always Use Prime Line for Voice Message to true means that whenever the messages button is pressed it will check the vm for line 1 no matter if there are other lines on the phone with other vm boxes configured.  It will simply always check the vm for line 1 when pressed no matter what.  To check vm for other lines you'd have to go off hook on the other line and then press the messages or dial the vm pilot number of course.  I am pretty confident about this as we have this in place now in one of our clusters.

 

We had this issue recently where the person wanted to press the messages button and have it go to line 1's vm as this was on a 7960 with a 7914 expansion module.  Every time they pressed the messages button it would go to what they thought was a seemingly random line which was actually a line that had a new vm waiting.

 

If this setting is left at the default of False then if the line that has new vm waiting is in use then I would assume it would go to the next line that is not in use that has a new vm.  I would also assume if there were no lines with new vm then line 1 would be used.  This would be what is meant by 'based on line status' is whether the line is in use or not.  If in use it will just go to the next line with new vm that is not in use.

 

Again, these are assumptions that seem to make sense as I haven't sat down and tested all of this.  Hope this helps.

 

Keith 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Reimers
Sent: Wednesday, September 27, 2006 3:54 PM
To: cisco-voip at puck.nether.net
Cc: Alan Surrette
Subject: [cisco-voip] question regarding 'prime line'

 

 

Question on answering lines with wireless headsets: 

Here's my configuration: 

We have a number of secretaries with their own personal line as #1 
They also have the main school number as their second/third button. 
They want to use wireless headsets and be able to be away from their desk and answer the main school line. 
My understanding is that they can only answer line#1 with most headsets-- since the headset does not have a way to select a line remotely to answer.

Therefore-- 
We've considered reprogramming their phones to  have the main # as the first button, and have their personal line as second or third.

That way, the wireless headset will answer the main line correctly: 

Upon inquiring of our vendor what the ramifications would be of making the main# the first button, the answer was: 
"When they hit Messages, they'll get the Vmbox for the system, not their own mailbox" 
because we've set the option below to "true" 

"Always Use Prime Line for Voice Message : This parameter specifies valid value as True or False. If the flag is set to True, when the message button is pressed, primary line gets chosen and becomes the active line. If the flag is set to False, phone automatically select a line with voice mail as the active line based on the line status.
This is a required field.
Default: false."  (Ours is "true") 

If I change that,  and someone has two lines on their phone with two different Vmboxes (ie, the school main# with a CallHandler set up for RNA on that line, their own personal line, and then shared lines for the principal, the AP, etc etc)

How will the Messages button function? 
What will happen when there are -no- active lines? Would that button take you to the CH for the main school line on button #1 ??

Would it take you to the next available 'on hook' line and the related mailbox? 
or would it be random -- such that they might find that the Messages button one time gets their VM, another time gets them the Principals vm on line #3, or AP on #4, etc?

Mainly, I guess I'm looking to find out if the approach of 'main line as button#1' to fix the headset issue is the right way to go, and if that IS the right direction, 

what does "based on the line status" mean with respect to the option above? 

Does this make sense to folks? I can re-explain if it's not clear... 

thanks, Tim 

 

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From db7td at gmx.de  Thu Sep 28 01:47:56 2006
From: db7td at gmx.de (db7td at gmx.de)
Date: Thu, 28 Sep 2006 07:47:56 +0200
Subject: [cisco-voip] Forward no answer external
Message-ID: <20060928054756.143590@gmx.net>

Hi,

I have a line with "forward no answer external" pointing to internal DN, but the forwarding never occurs. When setting "forward no answer internal" everything works fine.

How does Callmanager know whether a incoming call is internal or external? Is this the onnet/offnet setting for the gateway?


Dietmar

From Andrius-conf at elsis.lt  Thu Sep 28 02:40:37 2006
From: Andrius-conf at elsis.lt (Andrius Kislas)
Date: Thu, 28 Sep 2006 09:40:37 +0300
Subject: [cisco-voip] CDR IP Address Conversion
In-Reply-To: <s51ad740.029@comrcm01intgwia.mandtbank.com>
References: <s51ad740.029@comrcm01intgwia.mandtbank.com>
Message-ID: <451B6E65.4090101@elsis.lt>

perl version:

sub fromDecToIP {
    my $dec = $_[0];
    $dec = unpack('I', pack('i', $dec));

    if (!defined($dec) || ($dec==0) || $dec eq '') {
        return '';
    }

    if ($dec == 0){
        return '';
    }
    my $hex = sprintf("%X", $dec);
    if (length($hex) != 8) {
        $hex = "0$hex";
    }

    #print "hex=$hex\n";

    my $D = substr($hex, 0, 2);
    my $C = substr($hex, 2, 2);
    my $B = substr($hex, 4, 2);
    my $A = substr($hex, 6, 2);

    #print "$A $B $C $D\n";
    #print sprintf("%d", hex($A)) . "." . sprintf("%d", hex($B)) . "." .
sprintf("%d", hex($C)) . "." . sprintf("%d", hex($D));
    return sprintf("%d", hex($A)) . "." . sprintf("%d", hex($B)) . "." .
sprintf("%d", hex($C)) . "." . sprintf("%d", hex($D));
}



STEVEN CASPER wrote:
> Does anyone have a tool or formula they could share that can be used to
> convert the signed decimal value in the CDR database to an IP address?
> 
> Thanks!
> Steve
> 
> 
> 
> 
> Steve Casper
> Voice Technologies
> M&T Bank
> (410) 347-6026
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From Nick.Kassel at Charles-Stanley.co.uk  Thu Sep 28 02:55:47 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 28 Sep 2006 07:55:47 +0100
Subject: [cisco-voip] Forward no answer external
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC01451A22@CSEXCH01.charles-stanley.co.uk>

You need to make sure that you have set the No Answer Ring Duration to
the amount of seconds you require.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of db7td at gmx.de
Sent: Thursday, September 28, 2006 6:48 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Forward no answer external

Hi,

I have a line with "forward no answer external" pointing to internal DN,
but the forwarding never occurs. When setting "forward no answer
internal" everything works fine.

How does Callmanager know whether a incoming call is internal or
external? Is this the onnet/offnet setting for the gateway?


Dietmar
_______________________________________________
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 Sep 28 08:41:21 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 28 Sep 2006 08:41:21 -0400
Subject: [cisco-voip] Forward no answer external
References: <20060928054756.143590@gmx.net>
Message-ID: <001201c6e2fb$66d01760$47196883@cfs.uoguelph.ca>

Are you using Unity as your autoattendant? All Unity calls appear as internal.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: db7td at gmx.de 
  To: cisco-voip at puck.nether.net 
  Sent: Thursday, September 28, 2006 1:47 AM
  Subject: [cisco-voip] Forward no answer external


  Hi,

  I have a line with "forward no answer external" pointing to internal DN, but the forwarding never occurs. When setting "forward no answer internal" everything works fine.

  How does Callmanager know whether a incoming call is internal or external? Is this the onnet/offnet setting for the gateway?


  Dietmar
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From CORTIZ at broward.org  Thu Sep 28 09:34:51 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Thu, 28 Sep 2006 09:34:51 -0400
Subject: [cisco-voip] Less Route Patterns for Toll Free
In-Reply-To: <6A0744639B86DD459433108DBF101D26011892AD@MAIL01.sentinel.com>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A4761B@MREXVS01.bc.broward.cty>

We use 9.18[0678][0678]XXXXXXX.  This allows a few patterns that are not
toll free but not much concern on our side as our LD access codes are
invoked from the carrier.

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Wednesday, September 27, 2006 1:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Less Route Patterns for Toll Free

 
I am looking for a way in CallManager to reduce the 5 route patterns for
toll free.  Right no I use:
9.1[800]XXXXXXX
9.1[855]XXXXXXX
9.1[866]XXXXXXX
9.1[877]XXXXXXX
9.1[888]XXXXXXX

Any suggestions?


This email may contain proprietary and confidential information for the
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Any review, retransmission, dissemination, or other use of this
information by persons or entities other than 
the intended recipient is prohibited. If you are not the intended
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delete all copies. To the extent that opinions are expressed in this
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opinions of Sentinel Technologies or any of its affiliates, employees,
directors, officers or shareholders.

_______________________________________________
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From wsisk at cisco.com  Thu Sep 28 09:34:57 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 28 Sep 2006 09:34:57 -0400
Subject: [cisco-voip] Cisco IP Communicator and Realtek High Definition
 Audio
In-Reply-To: <006301c6e270$450667a0$cf1336e0$@tuke.sk>
References: <006301c6e270$450667a0$cf1336e0$@tuke.sk>
Message-ID: <451BCF81.8040108@cisco.com>

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From BRiley at jackhenry.com  Thu Sep 28 09:39:15 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Thu, 28 Sep 2006 08:39:15 -0500
Subject: [cisco-voip] MOH input CCME
Message-ID: <2B3745BC9D31514381A6FCC5E72D937316B28A@SMOMAIL.jhacorp.com>

 

 

I have a customer that is using Call manager express. They have recently
purchased a live MOH system to play a Musak stream. The device is made
by premier technologies. The only output is an RCA connection currently
but they have a RCA to RJ-11 adaptor. I am trying to find out exactly
what I need to connect the RJ-11 port from this MOH system into on my
CCME system. I asked the vendor if it was an analog connection and if my
device needed to provide dial tone. Their response was that it was a
5-ohm output with an amplifier and I needed to plug it into my phone
systems MOH input. So rather than going back and fourth with them I
decided to see if anyone has any experience with these type of devices
and what you would suggest. Thanks for your help. 

 

 

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 milay at cisco.com  Thu Sep 28 09:48:47 2006
From: milay at cisco.com (Mike Lay (milay))
Date: Thu, 28 Sep 2006 06:48:47 -0700
Subject: [cisco-voip] CDR IP Address Conversion
Message-ID: <413FABF5DCE22849AF86F442E472B78BD20B86@xmb-sjc-219.amer.cisco.com>

The time is stored as EPOCH time.:

http://www.perlservices.net/en/programs/epoch_converter/epoch_converter.
html

Mike 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of STEVEN CASPER
Sent: Wednesday, September 27, 2006 6:56 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CDR IP Address Conversion


Does anyone have a tool or formula they could share that can be used to
convert the signed decimal value in the CDR database to an IP address?

Thanks!
Steve




Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

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


From rratliff at cisco.com  Thu Sep 28 09:48:31 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 28 Sep 2006 09:48:31 -0400
Subject: [cisco-voip] MOH input CCME
In-Reply-To: <2B3745BC9D31514381A6FCC5E72D937316B28A@SMOMAIL.jhacorp.com>
References: <2B3745BC9D31514381A6FCC5E72D937316B28A@SMOMAIL.jhacorp.com>
Message-ID: <4CDC40EB-B049-4A6A-AE52-402703A6C1A3@cisco.com>

This doc is for CME 3.3.

http://www.cisco.com/en/US/products/sw/voicesw/ps4625/ 
products_configuration_guide_chapter09186a00804ad866.html#wp2175068

-Ryan

On Sep 28, 2006, at 9:39 AM, Bill Riley III wrote:





I have a customer that is using Call manager express. They have  
recently purchased a live MOH system to play a Musak stream. The  
device is made by premier technologies. The only output is an RCA  
connection currently but they have a RCA to RJ-11 adaptor. I am  
trying to find out exactly what I need to connect the RJ-11 port from  
this MOH system into on my CCME system. I asked the vendor if it was  
an analog connection and if my device needed to provide dial tone.  
Their response was that it was a 5-ohm output with an amplifier and I  
needed to plug it into my phone systems MOH input. So rather than  
going back and fourth with them I decided to see if anyone has any  
experience with these type of devices and what you would suggest.  
Thanks for your help.





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

From wsisk at cisco.com  Thu Sep 28 09:56:31 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 28 Sep 2006 09:56:31 -0400
Subject: [cisco-voip] MOH input CCME
In-Reply-To: <2B3745BC9D31514381A6FCC5E72D937316B28A@SMOMAIL.jhacorp.com>
References: <2B3745BC9D31514381A6FCC5E72D937316B28A@SMOMAIL.jhacorp.com>
Message-ID: <451BD48F.7050103@cisco.com>

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From aman.chugh at gmail.com  Thu Sep 28 10:22:07 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Thu, 28 Sep 2006 19:52:07 +0530
Subject: [cisco-voip] Default Qos behaviour
Message-ID: <d4c697ad0609280722x23c45251med7e815f83368ae5@mail.gmail.com>

All,

I would like to understand the default behaviour for QOS marking set by an
Ip phone for voice packets,In my scenario if dont have any qos settings
configured on the switch ,will the switch disregard/remove the marking set
by a cisco ipphone before passing them on next hop device,I ask this because
as I have configured CBWFQ with LLQ on my router but dont have any qos
settings on my switch where my phone connects ,I have not checked this out
with a sniffer just wanted to know if any one in the group knows about the
same.

Thanks for the help

Aman
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From jsteinberg at gmail.com  Thu Sep 28 11:05:42 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 28 Sep 2006 11:05:42 -0400
Subject: [cisco-voip] Configuring pickup groups
In-Reply-To: <451A9C9D.1010301@smalltime.com>
References: <45199679.40600@smalltime.com>
	<f1b65f880609270734i6c1d5d4dj22e6ed61bb59c634@mail.gmail.com>
	<451A9C9D.1010301@smalltime.com>
Message-ID: <f1b65f880609280805q88d975er4ad4a74c2019e51b@mail.gmail.com>

remove the pickup softkey and just leave opickup.

On 9/27/06, Robert Kulagowski <bob at smalltime.com> wrote:
>
> Justin Steinberg wrote:
> > You can do this with oPickup.
> >
> > Configure an pickup group "Everyone", add all phones to this group.
> > Configure an pickup group "Overhead night bell", add the DN of the night
> > bell to this group.
> >
> > Then add the park number for the "Overhead night bell" pickup group as a
> > current pickup group member of the "everyone" pickup group.
> >
> > Fwd the receptionist phone to a DN that is a member of "overhead night
> > bell" pickup group.
> >
> > Any phone in the "everyone" pickup group can dial opickup and answer the
> > night bell.
> >
> > You could put all phones, night bell dn, and pickup groups in the same
> > partition and that would be fine.
>
> Right; I tried this, but the issue is that any phone can still pickup
> any other ringing phone (because they're all in the same pickup group),
> which is not what I want.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From bob at smalltime.com  Thu Sep 28 11:33:43 2006
From: bob at smalltime.com (Robert Kulagowski)
Date: Thu, 28 Sep 2006 10:33:43 -0500
Subject: [cisco-voip] Configuring pickup groups
In-Reply-To: <f1b65f880609280805q88d975er4ad4a74c2019e51b@mail.gmail.com>
References: <45199679.40600@smalltime.com>	
	<f1b65f880609270734i6c1d5d4dj22e6ed61bb59c634@mail.gmail.com>	
	<451A9C9D.1010301@smalltime.com>
	<f1b65f880609280805q88d975er4ad4a74c2019e51b@mail.gmail.com>
Message-ID: <451BEB57.1050608@smalltime.com>

> remove the pickup softkey and just leave opickup.

Doesn't work; the opickup seems to have pickup as a subset; pressing 
opickup is still picking up a ringing phone, which wasn't what I wanted.

From rratliff at cisco.com  Thu Sep 28 11:53:34 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 28 Sep 2006 11:53:34 -0400
Subject: [cisco-voip] Default Qos behaviour
In-Reply-To: <d4c697ad0609280722x23c45251med7e815f83368ae5@mail.gmail.com>
References: <d4c697ad0609280722x23c45251med7e815f83368ae5@mail.gmail.com>
Message-ID: <052C7574-DDD7-40DF-AED8-24BAFA1537EF@cisco.com>

Depends on the switch hardware and whether qos is enabled or not  
globally on the switch.

The QOS SRND talks about this for the 6500 and 3550 switches (perhaps  
others too).

-Ryan

On Sep 28, 2006, at 10:22 AM, Aman Chugh wrote:

All,

I would like to understand the default behaviour for QOS marking set  
by an Ip phone for voice packets,In my scenario if dont have any qos  
settings configured on the switch ,will the switch disregard/remove  
the marking set by a cisco ipphone before passing them on next hop  
device,I ask this because as I have configured CBWFQ with LLQ on my  
router but dont have any qos settings on my switch where my phone  
connects ,I have not checked this out with a sniffer just wanted to  
know if any one in the group knows about the same.

Thanks for the help

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

From aman.chugh at gmail.com  Thu Sep 28 12:20:29 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Thu, 28 Sep 2006 21:50:29 +0530
Subject: [cisco-voip] Default Qos behaviour
In-Reply-To: <052C7574-DDD7-40DF-AED8-24BAFA1537EF@cisco.com>
References: <d4c697ad0609280722x23c45251med7e815f83368ae5@mail.gmail.com>
	<052C7574-DDD7-40DF-AED8-24BAFA1537EF@cisco.com>
Message-ID: <d4c697ad0609280920rd3260aby8064ed5916ba2157@mail.gmail.com>

Thanks for the reply ryan,I have a 3560 switch and qos is not enabled on any
ports or globaly.will the qos both layer 2 (COS) and layer 3 (DSCP ) be
removed/disregarded by the switch which is set by Ipphones.

Aman


On 9/28/06, Ryan Ratliff <rratliff at cisco.com> wrote:
>
> Depends on the switch hardware and whether qos is enabled or not
> globally on the switch.
>
> The QOS SRND talks about this for the 6500 and 3550 switches (perhaps
> others too).
>
> -Ryan
>
> On Sep 28, 2006, at 10:22 AM, Aman Chugh wrote:
>
> All,
>
> I would like to understand the default behaviour for QOS marking set
> by an Ip phone for voice packets,In my scenario if dont have any qos
> settings configured on the switch ,will the switch disregard/remove
> the marking set by a cisco ipphone before passing them on next hop
> device,I ask this because as I have configured CBWFQ with LLQ on my
> router but dont have any qos settings on my switch where my phone
> connects ,I have not checked this out with a sniffer just wanted to
> know if any one in the group knows about the same.
>
> Thanks for the help
>
> Aman
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From bcarter at sentinel.com  Thu Sep 28 12:27:58 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 28 Sep 2006 11:27:58 -0500
Subject: [cisco-voip] Default Qos behaviour
Message-ID: <6A0744639B86DD459433108DBF101D26011893E2@MAIL01.sentinel.com>

Ryan,

On this topic...The QoS SRND has some very good information/scenarios
for configuring Cat 6500s for QoS.  The problem is, there is multiple
scenarios and configuration samples for Cat OS, but 1 or 2 for Cat IOS.

Could you pass word along to the QoS SRND group that we would appreciate
more Cat 6500 IOS configuration samples?

Thanks 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Thursday, September 28, 2006 10:54 AM
To: Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Default Qos behaviour

Depends on the switch hardware and whether qos is enabled or not
globally on the switch.

The QOS SRND talks about this for the 6500 and 3550 switches (perhaps
others too).

-Ryan

On Sep 28, 2006, at 10:22 AM, Aman Chugh wrote:

All,

I would like to understand the default behaviour for QOS marking set by
an Ip phone for voice packets,In my scenario if dont have any qos
settings configured on the switch ,will the switch disregard/remove the
marking set by a cisco ipphone before passing them on next hop device,I
ask this because as I have configured CBWFQ with LLQ on my router but
dont have any qos settings on my switch where my phone connects ,I have
not checked this out with a sniffer just wanted to know if any one in
the group knows about the same.

Thanks for the help

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


This email may contain proprietary and confidential information for the sole use of the intended recipient. 
Any review, retransmission, dissemination, or other use of this information by persons or entities other than 
the intended recipient is prohibited. If you are not the intended recipient, please contact the sender and 
delete all copies. To the extent that opinions are expressed in this message, they are not necessarily the 
opinions of Sentinel Technologies or any of its affiliates, employees, directors, officers or shareholders.


From rratliff at cisco.com  Thu Sep 28 12:59:46 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 28 Sep 2006 12:59:46 -0400
Subject: [cisco-voip] Default Qos behaviour
In-Reply-To: <d4c697ad0609280920rd3260aby8064ed5916ba2157@mail.gmail.com>
References: <d4c697ad0609280722x23c45251med7e815f83368ae5@mail.gmail.com>
	<052C7574-DDD7-40DF-AED8-24BAFA1537EF@cisco.com>
	<d4c697ad0609280920rd3260aby8064ed5916ba2157@mail.gmail.com>
Message-ID: <2DABE874-DED8-458C-AA02-193104D1C760@cisco.com>

http://www.cisco.com/en/US/netsol/ns656/ 
networking_solutions_design_guidances_list.html#anchor9

Let me know if you can't find it.

-Ryan

On Sep 28, 2006, at 12:20 PM, Aman Chugh wrote:

Thanks for the reply ryan,I have a 3560 switch and qos is not enabled  
on any ports or globaly.will the qos both layer 2 (COS) and layer 3  
(DSCP ) be removed/disregarded by the switch which is set by Ipphones.

Aman


On 9/28/06, Ryan Ratliff <rratliff at cisco.com> wrote: Depends on the  
switch hardware and whether qos is enabled or not
globally on the switch.

The QOS SRND talks about this for the 6500 and 3550 switches (perhaps
others too).

-Ryan

On Sep 28, 2006, at 10:22 AM, Aman Chugh wrote:

All,

I would like to understand the default behaviour for QOS marking set
by an Ip phone for voice packets,In my scenario if dont have any qos
settings configured on the switch ,will the switch disregard/remove
the marking set by a cisco ipphone before passing them on next hop
device,I ask this because as I have configured CBWFQ with LLQ on my
router but dont have any qos settings on my switch where my phone
connects ,I have not checked this out with a sniffer just wanted to
know if any one in the group knows about the same.

Thanks for the help

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


From rratliff at cisco.com  Thu Sep 28 13:02:21 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 28 Sep 2006 13:02:21 -0400
Subject: [cisco-voip] Default Qos behaviour
In-Reply-To: <6A0744639B86DD459433108DBF101D26011893E2@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26011893E2@MAIL01.sentinel.com>
Message-ID: <F044E11A-7368-4EAB-8F95-8F90C6C12501@cisco.com>

"When configured as an access layer switch, the recommended software  
for the Supervisor is CatOS;
when configured as a distribution or core layer switch, the  
recommended software is Cisco IOS.
Furthermore, at the time of writing, Catalyst 6500 IOS does not yet  
support AutoQoS nor the
conditional-trust feature, therefore only CatOS examples of these  
models are given. However, both
AutoQoS VoIP and conditional trust have been committed for future  
releases of Catalyst 6500 IOS."

I take this to mean a later version of the SRND may well have config  
examples for IOS once the IOS is released that supports more of the  
features.

-Ryan

On Sep 28, 2006, at 12:27 PM, Carter, Bill wrote:

Ryan,

On this topic...The QoS SRND has some very good information/scenarios
for configuring Cat 6500s for QoS.  The problem is, there is multiple
scenarios and configuration samples for Cat OS, but 1 or 2 for Cat IOS.

Could you pass word along to the QoS SRND group that we would appreciate
more Cat 6500 IOS configuration samples?

Thanks

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Thursday, September 28, 2006 10:54 AM
To: Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Default Qos behaviour

Depends on the switch hardware and whether qos is enabled or not
globally on the switch.

The QOS SRND talks about this for the 6500 and 3550 switches (perhaps
others too).

-Ryan

On Sep 28, 2006, at 10:22 AM, Aman Chugh wrote:

All,

I would like to understand the default behaviour for QOS marking set by
an Ip phone for voice packets,In my scenario if dont have any qos
settings configured on the switch ,will the switch disregard/remove the
marking set by a cisco ipphone before passing them on next hop device,I
ask this because as I have configured CBWFQ with LLQ on my router but
dont have any qos settings on my switch where my phone connects ,I have
not checked this out with a sniffer just wanted to know if any one in
the group knows about the same.

Thanks for the help

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


This email may contain proprietary and confidential information for  
the sole use of the intended recipient.
Any review, retransmission, dissemination, or other use of this  
information by persons or entities other than
the intended recipient is prohibited. If you are not the intended  
recipient, please contact the sender and
delete all copies. To the extent that opinions are expressed in this  
message, they are not necessarily the
opinions of Sentinel Technologies or any of its affiliates,  
employees, directors, officers or shareholders.

From bcarter at sentinel.com  Thu Sep 28 13:14:12 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 28 Sep 2006 12:14:12 -0500
Subject: [cisco-voip] CME - Time of Day Routing
Message-ID: <6A0744639B86DD459433108DBF101D26011893E7@MAIL01.sentinel.com>

CME 3.3 - I have a CME issue.  During business hours, the main number
calls a hunt group with 2 users.  

During night hours, obviously there is no one available to answer the
call.  When calling in, the phone rings for 30-40 seconds, checking each
member of the hunt group, before hitting CUE and the Night/Closed
message.

Customer would like After hours calls to hit CUE Night/Closed message on
the first ring.  Can CME do time-of-day routing to accomplish this?


From bcarter at sentinel.com  Thu Sep 28 13:24:01 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 28 Sep 2006 12:24:01 -0500
Subject: [cisco-voip] Less Route Patterns for Toll Free
Message-ID: <6A0744639B86DD459433108DBF101D26011893EB@MAIL01.sentinel.com>

To the group....

This is what I am looking for. Is there a route pattern expression that
says 9.18[ any 2 occurrences of 0, any 2 occurrences of 5, any 2
occurrences of 6...etc. 

I don't want to use 9.@ with route filters.  Cisco documentation has
sufficently scared me away from 9. at .


-----Original Message-----
From: Ortiz, Carlos [mailto:CORTIZ at broward.org] 
Sent: Thursday, September 28, 2006 8:35 AM
To: Carter, Bill; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Less Route Patterns for Toll Free

We use 9.18[0678][0678]XXXXXXX.  This allows a few patterns that are not
toll free but not much concern on our side as our LD access codes are
invoked from the carrier.

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Wednesday, September 27, 2006 1:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Less Route Patterns for Toll Free

 
I am looking for a way in CallManager to reduce the 5 route patterns for
toll free.  Right no I use:
9.1[800]XXXXXXX
9.1[855]XXXXXXX
9.1[866]XXXXXXX
9.1[877]XXXXXXX
9.1[888]XXXXXXX

Any suggestions?


This email may contain proprietary and confidential information for the
sole use of the intended recipient. 
Any review, retransmission, dissemination, or other use of this
information by persons or entities other than the intended recipient is
prohibited. If you are not the intended recipient, please contact the
sender and delete all copies. To the extent that opinions are expressed
in this message, they are not necessarily the opinions of Sentinel
Technologies or any of its affiliates, employees, directors, officers or
shareholders.

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


This email may contain proprietary and confidential information for the sole use of the intended recipient. 
Any review, retransmission, dissemination, or other use of this information by persons or entities other than 
the intended recipient is prohibited. If you are not the intended recipient, please contact the sender and 
delete all copies. To the extent that opinions are expressed in this message, they are not necessarily the 
opinions of Sentinel Technologies or any of its affiliates, employees, directors, officers or shareholders.


From rratliff at cisco.com  Thu Sep 28 13:57:28 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 28 Sep 2006 13:57:28 -0400
Subject: [cisco-voip] Less Route Patterns for Toll Free
In-Reply-To: <6A0744639B86DD459433108DBF101D26011893EB@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26011893EB@MAIL01.sentinel.com>
Message-ID: <1BEFF2C8-46EB-4375-8C28-532749842B80@cisco.com>

The only way I can think of to do this is with the @ symbol and a  
route filter.  If you really really hate the NANP and don't want to  
use @ you can always rewrite your NANP file.  Be aware that it may be  
overwritten any time you upgrade CM though.

I will say that creating a route filter that matches only those area  
codes is not very painful.  You can always test with a separate  
partition only your test phone can access if you really want to put  
it through the paces and not risk affecting everyone else.

-Ryan

On Sep 28, 2006, at 1:24 PM, Carter, Bill wrote:

To the group....

This is what I am looking for. Is there a route pattern expression that
says 9.18[ any 2 occurrences of 0, any 2 occurrences of 5, any 2
occurrences of 6...etc.

I don't want to use 9.@ with route filters.  Cisco documentation has
sufficently scared me away from 9. at .


-----Original Message-----
From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Thursday, September 28, 2006 8:35 AM
To: Carter, Bill; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Less Route Patterns for Toll Free

We use 9.18[0678][0678]XXXXXXX.  This allows a few patterns that are not
toll free but not much concern on our side as our LD access codes are
invoked from the carrier.

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Wednesday, September 27, 2006 1:31 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Less Route Patterns for Toll Free


I am looking for a way in CallManager to reduce the 5 route patterns for
toll free.  Right no I use:
9.1[800]XXXXXXX
9.1[855]XXXXXXX
9.1[866]XXXXXXX
9.1[877]XXXXXXX
9.1[888]XXXXXXX

Any suggestions?


This email may contain proprietary and confidential information for the
sole use of the intended recipient.
Any review, retransmission, dissemination, or other use of this
information by persons or entities other than the intended recipient is
prohibited. If you are not the intended recipient, please contact the
sender and delete all copies. To the extent that opinions are expressed
in this message, they are not necessarily the opinions of Sentinel
Technologies or any of its affiliates, employees, directors, officers or
shareholders.

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


This email may contain proprietary and confidential information for  
the sole use of the intended recipient.
Any review, retransmission, dissemination, or other use of this  
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the intended recipient is prohibited. If you are not the intended  
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delete all copies. To the extent that opinions are expressed in this  
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_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip

From lelio at uoguelph.ca  Thu Sep 28 14:04:10 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 28 Sep 2006 14:04:10 -0400
Subject: [cisco-voip] Forward no answer external
References: <20060928054756.143590@gmx.net>
	<001201c6e2fb$66d01760$47196883@cfs.uoguelph.ca>
	<200609281956.44812.dietmar.romer@gmx.de>
Message-ID: <006501c6e328$7f9b9ee0$47196883@cfs.uoguelph.ca>

Then it's most likely a trunk configuration issue as described earlier...
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
"I can eat fifty eggs." "Nobody can eat fifty eggs."
  ----- Original Message ----- 
  From: Dietmar Romer 
  To: cisco-voip at puck.nether.net 
  Cc: Lelio Fulgenzi 
  Sent: Thursday, September 28, 2006 1:56 PM
  Subject: Re: [cisco-voip] Forward no answer external


  No, we have no unity. It is happening with direct calls. No attendant console 
  or something else between.

  Dietmar


  > Are you using Unity as your autoattendant? All Unity calls appear as
  > internal.
  >
  > ---------------------------------------------------------------------------
  >----- Lelio Fulgenzi, B.A.
  > Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
  > (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
  > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
  > "I can eat fifty eggs." "Nobody can eat fifty eggs."
  >   ----- Original Message -----
  >   From: db7td at gmx.de
  >   To: cisco-voip at puck.nether.net
  >   Sent: Thursday, September 28, 2006 1:47 AM
  >   Subject: [cisco-voip] Forward no answer external
  >
  >
  >   Hi,
  >
  >   I have a line with "forward no answer external" pointing to internal DN,
  > but the forwarding never occurs. When setting "forward no answer internal"
  > everything works fine.
  >
  >   How does Callmanager know whether a incoming call is internal or
  > external? Is this the onnet/offnet setting for the gateway?
  >
  >
  >   Dietmar
  >   _______________________________________________
  >   cisco-voip mailing list
  >   cisco-voip at puck.nether.net
  >   https://puck.nether.net/mailman/listinfo/cisco-voip

  -- 
  E-Mail: dietmar.romer at gmx.de
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From mfa at crec.ifas.ufl.edu  Thu Sep 28 16:14:32 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Thu, 28 Sep 2006 16:14:32 -0400
Subject: [cisco-voip] Deleting old domain reference in Unity
References: <mailman.3012.1159466656.76842.cisco-voip@puck.nether.net>
Message-ID: <044901c6e33a$b6e4bcd0$400cfb0a@crec.ifas.ufl.edu>

2 years ago we renamed our Windows 2003 domain (but Unity runs on Windows 
2000) from "lal.ufl.edu" to "crec.ifas.ufl.edu". The NetBIOS name was 
unchanged, at "CREC". Unity was rebooted per Microsoft domain-renaming 
instructions, and Unity itself wound up in the new domain OK. After some 
initial glitches, Unity seemed to work well for the last couple of years.

However, during recent housekeeping, we find we can't delete several 
subscribers, receiving the tool-tip-style error message: "The Domain 
controller for lal.ufl.edu cannot be contacted."  when hovering over the 
Delete icon.  Some of these subscribers were added long after the domain 
rename, FWIW.  The old domain name still appears in the first row of the 
ADDomain table in the UnityDB database.  I'm leaving it alone for now, but 
every fiber in my body cries out to delete that row.

Also, not unexpected, the top line in the DCGC tools table still shows the 
old domain.  The second line shows the new domain.  Help for DCGC says that 
"the first domain in the list is the one specified when the Message Store 
Configuration Wizard is run during Cisco Unity Installation."  Unity has 
been upgraded several times since the domain rename, and the Message Store 
Configuration Wizard was re-run sveral times since then as well, most 
recently when Unity 4.2(1) was installed.  In all cases the new domain was 
specified.  It looks like once a domain gets into Unity's ADDomain table, it 
stays there.

There is no reference to the old domain anywhere in the Unity Registry.

Bottom line (finally) is:  How do I get rid of this non-existent domain in 
Unity?

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL 


From rblayzor at inoc.net  Thu Sep 28 18:54:12 2006
From: rblayzor at inoc.net (Robert Blayzor)
Date: Thu, 28 Sep 2006 18:54:12 -0400
Subject: [cisco-voip] CME - Time of Day Routing
In-Reply-To: <6A0744639B86DD459433108DBF101D26011893E7@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26011893E7@MAIL01.sentinel.com>
Message-ID: <451C5294.3090002@inoc.net>

Carter, Bill wrote:
> Customer would like After hours calls to hit CUE Night/Closed message on
> the first ring.  Can CME do time-of-day routing to accomplish this?


Well yes and no.  The easiest way to do this is create an AA script to
do the ToD routing for you.  Have the phone answered by the AA, it can
check time of day and forward to extensions/hunt groups/voicemail
accordingly.

-- 
Robert Blayzor, BOFH
INOC, LLC
rblayzor\@(inoc.net|gmail.com)
PGP: 0x66F90BFC @ http://pgp.mit.edu
Key fingerprint = 6296 F715 038B 44C1 2720  292A 8580 500E 66F9 0BFC

My computer NEVER cras

From SPackett at fenwick.com  Thu Sep 28 19:41:58 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu, 28 Sep 2006 16:41:58 -0700
Subject: [cisco-voip] Website That Has List of CallManager Applications
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01D71390@SVCEXCH01.firm.fenwick.llp>

Does anyone recall the website that has a list of companies that works
on CallManager applications?  It's not a Cisco site and that's how I
found out about Berbee.  We're looking for some companies that might be
able to help us with a project.  Thanks in advance.
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From lelio at uoguelph.ca  Thu Sep 28 20:00:46 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 28 Sep 2006 20:00:46 -0400
Subject: [cisco-voip] Website That Has List of CallManager Applications
References: <01C8F21F30DE9740B53FD2EF05C3A8FC01D71390@SVCEXCH01.firm.fenwick.llp>
Message-ID: <002101c6e35b$390f4930$c901a8c0@deadfish>

Website That Has List of CallManager Applicationsnot sure if this is it, but i find it handy

https://www.ciptug.org/vendors.php

  ----- Original Message ----- 
  From: Stu Packett 
  To: cisco-voip at puck.nether.net 
  Sent: Thursday, September 28, 2006 7:41 PM
  Subject: [cisco-voip] Website That Has List of CallManager Applications


  Does anyone recall the website that has a list of companies that works on CallManager applications?  It's not a Cisco site and that's how I found out about Berbee.  We're looking for some companies that might be able to help us with a project.  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 EdwinKlaczko at sd54.k12.il.us  Thu Sep 28 20:48:47 2006
From: EdwinKlaczko at sd54.k12.il.us (Klaczko, Edwin)
Date: Thu, 28 Sep 2006 19:48:47 -0500
Subject: [cisco-voip] CER and BARS
Message-ID: <CBF954A4A67DCA45A593A8AD076CE8DE104C1584@info54.sd54.k12.il.us>



I am having trouble having BARS 4.0.10 and 4.0.11 backup my CER database.  Here's some background:

OS versions:? CCM 4.2.sr11? and CER 4.2sr11
software verisions:? CCM 4.1(3)sr1 and CER 1.3(1a) 
I am using the same version of BARS 4.0.11 on both, I had 4.0.9 on before and it had stopped working so I decided to reinstall the latest.

I installed BARS on my CCM publisher as a backup server and setup this server as a local data source. It is backing up the CCM database and CDR data fine.? I installed BARS on the CER publisher as a backup target, but when I try to add the CER as a data source from the CCM backup server I get the error message "Could not find BARS installed as a backup target on ..." 

I have tried BARS version 4.0.10 and the same thing happens.  I have tried the install multiple times and made sure the BackAdmin password is the same on both.  I do get a message when installing the Backup Target on the CER that the password does not match do I want to continue anyway.  But having done this several times I know the passwords are the same.  I even see a successful logon by my CCM publisher in the Security log on the CER server.  The install on the CER server completes without a problem, but I notice that there is no BARS service running.  Is there supposed to be a service running on the Backup Targets?  I have tried to create the Data Source with AV and CSA disabled on both servers still the same problem.  Any help would be appreciated.


Eddie Klaczko
Voip/Network Specialist
Schaumburg School District 54


From Ferdinand.Navarro at ficoh.com  Thu Sep 28 21:30:04 2006
From: Ferdinand.Navarro at ficoh.com (Navarro, Ferdinand)
Date: Thu, 28 Sep 2006 15:30:04 -1000
Subject: [cisco-voip] Website That Has List of CallManager Applications
Message-ID: <5040CF0E8EB7574CBBB147CAEAD0CAA1363110@f01mexcp03.ficoh.com>

 
Try here.
 
http://forums.cisco.com/eforum/servlet/IPCApps?page=IP_Phone_Apps
 
Just did a search on Google for Cisco Phone Applications and this was
the first hit.
 
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Thursday, September 28, 2006 2:01 PM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Website That Has List of CallManager
Applications


not sure if this is it, but i find it handy
 
https://www.ciptug.org/vendors.php
 

	----- Original Message ----- 
	From: Stu Packett <mailto:SPackett at fenwick.com>  
	To: cisco-voip at puck.nether.net 
	Sent: Thursday, September 28, 2006 7:41 PM
	Subject: [cisco-voip] Website That Has List of CallManager
Applications


	Does anyone recall the website that has a list of companies that
works on CallManager applications?  It's not a Cisco site and that's how
I found out about Berbee.  We're looking for some companies that might
be able to help us with a project.  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 Matt.Slaga at us.didata.com  Thu Sep 28 21:54:23 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Thu, 28 Sep 2006 21:54:23 -0400
Subject: [cisco-voip] Website That Has List of CallManager Applications
Message-ID: <C1FE15183DA37645BC0633BC604E44F0034AB009@USNAEXCH.na.didata.local>

The site you are probably thinking about is hotdispatch.com, but it has
been down for the past few months.  I think they may have pulled the
plug.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, September 28, 2006 7:42 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Website That Has List of CallManager Applications

 

Does anyone recall the website that has a list of companies that works
on CallManager applications?  It's not a Cisco site and that's how I
found out about Berbee.  We're looking for some companies that might be
able to help us with a project.  Thanks in advance.




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

This e-mail communication and any attachments may contain
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From SPackett at fenwick.com  Thu Sep 28 22:12:31 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu, 28 Sep 2006 19:12:31 -0700
Subject: [cisco-voip] Website That Has List of CallManager Applications
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC56F58E@SVCEXCH01.firm.fenwick.llp>

Hotdispatch.com - that's it, but too bad they're down.  They were a
great resource.  Thanks.

________________________________

From: Matt Slaga (US) [mailto:Matt.Slaga at us.didata.com] 
Sent: Thursday, September 28, 2006 6:54 PM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Website That Has List of CallManager
Applications



The site you are probably thinking about is hotdispatch.com, but it has
been down for the past few months.  I think they may have pulled the
plug.

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Thursday, September 28, 2006 7:42 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Website That Has List of CallManager Applications

 

Does anyone recall the website that has a list of companies that works
on CallManager applications?  It's not a Cisco site and that's how I
found out about Berbee.  We're looking for some companies that might be
able to help us with a project.  Thanks in advance.

________________________________

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

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From medley at mac.com  Thu Sep 28 16:34:02 2006
From: medley at mac.com (Tim Medley)
Date: Thu, 28 Sep 2006 13:34:02 -0700
Subject: [cisco-voip] Ciptug Conference?
Message-ID: <81F69EB4-829D-480A-937B-580B41EB3A24@mac.com>

In anyone going to the CIPTUG Conference? It's coming up in just a  
couple of weeks.

October 9 - 12 in sunny Phoenix Arizona.

We had an impromptu meetup last year by some of the folks on the list  
here, anyone interested in doing that again this year?

tm

Tim Medley
Moxie Enterprises
Email: tim.medley at moxieent.com
Phone: (704) 905-4604





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From CMStaffin at ra.rockwell.com  Fri Sep 29 02:15:52 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Fri, 29 Sep 2006 01:15:52 -0500
Subject: [cisco-voip] Ciptug Conference?
In-Reply-To: <81F69EB4-829D-480A-937B-580B41EB3A24@mac.com>
Message-ID: <OF8288AF0C.928547A1-ON862571F8.00224D95-862571F8.00225E23@ra.rockwell.com>

Tim,

I will be down there lwith 3 of my co-workers.  Let me know time and place

Thanks,
__________________________
Craig Staffin
Network Analyst
Global Network and Telecomunnications
Rockwell Automation
cmstaffin at ra.rockwell.com
(414) 382-2309
(4140 839-7608 Cell





Tim Medley <medley at mac.com> 
Sent by: cisco-voip-bounces at puck.nether.net
09/28/2006 03:34 PM

To
"cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
cc

Subject
[cisco-voip] Ciptug Conference?






In anyone going to the CIPTUG Conference? It's coming up in just a couple 
of weeks.

October 9 - 12 in sunny Phoenix Arizona.

We had an impromptu meetup last year by some of the folks on the list 
here, anyone interested in doing that again this year?

tm

Tim Medley
Moxie Enterprises
Email: tim.medley at moxieent.com
Phone: (704) 905-4604
 



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

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From nathan.a.reeves at gmail.com  Fri Sep 29 03:33:18 2006
From: nathan.a.reeves at gmail.com (Nathan Reeves)
Date: Fri, 29 Sep 2006 17:33:18 +1000
Subject: [cisco-voip] Issues with some calls into unity after user enters *
Message-ID: <62833d8a0609290033h6405b673xa5b4149c4e14f454@mail.gmail.com>

Seeing an issue where calls are passed over to Unity and the greeting
message is played.  When the users presses * to drop to the sign in
process the call gets dropped.

I've checked the call handler, and all looks good.  When calling the
unity pilot from a phone internally, I can press *.  When calling in
from some gateways, pressing * works.  But from one gateway it
doesn't.

Anyone got any ideas where to start troubleshooting?

From aman.chugh at gmail.com  Fri Sep 29 03:43:54 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Fri, 29 Sep 2006 13:13:54 +0530
Subject: [cisco-voip] Issues with some calls into unity after user
	enters *
In-Reply-To: <62833d8a0609290033h6405b673xa5b4149c4e14f454@mail.gmail.com>
References: <62833d8a0609290033h6405b673xa5b4149c4e14f454@mail.gmail.com>
Message-ID: <d4c697ad0609290043j56b46544w8de4696cdd7c60f@mail.gmail.com>

Might be a dtmf issue with this particular gateway or audio cut through not
happening,what protocol r you running on this gateway and what is the
interface to the pstn ,run a appropirate debug to check the disconnect
cause,also any thing getting generated on unity app log ,Also run unity port
status monitor for this call to  see if this dtmf is coming through to unity
or not.

Aman


On 9/29/06, Nathan Reeves <nathan.a.reeves at gmail.com> wrote:
>
> Seeing an issue where calls are passed over to Unity and the greeting
> message is played.  When the users presses * to drop to the sign in
> process the call gets dropped.
>
> I've checked the call handler, and all looks good.  When calling the
> unity pilot from a phone internally, I can press *.  When calling in
> from some gateways, pressing * works.  But from one gateway it
> doesn't.
>
> Anyone got any ideas where to start troubleshooting?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From nathan.a.reeves at gmail.com  Fri Sep 29 04:14:31 2006
From: nathan.a.reeves at gmail.com (Nathan Reeves)
Date: Fri, 29 Sep 2006 18:14:31 +1000
Subject: [cisco-voip] Issues with some calls into unity after user
	enters *
In-Reply-To: <d4c697ad0609290043j56b46544w8de4696cdd7c60f@mail.gmail.com>
References: <62833d8a0609290033h6405b673xa5b4149c4e14f454@mail.gmail.com>
	<d4c697ad0609290043j56b46544w8de4696cdd7c60f@mail.gmail.com>
Message-ID: <62833d8a0609290114y4334c3bbk3e67982986cbe53a@mail.gmail.com>

I've been running a unity port status monitor and I see the * come
through on PSTN.  It then steps through the cde scripts, but seems to
just drop through the script after the * keypress.

Running MGCP and E1's on all gateways.  No events on the Unity App
Log.  I'll start doing the debugs on the router.

A failed log looks like:

"13","2006/09/29","16:37:44","Conv","Call answered if needed"
"13","2006/09/29","16:37:44","Disp","Playing greeting for Call
Handler:  Opening Greeting"
"13","2006/09/29","16:37:44","Appl","phgreeting"
"13","2006/09/29","16:37:55","Conv","DTMF [*]"
"13","2006/09/29","16:37:55","Conv","Event NULL"
"13","2006/09/29","16:37:55","Conv","State -
d:\commserver\localize\scripts\phgreeting.cde!Play_BailAtGreeting"
"13","2006/09/29","16:37:55","Conv","Event NULL"
"13","2006/09/29","16:37:55","Appl","subsignin"
"13","2006/09/29","16:37:55","Disp","Subscriber Sign-In"
"13","2006/09/29","16:37:55","Conv","State -
d:\commserver\localize\scripts\subsignin.cde!AnswerPhone"
"13","2006/09/29","16:37:55","Conv","Event is [TrueEvent]"
"13","2006/09/29","16:37:55","Conv","State -
d:\commserver\localize\scripts\subsignin.cde!EnableEnrollment"
"13","2006/09/29","16:37:55","Conv","Event is [TrueEvent]"
"13","2006/09/29","16:37:55","Conv","State -
d:\commserver\localize\scripts\subsignin.cde!LoginUserWithIDAndPassword"
"13","2006/09/29","16:37:55","Appl","-->SubAuthenticate"
"13","2006/09/29","16:37:55","Conv","State -
d:\commserver\localize\scripts\subauthenticate.cde!GetIDFromUserInitialize"
"13","2006/09/29","16:37:55","Conv","Event NULL"
"13","2006/09/29","16:37:55","Conv","State -
d:\commserver\localize\scripts\subauthenticate.cde!GetReadyForIDFromUser"
"13","2006/09/29","16:37:55","Conv","Event NULL"
"13","2006/09/29","16:37:55","Conv","State -
d:\commserver\localize\scripts\subauthenticate.cde!GetIDFromUser"
"13","2006/09/29","16:37:55","Conv","Event is [TTStarEvent]"
"13","2006/09/29","16:37:55","Conv","State -
d:\commserver\localize\scripts\subauthenticate.cde!Goodbye"
"13","2006/09/29","16:37:55","Conv","Event NULL"
"13","2006/09/29","16:37:55","Conv","State -
d:\commserver\localize\scripts\subauthenticate.cde!GoodbyeGone"
"13","2006/09/29","16:37:55","Conv","Event is [HangupEvent]"
"13","2006/09/29","16:37:55","Appl","<--SubAuthenticate"
"13","2006/09/29","16:37:55","Conv","Event is [HangupEvent]"

A Successful * keypress looks like:
-----------------------------------------------------------------------------------------------
"12","2006/09/29","16:30:01","Conv","Call answered if needed"
"12","2006/09/29","16:30:01","Disp","Playing greeting for Call
Handler:  Opening Greeting"
"12","2006/09/29","16:30:01","Appl","phgreeting"
"12","2006/09/29","16:30:10","Conv","DTMF [*]"
"12","2006/09/29","16:30:10","Conv","Event NULL"
"12","2006/09/29","16:30:10","Conv","State -
d:\commserver\localize\scripts\phgreeting.cde!Play_BailAtGreeting"
"12","2006/09/29","16:30:10","Conv","Event NULL"
"12","2006/09/29","16:30:10","Appl","subsignin"
"12","2006/09/29","16:30:10","Disp","Subscriber Sign-In"
"12","2006/09/29","16:30:10","Conv","State -
d:\commserver\localize\scripts\subsignin.cde!AnswerPhone"
"12","2006/09/29","16:30:10","Conv","Event is [TrueEvent]"
"12","2006/09/29","16:30:10","Conv","State -
d:\commserver\localize\scripts\subsignin.cde!EnableEnrollment"
"12","2006/09/29","16:30:10","Conv","Event is [TrueEvent]"
"12","2006/09/29","16:30:10","Conv","State -
d:\commserver\localize\scripts\subsignin.cde!LoginUserWithIDAndPassword"
"12","2006/09/29","16:30:10","Appl","-->SubAuthenticate"
"12","2006/09/29","16:30:10","Conv","State -
d:\commserver\localize\scripts\subauthenticate.cde!GetIDFromUserInitialize"
"12","2006/09/29","16:30:10","Conv","Event is [TrueEvent]"
"12","2006/09/29","16:30:10","Conv","State -
d:\commserver\localize\scripts\subauthenticate.cde!CheckSignInCallerID"
"12","2006/09/29","16:30:10","Disp","Check for Cross-Box Login"
"12","2006/09/29","16:30:10","Conv","Event is [FalseEvent]"
"12","2006/09/29","16:30:10","Conv","State -
d:\commserver\localize\scripts\subauthenticate.cde!GetReadyForIDFromUser"
"12","2006/09/29","16:30:10","Conv","Event NULL"
"12","2006/09/29","16:30:10","Conv","State -
d:\commserver\localize\scripts\subauthenticate.cde!GetIDFromUser"
"12","2006/09/29","16:30:21","Appl","-->ConvUtilsNoEntry"
"12","2006/09/29","16:30:21","Conv","State -
d:\commserver\localize\scripts\convutilsnoentry.cde!NoEntry"
"12","2006/09/29","16:30:21","Conv","Event is [HangupEvent]"
"12","2006/09/29","16:30:21","Appl","<--ConvUtilsNoEntry"
"12","2006/09/29","16:30:21","Conv","Event is [HangupEvent]"
"12","2006/09/29","16:30:21","Appl","<--SubAuthenticate"
"12","2006/09/29","16:30:21","Conv","Event is [HangupEvent]"

On 9/29/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> Might be a dtmf issue with this particular gateway or audio cut through not
> happening,what protocol r you running on this gateway and what is the
> interface to the pstn ,run a appropirate debug to check the disconnect
> cause,also any thing getting generated on unity app log ,Also run unity port
> status monitor for this call to  see if this dtmf is coming through to unity
> or not.
>
> Aman
>
>
> On 9/29/06, Nathan Reeves <nathan.a.reeves at gmail.com> wrote:
> >
> > Seeing an issue where calls are passed over to Unity and the greeting
> > message is played.  When the users presses * to drop to the sign in
> > process the call gets dropped.
> >

From alfred.thyri at mii.at  Fri Sep 29 04:16:35 2006
From: alfred.thyri at mii.at (Thyri Alfred)
Date: Fri, 29 Sep 2006 10:16:35 +0200
Subject: [cisco-voip] Callmanager version 5.0 Upgrade
In-Reply-To: <18576071F1E2B047A6850107573475D42959DC@smiivie0123.at-work.ent>
References: <20DF280A12FBD1478CDCDEA366ACE077B5D56E@apollo.calore.local>
	<18576071F1E2B047A6850107573475D42959DC@smiivie0123.at-work.ent>
Message-ID: <18576071F1E2B047A6850107573475D4E552FB@smiivie0123.at-work.ent>

Update:

Since we decided to make a complete new configuration of our CCM (reason: our system grew over the years under different admins...), I made a new, clean CCM install, imported our phones and ran the DMA tool - without any troubles :-)

some hints for installing ccm5 with dma data:
- don't change ip or hostname, setup will fail
- copying the dam data by ftp: don't configure a ftp password. the installer doesn't send the configured password.

After installing 5.0(4) and importing the dma data I tried to activate the intermediate license on cco (https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationServlet - Cisco CallManager Version Upgrade) with no success.

I get get an error, because of an incorrect xml file.

regards,
Alfred.


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Thyri Alfred
Sent: Dienstag, 29. August 2006 22:00
To: Charles J. Boening; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Callmanager version 5.0 Upgrade

Hi,

Thanks for your response Charlie!

We do have a single server, and i tried to install it on the CCM itself.
I already contacted our local cisco system engineers, next step will be opening a tac case. I will do so more testing, since i still have to wait for the delivery of our new ccm hardware - so i still have some time :)

I will send an update after we fixed the problem...

So long,
Alfred.

-----Urspr?ngliche Nachricht-----
Von: Charles J. Boening [mailto:charlieb at cot.net]
Gesendet: Dienstag, 29. August 2006 17:19
An: Thyri Alfred; cisco-voip at puck.nether.net
Betreff: RE: [cisco-voip] Callmanager version 5.0 Upgrade

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

Alfred,

Your version should be supported by 5.0.4 DMA Tool.  I'd suggest opening a TAC case.  Maybe try a reboot if you can then install again.  If you have a publisher/subscriber, then I think you're supposed to run the tool on the publisher.  Also, you have to install the DMA Tool on the CCM server itself.  You can't install to a workstation or another server.

Good luck,
Charlie


> -----Original Message-----
> From: Thyri Alfred [mailto:alfred.thyri at mii.at]
> Sent: Tuesday, August 29, 2006 3:43 AM
> To: Charles J. Boening; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Callmanager version 5.0 Upgrade
> 
> Hi Charlie,
> 
> I tried to install the dma tool (5.0(4)) on our CCM 4.1(3)sr2 
> installation with bad luck :-(
> 
> In the log file I found something interesting:
> 13:54:15: DMAInstall.cpp: fnInitDMAInstallProperty:      Error: Init
> SZCMVER from default because of invalid szCCMBuildVersion = 4.1(3)
> 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         Function started
> 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         
> CCMInstalled = "False"
> 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         Function ended
> 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         Function started
> 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         SZCMVER = "2"
> 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         Function ended
> 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         Function started
> 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         REINSTALL = ""
> 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         Function ended
> 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         Function started
> 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         SZSETUPTYPE = ""
> 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         Function ended
> 
> It seems that our version is not supported by the dma tool, although 
> it should be supported.
> 
> I also tried to check with Upgrade Utility (version 4.2(2)), with the 
> result that everything is ok but the hardware should be upgraded. We 
> already ordered new hardware...
> 
> 
> Do you have any hints for me?? Somebody else?
> 
> cheers,
> Alfred.
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Charles J.
> Boening
> Sent: Dienstag, 29. August 2006 05:40
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callmanager version 5.0 Upgrade
> 
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Here's my experience with upgrading to CCM5.
> 
> I came from a CCM 4.0(2a) system running on an ICS7750 to CCM
> 5.0(3) running on my own supplied IBM server.  Cisco documentation 
> indicated my IBM server (8491G2U) was supported in this release.  I 
> actually purchased it based on Cisco documentation and specifications 
> just for this application.
> Turns out that the 8841G2U was supported by the installer but the 
> 8491G2U wasn't.  The only difference between these two is the 
> warranty.  One has a 1yr warranty and the one I had came with a 3yr 
> on-site warranty.  Other than that, there are no hardware differences.
> TAC opened a bug case and was supposed to fix the documentation.  I 
> think TAC said I'd have to wait for 5.0(4) which wouldn't be available 
> on DVD for a few more months or whatever it was.
> Being the nosey person I am, I snooped around and found where in the 
> installer to change it to recognize my machine type.
> I ended up making an ISO image and used a hex editor to change the 
> file the installer reads so my machine type would be supported.  :) 
> Please don't ask what file I changed as I'm sure Cisco wouldn't want 
> that information floating around.  I'm sure some of the more 
> industrious users out there will be able to figure it out on their 
> own.  For the rest,  I'm sure someone will post the information on 
> some message board or somewhere at some point.  :)
> 
> During my data export from CCM 4.x, I ended up having some data 
> inconsistencies in my 4.x database that caused export errors.  A call 
> to TAC and a few days later, they had found and fixed the problems.  I 
> was then able to successfully export out my old data using DMA 5.0.4.
> 
> Importing of Windows CCM data is done during the installation process.
> In my playing around, I had successfully installed CCM5 without 
> importing any data.  Once installed, I saw no provisions for importing 
> the data after the installation.
> So, with exported data in hand (actually on a local FTP server), I 
> went through the installer process and came up with some errors when 
> importing Windows DMA data.  I didn't see any documentation indicating 
> it, but it turns out the name of my new CCM had to match the name of 
> the machine I was coming from.  The IP address may have had to match 
> too but I'm not sure.  Per TAC request, I set both the machine name 
> and IP address so they were the same as my current production system.
> This wasn't a problem since I was building the new server on it's own 
> VLAN for testing.  Once this was done, the install went very smooth 
> and all my data was imported.  I was actually very impressed with the 
> installer and it's ease of use.  Not that there's much to it, but it 
> was all pretty easy and straight forward.
> 
> All in all I'd say the installation and upgrade went pretty well.  I 
> was a little disappointed with a couple of TAC people and their 
> slowness to respond.  At one point I waited over a week for a 
> response.  Fortunately I had enough other duties to keep me busy in 
> the meantime.  I think overall I spent about two to three weeks off 
> and on between messing with things on my own and waiting on TAC.
> 
> Post upgrade everything is running very well.  My Unity integration, 
> Attendant Console and everything else came over just fine.  I also 
> have a SIP trunk to an Asterisk box that came right up and is running 
> fine as well.  I'm using MGCP on a 1760V router to two PRIs.  I have 
> no FXO or FXS lines.
> We're actually a small rural telephone company so I have plenty of CO 
> lines available for fax machines and such.  :)
> 
> The only issue I've had so far is with shared line appearances.  In 
> the old system I had 3 extensions shared across seven phones.  I had 
> call forward busy from one extension to the other with the last 
> extension going to voicemail on busy.  All extensions go to voicemail 
> on no answer.  After the conversion the forwarding wasn't working 
> right.  If someone was on the first line, callers would go right to 
> voicemail or get a fast busy; I don't remember which.  In the end I 
> think I had to mess with the number of concurrent calls on the line.
> I increased the max calls for each phone on all extensions to the 
> number of shared lines
> + 1.  I did this on both 7940 and 7960 phones that shared the
> extensions.  I set the busy call trigger to 1 and everything seems to 
> be working ok.  In CCM5, there seems to be more of a distinction 
> between lines and phones.  You can edit lines directly without even 
> going through the phone config.  A little confusing at first but not 
> bad once you figure it all out.
> 
> One more issue I have is doing the 5.0(4a) update.  I'm currently at
> 5.0(4) but it seems that updates are now required to be signed and the 
> update posted on CCO isn't.  Oh well, another call to TAC I guess.  :)
> 
> We've been live for about two weeks now.  Everything has been running 
> very smooth and so far I'm pleased with CCM5.
> 
> Charlie
> 
> 
> 
> 
> 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Manoj Kalpage
> > Sent: Monday, August 28, 2006 7:04 PM
> > To: Matt Slaga (US)
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] Callmanager version 5.0 Upgrade
> > 
> > Thank you very much for the reply as always.
> > What is your recommendation to upgrade to Version 5.0? Does
> this work
> > smoothly? Anyone had any bad experience of upgrading to 5.0?
> >  
> > Best Regards,
> > Manoj
> > 
> >  
> > On 8/28/06, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote: 
> > 
> > 	Manoj-
> > 
> > 	 
> > 
> > 	CallManager 5.0 only runs on Linux.  While you can upgrade from
> 4.0
> > or 4.1 to 5.0, it requires a complete rebuild and restore
> process to
> > migrate the data from SQL to Informix under the new linux platform.
> > There is no 5.0 that runs on win2k.
> > 
> > 	 
> > 
> > 	 
> > 
> > 	
> > ________________________________
> > 
> > 
> > 	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
> Manoj Kalpage
> > 	Sent: Sunday, August 27, 2006 10:04 PM
> > 
> > 	
> > 	To: cisco-voip at puck.nether.net
> > 	
> > 	Subject: [cisco-voip] Callmanager version 5.0 Upgrade
> > 
> > 	
> > 
> > 	 
> > 
> > 	Hi All,
> > 
> > 	Please someone help me if I am wrong here.
> > 
> > 	My Understanding is CallManager 4.0 can be upgraded to
> Callamanger
> > 5.x keeping existing windows 2000 OS but new installation
> only work on
> > Linux.
> > 
> > 	 
> > 
> > 	Regards,
> > 
> > 	Manoj
> > 
> > 	 
> > 
> > 	
> > ________________________________
> > 
> > 
> > 	
> > 
> > 
> > 	
> > 	Disclaimer:
> > 	
> > 	This e-mail communication and any attachments may contain
> > 	confidential and privileged information and is for use by the
> > 	designated addressee(s) named above only. If you are not the 
> > 	intended addressee, you are hereby notified that you
> have received
> > 	this communication in error and that any use or reproduction of
> > 	this email or its contents is strictly prohibited and may be
> > 	unlawful. If you have received this communication in
> error, please
> > 	notify us immediately by replying to this message and
> deleting it
> > 	from your computer. Thank you.
> > 	
> > 
> > 
> > 
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> 
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> TcATbYEo8nOCBavtsYcvzpU=
> =qoKW
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> 
> _______________________________________________
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> https://puck.nether.net/mailman/listinfo/cisco-voip
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From aman.chugh at gmail.com  Fri Sep 29 04:33:47 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Fri, 29 Sep 2006 14:03:47 +0530
Subject: [cisco-voip] Issues with some calls into unity after user
	enters *
In-Reply-To: <62833d8a0609290114y4334c3bbk3e67982986cbe53a@mail.gmail.com>
References: <62833d8a0609290033h6405b673xa5b4149c4e14f454@mail.gmail.com>
	<d4c697ad0609290043j56b46544w8de4696cdd7c60f@mail.gmail.com>
	<62833d8a0609290114y4334c3bbk3e67982986cbe53a@mail.gmail.com>
Message-ID: <d4c697ad0609290133m1aab9854o1c674575bc8a6f12@mail.gmail.com>

Nathan, It looks to be Gw or ccm issue if no error getting generated on
Unity app log.Detailed ccm trace along with gw config can shed more light
into this.


Aman


On 9/29/06, Nathan Reeves <nathan.a.reeves at gmail.com> wrote:
>
> I've been running a unity port status monitor and I see the * come
> through on PSTN.  It then steps through the cde scripts, but seems to
> just drop through the script after the * keypress.
>
> Running MGCP and E1's on all gateways.  No events on the Unity App
> Log.  I'll start doing the debugs on the router.
>
> A failed log looks like:
>
> "13","2006/09/29","16:37:44","Conv","Call answered if needed"
> "13","2006/09/29","16:37:44","Disp","Playing greeting for Call
> Handler:  Opening Greeting"
> "13","2006/09/29","16:37:44","Appl","phgreeting"
> "13","2006/09/29","16:37:55","Conv","DTMF [*]"
> "13","2006/09/29","16:37:55","Conv","Event NULL"
> "13","2006/09/29","16:37:55","Conv","State -
> d:\commserver\localize\scripts\phgreeting.cde!Play_BailAtGreeting"
> "13","2006/09/29","16:37:55","Conv","Event NULL"
> "13","2006/09/29","16:37:55","Appl","subsignin"
> "13","2006/09/29","16:37:55","Disp","Subscriber Sign-In"
> "13","2006/09/29","16:37:55","Conv","State -
> d:\commserver\localize\scripts\subsignin.cde!AnswerPhone"
> "13","2006/09/29","16:37:55","Conv","Event is [TrueEvent]"
> "13","2006/09/29","16:37:55","Conv","State -
> d:\commserver\localize\scripts\subsignin.cde!EnableEnrollment"
> "13","2006/09/29","16:37:55","Conv","Event is [TrueEvent]"
> "13","2006/09/29","16:37:55","Conv","State -
> d:\commserver\localize\scripts\subsignin.cde!LoginUserWithIDAndPassword"
> "13","2006/09/29","16:37:55","Appl","-->SubAuthenticate"
> "13","2006/09/29","16:37:55","Conv","State -
>
> d:\commserver\localize\scripts\subauthenticate.cde!GetIDFromUserInitialize"
> "13","2006/09/29","16:37:55","Conv","Event NULL"
> "13","2006/09/29","16:37:55","Conv","State -
> d:\commserver\localize\scripts\subauthenticate.cde!GetReadyForIDFromUser"
> "13","2006/09/29","16:37:55","Conv","Event NULL"
> "13","2006/09/29","16:37:55","Conv","State -
> d:\commserver\localize\scripts\subauthenticate.cde!GetIDFromUser"
> "13","2006/09/29","16:37:55","Conv","Event is [TTStarEvent]"
> "13","2006/09/29","16:37:55","Conv","State -
> d:\commserver\localize\scripts\subauthenticate.cde!Goodbye"
> "13","2006/09/29","16:37:55","Conv","Event NULL"
> "13","2006/09/29","16:37:55","Conv","State -
> d:\commserver\localize\scripts\subauthenticate.cde!GoodbyeGone"
> "13","2006/09/29","16:37:55","Conv","Event is [HangupEvent]"
> "13","2006/09/29","16:37:55","Appl","<--SubAuthenticate"
> "13","2006/09/29","16:37:55","Conv","Event is [HangupEvent]"
>
> A Successful * keypress looks like:
>
> -----------------------------------------------------------------------------------------------
> "12","2006/09/29","16:30:01","Conv","Call answered if needed"
> "12","2006/09/29","16:30:01","Disp","Playing greeting for Call
> Handler:  Opening Greeting"
> "12","2006/09/29","16:30:01","Appl","phgreeting"
> "12","2006/09/29","16:30:10","Conv","DTMF [*]"
> "12","2006/09/29","16:30:10","Conv","Event NULL"
> "12","2006/09/29","16:30:10","Conv","State -
> d:\commserver\localize\scripts\phgreeting.cde!Play_BailAtGreeting"
> "12","2006/09/29","16:30:10","Conv","Event NULL"
> "12","2006/09/29","16:30:10","Appl","subsignin"
> "12","2006/09/29","16:30:10","Disp","Subscriber Sign-In"
> "12","2006/09/29","16:30:10","Conv","State -
> d:\commserver\localize\scripts\subsignin.cde!AnswerPhone"
> "12","2006/09/29","16:30:10","Conv","Event is [TrueEvent]"
> "12","2006/09/29","16:30:10","Conv","State -
> d:\commserver\localize\scripts\subsignin.cde!EnableEnrollment"
> "12","2006/09/29","16:30:10","Conv","Event is [TrueEvent]"
> "12","2006/09/29","16:30:10","Conv","State -
> d:\commserver\localize\scripts\subsignin.cde!LoginUserWithIDAndPassword"
> "12","2006/09/29","16:30:10","Appl","-->SubAuthenticate"
> "12","2006/09/29","16:30:10","Conv","State -
>
> d:\commserver\localize\scripts\subauthenticate.cde!GetIDFromUserInitialize"
> "12","2006/09/29","16:30:10","Conv","Event is [TrueEvent]"
> "12","2006/09/29","16:30:10","Conv","State -
> d:\commserver\localize\scripts\subauthenticate.cde!CheckSignInCallerID"
> "12","2006/09/29","16:30:10","Disp","Check for Cross-Box Login"
> "12","2006/09/29","16:30:10","Conv","Event is [FalseEvent]"
> "12","2006/09/29","16:30:10","Conv","State -
> d:\commserver\localize\scripts\subauthenticate.cde!GetReadyForIDFromUser"
> "12","2006/09/29","16:30:10","Conv","Event NULL"
> "12","2006/09/29","16:30:10","Conv","State -
> d:\commserver\localize\scripts\subauthenticate.cde!GetIDFromUser"
> "12","2006/09/29","16:30:21","Appl","-->ConvUtilsNoEntry"
> "12","2006/09/29","16:30:21","Conv","State -
> d:\commserver\localize\scripts\convutilsnoentry.cde!NoEntry"
> "12","2006/09/29","16:30:21","Conv","Event is [HangupEvent]"
> "12","2006/09/29","16:30:21","Appl","<--ConvUtilsNoEntry"
> "12","2006/09/29","16:30:21","Conv","Event is [HangupEvent]"
> "12","2006/09/29","16:30:21","Appl","<--SubAuthenticate"
> "12","2006/09/29","16:30:21","Conv","Event is [HangupEvent]"
>
> On 9/29/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> > Might be a dtmf issue with this particular gateway or audio cut through
> not
> > happening,what protocol r you running on this gateway and what is the
> > interface to the pstn ,run a appropirate debug to check the disconnect
> > cause,also any thing getting generated on unity app log ,Also run unity
> port
> > status monitor for this call to  see if this dtmf is coming through to
> unity
> > or not.
> >
> > Aman
> >
> >
> > On 9/29/06, Nathan Reeves <nathan.a.reeves at gmail.com> wrote:
> > >
> > > Seeing an issue where calls are passed over to Unity and the greeting
> > > message is played.  When the users presses * to drop to the sign in
> > > process the call gets dropped.
> > >
>
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From SCASPER at mandtbank.com  Fri Sep 29 06:42:24 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Fri, 29 Sep 2006 06:42:24 -0400
Subject: [cisco-voip]  SRST and Large Sites
Message-ID: <s51cc072.044@comrcm01intgwia.mandtbank.com>


Curious as to how or if anyone uses SRST for large sites (500+)
telephones. Does any one do this and if so how do they have it set up -
for instance all users to one 3845 router or are they spread across a
couple of routers.

Thanks!
Steve


Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026


From Matt.Slaga at us.didata.com  Fri Sep 29 08:06:16 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 29 Sep 2006 08:06:16 -0400
Subject: [cisco-voip] Ciptug Conference?
Message-ID: <C1FE15183DA37645BC0633BC604E44F0034AB0FE@USNAEXCH.na.didata.local>

I'll be there as well.  Are you running the show again?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Medley
Sent: Thursday, September 28, 2006 4:34 PM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] Ciptug Conference?

 

In anyone going to the CIPTUG Conference? It's coming up in just a
couple of weeks.

 

October 9 - 12 in sunny Phoenix Arizona.

 

We had an impromptu meetup last year by some of the folks on the list
here, anyone interested in doing that again this year?

 

tm

 

Tim Medley

Moxie Enterprises

Email: tim.medley at moxieent.com

Phone: (704) 905-4604

 

 





 




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From SCASPER at mandtbank.com  Fri Sep 29 08:10:03 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Fri, 29 Sep 2006 08:10:03 -0400
Subject: [cisco-voip] SRST and Large Sites
Message-ID: <s51cd4f7.056@comrcm01intgwia.mandtbank.com>

These are sites that are configured with Gig-E across a dark fiber ring.
Our strategy is a centralized cluster split across our 2 data center
locations. Perhaps I should look at locating backup CCM servers these
sites.

>>> "Jim McBurnett" <jim at tgasolutions.com> 09/29/06 6:49 AM >>>
Steven,
I have a few customers that are this size, and we recommended a CCM
server to sit there.
The control and phone upgrades would kill a WAN link.
DO you have a DS-3 here?
Or at least a frac ds-3?

Check out the SRND
www.cisco.com/go/SRND 
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of STEVEN CASPER
Sent: Friday, September 29, 2006 6:42 AM
To: cisco-voip at puck.nether.net 
Subject: [cisco-voip] SRST and Large Sites


Curious as to how or if anyone uses SRST for large sites (500+)
telephones. Does any one do this and if so how do they have it set up
-
for instance all users to one 3845 router or are they spread across a
couple of routers.

Thanks!
Steve


Steve Casper
Voice Technologies
M&T Bank
(410) 347-6026

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



From wsisk at cisco.com  Fri Sep 29 08:48:53 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Fri, 29 Sep 2006 08:48:53 -0400
Subject: [cisco-voip] CER and BARS
In-Reply-To: <CBF954A4A67DCA45A593A8AD076CE8DE104C1584@info54.sd54.k12.il.us>
References: <CBF954A4A67DCA45A593A8AD076CE8DE104C1584@info54.sd54.k12.il.us>
Message-ID: <26E61049-28DC-41DC-92A8-5530E1839910@cisco.com>

Edwin,

Are your servers members of a windows domain?
CSCsb02046 BARS may not work when servers are members of a windows  
domain

Otherwise, need to verify component services is working properly on  
your CER server.  start/programs/administrative tools/component  
services.  Is there a BARS component listed under "My Computer"?

/Wes

On Sep 28, 2006, at 8:48 PM, Klaczko, Edwin wrote:



I am having trouble having BARS 4.0.10 and 4.0.11 backup my CER  
database.  Here's some background:

OS versions:  CCM 4.2.sr11  and CER 4.2sr11
software verisions:  CCM 4.1(3)sr1 and CER 1.3(1a)
I am using the same version of BARS 4.0.11 on both, I had 4.0.9 on  
before and it had stopped working so I decided to reinstall the latest.

I installed BARS on my CCM publisher as a backup server and setup  
this server as a local data source. It is backing up the CCM database  
and CDR data fine.  I installed BARS on the CER publisher as a backup  
target, but when I try to add the CER as a data source from the CCM  
backup server I get the error message "Could not find BARS installed  
as a backup target on ..."

I have tried BARS version 4.0.10 and the same thing happens.  I have  
tried the install multiple times and made sure the BackAdmin password  
is the same on both.  I do get a message when installing the Backup  
Target on the CER that the password does not match do I want to  
continue anyway.  But having done this several times I know the  
passwords are the same.  I even see a successful logon by my CCM  
publisher in the Security log on the CER server.  The install on the  
CER server completes without a problem, but I notice that there is no  
BARS service running.  Is there supposed to be a service running on  
the Backup Targets?  I have tried to create the Data Source with AV  
and CSA disabled on both servers still the same problem.  Any help  
would be appreciated.


Eddie Klaczko
Voip/Network Specialist
Schaumburg School District 54

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From lelio at uoguelph.ca  Fri Sep 29 09:19:13 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 29 Sep 2006 09:19:13 -0400
Subject: [cisco-voip] Ciptug Conference?
References: <C1FE15183DA37645BC0633BC604E44F0034AB0FE@USNAEXCH.na.didata.local>
Message-ID: <008101c6e3c9$dbb7df30$c901a8c0@deadfish>

I asked a while ago and got a few responses. I think another get together would be great. Is anyone familiar with the Pheonix area that could suggest something? I have a friend who lived there and might ask him if there is a nice place we can meet. I'll also look up the hotel and see if there is a bar in there. Might not be as nice as the Vegas one, but what the heck....

  ----- Original Message ----- 
  From: Matt Slaga (US) 
  To: Tim Medley ; cisco-voip at puck-nether.net 
  Sent: Friday, September 29, 2006 8:06 AM
  Subject: Re: [cisco-voip] Ciptug Conference?


  I'll be there as well.  Are you running the show again?

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Medley
  Sent: Thursday, September 28, 2006 4:34 PM
  To: cisco-voip at puck-nether.net
  Subject: [cisco-voip] Ciptug Conference?

   

  In anyone going to the CIPTUG Conference? It's coming up in just a couple of weeks.

   

  October 9 - 12 in sunny Phoenix Arizona.

   

  We had an impromptu meetup last year by some of the folks on the list here, anyone interested in doing that again this year?

   

  tm

   

  Tim Medley

  Moxie Enterprises

  Email: tim.medley at moxieent.com

  Phone: (704) 905-4604

   

   





   



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  https://puck.nether.net/mailman/listinfo/cisco-voip
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From lelio at uoguelph.ca  Fri Sep 29 09:20:09 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 29 Sep 2006 09:20:09 -0400
Subject: [cisco-voip] Ciptug Conference?
Message-ID: <008e01c6e3c9$fcf14600$c901a8c0@deadfish>

Also, if people tell me/Tim what time you're showing up, we can plan things so as many people can attend. Perhaps something just before one of the CIPTUG events.

  ----- Original Message ----- 
  From: Lelio Fulgenzi 
  To: Matt Slaga (US) ; Tim Medley ; cisco-voip at puck-nether.net 
  Sent: Friday, September 29, 2006 9:19 AM
  Subject: Re: [cisco-voip] Ciptug Conference?


  I asked a while ago and got a few responses. I think another get together would be great. Is anyone familiar with the Pheonix area that could suggest something? I have a friend who lived there and might ask him if there is a nice place we can meet. I'll also look up the hotel and see if there is a bar in there. Might not be as nice as the Vegas one, but what the heck....

    ----- Original Message ----- 
    From: Matt Slaga (US) 
    To: Tim Medley ; cisco-voip at puck-nether.net 
    Sent: Friday, September 29, 2006 8:06 AM
    Subject: Re: [cisco-voip] Ciptug Conference?


    I'll be there as well.  Are you running the show again?

     


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

    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Medley
    Sent: Thursday, September 28, 2006 4:34 PM
    To: cisco-voip at puck-nether.net
    Subject: [cisco-voip] Ciptug Conference?

     

    In anyone going to the CIPTUG Conference? It's coming up in just a couple of weeks.

     

    October 9 - 12 in sunny Phoenix Arizona.

     

    We had an impromptu meetup last year by some of the folks on the list here, anyone interested in doing that again this year?

     

    tm

     

    Tim Medley

    Moxie Enterprises

    Email: tim.medley at moxieent.com

    Phone: (704) 905-4604

     

     





     



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


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



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


    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip
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From duncanw at otc.edu  Fri Sep 29 09:28:26 2006
From: duncanw at otc.edu (DUNCAN, W B.)
Date: Fri, 29 Sep 2006 08:28:26 -0500
Subject: [cisco-voip] Ciptug Conference?
In-Reply-To: <008101c6e3c9$dbb7df30$c901a8c0@deadfish>
Message-ID: <06D1B6D4926222458F803D0D3EDCCB7EACDF3C@EXM1.otc.edu>

I will be attending the conference as well so count me in.

 

Brian Duncan

Ozarks Technical Community College

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, September 29, 2006 8:19 AM
To: Matt Slaga (US); Tim Medley; cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] Ciptug Conference?

 

I asked a while ago and got a few responses. I think another get
together would be great. Is anyone familiar with the Pheonix area that
could suggest something? I have a friend who lived there and might ask
him if there is a nice place we can meet. I'll also look up the hotel
and see if there is a bar in there. Might not be as nice as the Vegas
one, but what the heck....

 

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

	From: Matt Slaga (US) <mailto:Matt.Slaga at us.didata.com>  

	To: Tim Medley <mailto:medley at mac.com>  ;
cisco-voip at puck-nether.net 

	Sent: Friday, September 29, 2006 8:06 AM

	Subject: Re: [cisco-voip] Ciptug Conference?

	 

	I'll be there as well.  Are you running the show again?

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Medley
	Sent: Thursday, September 28, 2006 4:34 PM
	To: cisco-voip at puck-nether.net
	Subject: [cisco-voip] Ciptug Conference?

	 

	In anyone going to the CIPTUG Conference? It's coming up in just
a couple of weeks.

	 

	October 9 - 12 in sunny Phoenix Arizona.

	 

	We had an impromptu meetup last year by some of the folks on the
list here, anyone interested in doing that again this year?

	 

	tm

	 

	

	Tim Medley

	Moxie Enterprises

	Email: tim.medley at moxieent.com

	Phone: (704) 905-4604

	 

	 

	 

	 

	
________________________________


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

	
________________________________


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

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From lelio at uoguelph.ca  Fri Sep 29 09:50:19 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 29 Sep 2006 09:50:19 -0400
Subject: [cisco-voip] ciptug get together time/place suggestion
Message-ID: <00c701c6e3ce$33d45780$c901a8c0@deadfish>

How more fitting a place can we find? Right in the Hyatt, a lounge called "Networks" !
    a.. http://phoenix.hyatt.com/hyatt/hotels/gallery/photos.jsp?hotelId=2199&start=14
    b.. http://phoenix.hyatt.com/hyatt/hotels/gallery/photos.jsp?hotelId=2199&start=14
    c.. http://phoenix.hyatt.com/hyatt/hotels/gallery/virtual-tour.jsp?icamp=propVirtualTour and click on Networks
They have plenty of specials during the week too!
    a.. http://www.thehyatteditor.com/images/editor/siteImages/images/hotels/phxrp/Hyatt%20Web-Networks.pdf
Since last year it seemed like the first day was a bit too rushed, how about we say Tuesday between 5pm and 6pm at Networks?



--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.     Network Analyst (CCS)
University of Guelph     Guelph, Ontario     N1G 2W1
(519) 824-4120 x56354     (519) 767-1060 FAX 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Sanity First : Number of days with less than
50 messages in my inbox at the end of the day:    buffer overrun
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From Matt.Slaga at us.didata.com  Fri Sep 29 10:37:10 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 29 Sep 2006 10:37:10 -0400
Subject: [cisco-voip] ciptug get together time/place suggestion
Message-ID: <C1FE15183DA37645BC0633BC604E44F0034E1968@USNAEXCH.na.didata.local>

I call the third seat on the left next to the network drop and the
lemons.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, September 29, 2006 9:50 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ciptug get together time/place suggestion

 

How more fitting a place can we find? Right in the Hyatt, a lounge
called "Networks" !

	*
http://phoenix.hyatt.com/hyatt/hotels/gallery/photos..jsp?hotelId=2199&s
tart=14
<http://phoenix.hyatt.com/hyatt/hotels/gallery/photos.jsp?hotelId=2199&s
tart=14> 
	*
http://phoenix.hyatt.com/hyatt/hotels/gallery/photos..jsp?hotelId=2199&s
tart=14
<http://phoenix.hyatt.com/hyatt/hotels/gallery/photos.jsp?hotelId=2199&s
tart=14> 
	*
http://phoenix.hyatt.com/hyatt/hotels/gallery/virtual-tour.jsp?icamp=pro
pVirtualTour and click on Networks

They have plenty of specials during the week too!

	*
http://www.thehyatteditor.com/images/editor/siteImages/images/hotels/phx
rp/Hyatt%20Web-Networks.pdf

Since last year it seemed like the first day was a bit too rushed, how
about we say Tuesday between 5pm and 6pm at Networks?

 

 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.     Network Analyst (CCS)
University of Guelph     Guelph, Ontario     N1G 2W1
(519) 824-4120 x56354     (519) 767-1060 FAX 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Sanity First : Number of days with less than
50 messages in my inbox at the end of the day:    buffer overrun




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

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From manoj.kalpage at gmail.com  Fri Sep 29 11:32:27 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Sat, 30 Sep 2006 00:32:27 +0900
Subject: [cisco-voip] DMVPN and QOS implementaion
Message-ID: <017f01c6e3dc$7c674210$6500a8c0@computec.lk>

Hi all,
We have hosted PBX system which is located in data centre and we have 
dedicated 1MB internet connection. At present we only have four remote sites 
and all of them have high speed ADSL connection for both their data and 
Voice. Each sites has 4 to 5 phones. We used to use PIX 515 at Data centre 
and PIX 501 at remote sites. As we were experiencing voice quality issue we 
moved to Cisco 2821 at Data centre and 800 series at remote sites. I have 
configured DMVPN using GRE over IPSec for our VPN network. all the tunnels 
are up and seems to be working fine so far but I am just wondering whether I 
got right QoS configuration at HeadEnd Router. As, I am a newbie for QoS, I 
have referred various cisco documentations to configure bellow DMVPN and QoS 
for our head end router. Since we don't have data transaction at HeadEnd 
site I think I can use 75% of bandwidth for voice it self.  Can some one 
help me figure out QoS requirment for my network environment.

Thank you in advanced,

Best regards
Manoj

---------------------------------------------------------------------------
Building configuration...

Current configuration : 5063 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname PBXLGATE01
!
boot-start-marker
boot-end-marker
!
logging buffered 51200 warnings
!
no aaa new-model
!
resource policy
!
ip cef

!
ip domain name yourdomain.com
!
!
crypto pki trustpoint TP-self-signed-2723000426
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-2723000426
 revocation-check none
 rsakeypair TP-self-signed-2723000426
!
!
crypto pki certificate chain TP-self-signed-2723000426
 certificate self-signed 01
  30820251 308201BA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
  31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
  quit
username xxxx privilege 15 secret 5 @#@#@@@GlPb96SyZxV6Q0
!
!
class-map match-all VOICE
 match ip dscp ef
class-map match-all SCAVENGER
 match ip dscp cs1
class-map match-any INTERNETWORK-CONTROL
 match ip dscp cs6
 match access-group name IKE
class-map match-any CALL-SIGNALING
 match ip dscp cs3
 match ip dscp af31
!
!
policy-map V3PN-EDGE
 class VOICE
  priority percent 55
 class CALL-SIGNALING
  bandwidth percent 5
 class INTERNETWORK-CONTROL
  bandwidth percent 5
 class SCAVENGER
  bandwidth percent 1
  queue-limit 1
 class class-default
  bandwidth percent 9
  queue-limit 16
!
!
crypto isakmp policy 10
 hash md5
 authentication pre-share
crypto isakmp key 6 G0G0G0G0 address 0.0.0.0 0.0.0.0
!
!
crypto ipsec transform-set PBXL esp-3des esp-md5-hmac
!
crypto ipsec profile PBXL
 set security-association lifetime seconds 120
 set transform-set PBXL
!
!
interface Tunnel0
 ip address 10.10.1.1 255.255.255.0
 no ip redirects
 ip mtu 1440
 ip nhrp authentication xxxxxxxx
 ip nhrp map multicast dynamic
 ip nhrp network-id 1
 ip tcp adjust-mss 1360
 qos pre-classify
 tunnel source FastEthernet0/0
 tunnel mode gre multipoint
 tunnel key 0
 tunnel protection ipsec profile PBXL
!

interface FastEthernet0/0
 description Connect to Verizon Network
 bandwidth 1000
 ip address 222.222.222.222 255.255.255.192
 ip nbar protocol-discovery
 ip nat outside
 ip virtual-reassembly
 duplex auto
 speed auto
 service-policy output V3PN-EDGE
!
interface FastEthernet0/1
 ip address 192.168.4.1 255.255.255.0
 ip nat inside
 ip virtual-reassembly
 duplex auto
 speed auto
!
router eigrp 90
 network 10.0.0.0
 network 172.16.0.0 0.0.0.255
 no auto-summary
!
ip route 0.0.0.0 0.0.0.0 222.222.222.222
!
!
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 1000
ip nat inside source list 1 interface FastEthernet0/0 overload
!
ip access-list extended IKE
 permit udp any eq isakmp any eq isakmp
!
access-list 1 permit 192.168.4.0 0.0.0.255
!
!
control-plane
!
!
line con 0
 login local
line aux 0
line vty 0 4
 access-class 23 in
 privilege level 15
 login local
 transport input ssh
!
scheduler allocate 20000 1000
end

PBXLGATE01# 


From duncanw at otc.edu  Fri Sep 29 11:50:46 2006
From: duncanw at otc.edu (DUNCAN, W B.)
Date: Fri, 29 Sep 2006 10:50:46 -0500
Subject: [cisco-voip] Missed Call Log
Message-ID: <06D1B6D4926222458F803D0D3EDCCB7EACE028@EXM1.otc.edu>

Is there some way to force the missed calls log to show only missed
calls for the primary line, rather than all lines on the phone?  I have
a number of users requesting this.  We are running 4.1(3) sr2.  

 

Thanks all,

Brian Duncan

Coordinator of Telecommunications

Ozarks Technical Community College

duncanw at otc.edu

 

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From mhardy at pacific.edu  Fri Sep 29 11:51:54 2006
From: mhardy at pacific.edu (Monica Hardy)
Date: Fri, 29 Sep 2006 08:51:54 -0700
Subject: [cisco-voip] Ciptug Conference?
In-Reply-To: <81F69EB4-829D-480A-937B-580B41EB3A24@mac.com>
References: <81F69EB4-829D-480A-937B-580B41EB3A24@mac.com>
Message-ID: <451CDEAA.936D.0050.0@pacific.edu>

I will be attending this year along with a co-worker.
 
 
 
-Monica

____________________________

Monica Hardy
University of the Pacific
Office of Information Technology
Network Engineer II - Telecom
209-946-7400 Helpdesk
209-946-3065 Direct

>>> Tim Medley <medley at mac.com> 9/28/2006 1:34 PM >>>
In anyone going to the CIPTUG Conference? It's coming up in just a
couple of weeks.

October 9 - 12 in sunny Phoenix Arizona.

We had an impromptu meetup last year by some of the folks on the list
here, anyone interested in doing that again this year?

tm


Tim Medley

Moxie Enterprises
Email: tim.medley at moxieent.com 
Phone: (704) 905-4604






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From charlieb at cot.net  Fri Sep 29 11:57:37 2006
From: charlieb at cot.net (Charles J. Boening)
Date: Fri, 29 Sep 2006 08:57:37 -0700
Subject: [cisco-voip] Callmanager version 5.0 Upgrade
Message-ID: <20DF280A12FBD1478CDCDEA366ACE077B5D8CD@apollo.calore.local>

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



Are you pasting the whole output from CCM5 or just the XML part?  Make sure you're pasting in just the XML part.  The directions say to copy all and paste but they lie.  :)

So, copy everything between and including the <LicenseUpgrade></LicenseUpgrade> tags only.

Good luck.

Charlie


 

> -----Original Message-----
> From: Thyri Alfred [mailto:alfred.thyri at mii.at] 
> Sent: Friday, September 29, 2006 1:17 AM
> To: cisco-voip at puck.nether.net
> Cc: Charles J. Boening
> Subject: RE: [cisco-voip] Callmanager version 5.0 Upgrade
> 
> Update:
> 
> Since we decided to make a complete new configuration of our 
> CCM (reason: our system grew over the years under different 
> admins...), I made a new, clean CCM install, imported our 
> phones and ran the DMA tool - without any troubles :-)
> 
> some hints for installing ccm5 with dma data:
> - don't change ip or hostname, setup will fail
> - copying the dam data by ftp: don't configure a ftp 
> password. the installer doesn't send the configured password.
> 
> After installing 5.0(4) and importing the dma data I tried to 
> activate the intermediate license on cco 
> (https://tools.cisco.com/SWIFT/Licensing/PrivateRegistrationSe
> rvlet - Cisco CallManager Version Upgrade) with no success.
> 
> I get get an error, because of an incorrect xml file.
> 
> regards,
> Alfred.
> 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Thyri Alfred
> Sent: Dienstag, 29. August 2006 22:00
> To: Charles J. Boening; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Callmanager version 5.0 Upgrade
> 
> Hi,
> 
> Thanks for your response Charlie!
> 
> We do have a single server, and i tried to install it on the 
> CCM itself.
> I already contacted our local cisco system engineers, next 
> step will be opening a tac case. I will do so more testing, 
> since i still have to wait for the delivery of our new ccm 
> hardware - so i still have some time :)
> 
> I will send an update after we fixed the problem...
> 
> So long,
> Alfred.
> 
> -----Urspr?ngliche Nachricht-----
> Von: Charles J. Boening [mailto:charlieb at cot.net]
> Gesendet: Dienstag, 29. August 2006 17:19
> An: Thyri Alfred; cisco-voip at puck.nether.net
> Betreff: RE: [cisco-voip] Callmanager version 5.0 Upgrade
> 
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
> 
> Alfred,
> 
> Your version should be supported by 5.0.4 DMA Tool.  I'd 
> suggest opening a TAC case.  Maybe try a reboot if you can 
> then install again.  If you have a publisher/subscriber, then 
> I think you're supposed to run the tool on the publisher.  
> Also, you have to install the DMA Tool on the CCM server 
> itself.  You can't install to a workstation or another server.
> 
> Good luck,
> Charlie
> 
> 
> > -----Original Message-----
> > From: Thyri Alfred [mailto:alfred.thyri at mii.at]
> > Sent: Tuesday, August 29, 2006 3:43 AM
> > To: Charles J. Boening; cisco-voip at puck.nether.net
> > Subject: RE: [cisco-voip] Callmanager version 5.0 Upgrade
> > 
> > Hi Charlie,
> > 
> > I tried to install the dma tool (5.0(4)) on our CCM 4.1(3)sr2 
> > installation with bad luck :-(
> > 
> > In the log file I found something interesting:
> > 13:54:15: DMAInstall.cpp: fnInitDMAInstallProperty:      Error: Init
> > SZCMVER from default because of invalid szCCMBuildVersion = 4.1(3)
> > 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         Function started
> > 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         
> > CCMInstalled = "False"
> > 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         Function ended
> > 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         Function started
> > 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         SZCMVER = "2"
> > 13:54:15: MSIUtils.cpp: fnMsiSetProperty:         Function ended
> > 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         Function started
> > 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         REINSTALL = ""
> > 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         Function ended
> > 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         Function started
> > 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         SZSETUPTYPE = ""
> > 13:54:15: MSIUtils.cpp: fnMsiGetProperty:         Function ended
> > 
> > It seems that our version is not supported by the dma tool, 
> although 
> > it should be supported.
> > 
> > I also tried to check with Upgrade Utility (version 
> 4.2(2)), with the 
> > result that everything is ok but the hardware should be 
> upgraded. We 
> > already ordered new hardware...
> > 
> > 
> > Do you have any hints for me?? Somebody else?
> > 
> > cheers,
> > Alfred.
> > 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Charles J.
> > Boening
> > Sent: Dienstag, 29. August 2006 05:40
> > To: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] Callmanager version 5.0 Upgrade
> > 
> > -----BEGIN PGP SIGNED MESSAGE-----
> > Hash: SHA1
> > 
> > Here's my experience with upgrading to CCM5.
> > 
> > I came from a CCM 4.0(2a) system running on an ICS7750 to CCM
> > 5.0(3) running on my own supplied IBM server.  Cisco documentation 
> > indicated my IBM server (8491G2U) was supported in this release.  I 
> > actually purchased it based on Cisco documentation and 
> specifications 
> > just for this application.
> > Turns out that the 8841G2U was supported by the installer but the 
> > 8491G2U wasn't.  The only difference between these two is the 
> > warranty.  One has a 1yr warranty and the one I had came with a 3yr 
> > on-site warranty.  Other than that, there are no hardware 
> differences.
> > TAC opened a bug case and was supposed to fix the documentation.  I 
> > think TAC said I'd have to wait for 5.0(4) which wouldn't 
> be available 
> > on DVD for a few more months or whatever it was.
> > Being the nosey person I am, I snooped around and found 
> where in the 
> > installer to change it to recognize my machine type.
> > I ended up making an ISO image and used a hex editor to change the 
> > file the installer reads so my machine type would be supported.  :) 
> > Please don't ask what file I changed as I'm sure Cisco 
> wouldn't want 
> > that information floating around.  I'm sure some of the more 
> > industrious users out there will be able to figure it out on their 
> > own.  For the rest,  I'm sure someone will post the information on 
> > some message board or somewhere at some point.  :)
> > 
> > During my data export from CCM 4.x, I ended up having some data 
> > inconsistencies in my 4.x database that caused export 
> errors.  A call 
> > to TAC and a few days later, they had found and fixed the 
> problems.  I 
> > was then able to successfully export out my old data using 
> DMA 5.0.4.
> > 
> > Importing of Windows CCM data is done during the 
> installation process.
> > In my playing around, I had successfully installed CCM5 without 
> > importing any data.  Once installed, I saw no provisions 
> for importing 
> > the data after the installation.
> > So, with exported data in hand (actually on a local FTP server), I 
> > went through the installer process and came up with some 
> errors when 
> > importing Windows DMA data.  I didn't see any documentation 
> indicating 
> > it, but it turns out the name of my new CCM had to match 
> the name of 
> > the machine I was coming from.  The IP address may have had 
> to match 
> > too but I'm not sure.  Per TAC request, I set both the machine name 
> > and IP address so they were the same as my current 
> production system.
> > This wasn't a problem since I was building the new server 
> on it's own 
> > VLAN for testing.  Once this was done, the install went very smooth 
> > and all my data was imported.  I was actually very 
> impressed with the 
> > installer and it's ease of use.  Not that there's much to 
> it, but it 
> > was all pretty easy and straight forward.
> > 
> > All in all I'd say the installation and upgrade went pretty 
> well.  I 
> > was a little disappointed with a couple of TAC people and their 
> > slowness to respond.  At one point I waited over a week for a 
> > response.  Fortunately I had enough other duties to keep me busy in 
> > the meantime.  I think overall I spent about two to three weeks off 
> > and on between messing with things on my own and waiting on TAC.
> > 
> > Post upgrade everything is running very well.  My Unity 
> integration, 
> > Attendant Console and everything else came over just fine.  I also 
> > have a SIP trunk to an Asterisk box that came right up and 
> is running 
> > fine as well.  I'm using MGCP on a 1760V router to two 
> PRIs.  I have 
> > no FXO or FXS lines.
> > We're actually a small rural telephone company so I have 
> plenty of CO 
> > lines available for fax machines and such.  :)
> > 
> > The only issue I've had so far is with shared line appearances.  In 
> > the old system I had 3 extensions shared across seven 
> phones.  I had 
> > call forward busy from one extension to the other with the last 
> > extension going to voicemail on busy.  All extensions go to 
> voicemail 
> > on no answer.  After the conversion the forwarding wasn't working 
> > right.  If someone was on the first line, callers would go right to 
> > voicemail or get a fast busy; I don't remember which.  In the end I 
> > think I had to mess with the number of concurrent calls on the line.
> > I increased the max calls for each phone on all extensions to the 
> > number of shared lines
> > + 1.  I did this on both 7940 and 7960 phones that shared the
> > extensions.  I set the busy call trigger to 1 and 
> everything seems to 
> > be working ok.  In CCM5, there seems to be more of a distinction 
> > between lines and phones.  You can edit lines directly without even 
> > going through the phone config.  A little confusing at 
> first but not 
> > bad once you figure it all out.
> > 
> > One more issue I have is doing the 5.0(4a) update.  I'm currently at
> > 5.0(4) but it seems that updates are now required to be 
> signed and the 
> > update posted on CCO isn't.  Oh well, another call to TAC I 
> guess.  :)
> > 
> > We've been live for about two weeks now.  Everything has 
> been running 
> > very smooth and so far I'm pleased with CCM5.
> > 
> > Charlie
> > 
> > 
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> > Manoj Kalpage
> > > Sent: Monday, August 28, 2006 7:04 PM
> > > To: Matt Slaga (US)
> > > Cc: cisco-voip at puck.nether.net
> > > Subject: Re: [cisco-voip] Callmanager version 5.0 Upgrade
> > > 
> > > Thank you very much for the reply as always.
> > > What is your recommendation to upgrade to Version 5.0? Does
> > this work
> > > smoothly? Anyone had any bad experience of upgrading to 5.0?
> > >  
> > > Best Regards,
> > > Manoj
> > > 
> > >  
> > > On 8/28/06, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote: 
> > > 
> > > 	Manoj-
> > > 
> > > 	 
> > > 
> > > 	CallManager 5.0 only runs on Linux.  While you can upgrade from
> > 4.0
> > > or 4.1 to 5.0, it requires a complete rebuild and restore
> > process to
> > > migrate the data from SQL to Informix under the new linux 
> platform.
> > > There is no 5.0 that runs on win2k.
> > > 
> > > 	 
> > > 
> > > 	 
> > > 
> > > 	
> > > ________________________________
> > > 
> > > 
> > > 	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
> > Manoj Kalpage
> > > 	Sent: Sunday, August 27, 2006 10:04 PM
> > > 
> > > 	
> > > 	To: cisco-voip at puck.nether.net
> > > 	
> > > 	Subject: [cisco-voip] Callmanager version 5.0 Upgrade
> > > 
> > > 	
> > > 
> > > 	 
> > > 
> > > 	Hi All,
> > > 
> > > 	Please someone help me if I am wrong here.
> > > 
> > > 	My Understanding is CallManager 4.0 can be upgraded to
> > Callamanger
> > > 5.x keeping existing windows 2000 OS but new installation
> > only work on
> > > Linux.
> > > 
> > > 	 
> > > 
> > > 	Regards,
> > > 
> > > 	Manoj
> > > 
> > > 	 
> > > 
> > > 	
> > > ________________________________
> > > 
> > > 
> > > 	
> > > 
> > > 
> > > 	
> > > 	Disclaimer:
> > > 	
> > > 	This e-mail communication and any attachments may contain
> > > 	confidential and privileged information and is for use by the
> > > 	designated addressee(s) named above only. If you are not the 
> > > 	intended addressee, you are hereby notified that you
> > have received
> > > 	this communication in error and that any use or reproduction of
> > > 	this email or its contents is strictly prohibited and may be
> > > 	unlawful. If you have received this communication in
> > error, please
> > > 	notify us immediately by replying to this message and
> > deleting it
> > > 	from your computer. Thank you.
> > > 	
> > > 
> > > 
> > > 
> > -----BEGIN PGP SIGNATURE-----
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> > 
> > iD8DBQFE87cBcGGHuFdGSWARAh5ZAJ4oAgw7f/rYe7uO1ERMvUv9JS48EgCfc/rQ
> > TcATbYEo8nOCBavtsYcvzpU=
> > =qoKW
> > -----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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> mMbEx/UzDLZDZXGRSkWbt/E=
> =NsnH
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> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
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From Scott.Voll at wesd.org  Fri Sep 29 12:14:17 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 29 Sep 2006 09:14:17 -0700
Subject: [cisco-voip] Issues with some calls into unity after userenters
	*
Message-ID: <407055A92CECCB499C922A2D35FC19A606B4422D@Apollo.wesd.org>

We have seen this issue also and it is a DTMF issue in our case.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Aman Chugh
Sent: Friday, September 29, 2006 12:44 AM
To: Nathan Reeves
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Issues with some calls into unity after
userenters *

 

Might be a dtmf issue with this particular gateway or audio cut through
not happening,what protocol r you running on this gateway and what is
the interface to the pstn ,run a appropirate debug to check the
disconnect cause,also any thing getting generated on unity app log ,Also
run unity port status monitor for this call to  see if this dtmf is
coming through to unity or not. 

 

Aman

 

On 9/29/06, Nathan Reeves <nathan.a.reeves at gmail.com> wrote: 

Seeing an issue where calls are passed over to Unity and the greeting
message is played.  When the users presses * to drop to the sign in 
process the call gets dropped.

I've checked the call handler, and all looks good.  When calling the
unity pilot from a phone internally, I can press *.  When calling in
from some gateways, pressing * works.  But from one gateway it 
doesn't.

Anyone got any ideas where to start troubleshooting?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

 

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From MLinsemier at apcapital.com  Fri Sep 29 12:19:51 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Fri, 29 Sep 2006 12:19:51 -0400
Subject: [cisco-voip] DMVPN and QOS implementaion
In-Reply-To: <017f01c6e3dc$7c674210$6500a8c0@computec.lk>
References: <017f01c6e3dc$7c674210$6500a8c0@computec.lk>
Message-ID: <AB0A35F2486F81468971AA68558BE71904F04427@smieln01winm01.micoanet.com>

Manoj,

Here is our head-end configuration that we use.  We have a 6MB Internet
link at this time which is also shared among browsing.  We are using a
PacketShaper to prioritize the IPSec traffic above others in the pipe
and using QoS in the routers to prioritize what's in the tunnel.  If
anyone has suggestions for my configuration as well, please advise.

class-map match-any Voice
 match ip dscp ef 
class-map match-any Control
 match ip dscp cs3 
 match ip dscp af31

policy-map DMVPN_QoS
 class Voice
  priority percent 70
 class Control
  bandwidth percent 5
 class class-default
  fair-queue
  random-detect
policy-map Shape-6M
 class class-default
  shape average 6144000
  service-policy DMVPN_QoS

interface GigabitEthernet0/1
bandwidth 6144
 ip address X.X.X.X X.X.X.X
 ip access-group 101 in
 no ip redirects
 no ip unreachables
 no ip proxy-arp
 ip nbar protocol-discovery
 ip flow ingress
 ip flow egress
 ip nat outside
 ip inspect IOS_FW in
 ip virtual-reassembly
 ip route-cache flow
 duplex auto
 speed auto
 no mop enabled
 service-policy output Shape-6M

Matt


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Manoj Kalpage
Sent: Friday, September 29, 2006 11:32 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] DMVPN and QOS implementaion

Hi all,
We have hosted PBX system which is located in data centre and we have
dedicated 1MB internet connection. At present we only have four remote
sites and all of them have high speed ADSL connection for both their
data and Voice. Each sites has 4 to 5 phones. We used to use PIX 515 at
Data centre and PIX 501 at remote sites. As we were experiencing voice
quality issue we moved to Cisco 2821 at Data centre and 800 series at
remote sites. I have configured DMVPN using GRE over IPSec for our VPN
network. all the tunnels are up and seems to be working fine so far but
I am just wondering whether I got right QoS configuration at HeadEnd
Router. As, I am a newbie for QoS, I have referred various cisco
documentations to configure bellow DMVPN and QoS for our head end
router. Since we don't have data transaction at HeadEnd site I think I
can use 75% of bandwidth for voice it self.  Can some one help me figure
out QoS requirment for my network environment.

Thank you in advanced,

Best regards
Manoj

------------------------------------------------------------------------
---
Building configuration...

Current configuration : 5063 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname PBXLGATE01
!
boot-start-marker
boot-end-marker
!
logging buffered 51200 warnings
!
no aaa new-model
!
resource policy
!
ip cef

!
ip domain name yourdomain.com
!
!
crypto pki trustpoint TP-self-signed-2723000426  enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2723000426
 revocation-check none
 rsakeypair TP-self-signed-2723000426
!
!
crypto pki certificate chain TP-self-signed-2723000426  certificate
self-signed 01
  30820251 308201BA A0030201 02020101 300D0609 2A864886 F70D0101
04050030
  31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D
43657274
  quit
username xxxx privilege 15 secret 5 @#@#@@@GlPb96SyZxV6Q0 !
!
class-map match-all VOICE
 match ip dscp ef
class-map match-all SCAVENGER
 match ip dscp cs1
class-map match-any INTERNETWORK-CONTROL  match ip dscp cs6  match
access-group name IKE class-map match-any CALL-SIGNALING  match ip dscp
cs3  match ip dscp af31 !
!
policy-map V3PN-EDGE
 class VOICE
  priority percent 55
 class CALL-SIGNALING
  bandwidth percent 5
 class INTERNETWORK-CONTROL
  bandwidth percent 5
 class SCAVENGER
  bandwidth percent 1
  queue-limit 1
 class class-default
  bandwidth percent 9
  queue-limit 16
!
!
crypto isakmp policy 10
 hash md5
 authentication pre-share
crypto isakmp key 6 G0G0G0G0 address 0.0.0.0 0.0.0.0 !
!
crypto ipsec transform-set PBXL esp-3des esp-md5-hmac !
crypto ipsec profile PBXL
 set security-association lifetime seconds 120  set transform-set PBXL !
!
interface Tunnel0
 ip address 10.10.1.1 255.255.255.0
 no ip redirects
 ip mtu 1440
 ip nhrp authentication xxxxxxxx
 ip nhrp map multicast dynamic
 ip nhrp network-id 1
 ip tcp adjust-mss 1360
 qos pre-classify
 tunnel source FastEthernet0/0
 tunnel mode gre multipoint
 tunnel key 0
 tunnel protection ipsec profile PBXL
!

interface FastEthernet0/0
 description Connect to Verizon Network
 bandwidth 1000
 ip address 222.222.222.222 255.255.255.192  ip nbar protocol-discovery
ip nat outside  ip virtual-reassembly  duplex auto  speed auto
service-policy output V3PN-EDGE !
interface FastEthernet0/1
 ip address 192.168.4.1 255.255.255.0
 ip nat inside
 ip virtual-reassembly
 duplex auto
 speed auto
!
router eigrp 90
 network 10.0.0.0
 network 172.16.0.0 0.0.0.255
 no auto-summary
!
ip route 0.0.0.0 0.0.0.0 222.222.222.222 !
!
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 1000 ip nat inside
source list 1 interface FastEthernet0/0 overload !
ip access-list extended IKE
 permit udp any eq isakmp any eq isakmp
!
access-list 1 permit 192.168.4.0 0.0.0.255 !
!
control-plane
!
!
line con 0
 login local
line aux 0
line vty 0 4
 access-class 23 in
 privilege level 15
 login local
 transport input ssh
!
scheduler allocate 20000 1000
end

PBXLGATE01# 

CONFIDENTIALITY STATEMENT
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be protected by one or more legal privileges. It is intended
solely for the use of the addressee identified above. If you
are not the intended recipient, any use, disclosure, copying
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this information block, the typed name of the sender, nor
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From medley at mac.com  Fri Sep 29 12:23:00 2006
From: medley at mac.com (Tim Medley)
Date: 29 Sep 2006 09:23:00 -0700
Subject: [cisco-voip] Ciptug Conference?
Message-ID: <3242366604.2412693@smtp.mac.com>

Yes, but we have an MC this year so I do not have to be everywhere.

Tm

-----Original Message-----
From: "Matt Slaga (US)" <Matt.Slaga at us.didata.com>
Date: Friday, Sep 29, 2006 5:07 am
Subject: RE: [cisco-voip] Ciptug Conference?

              
I'll be there as well.  Are you runningthe show again? 
  
   
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Medley
Sent: Thursday, September 28, 20064:34 PM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] CiptugConference?


From medley at mac.com  Fri Sep 29 12:25:00 2006
From: medley at mac.com (Tim Medley)
Date: 29 Sep 2006 09:25:00 -0700
Subject: [cisco-voip] Ciptug Conference?
Message-ID: <3242366706.2422953@smtp.mac.com>

I'll be in Phoenix the thursday before the conference until the Friday after.

Tm
-----Original Message-----
From: Lelio Fulgenzi <lelio at uoguelph.ca>
Date: Friday, Sep 29, 2006 6:20 am
Subject: Re: [cisco-voip] Ciptug Conference?

        
Also, if people tell me/Tim what time you're showing up, we can plan things so as many people can attend. Perhaps something just before one of the CIPTUG events. 
     ----- Original Message -----    From:   Lelio Fulgenzi      To: Matt Slaga (US) ; Tim Medley ; cisco-voip at puck-nether.net    Sent: Friday, September 29, 2006 9:19   AM   Subject: Re: [cisco-voip] Ciptug   Conference?   

   
I asked a while ago and got a few responses. I   think another get together would be great. Is anyone familiar with the Pheonix   area that could suggest something? I have a friend who lived there and might   ask him if there is a nice place we can meet. I'll also look up the hotel and   see if there is a bar in there. Might not be as nice as the Vegas one, but   what the heck....   
    


From medley at mac.com  Fri Sep 29 12:25:00 2006
From: medley at mac.com (Tim Medley)
Date: 29 Sep 2006 09:25:00 -0700
Subject: [cisco-voip] ciptug get together time/place suggestion
Message-ID: <3242366748.2427144@smtp.mac.com>

Lelio, that's a great find! 

-----Original Message-----
From: Lelio Fulgenzi <lelio at uoguelph.ca>
Date: Friday, Sep 29, 2006 6:52 am
Subject: [cisco-voip] ciptug get together time/place suggestion

This is a multi-part message in MIME format.

------


From SPackett at fenwick.com  Fri Sep 29 12:27:09 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Fri, 29 Sep 2006 09:27:09 -0700
Subject: [cisco-voip] Ciptug Conference?
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01D71398@SVCEXCH01.firm.fenwick.llp>

I'll be there, so count me in.  My flight arrives 6:00 PM PST.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, September 29, 2006 6:20 AM
To: Lelio Fulgenzi; Matt Slaga (US); Tim Medley;
cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] Ciptug Conference?


Also, if people tell me/Tim what time you're showing up, we can plan
things so as many people can attend. Perhaps something just before one
of the CIPTUG events.
 

	----- Original Message ----- 
	From: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  
	To: Matt Slaga (US) <mailto:Matt.Slaga at us.didata.com>  ; Tim
Medley <mailto:medley at mac.com>  ; cisco-voip at puck-nether.net 
	Sent: Friday, September 29, 2006 9:19 AM
	Subject: Re: [cisco-voip] Ciptug Conference?

	I asked a while ago and got a few responses. I think another get
together would be great. Is anyone familiar with the Pheonix area that
could suggest something? I have a friend who lived there and might ask
him if there is a nice place we can meet. I'll also look up the hotel
and see if there is a bar in there. Might not be as nice as the Vegas
one, but what the heck....
	 

		----- Original Message ----- 
		From: Matt Slaga (US) <mailto:Matt.Slaga at us.didata.com>

		To: Tim Medley <mailto:medley at mac.com>  ;
cisco-voip at puck-nether.net 
		Sent: Friday, September 29, 2006 8:06 AM
		Subject: Re: [cisco-voip] Ciptug Conference?


		I'll be there as well.  Are you running the show again?

		 

		
________________________________


		From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Medley
		Sent: Thursday, September 28, 2006 4:34 PM
		To: cisco-voip at puck-nether.net
		Subject: [cisco-voip] Ciptug Conference?

		 

		In anyone going to the CIPTUG Conference? It's coming up
in just a couple of weeks.

		 

		October 9 - 12 in sunny Phoenix Arizona.

		 

		We had an impromptu meetup last year by some of the
folks on the list here, anyone interested in doing that again this year?

		 

		tm

		 

		

		Tim Medley

		Moxie Enterprises

		Email: tim.medley at moxieent.com

		Phone: (704) 905-4604

		 

		 

		
		
		

		 

		
________________________________


		

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

		
________________________________


		

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

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From SPackett at fenwick.com  Fri Sep 29 12:32:58 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Fri, 29 Sep 2006 09:32:58 -0700
Subject: [cisco-voip] ciptug get together time/place suggestion
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01D7139A@SVCEXCH01.firm.fenwick.llp>

Perfect.  Count me in.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, September 29, 2006 6:50 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ciptug get together time/place suggestion


How more fitting a place can we find? Right in the Hyatt, a lounge
called "Networks" !

	*
http://phoenix.hyatt.com/hyatt/hotels/gallery/photos.jsp?hotelId=2199&st
art=14 
	*
http://phoenix.hyatt.com/hyatt/hotels/gallery/photos.jsp?hotelId=2199&st
art=14 
	*
http://phoenix.hyatt.com/hyatt/hotels/gallery/virtual-tour.jsp?icamp=pro
pVirtualTour and click on Networks

They have plenty of specials during the week too!

	*
http://www.thehyatteditor.com/images/editor/siteImages/images/hotels/phx
rp/Hyatt%20Web-Networks.pdf

Since last year it seemed like the first day was a bit too rushed, how
about we say Tuesday between 5pm and 6pm at Networks?
 
 
 
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.     Network Analyst (CCS)
University of Guelph     Guelph, Ontario     N1G 2W1
(519) 824-4120 x56354     (519) 767-1060 FAX 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Sanity First : Number of days with less than
50 messages in my inbox at the end of the day:    buffer overrun
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From aman.chugh at gmail.com  Fri Sep 29 12:46:49 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Fri, 29 Sep 2006 22:16:49 +0530
Subject: [cisco-voip] DMVPN and QOS implementaion
In-Reply-To: <017f01c6e3dc$7c674210$6500a8c0@computec.lk>
References: <017f01c6e3dc$7c674210$6500a8c0@computec.lk>
Message-ID: <d4c697ad0609290946i10d76ff2h9aa8ee31d4fca931@mail.gmail.com>

 Manoj,

Your Qos configuration looks ok,I had similar Issues V3PN setup ,some of the
things you need to consider for this kind of setup is

1.Use hardware encryption for VPN as software encryption adds unneccassry
delays
2.Check the Voice codec for your inter site calls and during  a bad
 call press the ? key on the phone to check MAX jitter and RXLOST values.
3. Do sh policy-map interface to check for drops ,if so see drops change
your priority queue bandwitdhs
3.There is a service parrametter in callmanager to remove
G.729ab,G.729bfrom cabablities when calls are established,I had
considerable improvement
in voice quality after changing this parrametter to true as by default its
false.I could not use g.729 for voice calls as voice sounded very choppy
,once i applied this change i am able to use g.729 for calls.

Hope this helps
Aman


On 9/29/06, Manoj Kalpage <manoj.kalpage at gmail.com> wrote:
>
> Hi all,
> We have hosted PBX system which is located in data centre and we have
> dedicated 1MB internet connection. At present we only have four remote
> sites
> and all of them have high speed ADSL connection for both their data and
> Voice. Each sites has 4 to 5 phones. We used to use PIX 515 at Data centre
> and PIX 501 at remote sites. As we were experiencing voice quality issue
> we
> moved to Cisco 2821 at Data centre and 800 series at remote sites. I have
> configured DMVPN using GRE over IPSec for our VPN network. all the tunnels
> are up and seems to be working fine so far but I am just wondering whether
> I
> got right QoS configuration at HeadEnd Router. As, I am a newbie for QoS,
> I
> have referred various cisco documentations to configure bellow DMVPN and
> QoS
> for our head end router. Since we don't have data transaction at HeadEnd
> site I think I can use 75% of bandwidth for voice it self.  Can some one
> help me figure out QoS requirment for my network environment.
>
> Thank you in advanced,
>
> Best regards
> Manoj
>
>
> ---------------------------------------------------------------------------
> Building configuration...
>
> Current configuration : 5063 bytes
> !
> version 12.4
> service timestamps debug datetime msec
> service timestamps log datetime msec
> no service password-encryption
> !
> hostname PBXLGATE01
> !
> boot-start-marker
> boot-end-marker
> !
> logging buffered 51200 warnings
> !
> no aaa new-model
> !
> resource policy
> !
> ip cef
>
> !
> ip domain name yourdomain.com
> !
> !
> crypto pki trustpoint TP-self-signed-2723000426
> enrollment selfsigned
> subject-name cn=IOS-Self-Signed-Certificate-2723000426
> revocation-check none
> rsakeypair TP-self-signed-2723000426
> !
> !
> crypto pki certificate chain TP-self-signed-2723000426
> certificate self-signed 01
> 30820251 308201BA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
> 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
> quit
> username xxxx privilege 15 secret 5 @#@#@@@GlPb96SyZxV6Q0
> !
> !
> class-map match-all VOICE
> match ip dscp ef
> class-map match-all SCAVENGER
> match ip dscp cs1
> class-map match-any INTERNETWORK-CONTROL
> match ip dscp cs6
> match access-group name IKE
> class-map match-any CALL-SIGNALING
> match ip dscp cs3
> match ip dscp af31
> !
> !
> policy-map V3PN-EDGE
> class VOICE
> priority percent 55
> class CALL-SIGNALING
> bandwidth percent 5
> class INTERNETWORK-CONTROL
> bandwidth percent 5
> class SCAVENGER
> bandwidth percent 1
> queue-limit 1
> class class-default
> bandwidth percent 9
> queue-limit 16
> !
> !
> crypto isakmp policy 10
> hash md5
> authentication pre-share
> crypto isakmp key 6 G0G0G0G0 address 0.0.0.0 0.0.0.0
> !
> !
> crypto ipsec transform-set PBXL esp-3des esp-md5-hmac
> !
> crypto ipsec profile PBXL
> set security-association lifetime seconds 120
> set transform-set PBXL
> !
> !
> interface Tunnel0
> ip address 10.10.1.1 255.255.255.0
> no ip redirects
> ip mtu 1440
> ip nhrp authentication xxxxxxxx
> ip nhrp map multicast dynamic
> ip nhrp network-id 1
> ip tcp adjust-mss 1360
> qos pre-classify
> tunnel source FastEthernet0/0
> tunnel mode gre multipoint
> tunnel key 0
> tunnel protection ipsec profile PBXL
> !
>
> interface FastEthernet0/0
> description Connect to Verizon Network
> bandwidth 1000
> ip address 222.222.222.222 255.255.255.192
> ip nbar protocol-discovery
> ip nat outside
> ip virtual-reassembly
> duplex auto
> speed auto
> service-policy output V3PN-EDGE
> !
> interface FastEthernet0/1
> ip address 192.168.4.1 255.255.255.0
> ip nat inside
> ip virtual-reassembly
> duplex auto
> speed auto
> !
> router eigrp 90
> network 10.0.0.0
> network 172.16.0.0 0.0.0.255
> no auto-summary
> !
> ip route 0.0.0.0 0.0.0.0 222.222.222.222
> !
> !
> ip http server
> ip http authentication local
> ip http secure-server
> ip http timeout-policy idle 600 life 86400 requests 1000
> ip nat inside source list 1 interface FastEthernet0/0 overload
> !
> ip access-list extended IKE
> permit udp any eq isakmp any eq isakmp
> !
> access-list 1 permit 192.168.4.0 0.0.0.255
> !
> !
> control-plane
> !
> !
> line con 0
> login local
> line aux 0
> line vty 0 4
> access-class 23 in
> privilege level 15
> login local
> transport input ssh
> !
> scheduler allocate 20000 1000
> end
>
> PBXLGATE01#
>
>
>
> _______________________________________________
> 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 Sep 29 12:58:56 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 29 Sep 2006 09:58:56 -0700
Subject: [cisco-voip] Long delay
Message-ID: <407055A92CECCB499C922A2D35FC19A606B4424D@Apollo.wesd.org>

I'm running cm 4.1.3sr2 and I have a remote site complaining of long
delays in transfers and answering.  Does anyone have any idea where to
start with this?

 

I have not been out to the remote site to personally check.

 

Scott

 

 

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From JHorton at jackhenry.com  Fri Sep 29 13:16:35 2006
From: JHorton at jackhenry.com (Josh Horton)
Date: Fri, 29 Sep 2006 12:16:35 -0500
Subject: [cisco-voip] DND issues - CME 4.0(3) - 7961 and 7941
Message-ID: <2B3745BC9D31514381A6FCC5E72D93731D277F@SMOMAIL.jhacorp.com>


I am having a problem with "DND" not going away after the dnd softkey is
pressed a second time. It will eventually go away, but it takes 3-5
minutes.

CME Version: 4.0(3)
Phones: 7961 and 7941
Protocol: SCCP
Phone Firmware: SCCP41.8-0-4SR1S

Has anyone else had any experience with this?

--
Josh Horton
Network Engineer
Matrix Network Services (R) 
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.


From CORTIZ at broward.org  Fri Sep 29 13:42:49 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 29 Sep 2006 13:42:49 -0400
Subject: [cisco-voip] ciptug get together time/place suggestion
In-Reply-To: <01C8F21F30DE9740B53FD2EF05C3A8FC01D7139A@SVCEXCH01.firm.fenwick.llp>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A47631@MREXVS01.bc.broward.cty>

I should be able to make it.  I'll be bringing a co-worker this year.

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Stu Packett
Sent: Friday, September 29, 2006 12:33 PM
To: Lelio Fulgenzi; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ciptug get together time/place suggestion

 

Perfect.  Count me in.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, September 29, 2006 6:50 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] ciptug get together time/place suggestion

How more fitting a place can we find? Right in the Hyatt, a lounge
called "Networks" !

	*
http://phoenix.hyatt.com/hyatt/hotels/gallery/photos.jsp?hotelId=2199&st
art=14 
	*
http://phoenix.hyatt.com/hyatt/hotels/gallery/photos.jsp?hotelId=2199&st
art=14 
	*
http://phoenix.hyatt.com/hyatt/hotels/gallery/virtual-tour.jsp?icamp=pro
pVirtualTour and click on Networks

They have plenty of specials during the week too!

	*
http://www.thehyatteditor.com/images/editor/siteImages/images/hotels/phx
rp/Hyatt%20Web-Networks.pdf

Since last year it seemed like the first day was a bit too rushed, how
about we say Tuesday between 5pm and 6pm at Networks?

 

 

 

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.     Network Analyst (CCS)
University of Guelph     Guelph, Ontario     N1G 2W1
(519) 824-4120 x56354     (519) 767-1060 FAX 
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 
Sanity First : Number of days with less than
50 messages in my inbox at the end of the day:    buffer overrun

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From CORTIZ at broward.org  Fri Sep 29 13:45:45 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 29 Sep 2006 13:45:45 -0400
Subject: [cisco-voip] Long delay
In-Reply-To: <407055A92CECCB499C922A2D35FC19A606B4424D@Apollo.wesd.org>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03A47632@MREXVS01.bc.broward.cty>

How heavy is the volume of calls at the site?  I have a site with 2
PRI's (about 160 users) that was getting inundated with calls yesterday.
They claim that they were having slow downs transferring calls when the
traffic was heavy. Not sure what would cause this but wondering if your
location was similar.

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, September 29, 2006 12:59 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Long delay

 

I'm running cm 4.1.3sr2 and I have a remote site complaining of long
delays in transfers and answering.  Does anyone have any idea where to
start with this?

 

I have not been out to the remote site to personally check.

 

Scott

 

 

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From Scott.Voll at wesd.org  Fri Sep 29 14:28:02 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 29 Sep 2006 11:28:02 -0700
Subject: [cisco-voip] Unity Tech Admin
Message-ID: <407055A92CECCB499C922A2D35FC19A606B442A4@Apollo.wesd.org>

I have two Unity servers both running unity 4.0.5.  one connected to my
CM 4.1.3sr2 and one to my old PBX.  Both are digitally networked and
working fine.

 

I have made myself an admin on the CM Unity box and then had to so some
utility to make myself an admin on the PBX Unity server.

 

I have now made a COS on the CM Unity server for our Helpdesk to
add/change/delete subscribers but now how do I do that on the PBX Unity
server?

 

Scott

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From keith.klevenski at rig.net  Fri Sep 29 14:57:03 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Fri, 29 Sep 2006 13:57:03 -0500
Subject: [cisco-voip] Long delay
Message-ID: <7F0CE0A1A0961E498A4132953534226662E1FD@HOUEXH02.rignet.org>

Are you having slow softkey response and delayed dial tone as well or is
this specifically related only to transfers?  I know in the past I've
had slow softkey response and delayed dial tone due to bandwidth
problems between the subscriber and publisher as they separated by a WAN
link.  The lack of dedicated bandwidth between the sub and pub caused
all kinds of weird problems like this.  Even with a local sub at the
site they were experiencing slow softkey response and delayed dial tone.

 

Keith 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, September 29, 2006 11:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Long delay

 

I'm running cm 4.1.3sr2 and I have a remote site complaining of long
delays in transfers and answering.  Does anyone have any idea where to
start with this?

 

I have not been out to the remote site to personally check.

 

Scott

 

 

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From WOCAMPO at broward.org  Fri Sep 29 15:02:48 2006
From: WOCAMPO at broward.org (Ocampo, Walter)
Date: Fri, 29 Sep 2006 15:02:48 -0400
Subject: [cisco-voip] Long delay
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E03A47632@MREXVS01.bc.broward.cty>
Message-ID: <F4DD4283A3EF194B84D774C60FB7F78601845816@UVEXVS01.bc.broward.cty>

Also, it will good to take a look a CDR and run a statistics for the
remote site.

 

Regards,

 

Walter Ocampo

System Network Analyst

Office of Information Technology 

540 S.E 3rd. Avenue, suite 300

Fort lauderdale, FL 33301

(954) 357-8041 Cell (954) 632 8228

wocampo at broward.org

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: Friday, September 29, 2006 1:46 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Long delay

 

How heavy is the volume of calls at the site?  I have a site with 2
PRI's (about 160 users) that was getting inundated with calls yesterday.
They claim that they were having slow downs transferring calls when the
traffic was heavy. Not sure what would cause this but wondering if your
location was similar.

 

Carlos

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, September 29, 2006 12:59 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Long delay

 

I'm running cm 4.1.3sr2 and I have a remote site complaining of long
delays in transfers and answering.  Does anyone have any idea where to
start with this?

 

I have not been out to the remote site to personally check.

 

Scott

 

 

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From aman.chugh at gmail.com  Fri Sep 29 15:30:50 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Sat, 30 Sep 2006 01:00:50 +0530
Subject: [cisco-voip] Long delay
In-Reply-To: <F4DD4283A3EF194B84D774C60FB7F78601845816@UVEXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E03A47632@MREXVS01.bc.broward.cty>
	<F4DD4283A3EF194B84D774C60FB7F78601845816@UVEXVS01.bc.broward.cty>
Message-ID: <d4c697ad0609291230r12623f3eoe4018e445ac451c7@mail.gmail.com>

Do the site experiencing heavy delays also see lot of ipphone
unregistrations .

Aman


On 9/30/06, Ocampo, Walter <WOCAMPO at broward.org> wrote:
>
>  Also, it will good to take a look a CDR and run a statistics for the
> remote site.
>
>
>
> Regards,
>
>
>
> Walter Ocampo
>
> System Network Analyst
>
> Office of Information Technology
>
> 540 S.E 3rd. Avenue, suite 300
>
> Fort lauderdale, FL 33301
>
> (954) 357-8041 Cell (954) 632 8228
>
> wocampo at broward.org
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Ortiz, Carlos
> *Sent:* Friday, September 29, 2006 1:46 PM
> *To:* Voll, Scott; cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Long delay
>
>
>
> How heavy is the volume of calls at the site?  I have a site with 2 PRI's
> (about 160 users) that was getting inundated with calls yesterday.  They
> claim that they were having slow downs transferring calls when the traffic
> was heavy. Not sure what would cause this but wondering if your location was
> similar.
>
>
>
> Carlos
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Voll, Scott
> *Sent:* Friday, September 29, 2006 12:59 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Long delay
>
>
>
> I'm running cm 4.1.3sr2 and I have a remote site complaining of long
> delays in transfers and answering.  Does anyone have any idea where to start
> with this?
>
>
>
> I have not been out to the remote site to personally check.
>
>
>
> Scott
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From msmith at laurinburg.org  Fri Sep 29 15:50:16 2006
From: msmith at laurinburg.org (Marcus Smith)
Date: Fri, 29 Sep 2006 15:50:16 -0400
Subject: [cisco-voip] MWI
Message-ID: <001e01c6e400$7c71bf00$3d00a8c0@laurinburg.org>

I have a little problem that will not go away.  A user phone MWI does not light up when there is a message for her.  I have checked everything to resolve this issue, but it is still there.  Even deleted the extension and recreated it, but still no luck, also, checked her profile sheet in CCM, everything is good.  what else can I check to resolve this issue.  Yes, the light does work  when incoming calls are received.

Marcus
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From keith.klevenski at rig.net  Fri Sep 29 15:54:57 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Fri, 29 Sep 2006 14:54:57 -0500
Subject: [cisco-voip] MWI
Message-ID: <7F0CE0A1A0961E498A4132953534226662E200@HOUEXH02.rignet.org>

Can you dial the MWI On extension from the phone and turn it on?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Marcus Smith
Sent: Friday, September 29, 2006 2:50 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MWI

 

I have a little problem that will not go away.  A user phone MWI does
not light up when there is a message for her.  I have checked everything
to resolve this issue, but it is still there.  Even deleted the
extension and recreated it, but still no luck, also, checked her profile
sheet in CCM, everything is good.  what else can I check to resolve this
issue.  Yes, the light does work  when incoming calls are received.

 

Marcus

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From aman.chugh at gmail.com  Fri Sep 29 15:56:56 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Sat, 30 Sep 2006 01:26:56 +0530
Subject: [cisco-voip] MWI
In-Reply-To: <7F0CE0A1A0961E498A4132953534226662E200@HOUEXH02.rignet.org>
References: <7F0CE0A1A0961E498A4132953534226662E200@HOUEXH02.rignet.org>
Message-ID: <d4c697ad0609291256s12dd00ev2c8409a10efec49e@mail.gmail.com>

Also,whats the status of MWI on that subscriber's page when you leave
him/her a message,does it generate anything in applog for the particular
subscriber.

Aman


On 9/30/06, Keith Klevenski <keith.klevenski at rig.net> wrote:
>
>  Can you dial the MWI On extension from the phone and turn it on?
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Marcus Smith
> *Sent:* Friday, September 29, 2006 2:50 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] MWI
>
>
>
> I have a little problem that will not go away.  A user phone MWI does not
> light up when there is a message for her.  I have checked everything to
> resolve this issue, but it is still there.  Even deleted the extension and
> recreated it, but still no luck, also, checked her profile sheet in CCM,
> everything is good.  what else can I check to resolve this issue.  Yes, the
> light does work  when incoming calls are received.
>
>
>
> Marcus
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From SPackett at fenwick.com  Fri Sep 29 15:59:53 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Fri, 29 Sep 2006 12:59:53 -0700
Subject: [cisco-voip] MWI
Message-ID: <01C8F21F30DE9740B53FD2EF05C3A8FC01D713A1@SVCEXCH01.firm.fenwick.llp>

I had a similar issue when I first rolled out Unity.  It turned out to
be a calling search space / partition issue.  You should look there
first.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Marcus Smith
Sent: Friday, September 29, 2006 12:50 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MWI


I have a little problem that will not go away.  A user phone MWI does
not light up when there is a message for her.  I have checked everything
to resolve this issue, but it is still there.  Even deleted the
extension and recreated it, but still no luck, also, checked her profile
sheet in CCM, everything is good.  what else can I check to resolve this
issue.  Yes, the light does work  when incoming calls are received.
 
Marcus
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From Scott.Voll at wesd.org  Fri Sep 29 16:07:04 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 29 Sep 2006 13:07:04 -0700
Subject: [cisco-voip] MWI
Message-ID: <407055A92CECCB499C922A2D35FC19A606B442EC@Apollo.wesd.org>

If you can, then just do a simple resync of the MWI.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Friday, September 29, 2006 12:55 PM
To: Marcus Smith; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MWI

 

Can you dial the MWI On extension from the phone and turn it on?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Marcus Smith
Sent: Friday, September 29, 2006 2:50 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MWI

 

I have a little problem that will not go away.  A user phone MWI does
not light up when there is a message for her.  I have checked everything
to resolve this issue, but it is still there.  Even deleted the
extension and recreated it, but still no luck, also, checked her profile
sheet in CCM, everything is good.  what else can I check to resolve this
issue.  Yes, the light does work  when incoming calls are received.

 

Marcus

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From Brett.Krueger at ficoh.com  Fri Sep 29 16:09:49 2006
From: Brett.Krueger at ficoh.com (Krueger, Brett)
Date: Fri, 29 Sep 2006 10:09:49 -1000
Subject: [cisco-voip] MWI
Message-ID: <4FFD4B847496F44EB7DCF9CFEF2E771F5D72C5@f01mexcp03.ficoh.com>

We actually have the same issue here.  Seems that its random throughout
our installs and refreshing the status of the MWI through Unity sets it
correctly.  Some sort of sync issue possibly?  Unity 4.0(5).  After our
plans of upgrading Unity, if its still persists i guess then I might go
as far as opening up a TAC case on it.
*shrug*
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Friday, September 29, 2006 9:55 AM
To: Marcus Smith; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MWI



Can you dial the MWI On extension from the phone and turn it on?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Marcus Smith
Sent: Friday, September 29, 2006 2:50 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MWI

 

I have a little problem that will not go away.  A user phone MWI does
not light up when there is a message for her.  I have checked everything
to resolve this issue, but it is still there.  Even deleted the
extension and recreated it, but still no luck, also, checked her profile
sheet in CCM, everything is good.  what else can I check to resolve this
issue.  Yes, the light does work  when incoming calls are received.

 

Marcus




E-MAIL CONFIDENTIALITY NOTICE:
The contents of this e-mail message and any attachments are intended solely for the addressee(s) and may contain confidential and/or legally privileged information.  If you are not the intended recipient of this message or if this message has been addressed to you in error, please immediately alert the sender by reply e-mail and then delete this message and any attachments.  If you are not the intended recipient, you are notified that any use, dissemination, distribution, copying, or storage of this message or any attachments is strictly prohibited.

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From Scott.Voll at wesd.org  Fri Sep 29 16:19:17 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 29 Sep 2006 13:19:17 -0700
Subject: [cisco-voip] MWI
Message-ID: <407055A92CECCB499C922A2D35FC19A606B442F5@Apollo.wesd.org>

We are a small deployment ~500 subscribers but we have it set to resync
each night.

 

Also, have you rebooted your exchange server and not your Unity server
after the exchange server? This can cause this kind of issue also.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Krueger, Brett
Sent: Friday, September 29, 2006 1:10 PM
To: Keith Klevenski; Marcus Smith; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MWI

 

We actually have the same issue here.  Seems that its random throughout
our installs and refreshing the status of the MWI through Unity sets it
correctly.  Some sort of sync issue possibly?  Unity 4.0(5).  After our
plans of upgrading Unity, if its still persists i guess then I might go
as far as opening up a TAC case on it.

*shrug*

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Klevenski
Sent: Friday, September 29, 2006 9:55 AM
To: Marcus Smith; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MWI

Can you dial the MWI On extension from the phone and turn it on?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Marcus Smith
Sent: Friday, September 29, 2006 2:50 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MWI

 

I have a little problem that will not go away.  A user phone MWI does
not light up when there is a message for her.  I have checked everything
to resolve this issue, but it is still there.  Even deleted the
extension and recreated it, but still no luck, also, checked her profile
sheet in CCM, everything is good.  what else can I check to resolve this
issue.  Yes, the light does work  when incoming calls are received.

 

Marcus

________________________________


E-MAIL CONFIDENTIALITY NOTICE:
The contents of this e-mail message and any attachments are intended
solely for the addressee(s) and may contain confidential and/or legally
privileged information. If you are not the intended recipient of this
message or if this message has been addressed to you in error, please
immediately alert the sender by reply e-mail and then delete this
message and any attachments. If you are not the intended recipient, you
are notified that any use, dissemination, distribution, copying, or
storage of this message or any attachments is strictly prohibited.

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From Scott.Voll at wesd.org  Fri Sep 29 17:26:07 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 29 Sep 2006 14:26:07 -0700
Subject: [cisco-voip] Unity Tech Admin
Message-ID: <407055A92CECCB499C922A2D35FC19A606B44313@Apollo.wesd.org>

Thanks

 

I couldn't remember the GrantUnityAccess utility.

 

Scott

 

________________________________

From: Rasim Duric [mailto:rduric at uoguelph.ca] 
Sent: Friday, September 29, 2006 11:42 AM
To: Voll, Scott
Subject: RE: [cisco-voip] Unity Tech Admin

 

You can create the same COS on the PBX Unity server and assign a
subscriber to it. Now, you'll need GrantUnityAccess for all HelpDesk
users to this particular subscriber. 

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, September 29, 2006 2:28 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Tech Admin

 

I have two Unity servers both running unity 4.0.5.  one connected to my
CM 4.1.3sr2 and one to my old PBX.  Both are digitally networked and
working fine.

 

I have made myself an admin on the CM Unity box and then had to so some
utility to make myself an admin on the PBX Unity server.

 

I have now made a COS on the CM Unity server for our Helpdesk to
add/change/delete subscribers but now how do I do that on the PBX Unity
server?

 

Scott

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From tad at lmteam.com  Sat Sep 30 05:55:21 2006
From: tad at lmteam.com (Tad A. Lagestee)
Date: Sat, 30 Sep 2006 04:55:21 -0500
Subject: [cisco-voip]  Mobility Manager Call Waiting on Cell Phones
Message-ID: <881C73F736582D40AF21F1ADAD6217F366D838@lmms01.lmteam.local>

I have a different problem with my Mobility install:

When you receive a call on your cell phone that was sent through
Mobility, and another person calls your "mobility number", Mobility DOES
NOT call your cell phone....  Your call waiting on your cell phone is
gone.  Imagine someone having an emergency and there is no way they
could get through to you, unless of course they call your cell phone
directly.  Kind of defeats the purpose.

---------------------------------------------
 
Sincerely,
 
Tad A. Lagestee


From rduric at uoguelph.ca  Sat Sep 30 10:54:57 2006
From: rduric at uoguelph.ca (Rasim Duric)
Date: Sat, 30 Sep 2006 10:54:57 -0400
Subject: [cisco-voip] Unity Tech Admin
In-Reply-To: <407055A92CECCB499C922A2D35FC19A606B44313@Apollo.wesd.org>
Message-ID: <BD89624EE83BD349BD31A08B453DE244D503B6@uogexb201.cfs.uoguelph.ca>

It's in the \commserver directory. You run it from the command line, for
example if you want to assign the EAdmin rights to a particular userID:

 

GrantUnityAccess -u NT4DOMAIN\userID -s EAdmin

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Friday, September 29, 2006 5:26 PM
To: Rasim Duric
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity Tech Admin

 

Thanks

 

I couldn't remember the GrantUnityAccess utility.

 

Scott

 

________________________________

From: Rasim Duric [mailto:rduric at uoguelph.ca] 
Sent: Friday, September 29, 2006 11:42 AM
To: Voll, Scott
Subject: RE: [cisco-voip] Unity Tech Admin

 

You can create the same COS on the PBX Unity server and assign a
subscriber to it. Now, you'll need GrantUnityAccess for all HelpDesk
users to this particular subscriber. 

 

Rasim Duric
Network Analyst (CCS)
University of Guelph
Guelph, N1G 2W1, ON
519-824-4120x53146
rduric at uoguelph.ca

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, September 29, 2006 2:28 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Tech Admin

 

I have two Unity servers both running unity 4.0.5.  one connected to my
CM 4.1.3sr2 and one to my old PBX.  Both are digitally networked and
working fine.

 

I have made myself an admin on the CM Unity box and then had to so some
utility to make myself an admin on the PBX Unity server.

 

I have now made a COS on the CM Unity server for our Helpdesk to
add/change/delete subscribers but now how do I do that on the PBX Unity
server?

 

Scott

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From manoj.kalpage at gmail.com  Sat Sep 30 12:18:10 2006
From: manoj.kalpage at gmail.com (Manoj Kalpage)
Date: Sun, 1 Oct 2006 01:18:10 +0900
Subject: [cisco-voip] DMVPN and QOS implementaion
References: <017f01c6e3dc$7c674210$6500a8c0@computec.lk>
	<d4c697ad0609290946i10d76ff2h9aa8ee31d4fca931@mail.gmail.com>
Message-ID: <01f701c6e4ac$07bde0b0$6500a8c0@computec.lk>

Aman,
Thank you very much for your reply. This is very helpfull. 
G.729 parameter change is very interesting. I am using GT.711 just because for the conference feature. Does G.729 support for conference call at remote sites?

Thanks,
Manoj

  ----- Original Message ----- 
  From: Aman Chugh 
  To: Manoj Kalpage 
  Cc: cisco-voip at puck.nether.net 
  Sent: Saturday, September 30, 2006 1:46 AM
  Subject: Re: [cisco-voip] DMVPN and QOS implementaion


   Manoj,

  Your Qos configuration looks ok,I had similar Issues V3PN setup ,some of the things you need to consider for this kind of setup is

  1.Use hardware encryption for VPN as software encryption adds unneccassry delays
  2.Check the Voice codec for your inter site calls and during  a bad  call press the ? key on the phone to check MAX jitter and RXLOST values.
  3. Do sh policy-map interface to check for drops ,if so see drops change your priority queue bandwitdhs
  3.There is a service parrametter in callmanager to remove G.729ab,G.729b from cabablities when calls are established,I had considerable improvement in voice quality after changing this parrametter to true as by default its false.I could not use g.729 for voice calls as voice sounded very choppy ,once i applied this change i am able to use g.729 for calls.

  Hope this helps
  Aman

   
  On 9/29/06, Manoj Kalpage <manoj.kalpage at gmail.com> wrote: 
    Hi all,
    We have hosted PBX system which is located in data centre and we have
    dedicated 1MB internet connection. At present we only have four remote sites 
    and all of them have high speed ADSL connection for both their data and
    Voice. Each sites has 4 to 5 phones. We used to use PIX 515 at Data centre
    and PIX 501 at remote sites. As we were experiencing voice quality issue we 
    moved to Cisco 2821 at Data centre and 800 series at remote sites. I have
    configured DMVPN using GRE over IPSec for our VPN network. all the tunnels
    are up and seems to be working fine so far but I am just wondering whether I 
    got right QoS configuration at HeadEnd Router. As, I am a newbie for QoS, I
    have referred various cisco documentations to configure bellow DMVPN and QoS
    for our head end router. Since we don't have data transaction at HeadEnd 
    site I think I can use 75% of bandwidth for voice it self.  Can some one
    help me figure out QoS requirment for my network environment.

    Thank you in advanced,

    Best regards
    Manoj

    --------------------------------------------------------------------------- 
    Building configuration...

    Current configuration : 5063 bytes
    !
    version 12.4
    service timestamps debug datetime msec
    service timestamps log datetime msec
    no service password-encryption
    !
    hostname PBXLGATE01 
    !
    boot-start-marker
    boot-end-marker
    !
    logging buffered 51200 warnings
    !
    no aaa new-model
    !
    resource policy
    !
    ip cef

    !
    ip domain name yourdomain.com 
    !
    !
    crypto pki trustpoint TP-self-signed-2723000426
    enrollment selfsigned
    subject-name cn=IOS-Self-Signed-Certificate-2723000426
    revocation-check none
    rsakeypair TP-self-signed-2723000426
    !
    !
    crypto pki certificate chain TP-self-signed-2723000426
    certificate self-signed 01
    30820251 308201BA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
    31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274 
    quit
    username xxxx privilege 15 secret 5 @#@#@@@GlPb96SyZxV6Q0
    !
    !
    class-map match-all VOICE
    match ip dscp ef
    class-map match-all SCAVENGER
    match ip dscp cs1
    class-map match-any INTERNETWORK-CONTROL 
    match ip dscp cs6
    match access-group name IKE
    class-map match-any CALL-SIGNALING
    match ip dscp cs3
    match ip dscp af31
    !
    !
    policy-map V3PN-EDGE
    class VOICE
    priority percent 55
    class CALL-SIGNALING 
    bandwidth percent 5
    class INTERNETWORK-CONTROL
    bandwidth percent 5
    class SCAVENGER
    bandwidth percent 1
    queue-limit 1
    class class-default
    bandwidth percent 9
    queue-limit 16
    !
    !
    crypto isakmp policy 10 
    hash md5
    authentication pre-share
    crypto isakmp key 6 G0G0G0G0 address 0.0.0.0 0.0.0.0
    !
    !
    crypto ipsec transform-set PBXL esp-3des esp-md5-hmac 
    !
    crypto ipsec profile PBXL
    set security-association lifetime seconds 120
    set transform-set PBXL
    !
    !
    interface Tunnel0
    ip address 10.10.1.1 255.255.255.0
    no ip redirects
    ip mtu 1440
    ip nhrp authentication xxxxxxxx
    ip nhrp map multicast dynamic
    ip nhrp network-id 1
    ip tcp adjust-mss 1360
    qos pre-classify
    tunnel source FastEthernet0/0 
    tunnel mode gre multipoint
    tunnel key 0
    tunnel protection ipsec profile PBXL
    !

    interface FastEthernet0/0
    description Connect to Verizon Network
    bandwidth 1000
    ip address 222.222.222.222 255.255.255.192
    ip nbar protocol-discovery
    ip nat outside
    ip virtual-reassembly
    duplex auto
    speed auto
    service-policy output V3PN-EDGE
    !
    interface FastEthernet0/1 
    ip address 192.168.4.1 255.255.255.0
    ip nat inside
    ip virtual-reassembly
    duplex auto
    speed auto
    !
    router eigrp 90
    network 10.0.0.0
    network 172.16.0.0 0.0.0.255
    no auto-summary
    !
    ip route 0.0.0.0 0.0.0.0 222.222.222.222
    !
    !
    ip http server
    ip http authentication local
    ip http secure-server
    ip http timeout-policy idle 600 life 86400 requests 1000
    ip nat inside source list 1 interface FastEthernet0/0 overload 
    !
    ip access-list extended IKE
    permit udp any eq isakmp any eq isakmp
    !
    access-list 1 permit 192.168.4.0 0.0.0.255
    !
    !
    control-plane
    !
    !
    line con 0
    login local
    line aux 0
    line vty 0 4
    access-class 23 in
    privilege level 15
    login local
    transport input ssh
    !
    scheduler allocate 20000 1000
    end

    PBXLGATE01#



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




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From rwolfe at Burwood.com  Sat Sep 30 14:17:33 2006
From: rwolfe at Burwood.com (Wolfe, Robert)
Date: Sat, 30 Sep 2006 13:17:33 -0500
Subject: [cisco-voip] Unity Directory Handler forwards callers straight to
	user's voicmailbox
Message-ID: <97F472A9F637E44B954D7FBABC800430FC88B3@bur-msg-02.Burwood.com>

Hello,

 

All my Unity Directory handlers are going straight to VM and not ringing
the lines of the subscribers.  Checked the DNA from the voicemail port
to the endpoint and it shows that it should go through...I also made the
phone a one-off number so there is no translation overlap or alternate
matches.

 

Any ideas would be greatly appreciated.  

 

Thanks,

 

-rob

 

  <http://www.burwood.com> 

Robert Wolfe
Network Consultant
Burwood Group, Inc.

IP:  312.327.4634
Mobile:  312.593.0605
E-Mail:  rwolfe <mailto:rwolfe at burwood.com> @burwood.com
<mailto:rwolfe at burwood.com> 

 

  <http://www.burwood.com> 

 

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From jrybak at annese.com  Sat Sep 30 14:59:30 2006
From: jrybak at annese.com (Jake Rybak)
Date: Sat, 30 Sep 2006 14:59:30 -0400
Subject: [cisco-voip] Encrypted VM ports
Message-ID: <000601c6e4c2$a1954320$3f0114ac@none>

Has anyone been able to get encrypted VM ports to work with CM 5.0(4a) and
Unity 4.2(1)?

 

I am getting the error message in the Unity event viewer "Unable to download
cert.tvl from tftp server" and the CM status is rejected or unknown. The
tftp server is definitely available as the Unity box and CM are on the same
vlan and pingable. I have deleted the integration on both sides, regenerated
the Unity cert.pem and uploaded it to all callmanagers several times. I have
upgraded to TSP version 8.1.3. The vm ports will register in non-secure
mode. Also, I have no problems with secure phones. I verified the cert is in
the tftp directory so I copied the cert to Unity manually and still no dice.
CallManger traces show that Unity is trying to download the cert but it
fails. 

 

TAC has been through the config and has not been able to figure it out.

 

 

Thanks,

 

jake

 

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From cshaffer at gmail.com  Sat Sep 30 17:34:03 2006
From: cshaffer at gmail.com (Curt Shaffer)
Date: Sat, 30 Sep 2006 16:34:03 -0500
Subject: [cisco-voip] router can't seem to handle even 1 call
Message-ID: <001401c6e4d8$27530960$171b10ac@betsy>

I have a 1760 here in my lab. As soon as I make 1 SIP based phone call to a
proxy on the WAN side if it the CPU spikes to 50 + % and the jitter jumps to
an unusable state with MOS down to 1.0 and less. When I hang the call up and
give it a couple seconds everything goes back to normal CPU to 5%, MOS to
4.40 jitter almost non existent. My question is why would this happen with
only 1 call? I was playing with NBAR and QoS but I removed all service,
policy and class maps completely and this still happens. Any ideas out
there?

 

 

Thanks

 

Curt

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From jiteto at gmail.com  Sat Sep 30 17:44:44 2006
From: jiteto at gmail.com (Ivan Zhitianov)
Date: Sun, 1 Oct 2006 00:44:44 +0300
Subject: [cisco-voip] IPCC - Unable to Route Call to Agent
Message-ID: <78747e9c0609301444w66810262ncee13f88cf1cb2ca@mail.gmail.com>

Hi all,

I'm trying to get ICD working on IPCC Express 4.5.

Everything seems fine but when I dial defined for the ICD Route Point I get
MOH followed by an error message that service is not running OK or just that
there are no free agents at the moment. Route Point and CTI Ports are
registered to Call Manger. All of the phones and CTI ports are in one
Calling Searching Space. RM JTAPI User and End-user has a control over Phone
and CTI Port.

Mean while on the IP Phone agent and/or Desktop Agent I get status changing
from Ready to Reserved again Ready and Reserved till I here the error
message.

What I'm missing?!

Thanks in advance!

Best Regards,
Ivan
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From erickbe at yahoo.com  Sat Sep 30 18:46:10 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Sat, 30 Sep 2006 15:46:10 -0700 (PDT)
Subject: [cisco-voip] Long delay
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E03A47632@MREXVS01.bc.broward.cty>
Message-ID: <20060930224610.9883.qmail@web37215.mail.mud.yahoo.com>

Check your QoS policies between the sites , and check for drops.  I recently had a client having similar type of problem (audio-cutover delays) during times of lots of call signalling activity. The issue was the signalling class-map had enough bandwidth allocated but the default queue-depth (64 packets) was not enough to handle the burst of lots of control traffic at busy times. Had to add the queue-limit command to class map and extend it to 256 packets which is working out for that setup fine now. 

----- Original Message ----
From: "Ortiz, Carlos" <CORTIZ at broward.org>
To: "Voll, Scott" <Scott.Voll at wesd.org>; cisco-voip at puck.nether.net
Sent: Friday, September 29, 2006 12:45:45 PM
Subject: Re: [cisco-voip] Long delay

              How heavy is the volume of calls at the site?  I have a site with 2 PRI?s (about 160 users) that was getting inundated with calls yesterday.  They claim that they were having slow downs transferring calls when the traffic was heavy. Not sure what would cause this but wondering if your location was similar.
     
   Carlos
     
         
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
 Sent: Friday, September 29, 2006 12:59 PM
 To: cisco-voip at puck.nether.net
 Subject: [cisco-voip] Long delay
   
    
   I?m running cm 4.1.3sr2 and I have a remote site complaining of long delays in transfers and answering.  Does anyone have any idea where to start with this?
     
   I have not been out to the remote site to personally check.
     
   Scott
     
     
   
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From aman.chugh at gmail.com  Sat Sep 30 21:53:07 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Sun, 1 Oct 2006 07:23:07 +0530
Subject: [cisco-voip] DMVPN and QOS implementaion
In-Reply-To: <AB0A35F2486F81468971AA68558BE71904F04661@smieln01winm01.micoanet.com>
References: <01f701c6e4ac$07bde0b0$6500a8c0@computec.lk>
	<AB0A35F2486F81468971AA68558BE71904F04661@smieln01winm01.micoanet.com>
Message-ID: <d4c697ad0609301853m577a7d4ai312330e76ba7fb79@mail.gmail.com>

Matthew,

I had a TAC case opened as I could not use g.729 for voice calls for
intersite calls over a site to site IPSEC tunnel,it would sound very
choppy,after changing this parametter ,I saw a considerable improvement and
now I am able to use g.729 for inter site calls.This is the service
parrameeter and this will affect all calls and need to same on both the
publisher and subscriber,If you r not seeing any Issues with g.729 over VPN
,I would not recommend changing it to true.

"Stirp G.729 Annex B (Silence Suppression) from Capabilities"

Manoj, You can use g.729 for conferencing if you use hardware conferrence
bridge ,Callmanager software conference bridge will only do g.711 conference
and if a g.729 participant wants to jump in then it would make use of a
transcorder if one is available.You can deploy dspfarm at each remote site
to be used for conferencing or transcoding.

Aman


On 9/30/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>  What was your reason for changing the parameter in CallManager (and what
> is the exact name?)  I am interested improving my g.729 Call Quality.
>
>
>
> Matt
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Manoj Kalpage
> *Sent:* Saturday, September 30, 2006 12:18 PM
> *To:* Aman Chugh
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] DMVPN and QOS implementaion
>
>
>
> Aman,
>
> Thank you very much for your reply. This is very helpfull.
>
> G.729 parameter change is very interesting. I am using GT.711 just because
> for the conference feature. Does G.729 support for conference call at
> remote sites?
>
>
>
> Thanks,
>
> Manoj
>
>
>
>  ----- Original Message -----
>
> *From:* Aman Chugh <aman.chugh at gmail.com>
>
> *To:* Manoj Kalpage <manoj.kalpage at gmail.com>
>
> *Cc:* cisco-voip at puck.nether.net
>
> *Sent:* Saturday, September 30, 2006 1:46 AM
>
> *Subject:* Re: [cisco-voip] DMVPN and QOS implementaion
>
>
>
>  Manoj,
>
>
>
> Your Qos configuration looks ok,I had similar Issues V3PN setup ,some of
> the things you need to consider for this kind of setup is
>
>
>
> 1.Use hardware encryption for VPN as software encryption adds unneccassry
> delays
>
> 2.Check the Voice codec for your inter site calls and during  a bad
>  call press the ? key on the phone to check MAX jitter and RXLOST values.
>
> 3. Do sh policy-map interface to check for drops ,if so see drops change
> your priority queue bandwitdhs
>
> 3.There is a service parrametter in callmanager to remove G.729ab,G.729bfrom cabablities when calls are established,I had considerable improvement
> in voice quality after changing this parrametter to true as by default its
> false.I could not use g.729 for voice calls as voice sounded very choppy
> ,once i applied this change i am able to use g.729 for calls.
>
>
>
> Hope this helps
>
> Aman
>
>
>
> On 9/29/06, *Manoj Kalpage* <manoj.kalpage at gmail.com> wrote:
>
> Hi all,
> We have hosted PBX system which is located in data centre and we have
> dedicated 1MB internet connection. At present we only have four remote
> sites
> and all of them have high speed ADSL connection for both their data and
> Voice. Each sites has 4 to 5 phones. We used to use PIX 515 at Data centre
> and PIX 501 at remote sites. As we were experiencing voice quality issue
> we
> moved to Cisco 2821 at Data centre and 800 series at remote sites. I have
> configured DMVPN using GRE over IPSec for our VPN network. all the tunnels
> are up and seems to be working fine so far but I am just wondering whether
> I
> got right QoS configuration at HeadEnd Router. As, I am a newbie for QoS,
> I
> have referred various cisco documentations to configure bellow DMVPN and
> QoS
> for our head end router. Since we don't have data transaction at HeadEnd
> site I think I can use 75% of bandwidth for voice it self.  Can some one
> help me figure out QoS requirment for my network environment.
>
> Thank you in advanced,
>
> Best regards
> Manoj
>
> ---------------------------------------------------------------------------
>
> Building configuration...
>
> Current configuration : 5063 bytes
> !
> version 12.4
> service timestamps debug datetime msec
> service timestamps log datetime msec
> no service password-encryption
> !
> hostname PBXLGATE01
> !
> boot-start-marker
> boot-end-marker
> !
> logging buffered 51200 warnings
> !
> no aaa new-model
> !
> resource policy
> !
> ip cef
>
> !
> ip domain name yourdomain.com
> !
> !
> crypto pki trustpoint TP-self-signed-2723000426
> enrollment selfsigned
> subject-name cn=IOS-Self-Signed-Certificate-2723000426
> revocation-check none
> rsakeypair TP-self-signed-2723000426
> !
> !
> crypto pki certificate chain TP-self-signed-2723000426
> certificate self-signed 01
> 30820251 308201BA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
> 31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
> quit
> username xxxx privilege 15 secret 5 @#@#@@@GlPb96SyZxV6Q0
> !
> !
> class-map match-all VOICE
> match ip dscp ef
> class-map match-all SCAVENGER
> match ip dscp cs1
> class-map match-any INTERNETWORK-CONTROL
> match ip dscp cs6
> match access-group name IKE
> class-map match-any CALL-SIGNALING
> match ip dscp cs3
> match ip dscp af31
> !
> !
> policy-map V3PN-EDGE
> class VOICE
> priority percent 55
> class CALL-SIGNALING
> bandwidth percent 5
> class INTERNETWORK-CONTROL
> bandwidth percent 5
> class SCAVENGER
> bandwidth percent 1
> queue-limit 1
> class class-default
> bandwidth percent 9
> queue-limit 16
> !
> !
> crypto isakmp policy 10
> hash md5
> authentication pre-share
> crypto isakmp key 6 G0G0G0G0 address 0.0.0.0 0.0.0.0
> !
> !
> crypto ipsec transform-set PBXL esp-3des esp-md5-hmac
> !
> crypto ipsec profile PBXL
> set security-association lifetime seconds 120
> set transform-set PBXL
> !
> !
> interface Tunnel0
> ip address 10.10.1.1 255.255.255.0
> no ip redirects
> ip mtu 1440
> ip nhrp authentication xxxxxxxx
> ip nhrp map multicast dynamic
> ip nhrp network-id 1
> ip tcp adjust-mss 1360
> qos pre-classify
> tunnel source FastEthernet0/0
> tunnel mode gre multipoint
> tunnel key 0
> tunnel protection ipsec profile PBXL
> !
>
> interface FastEthernet0/0
> description Connect to Verizon Network
> bandwidth 1000
> ip address 222.222.222.222 255.255.255.192
> ip nbar protocol-discovery
> ip nat outside
> ip virtual-reassembly
> duplex auto
> speed auto
> service-policy output V3PN-EDGE
> !
> interface FastEthernet0/1
> ip address 192.168.4.1 255.255.255.0
> ip nat inside
> ip virtual-reassembly
> duplex auto
> speed auto
> !
> router eigrp 90
> network 10.0.0.0
> network 172.16.0.0 0.0.0.255
> no auto-summary
> !
> ip route 0.0.0.0 0.0.0.0 222.222.222.222
> !
> !
> ip http server
> ip http authentication local
> ip http secure-server
> ip http timeout-policy idle 600 life 86400 requests 1000
> ip nat inside source list 1 interface FastEthernet0/0 overload
> !
> ip access-list extended IKE
> permit udp any eq isakmp any eq isakmp
> !
> access-list 1 permit 192.168.4.0 0.0.0.255
> !
> !
> control-plane
> !
> !
> line con 0
> login local
> line aux 0
> line vty 0 4
> access-class 23 in
> privilege level 15
> login local
> transport input ssh
> !
> scheduler allocate 20000 1000
> end
>
> PBXLGATE01#
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>  ------------------------------
>
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