From jonvoip at gmail.com  Thu Jun  1 00:32:42 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 31 May 2006 23:32:42 -0500
Subject: [cisco-voip] Need BugID for MoH issue... workaround included
Message-ID: <5d093f9a0605312132o3e67012ewc87ce5b94f1f2092@mail.gmail.com>

Found a problem on all versions of CallManager, with OS versions of
2.7 or higher. Was wondering if there was a BugID for this...

CCM Version:

 Confirmed on 3.3, 4.0 and 4.1

 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
 a '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.

 This fix has solved the problem every time so far.

From erickbe at yahoo.com  Thu Jun  1 01:08:25 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 31 May 2006 22:08:25 -0700 (PDT)
Subject: [cisco-voip] Mcafee 8.0 and CM CSA Conflict - McShield service
	terminating
In-Reply-To: <64CED1E0-B1DD-4F82-95AF-D93C79DD86DE@cisco.com>
Message-ID: <20060601050825.92659.qmail@web37213.mail.mud.yahoo.com>

It looks like applying patch 11 for Mcafee virusscan 8.0i enterprise clears this issue up, at least in this instance.

I tried to duplicate the issue on some lab servers with various mcafee versions and CSA and was unable to. Had customer update to patch 11 version - they were on patch 1, and I have not been able to recreate the issue. Before updating, I could recreate the errors and service crash by just stopping and starting the Mcshield service with CSA already started and it would have issues below each time. 

----- 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: Wednesday, May 24, 2006 10:53:54 AM
Subject: Re: [cisco-voip] Mcafee 8.0 and CM CSA Conflict - McShield service terminating

I see a similar TAC SR open currently but no bug filed.  If you can  
reproduce with the latest CSA please open a TAC case.

-Ryan

On May 24, 2006, at 10:56 AM, Erick Bergquist wrote:

I am seeing a conflict between McAfee VirusScan Enterprise v8.0.0  
patch version 1 and CSA v4.5.1 build 639.

When the network associates McShield service starts when CSA is  
started , the McShield service terminates unexpectedly a couple dozen  
times. This is causing other cisco call manager related services to  
get briefly hung up on the server also.

If I stop CSA, then start McShield, then start CSA after McShield is  
started all appears to be fine.

I have done several bug searches and come up empty. I am planning on  
applying latest CSA on call manager but can't find anything yet that  
confirms a newer CSA fixes this or not.



Was curious if anyone here  had ran into this or not..

Event Type:    Information
Event Source:    CSAgent
Event Category:    Local Event Mgr
Event ID:    256
Date:        5/24/2006
Time:        12:02:40 AM
User:        N/A
Computer:    CCM
Description:
[2006-05-24 00:02:40.575] [PID=8208] [AgentUI]: listener_th: It took  
too long (88672 millisec) to process event with code=650: type=EVTM  
EvSrcComp=6 EvDstComp=4 EvCode=AUI_AGENT_STATUS_UPDATE

Event Type:    Information
Event Source:    CSAgent
Event Category:    Local Event Mgr
Event ID:    256
Date:        5/24/2006
Time:        12:03:00 AM
User:        N/A
Computer:    CCM
Description:
[2006-05-24 00:03:00.826] [PID=8208] [AgentUI]: DIPC_get_unique:  
Failed to write to comp Csamanager on fixed channel Global 
\Csamanager_ch_00000000: 3006


Event Type:    Error
Event Source:    Service Control Manager
Event Category:    None
Event ID:    7031
Date:        5/24/2006
Time:        12:02:42 AM
User:        N/A
Computer:    CCM1
Description:
The Network Associates McShield service terminated unexpectedly. It  
has done this 26 time(s). The following corrective action will be  
taken in 0 milliseconds: No action.

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






From SechabaM at kss.co.za  Thu Jun  1 05:48:11 2006
From: SechabaM at kss.co.za (Sechaba Mokoena)
Date: Thu, 1 Jun 2006 11:48:11 +0200
Subject: [cisco-voip]  7912 IP Phone
Message-ID: <3E110434E2419D47883D119DD226ACA1010B02A7@kss-jhb-exch-01.kss.co.za>

To Whom It May Concern:

 

Does anyone know how to completely erase the configuration of a 7912 IP
Phone? I've noticed that if you move a 7912 from one Call Manager to
another the phone retains the IP address of the last Call Manager that
it registered with. I've also noticed that if you log into the phone
using Extension Mobility, it doesn't change the DN immediately; instead
the phone resets, and tries to register with the last Call Manager that
it registered with first. When the phone doesn't find the former Call
Manager it then registers with the new one. Is this common for this
particular model?

 

Regards,

 

Sechaba Mokoena | Systems Engineer 

KSS Technologies (Pty) LTD.

KSS Convergence - Doing more with less

(IP Phone) +27 11 5456300|(IP Phonedirect) | +27 11 5456331| Fax)+27 11
545 6400| Mobile +27 84 798 6803 |

(e-mail)sechabam at kss.co.za

(website)http://www.kss.co.za <blocked::http://www.kss.co.za> 

Block N, Central Park, 400 16th Road, Randjespark, Midrand, South Africa

 

 

Toll Free Support: 0861 6428 35 

 


This message, including any attachments is intended only for
the persons or entity to which it is addressed, and may 
contain privileged or confidential information. Retransmission, 
dissemination, copying, distribution, modification and / or 
publication of this message by a person other than the intended 
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From kthorngr at cisco.com  Thu Jun  1 07:56:47 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu, 1 Jun 2006 07:56:47 -0400
Subject: [cisco-voip] Need BugID for MoH issue... workaround included
In-Reply-To: <5d093f9a0605312132o3e67012ewc87ce5b94f1f2092@mail.gmail.com>
References: <5d093f9a0605312132o3e67012ewc87ce5b94f1f2092@mail.gmail.com>
Message-ID: <ad221a6696065bd2802e14c252105aaa@cisco.com>

Hi Jonathan,

I did not find any BugIDs for this issue.  I just tried it with both 
4.1(3) running on 2000.2.7sr5 and 4.2(1) with 2000.4.2sr3.  It worked 
fine in both clusters.  Are the servers in a Domain?

Could it be that steps similar to what is described in the below DDTS 
were used?

CSCec36827 - Need CCM tool to reset file/folder permissions back to 
defaults
Symptom:

CallManager services stop operating correctly after making permissions 
changes on files/folders.

Condition:

This problem may occur after manually changing file/folder permissions 
on the C:\Program Files
folder and selecting to "Reset permissions on all child objects and 
enable propagation of
inheritable permissions."

This essentially removes all of the permissions given to CCMCDR, 
CCMEML, CCMService, CCMServiceRW,
and CCMUser on the C:\Program Files\Cisco folder.

This bug was opened to request that a new tool be developed so that the 
file/folder permissions
can be easily restored to their defaults.

Workaround:

Manually give CCMCDR, CCMEML, CCMService, CCMServiceRW, and CCMUser 
read, write, and modify
permissions to the C:\Program Files\Cisco folder.

WARNING:  Be very careful when making permissions changes to 
files\folders on a CallManager server.
           Contact Cisco TAC before making such changes.


If you can still recreate the issue then it might be best to open a TAC 
case and provide remote access so the problem can be investigated.

Kevin

On Jun 1, 2006, at 12:32 AM, Jonathan Charles wrote:

> Found a problem on all versions of CallManager, with OS versions of
> 2.7 or higher. Was wondering if there was a BugID for this...
>
> CCM Version:
>
>  Confirmed on 3.3, 4.0 and 4.1
>
>  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
>  a '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.
>
>  This fix has solved the problem every time so far.
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From jaison.joy at gmail.com  Thu Jun  1 08:46:57 2006
From: jaison.joy at gmail.com (Jaison Joy)
Date: Thu, 1 Jun 2006 18:16:57 +0530
Subject: [cisco-voip] IPCC Express /CCM Issue.
Message-ID: <e4aded970606010546v4d069043pe716d5f8a20cc038@mail.gmail.com>

Hi ,

Background:
----------
I have an issue with the installation of the IPCC express After i finish the
software installation.  and the Server reboots and when it comes up next
time. It comes up first time with the webpage http://<ccm>/appadmin. it
gives me error "You are not authorized to view this page" Error 401.1

Under the event logs -> security, I see errors regarding unsuccessfull login
by the IIS Accounts.

After installing CRS in call manager. I have issues with the CTI manager and
Cisco TCD services not starting.
Every second their is an error about the CTI service Terminated.
I see the following errors.

Event Type: Error
Event Source: Cisco CallManager
Event Category: None
Event ID: 3
Date:  6/1/2006
Time:  5:02:46 AM
User:  N/A
Computer: DELL
Description:
Error: GetNewFileName-fopen in wb mode failed to update existing file.
szFileNumberName = c:\Program Files\Cisco\Trace\CTI\cti~num.bin.

Event Type: Error
Event Source: Service Control Manager
Event Category: None
Event ID: 7031
Date:  6/1/2006
Time:  4:57:41 AM
User:  N/A
Computer: DELL
Description:
The Cisco CTIManager service terminated unexpectedly.  It has done this 1
time(s).  The following corrective action will be taken in 60000
milliseconds: Restart the service.

Problem statement:
------------------
Issue 1:
What am i missing during the CRS installtion.

Issue 2:
After loading CRS why CRS CTI manager and TCD.

how can i solve these issues.

Software version:
----------------
IPCC Express 3.(03)a
Callmanager 3.3.4


Actions already performed:
-------------------------
1. Followed Microsoft kb.
http://support.microsoft.com/?kbid=896861.
2. Reconfigure all the services in the callmanager- admin utility.
3. Restarted the IIS.
4. Changed the execution permissions in IIS webpage.
5. Changed the Directory Security tried diffrent authentications.

Any idea how to solve the issue.

Thanks
Jaison
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From jaison.joy at gmail.com  Thu Jun  1 10:11:18 2006
From: jaison.joy at gmail.com (Jaison Joy)
Date: Thu, 1 Jun 2006 19:41:18 +0530
Subject: [cisco-voip] IPCC Express /CCM Issue.
In-Reply-To: <e4aded970606010546v4d069043pe716d5f8a20cc038@mail.gmail.com>
References: <e4aded970606010546v4d069043pe716d5f8a20cc038@mail.gmail.com>
Message-ID: <e4aded970606010711t142b0601p90cdefa2dc166662@mail.gmail.com>

Forgot to add the following performed troubleshooting steps.

I tried with the IP address i.e. <http://<ipaddress>/appadmin>, edited the
lmhost file. - same Error message

I found few documents in microsoft where it said it is due to permission
issues given to the folders theni changed their permission according to that
document - same Error message.

I also found a document where it asked me to do add a dword ="
disableloopbackcheck"
under "HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Control\Lsa"........same
Error message



On 6/1/06, Jaison Joy <jaison.joy at gmail.com> wrote:
>
>  Hi ,
>
> Background:
> ----------
> I have an issue with the installation of the IPCC express After i finish
> the software installation.  and the Server reboots and when it comes up next
> time. It comes up first time with the webpage http://<ccm>/appadmin. it
> gives me error "You are not authorized to view this page" Error 401.1
>
> Under the event logs -> security, I see errors regarding unsuccessfull
> login by the IIS Accounts.
>
> After installing CRS in call manager. I have issues with the CTI manager
> and Cisco TCD services not starting.
> Every second their is an error about the CTI service Terminated.
> I see the following errors.
>
> Event Type: Error
> Event Source: Cisco CallManager
> Event Category: None
> Event ID: 3
> Date:  6/1/2006
> Time:  5:02:46 AM
> User:  N/A
> Computer: DELL
> Description:
> Error: GetNewFileName-fopen in wb mode failed to update existing file.
> szFileNumberName = c:\Program Files\Cisco\Trace\CTI\cti~num.bin.
>
> Event Type: Error
> Event Source: Service Control Manager
> Event Category: None
> Event ID: 7031
> Date:  6/1/2006
> Time:  4:57:41 AM
> User:  N/A
> Computer: DELL
> Description:
> The Cisco CTIManager service terminated unexpectedly.  It has done this 1
> time(s).  The following corrective action will be taken in 60000
> milliseconds: Restart the service.
>
> Problem statement:
> ------------------
> Issue 1:
> What am i missing during the CRS installtion.
>
> Issue 2:
> After loading CRS why CRS CTI manager and TCD.
>
> how can i solve these issues.
>
> Software version:
> ----------------
> IPCC Express 3.(03)a
> Callmanager 3.3.4
>
>
> Actions already performed:
> -------------------------
> 1. Followed Microsoft kb.
> http://support.microsoft.com/?kbid=896861.
> 2. Reconfigure all the services in the callmanager- admin utility.
> 3. Restarted the IIS.
> 4. Changed the execution permissions in IIS webpage.
> 5. Changed the Directory Security tried diffrent authentications.
>
> Any idea how to solve the issue.
>
> Thanks
>  Jaison
>
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From Greg_Hinson at adp.com  Thu Jun  1 10:50:15 2006
From: Greg_Hinson at adp.com (Hinson, Greg)
Date: Thu, 1 Jun 2006 09:50:15 -0500
Subject: [cisco-voip] IPCC Express /CCM Issue.
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BB31119@EXCHANGEDS01.ds.ad.adp.com>

I have a similar problem, when I installed CRS4.x co-resident with
callmanager 4.1.3(sr2). The appadmin page won't load due to an internet
explorer error, which I presume to be related to an IIS permissions
issue. You said you found documentation on how to fix that?
 
 
 
-Greg
 
________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jaison Joy
Sent: Thursday, June 01, 2006 7:11 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC Express /CCM Issue.
 
Forgot to add the following performed troubleshooting steps.
 
I tried with the IP address i.e. <http://<ipaddress>/appadmin>, edited
the lmhost file. - same Error message  
 
I found few documents in microsoft where it said it is due to permission
issues given to the folders theni changed their permission according to
that document - same Error message.
 
I also found a document where it asked me to do add a dword ="
disableloopbackcheck"
under
"HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Control\Lsa"........same
Error message


 
On 6/1/06, Jaison Joy <jaison.joy at gmail.com> wrote: 
Hi ,
 
Background:
----------
I have an issue with the installation of the IPCC express After i finish
the software installation.  and the Server reboots and when it comes up
next time. It comes up first time with the webpage
http://<ccm>/appadmin. it gives me error "You are not authorized to view
this page" Error 401.1
Under the event logs -> security, I see errors regarding unsuccessfull
login by the IIS Accounts.
After installing CRS in call manager. I have issues with the CTI manager
and Cisco TCD services not starting.
Every second their is an error about the CTI service Terminated.
I see the following errors.
Event Type: Error
Event Source: Cisco CallManager
Event Category: None
Event ID: 3
Date:  6/1/2006
Time:  5:02:46 AM
User:  N/A
Computer: DELL
Description:
Error: GetNewFileName-fopen in wb mode failed to update existing file.
szFileNumberName = c:\Program Files\Cisco\Trace\CTI\cti~num.bin. 
 
Event Type: Error
Event Source: Service Control Manager
Event Category: None
Event ID: 7031
Date:  6/1/2006
Time:  4:57:41 AM
User:  N/A
Computer: DELL
Description:
The Cisco CTIManager service terminated unexpectedly.  It has done this
1 time(s).  The following corrective action will be taken in 60000
milliseconds: Restart the service. 
Problem statement:
------------------
Issue 1:
What am i missing during the CRS installtion.
Issue 2:
After loading CRS why CRS CTI manager and TCD. 
how can i solve these issues.
Software version:
----------------
IPCC Express 3.(03)a
Callmanager 3.3.4

Actions already performed:
-------------------------
1. Followed Microsoft kb.
http://support.microsoft.com/?kbid=896861 .
2. Reconfigure all the services in the callmanager- admin utility. 
3. Restarted the IIS.
4. Changed the execution permissions in IIS webpage.
5. Changed the Directory Security tried diffrent authentications. 

Any idea how to solve the issue.
 
Thanks
Jaison
 


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 ml at nesys.it  Thu Jun  1 10:52:29 2006
From: ml at nesys.it (Andrea Riela)
Date: Thu, 1 Jun 2006 16:52:29 +0200
Subject: [cisco-voip] about QoS group
Message-ID: <baa53d1221540d64a16bda317eedfe58@nesys.it>

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

hi folks,

just a QoS question: what is a qos group exactly, and when it's 
recommended to use?
Could you send me more informations?

Thanks for your support
Regards
Andrea
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Version: GnuPG v1.4.1 (Darwin)

iD8DBQFEfv8vMakHrsrHP9wRAi+0AKDsNO4SQQwEkEzpqz6MMDKKvDCLpgCfQtjG
dBmOoCabS93/Q8ESlPhtUjw=
=+5Bg
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From jaison.joy at gmail.com  Thu Jun  1 11:57:38 2006
From: jaison.joy at gmail.com (Jaison Joy)
Date: Thu, 1 Jun 2006 21:27:38 +0530
Subject: [cisco-voip] IPCC Express /CCM Issue.
In-Reply-To: <fd950f1e0a1b95a6da683447fb80760e@cisco.com>
References: <e4aded970606010546v4d069043pe716d5f8a20cc038@mail.gmail.com>
	<e4aded970606010711t142b0601p90cdefa2dc166662@mail.gmail.com>
	<fd950f1e0a1b95a6da683447fb80760e@cisco.com>
Message-ID: <e4aded970606010857t3730ea1sa52fc133e7cd0d89@mail.gmail.com>

It doesnt ask me for UserID or Password It just directly gives me the same
error again and again.

I agree as it could be some permission issue with IIS, But i am not hitting
the right spot of the issue.

Call manager Service is Running fine I can make calls.
CTI manager services was also running before i installed CRS, but after
installation of CRS, CTIManager and cisco TCD services starts and gets
terminated.
I get the same error with the callManager 4.0.1.

Thanks,
Jaison

On 6/1/06, Kevin Thorngren <kthorngr at cisco.com> wrote:
>
> Not sure why you are getting the 401 errors.  Since this is the first
> time you are logging in are you using the user ID of Administrator and
> password of ciscocisco?
>
> The first app error below looks like a permissions issue.  Do you have
> the CallManager service running on this node?  The CallManager server
> needs to run on all nodes you are running CTI Manager on.
>
> Kevin
> On Jun 1, 2006, at 10:11 AM, Jaison Joy wrote:
>
> > Forgot to add the following performed troubleshooting steps.
> >
> > I tried with the IP address i.e. <http://<ipaddress>/appadmin>,edited
> > the lmhost file. - same Error message
> >
> > I found few documents in microsoft where it said it is due to
> > permission issues given to the folders theni changed their permission
> > according to that document - same Error message.
> >
> > I also found a document where it asked me to do add a dword ="
> > disableloopbackcheck"
> > under
> > "HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Control\Lsa"........same
> > Error message
> >
> >
> >
> > On 6/1/06, Jaison Joy <jaison.joy at gmail.com> wrote: Hi ,
> >>
> >> Background:
> >> ----------
> >> I have an issue with the installation of the IPCC express After i
> >> finish the software installation. and the Server reboots and when it
> >> comes up next time. It comes up first time with the webpage
> >> http://<ccm>/appadmin. it gives me error "You are not authorized to
> >> view this page" Error 401.1
> >>
> >> Under the event logs -> security, I see errors regarding
> >> unsuccessfull login by the IIS Accounts.
> >>
> >> After installing CRS in call manager. I have issues with the CTI
> >> manager and Cisco TCD services not starting.
> >> Every second their is an error about the CTI service Terminated.
> >> I see the following errors.
> >>
> >> Event Type: Error
> >> Event Source: Cisco CallManager
> >> Event Category: None
> >> Event ID: 3
> >> Date: 6/1/2006
> >> Time: 5:02:46 AM
> >> User: N/A
> >> Computer: DELL
> >> Description:
> >> Error: GetNewFileName-fopen in wb mode failed to update existing
> >> file. szFileNumberName = c:\Program
> >> Files\Cisco\Trace\CTI\cti~num.bin.
> >>
> >> Event Type: Error
> >> Event Source: Service Control Manager
> >> Event Category: None
> >> Event ID: 7031
> >> Date: 6/1/2006
> >> Time: 4:57:41 AM
> >> User: N/A
> >> Computer: DELL
> >> Description:
> >> The Cisco CTIManager service terminated unexpectedly. It has done
> >> this 1 time(s). The following corrective action will be taken in
> >> 60000 milliseconds: Restart the service.
> >>
> >> Problem statement:
> >> ------------------
> >> Issue 1:
> >> What am i missing during the CRS installtion.
> >>
> >> Issue 2:
> >> After loading CRS why CRS CTI manager and TCD.
> >>
> >> how can i solve these issues.
> >>
> >> Software version:
> >> ----------------
> >> IPCC Express 3.(03)a
> >> Callmanager 3.3.4
> >>
> >>
> >> Actions already performed:
> >> -------------------------
> >> 1. Followed Microsoft kb.
> >> http://support.microsoft.com/?kbid=896861 .
> >> 2. Reconfigure all the services in the callmanager- admin utility.
> >> 3. Restarted the IIS.
> >> 4. Changed the execution permissions in IIS webpage.
> >> 5. Changed the Directory Security tried diffrent authentications.
> >>
> >> Any idea how to solve the issue.
> >>
> >> Thanks
> >> Jaison_______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From BurwellJ at firstcharter.com  Thu Jun  1 10:59:13 2006
From: BurwellJ at firstcharter.com (Jason Burwell)
Date: Thu, 1 Jun 2006 10:59:13 -0400
Subject: [cisco-voip] VT Advantage 2.0 is HERE!!
Message-ID: <697161D0107BAD40AFE9A86B06509AC90643E708@FCEXCLUSTER.FirstCharter.com>

VT Advantage 2.0 is finally available for download from Cisco!! I can't
wait to try it out.
 
 
 
Jason T. Burwell
Infrastructure/Telecommunications
First Charter Bank
10200 David Taylor Drive
Charlotte, NC 28262
ph-704-688-4664
fax-704-688-4901
www.firstcharter.com
 


**********************************************************************
The information transmitted is intended only for the person or entity to which it is 
addressed and may contain confidential and/or privileged material. Any review,
retransmission, dissemination or other use of, or taking of any action in reliance upon,
this information by persons or entities other than the intended recipient is prohibited.
If you received this in error, please contact the sender and delete the material from
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From rleetun at co.boulder.co.us  Thu Jun  1 12:24:49 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Thu, 1 Jun 2006 10:24:49 -0600
Subject: [cisco-voip] VT Advantage 2.0 is HERE!!
Message-ID: <963CE94787005149A52C17F33A028E12051D47DF@EVS1B.boco.co.boulder.co.us>

Awesome!!

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burwell
Sent: Thursday, June 01, 2006 8:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VT Advantage 2.0 is HERE!!


VT Advantage 2.0 is finally available for download from Cisco!! I can't
wait to try it out.
 
 
 
Jason T. Burwell
Infrastructure/Telecommunications
First Charter Bank
10200 David Taylor Drive
Charlotte, NC 28262
ph-704-688-4664
fax-704-688-4901
www.firstcharter.com
 


**********************************************************************
The information transmitted is intended only for the person or entity to
which it is 
addressed and may contain confidential and/or privileged material. Any
review,
retransmission, dissemination or other use of, or taking of any action
in reliance upon,
this information by persons or entities other than the intended
recipient is prohibited.
If you received this in error, please contact the sender and delete the
material from
any computer. 
**********************************************************************

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From rick at trutechsystems.net  Thu Jun  1 12:49:33 2006
From: rick at trutechsystems.net (Rick Truong)
Date: Thu, 1 Jun 2006 09:49:33 -0700
Subject: [cisco-voip] 7912 IP Phone
In-Reply-To: <3E110434E2419D47883D119DD226ACA1010B02A7@kss-jhb-exch-01.kss.co.za>
Message-ID: <200606011649.k51GnUtY089474@puck.nether.net>

Hi Sechaba,

 

Have you tried the following procedure?

 


Resetting to factory defaults 



Step 1 Press the Menu button. 

Step 2 Use the Navigation button to select Settings, and then press the
Select softkey. 

Step 3 Use the Navigation button to select Network Configuration, and then
press the Select softkey. 

Step 4 Perform either of these procedures: 

Procedure A: 

*	Press **2. The phone displays "Do you want to reset all system
settings to default values?" 

 

*	Press the Yes softkey. 


Procedure B: 

*	Press **#. If your phone displays "Enter Admin Password," enter your
password and then press the Enter softkey. Make sure that an unlocked
padlock icon appears in the upper-right corner of your LCD. 

 

*	Scroll to Erase Configuration. 

 

*	Press the Yes softkey and then press the Save softkey. 


The phone cycles through normal startup procedures.

 

* We buy and sell new/used networking equipment.  Please contact me if you
have any offers or are interested in buying *

 

** VG224 NEW IN BOX @ $3400 **

** VG248 NEW IN BOX @ $4800 **

 

Thanks,

 

Rick Truong

TruTech Systems

39263 Half Moon Circle

Murrieta, CA 92563

Office: +1.951.696.3688

Mobile: +1.951.440.8711

AIM: ricktrutek

E-Mail: rick at trutechsystems.net

WEB: http://www.trutechsystems.net 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sechaba Mokoena
Sent: Thursday, June 01, 2006 2:48 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7912 IP Phone

 

To Whom It May Concern:

 

Does anyone know how to completely erase the configuration of a 7912 IP
Phone? I've noticed that if you move a 7912 from one Call Manager to another
the phone retains the IP address of the last Call Manager that it registered
with. I've also noticed that if you log into the phone using Extension
Mobility, it doesn't change the DN immediately; instead the phone resets,
and tries to register with the last Call Manager that it registered with
first. When the phone doesn't find the former Call Manager it then registers
with the new one. Is this common for this particular model?

 

Regards,

 

Sechaba Mokoena | Systems Engineer 

KSS Technologies (Pty) LTD.

KSS Convergence - Doing more with less

(IP Phone) +27 11 5456300|(IP Phonedirect) | +27 11 5456331| Fax)+27 11 545
6400| Mobile +27 84 798 6803 |

(e-mail)sechabam at kss.co.za

(website) <blocked::http://www.kss.co.za> http://www.kss.co.za

Block N, Central Park, 400 16th Road, Randjespark, Midrand, South Africa

 

 

Toll Free Support: 0861 6428 35 

 

  _____  

This message, including any attachments is intended only for the persons or
entity to which it is addressed, and may contain privileged or confidential
information. Retransmission, dissemination, copying, distribution,
modification and / or publication of this message by a person other than the
intended recipient is strictly prohibited by the sender. If you have
received this message in error, or you feel it is unacceptable
communication, a chain letter or offensive material of any nature, please
delete the material and any copies from all systems and report it to:

info at kss.co.za

  _____  

 

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From sodonnell at CCSINET.com  Thu Jun  1 12:53:17 2006
From: sodonnell at CCSINET.com (Scott O'Donnell)
Date: Thu, 1 Jun 2006 12:53:17 -0400
Subject: [cisco-voip] Callmanager configured with multiple remote CUEs.
Message-ID: <17A170DC8DD5CC44BA56B24B366EB60B01DACFF1@E3K-VS1.ccsi.local>

Does any one have experience with a centralized CallManager install
using SRST with CUE at the remote branch locations?
 
Seems to me that it doesn't present a major problems but I haven't done
it yet.
 
Any thoughts are appreciated.
 
Scott
 
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From kthorngr at cisco.com  Thu Jun  1 13:00:13 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu, 1 Jun 2006 13:00:13 -0400
Subject: [cisco-voip] IPCC Express /CCM Issue.
In-Reply-To: <e4aded970606010857t3730ea1sa52fc133e7cd0d89@mail.gmail.com>
References: <e4aded970606010546v4d069043pe716d5f8a20cc038@mail.gmail.com>
	<e4aded970606010711t142b0601p90cdefa2dc166662@mail.gmail.com>
	<fd950f1e0a1b95a6da683447fb80760e@cisco.com>
	<e4aded970606010857t3730ea1sa52fc133e7cd0d89@mail.gmail.com>
Message-ID: <99fe95a250980f1601dc8420be2113d7@cisco.com>

Well, I don't have any clues off the top of my head.  I have not seen 
where installing co-resident CRS would cause these types of issues.  I 
would recommend opening a TAC case and providing remote access to the 
system.

Kevin
On Jun 1, 2006, at 11:57 AM, Jaison Joy wrote:

> It doesnt ask me for UserID or Password It just directly gives me the 
> same error again and again.
>
> I agree as it could be some permission issue with IIS, But i am not 
> hitting the right spot of the issue.
>
> Call manager Service is Running fine I can make calls.
> CTI manager services was also running before i installed CRS, but 
> after installation of CRS, CTIManager and cisco TCD services starts 
> and gets terminated.
> I get the same error with the callManager 4.0.1.
> ?
> Thanks,
> Jaison
>
> On 6/1/06, Kevin Thorngren <kthorngr at cisco.com> wrote: Not sure why 
> you are getting the 401 errors.??Since this is the first
>> time you are logging in are you using the user ID of Administrator and
>> password of ciscocisco?
>>
>> The first app error below looks like a permissions issue.??Do you have
>> the CallManager service running on this node???The CallManager server
>> needs to run on all nodes you are running CTI Manager on.
>>
>> Kevin
>> On Jun 1, 2006, at 10:11 AM, Jaison Joy wrote:
>>
>> > Forgot to add the following performed troubleshooting steps.
>> >
>> > I tried with the IP address i.e. 
>> <http://<ipaddress>/appadmin>,edited
>> > the lmhost file. - same Error message
>> >
>> > I found few documents in microsoft where it said it is due to
>> > permission issues given to the folders theni changed their 
>> permission
>> > according to that document - same Error message.
>> >
>> > I also found a document where it asked me to do add a dword ="
>> > disableloopbackcheck"
>> > under
>> > 
>> "HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Control\Lsa"........same
>> > Error message
>> >
>> >
>> >
>> > On 6/1/06, Jaison Joy <jaison.joy at gmail.com> wrote: Hi ,
>> >>
>> >> Background:
>> >> ----------
>>  >> I have an issue with the installation of the IPCC express After i
>> >> finish the software installation. and the Server reboots and when 
>> it
>> >> comes up next time. It comes up first time with the webpage
>> >> http://<ccm>/appadmin. it gives me error "You are not authorized to
>> >> view this page" Error 401.1
>> >>
>> >> Under the event logs -> security, I see errors regarding
>>  >> unsuccessfull login by the IIS Accounts.
>> >>
>> >> After installing CRS in call manager. I have issues with the CTI
>> >> manager and Cisco TCD services not starting.
>> >> Every second their is an error about the CTI service Terminated.
>> >> I see the following errors.
>> >>
>> >> Event Type: Error
>> >> Event Source: Cisco CallManager
>> >> Event Category: None
>> >> Event ID: 3
>> >> Date: 6/1/2006
>> >> Time: 5:02:46 AM
>> >> User: N/A
>> >> Computer: DELL
>> >> Description:
>> >> Error: GetNewFileName-fopen in wb mode failed to update existing
>> >> file. szFileNumberName = c:\Program
>> >> Files\Cisco\Trace\CTI\cti~num.bin.
>> >>
>> >> Event Type: Error
>> >> Event Source: Service Control Manager
>> >> Event Category: None
>> >> Event ID: 7031
>> >> Date: 6/1/2006
>> >> Time: 4:57:41 AM
>> >> User: N/A
>> >> Computer: DELL
>> >> Description:
>> >> The Cisco CTIManager service terminated unexpectedly. It has done
>> >> this 1 time(s). The following corrective action will be taken in
>> >> 60000 milliseconds: Restart the service.
>> >>
>> >> Problem statement:
>> >> ------------------
>> >> Issue 1:
>> >> What am i missing during the CRS installtion.
>> >>
>> >> Issue 2:
>> >> After loading CRS why CRS CTI manager and TCD.
>> >>
>> >> how can i solve these issues.
>> >>
>> >> Software version:
>> >> ----------------
>> >> IPCC Express 3.(03)a
>> >> Callmanager 3.3.4
>> >>
>> >>
>> >> Actions already performed:
>> >> -------------------------
>> >> 1. Followed Microsoft kb.
>> >> http://support.microsoft.com/?kbid=896861 .
>> >> 2. Reconfigure all the services in the callmanager- admin utility.
>> >> 3. Restarted the IIS.
>> >> 4. Changed the execution permissions in IIS webpage.
>> >> 5. Changed the Directory Security tried diffrent authentications.
>> >>
>> >> Any idea how to solve the issue.
>> >>
>> >> Thanks
>> >> Jaison_______________________________________________
>> > cisco-voip mailing list
>> > cisco-voip at puck.nether.net
>> > https://puck.nether.net/mailman/listinfo/cisco-voip
>>
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From mborchers at igillc.com  Thu Jun  1 13:19:08 2006
From: mborchers at igillc.com (Mark Borchers)
Date: Thu, 1 Jun 2006 12:19:08 -0500
Subject: [cisco-voip] Initial AA Config Problems with CME
Message-ID: <006201c6859f$814a9200$4336a8c0@Traveler>

I'd be grateful for pointers to get AA working in the architecture
described below.

We have a voice T1 coming into a CME (3.3) router at a parent site,
which I'll refer to as CME-main.  Call forwarding is currently 
configured and working to allow a remote site, also running CME,
to make inbound and outbound calls on the T1.

I configured AA on CME-remote with unsuccessful results:
- When making an inbound test call to CME-remote, I don't hear the 
AA welcome message, nor does the dial-by-extension option work.
- However, I can press the appropriate phone button and access
a hunt group on CME-remote.  These two observations made me think
that the queue service was starting but not the aa service.
- However, when I show application sessions during the test call,
I observe that the aa service *did* start.

Since the T1 terminates on CME-main, I have the dial-peer session
target for the aa service mapped to loopback.  Maybe somebody can
confirm if this is correct.  Any other light anyone can shed based
on a similar architecture would be appreciated.

TIA,
Mark


Relevant config on CME-remote:

application
 service queue flash:app-b-acd-2.1.0.0.tcl
  param queue-len 6
  param aa-hunt1 2201
  param number-of-hunt-grps 1
  param queue-manager-debugs 0
 !
 service aa flash:app-b-acd-aa-2.1.0.0.tcl
  paramspace english index 0
  paramspace english language en
  paramspace english location flash:
  param handoff-string aa
  param dial-by-extension-option 2
  param voice-mail 2200
  param welcome-prompt flash:_bacd_enter_dest.au
  param aa-pilot 3299
  param service-name queue
!
interface Loopback0
 ip address 10.0.0.4 255.255.255.255
 h323-gateway voip bind srcaddr 10.0.0.4
!
!
dial-peer voice 3200 voip
 description RTS IP Phones
 destination-pattern 32..
 session target ipv4:10.0.2.2
!
dial-peer voice 99 voip
 description Outgoing PBX calls to PSTN
 destination-pattern 99...........
 session target ipv4:10.0.2.2
 dtmf-relay h245-alphanumeric
!
dial-peer voice 55 voip
 description Outgoing PBX calls
 destination-pattern 5....
 session target ipv4:10.0.2.2
 dtmf-relay h245-alphanumeric
!
dial-peer voice 3299 voip
 service aa
 destination-pattern 3299
 session target ipv4:10.0.0.4
 incoming called-number 3299
!
!
!
!
telephony-service
 load 7910 P00403020214
 load 7960-7940 P00305000301
 max-ephones 12
 max-dn 24
 ip source-address 10.0.0.4 port 2000
 auto assign 1 to 3
 create cnf-files version-stamp Jan 01 2002 00:00:00
 max-conferences 12 gain -6
 call-forward pattern 9...........
 transfer-system full-consult
 transfer-pattern 32..
 secondary-dialtone 99
!



From Sarkany at nts.at  Thu Jun  1 13:33:34 2006
From: Sarkany at nts.at (August Sarkany)
Date: Thu, 1 Jun 2006 19:33:34 +0200
Subject: [cisco-voip] PSTN Infaces in Russia
Message-ID: <5A8C69748176434F92F1EECD7D21F72C64E50A@srvgraz03.nts.at>

Hi all,

 

we need information which telephony standards are available in Russia?

(analog, digital lines (ISDN, E1))

 

And which interface to use with?

 

Many thanks 

 

August

 

 

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From d_abelski at inbox.ru  Thu Jun  1 14:01:35 2006
From: d_abelski at inbox.ru (d_abelski at inbox.ru)
Date: Fri, 2 Jun 2006 00:01:35 +0600
Subject: [cisco-voip] PSTN Infaces in Russia
In-Reply-To: <5A8C69748176434F92F1EECD7D21F72C64E50A@srvgraz03.nts.at>
References: <5A8C69748176434F92F1EECD7D21F72C64E50A@srvgraz03.nts.at>
Message-ID: <683005969.20060602000135@inbox.ru>

ISDN PRI, E1 R1.5 national but there are converters, BRI, FXO, FXS...
 Id say all are availiable.. R1.5 is the legacy one that makes all the
 problems.
AS> Hi all,

AS> ?

AS> we need information which telephony standards areavailable in Russia?

AS> (analog, digital lines (ISDN, E1))

AS> ?

AS> And which interface to use with?

AS> ?

AS> Many thanks 

AS> ?

AS> August

AS> ?

AS> ?




AS> __________ ?????????? NOD32 1.1569 (20060531) __________

AS> ??? ????????? ????????? ???????????? ???????? NOD32.
AS> http://www.eset.com





-- 
Regards, Daniel                        mailto:d_abelski at inbox.ru


From ErasmuE4 at telkom.co.za  Thu Jun  1 17:09:35 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Thu, 1 Jun 2006 23:09:35 +0200
Subject: [cisco-voip] 7912 IP Phone
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D52F@CNTRRA20-XCS01.telkom.co.za>


prbably like most phones - settings, network settings, press **# to
unlock settings, scroll to erase config, save and wait.


erik

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Sechaba Mokoena
Sent: Thu 2006-06-01 11:48
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7912 IP Phone



To Whom It May Concern:



Does anyone know how to completely erase the configuration of a 7912
IP Phone? I've noticed that if you move a 7912 from one Call Manager
to another the phone retains the IP address of the last Call Manager
that it registered with. I've also noticed that if you log into the
phone using Extension Mobility, it doesn't change the DN immediately;
instead the phone resets, and tries to register with the last Call
Manager that it registered with first. When the phone doesn't find the
former Call Manager it then registers with the new one. Is this common
for this particular model?



Regards,



Sechaba Mokoena | Systems Engineer

KSS Technologies (Pty) LTD.

KSS Convergence - Doing more with less

(IP Phone) +27 11 5456300|(IP Phonedirect) | +27 11 5456331| Fax)+27
11 545 6400| Mobile +27 84 798 6803 |

(e-mail)sechabam at kss.co.za

(website)http://www.kss.co.za

Block N, Central Park, 400 16th Road, Randjespark, Midrand, South
Africa





Toll Free Support: 0861 6428 35



________________________________

This message, including any attachments is intended only for the
persons or entity to which it is addressed, and may contain privileged
or confidential information. Retransmission, dissemination, copying,
distribution, modification and / or publication of this message by a
person other than the intended recipient is strictly prohibited by the
sender. If you have received this message in error, or you feel it is
unacceptable communication, a chain letter or offensive material of
any nature, please delete the material and any copies from all systems
and report it to:
info at kss.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 ErasmuE4 at telkom.co.za  Thu Jun  1 17:37:06 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Thu, 1 Jun 2006 23:37:06 +0200
Subject: [cisco-voip] configure IOS MOH for remote site
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D532@CNTRRA20-XCS01.telkom.co.za>


Hi Anthony

I need to do exactly the same but could not get it working
immediately. I will go back to the lab again. I suspect it is because
my MOH in multicast mode was not working at my HQ. It was in unicast
mode. I did manage to get the MOH for CME fallback mode working and in
fallback mode it is working but when the site s wan link is up I can
not get MOH - just tone on hold. My branch is using g711 for voice in
the branch and G729 across the wan to HQ. Apart from getting the MOH
at HQ working - are there any special tricks - Do I need MOH at HQ in
G711 or G729 mode or does it not matter - I.E a g729 region for my MOH
to make sure the MOH signalling to fool the phones into listening to
the stream at the branch etc or does it work irrespective of the codec
at HQ.

Another question - the MOH sample file for a branch - is it a G711 or
G729 ??

The one thing that I did was to get the branch going on MOH when in
CME fallback mode (not SRST) is the command where you tie the MOH to a
loopback. I had to also add the fast etehr IP address in the command -
the same address as my cme skinny interface


erik

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Anthony
Kouloglou
Sent: Wed 2006-05-31 15:32
To: Goran Selthofer
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] configure IOS MOH for remote site


Hi Goran,
i made it work thanks to you and this link below

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_featur
e_guide09186a00802d1c31.html

but
when i change the .au file for a diferrent audio file, i can't stream
it to the phones unless i reload the router. Is there a way to reset
the music on hold server or something like that?

cheers
Akoul
Goran Selthofer wrote:

	anthony, this MoH is active always at multicast address...not
only for srst.
	that it why you should limit MoH server to one hop...because
we are practicaly cheating it with moh file from flash...
	let me explain:
	when you put someone on hold (ip phone or external caller
trough voice gateway), those devices expect to "hear" MOH on multicast
address (practically, they became memebers fo multicast group).
	So, instead receiveing direct moh from MoH server (remmember,
we limited it on one hop), the devices will hear MOH file from
router's flash which will be played on 239.1.1.1 port 16384 - exactly
where they are listening... :)

	believe me...it works fine for over 50 locations in my
installation :)

	cu,
	goran

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

		Thanks goran but,
		with this configuration, the MOH server is active only
when srst is active. I want that the MOH server is always active and
so:
		assign this IOS MOH to a media resource group list
that only the specific remote site uses.
		Is this possible?
	
		-akoul
	
		Goran Selthofer wrote:

			ccm-manager music-on-hold
			!
			call-manager-fallback
			???moh music-on-hold.au
			???multicast moh 239.1.1.1 port 16384 route
<IP address of loopback interface> <space> <IP address of some
physical interface>
			???
			you should copy file "music-on-hold.au" to
flash
			email me if you need this file
		
			this example is for g711 MoH stream to ip
phones.
			at the same time, you should configure one of
the servers in cluster as MoH multicast server but limit it to one
hop.
		
			cu,
			selt
		
		
			Anthony Kouloglou <akoul at dataways.gr>
<mailto:akoul at dataways.gr>  wrote:

				Hi all,
				is there documentation on how to
configure a IOS to act as MOH server
				(2851 series router). i just don't
want the remote site to stream to the
				local users music from the wan.
			
				central site: cluster with 2 ccm
4.1(3)sr3a
				remote sites: 3 each with 2851 series
router and srst.
			
				Thanks

_______________________________________________
				cisco-voip mailing list
				cisco-voip at puck.nether.net

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


________________________________

			Love cheap thrills? Enjoy PC-to-Phone calls to
30+ countries
<http://us.rd.yahoo.com/mail_us/taglines/postman9/*http://us.rd.yahoo.
com/evt=39666/*http://messenger.yahoo.com/>  for just 2??/min with
Yahoo! Messenger with Voice.




________________________________

	Love cheap thrills? Enjoy PC-to-Phone calls to 30+ countries
<http://us.rd.yahoo.com/mail_us/taglines/postman9/*http://us.rd.yahoo.
com/evt=39666/*http://messenger.yahoo.com/>  for just 2?/min with
Yahoo! Messenger with Voice.



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
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From Joe_Bell at adp.com  Thu Jun  1 19:26:44 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Thu, 1 Jun 2006 18:26:44 -0500
Subject: [cisco-voip] configure IOS MOH for remote site
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BB317BE@EXCHANGEDS01.ds.ad.adp.com>

Tone on hold tells you CCM is configured in such a way that it does not think MoH is deliverable to the end point, so you get tone.  If you get dead air, your network is not configured to deliver the multicast streams.  Codec will have nothing to do with the inclusion of a phone to the multicast group.  Unless of course the regions settings force the delivery of a g.729 multicast stream and the MoH server is not configured to support that natively.  You cannot transcode a multicast stream, so it must be sourced with the proper codec from the server.  To source a MoH multicast stream in g.729 format:  

 

Go into Service Parameter, select Cisco Voice IP Media Streaming App, scroll to the bottom and highlight both g.711 and g.729 codecs.  Do that with the control key and up date.

 

Verify CCM is handing out a multicast stream by using PerfMon as well.  With that tool, you can see exactly what CCM thinks it should do.  

 

Hope this helps,

 

Joe

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus (E)
Sent: Thursday, June 01, 2006 2:37 PM
To: Anthony Kouloglou; Goran Selthofer
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] configure IOS MOH for remote site

 

Hi Anthony

 

I need to do exactly the same but could not get it working immediately. I will go back to the lab again. I suspect it is because my MOH in multicast mode was not working at my HQ. It was in unicast mode. I did manage to get the MOH for CME fallback mode working and in fallback mode it is working but when the site s wan link is up I can not get MOH - just tone on hold. My branch is using g711 for voice in the branch and G729 across the wan to HQ. Apart from getting the MOH at HQ working - are there any special tricks - Do I need MOH at HQ in G711 or G729 mode or does it not matter - I.E a g729 region for my MOH to make sure the MOH signalling to fool the phones into listening to the stream at the branch etc or does it work irrespective of the codec at HQ.

 

Another question - the MOH sample file for a branch - is it a G711 or G729 ??

 

The one thing that I did was to get the branch going on MOH when in CME fallback mode (not SRST) is the command where you tie the MOH to a loopback. I had to also add the fast etehr IP address in the command - the same address as my cme skinny interface

 

 

erik

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Anthony Kouloglou
Sent: Wed 2006-05-31 15:32
To: Goran Selthofer
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] configure IOS MOH for remote site

Hi Goran,
i made it work thanks to you and this link below

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_feature_guide09186a00802d1c31.html

but
when i change the .au file for a diferrent audio file, i can't stream it to the phones unless i reload the router. Is there a way to reset the music on hold server or something like that?

cheers
Akoul
Goran Selthofer wrote: 

anthony, this MoH is active always at multicast address...not only for srst.
that it why you should limit MoH server to one hop...because we are practicaly cheating it with moh file from flash...
let me explain:
when you put someone on hold (ip phone or external caller trough voice gateway), those devices expect to "hear" MOH on multicast address (practically, they became memebers fo multicast group).
So, instead receiveing direct moh from MoH server (remmember, we limited it on one hop), the devices will hear MOH file from router's flash which will be played on 239.1.1.1 port 16384 - exactly where they are listening... :)

believe me...it works fine for over 50 locations in my installation :)

cu,
goran

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

Thanks goran but,
with this configuration, the MOH server is active only when srst is active. I want that the MOH server is always active and so:
assign this IOS MOH to a media resource group list that only the specific remote site uses.
Is this possible?

-akoul

Goran Selthofer wrote: 

ccm-manager music-on-hold
!
call-manager-fallback
???moh music-on-hold.au
???multicast moh 239.1.1.1 port 16384 route <IP address of loopback interface> <space> <IP address of some physical interface>
???
you should copy file "music-on-hold.au" to flash
email me if you need this file

this example is for g711 MoH stream to ip phones.
at the same time, you should configure one of the servers in cluster as MoH multicast server but limit it to one hop.

cu,
selt


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

Hi all,
is there documentation on how to configure a IOS to act as MOH server 
(2851 series router). i just don't want the remote site to stream to the 
local users music from the wan.

central site: cluster with 2 ccm 4.1(3)sr3a
remote sites: 3 each with 2851 series router and srst.

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

 

________________________________

Love cheap thrills? Enjoy PC-to-Phone calls to 30+ countries <http://us.rd.yahoo.com/mail_us/taglines/postman9/*http:/us.rd.yahoo.com/evt=39666/*http:/messenger.yahoo.com/>  for just 2??/min with Yahoo! Messenger with Voice. 

 

________________________________

Love cheap thrills? Enjoy PC-to-Phone calls to 30+ countries <http://us.rd.yahoo.com/mail_us/taglines/postman9/*http:/us.rd.yahoo.com/evt=39666/*http:/messenger.yahoo.com/>  for just 2?/min with Yahoo! Messenger with Voice. 

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From chris at chrisserafin.com  Fri Jun  2 00:21:48 2006
From: chris at chrisserafin.com (ChrisSerafin)
Date: Thu, 01 Jun 2006 23:21:48 -0500
Subject: [cisco-voip] about QoS group
In-Reply-To: <baa53d1221540d64a16bda317eedfe58@nesys.it>
References: <baa53d1221540d64a16bda317eedfe58@nesys.it>
Message-ID: <447FBCDC.1010006@chrisserafin.com>

I believe the QoS groups are related to QoS  policy propagation via BGP:

http://www.cisco.com/univercd/cc/td/doc/product/software/ios111/cc111/bgpprop.htm

cisco80211
Chris Serafin
IT Security / Cisco VoIP Engineer
chris at chriserafin.com
Cisco Security Clearinghouse - under creation - chrisserafin.com
 



Andrea Riela wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> hi folks,
>
> just a QoS question: what is a qos group exactly, and when it's 
> recommended to use?
> Could you send me more informations?
>
> Thanks for your support
> Regards
> Andrea
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.1 (Darwin)
>
> iD8DBQFEfv8vMakHrsrHP9wRAi+0AKDsNO4SQQwEkEzpqz6MMDKKvDCLpgCfQtjG
> dBmOoCabS93/Q8ESlPhtUjw=
> =+5Bg
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>   


From vignesh at novanet.net  Fri Jun  2 02:59:29 2006
From: vignesh at novanet.net (vignesh)
Date: Fri, 2 Jun 2006 12:29:29 +0530
Subject: [cisco-voip] suppressing outbound leg on ipipgw
Message-ID: <F5DA4C2928E99A4DAF41CC88A6C1B27603E694@server.NOVANET.local>

hello all 
i am using cisco ipipgw and free radius. for a single call i get 2
records. one for each leg. now i want to supress one of the leg mostly
outbound. i cant use gw-accounting suppress as both the legs are voip. i
even tried voice class aaa accounting suppress outbound leg but still no
luck any idea guys how can i do this. 
thanks in advance 

Vignesh


From lex at sandy.ru  Fri Jun  2 05:29:16 2006
From: lex at sandy.ru (Alex A. Pavlenko)
Date: Fri, 2 Jun 2006 13:29:16 +0400
Subject: [cisco-voip] Line group membership
Message-ID: <05b601c68627$04eec040$19477dc1@corporate.sandy.ru>

Hi!

First, let me explain the situation.We have a group of phones that serves all incoming calls
to our company. Operator after accepting a call forwards it to appropriate person. Those
phones organized as a line group with broadcast distribution algorithm in CCM.
Scenario is rather simple, however there is a big problem in it. Operators are humans
so they never stay on their sits for a whole day. In the periods of time when some operators
are busy and some are left customer calling to our company gets no answer and thinks
that he is ignored. All I want is to give a busy tone to customer during described periods
of time. The only thing I see to achieve this is to make operators to switch the phone off before
leaving. This is very ugly solution however. In fact it is sufficent to exclude phone
line number from a line group when operator leaves and include it back into line group
when operator returns, but only administrator can accomplish this task. Is there
a possibility, may be some kind of IP Phone service or anything else to let the phone user
to change Line group membershit by himself? Any other suggestions?

Thanks.

--
Alex Pavlenko
System Administrator
Sandy Info



From erickbe at yahoo.com  Fri Jun  2 08:28:05 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri, 2 Jun 2006 05:28:05 -0700 (PDT)
Subject: [cisco-voip] Line group membership
In-Reply-To: <05b601c68627$04eec040$19477dc1@corporate.sandy.ru>
Message-ID: <20060602122805.60995.qmail@web37211.mail.mud.yahoo.com>

Call Manager 4.2 has a hunt-group login/logout toggle where a user can press the "Hlog" softkey on their phone and toggle themselves logged in/out of hunt groups. This needs to be added to the softkey template user has assigned to their phone and their is a phone option that needs to be checked as well.

Also, if you can't go to 4.2 for some reason, you may want to consider attendant console. I know it needs software app on PC but it lets users go offline when they go away from their desk. 

Do you have VM or call center of some sort? You can set the forward hunt no answer/busy setting on hunt pilot to go to a auto-attendant or VM box when no operator is available instead of giving caller busy tone. To me, when I call a company and get busy signals it appears they may be having problems with their equipment and/or phone lines. 

Erick

----- Original Message ----
From: Alex A. Pavlenko <lex at sandy.ru>
To: cisco-voip at puck.nether.net
Sent: Friday, June 2, 2006 4:29:16 AM
Subject: [cisco-voip] Line group membership

Hi!

First, let me explain the situation.We have a group of phones that serves all incoming calls
to our company. Operator after accepting a call forwards it to appropriate person. Those
phones organized as a line group with broadcast distribution algorithm in CCM.
Scenario is rather simple, however there is a big problem in it. Operators are humans
so they never stay on their sits for a whole day. In the periods of time when some operators
are busy and some are left customer calling to our company gets no answer and thinks
that he is ignored. All I want is to give a busy tone to customer during described periods
of time. The only thing I see to achieve this is to make operators to switch the phone off before
leaving. This is very ugly solution however. In fact it is sufficent to exclude phone
line number from a line group when operator leaves and include it back into line group
when operator returns, but only administrator can accomplish this task. Is there
a possibility, may be some kind of IP Phone service or anything else to let the phone user
to change Line group membershit by himself? Any other suggestions?

Thanks.

--
Alex Pavlenko
System Administrator
Sandy Info


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





From ealeatherman at gmail.com  Fri Jun  2 08:29:46 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 2 Jun 2006 08:29:46 -0400
Subject: [cisco-voip] Line group membership
In-Reply-To: <05b601c68627$04eec040$19477dc1@corporate.sandy.ru>
References: <05b601c68627$04eec040$19477dc1@corporate.sandy.ru>
Message-ID: <94a1afde0606020529n78d5a14bp1722cdae91d19754@mail.gmail.com>

You could put extension mobility on the phones and have them log in to get a
profile that is a member of that line group.

CCM 4.2 has a feature like you are requesting,
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/prod_release_note09186a00805f1470.html#wp2440103



On 6/2/06, Alex A. Pavlenko <lex at sandy.ru> wrote:
>
> Hi!
>
> First, let me explain the situation.We have a group of phones that serves
> all incoming calls
> to our company. Operator after accepting a call forwards it to appropriate
> person. Those
> phones organized as a line group with broadcast distribution algorithm in
> CCM.
> Scenario is rather simple, however there is a big problem in it. Operators
> are humans
> so they never stay on their sits for a whole day. In the periods of time
> when some operators
> are busy and some are left customer calling to our company gets no answer
> and thinks
> that he is ignored. All I want is to give a busy tone to customer during
> described periods
> of time. The only thing I see to achieve this is to make operators to
> switch the phone off before
> leaving. This is very ugly solution however. In fact it is sufficent to
> exclude phone
> line number from a line group when operator leaves and include it back
> into line group
> when operator returns, but only administrator can accomplish this task. Is
> there
> a possibility, may be some kind of IP Phone service or anything else to
> let the phone user
> to change Line group membershit by himself? Any other suggestions?
>
> Thanks.
>
> --
> Alex Pavlenko
> System Administrator
> Sandy Info
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From francisco.sedano at minx.com  Fri Jun  2 10:00:25 2006
From: francisco.sedano at minx.com (Francisco Sedano)
Date: Fri, 2 Jun 2006 15:00:25 +0100
Subject: [cisco-voip] TFTP escape characters
Message-ID: <91E474CF2918114D82BE11A57295A2C0457BBD@EXCHANGE2005.minxtechnology.local>

Hi, friends.

 

I'm trying to configure a background image in a IPCommunicator
registered to a Callmanager Express. 

The problem I'm finding is that the IPCommunicator is asking to ftp
server for file Desktops\320x212x12\List.xml (note the backslashes). 

 

When I try to configure that filename in ios tftp server (with or
without an alias), IOS converts \320 to the letter P.

 

I've tried all combinations I can think of to escape that character, but
with no luck. Any ideas? 

7970 is perfect so far, since it asks for Desktops/320x212x12/List.xml

 

GW_VOZ_LAB#copy tftp://192.16.10.36/List.xml flash:Des

GW_VOZ_LAB#$/192.16.10.36/List.xml flash:Desktops\320x212x12\List.xml

Accessing tftp://192.16.10.36/List.xml...

Loading List.xml from 192.16.10.36 (via GigabitEthernet0/0): !

[OK - 147 bytes]

 

%Error opening flash:Desktops\320x212x12\List.xml (Illegal file
character)

GW_VOZ_LAB#

 

GW_VOZ_LAB(config)#$ flash:List.xml alias Desktops\320x212x12\List.xml

GW_VOZ_LAB(config)#do sh run | i tftp-server

tftp-server flash:Desktops/320x212x12/List.xml

tftp-server flash:Desktops/320x212x12/TN-homer1.png

tftp-server flash:Desktops/320x212x12/homer1.png

tftp-server flash:Desktops\320x212x12\List.xml

tftp-server flash:List.xml alias DesktopsPx212x12\List.xml

GW_VOZ_LAB(config)#

 

 

Francisco Sedano (CCIE #14859)
Senior IPT Consultant 
MINX Technology

main:     +44(0)870 600 6150
email:      francisco.sedano at minx.com <mailto:denise.duignan at minx.com> 
web:        www.minx.com <http://www.minx.com/> 

 
Privileged/Confidential Information may be contained in this message. 

If you are not the addressee indicated in this message (or responsible
for delivery of the message to such person), you may not copy or deliver
this message to anyone. 

In such case, you should destroy this message and kindly notify the
sender by reply email. 

Whilst every effort is made to ensure that emails do not contain viruses
this cannot be guaranteed and MINX Technology can accept no
responsibility or liability for any loss
or damage caused by the inadvertent transmission

 

 

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From Scott.Voll at wesd.org  Fri Jun  2 11:18:36 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 2 Jun 2006 08:18:36 -0700
Subject: [cisco-voip] VT Advantage 2.0 is HERE!!
Message-ID: <D713462ED535184D830F6F6301753E7E01E8F3BF@VISHNU.wesd.org>

So has anyone tried it with IPC. yet?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Leetun, Rob
Sent: Thursday, June 01, 2006 9:25 AM
To: Jason Burwell; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VT Advantage 2.0 is HERE!!

 

Awesome!!

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burwell
Sent: Thursday, June 01, 2006 8:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VT Advantage 2.0 is HERE!!

VT Advantage 2.0 is finally available for download from Cisco!! I can't
wait to try it out.

 

 

 

Jason T. Burwell

Infrastructure/Telecommunications

First Charter Bank

10200 David Taylor Drive

Charlotte, NC 28262

ph-704-688-4664

fax-704-688-4901

www.firstcharter.com

 



**********************************************************************
The information transmitted is intended only for the person or entity to
which it is 
addressed and may contain confidential and/or privileged material. Any
review,
retransmission, dissemination or other use of, or taking of any action
in reliance upon,
this information by persons or entities other than the intended
recipient is prohibited.
If you received this in error, please contact the sender and delete the
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From rleetun at co.boulder.co.us  Fri Jun  2 11:22:19 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Fri, 2 Jun 2006 09:22:19 -0600
Subject: [cisco-voip] VT Advantage 2.0 is HERE!!
Message-ID: <963CE94787005149A52C17F33A028E12051D47F6@EVS1B.boco.co.boulder.co.us>

I have it installed on a laptop that I connect to our environment
through VPN and a Verizon EDO card.  It is a bit rastery return video
feed, but the send video feed to the other video phone that is on the
lan is great.  For the most part it looks and works nice.  Thanks
Cisco!!

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Friday, June 02, 2006 9:19 AM
To: Leetun, Rob; Jason Burwell; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] VT Advantage 2.0 is HERE!!



So has anyone tried it with IPC. yet?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Leetun, Rob
Sent: Thursday, June 01, 2006 9:25 AM
To: Jason Burwell; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VT Advantage 2.0 is HERE!!

 

Awesome!!

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burwell
Sent: Thursday, June 01, 2006 8:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VT Advantage 2.0 is HERE!!

VT Advantage 2.0 is finally available for download from Cisco!! I can't
wait to try it out.

 

 

 

Jason T. Burwell

Infrastructure/Telecommunications

First Charter Bank

10200 David Taylor Drive

Charlotte, NC 28262

ph-704-688-4664

fax-704-688-4901

www.firstcharter.com

 



**********************************************************************
The information transmitted is intended only for the person or entity to
which it is 
addressed and may contain confidential and/or privileged material. Any
review,
retransmission, dissemination or other use of, or taking of any action
in reliance upon,
this information by persons or entities other than the intended
recipient is prohibited.
If you received this in error, please contact the sender and delete the
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From pmowry at getgds.com  Fri Jun  2 11:29:00 2006
From: pmowry at getgds.com (Patrick Mowry)
Date: Fri, 2 Jun 2006 10:29:00 -0500
Subject: [cisco-voip] A physical attendant console for CallManager?
In-Reply-To: <mailman.2901.1149256833.1478.cisco-voip@puck.nether.net>
Message-ID: <4C792A099259634C83F97067671B37FE021D8E40@GDSEXC.gds.pri>

Hello,
  I have a requirement for an attendant console that needs to be able to
monitor over 100 lines and they refuse to use a computer.  Does anyone
know of a 3rd party phone or sidecar that can do this with CallManger
4.1 or 5.0?

Thanks,

-Patrick


From BillS at tns.its.psu.edu  Fri Jun  2 11:47:10 2006
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Fri, 2 Jun 2006 11:47:10 -0400 
Subject: [cisco-voip] A physical attendant console for CallManager?
Message-ID: <07A14D6987A5F647B2415D72EDF63B5604F3D0A3@tnsex01.tns.its.psu.edu>

It's like refusing to use a mouse to control a graphical operating system.
You can do it, if you're really stubborn and patient, or you can do it the
right way and just learn to use a mouse.

Perhaps if you dress up the computer like a big telephone that will help...
 

> -----Original Message-----
> From: Patrick Mowry [mailto:pmowry at getgds.com] 
> Sent: Friday, June 02, 2006 11:29 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] A physical attendant console for CallManager?
> 
> Hello,
>   I have a requirement for an attendant console that needs to 
> be able to
> monitor over 100 lines and they refuse to use a computer.  Does anyone
> know of a 3rd party phone or sidecar that can do this with CallManger
> 4.1 or 5.0?
> 
> Thanks,
> 
> -Patrick
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

From BRogers at pasadenaisd.org  Fri Jun  2 12:29:00 2006
From: BRogers at pasadenaisd.org (Brad Rogers)
Date: Fri, 2 Jun 2006 11:29:00 -0500
Subject: [cisco-voip] 7920, ACS and AP 1200
Message-ID: <CB3DDB018C56BB42A32D5C63006B76ED01003B62@adminexchg03.pasadenaisd.org>

Does anyone have a configuration example for a 7920 on a 1200 series
access point and ACS?  I can get the phone to authenticate using 128 bit
WEP, but when I try to use LEAP authentication fails.  LEAP is enabled
in the ACS and I have setup a username and password.

 

Brad Rogers

Network Systems Analyst

Pasadena ISD - Network Services

 

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From Scott.Voll at wesd.org  Fri Jun  2 12:38:03 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 2 Jun 2006 09:38:03 -0700
Subject: [cisco-voip] VoIP through PIX
Message-ID: <D713462ED535184D830F6F6301753E7E01E8F3DF@VISHNU.wesd.org>

I have a Pix running 6.3.4 and call manager 4.1.3sr3a)on the inside of
the Pix.  I placed a 7960 (skinny) on the outside and it looks like it
reaches the TFTP server (nat'd address) and then gets the CM IP
addresses.  But the CM IP's are the inside addresses and it never
registers with the CM.

 

I have both the TFTP and SCCP fixups and the ACL's are allowing IP any
any to and from the CM's and phone.  Any idea what is wrong?

 

Thanks

 

Scott

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From SPackett at fenwick.com  Fri Jun  2 13:04:08 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Fri, 2 Jun 2006 10:04:08 -0700
Subject: [cisco-voip] 7920, ACS and AP 1200
Message-ID: <965C0255FD5828438E34F9A68ECA76E740B511@svcexch02.firm.fenwick.llp>

Are you also setting up the phone to use LEAP by entering the same LEAP
username and password to match the LEAP username and password in ACS?

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Rogers
Sent: Friday, June 02, 2006 9:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7920, ACS and AP 1200



Does anyone have a configuration example for a 7920 on a 1200 series
access point and ACS?  I can get the phone to authenticate using 128 bit
WEP, but when I try to use LEAP authentication fails.  LEAP is enabled
in the ACS and I have setup a username and password.

 

Brad Rogers

Network Systems Analyst

Pasadena ISD - Network Services

 

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From BRogers at pasadenaisd.org  Fri Jun  2 13:56:33 2006
From: BRogers at pasadenaisd.org (Brad Rogers)
Date: Fri, 2 Jun 2006 12:56:33 -0500
Subject: [cisco-voip] 7920, ACS and AP 1200
Message-ID: <CB3DDB018C56BB42A32D5C63006B76ED01003B64@adminexchg03.pasadenaisd.org>

Yes.

 

________________________________

From: Stu Packett [mailto:SPackett at fenwick.com] 
Sent: Friday, June 02, 2006 12:04 PM
To: Brad Rogers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] 7920, ACS and AP 1200

 

Are you also setting up the phone to use LEAP by entering the same LEAP
username and password to match the LEAP username and password in ACS?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Rogers
Sent: Friday, June 02, 2006 9:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7920, ACS and AP 1200

Does anyone have a configuration example for a 7920 on a 1200 series
access point and ACS?  I can get the phone to authenticate using 128 bit
WEP, but when I try to use LEAP authentication fails.  LEAP is enabled
in the ACS and I have setup a username and password.

 

Brad Rogers

Network Systems Analyst

Pasadena ISD - Network Services

 

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From CDAmore at jackhenry.com  Fri Jun  2 14:00:38 2006
From: CDAmore at jackhenry.com (Chris D'amore)
Date: Fri, 2 Jun 2006 13:00:38 -0500
Subject: [cisco-voip] 7920, ACS and AP 1200
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002A00566@smoxchg.jhacorp.com>

Are you seeing anything in the logs on the ACS? 

Chris

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Rogers
Sent: Friday, June 02, 2006 12:57 PM
To: Stu Packett; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7920, ACS and AP 1200

 

Yes.

 

________________________________

From: Stu Packett [mailto:SPackett at fenwick.com] 
Sent: Friday, June 02, 2006 12:04 PM
To: Brad Rogers; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] 7920, ACS and AP 1200

 

Are you also setting up the phone to use LEAP by entering the same LEAP
username and password to match the LEAP username and password in ACS?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brad Rogers
Sent: Friday, June 02, 2006 9:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7920, ACS and AP 1200

Does anyone have a configuration example for a 7920 on a 1200 series
access point and ACS?  I can get the phone to authenticate using 128 bit
WEP, but when I try to use LEAP authentication fails.  LEAP is enabled
in the ACS and I have setup a username and password.

 

Brad Rogers

Network Systems Analyst

Pasadena ISD - Network Services

 

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 jonvoip at gmail.com  Fri Jun  2 14:38:22 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 2 Jun 2006 13:38:22 -0500
Subject: [cisco-voip] VoIP through PIX
In-Reply-To: <D713462ED535184D830F6F6301753E7E01E8F3DF@VISHNU.wesd.org>
References: <D713462ED535184D830F6F6301753E7E01E8F3DF@VISHNU.wesd.org>
Message-ID: <5d093f9a0606021138h5a93b4a2seb722f9de2dcd0fa@mail.gmail.com>

The problem you are running into is that the configs on the CCM are
using the internal IP... So, the phone is trying to connect to IP
addresses it has no connectivity to.

Solution?

NAT the destination IP to the public static NAT IP.

So, wherever the phone is create a NAT translation so that that
destination can be reached by the phone.



Jonathan

On 6/2/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>
>
>
> I have a Pix running 6.3.4 and call manager 4.1.3sr3a)on the inside of the
> Pix.  I placed a 7960 (skinny) on the outside and it looks like it reaches
> the TFTP server (nat'd address) and then gets the CM IP addresses.  But the
> CM IP's are the inside addresses and it never registers with the CM.
>
>
>
> I have both the TFTP and SCCP fixups and the ACL's are allowing IP any any
> to and from the CM's and phone.  Any idea what is wrong?
>
>
>
> Thanks
>
>
>
> Scott
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From CORTIZ at broward.org  Fri Jun  2 15:01:41 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 2 Jun 2006 15:01:41 -0400
Subject: [cisco-voip] Speed Dial
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC33D4@MREXVS01.bc.broward.cty>

Is there a way to create a speed dial that only dials a portion of the
number but does not try to complete the call ?

 

i.e.

 

XXX - XXX - YYYY 

 

Where X= speed dial and Y= user entered digits?

 

I have a request where "I could do this in the old system" was
quoted.....

 

 

Carlos

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From ealeatherman at gmail.com  Fri Jun  2 15:52:09 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 2 Jun 2006 15:52:09 -0400
Subject: [cisco-voip] Speed Dial
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC33D4@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC33D4@MREXVS01.bc.broward.cty>
Message-ID: <94a1afde0606021252l2e26d79cgfc494c6a95dc6090@mail.gmail.com>

I think there is an enterprise setting that you can set to have callmanager
wait for more digits on a speed dial instead of instantly trying to connect
the call. Havent messed with it myself though.

I wish we could put pauses in the speed dial also. thats another "but the
old system could do that" thing that keeps coming up.

On 6/2/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
>
>  Is there a way to create a speed dial that only dials a portion of the
> number but does not try to complete the call ?
>
>
>
> i.e.
>
>
>
> XXX ? XXX ? YYYY
>
>
>
> Where X= speed dial and Y= user entered digits?
>
>
>
> I have a request where "I could do this in the old system" was quoted?..
>
>
>
>
>
> Carlos
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From Scott.Voll at wesd.org  Fri Jun  2 19:24:12 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 2 Jun 2006 16:24:12 -0700
Subject: [cisco-voip] TFTP
Message-ID: <D713462ED535184D830F6F6301753E7E01E8F44B@VISHNU.wesd.org>

What file does the phone get the CM name(IP) from via the TFTP server?
CM 4.1

 

TIA

 

Scott

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From toddnh65 at gmail.com  Fri Jun  2 19:27:33 2006
From: toddnh65 at gmail.com (Todd Franklin)
Date: Fri, 2 Jun 2006 19:27:33 -0400
Subject: [cisco-voip] Two Phones, Same Line
Message-ID: <6101bda50606021627x25380bb1j96d187a5a5913548@mail.gmail.com>

We have a police dispatch center with 7960s.  (Three, side by side).
We hired a new person to train.  Under our old traditional PBX, persons 1
and 2 could listen in on the same phone call.  (If a call came in on Line 1,
whoever answered would press the Line1 button, and person 2 would also press
Line1).  They could both listen and talk, although person 2 was expected to
just listen for training purposes.

Any way to do this with 7960s?

Thanks.
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From ealeatherman at gmail.com  Sat Jun  3 00:06:05 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Sat, 3 Jun 2006 00:06:05 -0400
Subject: [cisco-voip] Two Phones, Same Line
In-Reply-To: <6101bda50606021627x25380bb1j96d187a5a5913548@mail.gmail.com>
References: <6101bda50606021627x25380bb1j96d187a5a5913548@mail.gmail.com>
Message-ID: <94a1afde0606022106r72ef1dd1xac401c8151e67750@mail.gmail.com>

Can you do this with the Barge feature? I think it would play a tone when
the person barged in though so it might not be what you're after.

On 6/2/06, Todd Franklin <toddnh65 at gmail.com> wrote:
>
> We have a police dispatch center with 7960s.  (Three, side by side).
> We hired a new person to train.  Under our old traditional PBX, persons 1
> and 2 could listen in on the same phone call.  (If a call came in on Line 1,
> whoever answered would press the Line1 button, and person 2 would also press
> Line1).  They could both listen and talk, although person 2 was expected to
> just listen for training purposes.
>
> Any way to do this with 7960s?
>
> Thanks.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From scall at metreos.com  Sat Jun  3 10:09:38 2006
From: scall at metreos.com (Seth Call)
Date: Sat, 3 Jun 2006 09:09:38 -0500
Subject: [cisco-voip] TFTP
Message-ID: <6EE053013F00084096AF0F02CFDB88B876974F@securemail.metreos.com>

[PHONE DEVICE NAME].cnf.xml - the files are created dynamically by CCM
per TFTP request I believe.

 

Seth Call

Metreos Corporation

(512) 687-2006

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, June 02, 2006 6:24 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] TFTP

 

What file does the phone get the CM name(IP) from via the TFTP server?
CM 4.1

 

TIA

 

Scott

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From jason.aarons at us.didata.com  Sat Jun  3 14:34:58 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Sat, 3 Jun 2006 14:34:58 -0400
Subject: [cisco-voip] IPCC Express Cisco Agent Desktop locks up
Message-ID: <C1FE15183DA37645BC0633BC604E44F0020DFDF3@USNAEXCH.na.didata.local>

Problem:

Cisco Agent Desktop screen becomes sluggish/non-responsive. The cisco
client will start flashing, then the full screen comes up on the monitor
and will be a blank screen and can not get out of it or get to other
applications (ie AS/400 Screens, Internet Explorer, Outlook).  Requires
a reboot to regain control of PC.

 

The Agent Desktop client is not resetting on a new call.  Users still
see the previous call information.  Does not correlate to what is truly
the condition of the call

Cisco client screen disappears.  Can be in Ready or not ready, push
button to more to different status, hourglass appears but does not
release timely (could be up to one minute long).

 

Restarted all sub-systems and CTI Manager in CCMService.  Checked all
event logs, etc. Opened a P1 provided Cisco TAC with agent
debugs/sniffer traces and no resolution after 4 hours of
troubleshooting.  Rebooted box after hours and next day no problems
reported. Any ideas, seen this before?

 

IPCC 4.0(4)Build80

Agent 6.1(3)

MCS-7825-H2

OS Ver 2000.4.2sr3

 

In production 2 months no similar issues

 




-----------------------------------------
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From cidar at acelerate.com  Sat Jun  3 16:41:19 2006
From: cidar at acelerate.com (Cidar)
Date: Sat, 3 Jun 2006 16:41:19 -0400
Subject: [cisco-voip] caller id on as5350
Message-ID: <200606032014.k53KEKcm092318@puck.nether.net>

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From zthoran at yahoo.com  Sun Jun  4 09:52:38 2006
From: zthoran at yahoo.com (Zevier Thoran)
Date: Sun, 4 Jun 2006 06:52:38 -0700 (PDT)
Subject: [cisco-voip] Inbound Caller-id problem
Message-ID: <20060604135238.519.qmail@web37101.mail.mud.yahoo.com>

Hi,

On my Callmanager gateway configuration I have the "Significant
Digits" set to 4 under Inbound Calls. This works well for our 
DID numbers, but there is one problem - I can't get correct 
caller-id info on incoming calls. So do I need to change 
"Significant Digits" to 'All' or '10' for the sake of 
caller-id?
Please advise? Thanks - Zevier.




		
---------------------------------
How low will we go? Check out Yahoo! Messenger?s low  PC-to-Phone call rates.
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From rratliff at cisco.com  Sun Jun  4 10:00:27 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Sun, 4 Jun 2006 10:00:27 -0400
Subject: [cisco-voip] Inbound Caller-id problem
In-Reply-To: <20060604135238.519.qmail@web37101.mail.mud.yahoo.com>
References: <20060604135238.519.qmail@web37101.mail.mud.yahoo.com>
Message-ID: <3C4F49BD-FD91-4705-BFC4-8438DF3B565F@cisco.com>

Sig digits has nothing to do with caller name, only the called party  
number.

What type of gateway (h323, mgcp) is this and what type of connection  
does it have to your telco?

-Ryan

On Jun 4, 2006, at 9:52 AM, Zevier Thoran wrote:

Hi,
On my Callmanager gateway configuration I have the "Significant
Digits" set to 4 under Inbound Calls. This works well for our
DID numbers, but there is one problem - I can't get correct
caller-id info on incoming calls. So do I need to change
"Significant Digits" to 'All' or '10' for the sake of
caller-id?
Please advise? Thanks - Zevier.



How low will we go? Check out Yahoo! Messenger?s low PC-to-Phone call  
rates.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From randy_g_wells at hotmail.com  Sun Jun  4 20:20:10 2006
From: randy_g_wells at hotmail.com (Randy Wells)
Date: Sun, 4 Jun 2006 17:20:10 -0700
Subject: [cisco-voip] IPMA Redundancy
Message-ID: <BAY120-W5D8C90CAFE74BA8E2B7E6A9940@phx.gbl>

I have a couple of questions about IPMA and redundancy.  Assuming I have a CallManager Publisher and Subscriber, with the Subscriber being the first choice for call processing and the Publisher backing it up, in a cluster and the IPMA service enabled on both...
 
- Under the IPMA Service Parameters there are options for CTI Manager IP addresses and IPMA Server IP addresses.  The CTI Manager options are server specific and the IPMA options are cluster wide.  Assuming I want everything to connect to the Subscriber and failover to the Publisher, including the IPMA service, what would the CTI options look like on each server?
 
- When installing the IPMA assistant client, which server should I use to install this from?  Subscriber since this is the primary server hosting the service?
 
- What IP address should I use for the IPMA phone service?  If the subscriber, what happens to the phone service when the Subscriber goes offline?  How does this failure impact the phones?
 
- What should be expected when the Subscriber goes offline?  I understand the need for the translation pattern to deal with IPMA failure, but what should still work when the Subscriber fails and the Publisher takes over the IPMA service.
 
Thanks,
Randy
_________________________________________________________________
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From JOlson at mlnusa.com  Mon Jun  5 09:24:58 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Mon, 5 Jun 2006 09:24:58 -0400
Subject: [cisco-voip] Time of day call routing
Message-ID: <FE2D092F8B1516488B99342551AACFD409CD491D@mlnusa-exch1.mlnusa.com>

I am not sure if this a call manager question or a unity question.  I
have call handlers programmed for day greetings and night greeting.  I
need to determine how calls are routed based on time of day to the CTI
route points.  


Jeff

********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

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From rratliff at cisco.com  Mon Jun  5 09:34:27 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 5 Jun 2006 09:34:27 -0400
Subject: [cisco-voip] Time of day call routing
In-Reply-To: <FE2D092F8B1516488B99342551AACFD409CD491D@mlnusa-exch1.mlnusa.com>
References: <FE2D092F8B1516488B99342551AACFD409CD491D@mlnusa-exch1.mlnusa.com>
Message-ID: <C6736724-2ED5-4623-A2F5-84A54C4918BF@cisco.com>

If you have two different call handlers that require the calls to hit  
different CTI RP to get routed appropriately then you are going to  
use the CM time of day routing feature.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_administration_guide_chapter09186a00803edbaa.html

The other option is having one call handler in Unity with a schedule  
set up for open and closed greetings.  It's more complicated to get  
working Unity-wise but the CM config is very simple.

-Ryan

On Jun 5, 2006, at 9:24 AM, Olson, Jeffrey wrote:

I am not sure if this a call manager question or a unity question.  I  
have call handlers programmed for day greetings and night greeting.   
I need to determine how calls are routed based on time of day to the  
CTI route points.


Jeff

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders  
Network USA, Inc. which is confidential and legally privileged. The  
information is intended only for the use of the individual or entity  
named on this E-mail. If you are not the intended recipient, you are  
hereby notified that disclosing, copying, distributing, or taking any  
action in reliance on the contents of this information is strictly  
prohibited, and that the E-mail should be returned to this company  
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contact the sender immediately.


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


From cidar at acelerate.com  Mon Jun  5 10:45:41 2006
From: cidar at acelerate.com (Cidar Ramirez)
Date: Mon, 5 Jun 2006 08:45:41 -0600
Subject: [cisco-voip] caller id on as5350
In-Reply-To: <BAY122-F36249CFBC190B20D2BBCB7B0970@phx.gbl>
Message-ID: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAATb16yFsuxEqckYCsMTcHdMKAAAAQAAAAZFnDr7elQk+gIZGX/+vxAwEAAAAA@acelerate.com>

Anyone does do this before?
I'm not sure if the translations will work

Thanks

cidar

-----Mensaje original-----
De: Rajesh Dogra [mailto:razidogra at hotmail.com] 
Enviado el: Sunday, June 04, 2006 3:31 AM
Para: cidar at acelerate.com
Asunto: RE: [cisco-voip] caller id on as5350

Cidar,


You are right by Appling  Tranlations it would solve ur problem.

Regards

Razi


>From: "Cidar" <cidar at acelerate.com>
>Reply-To: Cidar <cidar at acelerate.com>
>To: cisco-voip at puck.nether.net
>Subject: [cisco-voip] caller id on as5350
>Date: Sat, 3 Jun 2006 16:41:19 -0400
>
>hi
>
>i have a as5350 whit 4 dial-peer working (3 for outbound calls and
>1 for inbound calls), the as5350 is connected to a switch using 2E1 whit r2
>signaling as the summary of the config below
>
>dial-peer voice 1000
>voip
>?destination-pattern 1..........
>?voice-class codec
>100
>?session target
>ipv4:x.x.x.x
>?tech-prefix?12
>?dtmf-relay
>h245-alphanumeric
>?no vad
>!
>dial-peer voice 2000
>voip
>?destination-pattern 54..........
>?voice-class codec
>100
>?session target
>ipv4:x.x.x.x
>?tech-prefix?25
>?dtmf-relay
>h245-alphanumeric
>?no vad
>!
>dial-peer voice 2000
>voip
>?destination-pattern 44..........
>?voice-class codec
>100
>?session target
>ipv4:x.x.x.x
>?tech-prefix?45
>?dtmf-relay
>h245-alphanumeric
>?no vad
>!
>dial-peer voice 4000
>pots
>?destination-pattern 010[2,6].....
>?port
>3/2:1
>?forward-digits 5
>!
>
>
>i will add a 5th dial-peer,
>this is for inbound traffic, but now i need to differenciate the traffic
>beetween both inboud prividers, i'm not using syslog or radius logs, i'm
>using the CDR of the 2E1 attached switch, so i ask if it is posible to put 
>a
>fix ani for each dial-peer on the as5350 ?
>do i need to use tranlations
>?
>I'm thinking that the solution could be modify the caller id or ANI
>before the call is send to the E1 for termination.
>
>this is the new
>dial-peer
>
>dial-peer voice 5000 pots
>?destination-pattern
>011[2,6].....
>?port 3/2:1
>?forward-digits 5
>!
>
>any
>advice ?
>thanks
>
>seedorf
>


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

_________________________________________________________________
Spice up your IM conversations. New, colorful and animated emoticons. Get 
chatting! http://server1.msn.co.in/SP05/emoticons/






From jason.aarons at us.didata.com  Mon Jun  5 10:54:09 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Mon, 5 Jun 2006 10:54:09 -0400
Subject: [cisco-voip] SBC Supertrunk T1 CAS (all channels show seized)
Message-ID: <C1FE15183DA37645BC0633BC604E44F002142CBF@USNAEXCH.na.didata.local>

Anything special for configuring a SBC T1 CAS Supertrunk?  All channels
show seized

 

 

BLW-2801#show voice port summary

                                     IN       OUT

PORT      CH   SIG-TYPE   ADMIN OPER STATUS   STATUS   EC

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

0/0/0     --  fxo-ls      up    dorm idle     on-hook  y

0/0/1     --  fxo-ls      up    dorm idle     on-hook  y

0/0/2     --  fxo-ls      up    dorm idle     on-hook  y

0/0/3     --  fxo-ls      up    dorm idle     on-hook  y

0/2/1:0   01  e&m-wnk     up    up   seized   idle     y

0/2/1:0   02  e&m-wnk     up    up   seized   idle     y

0/2/1:0   03  e&m-wnk     up    up   seized   idle     y

0/2/1:0   04  e&m-wnk     up    up   seized   idle     y

0/2/1:0   05  e&m-wnk     up    up   seized   idle     y

0/2/1:0   06  e&m-wnk     up    up   seized   idle     y

 

 

controller T1 0/2/1

 framing esf

 linecode b8zs

 ds0-group 0 timeslots 1-6 type e&m-wink-start

 description SUPERTRUNK

 

 

dial-peer voice 7075 voip

 destination-pattern 70..

 voice-class codec 1

 voice-class h323 1

 session target ipv4:10.168.32.12

 incoming called-number .

 dtmf-relay h245-alphanumeric

 no vad

!

dial-peer voice 210 pots

 destination-pattern 9.T

 incoming called-number .

 port 0/2/1:0




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From ErasmuE4 at telkom.co.za  Mon Jun  5 11:21:50 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Mon, 5 Jun 2006 17:21:50 +0200
Subject: [cisco-voip] (no subject)
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D539@CNTRRA20-XCS01.telkom.co.za>



Has anyone seen this message in the MS event viewer before. I was
informed by a customer that their IPt solution running on:

CCM 4-1-3 sr3a -- OS 2000-2-7
MCS 7835 platform
roughly 200 users, IPMA for 10 managers and extension mobility for
everyone - nothing fancy


went down because the server restarted. Unfortunately I was at another
branch and could not see exactly what happened. When I checked the
eventviewer later it had the following message.

The previous system shutdown at date/time was unexpected
<http://www.pcreview.co.uk/forums/thread-1440123.php>

A week earlier while I worked at the main site the server went down
once because the power failed - people pulling wires powered the
server off accidentally . In the case now - today - where the message
as shown above is present in the event viewer can it be power again or
is it a real problem with CCM hanging - rebooting etc.


any advice - much appreciated


. <http://www.pcreview.co.uk/forums/thread-1440123.php>


~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From rratliff at cisco.com  Mon Jun  5 11:42:24 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 5 Jun 2006 11:42:24 -0400
Subject: [cisco-voip] (no subject)
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F84D539@CNTRRA20-XCS01.telkom.co.za>
References: <472E195E692EE74BAE4E7B2880A78C1F84D539@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <482A42FE-81A7-405E-9398-6E9E07281209@cisco.com>

This could be anything from a power outage to a hardware problem  
causing the server to reboot itself.   Basically Windows did not shut  
down gracefully for some reason.

If this is an HP server you definitely need to get to the latest  
major OS release to get a newer BIOS.   There are known issues with  
the microcode in the bios that can cause random reboots.

-Ryan

On Jun 5, 2006, at 11:21 AM, Erik Erasmus ((E)) wrote:


Has anyone seen this message in the MS event viewer before. I was  
informed by a customer that their IPt solution running on:

CCM 4-1-3 sr3a -- OS 2000-2-7
MCS 7835 platform
roughly 200 users, IPMA for 10 managers and extension mobility for  
everyone - nothing fancy


went down because the server restarted. Unfortunately I was at  
another branch and could not see exactly what happened. When I  
checked the eventviewer later it had the following message.

The previous system shutdown at date/time was unexpected

A week earlier while I worked at the main site the server went down  
once because the power failed - people pulling wires powered the  
server off accidentally . In the case now - today - where the message  
as shown above is present in the event viewer can it be power again  
or is it a real problem with CCM hanging - rebooting etc.


any advice - much appreciated


.
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From lelio at uoguelph.ca  Mon Jun  5 12:01:50 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 5 Jun 2006 12:01:50 -0400
Subject: [cisco-voip] Time of day call routing
References: <FE2D092F8B1516488B99342551AACFD409CD491D@mlnusa-exch1.mlnusa.com>
Message-ID: <001601c688b9$5b604180$c801a8c0@dellectable>

Time of day call routingIf you're looking to use day and night greeting, I would use the schedules in Unity. We use them with little (or no) problems. Using the schedules in Unity allow you to affect everyone calling, not just those people who's search space you modify with timebased partitions.
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network 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: Olson, Jeffrey 
  To: cisco-voip at puck.nether.net 
  Sent: Monday, June 05, 2006 9:24 AM
  Subject: [cisco-voip] Time of day call routing


  I am not sure if this a call manager question or a unity question.  I have call handlers programmed for day greetings and night greeting.  I need to determine how calls are routed based on time of day to the CTI route points.  



  Jeff 

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

  CONFIDENTIALITY NOTICE:

  This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.





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From ErasmuE4 at telkom.co.za  Mon Jun  5 12:45:35 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Mon, 5 Jun 2006 18:45:35 +0200
Subject: [cisco-voip] (no subject) previous system shutdown at xxxxxx
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F84D540@CNTRRA20-XCS01.telkom.co.za>


Thanks Ryan

Appologies for - fotgetting the subject

Thanks for the advise - Will a BIOS upgrade come with the new OS --
ver 2000 ..... etc or is this something I have to do seperately


erik

________________________________

From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Mon 2006-06-05 17:42
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] (no subject)



This could be anything from a power outage to a hardware problem
causing the server to reboot itself.   Basically Windows did not shut
down gracefully for some reason.

If this is an HP server you definitely need to get to the latest
major OS release to get a newer BIOS.   There are known issues with
the microcode in the bios that can cause random reboots.

-Ryan

On Jun 5, 2006, at 11:21 AM, Erik Erasmus ((E)) wrote:


Has anyone seen this message in the MS event viewer before. I was
informed by a customer that their IPt solution running on:

CCM 4-1-3 sr3a -- OS 2000-2-7
MCS 7835 platform
roughly 200 users, IPMA for 10 managers and extension mobility for
everyone - nothing fancy


went down because the server restarted. Unfortunately I was at
another branch and could not see exactly what happened. When I
checked the eventviewer later it had the following message.

The previous system shutdown at date/time was unexpected

A week earlier while I worked at the main site the server went down
once because the power failed - people pulling wires powered the
server off accidentally . In the case now - today - where the message
as shown above is present in the event viewer can it be power again
or is it a real problem with CCM hanging - rebooting etc.


any advice - much appreciated


.
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip





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From rratliff at cisco.com  Mon Jun  5 12:48:26 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 5 Jun 2006 12:48:26 -0400
Subject: [cisco-voip] (no subject) previous system shutdown at xxxxxx
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F84D540@CNTRRA20-XCS01.telkom.co.za>
References: <472E195E692EE74BAE4E7B2880A78C1F84D540@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <5F767649-6BF8-433F-9898-366E33A48F82@cisco.com>

The major OS patch will include a bios upgrade.  The OS SR patches  
typically only contain patches from MS.

-Ryan

On Jun 5, 2006, at 12:45 PM, Erik Erasmus ((E)) wrote:

Thanks Ryan

Appologies for - fotgetting the subject

Thanks for the advise - Will a BIOS upgrade come with the new OS --  
ver 2000 ..... etc or is this something I have to do seperately


erik

From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Mon 2006-06-05 17:42
To: Erik Erasmus (E)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] (no subject)

This could be anything from a power outage to a hardware problem
causing the server to reboot itself.   Basically Windows did not shut
down gracefully for some reason.

If this is an HP server you definitely need to get to the latest
major OS release to get a newer BIOS.   There are known issues with
the microcode in the bios that can cause random reboots.

-Ryan

On Jun 5, 2006, at 11:21 AM, Erik Erasmus ((E)) wrote:


Has anyone seen this message in the MS event viewer before. I was
informed by a customer that their IPt solution running on:

CCM 4-1-3 sr3a -- OS 2000-2-7
MCS 7835 platform
roughly 200 users, IPMA for 10 managers and extension mobility for
everyone - nothing fancy


went down because the server restarted. Unfortunately I was at
another branch and could not see exactly what happened. When I
checked the eventviewer later it had the following message.

The previous system shutdown at date/time was unexpected

A week earlier while I worked at the main site the server went down
once because the power failed - people pulling wires powered the
server off accidentally . In the case now - today - where the message
as shown above is present in the event viewer can it be power again
or is it a real problem with CCM hanging - rebooting etc.


any advice - much appreciated


.
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~


From pyago at adomo.com  Mon Jun  5 14:39:19 2006
From: pyago at adomo.com (Paul Yago)
Date: Mon, 5 Jun 2006 11:39:19 -0700
Subject: [cisco-voip]  MWI Lamp Policy
Message-ID: <170BC64B51964949A28FE8A48EF1D4EB014A800F@corpexchsrv01.adomo.com>

Hello,

Is anyone familiar with the MWI lamp policy for Callmanager 4.1.3?
For 7940 and 7960, what is the "Prompt", and how do I configure this?

-Paul


From lelio at uoguelph.ca  Mon Jun  5 14:41:47 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 5 Jun 2006 14:41:47 -0400
Subject: [cisco-voip] MWI Lamp Policy
References: <170BC64B51964949A28FE8A48EF1D4EB014A800F@corpexchsrv01.adomo.com>
Message-ID: <00a101c688cf$b3cf6880$c801a8c0@dellectable>

I believe prompt means the 'You have voicemail' text you see on the phone. I 
don't think there is a way to change this from CallManager admin interface, 
but I'll bet someone knows the locale files containing the strings that you 
can modify manually. Problem with that is you have to remember to modify 
them again after any service or maintenance release or version upgrades, 
since they are likely overwritten.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network 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: "Paul Yago" <pyago at adomo.com>
To: <cisco-voip at puck.nether.net>
Cc: "Jacob Jensen" <jjensen at adomo.com>
Sent: Monday, June 05, 2006 2:39 PM
Subject: [cisco-voip] MWI Lamp Policy


> Hello,
>
> Is anyone familiar with the MWI lamp policy for Callmanager 4.1.3?
> For 7940 and 7960, what is the "Prompt", and how do I configure this?
>
> -Paul
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From it at cimgroup.com  Mon Jun  5 15:01:13 2006
From: it at cimgroup.com (IT)
Date: Mon, 5 Jun 2006 12:01:13 -0700
Subject: [cisco-voip] VT Advantage on an MCU
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0328EB3E@exchange.cimgroup.com>

I am looking for a video conference solution that will allow me to get
VTA users into my video conferences. A few months back, I tried out the
IPVC 3511 and a few Sony PCS-1's loaded with SCCP.  However, that test
was a failure, since the IPVC only showed one site at a time (based on
who was talking). This was entirely unacceptable, and I was told I would
need the 3540 to get the "brady bunch" effect, so I returned the 3511.
Now I am leaning towards the H.323 only video world. I am reloading all
the PCS-1's to H.323 firmware, and purchasing a sony h323 mcu software
upgrade for one of them. Using this configuration, sony has told me that
I will be able to have a 5-camera conference without a problem,
including the grid layout on-screen. That being said, I know that there
is a huge downside to this setup, none of my VTA users will be able to
join, correct?  Does anyone know of a way to get VT advantages to call
in to a h323 video conference hosted by a h323 mcu?

 

Any advice is truly appreciated.

 

Thanks,

Avidan

 

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From toddnh65 at gmail.com  Mon Jun  5 15:17:07 2006
From: toddnh65 at gmail.com (Todd Franklin)
Date: Mon, 5 Jun 2006 15:17:07 -0400
Subject: [cisco-voip] No More MTP Resources Available
Message-ID: <6101bda50606051217u4ccdc539q41742ec5a51863ad@mail.gmail.com>

Hey all.

We recently converted one of our gateways to H323 because we needed a lot of
POTS lines and Caller ID, and this appeared to be the only way to do it.

Ever since, my CCM in the Windows Event Viewer, shows lots of occurences of
this message: "No More MTP Resources Available."  Occasionally, after this
message, we will drop a call, or a phone (oddly enough, in another location
not attaced to the H323 gateway) will unregister without warning.

Is the MTP resource issue related to the H323 thing?  A friend told me I can
add more MTPs by simply installing the software on another NT/Win2x server.

Anyone?
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From scott.coner at us.didata.com  Mon Jun  5 15:30:17 2006
From: scott.coner at us.didata.com (Scott Coner (US))
Date: Mon, 5 Jun 2006 15:30:17 -0400
Subject: [cisco-voip] No More MTP Resources Available
Message-ID: <C1FE15183DA37645BC0633BC604E44F00214330C@USNAEXCH.na.didata.local>

Are you specifying G.711 in your H.323 dial peers?  Either with the
codec command or a 'voice-class'.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Monday, June 05, 2006 3:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No More MTP Resources Available

 

Hey all.

We recently converted one of our gateways to H323 because we needed a
lot of POTS lines and Caller ID, and this appeared to be the only way to
do it.

Ever since, my CCM in the Windows Event Viewer, shows lots of occurences
of this message: "No More MTP Resources Available."  Occasionally, after
this message, we will drop a call, or a phone (oddly enough, in another
location not attaced to the H323 gateway) will unregister without
warning. 

Is the MTP resource issue related to the H323 thing?  A friend told me I
can add more MTPs by simply installing the software on another NT/Win2x
server.

Anyone?




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This e-mail communication and any attachments may contain
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designated addressee(s) named above only.  If you are not the
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this communication in error and that any use or reproduction of
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From Scott.Voll at wesd.org  Mon Jun  5 16:06:28 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 5 Jun 2006 13:06:28 -0700
Subject: [cisco-voip] VT Advantage on an MCU
Message-ID: <D713462ED535184D830F6F6301753E7E01E8F4CE@VISHNU.wesd.org>

Could you not do the "brady bunch" idea on the 3511 at all, or just
because you wanted to use SCCP?

 

We have a 3510 that was configured to do the continuous presents with
h323.

 

I don't know about sccp to h323 but can't you use a IPIPGW or GK to do
this? 

 

Scott

 

PS>  make sure that your MCU supports all h323 vendors your going to
use.  I know the Polycom has features only polycom can use.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Monday, June 05, 2006 12:01 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VT Advantage on an MCU

 

I am looking for a video conference solution that will allow me to get
VTA users into my video conferences. A few months back, I tried out the
IPVC 3511 and a few Sony PCS-1's loaded with SCCP.  However, that test
was a failure, since the IPVC only showed one site at a time (based on
who was talking). This was entirely unacceptable, and I was told I would
need the 3540 to get the "brady bunch" effect, so I returned the 3511.
Now I am leaning towards the H.323 only video world. I am reloading all
the PCS-1's to H.323 firmware, and purchasing a sony h323 mcu software
upgrade for one of them. Using this configuration, sony has told me that
I will be able to have a 5-camera conference without a problem,
including the grid layout on-screen. That being said, I know that there
is a huge downside to this setup, none of my VTA users will be able to
join, correct?  Does anyone know of a way to get VT advantages to call
in to a h323 video conference hosted by a h323 mcu?

 

Any advice is truly appreciated.

 

Thanks,

Avidan

 

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From rratliff at cisco.com  Mon Jun  5 16:33:11 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 5 Jun 2006 16:33:11 -0400
Subject: [cisco-voip] VT Advantage on an MCU
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B0328EB3E@exchange.cimgroup.com>
References: <B487D4C1FE6D04439061018C6EBC109B0328EB3E@exchange.cimgroup.com>
Message-ID: <867C982A-8569-4B01-8B45-05D90C774F3F@cisco.com>

Continuous presence with SCCP requires the EMP module regardless of  
what IPVC gear you have.  For the 3511 you'd need a 3511E.  I'm not  
on the sales side so I have no idea what the cost difference between  
the two would be.

That said I have no idea why a VTA couldn't call into an H.323  
conference.  As long as CM can route the call (via gatekeeper or  
directly to the H.323 gateway) there should be no problem.

-Ryan

On Jun 5, 2006, at 3:01 PM, IT wrote:

I am looking for a video conference solution that will allow me to  
get VTA users into my video conferences. A few months back, I tried  
out the IPVC 3511 and a few Sony PCS-1?s loaded with SCCP.  However,  
that test was a failure, since the IPVC only showed one site at a  
time (based on who was talking). This was entirely unacceptable, and  
I was told I would need the 3540 to get the ?brady bunch? effect, so  
I returned the 3511.  Now I am leaning towards the H.323 only video  
world. I am reloading all the PCS-1?s to H.323 firmware, and  
purchasing a sony h323 mcu software upgrade for one of them. Using  
this configuration, sony has told me that I will be able to have a 5- 
camera conference without a problem, including the grid layout on- 
screen. That being said, I know that there is a huge downside to this  
setup, none of my VTA users will be able to join, correct?  Does  
anyone know of a way to get VT advantages to call in to a h323 video  
conference hosted by a h323 mcu?



Any advice is truly appreciated.



Thanks,

Avidan



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



From rratliff at cisco.com  Mon Jun  5 16:42:16 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 5 Jun 2006 16:42:16 -0400
Subject: [cisco-voip] No More MTP Resources Available
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F00214330C@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F00214330C@USNAEXCH.na.didata.local>
Message-ID: <7E3B9B73-FD2E-45D0-B215-2CFBECF359DB@cisco.com>

Additionally check to see if you have "MTP Required" set for your H. 
323 gateway.  This setting will cause CM to allocate an MTP resource  
for each call involving the gateway.  The calls won't fail if no MTPs  
are available but supplementary services (hold, transfer, etc) will  
not function for that call.

Also note that if no codec is specified on an H.323 voip dial-peer  
the gateway uses g.729.  In this case calls to phones would work but  
any time the gateway is conferenced or hits a g.711 only device a  
transcoder will be required.  In this case you can see reorder tone  
played to one or more parties.

-Ryan

On Jun 5, 2006, at 3:30 PM, Scott Coner ((US)) wrote:

Are you specifying G.711 in your H.323 dial peers?  Either with the  
codec command or a ?voice-class?.



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Monday, June 05, 2006 3:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] No More MTP Resources Available



Hey all.

We recently converted one of our gateways to H323 because we needed a  
lot of POTS lines and Caller ID, and this appeared to be the only way  
to do it.

Ever since, my CCM in the Windows Event Viewer, shows lots of  
occurences of this message: "No More MTP Resources Available."   
Occasionally, after this message, we will drop a call, or a phone  
(oddly enough, in another location not attaced to the H323 gateway)  
will unregister without warning.

Is the MTP resource issue related to the H323 thing?  A friend told  
me I can add more MTPs by simply installing the software on another  
NT/Win2x server.

Anyone?





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From pwalenta at wi.rr.com  Mon Jun  5 17:18:31 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Mon, 5 Jun 2006 16:18:31 -0500
Subject: [cisco-voip] VT Advantage on an MCU
In-Reply-To: <867C982A-8569-4B01-8B45-05D90C774F3F@cisco.com>
Message-ID: <00f901c688e5$99510570$06000100@walentawk1>

I actually just help set up a lab with this sort of thing.

VT advantage (on CME and CCM), GK and a Polycom MCU.

All of them could bridge to it just fine via H323. 

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
> Sent: Monday, June 05, 2006 3:33 PM
> To: IT
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] VT Advantage on an MCU
> 
> Continuous presence with SCCP requires the EMP module 
> regardless of what IPVC gear you have.  For the 3511 you'd 
> need a 3511E.  I'm not on the sales side so I have no idea 
> what the cost difference between the two would be.
> 
> That said I have no idea why a VTA couldn't call into an 
> H.323 conference.  As long as CM can route the call (via 
> gatekeeper or directly to the H.323 gateway) there should be 
> no problem.
> 
> -Ryan
> 
> On Jun 5, 2006, at 3:01 PM, IT wrote:
> 
> I am looking for a video conference solution that will allow 
> me to get VTA users into my video conferences. A few months 
> back, I tried out the IPVC 3511 and a few Sony PCS-1's loaded 
> with SCCP.  However, that test was a failure, since the IPVC 
> only showed one site at a time (based on who was talking). 
> This was entirely unacceptable, and I was told I would need 
> the 3540 to get the "brady bunch" effect, so I returned the 
> 3511.  Now I am leaning towards the H.323 only video world. I 
> am reloading all the PCS-1's to H.323 firmware, and 
> purchasing a sony h323 mcu software upgrade for one of them. 
> Using this configuration, sony has told me that I will be 
> able to have a 5- camera conference without a problem, 
> including the grid layout on- screen. That being said, I know 
> that there is a huge downside to this setup, none of my VTA 
> users will be able to join, correct?  Does anyone know of a 
> way to get VT advantages to call in to a h323 video 
> conference hosted by a h323 mcu?
> 
> 
> 
> Any advice is truly appreciated.
> 
> 
> 
> Thanks,
> 
> Avidan
> 
> 
> 
> _______________________________________________
> 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 Ryan.OConnell at nexinnovations.com  Mon Jun  5 18:56:43 2006
From: Ryan.OConnell at nexinnovations.com (OConnell, Ryan)
Date: Mon, 5 Jun 2006 18:56:43 -0400
Subject: [cisco-voip] Jtapi integration from CM to CUE version 2.3.1
Message-ID: <D9C2C0CEF6CC4248BF8F275D03E57F0C0D0CF357@MISEXCVS01.corp.nexinnovations.com>

Hello all,

 

I just setup CUE 2.3.1 into CM 4.1(3) as per this document

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_c
onfiguration_example09186a0080289ef0.shtml

 

I found that the CTI Ports and the CTI RP are registered in CM. But when
I make a call to the CUE Pilot, the call tries to go through to the
Pilot number for 2 seconds, then I get a slow busy. 

 

The call seems to be getting to CUE because I am seeing information in
the following debug come through

 

tcCUE# Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
ROUTE_FAILED:Route failed : All Call
ids=JTAPICallContact[id=64,type=Cisco JTAPI
Call,implId=6922/2,active=true,state=CALL_RECEIVED,inbound=true,handled=
false,locale=en_US,aborting=false,app=App[name=voicemail,type=Cisco
Script Application,id=2,desc=voice

Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
EXCEPTION:com.cisco.jtapi.InvalidPartyExceptionImpl: Request failed
because of an invalid destination. 

Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
EXCEPTION:      at
com.cisco.jtapi.ConnectionImpl.redirect(ConnectionImpl)

Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
EXCEPTION:      at
com.cisco.jtapi.ConnectionImpl.redirect(ConnectionImpl)

Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
EXCEPTION:      at
com.cisco.jtapi.ConnectionImpl.redirect(ConnectionImpl)

Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
EXCEPTION:      at
com.cisco.jtapi.ConnectionImpl.redirect(ConnectionImpl)

Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
EXCEPTION:      at
com.cisco.wf.subsystems.jtapi.TAPIPortGroup$30.run(TAPIPortGroup.java:17
532)

Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
EXCEPTION:      at
com.cisco.executor.impl.ParallelExecutorImpl$WorkOrder.run(ParallelExecu
torImpl.java:283)

Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
EXCEPTION:      at
com.cisco.executor.impl.WorkerPool$Worker.run(WorkerPool.java:398)

Jun  5 16:54:47 localhost java:   ERROR ccn SubsystemJtapi DBUG
EXCEPTION:      at java.lang.Thread.run(Thread.java:570)

 

The only thing I haven't tried yet is reboot the CTI services on CM. I
will try this tonight

 

Thoughts???

 

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

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

 

 

 

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From chris at chrisserafin.com  Mon Jun  5 19:03:33 2006
From: chris at chrisserafin.com (ChrisSerafin)
Date: Mon, 05 Jun 2006 18:03:33 -0500
Subject: [cisco-voip] Callmanager configured with multiple remote CUEs.
In-Reply-To: <17A170DC8DD5CC44BA56B24B366EB60B01DACFF1@E3K-VS1.ccsi.local>
References: <17A170DC8DD5CC44BA56B24B366EB60B01DACFF1@E3K-VS1.ccsi.local>
Message-ID: <4484B845.2070506@chrisserafin.com>

The majority of Cisco IPT experience I have is with a cluster of 
centralized CM's and everywhere using SRST.  Using an MPLS MCI backbone.

Chris Serafin
IT Security / Cisco VoIP Engineer
chris at chrisserafin.com





Scott O'Donnell wrote:
> Does any one have experience with a centralized CallManager install 
> using SRST with CUE at the remote branch locations?
>  
> Seems to me that it doesn't present a major problems but I haven't 
> done it yet.
>  
> Any thoughts are appreciated.
>  
> Scott
>  
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From chris at chrisserafin.com  Mon Jun  5 19:29:29 2006
From: chris at chrisserafin.com (ChrisSerafin)
Date: Mon, 05 Jun 2006 18:29:29 -0500
Subject: [cisco-voip] 7920
In-Reply-To: <OF9E75C594.89F8FA13-ONC125717F.004DC85C-C125717F.004DEE61@shearman.com>
References: <OF9E75C594.89F8FA13-ONC125717F.004DC85C-C125717F.004DEE61@shearman.com>
Message-ID: <4484BE59.70901@chrisserafin.com>

I know my regular PDA phone can change via USB to my laptop; but the 
charge is EXTRA SLOW.  This may be the case with the 7920's?  If you are 
going from charging the unit from the wall to the USB, the slow charge 
may not seem like it is working.....just a thought

Chris Serafin
IT Security / Cisco VoIP Engineer
chris at chrisserafin.com


Loris Vicinanza wrote:
> Hi, 7920's wireless phones can be charged via the desktop charger using a
> standard USB cable, but it doesn't seem to work. Any ideas?
>
> Thanks,
>
> Loris
>
>
>
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From it at cimgroup.com  Tue Jun  6 00:07:56 2006
From: it at cimgroup.com (IT)
Date: Mon, 5 Jun 2006 21:07:56 -0700
Subject: [cisco-voip] VT Advantage on an MCU
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0328EF79@exchange.cimgroup.com>

Yes, I was told that a 3511E would be necessary to do continuous
presence, and it was twice the price if I remember correctly.

If I want to do this all without a 3511E, my understanding is that I
will need a h323 gatekeeper. Can someone please drop me a few links
about setting up a gatekeeper? I inherited a vocaltec gk, but don't know
about integrating with CCM. Anyone with advice?

Thanks,
Avidan

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Monday, June 05, 2006 1:33 PM
To: IT
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VT Advantage on an MCU

Continuous presence with SCCP requires the EMP module regardless of  
what IPVC gear you have.  For the 3511 you'd need a 3511E.  I'm not  
on the sales side so I have no idea what the cost difference between  
the two would be.

That said I have no idea why a VTA couldn't call into an H.323  
conference.  As long as CM can route the call (via gatekeeper or  
directly to the H.323 gateway) there should be no problem.

-Ryan

On Jun 5, 2006, at 3:01 PM, IT wrote:

I am looking for a video conference solution that will allow me to  
get VTA users into my video conferences. A few months back, I tried  
out the IPVC 3511 and a few Sony PCS-1's loaded with SCCP.  However,  
that test was a failure, since the IPVC only showed one site at a  
time (based on who was talking). This was entirely unacceptable, and  
I was told I would need the 3540 to get the "brady bunch" effect, so  
I returned the 3511.  Now I am leaning towards the H.323 only video  
world. I am reloading all the PCS-1's to H.323 firmware, and  
purchasing a sony h323 mcu software upgrade for one of them. Using  
this configuration, sony has told me that I will be able to have a 5- 
camera conference without a problem, including the grid layout on- 
screen. That being said, I know that there is a huge downside to this  
setup, none of my VTA users will be able to join, correct?  Does  
anyone know of a way to get VT advantages to call in to a h323 video  
conference hosted by a h323 mcu?



Any advice is truly appreciated.



Thanks,

Avidan



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



From erickbe at yahoo.com  Tue Jun  6 01:02:06 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Mon, 5 Jun 2006 22:02:06 -0700 (PDT)
Subject: [cisco-voip] (no subject) OS / BIOS versions
In-Reply-To: <482A42FE-81A7-405E-9398-6E9E07281209@cisco.com>
Message-ID: <20060606050206.89877.qmail@web37204.mail.mud.yahoo.com>

Ryan,

Do the HP BIOS that come with 2000.2-7 release contain the random-reboot issues? I have seen a handful as of late but most seem to be power-related with some causes unknown. Working on the 2000.4-2 updates places but just curious about the comment. :)  I have one client who states that once they loaded 2000.4-2 their system is performing a lot better and smoother/quicker but only thing that changed was going from 2-7sr8 to 4-2 with latest SR. 

----- Original Message ----
From: Ryan Ratliff <rratliff at cisco.com>
To: Erik Erasmus <ErasmuE4 at telkom.co.za>
Cc: cisco-voip at puck.nether.net
Sent: Monday, June 5, 2006 10:42:24 AM
Subject: Re: [cisco-voip] (no subject)

This could be anything from a power outage to a hardware problem  
causing the server to reboot itself.   Basically Windows did not shut  
down gracefully for some reason.

If this is an HP server you definitely need to get to the latest  
major OS release to get a newer BIOS.   There are known issues with  
the microcode in the bios that can cause random reboots.

-Ryan

On Jun 5, 2006, at 11:21 AM, Erik Erasmus ((E)) wrote:


Has anyone seen this message in the MS event viewer before. I was  
informed by a customer that their IPt solution running on:

CCM 4-1-3 sr3a -- OS 2000-2-7
MCS 7835 platform
roughly 200 users, IPMA for 10 managers and extension mobility for  
everyone - nothing fancy


went down because the server restarted. Unfortunately I was at  
another branch and could not see exactly what happened. When I  
checked the eventviewer later it had the following message.

The previous system shutdown at date/time was unexpected

A week earlier while I worked at the main site the server went down  
once because the power failed - people pulling wires powered the  
server off accidentally . In the case now - today - where the message  
as shown above is present in the event viewer can it be power again  
or is it a real problem with CCM hanging - rebooting etc.


any advice - much appreciated


.
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From jonvoip at gmail.com  Tue Jun  6 02:18:58 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 6 Jun 2006 01:18:58 -0500
Subject: [cisco-voip] Callmanager configured with multiple remote CUEs.
In-Reply-To: <17A170DC8DD5CC44BA56B24B366EB60B01DACFF1@E3K-VS1.ccsi.local>
References: <17A170DC8DD5CC44BA56B24B366EB60B01DACFF1@E3K-VS1.ccsi.local>
Message-ID: <5d093f9a0606052318i697142c7k85d824f98f84be1a@mail.gmail.com>

Not a big problem... You just need to make sure you have the CCM
license for the CUE...

Then you create JTAPI connections to the CUE


Jonathan

On 6/1/06, Scott O'Donnell <sodonnell at ccsinet.com> wrote:
>
>
> Does any one have experience with a centralized CallManager install using
> SRST with CUE at the remote branch locations?
>
> Seems to me that it doesn't present a major problems but I haven't done it
> yet.
>
> Any thoughts are appreciated.
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From balbler at nts.at  Tue Jun  6 05:06:33 2006
From: balbler at nts.at (Bernhard Albler)
Date: Tue, 6 Jun 2006 11:06:33 +0200
Subject: [cisco-voip] SIP Tandem Call Routing in IOS / CME
Message-ID: <5A8C69748176434F92F1EECD7D21F72C64E5A2@srvgraz03.nts.at>

Hi everyone,

Does anyone know if and when SIP Tandem call routing will be supported?
(I can't get it working).
I have following scenario:

Cme connected via a SIP trunk to a service provider. Unfortunately the
service provider does not support REFER for transfers, so I am stuck.
Unless IOS could just make another outbound call and bridge the two
calls (original inbound and the new outbound call) internally.

Has someone gotten this to work? I can get forwarding to work, but not
transfer

Many thanks and best regards
bernhard


From lex at sandy.ru  Tue Jun  6 05:53:47 2006
From: lex at sandy.ru (Alex A. Pavlenko)
Date: Tue, 6 Jun 2006 13:53:47 +0400
Subject: [cisco-voip] Line group membership
References: <05b601c68627$04eec040$19477dc1@corporate.sandy.ru>
	<94a1afde0606020529n78d5a14bp1722cdae91d19754@mail.gmail.com>
Message-ID: <095701c6894f$1ad92d20$19477dc1@corporate.sandy.ru>

Hi!

Colleagues, thank you very much for quick and exact answers. I've tested Extension mobility - it's
all I need in this situation. CCM 4.2 is interesting but bound resource :(

--
Alex Pavlenko
System Administrator
Sandy Info



> You could put extension mobility on the phones and have them log in to get a
> profile that is a member of that line group.
>
> CCM 4.2 has a feature like you are requesting,
> http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/prod_release_note09186a00805f1470.html#wp2440103
>
>
>
> On 6/2/06, Alex A. Pavlenko <lex at sandy.ru> wrote:
>>
>> Hi!
>>
>> First, let me explain the situation.We have a group of phones that serves
>> all incoming calls
>> to our company. Operator after accepting a call forwards it to appropriate
>> person. Those
>> phones organized as a line group with broadcast distribution algorithm in
>> CCM.
>> Scenario is rather simple, however there is a big problem in it. Operators
>> are humans
>> so they never stay on their sits for a whole day. In the periods of time
>> when some operators
>> are busy and some are left customer calling to our company gets no answer
>> and thinks
>> that he is ignored. All I want is to give a busy tone to customer during
>> described periods
>> of time. The only thing I see to achieve this is to make operators to
>> switch the phone off before
>> leaving. This is very ugly solution however. In fact it is sufficent to
>> exclude phone
>> line number from a line group when operator leaves and include it back
>> into line group
>> when operator returns, but only administrator can accomplish this task. Is
>> there
>> a possibility, may be some kind of IP Phone service or anything else to
>> let the phone user
>> to change Line group membershit by himself? Any other suggestions?
>>
>> Thanks.
>>
>> --
>> Alex Pavlenko
>> System Administrator
>> Sandy Info
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
>
> -- 
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> 


From cisco-voip at ibh.net  Tue Jun  6 08:00:09 2006
From: cisco-voip at ibh.net (Andre Beck)
Date: Tue, 6 Jun 2006 14:00:09 +0200
Subject: [cisco-voip] No More MTP Resources Available
In-Reply-To: <6101bda50606051217u4ccdc539q41742ec5a51863ad@mail.gmail.com>
References: <6101bda50606051217u4ccdc539q41742ec5a51863ad@mail.gmail.com>
Message-ID: <20060606120009.GA4090@ibh.de>

Hi,

I just hit that issue last week ;)

On Mon, Jun 05, 2006 at 03:17:07PM -0400, Todd Franklin wrote:
> 
> We recently converted one of our gateways to H323 because we needed a lot of
> POTS lines and Caller ID, and this appeared to be the only way to do it.

If only Cisco would not go the way of supporting non-overlapping feature
sets on H.323 vs. MGCP. Either you chose MGCP and lack features that are
present with H.323 only or you chose H.323 lacking features that are there
with MGCP only...
 
> Ever since, my CCM in the Windows Event Viewer, shows lots of occurences of
> this message: "No More MTP Resources Available."  Occasionally, after this
> message, we will drop a call, or a phone (oddly enough, in another location
> not attaced to the H323 gateway) will unregister without warning.

This is the worst case. Way before that, you will notice invocations
of Hold or Transfer (and maybe Park and related features) will fail,
the phone's soft buttons just will not do what is expected.
 
> Is the MTP resource issue related to the H323 thing?

Yep. MTPs are required to provide features like Hold and Transfer on
calls coming into CCM via H.323, as H.323 itself doesn't provide them.
You have to enable it in the gateway/gatekeeper profile, though.

> A friend told me I can add more MTPs by simply installing the software
> on another NT/Win2x server.

First of all you should check how much MTPs are configured on the CCM
itself (Voice Streaming App Service Parameters). It defaults to 48,
while claiming the maximum is 512. Be careful with specifying the max,
my voice streaming service failed to start correctly when I just set
this to 512, it seems to overrun some other internal limits for MOH
ressources if done so. When configuring the number of MTP ressources,
keep in mind that the number you configure is essentially devided by
two, as one MTP is required for each stream direction of a call. So
the default of 48 actually allows you to terminate 24 H.323 calls only.

If you cannot provide enough MTP ressources on the CCM alone (by either
running in the total limit of 512 or even one of the interdependencies
mentioned above) you will have to add external ones. I've not done so
yet, as increasing to 128 was sufficient to terminate my max 60 B-channels,
but your installation might require more.

BTW, in Real Time Monitoring Tool/Performance you can nicely monitor
the MTP usage on your CCM up and down.

-- 
                  The _S_anta _C_laus _O_peration
  or "how to turn a complete illusion into a neverending money source"

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From rleetun at co.boulder.co.us  Tue Jun  6 08:31:55 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Tue, 6 Jun 2006 06:31:55 -0600
Subject: [cisco-voip] VT Advantage on an MCU
Message-ID: <963CE94787005149A52C17F33A028E12051D4803@EVS1B.boco.co.boulder.co.us>

Greetings, 

Here is a work around, obtain a gatekeeper (2851), buy a Sony PCS-TL50,
which can be your MCU and can host 10 connects.  I am in the process of
building this environment.

Cisco CallManager Administration Guide, Release 4.1.3
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_a
dministration_guide_chapter09186a00803ed6e7.html#wp1118192

Understanding Cisco IOS H.323 Gatekeeper Call Routing
http://www.cisco.com/warp/public/788/voip/gk-call-routing.html

Cisco IOS Voice Configuration Library
http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124tcg/v
cl.htm

Hope this helps!


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Monday, June 05, 2006 10:08 PM
To: Ryan Ratliff; IT
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VT Advantage on an MCU

Yes, I was told that a 3511E would be necessary to do continuous
presence, and it was twice the price if I remember correctly.

If I want to do this all without a 3511E, my understanding is that I
will need a h323 gatekeeper. Can someone please drop me a few links
about setting up a gatekeeper? I inherited a vocaltec gk, but don't know
about integrating with CCM. Anyone with advice?

Thanks,
Avidan

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Monday, June 05, 2006 1:33 PM
To: IT
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VT Advantage on an MCU

Continuous presence with SCCP requires the EMP module regardless of what
IPVC gear you have.  For the 3511 you'd need a 3511E.  I'm not on the
sales side so I have no idea what the cost difference between the two
would be.

That said I have no idea why a VTA couldn't call into an H.323
conference.  As long as CM can route the call (via gatekeeper or
directly to the H.323 gateway) there should be no problem.

-Ryan

On Jun 5, 2006, at 3:01 PM, IT wrote:

I am looking for a video conference solution that will allow me to get
VTA users into my video conferences. A few months back, I tried out the
IPVC 3511 and a few Sony PCS-1's loaded with SCCP.  However, that test
was a failure, since the IPVC only showed one site at a time (based on
who was talking). This was entirely unacceptable, and I was told I would
need the 3540 to get the "brady bunch" effect, so I returned the 3511.
Now I am leaning towards the H.323 only video world. I am reloading all
the PCS-1's to H.323 firmware, and purchasing a sony h323 mcu software
upgrade for one of them. Using this configuration, sony has told me that
I will be able to have a 5- camera conference without a problem,
including the grid layout on- screen. That being said, I know that there
is a huge downside to this setup, none of my VTA users will be able to
join, correct?  Does anyone know of a way to get VT advantages to call
in to a h323 video conference hosted by a h323 mcu?



Any advice is truly appreciated.



Thanks,

Avidan



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


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From rratliff at cisco.com  Tue Jun  6 09:12:07 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 6 Jun 2006 09:12:07 -0400
Subject: [cisco-voip] (no subject) OS / BIOS versions
In-Reply-To: <20060606050206.89877.qmail@web37204.mail.mud.yahoo.com>
References: <20060606050206.89877.qmail@web37204.mail.mud.yahoo.com>
Message-ID: <CF995E94-D01B-42B6-8161-FD0A5495F5D1@cisco.com>

Check the readme for 2000.2.7 and 2000.4.2 :)  There is indeed a bios  
upgrade and you want to be on the latest HP bios.

-Ryan

On Jun 6, 2006, at 1:02 AM, Erick Bergquist wrote:

Ryan,

Do the HP BIOS that come with 2000.2-7 release contain the random- 
reboot issues? I have seen a handful as of late but most seem to be  
power-related with some causes unknown. Working on the 2000.4-2  
updates places but just curious about the comment. :)  I have one  
client who states that once they loaded 2000.4-2 their system is  
performing a lot better and smoother/quicker but only thing that  
changed was going from 2-7sr8 to 4-2 with latest SR.

----- Original Message ----
From: Ryan Ratliff <rratliff at cisco.com>
To: Erik Erasmus <ErasmuE4 at telkom.co.za>
Cc: cisco-voip at puck.nether.net
Sent: Monday, June 5, 2006 10:42:24 AM
Subject: Re: [cisco-voip] (no subject)

This could be anything from a power outage to a hardware problem
causing the server to reboot itself.   Basically Windows did not shut
down gracefully for some reason.

If this is an HP server you definitely need to get to the latest
major OS release to get a newer BIOS.   There are known issues with
the microcode in the bios that can cause random reboots.

-Ryan

On Jun 5, 2006, at 11:21 AM, Erik Erasmus ((E)) wrote:


Has anyone seen this message in the MS event viewer before. I was
informed by a customer that their IPt solution running on:

CCM 4-1-3 sr3a -- OS 2000-2-7
MCS 7835 platform
roughly 200 users, IPMA for 10 managers and extension mobility for
everyone - nothing fancy


went down because the server restarted. Unfortunately I was at
another branch and could not see exactly what happened. When I
checked the eventviewer later it had the following message.

The previous system shutdown at date/time was unexpected

A week earlier while I worked at the main site the server went down
once because the power failed - people pulling wires powered the
server off accidentally . In the case now - today - where the message
as shown above is present in the event viewer can it be power again
or is it a real problem with CCM hanging - rebooting etc.


any advice - much appreciated


.
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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  Tue Jun  6 09:22:04 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 6 Jun 2006 09:22:04 -0400
Subject: [cisco-voip] No More MTP Resources Available
In-Reply-To: <20060606120009.GA4090@ibh.de>
References: <6101bda50606051217u4ccdc539q41742ec5a51863ad@mail.gmail.com>
	<20060606120009.GA4090@ibh.de>
Message-ID: <7707EEE2-697B-4192-886C-20D7C33E80B5@cisco.com>

What you describe regarding the supplementary services failing is a  
symptom of "MTP required" set on the gateway and you being out of MTP  
resources.   The only time that setting is required is when you are  
going to some 3rd party H.323 gateways and when you want to use  
outbound FastStart.  If you are going to a Cisco gateway and don't  
care about outbound FastStart then by all means don't require MTP and  
your calls will work fine.

Supplementary services do not require an MTP.

-Ryan

On Jun 6, 2006, at 8:00 AM, Andre Beck wrote:

Hi,

I just hit that issue last week ;)

On Mon, Jun 05, 2006 at 03:17:07PM -0400, Todd Franklin wrote:
>
> We recently converted one of our gateways to H323 because we needed  
> a lot of
> POTS lines and Caller ID, and this appeared to be the only way to  
> do it.

If only Cisco would not go the way of supporting non-overlapping feature
sets on H.323 vs. MGCP. Either you chose MGCP and lack features that are
present with H.323 only or you chose H.323 lacking features that are  
there
with MGCP only...

> Ever since, my CCM in the Windows Event Viewer, shows lots of  
> occurences of
> this message: "No More MTP Resources Available."  Occasionally,  
> after this
> message, we will drop a call, or a phone (oddly enough, in another  
> location
> not attaced to the H323 gateway) will unregister without warning.

This is the worst case. Way before that, you will notice invocations
of Hold or Transfer (and maybe Park and related features) will fail,
the phone's soft buttons just will not do what is expected.

> Is the MTP resource issue related to the H323 thing?

Yep. MTPs are required to provide features like Hold and Transfer on
calls coming into CCM via H.323, as H.323 itself doesn't provide them.
You have to enable it in the gateway/gatekeeper profile, though.

> A friend told me I can add more MTPs by simply installing the software
> on another NT/Win2x server.

First of all you should check how much MTPs are configured on the CCM
itself (Voice Streaming App Service Parameters). It defaults to 48,
while claiming the maximum is 512. Be careful with specifying the max,
my voice streaming service failed to start correctly when I just set
this to 512, it seems to overrun some other internal limits for MOH
ressources if done so. When configuring the number of MTP ressources,
keep in mind that the number you configure is essentially devided by
two, as one MTP is required for each stream direction of a call. So
the default of 48 actually allows you to terminate 24 H.323 calls only.

If you cannot provide enough MTP ressources on the CCM alone (by either
running in the total limit of 512 or even one of the interdependencies
mentioned above) you will have to add external ones. I've not done so
yet, as increasing to 128 was sufficient to terminate my max 60 B- 
channels,
but your installation might require more.

BTW, in Real Time Monitoring Tool/Performance you can nicely monitor
the MTP usage on your CCM up and down.

-- 
                   The _S_anta _C_laus _O_peration
   or "how to turn a complete illusion into a neverending money source"

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From gary.sullivan at waters.nestle.com  Tue Jun  6 09:30:21 2006
From: gary.sullivan at waters.nestle.com (Sullivan, Gary, GREENWICH,
	Information Services)
Date: Tue, 6 Jun 2006 09:30:21 -0400
Subject: [cisco-voip] Two Phones, Same Line
Message-ID: <E9B4C49E7DD9DA459F42493E5485677B038F4C47@usgcte00.nestle.com>

If there using a head set. get a y connector to plug in two head set in to one phone.
We do that in our call center.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net]On Behalf Of Todd Franklin
Sent: Friday, June 02, 2006 7:28 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Two Phones, Same Line


We have a police dispatch center with 7960s.  (Three, side by side).
We hired a new person to train.  Under our old traditional PBX, persons 1 and 2 could listen in on the same phone call.  (If a call came in on Line 1, whoever answered would press the Line1 button, and person 2 would also press Line1).  They could both listen and talk, although person 2 was expected to just listen for training purposes. 

Any way to do this with 7960s?

Thanks.


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From Timothy_Frazee at adp.com  Tue Jun  6 11:32:04 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Tue, 6 Jun 2006 10:32:04 -0500
Subject: [cisco-voip] shared line on ATA186
Message-ID: <26C0F29B-0EC7-4C39-9CEE-5319981D8755@mimectl>

All,

I have a question about setup a shared line on both port of a ATA186.

CCM version is 4.1.3sr3a

Load for the ATA is:  ATA030203SCCP051201A

Codec will be 711 on both ports with no possibilty of needing to xcode.

The reason is that I am setting up an overhead ringer on a Bogen Amp that has a bell input. I am not sure if I can dasey-chain the two amps together off of one ATA port. If i can, then i will just use one port and nevermind then :)

Thanks.


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 rratliff at cisco.com  Tue Jun  6 12:30:25 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 6 Jun 2006 12:30:25 -0400
Subject: [cisco-voip] shared line on ATA186
In-Reply-To: <26C0F29B-0EC7-4C39-9CEE-5319981D8755@mimectl>
References: <26C0F29B-0EC7-4C39-9CEE-5319981D8755@mimectl>
Message-ID: <6F441656-1621-4AB7-BF7D-CD88AD684500@cisco.com>

How about adding them to a broadcast line group?

-Ryan

On Jun 6, 2006, at 11:32 AM, Frazee, Timothy wrote:

All,

I have a question about setup a shared line on both port of a ATA186.

CCM version is 4.1.3sr3a

Load for the ATA is:  ATA030203SCCP051201A

Codec will be 711 on both ports with no possibilty of needing to xcode.

The reason is that I am setting up an overhead ringer on a Bogen Amp  
that has a bell input. I am not sure if I can dasey-chain the two  
amps together off of one ATA port. If i can, then i will just use one  
port and nevermind then :)

Thanks.

This message and any attachments are intended only for the use of the  
addressee and may contain information that is privileged and  
confidential. If the reader of the message is not the intended  
recipient or an authorized representative of the intended recipient,  
you are hereby notified that any dissemination of this communication  
is strictly prohibited. If you have received this communication in  
error, please notify us immediately by e-mail and delete the message  
and any attachments from your system.

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


From mario.velazquez at gmail.com  Tue Jun  6 12:37:12 2006
From: mario.velazquez at gmail.com (Mario Velazquez)
Date: Tue, 06 Jun 2006 11:37:12 -0500
Subject: [cisco-voip] SS7 ITP Configuration
Message-ID: <4485AF38.2080105@gmail.com>

Hi Group

Des anyone has a configuration or a guide for "A-link" termination?

I am using a PGW2200 (9.5.1) and a ITP_SG

Thanks

Mario

From rsingleton at novateck.com  Tue Jun  6 12:51:26 2006
From: rsingleton at novateck.com (Robert Singleton)
Date: Tue, 06 Jun 2006 11:51:26 -0500
Subject: [cisco-voip] shared line on ATA186
In-Reply-To: <26C0F29B-0EC7-4C39-9CEE-5319981D8755@mimectl>
References: <26C0F29B-0EC7-4C39-9CEE-5319981D8755@mimectl>
Message-ID: <200606061151390.SM00904@mail.leasenet.net>

On Tue, 2006-06-06 at 10:32 -0500, Frazee, Timothy wrote:
> All,
>  
> I have a question about setup a shared line on both port of a ATA186.
>  
> CCM version is 4.1.3sr3a
>  
> Load for the ATA is:  ATA030203SCCP051201A
>  
> Codec will be 711 on both ports with no possibilty of needing to
> xcode.
>  
> The reason is that I am setting up an overhead ringer on a Bogen Amp
> that has a bell input. I am not sure if I can dasey-chain the two amps
> together off of one ATA port. If i can, then i will just use one port
> and nevermind then :)

If I understand correctly, you need two ATA ports to ring
simultaneously. I'm going to make a presumption that you are needing to
ring a night bell so people can pick up a call from anywhere in the
building.

I would set up two extensions on the ATA ports, say 4201 and 4202 for
example. These extensions would be in a Call Pickup Group that the
potential call answering people would be able to access. Then create a
line group with the ATA extensions as members and the Distribution
Algorithm as "Broadcast". Associated that line group with a hunt list,
then a hunt pilot with the hunt list. The hunt pilot might be 4200, for
example.

Then, whenever you need the night bells to ring, send incoming calls to
4200. The Broadcast Hunt Group will make both extensions ring
simultaneously, so the night bell inputs to both paging systems would
ring. People can hit More Pickup (or More GPickUp plus the group number,
as is appropriate for their phone) to answer the incoming calls. 

Be sure that the Busy Trigger for the ATA extensions matches or exceeds
the number of incoming calls you need them to be able to ring for
simultaneously. By default, I think this is set to '2', so the third
incoming caller would get a busy signal.

Hope it helps!

Robert




From Timothy_Frazee at adp.com  Tue Jun  6 13:11:53 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Tue, 6 Jun 2006 12:11:53 -0500
Subject: [cisco-voip] shared line on ATA186
In-Reply-To: <1149612686.7338.42.camel@robert.morsco.com>
References: <26C0F29B-0EC7-4C39-9CEE-5319981D8755@mimectl>,
	<1149612686.7338.42.camel@robert.morsco.com>
Message-ID: <CBB6F027-0706-4A7C-B8F2-079D19F97C27@mimectl>

I am familiar with pickup groups and line group etc....

my question is: will the ATA freak out if both ports ring at the same time. I assume it wont as it would be a big deal.

the other question I had is if anyone knows if I can daisy chain two bogen amps connected to the bell input off of one ATA?

what is the ringer equalivance of a single port on an ATA? will it drive two bell inputs on two amps?




From: Robert Singleton
Sent: Tue 6/6/2006 11:51 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] shared line on ATA186


On Tue, 2006-06-06 at 10:32 -0500, Frazee, Timothy wrote:
> All,
>  
> I have a question about setup a shared line on both port of a ATA186.
>  
> CCM version is 4.1.3sr3a
>  
> Load for the ATA is:  ATA030203SCCP051201A
>  
> Codec will be 711 on both ports with no possibilty of needing to
> xcode.
>  
> The reason is that I am setting up an overhead ringer on a Bogen Amp
> that has a bell input. I am not sure if I can dasey-chain the two amps
> together off of one ATA port. If i can, then i will just use one port
> and nevermind then :)

If I understand correctly, you need two ATA ports to ring
simultaneously. I'm going to make a presumption that you are needing to
ring a night bell so people can pick up a call from anywhere in the
building.

I would set up two extensions on the ATA ports, say 4201 and 4202 for
example. These extensions would be in a Call Pickup Group that the
potential call answering people would be able to access. Then create a
line group with the ATA extensions as members and the Distribution
Algorithm as "Broadcast". Associated that line group with a hunt list,
then a hunt pilot with the hunt list. The hunt pilot might be 4200, for
example.

Then, whenever you need the night bells to ring, send incoming calls to
4200. The Broadcast Hunt Group will make both extensions ring
simultaneously, so the night bell inputs to both paging systems would
ring. People can hit More Pickup (or More GPickUp plus the group number,
as is appropriate for their phone) to answer the incoming calls. 

Be sure that the Busy Trigger for the ATA extensions matches or exceeds
the number of incoming calls you need them to be able to ring for
simultaneously. By default, I think this is set to '2', so the third
incoming caller would get a busy signal.

Hope it helps!

Robert



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


This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From cshaffer at gmail.com  Tue Jun  6 13:25:13 2006
From: cshaffer at gmail.com (Curt Shaffer)
Date: Tue, 6 Jun 2006 13:25:13 -0400
Subject: [cisco-voip] Training material
Message-ID: <009b01c6898e$2be05000$8800550a@acg.arlington.local>

Does anyone know of good emulation type training material for CCM and the
needed hardware to practice and learn about VoIP the Cisco way? I would like
to get some hands on but obviously can't afford to build out my own lab ;) I
have experience with Asterisk and I am 3Com VoIP certified, so general VoIP
concepts are really not that important, but I would like to get up to speed
on the Cisco specifics.

 

Thanks

 

Curt

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From newlook110 at yahoo.com  Tue Jun  6 13:52:19 2006
From: newlook110 at yahoo.com (new look)
Date: Tue, 6 Jun 2006 10:52:19 -0700 (PDT)
Subject: [cisco-voip] Reserving line problem at CCM
Message-ID: <20060606175219.5125.qmail@web61326.mail.yahoo.com>

Dear All,
  My company upgraded from PBX telephone system to IP Phones, i have a problem and hope you could help me with it, the problem is that on the old system (PBX) when a line was busy i could reserve this line so that when it is free a get a ring back,  i need to know how to configure this feature on the call manager so that  could reserve busy lines and get a ring back or indication that the line is now free, i'm using call manager 4.1(3) and a 2811 router with FXO ports as my VG, thanks in advance for your cooperation

		
---------------------------------
New Yahoo! Messenger with Voice. Call regular phones from your PC and save big.
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From lelio at uoguelph.ca  Tue Jun  6 14:27:15 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 6 Jun 2006 14:27:15 -0400
Subject: [cisco-voip] Reserving line problem at CCM
References: <20060606175219.5125.qmail@web61326.mail.yahoo.com>
Message-ID: <026901c68996$d605ec90$47196883@cfs.uoguelph.ca>

There is an option called "CallBack" that you should investigate. This does not reserve the line, but notifies the caller when the busy line is available.

--------------------------------------------------------------------------------
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: new look 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, June 06, 2006 1:52 PM
  Subject: [cisco-voip] Reserving line problem at CCM


  Dear All,
  My company upgraded from PBX telephone system to IP Phones, i have a problem and hope you could help me with it, the problem is that on the old system (PBX) when a line was busy i could reserve this line so that when it is free a get a ring back,  i need to know how to configure this feature on the call manager so that  could reserve busy lines and get a ring back or indication that the line is now free, i'm using call manager 4.1(3) and a 2811 router with FXO ports as my VG, thanks in advance for your cooperation


------------------------------------------------------------------------------
  New Yahoo! Messenger with Voice. Call regular phones from your PC and save big.


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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From jason.aarons at us.didata.com  Tue Jun  6 14:42:54 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 6 Jun 2006 14:42:54 -0400
Subject: [cisco-voip] Reserving line problem at CCM
Message-ID: <C1FE15183DA37645BC0633BC604E44F002187E0B@USNAEXCH.na.didata.local>

I haven't used it, but I understand ARC Console has camp-on capability.
Callback would not scale for a receptionist tracking many lines -jason

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, June 06, 2006 2:27 PM
To: new look; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Reserving line problem at CCM

 

There is an option called "CallBack" that you should investigate. This
does not reserve the line, but notifies the caller when the busy line is
available.

 

------------------------------------------------------------------------
--------
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: new look <mailto:newlook110 at yahoo.com>  

	To: cisco-voip at puck.nether.net 

	Sent: Tuesday, June 06, 2006 1:52 PM

	Subject: [cisco-voip] Reserving line problem at CCM

	 

	Dear All,

	My company upgraded from PBX telephone system to IP Phones, i
have a problem and hope you could help me with it, the problem is that
on the old system (PBX) when a line was busy i could reserve this line
so that when it is free a get a ring back,  i need to know how to
configure this feature on the call manager so that  could reserve busy
lines and get a ring back or indication that the line is now free, i'm
using call manager 4.1(3) and a 2811 router with FXO ports as my VG,
thanks in advance for your cooperation

	
________________________________


	New Yahoo! Messenger with Voice. Call regular phones from your
PC
<http://us.rd.yahoo.com/mail_us/taglines/postman5/*http:/us.rd.yahoo.com
/evt=39666/*http:/messenger.yahoo.com>  and save big. 

	
________________________________


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




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

This e-mail communication and any attachments may contain
confidential and privileged information and is for use by the
designated addressee(s) named above only.  If you are not the
intended addressee, you are hereby notified that you have received
this communication in error and that any use or reproduction of
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From jgrace at digitelusa.net  Tue Jun  6 14:42:27 2006
From: jgrace at digitelusa.net (James Grace)
Date: Tue, 6 Jun 2006 14:42:27 -0400
Subject: [cisco-voip] how can i get unity to send VM to smartphones
Message-ID: <200606061454765.SM01572@3729434770>

We have a few smart phones and would like to some get messages from unity or
unified messaging.  Is there a way to do this 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Scott.Voll at wesd.org  Tue Jun  6 15:07:09 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 6 Jun 2006 12:07:09 -0700
Subject: [cisco-voip] how can i get unity to send VM to smartphones
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE175@Apollo.wesd.org>

Traverse networks has an application that ties your cell phone in with
UM to play your messages over the Data network as a wav file.  Don't
know if that's what you were looking for.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, June 06, 2006 11:42 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how can i get unity to send VM to smartphones

 

We have a few smart phones and would like to some get messages from
unity or unified messaging.  Is there a way to do this 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

-------------- next part --------------
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From newlook110 at yahoo.com  Tue Jun  6 15:14:04 2006
From: newlook110 at yahoo.com (new look)
Date: Tue, 6 Jun 2006 12:14:04 -0700 (PDT)
Subject: [cisco-voip] Reserving line problem at CCM
In-Reply-To: <02c401c68999$68bebc90$47196883@cfs.uoguelph.ca>
Message-ID: <20060606191404.65350.qmail@web61325.mail.yahoo.com>

Dear All,
  Thanks very much for your support 
  The Problem I have is when all external lines on my voice gateway are used  by another users no one can make an external call.
  I want reseve any line and when any line become free I get a ring back
   
  and I thing that some communication must be done between the CCM and the  voice gateway
  Thanks
  Waleed


Lelio Fulgenzi <lelio at uoguelph.ca> wrote:          New Look - tell us more! 
   
   
  --------------------------------------------------------------------------------
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: Charlie Watts 
  To: Lelio Fulgenzi 
  Cc: new look 
  Sent: Tuesday, June 06, 2006 2:43 PM
  Subject: RE: [cisco-voip] Reserving line problem at CCM
  

Because he specified "fxo on a 2811" I assumed the calls were using that
link.

But I wasn't aware that QSIG could support CallBack. I suppose it makes
sense. Thanks for the pointer, I'll investigate.

Lelio Fulgenzi wrote:
> It sounded like he had a complete migration and was looking
> for a similar feature on the new system.
> 
> That being said, QSIG trunks supposedly allow for CallBack to
> work across PBXs.
> 
> --------------------------------------------------------------
> ------------------
> 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: Charlie Watts <mailto:cwatts at mercurypay.com>
> To: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  ; new
> look <mailto:newlook110 at yahoo.com>
> Sent: Tuesday, June 06, 2006 2:29 PM
> Subject: RE: [cisco-voip] Reserving line problem at CCM
> 
> But it doesn't work for off-system extensions ...
> 
> He doesn't actually want the "callback when user
> available" feature that
> CM support. He wants "callback when trunk becomes
> free". And I don't
> know that CM supports that. I'm looking forward to
> other responses from
> the list.
> 
> cisco-voip-bounces at puck.nether.net wrote:
> > There is an option called "CallBack" that you should
> > investigate. This does not reserve the line, but notifies the
> > caller when the busy line is available.
> >
> > --------------------------------------------------------------
>
> ------------------ > 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: new look <mailto:newlook110 at yahoo.com>
> > To: cisco-voip at puck.nether.net
> > Sent: Tuesday, June 06, 2006 1:52 PM
> > Subject: [cisco-voip] Reserving line problem at CCM >
> > Dear All,
> > My company upgraded from PBX telephone system to IP
> > Phones, i have a problem and hope you could help me with it,
> > the problem is that on the old system (PBX) when a line was
> > busy i could reserve this line so that when it is free a get
> > a ring back,  i need to know how to configure this feature on
> > the call manager so that  could reserve busy lines and get a
> > ring back or indication that the line is now free, i'm using
> > call manager 4.1(3) and a 2811 router with FXO ports as my
> > VG, thanks in advance for your cooperation
> >
> >
> > ________________________________
> >
> > New Yahoo! Messenger with Voice. Call regular phones > from
your
> PC >
<http://us.rd.yahoo.com/mail_us/taglines/postman5/*http://us.r
> d.yahoo.com/evt=39666/*http://messenger.yahoo.com>  and
> save > big.
> >
> >
> > ________________________________
> >
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> --
> Charlie Watts
> Information Technology
> Mercury Payment Systems
> 970 247 5557 x 3187
> cwatts at mercurypay.com
> http://www.mercurypay.com/



-- 
Charlie Watts
Information Technology
Mercury Payment Systems
970 247 5557 x 3187
cwatts at mercurypay.com
http://www.mercurypay.com/



		
---------------------------------
Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone.  Get Yahoo! Messenger with Voice
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From Scott.Voll at wesd.org  Tue Jun  6 15:15:09 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 6 Jun 2006 12:15:09 -0700
Subject: [cisco-voip] Centrex lines
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE179@Apollo.wesd.org>

Has anyone done a VoIP solution with POTS Centrex lines? 

 

IE> I have a site that current PBX integration they dial 3 digits
between phone on site.  They dial 9+3 digits to other sites.  And dial
9.9+10 digits for outside lines.  I have the outside and inside working
but the other sites are having an issue.  Any ideas?

 

CM 4.1.3sr3a and 2801 with 12 FXO running MGCP.

 

Thanks

 

Scott

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From jason.aarons at us.didata.com  Tue Jun  6 15:18:41 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 6 Jun 2006 15:18:41 -0400
Subject: [cisco-voip] how can i get unity to send VM to smartphones
Message-ID: <C1FE15183DA37645BC0633BC604E44F002187EEC@USNAEXCH.na.didata.local>

4.1(2) adds a Blackberry plug-in.

 

What codec are your using in Unity Unified Messaging?  Your Smartphone
needs a codec for G711ulaw or G729a most likely.  Not sure where you can
get those.

 

Is your SmartPhone Windows Mobile or Palm, Symbian, Linux, etc?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: Tuesday, June 06, 2006 2:42 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how can i get unity to send VM to smartphones

 

We have a few smart phones and would like to some get messages from
unity or unified messaging.  Is there a way to do this 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 




-----------------------------------------
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 jason.aarons at us.didata.com  Tue Jun  6 15:19:43 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 6 Jun 2006 15:19:43 -0400
Subject: [cisco-voip] Centrex lines
Message-ID: <C1FE15183DA37645BC0633BC604E44F002187EF2@USNAEXCH.na.didata.local>

DISASTER.  I did this once.  It sucked. Was a bad idea. Doesn't scale.

 

Each line needs it's own Calling Search Space/Partition.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, June 06, 2006 3:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Centrex lines

 

Has anyone done a VoIP solution with POTS Centrex lines? 

 

IE> I have a site that current PBX integration they dial 3 digits
between phone on site.  They dial 9+3 digits to other sites.  And dial
9.9+10 digits for outside lines.  I have the outside and inside working
but the other sites are having an issue.  Any ideas?

 

CM 4.1.3sr3a and 2801 with 12 FXO running MGCP.

 

Thanks

 

Scott




-----------------------------------------
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
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notify us immediately by replying to this message and deleting it
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From Scott.Voll at wesd.org  Tue Jun  6 15:34:22 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 6 Jun 2006 12:34:22 -0700
Subject: [cisco-voip] Centrex lines
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE184@Apollo.wesd.org>

Why does each line need its own CSS and Part?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Tuesday, June 06, 2006 12:20 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Centrex lines

 

DISASTER.  I did this once.  It sucked. Was a bad idea. Doesn't scale.

 

Each line needs it's own Calling Search Space/Partition.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, June 06, 2006 3:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Centrex lines

 

Has anyone done a VoIP solution with POTS Centrex lines? 

 

IE> I have a site that current PBX integration they dial 3 digits
between phone on site.  They dial 9+3 digits to other sites.  And dial
9.9+10 digits for outside lines.  I have the outside and inside working
but the other sites are having an issue.  Any ideas?

 

CM 4.1.3sr3a and 2801 with 12 FXO running MGCP.

 

Thanks

 

Scott

________________________________



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 newlook110 at yahoo.com  Tue Jun  6 15:37:53 2006
From: newlook110 at yahoo.com (new look)
Date: Tue, 6 Jun 2006 12:37:53 -0700 (PDT)
Subject: [cisco-voip] Reserving line problem at CCM
In-Reply-To: <02fa01c6899c$a49a58c0$47196883@cfs.uoguelph.ca>
Message-ID: <20060606193753.80722.qmail@web61318.mail.yahoo.com>

Ok,Thanks Please If you get any inforamtion please update  me
  I got this feature name at the PBX it is called Camping
   
  I have another Problem 
  I installed the Auto attendant its CTI route port is 5000
  and I add this command at all my voice ports
   
  connection plar opx 5000
   
  So when any one call any No. for my company the call will transfered to the AutoAttendant and the Autoattendant tell the caller to enter the extension no. he wants and every thing is ok
  The problem is when the called extension not answer and the caller hung up the call the ring for the called extension continue ringing
   
  So did any one face this problem or has asolution for this Issue
   
   
  Thanks
  Waleed

Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
      Ah - I see. I don't think that feature is available...however, there is a pre-empt option which people can use to override people's calls. Again, not what you were looking for, but we've got used to the fact that the PBX features just are not there.
   
  --------------------------------------------------------------------------------
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: new look 
  To: Lelio Fulgenzi 
  Sent: Tuesday, June 06, 2006 3:05 PM
  Subject: Re: [cisco-voip] Reserving line problem at CCM
  

  Dear All,
  Thanks very much for your support 
  The Problem I have is when all external lines on my voice gateway are used  by another users no one can make an external call.
  I want reseve any line and when any line become free I get a ring back
   
  and I thing that some communication must be done between the CCM and the  voice gateway
  Thanks
  Waleed
  
Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
          New Look - tell us more! 
   
   
  --------------------------------------------------------------------------------
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: Charlie Watts 
  To: Lelio Fulgenzi 
  Cc: new look 
  Sent: Tuesday, June 06, 2006 2:43 PM
  Subject: RE: [cisco-voip] Reserving line problem at CCM
  

Because he specified "fxo on a 2811" I assumed the calls were using that
link.

But I wasn't aware that QSIG could support CallBack. I suppose it makes
sense. Thanks for the pointer, I'll investigate.

Lelio Fulgenzi wrote:
> It sounded like he had a complete migration and was looking
> for a similar feature on the new system.
> 
> That being said, QSIG trunks supposedly allow for CallBack to
> work across PBXs.
> 
> --------------------------------------------------------------
> ------------------
> 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: Charlie Watts <mailto:cwatts at mercurypay.com>
> To: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  ; new
> look <mailto:newlook110 at yahoo.com>
> Sent: Tuesday, June 06, 2006 2:29 PM
> Subject: RE: [cisco-voip] Reserving line problem at CCM
> 
> But it doesn't work for off-system extensions ...
> 
> He doesn't actually want the "callback when user
> available" feature that
> CM support. He wants "callback when trunk becomes
> free". And I don't
> know that CM supports that. I'm looking forward to
> other responses from
> the list.
> 
> cisco-voip-bounces at puck.nether.net wrote:
> > There is an option called "CallBack" that you should
> > investigate. This does not reserve the line, but notifies the
> > caller when the busy line is available.
> >
> > --------------------------------------------------------------
>
> ------------------ > 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: new look <mailto:newlook110 at yahoo.com>
> > To: cisco-voip at puck.nether.net
> > Sent: Tuesday, June 06, 2006 1:52 PM
> > Subject: [cisco-voip] Reserving line problem at CCM >
> > Dear All,
> > My company upgraded from PBX telephone system to IP
> > Phones, i have a problem and hope you could help me with it,
> > the problem is that on the old system (PBX) when a line was
> > busy i could reserve this line so that when it is free a get
> > a ring back,  i need to know how to configure this feature on
> > the call manager so that  could reserve busy lines and get a
> > ring back or indication that the line is now free, i'm using
> > call manager 4.1(3) and a 2811 router with FXO ports as my
> > VG, thanks in advance for your cooperation
> >
> >
> > ________________________________
> >
> > New Yahoo! Messenger with Voice. Call regular phones > from
your
> PC >
<http://us.rd.yahoo.com/mail_us/taglines/postman5/*http://us.r
> d.yahoo.com/evt=39666/*http://messenger.yahoo.com>  and
> save > big.
> >
> >
> > ________________________________
> >
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> --
> Charlie Watts
> Information Technology
> Mercury Payment Systems
> 970 247 5557 x 3187
> cwatts at mercurypay.com
> http://www.mercurypay.com/



-- 
Charlie Watts
Information Technology
Mercury Payment Systems
970 247 5557 x 3187
cwatts at mercurypay.com
http://www.mercurypay.com/


    
---------------------------------
  Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone. Get Yahoo! Messenger with Voice

		
---------------------------------
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From rratliff at cisco.com  Tue Jun  6 15:47:29 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 6 Jun 2006 15:47:29 -0400
Subject: [cisco-voip] Reserving line problem at CCM
In-Reply-To: <20060606193753.80722.qmail@web61318.mail.yahoo.com>
References: <20060606193753.80722.qmail@web61318.mail.yahoo.com>
Message-ID: <733AB723-4EAF-4BA6-8D2F-E339C1EDB925@cisco.com>

Sounds like a disconnect supervision problem.

Check out http://www.cisco.com/en/US/tech/tk652/tk653/ 
technologies_tech_note09186a00800ae2d1.shtml.

-Ryan

On Jun 6, 2006, at 3:37 PM, new look wrote:

Ok,Thanks Please If you get any inforamtion please update  me
I got this feature name at the PBX it is called Camping

I have another Problem
I installed the Auto attendant its CTI route port is 5000
and I add this command at all my voice ports

connection plar opx 5000

So when any one call any No. for my company the call will transfered  
to the AutoAttendant and the Autoattendant tell the caller to enter  
the extension no. he wants and every thing is ok
The problem is when the called extension not answer and the caller  
hung up the call the ring for the called extension continue ringing

So did any one face this problem or has asolution for this Issue


Thanks
Waleed

Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
Ah - I see. I don't think that feature is available...however, there  
is a pre-empt option which people can use to override people's calls.  
Again, not what you were looking for, but we've got used to the fact  
that the PBX features just are not there.

------------------------------------------------------------------------ 
--------
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: new look
To: Lelio Fulgenzi
Sent: Tuesday, June 06, 2006 3:05 PM
Subject: Re: [cisco-voip] Reserving line problem at CCM

Dear All,
Thanks very much for your support
The Problem I have is when all external lines on my voice gateway are  
used  by another users no one can make an external call.
I want reseve any line and when any line become free I get a ring back

and I thing that some communication must be done between the CCM and  
the  voice gateway
Thanks
Waleed

Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
New Look - tell us more!


------------------------------------------------------------------------ 
--------
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: Charlie Watts
To: Lelio Fulgenzi
Cc: new look
Sent: Tuesday, June 06, 2006 2:43 PM
Subject: RE: [cisco-voip] Reserving line problem at CCM

Because he specified "fxo on a 2811" I assumed the calls were using that
link.

But I wasn't aware that QSIG could support CallBack. I suppose it makes
sense. Thanks for the pointer, I'll investigate.

Lelio Fulgenzi wrote:
 > It sounded like he had a complete migration and was looking
 > for a similar feature on the new system.
 >
 > That being said, QSIG trunks supposedly allow for CallBack to
 > work across PBXs.
 >
 > --------------------------------------------------------------
 > ------------------
 > 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: Charlie Watts <mailto:cwatts at mercurypay.com>
 > To: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  ; new
 > look <mailto:newlook110 at yahoo.com>
 > Sent: Tuesday, June 06, 2006 2:29 PM
 > Subject: RE: [cisco-voip] Reserving line problem at CCM
 >
 > But it doesn't work for off-system extensions ...
 >
 > He doesn't actually want the "callback when user
 > available" feature that
 > CM support. He wants "callback when trunk becomes
 > free". And I don't
 > know that CM supports that. I'm looking forward to
 > other responses from
 > the list.
 >
 > cisco-voip-bounces at puck.nether.net wrote:
 > > There is an option called "CallBack" that you should
 > > investigate. This does not reserve the line, but notifies the
 > > caller when the busy line is available.
 > >
 > > --------------------------------------------------------------
 >
 > ------------------ > 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: new look <mailto:newlook110 at yahoo.com>
 > > To: cisco-voip at puck.nether.net
 > > Sent: Tuesday, June 06, 2006 1:52 PM
 > > Subject: [cisco-voip] Reserving line problem at CCM >
 > > Dear All,
 > > My company upgraded from PBX telephone system to IP
 > > Phones, i have a problem and hope you could help me with it,
 > > the problem is that on the old system (PBX) when a line was
 > > busy i could reserve this line so that when it is free a get
 > > a ring back,  i need to know how to configure this feature on
 > > the call manager so that  could reserve busy lines and get a
 > > ring back or indication that the line is now free, i'm using
 > > call manager 4.1(3) and a 2811 router with FXO ports as my
 > > VG, thanks in advance for your cooperation
 > >
 > >
 > > ________________________________
 > >
 > > New Yahoo! Messenger with Voice. Call regular phones > from
your
 > PC >
<http://us.rd.yahoo.com/mail_us/taglines/postman5/*http://us.r
 > d.yahoo.com/evt=39666/*http://messenger.yahoo.com>  and
 > save > big.
 > >
 > >
 > > ________________________________
 > >
 > >
 > >
 > >
 > > _______________________________________________
 > > cisco-voip mailing list
 > > cisco-voip at puck.nether.net
 > > https://puck.nether.net/mailman/listinfo/cisco-voip
 >
 >
 >
 > --
 > Charlie Watts
 > Information Technology
 > Mercury Payment Systems
 > 970 247 5557 x 3187
 > cwatts at mercurypay.com
 > http://www.mercurypay.com/



-- 
Charlie Watts
Information Technology
Mercury Payment Systems
970 247 5557 x 3187
cwatts at mercurypay.com
http://www.mercurypay.com/


Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone. Get  
Yahoo! Messenger with Voice


Do you Yahoo!?
Get on board. You're invited to try the new Yahoo! Mail Beta.
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip


From newlook110 at yahoo.com  Tue Jun  6 15:56:39 2006
From: newlook110 at yahoo.com (new look)
Date: Tue, 6 Jun 2006 12:56:39 -0700 (PDT)
Subject: [cisco-voip] Auto Attendant Problem
Message-ID: <20060606195639.82580.qmail@web61325.mail.yahoo.com>

Dear All,
  I have CCM 4.1(3) and Cisco 2811 as Vioice GW
   
  I installed the Auto attendant its CTI route port is 5000
  and I add this command at all my voice ports at my voice GW
   
  connection plar opx 5000
   
  So when any one call any No. for my company the call will transfered to the AutoAttendant and the Autoattendant tell the caller to enter the extension no. he wants and every thing is ok
  The problem is when the called extension not answer and the caller hung up the call the ring for the called extension continue ringing
   
  So did any one face this problem or has asolution for this Issue
   
   
  Thanks
  Waleed


		
---------------------------------
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 Get on board. You're invited to try the new Yahoo! Mail Beta.
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From jason.aarons at us.didata.com  Tue Jun  6 16:10:59 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 6 Jun 2006 16:10:59 -0400
Subject: [cisco-voip] Centrex lines
Message-ID: <C1FE15183DA37645BC0633BC604E44F002188048@USNAEXCH.na.didata.local>

If your calls goes out another person's FXO port they can't receive
calls on that port.  Your calls need to be on your and only your Centrex
line.

 

Centrex usually has central office voicemail, this creates problems.

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, June 06, 2006 3:34 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Centrex lines

 

Why does each line need its own CSS and Part?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Tuesday, June 06, 2006 12:20 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Centrex lines

 

DISASTER.  I did this once.  It sucked. Was a bad idea. Doesn't scale.

 

Each line needs it's own Calling Search Space/Partition.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, June 06, 2006 3:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Centrex lines

 

Has anyone done a VoIP solution with POTS Centrex lines? 

 

IE> I have a site that current PBX integration they dial 3 digits
between phone on site.  They dial 9+3 digits to other sites.  And dial
9.9+10 digits for outside lines.  I have the outside and inside working
but the other sites are having an issue.  Any ideas?

 

CM 4.1.3sr3a and 2801 with 12 FXO running MGCP.

 

Thanks

 

Scott

________________________________



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From jwydra at Burwood.com  Tue Jun  6 16:17:11 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Tue, 6 Jun 2006 15:17:11 -0500
Subject: [cisco-voip] General QOS question
Message-ID: <97F472A9F637E44B954D7FBABC800430B84890@bur-msg-02.Burwood.com>

If you don't turn on trunking for an IP phone switch interface, then how
is COS set and/or recognized on ingress?

 

Jason Wydra

Consultant

burwood group, inc.

Phone: (312) 327-4677

Cell: (312) 493-2834

jwydra at burwood.com

 

 

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From David_Zilk at adp.com  Tue Jun  6 19:15:42 2006
From: David_Zilk at adp.com (Zilk, David)
Date: Tue, 6 Jun 2006 18:15:42 -0500
Subject: [cisco-voip] 7920
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BB79748@EXCHANGEDS01.ds.ad.adp.com>

To the best of my knowledge, the 7920 cradle will only charge the phone
if the power cord (the same one used to charge the phone directly) is
plugged into the back of the cradle.

The USB cable that can be connected to the cradle is only used for
configuration.

David


|-----Original Message-----
|From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
|bounces at puck.nether.net] On Behalf Of Loris Vicinanza
|Sent: Wednesday, May 31, 2006 7:11 AM
|To: cisco-voip at puck.nether.net
|Subject: [cisco-voip] 7920
|
|Hi, 7920's wireless phones can be charged via the desktop charger using
a
|standard USB cable, but it doesn't seem to work. Any ideas?
|
|Thanks,
|
|Loris
|
|
|
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|
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This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.


From commo_ssg_31f at yahoo.com  Tue Jun  6 20:07:04 2006
From: commo_ssg_31f at yahoo.com (ash AD)
Date: Tue, 6 Jun 2006 17:07:04 -0700 (PDT)
Subject: [cisco-voip] General QOS question
In-Reply-To: <97F472A9F637E44B954D7FBABC800430B84890@bur-msg-02.Burwood.com>
Message-ID: <20060607000704.58047.qmail@web50409.mail.yahoo.com>

Jason, 
   
      Trunking is not required to set QoS DSCP or COS settings. Those markings are not found in the ISL or 802.1q header; they are found in the ToS byte of the IP header. When Cisco IP phones encapsulate the voice bearer traffic within the IP packet they set the ToS byte by default to the EF QoS value. Most people allow Cisco switches to trust this value and not remark the VoIP packet. 
   
  Peter

"Wydra, Jason" <jwydra at Burwood.com> wrote:
                If you don?t turn on trunking for an IP phone switch interface, then how is COS set and/or recognized on ingress?
   
  Jason Wydra
  Consultant
  burwood group, inc.
  Phone: (312) 327-4677
  Cell: (312) 493-2834
  jwydra at burwood.com
   
   

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


		
---------------------------------
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From commo_ssg_31f at yahoo.com  Tue Jun  6 20:20:44 2006
From: commo_ssg_31f at yahoo.com (ash AD)
Date: Tue, 6 Jun 2006 17:20:44 -0700 (PDT)
Subject: [cisco-voip] Centrex lines
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F002188048@USNAEXCH.na.didata.local>
Message-ID: <20060607002044.5246.qmail@web50412.mail.yahoo.com>

Voll,
   
     Have you tried removing the MGCP controll and going to H.323. Group the FXOs into a trunk group and point your dial-peers to the group. You can still enable incoming PLAR for dedicated 1-to-1 lines.
   
  Peter

"Jason Aarons (US)" <jason.aarons at us.didata.com> wrote:
        v\:* {behavior:url(#default#VML);}  o\:* {behavior:url(#default#VML);}  w\:* {behavior:url(#default#VML);}  ..shape {behavior:url(#default#VML);}                If your calls goes out another person?s FXO port they can?t receive calls on that port.  Your calls need to be on your and only your Centrex line.
   
  Centrex usually has central office voicemail, this creates problems.
   
      
---------------------------------
  
  From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Tuesday, June 06, 2006 3:34 PM
To: Jason Aarons (US); cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Centrex lines

   
  Why does each line need its own CSS and Part?
   
  Scott
   
      
---------------------------------
  
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (US)
Sent: Tuesday, June 06, 2006 12:20 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Centrex lines

   
  DISASTER.  I did this once.  It sucked. Was a bad idea. Doesn?t scale.
   
  Each line needs it?s own Calling Search Space/Partition.
   
      
---------------------------------
  
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, June 06, 2006 3:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Centrex lines

   
  Has anyone done a VoIP solution with POTS Centrex lines? 
   
  IE> I have a site that current PBX integration they dial 3 digits between phone on site.  They dial 9+3 digits to other sites.  And dial 9.9+10 digits for outside lines.  I have the outside and inside working but the other sites are having an issue.  Any ideas?
   
  CM 4.1.3sr3a and 2801 with 12 FXO running MGCP.
   
  Thanks
   
  Scott

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

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

  
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From Joe_Bell at adp.com  Tue Jun  6 20:34:49 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Tue, 6 Jun 2006 19:34:49 -0500
Subject: [cisco-voip] General QOS question
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BB79790@EXCHANGEDS01.ds.ad.adp.com>

Actually, layer 2 QoS (a.k.a CoS settings) are in the 802.1q header.
Layer 3, (DSCP or PHB bits), are the 5 in the IP Precedence field.  If
your not using 802.1q, your not using layer 2 CoS.

 

The reason you see switch configs that are not set for full trunking and
instead have command sets like:

 

Interface fa0/1

 switchport  dynamic desirable  <-- negotiates a quasi-trunk

 switchport  voice vlan 100

 switchport  access vlan 200

 spanning-tree portfast

 

 

Is because the newer voice enabled switches are smart enough to
understand the voice vlan information and respect the tags coming down
from the phone, and there are still 802.1q tags, even though you have
not configured a true trunk.  As I said, the phone does insert an 802.1q
header and tags appropriately, but the port is not in a true trunking
mode.  The reason for this, is the new config keeps a lot of trunking
management protocols off the port and frees it up for voip traffic =
efficiency.

 

Older switches were configured as full trunks and you might still see
that out there.

 

interface FastEthernet0/1   <-- true trunking config

 switchport trunk encapsulation dot1q

 switchport trunk native vlan 100  <-- data and untagged vlan

 switchport mode trunk

 switchport voice vlan 200  <-- tagged 802.1q header with the phone
inserting layer 2 CoS and layer 3 DSCP tags

 spanning-tree portfast

 

These configs accomplish the same thing, but the "newer" switches config
is much more efficient.

 

Joe

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of ash AD
Sent: Tuesday, June 06, 2006 5:07 PM
To: Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] General QOS question

 

Jason, 

 

    Trunking is not required to set QoS DSCP or COS settings. Those
markings are not found in the ISL or 802.1q header; they are found in
the ToS byte of the IP header. When Cisco IP phones encapsulate the
voice bearer traffic within the IP packet they set the ToS byte by
default to the EF QoS value. Most people allow Cisco switches to trust
this value and not remark the VoIP packet. 

 

Peter

"Wydra, Jason" <jwydra at Burwood.com> wrote:

	If you don't turn on trunking for an IP phone switch interface,
then how is COS set and/or recognized on ingress?

	 

	Jason Wydra

	Consultant

	burwood group, inc.

	Phone: (312) 327-4677

	Cell: (312) 493-2834

	jwydra at burwood.com

	 

	 

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

 

________________________________

New Yahoo! Messenger with Voice. Call regular phones from your PC
<http://us.rd.yahoo.com/mail_us/taglines/postman5/*http:/us.rd.yahoo.com
/evt=39666/*http:/messenger.yahoo.com>  and save big.



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 commo_ssg_31f at yahoo.com  Tue Jun  6 20:35:40 2006
From: commo_ssg_31f at yahoo.com (ash AD)
Date: Tue, 6 Jun 2006 17:35:40 -0700 (PDT)
Subject: [cisco-voip] Training material
In-Reply-To: <009b01c6898e$2be05000$8800550a@acg.arlington.local>
Message-ID: <20060607003540.32712.qmail@web50406.mail.yahoo.com>

Curt,
   
     KnowledgeNet offer several great CCVP courses which are much more indepth than your typical CBT. They bring you into simualtion and show lots of live configuration. CBTNuggets is another place to look for great training for those without the equipment. A CallManager lab will cost you more than a small car so, a good CIPT 1 & 2 course can get you on a good track.
   
  Peter

Curt Shaffer <cshaffer at gmail.com> wrote:
                Does anyone know of good emulation type training material for CCM and the needed hardware to practice and learn about VoIP the Cisco way? I would like to get some hands on but obviously can?t afford to build out my own lab ;) I have experience with Asterisk and I am 3Com VoIP certified, so general VoIP concepts are really not that important, but I would like to get up to speed on the Cisco specifics.
   
  Thanks
   
  Curt

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


		
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From jwydra at Burwood.com  Tue Jun  6 22:40:10 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Tue, 6 Jun 2006 21:40:10 -0500
Subject: [cisco-voip] General QOS question
Message-ID: <97F472A9F637E44B954D7FBABC800430B84962@bur-msg-02.Burwood.com>

Thanks for all of the replies. Joe has the answer to my question. I
understand you can still do ToS without trunking but in order to have
CoS then you need 802.1Q which has the 802.1p header. The 802.1p header
is where you find CoS. Without trunking, there is no CoS period. So
basically, the newer switches/IOS look for it anyway even though the
switch port is not set for trunking. The IP phone still does what we
consider standard 802.1q trunking. The switch just sees the CoS
regardless.

 

I must say though that I have never seen the dynamic desirable command
used in implementations I've seen. Maybe I just don't remember.

 

Thanks again!

 

Jason

 

________________________________

From: Bell, Joe [mailto:Joe_Bell at adp.com] 
Sent: Tuesday, June 06, 2006 7:35 PM
To: ash AD; Wydra, Jason; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] General QOS question

 

Actually, layer 2 QoS (a.k.a CoS settings) are in the 802.1q header.
Layer 3, (DSCP or PHB bits), are the 5 in the IP Precedence field.  If
your not using 802.1q, your not using layer 2 CoS.

 

The reason you see switch configs that are not set for full trunking and
instead have command sets like:

 

Interface fa0/1

 switchport  dynamic desirable  <-- negotiates a quasi-trunk

 switchport  voice vlan 100

 switchport  access vlan 200

 spanning-tree portfast

 

 

Is because the newer voice enabled switches are smart enough to
understand the voice vlan information and respect the tags coming down
from the phone, and there are still 802.1q tags, even though you have
not configured a true trunk.  As I said, the phone does insert an 802.1q
header and tags appropriately, but the port is not in a true trunking
mode.  The reason for this, is the new config keeps a lot of trunking
management protocols off the port and frees it up for voip traffic =
efficiency.

 

Older switches were configured as full trunks and you might still see
that out there.

 

interface FastEthernet0/1   <-- true trunking config

 switchport trunk encapsulation dot1q

 switchport trunk native vlan 100  <-- data and untagged vlan

 switchport mode trunk

 switchport voice vlan 200  <-- tagged 802.1q header with the phone
inserting layer 2 CoS and layer 3 DSCP tags

 spanning-tree portfast

 

These configs accomplish the same thing, but the "newer" switches config
is much more efficient.

 

Joe

 

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of ash AD
Sent: Tuesday, June 06, 2006 5:07 PM
To: Wydra, Jason; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] General QOS question

 

Jason, 

 

    Trunking is not required to set QoS DSCP or COS settings. Those
markings are not found in the ISL or 802.1q header; they are found in
the ToS byte of the IP header. When Cisco IP phones encapsulate the
voice bearer traffic within the IP packet they set the ToS byte by
default to the EF QoS value. Most people allow Cisco switches to trust
this value and not remark the VoIP packet. 

 

Peter

"Wydra, Jason" <jwydra at Burwood.com> wrote:

	If you don't turn on trunking for an IP phone switch interface,
then how is COS set and/or recognized on ingress?

	 

	Jason Wydra

	Consultant

	burwood group, inc.

	Phone: (312) 327-4677

	Cell: (312) 493-2834

	jwydra at burwood.com

	 

	 

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

 

________________________________

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From jwydra at Burwood.com  Tue Jun  6 22:48:27 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Tue, 6 Jun 2006 21:48:27 -0500
Subject: [cisco-voip] Problem with CCM 3.3.3 install on VMware 5.5.1 build
	19175
Message-ID: <97F472A9F637E44B954D7FBABC800430B84966@bur-msg-02.Burwood.com>

OK, I'm on my third go around trying to install 3.3.3 on VMware (lab
environment). During the install, it very annoyingly does the familiar
"preparing to install" for about 15 minutes before it comes back and
tells me, "The system cannot read from the specified device". This is a
Windows Installer error by the way. I have 2000 advanced server with SP4
installed. Any ideas? I've installed 4.1.3  and Unity successfully on
the same machine. I have the registry edit.

 

Thanks,

 

Jason Wydra

Consultant

burwood group, inc.

Phone: (312) 327-4677

Cell: (312) 493-2834

jwydra at burwood.com

 

 

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From selt9 at yahoo.com  Wed Jun  7 03:51:27 2006
From: selt9 at yahoo.com (Goran Selthofer)
Date: Wed, 7 Jun 2006 00:51:27 -0700 (PDT)
Subject: [cisco-voip] Reserving line problem at CCM
In-Reply-To: <733AB723-4EAF-4BA6-8D2F-E339C1EDB925@cisco.com>
Message-ID: <20060607075127.63851.qmail@web52306.mail.yahoo.com>

there are many documents on cisco site regarding fxo answer supervison problem, but unfortunatelly none can solve that situation exactly as expected, or somene can update us with exact solution (no links) ?

Ryan Ratliff <rratliff at cisco.com> wrote: Sounds like a disconnect supervision problem.

Check out http://www.cisco.com/en/US/tech/tk652/tk653/ 
technologies_tech_note09186a00800ae2d1.shtml.

-Ryan

On Jun 6, 2006, at 3:37 PM, new look wrote:

Ok,Thanks Please If you get any inforamtion please update  me
I got this feature name at the PBX it is called Camping

I have another Problem
I installed the Auto attendant its CTI route port is 5000
and I add this command at all my voice ports

connection plar opx 5000

So when any one call any No. for my company the call will transfered  
to the AutoAttendant and the Autoattendant tell the caller to enter  
the extension no. he wants and every thing is ok
The problem is when the called extension not answer and the caller  
hung up the call the ring for the called extension continue ringing

So did any one face this problem or has asolution for this Issue


Thanks
Waleed

Lelio Fulgenzi  wrote:
Ah - I see. I don't think that feature is available...however, there  
is a pre-empt option which people can use to override people's calls.  
Again, not what you were looking for, but we've got used to the fact  
that the PBX features just are not there.

------------------------------------------------------------------------ 
--------
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: new look
To: Lelio Fulgenzi
Sent: Tuesday, June 06, 2006 3:05 PM
Subject: Re: [cisco-voip] Reserving line problem at CCM

Dear All,
Thanks very much for your support
The Problem I have is when all external lines on my voice gateway are  
used  by another users no one can make an external call.
I want reseve any line and when any line become free I get a ring back

and I thing that some communication must be done between the CCM and  
the  voice gateway
Thanks
Waleed

Lelio Fulgenzi  wrote:
New Look - tell us more!


------------------------------------------------------------------------ 
--------
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: Charlie Watts
To: Lelio Fulgenzi
Cc: new look
Sent: Tuesday, June 06, 2006 2:43 PM
Subject: RE: [cisco-voip] Reserving line problem at CCM

Because he specified "fxo on a 2811" I assumed the calls were using that
link.

But I wasn't aware that QSIG could support CallBack. I suppose it makes
sense. Thanks for the pointer, I'll investigate.

Lelio Fulgenzi wrote:
 > It sounded like he had a complete migration and was looking
 > for a similar feature on the new system.
 >
 > That being said, QSIG trunks supposedly allow for CallBack to
 > work across PBXs.
 >
 > --------------------------------------------------------------
 > ------------------
 > 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: Charlie Watts 
 > To: Lelio Fulgenzi   ; new
 > look 
 > Sent: Tuesday, June 06, 2006 2:29 PM
 > Subject: RE: [cisco-voip] Reserving line problem at CCM
 >
 > But it doesn't work for off-system extensions ...
 >
 > He doesn't actually want the "callback when user
 > available" feature that
 > CM support. He wants "callback when trunk becomes
 > free". And I don't
 > know that CM supports that. I'm looking forward to
 > other responses from
 > the list.
 >
 > cisco-voip-bounces at puck.nether.net wrote:
 > > There is an option called "CallBack" that you should
 > > investigate. This does not reserve the line, but notifies the
 > > caller when the busy line is available.
 > >
 > > --------------------------------------------------------------
 >
 > ------------------ > 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: new look 
 > > To: cisco-voip at puck.nether.net
 > > Sent: Tuesday, June 06, 2006 1:52 PM
 > > Subject: [cisco-voip] Reserving line problem at CCM >
 > > Dear All,
 > > My company upgraded from PBX telephone system to IP
 > > Phones, i have a problem and hope you could help me with it,
 > > the problem is that on the old system (PBX) when a line was
 > > busy i could reserve this line so that when it is free a get
 > > a ring back,  i need to know how to configure this feature on
 > > the call manager so that  could reserve busy lines and get a
 > > ring back or indication that the line is now free, i'm using
 > > call manager 4.1(3) and a 2811 router with FXO ports as my
 > > VG, thanks in advance for your cooperation
 > >
 > >
 > > ________________________________
 > >
 > > New Yahoo! Messenger with Voice. Call regular phones > from
your
 > PC >

 > d.yahoo.com/evt=39666/*http://messenger.yahoo.com>  and
 > save > big.
 > >
 > >
 > > ________________________________
 > >
 > >
 > >
 > >
 > > _______________________________________________
 > > cisco-voip mailing list
 > > cisco-voip at puck.nether.net
 > > https://puck.nether.net/mailman/listinfo/cisco-voip
 >
 >
 >
 > --
 > Charlie Watts
 > Information Technology
 > Mercury Payment Systems
 > 970 247 5557 x 3187
 > cwatts at mercurypay.com
 > http://www.mercurypay.com/



-- 
Charlie Watts
Information Technology
Mercury Payment Systems
970 247 5557 x 3187
cwatts at mercurypay.com
http://www.mercurypay.com/


Feel free to call! Free PC-to-PC calls. Low rates on PC-to-Phone. Get  
Yahoo! Messenger with Voice


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Get on board. You're invited to try the new Yahoo! Mail Beta.
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https://puck.nether.net/mailman/listinfo/cisco-voip

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From akoul at dataways.gr  Wed Jun  7 06:45:43 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Wed, 07 Jun 2006 13:45:43 +0300
Subject: [cisco-voip] Power consuption by 7941G-GE with POE swithes
Message-ID: <4486AE57.10406@dataways.gr>

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From chris.ellington at nsi1.com  Wed Jun  7 07:10:31 2006
From: chris.ellington at nsi1.com (Chris Ellington)
Date: Wed, 07 Jun 2006 07:10:31 -0400
Subject: [cisco-voip] General QOS question
In-Reply-To: <mailman.4441.1149648524.1478.cisco-voip@puck.nether.net>
Message-ID: <C0AC2C67.77A2%chris.ellington@nsi1.com>

Do you have links on the quasi trunking mode with the phones?  I've done a
quick search and don't find it on the Cisco site (now that's a surprise :D)

My understanding of the desirable command is that is will allow a trunk to
form if the other side requests it (say two switches are connected).  I have
had trouble in the past with both sides wanting a desirable mode in that
neither will 'request' the trunk so it stays in an access mode (that's been
a while ago though, so it's probably fixed).

Also, do you really save much in the way of overhead/processing by using
this quasi trunk?  Let's say I have a Sup1A in a 6500 running 384 ports, all
trunked to various voice vlans - does the extra processing really hurt?

I'm not trying to be difficult, rather just understand the options.  I also
have a team of people to convince; my next move is to go to the SRND and
take the recommendation from there since TAC seems to fall back to that when
I have troubles.

Thanks!

chris

-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com

> Thanks for all of the replies. Joe has the answer to my question. I
> understand you can still do ToS without trunking but in order to have
> CoS then you need 802.1Q which has the 802.1p header. The 802.1p header
> is where you find CoS. Without trunking, there is no CoS period. So
> basically, the newer switches/IOS look for it anyway even though the
> switch port is not set for trunking. The IP phone still does what we
> consider standard 802.1q trunking. The switch just sees the CoS
> regardless.
> 
>  
> 
> I must say though that I have never seen the dynamic desirable command
> used in implementations I've seen. Maybe I just don't remember.
> 
>  
> 
> Thanks again!
> 
>  
> 
> Jason
> 
>  
> 
> ________________________________
> 
> From: Bell, Joe [mailto:Joe_Bell at adp.com]
> Sent: Tuesday, June 06, 2006 7:35 PM
> To: ash AD; Wydra, Jason; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] General QOS question
> 
>  
> 
> Actually, layer 2 QoS (a.k.a CoS settings) are in the 802.1q header.
> Layer 3, (DSCP or PHB bits), are the 5 in the IP Precedence field.  If
> your not using 802.1q, your not using layer 2 CoS.
> 
>  
> 
> The reason you see switch configs that are not set for full trunking and
> instead have command sets like:
> 
>  
> 
> Interface fa0/1
> 
>  switchport  dynamic desirable  <-- negotiates a quasi-trunk
> 
>  switchport  voice vlan 100
> 
>  switchport  access vlan 200
> 
>  spanning-tree portfast
> 
>  
> 
>  
> 
> Is because the newer voice enabled switches are smart enough to
> understand the voice vlan information and respect the tags coming down
> from the phone, and there are still 802.1q tags, even though you have
> not configured a true trunk.  As I said, the phone does insert an 802.1q
> header and tags appropriately, but the port is not in a true trunking
> mode.  The reason for this, is the new config keeps a lot of trunking
> management protocols off the port and frees it up for voip traffic =
> efficiency.
> 
>  
> 
> Older switches were configured as full trunks and you might still see
> that out there.
> 
>  
> 
> interface FastEthernet0/1   <-- true trunking config
> 
>  switchport trunk encapsulation dot1q
> 
>  switchport trunk native vlan 100  <-- data and untagged vlan
> 
>  switchport mode trunk
> 
>  switchport voice vlan 200  <-- tagged 802.1q header with the phone
> inserting layer 2 CoS and layer 3 DSCP tags
> 
>  spanning-tree portfast
> 
>  
> 
> These configs accomplish the same thing, but the "newer" switches config
> is much more efficient.
> 
>  
> 
> Joe
> 
> 



From erickbe at yahoo.com  Wed Jun  7 08:21:18 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 7 Jun 2006 05:21:18 -0700 (PDT)
Subject: [cisco-voip] CCM 3.3(3) and RTMT
Message-ID: <20060607122118.37140.qmail@web37210.mail.mud.yahoo.com>

With CCM 3.3(3) with 1 publisher, and 2 subscriber servers the RTMT Callmanager Cluster info properties, the info for the 2nd subscriber does not report back in RTMT tool on the cluster info window. It saids "Error" next to most values but the first 2 fields in list show good data (registered phones).
 
The info from the other servers displays fine. If you launch RTMT from the problem subscriber and click the info button the RTMT server version is empty. 
 
So like it can't access data on this server, but this happens when you RTMT locally on the subscriber or remotely with IE over web. You can view all performance counters fine in RTMT on this server, it's just the cluster info window and RTMT version info, and processor summary window. 

Windows performance monitor appears fine, can pull up counters fine on all servers.

I have checked to make sure all the services were running, have restarted RIS DC service, etc and can't seem to find anything different between them yet. 
 
CCM 3.3(3)sr4a 
OS 2000.4-2sr7  
RTMT client version 3.3(0.21) 
RTMT server version is 3.3.2.0 
 
This has been an issue for awhile, even before OS updates from 2000.2-7 to 2000.4-2. Not sure if RTMT ever worked fine on this one server.

I've done several bug searches and can't find a good hit yet. I do realize 3.3(3) and RTMT is from the early days of RTMT so expect there may be some issues since RTMT in 4.x is completely different.

Anyone have any ideas?

Thanks, Erick


From jim at tgasolutions.com  Wed Jun  7 09:05:50 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Wed, 7 Jun 2006 09:05:50 -0400
Subject: [cisco-voip] VM After Scheduled Bus hrs.
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC15E3CA@exchange.TGASolutions.local>

Hi guys,

I am working on an issue where after business hours a phone without a
DID when someone transfers from Unity via the dial by name etc goes
directly to VM and does not ring the phone.

I know the issue is sitting right there on the tip of my tongue, but I
just an having a non-thinking moment.


Ideas anyone?
Thanks,
Jim



From erickbe at yahoo.com  Wed Jun  7 10:09:34 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 7 Jun 2006 07:09:34 -0700 (PDT)
Subject: [cisco-voip] VM After Scheduled Bus hrs.
In-Reply-To: <D2BBAE709B4D874DAC556C7EB9D938FC15E3CA@exchange.TGASolutions.local>
Message-ID: <20060607140934.67006.qmail@web37215.mail.mud.yahoo.com>

Is the unity subscriber call transfer set to transfer the call or go directly to the greeting? Is the transfer number correct?  Does the CSS on the voice mail ports have right partitions, etc to call that number?

If everythings looks fine config-wise, any errors in the application event log from time of call / transfer attempt?

----- Original Message ----
From: Jim McBurnett <jim at tgasolutions.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Wednesday, June 7, 2006 8:05:50 AM
Subject: [cisco-voip] VM After Scheduled Bus hrs.

Hi guys,

I am working on an issue where after business hours a phone without a
DID when someone transfers from Unity via the dial by name etc goes
directly to VM and does not ring the phone.

I know the issue is sitting right there on the tip of my tongue, but I
just an having a non-thinking moment.


Ideas anyone?
Thanks,
Jim


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





From PRuttman at foley.com  Wed Jun  7 10:22:55 2006
From: PRuttman at foley.com (Ruttman, Peter G.)
Date: Wed, 7 Jun 2006 09:22:55 -0500
Subject: [cisco-voip] how to load dsp firmware
Message-ID: <7E3064B7EFC16C41B185C41D141008B701A6DC21@FOLEYEX97.foleylaw.com>

TAC says we may be experiencing this bug:
 

CSCsd54344: DSP: Persistent echo heard on c5510 DSPs with 4.4.13 DSPware
or higher

Symptom:

Under certain circumstances, continual echo may be heard. The echo is
perisitent and the DSP echo canceller does not converge.

 

Conditions:

DSPware 4.4.13 or higher on C5510 DSPs are required to experience this
problem.

 

Workaround:

Downgrading DSPware to 4.4.12 or earlier resolves the echo issue.

 

TAC send is firmware and these instructions:

INSTRUCTIONS:

1. Put the file on the flash.

2. Reload the router.

3. Verify new DSP version ' sh voice dsp' (should be 4.4.18)

3. IMPORTANT: Remove file from flash after verification

 

However, I did the show voice dsp and saw no change in the dsp version
from before getting the new dsp firmware.  Can someone tell me if this
is the correct way to load firmware on an NM-HDV2 on a 3745 voice
gateway?

Thanks,

Pete Ruttman

CCNA  CCVP

Senior Network Engineer

Foley & Lardner LLP

 

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From Scott.Voll at wesd.org  Wed Jun  7 10:32:56 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 7 Jun 2006 07:32:56 -0700
Subject: [cisco-voip] General QOS question
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE256@Apollo.wesd.org>

Chris--

Quasi -- to some degree; having a likeness to something; resembling;
having some resemblance; etc.

It's not really a trunking mode.

The desirable automagically goes into the config when you have both
switchport access and voice on the same interface.  You don't have to
manually set it.

QoS at layer two is basically done by trusting the CoS, ToS, DSCP and
passing it along. IE> 

interface FastEthernet0/6
 switchport access vlan 2
 switchport mode dynamic desirable
 switchport voice vlan 110
 no ip address
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 priority-queue out
 mls qos trust device cisco-phone
 mls qos trust cos
 no mdix auto
 auto qos voip cisco-phone
 spanning-tree portfast

Hope that helps.

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: Wednesday, June 07, 2006 4:11 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] General QOS question

Do you have links on the quasi trunking mode with the phones?  I've done
a
quick search and don't find it on the Cisco site (now that's a surprise
:D)

My understanding of the desirable command is that is will allow a trunk
to
form if the other side requests it (say two switches are connected).  I
have
had trouble in the past with both sides wanting a desirable mode in that
neither will 'request' the trunk so it stays in an access mode (that's
been
a while ago though, so it's probably fixed).

Also, do you really save much in the way of overhead/processing by using
this quasi trunk?  Let's say I have a Sup1A in a 6500 running 384 ports,
all
trunked to various voice vlans - does the extra processing really hurt?

I'm not trying to be difficult, rather just understand the options.  I
also
have a team of people to convince; my next move is to go to the SRND and
take the recommendation from there since TAC seems to fall back to that
when
I have troubles.

Thanks!

chris

-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com

> Thanks for all of the replies. Joe has the answer to my question. I
> understand you can still do ToS without trunking but in order to have
> CoS then you need 802.1Q which has the 802.1p header. The 802.1p
header
> is where you find CoS. Without trunking, there is no CoS period. So
> basically, the newer switches/IOS look for it anyway even though the
> switch port is not set for trunking. The IP phone still does what we
> consider standard 802.1q trunking. The switch just sees the CoS
> regardless.
> 
>  
> 
> I must say though that I have never seen the dynamic desirable command
> used in implementations I've seen. Maybe I just don't remember.
> 
>  
> 
> Thanks again!
> 
>  
> 
> Jason
> 
>  
> 
> ________________________________
> 
> From: Bell, Joe [mailto:Joe_Bell at adp.com]
> Sent: Tuesday, June 06, 2006 7:35 PM
> To: ash AD; Wydra, Jason; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] General QOS question
> 
>  
> 
> Actually, layer 2 QoS (a.k.a CoS settings) are in the 802.1q header.
> Layer 3, (DSCP or PHB bits), are the 5 in the IP Precedence field.  If
> your not using 802.1q, your not using layer 2 CoS.
> 
>  
> 
> The reason you see switch configs that are not set for full trunking
and
> instead have command sets like:
> 
>  
> 
> Interface fa0/1
> 
>  switchport  dynamic desirable  <-- negotiates a quasi-trunk
> 
>  switchport  voice vlan 100
> 
>  switchport  access vlan 200
> 
>  spanning-tree portfast
> 
>  
> 
>  
> 
> Is because the newer voice enabled switches are smart enough to
> understand the voice vlan information and respect the tags coming down
> from the phone, and there are still 802.1q tags, even though you have
> not configured a true trunk.  As I said, the phone does insert an
802.1q
> header and tags appropriately, but the port is not in a true trunking
> mode.  The reason for this, is the new config keeps a lot of trunking
> management protocols off the port and frees it up for voip traffic =
> efficiency.
> 
>  
> 
> Older switches were configured as full trunks and you might still see
> that out there.
> 
>  
> 
> interface FastEthernet0/1   <-- true trunking config
> 
>  switchport trunk encapsulation dot1q
> 
>  switchport trunk native vlan 100  <-- data and untagged vlan
> 
>  switchport mode trunk
> 
>  switchport voice vlan 200  <-- tagged 802.1q header with the phone
> inserting layer 2 CoS and layer 3 DSCP tags
> 
>  spanning-tree portfast
> 
>  
> 
> These configs accomplish the same thing, but the "newer" switches
config
> is much more efficient.
> 
>  
> 
> Joe
> 
> 


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


From Scott.Voll at wesd.org  Wed Jun  7 10:41:09 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 7 Jun 2006 07:41:09 -0700
Subject: [cisco-voip] Power consuption by 7941G-GE with POE swithes
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE25A@Apollo.wesd.org>

Anthony-

 

This just came up on the list a couple of weeks ago and I ran it past my
SE.  The Gig phones require twice the power of the non-gig phones and
therefore you can only run 24 gig phones on a 48 port PoE switch....
Leave it to cisco to come up with such a great design....... NOT.
Basically I got the impression that if you were going to use all Gig
phones to buy only the 24 port switches.   Ci$co.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony
Kouloglou
Sent: Wednesday, June 07, 2006 3:46 AM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] Power consuption by 7941G-GE with POE swithes

 

Hi all,
i have implemented at a customer a WS-C3560G-48PS (POE) and several
7941G-GE phones.

1.How much power (in Watts) do they need when they are with inline power
from the switch? In the specifications of the phone it says: 
"48 VDC, 0.2 A-when using the in-line power over the network cable" (so
does it need 48x0.2=9.6watts to work?) . 
I must use the command "#power inline static max <milliwatts>" to
restrict the phone from drawning the maximum power which is now
12.9Watts (as shown by show cdp neaighbous detail) and so it reduces the
maximum number of phones the 3560 can support (after i put 27 phones it
could not give any more power)?
2. For a 7971, i also plugged the power cube but it continues to drawn
15.050Watts!! AC power adaptor cannot cover it's needs for power and so
it would not have to drawn any power from the POE switch?
3 Can i use a RPS that i have on site and configure it to give extra
power to the switch to support 48 phones of any kind?

Thanks a lot
Anthony kouloglou

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From rlatorre at unislumin.com  Wed Jun  7 11:19:56 2006
From: rlatorre at unislumin.com (Ryan LaTorre)
Date: Wed, 7 Jun 2006 11:19:56 -0400
Subject: [cisco-voip] how to load dsp firmware
Message-ID: <18082B35CF0EBD4C8D265874508A615C02C6587F@hermes.unislumin.com>

 
I've encountered this bug as well, and the instructions below seem
pretty close to what I do.
 
Is the filename "dsp_c5510_flex.rbf"? I believe the firmware must have
that exact name, and be installed on the root of the flash.
 
Also, I don't remove the file from the flash after the reboot, since a
subsequent reboot seems to then use the DSP firmware bundled in the IOS.
 
I hope this helps.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter
G.
Sent: Wednesday, June 07, 2006 10:23 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to load dsp firmware


TAC says we may be experiencing this bug:
 

CSCsd54344: DSP: Persistent echo heard on c5510 DSPs with 4.4.13 DSPware
or higher

Symptom:

Under certain circumstances, continual echo may be heard. The echo is
perisitent and the DSP echo canceller does not converge.

 

Conditions:

DSPware 4.4.13 or higher on C5510 DSPs are required to experience this
problem.

 

Workaround:

Downgrading DSPware to 4.4.12 or earlier resolves the echo issue.

 

TAC send is firmware and these instructions:

INSTRUCTIONS:

1. Put the file on the flash.

2. Reload the router.

3. Verify new DSP version ' sh voice dsp' (should be 4.4.18)

3. IMPORTANT: Remove file from flash after verification

 

However, I did the show voice dsp and saw no change in the dsp version
from before getting the new dsp firmware.  Can someone tell me if this
is the correct way to load firmware on an NM-HDV2 on a 3745 voice
gateway?

Thanks,

Pete Ruttman

CCNA  CCVP

Senior Network Engineer

Foley & Lardner LLP

 

The preceding email message may be confidential or protected by the
attorney-client privilege. It is not intended for transmission to, or
receipt by, any unauthorized persons. If you have received this message
in error, please (i) do not read it, (ii) reply to the sender that you
received the message in error, and (iii) erase or destroy the message.
Legal advice contained in the preceding message is solely for the
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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.


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 McLean.Pickett at ptgcorp.com  Wed Jun  7 11:25:16 2006
From: McLean.Pickett at ptgcorp.com (McLean Pickett)
Date: Wed, 7 Jun 2006 11:25:16 -0400
Subject: [cisco-voip] how to load dsp firmware
Message-ID: <36E88C352318BC468EC5EF07C76ACC9B0101338A@Z3EXCHANGE01.ptg-domain.com>

Pete -
 
I was given explicit direction not to remove the file from firmware as
the DSP's would revert to the old firmware. 
 
McLean

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter
G.
Sent: Wednesday, June 07, 2006 10:23 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to load dsp firmware


TAC says we may be experiencing this bug:
 

CSCsd54344: DSP: Persistent echo heard on c5510 DSPs with 4.4.13 DSPware
or higher

Symptom:

Under certain circumstances, continual echo may be heard. The echo is
perisitent and the DSP echo canceller does not converge.

 

Conditions:

DSPware 4.4.13 or higher on C5510 DSPs are required to experience this
problem.

 

Workaround:

Downgrading DSPware to 4.4.12 or earlier resolves the echo issue.

 

TAC send is firmware and these instructions:

INSTRUCTIONS:

1. Put the file on the flash.

2. Reload the router.

3. Verify new DSP version ' sh voice dsp' (should be 4.4.18)

3. IMPORTANT: Remove file from flash after verification

 

However, I did the show voice dsp and saw no change in the dsp version
from before getting the new dsp firmware.  Can someone tell me if this
is the correct way to load firmware on an NM-HDV2 on a 3745 voice
gateway?

Thanks,

Pete Ruttman

CCNA  CCVP

Senior Network Engineer

Foley & Lardner LLP

 

The preceding email message may be confidential or protected by the
attorney-client privilege. It is not intended for transmission to, or
receipt by, any unauthorized persons. If you have received this message
in error, please (i) do not read it, (ii) reply to the sender that you
received the message in error, and (iii) erase or destroy the message.
Legal advice contained in the preceding message is solely for the
benefit of the Foley & Lardner LLP client(s) represented by the Firm in
the particular matter that is the subject of this message, and may not
be relied upon by any other party. 


Internal Revenue Service regulations require that certain types of
written advice include a disclaimer. To the extent the preceding message
contains advice relating to a Federal tax issue, unless expressly stated
otherwise the advice is not intended or written to be used, and it
cannot be used by the recipient or any other taxpayer, for the purpose
of avoiding Federal tax penalties, and was not written to support the
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From Andy.Clements at eon-is.co.uk  Wed Jun  7 11:30:46 2006
From: Andy.Clements at eon-is.co.uk (Clements, Andy (E.ON IS UK))
Date: Wed, 7 Jun 2006 16:30:46 +0100
Subject: [cisco-voip] General QOS question
Message-ID: <4B304708E89C5E44ACD9A1E526C697870A87FB6F@CORWWPKSEXV03.corp.pg.eon.net>

Hi,

I do not think you need the dynamic desirable as I have configured edge
switch interfaces as switch mode access with the switch voice vlan
command and that still sends the voice vlan in a dot1q trunk and the
access vlan as the native untagged vlan.

[Example]
 switchport access vlan 505
 switchport mode access
 switchport voice vlan 405

I believe this "newer" configuration for voice is to allow the use of
802.1x as this wont work on a "trunked interface". Although I haven't
had experience of this.

By the way the configuration above is working for both Cisco and Nortel
IPT phones.

Regards Andy

-----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, June 07, 2006 3:33 PM
To: Chris Ellington; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] General QOS question

Chris--

Quasi -- to some degree; having a likeness to something; resembling;
having some resemblance; etc.

It's not really a trunking mode.

The desirable automagically goes into the config when you have both
switchport access and voice on the same interface.  You don't have to
manually set it.

QoS at layer two is basically done by trusting the CoS, ToS, DSCP and
passing it along. IE> 

interface FastEthernet0/6
 switchport access vlan 2
 switchport mode dynamic desirable
 switchport voice vlan 110
 no ip address
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 priority-queue out
 mls qos trust device cisco-phone
 mls qos trust cos
 no mdix auto
 auto qos voip cisco-phone
 spanning-tree portfast

Hope that helps.

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: Wednesday, June 07, 2006 4:11 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] General QOS question

Do you have links on the quasi trunking mode with the phones?  I've done
a
quick search and don't find it on the Cisco site (now that's a surprise
:D)

My understanding of the desirable command is that is will allow a trunk
to
form if the other side requests it (say two switches are connected).  I
have
had trouble in the past with both sides wanting a desirable mode in that
neither will 'request' the trunk so it stays in an access mode (that's
been
a while ago though, so it's probably fixed).

Also, do you really save much in the way of overhead/processing by using
this quasi trunk?  Let's say I have a Sup1A in a 6500 running 384 ports,
all
trunked to various voice vlans - does the extra processing really hurt?

I'm not trying to be difficult, rather just understand the options.  I
also
have a team of people to convince; my next move is to go to the SRND and
take the recommendation from there since TAC seems to fall back to that
when
I have troubles.

Thanks!

chris

-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com

> Thanks for all of the replies. Joe has the answer to my question. I
> understand you can still do ToS without trunking but in order to have
> CoS then you need 802.1Q which has the 802.1p header. The 802.1p
header
> is where you find CoS. Without trunking, there is no CoS period. So
> basically, the newer switches/IOS look for it anyway even though the
> switch port is not set for trunking. The IP phone still does what we
> consider standard 802.1q trunking. The switch just sees the CoS
> regardless.
> 
>  
> 
> I must say though that I have never seen the dynamic desirable command
> used in implementations I've seen. Maybe I just don't remember.
> 
>  
> 
> Thanks again!
> 
>  
> 
> Jason
> 
>  
> 
> ________________________________
> 
> From: Bell, Joe [mailto:Joe_Bell at adp.com]
> Sent: Tuesday, June 06, 2006 7:35 PM
> To: ash AD; Wydra, Jason; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] General QOS question
> 
>  
> 
> Actually, layer 2 QoS (a.k.a CoS settings) are in the 802.1q header.
> Layer 3, (DSCP or PHB bits), are the 5 in the IP Precedence field.  If
> your not using 802.1q, your not using layer 2 CoS.
> 
>  
> 
> The reason you see switch configs that are not set for full trunking
and
> instead have command sets like:
> 
>  
> 
> Interface fa0/1
> 
>  switchport  dynamic desirable  <-- negotiates a quasi-trunk
> 
>  switchport  voice vlan 100
> 
>  switchport  access vlan 200
> 
>  spanning-tree portfast
> 
>  
> 
>  
> 
> Is because the newer voice enabled switches are smart enough to
> understand the voice vlan information and respect the tags coming down
> from the phone, and there are still 802.1q tags, even though you have
> not configured a true trunk.  As I said, the phone does insert an
802.1q
> header and tags appropriately, but the port is not in a true trunking
> mode.  The reason for this, is the new config keeps a lot of trunking
> management protocols off the port and frees it up for voip traffic =
> efficiency.
> 
>  
> 
> Older switches were configured as full trunks and you might still see
> that out there.
> 
>  
> 
> interface FastEthernet0/1   <-- true trunking config
> 
>  switchport trunk encapsulation dot1q
> 
>  switchport trunk native vlan 100  <-- data and untagged vlan
> 
>  switchport mode trunk
> 
>  switchport voice vlan 200  <-- tagged 802.1q header with the phone
> inserting layer 2 CoS and layer 3 DSCP tags
> 
>  spanning-tree portfast
> 
>  
> 
> These configs accomplish the same thing, but the "newer" switches
config
> is much more efficient.
> 
>  
> 
> Joe
> 
> 


_______________________________________________
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 Jun  7 11:34:11 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 7 Jun 2006 10:34:11 -0500
Subject: [cisco-voip] Training material
In-Reply-To: <20060607003540.32712.qmail@web50406.mail.yahoo.com>
References: <009b01c6898e$2be05000$8800550a@acg.arlington.local>
	<20060607003540.32712.qmail@web50406.mail.yahoo.com>
Message-ID: <5d093f9a0606070834s3871bd40mdb3244ba7b36cf27@mail.gmail.com>

That is not true.

I built a CCM/Unity lab for under $1800..

I built a server with 600GB HD space, 4GB RAM and a P4 dual... Loaded
VMWare GSX and built a CCM cluster with Unity...

Granted, I then added a bunch of phones and a 1760 gateway... (I also
had a bunch of routers (2610 and a pile of 2500s) and a 3550 from my
R&S days).


Jonathan

On 6/6/06, ash AD <commo_ssg_31f at yahoo.com> wrote:
>
> Curt,
>
>    KnowledgeNet offer several great CCVP courses which are much more indepth
> than your typical CBT. They bring you into simualtion and show lots of live
> configuration. CBTNuggets is another place to look for great training for
> those without the equipment. A CallManager lab will cost you more than a
> small car so, a good CIPT 1 & 2 course can get you on a good track.
>
> Peter
>
>
> Curt Shaffer <cshaffer at gmail.com> wrote:
>
>
>
> Does anyone know of good emulation type training material for CCM and the
> needed hardware to practice and learn about VoIP the Cisco way? I would like
> to get some hands on but obviously can't afford to build out my own lab ;) I
> have experience with Asterisk and I am 3Com VoIP certified, so general VoIP
> concepts are really not that important, but I would like to get up to speed
> on the Cisco specifics.
>
> Thanks
>
> Curt
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>  ________________________________
> How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call rates.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From mounir.mohamed at gmail.com  Wed Jun  7 11:41:53 2006
From: mounir.mohamed at gmail.com (Mounir Mohamed)
Date: Wed, 7 Jun 2006 18:41:53 +0300
Subject: [cisco-voip] General QOS question
In-Reply-To: <97F472A9F637E44B954D7FBABC800430B84890@bur-msg-02.Burwood.com>
References: <97F472A9F637E44B954D7FBABC800430B84890@bur-msg-02.Burwood.com>
Message-ID: <fca5f5380606070841xdaaba20n200eec2a570db12@mail.gmail.com>

Dear jason,

The IP phone switch port must be configured as a trunk..Why? Because the COS
marking attached to the 3bits User priority of TCI 1byte of the ethernet
frame (802.1Q), or to the VLAN-ID 3 bits in the ISL header so if the port
not configured as trunk so the Filed that contain the COS will not be found
this is whay the IP Phone port must be configured as trunk port.


Best Regards,
Mounir Mohamed



On 6/6/06, Wydra, Jason <jwydra at burwood.com> wrote:
>
>   If you don't turn on trunking for an IP phone switch interface, then how
> is COS set and/or recognized on ingress?
>
>
>
> *Jason Wydra*
>
> *Consultant***
>
> *burwood* *group, inc.*
>
> *Phone: (312) 327-4677***
>
> *Cell: (312) 493-2834*
>
> jwydra at burwood.com
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Best Reagrds,
Mounir Mohamed
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From rleetun at co.boulder.co.us  Wed Jun  7 11:42:23 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Wed, 7 Jun 2006 09:42:23 -0600
Subject: [cisco-voip] IPCC installed on Vmware for testing environment
Message-ID: <963CE94787005149A52C17F33A028E12051D4820@EVS1B.boco.co.boulder.co.us>

Good morning,

Has anyone had any success of putting IPCC 3.5.3 on Vmware?  I am trying
to create a testing environment for our team.

Thanks.

Rob



From jwydra at Burwood.com  Wed Jun  7 11:52:38 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Wed, 7 Jun 2006 10:52:38 -0500
Subject: [cisco-voip] Training material
Message-ID: <97F472A9F637E44B954D7FBABC800430B84A56@bur-msg-02.Burwood.com>

I built mine for around $2500. I have four 2600s from Ebay with various
netmods, an ATA186 plus VMware running 2 CCMS, Unity and IPCC Express (2
clusters). The 2600s go for around $150 on Ebay. I don't have any XM
series though. I'm using 1 2600 as a Frame switch and a Gatekeeper. I
have 2 7960's and one 7940. I also bought some cheap analog phones for 9
bucks each.

By the way, just because I'm lazy and don't feel like looking it up,
does anyone know off the top of their head what the bare requirements
are for CME? Platform/memory?

The only thing I'm missing in my lab is a switch with advanced image,
6509, VG248, ISR router. These are all big ticket items that would
definitely bump up the overall cost. Especially if you're crazy enough
to by a 6509 chassis.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Wednesday, June 07, 2006 10:34 AM
To: ash AD
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Training material

That is not true.

I built a CCM/Unity lab for under $1800..

I built a server with 600GB HD space, 4GB RAM and a P4 dual... Loaded
VMWare GSX and built a CCM cluster with Unity...

Granted, I then added a bunch of phones and a 1760 gateway... (I also
had a bunch of routers (2610 and a pile of 2500s) and a 3550 from my
R&S days).


Jonathan

On 6/6/06, ash AD <commo_ssg_31f at yahoo.com> wrote:
>
> Curt,
>
>    KnowledgeNet offer several great CCVP courses which are much more
indepth
> than your typical CBT. They bring you into simualtion and show lots of
live
> configuration. CBTNuggets is another place to look for great training
for
> those without the equipment. A CallManager lab will cost you more than
a
> small car so, a good CIPT 1 & 2 course can get you on a good track.
>
> Peter
>
>
> Curt Shaffer <cshaffer at gmail.com> wrote:
>
>
>
> Does anyone know of good emulation type training material for CCM and
the
> needed hardware to practice and learn about VoIP the Cisco way? I
would like
> to get some hands on but obviously can't afford to build out my own
lab ;) I
> have experience with Asterisk and I am 3Com VoIP certified, so general
VoIP
> concepts are really not that important, but I would like to get up to
speed
> on the Cisco specifics.
>
> Thanks
>
> Curt
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>  ________________________________
> How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call
rates.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From jonvoip at gmail.com  Wed Jun  7 12:21:24 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 7 Jun 2006 11:21:24 -0500
Subject: [cisco-voip] Training material
In-Reply-To: <4486F4D9.2070108@munzani.com>
References: <009b01c6898e$2be05000$8800550a@acg.arlington.local>
	<20060607003540.32712.qmail@web50406.mail.yahoo.com>
	<5d093f9a0606070834s3871bd40mdb3244ba7b36cf27@mail.gmail.com>
	<4486F4D9.2070108@munzani.com>
Message-ID: <5d093f9a0606070921o2b8be9fcte9b35914fa6eb583@mail.gmail.com>

CCM is not licensed... just a reg hack and a license key which can be
found here:

http://www.cisco.com/warp/public/788/AVVID/productkey.html

As far as the Unity license goes, yes, I did grab it from work... and
I had to hack the MAC address of the windows box to make it match
(can't do that with CCM5...)...



Jonathan

On 6/7/06, Sam Munzani <sam at munzani.com> wrote:
>
>  Jonathan,
>
>  You missed biggest item. The licenses for CCM and Unity. You might be
> fortunate enough to use licenses from your work but everybody may not have
> same luxury :-(
>
>  Sam
>
>
>  That is not true.
>
> I built a CCM/Unity lab for under $1800..
>
> I built a server with 600GB HD space, 4GB RAM and a P4 dual... Loaded
> VMWare GSX and built a CCM cluster with Unity...
>
> Granted, I then added a bunch of phones and a 1760 gateway... (I also
> had a bunch of routers (2610 and a pile of 2500s) and a 3550 from my
> R&S days).
>
>
> Jonathan
>
> On 6/6/06, ash AD <commo_ssg_31f at yahoo.com> wrote:
>
>
>  Curt,
>
>  KnowledgeNet offer several great CCVP courses which are much more indepth
> than your typical CBT. They bring you into simualtion and show lots of live
> configuration. CBTNuggets is another place to look for great training for
> those without the equipment. A CallManager lab will cost you more than a
> small car so, a good CIPT 1 & 2 course can get you on a good track.
>
> Peter
>
>
> Curt Shaffer <cshaffer at gmail.com> wrote:
>
>
>
> Does anyone know of good emulation type training material for CCM and the
> needed hardware to practice and learn about VoIP the Cisco way? I would like
> to get some hands on but obviously can't afford to build out my own lab ;) I
> have experience with Asterisk and I am 3Com VoIP certified, so general VoIP
> concepts are really not that important, but I would like to get up to speed
> on the Cisco specifics.
>
> Thanks
>
> Curt
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>  ________________________________
> How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call rates.
>
>
> _______________________________________________
> 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 jwydra at Burwood.com  Wed Jun  7 12:27:14 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Wed, 7 Jun 2006 11:27:14 -0500
Subject: [cisco-voip] Training material
Message-ID: <97F472A9F637E44B954D7FBABC800430B84A81@bur-msg-02.Burwood.com>

Well, at the very least, Unity does give you limited functionality with
the default license.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Wednesday, June 07, 2006 11:21 AM
To: sam at munzani.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Training material

CCM is not licensed... just a reg hack and a license key which can be
found here:

http://www.cisco.com/warp/public/788/AVVID/productkey.html

As far as the Unity license goes, yes, I did grab it from work... and
I had to hack the MAC address of the windows box to make it match
(can't do that with CCM5...)...



Jonathan

On 6/7/06, Sam Munzani <sam at munzani.com> wrote:
>
>  Jonathan,
>
>  You missed biggest item. The licenses for CCM and Unity. You might be
> fortunate enough to use licenses from your work but everybody may not
have
> same luxury :-(
>
>  Sam
>
>
>  That is not true.
>
> I built a CCM/Unity lab for under $1800..
>
> I built a server with 600GB HD space, 4GB RAM and a P4 dual... Loaded
> VMWare GSX and built a CCM cluster with Unity...
>
> Granted, I then added a bunch of phones and a 1760 gateway... (I also
> had a bunch of routers (2610 and a pile of 2500s) and a 3550 from my
> R&S days).
>
>
> Jonathan
>
> On 6/6/06, ash AD <commo_ssg_31f at yahoo.com> wrote:
>
>
>  Curt,
>
>  KnowledgeNet offer several great CCVP courses which are much more
indepth
> than your typical CBT. They bring you into simualtion and show lots of
live
> configuration. CBTNuggets is another place to look for great training
for
> those without the equipment. A CallManager lab will cost you more than
a
> small car so, a good CIPT 1 & 2 course can get you on a good track.
>
> Peter
>
>
> Curt Shaffer <cshaffer at gmail.com> wrote:
>
>
>
> Does anyone know of good emulation type training material for CCM and
the
> needed hardware to practice and learn about VoIP the Cisco way? I
would like
> to get some hands on but obviously can't afford to build out my own
lab ;) I
> have experience with Asterisk and I am 3Com VoIP certified, so general
VoIP
> concepts are really not that important, but I would like to get up to
speed
> on the Cisco specifics.
>
> Thanks
>
> Curt
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>  ________________________________
> How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call
rates.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>  _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From jonvoip at gmail.com  Wed Jun  7 12:33:39 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 7 Jun 2006 11:33:39 -0500
Subject: [cisco-voip] Training material
In-Reply-To: <97F472A9F637E44B954D7FBABC800430B84A81@bur-msg-02.Burwood.com>
References: <97F472A9F637E44B954D7FBABC800430B84A81@bur-msg-02.Burwood.com>
Message-ID: <5d093f9a0606070933m3fbbf631o79233ad96901b1db@mail.gmail.com>

And you can get a two-port demo license from Cisco for free...

On 6/7/06, Wydra, Jason <jwydra at burwood.com> wrote:
> Well, at the very least, Unity does give you limited functionality with
> the default license.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Wednesday, June 07, 2006 11:21 AM
> To: sam at munzani.com
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Training material
>
> CCM is not licensed... just a reg hack and a license key which can be
> found here:
>
> http://www.cisco.com/warp/public/788/AVVID/productkey.html
>
> As far as the Unity license goes, yes, I did grab it from work... and
> I had to hack the MAC address of the windows box to make it match
> (can't do that with CCM5...)...
>
>
>
> Jonathan
>
> On 6/7/06, Sam Munzani <sam at munzani.com> wrote:
> >
> >  Jonathan,
> >
> >  You missed biggest item. The licenses for CCM and Unity. You might be
> > fortunate enough to use licenses from your work but everybody may not
> have
> > same luxury :-(
> >
> >  Sam
> >
> >
> >  That is not true.
> >
> > I built a CCM/Unity lab for under $1800..
> >
> > I built a server with 600GB HD space, 4GB RAM and a P4 dual... Loaded
> > VMWare GSX and built a CCM cluster with Unity...
> >
> > Granted, I then added a bunch of phones and a 1760 gateway... (I also
> > had a bunch of routers (2610 and a pile of 2500s) and a 3550 from my
> > R&S days).
> >
> >
> > Jonathan
> >
> > On 6/6/06, ash AD <commo_ssg_31f at yahoo.com> wrote:
> >
> >
> >  Curt,
> >
> >  KnowledgeNet offer several great CCVP courses which are much more
> indepth
> > than your typical CBT. They bring you into simualtion and show lots of
> live
> > configuration. CBTNuggets is another place to look for great training
> for
> > those without the equipment. A CallManager lab will cost you more than
> a
> > small car so, a good CIPT 1 & 2 course can get you on a good track.
> >
> > Peter
> >
> >
> > Curt Shaffer <cshaffer at gmail.com> wrote:
> >
> >
> >
> > Does anyone know of good emulation type training material for CCM and
> the
> > needed hardware to practice and learn about VoIP the Cisco way? I
> would like
> > to get some hands on but obviously can't afford to build out my own
> lab ;) I
> > have experience with Asterisk and I am 3Com VoIP certified, so general
> VoIP
> > concepts are really not that important, but I would like to get up to
> speed
> > on the Cisco specifics.
> >
> > Thanks
> >
> > Curt
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >  ________________________________
> > How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call
> rates.
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >
> >  _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

From Matt.Slaga at us.didata.com  Wed Jun  7 13:16:03 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 7 Jun 2006 13:16:03 -0400
Subject: [cisco-voip] Training material
Message-ID: <C1FE15183DA37645BC0633BC604E44F0021C5F32@USNAEXCH.na.didata.local>

Just change your MAC address in windows to match whatever unity license
you already have. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wydra, Jason
Sent: Wednesday, June 07, 2006 12:27 PM
To: Jonathan Charles; sam at munzani.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Training material

Well, at the very least, Unity does give you limited functionality with
the default license.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Wednesday, June 07, 2006 11:21 AM
To: sam at munzani.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Training material

CCM is not licensed... just a reg hack and a license key which can be
found here:

http://www.cisco.com/warp/public/788/AVVID/productkey.html

As far as the Unity license goes, yes, I did grab it from work... and I
had to hack the MAC address of the windows box to make it match (can't
do that with CCM5...)...



Jonathan

On 6/7/06, Sam Munzani <sam at munzani.com> wrote:
>
>  Jonathan,
>
>  You missed biggest item. The licenses for CCM and Unity. You might be

> fortunate enough to use licenses from your work but everybody may not
have
> same luxury :-(
>
>  Sam
>
>
>  That is not true.
>
> I built a CCM/Unity lab for under $1800..
>
> I built a server with 600GB HD space, 4GB RAM and a P4 dual... Loaded 
> VMWare GSX and built a CCM cluster with Unity...
>
> Granted, I then added a bunch of phones and a 1760 gateway... (I also 
> had a bunch of routers (2610 and a pile of 2500s) and a 3550 from my 
> R&S days).
>
>
> Jonathan
>
> On 6/6/06, ash AD <commo_ssg_31f at yahoo.com> wrote:
>
>
>  Curt,
>
>  KnowledgeNet offer several great CCVP courses which are much more
indepth
> than your typical CBT. They bring you into simualtion and show lots of
live
> configuration. CBTNuggets is another place to look for great training
for
> those without the equipment. A CallManager lab will cost you more than
a
> small car so, a good CIPT 1 & 2 course can get you on a good track.
>
> Peter
>
>
> Curt Shaffer <cshaffer at gmail.com> wrote:
>
>
>
> Does anyone know of good emulation type training material for CCM and
the
> needed hardware to practice and learn about VoIP the Cisco way? I
would like
> to get some hands on but obviously can't afford to build out my own
lab ;) I
> have experience with Asterisk and I am 3Com VoIP certified, so general
VoIP
> concepts are really not that important, but I would like to get up to
speed
> on the Cisco specifics.
>
> Thanks
>
> Curt
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>  ________________________________
> How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call
rates.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>  _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


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

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From cshaffer at gmail.com  Wed Jun  7 13:40:18 2006
From: cshaffer at gmail.com (Curt Shaffer)
Date: Wed, 7 Jun 2006 13:40:18 -0400
Subject: [cisco-voip] Training material
In-Reply-To: <5d093f9a0606070921o2b8be9fcte9b35914fa6eb583@mail.gmail.com>
Message-ID: <014d01c68a59$72375870$8800550a@acg.arlington.local>


That is good news. I guess my next question is what I need for a lab as far
as routers/switches and gateways go. I have two Cisco 4000 routers, a couple
25xx routers, a 1900 and a 2524 switch. I can get some Cisco IP phones, I
was planning on doing that anyway to work with my current Asterisk system.
Is this hardware enough?

I am basically looking for guidelines for a basic lab to get the hands on
needed to install/support Cisco based VoIP systems. I would like it to touch
multiple areas like routers, switches, FXS/FXO ports, FoIP if possible and
any of the other features needed. I plan on taking the training and the
certification but I A. Need to get my CCNA (in progress) and B. I am more of
a hands on learner and would probably pass the exam easier if I could not
only learn but do it too.

Thanks to all of you for your help!

Curt

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan Charles
Sent: Wednesday, June 07, 2006 12:21 PM
To: sam at munzani.com
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Training material

CCM is not licensed... just a reg hack and a license key which can be
found here:

http://www.cisco.com/warp/public/788/AVVID/productkey.html

As far as the Unity license goes, yes, I did grab it from work... and
I had to hack the MAC address of the windows box to make it match
(can't do that with CCM5...)...



Jonathan

On 6/7/06, Sam Munzani <sam at munzani.com> wrote:
>
>  Jonathan,
>
>  You missed biggest item. The licenses for CCM and Unity. You might be
> fortunate enough to use licenses from your work but everybody may not have
> same luxury :-(
>
>  Sam
>
>
>  That is not true.
>
> I built a CCM/Unity lab for under $1800..
>
> I built a server with 600GB HD space, 4GB RAM and a P4 dual... Loaded
> VMWare GSX and built a CCM cluster with Unity...
>
> Granted, I then added a bunch of phones and a 1760 gateway... (I also
> had a bunch of routers (2610 and a pile of 2500s) and a 3550 from my
> R&S days).
>
>
> Jonathan
>
> On 6/6/06, ash AD <commo_ssg_31f at yahoo.com> wrote:
>
>
>  Curt,
>
>  KnowledgeNet offer several great CCVP courses which are much more indepth
> than your typical CBT. They bring you into simualtion and show lots of
live
> configuration. CBTNuggets is another place to look for great training for
> those without the equipment. A CallManager lab will cost you more than a
> small car so, a good CIPT 1 & 2 course can get you on a good track.
>
> Peter
>
>
> Curt Shaffer <cshaffer at gmail.com> wrote:
>
>
>
> Does anyone know of good emulation type training material for CCM and the
> needed hardware to practice and learn about VoIP the Cisco way? I would
like
> to get some hands on but obviously can't afford to build out my own lab ;)
I
> have experience with Asterisk and I am 3Com VoIP certified, so general
VoIP
> concepts are really not that important, but I would like to get up to
speed
> on the Cisco specifics.
>
> Thanks
>
> Curt
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>  ________________________________
> How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call
rates.
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>  _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From Joe_Bell at adp.com  Wed Jun  7 13:49:00 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Wed, 7 Jun 2006 12:49:00 -0500
Subject: [cisco-voip] General QOS question
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BB79BCE@EXCHANGEDS01.ds.ad.adp.com>

Andy is right, you don't need desireable.  An oversight on my part.
His config below is what we use for our newer switches, while the older,
non-IPT switches still need to be configured as a trunk because their
stupid.  There were a number of questions that came up from this string,
so I'll try to answer a couple of them.  

First, there was a question as to what switch port commands
automatically form a trunk and which doesn't.  Here is a list:

---

If you enter access mode, the interface goes into permanent nontrunking
mode and negotiates to convert the link into a nontrunk link even if the
neighboring interface does not agree to the change. 

If you enter trunk mode, the interface goes into permanent trunking mode
and negotiates to convert the link into a trunk link even if the
neighboring interface does not agree to the change. 

If you enter dynamic auto mode, the interface converts the link to a
trunk link if the neighboring interface is set to trunk or desirable
mode. <--  So two autos do not form a trunk.

If you enter dynamic desirable mode, the interface becomes a trunk
interface if the neighboring interface is set to trunk, desirable, or
auto mode. 

---

Second, some info on the quasi-nature of the trunk in a voice enabled
switch.  The big reason why you do not want this to be a real trunk, is
because you don't want to run spanning tree for every VLAN, which is
what trunks do.  The phone really is only concerned about 2 VLAN's
(voice and data) and one of which is untagged (from the phones point of
view) with 802.1q frames.  Keeping that processing off the phone is
apparently a big hit.  Second, the CAM table on a phone has room for 32
addresses, not nearly enough for a true trunk.  If it acted like a real
trunk, it would be deleting and adding addresses all the time.  The main
reason for a switch in the phone is tagging = QoS to protect voice from
the PC behind it.  To that end, it's a very limited switch set and so
the newer switches do not form a true trunk.  If it were a true trunk,
you could not run a sniffer on your PC behind the phone, and see traffic
to and from the phone as they are on two separate VLAN's.

There is a good discussion on the capabilities of the switch in the
phone in the Cisco IPT Troubleshooting handbook.

Hope this helps,

Joe




-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Clements, Andy
(E.ON IS UK)
Sent: Wednesday, June 07, 2006 8:31 AM
To: Voll, Scott; Chris Ellington; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] General QOS question

Hi,

I do not think you need the dynamic desirable as I have configured edge
switch interfaces as switch mode access with the switch voice vlan
command and that still sends the voice vlan in a dot1q trunk and the
access vlan as the native untagged vlan.

[Example]
 switchport access vlan 505
 switchport mode access
 switchport voice vlan 405

I believe this "newer" configuration for voice is to allow the use of
802.1x as this wont work on a "trunked interface". Although I haven't
had experience of this.

By the way the configuration above is working for both Cisco and Nortel
IPT phones.

Regards Andy

-----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, June 07, 2006 3:33 PM
To: Chris Ellington; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] General QOS question

Chris--

Quasi -- to some degree; having a likeness to something; resembling;
having some resemblance; etc.

It's not really a trunking mode.

The desirable automagically goes into the config when you have both
switchport access and voice on the same interface.  You don't have to
manually set it.

QoS at layer two is basically done by trusting the CoS, ToS, DSCP and
passing it along. IE> 

interface FastEthernet0/6
 switchport access vlan 2
 switchport mode dynamic desirable
 switchport voice vlan 110
 no ip address
 srr-queue bandwidth share 10 10 60 20
 srr-queue bandwidth shape  10  0  0  0
 priority-queue out
 mls qos trust device cisco-phone
 mls qos trust cos
 no mdix auto
 auto qos voip cisco-phone
 spanning-tree portfast

Hope that helps.

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: Wednesday, June 07, 2006 4:11 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] General QOS question

Do you have links on the quasi trunking mode with the phones?  I've done
a
quick search and don't find it on the Cisco site (now that's a surprise
:D)

My understanding of the desirable command is that is will allow a trunk
to
form if the other side requests it (say two switches are connected).  I
have
had trouble in the past with both sides wanting a desirable mode in that
neither will 'request' the trunk so it stays in an access mode (that's
been
a while ago though, so it's probably fixed).

Also, do you really save much in the way of overhead/processing by using
this quasi trunk?  Let's say I have a Sup1A in a 6500 running 384 ports,
all
trunked to various voice vlans - does the extra processing really hurt?

I'm not trying to be difficult, rather just understand the options.  I
also
have a team of people to convince; my next move is to go to the SRND and
take the recommendation from there since TAC seems to fall back to that
when
I have troubles.

Thanks!

chris

-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com

> Thanks for all of the replies. Joe has the answer to my question. I
> understand you can still do ToS without trunking but in order to have
> CoS then you need 802.1Q which has the 802.1p header. The 802.1p
header
> is where you find CoS. Without trunking, there is no CoS period. So
> basically, the newer switches/IOS look for it anyway even though the
> switch port is not set for trunking. The IP phone still does what we
> consider standard 802.1q trunking. The switch just sees the CoS
> regardless.
> 
>  
> 
> I must say though that I have never seen the dynamic desirable command
> used in implementations I've seen. Maybe I just don't remember.
> 
>  
> 
> Thanks again!
> 
>  
> 
> Jason
> 
>  
> 
> ________________________________
> 
> From: Bell, Joe [mailto:Joe_Bell at adp.com]
> Sent: Tuesday, June 06, 2006 7:35 PM
> To: ash AD; Wydra, Jason; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] General QOS question
> 
>  
> 
> Actually, layer 2 QoS (a.k.a CoS settings) are in the 802.1q header.
> Layer 3, (DSCP or PHB bits), are the 5 in the IP Precedence field.  If
> your not using 802.1q, your not using layer 2 CoS.
> 
>  
> 
> The reason you see switch configs that are not set for full trunking
and
> instead have command sets like:
> 
>  
> 
> Interface fa0/1
> 
>  switchport  dynamic desirable  <-- negotiates a quasi-trunk
> 
>  switchport  voice vlan 100
> 
>  switchport  access vlan 200
> 
>  spanning-tree portfast
> 
>  
> 
>  
> 
> Is because the newer voice enabled switches are smart enough to
> understand the voice vlan information and respect the tags coming down
> from the phone, and there are still 802.1q tags, even though you have
> not configured a true trunk.  As I said, the phone does insert an
802.1q
> header and tags appropriately, but the port is not in a true trunking
> mode.  The reason for this, is the new config keeps a lot of trunking
> management protocols off the port and frees it up for voip traffic =
> efficiency.
> 
>  
> 
> Older switches were configured as full trunks and you might still see
> that out there.
> 
>  
> 
> interface FastEthernet0/1   <-- true trunking config
> 
>  switchport trunk encapsulation dot1q
> 
>  switchport trunk native vlan 100  <-- data and untagged vlan
> 
>  switchport mode trunk
> 
>  switchport voice vlan 200  <-- tagged 802.1q header with the phone
> inserting layer 2 CoS and layer 3 DSCP tags
> 
>  spanning-tree portfast
> 
>  
> 
> These configs accomplish the same thing, but the "newer" switches
config
> is much more efficient.
> 
>  
> 
> Joe
> 
> 


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From cidar at acelerate.com  Wed Jun  7 15:37:01 2006
From: cidar at acelerate.com (Cidar Ramirez)
Date: Wed, 7 Jun 2006 13:37:01 -0600
Subject: [cisco-voip] voip billing
In-Reply-To: <mailman.4689.1149702560.1478.cisco-voip@puck.nether.net>
Message-ID: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAATb16yFsuxEqckYCsMTcHdMKAAAAQAAAATTpIbDeX6kCYeCA3K8IkSAEAAAAA@acelerate.com>

Anybody knows about a open source billing software for cisco as5300 cdrs ?

Thanks

Cid



From Ryan.OConnell at nexinnovations.com  Wed Jun  7 15:50:28 2006
From: Ryan.OConnell at nexinnovations.com (OConnell, Ryan)
Date: Wed, 7 Jun 2006 15:50:28 -0400
Subject: [cisco-voip] Quick question regarding Caller ID
Message-ID: <D9C2C0CEF6CC4248BF8F275D03E57F0C0D0CF602@MISEXCVS01.corp.nexinnovations.com>

Hello all,

 

Does anyone know if there are any plans to support Caller ID on FXO
ports using MGCP any time soon? Or is h.323 the only way

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

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

 

 

 

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From Matt.Slaga at us.didata.com  Wed Jun  7 16:02:49 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 7 Jun 2006 16:02:49 -0400
Subject: [cisco-voip] how to load dsp firmware
Message-ID: <C1FE15183DA37645BC0633BC604E44F0021C637C@USNAEXCH.na.didata.local>

DSPs are not user upgradeable, they must be replaced.  Open a TAC case
referencing your bug ID.and see if they can replace them.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of McLean Pickett
Sent: Wednesday, June 07, 2006 11:25 AM
To: Ruttman, Peter G.; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] how to load dsp firmware


Pete -
 
I was given explicit direction not to remove the file from firmware as
the DSP's would revert to the old firmware. 
 
McLean

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter
G.
Sent: Wednesday, June 07, 2006 10:23 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to load dsp firmware


TAC says we may be experiencing this bug:
 

CSCsd54344: DSP: Persistent echo heard on c5510 DSPs with 4.4.13 DSPware
or higher

Symptom:

Under certain circumstances, continual echo may be heard. The echo is
perisitent and the DSP echo canceller does not converge.

 

Conditions:

DSPware 4.4.13 or higher on C5510 DSPs are required to experience this
problem.

 

Workaround:

Downgrading DSPware to 4.4.12 or earlier resolves the echo issue.

 

TAC send is firmware and these instructions:

INSTRUCTIONS:

1. Put the file on the flash.

2. Reload the router.

3. Verify new DSP version ' sh voice dsp' (should be 4.4.18)

3. IMPORTANT: Remove file from flash after verification

 

However, I did the show voice dsp and saw no change in the dsp version
from before getting the new dsp firmware.  Can someone tell me if this
is the correct way to load firmware on an NM-HDV2 on a 3745 voice
gateway?

Thanks,

Pete Ruttman

CCNA  CCVP

Senior Network Engineer

Foley & Lardner LLP

 

The preceding email message may be confidential or protected by the
attorney-client privilege. It is not intended for transmission to, or
receipt by, any unauthorized persons. If you have received this message
in error, please (i) do not read it, (ii) reply to the sender that you
received the message in error, and (iii) erase or destroy the message.
Legal advice contained in the preceding message is solely for the
benefit of the Foley & Lardner LLP client(s) represented by the Firm in
the particular matter that is the subject of this message, and may not
be relied upon by any other party. 


Internal Revenue Service regulations require that certain types of
written advice include a disclaimer. To the extent the preceding message
contains advice relating to a Federal tax issue, unless expressly stated
otherwise the advice is not intended or written to be used, and it
cannot be used by the recipient or any other taxpayer, for the purpose
of avoiding Federal tax penalties, and was not written to support the
promotion or marketing of any transaction or matter discussed herein.




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From erickbe at yahoo.com  Wed Jun  7 20:40:44 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 7 Jun 2006 17:40:44 -0700 (PDT)
Subject: [cisco-voip] how to load dsp firmware
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0021C637C@USNAEXCH.na.didata.local>
Message-ID: <20060608004044.21658.qmail@web37203.mail.mud.yahoo.com>

On some modules and DSP models, and IOS versions the DSPWare can be changed by putting special files on the router flash the router looks for but they need to stay on flash to be used. I have personally only seen this sort of fix work once in my experiences where this was needed. The file does need to stay on the flash to be used from my experiences but different hardware/PVDMs may be different.

In the instructions below and in bug id, it states to downgrade to 4.4.12 or earlier , but in the instructions for the verify step it states it should be 4.4.18 version. That doesn't match up.

Did the file get transferred to the router right - it's not corrupt, etc. I would let the engineer know that the special file did not work for you. 

What is your actual echo issue you are experiencing? What version IOS? What type of voice port configuration (echo cancelling commands, input gain, output atten, etc)?  Is there a certain call-flow the echo occurs? Who hears the echo (calling party, called party, both?). Only external calls or do you have it on internal calls also (IP to IP Phone)? Echo can be a very complex thing to fix at times.

----- Original Message ----
From: Matt Slaga (US) <Matt.Slaga at us.didata.com>
To: McLean Pickett <McLean.Pickett at ptgcorp.com>; "Ruttman, Peter G." <PRuttman at foley.com>; cisco-voip at puck.nether.net
Sent: Wednesday, June 7, 2006 3:02:49 PM
Subject: Re: [cisco-voip] how to load dsp firmware

Message       DSPs are not user upgradeable, they must be   replaced.  Open a TAC case referencing your bug ID.and see if they can   replace them.

      From: cisco-voip-bounces at puck.nether.net   [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of McLean   Pickett
Sent: Wednesday, June 07, 2006 11:25 AM
To: Ruttman,   Peter G.; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] how to   load dsp firmware


  
  Pete -
   
  I was given explicit direction not to remove the file from firmware   as the DSP's would revert to the old firmware. 
   
  McLean

      From: cisco-voip-bounces at puck.nether.net   [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ruttman, Peter   G.
Sent: Wednesday, June 07, 2006 10:23 AM
To:   cisco-voip at puck.nether.net
Subject: [cisco-voip] how to load dsp   firmware


  
  TAC says we may be experiencing this   bug:
   
    CSCsd54344: DSP: Persistent echo heard on c5510 DSPs with 4.4.13 DSPware or   higher
  Symptom:
  Under certain circumstances, continual echo may be heard. The echo is   perisitent and the DSP echo canceller does not converge.
   
  Conditions:
  DSPware 4.4.13 or higher on C5510 DSPs are required to experience this   problem.
   
  Workaround:
  Downgrading DSPware to 4.4.12 or earlier resolves the echo issue.
   
  TAC send is firmware and   these instructions:
  INSTRUCTIONS:
  1. Put the file on the flash.
  2. Reload the router.
  3. Verify new DSP version ' sh voice dsp' (should be   4.4.18)
  3. IMPORTANT: Remove file from flash after   verification
   
  However, I did the show voice   dsp and saw no change in the dsp version from before getting the new dsp   firmware.  Can someone tell me if this is the correct way to load firmware   on an NM-HDV2 on a 3745 voice gateway?
  Thanks,
  Pete   Ruttman

    CCNA    CCVP
  Senior Network   Engineer
  Foley & Lardner LLP

   

The preceding email message may be confidential or   protected by the attorney-client privilege. It is not intended for transmission   to, or receipt by, any unauthorized persons. If you have received this message   in error, please (i) do not read it, (ii) reply to the sender that you received   the message in error, and (iii) erase or destroy the message. Legal advice   contained in the preceding message is solely for the benefit of the Foley &   Lardner LLP client(s) represented by the Firm in the particular matter that is   the subject of this message, and may not be relied upon by any other party.   


Internal Revenue Service regulations require that certain types of   written advice include a disclaimer. To the extent the preceding message   contains advice relating to a Federal tax issue, unless expressly stated   otherwise the advice is not intended or written to be used, and it cannot be   used by the recipient or any other taxpayer, for the purpose of avoiding Federal   tax penalties, and was not written to support the promotion or marketing of any   transaction or matter discussed herein.



  
  Disclaimer:
  
  This e-mail communication and any attachments may contain
  confidential and privileged information and is for use by the
  designated addressee(s) named above only.  If you are not the
  intended addressee, you are hereby notified that you have received
  this communication in error and that any use or reproduction of
  this email or its contents is strictly prohibited and may be
  unlawful.  If you have received this communication in error, please
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From erickbe at yahoo.com  Wed Jun  7 20:43:30 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 7 Jun 2006 17:43:30 -0700 (PDT)
Subject: [cisco-voip] CCM 3.3(3) and RTMT
In-Reply-To: <20060607122118.37140.qmail@web37210.mail.mud.yahoo.com>
Message-ID: <20060608004330.73601.qmail@web37205.mail.mud.yahoo.com>

Does anyone know if there is a hidden knob somewhere in 3.3(3) where you can set the RIS Data Collection parameter  if it even exists in 3.3 that is.  I know it is 4.x under RIS service parameters. 

----- Original Message ----
From: Erick Bergquist <erickbe at yahoo.com>
To: ciscovoip <cisco-voip at puck.nether.net>
Sent: Wednesday, June 7, 2006 7:21:18 AM
Subject: [cisco-voip] CCM 3.3(3) and RTMT

With CCM 3.3(3) with 1 publisher, and 2 subscriber servers the RTMT Callmanager Cluster info properties, the info for the 2nd subscriber does not report back in RTMT tool on the cluster info window. It saids "Error" next to most values but the first 2 fields in list show good data (registered phones).
 
The info from the other servers displays fine. If you launch RTMT from the problem subscriber and click the info button the RTMT server version is empty. 
 
So like it can't access data on this server, but this happens when you RTMT locally on the subscriber or remotely with IE over web. You can view all performance counters fine in RTMT on this server, it's just the cluster info window and RTMT version info, and processor summary window. 

Windows performance monitor appears fine, can pull up counters fine on all servers.

I have checked to make sure all the services were running, have restarted RIS DC service, etc and can't seem to find anything different between them yet. 
 
CCM 3.3(3)sr4a 
OS 2000.4-2sr7  
RTMT client version 3.3(0.21) 
RTMT server version is 3.3.2.0 
 
This has been an issue for awhile, even before OS updates from 2000.2-7 to 2000.4-2. Not sure if RTMT ever worked fine on this one server.

I've done several bug searches and can't find a good hit yet. I do realize 3.3(3) and RTMT is from the early days of RTMT so expect there may be some issues since RTMT in 4.x is completely different.

Anyone have any ideas?

Thanks, Erick

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From erickbe at yahoo.com  Wed Jun  7 21:03:39 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 7 Jun 2006 18:03:39 -0700 (PDT)
Subject: [cisco-voip] how to load dsp firmware
In-Reply-To: <7E3064B7EFC16C41B185C41D141008B701A6DC21@FOLEYEX97.foleylaw.com>
Message-ID: <20060608010339.61925.qmail@web37212.mail.mud.yahoo.com>

Here's another idea.. if you recently upgraded (the bug id has 12.4(7) as first found version) and then started having the issue, maybe you can downgrade the IOS back to where you were until a fixed IOS is available with that version DSPware. 

12.4(8) is out , but I do not know what DSPware version it has. Does anyone know if there is a IOS / DSPWare version matrix anywhere?  I have had good, bad, and very bad results with 12.4 versions so far so am cautious when using them and keep my fingers and toes crossed. 

----- Original Message ----
From: "Ruttman, Peter G." <PRuttman at foley.com>
To: cisco-voip at puck.nether.net
Sent: Wednesday, June 7, 2006 9:22:55 AM
Subject: [cisco-voip] how to load dsp firmware

Message      TAC says we may be experiencing this  bug:
  
  CSCsd54344: DSP: Persistent echo heard on c5510 DSPs with 4.4.13 DSPware or  higher
 Symptom:
 Under certain circumstances, continual echo may be heard. The echo is  perisitent and the DSP echo canceller does not converge.
  
 Conditions:
 DSPware 4.4.13 or higher on C5510 DSPs are required to experience this  problem.
  
 Workaround:
 Downgrading DSPware to 4.4.12 or earlier resolves the echo issue.
  
 TAC send is firmware and  these instructions:
 INSTRUCTIONS:
 1. Put the file on the flash.
 2. Reload the router.
 3. Verify new DSP version ' sh voice dsp' (should be  4.4.18)
 3. IMPORTANT: Remove file from flash after  verification
  
 However, I did the show voice  dsp and saw no change in the dsp version from before getting the new dsp  firmware.  Can someone tell me if this is the correct way to load firmware  on an NM-HDV2 on a 3745 voice gateway?
 Thanks,
 Pete  Ruttman

  CCNA   CCVP
 Senior Network  Engineer
 Foley & Lardner LLP

  

The preceding email message may be confidential or protected by the attorney-client privilege. It is not intended for transmission to, or receipt by, any unauthorized persons. If you have received this message in error, please (i) do not read it, (ii) reply to the sender that you received the message in error, and (iii) erase or destroy the message. Legal advice contained in the preceding message is solely for the benefit of the Foley & Lardner LLP client(s) represented by the Firm in the particular matter that is the subject of this message, and may not be relied upon by any other party. 

 
Internal Revenue Service regulations require that certain types of written advice include a disclaimer. To the extent the preceding message contains advice relating to a Federal tax issue, unless expressly stated otherwise the advice is not intended or written to be used, and it cannot be used by the recipient or any other taxpayer, for the purpose of avoiding Federal tax penalties, and was not written to support the promotion or marketing of any transaction or matter discussed herein.
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From jose.mujica at netxar.com  Wed Jun  7 22:22:27 2006
From: jose.mujica at netxar.com (Jose Mujica)
Date: Wed, 7 Jun 2006 22:22:27 -0400
Subject: [cisco-voip] Attendant Console / CallManager 4.1(3) with Line Group
	Problems
Message-ID: <20060608021941.M23570@netxar.com>

Hi. We have a client that has 2 operators and they are inserted in a line
group with a hunt pilot [not an AC Pilot] in order to reach them TopDown.. We
installed the Attendant Console version 1.4(1 ES11) in the first operator
because she would like to see line states of other phones and everytime a
calls come in it shows up in the Console fine but when the CCM takes the call
back in order to send it to the 2nd operator because of RNA I received a
message on the Attendant Console saying:

Failed to make a new call.

And we didn't make a new call at all... Any ideas?

CallManager version is 4.1(3)sr3a. I already restarted the TCD service in both
callmanagers but the problem continues.. I will probably open a TAC case but
was wondering if someone got the same problem before.

Thanks in advanced,
-Jose

From jgrace at digitelusa.net  Wed Jun  7 22:38:52 2006
From: jgrace at digitelusa.net (James Grace)
Date: Wed, 7 Jun 2006 22:38:52 -0400
Subject: [cisco-voip] how to increase the rings to an agent
Message-ID: <200606072250718.SM01572@3729434770>

We are running crs 4.0.3  and need to know how to increase the about of ring
to agent before it rolls to the next

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From bababooey at cox.net  Wed Jun  7 23:57:33 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Wed, 7 Jun 2006 23:57:33 -0400
Subject: [cisco-voip] PersonalFastDials service
References: <mailman.4803.1149727251.1478.cisco-voip@puck.nether.net>
Message-ID: <000301c68aaf$acf84720$0f0110ac@corporation2>

How can I install the Personal Address Book, which installs two ip srevices, 
address book and PersonalFastDials service.

I need to activate the myfastdials option to allow user access the the 99 
abbr dials.

I cannot find how to install it?

Thanks
Bob



From lelio at uoguelph.ca  Thu Jun  8 00:03:10 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 8 Jun 2006 00:03:10 -0400
Subject: [cisco-voip] PersonalFastDials service
References: <mailman.4803.1149727251.1478.cisco-voip@puck.nether.net>
	<000301c68aaf$acf84720$0f0110ac@corporation2>
Message-ID: <00e601c68ab0$750225b0$c801a8c0@dellectable>

The AbbrDial softkey/feature allows a user to access one of the 99 "Speed 
Dials" that they program through the CCMUser pages. I had hoped that the 
FastDial service would allow the user to populate these over the phone, but 
alas, no luck. The SpeedDials and the FastDials are mutually exclusive. As 
far as I can tell from my testing.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network 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: "Bob A. Bowie" <bababooey at cox.net>
To: <cisco-voip at puck.nether.net>
Sent: Wednesday, June 07, 2006 11:57 PM
Subject: [cisco-voip] PersonalFastDials service


> How can I install the Personal Address Book, which installs two ip 
> srevices,
> address book and PersonalFastDials service.
>
> I need to activate the myfastdials option to allow user access the the 99
> abbr dials.
>
> I cannot find how to install it?
>
> Thanks
> Bob
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From jgrace at digitelusa.net  Thu Jun  8 00:10:44 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu, 8 Jun 2006 00:10:44 -0400
Subject: [cisco-voip] remove record files from crs 4.0.3
Message-ID: <200606080022828.SM01572@3729434770>

We are recording all answered call.  Is there any way to move the file from
the crs server and convert them in bulk.

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From smgustafson at gmail.com  Thu Jun  8 00:50:58 2006
From: smgustafson at gmail.com (Steve G)
Date: Wed, 7 Jun 2006 21:50:58 -0700
Subject: [cisco-voip] inbound call tracking
Message-ID: <8a1f34b50606072150x4a6bfee7i3c6976e263896814@mail.gmail.com>

I have a client with a unique request to track the number of calls each or
their salesmen receive in a day.  So, a call would come into the store for
"sales", the receptionist would transfer to the "sales" huntpilot which
would broadcast ring a number of phones.  The salesman would pick-up a
ringing phone (not necessarily one that is assigned to them) and enter his
code and proceed with helping the customer.

This would kind-of be like a client matter code situation, but it would be
for incoming as opposed to outgoing route patterns.

If anyone has any creative ways of doing this in an automated fashion, I
would like to hear from you.

Thanks,
Steve
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From kthorngr at cisco.com  Thu Jun  8 06:45:30 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu, 8 Jun 2006 06:45:30 -0400
Subject: [cisco-voip] inbound call tracking
In-Reply-To: <8a1f34b50606072150x4a6bfee7i3c6976e263896814@mail.gmail.com>
References: <8a1f34b50606072150x4a6bfee7i3c6976e263896814@mail.gmail.com>
Message-ID: <28f44d2ddd6310df809b995518ea4ebe@cisco.com>

Call Pickup is the only way to answer a call ringing on another phone.   
You might be able to utilize CDR records to accomplish the accounting  
you want to do.  For example from the 4.1(3) docs:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_administration_guide_chapter09186a00803edafe.html#wp1039295

Kevin
On Jun 8, 2006, at 12:50 AM, Steve G wrote:

> I have a client with a unique request to track the number of calls  
> each or their salesmen receive in a day.? So, a call would come into  
> the store for "sales", the receptionist would transfer to the "sales"  
> huntpilot which would broadcast ring a number of phones.? The salesman  
> would pick-up a ringing phone (not necessarily one that is assigned?to  
> them)?and enter his code and proceed with helping the customer.
> ?
> This would kind-of be like a client matter code situation, but it  
> would be for incoming as opposed to outgoing route patterns.
> ?
> If anyone has any creative ways of doing this in an automated fashion,  
> I would like to hear from you.
> ?
> Thanks,
> Steve_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From smgustafson at gmail.com  Thu Jun  8 09:27:06 2006
From: smgustafson at gmail.com (Steve G)
Date: Thu, 8 Jun 2006 06:27:06 -0700
Subject: [cisco-voip] inbound call tracking
In-Reply-To: <28f44d2ddd6310df809b995518ea4ebe@cisco.com>
References: <8a1f34b50606072150x4a6bfee7i3c6976e263896814@mail.gmail.com>
	<28f44d2ddd6310df809b995518ea4ebe@cisco.com>
Message-ID: <8a1f34b50606080627s15945b04gae364001db2d36df@mail.gmail.com>

Thanks Kevin, I'm familiar with Call Pickup.  This is a show room scenario
with a bunch of phones that are not tied to any individual sales person.  So
they ring in a broadcast linegroup.  We can't tell who is picking up the
call on any given phone in the showroom.  I am going to assume that this is
not going to be possible.  Just wanted to inquire to others.

Thanks,
Steve


On 6/8/06, Kevin Thorngren <kthorngr at cisco.com> wrote:
>
> Call Pickup is the only way to answer a call ringing on another phone.
> You might be able to utilize CDR records to accomplish the accounting
> you want to do.  For example from the 4.1(3) docs:
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/
> products_administration_guide_chapter09186a00803edafe.html#wp1039295
>
> Kevin
> On Jun 8, 2006, at 12:50 AM, Steve G wrote:
>
> > I have a client with a unique request to track the number of calls
> > each or their salesmen receive in a day. So, a call would come into
> > the store for "sales", the receptionist would transfer to the "sales"
> > huntpilot which would broadcast ring a number of phones. The salesman
> > would pick-up a ringing phone (not necessarily one that is assignedto
> > them)and enter his code and proceed with helping the customer.
> >
> > This would kind-of be like a client matter code situation, but it
> > would be for incoming as opposed to outgoing route patterns.
> >
> > If anyone has any creative ways of doing this in an automated fashion,
> > I would like to hear from you.
> >
> > Thanks,
> > Steve_______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From ppauly at gmail.com  Thu Jun  8 11:10:51 2006
From: ppauly at gmail.com (Peter Pauly)
Date: Thu, 08 Jun 2006 11:10:51 -0400
Subject: [cisco-voip] Unity not using newly added VM ports
Message-ID: <44883DFB.3080403@gmail.com>

I'm running Unity 4.0(3) SR 1 and just expanded the number of voice mail
ports from 32 to 96. The licensing application shows that
LicVoicePortsMax is 96. In the Manage Integrations app, I added the
additional ports and checked all the boxes. I also added ports in Call
Manager. Unity has been rebooted since the ports were added, but it
still won't use the additional ports. I max out the 32 ports a few times
during the day and get a busy signal. Status Monitor shows only the old
ports being used. The new ports are displayed but are idle.

The only odd thing about my installation is that I have two
integrations, ports 1-8 are on Centrex. Ports 9 and up are on Call Manager.

Is there some trick to get Unity to start using the new ports? Did I
leave something out?

From rratliff at cisco.com  Thu Jun  8 11:17:44 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 8 Jun 2006 11:17:44 -0400
Subject: [cisco-voip] Unity not using newly added VM ports
In-Reply-To: <44883DFB.3080403@gmail.com>
References: <44883DFB.3080403@gmail.com>
Message-ID: <D604CA0B-A851-4359-9563-DB63527304DB@cisco.com>

Sounds like your problem is not with Unity but with CallManager.   If  
you are on CM 3.3 or earlier make sure port 32 is CFNA, CFB to port  
33 and so on.  If you are on CM 4.x make sure your line group  
includes the new vm ports.

Also make sure ports 33+ show registered in CallManager.
-Ryan

On Jun 8, 2006, at 11:10 AM, Peter Pauly wrote:

I'm running Unity 4.0(3) SR 1 and just expanded the number of voice mail
ports from 32 to 96. The licensing application shows that
LicVoicePortsMax is 96. In the Manage Integrations app, I added the
additional ports and checked all the boxes. I also added ports in Call
Manager. Unity has been rebooted since the ports were added, but it
still won't use the additional ports. I max out the 32 ports a few times
during the day and get a busy signal. Status Monitor shows only the old
ports being used. The new ports are displayed but are idle.

The only odd thing about my installation is that I have two
integrations, ports 1-8 are on Centrex. Ports 9 and up are on Call  
Manager.

Is there some trick to get Unity to start using the new ports? Did I
leave something out?
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From ppauly at gmail.com  Thu Jun  8 11:31:16 2006
From: ppauly at gmail.com (Peter Pauly)
Date: Thu, 08 Jun 2006 11:31:16 -0400
Subject: [cisco-voip] Unity not using newly added VM ports
In-Reply-To: <D604CA0B-A851-4359-9563-DB63527304DB@cisco.com>
References: <44883DFB.3080403@gmail.com>
	<D604CA0B-A851-4359-9563-DB63527304DB@cisco.com>
Message-ID: <448842C4.7020808@gmail.com>

Ryan Ratliff wrote:
If you are on CM 4.x make sure your line group includes the
> new vm ports.

Thanks - the line group was the piece I was missing.

From jonvoip at gmail.com  Thu Jun  8 12:52:01 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 8 Jun 2006 11:52:01 -0500
Subject: [cisco-voip] CM 5.0
In-Reply-To: <20060608153002.490.qmail@web52309.mail.yahoo.com>
References: <5d093f9a0606080816w5eabfeedr4b9583c159ed8c2@mail.gmail.com>
	<20060608153002.490.qmail@web52309.mail.yahoo.com>
Message-ID: <5d093f9a0606080952s43dba29ds7f1082ff67498f45@mail.gmail.com>

Install Windows 2003 and install VMWare and follow the doc...



Jonathan

On 6/8/06, Goran Selthofer <selt9 at yahoo.com> wrote:
> jonathan, i have tried it on IMB eSeries with 16GB RAM and 160GB HDD, both
> IDE and SCSI... the same error... "not supported HW..."
> my CCM is version 5.0.2.9001-59
> any hope?
>
>
> Jonathan Charles <jonvoip at gmail.com> wrote:
>
>  I have followed that procedure at least 9 times and all have been
> succesful (failed for other reasons (mainly a faulty DMA backup)).
>
> You need to give it a minimum of 1GB of RAM (I did 2GB) and a 72GB
> drive... use the LSI SCSI drive.
>
> It has worked every time for 5.0(2)...
>
>
>
> Jonathan
>
> On 6/8/06, Goran Selthofer wrote:
> > please see the last message on that forum...
> > the procedure is not correct for ccm 5.0.2 ... because I have tried
> several
> > times and failed...
> >
> >
> > Jonathan Charles wrote:
> >
> > Apologies, this was posted, but I thought to the group, apparently it
> > was sent only to me (apologies in advance if the intent was to not
> > share it)...
> >
> >
> http://www.vmware.com/community/thread.jspa?messageID=399465??
> >
> >
> > BTW, I didn't have to do the boot off of red hat cd thingy... However,
> > it definitely took a long time for core services to start after bootup
> > (about 10 mins).
> >
> > Is there any way to get a demo license for lab use? Just five phones
> > or something...
> >
> >
> >
> > Jonathan
> >
> > On 6/7/06, Goran Selthofer wrote:
> > > that is right!
> > > vmware has this fix mac address format and you can not change it.
> > > you have to licence it with demo licence made just for vmaware
> > > installations...
> > > regarding the procedure of mac changing, you should change it by editing
> > vmx
> > > file...
> > >
> > > stil, noone answered to my question about exact procedure for vmware
> > > installation of CM 5.0.2 version?
> > > thanks
> > >
> > >
> > > Jonathan Charles wrote:
> > >
> > > OK, so I can get CCM5 running on VMWare, but how do I change the MAC
> > > address of the VM so I can license this thing?
> > >
> > > I have a license file but I am apparently very limited on what MAC
> > > addresses I can specify for VMWare.
> > >
> > > Apparently, VMWare only lets you assign MACs of this format:
> > >
> > > 00:50:56:XX:YY:ZZ
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On 5/30/06, Frazee, Timothy wrote:
> > > >
> > > >
> > > >
> > > >
> > > > I am running CCM 5.0.1 on GSX 3.2.0.
> > > >
> > > >
> > > >
> > > > The problem I have is I have tried to duplicate the install to write a
> > > > procedure, and it completes the install with no errors. When I try to
> > boot
> > > > the newly installed system, it freaks out at GRUB and gives me an
> error.
> > I
> > > > have looked up the error code and it describes a problem loading the
> > > > partition with GRUB. There seems to be no "repair" type of option.
> > > >
> > > >
> > > >
> > > > I have successfully installed 5.0.1 on vmware Workstation
> > > 5.5.latestversion
> > > > many times. Unfortunately the install fails when I try to make a new
> > > legacy
> > > > vm that I can move to GSX.
> > > >
> > > >
> > > >
> > > > I have a feeling that when this is released to "public" it will be so
> > > > cranked down, it will be hard to install in vmware without some
> > tweaking.
> > > >
> > > >
> > > >
> > > > I have not tried to upgrade a 5.0.1 running in vmware with the
> > > downloadable
> > > > files from cco.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Timothy Frazee
> > > >
> > > > Network Solutions
> > > >
> > > > 314.249.8697 cellular phone
> > > >
> > > > ________________________________
> > > >
> > > >
> > > > From: Goran Selthofer [mailto:selt9 at yahoo.com]
> > > > Sent: Tuesday, May 30, 2006 7:32 AM
> > > > To: Frazee, Timothy; Voll, Scott; ciscovoip
> > > >
> > > > Subject: Re: [cisco-voip] CM 5.0
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Timothy,
> > > >
> > > > please give us the exact CCM version you have?
> > > > I have tried to run 5.0.2 on VMWARE GSX and Workstation and I have
> > failed
> > > > both times with the same code:
> > > > something like "Not supported hardware..."
> > > >
> > > > also, some procedure will be highly appreciated :)
> > > >
> > > > Thanks,
> > > > Goran
> > > >
> > > > "Frazee, Timothy" wrote:
> > > >
> > > >
> > > > I have it running on vmware GSX server 3.2.0. Works well. I troubles
> > > getting
> > > > the installer to not freak out. You have to be real specific and
> emulate
> > a
> > > > red hat enterprise server 3. When I selected RHES 4, the installer
> would
> > > > hang.
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > 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
> > > > Voll, Scott
> > > > Sent: Friday, May 26, 2006 9:47 AM
> > > > To: ciscovoip
> > > > Subject: [cisco-voip] CM 5.0
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Has anyone had the opportunity to get CM 5.x running on non-cisco
> > servers?
> > > > I'm thinking I would really like to put it in a lab environment.
> > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > Scott
> > > >
> > > > ________________________________
> > > >
> > > >
> > > > This message and any attachments are intended only for the use of the
> > > > addressee and may contain information that is privileged and
> > confidential.
> > > > If the reader of the message is not the intended recipient or an
> > > authorized
> > > > representative of the intended recipient, you are hereby notified that
> > any
> > > > dissemination of this communication is strictly prohibited. If you
> have
> > > > received this communication in error, please notify us immediately by
> > > e-mail
> > > > and delete the message and any attachments from your system.
> > > >
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> > > > ________________________________
> > > >
> > > >
> > > > New Yahoo! Messenger with Voice. Call regular phones from your PC and
> > save
> > > > big.
> > > >
> > > >
> > > >
> > > > ________________________________
> > > > This message and any attachments are intended only for the use of the
> > > > addressee and may contain information that is privileged and
> > confidential.
> > > > If the reader of the message is not the intended recipient or an
> > > authorized
> > > > representative of the intended recipient, you are hereby notified that
> > any
> > > > dissemination of this communication is strictly prohibited. If you
> have
> > > > received this communication in error, please notify us immediately by
> > > e-mail
> > > > and delete the message and any attachments from your system.
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> > > >
> > >
> > >
> > >
> > >
> > > __________________________________________________
> > > Do You Yahoo!?
> > > Tired of spam? Yahoo! Mail has the best spam protection around
> > > http://mail.yahoo.com
> >
> >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam? Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
>
>
>
>
>  __________________________________________________
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com


From cisco-voip at ibh.net  Thu Jun  8 13:22:17 2006
From: cisco-voip at ibh.net (Andre Beck)
Date: Thu, 8 Jun 2006 19:22:17 +0200
Subject: [cisco-voip] CME 4.0, SCCP ATA & fax pass-through
Message-ID: <20060608172217.GA20773@ibh.de>

Hi,

in a quite simple CME installation (3.4) I have to support several ATAs
which have fax machines connected. I set this up in "fax mode" as Cisco
calls it, meaning the ATA is hardwired to G.711 only. Sadly enough, there
are fax issues anyway. I assume they are due to echo cancellation and
maybe other signal processing side effects still present even with G.711
calls. Fax pass-through is supposed to fix this by detecting a fax call
and switching over to a special variant of G.711, disabling echo cancel,
reducing the packet size to 10ms and allowing redundancy (sending every
packet twice). Formerly this was not supported with CME and SCCP ATAs,
but with the H.323 ATA load going EOL next year, Cisco announces they
have implemented fax pass-through with SCCP ATAs on CME in CME 4.0.

So I installed 12.4(4)XC, bumped CME up to 4.0 and tried to get fax
pass-through. But I seem to fail, the codec in use is still G.711 ?-Law
instead of the a-Law I request with a modem passthrough nse statement
in voice service voip.

Anyone knows how this is supposed to be configured? Scanning the docs
for examples found no congruent results so far.

TIA,
Andre.
-- 
                  The _S_anta _C_laus _O_peration
  or "how to turn a complete illusion into a neverending money source"

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From ppauly at gmail.com  Thu Jun  8 13:31:46 2006
From: ppauly at gmail.com (Peter Pauly)
Date: Thu, 08 Jun 2006 13:31:46 -0400
Subject: [cisco-voip] Preventing Hook-flash on ATA-186
Message-ID: <44885F02.8090905@gmail.com>

We have a lot of ATA-186's (controlled from Call Manager, not SIP) that
are used to play a story to children on a 2500 set. The ATA's are setup
in Call Manager with a special calling search space and translation so
that when they pick up the phone, they get a story played to them from
Unity.

The problem is, that little kiddies (bless their hearts) bang on the
hook or else otherwise play with the phone and manage to get several
calls going at the same time. I can see simultaneous calls going on in
Call Viewer and it eats up the voicemail ports. The story plays for
around 5 minutes, so each port is tied up for that long.

Is there a way to prevent hook-flash on the ATA-186's? There's a setting
in CM called "Maximum Number of Calls" but I beleive that is inbound
calls to the phone.

From mfong at pacific.edu  Thu Jun  8 14:02:38 2006
From: mfong at pacific.edu (Melvin Fong)
Date: Thu, 08 Jun 2006 11:02:38 -0700
Subject: [cisco-voip] Preventing Hook-flash on ATA-186
Message-ID: <448803CE0200004E00009379@gwsmtp.pacific.edu>

I believe if you set the maximum calls to 1 it limits you to one call
appearance and you cannot do a switch hook to get additional dial tone,
I know this works if you limit a line on a 7960.
The Busy limit controls the incoming call behavior.

>>> Peter Pauly <ppauly at gmail.com> 6/8/06 10:31:46 >>>
We have a lot of ATA-186's (controlled from Call Manager, not SIP)
that
are used to play a story to children on a 2500 set. The ATA's are
setup
in Call Manager with a special calling search space and translation so
that when they pick up the phone, they get a story played to them from
Unity.

The problem is, that little kiddies (bless their hearts) bang on the
hook or else otherwise play with the phone and manage to get several
calls going at the same time. I can see simultaneous calls going on in
Call Viewer and it eats up the voicemail ports. The story plays for
around 5 minutes, so each port is tied up for that long.

Is there a way to prevent hook-flash on the ATA-186's? There's a
setting
in CM called "Maximum Number of Calls" but I beleive that is inbound
calls to the phone.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip

From bababooey at cox.net  Thu Jun  8 15:33:48 2006
From: bababooey at cox.net (bababooey at cox.net)
Date: Thu, 8 Jun 2006 15:33:48 -0400
Subject: [cisco-voip] IP Phones/various firmware loads
Message-ID: <26899903.1149795228235.JavaMail.root@eastrmwml02.mgt.cox.net>

Is it possbile to have different firmware loads for various models of the IP Phones in CCM 4.1.  I have mostly 7960 and 7970 but I want to keep the remote site 7970 at current revision. 

Also, I have a 7970 that is stuck on the upgrade splash screen I have tried erase and factory reset to no avail, does anyone have other tips to bring a 7970 back to life.

THanks
Bob


From lelio at uoguelph.ca  Thu Jun  8 15:37:52 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Thu, 8 Jun 2006 15:37:52 -0400
Subject: [cisco-voip] IP Phones/various firmware loads
References: <26899903.1149795228235.JavaMail.root@eastrmwml02.mgt.cox.net>
Message-ID: <001301c68b33$0b79ea50$47196883@cfs.uoguelph.ca>

As far as I know, the only way to have a device on a different load than the device defaults is to put the firmware you want into the device configuration page of the phones you want to keep at a different version. It's worked well in the past, and now, you can even present a list of devices that are not using the device defaults by going to Device > Device Settings > Firmware Load Information and looking to the right hand side for any devices that are not using the default load. 

--------------------------------------------------------------------------------
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: bababooey at cox.net 
  To: cisco-voip at puck.nether.net 
  Sent: Thursday, June 08, 2006 3:33 PM
  Subject: [cisco-voip] IP Phones/various firmware loads


  Is it possbile to have different firmware loads for various models of the IP Phones in CCM 4.1.  I have mostly 7960 and 7970 but I want to keep the remote site 7970 at current revision. 

  Also, I have a 7970 that is stuck on the upgrade splash screen I have tried erase and factory reset to no avail, does anyone have other tips to bring a 7970 back to life.

  THanks
  Bob

  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Ryan.OConnell at nexinnovations.com  Thu Jun  8 15:49:09 2006
From: Ryan.OConnell at nexinnovations.com (OConnell, Ryan)
Date: Thu, 8 Jun 2006 15:49:09 -0400
Subject: [cisco-voip] Static noise on FXO ports
Message-ID: <D9C2C0CEF6CC4248BF8F275D03E57F0C0D0CF7F2@MISEXCVS01.corp.nexinnovations.com>

Hello all,

 

I've been battling this issue with what was suppose to be a simple
little CME install for 10 people for a little while now and I'm at my
wits end. My customer has a 2801 router with VIC2-4FXO a single PVDM2-8
DSP and an 8 port powered Etherswitch module. I have turned vad on/off,
comfort noise on/off, I have played with input gain, output gain, and
impedance settings. I tested the pots lines with a regular analog phone
I get no noise at all. When making a call as soon as the call is
connected the static starts. Internal IP to IP phone calls are great and
calls over the WAN are great. TAC had me do a procedure where I captured
the voice to a wav file before it went through the DSP and after it went
through the DSP and the static can not be heard. I also got the TELCO in
to test the lines for any nuances and I had the customer ground the rack
the 2801 was mounted into. Lastly TAC shipped me all brand new equipment
but the problem still persists after I swapped it out.

 

I'm running 12.3(14T) IOS. This customer has a second site with
identical hardware and IOS and is not experiencing the problem. 

 

It has to be the phone lines but the Telco says they are fine and since
I can plug a regular phone into the line and get no noise it is hard to
argue with them.

 

 

Thoughts?

 

 

 

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

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

 

 

 

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From MLinsemier at apcapital.com  Thu Jun  8 15:56:40 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Thu, 8 Jun 2006 15:56:40 -0400
Subject: [cisco-voip] Static noise on FXO ports
Message-ID: <AB0A35F2486F81468971AA68558BE71903DCEED0@smieln01winm01.micoanet.com>

Ryan,

 

Is there any chance that you can take the problematic hardware to a
different location (along with a few phones) and test it there?  If you
were able to take that equipment to your office or house and test
standard POTS lines there and don't have a problem, you will know right
away that it is Telco or something 'outside' the router causing the
interference.

 

Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of OConnell, Ryan
Sent: Thursday, June 08, 2006 3:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Static noise on FXO ports

 

Hello all,

 

I've been battling this issue with what was suppose to be a simple
little CME install for 10 people for a little while now and I'm at my
wits end. My customer has a 2801 router with VIC2-4FXO a single PVDM2-8
DSP and an 8 port powered Etherswitch module. I have turned vad on/off,
comfort noise on/off, I have played with input gain, output gain, and
impedance settings. I tested the pots lines with a regular analog phone
I get no noise at all. When making a call as soon as the call is
connected the static starts. Internal IP to IP phone calls are great and
calls over the WAN are great. TAC had me do a procedure where I captured
the voice to a wav file before it went through the DSP and after it went
through the DSP and the static can not be heard. I also got the TELCO in
to test the lines for any nuances and I had the customer ground the rack
the 2801 was mounted into. Lastly TAC shipped me all brand new equipment
but the problem still persists after I swapped it out.

 

I'm running 12.3(14T) IOS. This customer has a second site with
identical hardware and IOS and is not experiencing the problem. 

 

It has to be the phone lines but the Telco says they are fine and since
I can plug a regular phone into the line and get no noise it is hard to
argue with them.

 

 

Thoughts?

 

 

 

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

http://www.nexinnovations.com <http://www.nexinnovations.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.

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

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From jim at tgasolutions.com  Thu Jun  8 16:09:27 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Thu, 8 Jun 2006 16:09:27 -0400
Subject: [cisco-voip] Static noise on FXO ports
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC15E4B1@exchange.TGASolutions.local>

, look for any resistance impacting devices-- Carbon blocks etc..
between where you plug in the phone and the CME..
is it a short run from the jack to the CME router?
 
Later,
Jim
 
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of OConnell, Ryan
Sent: Thursday, June 08, 2006 3:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Static noise on FXO ports



Hello all,

 

I've been battling this issue with what was suppose to be a simple
little CME install for 10 people for a little while now and I'm at my
wits end. My customer has a 2801 router with VIC2-4FXO a single PVDM2-8
DSP and an 8 port powered Etherswitch module. I have turned vad on/off,
comfort noise on/off, I have played with input gain, output gain, and
impedance settings. I tested the pots lines with a regular analog phone
I get no noise at all. When making a call as soon as the call is
connected the static starts. Internal IP to IP phone calls are great and
calls over the WAN are great. TAC had me do a procedure where I captured
the voice to a wav file before it went through the DSP and after it went
through the DSP and the static can not be heard. I also got the TELCO in
to test the lines for any nuances and I had the customer ground the rack
the 2801 was mounted into. Lastly TAC shipped me all brand new equipment
but the problem still persists after I swapped it out.

 

I'm running 12.3(14T) IOS. This customer has a second site with
identical hardware and IOS and is not experiencing the problem. 

 

It has to be the phone lines but the Telco says they are fine and since
I can plug a regular phone into the line and get no noise it is hard to
argue with them.

 

 

Thoughts?

 

 

 

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

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

 

 

 

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From jgrace at digitelusa.net  Thu Jun  8 15:58:11 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu, 8 Jun 2006 15:58:11 -0400
Subject: [cisco-voip] how to add addition address books to ipphone directorys
Message-ID: <200606081610781.SM01572@3729434770>

how to add addition address books to ipphone directorys.  We want to create
one call remote user and has there cell phone number.  I know I could add a
user to cm with their cell # .  but that will fall into the corp dir.

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Ryan.OConnell at nexinnovations.com  Thu Jun  8 17:05:02 2006
From: Ryan.OConnell at nexinnovations.com (OConnell, Ryan)
Date: Thu, 8 Jun 2006 17:05:02 -0400
Subject: [cisco-voip] Static noise on FXO ports
Message-ID: <D9C2C0CEF6CC4248BF8F275D03E57F0C0D0CF831@MISEXCVS01.corp.nexinnovations.com>

The phones are directly connected to the powered ESM on the router. I
have ruled out internal cabling issues by directly connecting phones to
the ESM. Also IP to IP calls work fine it is only PSTN calls through the
FXO ports. Another interesting thing that I have noticed is that the
static seems to only happen as I am talking into the IP phone. If the
PSTN person is talking and I am listening I hear no static. The moment I
speak into the IP phone I hear the background static in my ear. The PSTN
person does not hear the static. If I have a call connected and do not
speak into the set it sound clean. If I breath into the set I hear
static, it is continuous as I speak or add noise to the mouth-piece.

 

Thank

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

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

 

 

________________________________

From: Jim McBurnett [mailto:jim at tgasolutions.com] 
Sent: June 8, 2006 2:09 PM
To: OConnell, Ryan; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Static noise on FXO ports

 

, look for any resistance impacting devices-- Carbon blocks etc..
between where you plug in the phone and the CME..

is it a short run from the jack to the CME router?

 

Later,

Jim

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of OConnell, Ryan
Sent: Thursday, June 08, 2006 3:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Static noise on FXO ports

Hello all,

 

I've been battling this issue with what was suppose to be a simple
little CME install for 10 people for a little while now and I'm at my
wits end. My customer has a 2801 router with VIC2-4FXO a single PVDM2-8
DSP and an 8 port powered Etherswitch module. I have turned vad on/off,
comfort noise on/off, I have played with input gain, output gain, and
impedance settings. I tested the pots lines with a regular analog phone
I get no noise at all. When making a call as soon as the call is
connected the static starts. Internal IP to IP phone calls are great and
calls over the WAN are great. TAC had me do a procedure where I captured
the voice to a wav file before it went through the DSP and after it went
through the DSP and the static can not be heard. I also got the TELCO in
to test the lines for any nuances and I had the customer ground the rack
the 2801 was mounted into. Lastly TAC shipped me all brand new equipment
but the problem still persists after I swapped it out.

 

I'm running 12.3(14T) IOS. This customer has a second site with
identical hardware and IOS and is not experiencing the problem. 

 

It has to be the phone lines but the Telco says they are fine and since
I can plug a regular phone into the line and get no noise it is hard to
argue with them.

 

 

Thoughts?

 

 

 

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

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

 

 

 

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From Ryan.OConnell at nexinnovations.com  Thu Jun  8 17:09:54 2006
From: Ryan.OConnell at nexinnovations.com (OConnell, Ryan)
Date: Thu, 8 Jun 2006 17:09:54 -0400
Subject: [cisco-voip] how to add addition address books to ipphone
	directorys
Message-ID: <D9C2C0CEF6CC4248BF8F275D03E57F0C0D0CF834@MISEXCVS01.corp.nexinnovations.com>

Just edit the Directories XML file to call up another, xml directory. I
used the xml file for CCME My Fastdials service as a template. Then I
edited the Directories xml file in CM to call that FASTDIALS.xml file
which contained all the numbers the company asked for as a separate line
or menu Item.

 

Ryan

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

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

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of James Grace
Sent: June 8, 2006 1:58 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] how to add addition address books to ipphone
directorys

 

how to add addition address books to ipphone directorys.  We want to
create one call remote user and has there cell phone number.  I know I
could add a user to cm with their cell # .  but that will fall into the
corp dir.

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

-------------- next part --------------
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From MLinsemier at apcapital.com  Thu Jun  8 17:12:02 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Thu, 8 Jun 2006 17:12:02 -0400
Subject: [cisco-voip] Static noise on FXO ports
Message-ID: <AB0A35F2486F81468971AA68558BE71903DCEF38@smieln01winm01.micoanet.com>

What firmware loads are you using on the phones?  Have you tried a newer
or older load?

 

Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of OConnell, Ryan
Sent: Thursday, June 08, 2006 5:05 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Static noise on FXO ports

 

The phones are directly connected to the powered ESM on the router. I
have ruled out internal cabling issues by directly connecting phones to
the ESM. Also IP to IP calls work fine it is only PSTN calls through the
FXO ports. Another interesting thing that I have noticed is that the
static seems to only happen as I am talking into the IP phone. If the
PSTN person is talking and I am listening I hear no static. The moment I
speak into the IP phone I hear the background static in my ear. The PSTN
person does not hear the static. If I have a call connected and do not
speak into the set it sound clean. If I breath into the set I hear
static, it is continuous as I speak or add noise to the mouth-piece.

 

Thank

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

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

 

 

________________________________

From: Jim McBurnett [mailto:jim at tgasolutions.com] 
Sent: June 8, 2006 2:09 PM
To: OConnell, Ryan; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Static noise on FXO ports

 

, look for any resistance impacting devices-- Carbon blocks etc..
between where you plug in the phone and the CME..

is it a short run from the jack to the CME router?

 

Later,

Jim

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of OConnell, Ryan
Sent: Thursday, June 08, 2006 3:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Static noise on FXO ports

Hello all,

 

I've been battling this issue with what was suppose to be a simple
little CME install for 10 people for a little while now and I'm at my
wits end. My customer has a 2801 router with VIC2-4FXO a single PVDM2-8
DSP and an 8 port powered Etherswitch module. I have turned vad on/off,
comfort noise on/off, I have played with input gain, output gain, and
impedance settings. I tested the pots lines with a regular analog phone
I get no noise at all. When making a call as soon as the call is
connected the static starts. Internal IP to IP phone calls are great and
calls over the WAN are great. TAC had me do a procedure where I captured
the voice to a wav file before it went through the DSP and after it went
through the DSP and the static can not be heard. I also got the TELCO in
to test the lines for any nuances and I had the customer ground the rack
the 2801 was mounted into. Lastly TAC shipped me all brand new equipment
but the problem still persists after I swapped it out.

 

I'm running 12.3(14T) IOS. This customer has a second site with
identical hardware and IOS and is not experiencing the problem. 

 

It has to be the phone lines but the Telco says they are fine and since
I can plug a regular phone into the line and get no noise it is hard to
argue with them.

 

 

Thoughts?

 

 

 

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

http://www.nexinnovations.com <http://www.nexinnovations.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.

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

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From Brian.Henry at apptis.com  Thu Jun  8 18:30:20 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu, 8 Jun 2006 18:30:20 -0400
Subject: [cisco-voip] IP Phones/various firmware loads
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC0283821D@CH-EXCH-02.intelltech.apptis.inc>

You can have different loads for phone model on various endpoints.
7960-1 - 8.0(1)
7960-2 - 8.0(2)
7960-3 - 8.0(3)
 
Just make sure you have all the files in the TFTP dir and go to the phone(device) and there is a place that you can put in the name of the phone load that you want.
 
Part-2
What version of CCM? and OSver? 2000.7 or 2000.4.X?
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of bababooey at cox.net
Sent: Thu 6/8/2006 3:33 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IP Phones/various firmware loads



Is it possbile to have different firmware loads for various models of the IP Phones in CCM 4.1.  I have mostly 7960 and 7970 but I want to keep the remote site 7970 at current revision.

Also, I have a 7970 that is stuck on the upgrade splash screen I have tried erase and factory reset to no avail, does anyone have other tips to bring a 7970 back to life.

THanks
Bob

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


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From kburns at oneipvoice.com  Thu Jun  8 21:17:21 2006
From: kburns at oneipvoice.com (Keith Burns)
Date: Thu, 08 Jun 2006 19:17:21 -0600
Subject: [cisco-voip] Voice clipping on Cisco PRI
Message-ID: <C0AE2841.A5E%kburns@oneipvoice.com>

Hi,

I have a customer with a PBX connected to a 2801. He says that when we both
talk (I am calling in from PSTN) he hears my voice clipping, yet I hear his
fine.

I was wondering if someone has seen this before.

Here are some configlets and version:

<snip>
isdn switch-type primary-ni
!
voice-card 0
!
!
!
voice service voip 
 fax protocol pass-through g711ulaw
 modem passthrough nse codec g711ulaw
!
controller T1 0/3/0
 framing esf
 clock source internal
 linecode b8zs
 pri-group timeslots 1-8,24
!
class-map match-all voip
 match access-group 105
!
!
policy-map llq
 class voip
  priority percent 100
 class class-default
  fair-queue
!
<snip>
interface Serial0/1/0:0.16 point-to-point
 ip address 10.20.121.1 255.255.255.252
 no cdp enable
 frame-relay interface-dlci 16 IETF protocol ip 10.20.121.2
  class T1-throttledown
!
interface Serial0/3/0:23
 no ip address
 isdn switch-type primary-ni
 isdn protocol-emulate network
 isdn incoming-voice voice
 no cdp enable
!
<snip>
map-class frame-relay T1-throttledown
 frame-relay traffic-rate 1544000 1544000
 service-policy output llq
!
access-list 105 permit ip any any precedence critical
access-list 105 permit ip any any dscp ef
!
voice-port 0/3/0:23
!
dial-peer voice 1000 voip
 application session
 destination-pattern .T
 session protocol sipv2
 session target ipv4:10.50.0.40
 codec g711ulaw
 ip qos dscp ef signaling
 no vad
!
dial-peer voice 103 pots
 application session
 destination-pattern <snip>75..
 direct-inward-dial
 port 0/3/0:23
 prefix 75
!

accurate-pri#sh ver
Cisco IOS Software, 2801 Software (C2801-IPVOICE-M), Version 12.3(11)T5,
RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2005 by Cisco Systems, Inc.
Compiled Sat 02-Apr-05 21:22 by yiyan

ROM: System Bootstrap, Version 12.3(8r)T8, RELEASE SOFTWARE (fc1)

accurate-pri uptime is 3 weeks, 5 days, 2 hours, 39 minutes
System returned to ROM by reload at 22:37:40 UTC Sat May 13 2006
System image file is "flash:c2801-ipvoice-mz.123-11.T5.bin"

Cisco 2801 (revision 5.0) with 115712K/15360K bytes of memory.
Processor board ID FTX0946Y355
2 FastEthernet interfaces
10 Serial interfaces
2 Channelized T1/PRI ports
1 DSP, 8 Voice resources
DRAM configuration is 64 bits wide with parity disabled.
191K bytes of NVRAM.
62720K bytes of ATA CompactFlash (Read/Write)

Configuration register is 0x2102

!
Any help greatly appreciated!

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From Ryan.OConnell at nexinnovations.com  Thu Jun  8 21:25:21 2006
From: Ryan.OConnell at nexinnovations.com (OConnell, Ryan)
Date: Thu, 8 Jun 2006 21:25:21 -0400
Subject: [cisco-voip] Voice clipping on Cisco PRI
Message-ID: <D9C2C0CEF6CC4248BF8F275D03E57F0C0D0CF893@MISEXCVS01.corp.nexinnovations.com>

Keith try this 

 

 

network-clock-select 1 T1 0/0/0

 

 

 

Ryan O'Connell 

Senior Consultant CCIE Voice (13382)

 

NexInnovations Inc.

3636 - 23rd Street N.E.

Calgary, AB. T2E 8Z5

 

Desk (403) 291-7293

Cell: (403) 990-1275

Fax: (403) 291-7281

 

Ryan.OConnell at nexinnovations.com

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

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Keith Burns
Sent: June 8, 2006 7:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Voice clipping on Cisco PRI

 

Hi,

I have a customer with a PBX connected to a 2801. He says that when we
both talk (I am calling in from PSTN) he hears my voice clipping, yet I
hear his fine.

I was wondering if someone has seen this before.

Here are some configlets and version:

<snip>
isdn switch-type primary-ni
!
voice-card 0
!
!
!
voice service voip 
 fax protocol pass-through g711ulaw
 modem passthrough nse codec g711ulaw
!
controller T1 0/3/0
 framing esf
 clock source internal
 linecode b8zs
 pri-group timeslots 1-8,24
!
class-map match-all voip
 match access-group 105
!
!
policy-map llq
 class voip
  priority percent 100
 class class-default
  fair-queue
!
<snip>
interface Serial0/1/0:0.16 point-to-point
 ip address 10.20.121.1 255.255.255.252
 no cdp enable
 frame-relay interface-dlci 16 IETF protocol ip 10.20.121.2
  class T1-throttledown
!
interface Serial0/3/0:23
 no ip address
 isdn switch-type primary-ni
 isdn protocol-emulate network
 isdn incoming-voice voice
 no cdp enable
!
<snip>
map-class frame-relay T1-throttledown
 frame-relay traffic-rate 1544000 1544000
 service-policy output llq
!
access-list 105 permit ip any any precedence critical
access-list 105 permit ip any any dscp ef
!
voice-port 0/3/0:23
!
dial-peer voice 1000 voip
 application session
 destination-pattern .T
 session protocol sipv2
 session target ipv4:10.50.0.40
 codec g711ulaw
 ip qos dscp ef signaling
 no vad
!
dial-peer voice 103 pots
 application session
 destination-pattern <snip>75..
 direct-inward-dial
 port 0/3/0:23
 prefix 75
!

accurate-pri#sh ver
Cisco IOS Software, 2801 Software (C2801-IPVOICE-M), Version 12.3(11)T5,
RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2005 by Cisco Systems, Inc.
Compiled Sat 02-Apr-05 21:22 by yiyan

ROM: System Bootstrap, Version 12.3(8r)T8, RELEASE SOFTWARE (fc1)

accurate-pri uptime is 3 weeks, 5 days, 2 hours, 39 minutes
System returned to ROM by reload at 22:37:40 UTC Sat May 13 2006
System image file is "flash:c2801-ipvoice-mz.123-11.T5.bin"

Cisco 2801 (revision 5.0) with 115712K/15360K bytes of memory.
Processor board ID FTX0946Y355
2 FastEthernet interfaces
10 Serial interfaces
2 Channelized T1/PRI ports
1 DSP, 8 Voice resources
DRAM configuration is 64 bits wide with parity disabled.
191K bytes of NVRAM.
62720K bytes of ATA CompactFlash (Read/Write)

Configuration register is 0x2102

!
Any help greatly appreciated!

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From David.Grullon at co.galveston.tx.us  Thu Jun  8 22:22:56 2006
From: David.Grullon at co.galveston.tx.us (Grullon, David)
Date: Thu, 8 Jun 2006 21:22:56 -0500
Subject: [cisco-voip] CCM SQL
Message-ID: <B8A79694796F4D45B086E36DCFC1005901C55AD2@SVREXC2K3.gc.pri>

While Pub to Sub SQL db
Login failed for user 'sa' Reason Not associated with trusted SQL server connection

Thanks
Dave
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From bababooey at cox.net  Fri Jun  9 00:37:56 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Fri, 9 Jun 2006 00:37:56 -0400
Subject: [cisco-voip] IP Phones/various firmware loads
References: <2F09B2162CCDEA4EA05C089B24FBC2BC0283821D@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <000701c68b7e$7b2f10a0$0f0110ac@corporation2>

[cisco-voip] IP Phones/various firmware loadsCCM 4.1.3sr2, OServ 2000.7sr8
  ----- Original Message ----- 
  From: Brian Henry 
  To: bababooey at cox.net ; cisco-voip at puck.nether.net 
  Sent: Thursday, June 08, 2006 6:30 PM
  Subject: RE: [cisco-voip] IP Phones/various firmware loads


  You can have different loads for phone model on various endpoints.
  7960-1 - 8.0(1)
  7960-2 - 8.0(2)
  7960-3 - 8.0(3)

  Just make sure you have all the files in the TFTP dir and go to the phone(device) and there is a place that you can put in the name of the phone load that you want.

  Part-2
  What version of CCM? and OSver? 2000.7 or 2000.4.X?

  Brian


------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net on behalf of bababooey at cox.net
  Sent: Thu 6/8/2006 3:33 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] IP Phones/various firmware loads


  Is it possbile to have different firmware loads for various models of the IP Phones in CCM 4.1.  I have mostly 7960 and 7970 but I want to keep the remote site 7970 at current revision.

  Also, I have a 7970 that is stuck on the upgrade splash screen I have tried erase and factory reset to no avail, does anyone have other tips to bring a 7970 back to life.

  THanks
  Bob

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

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From Brian.Henry at apptis.com  Fri Jun  9 04:58:10 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Fri, 9 Jun 2006 04:58:10 -0400
Subject: [cisco-voip] IP Phones/various firmware loads
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC0283821E@CH-EXCH-02.intelltech.apptis.inc>

Honestly I do not know which one of these fixed the issue because I took the 7970 from a live environment to a lab where I had 4.1(3)sr3a and OS 2004.4.2sr6(or 7) and the latest device pack.
The live environment was 4.1(3)sr2, OS 2007.2.X(sr7) with not the latest device pack.
 
However the phone was doing the same thing that you were describing but when I moved it to the lab it came right up, after it put the latest phone load on of course. 
 
Brian

________________________________

From: Bob A. Bowie [mailto:bababooey at cox.net]
Sent: Fri 6/9/2006 12:37 AM
To: Brian Henry; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IP Phones/various firmware loads


CCM 4.1.3sr2, OServ 2000.7sr8

	----- Original Message ----- 
	From: Brian Henry <mailto:Brian.Henry at apptis.com>  
	To: bababooey at cox.net ; cisco-voip at puck.nether.net 
	Sent: Thursday, June 08, 2006 6:30 PM
	Subject: RE: [cisco-voip] IP Phones/various firmware loads

	You can have different loads for phone model on various endpoints.
	7960-1 - 8.0(1)
	7960-2 - 8.0(2)
	7960-3 - 8.0(3)
	 
	Just make sure you have all the files in the TFTP dir and go to the phone(device) and there is a place that you can put in the name of the phone load that you want.
	 
	Part-2
	What version of CCM? and OSver? 2000.7 or 2000.4.X?
	 
	Brian

________________________________

	From: cisco-voip-bounces at puck.nether.net on behalf of bababooey at cox.net
	Sent: Thu 6/8/2006 3:33 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] IP Phones/various firmware loads
	
	

	Is it possbile to have different firmware loads for various models of the IP Phones in CCM 4.1.  I have mostly 7960 and 7970 but I want to keep the remote site 7970 at current revision.
	
	Also, I have a 7970 that is stuck on the upgrade splash screen I have tried erase and factory reset to no avail, does anyone have other tips to bring a 7970 back to life.
	
	THanks
	Bob
	
	_______________________________________________
	cisco-voip mailing list
	cisco-voip at puck.nether.net
	https://puck.nether.net/mailman/listinfo/cisco-voip
	

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From loris.vicinanza at shearman.com  Fri Jun  9 06:40:07 2006
From: loris.vicinanza at shearman.com (Loris Vicinanza)
Date: Fri, 9 Jun 2006 12:40:07 +0200
Subject: [cisco-voip] attendant console question
Message-ID: <OF3E9FBDA9.1D1C8D5D-ONC1257188.002F2A11-C1257188.003A9BBE@shearman.com>

Hi all,

I have a receptionist that receives all incoming calls from our main office
number, and I would like her to monitor all user's phone lines (50 phones)
and eventually transfer calls if the user line is free. She currently has a
7970 phone with two expansion modules, from what I understand the attendant
console is tied to the modules and the 7970 phone cannot have more than two
module connected-
I have a couple questions: is there a way or a different phone which
supports more than two modules?and is there a way to make the attendant
console not dependant from the number of modules the receptionist has?

thanks in advance!



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

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

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

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

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

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


From Matt.Slaga at us.didata.com  Fri Jun  9 07:34:02 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 9 Jun 2006 07:34:02 -0400
Subject: [cisco-voip] attendant console question
Message-ID: <C1FE15183DA37645BC0633BC604E44F0021FDF55@USNAEXCH.na.didata.local>

I would try Arc Solutions' Arc Enterprise Console.  Way better than
Cisco's out of the box solution.  It will give you the functionality you
are looking for without all the extra sidecars (any, for that matter).

http://www.arcsolutions.com



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Loris Vicinanza
Sent: Friday, June 09, 2006 6:40 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] attendant console question

Hi all,

I have a receptionist that receives all incoming calls from our main
office number, and I would like her to monitor all user's phone lines
(50 phones) and eventually transfer calls if the user line is free. She
currently has a 7970 phone with two expansion modules, from what I
understand the attendant console is tied to the modules and the 7970
phone cannot have more than two module connected- I have a couple
questions: is there a way or a different phone which supports more than
two modules?and is there a way to make the attendant console not
dependant from the number of modules the receptionist has?

thanks in advance!



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

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

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

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

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

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

_______________________________________________
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
this communication in error and that any use or reproduction of
this email or its contents is strictly prohibited and may be
unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.



From Matt.Slaga at us.didata.com  Fri Jun  9 07:35:12 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 9 Jun 2006 07:35:12 -0400
Subject: [cisco-voip] CCM SQL
Message-ID: <C1FE15183DA37645BC0633BC604E44F0021FDF56@USNAEXCH.na.didata.local>

What version of CallManager?  4.x does not enable SA login by default,
only windows authentication.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Grullon, David
Sent: Thursday, June 08, 2006 10:23 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM SQL



While Pub to Sub SQL db 
Login failed for user 'sa' Reason Not associated with trusted SQL server
connection 

Thanks 
Dave 




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

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From JOlson at mlnusa.com  Fri Jun  9 08:59:44 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Fri, 9 Jun 2006 08:59:44 -0400
Subject: [cisco-voip] IPCC 3.5 to 4.0
Message-ID: <FE2D092F8B1516488B99342551AACFD409CD4932@mlnusa-exch1.mlnusa.com>

I need to upgrade my IPCC to version 4.04 from version 3.5.  I know that the CRA's name in 3.5 is changed to CRS.  Will I be able to open my 3.5 scripts in 4.04?  Also does anyone have a link that defines the upgrade matrix.  (what versions of call manager and unity I need to have in order for my upgrade to be successful)

Thanks
Jeff

********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

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From Timothy_Frazee at adp.com  Fri Jun  9 10:11:29 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Fri, 9 Jun 2006 09:11:29 -0500
Subject: [cisco-voip] inside verses outside call and ring cadence
Message-ID: <C0427FD6-73B8-4999-B9C3-B55318635510@mimectl>

 all,

when a call comes in from outside, how does ccm determine to change the cadence on the ip phone?

is this the offnet and on-net thing?

how would some calls not ring-ring (external) and some would ring (internal) when they all come from outside in.

running 4.1.3sr2 with a 2821 and 11 FXO to pots via H323.

thanks.


This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From Scott.Voll at wesd.org  Fri Jun  9 10:33:12 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 9 Jun 2006 07:33:12 -0700
Subject: [cisco-voip] IPCC 3.5 to 4.0
References: <FE2D092F8B1516488B99342551AACFD409CD4932@mlnusa-exch1.mlnusa.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A605D75A17@Apollo.wesd.org>

my scripts worked just fine going from 3.5 to 4.0.  the parameters need " marks but that was my only problem.
 
IPCCx (crs, cra, what ever you would like to call it) 4.x works with 4.x I believe.  Unity should not matter.  I'm out of the office.  but if you search CCO for IPCCx metrix you should find all the details.
 
Scott

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Olson, Jeffrey
Sent: Fri 6/9/2006 5:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC 3.5 to 4.0



I need to upgrade my IPCC to version 4.04 from version 3.5.  I know that the CRA's name in 3.5 is changed to CRS.  Will I be able to open my 3.5 scripts in 4.04?  Also does anyone have a link that defines the upgrade matrix.  (what versions of call manager and unity I need to have in order for my upgrade to be successful)

Thanks 
Jeff 

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

 

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From CORTIZ at broward.org  Fri Jun  9 10:56:59 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 9 Jun 2006 10:56:59 -0400
Subject: [cisco-voip] IPCC 3.5 to 4.0
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC344E@MREXVS01.bc.broward.cty>

What do you mean parameters need " marks ??  I am looking to upgrade to
4.X as well.  From what I read 4.X requires 4.1(2) CM minimum:

 

Matrix:

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/ccmcomp.h
tm

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, June 09, 2006 10:33 AM
To: Olson, Jeffrey; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC 3.5 to 4.0

 

my scripts worked just fine going from 3.5 to 4.0.  the parameters need
" marks but that was my only problem.

 

IPCCx (crs, cra, what ever you would like to call it) 4.x works with 4.x
I believe.  Unity should not matter.  I'm out of the office.  but if you
search CCO for IPCCx metrix you should find all the details.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Olson, Jeffrey
Sent: Fri 6/9/2006 5:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC 3.5 to 4.0

I need to upgrade my IPCC to version 4.04 from version 3.5.  I know that
the CRA's name in 3.5 is changed to CRS.  Will I be able to open my 3.5
scripts in 4.04?  Also does anyone have a link that defines the upgrade
matrix.  (what versions of call manager and unity I need to have in
order for my upgrade to be successful)

Thanks 
Jeff 

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders
Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
hereby notified that disclosing, copying, distributing, or taking any
action in reliance on the contents of this information is strictly
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immediately. If you have received this E-mail in error, please contact
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From JAbbey at regattausa.com  Fri Jun  9 10:55:58 2006
From: JAbbey at regattausa.com (Joshua Abbey)
Date: Fri, 9 Jun 2006 10:55:58 -0400
Subject: [cisco-voip] blocking a specific unwanted inbound call
In-Reply-To: <mailman.5429.1149862296.1478.cisco-voip@puck.nether.net>
Message-ID: <OFE6464A14.BAF962C4-ON85257188.0051A51D-85257188.0052080A@regattausa.com>

Is there any way for a CCM 4.0sr2a, with MGCP 2651xm gateways to 
selectively block or drop calls inbound from a specific phone number? 
We've been receiving unwanted calls from an ex-employee, just calling and 
hanging up, calling and hanging up, and it is apparently not threatening 
enough for the police to do anything.  Does anyone know if this is doable?

Regards,
-jla

Joshua L. Abbey
Director - IT & Network Technology
Regatta USA LLC
212.589-5239 (o)
908.601.5993 (c)
212-827-3039 (f)
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From jfaulhaber at avodaq.com  Fri Jun  9 11:19:34 2006
From: jfaulhaber at avodaq.com (Jens Faulhaber)
Date: Fri, 9 Jun 2006 17:19:34 +0200
Subject: [cisco-voip] CCM Jatpi controlled single T1 CAS Ports
Message-ID: <F666605263F0EA40BC95239B1ADE10BC0A816F@sta-exch-1.avodaq.com>

Hi all,

I need help :-)

 

I need every single channel on a T1/E1 CAS to be controlled by CCM
JTAPI.

Every single channel should have its own number.

The Issue is that we have an IVR connected over T1or E1 CAS and need to
transfer the calls to the IVR back to CCM ....

Unfortunately the IVR is not capable of transferring the calls, so a
jtapi user on CCM should do this.

 

Any ideas? 

Best regards

Jens  

 

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From bababooey at cox.net  Fri Jun  9 11:55:58 2006
From: bababooey at cox.net (bababooey at cox.net)
Date: Fri, 9 Jun 2006 11:55:58 -0400
Subject: [cisco-voip] vmware gsx server
Message-ID: <20861400.1149868558357.JavaMail.root@eastrmwml02.mgt.cox.net>

Has anyone utilized the gsx server to run the call mamanger environment in a vmware test bed.

Thanks,


From lelio at uoguelph.ca  Fri Jun  9 13:06:38 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 9 Jun 2006 13:06:38 -0400
Subject: [cisco-voip] 3700 vs 3800 for SRST router
Message-ID: <000c01c68be7$1261bfa0$47196883@cfs.uoguelph.ca>

Just wondering what the main difference between the 3700 and 3800 series routers are. We're looking for a SRST device that will also have some T1 trunks and perhaps some analog FXO ports as well. I know that the different models within each series have different maximum supported devices, it's likely I'll either get a 3845 or 3745 in order to get the SRST density I need.

--------------------------------------------------------------------------------
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 Matt.Slaga at us.didata.com  Fri Jun  9 13:16:34 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 9 Jun 2006 13:16:34 -0400
Subject: [cisco-voip] blocking a specific unwanted inbound call
Message-ID: <C1FE15183DA37645BC0633BC604E44F00223C6ED@USNAEXCH.na.didata.local>

The easiest way is to call the phone company and request it be blocked
from their end.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joshua Abbey
Sent: Friday, June 09, 2006 10:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] blocking a specific unwanted inbound call



Is there any way for a CCM 4.0sr2a, with MGCP 2651xm gateways to
selectively block or drop calls inbound from a specific phone number?
We've been receiving unwanted calls from an ex-employee, just calling
and hanging up, calling and hanging up, and it is apparently not
threatening enough for the police to do anything.  Does anyone know if
this is doable? 

Regards,
-jla

Joshua L. Abbey
Director - IT & Network Technology
Regatta USA LLC
212.589-5239 (o)
908.601.5993 (c)
212-827-3039 (f)



-----------------------------------------
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 CORTIZ at broward.org  Fri Jun  9 13:17:17 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 9 Jun 2006 13:17:17 -0400
Subject: [cisco-voip] 3700 vs 3800 for SRST router
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC3451@MREXVS01.bc.broward.cty>

Lelio,

 

I would stick to the newer ISR models (2800, 3800) if possible.  I now
strictly buy these and have been pretty happy with them?  How many SRST
users do you need to support?

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, June 09, 2006 1:07 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 3700 vs 3800 for SRST router

 

Just wondering what the main difference between the 3700 and 3800 series
routers are. We're looking for a SRST device that will also have some T1
trunks and perhaps some analog FXO ports as well. I know that the
different models within each series have different maximum supported
devices, it's likely I'll either get a 3845 or 3745 in order to get the
SRST density I need.

 

------------------------------------------------------------------------
--------
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 JOlson at mlnusa.com  Fri Jun  9 13:32:39 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Fri, 9 Jun 2006 13:32:39 -0400
Subject: [cisco-voip] Unity Java authentication
Message-ID: <FE2D092F8B1516488B99342551AACFD41C6340FB@mlnusa-exch1.mlnusa.com>

I am running Unity 4.0 Build 4.0(4).  When I log in to Unity I am able
to authenticate with the first authentication window but when the second
window comes up (I believe it is a Java window) I am not able to
authenticate.  I am able to authenticate from any other PCs in my
organization but not mine.  By clicking cancel on this authentication
window I am not able to play recordings or record scripts.  I have
uninstalled Java and reinstalled it on my PC and it didn't correct the
issue.  Anyone have any ideas how I can correct this problem?


Jeff

********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

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From scott.coner at us.didata.com  Fri Jun  9 13:57:56 2006
From: scott.coner at us.didata.com (Scott Coner (US))
Date: Fri, 9 Jun 2006 13:57:56 -0400
Subject: [cisco-voip] Unity Java authentication
References: <FE2D092F8B1516488B99342551AACFD41C6340FB@mlnusa-exch1.mlnusa.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00C91C1@USNAEXCH.na.didata.local>

Are you accessing Unity using an IP address or a host name?  Can your PC properly resolve the host name to an IP address?  There was a similar issue to this with 4.0(3) and name resolution but I thought it was resolved in 4.0(4).

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Olson, Jeffrey
Sent: Fri 6/9/2006 1:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Java authentication



I am running Unity 4.0 Build 4.0(4).  When I log in to Unity I am able to authenticate with the first authentication window but when the second window comes up (I believe it is a Java window) I am not able to authenticate.  I am able to authenticate from any other PCs in my organization but not mine.  By clicking cancel on this authentication window I am not able to play recordings or record scripts.  I have uninstalled Java and reinstalled it on my PC and it didn't correct the issue.  Anyone have any ideas how I can correct this problem?


Jeff 

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

 




-----------------------------------------
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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notify us immediately by replying to this message and deleting it
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From lelio at uoguelph.ca  Fri Jun  9 14:02:49 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 9 Jun 2006 14:02:49 -0400
Subject: [cisco-voip] 3700 vs 3800 for SRST router
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC3451@MREXVS01.bc.broward.cty>
Message-ID: <003c01c68bee$ec635d10$47196883@cfs.uoguelph.ca>

Anywhere between 250 and 400 users. There are plans for expansion after the migration of this remote site, so we want to make sure we are ready for it. After they see the cost, they may say that the expansion does not require SRST. ;)

So the 3800 series are newer boxes then?

--------------------------------------------------------------------------------
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: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Friday, June 09, 2006 1:17 PM
  Subject: RE: [cisco-voip] 3700 vs 3800 for SRST router


  Lelio,

   

  I would stick to the newer ISR models (2800, 3800) if possible.  I now strictly buy these and have been pretty happy with them?  How many SRST users do you need to support?

   

   

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Friday, June 09, 2006 1:07 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] 3700 vs 3800 for SRST router

   

  Just wondering what the main difference between the 3700 and 3800 series routers are. We're looking for a SRST device that will also have some T1 trunks and perhaps some analog FXO ports as well. I know that the different models within each series have different maximum supported devices, it's likely I'll either get a 3845 or 3745 in order to get the SRST density I need.

   

  --------------------------------------------------------------------------------
  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 Eric.A.Fretz at L-3Com.com  Fri Jun  9 14:03:30 2006
From: Eric.A.Fretz at L-3Com.com (Fretz, EA Eric @ IS)
Date: Fri, 9 Jun 2006 13:03:30 -0500
Subject: [cisco-voip] 3700 vs 3800 for SRST router
Message-ID: <F2E442781002EA41B3DB013749748D3033A1A2@gvlexch02.is.l-3com.com>

3800 series can support PVDM chips directly on the motherboard, which
means the 4 built-in WIC slots can also function as VWIC slots.  On the
3700 series, you have to use a NM-HDV  type of netmodule to get DSPs
into the chassis.  This is important because on the 3845, you only have
to buy the router (w/ voice package) and the VWIC cards.  On the 3745,
you'd have to buy the netmodule cards, as well.  This actually makes the
3845 come out a bit cheaper.  According to this page, the 3700 series is
EOL after Mar 7, 2007:
http://www.cisco.com/en/US/products/hw/routers/ps282/prod_eol_notice0900
aecd80444877.html
 
The 3845 sports a faster main CPU, which will have an impact on the
quantity of MGPC/SCCP devices you can support.
 
Unless you are getting a 3745 used or you already have one, save
yourself the effort and get a 3845.

	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	Sent: Friday, June 09, 2006 12:07 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] 3700 vs 3800 for SRST router
	
	
	Just wondering what the main difference between the 3700 and
3800 series routers are. We're looking for a SRST device that will also
have some T1 trunks and perhaps some analog FXO ports as well. I know
that the different models within each series have different maximum
supported devices, it's likely I'll either get a 3845 or 3745 in order
to get the SRST density I need.
	 
	
------------------------------------------------------------------------
--------
	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 lelio at uoguelph.ca  Fri Jun  9 14:08:55 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 9 Jun 2006 14:08:55 -0400
Subject: [cisco-voip] 3700 vs 3800 for SRST router
References: <F2E442781002EA41B3DB013749748D3033A1A2@gvlexch02.is.l-3com.com>
Message-ID: <004c01c68bef$c57176f0$47196883@cfs.uoguelph.ca>

MessageWell, that settles it. 3845s for everyone!

--------------------------------------------------------------------------------
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: Fretz, EA Eric @ IS 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Friday, June 09, 2006 2:03 PM
  Subject: RE: [cisco-voip] 3700 vs 3800 for SRST router


  3800 series can support PVDM chips directly on the motherboard, which means the 4 built-in WIC slots can also function as VWIC slots.  On the 3700 series, you have to use a NM-HDV  type of netmodule to get DSPs into the chassis.  This is important because on the 3845, you only have to buy the router (w/ voice package) and the VWIC cards.  On the 3745, you'd have to buy the netmodule cards, as well.  This actually makes the 3845 come out a bit cheaper.  According to this page, the 3700 series is EOL after Mar 7, 2007:
  http://www.cisco.com/en/US/products/hw/routers/ps282/prod_eol_notice0900aecd80444877.html

  The 3845 sports a faster main CPU, which will have an impact on the quantity of MGPC/SCCP devices you can support.

  Unless you are getting a 3745 used or you already have one, save yourself the effort and get a 3845.
    -----Original Message-----
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
    Sent: Friday, June 09, 2006 12:07 PM
    To: cisco-voip at puck.nether.net
    Subject: [cisco-voip] 3700 vs 3800 for SRST router


    Just wondering what the main difference between the 3700 and 3800 series routers are. We're looking for a SRST device that will also have some T1 trunks and perhaps some analog FXO ports as well. I know that the different models within each series have different maximum supported devices, it's likely I'll either get a 3845 or 3745 in order to get the SRST density I need.

    --------------------------------------------------------------------------------
    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  Fri Jun  9 14:18:33 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 9 Jun 2006 11:18:33 -0700
Subject: [cisco-voip] Broadcast group -- Pickup
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE701@Apollo.wesd.org>

If I setup a Broadcast Hunt list, can I setup some phones with the
pickup function to pickup that line?

 

Thanks

 

Scott

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From JOlson at mlnusa.com  Fri Jun  9 14:23:44 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Fri, 9 Jun 2006 14:23:44 -0400
Subject: [cisco-voip] Unity Java authentication
Message-ID: <FE2D092F8B1516488B99342551AACFD41C634114@mlnusa-exch1.mlnusa.com>

I have been accessing Unity using an IP address.  In the past I was able
to access it without a problem.  Other people in my organization have no
problem accessing it via IP and are able to play and change recordings.

________________________________

From: Scott Coner (US) [mailto:scott.coner at us.didata.com] 
Sent: Friday, June 09, 2006 1:58 PM
To: Olson, Jeffrey; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Unity Java authentication


Are you accessing Unity using an IP address or a host name?  Can your PC
properly resolve the host name to an IP address?  There was a similar
issue to this with 4.0(3) and name resolution but I thought it was
resolved in 4.0(4).

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Olson, Jeffrey
Sent: Fri 6/9/2006 1:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Java authentication



I am running Unity 4.0 Build 4.0(4).  When I log in to Unity I am able
to authenticate with the first authentication window but when the second
window comes up (I believe it is a Java window) I am not able to
authenticate.  I am able to authenticate from any other PCs in my
organization but not mine.  By clicking cancel on this authentication
window I am not able to play recordings or record scripts.  I have
uninstalled Java and reinstalled it on my PC and it didn't correct the
issue.  Anyone have any ideas how I can correct this problem?


Jeff 

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders
Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
hereby notified that disclosing, copying, distributing, or taking any
action in reliance on the contents of this information is strictly
prohibited, and that the E-mail should be returned to this company
immediately. If you have received this E-mail in error, please contact
the sender immediately.

 

________________________________



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 bcarter at sentinel.com  Fri Jun  9 14:24:45 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Fri, 9 Jun 2006 13:24:45 -0500
Subject: [cisco-voip] Metreos SCCP Proxy
Message-ID: <6A0744639B86DD459433108DBF101D26FC620E@MAIL01.sentinel.com>

 
An thought or experiences with the Metreos SCCP Proxy server.  We are
thinking of placing IP Phones in our home offices and using the SCCP
proxy instead of DMVPN.


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 Matt.Slaga at us.didata.com  Fri Jun  9 14:28:38 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 9 Jun 2006 14:28:38 -0400
Subject: [cisco-voip] Metreos SCCP Proxy
Message-ID: <C1FE15183DA37645BC0633BC604E44F00223C83B@USNAEXCH.na.didata.local>

FYI...  Metreos will soon be a Cisco part #:

Cisco buys communications software developer tools 

By Marguerite Reardon
http://news.com.com/Cisco+buys+communications+software+developer+tools/2
100-1033_3-6081822.html 

Story last modified Thu Jun 08 23:39:59 PDT 2006 



Cisco Systems plans to announce Friday that it's spending $47.8 million
in cash to acquire two start-ups whose technology will help large
companies merge their business software applications with their IP
communications platforms. 
Cisco said it will pay $28 million in cash for privately held Metreos,
an Austin, Texas-based company with 19 employees. It will dish out
another $19.8 million in cash for a 26-person company called Audium
based in New York City.

Software developed by Metreos and Audium will be used by Cisco's
third-party technology partners, systems integrators, value-added
resellers and customers to build and deliver applications on Cisco's
Unified Communications System, a suite of voice, data and video products
and applications that integrate communications with IT infrastructure.

"The technology from these companies act as middleware linking the
business applications with communications so their business processes
can be connected to actual people," said Zoltan Poleretzky, director of
business development for Cisco's Unified Communications products. "The
great thing about these tools is that no one has to be a Java developer
to do it."

For example, a company's enterprise resource planning software could be
linked to its unified communications infrastructure so that if an issue
comes up in its supply chain it can notify all the appropriate people
simultaneously and immediately.

When the deals close, Metreos and Audium products groups will be
integrated into Cisco's Voice Technology Group, reporting to Don
Proctor, senior vice president of Cisco's Voice Technology Group. The
acquisitions are expected to close by the end of June. 

Cisco has already been working separately with each company. It has
resold Audium's product for more than a year, and Metrios is already a
member of Cisco's developer program.



Copyright (c)1995-2006 CNET Networks, Inc. All rights reserved. 

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Friday, June 09, 2006 2:25 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Metreos SCCP Proxy

 
An thought or experiences with the Metreos SCCP Proxy server.  We are
thinking of placing IP Phones in our home offices and using the SCCP
proxy instead of DMVPN.


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 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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This e-mail communication and any attachments may contain
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From lelio at uoguelph.ca  Fri Jun  9 14:32:20 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 9 Jun 2006 14:32:20 -0400
Subject: [cisco-voip] Metreos SCCP Proxy
References: <6A0744639B86DD459433108DBF101D26FC620E@MAIL01.sentinel.com>
Message-ID: <00a601c68bf3$0b20ca90$47196883@cfs.uoguelph.ca>

We are seriously considering the SCCP proxy for home offices as well where there is only one PC and one phone. Anything else and a small router might be better. We wanted to use it for softphone from off-site as well as wireless phones. They were not eager to provide any demo/loaner, but now that Cisco is going to buy them, we might get better response. Let's just hope Cisco doesn't can the product. I doubt they will, supposedly they have a client with a large install base already using it. It was a custom app that they decided to market. 

http://newsroom.cisco.com/dlls/2006/corp_060806c.html?CMP=ILC-001


--------------------------------------------------------------------------------
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: Friday, June 09, 2006 2:24 PM
  Subject: [cisco-voip] Metreos SCCP Proxy



  An thought or experiences with the Metreos SCCP Proxy server.  We are
  thinking of placing IP Phones in our home offices and using the SCCP
  proxy instead of DMVPN.


  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.

  _______________________________________________
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  cisco-voip at puck.nether.net
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From lelio at uoguelph.ca  Fri Jun  9 15:23:27 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 9 Jun 2006 15:23:27 -0400
Subject: [cisco-voip] PVDMs! What are they good for?
Message-ID: <012901c68bfa$2f5f56e0$47196883@cfs.uoguelph.ca>

Can anyone explain briefly what the PVDMs are used for on the routers? Do I need these if I am going to be running T1s, FXOs and SRST or do I need them just for conferencing?

--------------------------------------------------------------------------------
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 Greg_Hinson at adp.com  Fri Jun  9 16:40:01 2006
From: Greg_Hinson at adp.com (Hinson, Greg)
Date: Fri, 9 Jun 2006 15:40:01 -0500
Subject: [cisco-voip] PVDMs! What are they good for?
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC05F1F@EXCHANGEDS01.ds.ad.adp.com>

PVDM's are your dsp resources that are used for transcoding and
conference bridging. Depending on the hardware installed, you'll either
have PVDM or PVDMII modules. (PVDMII has twice the capacity)
 
-Greg
 
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, June 09, 2006 12:23 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PVDMs! What are they good for?
 
Can anyone explain briefly what the PVDMs are used for on the routers?
Do I need these if I am going to be running T1s, FXOs and SRST or do I
need them just for conferencing?
 
------------------------------------------------------------------------
--------
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 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 MLinsemier at apcapital.com  Fri Jun  9 17:06:23 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Fri, 9 Jun 2006 17:06:23 -0400
Subject: [cisco-voip] 7960 Phone - connected via the 10/100PC
Message-ID: <AB0A35F2486F81468971AA68558BE71903EDA3D6@smieln01winm01.micoanet.com>

I was wondering if anyone knew the potential effects of a phone running
connected to the 10/100PC port rather then the correct 10/100SW port.
We are in the second phase of our Teleworker rollout (replacing the old
routers with new Cisco 870's) and I am finding that our helpdesk that
originally helped many users setup their home setup had their phones
connected to the older 831/837 routers via their 10/100PC port.  These
happen to be some of the same people that were also complaining about
voice quality (even though the policy-maps showed that voice was being
prioritized correctly).  Has anyone run across this? 

 

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

 


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

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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 lelio at uoguelph.ca  Fri Jun  9 17:13:35 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 9 Jun 2006 17:13:35 -0400
Subject: [cisco-voip] 7960 Phone - connected via the 10/100PC
References: <AB0A35F2486F81468971AA68558BE71903EDA3D6@smieln01winm01.micoanet.com>
Message-ID: <001301c68c09$919d5780$47196883@cfs.uoguelph.ca>

I heard that there was a hospital out there that used both ports of the phone plugged into a switch and used this as a backup link configuration along with the power cube.

Totally unsupported I believe as well.

--------------------------------------------------------------------------------
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: Linsemier, Matthew 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, June 09, 2006 5:06 PM
  Subject: [cisco-voip] 7960 Phone - connected via the 10/100PC


  I was wondering if anyone knew the potential effects of a phone running connected to the 10/100PC port rather then the correct 10/100SW port.  We are in the second phase of our Teleworker rollout (replacing the old routers with new Cisco 870's) and I am finding that our helpdesk that originally helped many users setup their home setup had their phones connected to the older 831/837 routers via their 10/100PC port.  These happen to be some of the same people that were also complaining about voice quality (even though the policy-maps showed that voice was being prioritized correctly).  Has anyone run across this? 

   

  Matthew M. Linsemier, CCSP/CCNP/CCDA
  Senior Network Engineer
  APCapital
  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.



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From Joe_Bell at adp.com  Fri Jun  9 17:34:33 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Fri, 9 Jun 2006 16:34:33 -0500
Subject: [cisco-voip] 7960 Phone - connected via the 10/100PC
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC05F92@EXCHANGEDS01.ds.ad.adp.com>

The major difference between the two ports is the PC port does not
accept PoE.  The Cisco Troubleshooting handbook (version 1, pg 164)
states the PC port can be used for switch redundancy (use a cross over
cable from PC port to he second switch) so that should the PoE switch
fails, you still have network access.  You need an external power supply
for that option.  

 

How does loop avoidance work in that scenario?  Well the Phone switch
does not run STP, but it does pass BPDU's, so the switches perform the
blocking forwarding etc.

 

Joe

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Friday, June 09, 2006 2:14 PM
To: Linsemier, Matthew; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7960 Phone - connected via the 10/100PC

 

I heard that there was a hospital out there that used both ports of the
phone plugged into a switch and used this as a backup link configuration
along with the power cube.

 

Totally unsupported I believe as well.

 

------------------------------------------------------------------------
--------
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: Linsemier, Matthew <mailto:MLinsemier at apcapital.com>  

	To: cisco-voip at puck.nether.net 

	Sent: Friday, June 09, 2006 5:06 PM

	Subject: [cisco-voip] 7960 Phone - connected via the 10/100PC

	 

	I was wondering if anyone knew the potential effects of a phone
running connected to the 10/100PC port rather then the correct 10/100SW
port.  We are in the second phase of our Teleworker rollout (replacing
the old routers with new Cisco 870's) and I am finding that our helpdesk
that originally helped many users setup their home setup had their
phones connected to the older 831/837 routers via their 10/100PC port.
These happen to be some of the same people that were also complaining
about voice quality (even though the policy-maps showed that voice was
being prioritized correctly).  Has anyone run across this? 

	 

	Matthew M. Linsemier, CCSP/CCNP/CCDA
	Senior Network Engineer
	APCapital
	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.

	
________________________________

	
________________________________


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



This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From ErasmuE4 at telkom.co.za  Fri Jun  9 17:44:35 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Fri, 9 Jun 2006 23:44:35 +0200
Subject: [cisco-voip] blocking a specific unwanted inbound call
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F7AF73E@CNTRRA20-XCS01.telkom.co.za>


I guess it will also depend on the callers ID which can change - might
not always call from the same source







________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joshua Abbey
Sent: 09 June 2006 04:56 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] blocking a specific unwanted inbound call




Is there any way for a CCM 4.0sr2a, with MGCP 2651xm gateways to
selectively block or drop calls inbound from a specific phone number?
We've been receiving unwanted calls from an ex-employee, just calling
and hanging up, calling and hanging up, and it is apparently not
threatening enough for the police to do anything.  Does anyone know if
this is doable?

Regards,
-jla

Joshua L. Abbey
Director - IT & Network Technology
Regatta USA LLC
212.589-5239 (o)
908.601.5993 (c)
212-827-3039 (f)



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From JAbbey at regattausa.com  Fri Jun  9 17:45:10 2006
From: JAbbey at regattausa.com (Joshua Abbey)
Date: Fri, 9 Jun 2006 17:45:10 -0400
Subject: [cisco-voip] blocking a specific unwanted inbound call
Message-ID: <OFBC7BBB94.48937310-ON85257188.00777DFD@regattausa.com>

He is, for the moment, calling from the same number

 
Regards,
-jla
Director, IT & Network Technology
Regatta USA, LLC
212.589.5239 (o)
908.601.5993 (c)
212.827.3039 (f)


----- Original Message -----
From: "Erik Erasmus \(E\)" [ErasmuE4 at telkom.co.za]
Sent: 06/09/2006 05:44 PM
To: Joshua Abbey; <cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] blocking a specific unwanted inbound call




I guess it will also depend on the callers ID which can change - might
not always call from the same source










________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joshua Abbey
Sent: 09 June 2006 04:56 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] blocking a specific unwanted inbound call





Is there any way for a CCM 4.0sr2a, with MGCP 2651xm gateways to
selectively block or drop calls inbound from a specific phone number?
We've been receiving unwanted calls from an ex-employee, just calling
and hanging up, calling and hanging up, and it is apparently not
threatening enough for the police to do anything.  Does anyone know if
this is doable?


Regards,
-jla

Joshua L. Abbey
Director - IT & Network Technology
Regatta USA LLC
212.589-5239 (o)
908.601.5993 (c)
212-827-3039 (f)



~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at

http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
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From ErasmuE4 at telkom.co.za  Fri Jun  9 17:50:34 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Fri, 9 Jun 2006 23:50:34 +0200
Subject: [cisco-voip] PVDMs! What are they good for?
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F7AF73F@CNTRRA20-XCS01.telkom.co.za>


Lelio



I am implementing a solution now where we used PVDMs as follows:



1.	to handle the E1 PRI lines - probably the equivalent of your
T1s.
2.	at remote branches to handle the ISDN basic rate line to the
telco for local breakout



and soon we will have to add DSPS - PVDMs to allow conferencing in
branches and also to HQ across the IP cloud. As far as I know the Call
Manager only supports G711 conferences so if a branch connects to HQ
across an IP wan in a centralised CCM model via G729 you will need
DSPs at HQ to mix the G729 and G711 calls in a conference



also see the DSP calculator on www.cisco.com <http://www.cisco.com/>
- as clear as MUD in some cases as to how many DSPs to buy - but once
you playe with it online and get used to it it is actually very
usefull. DSPs are expensive so it is important to plan and get it
right first time





erik erasmus



________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio
Fulgenzi
Sent: 09 June 2006 09:23 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PVDMs! What are they good for?



Can anyone explain briefly what the PVDMs are used for on the routers?
Do I need these if I am going to be running T1s, FXOs and SRST or do I
need them just for conferencing?



----------------------------------------------------------------------
----------
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 e-mail and its contents are subject to the Telkom SA Limited
e-mail legal notice available at
http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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From ErasmuE4 at telkom.co.za  Fri Jun  9 18:00:45 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Sat, 10 Jun 2006 00:00:45 +0200
Subject: [cisco-voip] Call Manager - MCS7838 100% CPU util
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F7AF741@CNTRRA20-XCS01.telkom.co.za>


Has anyone seen a problem as described here:



I recently worked on a new CCM install. Small network with 200 users.
Was using remote desktop to CCM 4.1.3 SR3a and OS 2000-4-2 and
security agent.



Added a couple of phones in the administration of CCM. When clicking
on the update button one gets the annoying jave proxy applet da da da
... message in fine print at the bottom of the screen and a dely
before the CCM show update completed and allows you to move on. This
time I probably clicked on the next link too fast and things locked
up. I went o the console and noticed tha the CPU util was running at
100%. Stopped the Iexplore service and things normallised. During this
time users started struggling to get dialtone etc because of the 100%
cpu util.



This is BAD and extremely worrying - how can simple administration
cause something like this to happen ?? Any ideas will e appreciated -
do ii need to load any fixes . As far as possible this installation
and the upgrades from 2000-2-7 was done by the book so it is hard to
explain. The hardware is also brand new MCS 7835 H DL380 HP





???



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From ealeatherman at gmail.com  Fri Jun  9 20:16:46 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 9 Jun 2006 20:16:46 -0400
Subject: [cisco-voip] Long Distance woes
Message-ID: <94a1afde0606091716g22c48512i3828ecdb90c4f0dd@mail.gmail.com>

Was wondering if anyone is familiar with how long distance carriers/service
interacts with local service...

Working with service provider at one of our branch campuses which we just
converted to a callmanager system this past week. Users can call long
distance numbers within our area code (304) just fine, but cannot call
outside of the area code. The dial pattern is the same (9.1[2-9]XX[2-9]XX
XXXX) for either case (exact same route pattern actually), I just strip the
9 and send 1 + 10 digits to the phone company. Can see this in CCM trace.
System uses 2 PRI T1's.

If users call an out of state number, they get fast busy. The local carrier
is looking at the problem now but initially they just told me it wasnt there
problem, as they just pass the digits to the long distance carrier if the
number starts with "1". It seems to me that maybe the lines are setup
incorrectly as far as what long distance provider it should use.. but I dont
know enough about the process to make a good guess.. I've not worked with
this particular local service provider before so I dont have the same
"faith" if you could call it that, as I do with the carrier we have at the
main campus which I have dealt with before.

Caller ID also does not come across, but I think this is a seperate issue
which we can hopefully address after the long distance is working.

Any ideas? I'm curious if there is anything more I can look at from my end.

-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From abdul_zu at yahoo.com  Sat Jun 10 01:39:03 2006
From: abdul_zu at yahoo.com (Abdul Lateef)
Date: Fri, 9 Jun 2006 22:39:03 -0700 (PDT)
Subject: [cisco-voip] Hardware Suggestion
Message-ID: <20060610053903.14732.qmail@web34509.mail.mud.yahoo.com>

Hi All,

I would to know your suggestion for cisco hardware to
work well as VoIP Gatekeeper with fully SIP
supporting.
Here is a small feautre which we are looking from
cisco gatekeeper.


1- 10,000 Cuncurent calls
2- SIP Proxy Server
3- H323 AAA
4- NAT Client Support
5- Call Control features (Maximum Talk Time, etc)
6- Call Routing Managment
7- Unlimited RAS registration
8- Fully Radius Support

Some of my friend suggest to go with 2651XM, Please
advice me there any other powerfull hardware is
avialable from cisco?

Regards,
Abdul Lateef


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

From tgeorgiev at is-bg.net  Sat Jun 10 04:33:23 2006
From: tgeorgiev at is-bg.net (Teodor Georgiev)
Date: Sat, 10 Jun 2006 11:33:23 +0300
Subject: [cisco-voip] Hardware Suggestion
References: <20060610053903.14732.qmail@web34509.mail.mud.yahoo.com>
Message-ID: <002401c68c68$8994de00$978e4bd9@tg>


    Abdul,

1. Your feature list is anything, but "small" and humble :)

2. Your friend is wrong. 2651 will not handle this capacity, neither does
possess the requested features.

3. The Cisco gatekeeper has very basic functions. You better look at Cisco
IP-IP gateway. Which is not
really a "gateway", but a special IOS image (simply said).

4. I would suggest you to look at carrier-grade softswitches like MERA MVTS,
NexTone, NetCentrex or Teles.
   But even with them is a challenge to meet the capacity requested by you
(10,000 calls)

----- Original Message -----
From: "Abdul Lateef" <abdul_zu at yahoo.com>
To: <cisco-voip at puck.nether.net>
Sent: Saturday, June 10, 2006 8:39 AM
Subject: [cisco-voip] Hardware Suggestion


> Hi All,
>
> I would to know your suggestion for cisco hardware to
> work well as VoIP Gatekeeper with fully SIP
> supporting.
> Here is a small feautre which we are looking from
> cisco gatekeeper.
>
>
> 1- 10,000 Cuncurent calls
> 2- SIP Proxy Server
> 3- H323 AAA
> 4- NAT Client Support
> 5- Call Control features (Maximum Talk Time, etc)
> 6- Call Routing Managment
> 7- Unlimited RAS registration
> 8- Fully Radius Support
>
> Some of my friend suggest to go with 2651XM, Please
> advice me there any other powerfull hardware is
> avialable from cisco?
>
> Regards,
> Abdul Lateef
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> __________ NOD32 1.1523 (20060505) Information __________
>
> This message was checked by NOD32 antivirus system.
> http://www.nod32.com
>
>


From jim at tgasolutions.com  Sat Jun 10 09:05:46 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Sat, 10 Jun 2006 09:05:46 -0400
Subject: [cisco-voip] PVDMs! What are they good for?
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC15E54E@exchange.TGASolutions.local>

Try this:
http://www.cisco.com/en/US/products/hw/modules/ps3115/products_qanda_ite
m0900aecd8016c6ad.shtml
 
or 
http://tinyurl.com/hsnsu
 
Later,
J

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
Sent: Friday, June 09, 2006 5:51 PM
To: Lelio Fulgenzi; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PVDMs! What are they good for?



Lelio

 

I am implementing a solution now where we used PVDMs as follows:

 

1.	to handle the E1 PRI lines - probably the equivalent of your
T1s. 
2.	at remote branches to handle the ISDN basic rate line to the
telco for local breakout 

 

and soon we will have to add DSPS - PVDMs to allow conferencing in
branches and also to HQ across the IP cloud. As far as I know the Call
Manager only supports G711 conferences so if a branch connects to HQ
across an IP wan in a centralised CCM model via G729 you will need DSPs
at HQ to mix the G729 and G711 calls in a conference

 

also see the DSP calculator on www.cisco.com <http://www.cisco.com/>  -
as clear as MUD in some cases as to how many DSPs to buy - but once you
playe with it online and get used to it it is actually very usefull.
DSPs are expensive so it is important to plan and get it right first
time

 

 

erik erasmus

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: 09 June 2006 09:23 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] PVDMs! What are they good for?

 

Can anyone explain briefly what the PVDMs are used for on the routers?
Do I need these if I am going to be running T1s, FXOs and SRST or do I
need them just for conferencing?

 

------------------------------------------------------------------------
--------
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 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 lelio at uoguelph.ca  Sat Jun 10 09:23:07 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Sat, 10 Jun 2006 09:23:07 -0400
Subject: [cisco-voip] PVDMs! What are they good for?
References: <D2BBAE709B4D874DAC556C7EB9D938FC15E54E@exchange.TGASolutions.local>
Message-ID: <00a701c68c91$032dd640$c801a8c0@dellectable>

Thanks - that's what I was looking for to help me understand.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network 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: Jim McBurnett 
  To: Erik Erasmus (E) ; Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Saturday, June 10, 2006 9:05 AM
  Subject: RE: [cisco-voip] PVDMs! What are they good for?


  Try this:
  http://www.cisco.com/en/US/products/hw/modules/ps3115/products_qanda_item0900aecd8016c6ad.shtml

  or 
  http://tinyurl.com/hsnsu

  Later,
  J



------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus (E)
  Sent: Friday, June 09, 2006 5:51 PM
  To: Lelio Fulgenzi; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] PVDMs! What are they good for?


  Lelio

   

  I am implementing a solution now where we used PVDMs as follows:

   

    1.. to handle the E1 PRI lines - probably the equivalent of your T1s. 
    2.. at remote branches to handle the ISDN basic rate line to the telco for local breakout 
   

  and soon we will have to add DSPS - PVDMs to allow conferencing in branches and also to HQ across the IP cloud. As far as I know the Call Manager only supports G711 conferences so if a branch connects to HQ across an IP wan in a centralised CCM model via G729 you will need DSPs at HQ to mix the G729 and G711 calls in a conference

   

  also see the DSP calculator on www.cisco.com - as clear as MUD in some cases as to how many DSPs to buy - but once you playe with it online and get used to it it is actually very usefull. DSPs are expensive so it is important to plan and get it right first time

   

   

  erik erasmus

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: 09 June 2006 09:23 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] PVDMs! What are they good for?

   

  Can anyone explain briefly what the PVDMs are used for on the routers? Do I need these if I am going to be running T1s, FXOs and SRST or do I need them just for conferencing?

   

  --------------------------------------------------------------------------------
  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 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  Sat Jun 10 13:06:13 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Sat, 10 Jun 2006 13:06:13 -0400
Subject: [cisco-voip] blocking a specific unwanted inbound call
In-Reply-To: <OFBC7BBB94.48937310-ON85257188.00777DFD@regattausa.com>
References: <OFBC7BBB94.48937310-ON85257188.00777DFD@regattausa.com>
Message-ID: <3039806322777429a66237b1a7ad13c0@cisco.com>

Try creating a Route Pattern in CCM with the specific pattern that 
matches the caller id of the unwanted call.  Change the Route Pattern 
to block the call.  You will need to select a GW but the call will not 
go to that GW, the call will be disconnected.  Make sure the Route 
Pattern is in a Partition the GW can call.  Be careful that the 
Partition is not callable from the phone's CSS as it would block your 
users from calling that person (maybe that is desirable too :-)

Kevin
On Jun 9, 2006, at 5:45 PM, Joshua Abbey wrote:

> He is, for the moment, calling from the same number
>
> Regards,
> -jla
> Director, IT & Network Technology
> Regatta USA, LLC
> 212.589.5239 (o)
> 908.601.5993 (c)
> 212.827.3039 (f)
>
> ?----- Original Message -----
> ? From: "Erik Erasmus \(E\)" [ErasmuE4 at telkom.co.za]
> ? Sent: 06/09/2006 05:44 PM
> ? To: Joshua Abbey; <cisco-voip at puck.nether.net>
> ? Subject: RE: [cisco-voip] blocking a specific unwanted inbound call
>
>
> I guess it will also depend on the callers ID which can change ? might 
> not always call from the same source
> ?
> ?
> ?
>
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joshua Abbey
> Sent: 09 June 2006 04:56 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] blocking a specific unwanted inbound call
> ?
>
> Is there any way for a CCM 4.0sr2a, with MGCP 2651xm gateways to 
> selectively block or drop calls inbound from a specific phone number? 
> ?We've been receiving unwanted calls from an ex-employee, just calling 
> and hanging up, calling and hanging up, and it is apparently not 
> threatening enough for the police to do anything. ?Does anyone know if 
> this is doable?
>
> Regards,
>  -jla
>
>  Joshua L. Abbey
>  Director - IT & Network Technology
>  Regatta USA LLC
>  212.589-5239 (o)
>  908.601.5993 (c)
>  212-827-3039 (f)
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>  This e-mail and its contents are subject to the Telkom SA Limited
>  e-mail legal notice available at
>  http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
>  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From Brian.Henry at apptis.com  Sat Jun 10 13:49:53 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Sat, 10 Jun 2006 13:49:53 -0400
Subject: [cisco-voip] blocking a specific unwanted inbound call
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC0283821F@CH-EXCH-02.intelltech.apptis.inc>

If you have an H323 gateway you can use answer-address to match upon the ANI on a dial-peer and then drop the call.
 
Route-Patterns ONLY match upon DNIS, the number that you are calling and cannot be done in MGCP.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Kevin Thorngren
Sent: Sat 6/10/2006 1:06 PM
To: Joshua Abbey
Cc: cisco-voip
Subject: Re: [cisco-voip] blocking a specific unwanted inbound call


Try creating a Route Pattern in CCM with the specific pattern that matches the caller id of the unwanted call. Change the Route Pattern to block the call. You will need to select a GW but the call will not go to that GW, the call will be disconnected. Make sure the Route Pattern is in a Partition the GW can call. Be careful that the Partition is not callable from the phone's CSS as it would block your users from calling that person (maybe that is desirable too :-) 

Kevin 
On Jun 9, 2006, at 5:45 PM, Joshua Abbey wrote: 


	He is, for the moment, calling from the same number 

	Regards, 
	-jla 
	Director, IT & Network Technology 
	Regatta USA, LLC 
	212.589.5239 (o) 
	908.601.5993 (c) 
	212.827.3039 (f) 

	 ----- Original Message ----- 
	  From: "Erik Erasmus \(E\)" [ErasmuE4 at telkom.co.za] 
	  Sent: 06/09/2006 05:44 PM 
	  To: Joshua Abbey; <cisco-voip at puck.nether.net> 
	  Subject: RE: [cisco-voip] blocking a specific unwanted inbound call 

	I guess it will also depend on the callers ID which can change - might not always call from the same source 
	
	
	

	From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Joshua Abbey 
	Sent: 09 June 2006 04:56 PM 
	To: cisco-voip at puck.nether.net 
	Subject: [cisco-voip] blocking a specific unwanted inbound call 
	

	Is there any way for a CCM 4.0sr2a, with MGCP 2651xm gateways to selectively block or drop calls inbound from a specific phone number?  We've been receiving unwanted calls from an ex-employee, just calling and hanging up, calling and hanging up, and it is apparently not threatening enough for the police to do anything.  Does anyone know if this is doable? 

	Regards, 
	-jla 

	Joshua L. Abbey 
	Director - IT & Network Technology 
	Regatta USA LLC 
	212.589-5239 (o) 
	908.601.5993 (c) 
	212-827-3039 (f) 
	~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 
	This e-mail and its contents are subject to the Telkom SA Limited 
	e-mail legal notice available at 
	http://www.telkom.co.za/TelkomEMailLegalNotice.PDF 
	~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 
	_______________________________________________ 
	cisco-voip mailing list 
	cisco-voip at puck.nether.net 
	https://puck.nether.net/mailman/listinfo/cisco-voip 

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From jim at tgasolutions.com  Sat Jun 10 18:31:46 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Sat, 10 Jun 2006 18:31:46 -0400
Subject: [cisco-voip] PVDMs! What are they good for?
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC15E54F@exchange.TGASolutions.local>

Glad I could help..
 
Later,
Jim

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Saturday, June 10, 2006 9:23 AM
To: Jim McBurnett; Erik Erasmus (E); cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] PVDMs! What are they good for?


Thanks - that's what I was looking for to help me understand.
 
------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network 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: Jim McBurnett <mailto:jim at tgasolutions.com>  
	To: Erik Erasmus (E) <mailto:ErasmuE4 at telkom.co.za>  ; Lelio
Fulgenzi <mailto:lelio at uoguelph.ca>  ; cisco-voip at puck.nether.net 
	Sent: Saturday, June 10, 2006 9:05 AM
	Subject: RE: [cisco-voip] PVDMs! What are they good for?

	Try this:
	
http://www.cisco.com/en/US/products/hw/modules/ps3115/products_qanda_ite
m0900aecd8016c6ad.shtml
	 
	or 
	http://tinyurl.com/hsnsu
	 
	Later,
	J

________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Erik Erasmus
(E)
	Sent: Friday, June 09, 2006 5:51 PM
	To: Lelio Fulgenzi; cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] PVDMs! What are they good for?
	
	

	Lelio

	 

	I am implementing a solution now where we used PVDMs as follows:

	 

	1.	to handle the E1 PRI lines - probably the equivalent of
your T1s. 
	2.	at remote branches to handle the ISDN basic rate line to
the telco for local breakout 

	 

	and soon we will have to add DSPS - PVDMs to allow conferencing
in branches and also to HQ across the IP cloud. As far as I know the
Call Manager only supports G711 conferences so if a branch connects to
HQ across an IP wan in a centralised CCM model via G729 you will need
DSPs at HQ to mix the G729 and G711 calls in a conference

	 

	also see the DSP calculator on www.cisco.com
<http://www.cisco.com/>  - as clear as MUD in some cases as to how many
DSPs to buy - but once you playe with it online and get used to it it is
actually very usefull. DSPs are expensive so it is important to plan and
get it right first time

	 

	 

	erik erasmus

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	Sent: 09 June 2006 09:23 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] PVDMs! What are they good for?

	 

	Can anyone explain briefly what the PVDMs are used for on the
routers? Do I need these if I am going to be running T1s, FXOs and SRST
or do I need them just for conferencing?

	 

	
------------------------------------------------------------------------
--------
	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 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 kburns at oneipvoice.com  Sat Jun 10 18:32:24 2006
From: kburns at oneipvoice.com (Keith Burns)
Date: Sat, 10 Jun 2006 16:32:24 -0600
Subject: [cisco-voip] blocking a specific unwanted inbound call
In-Reply-To: <2F09B2162CCDEA4EA05C089B24FBC2BC0283821F@CH-EXCH-02.intelltech.apptis.inc>
Message-ID: <C0B0A498.AD9%kburns@oneipvoice.com>

I think you need to think through the legal implications.

You are better off going through your upstream provider and filing the
appropriate grievances.

A more appropriate solution would be to allow those calls in but forward
them to a specific endpoint (SCCP ?) answered by your legal dept.

IMHO.




On 6/10/06 11:49 AM, "Brian Henry" <Brian.Henry at apptis.com> wrote:

> If you have an H323 gateway you can use answer-address to match upon the ANI
> on a dial-peer and then drop the call.
>  
> Route-Patterns ONLY match upon DNIS, the number that you are calling and
> cannot be done in MGCP.
>  
> Brian
> 
> 
> From: cisco-voip-bounces at puck.nether.net on behalf of Kevin Thorngren
> Sent: Sat 6/10/2006 1:06 PM
> To: Joshua Abbey
> Cc: cisco-voip
> Subject: Re: [cisco-voip] blocking a specific unwanted inbound call
> 
> Try creating a Route Pattern in CCM with the specific pattern that matches the
> caller id of the unwanted call. Change the Route Pattern to block the call.
> You will need to select a GW but the call will not go to that GW, the call
> will be disconnected. Make sure the Route Pattern is in a Partition the GW can
> call. Be careful that the Partition is not callable from the phone's CSS as it
> would block your users from calling that person (maybe that is desirable too
> :-) 
> 
> Kevin 
> On Jun 9, 2006, at 5:45 PM, Joshua Abbey wrote:
> 
>>  
>> He is, for the moment, calling from the same number
>> 
>>  
>> Regards, 
>>  
>> -jla 
>>  
>> Director, IT & Network Technology
>>  
>> Regatta USA, LLC
>>  
>> 212.589.5239 (o)
>>  
>> 908.601.5993 (c)
>>  
>> 212.827.3039 (f)
>> 
>>  
>>  ----- Original Message -----
>>  
>>   From: "Erik Erasmus \(E\)"  [ErasmuE4 at telkom.co.za]
>>  
>>   Sent: 06/09/2006 05:44 PM
>>  
>>   To: Joshua Abbey;  <cisco-voip at puck.nether.net>
>>  
>>   Subject: RE: [cisco-voip] blocking a  specific unwanted inbound call
>> 
>>  
>>  
>> I guess it will also depend on the  callers ID which can change ? might not
>> always call from the same  source
>>  
>>  
>>  
>> 
>>  
>> From:  cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net]  On Behalf Of Joshua Abbey
>>  
>> Sent: 09 June 2006  04:56 PM
>>  
>> To:  cisco-voip at puck.nether.net
>>  
>> Subject: [cisco-voip]  blocking a specific unwanted inbound call
>>  
>> 
>>  
>> Is there any way for a CCM 4.0sr2a, with MGCP 2651xm gateways to  selectively
>> block or drop calls inbound from a specific phone number?   We've been
>> receiving unwanted calls from an ex-employee, just calling  and hanging up,
>> calling and hanging up, and it is apparently not threatening  enough for the
>> police to do anything.  Does anyone know if this is  doable?
>> 
>>  
>> Regards, 
>>  
>> -jla 
>> 
>>  
>> Joshua L. Abbey 
>>  
>> Director - IT & Network Technology
>>  
>> Regatta USA LLC 
>>  
>> 212.589-5239 (o)
>>  
>> 908.601.5993 (c)
>>  
>> 212-827-3039 (f)
>>  
>> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>>  
>> This e-mail and its contents are subject to the Telkom SA Limited
>>  
>> e-mail legal notice available at
>>  
>> http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
>>  
>> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>>  
>> _______________________________________________
>>  
>> cisco-voip mailing list
>>  
>> cisco-voip at puck.nether.net
>>  
>> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip






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From kburns at oneipvoice.com  Sat Jun 10 18:38:42 2006
From: kburns at oneipvoice.com (Keith Burns)
Date: Sat, 10 Jun 2006 16:38:42 -0600
Subject: [cisco-voip] Hardware Suggestion
In-Reply-To: <002401c68c68$8994de00$978e4bd9@tg>
Message-ID: <C0B0A612.ADE%kburns@oneipvoice.com>

Agreed.

H323 and SIP ? If you are doing H323 O&T and translating back to SIP I would
recommend an SBC (look at ACME Packets, very solid... Stay away from
Kagoor/Juniper in my humble opinion).

10,000 concurrent calls is probably too much for CM from anecdotal evidence.

What are you trying to do? I would suggest a carrier-grade feature server
like Sylantro or Broadsoft for something of this magnitude.

Is it possible to stay full SIP to the endpoints? Or are you doing a
PSTN-access application for H323 enterprise call servers?


On 6/10/06 2:33 AM, "Teodor Georgiev" <tgeorgiev at is-bg.net> wrote:

> 
>     Abdul,
> 
> 1. Your feature list is anything, but "small" and humble :)
> 
> 2. Your friend is wrong. 2651 will not handle this capacity, neither does
> possess the requested features.
> 
> 3. The Cisco gatekeeper has very basic functions. You better look at Cisco
> IP-IP gateway. Which is not
> really a "gateway", but a special IOS image (simply said).
> 
> 4. I would suggest you to look at carrier-grade softswitches like MERA MVTS,
> NexTone, NetCentrex or Teles.
>    But even with them is a challenge to meet the capacity requested by you
> (10,000 calls)
> 
> ----- Original Message -----
> From: "Abdul Lateef" <abdul_zu at yahoo.com>
> To: <cisco-voip at puck.nether.net>
> Sent: Saturday, June 10, 2006 8:39 AM
> Subject: [cisco-voip] Hardware Suggestion
> 
> 
>> Hi All,
>> 
>> I would to know your suggestion for cisco hardware to
>> work well as VoIP Gatekeeper with fully SIP
>> supporting.
>> Here is a small feautre which we are looking from
>> cisco gatekeeper.
>> 
>> 
>> 1- 10,000 Cuncurent calls
>> 2- SIP Proxy Server
>> 3- H323 AAA
>> 4- NAT Client Support
>> 5- Call Control features (Maximum Talk Time, etc)
>> 6- Call Routing Managment
>> 7- Unlimited RAS registration
>> 8- Fully Radius Support
>> 
>> Some of my friend suggest to go with 2651XM, Please
>> advice me there any other powerfull hardware is
>> avialable from cisco?
>> 
>> Regards,
>> Abdul Lateef
>> 
>> 
>> __________________________________________________
>> Do You Yahoo!?
>> Tired of spam?  Yahoo! Mail has the best spam protection around
>> http://mail.yahoo.com
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> 
>> __________ NOD32 1.1523 (20060505) Information __________
>> 
>> This message was checked by NOD32 antivirus system.
>> http://www.nod32.com
>> 
>> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 



Keith Burns | Director of Product Development | OneIPVoice | 720.891.4279




From kburns at oneipvoice.com  Sat Jun 10 18:43:03 2006
From: kburns at oneipvoice.com (Keith Burns)
Date: Sat, 10 Jun 2006 16:43:03 -0600
Subject: [cisco-voip] PVDMs! What are they good for?
In-Reply-To: <472E195E692EE74BAE4E7B2880A78C1F7AF73F@CNTRRA20-XCS01.telkom.co.za>
Message-ID: <C0B0A717.AE0%kburns@oneipvoice.com>

Perhaps look at convedia ? Media translation is a hairy business... You will
find that you need to stay on top of the user use cases to make sure a G729
caller tries to call a G711 call without hitting a translator.


On 6/9/06 3:50 PM, "Erik Erasmus (E)" <ErasmuE4 at telkom.co.za> wrote:

> Lelio
>  
> I am implementing a solution now where we used PVDMs as follows:
>  
> 1. to handle the E1 PRI lines ? probably the equivalent of your T1s.
> 2. at remote branches to handle the ISDN basic rate line to the telco for
> local breakout 
>  
> and soon we will have to add DSPS ? PVDMs to allow conferencing in branches
> and also to HQ across the IP cloud. As far as I know the Call Manager only
> supports G711 conferences so if a branch connects to HQ across an IP wan in a
> centralised CCM model via G729 you will need DSPs at HQ to mix the G729 and
> G711 calls in a conference
>  
> also see the DSP calculator on www.cisco.com <http://www.cisco.com/>  ? as
> clear as MUD in some cases as to how many DSPs to buy ? but once you playe
> with it online and get used to it it is actually very usefull. DSPs are
> expensive so it is important to plan and get it right first time
>  
>  
> erik erasmus
>  
> 
> 
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
> Sent: 09 June 2006 09:23 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] PVDMs! What are they good for?
>  
> 
> Can anyone explain briefly what the PVDMs are used for on the routers? Do I
> need these if I am going to be running T1s, FXOs and SRST or do I need them
> just for conferencing?
> 
>  
> 
> ------------------------------------------------------------------------------
> --
> 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 e-mail and its contents are subject to the Telkom SA Limited
> e-mail legal notice available at
> http://www.telkom.co.za/TelkomEMailLegalNotice.PDF
> ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip




Keith Burns | Director of Product Development | OneIPVoice | 720.891.4279


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From jonvoip at gmail.com  Sat Jun 10 22:41:40 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 10 Jun 2006 21:41:40 -0500
Subject: [cisco-voip] vmware gsx server
In-Reply-To: <20861400.1149868558357.JavaMail.root@eastrmwml02.mgt.cox.net>
References: <20861400.1149868558357.JavaMail.root@eastrmwml02.mgt.cox.net>
Message-ID: <5d093f9a0606101941o8b7f560w93b00f7c48de4937@mail.gmail.com>

Yes.

To be honest, that is all I have been using... I am running
CallManager 3.3, 4.0, 4.1, 4.2 and 5.0(2) and Unity 4.0(4), 4.0(5),
4.1(1) and 4.2(1), as well as CRA 3.5 and 4.0 all on VMWare GSX 3.1

I also have some test bed client machines (for BearShare and client apps)...

To be honest, I don't know how you guys run labs without VMWare...



Jonathan

On 6/9/06, bababooey at cox.net <bababooey at cox.net> wrote:
> Has anyone utilized the gsx server to run the call mamanger environment in a vmware test bed.
>
> Thanks,
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From newlook110 at yahoo.com  Sun Jun 11 15:36:47 2006
From: newlook110 at yahoo.com (new look)
Date: Sun, 11 Jun 2006 12:36:47 -0700 (PDT)
Subject: [cisco-voip] Auto Attendant Problem
Message-ID: <20060611193647.13097.qmail@web61316.mail.yahoo.com>

Dear All,
  I have CCM and installed Auto Attendant and ever think is ok,
  The problem I have is when any call my company he hear this default auto attendant welcome message and as you know the message asked the caller to enter 1 for entering the ext. no., 2 for entering the name and 0 for operator, and when the customer entered 1 
  The message asked him to enter the ext. no. than pound key (#) and this is very long.
   
  So, I need when the customer call my company he hears the message says welcome
  Please enter the ext. or zero for operator (I already have this message recorded and I know how to upload it to the call manager) But I need to change the procedure of the Auto Attendant for this new message.
   
  Thanks 

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From bzayed80 at yahoo.com  Sun Jun 11 15:47:18 2006
From: bzayed80 at yahoo.com (Bassem Zayed)
Date: Sun, 11 Jun 2006 12:47:18 -0700 (PDT)
Subject: [cisco-voip] Auto Attendant customization
Message-ID: <20060611194718.38080.qmail@web31113.mail.mud.yahoo.com>

Hi all, 
   
  i have cisco call manager v4.02. I successfully installed the auto attendant. it works fine and the calls are forwarded to the right user or phone. 
   
  The problem is when a user who calls my number and the Auto attendant answer, he has to follow the sequence of procedure " first press 1 to enter extention or 2 to enter name or 3 to for operator" . Then if  i pressed 1 i need to press the extention then hte pound key"#". the problem is i need to change this sequence. 
   
  I have created my own custom msg which only a greeting for the caller and to dial the extention number directly ex."160" or the operator "0". I did this because not all the people who calls this number can follow this kind of procedure!!. 
   
  The question is how can i change the steps a caller has to follow when calls and answered by the autoattendant.
   
  Thank you all and have a nice day.
   
    

 __________________________________________________
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From bzayed80 at yahoo.com  Sun Jun 11 16:08:23 2006
From: bzayed80 at yahoo.com (Bassem Zayed)
Date: Sun, 11 Jun 2006 13:08:23 -0700 (PDT)
Subject: [cisco-voip] testing
Message-ID: <20060611200823.75150.qmail@web31110.mail.mud.yahoo.com>

testing
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From Scott.Voll at wesd.org  Sun Jun 11 16:42:59 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Sun, 11 Jun 2006 13:42:59 -0700
Subject: [cisco-voip] IPCC 3.5 to 4.0
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC344E@MREXVS01.bc.broward.cty>
Message-ID: <407055A92CECCB499C922A2D35FC19A605D75A18@Apollo.wesd.org>

sorry, my bad.  4.1
 
On my scripts I have parameters you can change from the application menu.  they came in after the upgrade with no " marks.  if you change the parameter you must have the quote marks.
 
Scott

________________________________

From: Ortiz, Carlos [mailto:CORTIZ at broward.org]
Sent: Fri 6/9/2006 7:56 AM
To: Voll, Scott; Olson, Jeffrey; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCC 3.5 to 4.0



What do you mean parameters need " marks ??  I am looking to upgrade to 4.X as well.  From what I read 4.X requires 4.1(2) CM minimum:

 

Matrix:

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/ccmcomp.htm

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, June 09, 2006 10:33 AM
To: Olson, Jeffrey; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCC 3.5 to 4.0

 

my scripts worked just fine going from 3.5 to 4.0.  the parameters need " marks but that was my only problem.

 

IPCCx (crs, cra, what ever you would like to call it) 4.x works with 4.x I believe.  Unity should not matter.  I'm out of the office.  but if you search CCO for IPCCx metrix you should find all the details.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Olson, Jeffrey
Sent: Fri 6/9/2006 5:59 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC 3.5 to 4.0

I need to upgrade my IPCC to version 4.04 from version 3.5.  I know that the CRA's name in 3.5 is changed to CRS.  Will I be able to open my 3.5 scripts in 4.04?  Also does anyone have a link that defines the upgrade matrix.  (what versions of call manager and unity I need to have in order for my upgrade to be successful)

Thanks 
Jeff 

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

CONFIDENTIALITY NOTICE:

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From Scott.Voll at wesd.org  Sun Jun 11 16:53:18 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Sun, 11 Jun 2006 13:53:18 -0700
Subject: [cisco-voip] Long Distance woes
References: <94a1afde0606091716g22c48512i3828ecdb90c4f0dd@mail.gmail.com>
Message-ID: <407055A92CECCB499C922A2D35FC19A605D75A19@Apollo.wesd.org>

have you tried changing the isdn to national, or unknown or something along those lines?
 
Scott

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Ed Leatherman
Sent: Fri 6/9/2006 5:16 PM
To: ciscovoip
Subject: [cisco-voip] Long Distance woes


Was wondering if anyone is familiar with how long distance carriers/service interacts with local service...

Working with service provider at one of our branch campuses which we just converted to a callmanager system this past week. Users can call long distance numbers within our area code (304) just fine, but cannot call outside of the area code. The dial pattern is the same (9.1[2-9]XX[2-9]XX XXXX) for either case (exact same route pattern actually), I just strip the 9 and send 1 + 10 digits to the phone company. Can see this in CCM trace. System uses 2 PRI T1's.

If users call an out of state number, they get fast busy. The local carrier is looking at the problem now but initially they just told me it wasnt there problem, as they just pass the digits to the long distance carrier if the number starts with "1". It seems to me that maybe the lines are setup incorrectly as far as what long distance provider it should use.. but I dont know enough about the process to make a good guess.. I've not worked with this particular local service provider before so I dont have the same "faith" if you could call it that, as I do with the carrier we have at the main campus which I have dealt with before.

Caller ID also does not come across, but I think this is a seperate issue which we can hopefully address after the long distance is working.

Any ideas? I'm curious if there is anything more I can look at from my end.

-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 
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From hsalama_us at yahoo.com  Sun Jun 11 19:39:10 2006
From: hsalama_us at yahoo.com (Hassan Salama)
Date: Sun, 11 Jun 2006 16:39:10 -0700 (PDT)
Subject: [cisco-voip] Long Distance woes
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605D75A19@Apollo.wesd.org>
Message-ID: <20060611233910.14187.qmail@web34606.mail.mud.yahoo.com>


Hello,

Regarding your issue of long distance call, the number
handling could varies from country to country as this
is telecom standard.

But you may try the translation rule to modify Called
party number type to become national [a must for long
distance within a country], and stip 91 from the
begeinning of the number [i.e. send 10 digits without
1], you may give another try with stipping 9 digit
only

Thanks
 
--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> have you tried changing the isdn to national, or
> unknown or something along those lines?
>  
> Scott
> 
> ________________________________
> 
> From: cisco-voip-bounces at puck.nether.net on behalf
> of Ed Leatherman
> Sent: Fri 6/9/2006 5:16 PM
> To: ciscovoip
> Subject: [cisco-voip] Long Distance woes
> 
> 
> Was wondering if anyone is familiar with how long
> distance carriers/service interacts with local
> service...
> 
> Working with service provider at one of our branch
> campuses which we just converted to a callmanager
> system this past week. Users can call long distance
> numbers within our area code (304) just fine, but
> cannot call outside of the area code. The dial
> pattern is the same (9.1[2-9]XX[2-9]XX XXXX) for
> either case (exact same route pattern actually), I
> just strip the 9 and send 1 + 10 digits to the phone
> company. Can see this in CCM trace. System uses 2
> PRI T1's.
> 
> If users call an out of state number, they get fast
> busy. The local carrier is looking at the problem
> now but initially they just told me it wasnt there
> problem, as they just pass the digits to the long
> distance carrier if the number starts with "1". It
> seems to me that maybe the lines are setup
> incorrectly as far as what long distance provider it
> should use.. but I dont know enough about the
> process to make a good guess.. I've not worked with
> this particular local service provider before so I
> dont have the same "faith" if you could call it
> that, as I do with the carrier we have at the main
> campus which I have dealt with before.
> 
> Caller ID also does not come across, but I think
> this is a seperate issue which we can hopefully
> address after the long distance is working.
> 
> Any ideas? I'm curious if there is anything more I
> can look at from my end.
> 
> -- 
> Ed Leatherman
> IP Telephony Coordinator
> 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
> 


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

From ealeatherman at gmail.com  Sun Jun 11 21:03:31 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Sun, 11 Jun 2006 21:03:31 -0400
Subject: [cisco-voip] Long Distance woes
In-Reply-To: <20060611233910.14187.qmail@web34606.mail.mud.yahoo.com>
References: <407055A92CECCB499C922A2D35FC19A605D75A19@Apollo.wesd.org>
	<20060611233910.14187.qmail@web34606.mail.mud.yahoo.com>
Message-ID: <94a1afde0606111803m3d004147k58f0280f7fe3c4bd@mail.gmail.com>

I'm going to try changing the numbering plan tomorrow to national and see if
that helps, thanks for the suggestion!

On 6/11/06, Hassan Salama <hsalama_us at yahoo.com> wrote:

> Hello,
>
> Regarding your issue of long distance call, the number
> handling could varies from country to country as this
> is telecom standard.
>
> But you may try the translation rule to modify Called
> party number type to become national [a must for long
> distance within a country], and stip 91 from the
> begeinning of the number [i.e. send 10 digits without
> 1], you may give another try with stipping 9 digit
> only
>
> Thanks
>
> --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
>
> > have you tried changing the isdn to national, or
> > unknown or something along those lines?
> >
> > Scott
> >
> > ________________________________
> >
> > From: cisco-voip-bounces at puck.nether.net on behalf
> > of Ed Leatherman
> > Sent: Fri 6/9/2006 5:16 PM
> > To: ciscovoip
> > Subject: [cisco-voip] Long Distance woes
> >
> >
> > Was wondering if anyone is familiar with how long
> > distance carriers/service interacts with local
> > service...
> >
> > Working with service provider at one of our branch
> > campuses which we just converted to a callmanager
> > system this past week. Users can call long distance
> > numbers within our area code (304) just fine, but
> > cannot call outside of the area code. The dial
> > pattern is the same (9.1[2-9]XX[2-9]XX XXXX) for
> > either case (exact same route pattern actually), I
> > just strip the 9 and send 1 + 10 digits to the phone
> > company. Can see this in CCM trace. System uses 2
> > PRI T1's.
> >
> > If users call an out of state number, they get fast
> > busy. The local carrier is looking at the problem
> > now but initially they just told me it wasnt there
> > problem, as they just pass the digits to the long
> > distance carrier if the number starts with "1". It
> > seems to me that maybe the lines are setup
> > incorrectly as far as what long distance provider it
> > should use.. but I dont know enough about the
> > process to make a good guess.. I've not worked with
> > this particular local service provider before so I
> > dont have the same "faith" if you could call it
> > that, as I do with the carrier we have at the main
> > campus which I have dealt with before.
> >
> > Caller ID also does not come across, but I think
> > this is a seperate issue which we can hopefully
> > address after the long distance is working.
> >
> > Any ideas? I'm curious if there is anything more I
> > can look at from my end.
> >
> > --
> > Ed Leatherman
> > IP Telephony Coordinator
> > 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
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From bababooey at cox.net  Sun Jun 11 22:04:23 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Sun, 11 Jun 2006 22:04:23 -0400
Subject: [cisco-voip] 7970G - Brightness display/display not active
References: <mailman.6202.1150069158.1478.cisco-voip@puck.nether.net>
Message-ID: <000301c68dc4$86acb610$0f0110ac@corporation2>

1.  The brightness display on my 7970 does not allow for upward adjustment,
the bar is currently set less than halfway up?

2.  How can I set the default phone config to change the default to have the 
phone display active always,
the default is to blank the display on Sunday???

Thanks 



From CMStaffin at ra.rockwell.com  Sun Jun 11 22:10:28 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Sun, 11 Jun 2006 21:10:28 -0500
Subject: [cisco-voip] 7970G - Brightness display/display not active
In-Reply-To: <000301c68dc4$86acb610$0f0110ac@corporation2>
Message-ID: <OFD965A43A.EDC6974C-ON8625718B.000BD98A-8625718B.000BDF27@ra.rockwell.com>

Bob,

What switch is this connected to.  If it is an older switch that can not
supply the full 15 watts the display will be dim.  This can be corrected
with a 3560 or a power brick.

Also if you log inot callmanager you can change when the display will be
active or not

Craig




                                                                           
             "Bob A. Bowie"                                                
             <bababooey at cox.ne                                             
             t>                                                         To 
             Sent by:                  <cisco-voip at puck.nether.net>        
             cisco-voip-bounce                                          cc 
             s at puck.nether.net                                             
                                                                   Subject 
                                       [cisco-voip] 7970G - Brightness     
             06/11/2006 09:04          display/display not active          
             PM                                                            
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           




1.  The brightness display on my 7970 does not allow for upward adjustment,
the bar is currently set less than halfway up?

2.  How can I set the default phone config to change the default to have
the
phone display active always,
the default is to blank the display on Sunday???

Thanks


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



From bababooey at cox.net  Sun Jun 11 23:15:03 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Sun, 11 Jun 2006 23:15:03 -0400
Subject: [cisco-voip] 7970G - Brightness display/display not active
References: <OFD965A43A.EDC6974C-ON8625718B.000BD98A-8625718B.000BDF27@ra.rockwell.com>
Message-ID: <000301c68dce$661920a0$0f0110ac@corporation2>

It is a 3550, does it supply the 15watts?

Yes, but how can I make the default when creating the phone to change the 
display to be active all the time.
So I don't have to go down and uncheck it every time.

THanks


----- Original Message ----- 
From: "Craig M Staffin" <CMStaffin at ra.rockwell.com>
To: "Bob A. Bowie" <bababooey at cox.net>
Cc: <cisco-voip at puck.nether.net>; <cisco-voip-bounces at puck.nether.net>
Sent: Sunday, June 11, 2006 10:10 PM
Subject: Re: [cisco-voip] 7970G - Brightness display/display not active


> Bob,
>
> What switch is this connected to.  If it is an older switch that can not
> supply the full 15 watts the display will be dim.  This can be corrected
> with a 3560 or a power brick.
>
> Also if you log inot callmanager you can change when the display will be
> active or not
>
> Craig
>
>
>
>
>
>             "Bob A. Bowie"
>             <bababooey at cox.ne
>             t>                                                         To
>             Sent by:                  <cisco-voip at puck.nether.net>
>             cisco-voip-bounce                                          cc
>             s at puck.nether.net
>                                                                   Subject
>                                       [cisco-voip] 7970G - Brightness
>             06/11/2006 09:04          display/display not active
>             PM
>
>
>
>
>
>
>
>
>
> 1.  The brightness display on my 7970 does not allow for upward 
> adjustment,
> the bar is currently set less than halfway up?
>
> 2.  How can I set the default phone config to change the default to have
> the
> phone display active always,
> the default is to blank the display on Sunday???
>
> Thanks
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
> 



From CMStaffin at ra.rockwell.com  Mon Jun 12 00:26:06 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Sun, 11 Jun 2006 23:26:06 -0500
Subject: [cisco-voip] 7970G - Brightness display/display not active
In-Reply-To: <000301c68dce$661920a0$0f0110ac@corporation2>
Message-ID: <OF6EBF193A.8342FEBC-ON8625718B.00185118-8625718B.00184A02@ra.rockwell.com>

I can not remember what the exact setting is called but there is a setting
under enterprise perameters




                                                                           
             "Bob A. Bowie"                                                
             <bababooey at cox.ne                                             
             t>                                                         To 
             Sent by:                  "Craig M Staffin"                   
             cisco-voip-bounce         <CMStaffin at ra.rockwell.com>         
             s at puck.nether.net                                          cc 
                                       cisco-voip-bounces at puck.nether.net, 
                                       cisco-voip at puck.nether.net          
             06/11/2006 10:15                                      Subject 
             PM                        Re: [cisco-voip] 7970G - Brightness 
                                       display/display not active          
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           




It is a 3550, does it supply the 15watts?

Yes, but how can I make the default when creating the phone to change the
display to be active all the time.
So I don't have to go down and uncheck it every time.

THanks


----- Original Message -----
From: "Craig M Staffin" <CMStaffin at ra.rockwell.com>
To: "Bob A. Bowie" <bababooey at cox.net>
Cc: <cisco-voip at puck.nether.net>; <cisco-voip-bounces at puck.nether.net>
Sent: Sunday, June 11, 2006 10:10 PM
Subject: Re: [cisco-voip] 7970G - Brightness display/display not active


> Bob,
>
> What switch is this connected to.  If it is an older switch that can not
> supply the full 15 watts the display will be dim.  This can be corrected
> with a 3560 or a power brick.
>
> Also if you log inot callmanager you can change when the display will be
> active or not
>
> Craig
>
>
>
>
>
>             "Bob A. Bowie"
>             <bababooey at cox.ne
>             t>                                                         To
>             Sent by:                  <cisco-voip at puck.nether.net>
>             cisco-voip-bounce                                          cc
>             s at puck.nether.net
>                                                                   Subject
>                                       [cisco-voip] 7970G - Brightness
>             06/11/2006 09:04          display/display not active
>             PM
>
>
>
>
>
>
>
>
>
> 1.  The brightness display on my 7970 does not allow for upward
> adjustment,
> the bar is currently set less than halfway up?
>
> 2.  How can I set the default phone config to change the default to have
> the
> phone display active always,
> the default is to blank the display on Sunday???
>
> 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 bababooey at cox.net  Mon Jun 12 01:03:01 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Mon, 12 Jun 2006 01:03:01 -0400
Subject: [cisco-voip] Cisco 3550 POE
References: <OF6EBF193A.8342FEBC-ON8625718B.00185118-8625718B.00184A02@ra.rockwell.com>
Message-ID: <000301c68ddd$7c18d990$0f0110ac@corporation2>

Does the Cisco 3550-24pwr support 24 ports at 15.4W when just plugged into 
110v outlet?
Or is the Cisco RPS required, it is not clear on the data sheet?  If so, 
does the Cisco 3560 improve upon this?



From jordyl at yahoo.com  Mon Jun 12 05:13:28 2006
From: jordyl at yahoo.com (Jordy Leung)
Date: Mon, 12 Jun 2006 02:13:28 -0700 (PDT)
Subject: [cisco-voip] How Does ICM Control The Voice Call Session in IPCC
	7.0?
Message-ID: <20060612091328.67923.qmail@web37201.mail.mud.yahoo.com>

Dear all,

I'm doing analysis on IPCC 7.0 architecture.  I want to ask experts out there that anyone knows how the call is controlled in the IPCC system?  When the call goes to ICM thru dial number, will ICM assign any unique ID like session ID or even session object like what other web application servers do?  If there is any unique ID, can I use this ID and control this call routing amount different Agent/IP Phones by using CTIOS API (or other API)?
 
Jordy Leung




From hsalama_us at yahoo.com  Mon Jun 12 05:36:47 2006
From: hsalama_us at yahoo.com (Hassan Salama)
Date: Mon, 12 Jun 2006 02:36:47 -0700 (PDT)
Subject: [cisco-voip] inside verses outside call and ring cadence
In-Reply-To: <C0427FD6-73B8-4999-B9C3-B55318635510@mimectl>
Message-ID: <20060612093647.29257.qmail@web34609.mail.mud.yahoo.com>

Hello,

      When the call comming across access gateway the
cadence can be configured on the voice port. meanwhile
when the call purely H.323, i believe that it should
be configured from the call agent(call manager)

for callmanager :
under device pool configuration, you can modify
"network locale"
Reference:
Network Locale
 From the drop-down list box, choose the locale that
is associated with the phone. The network locale
contains a definition of the tones and cadences that
are used by the phones and gateways in the device pool
in a specific geographic area.
 
 
Thanks

--- "Frazee, Timothy" <Timothy_Frazee at adp.com> wrote:

>  all,
> 
> when a call comes in from outside, how does ccm
> determine to change the cadence on the ip phone?
> 
> is this the offnet and on-net thing?
> 
> how would some calls not ring-ring (external) and
> some would ring (internal) when they all come from
> outside in.
> 
> running 4.1.3sr2 with a 2821 and 11 FXO to pots via
> H323.
> 
> thanks.
> 
> 
> This message and any attachments are intended only
> for the use of the addressee and may contain
> information that is privileged and confidential. If
> the reader of the message is not the intended
> recipient or an authorized representative of the
> intended recipient, you are hereby notified that any
> dissemination of this communication is strictly
> prohibited. If you have received this communication
> in error, please notify us immediately by e-mail and
> delete the message and any attachments from your
> system.
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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From newlook110 at yahoo.com  Mon Jun 12 06:06:36 2006
From: newlook110 at yahoo.com (new look)
Date: Mon, 12 Jun 2006 03:06:36 -0700 (PDT)
Subject: [cisco-voip] Auto Attendant Problem
Message-ID: <20060612100636.55042.qmail@web61314.mail.yahoo.com>

Dear All,
  I have CCM and installed Auto Attendant and ever think is ok,
  The problem I have is when any call my company he hear this default auto attendant welcome message and as you know the message asked the caller to enter 1 for entering the ext. no., 2 for entering the name and 0 for operator, and when the customer entered 1 
  The message asked him to enter the ext. no. than pound key (#) and this is very long.
   
  So, I need when the customer call my company he hears the message says welcome
  Please enter the ext. or zero for operator (I already have this message recorded and I know how to upload it to the call manager) But I need to change the procedure of the Auto Attendant for this new message.
   
  Thanks

 __________________________________________________
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From abdul_zu at yahoo.com  Mon Jun 12 06:33:05 2006
From: abdul_zu at yahoo.com (Abdul Lateef)
Date: Mon, 12 Jun 2006 03:33:05 -0700 (PDT)
Subject: [cisco-voip] VoIP Gatekeeper
Message-ID: <20060612103305.54329.qmail@web34515.mail.mud.yahoo.com>

Hi All,

Could any one suggest me cisco model to work as
Gatekeeper
with full H.323 and SIP support.

I want more than 50 E1/T1 supports.

I will appricated for your kind of repliese


Thank You
Abul Lateef

__________________________________________________
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From hsalama_us at yahoo.com  Mon Jun 12 07:34:12 2006
From: hsalama_us at yahoo.com (Hassan Salama)
Date: Mon, 12 Jun 2006 04:34:12 -0700 (PDT)
Subject: [cisco-voip] VoIP Gatekeeper
In-Reply-To: <20060612103305.54329.qmail@web34515.mail.mud.yahoo.com>
Message-ID: <20060612113412.88943.qmail@web34605.mail.mud.yahoo.com>

I think for such capacity of 50 E1/T1, it is beeter to
go for softswitch/PSTN Gateway solution

--- Abdul Lateef <abdul_zu at yahoo.com> wrote:

> Hi All,
> 
> Could any one suggest me cisco model to work as
> Gatekeeper
> with full H.323 and SIP support.
> 
> I want more than 50 E1/T1 supports.
> 
> I will appricated for your kind of repliese
> 
> 
> Thank You
> Abul Lateef
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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From cisco-voip at ibh.net  Mon Jun 12 08:06:47 2006
From: cisco-voip at ibh.net (Andre Beck)
Date: Mon, 12 Jun 2006 14:06:47 +0200
Subject: [cisco-voip] network-clock-select vs. c3745
Message-ID: <20060612120647.GA9756@ibh.de>

Hi,

hunting fax issues, I tried to correct the clocking of a certain c3745
that is fitted with several NM-HD-2Vs bearing VIC2-2BRI-NT/TE and
VIC-4FXS/DID WICs. IOS is 12.4(8), aka the latest stable one. How
should I interpret this dialog:

phonebox-tz1(config)#network-clock-select ?
  <1-8>  priority

phonebox-tz1(config)#network-clock-select 1 ?
% Unrecognized command
phonebox-tz1(config)#network-clock-select 1 bri1/0
                                    ^
% Invalid input detected at '^' marker.


I can configure network-clock participation, but I cannot select a
source. This is supposed to work on this platform since 12.3 main-
line...

TIA,
Andre.
-- 
                  The _S_anta _C_laus _O_peration
  or "how to turn a complete illusion into a neverending money source"

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From hsalama_us at yahoo.com  Mon Jun 12 08:40:42 2006
From: hsalama_us at yahoo.com (Hassan Salama)
Date: Mon, 12 Jun 2006 05:40:42 -0700 (PDT)
Subject: [cisco-voip] network-clock-select vs. c3745
In-Reply-To: <20060612120647.GA9756@ibh.de>
Message-ID: <20060612124042.34283.qmail@web34612.mail.mud.yahoo.com>

Hi,
   could you have a look on this document:
http://www.cisco.com/en/US/partner/products/hw/routers/ps259/products_tech_note09186a008031a072.shtml

[it may help if you didn't read it before]

Thanks and regards,

--- Andre Beck <cisco-voip at ibh.net> wrote:

> Hi,
> 
> hunting fax issues, I tried to correct the clocking
> of a certain c3745
> that is fitted with several NM-HD-2Vs bearing
> VIC2-2BRI-NT/TE and
> VIC-4FXS/DID WICs. IOS is 12.4(8), aka the latest
> stable one. How
> should I interpret this dialog:
> 
> phonebox-tz1(config)#network-clock-select ?
>   <1-8>  priority
> 
> phonebox-tz1(config)#network-clock-select 1 ?
> % Unrecognized command
> phonebox-tz1(config)#network-clock-select 1 bri1/0
>                                     ^
> % Invalid input detected at '^' marker.
> 
> 
> I can configure network-clock participation, but I
> cannot select a
> source. This is supposed to work on this platform
> since 12.3 main-
> line...
> 
> TIA,
> Andre.
> -- 
>                   The _S_anta _C_laus _O_peration
>   or "how to turn a complete illusion into a
> neverending money source"
> 
> -> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr.
> Horn GmbH, Dresden <-
> _______________________________________________
> 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  Mon Jun 12 09:06:26 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Mon, 12 Jun 2006 09:06:26 -0400
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
Message-ID: <AB0A35F2486F81468971AA68558BE71903EDA48F@smieln01winm01.micoanet.com>

Has anyone run into a One Way Audio issue with phone load 8.0(3) on a
7960?  I had piloted the phone load with about 20 phones and didn't hear
of any reported problems.  Now after I have pushed the load to around
200+ phones I have users calling into the helpdesk reporting problems
under the following circumstances.  Please note that the calls are
internal and on the same LAN (even logical VLAN).

 

1)       User A calls User B

2)       User B can hear User A talking, however User A cannot hear User
B.

3)       User A hangs up and calls User B again and there are no issues.

 

I haven't had any issues to date with the older, 8.0(2) (or earlier)
phone loads.  I am thinking that I am going to roll everyone back, but I
wanted to see if anyone is having the issue while also warning people of
the problem. 

 

Sincerely,

 

Matthew M. Linsemier, CCSP/CCNP/CCDA
Senior Network Engineer
APCapital
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.

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From cisco-voip at ibh.net  Mon Jun 12 09:28:35 2006
From: cisco-voip at ibh.net (Andre Beck)
Date: Mon, 12 Jun 2006 15:28:35 +0200
Subject: [cisco-voip] network-clock-select vs. c3745
In-Reply-To: <20060612124042.34283.qmail@web34612.mail.mud.yahoo.com>
References: <20060612120647.GA9756@ibh.de>
	<20060612124042.34283.qmail@web34612.mail.mud.yahoo.com>
Message-ID: <20060612132835.GB9756@ibh.de>

Re,

On Mon, Jun 12, 2006 at 05:40:42AM -0700, Hassan Salama wrote:
> Hi,
>    could you have a look on this document:
> http://www.cisco.com/en/US/partner/products/hw/routers/ps259/products_tech_note09186a008031a072.shtml
> 
> [it may help if you didn't read it before]

It doesn't state anything unexpected. However, I've tried the
network-clock-select on a 28xx that happens to have no voice card
at all and there it behaves the same way, though on another 28xx with
voice BRIs the command is dealt with correctly. It is as well correctly
handled on yet another 3745 with E1 MFTs so the actual problem seems
to be that Cisco forgot to support BRI clock sources on the 3745 platform
(at least). Great, fax wreaks havoc and no way to fix it...

Andre.
-- 
                  The _S_anta _C_laus _O_peration
  or "how to turn a complete illusion into a neverending money source"

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From zephy316 at gmail.com  Mon Jun 12 10:50:05 2006
From: zephy316 at gmail.com (Israel Lang)
Date: Mon, 12 Jun 2006 10:50:05 -0400
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
In-Reply-To: <AB0A35F2486F81468971AA68558BE71903EDA48F@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE71903EDA48F@smieln01winm01.micoanet.com>
Message-ID: <91e236d30606120750p60961adeg6ee68422af981a0@mail.gmail.com>

Sounds like a classic partition/CSS issue to me. I would check to make sure
that phone A and phone B have each others partitions in their CSS. Sounds to
me like B does and A doesn't. Could also be a routing issue.

On 6/12/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>  Has anyone run into a One Way Audio issue with phone load 8.0(3) on a
> 7960?  I had piloted the phone load with about 20 phones and didn't hear of
> any reported problems.  Now after I have pushed the load to around 200+
> phones I have users calling into the helpdesk reporting problems under the
> following circumstances.  Please note that the calls are internal and on the
> same LAN (even logical VLAN).
>
>
>
> 1)       User A calls User B
>
> 2)       User B can hear User A talking, however User A cannot hear User
> B.
>
> 3)       User A hangs up and calls User B again and there are no issues.
>
>
>
> I haven't had any issues to date with the older, 8.0(2) (or earlier) phone
> loads.  I am thinking that I am going to roll everyone back, but I wanted to
> see if anyone is having the issue while also warning people of the problem.
>
>
>
> Sincerely,
>
>
>
> Matthew M. Linsemier, CCSP/CCNP/CCDA
> Senior Network Engineer
> APCapital
> 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
>
>
>


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

Replace the _ with an @ symbol
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From MLinsemier at apcapital.com  Mon Jun 12 10:58:44 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Mon, 12 Jun 2006 10:58:44 -0400
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
Message-ID: <AB0A35F2486F81468971AA68558BE71903EDA560@smieln01winm01.micoanet.com>

As stipulated below, phones are on the same logical VLAN so no routing
is taking place other then call setup and teardown to the Call Managers.
Additionally, I know that it has nothing to do with the CSS because if
this was the issue the call would not be connected in the first place
and users would hear fast busy.

 

This problem started specifically with the 8.0(3) phone load.  The only
other change that as been made has been the OS2.4sr7 patch, which I find
unlikely to be causing this problem.

 

Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Monday, June 12, 2006 10:50 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

 

Sounds like a classic partition/CSS issue to me. I would check to make
sure that phone A and phone B have each others partitions in their CSS.
Sounds to me like B does and A doesn't. Could also be a routing issue.

On 6/12/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:

Has anyone run into a One Way Audio issue with phone load 8.0(3) on a
7960?  I had piloted the phone load with about 20 phones and didn't hear
of any reported problems.  Now after I have pushed the load to around
200+ phones I have users calling into the helpdesk reporting problems
under the following circumstances.  Please note that the calls are
internal and on the same LAN (even logical VLAN).

 

1)       User A calls User B

2)       User B can hear User A talking, however User A cannot hear User
B.

3)       User A hangs up and calls User B again and there are no issues.

 

I haven't had any issues to date with the older, 8.0(2) (or earlier)
phone loads.  I am thinking that I am going to roll everyone back, but I
wanted to see if anyone is having the issue while also warning people of
the problem. 

 

Sincerely,

 

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

 

________________________________

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________________________________


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-- 
Have a great day!!!!!!!!!!
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Replace the _ with an @ symbol 


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From jared at puck.nether.net  Mon Jun 12 11:17:15 2006
From: jared at puck.nether.net (jared at puck.nether.net)
Date: Mon, 12 Jun 2006 10:17:15 -0500
Subject: [cisco-voip] Test
Message-ID: <200606121518.k5CFI9jW062336@npd.nether.net>

Your message could not be delivered

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From ekadapuram at mlnusa.com  Mon Jun 12 10:35:58 2006
From: ekadapuram at mlnusa.com (Kadapuram, Eby)
Date: Mon, 12 Jun 2006 10:35:58 -0400
Subject: [cisco-voip] Serial Number from Call Manager
Message-ID: <E81220B2F3DF934EAE40B50283BE49690378728B@mlnusa-exch1.mlnusa.com>

Hi,
  Is it possible to get the phone's serial number from call manager
using BAT tool? I know how to get it one by one but it will take a long
time to go to each phone and copying it. Any Ideas?

Thanks,

Eby 


********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.



From smunzani at comcast.net  Mon Jun 12 11:40:09 2006
From: smunzani at comcast.net (Sam Munzani)
Date: Mon, 12 Jun 2006 10:40:09 -0500
Subject: [cisco-voip] vmware gsx server
In-Reply-To: <5d093f9a0606101941o8b7f560w93b00f7c48de4937@mail.gmail.com>
References: <20861400.1149868558357.JavaMail.root@eastrmwml02.mgt.cox.net>
	<5d093f9a0606101941o8b7f560w93b00f7c48de4937@mail.gmail.com>
Message-ID: <448D8AD9.7080803@comcast.net>

Did anybody try ESX server with CCM and Unity? I have access to my 
company's lab where we have ESX server on very powerful PC.

Thanks,
Sam
> Yes.
>
> To be honest, that is all I have been using... I am running
> CallManager 3.3, 4.0, 4.1, 4.2 and 5.0(2) and Unity 4.0(4), 4.0(5),
> 4.1(1) and 4.2(1), as well as CRA 3.5 and 4.0 all on VMWare GSX 3.1
>
> I also have some test bed client machines (for BearShare and client apps)...
>
> To be honest, I don't know how you guys run labs without VMWare...
>
>
>
> Jonathan
>
> On 6/9/06, bababooey at cox.net <bababooey at cox.net> wrote:
>   
>> Has anyone utilized the gsx server to run the call mamanger environment in a vmware test bed.
>>
>> 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
>
>   

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From Scott.Voll at wesd.org  Mon Jun 12 12:09:33 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 12 Jun 2006 09:09:33 -0700
Subject: [cisco-voip] Wireless headset
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE7F6@Apollo.wesd.org>

I have a user demo'ing a wireless headset that has a receiver lifter to
remotely answer the phone.  So if you plug the phones phone jack to the
wireless headset you get no audio.  But if you plug it into the headset
jack it only works until you lift the receiver the first time then you
manually have to hit the headset button again.  I'm looking  to do this
a little more automagically.

 

Any way to keep the headset always on?  Service parameter?  Auto answer
on the line doesn't work with the wireless setup she has.

 

Any better solutions?  One that really ties in with IP Phones?

 

We just wait to remotely pick up the call(s) with no phone intervention.

 

Scott

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From jburton at NETechCorp.com  Mon Jun 12 12:30:35 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Mon, 12 Jun 2006 12:30:35 -0400
Subject: [cisco-voip] vmware gsx server
In-Reply-To: <448D8AD9.7080803@comcast.net>
References: <20861400.1149868558357.JavaMail.root@eastrmwml02.mgt.cox.net><5d093f9a0606101941o8b7f560w93b00f7c48de4937@mail.gmail.com>,
	<448D8AD9.7080803@comcast.net>
Message-ID: <376E9303-3B3A-4EA3-82D6-A365FE405B55@mimectl>

I have used ESX server and have had no problems with it.  Should work fine.



From: Sam Munzani
Sent: Mon 6/12/2006 11:40 AM
To: Jonathan Charles
Cc: cisco-voip at puck.nether.net; bababooey at cox.net
Subject: Re: [cisco-voip] vmware gsx server


Did anybody try ESX server with CCM and Unity? I have access to my company's lab where we have ESX server on very powerful PC.

Thanks,
Sam

Yes.

To be honest, that is all I have been using... I am running
CallManager 3.3, 4.0, 4.1, 4.2 and 5.0(2) and Unity 4.0(4), 4.0(5),
4.1(1) and 4.2(1), as well as CRA 3.5 and 4.0 all on VMWare GSX 3.1

I also have some test bed client machines (for BearShare and client apps)...

To be honest, I don't know how you guys run labs without VMWare...



Jonathan

On 6/9/06, bababooey at cox.net mailto:bababooey at cox.net wrote:
  
Has anyone utilized the gsx server to run the call mamanger environment in a vmware test bed.

Thanks,

_______________________________________________
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From abdul_zu at yahoo.com  Mon Jun 12 13:07:52 2006
From: abdul_zu at yahoo.com (Abdul Lateef)
Date: Mon, 12 Jun 2006 10:07:52 -0700 (PDT)
Subject: [cisco-voip] VoIP Gatekeeper
In-Reply-To: <20060612113412.88943.qmail@web34605.mail.mud.yahoo.com>
Message-ID: <20060612170752.7881.qmail@web34514.mail.mud.yahoo.com>


Hi,

could you please suggest me which model of cisco will
be fit for softswitch?

I found Cisco BTS 10200 Softswitch, but i am not so
familer with cisco products, Please advice me this BTS
can help us to do more than 50 E1/T1 with unlimited
user registration.


Thank You
Abdul Lateef

--- Hassan Salama <hsalama_us at yahoo.com> wrote:

> I think for such capacity of 50 E1/T1, it is beeter
> to
> go for softswitch/PSTN Gateway solution
> 
> --- Abdul Lateef <abdul_zu at yahoo.com> wrote:
> 
> > Hi All,
> > 
> > Could any one suggest me cisco model to work as
> > Gatekeeper
> > with full H.323 and SIP support.
> > 
> > I want more than 50 E1/T1 supports.
> > 
> > I will appricated for your kind of repliese
> > 
> > 
> > Thank You
> > Abul Lateef
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> > protection around 
> > http://mail.yahoo.com 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From Timothy_Frazee at adp.com  Mon Jun 12 13:11:28 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Mon, 12 Jun 2006 12:11:28 -0500
Subject: [cisco-voip] Serial Number from Call Manager
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC0648E@EXCHANGEDS01.ds.ad.adp.com>

Look at a product called Phone Inventory from Syn-Apps. Very slick tool.

You need to install this on another server, and you will need to request
a trial key. But it is worth it. I have used it many times.

(load it up in vmware, then you can take this tool with you on different
jobs/customers)

I am not sure if you can install it under xp pro, as it needs IIS and
stuff.

Here is the direct link.

http://www.syn-apps.net/PhoneInventory.aspx



Timothy Frazee 
Network Solutions
314.249.8697 cellular phone

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kadapuram, Eby
Sent: Monday, June 12, 2006 9:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Serial Number from Call Manager

Hi,
  Is it possible to get the phone's serial number from call manager
using BAT tool? I know how to get it one by one but it will take a long
time to go to each phone and copying it. Any Ideas?

Thanks,

Eby 


********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders
Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
hereby notified that disclosing, copying, distributing, or taking any
action in reliance on the contents of this information is strictly
prohibited, and that the E-mail should be returned to this company
immediately. If you have received this E-mail in error, please contact
the sender immediately.


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


This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.


From Scott.Voll at wesd.org  Mon Jun 12 13:23:47 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 12 Jun 2006 10:23:47 -0700
Subject: [cisco-voip] Serial Number from Call Manager
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE843@Apollo.wesd.org>

Variphy can do it also.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Frazee, Timothy
Sent: Monday, June 12, 2006 10:11 AM
To: Kadapuram, Eby; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Serial Number from Call Manager

Look at a product called Phone Inventory from Syn-Apps. Very slick tool.

You need to install this on another server, and you will need to request
a trial key. But it is worth it. I have used it many times.

(load it up in vmware, then you can take this tool with you on different
jobs/customers)

I am not sure if you can install it under xp pro, as it needs IIS and
stuff.

Here is the direct link.

http://www.syn-apps.net/PhoneInventory.aspx



Timothy Frazee 
Network Solutions
314.249.8697 cellular phone

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Kadapuram, Eby
Sent: Monday, June 12, 2006 9:36 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Serial Number from Call Manager

Hi,
  Is it possible to get the phone's serial number from call manager
using BAT tool? I know how to get it one by one but it will take a long
time to go to each phone and copying it. Any Ideas?

Thanks,

Eby 


********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders
Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
hereby notified that disclosing, copying, distributing, or taking any
action in reliance on the contents of this information is strictly
prohibited, and that the E-mail should be returned to this company
immediately. If you have received this E-mail in error, please contact
the sender immediately.


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


This message and any attachments are intended only for the use of the
addressee and may contain information that is privileged and
confidential. If the reader of the message is not the intended recipient
or an authorized representative of the intended recipient, you are
hereby notified that any dissemination of this communication is strictly
prohibited. If you have received this communication in error, please
notify us immediately by e-mail and delete the message and any
attachments from your system.

_______________________________________________
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From rsingleton at novateck.com  Mon Jun 12 14:20:24 2006
From: rsingleton at novateck.com (Robert Singleton)
Date: Mon, 12 Jun 2006 13:20:24 -0500
Subject: [cisco-voip] Serial Number from Call Manager
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605CEE843@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A605CEE843@Apollo.wesd.org>
Message-ID: <200606121320703.SM00900@mail.leasenet.net>

I use a handful of Linux command line tools to inventory my phones. No
software or licenses, trial or otherwise, to buy...

None of what I do necessarily requires Linux only, but I think it's
simpler that way.

I use nmap to get a list of IP addresses from the telephone subnets. A
one-liner script uses lynx, a text mode http browser, to gather the
required page from each telephone. I have to make two runs through it
because until I run through the first time, I can't reasonably tell
which IPs are ATA's rather than 79XX phones. The ATA's web page is in
the form of http://<ip address>/dev and the phones are http://<ip
address>.

Then I use awk to boil down that data into a CSV file. I then generate
my reports off the CSV or hand it to users who "need" Excel to sort and
filter data.

I have a single script that runs through this process in about 10
minutes for 800 or so phones. Any interested in it is welcome to it and
more detailed notes.

I use the same technique to get information from http://<ip
address>/PortInformation?1 page which shows what switch and port a given
telephone is connected to, which could also be helpful to an inventory.

While running it from a Linux machine simplifies it for me, I think
anyone who wants to use other tools should be able to do it. The two key
components for getting the raw data is nmap and lynx, both of which are
available for other operating systems, including Windows.

http://www.insecure.org/nmap/
http://lynx.browser.org/

Other UN*X command line tools are available to Windows users from
packages such as Cygwin.

http://www.cygwin.com/




From Scott.Voll at wesd.org  Mon Jun 12 15:41:47 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 12 Jun 2006 12:41:47 -0700
Subject: [cisco-voip] H323 SRST
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE8AF@Apollo.wesd.org>

Can you use a H323 gateway for SRST?

 

What would the config look like if you can?

 

Thanks

 

Scott

 

CM 4.1.3sr3a 2801 running h323 for caller id on FXO ports.

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From rratliff at cisco.com  Mon Jun 12 15:45:38 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Mon, 12 Jun 2006 15:45:38 -0400
Subject: [cisco-voip] H323 SRST
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605CEE8AF@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A605CEE8AF@Apollo.wesd.org>
Message-ID: <34522B2B-7B9D-440A-A5C2-ADB7DE3F6652@cisco.com>

Yes you can use an H.323 gateway for SRST.

The config would look exactly the same as an MGCP router configured  
for SRST but no MGCP commands.

call-manager-fallback is not dependant on MGCP.

-Ryan

On Jun 12, 2006, at 3:41 PM, Voll, Scott wrote:

Can you use a H323 gateway for SRST?



What would the config look like if you can?



Thanks



Scott



CM 4.1.3sr3a 2801 running h323 for caller id on FXO ports.

_______________________________________________
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  Mon Jun 12 16:00:39 2006
From: mfa at crec.ifas.ufl.edu (Mike Armstrong)
Date: Mon, 12 Jun 2006 16:00:39 -0400
Subject: [cisco-voip] Wireless headset
References: <mailman.6524.1150141317.1478.cisco-voip@puck.nether.net>
Message-ID: <00b501c68e5a$e0c5d460$400cfb0a@crec.ifas.ufl.edu>

One of our users has a Plantronics C555 with an HL10 handset lifter.  The 
base station is plugged into the handset plug, and the handset cord has been 
removed altogether.  When she pushes the button on the earpiece, the handset 
lifts and she talks.  I think this is about as automagic as it gets, albeit 
a little Rube Goldberg - ish.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL


> Date: Mon, 12 Jun 2006 09:09:33 -0700
> From: "Voll, Scott" <Scott.Voll at wesd.org>
> Subject: [cisco-voip] Wireless headset
> To: "cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
> Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE7F6 at Apollo.wesd.org>
> Content-Type: text/plain; charset="us-ascii"
>
> I have a user demo'ing a wireless headset that has a receiver lifter to
> remotely answer the phone.  So if you plug the phones phone jack to the
> wireless headset you get no audio.  But if you plug it into the headset
> jack it only works until you lift the receiver the first time then you
> manually have to hit the headset button again.  I'm looking  to do this
> a little more automagically.
>
>
>
> Any way to keep the headset always on?  Service parameter?  Auto answer
> on the line doesn't work with the wireless setup she has.
>
>
>
> Any better solutions?  One that really ties in with IP Phones?
>
>
>
> We just wait to remotely pick up the call(s) with no phone intervention.
>
>
>
> Scott


From Scott.Voll at wesd.org  Mon Jun 12 16:02:23 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 12 Jun 2006 13:02:23 -0700
Subject: [cisco-voip] Wireless headset
Message-ID: <407055A92CECCB499C922A2D35FC19A605D631B7@Apollo.wesd.org>

She has it setup the same but not getting audio.....    Will see if we can get a different demo.   Thx


--------------------------
Sent from my BlackBerry Wireless Handheld


-----Original Message-----
From: Mike Armstrong <mfa at crec.ifas.ufl.edu>
To: cisco-voip at puck.nether.net <cisco-voip at puck.nether.net>; Voll, Scott <Scott.Voll at wesd.org>
Sent: Mon Jun 12 13:00:39 2006
Subject: Re: [cisco-voip] Wireless headset

One of our users has a Plantronics C555 with an HL10 handset lifter.  The 
base station is plugged into the handset plug, and the handset cord has been 
removed altogether.  When she pushes the button on the earpiece, the handset 
lifts and she talks.  I think this is about as automagic as it gets, albeit 
a little Rube Goldberg - ish.

Mike Armstrong
UF/IFAS CREC
Lake Alfred, FL


> Date: Mon, 12 Jun 2006 09:09:33 -0700
> From: "Voll, Scott" <Scott.Voll at wesd.org>
> Subject: [cisco-voip] Wireless headset
> To: "cisco-voip at puck-nether.net" <cisco-voip at puck.nether.net>
> Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE7F6 at Apollo.wesd.org>
> Content-Type: text/plain; charset="us-ascii"
>
> I have a user demo'ing a wireless headset that has a receiver lifter to
> remotely answer the phone.  So if you plug the phones phone jack to the
> wireless headset you get no audio.  But if you plug it into the headset
> jack it only works until you lift the receiver the first time then you
> manually have to hit the headset button again.  I'm looking  to do this
> a little more automagically.
>
>
>
> Any way to keep the headset always on?  Service parameter?  Auto answer
> on the line doesn't work with the wireless setup she has.
>
>
>
> Any better solutions?  One that really ties in with IP Phones?
>
>
>
> We just wait to remotely pick up the call(s) with no phone intervention.
>
>
>
> Scott

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From Greg_Hinson at adp.com  Mon Jun 12 16:06:24 2006
From: Greg_Hinson at adp.com (Hinson, Greg)
Date: Mon, 12 Jun 2006 15:06:24 -0500
Subject: [cisco-voip] H323 SRST
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC06703@EXCHANGEDS01.ds.ad.adp.com>

The only point of mgcp fallback is to alert the router that the
call-agent(call manager) is no longer available and to permit the
registered devices to use the H.323 dial-peers. Fallback relies on the
command call application alternate default(12.3.11 or below) or
application -->, global -->, service alternate default(12.3.14 or newer)
to allow the router to rely on its default application (h.323) if MGCP
is unavailable.
 
The phones register the same, whether its H.323 or MGCP
 
-Greg
 
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, June 12, 2006 12:42 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] H323 SRST
 
Can you use a H323 gateway for SRST?
 
What would the config look like if you can?
 
Thanks
 
Scott
 
CM 4.1.3sr3a 2801 running h323 for caller id on FXO ports.


This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From Scott.Voll at wesd.org  Mon Jun 12 16:49:13 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 12 Jun 2006 13:49:13 -0700
Subject: [cisco-voip] h323 fxo connect faster
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE8C9@Apollo.wesd.org>

I have a h323 gateway with dial-peers as follows:

 

dial-peer voice 5020 pots

 preference 4

 answer-address 6331

 destination-pattern 9[2-9]..[2-9]......

 incoming called-number .

 direct-inward-dial

 port 0/2/0

 forward-digits all

 

but it acts like it is going through a interdigit time out.  How can I
make it dial quicker?

 

Thanks

 

Scott

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From jgrace at digitelusa.net  Mon Jun 12 16:43:16 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon, 12 Jun 2006 16:43:16 -0400
Subject: [cisco-voip] why will mgcp gw try to use dial peer
Message-ID: <200606121655562.SM01556@3729434770>

We have a mgcp gw that was trying to use a dial peer, we had set up for srst
mode.  All I could see is that when this was happening, the pots dial peer
was on top of the mgcp config.  When we move or reconfigure the dial peer
after all mgcp configs were up and running, they were place at the bottom of
the config and now it works.  What made this happen.  

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Joe_Bell at adp.com  Mon Jun 12 16:56:01 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Mon, 12 Jun 2006 15:56:01 -0500
Subject: [cisco-voip] h323 fxo connect faster
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC067C0@EXCHANGEDS01.ds.ad.adp.com>

This is a good practice for all dial-peers with deterministic lengths.
Put the regular expression for start of string ( a carrot, ^) and end of
string ( a dollar sign, $).  Notice I said deterministic lengths, so .T
or any other unknown length dial-peer acts very goofy when you do this. 

 

 

Your destination pattern becomes this:

 

dial-peer voice 5020 pots

 preference 4

 answer-address 6331

 destination-pattern ^9[2-9]..[2-9]......$

 incoming called-number .

 direct-inward-dial

 port 0/2/0

 forward-digits all

 

This will speed up the match process because your destination pattern is
now specific.

 

Joe

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, June 12, 2006 1:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] h323 fxo connect faster

 

I have a h323 gateway with dial-peers as follows:

 

dial-peer voice 5020 pots

 preference 4

 answer-address 6331

 destination-pattern 9[2-9]..[2-9]......

 incoming called-number .

 direct-inward-dial

 port 0/2/0

 forward-digits all

 

but it acts like it is going through a interdigit time out.  How can I
make it dial quicker?

 

Thanks

 

Scott



This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From Scott.Voll at wesd.org  Mon Jun 12 17:22:31 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 12 Jun 2006 14:22:31 -0700
Subject: [cisco-voip] h323 fxo connect faster
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE8EF@Apollo.wesd.org>


I just did that and I'm still going slow.

I have pots dial peers with the following destination patterns and still having issues.

?destination-pattern ^9[2-9]..[2-9]......$
?destination-pattern ^91[2-9]..[2-9]......$
?destination-pattern ^9...$

Any other ideas?

I also included the debug dial-peer if anyone can give me any direction.? I have 12 dial peers with the same destination pattern as I want it to use one of the 12 ports to dial out.? 

IE>
dial-peer voice 5001 pots
?preference 10
?answer-address 6331
?destination-pattern ^9[2-9]..[2-9]......$
?incoming called-number .
?direct-inward-dial
?port 0/0/1
?forward-digits all
!
dial-peer voice 5002 pots
?preference 10
?answer-address 6331
?destination-pattern ^9[2-9]..[2-9]......$
?incoming called-number .
?direct-inward-dial
?port 0/0/2
?forward-digits all


Is there a different way to do that?


Scott







Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,
?? Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
?? Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911
Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE
Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=95035084911, Expanded String=95035084911, Calling Number=
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5000 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5001 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5002 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5003 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5010 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5011 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5012 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5013 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5020 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5021 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5022 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5023 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5100 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5101 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5102 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5103 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5110 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5111 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5112 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5113 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5120 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5121 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5122 Is Matched
Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5123 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5200 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5201 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5202 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5203 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5210 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5211 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5212 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5213 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5220 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5221 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5222 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5223 Is Matched
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ANSWER; Calling Number=6343
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Result=NO_MATCH(-1) After All Match Rules Attempt
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,
?? Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
?? Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=95035084911, Expanded String=95035084911, Calling Number=
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5000 Is Matched
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5001 Is Matched
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5002 Is Matched
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5003 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5010 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5011 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5012 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5013 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5020 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5021 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5022 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5023 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5100 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5101 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5102 Is Matched
Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5103 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5110 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5111 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5112 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5113 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5120 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5121 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5122 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5123 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5200 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5201 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5202 Is Matched
Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5203 Is Matched
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5210 Is Matched
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5211 Is Matched
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5212 Is Matched
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5213 Is Matched
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5220 Is Matched
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5221 Is Matched
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5222 Is Matched
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Re
Schlador#sult=Success(0); Incoming Dial-peer=5223 Is Matched
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ANSWER; Calling Number=6343
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Result=NO_MATCH(-1) After All Match Rules Attempt
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926
Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchPeersCore:
?? Calling Number=, Called Number=95035084911, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchPeersCore:
?? Match Rule=DP_MATCH_DEST; Called Number=95035084911
Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=95035084911, Expanded String=95035084911, Calling Number=
?? Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5000 Is Matched
Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5001 Is Matched
Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5002 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5003 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5010 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5011 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5012 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5013 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5020 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5021 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5022 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5023 Is Matched
Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/dpMatchPeersCore:
?? Result=Success(0) after DP_MATCH_DEST
Jun 12 21:13:46.385: //-1/80CBD3252300/DPM/dpMatchPeersMoreArg:
?? Result=SUCCESS(0)
?? List of Matched Outgoing Dial-peer(s):
???? 1: Dial-peer Tag=5023
???? 2: Dial-peer Tag=5022
???? 3: Dial-peer Tag=5021
???? 4: Dial-peer Tag=5020
???? 5: Dial-peer Tag=5013
???? 6: Dial-peer Tag=5012
???? 7: Dial-peer Tag=5011
???? 8: Dial-peer Tag=5010
???? 9: Dial-peer Tag=5003
???? 10: Dial-peer Tag=5001
???? 11: Dial-peer Tag=5002
???? 12: Dial-peer Tag=5000
________________________________________
From: Bell, Joe [mailto:Joe_Bell at adp.com] 
Sent: Monday, June 12, 2006 1:56 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] h323 fxo connect faster

This is a good practice for all dial-peers with deterministic lengths.? Put the regular expression for start of string ( a carrot, ^) and end of string ( a dollar sign, $).? Notice I said deterministic lengths, so .T or any other unknown length dial-peer acts very goofy when you do this. 


Your destination pattern becomes this:

dial-peer voice 5020 pots
?preference 4
?answer-address 6331
?destination-pattern ^9[2-9]..[2-9]......$
?incoming called-number .
?direct-inward-dial
?port 0/2/0
?forward-digits all

This will speed up the match process because your destination pattern is now specific.

Joe

________________________________________
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, June 12, 2006 1:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] h323 fxo connect faster

I have a h323 gateway with dial-peers as follows:

dial-peer voice 5020 pots
?preference 4
?answer-address 6331
?destination-pattern 9[2-9]..[2-9]......
?incoming called-number .
?direct-inward-dial
?port 0/2/0
?forward-digits all

but it acts like it is going through a interdigit time out.? How can I make it dial quicker?

Thanks

Scott
________________________________________
This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. 


From Scott.Voll at wesd.org  Mon Jun 12 17:21:04 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 12 Jun 2006 14:21:04 -0700
Subject: [cisco-voip] h323 fxo connect faster
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE8ED@Apollo.wesd.org>

I just did that and I'm still going slow.

 

I have pots dial peers with the following destination patterns and still
having issues.

 

 destination-pattern ^9[2-9]..[2-9]......$

 destination-pattern ^91[2-9]..[2-9]......$

 destination-pattern ^9...$

 

Any other ideas?

 

I also included the debug dial-peer if anyone can give me any direction.
I have 12 dial peers with the same destination pattern as I want it to
use one of the 12 ports to dial out.  

 

IE>

dial-peer voice 5001 pots

 preference 10

 answer-address 6331

 destination-pattern ^9[2-9]..[2-9]......$

 incoming called-number .

 direct-inward-dial

 port 0/0/1

 forward-digits all

!

dial-peer voice 5002 pots

 preference 10

 answer-address 6331

 destination-pattern ^9[2-9]..[2-9]......$

 incoming called-number .

 direct-inward-dial

 port 0/0/2

 forward-digits all

 

 

Is there a different way to do that?

 

 

Scott

 

 

 

 

 

 

 

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=95035084911, Expanded String=95035084911, Calling Number=

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5000 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5001 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5002 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5003 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5010 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5011 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5012 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5013 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5020 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5021 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5022 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5023 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5100 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5101 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5102 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5103 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5110 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5111 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5112 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5113 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5120 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5121 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5122 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5123 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5200 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5201 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5202 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5203 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5210 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5211 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5212 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5213 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5220 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5221 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5222 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5223 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_ANSWER; Calling Number=6343

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=, Expanded String=, Calling Number=6343T

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:

   Result=-1

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=, Expanded String=, Calling Number=6343T

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchCore:

   Result=-1

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

Jun 12 21:13:46.357:
//-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=95035084911, Expanded String=95035084911, Calling Number=

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5000 Is Matched

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5001 Is Matched

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5002 Is Matched

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5003 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5010 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5011 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5012 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5013 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5020 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5021 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5022 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5023 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5100 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5101 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5102 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5103 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5110 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5111 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5112 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5113 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5120 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5121 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5122 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5123 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5200 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5201 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5202 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5203 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5210 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5211 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5212 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5213 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5220 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5221 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5222 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Re

Schlador#sult=Success(0); Incoming Dial-peer=5223 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_ANSWER; Calling Number=6343

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=, Expanded String=, Calling Number=6343T

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:

   Result=-1

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=, Expanded String=, Calling Number=6343T

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:

   Result=-1

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

Jun 12 21:13:46.373:
//-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=95035084911, Peer Info
Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=95035084911

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=95035084911, Expanded String=95035084911, Calling Number=

   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5000 Is Matched

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5001 Is Matched

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5002 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5003 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5010 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5011 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5012 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5013 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5020 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5021 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5022 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5023 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jun 12 21:13:46.385: //-1/80CBD3252300/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=5023

     2: Dial-peer Tag=5022

     3: Dial-peer Tag=5021

     4: Dial-peer Tag=5020

     5: Dial-peer Tag=5013

     6: Dial-peer Tag=5012

     7: Dial-peer Tag=5011

     8: Dial-peer Tag=5010

     9: Dial-peer Tag=5003

     10: Dial-peer Tag=5001

     11: Dial-peer Tag=5002

     12: Dial-peer Tag=5000

________________________________

From: Bell, Joe [mailto:Joe_Bell at adp.com] 
Sent: Monday, June 12, 2006 1:56 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] h323 fxo connect faster

 

This is a good practice for all dial-peers with deterministic lengths.
Put the regular expression for start of string ( a carrot, ^) and end of
string ( a dollar sign, $).  Notice I said deterministic lengths, so .T
or any other unknown length dial-peer acts very goofy when you do this. 

 

 

Your destination pattern becomes this:

 

dial-peer voice 5020 pots

 preference 4

 answer-address 6331

 destination-pattern ^9[2-9]..[2-9]......$

 incoming called-number .

 direct-inward-dial

 port 0/2/0

 forward-digits all

 

This will speed up the match process because your destination pattern is
now specific.

 

Joe

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, June 12, 2006 1:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] h323 fxo connect faster

 

I have a h323 gateway with dial-peers as follows:

 

dial-peer voice 5020 pots

 preference 4

 answer-address 6331

 destination-pattern 9[2-9]..[2-9]......

 incoming called-number .

 direct-inward-dial

 port 0/2/0

 forward-digits all

 

but it acts like it is going through a interdigit time out.  How can I
make it dial quicker?

 

Thanks

 

Scott

________________________________

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From Joe_Bell at adp.com  Mon Jun 12 17:50:00 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Mon, 12 Jun 2006 16:50:00 -0500
Subject: [cisco-voip] h323 fxo connect faster
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC06846@EXCHANGEDS01.ds.ad.adp.com>

I can think of only one thing left to try. Set your dial-peer
preferences sequential (1-11) instead of all 10.  That way the election
process for h323 and which dial-peer / ports to use is removed and your
still using all of your ports.  I believe what your are running into is
the inherent weakness of h323 and that is speed of setup to like
dial-peers or even across networks.  Setting up a call with lots of
decision points in h323 is inherently message intensive.  If that
doesn't work, I don't know.

 

Joe

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, June 12, 2006 2:21 PM
To: Bell, Joe; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] h323 fxo connect faster

 

I just did that and I'm still going slow.

 

I have pots dial peers with the following destination patterns and still
having issues.

 

 destination-pattern ^9[2-9]..[2-9]......$

 destination-pattern ^91[2-9]..[2-9]......$

 destination-pattern ^9...$

 

Any other ideas?

 

I also included the debug dial-peer if anyone can give me any direction.
I have 12 dial peers with the same destination pattern as I want it to
use one of the 12 ports to dial out.  

 

IE>

dial-peer voice 5001 pots

 preference 10

 answer-address 6331

 destination-pattern ^9[2-9]..[2-9]......$

 incoming called-number .

 direct-inward-dial

 port 0/0/1

 forward-digits all

!

dial-peer voice 5002 pots

 preference 10

 answer-address 6331

 destination-pattern ^9[2-9]..[2-9]......$

 incoming called-number .

 direct-inward-dial

 port 0/0/2

 forward-digits all

 

 

Is there a different way to do that?

 

 

Scott

 

 

 

 

 

 

 

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=95035084911, Expanded String=95035084911, Calling Number=

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5000 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5001 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5002 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5003 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5010 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5011 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5012 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5013 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5020 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5021 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5022 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5023 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5100 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5101 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5102 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5103 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5110 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5111 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5112 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5113 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5120 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5121 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5122 Is Matched

Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5123 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5200 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5201 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5202 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5203 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5210 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5211 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5212 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5213 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5220 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5221 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5222 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5223 Is Matched

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_ANSWER; Calling Number=6343

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=, Expanded String=, Calling Number=6343T

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:

   Result=-1

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=, Expanded String=, Calling Number=6343T

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchCore:

   Result=-1

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

Jun 12 21:13:46.357:
//-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=95035084911, Expanded String=95035084911, Calling Number=

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5000 Is Matched

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5001 Is Matched

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5002 Is Matched

Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5003 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5010 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5011 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5012 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5013 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5020 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5021 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5022 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5023 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5100 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5101 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5102 Is Matched

Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5103 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5110 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5111 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5112 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5113 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5120 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5121 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5122 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5123 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5200 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5201 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5202 Is Matched

Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5203 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5210 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5211 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5212 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5213 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5220 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5221 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Incoming Dial-peer=5222 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:

   Re

Schlador#sult=Success(0); Incoming Dial-peer=5223 Is Matched

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_ANSWER; Calling Number=6343

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=, Expanded String=, Calling Number=6343T

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:

   Result=-1

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:

   Is Incoming=TRUE, Number Expansion=FALSE

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=, Expanded String=, Calling Number=6343T

   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:

   Result=-1

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392

Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

Jun 12 21:13:46.373:
//-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=95035084911, Peer Info
Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=95035084911

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchCore:

   Dial String=95035084911, Expanded String=95035084911, Calling Number=

   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5000 Is Matched

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5001 Is Matched

Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5002 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5003 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5010 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5011 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5012 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5013 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5020 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5021 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5022 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:

   Result=Success(0); Outgoing Dial-peer=5023 Is Matched

Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

Jun 12 21:13:46.385: //-1/80CBD3252300/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=5023

     2: Dial-peer Tag=5022

     3: Dial-peer Tag=5021

     4: Dial-peer Tag=5020

     5: Dial-peer Tag=5013

     6: Dial-peer Tag=5012

     7: Dial-peer Tag=5011

     8: Dial-peer Tag=5010

     9: Dial-peer Tag=5003

     10: Dial-peer Tag=5001

     11: Dial-peer Tag=5002

     12: Dial-peer Tag=5000

________________________________

From: Bell, Joe [mailto:Joe_Bell at adp.com] 
Sent: Monday, June 12, 2006 1:56 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] h323 fxo connect faster

 

This is a good practice for all dial-peers with deterministic lengths.
Put the regular expression for start of string ( a carrot, ^) and end of
string ( a dollar sign, $).  Notice I said deterministic lengths, so .T
or any other unknown length dial-peer acts very goofy when you do this. 

 

 

Your destination pattern becomes this:

 

dial-peer voice 5020 pots

 preference 4

 answer-address 6331

 destination-pattern ^9[2-9]..[2-9]......$

 incoming called-number .

 direct-inward-dial

 port 0/2/0

 forward-digits all

 

This will speed up the match process because your destination pattern is
now specific.

 

Joe

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, June 12, 2006 1:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] h323 fxo connect faster

 

I have a h323 gateway with dial-peers as follows:

 

dial-peer voice 5020 pots

 preference 4

 answer-address 6331

 destination-pattern 9[2-9]..[2-9]......

 incoming called-number .

 direct-inward-dial

 port 0/2/0

 forward-digits all

 

but it acts like it is going through a interdigit time out.  How can I
make it dial quicker?

 

Thanks

 

Scott

________________________________

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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 bzayed80 at yahoo.com  Mon Jun 12 20:36:32 2006
From: bzayed80 at yahoo.com (Bassem Zayed)
Date: Mon, 12 Jun 2006 17:36:32 -0700 (PDT)
Subject: [cisco-voip] need script file for autoattendant
Message-ID: <20060613003632.52004.qmail@web31102.mail.mud.yahoo.com>

hi everyone,
   
  I need to change the script file "aa_sample1.aef". I need to change the way the autoattendant  responds. i've already created the messages to greet the caller and i know how to insert them. I tried the Cisco unity express editor to edit the aa_sample1 but i'm not good at it!!. I really tried to edit the example but no good.
   
  I need the auto att to:
  1)greet the caller then tell him to just enter the (extention of the employer or 0 for the operator)
  2)if the extention is:
   "busy, not valid" it should continue as the default in the autoattendant.
   
  if you need me to mail you the example plz tell me so i can attach it.
   
  Its just taking me forever and i need help on this issue,please.
  Thank you all and have a nice day.
   
  Bassem.
   
   
   

 __________________________________________________
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From cidar at acelerate.com  Mon Jun 12 21:30:46 2006
From: cidar at acelerate.com (Cidar Ramirez)
Date: Mon, 12 Jun 2006 19:30:46 -0600
Subject: [cisco-voip] voip billing...solved
Message-ID: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAATb16yFsuxEqckYCsMTcHdMKAAAAQAAAA28vjezKGKUiVpe3mETikbwEAAAAA@acelerate.com>

The answer is freeradius + postgresql

I'm mounting that using 1.1.2 version 

Cidar

-----Mensaje original-----
De: Cidar Ramirez [mailto:cidar at acelerate.com] 
Enviado el: Wednesday, June 07, 2006 1:37 PM
Para: 'cisco-voip at puck.nether.net'
Asunto: voip billing

Anybody knows about a open source billing software for cisco as5300 cdrs ?

Thanks

Cid



From bcarter at sentinel.com  Mon Jun 12 22:42:10 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 12 Jun 2006 21:42:10 -0500
Subject: [cisco-voip] h323 fxo connect faster
Message-ID: <6A0744639B86DD459433108DBF101D26FC62BD@MAIL01.sentinel.com>

 Trunk Groups

-----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, June 12, 2006 4:23 PM
To: Bell, Joe; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] h323 fxo connect faster


I just did that and I'm still going slow.

I have pots dial peers with the following destination patterns and still having issues.

?destination-pattern ^9[2-9]..[2-9]......$
?destination-pattern ^91[2-9]..[2-9]......$
?destination-pattern ^9...$

Any other ideas?

I also included the debug dial-peer if anyone can give me any direction.? I have 12 dial peers with the same destination pattern as I want it to use one of the 12 ports to dial out.? 

IE>
dial-peer voice 5001 pots
?preference 10
?answer-address 6331
?destination-pattern ^9[2-9]..[2-9]......$
?incoming called-number .
?direct-inward-dial
?port 0/0/1
?forward-digits all
!
dial-peer voice 5002 pots
?preference 10
?answer-address 6331
?destination-pattern ^9[2-9]..[2-9]......$
?incoming called-number .
?direct-inward-dial
?port 0/0/2
?forward-digits all


Is there a different way to do that?


Scott







Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,
?? Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
?? Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911 Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=95035084911, Expanded String=95035084911, Calling Number=
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5000 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5001 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5002 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5003 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5010 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5011 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5012 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5013 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5020 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5021 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5022 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5023 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5100 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5101 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5102 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5103 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5110 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5111 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5112 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5113 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5120 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5121 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5122 Is Matched Jun 12 21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5123 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5200 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5201 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5202 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5203 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5210 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5211 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5212 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5213 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5220 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5221 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5222 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5223 Is Matched Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ANSWER; Calling Number=6343 Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343 Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Result=NO_MATCH(-1) After All Match Rules Attempt Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,
?? Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
?? Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911 Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=95035084911, Expanded String=95035084911, Calling Number=
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5000 Is Matched Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5001 Is Matched Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5002 Is Matched Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5003 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5010 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5011 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5012 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5013 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5020 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5021 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5022 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5023 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5100 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5101 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5102 Is Matched Jun 12 21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5103 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5110 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5111 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5112 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5113 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5120 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5121 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5122 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5123 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5200 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5201 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5202 Is Matched Jun 12 21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5203 Is Matched Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5210 Is Matched Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5211 Is Matched Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5212 Is Matched Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5213 Is Matched Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5220 Is Matched Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5221 Is Matched Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5222 Is Matched Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Re
Schlador#sult=Success(0); Incoming Dial-peer=5223 Is Matched Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ANSWER; Calling Number=6343 Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343 Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Result=NO_MATCH(-1) After All Match Rules Attempt Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926
Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchPeersCore:
?? Calling Number=, Called Number=95035084911, Peer Info Type=DIALPEER_INFO_SPEECH Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchPeersCore:
?? Match Rule=DP_MATCH_DEST; Called Number=95035084911 Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=95035084911, Expanded String=95035084911, Calling Number=
?? Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5000 Is Matched Jun 12 21:
13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5001 Is Matched Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5002 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5003 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5010 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5011 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5012 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5013 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5020 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5021 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5022 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5023 Is Matched Jun 12 21:13:46.381: //-1/80CBD3252300/DPM/dpMatchPeersCore:
?? Result=Success(0) after DP_MATCH_DEST Jun 12 21:13:46.385: //-1/80CBD3252300/DPM/dpMatchPeersMoreArg:
?? Result=SUCCESS(0)
?? List of Matched Outgoing Dial-peer(s):
???? 1: Dial-peer Tag=5023
???? 2: Dial-peer Tag=5022
???? 3: Dial-peer Tag=5021
???? 4: Dial-peer Tag=5020
???? 5: Dial-peer Tag=5013
???? 6: Dial-peer Tag=5012
???? 7: Dial-peer Tag=5011
???? 8: Dial-peer Tag=5010
???? 9: Dial-peer Tag=5003
???? 10: Dial-peer Tag=5001
???? 11: Dial-peer Tag=5002
???? 12: Dial-peer Tag=5000
________________________________________
From: Bell, Joe [mailto:Joe_Bell at adp.com]
Sent: Monday, June 12, 2006 1:56 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] h323 fxo connect faster

This is a good practice for all dial-peers with deterministic lengths.? Put the regular expression for start of string ( a carrot, ^) and end of string ( a dollar sign, $).? Notice I said deterministic lengths, so .T or any other unknown length dial-peer acts very goofy when you do this. 


Your destination pattern becomes this:

dial-peer voice 5020 pots
?preference 4
?answer-address 6331
?destination-pattern ^9[2-9]..[2-9]......$
?incoming called-number .
?direct-inward-dial
?port 0/2/0
?forward-digits all

This will speed up the match process because your destination pattern is now specific.

Joe

________________________________________
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, June 12, 2006 1:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] h323 fxo connect faster

I have a h323 gateway with dial-peers as follows:

dial-peer voice 5020 pots
?preference 4
?answer-address 6331
?destination-pattern 9[2-9]..[2-9]......
?incoming called-number .
?direct-inward-dial
?port 0/2/0
?forward-digits all

but it acts like it is going through a interdigit time out.? How can I make it dial quicker?

Thanks

Scott
________________________________________
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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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 
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opinions of Sentinel Technologies or any of its affiliates, employees, directors, officers or shareholders.


From nathan.a.reeves at gmail.com  Mon Jun 12 23:34:18 2006
From: nathan.a.reeves at gmail.com (Nathan Reeves)
Date: Tue, 13 Jun 2006 11:34:18 +0800
Subject: [cisco-voip] 7936 Booting issues after CCM Upgrade
Message-ID: <62833d8a0606122034s3599f072i7d166ffe4ba475a8@mail.gmail.com>

Upgraded our CCM to 4.1.3(Sr3a) from 4.1.3(sr2) without any real issues.
Just been  finding that the 7935 and 7936 phones we've got seem to be having
issues since.

Main issue at the moment is 7936 unity which either keep rebooting
continuously or never actually boot up but come up with an error message
about not being able to contact the TFTP Server.

Anyone else seen issues like these?  Any ideas if there's a way of resetting
the 7936's to factory default without using the menu option?  I'm not able
to get to the menu to reset the unit, but want to try and do it to see if it
solves my issue

TIA
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From ROHARVEY at flemingc.on.ca  Tue Jun 13 09:31:58 2006
From: ROHARVEY at flemingc.on.ca (Robert Harvey)
Date: Tue, 13 Jun 2006 09:31:58 -0400
Subject: [cisco-voip] Serial Number from Call Manager (Frazee, Timothy)
Message-ID: <s48e8617.043@freedom7.flemingc.on.ca>

Hi Eby,

I'm not sure about CCM5.x, but with all previous versions I have just
queried the dbo.device table via an ODBC connection to the publisher. 
This is incredibly easy, flexible and the price is right :)  Email me if
you want further instruction on the process.

Robert.


Hi,
  Is it possible to get the phone's serial number from call manager
using BAT tool? I know how to get it one by one but it will take a
long
time to go to each phone and copying it. Any Ideas?

Thanks,

Eby 

From rratliff at cisco.com  Tue Jun 13 09:45:56 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 13 Jun 2006 09:45:56 -0400
Subject: [cisco-voip] Serial Number from Call Manager (Frazee, Timothy)
In-Reply-To: <s48e8617.043@freedom7.flemingc.on.ca>
References: <s48e8617.043@freedom7.flemingc.on.ca>
Message-ID: <399D7AD8-B7BA-44BA-9D2D-92E4E4E2D5D6@cisco.com>

Are you getting the mac address or serial number?

Serial numbers are not stored in the database.

-Ryan

On Jun 13, 2006, at 9:31 AM, Robert Harvey wrote:

Hi Eby,

I'm not sure about CCM5.x, but with all previous versions I have just
queried the dbo.device table via an ODBC connection to the publisher.
This is incredibly easy, flexible and the price is right :)  Email me if
you want further instruction on the process.

Robert.


Hi,
   Is it possible to get the phone's serial number from call manager
using BAT tool? I know how to get it one by one but it will take a
long
time to go to each phone and copying it. Any Ideas?

Thanks,

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


From JOlson at mlnusa.com  Tue Jun 13 10:07:54 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Tue, 13 Jun 2006 10:07:54 -0400
Subject: [cisco-voip] CRA version 3.5
Message-ID: <FE2D092F8B1516488B99342551AACFD41CAF3201@mlnusa-exch1.mlnusa.com>

I want to learn more about creating scripts in CRA 3.5
Does anyone have any scripts that I could use for learning purposes?



Thanks 
Jeff 

********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

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From jburton at NETechCorp.com  Tue Jun 13 10:42:47 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Tue, 13 Jun 2006 10:42:47 -0400
Subject: [cisco-voip] AD plugin on a W2k3 Domain
Message-ID: <9F9D5C20-D5F0-4FC4-9C48-2C31205DDA76@mimectl>

 I'm trying to install the AD plugin and keep getting the following error:

Error:Setup failed to connect. Please enter your Host Name, Port Number, Directory Administrator DN and Password again!

I've verified these 100 times over and still no luck.  This is the first time I've tried this to a widnows 2003 domain.  Any ideas?

Thanks in advance!!
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From Scott.Voll at wesd.org  Tue Jun 13 10:50:21 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 13 Jun 2006 07:50:21 -0700
Subject: [cisco-voip] AD plugin on a W2k3 Domain
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org>

Is there any pix between the CM and AD?  If so, we ran into a problem
with the fixup 389.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Tuesday, June 13, 2006 7:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] AD plugin on a W2k3 Domain

 

 I'm trying to install the AD plugin and keep getting the following
error:

 

Error:Setup failed to connect. Please enter your Host Name, Port Number,
Directory Administrator DN and Password again!

 

I've verified these 100 times over and still no luck.  This is the first
time I've tried this to a widnows 2003 domain.  Any ideas?

 

Thanks in advance!!

 

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From jburton at NETechCorp.com  Tue Jun 13 10:52:10 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Tue, 13 Jun 2006 10:52:10 -0400
Subject: [cisco-voip] AD plugin on a W2k3 Domain
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org>
Message-ID: <04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>

Nope, just a router on a stick.



From: Voll, Scott
Sent: Tue 6/13/2006 10:50 AM
To: Jason Burton; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] AD plugin on a W2k3 Domain


Is there any pix between the CM and AD?  If so, we ran into a problem with the fixup 389.




From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Tuesday, June 13, 2006 7:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] AD plugin on a W2k3 Domain

 I'm trying to install the AD plugin and keep getting the following error:

Error:Setup failed to connect. Please enter your Host Name, Port Number, Directory Administrator DN and Password again!

I've verified these 100 times over and still no luck.  This is the first time I've tried this to a widnows 2003 domain.  Any ideas?

Thanks in advance!!
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From rratliff at cisco.com  Tue Jun 13 10:56:23 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 13 Jun 2006 10:56:23 -0400
Subject: [cisco-voip] AD plugin on a W2k3 Domain
In-Reply-To: <04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>
References: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org>
	<04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>
Message-ID: <F2C5385D-60B3-47A5-8231-06B37A0905E3@cisco.com>

Get a sniffer capture, look at the ldap packets and see what's going on.

-Ryan

On Jun 13, 2006, at 10:52 AM, Jason Burton wrote:

Nope, just a router on a stick.

From: Voll, Scott
Sent: Tue 6/13/2006 10:50 AM
To: Jason Burton; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] AD plugin on a W2k3 Domain

Is there any pix between the CM and AD?  If so, we ran into a problem  
with the fixup 389.


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Tuesday, June 13, 2006 7:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] AD plugin on a W2k3 Domain


  I'm trying to install the AD plugin and keep getting the following  
error:


Error:Setup failed to connect. Please enter your Host Name, Port  
Number, Directory Administrator DN and Password again!


I've verified these 100 times over and still no luck.  This is the  
first time I've tried this to a widnows 2003 domain.  Any ideas?


Thanks in advance!!


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


From kthorngr at cisco.com  Tue Jun 13 11:01:41 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Tue, 13 Jun 2006 11:01:41 -0400
Subject: [cisco-voip] AD plugin on a W2k3 Domain
In-Reply-To: <04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>
References: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org>
	<04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>
Message-ID: <3f868042bd2ac6279fd820f9fda963ef@cisco.com>

The quickest way to isolate the issue is with a packet capture of the 
Publisher when doing the AD Plugin.  Looking at the LDAP packets should 
help you see what the failure is.

Kevin
On Jun 13, 2006, at 10:52 AM, Jason Burton wrote:

> Nope, just a router on a stick.
>
> From: Voll, Scott
> Sent: Tue 6/13/2006 10:50 AM
> To: Jason Burton; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] AD plugin on a W2k3 Domain
>
> Is there any pix between the CM and AD?? If so, we ran into a problem 
> with the fixup 389.
> ?
>
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
> Sent: Tuesday, June 13, 2006 7:43 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] AD plugin on a W2k3 Domain
> ?
> ?I'm trying to install the AD plugin and keep getting the following 
> error:
> ?
> Error:Setup failed to connect. Please enter your Host Name, Port 
> Number, Directory Administrator DN and Password again!
> ?
> I've verified these 100 times over and still no luck.? This is the 
> first time I've tried this to a widnows 2003 domain.? Any ideas?
> ?
> 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 jburton at NETechCorp.com  Tue Jun 13 11:08:29 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Tue, 13 Jun 2006 11:08:29 -0400
Subject: [cisco-voip] AD plugin on a W2k3 Domain
In-Reply-To: <3f868042bd2ac6279fd820f9fda963ef@cisco.com>
References: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org>
	<04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>,
	<3f868042bd2ac6279fd820f9fda963ef@cisco.com>
Message-ID: <4C6759F1-6EF6-466D-A19E-D76002178F7B@mimectl>

Thanks, downloading Ethereal now.



From: Kevin Thorngren
Sent: Tue 6/13/2006 11:01 AM
To: Jason Burton
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain


The quickest way to isolate the issue is with a packet capture of the 
Publisher when doing the AD Plugin.  Looking at the LDAP packets should 
help you see what the failure is.

Kevin
On Jun 13, 2006, at 10:52 AM, Jason Burton wrote:

> Nope, just a router on a stick.
>
> From: Voll, Scott
> Sent: Tue 6/13/2006 10:50 AM
> To: Jason Burton; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] AD plugin on a W2k3 Domain
>
> Is there any pix between the CM and AD?  If so, we ran into a problem 
> with the fixup 389.
>  
>
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
> Sent: Tuesday, June 13, 2006 7:43 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] AD plugin on a W2k3 Domain
>  
>  I'm trying to install the AD plugin and keep getting the following 
> error:
>  
> Error:Setup failed to connect. Please enter your Host Name, Port 
> Number, Directory Administrator DN and Password again!
>  
> I've verified these 100 times over and still no luck.  This is the 
> first time I've tried this to a widnows 2003 domain.  Any ideas?
>  
> 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 JHorton at jackhenry.com  Tue Jun 13 11:38:49 2006
From: JHorton at jackhenry.com (Josh Horton)
Date: Tue, 13 Jun 2006 10:38:49 -0500
Subject: [cisco-voip] Extension Mobility Conversion
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002A9585F@smoxchg.jhacorp.com>

Just in case someone else has the same question... 

Configuring EM for everyone in the organization was not as hard as I had
made it out to be. I simply created User device profiles with the same
extension as the user's phone in the same partition (shared line). 

This way, everyone "could" use EM but no one "had" to. 

Here are the steps I used: 
1) Setup Extension Mobility per Cisco instructions 
2) Use Bat to subscribe every phone to Extension Mobility service. 
3) Export existing phones using BAT. 
4) Use data from exported phones to populate user device profile BAT
template. 
5) Import User Device Profiles with BAT. 

Josh



________________________________________
From: Josh Horton 
Sent: Tuesday, May 23, 2006 3:48 PM
To: cisco-voip at puck.nether.net
Subject: Extension Mobility Conversion

Can someone point me in the right direction or give me some advice
regarding moving everyone on a CCM system to extension mobility? 

My customer has ~150 phones and they were initially setup without
extension mobility. Now, they want to move everyone in the organization
to EM. 
* Can I export the phones in BAT, modify the CSV file and import back
into the system as User Device Profiles?
* Are there any pitfalls when moving everyone to EM?


Thanks,

Josh Horton
Network Engineer
Matrix Network Services (R)
Jack Henry & Associates
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 rratliff at cisco.com  Tue Jun 13 11:41:51 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 13 Jun 2006 11:41:51 -0400
Subject: [cisco-voip] Extension Mobility Conversion
In-Reply-To: <13591CB4B2359F4D96E2606DD6B840A002A9585F@smoxchg.jhacorp.com>
References: <13591CB4B2359F4D96E2606DD6B840A002A9585F@smoxchg.jhacorp.com>
Message-ID: <B2AE9CE5-6676-42EB-9099-4FF4CA3D258D@cisco.com>

Don't forget to subscribe your UDPs to the EM service as well or they  
won't be able to log out :)

Glad to hear it was painless.

-Ryan

On Jun 13, 2006, at 11:38 AM, Josh Horton wrote:

Just in case someone else has the same question...

Configuring EM for everyone in the organization was not as hard as I had
made it out to be. I simply created User device profiles with the same
extension as the user's phone in the same partition (shared line).

This way, everyone "could" use EM but no one "had" to.

Here are the steps I used:
1) Setup Extension Mobility per Cisco instructions
2) Use Bat to subscribe every phone to Extension Mobility service.
3) Export existing phones using BAT.
4) Use data from exported phones to populate user device profile BAT
template.
5) Import User Device Profiles with BAT.

Josh



________________________________________
From: Josh Horton
Sent: Tuesday, May 23, 2006 3:48 PM
To: cisco-voip at puck.nether.net
Subject: Extension Mobility Conversion

Can someone point me in the right direction or give me some advice
regarding moving everyone on a CCM system to extension mobility?

My customer has ~150 phones and they were initially setup without
extension mobility. Now, they want to move everyone in the organization
to EM.
* Can I export the phones in BAT, modify the CSV file and import back
into the system as User Device Profiles?
* Are there any pitfalls when moving everyone to EM?


Thanks,

Josh Horton
Network Engineer
Matrix Network Services (R)
Jack Henry & Associates
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
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jburton at NETechCorp.com  Tue Jun 13 11:46:27 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Tue, 13 Jun 2006 11:46:27 -0400
Subject: [cisco-voip] AD plugin on a W2k3 Domain
In-Reply-To: <4C6759F1-6EF6-466D-A19E-D76002178F7B@mimectl>
References: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org><04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>,
	<3f868042bd2ac6279fd820f9fda963ef@cisco.com>,
	<4C6759F1-6EF6-466D-A19E-D76002178F7B@mimectl>
Message-ID: <AF6E4B63-7833-4B45-B013-483BD0370B3E@mimectl>

Well after doing the packet capture I'm getting invalid credentials error in the Bind result.

It's LDAPErr DSID-0c09334.  What is the correct format for AD Admin DN?

CN=USER, OU=Container, DC=domain, DC=Domain?



From: Jason Burton
Sent: Tue 6/13/2006 11:08 AM
To: Kevin Thorngren
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain


Thanks, downloading Ethereal now.



From: Kevin Thorngren
Sent: Tue 6/13/2006 11:01 AM
To: Jason Burton
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain


The quickest way to isolate the issue is with a packet capture of the 
Publisher when doing the AD Plugin.  Looking at the LDAP packets should 
help you see what the failure is.

Kevin
On Jun 13, 2006, at 10:52 AM, Jason Burton wrote:

> Nope, just a router on a stick.
>
> From: Voll, Scott
> Sent: Tue 6/13/2006 10:50 AM
> To: Jason Burton; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] AD plugin on a W2k3 Domain
>
> Is there any pix between the CM and AD?  If so, we ran into a problem 
> with the fixup 389.
>  
>
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
> Sent: Tuesday, June 13, 2006 7:43 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] AD plugin on a W2k3 Domain
>  
>  I'm trying to install the AD plugin and keep getting the following 
> error:
>  
> Error:Setup failed to connect. Please enter your Host Name, Port 
> Number, Directory Administrator DN and Password again!
>  
> I've verified these 100 times over and still no luck.  This is the 
> first time I've tried this to a widnows 2003 domain.  Any ideas?
>  
> 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 rratliff at cisco.com  Tue Jun 13 11:51:01 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 13 Jun 2006 11:51:01 -0400
Subject: [cisco-voip] AD plugin on a W2k3 Domain
In-Reply-To: <AF6E4B63-7833-4B45-B013-483BD0370B3E@mimectl>
References: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org><04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>,
	<3f868042bd2ac6279fd820f9fda963ef@cisco.com>,
	<4C6759F1-6EF6-466D-A19E-D76002178F7B@mimectl>
	<AF6E4B63-7833-4B45-B013-483BD0370B3E@mimectl>
Message-ID: <F4CB1CFE-C1D4-40BE-8B20-64F829C79DE3@cisco.com>

Try CN=user, CN=container,DC=domain,DC=com.


-Ryan

On Jun 13, 2006, at 11:46 AM, Jason Burton wrote:

Well after doing the packet capture I'm getting invalid credentials  
error in the Bind result.

It's LDAPErr DSID-0c09334.  What is the correct format for AD Admin DN?

CN=USER, OU=Container, DC=domain, DC=Domain?

From: Jason Burton
Sent: Tue 6/13/2006 11:08 AM
To: Kevin Thorngren
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain

Thanks, downloading Ethereal now.

From: Kevin Thorngren
Sent: Tue 6/13/2006 11:01 AM
To: Jason Burton
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain

The quickest way to isolate the issue is with a packet capture of the  
Publisher when doing the AD Plugin. Looking at the LDAP packets  
should help you see what the failure is. Kevin On Jun 13, 2006, at  
10:52 AM, Jason Burton wrote: > Nope, just a router on a stick. > >  
From: Voll, Scott > Sent: Tue 6/13/2006 10:50 AM > To: Jason Burton;  
cisco-voip at puck.nether.net > Subject: RE: [cisco-voip] AD plugin on a  
W2k3 Domain > > Is there any pix between the CM and AD?  If so, we  
ran into a problem > with the fixup 389. >   > > From: cisco-voip- 
bounces at puck.nether.net > [mailto:cisco-voip-bounces at puck.nether.net]  
On Behalf Of Jason Burton > Sent: Tuesday, June 13, 2006 7:43 AM >  
To: cisco-voip at puck.nether.net > Subject: [cisco-voip] AD plugin on a  
W2k3 Domain >   >  I'm trying to install the AD plugin and keep  
getting the following > error: >   > Error:Setup failed to connect.  
Please enter your Host Name, Port > Number, Directory Administrator  
DN and Password again! >   > I've verified these 100 times over and  
still no luck.  This is the > first time I've tried this to a widnows  
2003 domain.  Any ideas? >   > Thanks in advance!! >   
_______________________________________________ > cisco-voip mailing  
list > cisco-voip at puck.nether.net > https://puck.nether.net/mailman/ 
listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From jwydra at Burwood.com  Tue Jun 13 11:52:11 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Tue, 13 Jun 2006 10:52:11 -0500
Subject: [cisco-voip] Can't load 3.3.3 on VMware
Message-ID: <97F472A9F637E44B954D7FBABC800430BC682A@bur-msg-02.Burwood.com>

OK. So I created an ISO from a 3.3.3 Cisco installation disc. I
successfully upgraded the OS to 2000.4.2 using the download from Cisco.
The reg hack is in place. After the update the Spirian key shows
2000.4.2. I mounted the ISO and ran it. It keeps coming back and telling
me that my OS version is not up to date. WTFO??????

 

Jason 

 

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From jburton at NETechCorp.com  Tue Jun 13 11:55:07 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Tue, 13 Jun 2006 11:55:07 -0400
Subject: [cisco-voip] AD plugin on a W2k3 Domain
In-Reply-To: <F4CB1CFE-C1D4-40BE-8B20-64F829C79DE3@cisco.com>
References: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org><04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>,
	<3f868042bd2ac6279fd820f9fda963ef@cisco.com>, 
	<4C6759F1-6EF6-466D-A19E-D76002178F7B@mimectl>
	<AF6E4B63-7833-4B45-B013-483BD0370B3E@mimectl>,
	<F4CB1CFE-C1D4-40BE-8B20-64F829C79DE3@cisco.com>
Message-ID: <2A0FAAA5-90C3-4787-B141-23C72BC63541@mimectl>

Nope same result.



From: Ryan Ratliff
Sent: Tue 6/13/2006 11:51 AM
To: Jason Burton
Cc: Kevin Thorngren; cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain


Try CN=user, CN=container,DC=domain,DC=com.


-Ryan

On Jun 13, 2006, at 11:46 AM, Jason Burton wrote:

Well after doing the packet capture I'm getting invalid credentials  
error in the Bind result.

It's LDAPErr DSID-0c09334.  What is the correct format for AD Admin DN?

CN=USER, OU=Container, DC=domain, DC=Domain?

From: Jason Burton
Sent: Tue 6/13/2006 11:08 AM
To: Kevin Thorngren
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain

Thanks, downloading Ethereal now.

From: Kevin Thorngren
Sent: Tue 6/13/2006 11:01 AM
To: Jason Burton
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain

The quickest way to isolate the issue is with a packet capture of the  
Publisher when doing the AD Plugin. Looking at the LDAP packets  
should help you see what the failure is. Kevin On Jun 13, 2006, at  
10:52 AM, Jason Burton wrote: > Nope, just a router on a stick. > >  
From: Voll, Scott > Sent: Tue 6/13/2006 10:50 AM > To: Jason Burton;  
cisco-voip at puck.nether.net > Subject: RE: [cisco-voip] AD plugin on a  
W2k3 Domain > > Is there any pix between the CM and AD?  If so, we  
ran into a problem > with the fixup 389. >   > > From: cisco-voip- 
bounces at puck.nether.net > [mailto:cisco-voip-bounces at puck.nether.net]  
On Behalf Of Jason Burton > Sent: Tuesday, June 13, 2006 7:43 AM >  
To: cisco-voip at puck.nether.net > Subject: [cisco-voip] AD plugin on a  
W2k3 Domain >   >  I'm trying to install the AD plugin and keep  
getting the following > error: >   > Error:Setup failed to connect.  
Please enter your Host Name, Port > Number, Directory Administrator  
DN and Password again! >   > I've verified these 100 times over and  
still no luck.  This is the > first time I've tried this to a widnows  
2003 domain.  Any ideas? >   > 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 JHorton at jackhenry.com  Tue Jun 13 11:56:28 2006
From: JHorton at jackhenry.com (Josh Horton)
Date: Tue, 13 Jun 2006 10:56:28 -0500
Subject: [cisco-voip] Extension Mobility Conversion
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002A95887@smoxchg.jhacorp.com>

Ryan,

Is there a way to subscribe the UDPs to the EM service in bulk after
they have been imported? 

I couldn't find a way. Had to delete, modify the UDP templates and
re-import.

Josh



-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, June 13, 2006 10:42 AM
To: Josh Horton
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Extension Mobility Conversion

Don't forget to subscribe your UDPs to the EM service as well or they  
won't be able to log out :)

Glad to hear it was painless.

-Ryan

On Jun 13, 2006, at 11:38 AM, Josh Horton wrote:

Just in case someone else has the same question...

Configuring EM for everyone in the organization was not as hard as I had
made it out to be. I simply created User device profiles with the same
extension as the user's phone in the same partition (shared line).

This way, everyone "could" use EM but no one "had" to.

Here are the steps I used:
1) Setup Extension Mobility per Cisco instructions
2) Use Bat to subscribe every phone to Extension Mobility service.
3) Export existing phones using BAT.
4) Use data from exported phones to populate user device profile BAT
template.
5) Import User Device Profiles with BAT.

Josh



________________________________________
From: Josh Horton
Sent: Tuesday, May 23, 2006 3:48 PM
To: cisco-voip at puck.nether.net
Subject: Extension Mobility Conversion

Can someone point me in the right direction or give me some advice
regarding moving everyone on a CCM system to extension mobility?

My customer has ~150 phones and they were initially setup without
extension mobility. Now, they want to move everyone in the organization
to EM.
* Can I export the phones in BAT, modify the CSV file and import back
into the system as User Device Profiles?
* Are there any pitfalls when moving everyone to EM?


Thanks,

Josh Horton
Network Engineer
Matrix Network Services (R)
Jack Henry & Associates
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
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 Scott.Voll at wesd.org  Tue Jun 13 11:57:27 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 13 Jun 2006 08:57:27 -0700
Subject: [cisco-voip] 3560 port shut down
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE9A2@Apollo.wesd.org>

I have 3560's running 12.2(25)SEE1 IOS and when you plug in a device,
then plug in another device, then go back to the first device or a third
device the port does not work.

 

Is this a security feature and if so, how do you turn it off?

 

Thanks

 

Scott

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From netsecuredata at gmail.com  Tue Jun 13 12:00:44 2006
From: netsecuredata at gmail.com (Jorge Evangelista)
Date: Tue, 13 Jun 2006 11:00:44 -0500
Subject: [cisco-voip] Download CCM 5
Message-ID: <de85c96f0606130900y40776d8dibe285287a120a83b@mail.gmail.com>

Anyone knows where I can download Cisco Call Manager 5.0 for linux. I
want to test CCM.
Is there some link at Internet


-- 
"The network is the computer"

From rratliff at cisco.com  Tue Jun 13 12:04:51 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 13 Jun 2006 12:04:51 -0400
Subject: [cisco-voip] Extension Mobility Conversion
In-Reply-To: <13591CB4B2359F4D96E2606DD6B840A002A95887@smoxchg.jhacorp.com>
References: <13591CB4B2359F4D96E2606DD6B840A002A95887@smoxchg.jhacorp.com>
Message-ID: <A863F4D1-F55D-4BC8-AF76-5998E7169454@cisco.com>

I imagine BAT would do it much the same way you did with the other  
phones.

If not then you'd have to go to the UDP page in CCMAdmin and  
subscribe there.

-Ryan

On Jun 13, 2006, at 11:56 AM, Josh Horton wrote:

Ryan,

Is there a way to subscribe the UDPs to the EM service in bulk after
they have been imported?

I couldn't find a way. Had to delete, modify the UDP templates and
re-import.

Josh



-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, June 13, 2006 10:42 AM
To: Josh Horton
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Extension Mobility Conversion

Don't forget to subscribe your UDPs to the EM service as well or they
won't be able to log out :)

Glad to hear it was painless.

-Ryan

On Jun 13, 2006, at 11:38 AM, Josh Horton wrote:

Just in case someone else has the same question...

Configuring EM for everyone in the organization was not as hard as I had
made it out to be. I simply created User device profiles with the same
extension as the user's phone in the same partition (shared line).

This way, everyone "could" use EM but no one "had" to.

Here are the steps I used:
1) Setup Extension Mobility per Cisco instructions
2) Use Bat to subscribe every phone to Extension Mobility service.
3) Export existing phones using BAT.
4) Use data from exported phones to populate user device profile BAT
template.
5) Import User Device Profiles with BAT.

Josh



________________________________________
From: Josh Horton
Sent: Tuesday, May 23, 2006 3:48 PM
To: cisco-voip at puck.nether.net
Subject: Extension Mobility Conversion

Can someone point me in the right direction or give me some advice
regarding moving everyone on a CCM system to extension mobility?

My customer has ~150 phones and they were initially setup without
extension mobility. Now, they want to move everyone in the organization
to EM.
* Can I export the phones in BAT, modify the CSV file and import back
into the system as User Device Profiles?
* Are there any pitfalls when moving everyone to EM?


Thanks,

Josh Horton
Network Engineer
Matrix Network Services (R)
Jack Henry & Associates
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
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 Scott.Voll at wesd.org  Tue Jun 13 12:07:52 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 13 Jun 2006 09:07:52 -0700
Subject: [cisco-voip] Download CCM 5
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEE9B1@Apollo.wesd.org>

If you have a SAU contract you can order it on the PUT tool.  Otherwise
I don't know of another way.

Scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jorge
Evangelista
Sent: Tuesday, June 13, 2006 9:01 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Download CCM 5

Anyone knows where I can download Cisco Call Manager 5.0 for linux. I
want to test CCM.
Is there some link at Internet


-- 
"The network is the computer"
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From CORTIZ at broward.org  Tue Jun 13 12:09:43 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue, 13 Jun 2006 12:09:43 -0400
Subject: [cisco-voip] Attendant Console Login Causes reboot
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC347E@MREXVS01.bc.broward.cty>

I had attendant console installed on 5 PC's at a location.  3 of them
are working fine.  2 other machines reboot as soon as the user logs in
to Attendant Console.  We are having the PC's checked out (no solution
found yet), but I was wondering if anyone has seen something similar
before.

 

 

Carlos

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From Timothy_Frazee at adp.com  Tue Jun 13 12:13:38 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Tue, 13 Jun 2006 11:13:38 -0500
Subject: [cisco-voip] Download CCM 5
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC06CC9@EXCHANGEDS01.ds.ad.adp.com>

Officially, you need to talk to your friendly Cisco sales rep.

Timothy Frazee 
Network Solutions
314.249.8697 cellular phone

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jorge
Evangelista
Sent: Tuesday, June 13, 2006 11:01 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Download CCM 5

Anyone knows where I can download Cisco Call Manager 5.0 for linux. I
want to test CCM.
Is there some link at Internet


-- 
"The network is the computer"
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.


From Timothy_Frazee at adp.com  Tue Jun 13 12:18:14 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Tue, 13 Jun 2006 11:18:14 -0500
Subject: [cisco-voip] Can't load 3.3.3 on VMware
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC06CDA@EXCHANGEDS01.ds.ad.adp.com>

Can you post what you put in for your reg hack?

 

This is what I always put in:

 

Windows Registry Editor Version 5.00

[HKEY_LOCAL_MACHINE\SOFTWARE\Cisco Systems\Model]
"Hardware"="MCS7835"
"Memory"="1024"
"Speed"="733"
[HKEY_LOCAL_MACHINE\SOFTWARE\Spirian\System Info\OS Image] 
"Version"="2000.2.4"

 

 

Timothy Frazee 

Network Solutions

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wydra, Jason
Sent: Tuesday, June 13, 2006 10:52 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Can't load 3.3.3 on VMware

 

OK. So I created an ISO from a 3.3.3 Cisco installation disc. I
successfully upgraded the OS to 2000.4.2 using the download from Cisco.
The reg hack is in place. After the update the Spirian key shows
2000.4.2. I mounted the ISO and ran it. It keeps coming back and telling
me that my OS version is not up to date. WTFO??????

 

Jason 

 



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 kthorngr at cisco.com  Tue Jun 13 12:46:06 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Tue, 13 Jun 2006 12:46:06 -0400
Subject: [cisco-voip] AD plugin on a W2k3 Domain
In-Reply-To: <2A0FAAA5-90C3-4787-B141-23C72BC63541@mimectl>
References: <407055A92CECCB499C922A2D35FC19A605CEE960@Apollo.wesd.org><04BDB0D5-AFB8-4BDC-8CDF-5340D0E58197@mimectl>,
	<3f868042bd2ac6279fd820f9fda963ef@cisco.com>,
	<4C6759F1-6EF6-466D-A19E-D76002178F7B@mimectl>
	<AF6E4B63-7833-4B45-B013-483BD0370B3E@mimectl>,
	<F4CB1CFE-C1D4-40BE-8B20-64F829C79DE3@cisco.com>
	<2A0FAAA5-90C3-4787-B141-23C72BC63541@mimectl>
Message-ID: <a3f12d386faa709d9e47a941a0ad2404@cisco.com>

The format you use depends on the Object Class of each Container.

Object Class of the container using the following steps:
- Open Active Directory Users and Computers
- Click on View and check mark Advanced Features
- Right click on each container and select Properties
- Select the Object tab

Note the object class for each container and use that to build your 
Admin DN.

Kevin

On Jun 13, 2006, at 11:55 AM, Jason Burton wrote:

> Nope same result.
>
> From: Ryan Ratliff
> Sent: Tue 6/13/2006 11:51 AM
> To: Jason Burton
> Cc: Kevin Thorngren; cisco-voip Voip
> Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain
>
> Try CN=user, CN=container,DC=domain,DC=com.
>
>
> -Ryan
>
> On Jun 13, 2006, at 11:46 AM, Jason Burton wrote:
>
> Well after doing the packet capture I'm getting invalid credentials
> error in the Bind result.
>
> It's LDAPErr DSID-0c09334.  What is the correct format for AD Admin DN?
>
> CN=USER, OU=Container, DC=domain, DC=Domain?
>
> From: Jason Burton
> Sent: Tue 6/13/2006 11:08 AM
> To: Kevin Thorngren
> Cc: cisco-voip Voip
> Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain
>
> Thanks, downloading Ethereal now.
>
> From: Kevin Thorngren
> Sent: Tue 6/13/2006 11:01 AM
> To: Jason Burton
> Cc: cisco-voip Voip
> Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain
>
> The quickest way to isolate the issue is with a packet capture of the
> Publisher when doing the AD Plugin. Looking at the LDAP packets
> should help you see what the failure is. Kevin On Jun 13, 2006, at
> 10:52 AM, Jason Burton wrote: > Nope, just a router on a stick. > >
> From: Voll, Scott > Sent: Tue 6/13/2006 10:50 AM > To: Jason Burton;
> cisco-voip at puck.nether.net > Subject: RE: [cisco-voip] AD plugin on a
> W2k3 Domain > > Is there any pix between the CM and AD?  If so, we
> ran into a problem > with the fixup 389. >   > > From: cisco-voip-
> bounces at puck.nether.net > [mailto:cisco-voip-bounces at puck.nether.net]
> On Behalf Of Jason Burton > Sent: Tuesday, June 13, 2006 7:43 AM >
> To: cisco-voip at puck.nether.net > Subject: [cisco-voip] AD plugin on a
> W2k3 Domain >   >  I'm trying to install the AD plugin and keep
> getting the following > error: >   > Error:Setup failed to connect.
> Please enter your Host Name, Port > Number, Directory Administrator
> DN and Password again! >   > I've verified these 100 times over and
> still no luck.  This is the > first time I've tried this to a widnows
> 2003 domain.  Any ideas? >   > Thanks in advance!! >
> _______________________________________________ > cisco-voip mailing
> list > cisco-voip at puck.nether.net > https://puck.nether.net/mailman/
> listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From DylanV at semaphore.com  Tue Jun 13 12:55:54 2006
From: DylanV at semaphore.com (Dylan Vanderhoof)
Date: Tue, 13 Jun 2006 09:55:54 -0700
Subject: [cisco-voip] BAT line update question
Message-ID: <F6F72F553C7E754A945CB88CD8804B42417C44@exchange01.semaphore.lan>

So it turns out one of the engineers on a recent customer project
(probably me) imported about half the phones into a 4.1 Callmanager with
the Display (Caller ID) field on the lines the DN, not the user's name
like it should have been.

It doesn't appear that BAT can update lines with a CSV file like it can
import.  Is there a way to do this?  I can easily take a phone export
file and change the correct field, or similar, but I can't afford to
drop all the phones and reimport, since this is an IPCC install it would
take forever to reimport everybody.

Any way to fix this without manually clicking on each line and changing
the Display field?

Thanks,
Dylan


From Timothy_Frazee at adp.com  Tue Jun 13 13:01:17 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Tue, 13 Jun 2006 12:01:17 -0500
Subject: [cisco-voip] upgrade ccm 5 from .1 to .2 or .3
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC06D96@EXCHANGEDS01.ds.ad.adp.com>

Has anyone had problems upgrading a 5.0.1 to .2 using the ftp method?

 

It keeps erroring out and was wondering if anyone had any ideas or
things to look at?

 

Running 5.0.1 in vmware GSX server.

 

Timothy Frazee 

Network Solutions

Automatic Data Processing, Inc.

Dealer Services

314.249.8697 cellular phone

847.273.5950 desktop facsimile

847.273.3050 desktop ip phone

 

 

 



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 Matt.Slaga at us.didata.com  Tue Jun 13 13:04:23 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 13 Jun 2006 13:04:23 -0400
Subject: [cisco-voip] upgrade ccm 5 from .1 to .2 or .3
Message-ID: <C1FE15183DA37645BC0633BC604E44F00228E27E@USNAEXCH.na.didata.local>

The only luck I have had upgrading is putting the image on a DVD and
choosing that method.  We have upgraded from 5.0(1) to 5.0(2) and
recently to 5.0(4).  FTP did not work on any of them.

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Frazee, Timothy
Sent: Tuesday, June 13, 2006 1:01 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] upgrade ccm 5 from .1 to .2 or .3



Has anyone had problems upgrading a 5.0.1 to .2 using the ftp method?

 

It keeps erroring out and was wondering if anyone had any ideas or
things to look at?

 

Running 5.0.1 in vmware GSX server.

 

Timothy Frazee 

Network Solutions

Automatic Data Processing, Inc.

Dealer Services

314.249.8697 cellular phone

847.273.5950 desktop facsimile

847.273.3050 desktop ip phone

 

 

 

________________________________

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 jwydra at Burwood.com  Tue Jun 13 13:13:42 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Tue, 13 Jun 2006 12:13:42 -0500
Subject: [cisco-voip] Can't load 3.3.3 on VMware
Message-ID: <97F472A9F637E44B954D7FBABC800430BC6884@bur-msg-02.Burwood.com>

That's exactly what I used. And when I ran the OS upgrade it changed the
OS image key to 2000.4.2

 

Thanks,

Jason

 

________________________________

From: Frazee, Timothy [mailto:Timothy_Frazee at adp.com] 
Sent: Tuesday, June 13, 2006 11:18 AM
To: Wydra, Jason; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Can't load 3.3.3 on VMware

 

Can you post what you put in for your reg hack?

 

This is what I always put in:

 

Windows Registry Editor Version 5.00

[HKEY_LOCAL_MACHINE\SOFTWARE\Cisco Systems\Model]
"Hardware"="MCS7835"
"Memory"="1024"
"Speed"="733"
[HKEY_LOCAL_MACHINE\SOFTWARE\Spirian\System Info\OS Image] 
"Version"="2000.2.4"

 

 

Timothy Frazee 

Network Solutions

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wydra, Jason
Sent: Tuesday, June 13, 2006 10:52 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Can't load 3.3.3 on VMware

 

OK. So I created an ISO from a 3.3.3 Cisco installation disc. I
successfully upgraded the OS to 2000.4.2 using the download from Cisco.
The reg hack is in place. After the update the Spirian key shows
2000.4.2. I mounted the ISO and ran it. It keeps coming back and telling
me that my OS version is not up to date. WTFO??????

 

Jason 

 

________________________________

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
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notify us immediately by e-mail and delete the message and any
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From Scott.Voll at wesd.org  Tue Jun 13 13:41:16 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 13 Jun 2006 10:41:16 -0700
Subject: [cisco-voip] 79x1 vs 79x0
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEEA10@Apollo.wesd.org>

I have resently gotten a 79x1 phone and it has the feature of
remembering numbers as you dial and then you can scroll down and dial.
Is this a new feature only in the 79x1's or do I just need to upgrade
the firmware on the 79x0 to get this feature also?

 

CM 4.1.3sr2

 

Thanks

 

Scott

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From rratliff at cisco.com  Tue Jun 13 13:46:03 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 13 Jun 2006 13:46:03 -0400
Subject: [cisco-voip] 79x1 vs 79x0
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605CEEA10@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A605CEEA10@Apollo.wesd.org>
Message-ID: <5C6C8B8B-CD16-41B0-A4D8-E450277EC142@cisco.com>

This behavior is also present in the 7970.  The 79[46]1 phones are  
the same hardware platform as the 7970 so they behave the same.  This  
feature is not available on the 40 or 60.

-Ryan

On Jun 13, 2006, at 1:41 PM, Voll, Scott wrote:

I have resently gotten a 79x1 phone and it has the feature of  
remembering numbers as you dial and then you can scroll down and  
dial.  Is this a new feature only in the 79x1?s or do I just need to  
upgrade the firmware on the 79x0 to get this feature also?



CM 4.1.3sr2



Thanks



Scott

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



From marascio at metreos.com  Tue Jun 13 13:50:25 2006
From: marascio at metreos.com (Louis R. Marascio)
Date: Tue, 13 Jun 2006 12:50:25 -0500
Subject: [cisco-voip] Download CCM 5
Message-ID: <6EE053013F00084096AF0F02CFDB88B8769BDB@securemail.metreos.com>

Also, unlike earlier versions of CCM, 5.0 is license enforced so you
will also need a valid license to play with it.

Best regards,

Louis

---
Louis R. Marascio
Metreos Corporation
t: +1 (512) 687 2005
m: +1 (512) 964 4569
e: marascio at metreos.com 


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Frazee, Timothy
Sent: Tuesday, June 13, 2006 11:14 AM
To: Jorge Evangelista; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Download CCM 5

Officially, you need to talk to your friendly Cisco sales rep.

Timothy Frazee 
Network Solutions
314.249.8697 cellular phone

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jorge
Evangelista
Sent: Tuesday, June 13, 2006 11:01 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Download CCM 5

Anyone knows where I can download Cisco Call Manager 5.0 for linux. I
want to test CCM.
Is there some link at Internet


-- 
"The network is the computer"
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


This message and any attachments are intended only for the use of the
addressee and may contain information that is privileged and
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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
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From rick at trutechsystems.net  Tue Jun 13 13:54:04 2006
From: rick at trutechsystems.net (Rick Truong)
Date: Tue, 13 Jun 2006 10:54:04 -0700
Subject: [cisco-voip] 79x1 vs 79x0
In-Reply-To: <5C6C8B8B-CD16-41B0-A4D8-E450277EC142@cisco.com>
Message-ID: <200606131754.k5DHsAEs010575@puck.nether.net>

Hi Scott,

The feature is available on the 79x0's.  I have used it.

* We buy and sell new/used networking equipment.  Please contact me if you
have any offers or are interested in buying *

** VG224 NEW IN BOX @ $3600/each **
** VG248 NEW IN BOX @ $4900/each **
 
Thanks,
 
Rick Truong
TruTech Systems
39263 Half Moon Circle
Murrieta, CA 92563
Office: +1.951.696.3688
Mobile: +1.951.440.8711
AIM: ricktrutek
E-Mail: rick at trutechsystems.net
WEB: http://www.trutechsystems.net 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Tuesday, June 13, 2006 10:46 AM
To: Voll, Scott
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 79x1 vs 79x0

This behavior is also present in the 7970.  The 79[46]1 phones are  
the same hardware platform as the 7970 so they behave the same.  This  
feature is not available on the 40 or 60.

-Ryan

On Jun 13, 2006, at 1:41 PM, Voll, Scott wrote:

I have resently gotten a 79x1 phone and it has the feature of  
remembering numbers as you dial and then you can scroll down and  
dial.  Is this a new feature only in the 79x1's or do I just need to  
upgrade the firmware on the 79x0 to get this feature also?



CM 4.1.3sr2



Thanks



Scott

_______________________________________________
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 Timothy_Frazee at adp.com  Tue Jun 13 14:43:57 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Tue, 13 Jun 2006 13:43:57 -0500
Subject: [cisco-voip] Can't load 3.3.3 on VMware
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC06ED1@EXCHANGEDS01.ds.ad.adp.com>

This may sound stupid, but I put 2000.2.7 on first then 2000.4.2. always
worked for me.

 

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 Wydra, Jason
Sent: Tuesday, June 13, 2006 12:14 PM
To: Frazee, Timothy; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Can't load 3.3.3 on VMware

 

That's exactly what I used. And when I ran the OS upgrade it changed the
OS image key to 2000.4.2

 

Thanks,

Jason

 

________________________________

From: Frazee, Timothy [mailto:Timothy_Frazee at adp.com] 
Sent: Tuesday, June 13, 2006 11:18 AM
To: Wydra, Jason; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Can't load 3.3.3 on VMware

 

Can you post what you put in for your reg hack?

 

This is what I always put in:

 

Windows Registry Editor Version 5.00

[HKEY_LOCAL_MACHINE\SOFTWARE\Cisco Systems\Model]
"Hardware"="MCS7835"
"Memory"="1024"
"Speed"="733"
[HKEY_LOCAL_MACHINE\SOFTWARE\Spirian\System Info\OS Image] 
"Version"="2000.2.4"

 

 

Timothy Frazee 

Network Solutions

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wydra, Jason
Sent: Tuesday, June 13, 2006 10:52 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Can't load 3.3.3 on VMware

 

OK. So I created an ISO from a 3.3.3 Cisco installation disc. I
successfully upgraded the OS to 2000.4.2 using the download from Cisco.
The reg hack is in place. After the update the Spirian key shows
2000.4.2. I mounted the ISO and ran it. It keeps coming back and telling
me that my OS version is not up to date. WTFO??????

 

Jason 

 

________________________________

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. 



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 zthoran at yahoo.com  Tue Jun 13 15:29:37 2006
From: zthoran at yahoo.com (Zevier Thoran)
Date: Tue, 13 Jun 2006 12:29:37 -0700 (PDT)
Subject: [cisco-voip] IPCC Express / Supervisor Desktop
Message-ID: <20060613192937.10074.qmail@web37109.mail.mud.yahoo.com>

I AM NOT ABLE TO  HEAR RECORDED CONVERSTATIONS  FROM CISCO SUPERVISOR DESKTOP DURRING PLAYBACK.
ALL HELP IS WELCOME.
THANKS
-ZEVIER.


 __________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
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From BurwellJ at firstcharter.com  Tue Jun 13 21:53:31 2006
From: BurwellJ at firstcharter.com (Jason Burwell)
Date: Tue, 13 Jun 2006 21:53:31 -0400
Subject: [cisco-voip] 4ESS vs. 5ESS NI2-Calling party Name/Number Help
Message-ID: <697161D0107BAD40AFE9A86B06509AC90643E75F@FCEXCLUSTER.FirstCharter.com>

OK, I'm hoping someone out there can help me....
 
Note-Below I am speaking of the portion of at&t that was AT&T before the
SBC merger.
 
I recently installed an Inbound only T1 into my CallManager to carry
inbound 800# traffic from at&t. Of course the circuit is an at&t
circuit. The T is currently provisioned with 4ESS signaling. at&t says
they are sending Calling Party Name and Number info. The problem is
Callmanager will not decode Calling Party Name with 4ESS signaling- only
Calling Number. It seems the only way I can decode Calling Party Name
and Number is with NI2 which I am told is a form of 5ESS signaling. I
simply put in an order to have the circuit changed to NI2 but now at&t
is telling me that they cannot send Calling Party Name OR Number with
NI2. This makes absolutely NO sense to me as I have another (much
smaller) CLEC in the area that can deliver the Name and Number on NI2
with NO problem. I can't understand why someone as large as at&t can't
seem to provide this simple service. It seems logical that 4ESS
signaling would be "more dated" than 5ESS anyway, although that may not
be correct. My question here is, does anyone know why they are telling
me this is not possible? Am I missing something or do I have a tech that
dosen't know what they are talking about?
 
Any input will be greatly appreciated.
 
Thanks!
Jason


**********************************************************************
The information transmitted is intended only for the person or entity to which it is 
addressed and may contain confidential and/or privileged material. Any review,
retransmission, dissemination or other use of, or taking of any action in reliance upon,
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From jwydra at Burwood.com  Wed Jun 14 04:55:02 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Wed, 14 Jun 2006 03:55:02 -0500
Subject: [cisco-voip] IPCC Express / Supervisor Desktop
Message-ID: <97F472A9F637E44B954D7FBABC800430BC6A4F@bur-msg-02.Burwood.com>

DID YOU TURN UP THE VOLUME??????

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Zevier Thoran
Sent: Tuesday, June 13, 2006 2:30 PM
To: Cisco Voip
Subject: [cisco-voip] IPCC Express / Supervisor Desktop

 

I AM NOT ABLE TO  HEAR RECORDED CONVERSTATIONS  FROM CISCO SUPERVISOR
DESKTOP DURRING PLAYBACK.
ALL HELP IS WELCOME.
THANKS
-ZEVIER.

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

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From rvillagran at coasin.cl  Wed Jun 14 10:38:24 2006
From: rvillagran at coasin.cl (Rodrigo Villagran)
Date: Wed, 14 Jun 2006 10:38:24 -0400
Subject: [cisco-voip] h323 fxo connect faster
In-Reply-To: <6A0744639B86DD459433108DBF101D26FC62BD@MAIL01.sentinel.com>
Message-ID: <200606141438.k5EEc4xH023767@mail.coasin.cl>

Try this:

voice-port 1/1/0
 timeouts interdigit 3

with this parameter the port wait just 3 sec before dial the numbers.

Regards
Rodrigo V.

-----Mensaje original-----
De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de Carter, Bill
Enviado el: lunes, 12 de junio de 2006 22:42
Para: Voll, Scott; cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] h323 fxo connect faster

 Trunk Groups

-----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, June 12, 2006 4:23 PM
To: Bell, Joe; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] h323 fxo connect faster


I just did that and I'm still going slow.

I have pots dial peers with the following destination patterns and still
having issues.

?destination-pattern ^9[2-9]..[2-9]......$
?destination-pattern ^91[2-9]..[2-9]......$
?destination-pattern ^9...$

Any other ideas?

I also included the debug dial-peer if anyone can give me any direction.? I
have 12 dial peers with the same destination pattern as I want it to use one
of the 12 ports to dial out.? 

IE>
dial-peer voice 5001 pots
?preference 10
?answer-address 6331
?destination-pattern ^9[2-9]..[2-9]......$
?incoming called-number .
?direct-inward-dial
?port 0/0/1
?forward-digits all
!
dial-peer voice 5002 pots
?preference 10
?answer-address 6331
?destination-pattern ^9[2-9]..[2-9]......$
?incoming called-number .
?direct-inward-dial
?port 0/0/2
?forward-digits all


Is there a different way to do that?


Scott







Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,
?? Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,
?? Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.345: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911 Jun 12
21:13:46.345: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.345:
//-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=95035084911, Expanded String=95035084911, Calling Number=
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun
12 21:13:46.345: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5000 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5001 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5002 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5003 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5010 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5011 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5012 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5013 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5020 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5021 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5022 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5023 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5100 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5101 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5102 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5103 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5110 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5111 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5112 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5113 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5120 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5121 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5122 Is Matched Jun 12
21:13:46.349: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5123 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5200 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5201 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5202 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5203 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5210 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5211 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5212 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5213 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5220 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5221 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5222 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5223 Is Matched Jun 12
21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ANSWER; Calling Number=6343 Jun 12 21:13:46.353:
//-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.353:
//-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun
12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.353: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343 Jun 12 21:13:46.353:
//-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.353:
//-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun
12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Result=NO_MATCH(-1) After All Match Rules Attempt Jun 12 21:13:46.357:
//-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Calling Number=6343, Called Number=95035084911, Voice-Interface=0x0,
?? Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,
?? Peer Info Type=DIALPEER_INFO_SPEECH
Jun 12 21:13:46.357: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=95035084911 Jun 12
21:13:46.357: //-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.357:
//-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=95035084911, Expanded String=95035084911, Calling Number=
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun
12 21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5000 Is Matched Jun 12
21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5001 Is Matched Jun 12
21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5002 Is Matched Jun 12
21:13:46.357: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5003 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5010 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5011 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5012 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5013 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5020 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5021 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5022 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5023 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5100 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5101 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5102 Is Matched Jun 12
21:13:46.361: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5103 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5110 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5111 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5112 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5113 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5120 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5121 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5122 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5123 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5200 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5201 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5202 Is Matched Jun 12
21:13:46.365: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5203 Is Matched Jun 12
21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5210 Is Matched Jun 12
21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5211 Is Matched Jun 12
21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5212 Is Matched Jun 12
21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5213 Is Matched Jun 12
21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5220 Is Matched Jun 12
21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5221 Is Matched Jun 12
21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Incoming Dial-peer=5222 Is Matched Jun 12
21:13:46.369: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Re
Schlador#sult=Success(0); Incoming Dial-peer=5223 Is Matched Jun 12
21:13:46.369: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.369: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ANSWER; Calling Number=6343 Jun 12 21:13:46.373:
//-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.373:
//-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun
12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Match Rule=DP_MATCH_ORIGINATE; Calling Number=6343 Jun 12 21:13:46.373:
//-1/80CBD3252300/DPM/dpMatchPeertype:
?? Is Incoming=TRUE, Number Expansion=FALSE Jun 12 21:13:46.373:
//-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=, Expanded String=, Calling Number=6343T
?? Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH Jun
12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchCore:
?? Result=-1
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpMatchPeertype:exit at 5392
Jun 12 21:13:46.373: //-1/80CBD3252300/DPM/dpAssociateIncomingPeerCore:
?? Result=NO_MATCH(-1) After All Match Rules Attempt Jun 12 21:13:46.373:
//-1/80CBD3252300/DPM/dpAssociateIncomingPeerSPI:exit at 5926
Jun 12 21:13:46.377: //-1/80CBD3252300/DPM/dpMatchPeersCore:
?? Calling Number=, Called Number=95035084911, Peer Info
Type=DIALPEER_INFO_SPEECH Jun 12 21:13:46.377:
//-1/80CBD3252300/DPM/dpMatchPeersCore:
?? Match Rule=DP_MATCH_DEST; Called Number=95035084911 Jun 12 21:13:46.377:
//-1/80CBD3252300/DPM/dpMatchCore:
?? Dial String=95035084911, Expanded String=95035084911, Calling Number=
?? Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH Jun
12 21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5000 Is Matched Jun 12 21:
13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5001 Is Matched Jun 12
21:13:46.377: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5002 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5003 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5010 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5011 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5012 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5013 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5020 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5021 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5022 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/MatchNextPeer:
?? Result=Success(0); Outgoing Dial-peer=5023 Is Matched Jun 12
21:13:46.381: //-1/80CBD3252300/DPM/dpMatchPeersCore:
?? Result=Success(0) after DP_MATCH_DEST Jun 12 21:13:46.385:
//-1/80CBD3252300/DPM/dpMatchPeersMoreArg:
?? Result=SUCCESS(0)
?? List of Matched Outgoing Dial-peer(s):
???? 1: Dial-peer Tag=5023
???? 2: Dial-peer Tag=5022
???? 3: Dial-peer Tag=5021
???? 4: Dial-peer Tag=5020
???? 5: Dial-peer Tag=5013
???? 6: Dial-peer Tag=5012
???? 7: Dial-peer Tag=5011
???? 8: Dial-peer Tag=5010
???? 9: Dial-peer Tag=5003
???? 10: Dial-peer Tag=5001
???? 11: Dial-peer Tag=5002
???? 12: Dial-peer Tag=5000
________________________________________
From: Bell, Joe [mailto:Joe_Bell at adp.com]
Sent: Monday, June 12, 2006 1:56 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] h323 fxo connect faster

This is a good practice for all dial-peers with deterministic lengths.? Put
the regular expression for start of string ( a carrot, ^) and end of string
( a dollar sign, $).? Notice I said deterministic lengths, so .T or any
other unknown length dial-peer acts very goofy when you do this. 


Your destination pattern becomes this:

dial-peer voice 5020 pots
?preference 4
?answer-address 6331
?destination-pattern ^9[2-9]..[2-9]......$
?incoming called-number .
?direct-inward-dial
?port 0/2/0
?forward-digits all

This will speed up the match process because your destination pattern is now
specific.

Joe

________________________________________
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, June 12, 2006 1:49 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] h323 fxo connect faster

I have a h323 gateway with dial-peers as follows:

dial-peer voice 5020 pots
?preference 4
?answer-address 6331
?destination-pattern 9[2-9]..[2-9]......
?incoming called-number .
?direct-inward-dial
?port 0/2/0
?forward-digits all

but it acts like it is going through a interdigit time out.? How can I make
it dial quicker?

Thanks

Scott
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From monika.kapil at evalueserve.com  Wed Jun 14 10:44:04 2006
From: monika.kapil at evalueserve.com (Monika Kapil)
Date: Wed, 14 Jun 2006 20:14:04 +0530
Subject: [cisco-voip] FW:  IPCC forums
Message-ID: <A99CFB142B0B45489B4123C28ED33E3C0CC2BA3B@EVSEMAIL.Evalueserve.com>

Hi All,

 

I am using IPCC Express 3.5 for call recording. The server records the
calls but retains them for 7 days only. 

 

I am not able to find the option in the web interface where I can
increase the retention time of calls to 3 weeks.

 

Kindly help.

 

Regards,

Monika

	 

			 

			
			 



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From rvillagran at coasin.cl  Wed Jun 14 10:47:02 2006
From: rvillagran at coasin.cl (Rodrigo Villagran)
Date: Wed, 14 Jun 2006 10:47:02 -0400
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
In-Reply-To: <AB0A35F2486F81468971AA68558BE71903EDA560@smieln01winm01.micoanet.com>
Message-ID: <200606141446.k5EEkgh2027819@mail.coasin.cl>

If you press the "?" button twice there are some statistics of the call, are
the packets sent increasing?? (please do it in both phones), anyway it
sounds like a bug to me.. Those number are in monitor mode (or shared line)
could be a bug related with that.

 

Rodrigo

 

  _____  

De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de Linsemier, Matthew
Enviado el: lunes, 12 de junio de 2006 10:59
Para: Israel Lang
CC: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

 

As stipulated below, phones are on the same logical VLAN so no routing is
taking place other then call setup and teardown to the Call Managers.
Additionally, I know that it has nothing to do with the CSS because if this
was the issue the call would not be connected in the first place and users
would hear fast busy.

 

This problem started specifically with the 8.0(3) phone load.  The only
other change that as been made has been the OS2.4sr7 patch, which I find
unlikely to be causing this problem.

 

Matt

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Monday, June 12, 2006 10:50 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

 

Sounds like a classic partition/CSS issue to me. I would check to make sure
that phone A and phone B have each others partitions in their CSS. Sounds to
me like B does and A doesn't. Could also be a routing issue.

On 6/12/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:

Has anyone run into a One Way Audio issue with phone load 8.0(3) on a 7960?
I had piloted the phone load with about 20 phones and didn't hear of any
reported problems.  Now after I have pushed the load to around 200+ phones I
have users calling into the helpdesk reporting problems under the following
circumstances.  Please note that the calls are internal and on the same LAN
(even logical VLAN).

 

1)       User A calls User B

2)       User B can hear User A talking, however User A cannot hear User B.

3)       User A hangs up and calls User B again and there are no issues.

 

I haven't had any issues to date with the older, 8.0(2) (or earlier) phone
loads.  I am thinking that I am going to roll everyone back, but I wanted to
see if anyone is having the issue while also warning people of the problem. 

 

Sincerely,

 

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

 

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From Keith.Allan at 5i.co.uk  Mon Jun 12 16:42:29 2006
From: Keith.Allan at 5i.co.uk (Keith Allan)
Date: Mon, 12 Jun 2006 21:42:29 +0100
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
Message-ID: <41A3D72C0164E446891A5184585B704607E4BE@MAIL1.5idomain.local>

We are hitting the same problem following the upgrade from firmware
7-2-3 to 8-0-2 after upgrading to 4-1-3sr2.  The problem is
intermittent.  We have now downgraded some of our phones back to 7-2-3
to see if it is the firmware.  We are also on a single subnet and did
not have this problem prior to the upgrade.

 

Keith.

 

  _____  

From: Linsemier, Matthew [mailto:MLinsemier at apcapital.com] 
Sent: 12 June 2006 15:59
To: Israel Lang
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

 

As stipulated below, phones are on the same logical VLAN so no routing
is taking place other then call setup and teardown to the Call Managers.
Additionally, I know that it has nothing to do with the CSS because if
this was the issue the call would not be connected in the first place
and users would hear fast busy.

 

This problem started specifically with the 8.0(3) phone load.  The only
other change that as been made has been the OS2.4sr7 patch, which I find
unlikely to be causing this problem.

 

Matt

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Monday, June 12, 2006 10:50 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

 

Sounds like a classic partition/CSS issue to me. I would check to make
sure that phone A and phone B have each others partitions in their CSS.
Sounds to me like B does and A doesn't. Could also be a routing issue.

On 6/12/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:

Has anyone run into a One Way Audio issue with phone load 8.0(3) on a
7960?  I had piloted the phone load with about 20 phones and didn't hear
of any reported problems.  Now after I have pushed the load to around
200+ phones I have users calling into the helpdesk reporting problems
under the following circumstances.  Please note that the calls are
internal and on the same LAN (even logical VLAN).

 

1)       User A calls User B

2)       User B can hear User A talking, however User A cannot hear User
B.

3)       User A hangs up and calls User B again and there are no issues.

 

I haven't had any issues to date with the older, 8.0(2) (or earlier)
phone loads.  I am thinking that I am going to roll everyone back, but I
wanted to see if anyone is having the issue while also warning people of
the problem. 

 

Sincerely,

 

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

 

  _____  

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From tennille.spence at gsa.gov  Wed Jun 14 11:27:12 2006
From: tennille.spence at gsa.gov (tennille.spence at gsa.gov)
Date: Wed, 14 Jun 2006 10:27:12 -0500
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
In-Reply-To: <41A3D72C0164E446891A5184585B704607E4BE@MAIL1.5idomain.local>
Message-ID: <OFE20690C3.3BAB6294-ON8625718D.0053E34A-8625718D.0054E373@gsa.gov>


I have the same intermittent one way audio. When the issue started over a
year ago, we were running some version of 6 and upgraded to 7-2-3  on our
7940s and 7960s. The upgrade did nothing to correct the issue. Our 7935s
recently began doing this as well. My biggest challenge is capturing or
even trying to recreate the issue. Have you had any luck in capturing a
trace when this was happening and if so, did Cisco TAC recognize anything
unusual?

- Tennille


                                                                           
                                                                           
             "Keith Allan"                                                 
             <Keith.Allan at 5i.c                                          To 
             o.uk>                     "Linsemier, Matthew"                
             Sent by:                  <MLinsemier at apcapital.com>, "Israel 
             cisco-voip-bounce         Lang" <zephy316 at gmail.com>          
             s at puck.nether.net                                          cc 
                                       cisco-voip at puck.nether.net          
                                                                   Subject 
             06/12/2006 03:42          Re: [cisco-voip] Phone Load 8.0(3)  
             PM                        - One Way Audio                     
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           




We are hitting the same problem following the upgrade from firmware 7-2-3
to 8-0-2 after upgrading to 4-1-3sr2.  The problem is intermittent.  We
have now downgraded some of our phones back to 7-2-3 to see if it is the
firmware.  We are also on a single subnet and did not have this problem
prior to the upgrade.

Keith.


From: Linsemier, Matthew [mailto:MLinsemier at apcapital.com]
Sent: 12 June 2006 15:59
To: Israel Lang
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

As stipulated below, phones are on the same logical VLAN so no routing is
taking place other then call setup and teardown to the Call Managers.
Additionally, I know that it has nothing to do with the CSS because if this
was the issue the call would not be connected in the first place and users
would hear fast busy.

This problem started specifically with the 8.0(3) phone load.  The only
other change that as been made has been the OS2.4sr7 patch, which I find
unlikely to be causing this problem.

Matt


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Monday, June 12, 2006 10:50 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

Sounds like a classic partition/CSS issue to me. I would check to make sure
that phone A and phone B have each others partitions in their CSS. Sounds
to me like B does and A doesn't. Could also be a routing issue.
On 6/12/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:


Has anyone run into a One Way Audio issue with phone load 8.0(3) on a 7960?
I had piloted the phone load with about 20 phones and didn't hear of any
reported problems.  Now after I have pushed the load to around 200+ phones
I have users calling into the helpdesk reporting problems under the
following circumstances.  Please note that the calls are internal and on
the same LAN (even logical VLAN).





      1)       User A calls User B


      2)       User B can hear User A talking, however User A cannot hear
      User B.


      3)       User A hangs up and calls User B again and there are no
      issues.





I haven't had any issues to date with the older, 8.0(2) (or earlier) phone
loads.  I am thinking that I am going to roll everyone back, but I wanted
to see if anyone is having the issue while also warning people of the
problem.





Sincerely,





Matthew M. Linsemier, CCSP/CCNP/CCDA
Senior Network Engineer
APCapital
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.


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



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

Replace the _ with an @ symbol



CONFIDENTIALITY STATEMENT


This communication and any attachments are CONFIDENTIAL and may be
protected by one or more legal privileges. It is intended solely for the
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recipient, any use, disclosure, copying or distribution of this
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contrary is included in this message. If you have received this
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From Matt.Slaga at us.didata.com  Wed Jun 14 11:33:16 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 14 Jun 2006 11:33:16 -0400
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
Message-ID: <C1FE15183DA37645BC0633BC604E44F0022C87DE@USNAEXCH.na.didata.local>

This is where QRT becomes handy.  Of course, this requires a small
amount of training or at least introduction of the tool to the end users
involved.
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
tennille.spence at gsa.gov
Sent: Wednesday, June 14, 2006 11:27 AM
To: Keith.Allan at 5i.co.uk
Cc: cisco-voip-bounces at puck.nether.net; Linsemier, Matthew;
cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio


I have the same intermittent one way audio. When the issue started over
a year ago, we were running some version of 6 and upgraded to 7-2-3  on
our 7940s and 7960s. The upgrade did nothing to correct the issue. Our
7935s recently began doing this as well. My biggest challenge is
capturing or even trying to recreate the issue. Have you had any luck in
capturing a trace when this was happening and if so, did Cisco TAC
recognize anything unusual?

- Tennille


 

 

             "Keith Allan"

             <Keith.Allan at 5i.c
To 
             o.uk>                     "Linsemier, Matthew"

             Sent by:                  <MLinsemier at apcapital.com>,
"Israel 
             cisco-voip-bounce         Lang" <zephy316 at gmail.com>

             s at puck.nether.net
cc 
                                       cisco-voip at puck.nether.net

 
Subject 
             06/12/2006 03:42          Re: [cisco-voip] Phone Load
8.0(3)  
             PM                        - One Way Audio

 

 

 

 

 

 





We are hitting the same problem following the upgrade from firmware
7-2-3 to 8-0-2 after upgrading to 4-1-3sr2.  The problem is
intermittent.  We have now downgraded some of our phones back to 7-2-3
to see if it is the firmware.  We are also on a single subnet and did
not have this problem prior to the upgrade.

Keith.


From: Linsemier, Matthew [mailto:MLinsemier at apcapital.com]
Sent: 12 June 2006 15:59
To: Israel Lang
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

As stipulated below, phones are on the same logical VLAN so no routing
is taking place other then call setup and teardown to the Call Managers.
Additionally, I know that it has nothing to do with the CSS because if
this was the issue the call would not be connected in the first place
and users would hear fast busy.

This problem started specifically with the 8.0(3) phone load.  The only
other change that as been made has been the OS2.4sr7 patch, which I find
unlikely to be causing this problem.

Matt


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Monday, June 12, 2006 10:50 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

Sounds like a classic partition/CSS issue to me. I would check to make
sure that phone A and phone B have each others partitions in their CSS.
Sounds to me like B does and A doesn't. Could also be a routing issue.
On 6/12/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:


Has anyone run into a One Way Audio issue with phone load 8.0(3) on a
7960?
I had piloted the phone load with about 20 phones and didn't hear of any
reported problems.  Now after I have pushed the load to around 200+
phones I have users calling into the helpdesk reporting problems under
the following circumstances.  Please note that the calls are internal
and on the same LAN (even logical VLAN).





      1)       User A calls User B


      2)       User B can hear User A talking, however User A cannot
hear
      User B.


      3)       User A hangs up and calls User B again and there are no
      issues.





I haven't had any issues to date with the older, 8.0(2) (or earlier)
phone loads.  I am thinking that I am going to roll everyone back, but I
wanted to see if anyone is having the issue while also warning people of
the problem.





Sincerely,





Matthew M. Linsemier, CCSP/CCNP/CCDA
Senior Network Engineer
APCapital
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



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

Replace the _ with an @ symbol



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
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name of the sender, nor anything else in this message is intended to
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contrary is included in this message. If you have received this
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communication from your computer. Thank you.
_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip



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From MLinsemier at apcapital.com  Wed Jun 14 11:30:32 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Wed, 14 Jun 2006 11:30:32 -0400
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
Message-ID: <AB0A35F2486F81468971AA68558BE71903EDABFB@smieln01winm01.micoanet.com>

This just started with us in 8.0(3).  We were running 8.0(1) and I
believe 8.0(2) on several phones for a while without incident.  It is
tough to say because I know that 8.0(2) and 8.0(3) were released close
to each other so I am trying to remember if I ever reset the phones to
upgrade from 8.0(1).

I am also trying to discern if the problem started before or after the
OS4.2sr7 upgrade.

I am going to reset all of the phones back to 8.0(2) over the weekend.
I will get back to everyone.

Matt

-----Original Message-----
From: tennille.spence at gsa.gov [mailto:tennille.spence at gsa.gov] 
Sent: Wednesday, June 14, 2006 11:27 AM
To: Keith.Allan at 5i.co.uk
Cc: cisco-voip at puck.nether.net; cisco-voip-bounces at puck.nether.net;
Linsemier, Matthew; Israel Lang
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio


I have the same intermittent one way audio. When the issue started over
a
year ago, we were running some version of 6 and upgraded to 7-2-3  on
our
7940s and 7960s. The upgrade did nothing to correct the issue. Our 7935s
recently began doing this as well. My biggest challenge is capturing or
even trying to recreate the issue. Have you had any luck in capturing a
trace when this was happening and if so, did Cisco TAC recognize
anything
unusual?

- Tennille


 

 

             "Keith Allan"

             <Keith.Allan at 5i.c
To 
             o.uk>                     "Linsemier, Matthew"

             Sent by:                  <MLinsemier at apcapital.com>,
"Israel 
             cisco-voip-bounce         Lang" <zephy316 at gmail.com>

             s at puck.nether.net
cc 
                                       cisco-voip at puck.nether.net

 
Subject 
             06/12/2006 03:42          Re: [cisco-voip] Phone Load
8.0(3)  
             PM                        - One Way Audio

 

 

 

 

 

 





We are hitting the same problem following the upgrade from firmware
7-2-3
to 8-0-2 after upgrading to 4-1-3sr2.  The problem is intermittent.  We
have now downgraded some of our phones back to 7-2-3 to see if it is the
firmware.  We are also on a single subnet and did not have this problem
prior to the upgrade.

Keith.


From: Linsemier, Matthew [mailto:MLinsemier at apcapital.com]
Sent: 12 June 2006 15:59
To: Israel Lang
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

As stipulated below, phones are on the same logical VLAN so no routing
is
taking place other then call setup and teardown to the Call Managers.
Additionally, I know that it has nothing to do with the CSS because if
this
was the issue the call would not be connected in the first place and
users
would hear fast busy.

This problem started specifically with the 8.0(3) phone load.  The only
other change that as been made has been the OS2.4sr7 patch, which I find
unlikely to be causing this problem.

Matt


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Monday, June 12, 2006 10:50 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

Sounds like a classic partition/CSS issue to me. I would check to make
sure
that phone A and phone B have each others partitions in their CSS.
Sounds
to me like B does and A doesn't. Could also be a routing issue.
On 6/12/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:


Has anyone run into a One Way Audio issue with phone load 8.0(3) on a
7960?
I had piloted the phone load with about 20 phones and didn't hear of any
reported problems.  Now after I have pushed the load to around 200+
phones
I have users calling into the helpdesk reporting problems under the
following circumstances.  Please note that the calls are internal and on
the same LAN (even logical VLAN).





      1)       User A calls User B


      2)       User B can hear User A talking, however User A cannot
hear
      User B.


      3)       User A hangs up and calls User B again and there are no
      issues.





I haven't had any issues to date with the older, 8.0(2) (or earlier)
phone
loads.  I am thinking that I am going to roll everyone back, but I
wanted
to see if anyone is having the issue while also warning people of the
problem.





Sincerely,





Matthew M. Linsemier, CCSP/CCNP/CCDA
Senior Network Engineer
APCapital
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



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

Replace the _ with an @ symbol



CONFIDENTIALITY STATEMENT


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

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the intended recipient, any use, disclosure, copying or 
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From rleetun at co.boulder.co.us  Wed Jun 14 12:20:35 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Wed, 14 Jun 2006 10:20:35 -0600
Subject: [cisco-voip] Failure of CM during defrag of the HD,
	which phones did not failover to another CM
Message-ID: <963CE94787005149A52C17F33A028E12051D485F@EVS1B.boco.co.boulder.co.us>

Good morning,

Have any of you seen phones not failover to another CallManager during
the defragging of the call manager's hard drive (HD)?  The HD on the
CallManager is very fragmented, which the SA tried to run defrag on it.
The defrag failed, which it took down the phones with it.  Should have
they forced the phones over to another CallManager and then run defrag,
or does it matter?  I would lean towards forcing the phones to another
CallManager before running defrag.

Ideas?

Thanks in advance!

Rob
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From jsteinberg at gmail.com  Wed Jun 14 12:32:37 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 14 Jun 2006 12:32:37 -0400
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
In-Reply-To: <AB0A35F2486F81468971AA68558BE71903EDABFB@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE71903EDABFB@smieln01winm01.micoanet.com>
Message-ID: <f1b65f880606140932k2b0526fds25da0529734eb8e9@mail.gmail.com>

I had a very similiar issue when I used 6.0(5).  Sporadically audio
cut through was delayed for about 10 seconds at the beginning of the
call.   Many users hung up because of the one way audio.  There was no
rhyme or reason or pattern that I could determine.  I assume it was a
bug in the 7960 fw code.

I upgraded to 7.2.2 and have not had the problem since.

Justin



On 6/14/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
> This just started with us in 8.0(3).  We were running 8.0(1) and I
> believe 8.0(2) on several phones for a while without incident.  It is
> tough to say because I know that 8.0(2) and 8.0(3) were released close
> to each other so I am trying to remember if I ever reset the phones to
> upgrade from 8.0(1).
>
> I am also trying to discern if the problem started before or after the
> OS4.2sr7 upgrade.
>
> I am going to reset all of the phones back to 8.0(2) over the weekend.
> I will get back to everyone.
>
> Matt
>
> -----Original Message-----
> From: tennille.spence at gsa.gov [mailto:tennille.spence at gsa.gov]
> Sent: Wednesday, June 14, 2006 11:27 AM
> To: Keith.Allan at 5i.co.uk
> Cc: cisco-voip at puck.nether.net; cisco-voip-bounces at puck.nether.net;
> Linsemier, Matthew; Israel Lang
> Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio
>
>
> I have the same intermittent one way audio. When the issue started over
> a
> year ago, we were running some version of 6 and upgraded to 7-2-3  on
> our
> 7940s and 7960s. The upgrade did nothing to correct the issue. Our 7935s
> recently began doing this as well. My biggest challenge is capturing or
> even trying to recreate the issue. Have you had any luck in capturing a
> trace when this was happening and if so, did Cisco TAC recognize
> anything
> unusual?
>
> - Tennille
>
>
>
>
>
>
>             "Keith Allan"
>
>             <Keith.Allan at 5i.c
> To
>             o.uk>                     "Linsemier, Matthew"
>
>             Sent by:                  <MLinsemier at apcapital.com>,
> "Israel
>             cisco-voip-bounce         Lang" <zephy316 at gmail.com>
>
>             s at puck.nether.net
> cc
>                                       cisco-voip at puck.nether.net
>
>
> Subject
>             06/12/2006 03:42          Re: [cisco-voip] Phone Load
> 8.0(3)
>             PM                        - One Way Audio
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> We are hitting the same problem following the upgrade from firmware
> 7-2-3
> to 8-0-2 after upgrading to 4-1-3sr2.  The problem is intermittent.  We
> have now downgraded some of our phones back to 7-2-3 to see if it is the
> firmware.  We are also on a single subnet and did not have this problem
> prior to the upgrade.
>
> Keith.
>
>
> From: Linsemier, Matthew [mailto:MLinsemier at apcapital.com]
> Sent: 12 June 2006 15:59
> To: Israel Lang
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio
>
> As stipulated below, phones are on the same logical VLAN so no routing
> is
> taking place other then call setup and teardown to the Call Managers.
> Additionally, I know that it has nothing to do with the CSS because if
> this
> was the issue the call would not be connected in the first place and
> users
> would hear fast busy.
>
> This problem started specifically with the 8.0(3) phone load.  The only
> other change that as been made has been the OS2.4sr7 patch, which I find
> unlikely to be causing this problem.
>
> Matt
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
> Sent: Monday, June 12, 2006 10:50 AM
> To: Linsemier, Matthew
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio
>
> Sounds like a classic partition/CSS issue to me. I would check to make
> sure
> that phone A and phone B have each others partitions in their CSS.
> Sounds
> to me like B does and A doesn't. Could also be a routing issue.
> On 6/12/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
> Has anyone run into a One Way Audio issue with phone load 8.0(3) on a
> 7960?
> I had piloted the phone load with about 20 phones and didn't hear of any
> reported problems.  Now after I have pushed the load to around 200+
> phones
> I have users calling into the helpdesk reporting problems under the
> following circumstances.  Please note that the calls are internal and on
> the same LAN (even logical VLAN).
>
>
>
>
>
>      1)       User A calls User B
>
>
>      2)       User B can hear User A talking, however User A cannot
> hear
>      User B.
>
>
>      3)       User A hangs up and calls User B again and there are no
>      issues.
>
>
>
>
>
> I haven't had any issues to date with the older, 8.0(2) (or earlier)
> phone
> loads.  I am thinking that I am going to roll everyone back, but I
> wanted
> to see if anyone is having the issue while also warning people of the
> problem.
>
>
>
>
>
> Sincerely,
>
>
>
>
>
> Matthew M. Linsemier, CCSP/CCNP/CCDA
> Senior Network Engineer
> APCapital
> 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
>
>
>
> --
> Have a great day!!!!!!!!!!
> zephy316_gmail.com
>
> Replace the _ with an @ symbol
>
>
>
> 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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> 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
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> --------------------------------------------------------------------------------
>
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>

From rratliff at cisco.com  Wed Jun 14 12:37:10 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 14 Jun 2006 12:37:10 -0400
Subject: [cisco-voip] Phone Load 8.0(3) - One Way Audio
In-Reply-To: <AB0A35F2486F81468971AA68558BE71903EDABFB@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE71903EDABFB@smieln01winm01.micoanet.com>
Message-ID: <FC0F3176-1539-48E0-BDEC-96AD0E1BC5EE@cisco.com>

As an aside you may want to check out CSCse47327 if you are still on  
2000.4.2sr7.

-Ryan

On Jun 14, 2006, at 11:30 AM, Linsemier, Matthew wrote:

This just started with us in 8.0(3).  We were running 8.0(1) and I
believe 8.0(2) on several phones for a while without incident.  It is
tough to say because I know that 8.0(2) and 8.0(3) were released close
to each other so I am trying to remember if I ever reset the phones to
upgrade from 8.0(1).

I am also trying to discern if the problem started before or after the
OS4.2sr7 upgrade.

I am going to reset all of the phones back to 8.0(2) over the weekend.
I will get back to everyone.

Matt

-----Original Message-----
From: tennille.spence at gsa.gov [mailto:tennille.spence at gsa.gov]
Sent: Wednesday, June 14, 2006 11:27 AM
To: Keith.Allan at 5i.co.uk
Cc: cisco-voip at puck.nether.net; cisco-voip-bounces at puck.nether.net;
Linsemier, Matthew; Israel Lang
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio


I have the same intermittent one way audio. When the issue started over
a
year ago, we were running some version of 6 and upgraded to 7-2-3  on
our
7940s and 7960s. The upgrade did nothing to correct the issue. Our 7935s
recently began doing this as well. My biggest challenge is capturing or
even trying to recreate the issue. Have you had any luck in capturing a
trace when this was happening and if so, did Cisco TAC recognize
anything
unusual?

- Tennille






              "Keith Allan"

              <Keith.Allan at 5i.c
To
              o.uk>                     "Linsemier, Matthew"

              Sent by:                  <MLinsemier at apcapital.com>,
"Israel
              cisco-voip-bounce         Lang" <zephy316 at gmail.com>

              s at puck.nether.net
cc
                                        cisco-voip at puck.nether.net


Subject
              06/12/2006 03:42          Re: [cisco-voip] Phone Load
8.0(3)
              PM                        - One Way Audio

















We are hitting the same problem following the upgrade from firmware
7-2-3
to 8-0-2 after upgrading to 4-1-3sr2.  The problem is intermittent.  We
have now downgraded some of our phones back to 7-2-3 to see if it is the
firmware.  We are also on a single subnet and did not have this problem
prior to the upgrade.

Keith.


From: Linsemier, Matthew [mailto:MLinsemier at apcapital.com]
Sent: 12 June 2006 15:59
To: Israel Lang
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

As stipulated below, phones are on the same logical VLAN so no routing
is
taking place other then call setup and teardown to the Call Managers.
Additionally, I know that it has nothing to do with the CSS because if
this
was the issue the call would not be connected in the first place and
users
would hear fast busy.

This problem started specifically with the 8.0(3) phone load.  The only
other change that as been made has been the OS2.4sr7 patch, which I find
unlikely to be causing this problem.

Matt


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Israel Lang
Sent: Monday, June 12, 2006 10:50 AM
To: Linsemier, Matthew
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone Load 8.0(3) - One Way Audio

Sounds like a classic partition/CSS issue to me. I would check to make
sure
that phone A and phone B have each others partitions in their CSS.
Sounds
to me like B does and A doesn't. Could also be a routing issue.
On 6/12/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:


Has anyone run into a One Way Audio issue with phone load 8.0(3) on a
7960?
I had piloted the phone load with about 20 phones and didn't hear of any
reported problems.  Now after I have pushed the load to around 200+
phones
I have users calling into the helpdesk reporting problems under the
following circumstances.  Please note that the calls are internal and on
the same LAN (even logical VLAN).





       1)       User A calls User B


       2)       User B can hear User A talking, however User A cannot
hear
       User B.


       3)       User A hangs up and calls User B again and there are no
       issues.





I haven't had any issues to date with the older, 8.0(2) (or earlier)
phone
loads.  I am thinking that I am going to roll everyone back, but I
wanted
to see if anyone is having the issue while also warning people of the
problem.





Sincerely,





Matthew M. Linsemier, CCSP/CCNP/CCDA
Senior Network Engineer
APCapital
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
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name of the sender, nor anything else in this message is intended to
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contrary is included in this message. If you have received this
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communication from your computer. Thank you.


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



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

Replace the _ with an @ symbol



CONFIDENTIALITY STATEMENT


This communication and any attachments are CONFIDENTIAL and may be
protected by one or more legal privileges. It is intended solely for the
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recipient, any use, disclosure, copying or distribution of this
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constitute an electronic signature unless a specific statement to the
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communication in error, please immediately contact me and delete this
communication from your computer. Thank you.
_______________________________________________
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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
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the intended recipient, any use, disclosure, copying or
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From rratliff at cisco.com  Wed Jun 14 12:38:27 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 14 Jun 2006 12:38:27 -0400
Subject: [cisco-voip] Failure of CM during defrag of the HD,
	which phones did not failover to another CM
In-Reply-To: <963CE94787005149A52C17F33A028E12051D485F@EVS1B.boco.co.boulder.co.us>
References: <963CE94787005149A52C17F33A028E12051D485F@EVS1B.boco.co.boulder.co.us>
Message-ID: <8CA67603-0F05-4099-B1F6-E6B7D581D13F@cisco.com>

On a Windows box I try never to run defrag except from safe mode or  
else it takes forever.

I'd fully expect defrag to cause a tremendous amount of disk io which  
can easily crash CCM.

-Ryan

On Jun 14, 2006, at 12:20 PM, Leetun, Rob wrote:

Good morning,

Have any of you seen phones not failover to another CallManager  
during the defragging of the call manager's hard drive (HD)?  The HD  
on the CallManager is very fragmented, which the SA tried to run  
defrag on it.  The defrag failed, which it took down the phones with  
it.  Should have they forced the phones over to another CallManager  
and then run defrag, or does it matter?  I would lean towards forcing  
the phones to another CallManager before running defrag.

Ideas?

Thanks in advance!

Rob

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


From cj.alan at gmail.com  Wed Jun 14 13:14:39 2006
From: cj.alan at gmail.com (Alan Su)
Date: Thu, 15 Jun 2006 01:14:39 +0800
Subject: [cisco-voip]  CCM Cross WAN Clustering
Message-ID: <8f9f19f00606141014h7fc278a8pfdb7e9bf488699ee@mail.gmail.com>

Dear All,

>From cisco document, the clustered CCM servers should meet the requirement
of 40ms round-trip time for database replication. In my case, the 2 sites
connected by IPLC with about 50ms. So I did some lab test to make sure it
will be working in my environment. But I found out that even in 200ms
round-trip time lab, the replication is still working!! I'm wondering that
is it a normal case or I misconfigure something. Does anybody have any clue
about this?

Thanks.

Regards,

Alan
-------------- next part --------------
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From Matt.Slaga at us.didata.com  Wed Jun 14 13:35:16 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Wed, 14 Jun 2006 13:35:16 -0400
Subject: [cisco-voip] CCM Cross WAN Clustering
Message-ID: <C1FE15183DA37645BC0633BC604E44F0022C8A0C@USNAEXCH.na.didata.local>

While it will work, Cisco will not support anything over 40ms.  This
gets to be an issue with user response times at the telepad and the
potential for SQL replication issues. 
 
 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Alan Su
Sent: Wednesday, June 14, 2006 1:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM Cross WAN Clustering


Dear All,
 
From cisco document, the clustered CCM servers should meet the
requirement of 40ms round-trip time for database replication. In my
case, the 2 sites connected by IPLC with about 50ms. So I did some lab
test to make sure it will be working in my environment. But I found out
that even in 200ms round-trip time lab, the replication is still
working!! I'm wondering that is it a normal case or I misconfigure
something. Does anybody have any clue about this? 
 
Thanks.
 
Regards,
 
Alan



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From CMStaffin at ra.rockwell.com  Wed Jun 14 14:01:55 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Wed, 14 Jun 2006 13:01:55 -0500
Subject: [cisco-voip] CCM Cross WAN Clustering
In-Reply-To: <8f9f19f00606141014h7fc278a8pfdb7e9bf488699ee@mail.gmail.com>
Message-ID: <OFC0A89A33.0BEA035B-ON8625718D.0062F4F1-8625718D.0062FB73@ra.rockwell.com>

Alan,

The 40ms RTT is due to MS SQL.  These are numbers published by MS.  In a
lab envirment this will most likely work fine due to very little database
replication and changes.  However put 5000 phones on it and there is
considerably more network traffic between servers.

Craig




                                                                           
             "Alan Su"                                                     
             <cj.alan at gmail.co                                             
             m>                                                         To 
             Sent by:                  cisco-voip at puck.nether.net          
             cisco-voip-bounce                                          cc 
             s at puck.nether.net                                             
                                                                   Subject 
                                       [cisco-voip]  CCM Cross WAN         
             06/14/2006 12:14          Clustering                          
             PM                                                            
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           




Dear All,

>From cisco document, the clustered CCM servers should meet the requirement
of 40ms round-trip time for database replication. In my case, the 2 sites
connected by IPLC with about 50ms. So I did some lab test to make sure it
will be working in my environment. But I found out that even in 200ms
round-trip time lab, the replication is still working!! I'm wondering that
is it a normal case or I misconfigure something. Does anybody have any clue
about this?

Thanks.

Regards,

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



From lelio at uoguelph.ca  Wed Jun 14 14:30:19 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 14 Jun 2006 15:30:19 -0300
Subject: [cisco-voip] Remote Subscriber vs SRST: (was CCM Cross WAN
	Clustering)
References: <OFC0A89A33.0BEA035B-ON8625718D.0062F4F1-8625718D.0062FB73@ra.rockwell.com>
Message-ID: <001001c68fe0$9772b080$4ba2ad81@dellectable>

Just wondering what peoples thoughts are out there regarding remote 
subscribers vs SRST.

For myself, the logistical nightmare of any upgrade seems to outweigh the 
downfalls of SRST's lack(?) of features and even then, it seems like SRST is 
gaining in supported volumes as well as feature offering.

What pros to a remote subscriber am I missing other than full feature 
support (aside from those requiring write access like forwarding, 
unforwarding)?

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network 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 ealeatherman at gmail.com  Wed Jun 14 15:14:33 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 14 Jun 2006 15:14:33 -0400
Subject: [cisco-voip] Long Distance woes
In-Reply-To: <94a1afde0606111803m3d004147k58f0280f7fe3c4bd@mail.gmail.com>
References: <407055A92CECCB499C922A2D35FC19A605D75A19@Apollo.wesd.org>
	<20060611233910.14187.qmail@web34606.mail.mud.yahoo.com>
	<94a1afde0606111803m3d004147k58f0280f7fe3c4bd@mail.gmail.com>
Message-ID: <94a1afde0606141214hb59a148wa7dc65fbe0a6d187@mail.gmail.com>

While that didnt fix my problem, I did get it setup to match the telco's
settings, which sounds like a good idea anyway :)

Turns out the PRI's were setup with the wrong long distance code at the CO,
so they were sending calls to AT&T instead of Sprint. Was a good
troubleshooting exercise for me though.

On 6/11/06, Ed Leatherman <ealeatherman at gmail.com > wrote:
>
> I'm going to try changing the numbering plan tomorrow to national and see
> if that helps, thanks for the suggestion!
>
>
> On 6/11/06, Hassan Salama < hsalama_us at yahoo.com> wrote:
>
> > Hello,
> >
> > Regarding your issue of long distance call, the number
> > handling could varies from country to country as this
> > is telecom standard.
> >
> > But you may try the translation rule to modify Called
> > party number type to become national [a must for long
> > distance within a country], and stip 91 from the
> > begeinning of the number [ i.e. send 10 digits without
> > 1], you may give another try with stipping 9 digit
> > only
> >
> > Thanks
> >
> > --- "Voll, Scott" < Scott.Voll at wesd.org> wrote:
> >
> > > have you tried changing the isdn to national, or
> > > unknown or something along those lines?
> > >
> > > Scott
> > >
> > > ________________________________
> > >
> > > From: cisco-voip-bounces at puck.nether.net on behalf
> > > of Ed Leatherman
> > > Sent: Fri 6/9/2006 5:16 PM
> > > To: ciscovoip
> > > Subject: [cisco-voip] Long Distance woes
> > >
> > >
> > > Was wondering if anyone is familiar with how long
> > > distance carriers/service interacts with local
> > > service...
> > >
> > > Working with service provider at one of our branch
> > > campuses which we just converted to a callmanager
> > > system this past week. Users can call long distance
> > > numbers within our area code (304) just fine, but
> > > cannot call outside of the area code. The dial
> > > pattern is the same (9.1[2-9]XX[2-9]XX XXXX) for
> > > either case (exact same route pattern actually), I
> > > just strip the 9 and send 1 + 10 digits to the phone
> > > company. Can see this in CCM trace. System uses 2
> > > PRI T1's.
> > >
> > > If users call an out of state number, they get fast
> > > busy. The local carrier is looking at the problem
> > > now but initially they just told me it wasnt there
> > > problem, as they just pass the digits to the long
> > > distance carrier if the number starts with "1". It
> > > seems to me that maybe the lines are setup
> > > incorrectly as far as what long distance provider it
> > > should use.. but I dont know enough about the
> > > process to make a good guess.. I've not worked with
> > > this particular local service provider before so I
> > > dont have the same "faith" if you could call it
> > > that, as I do with the carrier we have at the main
> > > campus which I have dealt with before.
> > >
> > > Caller ID also does not come across, but I think
> > > this is a seperate issue which we can hopefully
> > > address after the long distance is working.
> > >
> > > Any ideas? I'm curious if there is anything more I
> > > can look at from my end.
> > >
> > > --
> > > Ed Leatherman
> > > IP Telephony Coordinator
> > > 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
> > >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From ErasmuE4 at telkom.co.za  Wed Jun 14 15:47:54 2006
From: ErasmuE4 at telkom.co.za (Erik Erasmus (E))
Date: Wed, 14 Jun 2006 21:47:54 +0200
Subject: [cisco-voip] Remote Subscriber vs SRST: (was CCM Cross
	WANClustering)
Message-ID: <472E195E692EE74BAE4E7B2880A78C1F7AF7DB@CNTRRA20-XCS01.telkom.co.za>


You can also look at Call Manager Express in Fallback mode

Rune phones to central Call Manager and if the wan link drops which
should not be too often the phone will register to the CME in fallback
mode. Fairly new - gives you the added benefits of more features in
fallback mode compared with SRST and also the ability to PRE configure
additional features which can be usefull when a wan link goes down and
the phone go to fallback to the local branch.

A Chapter in the CME administration guide for unified CME explains how
it works.


Erik erasmus

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio
Fulgenzi
Sent: 14 June 2006 08:30 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Remote Subscriber vs SRST: (was CCM Cross
WANClustering)

Just wondering what peoples thoughts are out there regarding remote
subscribers vs SRST.

For myself, the logistical nightmare of any upgrade seems to outweigh
the
downfalls of SRST's lack(?) of features and even then, it seems like
SRST is
gaining in supported volumes as well as feature offering.

What pros to a remote subscriber am I missing other than full feature
support (aside from those requiring write access like forwarding,
unforwarding)?

----------------------------------------------------------------------
----------
Lelio Fulgenzi, B.A.
Network 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

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~


From shefys at gmail.com  Wed Jun 14 21:10:23 2006
From: shefys at gmail.com (Yuri Schefer)
Date: Thu, 15 Jun 2006 12:10:23 +1100
Subject: [cisco-voip] incoming calls
Message-ID: <1826487312.20060615121023@gmail.com>

Hi All!

I have some troubles in network. I have the CCM 4.1(3)sr3a cluster and
about 150 IP Phones(7961G/7971G). Sometimes I or my colleagues can't receive incoming calls, but
outgoing calls work. I can resolve this problem by updating line info
on phone profile, but why it happen? Also I tried to reboot all
phones, but this problem keep stayed. LAN based on Cat6500/3750, QoS policies is set.

-- 
WBR, Yuri


From Michael_Davis at eLoyalty.com  Wed Jun 14 22:24:10 2006
From: Michael_Davis at eLoyalty.com (Davis, Michael)
Date: Wed, 14 Jun 2006 21:24:10 -0500
Subject: [cisco-voip] CCM Cross WAN Clustering
References: <8f9f19f00606141014h7fc278a8pfdb7e9bf488699ee@mail.gmail.com>
Message-ID: <8490B0F13023C84C9472B224C27715584BD08C@elhqex01.eloyaltyco.com>

In larger, high volume, or IPCC (Ent) or (Express) environments,  there can also be serious issues with CTI Manager as well as call processing synchronization across a slower or unreliable links.  Situations where there are devices controlled by a subscriber on one side of the link are talking to devices controlled by different subscribers on the other side of the "slow" link.
 
I think for this reason, Cisco sets forth the 40ms RTT rule. I've seen situations where a functioning split custer link became over subscribed, resulting in the need for CCM and/or CTIM restarts,  Random or all Route Points unregistering,  JTAPI failures, etc.
 
________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Alan Su
Sent: Wed 6/14/2006 12:14 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM Cross WAN Clustering


Dear All,
 
>From cisco document, the clustered CCM servers should meet the requirement of 40ms round-trip time for database replication. In my case, the 2 sites connected by IPLC with about 50ms. So I did some lab test to make sure it will be working in my environment. But I found out that even in 200ms round-trip time lab, the replication is still working!! I'm wondering that is it a normal case or I misconfigure something. Does anybody have any clue about this? 
 
Thanks.
 
Regards,
 
Alan


From erickbe at yahoo.com  Wed Jun 14 23:30:05 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 14 Jun 2006 20:30:05 -0700 (PDT)
Subject: [cisco-voip] ISDN PRI issue - busy calls
Message-ID: <20060615033005.24570.qmail@web37215.mail.mud.yahoo.com>

Anyone seen this before? Out of the blue, inbound calls are busy. MGCP Controlled gateway with 12.3(11)T2 IOS. 3745.  I am aware of other issues in this release and am updating, but I've not seen the behavior below before.

Debug isdn q931 was a constant flood of just SETUP requests received. Thats the only q931 message, only the SETUP. Nothing else. 

ISDN was multi-frame established, mgcp was registered, MGCP keepalives occurring. I turned on MGCP debugs also and saw only keepalives and no MGCP Call setup requests, etc during constant q931 SETUPs. I'm going on all the B's were in use since we had constant stream of SETUP messages on different channels that caused busy. The router show voice port, show voice call summary commands showed no activity on the voice ports either during the problem time. 

Nothing in router log for 1-2 months , as it is just  a voice gateway and things have been stable. PRI is error free, no errors. 

I shutdown the D-channel on the router and brought it back up and that cleared the issue. I know this resets MGCP and re-registers but maybe D-channel reestablishing cleared something on telco side.



From raphael at alter.it  Thu Jun 15 04:12:52 2006
From: raphael at alter.it (Raphael Pennisi)
Date: Thu, 15 Jun 2006 10:12:52 +0200
Subject: [cisco-voip] How to retrieve MeetMe Numbers
Message-ID: <1150359173.10126.3.camel@penny>


Dear all,
does somebody knows how to list all conferences Meet Me Number using
AXL?  

Thank you,
Raphael Pennisi
(alter.net Srl)


From Nick.Kassel at Charles-Stanley.co.uk  Thu Jun 15 05:49:10 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 15 Jun 2006 10:49:10 +0100
Subject: [cisco-voip] Unity Subscriber Usage Info
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648438@CSEXCH01.charles-stanley.co.uk>

Does any one know if there is a tool or app that will report on Unity
Subscriber Usage, I have a situation where we have run out of licences
and I would like to find out if any mailboxes have not been used within
a certain period so I can re use the licence.

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From tomlemay at adelphia.net  Thu Jun 15 06:14:06 2006
From: tomlemay at adelphia.net (Tom LeMay)
Date: Thu, 15 Jun 2006 06:14:06 -0400
Subject: [cisco-voip] 7935 Directory Service Can Not Find Server
Message-ID: <OKEPKJBEODBDLLBBOMMEMENBCKAA.tomlemay@adelphia.net>

Recently, we modified the xmldirectory.asp file so that we could merge two
separate call manager clusters together for name and extension lookup.
Everything was working fine except now if someone pushes the directory
service button on a 7935 and types in a name to search on it returns "server
not found."  The model 7940/7960/7970 phones are all functioning normally
with regard to the directory service feature.  I checked the directory
service on the publisher and it is started and running.  I do not see any
alarms or errors in the event viewer which would be associated with a
failure of some sort.  Any suggestions would be appreciated.  Thanks, Tom


From ml at nesys.it  Thu Jun 15 06:28:36 2006
From: ml at nesys.it (Andrea Riela)
Date: Thu, 15 Jun 2006 12:28:36 +0200
Subject: [cisco-voip] about CCM and Genesys IPCC
Message-ID: <172637b3cb2dc02f3eccf54438167848@nesys.it>

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

Hi folks,

someone using CCM cluster with Genesys IP Contact Center?
How JTAPI interface works on CCM cluster? all servers, Publisher and n 
Subscribers, forward calls to IPCC?
I know that T-Server Genesys interface (module that talks with Cisco 
JTAPI) works with redundancy, and drops informations on T-Server 
secondary, but I don't know how CCM works on cluster system ...

Any advice will be appreciated
Regards
Andrea
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Version: GnuPG v1.4.1 (Darwin)

iD8DBQFEkTZUMakHrsrHP9wRArbwAKDFTDaZNq6/7cQ/15/nJjroOHFxMQCgq+6r
uoLsbpRnsyx3vLe7SoTT81c=
=uWwQ
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From CarlosL at mrs.es  Thu Jun 15 10:04:40 2006
From: CarlosL at mrs.es (Carlos Lopez Fernandez)
Date: Thu, 15 Jun 2006 16:04:40 +0200
Subject: [cisco-voip] - CRA
Message-ID: <687262A118FB1146BFF85FB25047267914AA3B@mrsalow3s001.mrs.local>

I?ve installed CRA ina Call Manager 4.1, when the instalation begins it says: "Your license entities you to install: Automated attendant ". When I go to the CRA webadmin, I don?t have the scripts menu, and If I submit a personal script, when  I try to add an application, yit says (log) that "ApplicationLicenseViolationException: user-defined scripts are not allowed with Extended Services". what is the license I need to see the Scripts menu in the CRA web admin. 

Thanks In advance
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From BRiley at jackhenry.com  Thu Jun 15 10:22:26 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Thu, 15 Jun 2006 09:22:26 -0500
Subject: [cisco-voip] - CRA
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002AD132C@smoxchg.jhacorp.com>

What scripts do you want to see? Are you trying to add TAPS?

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carlos Lopez Fernandez
Sent: Thursday, June 15, 2006 9:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] - CRA

 

I?ve installed CRA ina Call Manager 4.1, when the instalation begins it says: "Your license entities you to install: Automated attendant ". When I go to the CRA webadmin, I don?t have the scripts menu, and If I submit a personal script, when  I try to add an application, yit says (log) that "ApplicationLicenseViolationException: user-defined scripts are not allowed with Extended Services". what is the license I need to see the Scripts menu in the CRA web admin. 

Thanks In advance 

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 BRiley at jackhenry.com  Thu Jun 15 10:25:30 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Thu, 15 Jun 2006 09:25:30 -0500
Subject: [cisco-voip] - CRA
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002AD1331@smoxchg.jhacorp.com>

If you are trying to change the AA script you will not be able to. All you can do is change the voice prompt recording. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carlos Lopez Fernandez
Sent: Thursday, June 15, 2006 9:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] - CRA

 

I?ve installed CRA ina Call Manager 4.1, when the instalation begins it says: "Your license entities you to install: Automated attendant ". When I go to the CRA webadmin, I don?t have the scripts menu, and If I submit a personal script, when  I try to add an application, yit says (log) that "ApplicationLicenseViolationException: user-defined scripts are not allowed with Extended Services". what is the license I need to see the Scripts menu in the CRA web admin. 

Thanks In advance 

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 bababooey at cox.net  Thu Jun 15 10:34:07 2006
From: bababooey at cox.net (bababooey at cox.net)
Date: Thu, 15 Jun 2006 10:34:07 -0400
Subject: [cisco-voip] OS 2000.4.1
Message-ID: <6198557.1150382047135.JavaMail.root@eastrmwml01.mgt.cox.net>

Is there anywhere I can download the 2000.4.1 OS install, my original cd is mia?

Thanks,


From rratliff at cisco.com  Thu Jun 15 10:40:30 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 15 Jun 2006 10:40:30 -0400
Subject: [cisco-voip] OS 2000.4.1
In-Reply-To: <6198557.1150382047135.JavaMail.root@eastrmwml01.mgt.cox.net>
References: <6198557.1150382047135.JavaMail.root@eastrmwml01.mgt.cox.net>
Message-ID: <3774DA8A-477A-4BD6-BDEF-87F652621572@cisco.com>

No you need to either order a new set via the PUT tool (www.cisco.com/ 
upgrade) or call TAC and ask for an RMA.

-Ryan

On Jun 15, 2006, at 10:34 AM, <bababooey at cox.net> <bababooey at cox.net>  
wrote:

Is there anywhere I can download the 2000.4.1 OS install, my original  
cd is mia?

Thanks,

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


From ealeatherman at gmail.com  Thu Jun 15 11:25:31 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 15 Jun 2006 11:25:31 -0400
Subject: [cisco-voip] debug isdn q931
Message-ID: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>

I'm fairly certain this is something I am doing wrong but I cant find it for
the life of me..

I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
conferencing. identical firmware loads and configurations, other than 1 has
an FXO card that the other doesn't.

I'm executing on both routers from seperate SSH sessions:
# term mon
# debug isdn q931

On router A, I can see all the Q931 debug messages go by. On router B, I see
nothing on the SSH session however I know calls are being placed on it ( I
can see the channels going in service etc from RTMT). Could I have something
turned off on router B that doesnt show up in the config?

-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From mborchers at igillc.com  Thu Jun 15 11:28:45 2006
From: mborchers at igillc.com (Mark Borchers)
Date: Thu, 15 Jun 2006 10:28:45 -0500
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
Message-ID: <001501c69090$6776d810$4336a8c0@Traveler>

Have you set your logging level high enough to show debug traffic?  See the
logging console command args.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Thursday, June 15, 2006 10:26 AM
To: ciscovoip
Cc: jeff.davis at mail.wvu.edu
Subject: [cisco-voip] debug isdn q931


I'm fairly certain this is something I am doing wrong but I cant find it for
the life of me..

I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
conferencing. identical firmware loads and configurations, other than 1 has
an FXO card that the other doesn't. 

I'm executing on both routers from seperate SSH sessions:
# term mon
# debug isdn q931

On router A, I can see all the Q931 debug messages go by. On router B, I see
nothing on the SSH session however I know calls are being placed on it ( I
can see the channels going in service etc from RTMT). Could I have something
turned off on router B that doesnt show up in the config? 

-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 

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From jsteinberg at gmail.com  Thu Jun 15 11:32:12 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 15 Jun 2006 11:32:12 -0400
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
References: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
Message-ID: <f1b65f880606150832p357de4e3g41429d521de5e330@mail.gmail.com>

I believe you should check for this config:

Router(config)#logging monitor 7

Justin


On 6/15/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
> I'm fairly certain this is something I am doing wrong but I cant find it for
> the life of me..
>
> I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
> conferencing. identical firmware loads and configurations, other than 1 has
> an FXO card that the other doesn't.
>
> I'm executing on both routers from seperate SSH sessions:
> # term mon
> # debug isdn q931
>
> On router A, I can see all the Q931 debug messages go by. On router B, I see
> nothing on the SSH session however I know calls are being placed on it ( I
> can see the channels going in service etc from RTMT). Could I have something
> turned off on router B that doesnt show up in the config?
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From ealeatherman at gmail.com  Thu Jun 15 11:37:02 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 15 Jun 2006 11:37:02 -0400
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <001501c69090$6776d810$4336a8c0@Traveler>
References: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
	<001501c69090$6776d810$4336a8c0@Traveler>
Message-ID: <94a1afde0606150837u3e3ffeaagd36b460afdca9333@mail.gmail.com>

Just had the defaults on both routers, however I tried the "logging console
7" on the problem router, which looks like it should display debuging
messages and it still shows nothing.

Just for grins and giggles I turned on debug cdp packets, and i am getting
debug messages on both routers.. so its something with the ISDN that is
causing it to not display.

On 6/15/06, Mark Borchers <mborchers at igillc.com> wrote:
>
>  Have you set your logging level high enough to show debug traffic?  See
> the logging console command args.
>
>  -----Original Message-----
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Ed Leatherman
> *Sent:* Thursday, June 15, 2006 10:26 AM
> *To:* ciscovoip
> *Cc:* jeff.davis at mail.wvu.edu
> *Subject:* [cisco-voip] debug isdn q931
>
> I'm fairly certain this is something I am doing wrong but I cant find it
> for the life of me..
>
> I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
> conferencing. identical firmware loads and configurations, other than 1 has
> an FXO card that the other doesn't.
>
> I'm executing on both routers from seperate SSH sessions:
> # term mon
> # debug isdn q931
>
> On router A, I can see all the Q931 debug messages go by. On router B, I
> see nothing on the SSH session however I know calls are being placed on it (
> I can see the channels going in service etc from RTMT). Could I have
> something turned off on router B that doesnt show up in the config?
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From hsalama_us at yahoo.com  Thu Jun 15 11:39:45 2006
From: hsalama_us at yahoo.com (Hassan Salama)
Date: Thu, 15 Jun 2006 08:39:45 -0700 (PDT)
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <001501c69090$6776d810$4336a8c0@Traveler>
Message-ID: <20060615153945.56253.qmail@web34615.mail.mud.yahoo.com>

did you execute term monitor?

#term mon

Thanks
--- Mark Borchers <mborchers at igillc.com> wrote:

> Have you set your logging level high enough to show
> debug traffic?  See the
> logging console command args.
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ed Leatherman
> Sent: Thursday, June 15, 2006 10:26 AM
> To: ciscovoip
> Cc: jeff.davis at mail.wvu.edu
> Subject: [cisco-voip] debug isdn q931
> 
> 
> I'm fairly certain this is something I am doing
> wrong but I cant find it for
> the life of me..
> 
> I have 2 ISR routers, 3825 with some T1 interfaces
> and DSPs for
> conferencing. identical firmware loads and
> configurations, other than 1 has
> an FXO card that the other doesn't. 
> 
> I'm executing on both routers from seperate SSH
> sessions:
> # term mon
> # debug isdn q931
> 
> On router A, I can see all the Q931 debug messages
> go by. On router B, I see
> nothing on the SSH session however I know calls are
> being placed on it ( I
> can see the channels going in service etc from
> RTMT). Could I have something
> turned off on router B that doesnt show up in the
> config? 
> 
> -- 
> Ed Leatherman
> IP Telephony Coordinator
> 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
> 


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

From rratliff at cisco.com  Thu Jun 15 11:39:03 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 15 Jun 2006 11:39:03 -0400
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <94a1afde0606150837u3e3ffeaagd36b460afdca9333@mail.gmail.com>
References: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
	<001501c69090$6776d810$4336a8c0@Traveler>
	<94a1afde0606150837u3e3ffeaagd36b460afdca9333@mail.gmail.com>
Message-ID: <32B6A5FB-933C-4D73-BF18-70F7387EF229@cisco.com>

Are both of them PRIs for sure?  Perhaps the one not showing q931  
debugs is actually a channel-group (CAS) and not a PRI?

-Ryan

On Jun 15, 2006, at 11:37 AM, Ed Leatherman wrote:

Just had the defaults on both routers, however I tried the "logging  
console 7" on the problem router, which looks like it should display  
debuging messages and it still shows nothing.

Just for grins and giggles I turned on debug cdp packets, and i am  
getting debug messages on both routers.. so its something with the  
ISDN that is causing it to not display.

On 6/15/06, Mark Borchers <mborchers at igillc.com> wrote:
Have you set your logging level high enough to show debug traffic?   
See the logging console command args.
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip- 
bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Thursday, June 15, 2006 10:26 AM
To: ciscovoip
Cc: jeff.davis at mail.wvu.edu
Subject: [cisco-voip] debug isdn q931

I'm fairly certain this is something I am doing wrong but I cant find  
it for the life of me..

I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for  
conferencing. identical firmware loads and configurations, other than  
1 has an FXO card that the other doesn't.

I'm executing on both routers from seperate SSH sessions:
# term mon
# debug isdn q931

On router A, I can see all the Q931 debug messages go by. On router  
B, I see nothing on the SSH session however I know calls are being  
placed on it ( I can see the channels going in service etc from  
RTMT). Could I have something turned off on router B that doesnt show  
up in the config?

-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From hsalama_us at yahoo.com  Thu Jun 15 11:41:33 2006
From: hsalama_us at yahoo.com (Hassan Salama)
Date: Thu, 15 Jun 2006 08:41:33 -0700 (PDT)
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <20060615153945.56253.qmail@web34615.mail.mud.yahoo.com>
Message-ID: <20060615154133.15118.qmail@web34601.mail.mud.yahoo.com>

sorry, 
i didn't noticed that you had already did:)

--- Hassan Salama <hsalama_us at yahoo.com> wrote:

> did you execute term monitor?
> 
> #term mon
> 
> Thanks
> --- Mark Borchers <mborchers at igillc.com> wrote:
> 
> > Have you set your logging level high enough to
> show
> > debug traffic?  See the
> > logging console command args.
> > 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Ed Leatherman
> > Sent: Thursday, June 15, 2006 10:26 AM
> > To: ciscovoip
> > Cc: jeff.davis at mail.wvu.edu
> > Subject: [cisco-voip] debug isdn q931
> > 
> > 
> > I'm fairly certain this is something I am doing
> > wrong but I cant find it for
> > the life of me..
> > 
> > I have 2 ISR routers, 3825 with some T1 interfaces
> > and DSPs for
> > conferencing. identical firmware loads and
> > configurations, other than 1 has
> > an FXO card that the other doesn't. 
> > 
> > I'm executing on both routers from seperate SSH
> > sessions:
> > # term mon
> > # debug isdn q931
> > 
> > On router A, I can see all the Q931 debug messages
> > go by. On router B, I see
> > nothing on the SSH session however I know calls
> are
> > being placed on it ( I
> > can see the channels going in service etc from
> > RTMT). Could I have something
> > turned off on router B that doesnt show up in the
> > config? 
> > 
> > -- 
> > Ed Leatherman
> > IP Telephony Coordinator
> > 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
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jwydra at Burwood.com  Thu Jun 15 11:56:03 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Thu, 15 Jun 2006 10:56:03 -0500
Subject: [cisco-voip] BAT Export/Import of user database
Message-ID: <97F472A9F637E44B954D7FBABC800430BC6EC4@bur-msg-02.Burwood.com>

I took an export of the CCM user database using BAT (3.3.4sr2). Problem
is that the passwords for the user accounts are encrypted and when I try
to import into another CallManager (4.1.3) it tells me the length of the
password is too long. So basically the new CallManager is seeing the
password field as the actual password instead of the encrypted password.
Is there a way around this? 

 

Jason 

 

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From ealeatherman at gmail.com  Thu Jun 15 14:02:49 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 15 Jun 2006 14:02:49 -0400
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <32B6A5FB-933C-4D73-BF18-70F7387EF229@cisco.com>
References: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
	<001501c69090$6776d810$4336a8c0@Traveler>
	<94a1afde0606150837u3e3ffeaagd36b460afdca9333@mail.gmail.com>
	<32B6A5FB-933C-4D73-BF18-70F7387EF229@cisco.com>
Message-ID: <94a1afde0606151102g3d74c5b3td00661ccb900fb10@mail.gmail.com>

They are definately both PRI

Isnt there some configuration that allows multiple PRI's to share a
D-channel? I specifically told the telco we didnt want this, and i'd be
surprised if it would work without any extra configuration on my side, but
thats the only other thing I could think of.

#sh isdn status
Global ISDN Switchtype = primary-4ess

%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 1. Layer 3 output may not
apply

ISDN Serial0/0/0:23 interface
        dsl 1, interface ISDN Switchtype = primary-ni
        L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
    Layer 1 Status:
        ACTIVE
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
    Layer 3 Status:
        0 Active Layer 3 Call(s)
    Active dsl 1 CCBs = 0
    The Free Channel Mask:  0x807FFFFF
    Number of L2 Discards = 0, L2 Session ID = 203
    Total Allocated ISDN CCBs = 0

On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote:
>
> Are both of them PRIs for sure?  Perhaps the one not showing q931
> debugs is actually a channel-group (CAS) and not a PRI?
>
> -Ryan
>
> On Jun 15, 2006, at 11:37 AM, Ed Leatherman wrote:
>
> Just had the defaults on both routers, however I tried the "logging
> console 7" on the problem router, which looks like it should display
> debuging messages and it still shows nothing.
>
> Just for grins and giggles I turned on debug cdp packets, and i am
> getting debug messages on both routers.. so its something with the
> ISDN that is causing it to not display.
>
> On 6/15/06, Mark Borchers <mborchers at igillc.com> wrote:
> Have you set your logging level high enough to show debug traffic?
> See the logging console command args.
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip-
> bounces at puck.nether.net] On Behalf Of Ed Leatherman
> Sent: Thursday, June 15, 2006 10:26 AM
> To: ciscovoip
> Cc: jeff.davis at mail.wvu.edu
> Subject: [cisco-voip] debug isdn q931
>
> I'm fairly certain this is something I am doing wrong but I cant find
> it for the life of me..
>
> I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
> conferencing. identical firmware loads and configurations, other than
> 1 has an FXO card that the other doesn't.
>
> I'm executing on both routers from seperate SSH sessions:
> # term mon
> # debug isdn q931
>
> On router A, I can see all the Q931 debug messages go by. On router
> B, I see nothing on the SSH session however I know calls are being
> placed on it ( I can see the channels going in service etc from
> RTMT). Could I have something turned off on router B that doesnt show
> up in the config?
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From rratliff at cisco.com  Thu Jun 15 14:03:50 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 15 Jun 2006 14:03:50 -0400
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <94a1afde0606151102g3d74c5b3td00661ccb900fb10@mail.gmail.com>
References: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
	<001501c69090$6776d810$4336a8c0@Traveler>
	<94a1afde0606150837u3e3ffeaagd36b460afdca9333@mail.gmail.com>
	<32B6A5FB-933C-4D73-BF18-70F7387EF229@cisco.com>
	<94a1afde0606151102g3d74c5b3td00661ccb900fb10@mail.gmail.com>
Message-ID: <06BF1548-7DC9-4201-9EC1-5C13AFB8A0A7@cisco.com>

nfas is the protocol that lets you bind multiple PRIs to the same D- 
chan.  Definitely not supported with MGCP though.

How about if you turn on 'deb mgcp packet' along with your q931 debug?

-Ryan

On Jun 15, 2006, at 2:02 PM, Ed Leatherman wrote:

They are definately both PRI

Isnt there some configuration that allows multiple PRI's to share a D- 
channel? I specifically told the telco we didnt want this, and i'd be  
surprised if it would work without any extra configuration on my  
side, but thats the only other thing I could think of.

#sh isdn status
Global ISDN Switchtype = primary-4ess

%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 1. Layer 3 output  
may not apply

ISDN Serial0/0/0:23 interface
         dsl 1, interface ISDN Switchtype = primary-ni
         L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
     Layer 1 Status:
         ACTIVE
     Layer 2 Status:
         TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
     Layer 3 Status:
         0 Active Layer 3 Call(s)
     Active dsl 1 CCBs = 0
     The Free Channel Mask:  0x807FFFFF
     Number of L2 Discards = 0, L2 Session ID = 203
     Total Allocated ISDN CCBs = 0

On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote: Are both of them  
PRIs for sure?  Perhaps the one not showing q931
debugs is actually a channel-group (CAS) and not a PRI?

-Ryan

On Jun 15, 2006, at 11:37 AM, Ed Leatherman wrote:

Just had the defaults on both routers, however I tried the "logging
console 7" on the problem router, which looks like it should display
debuging messages and it still shows nothing.

Just for grins and giggles I turned on debug cdp packets, and i am
getting debug messages on both routers.. so its something with the
ISDN that is causing it to not display.

On 6/15/06, Mark Borchers <mborchers at igillc.com> wrote:
Have you set your logging level high enough to show debug traffic?
See the logging console command args.
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip-
bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Thursday, June 15, 2006 10:26 AM
To: ciscovoip
Cc: jeff.davis at mail.wvu.edu
Subject: [cisco-voip] debug isdn q931

I'm fairly certain this is something I am doing wrong but I cant find
it for the life of me..

I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
conferencing. identical firmware loads and configurations, other than
1 has an FXO card that the other doesn't.

I'm executing on both routers from seperate SSH sessions:
# term mon
# debug isdn q931

On router A, I can see all the Q931 debug messages go by. On router
B, I see nothing on the SSH session however I know calls are being
placed on it ( I can see the channels going in service etc from
RTMT). Could I have something turned off on router B that doesnt show
up in the config?

--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations



--
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip




-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations


From ealeatherman at gmail.com  Thu Jun 15 14:21:44 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Thu, 15 Jun 2006 14:21:44 -0400
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <06BF1548-7DC9-4201-9EC1-5C13AFB8A0A7@cisco.com>
References: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
	<001501c69090$6776d810$4336a8c0@Traveler>
	<94a1afde0606150837u3e3ffeaagd36b460afdca9333@mail.gmail.com>
	<32B6A5FB-933C-4D73-BF18-70F7387EF229@cisco.com>
	<94a1afde0606151102g3d74c5b3td00661ccb900fb10@mail.gmail.com>
	<06BF1548-7DC9-4201-9EC1-5C13AFB8A0A7@cisco.com>
Message-ID: <94a1afde0606151121j3206ffc7k1a7a0f1217c1fa50@mail.gmail.com>

debug MGCP packet appears to work fine.

On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote:
>
> nfas is the protocol that lets you bind multiple PRIs to the same D-
> chan.  Definitely not supported with MGCP though.
>
> How about if you turn on 'deb mgcp packet' along with your q931 debug?
>
> -Ryan
>
> On Jun 15, 2006, at 2:02 PM, Ed Leatherman wrote:
>
> They are definately both PRI
>
> Isnt there some configuration that allows multiple PRI's to share a D-
> channel? I specifically told the telco we didnt want this, and i'd be
> surprised if it would work without any extra configuration on my
> side, but thats the only other thing I could think of.
>
> #sh isdn status
> Global ISDN Switchtype = primary-4ess
>
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 1. Layer 3 output
> may not apply
>
> ISDN Serial0/0/0:23 interface
>          dsl 1, interface ISDN Switchtype = primary-ni
>          L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
>      Layer 1 Status:
>          ACTIVE
>      Layer 2 Status:
>          TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
>      Layer 3 Status:
>          0 Active Layer 3 Call(s)
>      Active dsl 1 CCBs = 0
>      The Free Channel Mask:  0x807FFFFF
>      Number of L2 Discards = 0, L2 Session ID = 203
>      Total Allocated ISDN CCBs = 0
>
> On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote: Are both of them
> PRIs for sure?  Perhaps the one not showing q931
> debugs is actually a channel-group (CAS) and not a PRI?
>
> -Ryan
>
> On Jun 15, 2006, at 11:37 AM, Ed Leatherman wrote:
>
> Just had the defaults on both routers, however I tried the "logging
> console 7" on the problem router, which looks like it should display
> debuging messages and it still shows nothing.
>
> Just for grins and giggles I turned on debug cdp packets, and i am
> getting debug messages on both routers.. so its something with the
> ISDN that is causing it to not display.
>
> On 6/15/06, Mark Borchers <mborchers at igillc.com> wrote:
> Have you set your logging level high enough to show debug traffic?
> See the logging console command args.
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip-
> bounces at puck.nether.net] On Behalf Of Ed Leatherman
> Sent: Thursday, June 15, 2006 10:26 AM
> To: ciscovoip
> Cc: jeff.davis at mail.wvu.edu
> Subject: [cisco-voip] debug isdn q931
>
> I'm fairly certain this is something I am doing wrong but I cant find
> it for the life of me..
>
> I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
> conferencing. identical firmware loads and configurations, other than
> 1 has an FXO card that the other doesn't.
>
> I'm executing on both routers from seperate SSH sessions:
> # term mon
> # debug isdn q931
>
> On router A, I can see all the Q931 debug messages go by. On router
> B, I see nothing on the SSH session however I know calls are being
> placed on it ( I can see the channels going in service etc from
> RTMT). Could I have something turned off on router B that doesnt show
> up in the config?
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From jgrace at digitelusa.net  Thu Jun 15 14:46:30 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu, 15 Jun 2006 14:46:30 -0400
Subject: [cisco-voip] what the best way to create a back door for VM
Message-ID: <20060615145878.SM01464@3729434770>

Ive seen site the has a back door number config on there system.  When you
call it,  it goes direct to "please enter your ID"

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From rratliff at cisco.com  Thu Jun 15 15:02:55 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 15 Jun 2006 15:02:55 -0400
Subject: [cisco-voip] what the best way to create a back door for VM
In-Reply-To: <20060615145878.SM01464@3729434770>
References: <20060615145878.SM01464@3729434770>
Message-ID: <6A4ECBEE-5F56-4473-9840-9267A323A611@cisco.com>

Just take a spare DID, route it through a CTI RP that is CFA into vm.
Then in Unity make a call routing rule that sends callers to  
subscriber sign in.

-Ryan

On Jun 15, 2006, at 2:46 PM, James Grace wrote:

Ive seen site the has a back door number config on there system.   
When you call it,  it goes direct to ?please enter your ID?



James D. Grace

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation



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



From Scott.Voll at wesd.org  Thu Jun 15 16:48:16 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu, 15 Jun 2006 13:48:16 -0700
Subject: [cisco-voip] Unity Subscriber Usage Info
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEED96@Apollo.wesd.org>

Try the Subscriber information dump.

 

Click the time of last call or set for first time enrollment.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Thursday, June 15, 2006 2:49 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Subscriber Usage Info

 

Does any one know if there is a tool or app that will report on Unity
Subscriber Usage, I have a situation where we have run out of licences
and I would like to find out if any mailboxes have not been used within
a certain period so I can re use the licence.

 


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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
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Authorised and Regulated by the Financial Services Authority, Member of
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From Joe.Pollere at us.didata.com  Thu Jun 15 18:58:18 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Thu, 15 Jun 2006 18:58:18 -0400
Subject: [cisco-voip] Unity Subscriber Usage Info
Message-ID: <C65033A817B5734392E55CB9DF44610405692B7E@USNAEXCH.na.didata.local>

Exchange Message Store Manager might give you what you need.

http://www.ciscounitytools.com/App_MSM.htm

?
Joe

________________________________________
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Thursday, June 15, 2006 5:49 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity Subscriber Usage Info

Does any one know if there is a tool or app that will report on Unity Subscriber Usage, I have a situation where we have run out of licences and I would like to find out if any mailboxes have not been used within a certain period so I can re use the licence.


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From wael733 at hotmail.com  Thu Jun 15 19:53:33 2006
From: wael733 at hotmail.com (wael ahmed el mezain)
Date: Fri, 16 Jun 2006 00:53:33 +0100
Subject: [cisco-voip] debug isdn q931
Message-ID: <BAY105-W61133933EEB2EBDAA7E6BE7820@phx.gbl>

yes, "debug MGCP packet" plus "sh mgcp connection" during the call if you can make a call and "sh call .... " & "sh voice ..." commands

and may "debug call treatment action" help also


-----------------------------------------------
> Date: Thu, 15 Jun 2006 14:21:44 -0400
> From: ealeatherman at gmail.com
> To: rratliff at cisco.com
> CC: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] debug isdn q931
> 
> debug MGCP packet appears to work fine.
> On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote:
> nfas is the protocol that lets you bind multiple PRIs to the same D-
> chan.  Definitely not supported with MGCP though.
> How about if you turn on 'deb mgcp packet' along with your q931 debug?
> -Ryan
> On Jun 15, 2006, at 2:02 PM, Ed Leatherman wrote:
> They are definately both PRI
> Isnt there some configuration that allows multiple PRI's to share a D-
> channel? I specifically told the telco we didnt want this, and i'd be
> surprised if it would work without any extra configuration on my
> side, but thats the only other thing I could think of.
> #sh isdn status
> Global ISDN Switchtype = primary-4ess
> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 1. Layer 3 output
> may not apply
> ISDN Serial0/0/0:23 interface
>          dsl 1, interface ISDN Switchtype = primary-ni
>          L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
>      Layer 1 Status:
>          ACTIVE
>      Layer 2 Status:
>          TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
>      Layer 3 Status:
>          0 Active Layer 3 Call(s)
>      Active dsl 1 CCBs = 0
>      The Free Channel Mask:  0x807FFFFF
>      Number of L2 Discards = 0, L2 Session ID = 203
>      Total Allocated ISDN CCBs = 0
> On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote: Are both of them
> PRIs for sure?  Perhaps the one not showing q931
> debugs is actually a channel-group (CAS) and not a PRI?
> -Ryan
> On Jun 15, 2006, at 11:37 AM, Ed Leatherman wrote:
> Just had the defaults on both routers, however I tried the "logging
> console 7" on the problem router, which looks like it should display
> debuging messages and it still shows nothing.
> Just for grins and giggles I turned on debug cdp packets, and i am
> getting debug messages on both routers.. so its something with the
> ISDN that is causing it to not display.
> On 6/15/06, Mark Borchers <mborchers at igillc.com> wrote:
> Have you set your logging level high enough to show debug traffic?
> See the logging console command args.
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip-
> bounces at puck.nether.net ] On Behalf Of Ed Leatherman
> Sent: Thursday, June 15, 2006 10:26 AM
> To: ciscovoip
> Cc: jeff.davis at mail.wvu.edu
> Subject: [cisco-voip] debug isdn q931
> I'm fairly certain this is something I am doing wrong but I cant find
> it for the life of me..
> I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
> conferencing. identical firmware loads and configurations, other than
> 1 has an FXO card that the other doesn't.
> I'm executing on both routers from seperate SSH sessions:
> # term mon
> # debug isdn q931
> On router A, I can see all the Q931 debug messages go by. On router
> B, I see nothing on the SSH session however I know calls are being
> placed on it ( I can see the channels going in service etc from
> RTMT). Could I have something turned off on router B that doesnt show
> up in the config?
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations

_________________________________________________________________
Express yourself instantly with MSN Messenger! Download today it's FREE!
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

From netwirker at gmail.com  Fri Jun 16 00:59:23 2006
From: netwirker at gmail.com (netwirker)
Date: Fri, 16 Jun 2006 00:59:23 -0400
Subject: [cisco-voip] CME, CUE and T1 trunk with 3845
Message-ID: <44923a2f.0aad4049.2b0a.4931@mx.gmail.com>

Hi guys,

I will set up CME, CUE and install one PSTN PRI T1 in one Cisco 3845 based on customer request.
I didn't configure these services in one router before and wondering if Cisco router can stand
the burden and performance well. As my experience CUE and T1 link can't install in one router otherwise
serious voice quality problem will happen. What's your idea, any suggestion to optimize performance?
Client want to set up 120 phone on CME and 200 mailbox on CUE.

Btw, what T1 module I need install for 3845 if I need DID feature?

Thanks a lot!  

Andrew Xue
netwirker at gmail.com



From jonvoip at gmail.com  Fri Jun 16 01:18:20 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 16 Jun 2006 00:18:20 -0500
Subject: [cisco-voip] vmware gsx server
In-Reply-To: <376E9303-3B3A-4EA3-82D6-A365FE405B55@mimectl>
References: <20861400.1149868558357.JavaMail.root@eastrmwml02.mgt.cox.net>
	<5d093f9a0606101941o8b7f560w93b00f7c48de4937@mail.gmail.com>
	<448D8AD9.7080803@comcast.net>
	<376E9303-3B3A-4EA3-82D6-A365FE405B55@mimectl>
Message-ID: <5d093f9a0606152218n413da11jbb0bcf58cab15f35@mail.gmail.com>

My VMWare images are 20GB+

MY CCM5 image is 72GB.

All I do is build a VM at 20GB with SCSI, installl Win2K, add SNMP,
apply my patented reg hack, install OS2.7, then install CCM... No
biggie.



Jonathan

On 6/12/06, Jason Burton <jburton at netechcorp.com> wrote:
>
>
>
> I have used ESX server and have had no problems with it.  Should work fine.
>
>  ________________________________
>  From: Sam Munzani
> Sent: Mon 6/12/2006 11:40 AM
> To: Jonathan Charles
> Cc: cisco-voip at puck.nether.net; bababooey at cox.net
>
> Subject: Re: [cisco-voip] vmware gsx server
>
>
> Did anybody try ESX server with CCM and Unity? I have access to my company's
> lab where we have ESX server on very powerful PC.
>
> Thanks,
> Sam
>
>
> Yes. To be honest, that is all I have been using... I am running CallManager
> 3.3, 4.0, 4.1, 4.2 and 5.0(2) and Unity 4.0(4), 4.0(5), 4.1(1) and 4.2(1),
> as well as CRA 3.5 and 4.0 all on VMWare GSX 3.1
>  I also have some test bed client machines (for BearShare and client
> apps)... To be honest, I don't know how you guys run labs without VMWare...
> Jonathan On 6/9/06, bababooey at cox.net mailto:bababooey at cox.net wrote:
>
> Has anyone utilized the gsx server to run the call mamanger environment in a
> vmware test bed.
>
> 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 bloemhard_johan at silverfalls.k12.or.us  Fri Jun 16 02:42:29 2006
From: bloemhard_johan at silverfalls.k12.or.us (Johan Bloemhard)
Date: Thu, 15 Jun 2006 23:42:29 -0700
Subject: [cisco-voip] AD plugin on a W2k3 Domain
Message-ID: <ED9CD6C06120064590FC3BBA964D32A7EB10A2@msexch1.silverfalls.k12.or.us>

Are you using SSL or rather LDAPS?  I know that AD 2003 has some
differences in what you can do through LDAP-requiring LDAPS for some
operations.

 

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 Jason Burton
Sent: Tuesday, June 13, 2006 8:55 AM
To: Ryan Ratliff
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain
Importance: Low

 

Nope same result.

 

________________________________

From: Ryan Ratliff
Sent: Tue 6/13/2006 11:51 AM
To: Jason Burton
Cc: Kevin Thorngren; cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain

Try CN=user, CN=container,DC=domain,DC=com.
 
 
-Ryan
 
On Jun 13, 2006, at 11:46 AM, Jason Burton wrote:
 
Well after doing the packet capture I'm getting invalid credentials  
error in the Bind result.
 
It's LDAPErr DSID-0c09334.  What is the correct format for AD Admin DN?
 
CN=USER, OU=Container, DC=domain, DC=Domain?
 
From: Jason Burton
Sent: Tue 6/13/2006 11:08 AM
To: Kevin Thorngren
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain
 
Thanks, downloading Ethereal now.
 
From: Kevin Thorngren
Sent: Tue 6/13/2006 11:01 AM
To: Jason Burton
Cc: cisco-voip Voip
Subject: Re: [cisco-voip] AD plugin on a W2k3 Domain
 
The quickest way to isolate the issue is with a packet capture of the  
Publisher when doing the AD Plugin. Looking at the LDAP packets  
should help you see what the failure is. Kevin On Jun 13, 2006, at  
10:52 AM, Jason Burton wrote: > Nope, just a router on a stick. > >  
From: Voll, Scott > Sent: Tue 6/13/2006 10:50 AM > To: Jason Burton;  
cisco-voip at puck.nether.net > Subject: RE: [cisco-voip] AD plugin on a  
W2k3 Domain > > Is there any pix between the CM and AD?  If so, we  
ran into a problem > with the fixup 389. >   > > From: cisco-voip- 
bounces at puck.nether.net > [mailto:cisco-voip-bounces at puck.nether.net]  
On Behalf Of Jason Burton > Sent: Tuesday, June 13, 2006 7:43 AM >  
To: cisco-voip at puck.nether.net > Subject: [cisco-voip] AD plugin on a  
W2k3 Domain >   >  I'm trying to install the AD plugin and keep  
getting the following > error: >   > Error:Setup failed to connect.  
Please enter your Host Name, Port > Number, Directory Administrator  
DN and Password again! >   > I've verified these 100 times over and  
still no luck.  This is the > first time I've tried this to a widnows  
2003 domain.  Any ideas? >   > 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
 

______________________
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 jcampbell at ci.davidson.nc.us  Fri Jun 16 08:22:33 2006
From: jcampbell at ci.davidson.nc.us (Campbell, Jeremy P.)
Date: Fri, 16 Jun 2006 08:22:33 -0400
Subject: [cisco-voip] ip communicator handset recommendations?
Message-ID: <F81B53847D3DC945A55B555B5555CC79AF74F1@davnc03.ci.davidson.nc.us>

Can anyone recommend a USB handset to use in conjunction with IP
Communicator?  If such a thing exists...
 
Thanks,
Jeremy
 
________________________________________ 
Jeremy P. Campbell 
Information Systems Director 
Town of Davidson, NC 
704-940-9632 - voice direct 
704-892-7591 - voice main 
704-892-3971 - fax 
jcampbell at ci.davidson.nc.us
 
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From rleetun at co.boulder.co.us  Fri Jun 16 08:28:39 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Fri, 16 Jun 2006 06:28:39 -0600
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
Message-ID: <963CE94787005149A52C17F33A028E12051D4888@EVS1B.boco.co.boulder.co.us>

Let's take this communication in another direction...  This is very
interesting to read all the comments and questions about Vmware.  When
we purchased our environment from Cisco.  We had problems with TAC
support after the install because our environment was not standard in
their opinion.  I have two questions: 

1. Does Cisco support the Vmware environment?  

2. And if so, what are the parameters for support and guidance?

3. And, what products do they support, e.g. Windows?

This could save us $$ when it comes to replacing servers!

Thanks.

Rob
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From kthorngr at cisco.com  Fri Jun 16 08:37:25 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri, 16 Jun 2006 08:37:25 -0400
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
In-Reply-To: <963CE94787005149A52C17F33A028E12051D4888@EVS1B.boco.co.boulder.co.us>
References: <963CE94787005149A52C17F33A028E12051D4888@EVS1B.boco.co.boulder.co.us>
Message-ID: <423ec73646e14e57df9264340bf84926@cisco.com>

Cisco does not support VMWare for CallManager, etc.  The only Cisco 
supported installations are the ones described in the docs on CCO.

Kevin
On Jun 16, 2006, at 8:28 AM, Leetun, Rob wrote:

> Let's take this communication in another direction?? This is very 
> interesting to read all the comments and questions about Vmware.? When 
> we purchased our environment from Cisco.? We had problems with TAC 
> support after the install because our environment was not standard in 
> their opinion.? I have two questions:
>
> 1. Does Cisco support the Vmware environment??
>
> 2. And if so, what are the parameters for support and guidance?
>
> 3. And, what products do they support, e.g. Windows?
>
> This could save us $$ when it comes to replacing servers!
>
> Thanks.
>
> Rob
> _______________________________________________
> 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 Jun 16 08:38:41 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Fri, 16 Jun 2006 07:38:41 -0500
Subject: [cisco-voip] ip communicator handset recommendations?
In-Reply-To: <F81B53847D3DC945A55B555B5555CC79AF74F1@davnc03.ci.davidson.nc.us>
Message-ID: <OFCE5F8E68.B20F989E-ON8625718F.004563E3-8625718F.004563A1@ra.rockwell.com>

Plantronics Road Warrior

It has a built in DSP to offload that work from the main CPU.  Works nice
as even if the machine gets busy the sound quality does not cut out.

Craig




                                                                           
             "Campbell, Jeremy                                             
             P."                                                           
             <jcampbell at ci.dav                                          To 
             idson.nc.us>              <cisco-voip at puck.nether.net>        
             Sent by:                                                   cc 
             cisco-voip-bounce                                             
             s at puck.nether.net                                     Subject 
                                       [cisco-voip] ip communicator        
                                       handset recommendations?            
             06/16/2006 07:22                                              
             AM                                                            
                                                                           
                                                                           
                                                                           
                                                                           




Can anyone recommend a USB handset to use in conjunction with IP
Communicator?  If such a thing exists?





Thanks,


Jeremy





________________________________________
Jeremy P. Campbell
Information Systems Director
Town of Davidson, NC
704-940-9632 - voice direct
704-892-7591 - voice main
704-892-3971 - fax


jcampbell at ci.davidson.nc.us


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




From reports at cmreports.com  Fri Jun 16 08:39:04 2006
From: reports at cmreports.com (Reports@CMReports.com)
Date: Fri, 16 Jun 2006 08:39:04 -0400
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
References: <963CE94787005149A52C17F33A028E12051D4888@EVS1B.boco.co.boulder.co.us>
Message-ID: <090f01c69141$dc432ed0$5100ae0c@HP8240>

Vmware Supported by Cisco...Thoughts??1)  No.

2)  See #1.

3)  4.x train = Windows, 5.x train = Red Hat Linux, 6.x train = Windows and Red Hat


  ----- Original Message ----- 
  From: Leetun, Rob 
  To: cisco-voip at puck.nether.net 
  Sent: Friday, June 16, 2006 8:28 AM
  Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??


  Let's take this communication in another direction.  This is very interesting to read all the comments and questions about Vmware.  When we purchased our environment from Cisco.  We had problems with TAC support after the install because our environment was not standard in their opinion.  I have two questions: 

  1. Does Cisco support the Vmware environment?  

  2. And if so, what are the parameters for support and guidance? 

  3. And, what products do they support, e.g. Windows? 

  This could save us $$ when it comes to replacing servers! 

  Thanks. 

  Rob 



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


  _______________________________________________
  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  Fri Jun 16 08:38:29 2006
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Fri, 16 Jun 2006 08:38:29 -0400
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
Message-ID: <07A14D6987A5F647B2415D72EDF63B5605550238@tnsex01.tns.its.psu.edu>

This seems like an insane direction to go if you're talking about a
production environment.  I doubt you are, but you didn't specify "test".  It
would not be supported, your whole cluster would crash if the host machine
went down, realtime processing would suffer, etc.......
 
Because we've been running Cisco AVVID stuff for about five years now we
have been able to just use outdated hardware that's been removed from
production for our test environment.  It's a little slower, but overall,
probably no slower than emulated machines in VMware.  (Probably faster,
actually...)

---
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/ <http://tns.its.psu.edu/> 



 


  _____  

From: Leetun, Rob [mailto:rleetun at co.boulder.co.us] 
Sent: Friday, June 16, 2006 8:29 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??



Let's take this communication in another direction...  This is very
interesting to read all the comments and questions about Vmware.  When we
purchased our environment from Cisco.  We had problems with TAC support
after the install because our environment was not standard in their opinion.
I have two questions: 

1. Does Cisco support the Vmware environment?  

2. And if so, what are the parameters for support and guidance? 

3. And, what products do they support, e.g. Windows? 

This could save us $$ when it comes to replacing servers! 

Thanks. 

Rob 

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From rleetun at co.boulder.co.us  Fri Jun 16 08:41:23 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Fri, 16 Jun 2006 06:41:23 -0600
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
Message-ID: <963CE94787005149A52C17F33A028E12051D488B@EVS1B.boco.co.boulder.co.us>

Thanks for the response Kevin.  Other than for the obvious reason I
stated earlier.  Why is everyone on this site installing 4.1.3 and etc.
on Vmware if it is not supported?  When will Cisco support Vmware?
 
Rob

________________________________

From: Kevin Thorngren [mailto:kthorngr at cisco.com] 
Sent: Friday, June 16, 2006 6:37 AM
To: Leetun, Rob
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??


Cisco does not support VMWare for CallManager, etc. The only Cisco
supported installations are the ones described in the docs on CCO. 

Kevin 
On Jun 16, 2006, at 8:28 AM, Leetun, Rob wrote: 


	Let's take this communication in another direction...  This is
very interesting to read all the comments and questions about Vmware.
When we purchased our environment from Cisco.  We had problems with TAC
support after the install because our environment was not standard in
their opinion.  I have two questions: 

	1. Does Cisco support the Vmware environment?  

	2. And if so, what are the parameters for support and guidance? 

	3. And, what products do they support, e.g. Windows? 

	This could save us $$ when it comes to replacing servers! 

	Thanks. 

	Rob 
	_______________________________________________ 
	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  Fri Jun 16 08:50:45 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 16 Jun 2006 08:50:45 -0400
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
In-Reply-To: <963CE94787005149A52C17F33A028E12051D488B@EVS1B.boco.co.boulder.co.us>
References: <963CE94787005149A52C17F33A028E12051D488B@EVS1B.boco.co.boulder.co.us>
Message-ID: <94a1afde0606160550t43f7168bg4e8a090b9335e245@mail.gmail.com>

I think most are setting it up for test labs or demo boxes.. we had IPCC
demo'ed to us the first time around running on VMware installed on a little
shuttle PC (with CCM and agent desktop on a different VM's). The engineer
only had to lug around 1 small PC to run the demo.

On 6/16/06, Leetun, Rob <rleetun at co.boulder.co.us> wrote:
>
>  Thanks for the response Kevin.  Other than for the obvious reason I
> stated earlier.  Why is everyone on this site installing 4.1.3 and etc. on
> Vmware if it is not supported?  When will Cisco support Vmware?
>
> Rob
>
>  ------------------------------
> *From:* Kevin Thorngren [mailto:kthorngr at cisco.com]
> *Sent:* Friday, June 16, 2006 6:37 AM
> *To:* Leetun, Rob
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??
>
>  Cisco does not support VMWare for CallManager, etc. The only Cisco
> supported installations are the ones described in the docs on CCO.
>
> Kevin
> On Jun 16, 2006, at 8:28 AM, Leetun, Rob wrote:
>
>  Let's take this communication in another direction?  This is very
> interesting to read all the comments and questions about Vmware.  When we
> purchased our environment from Cisco.  We had problems with TAC support
> after the install because our environment was not standard in their
> opinion.  I have two questions:
>
> 1. Does Cisco support the Vmware environment?
>
> 2. And if so, what are the parameters for support and guidance?
>
> 3. And, what products do they support, e.g. Windows?
>
> This could save us $$ when it comes to replacing servers!
>
> Thanks.
>
> Rob
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From sean at northrock.bm  Fri Jun 16 08:50:25 2006
From: sean at northrock.bm (Sean Watkins)
Date: Fri, 16 Jun 2006 09:50:25 -0300
Subject: [cisco-voip] hunt-group report on CME 3.4?
Message-ID: <B146219F50F53348A6EF98F14B4202C688E94F@sumo2.northrock.local>

Hi,

Is anyone on this list using CME 3.4, and the hunt-group report commands? I
have noticed it decides at times, not to include all extensions; anyone seen
this or know any work arounds?

Thanks,
Sean

From CORTIZ at broward.org  Fri Jun 16 09:00:09 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 16 Jun 2006 09:00:09 -0400
Subject: [cisco-voip] Unity Timers
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC34BD@MREXVS01.bc.broward.cty>

Is there a way to increase the amount of time Unity waits between
buttons when accepting digits from callers?

 

That is to say that, if someone is trying to dial extension 7981 and
they hesitate a fraction of a second too long between 9 and 8 it doesn't
let them finish entering the extension.  I am just curious if it's
possible even if it's not recommended.

 

Carlos

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From kthorngr at cisco.com  Fri Jun 16 09:14:59 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri, 16 Jun 2006 09:14:59 -0400
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
In-Reply-To: <963CE94787005149A52C17F33A028E12051D488B@EVS1B.boco.co.boulder.co.us>
References: <963CE94787005149A52C17F33A028E12051D488B@EVS1B.boco.co.boulder.co.us>
Message-ID: <a40fb8757879aadd50e31058a01e3111@cisco.com>

I am not privy to any discussions in Cisco regarding vmware but I can 
give you my opinion on why vmware is not supported today.  I am also 
not aware of any discussions for future support and would guess that it 
is not on the roadmap.  Even if we were to open up support just for 
demo/lab purposes we would end up with customers deploying in a 
production environment and expecting support when things don't work as 
expected.

For us to support vmware we would need to verify the operation and 
performance of each version of CallManager (possibly including Service 
Releases and Engineering Specials) on vmware.  This would, at a 
minimum, double our testing effort which would cause major delays in 
getting product and updates out for our customers.  In addition we 
would need to create design guidelines for using vmware which would 
have an impact on product release timeframes.

 From a TAC support perspective; vmware would increase the complexity of 
troubleshooting operational and performance issues.  Currently we 
support the CallManager OS.  Adding vmware to the mix would greatly 
increase the complexity of now having to look at vmware and the host 
server's OS as potential issues.

Since we don't test vmware we can't provide nor guarantee any sort of 
performance levels for running CallManager on vmware.  I would highly 
recommend NOT using vmware in a production environment.

Again this is just my opinion and not official word from Cisco :-)

Kevin

On Jun 16, 2006, at 8:41 AM, Leetun, Rob wrote:

> Thanks for the response Kevin.? Other than for the obvious reason I 
> stated earlier.? Why is everyone on this site installing 4.1.3 and 
> etc. on Vmware if it is not supported?? When will Cisco support 
> Vmware?
> ?
> Rob
>
> From: Kevin Thorngren [mailto:kthorngr at cisco.com]
> Sent: Friday, June 16, 2006 6:37 AM
> To: Leetun, Rob
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??
>
> Cisco does not support VMWare for CallManager, etc. The only Cisco 
> supported installations are the ones described in the docs on CCO.
>
> Kevin
> On Jun 16, 2006, at 8:28 AM, Leetun, Rob wrote:
>
>> Let's take this communication in another direction?? This is very 
>> interesting to read all the comments and questions about Vmware.? 
>> When we purchased our environment from Cisco.? We had problems with 
>> TAC support after the install because our environment was not 
>> standard in their opinion.? I have two questions:
>>
>> 1. Does Cisco support the Vmware environment??
>>
>> 2. And if so, what are the parameters for support and guidance?
>>
>> 3. And, what products do they support, e.g. Windows?
>>
>> This could save us $$ when it comes to replacing servers!
>>
>> Thanks.
>>
>> Rob
>> _______________________________________________
>> 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  Fri Jun 16 09:09:07 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri, 16 Jun 2006 09:09:07 -0400
Subject: [cisco-voip] is there a way for on call to ringback like parked call
Message-ID: <20060616092178.SM01464@3729434770>

is there a way for on call to ringback like parked call

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From kthorngr at cisco.com  Fri Jun 16 09:30:36 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri, 16 Jun 2006 09:30:36 -0400
Subject: [cisco-voip] is there a way for on call to ringback like parked
	call
In-Reply-To: <20060616092178.SM01464@3729434770>
References: <20060616092178.SM01464@3729434770>
Message-ID: <cd358261051a64f782ca44728c5808b2@cisco.com>

I believe your are referring to a new feature introduced in CCM 
4.2(1)sr1 called "Call Hold Reversion".

Kevin
On Jun 16, 2006, at 9:09 AM, James Grace wrote:

> is there a way for on call to ringback like parked call
> ?
> James D. Grace
> CCNP CCNA MCSE MCDBA
> Sr. System Engineer / Professional Svc.
> Digitel Corporation
> ?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From jwydra at Burwood.com  Fri Jun 16 10:15:04 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Fri, 16 Jun 2006 09:15:04 -0500
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
Message-ID: <97F472A9F637E44B954D7FBABC800430BC7189@bur-msg-02.Burwood.com>

Lab only here. Never seen it in production. Think of all of the money
that HP, IBM and Cisco would lose if they even thought about supporting
VMware. Not to mention the obvious single point of failure. I suppose
you could have a hot standby VMware machine that would take over
whenever any of the components failed. Still very scary though if you
ask me.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Friday, June 16, 2006 7:51 AM
To: Leetun, Rob
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??

 

I think most are setting it up for test labs or demo boxes.. we had IPCC
demo'ed to us the first time around running on VMware installed on a
little shuttle PC (with CCM and agent desktop on a different VM's). The
engineer only had to lug around 1 small PC to run the demo. 

On 6/16/06, Leetun, Rob <rleetun at co.boulder.co.us> wrote:

Thanks for the response Kevin.  Other than for the obvious reason I
stated earlier.  Why is everyone on this site installing 4.1.3 and etc.
on Vmware if it is not supported?  When will Cisco support Vmware?

 

Rob

 

________________________________

From: Kevin Thorngren [mailto:kthorngr at cisco.com] 
Sent: Friday, June 16, 2006 6:37 AM
To: Leetun, Rob
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??

Cisco does not support VMWare for CallManager, etc. The only Cisco
supported installations are the ones described in the docs on CCO. 

 

Kevin 

On Jun 16, 2006, at 8:28 AM, Leetun, Rob wrote: 

 

Let's take this communication in another direction...  This is very
interesting to read all the comments and questions about Vmware.  When
we purchased our environment from Cisco.  We had problems with TAC
support after the install because our environment was not standard in
their opinion.  I have two questions: 

 

1. Does Cisco support the Vmware environment?  

 

2. And if so, what are the parameters for support and guidance? 

 

3. And, what products do they support, e.g. Windows? 

 

This could save us $$ when it comes to replacing servers! 

 

Thanks. 

 

Rob 

_______________________________________________ 

cisco-voip mailing list 

cisco-voip at puck.nether.net 

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


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






-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 

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From jwydra at Burwood.com  Fri Jun 16 10:20:07 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Fri, 16 Jun 2006 09:20:07 -0500
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
Message-ID: <97F472A9F637E44B954D7FBABC800430BC7190@bur-msg-02.Burwood.com>

Does anybody know of a version of VMware that will run on Windows
Mobile?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wydra, Jason
Sent: Friday, June 16, 2006 9:15 AM
To: Ed Leatherman; Leetun, Rob
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??

 

Lab only here. Never seen it in production. Think of all of the money
that HP, IBM and Cisco would lose if they even thought about supporting
VMware. Not to mention the obvious single point of failure. I suppose
you could have a hot standby VMware machine that would take over
whenever any of the components failed. Still very scary though if you
ask me.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
Sent: Friday, June 16, 2006 7:51 AM
To: Leetun, Rob
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??

 

I think most are setting it up for test labs or demo boxes.. we had IPCC
demo'ed to us the first time around running on VMware installed on a
little shuttle PC (with CCM and agent desktop on a different VM's). The
engineer only had to lug around 1 small PC to run the demo. 

On 6/16/06, Leetun, Rob <rleetun at co.boulder.co.us> wrote:

Thanks for the response Kevin.  Other than for the obvious reason I
stated earlier.  Why is everyone on this site installing 4.1.3 and etc.
on Vmware if it is not supported?  When will Cisco support Vmware?

 

Rob

 

________________________________

From: Kevin Thorngren [mailto:kthorngr at cisco.com] 
Sent: Friday, June 16, 2006 6:37 AM
To: Leetun, Rob
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??

Cisco does not support VMWare for CallManager, etc. The only Cisco
supported installations are the ones described in the docs on CCO. 

 

Kevin 

On Jun 16, 2006, at 8:28 AM, Leetun, Rob wrote: 

 

Let's take this communication in another direction...  This is very
interesting to read all the comments and questions about Vmware.  When
we purchased our environment from Cisco.  We had problems with TAC
support after the install because our environment was not standard in
their opinion.  I have two questions: 

 

1. Does Cisco support the Vmware environment?  

 

2. And if so, what are the parameters for support and guidance? 

 

3. And, what products do they support, e.g. Windows? 

 

This could save us $$ when it comes to replacing servers! 

 

Thanks. 

 

Rob 

_______________________________________________ 

cisco-voip mailing list 

cisco-voip at puck.nether.net 

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


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




-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 

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From ealeatherman at gmail.com  Fri Jun 16 10:27:43 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 16 Jun 2006 10:27:43 -0400
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <94a1afde0606151121j3206ffc7k1a7a0f1217c1fa50@mail.gmail.com>
References: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
	<001501c69090$6776d810$4336a8c0@Traveler>
	<94a1afde0606150837u3e3ffeaagd36b460afdca9333@mail.gmail.com>
	<32B6A5FB-933C-4D73-BF18-70F7387EF229@cisco.com>
	<94a1afde0606151102g3d74c5b3td00661ccb900fb10@mail.gmail.com>
	<06BF1548-7DC9-4201-9EC1-5C13AFB8A0A7@cisco.com>
	<94a1afde0606151121j3206ffc7k1a7a0f1217c1fa50@mail.gmail.com>
Message-ID: <94a1afde0606160727t152e66ccwc9bc0b1cfc33cf2d@mail.gmail.com>

sh debug shows this, I dont see the message about "condition 1" on the
router that will properly displaying debug messages.

The following ISDN debugs are enabled on all DSLs:

debug isdn error is             ON.
debug isdn q931 is              ON.   (filter is ON)



Condition 1: interface Se0/0/0:23 (1 flags triggered)
        Flags: Se0/0/0:23



On 6/15/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> debug MGCP packet appears to work fine.
>
>
> On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote:
> >
> > nfas is the protocol that lets you bind multiple PRIs to the same D-
> > chan.  Definitely not supported with MGCP though.
> >
> > How about if you turn on 'deb mgcp packet' along with your q931 debug?
> >
> > -Ryan
> >
> > On Jun 15, 2006, at 2:02 PM, Ed Leatherman wrote:
> >
> > They are definately both PRI
> >
> > Isnt there some configuration that allows multiple PRI's to share a D-
> > channel? I specifically told the telco we didnt want this, and i'd be
> > surprised if it would work without any extra configuration on my
> > side, but thats the only other thing I could think of.
> >
> > #sh isdn status
> > Global ISDN Switchtype = primary-4ess
> >
> > %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 1. Layer 3 output
> > may not apply
> >
> > ISDN Serial0/0/0:23 interface
> >          dsl 1, interface ISDN Switchtype = primary-ni
> >          L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
> >      Layer 1 Status:
> >          ACTIVE
> >      Layer 2 Status:
> >          TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
> >      Layer 3 Status:
> >          0 Active Layer 3 Call(s)
> >      Active dsl 1 CCBs = 0
> >      The Free Channel Mask:  0x807FFFFF
> >      Number of L2 Discards = 0, L2 Session ID = 203
> >      Total Allocated ISDN CCBs = 0
> >
> > On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote: Are both of them
> > PRIs for sure?  Perhaps the one not showing q931
> > debugs is actually a channel-group (CAS) and not a PRI?
> >
> > -Ryan
> >
> > On Jun 15, 2006, at 11:37 AM, Ed Leatherman wrote:
> >
> > Just had the defaults on both routers, however I tried the "logging
> > console 7" on the problem router, which looks like it should display
> > debuging messages and it still shows nothing.
> >
> > Just for grins and giggles I turned on debug cdp packets, and i am
> > getting debug messages on both routers.. so its something with the
> > ISDN that is causing it to not display.
> >
> > On 6/15/06, Mark Borchers <mborchers at igillc.com> wrote:
> > Have you set your logging level high enough to show debug traffic?
> > See the logging console command args.
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip-
> > bounces at puck.nether.net ] On Behalf Of Ed Leatherman
> > Sent: Thursday, June 15, 2006 10:26 AM
> > To: ciscovoip
> > Cc: jeff.davis at mail.wvu.edu
> > Subject: [cisco-voip] debug isdn q931
> >
> > I'm fairly certain this is something I am doing wrong but I cant find
> > it for the life of me..
> >
> > I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
> > conferencing. identical firmware loads and configurations, other than
> > 1 has an FXO card that the other doesn't.
> >
> > I'm executing on both routers from seperate SSH sessions:
> > # term mon
> > # debug isdn q931
> >
> > On router A, I can see all the Q931 debug messages go by. On router
> > B, I see nothing on the SSH session however I know calls are being
> > placed on it ( I can see the channels going in service etc from
> > RTMT). Could I have something turned off on router B that doesnt show
> > up in the config?
> >
> > --
> > Ed Leatherman
> > IP Telephony Coordinator
> > West Virginia University
> > Telecommunications and Network Operations
> >
> >
> >
> > --
> > Ed Leatherman
> > IP Telephony Coordinator
> > West Virginia University
> > Telecommunications and Network Operations
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >
> > --
> > Ed Leatherman
> > IP Telephony Coordinator
> > West Virginia University
> > Telecommunications and Network Operations
> >
> >
>
>
> --
>
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From lelio at uoguelph.ca  Fri Jun 16 10:46:29 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Fri, 16 Jun 2006 10:46:29 -0400
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
References: <963CE94787005149A52C17F33A028E12051D488B@EVS1B.boco.co.boulder.co.us>
	<a40fb8757879aadd50e31058a01e3111@cisco.com>
Message-ID: <02b201c69153$a78e9410$c901a8c0@EATIUSBIRDIUS>

I think VMWare (especially the ESX verison) has definate advantages, even in a production environment. As far as I know the underlying OS is as transparent as it can get to the client O/S and application. There are many educational institutions which are experimenting with this sort of datacentre. I'm not sure the single point of failure argument stands either, from what I gather, setting up two ESX servers with shared data stores is quite fault tolerant. Then again, I'm sure we'd all get at least two VMware servers, seperating our pubs and subs equally.

I think Cisco should at least investigate a version of CallManager, et al, for the lab so that we can effectively and effeciently build our labs to properly support our deployments. There are many institutions which lease their servers so when they replace them they don't have anything to migrate to the lab.

Then again, I've been waiting a long time for CSCdr03425 to be fixed, so I doubt VMware support is going to be done anytime soon. ;)



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

            CSCdr03425 Bug Details 
             
     
            Headline  AVVID: Allow additional line appearances to be forwarded from phone 
            Product  ciscocm 
            Feature  OTHERS  Duplicate of   
            Severity  6  Severity help Status  New  Status help 
            First Found-in Version  2.4(2b) First Fixed-in Version    Version help 
     

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

  ----- Original Message ----- 
  From: Kevin Thorngren 
  To: Leetun, Rob 
  Cc: cisco-voip at puck.nether.net 
  Sent: Friday, June 16, 2006 9:14 AM
  Subject: Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??


  I am not privy to any discussions in Cisco regarding vmware but I can give you my opinion on why vmware is not supported today. I am also not aware of any discussions for future support and would guess that it is not on the roadmap. Even if we were to open up support just for demo/lab purposes we would end up with customers deploying in a production environment and expecting support when things don't work as expected.

  For us to support vmware we would need to verify the operation and performance of each version of CallManager (possibly including Service Releases and Engineering Specials) on vmware. This would, at a minimum, double our testing effort which would cause major delays in getting product and updates out for our customers. In addition we would need to create design guidelines for using vmware which would have an impact on product release timeframes. 

  From a TAC support perspective; vmware would increase the complexity of troubleshooting operational and performance issues. Currently we support the CallManager OS. Adding vmware to the mix would greatly increase the complexity of now having to look at vmware and the host server's OS as potential issues.

  Since we don't test vmware we can't provide nor guarantee any sort of performance levels for running CallManager on vmware. I would highly recommend NOT using vmware in a production environment.

  Again this is just my opinion and not official word from Cisco :-)

  Kevin

  On Jun 16, 2006, at 8:41 AM, Leetun, Rob wrote:


    Thanks for the response Kevin.  Other than for the obvious reason I stated earlier.  Why is everyone on this site installing 4.1.3 and etc. on Vmware if it is not supported?  When will Cisco support Vmware?
     
    Rob

    From: Kevin Thorngren [mailto:kthorngr at cisco.com]
    Sent: Friday, June 16, 2006 6:37 AM
    To: Leetun, Rob
    Cc: cisco-voip at puck.nether.net
    Subject: Re: [cisco-voip] Vmware Supported by Cisco...Thoughts??

    Cisco does not support VMWare for CallManager, etc. The only Cisco supported installations are the ones described in the docs on CCO.

    Kevin
    On Jun 16, 2006, at 8:28 AM, Leetun, Rob wrote:


      Let's take this communication in another direction?  This is very interesting to read all the comments and questions about Vmware.  When we purchased our environment from Cisco.  We had problems with TAC support after the install because our environment was not standard in their opinion.  I have two questions: 

      1. Does Cisco support the Vmware environment?  

      2. And if so, what are the parameters for support and guidance? 

      3. And, what products do they support, e.g. Windows? 

      This could save us $$ when it comes to replacing servers! 

      Thanks. 

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


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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From kthorngr at cisco.com  Fri Jun 16 11:08:17 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Fri, 16 Jun 2006 11:08:17 -0400
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <94a1afde0606160727t152e66ccwc9bc0b1cfc33cf2d@mail.gmail.com>
References: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
	<001501c69090$6776d810$4336a8c0@Traveler>
	<94a1afde0606150837u3e3ffeaagd36b460afdca9333@mail.gmail.com>
	<32B6A5FB-933C-4D73-BF18-70F7387EF229@cisco.com>
	<94a1afde0606151102g3d74c5b3td00661ccb900fb10@mail.gmail.com>
	<06BF1548-7DC9-4201-9EC1-5C13AFB8A0A7@cisco.com>
	<94a1afde0606151121j3206ffc7k1a7a0f1217c1fa50@mail.gmail.com>
	<94a1afde0606160727t152e66ccwc9bc0b1cfc33cf2d@mail.gmail.com>
Message-ID: <1fd21c6de067a35f3d3324ef628ede60@cisco.com>

Interesting, I just tried this in the lab.  Per the documentation you  
would expect to see debug traffic for only the specified interface(s)  
in the condition.  But I see that having a condition on the serial  
interface seems to filter out the isdn debugs for all interfaces.  Not  
sure if this is a bug or if there are good reasons why this is not  
working.  Maybe other conditions need to apply for the filter to work.

You can type "no debug condition 1" to remove the condition.  You will  
see a warning message if this is the last condition you have.   You can  
type yes to the answer to remove the last condition.

Here is the docs describing this:
http://www.cisco.com/en/US/products/sw/iosswrel/ps5187/ 
products_command_reference_chapter09186a008017cfa1.html

Kevin
On Jun 16, 2006, at 10:27 AM, Ed Leatherman wrote:

> sh debug shows this, I dont see the message about "condition 1" on the  
> router that will properly displaying debug messages.
>
> The following ISDN debugs are enabled on all DSLs:
>
> debug isdn error is???????????? ON.
> debug isdn q931 is????????????? ON.?? (filter is ON)
>
>
>
> Condition 1: interface Se0/0/0:23 (1 flags triggered)
> ??????? Flags: Se0/0/0:23
>
>
>
> On 6/15/06, Ed Leatherman <ealeatherman at gmail.com> wrote:debug MGCP  
> packet appears to work fine.
>>
>>
>> On 6/15/06, Ryan Ratliff < rratliff at cisco.com> wrote:
>>> chan.??Definitely not supported with MGCP though.
>>>
>>> How about if you turn on 'deb mgcp packet' along with your q931  
>>> debug?
>>>
>>> -Ryan
>>>
>>>  On Jun 15, 2006, at 2:02 PM, Ed Leatherman wrote:
>>>
>>> They are definately both PRI
>>>
>>> Isnt there some configuration that allows multiple PRI's to share a  
>>> D-
>>> channel? I specifically told the telco we didnt want this, and i'd be
>>> surprised if it would work without any extra configuration on my
>>> side, but thats the only other thing I could think of.
>>>
>>> #sh isdn status
>>> Global ISDN Switchtype = primary-4ess
>>>
>>> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 1. Layer 3 output
>>> may not apply
>>>
>>>  ISDN Serial0/0/0:23 interface
>>> ???????? dsl 1, interface ISDN Switchtype = primary-ni
>>> ???????? L2 Protocol = Q.921 0x0000??L3 Protocol(s) = CCM MANAGER  
>>> 0x0003
>>> ???? Layer 1 Status:
>>> ???????? ACTIVE
>>> ???? Layer 2 Status:
>>> ???????? TEI = 0, Ces = 1, SAPI = 0, State =  
>>> MULTIPLE_FRAME_ESTABLISHED
>>> ???? Layer 3 Status:
>>> ???????? 0 Active Layer 3 Call(s)
>>> ???? Active dsl 1 CCBs = 0
>>> ???? The Free Channel Mask:??0x807FFFFF
>>> ???? Number of L2 Discards = 0, L2 Session ID = 203
>>> ???? Total Allocated ISDN CCBs = 0
>>>
>>> On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote: Are both of them
>>> PRIs for sure???Perhaps the one not showing q931
>>> debugs is actually a channel-group (CAS) and not a PRI?
>>>
>>> -Ryan
>>>
>>> On Jun 15, 2006, at 11:37 AM, Ed Leatherman wrote:
>>>
>>> Just had the defaults on both routers, however I tried the "logging
>>> console 7" on the problem router, which looks like it should display
>>> debuging messages and it still shows nothing.
>>>
>>> Just for grins and giggles I turned on debug cdp packets, and i am
>>> getting debug messages on both routers.. so its something with the
>>> ISDN that is causing it to not display.
>>>
>>> On 6/15/06, Mark Borchers <mborchers at igillc.com> wrote:
>>> Have you set your logging level high enough to show debug traffic?
>>> See the logging console command args.
>>> -----Original Message-----
>>> From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip-
>>> bounces at puck.nether.net ] On Behalf Of Ed Leatherman
>>> Sent: Thursday, June 15, 2006 10:26 AM
>>> To: ciscovoip
>>> Cc: jeff.davis at mail.wvu.edu
>>> Subject: [cisco-voip] debug isdn q931
>>>
>>> I'm fairly certain this is something I am doing wrong but I cant find
>>> it for the life of me..
>>>
>>> I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
>>> conferencing. identical firmware loads and configurations, other than
>>> 1 has an FXO card that the other doesn't.
>>>
>>>  I'm executing on both routers from seperate SSH sessions:
>>> # term mon
>>> # debug isdn q931
>>>
>>> On router A, I can see all the Q931 debug messages go by. On router
>>> B, I see nothing on the SSH session however I know calls are being
>>> placed on it ( I can see the channels going in service etc from
>>> RTMT). Could I have something turned off on router B that doesnt show
>>> up in the config?
>>>
>>> --
>>> Ed Leatherman
>>> IP Telephony Coordinator
>>>  West Virginia University
>>> Telecommunications and Network Operations
>>>
>>>
>>>
>>> --
>>> Ed Leatherman
>>> IP Telephony Coordinator
>>> West Virginia University
>>> Telecommunications and Network Operations
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>>  https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>>
>>>
>>> --
>>> Ed Leatherman
>>> IP Telephony Coordinator
>>> West Virginia University
>>> Telecommunications and Network Operations
>>>
>>
>>
>>
>> --
>>
>> Ed Leatherman
>> IP Telephony Coordinator
>> West Virginia University
>> Telecommunications and Network Operations
>
>
> -- 
> Ed Leatherman
> IP Telephony Coordinator
> 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 tednugent69 at yahoo.com  Fri Jun 16 11:49:01 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 16 Jun 2006 08:49:01 -0700 (PDT)
Subject: [cisco-voip] Vmware Supported by Cisco...Thoughts??
In-Reply-To: <02b201c69153$a78e9410$c901a8c0@EATIUSBIRDIUS>
Message-ID: <20060616154901.14353.qmail@web36110.mail.mud.yahoo.com>

My lab is and has been running on VMWare for almost 2
years now. I have most versions of CM/IPCC and Unity
setup as VM in my environment which makes customer
recreates a breeze! If I need a particular version to
recreate a scenario I simply spark up that VM, if I
need to test a patch or upgrade scenario I roll it out
on VMs before going to the customer site and if it
bombs I roll back the changes and start all over. Also
most of my VM are stored on an external USB HD so
they're completely portable and I can move them from
my home lab into the office lab without a hitch.
It works flawlessly in the lab providing you have
enough RAM and does in a production environment as
well (I know of at least 2 customers running it now in
a small environment under 500 phones), however we do
not support them nor would Cisco if they knew... TAC
has a hard time keeping up with all they have on their
plates now and support another 3rd party app like
VMWare would be a nightmare, it's not for your average
CM customer. Furthermore, according to some
conversations that I've had with DE's in the BU they
have no intentions of ever supporting Cisco IPT on
VMWare so you?re on your own. 
I do however see an opportunity for CM5.x to be
supported on VMWare since it will be appliance based
meaning you?re locked out of the Linux OS and this
would significantly reduce an sort of VM support
issues that might arise like drivers and disk space
issue but course you still have the fault tolerance
and other design consideration to deal with. I
planning on bringing CM5 into my VMWare lab the minute
I get my hands on a copy. It obviously needs to be
initially installed on an MCS server since, as I
understand it, it?s a single DVD that loads the OS and
CM at the sametime and does a hardware check during
install, however once it?s loaded I can just ghost off
a copy and use P2V to pull it into VM. The hurdle will
be the licensing since it?s tied to a MAC address and
I?m familiar with the CM5 eval licensing yet and what
it allows.


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

From ealeatherman at gmail.com  Fri Jun 16 13:01:49 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 16 Jun 2006 13:01:49 -0400
Subject: [cisco-voip] debug isdn q931
In-Reply-To: <1fd21c6de067a35f3d3324ef628ede60@cisco.com>
References: <94a1afde0606150825i6b719371n2d459f0307501532@mail.gmail.com>
	<001501c69090$6776d810$4336a8c0@Traveler>
	<94a1afde0606150837u3e3ffeaagd36b460afdca9333@mail.gmail.com>
	<32B6A5FB-933C-4D73-BF18-70F7387EF229@cisco.com>
	<94a1afde0606151102g3d74c5b3td00661ccb900fb10@mail.gmail.com>
	<06BF1548-7DC9-4201-9EC1-5C13AFB8A0A7@cisco.com>
	<94a1afde0606151121j3206ffc7k1a7a0f1217c1fa50@mail.gmail.com>
	<94a1afde0606160727t152e66ccwc9bc0b1cfc33cf2d@mail.gmail.com>
	<1fd21c6de067a35f3d3324ef628ede60@cisco.com>
Message-ID: <94a1afde0606161001y10e4e965id87a8c136028b607@mail.gmail.com>

Kevin,

Removing the condition cleared that up, i'm now seeing q931 debugs on that
router. Thanks a bunch! I'll read through that document to try to understand
what that was for and how it got set (not something I did knowingly)

Have a good weekend
Ed

On 6/16/06, Kevin Thorngren <kthorngr at cisco.com> wrote:
>
> Interesting, I just tried this in the lab.  Per the documentation you
> would expect to see debug traffic for only the specified interface(s)
> in the condition.  But I see that having a condition on the serial
> interface seems to filter out the isdn debugs for all interfaces.  Not
> sure if this is a bug or if there are good reasons why this is not
> working.  Maybe other conditions need to apply for the filter to work.
>
> You can type "no debug condition 1" to remove the condition.  You will
> see a warning message if this is the last condition you have.   You can
> type yes to the answer to remove the last condition.
>
> Here is the docs describing this:
> http://www.cisco.com/en/US/products/sw/iosswrel/ps5187/
> products_command_reference_chapter09186a008017cfa1.html
>
> Kevin
> On Jun 16, 2006, at 10:27 AM, Ed Leatherman wrote:
>
> > sh debug shows this, I dont see the message about "condition 1" on the
> > router that will properly displaying debug messages.
> >
> > The following ISDN debugs are enabled on all DSLs:
> >
> > debug isdn error is ON.
> > debug isdn q931 is ON. (filter is ON)
> >
> >
> >
> > Condition 1: interface Se0/0/0:23 (1 flags triggered)
> > Flags: Se0/0/0:23
> >
> >
> >
> > On 6/15/06, Ed Leatherman <ealeatherman at gmail.com> wrote:debug MGCP
> > packet appears to work fine.
> >>
> >>
> >> On 6/15/06, Ryan Ratliff < rratliff at cisco.com> wrote:
> >>> chan.Definitely not supported with MGCP though.
> >>>
> >>> How about if you turn on 'deb mgcp packet' along with your q931
> >>> debug?
> >>>
> >>> -Ryan
> >>>
> >>>  On Jun 15, 2006, at 2:02 PM, Ed Leatherman wrote:
> >>>
> >>> They are definately both PRI
> >>>
> >>> Isnt there some configuration that allows multiple PRI's to share a
> >>> D-
> >>> channel? I specifically told the telco we didnt want this, and i'd be
> >>> surprised if it would work without any extra configuration on my
> >>> side, but thats the only other thing I could think of.
> >>>
> >>> #sh isdn status
> >>> Global ISDN Switchtype = primary-4ess
> >>>
> >>> %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 1. Layer 3 output
> >>> may not apply
> >>>
> >>>  ISDN Serial0/0/0:23 interface
> >>> dsl 1, interface ISDN Switchtype = primary-ni
> >>> L2 Protocol = Q.921 0x0000L3 Protocol(s) = CCM MANAGER
> >>> 0x0003
> >>> Layer 1 Status:
> >>> ACTIVE
> >>> Layer 2 Status:
> >>> TEI = 0, Ces = 1, SAPI = 0, State =
> >>> MULTIPLE_FRAME_ESTABLISHED
> >>> Layer 3 Status:
> >>> 0 Active Layer 3 Call(s)
> >>> Active dsl 1 CCBs = 0
> >>> The Free Channel Mask:0x807FFFFF
> >>> Number of L2 Discards = 0, L2 Session ID = 203
> >>> Total Allocated ISDN CCBs = 0
> >>>
> >>> On 6/15/06, Ryan Ratliff <rratliff at cisco.com> wrote: Are both of them
> >>> PRIs for sure?Perhaps the one not showing q931
> >>> debugs is actually a channel-group (CAS) and not a PRI?
> >>>
> >>> -Ryan
> >>>
> >>> On Jun 15, 2006, at 11:37 AM, Ed Leatherman wrote:
> >>>
> >>> Just had the defaults on both routers, however I tried the "logging
> >>> console 7" on the problem router, which looks like it should display
> >>> debuging messages and it still shows nothing.
> >>>
> >>> Just for grins and giggles I turned on debug cdp packets, and i am
> >>> getting debug messages on both routers.. so its something with the
> >>> ISDN that is causing it to not display.
> >>>
> >>> On 6/15/06, Mark Borchers <mborchers at igillc.com> wrote:
> >>> Have you set your logging level high enough to show debug traffic?
> >>> See the logging console command args.
> >>> -----Original Message-----
> >>> From: cisco-voip-bounces at puck.nether.net [mailto: cisco-voip-
> >>> bounces at puck.nether.net ] On Behalf Of Ed Leatherman
> >>> Sent: Thursday, June 15, 2006 10:26 AM
> >>> To: ciscovoip
> >>> Cc: jeff.davis at mail.wvu.edu
> >>> Subject: [cisco-voip] debug isdn q931
> >>>
> >>> I'm fairly certain this is something I am doing wrong but I cant find
> >>> it for the life of me..
> >>>
> >>> I have 2 ISR routers, 3825 with some T1 interfaces and DSPs for
> >>> conferencing. identical firmware loads and configurations, other than
> >>> 1 has an FXO card that the other doesn't.
> >>>
> >>>  I'm executing on both routers from seperate SSH sessions:
> >>> # term mon
> >>> # debug isdn q931
> >>>
> >>> On router A, I can see all the Q931 debug messages go by. On router
> >>> B, I see nothing on the SSH session however I know calls are being
> >>> placed on it ( I can see the channels going in service etc from
> >>> RTMT). Could I have something turned off on router B that doesnt show
> >>> up in the config?
> >>>
> >>> --
> >>> Ed Leatherman
> >>> IP Telephony Coordinator
> >>>  West Virginia University
> >>> Telecommunications and Network Operations
> >>>
> >>>
> >>>
> >>> --
> >>> Ed Leatherman
> >>> IP Telephony Coordinator
> >>> West Virginia University
> >>> Telecommunications and Network Operations
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip at puck.nether.net
> >>>  https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>
> >>>
> >>>
> >>>
> >>> --
> >>> Ed Leatherman
> >>> IP Telephony Coordinator
> >>> West Virginia University
> >>> Telecommunications and Network Operations
> >>>
> >>
> >>
> >>
> >> --
> >>
> >> Ed Leatherman
> >> IP Telephony Coordinator
> >> West Virginia University
> >> Telecommunications and Network Operations
> >
> >
> > --
> > Ed Leatherman
> > IP Telephony Coordinator
> > West Virginia University
> > Telecommunications and Network
> > Operations_______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From voiprsvp at yahoo.com  Fri Jun 16 09:27:05 2006
From: voiprsvp at yahoo.com (Jay Oq)
Date: Fri, 16 Jun 2006 06:27:05 -0700 (PDT)
Subject: [cisco-voip] QoS 101 questions
Message-ID: <20060616132705.20887.qmail@web55706.mail.re3.yahoo.com>

Greetings all, I'm definitely no QoS expert and my Cisco experience is a little above newbie (4 years exp) and I now have a dubious task of VoIP QoS'ing. Anyhow currently I have this interface on my 3800:

interface Serial1/0
description Frame Relay Circuit
no ip address
service-policy Voice
enscapsulation frame-relay IETF
no ip mroute-cache
dsu bandwidth 44210
frame-relay lmi-type ansi

Would it make sense to do the following on that interface for QoS? I'm trying to set up QoS for a DS3 that is split. 20mb is hard linked, 25mb is what I'm pushing to the Internet but it's burstable to a full DS3. I'd like to allocate 75% of that 25mb for strictly voice... The rest for whatever else.

class-map X
match input-interface LAN
class-map class-default
match any 
class-map voice
match access-group 101

policy-map VOICEDATA
class voice
priority 19200
Precedence = 5
class data
bandwidth 6400

interface Serial1/0
bandwidth DS3
no ip directed-broadcast
service-policy output VOICEDATA
!
access-list 101 permit ip any any precedence critical

Does that make sense or is there an easier method to do this? Any help is greatly appreciated

 		
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From voiprsvp at yahoo.com  Fri Jun 16 11:18:56 2006
From: voiprsvp at yahoo.com (Jay Oq)
Date: Fri, 16 Jun 2006 08:18:56 -0700 (PDT)
Subject: [cisco-voip] QoS question
Message-ID: <20060616151857.85659.qmail@web55703.mail.re3.yahoo.com>

Greetings all, I'm definitely no QoS expert and my Cisco experience is a little above newbie (4 years exp) and I now have a dubious task of VoIP QoS'ing. Anyhow currently I have

interface Serial1/0
description Frame Relay Circuit
no ip address
service-policy Voice
enscapsulation frame-relay IETF
no ip mroute-cache
dsu bandwidth 44210
frame-relay lmi-type ansi

Would it make sense to do the following on that interface for QoS? I'm trying to set up QoS for a DS3 that is split. 20mb is hard linked, 25mb is what I'm pushing to the Internet but it's burstable to a full DS3. I'd like to allocate 75% of that 25mb for strictly voice... The rest for whatever else.

policy-map parent
shape average 18432000
service-policy voippolicy
!
int serial1/0
service-policy output parent


class-map X
match input-interface LAN
class-map class-default
match any
class-map voice
match access-group 101

policy-map VOICEDATA
class voice
priority 19200
Precedence = 5
class data
bandwidth 6400

interface Serial1/0
bandwidth DS3
no ip directed-broadcast
service-policy output VOICEDATA
!
access-list 101 permit ip any any precedence critical

Does that make sense or is there an easier method to do this? Any help is greatly appreciated
 		
---------------------------------
Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls.  Great rates starting at 1?/min.
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From jgrace at digitelusa.net  Fri Jun 16 14:02:56 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri, 16 Jun 2006 14:02:56 -0400
Subject: [cisco-voip] what is the best cordless head set for cisco 7940-60
Message-ID: <200606161415359.SM01464@3729434770>

what is the best cordless head set for cisco 7940-60

we are looking to deploy sets for our callcenter 20 ip phones.  I just want
to get an ideal of what to look for.  Plantronic is what im hearing, but
what would be the best model

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From don_miguel at chile.com  Fri Jun 16 14:19:50 2006
From: don_miguel at chile.com (miguel)
Date: Fri, 16 Jun 2006 14:19:50 -0400
Subject: [cisco-voip] RTP Call simulation
Message-ID: <4492F646.8000300@chile.com>

Hello guys!
    is it posible to test the RTP connectivity between two routers? I 
mean, one router send a sound or something with RTP to another?

Many thanks
Miguel


From ealeatherman at gmail.com  Fri Jun 16 14:32:23 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Fri, 16 Jun 2006 14:32:23 -0400
Subject: [cisco-voip] RTP Call simulation
In-Reply-To: <4492F646.8000300@chile.com>
References: <4492F646.8000300@chile.com>
Message-ID: <94a1afde0606161132t7b6c7ceeg4d6c54d035fc142c@mail.gmail.com>

Miguel,

You might want to look into IP SLA on the IOS.. I havent used it myself so
ymmv but it might be what you are looking for:

http://www.cisco.com/en/US/products/ps6602/products_qanda_item0900aecd8017bd5a.shtml
http://www.cisco.com/en/US/products/ps6441/products_feature_guide09186a0080556c4d.html

Hope that helps

Ed


On 6/16/06, miguel <don_miguel at chile.com> wrote:
>
> Hello guys!
>     is it posible to test the RTP connectivity between two routers? I
> mean, one router send a sound or something with RTP to another?
>
> Many thanks
> Miguel
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From JOlson at mlnusa.com  Fri Jun 16 17:04:00 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Fri, 16 Jun 2006 17:04:00 -0400
Subject: [cisco-voip] IPCC partial service
Message-ID: <FE2D092F8B1516488B99342551AACFD41CEE8E07@mlnusa-exch1.mlnusa.com>

I am running IPCC 3.5(3) and when I view the engine status under system
the JTAPI Subsystem is running in "PARTIAL_SERVICE"  

I pulled the following 3 lines out of the trace files that reference a
partial status.

Any ideas what could be causing this and how it could be corrected?

 

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=A number of CTI ports are OOS -
TPG[id=0,state=PARTIAL_SERVICE]:Ports[85023]

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=Some group can only be initialized partially -
TPG[id=0,state=IN_SERVICE]:Ports[85009]

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=Some group can only be initialized partially -
TPG[id=0,state=IN_SERVICE]:Ports[85041]

 

Thanks

Jeff


********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

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From Matt.Slaga at us.didata.com  Fri Jun 16 17:24:58 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 16 Jun 2006 17:24:58 -0400
Subject: [cisco-voip] IPCC partial service
Message-ID: <C1FE15183DA37645BC0633BC604E44F00235061D@USNAEXCH.na.didata.local>

Partial service is generally caused by either the JTAPI user not being
associated with all CTI ports or one of the CTI ports has a duplicate
directory number somewhere.
 
I would check the route plan report for the numbers listed below, 85023,
85009 and 85041.  Also check that the jtapi user is associated with
those ports.
 
Thanks,
Matt

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Olson, Jeffrey
Sent: Friday, June 16, 2006 5:04 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC partial service



I am running IPCC 3.5(3) and when I view the engine status under system
the JTAPI Subsystem is running in "PARTIAL_SERVICE"  

I pulled the following 3 lines out of the trace files that reference a
partial status.

Any ideas what could be causing this and how it could be corrected?

 

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=A number of CTI ports are OOS -
TPG[id=0,state=PARTIAL_SERVICE]:Ports[85023]

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=Some group can only be initialized partially -
TPG[id=0,state=IN_SERVICE]:Ports[85009]

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=Some group can only be initialized partially -
TPG[id=0,state=IN_SERVICE]:Ports[85041]

 

Thanks

Jeff

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders
Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
hereby notified that disclosing, copying, distributing, or taking any
action in reliance on the contents of this information is strictly
prohibited, and that the E-mail should be returned to this company
immediately. If you have received this E-mail in error, please contact
the sender immediately.

 




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From jwydra at Burwood.com  Fri Jun 16 17:37:02 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Fri, 16 Jun 2006 16:37:02 -0500
Subject: [cisco-voip] IPCC partial service
Message-ID: <142801c6918d$07a4a6d0$280111ac@Burwood.com>

Ditto

-----Original Message-----
From: "Matt Slaga (US)"<Matt.Slaga at us.didata.com>
Sent: 6/16/2006 4:26:49 PM
To: "Olson, Jeffrey"<JOlson at mlnusa.com>, "cisco-voip at puck.nether.net"<cisco-voip at puck.nether.net>
Cc: 
Subject: Re: [cisco-voip] IPCC partial service

Partial service is generally caused by either the JTAPI user not being
associated with all CTI ports or one of the CTI ports has a duplicate
directory number somewhere.
 
I would check the route plan report for the numbers listed below, 85023,
85009 and 85041.  Also check that the jtapi user is associated with
those ports.
 
Thanks,
Matt

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Olson, Jeffrey
Sent: Friday, June 16, 2006 5:04 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCC partial service



I am running IPCC 3.5(3) and when I view the engine status under system
the JTAPI Subsystem is running in "PARTIAL_SERVICE"  

I pulled the following 3 lines out of the trace files that reference a
partial status.

Any ideas what could be causing this and how it could be corrected?

 

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=A number of CTI ports are OOS -
TPG[id=0,state=PARTIAL_SERVICE]:Ports[85023]

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=Some group can only be initialized partially -
TPG[id=0,state=IN_SERVICE]:Ports[85009]

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=Some group can only be initialized partially -
TPG[id=0,state=IN_SERVICE]:Ports[85041]

 

Thanks

Jeff

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders
Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
hereby notified that disclosing, copying, distributing, or taking any
action in reliance on the contents of this information is strictly
prohibited, and that the E-mail should be returned to this company
immediately. If you have received this E-mail in error, please contact
the sender immediately.

 




-----------------------------------------
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 jwydra at Burwood.com  Fri Jun 16 17:37:04 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Fri, 16 Jun 2006 16:37:04 -0500
Subject: [cisco-voip] IPCC partial service
Message-ID: <142901c6918d$08fe5b39$280111ac@Burwood.com>

Partial service means that at least one of your cti ports in ccm are not registered. First place to look is ccm.

-----Original Message-----
From: "Olson, Jeffrey"<JOlson at mlnusa.com>
Sent: 6/16/2006 4:08:04 PM
To: "cisco-voip at puck.nether.net"<cisco-voip at puck.nether.net>
Cc: 
Subject: [cisco-voip] IPCC partial service

I am running IPCC 3.5(3) and when I view the engine status under system
the JTAPI Subsystem is running in "PARTIAL_SERVICE"  

I pulled the following 3 lines out of the trace files that reference a
partial status.

Any ideas what could be causing this and how it could be corrected?

 

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=A number of CTI ports are OOS -
TPG[id=0,state=PARTIAL_SERVICE]:Ports[85023]

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=Some group can only be initialized partially -
TPG[id=0,state=IN_SERVICE]:Ports[85009]

%MIVR-SS_TEL-2-SS_PARTIAL_SERVICE:JTAPI subsystem in partial service:
Failure reason=Some group can only be initialized partially -
TPG[id=0,state=IN_SERVICE]:Ports[85041]

 

Thanks

Jeff


********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.



From smunzani at comcast.net  Fri Jun 16 18:09:24 2006
From: smunzani at comcast.net (Sam Munzani)
Date: Fri, 16 Jun 2006 17:09:24 -0500
Subject: [cisco-voip] RTP Call simulation
In-Reply-To: <4492F646.8000300@chile.com>
References: <4492F646.8000300@chile.com>
Message-ID: <44932C14.4050909@comcast.net>

What's goal? If you need complete call simulations for VoIP testing 
WinSIP or Win323 softwares work great. They are commercial softwares but 
you can simulate 1000+ voice calls between 2 end points.

http://www.touchstone-inc.com/

Thanks,
sam
> Hello guys!
>     is it posible to test the RTP connectivity between two routers? I 
> mean, one router send a sound or something with RTP to another?
>
> Many thanks
> Miguel
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>   


From chris at chrisserafin.com  Fri Jun 16 20:22:10 2006
From: chris at chrisserafin.com (ChrisSerafin)
Date: Fri, 16 Jun 2006 19:22:10 -0500
Subject: [cisco-voip] 3560 port shut down
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605CEE9A2@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A605CEE9A2@Apollo.wesd.org>
Message-ID: <44934B32.9050605@chrisserafin.com>

Could you post a sanitized config?

Chris Serafin
IT Security / Cisco VoIP Engineer
chris at chrisserafin.com
Cell - 708-307-2413
www.chrisserafin.com - Cisco Security Clearinghouse - under construction


Voll, Scott wrote:
>
> I have 3560?s running 12.2(25)SEE1 IOS and when you plug in a device, 
> then plug in another device, then go back to the first device or a 
> third device the port does not work.
>
> Is this a security feature and if so, how do you turn it off?
>
> Thanks
>
> Scott
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>   


From vnthomas2 at gmail.com  Sat Jun 17 08:20:29 2006
From: vnthomas2 at gmail.com (Varghese Thomas)
Date: Sat, 17 Jun 2006 08:20:29 -0400
Subject: [cisco-voip] Opportunity for VoIP technical specialist
Message-ID: <375936670606170520p5c15ab2cu581bad712b509c4e@mail.gmail.com>

One of the major system integrator is looking for Cisco VoIP technical
specialists. If interested, please send your resume to me. Location, NYC.
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From jrybak at annese.com  Sat Jun 17 16:10:46 2006
From: jrybak at annese.com (Jake Rybak)
Date: Sat, 17 Jun 2006 16:10:46 -0400
Subject: [cisco-voip] sccp ip precedence
Message-ID: <000001c6924a$309e1720$91da4560$@com>

I have a question regarding the command: sccp ip precedence value

 

 

The documentation states "value = IP precedence value. Range is from 1
(highest) to 7 (lowest)." The following example sets IP precedence to the
highest possible value: 

Router# sccp ip precedence 1

 

This is obviously a typo, no?

 

The default value makes it appear as though it would refer to media streams.


 

This docs suggests it is the actual signaling.

 

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_refe
rence_chapter09186a008045733b.html#wp1245396

 

This doc suggests voice streams.

 

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_access/vg_200/config
/dsp.htm#xtocid273628

 

I have always assumed it was the voice stream because of the default value
of 5.

 

Any clarification would be appreciated.

 

Thanks,

 

jake

 

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From tednugent69 at yahoo.com  Sat Jun 17 23:48:49 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sat, 17 Jun 2006 20:48:49 -0700 (PDT)
Subject: [cisco-voip] sccp ip precedence
In-Reply-To: <000001c6924a$309e1720$91da4560$@com>
Message-ID: <20060618034849.94795.qmail@web36101.mail.mud.yahoo.com>

Yes, that?s obviously a typo, if this is your first
day with Cisco docs then you might want to get used to
it. If maybe you can give an idea of what you?re
trying to accomplish then maybe we can try and give
you some guidance. By default at least in the Cisco
world you can expect voice control to be tagged as
IPPrec of 3 (AF31 or CS3) and media (rtp) as IP Prec 5
(EF). Unless you have a reason to mess with these
values then why bother? Again without knowing what
you?re trying to accomplish then it makes it difficult
to give you any sort of advice. BTW, IP Prec is old
school, the rest of the world uses DSCP?


--- Jake Rybak <jrybak at annese.com> wrote:

> I have a question regarding the command: sccp ip
> precedence value
> 
>  
> 
>  
> 
> The documentation states "value = IP precedence
> value. Range is from 1
> (highest) to 7 (lowest)." The following example sets
> IP precedence to the
> highest possible value: 
> 
> Router# sccp ip precedence 1
> 
>  
> 
> This is obviously a typo, no?
> 
>  
> 
> The default value makes it appear as though it would
> refer to media streams.
> 
> 
>  
> 
> This docs suggests it is the actual signaling.
> 
>  
> 
>
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_refe
> rence_chapter09186a008045733b.html#wp1245396
> 
>  
> 
> This doc suggests voice streams.
> 
>  
> 
>
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_access/vg_200/config
> /dsp.htm#xtocid273628
> 
>  
> 
> I have always assumed it was the voice stream
> because of the default value
> of 5.
> 
>  
> 
> Any clarification would be appreciated.
> 
>  
> 
> Thanks,
> 
>  
> 
> jake
> 
>  
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jrybak at annese.com  Sun Jun 18 05:00:45 2006
From: jrybak at annese.com (Jake Rybak)
Date: Sun, 18 Jun 2006 05:00:45 -0400
Subject: [cisco-voip] sccp ip precedence
In-Reply-To: <20060618034849.94795.qmail@web36101.mail.mud.yahoo.com>
References: <000001c6924a$309e1720$91da4560$@com>
	<20060618034849.94795.qmail@web36101.mail.mud.yahoo.com>
Message-ID: <000301c692b5$c229c110$467d4330$@com>

Thanks. I just wanted to be 100% positive that it is changing the voice
stream and not signaling. I guess I could have just captured the packets but
I wasn't in place that it was possible.

As you know, MGCP and H323 you can choose signaling or media. 

So being registered to a CM 3.3, the CM service parameter would override
this anyway? Not that you would want to.




-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ted Nugent
Sent: Saturday, June 17, 2006 23:49
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] sccp ip precedence

Yes, that's obviously a typo, if this is your first day with Cisco docs then
you might want to get used to it. If maybe you can give an idea of what
you're trying to accomplish then maybe we can try and give you some
guidance. By default at least in the Cisco world you can expect voice
control to be tagged as IPPrec of 3 (AF31 or CS3) and media (rtp) as IP Prec
5 (EF). Unless you have a reason to mess with these values then why bother?
Again without knowing what you're trying to accomplish then it makes it
difficult to give you any sort of advice. BTW, IP Prec is old school, the
rest of the world uses DSCP?


--- Jake Rybak <jrybak at annese.com> wrote:

> I have a question regarding the command: sccp ip precedence value
> 
>  
> 
>  
> 
> The documentation states "value = IP precedence value. Range is from 1
> (highest) to 7 (lowest)." The following example sets IP precedence to 
> the highest possible value:
> 
> Router# sccp ip precedence 1
> 
>  
> 
> This is obviously a typo, no?
> 
>  
> 
> The default value makes it appear as though it would refer to media 
> streams.
> 
> 
>  
> 
> This docs suggests it is the actual signaling.
> 
>  
> 
>
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_refe
> rence_chapter09186a008045733b.html#wp1245396
> 
>  
> 
> This doc suggests voice streams.
> 
>  
> 
>
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_access/vg_200/config
> /dsp.htm#xtocid273628
> 
>  
> 
> I have always assumed it was the voice stream because of the default 
> value of 5.
> 
>  
> 
> Any clarification would be appreciated.
> 
>  
> 
> Thanks,
> 
>  
> 
> jake
> 
>  
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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


From tednugent69 at yahoo.com  Sun Jun 18 11:12:52 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sun, 18 Jun 2006 08:12:52 -0700 (PDT)
Subject: [cisco-voip] sccp ip precedence
In-Reply-To: <000301c692b5$c229c110$467d4330$@com>
Message-ID: <20060618151252.27141.qmail@web36111.mail.mud.yahoo.com>

Since SCCP is a signaling protocol I would guess that
this is certainly for signaling and not media.




--- Jake Rybak <jrybak at annese.com> wrote:

> Thanks. I just wanted to be 100% positive that it is
> changing the voice
> stream and not signaling. I guess I could have just
> captured the packets but
> I wasn't in place that it was possible.
> 
> As you know, MGCP and H323 you can choose signaling
> or media. 
> 
> So being registered to a CM 3.3, the CM service
> parameter would override
> this anyway? Not that you would want to.
> 
> 
> 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ted Nugent
> Sent: Saturday, June 17, 2006 23:49
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] sccp ip precedence
> 
> Yes, that's obviously a typo, if this is your first
> day with Cisco docs then
> you might want to get used to it. If maybe you can
> give an idea of what
> you're trying to accomplish then maybe we can try
> and give you some
> guidance. By default at least in the Cisco world you
> can expect voice
> control to be tagged as IPPrec of 3 (AF31 or CS3)
> and media (rtp) as IP Prec
> 5 (EF). Unless you have a reason to mess with these
> values then why bother?
> Again without knowing what you're trying to
> accomplish then it makes it
> difficult to give you any sort of advice. BTW, IP
> Prec is old school, the
> rest of the world uses DSCP?
> 
> 
> --- Jake Rybak <jrybak at annese.com> wrote:
> 
> > I have a question regarding the command: sccp ip
> precedence value
> > 
> >  
> > 
> >  
> > 
> > The documentation states "value = IP precedence
> value. Range is from 1
> > (highest) to 7 (lowest)." The following example
> sets IP precedence to 
> > the highest possible value:
> > 
> > Router# sccp ip precedence 1
> > 
> >  
> > 
> > This is obviously a typo, no?
> > 
> >  
> > 
> > The default value makes it appear as though it
> would refer to media 
> > streams.
> > 
> > 
> >  
> > 
> > This docs suggests it is the actual signaling.
> > 
> >  
> > 
> >
>
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_refe
> > rence_chapter09186a008045733b.html#wp1245396
> > 
> >  
> > 
> > This doc suggests voice streams.
> > 
> >  
> > 
> >
>
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_access/vg_200/config
> > /dsp.htm#xtocid273628
> > 
> >  
> > 
> > I have always assumed it was the voice stream
> because of the default 
> > value of 5.
> > 
> >  
> > 
> > Any clarification would be appreciated.
> > 
> >  
> > 
> > Thanks,
> > 
> >  
> > 
> > jake
> > 
> >  
> > 
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jrybak at annese.com  Sun Jun 18 13:02:36 2006
From: jrybak at annese.com (Jake Rybak)
Date: Sun, 18 Jun 2006 13:02:36 -0400
Subject: [cisco-voip] sccp ip precedence
In-Reply-To: <20060618151252.27141.qmail@web36111.mail.mud.yahoo.com>
References: <000301c692b5$c229c110$467d4330$@com>
	<20060618151252.27141.qmail@web36111.mail.mud.yahoo.com>
Message-ID: <000001c692f9$11f1d050$35d570f0$@com>

With a default value of 5? So what you saying is that this command should
always be changed from the default?

I do not have a dspfarm in my lab so the router will not initiate any sccp
traffic to callmanager. I cannot get a capture.

My goal is to find out if this default value should be changed so the
packets will map to cs3.

-----Original Message-----
From: Ted Nugent [mailto:tednugent69 at yahoo.com] 
Sent: Sunday, June 18, 2006 11:13
To: Jake Rybak; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] sccp ip precedence

Since SCCP is a signaling protocol I would guess that this is certainly for
signaling and not media.




--- Jake Rybak <jrybak at annese.com> wrote:

> Thanks. I just wanted to be 100% positive that it is changing the 
> voice stream and not signaling. I guess I could have just captured the 
> packets but I wasn't in place that it was possible.
> 
> As you know, MGCP and H323 you can choose signaling or media.
> 
> So being registered to a CM 3.3, the CM service parameter would 
> override this anyway? Not that you would want to.
> 
> 
> 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ted Nugent
> Sent: Saturday, June 17, 2006 23:49
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] sccp ip precedence
> 
> Yes, that's obviously a typo, if this is your first day with Cisco 
> docs then you might want to get used to it. If maybe you can give an 
> idea of what you're trying to accomplish then maybe we can try and 
> give you some guidance. By default at least in the Cisco world you can 
> expect voice control to be tagged as IPPrec of 3 (AF31 or CS3) and 
> media (rtp) as IP Prec
> 5 (EF). Unless you have a reason to mess with these values then why 
> bother?
> Again without knowing what you're trying to accomplish then it makes 
> it difficult to give you any sort of advice. BTW, IP Prec is old 
> school, the rest of the world uses DSCP?
> 
> 
> --- Jake Rybak <jrybak at annese.com> wrote:
> 
> > I have a question regarding the command: sccp ip
> precedence value
> > 
> >  
> > 
> >  
> > 
> > The documentation states "value = IP precedence
> value. Range is from 1
> > (highest) to 7 (lowest)." The following example
> sets IP precedence to
> > the highest possible value:
> > 
> > Router# sccp ip precedence 1
> > 
> >  
> > 
> > This is obviously a typo, no?
> > 
> >  
> > 
> > The default value makes it appear as though it
> would refer to media
> > streams.
> > 
> > 
> >  
> > 
> > This docs suggests it is the actual signaling.
> > 
> >  
> > 
> >
>
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_refe
> > rence_chapter09186a008045733b.html#wp1245396
> > 
> >  
> > 
> > This doc suggests voice streams.
> > 
> >  
> > 
> >
>
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_access/vg_200/config
> > /dsp.htm#xtocid273628
> > 
> >  
> > 
> > I have always assumed it was the voice stream
> because of the default
> > value of 5.
> > 
> >  
> > 
> > Any clarification would be appreciated.
> > 
> >  
> > 
> > Thanks,
> > 
> >  
> > 
> > jake
> > 
> >  
> > 
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around 
> http://mail.yahoo.com _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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


From tednugent69 at yahoo.com  Sun Jun 18 13:45:43 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sun, 18 Jun 2006 10:45:43 -0700 (PDT)
Subject: [cisco-voip] sccp ip precedence
In-Reply-To: <000001c692f9$11f1d050$35d570f0$@com>
Message-ID: <20060618174543.92046.qmail@web36105.mail.mud.yahoo.com>

I can probably lab it this week sometime and get back
to you. You're right though the default being 5 does
not make a lot of sense.


--- Jake Rybak <jrybak at annese.com> wrote:

> With a default value of 5? So what you saying is
> that this command should
> always be changed from the default?
> 
> I do not have a dspfarm in my lab so the router will
> not initiate any sccp
> traffic to callmanager. I cannot get a capture.
> 
> My goal is to find out if this default value should
> be changed so the
> packets will map to cs3.
> 
> -----Original Message-----
> From: Ted Nugent [mailto:tednugent69 at yahoo.com] 
> Sent: Sunday, June 18, 2006 11:13
> To: Jake Rybak; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] sccp ip precedence
> 
> Since SCCP is a signaling protocol I would guess
> that this is certainly for
> signaling and not media.
> 
> 
> 
> 
> --- Jake Rybak <jrybak at annese.com> wrote:
> 
> > Thanks. I just wanted to be 100% positive that it
> is changing the 
> > voice stream and not signaling. I guess I could
> have just captured the 
> > packets but I wasn't in place that it was
> possible.
> > 
> > As you know, MGCP and H323 you can choose
> signaling or media.
> > 
> > So being registered to a CM 3.3, the CM service
> parameter would 
> > override this anyway? Not that you would want to.
> > 
> > 
> > 
> > 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Ted Nugent
> > Sent: Saturday, June 17, 2006 23:49
> > To: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] sccp ip precedence
> > 
> > Yes, that's obviously a typo, if this is your
> first day with Cisco 
> > docs then you might want to get used to it. If
> maybe you can give an 
> > idea of what you're trying to accomplish then
> maybe we can try and 
> > give you some guidance. By default at least in the
> Cisco world you can 
> > expect voice control to be tagged as IPPrec of 3
> (AF31 or CS3) and 
> > media (rtp) as IP Prec
> > 5 (EF). Unless you have a reason to mess with
> these values then why 
> > bother?
> > Again without knowing what you're trying to
> accomplish then it makes 
> > it difficult to give you any sort of advice. BTW,
> IP Prec is old 
> > school, the rest of the world uses DSCP?
> > 
> > 
> > --- Jake Rybak <jrybak at annese.com> wrote:
> > 
> > > I have a question regarding the command: sccp ip
> > precedence value
> > > 
> > >  
> > > 
> > >  
> > > 
> > > The documentation states "value = IP precedence
> > value. Range is from 1
> > > (highest) to 7 (lowest)." The following example
> > sets IP precedence to
> > > the highest possible value:
> > > 
> > > Router# sccp ip precedence 1
> > > 
> > >  
> > > 
> > > This is obviously a typo, no?
> > > 
> > >  
> > > 
> > > The default value makes it appear as though it
> > would refer to media
> > > streams.
> > > 
> > > 
> > >  
> > > 
> > > This docs suggests it is the actual signaling.
> > > 
> > >  
> > > 
> > >
> >
>
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_refe
> > > rence_chapter09186a008045733b.html#wp1245396
> > > 
> > >  
> > > 
> > > This doc suggests voice streams.
> > > 
> > >  
> > > 
> > >
> >
>
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_access/vg_200/config
> > > /dsp.htm#xtocid273628
> > > 
> > >  
> > > 
> > > I have always assumed it was the voice stream
> > because of the default
> > > value of 5.
> > > 
> > >  
> > > 
> > > Any clarification would be appreciated.
> > > 
> > >  
> > > 
> > > Thanks,
> > > 
> > >  
> > > 
> > > jake
> > > 
> > >  
> > > 
> > > >
> _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > 
> > 
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> > http://mail.yahoo.com
> _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com 
> 
> 


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

From don_miguel at chile.com  Sun Jun 18 23:48:14 2006
From: don_miguel at chile.com (miguel)
Date: Sun, 18 Jun 2006 23:48:14 -0400
Subject: [cisco-voip] RTP Call simulation
In-Reply-To: <4492F646.8000300@chile.com>
References: <4492F646.8000300@chile.com>
Message-ID: <44961E7E.3030204@chile.com>

Thanks for the answer! it really help the doc.

The problem with use simulation by software, is I need full accesss to 
the LAN, and it is no useful where the PBX is connected to a GW (and of 
course, I can not modify the config).

Many thanks!

miguel escribi?:
> Hello guys!
>     is it posible to test the RTP connectivity between two routers? I 
> mean, one router send a sound or something with RTP to another?
>
> Many thanks
> Miguel
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>   


From jfaulhaber at avodaq.com  Mon Jun 19 04:28:47 2006
From: jfaulhaber at avodaq.com (Jens Faulhaber)
Date: Mon, 19 Jun 2006 10:28:47 +0200
Subject: [cisco-voip] VoiceView Express and IP Communicator
Message-ID: <F666605263F0EA40BC95239B1ADE10BC0A8356@sta-exch-1.avodaq.com>

Hi folks,

does anyone know if Cisco IP Communicator supports Cisco
VoiceViewExpress?

I could not find this particular Info.

Cheers Jens

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From james.dust at charles-stanley.co.uk  Mon Jun 19 09:19:43 2006
From: james.dust at charles-stanley.co.uk (James Dust)
Date: Mon, 19 Jun 2006 14:19:43 +0100
Subject: [cisco-voip] Enter String Message
Message-ID: <FDEEDC07A7DA6E41BCB819487E7C85AED8D1F9@CSEXCH01.charles-stanley.co.uk>

Could someone please tell me what the definition of the error message
'enter string' on a Cisco 7970 please?
 
We will from time to time get this error and the only way we can find to
resolve it is to give the phone a cold boot.
 
Many Thanks.
 
James Dust | Network & Communications
Charles Stanley Stockbrokers | 25 Luke Street London EC2A 4AR
Tel: 020 7149 6314| Fax: 020 7150 9314 | Mob: 07989 491024
E-mail: james.dust at charles-stanley.co.uk
 
***********************************************************************************************

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
e-mail, is prohibited and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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

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

Charles Stanley Sutherlands and 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 SPackett at fenwick.com  Mon Jun 19 11:13:08 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Mon, 19 Jun 2006 08:13:08 -0700
Subject: [cisco-voip] Semi-OT: Where to sell or trade legacy telco equipment?
Message-ID: <965C0255FD5828438E34F9A68ECA76E7778D4E@svcexch02.firm.fenwick.llp>

Sorry for the semi-OT post, but we're ready to pull out our legacy
voicemail system, an Octel 250 (we actually have 2 of these).  For those
of you who have done migrations before, where do you sell or trade your
legacy telco equipment?  We're thinking about putting a package deal
where we just want to get rid of two Octel 250 units and two Cisco
VG248s since we're never going to use them again.

Thanks in advance.
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From Scott.Voll at wesd.org  Mon Jun 19 12:35:20 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 19 Jun 2006 09:35:20 -0700
Subject: [cisco-voip] IPCCx ???'s
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEEF53@Apollo.wesd.org>

I have IPCCx 4.0.1sr1build 55.  I upgraded my CM's and IPCCx to OS ver
4.2.sr7a.

 

This morning when I came in the agents could not log in.  Opened a TAC
case and they had me move the CRS 3.5 license files out and then agents
could log in. (the Licensing and resource manager service failed type
error)

 

Now I'm dealing with the supervisors can select a team.  Any idea how to
fix it?  The Supervisors were all removed from the teams so I added them
back and clicked update, but I'm still having the same problem.  This CM
and IPCCx is integrated with AD 2003.

 

TAC says that 4.2.sr7a is not supported on IPCCx.  I'm trying to find
anywhere it says this other then the metrix.  No note of this is in the
release notes I could find.

 

Thanks

 

Scott

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From lelio at uoguelph.ca  Mon Jun 19 12:43:27 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 19 Jun 2006 12:43:27 -0400
Subject: [cisco-voip] IPCCx ???'s
References: <407055A92CECCB499C922A2D35FC19A605CEEF53@Apollo.wesd.org>
Message-ID: <00c701c693bf$7d14f580$47196883@cfs.uoguelph.ca>

I've referred to this document in the past. It seems to be the same file name all the time since everytime I go back, it's always updated. This one seems to be updated June 12, 2006. sr7a does not seem to be on the list. :(


http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1225/ccmigration_09186a00803d82f5.pdf

      Cisco CRS 4.0(1) SR1 Solution Set (Sheet 1 of 2)CallManager
     OS Release
     Hardware
     IP Phones for Cisco Agent Desktop
     IP Phones for Cisco IP Phone Agent
     
      4.1(2) - JTAPI 2.1(2.3)*

         SR1 - JTAPI 2.1(2.5)

      4.1(3) - JTAPI 2.1(3.3)*- JTAPI 2.1(3.9)* SR2 - JTAPI 2.1(3.11)* SR3a - JTAPI 2.1(3.14)

      4.2(1) - JTAPI 2.2(1.5) SR1 - JTAPI 2.2(1.5)
     OS 2000.2.5 SR6   SR7 SR8

      OS 2000.2.6 SR1 SR2 SR3   SR4   SR5   SR6

      OS 2000.2.7* SR1*   SR2*   SR3   SR6

         SR7* 

      SR8*

      OS 2000.4.11 

      SR11 

      SR21 SR31 SR41 SR51*OS 2000.4.1a1

      OS 2000.4.2*

         SR1* SR2* SR3* SR4 SR5a 

      SR6 SR7
     

 
--------------------------------------------------------------------------------
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: Monday, June 19, 2006 12:35 PM
  Subject: [cisco-voip] IPCCx ???'s


  I have IPCCx 4.0.1sr1build 55.  I upgraded my CM's and IPCCx to OS ver 4.2.sr7a.

   

  This morning when I came in the agents could not log in.  Opened a TAC case and they had me move the CRS 3.5 license files out and then agents could log in. (the Licensing and resource manager service failed type error)

   

  Now I'm dealing with the supervisors can select a team.  Any idea how to fix it?  The Supervisors were all removed from the teams so I added them back and clicked update, but I'm still having the same problem.  This CM and IPCCx is integrated with AD 2003.

   

  TAC says that 4.2.sr7a is not supported on IPCCx.  I'm trying to find anywhere it says this other then the metrix.  No note of this is in the release notes I could find.

   

  Thanks

   

  Scott



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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Scott.Voll at wesd.org  Mon Jun 19 12:47:29 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 19 Jun 2006 09:47:29 -0700
Subject: [cisco-voip] IPCCx ???'s
Message-ID: <407055A92CECCB499C922A2D35FC19A605CEEF5C@Apollo.wesd.org>

Sure would be nice if they put that in the release notes of the SR!

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Monday, June 19, 2006 9:43 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCCx ???'s

 

I've referred to this document in the past. It seems to be the same file
name all the time since everytime I go back, it's always updated. This
one seems to be updated June 12, 2006. sr7a does not seem to be on the
list. :(

 

 

http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1225/c
cmigration_09186a00803d82f5.pdf

Cisco CRS 4.0(1) SR1 Solution Set (Sheet 1 of 2)CallManager

OS Release

Hardware

IP Phones for Cisco Agent Desktop

IP Phones for Cisco IP Phone Agent

4.1(2) - JTAPI 2.1(2.3)*

   SR1 - JTAPI 2.1(2.5)

4.1(3) - JTAPI 2.1(3.3)*- JTAPI 2.1(3.9)* SR2 - JTAPI 2.1(3.11)* SR3a -
JTAPI 2.1(3.14)

4.2(1) - JTAPI 2.2(1.5) SR1 - JTAPI 2.2(1.5)

OS 2000.2.5 SR6   SR7 SR8

OS 2000.2.6 SR1 SR2 SR3   SR4   SR5   SR6

OS 2000.2.7* SR1*   SR2*   SR3   SR6

   SR7* 

SR8*

OS 2000.4.11 

SR11 

SR21 SR31 SR41 SR51*OS 2000.4.1a1

OS 2000.4.2*

   SR1* SR2* SR3* SR4 SR5a 

SR6 SR7

 

------------------------------------------------------------------------
--------
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 <mailto:Scott.Voll at wesd.org>  

	To: cisco-voip at puck.nether.net 

	Sent: Monday, June 19, 2006 12:35 PM

	Subject: [cisco-voip] IPCCx ???'s

	 

	I have IPCCx 4.0.1sr1build 55.  I upgraded my CM's and IPCCx to
OS ver 4.2.sr7a.

	 

	This morning when I came in the agents could not log in.  Opened
a TAC case and they had me move the CRS 3.5 license files out and then
agents could log in. (the Licensing and resource manager service failed
type error)

	 

	Now I'm dealing with the supervisors can select a team.  Any
idea how to fix it?  The Supervisors were all removed from the teams so
I added them back and clicked update, but I'm still having the same
problem.  This CM and IPCCx is integrated with AD 2003.

	 

	TAC says that 4.2.sr7a is not supported on IPCCx.  I'm trying to
find anywhere it says this other then the metrix.  No note of this is in
the release notes I could find.

	 

	Thanks

	 

	Scott

	
________________________________


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

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From lelio at uoguelph.ca  Mon Jun 19 12:50:22 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 19 Jun 2006 12:50:22 -0400
Subject: [cisco-voip] IPCCx ???'s
References: <407055A92CECCB499C922A2D35FC19A605CEEF5C@Apollo.wesd.org>
Message-ID: <00df01c693c0$747bc470$47196883@cfs.uoguelph.ca>

Even if they did, you could not count in it's accuracy. I typically go by what's there and if it's not there, ask whether it's a mistake or if it was left out on purpose. I open documentation cases with the TAC all the time.

--------------------------------------------------------------------------------
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: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Monday, June 19, 2006 12:47 PM
  Subject: RE: [cisco-voip] IPCCx ???'s


  Sure would be nice if they put that in the release notes of the SR!

   


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

  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
  Sent: Monday, June 19, 2006 9:43 AM
  To: Voll, Scott; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] IPCCx ???'s

   

  I've referred to this document in the past. It seems to be the same file name all the time since everytime I go back, it's always updated. This one seems to be updated June 12, 2006. sr7a does not seem to be on the list. :(

   

   

  http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1225/ccmigration_09186a00803d82f5.pdf

        Cisco CRS 4.0(1) SR1 Solution Set (Sheet 1 of 2)CallManager
       OS Release
       Hardware
       IP Phones for Cisco Agent Desktop
       IP Phones for Cisco IP Phone Agent
       
        4.1(2) - JTAPI 2.1(2.3)*

           SR1 - JTAPI 2.1(2.5)

        4.1(3) - JTAPI 2.1(3.3)*- JTAPI 2.1(3.9)* SR2 - JTAPI 2.1(3.11)* SR3a - JTAPI 2.1(3.14)

        4.2(1) - JTAPI 2.2(1.5) SR1 - JTAPI 2.2(1.5)
       OS 2000.2.5 SR6   SR7 SR8

        OS 2000.2.6 SR1 SR2 SR3   SR4   SR5   SR6

        OS 2000.2.7* SR1*   SR2*   SR3   SR6

           SR7* 

        SR8*

        OS 2000.4.11 

        SR11 

        SR21 SR31 SR41 SR51*OS 2000.4.1a1

        OS 2000.4.2*

           SR1* SR2* SR3* SR4 SR5a 

        SR6 SR7
       

   

  --------------------------------------------------------------------------------
  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: Monday, June 19, 2006 12:35 PM

    Subject: [cisco-voip] IPCCx ???'s

     

    I have IPCCx 4.0.1sr1build 55.  I upgraded my CM's and IPCCx to OS ver 4.2.sr7a.

     

    This morning when I came in the agents could not log in.  Opened a TAC case and they had me move the CRS 3.5 license files out and then agents could log in. (the Licensing and resource manager service failed type error)

     

    Now I'm dealing with the supervisors can select a team.  Any idea how to fix it?  The Supervisors were all removed from the teams so I added them back and clicked update, but I'm still having the same problem.  This CM and IPCCx is integrated with AD 2003.

     

    TAC says that 4.2.sr7a is not supported on IPCCx.  I'm trying to find anywhere it says this other then the metrix.  No note of this is in the release notes I could find.

     

    Thanks

     

    Scott


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

    _______________________________________________
    cisco-voip mailing list
    cisco-voip at puck.nether.net
    https://puck.nether.net/mailman/listinfo/cisco-voip
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From tennille.spence at gsa.gov  Mon Jun 19 14:14:40 2006
From: tennille.spence at gsa.gov (tennille.spence at gsa.gov)
Date: Mon, 19 Jun 2006 13:14:40 -0500
Subject: [cisco-voip] Call History Help
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605CEEF53@Apollo.wesd.org>
Message-ID: <OF8A348101.16FA0D35-ON86257192.0063B0E7-86257192.00643839@gsa.gov>


Is there a way to remove the "clear" call history softkey from a phone or
to add a confirmation to the clearing process? Or is there any way to
retrieve call history once it has been cleared or after a phone reset? I
have someone with call history that keeps "disappearing". I am running
phone load 7.2.3.

- Tennille



From MLinsemier at apcapital.com  Mon Jun 19 14:20:35 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Mon, 19 Jun 2006 14:20:35 -0400
Subject: [cisco-voip] Call History Help
Message-ID: <AB0A35F2486F81468971AA68558BE71904037067@smieln01winm01.micoanet.com>

I have looked for a way to store the call history in CallManager 3.x and
4.x to no avail.  Apparently this isn't stored anyplace but on the
phone.  A lot of peers that I have talked with have been requesting this
feature from Cisco or third party for years.  Every time I do a firmware
upgrade or cycle the switches for maintenance, my employees loose their
call history.

Another big one for us was the request on clearing just a single entry,
rather then the entire history.

-Matt

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
tennille.spence at gsa.gov
Sent: Monday, June 19, 2006 2:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call History Help


Is there a way to remove the "clear" call history softkey from a phone
or
to add a confirmation to the clearing process? Or is there any way to
retrieve call history once it has been cleared or after a phone reset? I
have someone with call history that keeps "disappearing". I am running
phone load 7.2.3.

- Tennille


_______________________________________________
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.

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



From tennille.spence at gsa.gov  Mon Jun 19 14:25:37 2006
From: tennille.spence at gsa.gov (tennille.spence at gsa.gov)
Date: Mon, 19 Jun 2006 13:25:37 -0500
Subject: [cisco-voip] Call History Help
In-Reply-To: <AB0A35F2486F81468971AA68558BE71904037067@smieln01winm01.micoanet.com>
Message-ID: <OFCDB895F3.66C56B37-ON86257192.00650790-86257192.00653921@gsa.gov>


That's what I was afraid of. I think the lady reporting this issue is
accidentally hitting the clear key without noticing it and since it doesn't
bring up a confirmation it is gone before you know it. I tried going into
call manager to change the softkey layout but this particular one is not
there.


                                                                           
                                                                           
             "Linsemier,                                                   
             Matthew"                                                   To 
             <MLinsemier at apcap         tennille.spence at gsa.gov,            
             ital.com>                 cisco-voip at puck.nether.net          
                                                                        cc 
             06/19/2006 01:20                                              
             PM                                                    Subject 
                                       RE: [cisco-voip] Call History Help  
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           




I have looked for a way to store the call history in CallManager 3.x and
4.x to no avail.  Apparently this isn't stored anyplace but on the
phone.  A lot of peers that I have talked with have been requesting this
feature from Cisco or third party for years.  Every time I do a firmware
upgrade or cycle the switches for maintenance, my employees loose their
call history.

Another big one for us was the request on clearing just a single entry,
rather then the entire history.

-Matt

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
tennille.spence at gsa.gov
Sent: Monday, June 19, 2006 2:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Call History Help


Is there a way to remove the "clear" call history softkey from a phone
or
to add a confirmation to the clearing process? Or is there any way to
retrieve call history once it has been cleared or after a phone reset? I
have someone with call history that keeps "disappearing". I am running
phone load 7.2.3.

- Tennille


_______________________________________________
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.

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







From MillerSW at howrey.com  Mon Jun 19 14:29:54 2006
From: MillerSW at howrey.com (Miller, Steve W.)
Date: Mon, 19 Jun 2006 14:29:54 -0400
Subject: [cisco-voip] Different rings on the same phone set?
Message-ID: <FF5F9DB1ADC7BB42BFC2C6C1F6638F05E8130E@HQEXCHVS1.DC.howrey.net>

Can you have different rings for different numbers on the same 7960? If
so, how?  Thank you!
 

Steve W. Miller 
Telecom Engineer 
HOWREY, LLP 
1299 Pennsylvania Ave, NW 
Office IT103 (on B3 Level) 
Washington, DC 20004 
202-383-6949 (11-6949) 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, June 19, 2006 12:47 PM
To: Lelio Fulgenzi; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCCx ???'s



Sure would be nice if they put that in the release notes of the SR!

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Monday, June 19, 2006 9:43 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IPCCx ???'s

 

I've referred to this document in the past. It seems to be the same file
name all the time since everytime I go back, it's always updated. This
one seems to be updated June 12, 2006. sr7a does not seem to be on the
list. :(

 

 

http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1225/c
cmigration_09186a00803d82f5.pdf

Cisco CRS 4.0(1) SR1 Solution Set (Sheet 1 of 2)CallManager

OS Release

Hardware

IP Phones for Cisco Agent Desktop

IP Phones for Cisco IP Phone Agent

4.1(2) - JTAPI 2.1(2.3)*

   SR1 - JTAPI 2.1(2.5)

4.1(3) - JTAPI 2.1(3.3)*- JTAPI 2.1(3.9)* SR2 - JTAPI 2.1(3.11)* SR3a -
JTAPI 2.1(3.14)

4.2(1) - JTAPI 2.2(1.5) SR1 - JTAPI 2.2(1.5)

OS 2000.2.5 SR6   SR7 SR8

OS 2000.2.6 SR1 SR2 SR3   SR4   SR5   SR6

OS 2000.2.7* SR1*   SR2*   SR3   SR6

   SR7* 

SR8*

OS 2000.4.11 

SR11 

SR21 SR31 SR41 SR51*OS 2000.4.1a1

OS 2000.4.2*

   SR1* SR2* SR3* SR4 SR5a 

SR6 SR7

 

------------------------------------------------------------------------
--------
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 <mailto:Scott.Voll at wesd.org>  

	To: cisco-voip at puck.nether.net 

	Sent: Monday, June 19, 2006 12:35 PM

	Subject: [cisco-voip] IPCCx ???'s

	 

	I have IPCCx 4.0.1sr1build 55.  I upgraded my CM's and IPCCx to
OS ver 4.2.sr7a.

	 

	This morning when I came in the agents could not log in.  Opened
a TAC case and they had me move the CRS 3.5 license files out and then
agents could log in. (the Licensing and resource manager service failed
type error)

	 

	Now I'm dealing with the supervisors can select a team.  Any
idea how to fix it?  The Supervisors were all removed from the teams so
I added them back and clicked update, but I'm still having the same
problem.  This CM and IPCCx is integrated with AD 2003.

	 

	TAC says that 4.2.sr7a is not supported on IPCCx.  I'm trying to
find anywhere it says this other then the metrix.  No note of this is in
the release notes I could find.

	 

	Thanks

	 

	Scott

	
________________________________


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

-------------------------------------------------------------------------------------------------
This email and any attachments contain information from the law firm of Howrey LLP, which may be confidential and/or privileged. 
The information is intended to be for the use of the individual or entity named on this email. 
If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this email is prohibited. 
If you receive this email in error, please notify us by reply email immediately so that we can arrange for the retrieval of the original documents at no cost to you.
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From lelio at uoguelph.ca  Mon Jun 19 14:37:10 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 19 Jun 2006 14:37:10 -0400
Subject: [cisco-voip] Different rings on the same phone set?
References: <FF5F9DB1ADC7BB42BFC2C6C1F6638F05E8130E@HQEXCHVS1.DC.howrey.net>
Message-ID: <021701c693cf$603a1390$47196883@cfs.uoguelph.ca>

Can you be more specific? Are you looking for different ring tones? If so, yes. Just follow the prompts in the ring settings section of the settings.

If you're looking for different number of rings to voicemail, you need 4.1(x) I believe.

--------------------------------------------------------------------------------
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: Miller, Steve W. 
  To: Voll, Scott ; Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Monday, June 19, 2006 2:29 PM
  Subject: Different rings on the same phone set?


  Can you have different rings for different numbers on the same 7960? If so, how?  Thank you!

  Steve W. Miller 
  Telecom Engineer 
  HOWREY, LLP 
  1299 Pennsylvania Ave, NW 
  Office IT103 (on B3 Level) 
  Washington, DC 20004 
  202-383-6949 (11-6949) 
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From it at cimgroup.com  Mon Jun 19 14:46:10 2006
From: it at cimgroup.com (IT)
Date: Mon, 19 Jun 2006 11:46:10 -0700
Subject: [cisco-voip] call trigger vbscript
Message-ID: <B487D4C1FE6D04439061018C6EBC109B033A643E@exchange.cimgroup.com>

	Does anyone know how I can create a trigger for a specific
extension being called? We have a help desk extension, and I would like
to be able to create a vbscript (or any other script), that runs when
this extension is called internally, which will automatically create a
"phone ticket" in our trouble ticketing system.  The script will be
simple, I just need to know how I can trigger it...

	 

	Thanks,

	Avidan

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From MillerSW at howrey.com  Mon Jun 19 14:47:36 2006
From: MillerSW at howrey.com (Miller, Steve W.)
Date: Mon, 19 Jun 2006 14:47:36 -0400
Subject: [cisco-voip] Different rings on the same phone set?
Message-ID: <FF5F9DB1ADC7BB42BFC2C6C1F6638F05E81339@HQEXCHVS1.DC.howrey.net>

Ring settings.  Thank you!
 

Steve W. Miller 
Telecom Engineer 
HOWREY, LLP 
1299 Pennsylvania Ave, NW 
Office IT103 (on B3 Level) 
Washington, DC 20004 
202-383-6949 (11-6949) 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Monday, June 19, 2006 2:37 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Different rings on the same phone set?


Can you be more specific? Are you looking for different ring tones? If
so, yes. Just follow the prompts in the ring settings section of the
settings.
 
If you're looking for different number of rings to voicemail, you need
4.1(x) I believe.
 
------------------------------------------------------------------------
--------
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: Miller, Steve W. <mailto:MillerSW at howrey.com>  
	To: Voll, Scott <mailto:Scott.Voll at wesd.org>  ; Lelio Fulgenzi
<mailto:lelio at uoguelph.ca>  ; cisco-voip at puck.nether.net 
	Sent: Monday, June 19, 2006 2:29 PM
	Subject: Different rings on the same phone set?

	Can you have different rings for different numbers on the same
7960? If so, how?  Thank you!
	 

	Steve W. Miller 
	Telecom Engineer 
	HOWREY, LLP 
	1299 Pennsylvania Ave, NW 
	Office IT103 (on B3 Level) 
	Washington, DC 20004 
	202-383-6949 (11-6949) 

-------------------------------------------------------------------------------------------------
This email and any attachments contain information from the law firm of Howrey LLP, which may be confidential and/or privileged. 
The information is intended to be for the use of the individual or entity named on this email. 
If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this email is prohibited. 
If you receive this email in error, please notify us by reply email immediately so that we can arrange for the retrieval of the original documents at no cost to you.
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From BillS at tns.its.psu.edu  Mon Jun 19 14:51:10 2006
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Mon, 19 Jun 2006 14:51:10 -0400
Subject: [cisco-voip] call trigger vbscript
Message-ID: <07A14D6987A5F647B2415D72EDF63B5605550404@tnsex01.tns.its.psu.edu>

Two ways I can think of:
 
Use the JTAPI interface to monitor calls coming to your phones and fire off
the script on an incoming call.  (A good example that I refer to now and
then can be found here:
http://www-rocq.inria.fr/who/Philippe.Sultan/iptelJavaCiscoEng.html
<http://www-rocq.inria.fr/who/Philippe.Sultan/iptelJavaCiscoEng.html> )
 
Send all the calls through an IPCC script.  Then you'd also have the ability
to set up automatic call distribution for your helpdesk, should you decide
to do that.  (Don't have a link for this.)
 

---
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/ <http://tns.its.psu.edu/> 



 


  _____  

From: IT [mailto:it at cimgroup.com] 
Sent: Monday, June 19, 2006 2:46 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] call trigger vbscript



Does anyone know how I can create a trigger for a specific extension being
called? We have a help desk extension, and I would like to be able to create
a vbscript (or any other script), that runs when this extension is called
internally, which will automatically create a "phone ticket" in our trouble
ticketing system.  The script will be simple, I just need to know how I can
trigger it...

 

Thanks,

Avidan

-------------- next part --------------
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From lelio at uoguelph.ca  Mon Jun 19 14:52:26 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 19 Jun 2006 14:52:26 -0400
Subject: [cisco-voip] Different rings on the same phone set?
References: <FF5F9DB1ADC7BB42BFC2C6C1F6638F05E81339@HQEXCHVS1.DC.howrey.net>
Message-ID: <025601c693d1$82584a30$47196883@cfs.uoguelph.ca>

The key is to set the default ring type then associate it to the lines you want. Anything else will have to be one of the alternate ring types.

I'm hoping to clear up (unless someone already has and is willing to share) whether or not I can simply copy the ringlist.xml file to the alternate ringlist.xml file and have it supported by TAC. This way, you can use all ringtypes on all lines.

Another note is that as far as I can tell, you can only have 6 different ring types. If you have more lines (like on a 7914), when you try to set the 7th different ring type it won't let you. Can't remember what happens though.

--------------------------------------------------------------------------------
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: Miller, Steve W. 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Monday, June 19, 2006 2:47 PM
  Subject: RE: [cisco-voip] Different rings on the same phone set?


  Ring settings.  Thank you!

  Steve W. Miller 
  Telecom Engineer 
  HOWREY, LLP 
  1299 Pennsylvania Ave, NW 
  Office IT103 (on B3 Level) 
  Washington, DC 20004 
  202-383-6949 (11-6949) 





------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Monday, June 19, 2006 2:37 PM
  To: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Different rings on the same phone set?


  Can you be more specific? Are you looking for different ring tones? If so, yes. Just follow the prompts in the ring settings section of the settings.

  If you're looking for different number of rings to voicemail, you need 4.1(x) I believe.

  --------------------------------------------------------------------------------
  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: Miller, Steve W. 
    To: Voll, Scott ; Lelio Fulgenzi ; cisco-voip at puck.nether.net 
    Sent: Monday, June 19, 2006 2:29 PM
    Subject: Different rings on the same phone set?


    Can you have different rings for different numbers on the same 7960? If so, how?  Thank you!

    Steve W. Miller 
    Telecom Engineer 
    HOWREY, LLP 
    1299 Pennsylvania Ave, NW 
    Office IT103 (on B3 Level) 
    Washington, DC 20004 
    202-383-6949 (11-6949) 




        -------------------------------------------------------------------------------------------------
        This email and any attachments contain information from the law firm of Howrey LLP, which may be confidential and/or privileged. 
        The information is intended to be for the use of the individual or entity named on this email. 
        If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this email is prohibited. 
        If you receive this email in error, please notify us by reply email immediately so that we can arrange for the retrieval of the original documents at no cost to you. 
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From MillerSW at howrey.com  Mon Jun 19 15:28:09 2006
From: MillerSW at howrey.com (Miller, Steve W.)
Date: Mon, 19 Jun 2006 15:28:09 -0400
Subject: [cisco-voip] Different rings on the same phone set?
Message-ID: <FF5F9DB1ADC7BB42BFC2C6C1F6638F05E813A1@HQEXCHVS1.DC.howrey.net>

On the line settings on our Call Manager, we have:
 
Ring settings IDLE
Ring settings ACTIVE
 
and under each of these fields, we have:
 
disable
Flash Only
Ring Once
Ring
 
But I don't see anything about ring TYPE.  Where do you go to see what
rings are available?  Thank you!
 

Steve W. Miller 
Telecom Engineer 
HOWREY, LLP 
1299 Pennsylvania Ave, NW 
Office IT103 (on B3 Level) 
Washington, DC 20004 
202-383-6949 (11-6949) 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Monday, June 19, 2006 2:37 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Different rings on the same phone set?


Can you be more specific? Are you looking for different ring tones? If
so, yes. Just follow the prompts in the ring settings section of the
settings.
 
If you're looking for different number of rings to voicemail, you need
4.1(x) I believe.
 
------------------------------------------------------------------------
--------
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: Miller, Steve W. <mailto:MillerSW at howrey.com>  
	To: Voll, Scott <mailto:Scott.Voll at wesd.org>  ; Lelio Fulgenzi
<mailto:lelio at uoguelph.ca>  ; cisco-voip at puck.nether.net 
	Sent: Monday, June 19, 2006 2:29 PM
	Subject: Different rings on the same phone set?

	Can you have different rings for different numbers on the same
7960? If so, how?  Thank you!
	 

	Steve W. Miller 
	Telecom Engineer 
	HOWREY, LLP 
	1299 Pennsylvania Ave, NW 
	Office IT103 (on B3 Level) 
	Washington, DC 20004 
	202-383-6949 (11-6949) 

-------------------------------------------------------------------------------------------------
This email and any attachments contain information from the law firm of Howrey LLP, which may be confidential and/or privileged. 
The information is intended to be for the use of the individual or entity named on this email. 
If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this email is prohibited. 
If you receive this email in error, please notify us by reply email immediately so that we can arrange for the retrieval of the original documents at no cost to you.
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From lelio at uoguelph.ca  Mon Jun 19 15:32:13 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 19 Jun 2006 15:32:13 -0400
Subject: [cisco-voip] Different rings on the same phone set?
References: <FF5F9DB1ADC7BB42BFC2C6C1F6638F05E813A1@HQEXCHVS1.DC.howrey.net>
Message-ID: <02c701c693d7$128e5f90$47196883@cfs.uoguelph.ca>

These must be set on the phone.

--------------------------------------------------------------------------------
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: Miller, Steve W. 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Monday, June 19, 2006 3:28 PM
  Subject: RE: [cisco-voip] Different rings on the same phone set?


  On the line settings on our Call Manager, we have:

  Ring settings IDLE
  Ring settings ACTIVE

  and under each of these fields, we have:

  disable
  Flash Only
  Ring Once
  Ring

  But I don't see anything about ring TYPE.  Where do you go to see what rings are available?  Thank you!

  Steve W. Miller 
  Telecom Engineer 
  HOWREY, LLP 
  1299 Pennsylvania Ave, NW 
  Office IT103 (on B3 Level) 
  Washington, DC 20004 
  202-383-6949 (11-6949) 





------------------------------------------------------------------------------
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
  Sent: Monday, June 19, 2006 2:37 PM
  To: cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Different rings on the same phone set?


  Can you be more specific? Are you looking for different ring tones? If so, yes. Just follow the prompts in the ring settings section of the settings.

  If you're looking for different number of rings to voicemail, you need 4.1(x) I believe.

  --------------------------------------------------------------------------------
  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: Miller, Steve W. 
    To: Voll, Scott ; Lelio Fulgenzi ; cisco-voip at puck.nether.net 
    Sent: Monday, June 19, 2006 2:29 PM
    Subject: Different rings on the same phone set?


    Can you have different rings for different numbers on the same 7960? If so, how?  Thank you!

    Steve W. Miller 
    Telecom Engineer 
    HOWREY, LLP 
    1299 Pennsylvania Ave, NW 
    Office IT103 (on B3 Level) 
    Washington, DC 20004 
    202-383-6949 (11-6949) 




        -------------------------------------------------------------------------------------------------
        This email and any attachments contain information from the law firm of Howrey LLP, which may be confidential and/or privileged. 
        The information is intended to be for the use of the individual or entity named on this email. 
        If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this email is prohibited. 
        If you receive this email in error, please notify us by reply email immediately so that we can arrange for the retrieval of the original documents at no cost to you. 
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From MillerSW at howrey.com  Mon Jun 19 15:35:52 2006
From: MillerSW at howrey.com (Miller, Steve W.)
Date: Mon, 19 Jun 2006 15:35:52 -0400
Subject: [cisco-voip] Different rings on the same phone set?
Message-ID: <FF5F9DB1ADC7BB42BFC2C6C1F6638F05E813B5@HQEXCHVS1.DC.howrey.net>

Oh, OK.  There doesn't seem to be a list of different rings when I
access the ring types from the settings on the phone.  I just see
"default" and then "chirp".
 

Steve W. Miller 
Telecom Engineer 
HOWREY, LLP 
1299 Pennsylvania Ave, NW 
Office IT103 (on B3 Level) 
Washington, DC 20004 
202-383-6949 (11-6949) 

 

________________________________

From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
Sent: Monday, June 19, 2006 3:32 PM
To: Miller, Steve W.; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Different rings on the same phone set?


These must be set on the phone.
 
------------------------------------------------------------------------
--------
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: Miller, Steve W. <mailto:MillerSW at howrey.com>  
	To: Lelio Fulgenzi <mailto:lelio at uoguelph.ca>  ;
cisco-voip at puck.nether.net 
	Sent: Monday, June 19, 2006 3:28 PM
	Subject: RE: [cisco-voip] Different rings on the same phone set?

	On the line settings on our Call Manager, we have:
	 
	Ring settings IDLE
	Ring settings ACTIVE
	 
	and under each of these fields, we have:
	 
	disable
	Flash Only
	Ring Once
	Ring
	 
	But I don't see anything about ring TYPE.  Where do you go to
see what rings are available?  Thank you!
	 

	Steve W. Miller 
	Telecom Engineer 
	HOWREY, LLP 
	1299 Pennsylvania Ave, NW 
	Office IT103 (on B3 Level) 
	Washington, DC 20004 
	202-383-6949 (11-6949) 

	 

________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	Sent: Monday, June 19, 2006 2:37 PM
	To: cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Different rings on the same phone set?
	
	
	Can you be more specific? Are you looking for different ring
tones? If so, yes. Just follow the prompts in the ring settings section
of the settings.
	 
	If you're looking for different number of rings to voicemail,
you need 4.1(x) I believe.
	 
	
------------------------------------------------------------------------
--------
	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: Miller, Steve W. <mailto:MillerSW at howrey.com>  
		To: Voll, Scott <mailto:Scott.Voll at wesd.org>  ; Lelio
Fulgenzi <mailto:lelio at uoguelph.ca>  ; cisco-voip at puck.nether.net 
		Sent: Monday, June 19, 2006 2:29 PM
		Subject: Different rings on the same phone set?

		Can you have different rings for different numbers on
the same 7960? If so, how?  Thank you!
		 

		Steve W. Miller 
		Telecom Engineer 
		HOWREY, LLP 
		1299 Pennsylvania Ave, NW 
		Office IT103 (on B3 Level) 
		Washington, DC 20004 
		202-383-6949 (11-6949) 




------------------------------------------------------------------------
-------------------------
This email and any attachments contain information from the law firm of
Howrey LLP, which may be confidential and/or privileged. 
The information is intended to be for the use of the individual or
entity named on this email. 
If you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this email is
prohibited. 
If you receive this email in error, please notify us by reply email
immediately so that we can arrange for the retrieval of the original
documents at no cost to you.	

-------------- next part --------------
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From lelio at uoguelph.ca  Mon Jun 19 15:45:24 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 19 Jun 2006 15:45:24 -0400
Subject: [cisco-voip] Different rings on the same phone set?
References: <FF5F9DB1ADC7BB42BFC2C6C1F6638F05E813B5@HQEXCHVS1.DC.howrey.net>
Message-ID: <02de01c693d8$e956b2b0$47196883@cfs.uoguelph.ca>

This can be one of two things, the default setting with no changes, or an error in loading the ring list file. Try this: Settings > Ring Type > Select (softkey) and you should see the list of ring types. If not, try reloading the phone. If that doesn't work, you'll have to start troubleshooting. Perhaps the file is corrupt. You should have two files (attached) in your TFTP directory. Mind you, the last selection is custom to us.

 
--------------------------------------------------------------------------------
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: Miller, Steve W. 
  To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
  Sent: Monday, June 19, 2006 3:35 PM
  Subject: RE: [cisco-voip] Different rings on the same phone set?


  Oh, OK.  There doesn't seem to be a list of different rings when I access the ring types from the settings on the phone.  I just see "default" and then "chirp".

  Steve W. Miller 
  Telecom Engineer 
  HOWREY, LLP 
  1299 Pennsylvania Ave, NW 
  Office IT103 (on B3 Level) 
  Washington, DC 20004 
  202-383-6949 (11-6949) 





------------------------------------------------------------------------------
  From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca] 
  Sent: Monday, June 19, 2006 3:32 PM
  To: Miller, Steve W.; cisco-voip at puck.nether.net
  Subject: Re: [cisco-voip] Different rings on the same phone set?


  These must be set on the phone.

  --------------------------------------------------------------------------------
  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: Miller, Steve W. 
    To: Lelio Fulgenzi ; cisco-voip at puck.nether.net 
    Sent: Monday, June 19, 2006 3:28 PM
    Subject: RE: [cisco-voip] Different rings on the same phone set?


    On the line settings on our Call Manager, we have:

    Ring settings IDLE
    Ring settings ACTIVE

    and under each of these fields, we have:

    disable
    Flash Only
    Ring Once
    Ring

    But I don't see anything about ring TYPE.  Where do you go to see what rings are available?  Thank you!

    Steve W. Miller 
    Telecom Engineer 
    HOWREY, LLP 
    1299 Pennsylvania Ave, NW 
    Office IT103 (on B3 Level) 
    Washington, DC 20004 
    202-383-6949 (11-6949) 





----------------------------------------------------------------------------
    From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
    Sent: Monday, June 19, 2006 2:37 PM
    To: cisco-voip at puck.nether.net
    Subject: Re: [cisco-voip] Different rings on the same phone set?


    Can you be more specific? Are you looking for different ring tones? If so, yes. Just follow the prompts in the ring settings section of the settings.

    If you're looking for different number of rings to voicemail, you need 4.1(x) I believe.

    --------------------------------------------------------------------------------
    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: Miller, Steve W. 
      To: Voll, Scott ; Lelio Fulgenzi ; cisco-voip at puck.nether.net 
      Sent: Monday, June 19, 2006 2:29 PM
      Subject: Different rings on the same phone set?


      Can you have different rings for different numbers on the same 7960? If so, how?  Thank you!

      Steve W. Miller 
      Telecom Engineer 
      HOWREY, LLP 
      1299 Pennsylvania Ave, NW 
      Office IT103 (on B3 Level) 
      Washington, DC 20004 
      202-383-6949 (11-6949) 




          -------------------------------------------------------------------------------------------------
          This email and any attachments contain information from the law firm of Howrey LLP, which may be confidential and/or privileged. 
          The information is intended to be for the use of the individual or entity named on this email. 
          If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this email is prohibited. 
          If you receive this email in error, please notify us by reply email immediately so that we can arrange for the retrieval of the original documents at no cost to you. 
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From dphillips at sapc.edu  Mon Jun 19 16:02:25 2006
From: dphillips at sapc.edu (Phillips, Danny)
Date: Mon, 19 Jun 2006 16:02:25 -0400
Subject: [cisco-voip] Upgrading Unity
Message-ID: <18F0B5EFCCE1A449B07A3A9DF61DC07308ACD0@xchange.sapc.edu>

I am running Unity 4.0(2) on a Windows 2000 Server with Exchange 2003. I
want to upgrade the server to Windows 2003, Exchange 2003 and the latest
version of unity. Is there a good migration guide out there for doing
this and or does anyone have any tips to make this as painless as
possible. I do have another server that has the same sepcs. as the
production unity server.

 

 

Thanks

 

 

`````````````````````````````````````````````````````````````````````

Danny Phillips

Network Administrator

St. Andrews Presbyterian College

Telephone 910-277-5079

http://www.sapc.edu <http://www.sapc.edu/> 

dphillips at sapc.edu

 

``````````````````````````````````````````````````````````````````````

 

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From sodonnell at CCSINET.com  Mon Jun 19 16:06:52 2006
From: sodonnell at CCSINET.com (Scott O'Donnell)
Date: Mon, 19 Jun 2006 16:06:52 -0400
Subject: [cisco-voip] CME Questions for Rings.
Message-ID: <17A170DC8DD5CC44BA56B24B366EB60B01EDFB93@E3K-VS1.ccsi.local>

Is it possible to have different line appearances on a CME phone use
different ringer files?
 
I know it's tied to the firmware version on the phone but is there a
version that is supported for CME that will do this?
 
 
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From Joe_Bell at adp.com  Mon Jun 19 16:19:52 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Mon, 19 Jun 2006 15:19:52 -0500
Subject: [cisco-voip] Upgrading Unity
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC884BD@EXCHANGEDS01.ds.ad.adp.com>

The Unity Install and upgrade guides are all listed here:

 

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_installation_
guides_list.html

 

You want to take a real close look into going to Windows 2003.  There
are some major pitfalls in doing that and you should be very clear on
what those are before doing this upgrade.  Namely, the changes in Active
Directory schema's and how Exchange 2003 interoperates with Windows
2003.  Many Cisco doc's warn admin's away from using Windows 2003 for
these reasons amongst others.  It can be done, just do lots of research
before you begin the upgrade.

 

Joe

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Phillips, Danny
Sent: Monday, June 19, 2006 1:02 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Upgrading Unity

 

I am running Unity 4.0(2) on a Windows 2000 Server with Exchange 2003. I
want to upgrade the server to Windows 2003, Exchange 2003 and the latest
version of unity. Is there a good migration guide out there for doing
this and or does anyone have any tips to make this as painless as
possible. I do have another server that has the same sepcs. as the
production unity server.

 

 

Thanks

 

 

`````````````````````````````````````````````````````````````````````

Danny Phillips

Network Administrator

St. Andrews Presbyterian College

Telephone 910-277-5079

http://www.sapc.edu <http://www.sapc.edu/> 

dphillips at sapc.edu

 

``````````````````````````````````````````````````````````````````````

 



This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
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From it at cimgroup.com  Mon Jun 19 16:40:20 2006
From: it at cimgroup.com (IT)
Date: Mon, 19 Jun 2006 13:40:20 -0700
Subject: [cisco-voip] call trigger vbscript
Message-ID: <B487D4C1FE6D04439061018C6EBC109B033A6594@exchange.cimgroup.com>

Does anyone have a simple JTAPI example for capturing an event such as
an incoming call?

 

Thanks,

Avidan

 

________________________________

From: Simon, Bill [mailto:BillS at tns.its.psu.edu] 
Sent: Monday, June 19, 2006 11:51 AM
To: IT; 'cisco-voip at puck.nether.net'
Subject: RE: [cisco-voip] call trigger vbscript

 

Two ways I can think of:

 

Use the JTAPI interface to monitor calls coming to your phones and fire
off the script on an incoming call.  (A good example that I refer to now
and then can be found here:
http://www-rocq.inria.fr/who/Philippe.Sultan/iptelJavaCiscoEng.html)

 

Send all the calls through an IPCC script.  Then you'd also have the
ability to set up automatic call distribution for your helpdesk, should
you decide to do that.  (Don't have a link for this.)

 

---
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/

 

	 

	
________________________________


	From: IT [mailto:it at cimgroup.com] 
	Sent: Monday, June 19, 2006 2:46 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] call trigger vbscript

		Does anyone know how I can create a trigger for a
specific extension being called? We have a help desk extension, and I
would like to be able to create a vbscript (or any other script), that
runs when this extension is called internally, which will automatically
create a "phone ticket" in our trouble ticketing system.  The script
will be simple, I just need to know how I can trigger it...

		 

		Thanks,

		Avidan

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From cenders at homesbyavi.com  Mon Jun 19 16:57:04 2006
From: cenders at homesbyavi.com (Corbett Enders)
Date: Mon, 19 Jun 2006 14:57:04 -0600
Subject: [cisco-voip] Unregistered remote handset in CM -> forward call out
	PSTN
References: <mailman.9011.1150743082.1478.cisco-voip@puck.nether.net>
Message-ID: <24DFD2417491E547AC80AE140706C485039D9F93@EX001.homesbyavi.local>

Have the situation where we have a number of remote sites using 2811s
and SRST (using FXO boards for analog telco lines).

If the WAN gets dropped, any calls outoing leave via the analog line. If
we call the unpublished analog number, calls ring in on the primary
phone. 

However, we'd like anyone trying to dial the extension from head office
to leave head office via the PRI and ring through to the unpublished
analog number on the FXO port.  Doesn't seem to be a field for this. The
field "Forward No Coverage" seems to only apply to CTI route points...

Any ideas?


From jsteinberg at gmail.com  Mon Jun 19 17:19:05 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Mon, 19 Jun 2006 17:19:05 -0400
Subject: [cisco-voip] Unregistered remote handset in CM -> forward call
	out PSTN
In-Reply-To: <24DFD2417491E547AC80AE140706C485039D9F93@EX001.homesbyavi.local>
References: <mailman.9011.1150743082.1478.cisco-voip@puck.nether.net>
	<24DFD2417491E547AC80AE140706C485039D9F93@EX001.homesbyavi.local>
Message-ID: <f1b65f880606191419t12613159j572e98abb86be0e2@mail.gmail.com>

With CallManager 4.2(1) you have new fields on the DN for:
Forward Unregistered Internal & Forward Unregistered External

I'm not sure what flavors of CCM 5.x have those fields or whether this
can be accomplished in 4.1 or below via some type of HG config.

Justin



On 6/19/06, Corbett Enders <cenders at homesbyavi.com> wrote:
> Have the situation where we have a number of remote sites using 2811s
> and SRST (using FXO boards for analog telco lines).
>
> If the WAN gets dropped, any calls outoing leave via the analog line. If
> we call the unpublished analog number, calls ring in on the primary
> phone.
>
> However, we'd like anyone trying to dial the extension from head office
> to leave head office via the PRI and ring through to the unpublished
> analog number on the FXO port.  Doesn't seem to be a field for this. The
> field "Forward No Coverage" seems to only apply to CTI route points...
>
> Any ideas?
>
> _______________________________________________
> 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  Mon Jun 19 17:21:14 2006
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Mon, 19 Jun 2006 17:21:14 -0400
Subject: [cisco-voip] OS 2000.4.2 upgrade
Message-ID: <07A14D6987A5F647B2415D72EDF63B5605550449@tnsex01.tns.its.psu.edu>

Is there any trouble to be had by upgrading the OS of an IPT cluster on
different days?

Our cluster is currently at 2000.2.7sr4.

So, for example, I would upgrade the CCM publisher during the maintenance
window on Wednesday, subscribers on Thursday and IPCC servers on Friday...

(The problem is that there is not enough of a window allotted to do all the
upgrades on the same day.)

I know that CCM application upgrades have to be done all at once, but does
the OS version matter in the same way? 

---
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/

From lelio at uoguelph.ca  Mon Jun 19 17:24:09 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 19 Jun 2006 17:24:09 -0400
Subject: [cisco-voip] OS 2000.4.2 upgrade
References: <07A14D6987A5F647B2415D72EDF63B5605550449@tnsex01.tns.its.psu.edu>
Message-ID: <004c01c693e6$b43dada0$c801a8c0@EATIUSBIRDIUS>

I've asked that question on the list before, and consesus seems to be that it is OK.

...he says, ducking for cover...
  ----- Original Message ----- 
  From: Simon, Bill 
  To: 'cisco-voip at puck.nether.net' 
  Sent: Monday, June 19, 2006 5:21 PM
  Subject: [cisco-voip] OS 2000.4.2 upgrade


  Is there any trouble to be had by upgrading the OS of an IPT cluster on
  different days?

  Our cluster is currently at 2000.2.7sr4.

  So, for example, I would upgrade the CCM publisher during the maintenance
  window on Wednesday, subscribers on Thursday and IPCC servers on Friday...

  (The problem is that there is not enough of a window allotted to do all the
  upgrades on the same day.)

  I know that CCM application upgrades have to be done all at once, but does
  the OS version matter in the same way? 

  ---
  Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
  http://tns.its.psu.edu/
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From Scott.Voll at wesd.org  Mon Jun 19 17:48:32 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 19 Jun 2006 14:48:32 -0700
Subject: [cisco-voip] IPCCx ???'s
Message-ID: <407055A92CECCB499C922A2D35FC19A605D631C4@Apollo.wesd.org>

Just to go full circle.

 

For the record 4.2sr7a is supported. The doc's just haven't been updated
yet.

 

Since I'm integrated with AD it just took time for everything to resync
so I'm back  up and running.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Monday, June 19, 2006 9:35 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] IPCCx ???'s

 

I have IPCCx 4.0.1sr1build 55.  I upgraded my CM's and IPCCx to OS ver
4.2.sr7a.

 

This morning when I came in the agents could not log in.  Opened a TAC
case and they had me move the CRS 3.5 license files out and then agents
could log in. (the Licensing and resource manager service failed type
error)

 

Now I'm dealing with the supervisors can select a team.  Any idea how to
fix it?  The Supervisors were all removed from the teams so I added them
back and clicked update, but I'm still having the same problem.  This CM
and IPCCx is integrated with AD 2003.

 

TAC says that 4.2.sr7a is not supported on IPCCx.  I'm trying to find
anywhere it says this other then the metrix.  No note of this is in the
release notes I could find.

 

Thanks

 

Scott

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From JOlson at mlnusa.com  Mon Jun 19 16:46:10 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Mon, 19 Jun 2006 16:46:10 -0400
Subject: [cisco-voip] Attendant Console online/offline
Message-ID: <FE2D092F8B1516488B99342551AACFD41CEE9216@mlnusa-exch1.mlnusa.com>

I have attendant console installed on a few different receptionists PCs
and it doesn't seem to matter if it is in the online or offline state.
Both ways it seems to have the same functionality.  What am I missing
with this?  Is there a setting in call manager that I am missing?

 

Jeff


********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

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From jim at tgasolutions.com  Mon Jun 19 20:42:55 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Mon, 19 Jun 2006 20:42:55 -0400
Subject: [cisco-voip] CME Questions for Rings.
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC15E986@exchange.TGASolutions.local>

The distintive ring tone selection exists in CME 4.. but I have not
played with it..
 
Jim

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Scott O'Donnell
Sent: Monday, June 19, 2006 4:07 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME Questions for Rings.


Is it possible to have different line appearances on a CME phone use
different ringer files?
 
I know it's tied to the firmware version on the phone but is there a
version that is supported for CME that will do this?
 
 
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From jgrace at digitelusa.net  Mon Jun 19 20:48:24 2006
From: jgrace at digitelusa.net (James Grace)
Date: Mon, 19 Jun 2006 20:48:24 -0400
Subject: [cisco-voip] crs 4.0.4
Message-ID: <200606192100890.SM01452@3729434770>

We have a coresident cm 4.2 and ipcc express 4.0.4

 For some reason the realtime reports do not start over each day.  We have
to manually reset the data each morning.  What could cause this 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From netwirker at gmail.com  Tue Jun 20 01:19:09 2006
From: netwirker at gmail.com (netwirker)
Date: Tue, 20 Jun 2006 01:19:09 -0400
Subject: [cisco-voip] CME, CUE and T1 trunk with 3845
Message-ID: <449784c6.2f4110a3.4feb.ffffe4f8@mx.gmail.com>

Hi Jim,

I just want to confirm in your case CUE, CME and T1(or E1) line running in one router?
I am very concern about this...
  

======= 2006-06-16 07:32:09 ????????=======

>Don't run Debugs on the TFTP server and u should be ok....
>I have a few 2851s doing this.. With near max phone counts..
>3845 ? Not a 25?  
>
>Vwic2-1mftt1/e1
>
>Voice quality? Never seen that except on a poor design or misconfig..
>
>J
>
>-----Original Message-----
>From: cisco-voip-bounces at puck.nether.net
>[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of netwirker
>Sent: Friday, June 16, 2006 12:59 AM
>To: cisco-voip
>Subject: [cisco-voip] CME, CUE and T1 trunk with 3845
>
>Hi guys,
>
>I will set up CME, CUE and install one PSTN PRI T1 in one Cisco 3845
>based on customer request.
>I didn't configure these services in one router before and wondering if
>Cisco router can stand the burden and performance well. As my experience
>CUE and T1 link can't install in one router otherwise serious voice
>quality problem will happen. What's your idea, any suggestion to
>optimize performance?
>Client want to set up 120 phone on CME and 200 mailbox on CUE.
>
>Btw, what T1 module I need install for 3845 if I need DID feature?
>
>Thanks a lot!  
>
>Andrew Xue
>netwirker at gmail.com
>




From hansel.gunawan at multipolar.co.id  Tue Jun 20 02:26:09 2006
From: hansel.gunawan at multipolar.co.id (Hansel Gideon Gunawan)
Date: Tue, 20 Jun 2006 13:26:09 +0700
Subject: [cisco-voip] Installing IPCC Express 4.5
Message-ID: <FCF1EA6D9A9506418E4DC24F01E0D14A05040713@mpcserver.multipolar.com>

Hi all,

 

I am wondering if we can install IPCC Express 4.5 in a generic server.
Will the registry tweak for previous version of IPCCX works? I have
searched the archive and cant find anything related to this version of
IPCC Express.

 

 

Regards,

 

Hansel Gideon Gunawan

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From rratliff at cisco.com  Tue Jun 20 09:17:27 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 20 Jun 2006 09:17:27 -0400
Subject: [cisco-voip] Attendant Console online/offline
In-Reply-To: <FE2D092F8B1516488B99342551AACFD41CEE9216@mlnusa-exch1.mlnusa.com>
References: <FE2D092F8B1516488B99342551AACFD41CEE9216@mlnusa-exch1.mlnusa.com>
Message-ID: <22FA429C-16AE-4F49-A398-14061BAB8FE6@cisco.com>

How do you have your AC hunt group configured?  If you set it to go  
to UserA-Line1, etc then the attendant must be logged in and online  
to receive calls.  If you set it to go to a dn/partition specifically  
then the AC console status will have no bearing on how calls are  
extended.

-Ryan

On Jun 19, 2006, at 4:46 PM, Olson, Jeffrey wrote:

I have attendant console installed on a few different receptionists  
PCs and it doesn?t seem to matter if it is in the online or offline  
state.  Both ways it seems to have the same functionality.  What am I  
missing with this?  Is there a setting in call manager that I am  
missing?



Jeff

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders  
Network USA, Inc. which is confidential and legally privileged. The  
information is intended only for the use of the individual or entity  
named on this E-mail. If you are not the intended recipient, you are  
hereby notified that disclosing, copying, distributing, or taking any  
action in reliance on the contents of this information is strictly  
prohibited, and that the E-mail should be returned to this company  
immediately. If you have received this E-mail in error, please  
contact the sender immediately.


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



From LUNDJC at uwec.edu  Tue Jun 20 09:50:46 2006
From: LUNDJC at uwec.edu (Lund, John C.)
Date: Tue, 20 Jun 2006 08:50:46 -0500
Subject: [cisco-voip] Background Image on 7970 - How to extract it from the
	phone
Message-ID: <D0AEBED8692AB14791F4FA1CE121CC6D031832F0@PEPSI.uwec.edu>

Is there any way to retrieve a background image from a 7970?  It does
not exist on tftp and only exists as a background image on one of our
7970's.  We are still looking for the original but will probably not be
able to locate it.
 
John
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From dphillips at sapc.edu  Tue Jun 20 10:19:35 2006
From: dphillips at sapc.edu (Phillips, Danny)
Date: Tue, 20 Jun 2006 10:19:35 -0400
Subject: [cisco-voip] Upgrading Unity
Message-ID: <18F0B5EFCCE1A449B07A3A9DF61DC07308ACD6@xchange.sapc.edu>

Thanks, I will keep that in mind

 

Danny

 

________________________________

From: Bell, Joe [mailto:Joe_Bell at adp.com] 
Sent: Monday, June 19, 2006 4:20 PM
To: Phillips, Danny; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Upgrading Unity

 

The Unity Install and upgrade guides are all listed here:

 

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_installation_
guides_list.html

 

You want to take a real close look into going to Windows 2003.  There
are some major pitfalls in doing that and you should be very clear on
what those are before doing this upgrade.  Namely, the changes in Active
Directory schema's and how Exchange 2003 interoperates with Windows
2003.  Many Cisco doc's warn admin's away from using Windows 2003 for
these reasons amongst others.  It can be done, just do lots of research
before you begin the upgrade.

 

Joe

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Phillips, Danny
Sent: Monday, June 19, 2006 1:02 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Upgrading Unity

 

I am running Unity 4.0(2) on a Windows 2000 Server with Exchange 2003. I
want to upgrade the server to Windows 2003, Exchange 2003 and the latest
version of unity. Is there a good migration guide out there for doing
this and or does anyone have any tips to make this as painless as
possible. I do have another server that has the same sepcs. as the
production unity server.

 

 

Thanks

 

 

`````````````````````````````````````````````````````````````````````

Danny Phillips

Network Administrator

St. Andrews Presbyterian College

Telephone 910-277-5079

http://www.sapc.edu <http://www.sapc.edu/> 

dphillips at sapc.edu

 

``````````````````````````````````````````````````````````````````````

 

________________________________

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 Ruben.Montes at eu.didata.com  Tue Jun 20 10:24:22 2006
From: Ruben.Montes at eu.didata.com (Ruben Montes (EU))
Date: Tue, 20 Jun 2006 16:24:22 +0200
Subject: [cisco-voip] Unity integration with Lotus Domino
Message-ID: <FA233791F398B848B978464507739E8F0C5D9C@EUBEBRUSVEX1.eu.didata.local>


Hello,

has anyone done successfully an integration of Cisco Unity and Lotus Domino for Unified Messaging? I have some questions regarding the changes that should be done in the Domino server.

Thanks in advance,

Ruben
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From tim.reimers at asheville.k12.nc.us  Tue Jun 20 10:42:38 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Tue, 20 Jun 2006 10:42:38 -0400
Subject: [cisco-voip] PRI line option question-
Message-ID: <050944EE48FF384F9C663F7048C47DA704143122@webmail.asheville.k12.nc.us>


Hi everyone-

I wanted to ask a question -


Our ISDN PRI provider has started billing us (belatedly) for an option
they  have provisioned on our line some time ago, and not told us.

It's a caller ID option called "Calling party number to name"

Before I decide that I don't want this, and they should take it off--

Does Cisco CCM support this?

Here's the details:

Cisco 2651XM running c2600-ipvoice-mz.123-8.T3
with two PRIs attached, 30 out of 47 channels live.
The gateway is in H.323 mode, with the appropriate (working fine) config
in CCM.

when you get an outside call, you get just numbers, not names..

So Bell is providing us names with our Caller ID, but I guess our
equipment is either incapable of showing that, or it's ignoring it.


Any thoughts?

Can I configure this to work, so that I can see what info -is- available
from various people?


thanks, Tim
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From b.turnbow at twt.it  Tue Jun 20 10:45:11 2006
From: b.turnbow at twt.it (Brian Turnbow)
Date: Tue, 20 Jun 2006 16:45:11 +0200
Subject: [cisco-voip] Dial collection
Message-ID: <52964DBFB59FA34F98E96921BE21C90601465A18@twt-exch.milano.twt>


Hello all,

I have a 2821 with a E1 multiflex card connected to a PBX. 
When analog lines of the pbx dial out I receive both the in band dialed
digits and isdn info messages from the pbx. 
This causes the router to send out in h323 the incorrect numer . For
example  1234 gets sent out as 11223344
I have been looking for a way to disable the in band digit collection,
but have yet to find out how. 
Can any one point me in the right direction ?

Thanks
Brian
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From rratliff at cisco.com  Tue Jun 20 10:52:15 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 20 Jun 2006 10:52:15 -0400
Subject: [cisco-voip] PRI line option question-
In-Reply-To: <050944EE48FF384F9C663F7048C47DA704143122@webmail.asheville.k12.nc.us>
References: <050944EE48FF384F9C663F7048C47DA704143122@webmail.asheville.k12.nc.us>
Message-ID: <F3F50B61-A22B-444A-AD2F-CCCF9EB5FC54@cisco.com>

You need to get a 'debug isdn q931' for an inbound call to see where  
the name is appearing and why CM isn't showing it.

-Ryan

On Jun 20, 2006, at 10:42 AM, Tim Reimers wrote:


Hi everyone-

I wanted to ask a question -


Our ISDN PRI provider has started billing us (belatedly) for an  
option they  have provisioned on our line some time ago, and not told  
us.

It's a caller ID option called "Calling party number to name"

Before I decide that I don't want this, and they should take it off--

Does Cisco CCM support this?

Here's the details:

Cisco 2651XM running c2600-ipvoice-mz.123-8.T3
with two PRIs attached, 30 out of 47 channels live.
The gateway is in H.323 mode, with the appropriate (working fine)  
config in CCM.

when you get an outside call, you get just numbers, not names..

So Bell is providing us names with our Caller ID, but I guess our  
equipment is either incapable of showing that, or it's ignoring it.


Any thoughts?

Can I configure this to work, so that I can see what info -is-  
available from various people?


thanks, Tim

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


From kfedor at amt.ru  Tue Jun 20 10:53:46 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Tue, 20 Jun 2006 18:53:46 +0400
Subject: [cisco-voip] Dial collection
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC707BE4755@amthqex01.amt.ru>

Hi, Brian.

 

ISDN Info - so you PBX send digits over overlap.

 

Whai is you scenario ? Two state dialing ??

 

Can you send a snip ?of

Debug isdn q931 

 

And D channel configuration.

-----------------------
Sincerely Yours,
Konstantin Fedorov

  _____  

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Turnbow
Sent: Tuesday, June 20, 2006 6:45 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Dial collection

 

 

Hello all, 

I have a 2821 with a E1 multiflex card connected to a PBX. 
When analog lines of the pbx dial out I receive both the in band dialed digits and isdn info messages from the pbx. 
This causes the router to send out in h323 the incorrect numer . For example  1234 gets sent out as 11223344 
I have been looking for a way to disable the in band digit collection, but have yet to find out how. 
Can any one point me in the right direction ? 

Thanks 
Brian 

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From b.turnbow at twt.it  Tue Jun 20 11:02:01 2006
From: b.turnbow at twt.it (Brian Turnbow)
Date: Tue, 20 Jun 2006 17:02:01 +0200
Subject: [cisco-voip] Dial collection
In-Reply-To: <8B5577ED1D5D2F4AA0E4651EC41C6DC707BE4755@amthqex01.amt.ru>
Message-ID: <52964DBFB59FA34F98E96921BE21C90601465A19@twt-exch.milano.twt>

Hi 
yes and with two state dialing , The end users digits a code to access the voip line and receives tone fron the router ( no direct dial in) 
Below is an ISDN setup, and everything looks ok. 
However on the outboud h323 i See  00223366552299443300 as the called number. 
ISDN phones send correct on the pbx, it is only the analog lines that have this problem. 
 
 
*Jun 20 14:59:16.151: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref = 0x018E
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98385
                Exclusive, Channel 5
        Progress Ind i = 0x8083 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '219'
                Plan:ISDN, Type:Unknown
        Low Layer Compat i = 0x9090A3
*Jun 20 14:59:16.159: ISDN Se0/0/0:15 Q931: TX -> SETUP_ACK pd = 8  callref = 0x818E
        Channel ID i = 0xA98385
                Exclusive, Channel 5
*Jun 20 14:59:16.991: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '0'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:17.779: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '2'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:18.495: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '3'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:19.199: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '6'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:19.795: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '5'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:20.387: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '2'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:21.087: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '9'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:21.591: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '4'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:22.391: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '3'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:22.983: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '0'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:27.987: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x818E

interface Serial0/0/0:15
 no ip address
 isdn switch-type primary-net5
 isdn overlap-receiving T302 5000
 isdn network-failure-cause 16
 isdn protocol-emulate network
 isdn incoming-voice voice
 isdn send-alerting
 no cdp enable
 
 
 
thanks
Brian

________________________________

From: Fedorov, Konstantin [mailto:kfedor at amt.ru] 
Sent: marted? 20 giugno 2006 16.54
To: Brian Turnbow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Dial collection



Hi, Brian.

 

ISDN Info - so you PBX send digits over overlap.

 

Whai is you scenario ? Two state dialing ??

 

Can you send a snip  of

Debug isdn q931 

 

And D channel configuration.

-----------------------
Sincerely Yours,
Konstantin Fedorov

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Turnbow
Sent: Tuesday, June 20, 2006 6:45 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Dial collection

 

 

Hello all, 

I have a 2821 with a E1 multiflex card connected to a PBX. 
When analog lines of the pbx dial out I receive both the in band dialed digits and isdn info messages from the pbx. 
This causes the router to send out in h323 the incorrect numer . For example  1234 gets sent out as 11223344 
I have been looking for a way to disable the in band digit collection, but have yet to find out how. 
Can any one point me in the right direction ? 

Thanks 
Brian 

-------------- next part --------------
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From kfedor at amt.ru  Tue Jun 20 11:11:59 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Tue, 20 Jun 2006 19:11:59 +0400
Subject: [cisco-voip] Dial collection
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC707BE4756@amthqex01.amt.ru>

Hi,

 

The best way disable overlap send on the PBX.

 

But try remove 

isdn overlap-receiving T302 5000

 

-----------------------
Sincerely Yours,
Konstantin Fedorov

  _____  

From: Brian Turnbow [mailto:b.turnbow at twt.it] 
Sent: Tuesday, June 20, 2006 7:02 PM
To: Fedorov, Konstantin; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Dial collection

 

Hi 

yes and with two state dialing , The end users digits a code to access the voip line and receives tone fron the router ( no direct dial in) 

Below is an ISDN setup, and everything looks ok. 

However on the outboud h323 i See  00223366552299443300 as the called number. 

ISDN phones send correct on the pbx, it is only the analog lines that have this problem. 

 

 

*Jun 20 14:59:16.151: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref = 0x018E
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98385
                Exclusive, Channel 5
        Progress Ind i = 0x8083 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '219'
                Plan:ISDN, Type:Unknown
        Low Layer Compat i = 0x9090A3
*Jun 20 14:59:16.159: ISDN Se0/0/0:15 Q931: TX -> SETUP_ACK pd = 8  callref = 0x818E
        Channel ID i = 0xA98385
                Exclusive, Channel 5
*Jun 20 14:59:16.991: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '0'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:17.779: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '2'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:18.495: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '3'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:19.199: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '6'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:19.795: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '5'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:20.387: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '2'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:21.087: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '9'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:21.591: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '4'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:22.391: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '3'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:22.983: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8  callref = 0x018E
        Called Party Number i = 0x81, '0'
                Plan:ISDN, Type:Unknown
*Jun 20 14:59:27.987: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x818E

interface Serial0/0/0:15
 no ip address
 isdn switch-type primary-net5
 isdn overlap-receiving T302 5000
 isdn network-failure-cause 16
 isdn protocol-emulate network
 isdn incoming-voice voice
 isdn send-alerting
 no cdp enable

 

 

 

thanks

Brian

 

  _____  

From: Fedorov, Konstantin [mailto:kfedor at amt.ru] 
Sent: marted? 20 giugno 2006 16.54
To: Brian Turnbow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Dial collection

Hi, Brian.

 

ISDN Info - so you PBX send digits over overlap.

 

Whai is you scenario ? Two state dialing ??

 

Can you send a snip  of

Debug isdn q931 

 

And D channel configuration.

-----------------------
Sincerely Yours,
Konstantin Fedorov

  _____  

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Brian Turnbow
Sent: Tuesday, June 20, 2006 6:45 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Dial collection

 

 

Hello all, 

I have a 2821 with a E1 multiflex card connected to a PBX. 
When analog lines of the pbx dial out I receive both the in band dialed digits and isdn info messages from the pbx. 
This causes the router to send out in h323 the incorrect numer . For example  1234 gets sent out as 11223344 
I have been looking for a way to disable the in band digit collection, but have yet to find out how. 
Can any one point me in the right direction ? 

Thanks 
Brian 

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From rratliff at cisco.com  Tue Jun 20 11:14:13 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 20 Jun 2006 11:14:13 -0400
Subject: [cisco-voip] Dial collection
In-Reply-To: <8B5577ED1D5D2F4AA0E4651EC41C6DC707BE4756@amthqex01.amt.ru>
References: <8B5577ED1D5D2F4AA0E4651EC41C6DC707BE4756@amthqex01.amt.ru>
Message-ID: <1024A90A-0E6D-4E14-807D-C4983D6C82B3@cisco.com>

Disabling overlap receiving will probably break the digital pbx  
phones ability to dial though :)

You can try disabling dtmf-relay on the inbound pots dial-peer but  
this may not work.

-Ryan

On Jun 20, 2006, at 11:11 AM, Fedorov, Konstantin wrote:

Hi,



The best way disable overlap send on the PBX.



But try remove

isdn overlap-receiving T302 5000



-----------------------
Sincerely Yours,
Konstantin Fedorov

From: Brian Turnbow [mailto:b.turnbow at twt.it]
Sent: Tuesday, June 20, 2006 7:02 PM
To: Fedorov, Konstantin; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Dial collection



Hi

yes and with two state dialing , The end users digits a code to  
access the voip line and receives tone fron the router ( no direct  
dial in)

Below is an ISDN setup, and everything looks ok.

However on the outboud h323 i See  00223366552299443300 as the called  
number.

ISDN phones send correct on the pbx, it is only the analog lines that  
have this problem.





*Jun 20 14:59:16.151: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8   
callref = 0x018E
         Bearer Capability i = 0x9090A3
                 Standard = CCITT
                 Transer Capability = 3.1kHz Audio
                 Transfer Mode = Circuit
                 Transfer Rate = 64 kbit/s
         Channel ID i = 0xA98385
                 Exclusive, Channel 5
         Progress Ind i = 0x8083 - Origination address is non-ISDN
         Calling Party Number i = 0x0181, '219'
                 Plan:ISDN, Type:Unknown
         Low Layer Compat i = 0x9090A3
*Jun 20 14:59:16.159: ISDN Se0/0/0:15 Q931: TX -> SETUP_ACK pd = 8   
callref = 0x818E
         Channel ID i = 0xA98385
                 Exclusive, Channel 5
*Jun 20 14:59:16.991: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '0'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:17.779: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '2'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:18.495: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '3'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:19.199: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '6'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:19.795: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '5'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:20.387: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '2'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:21.087: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '9'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:21.591: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '4'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:22.391: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '3'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:22.983: ISDN Se0/0/0:15 Q931: RX <- INFORMATION pd = 8   
callref = 0x018E
         Called Party Number i = 0x81, '0'
                 Plan:ISDN, Type:Unknown
*Jun 20 14:59:27.987: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8   
callref = 0x818E

interface Serial0/0/0:15
  no ip address
  isdn switch-type primary-net5
  isdn overlap-receiving T302 5000
  isdn network-failure-cause 16
  isdn protocol-emulate network
  isdn incoming-voice voice
  isdn send-alerting
  no cdp enable







thanks

Brian



From: Fedorov, Konstantin [mailto:kfedor at amt.ru]
Sent: marted? 20 giugno 2006 16.54
To: Brian Turnbow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Dial collection

Hi, Brian.



ISDN Info ? so you PBX send digits over overlap.



Whai is you scenario ? Two state dialing ??



Can you send a snip  of

Debug isdn q931



And D channel configuration.

-----------------------
Sincerely Yours,
Konstantin Fedorov

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Brian Turnbow
Sent: Tuesday, June 20, 2006 6:45 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Dial collection





Hello all,

I have a 2821 with a E1 multiflex card connected to a PBX.
When analog lines of the pbx dial out I receive both the in band  
dialed digits and isdn info messages from the pbx.
This causes the router to send out in h323 the incorrect numer . For  
example  1234 gets sent out as 11223344
I have been looking for a way to disable the in band digit  
collection, but have yet to find out how.
Can any one point me in the right direction ?

Thanks
Brian

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



From kfedor at amt.ru  Tue Jun 20 11:22:55 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Tue, 20 Jun 2006 19:22:55 +0400
Subject: [cisco-voip] Dial collection
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC70648ECE0@amthqex01.amt.ru>

Yes, 
I also think about this, a little rash/
But dtmf-relay not pertain to the pots dial-peer.
 
How I understand ISDN open b-channel and phone work on the tone 
So GW receive digits over in-band and out-of-band.

Brian ? is this possible to send connect instead setup_ack

If yes you can configure following
Dial-peer voice xxx pots
Incoming called-number 219
Port 0/0/0:D
No direct-inward-dial


Or try remove Progress Ind i = 0x8083 - Origination address is non-ISDN
On the PBX


-----------------------
Sincerely Yours,
Konstantin Fedorov

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, June 20, 2006 7:14 PM
To: Fedorov, Konstantin
Cc: Brian Turnbow; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Dial collection

Disabling overlap receiving will probably break the digital pbx  
phones ability to dial though :)

You can try disabling dtmf-relay on the inbound pots dial-peer but  
this may not work.

-Ryan




From jcc at unc.edu  Tue Jun 20 11:22:39 2006
From: jcc at unc.edu (Justin Church)
Date: Tue, 20 Jun 2006 11:22:39 -0400
Subject: [cisco-voip] CM 5.0 SIP Woes
Message-ID: <449812BF.5050101@unc.edu>

Recently installed CM 5.0(3), and I'm not able to register any SIP 
devices, neither a 7940 or a 3rd party SIP endpoint.  On the 7940, I 
continually get a 'File not Found' on the Application FW load.  I have 
installed the cmterm-7940-7960-8.3.00-sip.cop SIP load.  The 
cmterm-7940-7960-sip-8.3-readme states that "The 8.3 firmware image 
within the installation program is named P0S3-08-3-00..."; however, when 
the image is inflated on the CM server, the .bin is called 
P003-08-3-00.bin, NOT P0S3-08-3-00.bin.  I've tried using both 
P0S3-08-3-00 and P003-08-3-00 as the name of the default SIP load for 
the 7940, but both produce 'File not Found' on the phones.  I can 
successfully retrieve P003-08-3-00.bin with a tftp client, so I'm not 
sure what the issue is.  For the 3rd party SIP endpoint, I'm trying to 
register the x-ten soft client.  From looking at the client traces, it 
appears that authentication is working; however, registration is 
rejected by the CM with a 'Line not configured' message, even though 
I've configured a line in the 3rd party SIP (Advanced) declaration of my 
endpoint.  Suggestions on either of these?

Thanks.

-jc

From Ruben.Montes at eu.didata.com  Tue Jun 20 11:27:52 2006
From: Ruben.Montes at eu.didata.com (Ruben Montes (EU))
Date: Tue, 20 Jun 2006 17:27:52 +0200
Subject: [cisco-voip] IPMA- Call transfer to the manager
References: <8B5577ED1D5D2F4AA0E4651EC41C6DC707BE4756@amthqex01.amt.ru>
Message-ID: <FA233791F398B848B978464507739E8F0C5D9D@EUBEBRUSVEX1.eu.didata.local>


Hi,

I am using IPMA in Proxy mode. When the manager's assistant receives a call that wasd detined to the manager from the proxy line and forwards it to the manager, the end user can see in his phone display the number of the proxy line.
Is it possible to change this behaviour?

Regards,

Ruben

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From Jbrittain at ci.hickory.nc.us  Tue Jun 20 11:43:00 2006
From: Jbrittain at ci.hickory.nc.us (Jeff Brittain)
Date: Tue, 20 Jun 2006 11:43:00 -0400
Subject: [cisco-voip] Phone resets after upgrade
Message-ID: <D51ACD88CC49D4499543831A1D17D93302E1AF@HKYEXCH.ci.hickory.nc.us>

We recently upgraded to CCM 4.1(3)sr3a.  We have 2 phones (that we know about!) that reset multiple times each day and drop the call.  One is a 7960 and one is a 7940.  We have talked to Cisco TAC multiple times, but are still having the same problem.  They both have multiple lines, but a lot of the other phones that work normally do as well!  We have applied the TAC recommended patches on CCM and the phones (firmware), we've capture packets and sent the data to them for analysis, etc.  These users are getting very frustrated since they never know when the phone will reset.  It seems to always be while the phone is active (during a call).  We have moved the phones into a different powered switch, deleted them and added them back, replaced the 7940 phone itself - but, it is still happening.  I'm glad it's only 2 phones, but why??
 
Does anyone have any ideas before these users become violent!!??
 
Thanks.
 
Jeff Brittain
City of Hickory, NC

 

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From lelio at uoguelph.ca  Tue Jun 20 11:56:56 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 20 Jun 2006 11:56:56 -0400
Subject: [cisco-voip] Phone resets after upgrade
References: <D51ACD88CC49D4499543831A1D17D93302E1AF@HKYEXCH.ci.hickory.nc.us>
Message-ID: <027101c69482$28c953e0$47196883@cfs.uoguelph.ca>

Have you checked that there is no IP address conflict? Also, what is the phone's perspective on why the reset happened? That is: Settings > Status > Network Statistics. On the third line you should see a code of sometype. Here's something I cut/paste a while ago. Not sure how up to date this list is.

a.. Failback-phone closed due failback to a higher priority Cisco CallManager. 
a.. Phone-Loading-phone closed to upgrade software. This message appears only if TFTP fails during software upgrade. 
a.. Phone-Keypad-phone closed due to a **#** reset. 
a.. Phone-Re-IP-phone closed due to a duplicate IP address condition. 
a.. Reset-Reset-phone closed due to receiving a Reset/Reset from Cisco CallManager administration. 
a.. Reset-Restart-phone closed due to receiving a Reset/Restart from Cisco CallManager administration. 
a.. All-CMs-Bad-phone detected a state in which all Cisco CallManager servers failed their keepalives. 
a.. Phone-Reg-Rej-phone closed due to receiving a registration reject. 
a.. Phone-Initialized-phone has not experienced a connection close since the hardware reset or since it was powered on. 

--------------------------------------------------------------------------------
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: Jeff Brittain 
  To: cisco-voip at puck.nether.net 
  Sent: Tuesday, June 20, 2006 11:43 AM
  Subject: [cisco-voip] Phone resets after upgrade


  We recently upgraded to CCM 4.1(3)sr3a.  We have 2 phones (that we know about!) that reset multiple times each day and drop the call.  One is a 7960 and one is a 7940.  We have talked to Cisco TAC multiple times, but are still having the same problem.  They both have multiple lines, but a lot of the other phones that work normally do as well!  We have applied the TAC recommended patches on CCM and the phones (firmware), we've capture packets and sent the data to them for analysis, etc.  These users are getting very frustrated since they never know when the phone will reset.  It seems to always be while the phone is active (during a call).  We have moved the phones into a different powered switch, deleted them and added them back, replaced the 7940 phone itself - but, it is still happening.  I'm glad it's only 2 phones, but why??

  Does anyone have any ideas before these users become violent!!??

  Thanks.

  Jeff Brittain
  City of Hickory, NC



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


  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From MLinsemier at apcapital.com  Tue Jun 20 12:02:22 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Tue, 20 Jun 2006 12:02:22 -0400
Subject: [cisco-voip] Phone resets after upgrade
Message-ID: <AB0A35F2486F81468971AA68558BE719040372CC@smieln01winm01.micoanet.com>

When you installed CCM 4.1(3)sr3a, did you install the latest device
pack as well?  The service releases install new phone firmware (that was
current at the date of the service pack) and this may have introduced
problems with the phone if they were upgraded / downgraded.  The most
current device pack is ciscocm.4-1-DevPack-28.2.exe.  I would consider
installing this and see if the problems are resolved.  You can also
selectively upgrade just your 7940/60 firmware as well. Check the
version on the phone of what you are running and download the
appropriate files on CCO.

 

As a side note, CCM 4.1(3)sr3b was released so you may want to look at
the bug that was fixed and determined whether you need to upgrade.

 

Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jeff Brittain
Sent: Tuesday, June 20, 2006 11:43 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Phone resets after upgrade

 

We recently upgraded to CCM 4.1(3)sr3a.  We have 2 phones (that we know
about!) that reset multiple times each day and drop the call.  One is a
7960 and one is a 7940.  We have talked to Cisco TAC multiple times, but
are still having the same problem.  They both have multiple lines, but a
lot of the other phones that work normally do as well!  We have applied
the TAC recommended patches on CCM and the phones (firmware), we've
capture packets and sent the data to them for analysis, etc.  These
users are getting very frustrated since they never know when the phone
will reset.  It seems to always be while the phone is active (during a
call).  We have moved the phones into a different powered switch,
deleted them and added them back, replaced the 7940 phone itself - but,
it is still happening.  I'm glad it's only 2 phones, but why??

 

Does anyone have any ideas before these users become violent!!??

 

Thanks.

 

Jeff Brittain

City of Hickory, NC

	 


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

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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 
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From tomlemay at adelphia.net  Tue Jun 20 12:42:00 2006
From: tomlemay at adelphia.net (tomlemay at adelphia.net)
Date: Tue, 20 Jun 2006 12:42:00 -0400
Subject: [cisco-voip] 7935\7936 IP Conf Unit & XML Corp Directory
Message-ID: <3136433.1150821720534.JavaMail.root@web25>

Recently, we had a request to "merge or expand" two separate Call Manager directories from two separate clusters.  A search in the discussion forum on the CCO web page identified the XML script below to accomplish this request.  The only problem is the 7940\60\70g IP phones can all access both diretories from the directories button on the phone.  no problem.  But, users on the 7935\7936 conference units press the corporate directory button, key in a name and press the search button; they get "Server not found."  Can anyone with XML experience look atthe script below and explain what error is present or code that is missing which would prevent the 7935\36 units from finding the server?  Thanks in advance, Tom   

"I have done this in our environment. It is fairly easy. Just find the xmldirectory.asp file on your Call Manager and add a Menuitem to it. Here is what mine looks like. You won't need any type of authentication, just the URL. In this example I change the name "Corporate Directory" to "Balt Corporate Directory" and added "NY Corporate Directory" 

%> 
<!-- #include file="_top.asp" --> 
<% 

function getBaseURL() 
{ 
var scriptName = Request.ServerVariables("SCRIPT_NAME") ; 
var endloc = String(scriptName).lastIndexOf("/") ; 
var baseURL = "http://" + 
String(Request.ServerVariables("SERVER_NAME")) + 
String(scriptName).substr(0, endloc + 1) ; 
return baseURL ; 
} 
try 
{ 
var name = String(Request.QueryString("Name").Item); 
Response.ContentType = "text/xml"; 
Response.Buffer = true; 
//Response.Write("<CiscoIPPhoneMenu>\r\n<Prompt>Select a directory</Prompt>\r\n<MenuItem>\r\n<Name>Balt Corporate Directory</Name>\r\n<URL>" + getBaseURL() + "xmldirectoryinput.asp</URL>\r\n</MenuItem>\r\n</CiscoIPPhoneMenu>"); 
%> 
<CiscoIPPhoneMenu> 
<Prompt><% = outputString( dictionary.xmldirectoryMsgPagePrompt, "dictionary.xmldirectoryMsgPagePrompt" ) %></Prompt> 
<MenuItem> 
<Name><% = outputString( dictionary.lblCorporateDirectory, "dictionary.lblCorporateDirectory" ) %></Name> 
<URL><% = getBaseURL() %>xmldirectoryinput.asp</URL> 
</MenuItem> 
<MenuItem> 
<Name>NY Corporate Directory</Name> 
<URL>http://10.10.10.10/CCMCIP/xmldirectoryinput.asp</URL> 
</MenuItem> 

</CiscoIPPhoneMenu> 
<% 
} 
catch (err) 
{ 
Response.Clear(); 
Response.Write(dictionary.msgGeneralError + " ( "+ name + " )"); 
Response.AppendToLog(name + ":" + err.number + ":" + err.description); 
} 
Response.Flush(); 
%>" 



From janderson at enventis.com  Tue Jun 20 12:50:06 2006
From: janderson at enventis.com (Jeff Anderson)
Date: Tue, 20 Jun 2006 11:50:06 -0500
Subject: [cisco-voip] VG224 / SCCP / Modem
Message-ID: <20060620115006658.00000003156@janderson-3948>

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From b.turnbow at twt.it  Tue Jun 20 12:56:02 2006
From: b.turnbow at twt.it (Brian Turnbow)
Date: Tue, 20 Jun 2006 18:56:02 +0200
Subject: [cisco-voip] Dial collection
In-Reply-To: <8B5577ED1D5D2F4AA0E4651EC41C6DC70648ECE0@amthqex01.amt.ru>
Message-ID: <52964DBFB59FA34F98E96921BE21C90601465A1B@twt-exch.milano.twt>

Hi 
Thanks for the replies
I can't change anything on the PBX unfortunately, the company that manages it doesn't even 
Know the difference between enblock and overlap.....
The in and out of band signalling is the killer. We use some Patton gear for BRI lines and they 
Have a great little command that diasables inband dial collection on digital lines.
I was hoping that cisco had something similar , but I guess I'm SOL. 
If I send out a connect instead of setup_ack I think I'll break the Digital calls...

Brian

-----Original Message-----
From: Fedorov, Konstantin [mailto:kfedor at amt.ru] 
Sent: marted? 20 giugno 2006 17.23
To: Ryan Ratliff
Cc: Brian Turnbow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Dial collection

Yes, 
I also think about this, a little rash/
But dtmf-relay not pertain to the pots dial-peer.
 
How I understand ISDN open b-channel and phone work on the tone 
So GW receive digits over in-band and out-of-band.

Brian ? is this possible to send connect instead setup_ack

If yes you can configure following
Dial-peer voice xxx pots
Incoming called-number 219
Port 0/0/0:D
No direct-inward-dial


Or try remove Progress Ind i = 0x8083 - Origination address is non-ISDN
On the PBX


-----------------------
Sincerely Yours,
Konstantin Fedorov

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Tuesday, June 20, 2006 7:14 PM
To: Fedorov, Konstantin
Cc: Brian Turnbow; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Dial collection

Disabling overlap receiving will probably break the digital pbx  
phones ability to dial though :)

You can try disabling dtmf-relay on the inbound pots dial-peer but  
this may not work.

-Ryan




From rratliff at cisco.com  Tue Jun 20 12:58:20 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 20 Jun 2006 12:58:20 -0400
Subject: [cisco-voip] Dial collection
In-Reply-To: <52964DBFB59FA34F98E96921BE21C90601465A1B@twt-exch.milano.twt>
References: <52964DBFB59FA34F98E96921BE21C90601465A1B@twt-exch.milano.twt>
Message-ID: <04570BC5-28B4-415D-BA87-36E0B2E07CF8@cisco.com>

Have you opened a TAC SR yet?  None of us TAC folks on this list are  
gateway experts so there very well may be a way to do it.

-Ryan

On Jun 20, 2006, at 12:56 PM, Brian Turnbow wrote:

Hi
Thanks for the replies
I can't change anything on the PBX unfortunately, the company that  
manages it doesn't even
Know the difference between enblock and overlap.....
The in and out of band signalling is the killer. We use some Patton  
gear for BRI lines and they
Have a great little command that diasables inband dial collection on  
digital lines.
I was hoping that cisco had something similar , but I guess I'm SOL.
If I send out a connect instead of setup_ack I think I'll break the  
Digital calls...

Brian

-----Original Message-----
From: Fedorov, Konstantin [mailto:kfedor at amt.ru]
Sent: marted? 20 giugno 2006 17.23
To: Ryan Ratliff
Cc: Brian Turnbow; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Dial collection

Yes,
I also think about this, a little rash/
But dtmf-relay not pertain to the pots dial-peer.

How I understand ISDN open b-channel and phone work on the tone
So GW receive digits over in-band and out-of-band.

Brian ? is this possible to send connect instead setup_ack

If yes you can configure following
Dial-peer voice xxx pots
Incoming called-number 219
Port 0/0/0:D
No direct-inward-dial


Or try remove Progress Ind i = 0x8083 - Origination address is non-ISDN
On the PBX


-----------------------
Sincerely Yours,
Konstantin Fedorov

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Tuesday, June 20, 2006 7:14 PM
To: Fedorov, Konstantin
Cc: Brian Turnbow; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Dial collection

Disabling overlap receiving will probably break the digital pbx
phones ability to dial though :)

You can try disabling dtmf-relay on the inbound pots dial-peer but
this may not work.

-Ryan



From jsteinberg at gmail.com  Tue Jun 20 13:30:25 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 20 Jun 2006 13:30:25 -0400
Subject: [cisco-voip] Phone resets after upgrade
In-Reply-To: <AB0A35F2486F81468971AA68558BE719040372CC@smieln01winm01.micoanet.com>
References: <AB0A35F2486F81468971AA68558BE719040372CC@smieln01winm01.micoanet.com>
Message-ID: <f1b65f880606201030t285316f1hcbe15193d390e0a9@mail.gmail.com>

If it's just happening to two phones, just as a test I would try
putting those two user's phones on an external power brick to see if
that clears things up.

Justin



On 6/20/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
> When you installed CCM 4.1(3)sr3a, did you install the latest device pack as
> well?  The service releases install new phone firmware (that was current at
> the date of the service pack) and this may have introduced problems with the
> phone if they were upgraded / downgraded.  The most current device pack is
> ciscocm.4-1-DevPack-28.2.exe.  I would consider installing this and see if
> the problems are resolved.  You can also selectively upgrade just your
> 7940/60 firmware as well. Check the version on the phone of what you are
> running and download the appropriate files on CCO.
>
>
>
> As a side note, CCM 4.1(3)sr3b was released so you may want to look at the
> bug that was fixed and determined whether you need to upgrade.
>
>
>
> Matt
>
>
> ________________________________
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Jeff Brittain
>
> Sent: Tuesday, June 20, 2006 11:43 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Phone resets after upgrade
>
>
>
> Sent: Tuesday, June 20, 2006 11:43 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Phone resets after upgrade
>
>
>
>
>
>
>
> We recently upgraded to CCM 4.1(3)sr3a.  We have 2 phones (that we know
> about!) that reset multiple times each day and drop the call.  One is a 7960
> and one is a 7940.  We have talked to Cisco TAC multiple times, but are
> still having the same problem.  They both have multiple lines, but a lot of
> the other phones that work normally do as well!  We have applied the TAC
> recommended patches on CCM and the phones (firmware), we've capture packets
> and sent the data to them for analysis, etc.  These users are getting very
> frustrated since they never know when the phone will reset.  It seems to
> always be while the phone is active (during a call).  We have moved the
> phones into a different powered switch, deleted them and added them back,
> replaced the 7940 phone itself - but, it is still happening.  I'm glad it's
> only 2 phones, but why??
>
>
>
>
>
> Does anyone have any ideas before these users become violent!!??
>
>
>
>
>
> Thanks.
>
>
>
>
>
> Jeff Brittain
>
>
> City of Hickory, NC
>
>
>
>
>
> ________________________________
> ________________________________
>
>
>
>
> CONFIDENTIALITY STATEMENT
>
> This communication and any attachments are CONFIDENTIAL and may be protected
> by one or more legal privileges. It is intended solely for the use of the
> addressee identified above. If you are not the intended recipient, any use,
> disclosure, copying or distribution of this communication is UNAUTHORIZED.
> Neither this information block, the typed name of the sender, nor anything
> else in this message is intended to constitute an electronic signature
> unless a specific statement to the contrary is included in this message. If
> you have received this communication in error, please immediately contact me
> and delete this communication from your computer. Thank you.
>
> ________________________________
> ________________________________
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From MLinsemier at apcapital.com  Tue Jun 20 13:55:28 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Tue, 20 Jun 2006 13:55:28 -0400
Subject: [cisco-voip] Phone resets after upgrade
Message-ID: <AB0A35F2486F81468971AA68558BE7190403734D@smieln01winm01.micoanet.com>

You could also move swap the phones out with empty cubicle phones (if
you have any) and see if the problem goes away.

Matt

-----Original Message-----
From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Tuesday, June 20, 2006 1:30 PM
To: Linsemier, Matthew
Cc: Jeff Brittain; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Phone resets after upgrade

If it's just happening to two phones, just as a test I would try
putting those two user's phones on an external power brick to see if
that clears things up.

Justin



On 6/20/06, Linsemier, Matthew <MLinsemier at apcapital.com> wrote:
>
>
>
> When you installed CCM 4.1(3)sr3a, did you install the latest device
pack as
> well?  The service releases install new phone firmware (that was
current at
> the date of the service pack) and this may have introduced problems
with the
> phone if they were upgraded / downgraded.  The most current device
pack is
> ciscocm.4-1-DevPack-28.2.exe.  I would consider installing this and
see if
> the problems are resolved.  You can also selectively upgrade just your
> 7940/60 firmware as well. Check the version on the phone of what you
are
> running and download the appropriate files on CCO.
>
>
>
> As a side note, CCM 4.1(3)sr3b was released so you may want to look at
the
> bug that was fixed and determined whether you need to upgrade.
>
>
>
> Matt
>
>
> ________________________________
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Jeff Brittain
>
> Sent: Tuesday, June 20, 2006 11:43 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Phone resets after upgrade
>
>
>
> Sent: Tuesday, June 20, 2006 11:43 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Phone resets after upgrade
>
>
>
>
>
>
>
> We recently upgraded to CCM 4.1(3)sr3a.  We have 2 phones (that we
know
> about!) that reset multiple times each day and drop the call.  One is
a 7960
> and one is a 7940.  We have talked to Cisco TAC multiple times, but
are
> still having the same problem.  They both have multiple lines, but a
lot of
> the other phones that work normally do as well!  We have applied the
TAC
> recommended patches on CCM and the phones (firmware), we've capture
packets
> and sent the data to them for analysis, etc.  These users are getting
very
> frustrated since they never know when the phone will reset.  It seems
to
> always be while the phone is active (during a call).  We have moved
the
> phones into a different powered switch, deleted them and added them
back,
> replaced the 7940 phone itself - but, it is still happening.  I'm glad
it's
> only 2 phones, but why??
>
>
>
>
>
> Does anyone have any ideas before these users become violent!!??
>
>
>
>
>
> Thanks.
>
>
>
>
>
> Jeff Brittain
>
>
> City of Hickory, NC
>
>
>
>
>
> ________________________________
> ________________________________
>
>
>
>
> CONFIDENTIALITY STATEMENT
>
> This communication and any attachments are CONFIDENTIAL and may be
protected
> by one or more legal privileges. It is intended solely for the use of
the
> addressee identified above. If you are not the intended recipient, any
use,
> disclosure, copying or distribution of this communication is
UNAUTHORIZED.
> Neither this information block, the typed name of the sender, nor
anything
> else in this message is intended to constitute an electronic signature
> unless a specific statement to the contrary is included in this
message. If
> you have received this communication in error, please immediately
contact me
> and delete this communication from your computer. Thank you.
>
> ________________________________
> ________________________________
>
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
--------------------------------------------------------------------------------

CONFIDENTIALITY STATEMENT

This communication and any attachments are CONFIDENTIAL and may be 
protected by one or more legal privileges. It is intended solely
for the use of the addressee identified above. If you are not 
the intended recipient, any use, disclosure, copying or 
distribution of this communication is UNAUTHORIZED. Neither this 
information block, the typed name of the sender, nor anything 
else in this message is intended to constitute an electronic 
signature unless a specific statement to the contrary is 
included in this message. If you have received this communication in 
error, please immediately contact me and delete this communication from
your computer. Thank you.

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



From Johnk at fiu.edu  Tue Jun 20 14:28:45 2006
From: Johnk at fiu.edu (John Kakalik)
Date: Tue, 20 Jun 2006 14:28:45 -0400
Subject: [cisco-voip] John Kakalik/UTS/FIU is out of the office.
Message-ID: <OFA05AAF8F.5C42A614-ON85257193.006582A4-85257193.006582A4@fiu.edu>





I will be out of the office starting  06/20/2006 and will not return until
06/28/2006.

I will do my best  to respond to your message when I return.  Be Well!
Sincerely,

John

Take a moment to thank those that serve and protect us and to pray for
their safety.   God Bless America!


From jgrace at digitelusa.net  Tue Jun 20 14:24:29 2006
From: jgrace at digitelusa.net (James Grace)
Date: Tue, 20 Jun 2006 14:24:29 -0400
Subject: [cisco-voip] unable to create pri timeslot not enough resource
Message-ID: <200606201436531.SM01452@3729434770>

Have a 2811 with 48 and 16 dsp chip.  Configuring as a mgcp with srst
fallback.. for some reason we are not able add the t1 pri config   says not
enough resource.   Whats the cmd to show how many resources im using for
what.  This router has max dn 150 and max ephone 30 for the fallback setup 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From tednugent69 at yahoo.com  Tue Jun 20 14:55:29 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Tue, 20 Jun 2006 11:55:29 -0700 (PDT)
Subject: [cisco-voip] unable to create pri timeslot not enough resource
In-Reply-To: <200606201436531.SM01452@3729434770>
Message-ID: <20060620185529.66021.qmail@web36108.mail.mud.yahoo.com>

Try network-clock-participate wic x

and
I don't have an ISR in front of me but you can try
"sho voice dsp ?" and "sho diag" to see howmany PVDMs
you have installed.

--- James Grace <jgrace at digitelusa.net> wrote:

> Have a 2811 with 48 and 16 dsp chip.  Configuring as
> a mgcp with srst
> fallback.. for some reason we are not able add the
> t1 pri config   says not
> enough resource.   Whats the cmd to show how many
> resources im using for
> what.  This router has max dn 150 and max ephone 30
> for the fallback setup 
> 
>  
> 
> James D. Grace 
> 
> CCNP CCNA MCSE MCDBA
> 
> Sr. System Engineer / Professional Svc.
> 
> Digitel Corporation
> 
>  
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jim at tgasolutions.com  Tue Jun 20 15:21:22 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Tue, 20 Jun 2006 15:21:22 -0400
Subject: [cisco-voip] 7940 vs 41 time
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC15E9EF@exchange.TGASolutions.local>

Has anyone seen an issue where 7941s do not take daylight savings time
when it is configured on the router?
The 7940's 7912's and all other phones are correct.

Thanks,
Jim


From phobie at axfr.org  Tue Jun 20 17:02:52 2006
From: phobie at axfr.org (riki)
Date: Tue, 20 Jun 2006 23:02:52 +0200
Subject: [cisco-voip] John Kakalik/UTS/FIU is out of the office.
In-Reply-To: <OFA05AAF8F.5C42A614-ON85257193.006582A4-85257193.006582A4@fiu.edu>
References: <OFA05AAF8F.5C42A614-ON85257193.006582A4-85257193.006582A4@fiu.edu>
Message-ID: <4498627C.4050001@axfr.org>

Is your house empy too? :)

John Kakalik wrote:
> 
> 
> 
> I will be out of the office starting  06/20/2006 and will not return until
> 06/28/2006.
> 
> I will do my best  to respond to your message when I return.  Be Well!
> Sincerely,
> 
> John
> 
> Take a moment to thank those that serve and protect us and to pray for
> their safety.   God Bless America!
> 
> _______________________________________________
> 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  Tue Jun 20 18:29:37 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 20 Jun 2006 18:29:37 -0400
Subject: [cisco-voip] VG224 / SCCP / Modem
References: <20060620115006658.00000003156@janderson-3948>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E76A@USNAEXCH.na.didata.local>

You can plug a cheap analog phone into that VG224 and after the same x minutes get disconnected? Verify you can reproduce the problem.
 
Open a TAC case, sounds like you need some debugs, SDL traces, takes -jason

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
Sent: Tue 6/20/2006 12:50 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] VG224 / SCCP / Modem


Question:
 
I have a client who is using a VG224 that is configured to use SCCP on CCM 4.1. The majority of the ports are used for modems but there are devices that need skinny features as well. The problem is the modem calls fail after a random amount of time. It could be anywhere from 5 - 30 minutes. That same modem plugged directly into a 1FB will have no reliability issues.
 
I have searched Cisco's website and their forums looking for some configuration guidelines for modem usage and found nothing. Does anyone have any advice to offer?
 
Thanks,
Jeff Anderson 


 



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Disclaimer:

This e-mail communication and any attachments may contain
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From woody at pch.net  Tue Jun 20 18:44:03 2006
From: woody at pch.net (Bill Woodcock)
Date: Tue, 20 Jun 2006 15:44:03 -0700 (PDT)
Subject: [cisco-voip] Background Image on 7970 - How to extract it from
 the phone
In-Reply-To: <D0AEBED8692AB14791F4FA1CE121CC6D031832F0@PEPSI.uwec.edu>
References: <D0AEBED8692AB14791F4FA1CE121CC6D031832F0@PEPSI.uwec.edu>
Message-ID: <Pine.SOC.4.61.0606201543420.26566@paixhost.pch.net>

    > Is there any way to retrieve a background image from a 7970?  It does
    > not exist on tftp and only exists as a background image on one of our
    > 7970's.  We are still looking for the original but will probably not be
    > able to locate it.

They're not very big, so in the worst case, you could take a digital photo 
of it, and recreate it in photoshop.

                                -Bill


From scall at metreos.com  Tue Jun 20 19:49:55 2006
From: scall at metreos.com (Seth Call)
Date: Tue, 20 Jun 2006 18:49:55 -0500
Subject: [cisco-voip] Background Image on 7970 - How to extract it from
	the phone
Message-ID: <6EE053013F00084096AF0F02CFDB88B8769F7A@securemail.metreos.com>

http://ipphoneIP/CGI/Screenshot   You have to have a user associated
with the phone in ccmadmin--you will be prompted for this user's
credentials when you go to this URL.

I've noticed browsers treat the PNG incorrectly sometimes.  For
instance, this will spit out a bunch of binary data to Firefox, and so
you have to actually View Source -> Copy All, and paste into a new file
called whatever.png.  I assume that's because there is not the correct
header on the image, but I'm not sure.

Regards,
Seth Call
Metreos Corporation
(512) 687-2006


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Bill Woodcock
Sent: Tuesday, June 20, 2006 5:44 PM
To: Lund, John C.
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Background Image on 7970 - How to extract it
from the phone

    > Is there any way to retrieve a background image from a 7970?  It
does
    > not exist on tftp and only exists as a background image on one of
our
    > 7970's.  We are still looking for the original but will probably
not be
    > able to locate it.

They're not very big, so in the worst case, you could take a digital
photo 
of it, and recreate it in photoshop.

                                -Bill

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



From jburton at NETechCorp.com  Tue Jun 20 20:30:42 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Tue, 20 Jun 2006 20:30:42 -0400
Subject: [cisco-voip] Unity integration with Lotus Domino
In-Reply-To: <FA233791F398B848B978464507739E8F0C5D9C@EUBEBRUSVEX1.eu.didata.local>
References: <FA233791F398B848B978464507739E8F0C5D9C@EUBEBRUSVEX1.eu.didata.local>
Message-ID: <742E6B51-A582-40B6-92D8-35909CC73775@mimectl>

Just did one this week.  What would like to know?



From: Ruben Montes (EU)
Sent: Tue 6/20/2006 10:24 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity integration with Lotus Domino




Hello,

has anyone done successfully an integration of Cisco Unity and Lotus Domino for Unified Messaging? I have some questions regarding the changes that should be done in the Domino server.

Thanks in advance,

Ruben 
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From jburton at NETechCorp.com  Tue Jun 20 20:36:59 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Tue, 20 Jun 2006 20:36:59 -0400
Subject: [cisco-voip] PRI line option question-
In-Reply-To: <F3F50B61-A22B-444A-AD2F-CCCF9EB5FC54@cisco.com>
References: <050944EE48FF384F9C663F7048C47DA704143122@webmail.asheville.k12.nc.us>, 
	<F3F50B61-A22B-444A-AD2F-CCCF9EB5FC54@cisco.com>
Message-ID: <54FE69CA-C2C3-4C0F-AB5B-39C05C26F3A8@mimectl>

H.323 doesn't support calling name for PRI.  If you do a debug you can see the calling name come in the facility ie, but CM doesn't support this.



From: Ryan Ratliff
Sent: Tue 6/20/2006 10:52 AM
To: Tim Reimers
Cc: Kelly Looney; cisco-voip at puck.nether.net; Alan Surrette
Subject: Re: [cisco-voip] PRI line option question-


You need to get a 'debug isdn q931' for an inbound call to see where  
the name is appearing and why CM isn't showing it.

-Ryan

On Jun 20, 2006, at 10:42 AM, Tim Reimers wrote:


Hi everyone-

I wanted to ask a question -


Our ISDN PRI provider has started billing us (belatedly) for an  
option they  have provisioned on our line some time ago, and not told  
us.

It's a caller ID option called "Calling party number to name"

Before I decide that I don't want this, and they should take it off--

Does Cisco CCM support this?

Here's the details:

Cisco 2651XM running c2600-ipvoice-mz.123-8.T3
with two PRIs attached, 30 out of 47 channels live.
The gateway is in H.323 mode, with the appropriate (working fine)  
config in CCM.

when you get an outside call, you get just numbers, not names..

So Bell is providing us names with our Caller ID, but I guess our  
equipment is either incapable of showing that, or it's ignoring it.


Any thoughts?

Can I configure this to work, so that I can see what info -is-  
available from various people?


thanks, Tim

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

_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip
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From BDKusa at sgh.com  Tue Jun 20 21:05:29 2006
From: BDKusa at sgh.com (Ben D. Kusa)
Date: Tue, 20 Jun 2006 21:05:29 -0400
Subject: [cisco-voip] PRI line option question-
References: <050944EE48FF384F9C663F7048C47DA704143122@webmail.asheville.k12.nc.us>,
	<F3F50B61-A22B-444A-AD2F-CCCF9EB5FC54@cisco.com>
	<54FE69CA-C2C3-4C0F-AB5B-39C05C26F3A8@mimectl>
Message-ID: <8A9E98B05F39B14E82E546ECF9A81969012181A1@exchange.sgh.com>

I think h323 does support naming if it comes via display ie. Maybe telco
can change it, but I have never had luck with that.

 

 

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Tuesday, June 20, 2006 8:37 PM
To: Ryan Ratliff; Tim Reimers
Cc: Kelly Looney; cisco-voip at puck.nether.net; Alan Surrette
Subject: Re: [cisco-voip] PRI line option question-

 

H.323 doesn't support calling name for PRI.  If you do a debug you can
see the calling name come in the facility ie, but CM doesn't support
this.

 

________________________________

From: Ryan Ratliff
Sent: Tue 6/20/2006 10:52 AM
To: Tim Reimers
Cc: Kelly Looney; cisco-voip at puck.nether.net; Alan Surrette
Subject: Re: [cisco-voip] PRI line option question-

You need to get a 'debug isdn q931' for an inbound call to see where  
the name is appearing and why CM isn't showing it.
 
-Ryan
 
On Jun 20, 2006, at 10:42 AM, Tim Reimers wrote:
 
 
Hi everyone-
 
I wanted to ask a question -
 
 
Our ISDN PRI provider has started billing us (belatedly) for an  
option they  have provisioned on our line some time ago, and not told  
us.
 
It's a caller ID option called "Calling party number to name"
 
Before I decide that I don't want this, and they should take it off--
 
Does Cisco CCM support this?
 
Here's the details:
 
Cisco 2651XM running c2600-ipvoice-mz.123-8.T3
with two PRIs attached, 30 out of 47 channels live.
The gateway is in H.323 mode, with the appropriate (working fine)  
config in CCM.
 
when you get an outside call, you get just numbers, not names..
 
So Bell is providing us names with our Caller ID, but I guess our  
equipment is either incapable of showing that, or it's ignoring it.
 
 
Any thoughts?
 
Can I configure this to work, so that I can see what info -is-  
available from various people?
 
 
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 cisco at dignans.com  Tue Jun 20 22:01:10 2006
From: cisco at dignans.com (Andrew Dignan)
Date: Tue, 20 Jun 2006 21:01:10 -0500 (CDT)
Subject: [cisco-voip] VG224 / SCCP / Modem
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F013E76A@USNAEXCH.na.didata.local>
References: <20060620115006658.00000003156@janderson-3948>
	<C1FE15183DA37645BC0633BC604E44F013E76A@USNAEXCH.na.didata.local>
Message-ID: <5973.24.234.204.56.1150855270.squirrel@www.dignans.com>

The VG224 using the SCCP image does not support fax/modems:

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/prod_release_note09186a008041df56.html

------------------------------------------------
?Fax and modems are not supported on FXS ports with SCCP enabled. FXS
ports with either fax or modems attached should be configured in H.323 or
SIP mode.
-------------------------------------------------

andy dignan

> You can plug a cheap analog phone into that VG224 and after the same x
> minutes get disconnected? Verify you can reproduce the problem.
>
> Open a TAC case, sounds like you need some debugs, SDL traces, takes
> -jason
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
> Sent: Tue 6/20/2006 12:50 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] VG224 / SCCP / Modem
>
>
> Question:
>
> I have a client who is using a VG224 that is configured to use SCCP on CCM
> 4.1. The majority of the ports are used for modems but there are devices
> that need skinny features as well. The problem is the modem calls fail
> after a random amount of time. It could be anywhere from 5 - 30 minutes.
> That same modem plugged directly into a 1FB will have no reliability
> issues.
>
> I have searched Cisco's website and their forums looking for some
> configuration guidelines for modem usage and found nothing. Does anyone
> have any advice to offer?
>
> Thanks,
> Jeff Anderson
>
>
>
>
>
>
> -----------------------------------------
> 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 jim at tgasolutions.com  Tue Jun 20 23:31:08 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Tue, 20 Jun 2006 23:31:08 -0400
Subject: [cisco-voip] 7940 vs 41 time
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC15EA2E@exchange.TGASolutions.local>

Ok.. Found it...

The CME 4 docs don't say it, but the 41 and 61 work the same as the
7970/1 in regards to the time-zone setting in the telephony service...

 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jim McBurnett
Sent: Tuesday, June 20, 2006 3:21 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7940 vs 41 time

Has anyone seen an issue where 7941s do not take daylight savings time
when it is configured on the router?
The 7940's 7912's and all other phones are correct.

Thanks,
Jim

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


From erickbe at yahoo.com  Tue Jun 20 23:47:52 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Tue, 20 Jun 2006 20:47:52 -0700 (PDT)
Subject: [cisco-voip] Background Image on 7970 - How to extract it from
	the phone
In-Reply-To: <Pine.SOC.4.61.0606201543420.26566@paixhost.pch.net>
Message-ID: <20060621034752.54839.qmail@web37215.mail.mud.yahoo.com>

If you have a history of BARS backup files from your system, maybe you can search through the tar files for the image file.


----- Original Message ----
From: Bill Woodcock <woody at pch.net>
To: "Lund, John C." <LUNDJC at uwec.edu>
Cc: cisco-voip at puck.nether.net
Sent: Tuesday, June 20, 2006 5:44:03 PM
Subject: Re: [cisco-voip] Background Image on 7970 - How to extract it from the phone

    > Is there any way to retrieve a background image from a 7970?  It does
    > not exist on tftp and only exists as a background image on one of our
    > 7970's.  We are still looking for the original but will probably not be
    > able to locate it.

They're not very big, so in the worst case, you could take a digital photo 
of it, and recreate it in photoshop.

                                -Bill

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





From cm at netuse.de  Wed Jun 21 03:43:29 2006
From: cm at netuse.de (Christoph Moormann)
Date: Wed, 21 Jun 2006 09:43:29 +0200
Subject: [cisco-voip] IPCC Express + Database and MWI+SNMP Questions
Message-ID: <4498F8A1.2040003@netuse.de>

Hi,
does anyone know, what Oracle Database is supported by the current IPCC 
Express Version?

Besides, can anyone tell me whether it's possible to work against a 
mysql DB?

Another question. Can i query the state of the MWI on a given phone via 
SNMP?

If possible send me the corresponding OIDs.

Regards
Christoph
-- 
Christoph Moormann
NetUSE AG
Dr.-Hell-Strasse, 24107 Kiel, Federal Republic of Germany
Telefon: +49 431 2390-400 --  Telefax: +49 431 2390-499


From jfaulhaber at avodaq.com  Wed Jun 21 08:07:03 2006
From: jfaulhaber at avodaq.com (Jens Faulhaber)
Date: Wed, 21 Jun 2006 14:07:03 +0200
Subject: [cisco-voip] call preservation on 3rd Party SBC
Message-ID: <F666605263F0EA40BC95239B1ADE10BC0A845B@sta-exch-1.avodaq.com>

Hi all,

anyone have an idea if CallManager 4.1. SP3 in connection with NetPoint
SBC (IPtoIP GW) supports h323 call preservation?

Cheers

Jens

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From tim.reimers at asheville.k12.nc.us  Wed Jun 21 08:25:29 2006
From: tim.reimers at asheville.k12.nc.us (Tim Reimers)
Date: Wed, 21 Jun 2006 08:25:29 -0400
Subject: [cisco-voip] Attendant Console online/offline
In-Reply-To: <FE2D092F8B1516488B99342551AACFD41CEE9216@mlnusa-exch1.mlnusa.com>
Message-ID: <050944EE48FF384F9C663F7048C47DA70414319B@webmail.asheville.k12.nc.us>

I'm with you on this one--
 
have seen the same behaviour---
Does yours also not seem to be up to the moment on who's on the phone
and who is not?
 
Mine doesn't seem to always know if someone is on the phone-- even with
directory reload set to 2 seconds...
 
I never can tell when our helpdesk is on the phone or not...or my
Director...

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Olson, Jeffrey
Sent: Monday, June 19, 2006 4:46 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Attendant Console online/offline



I have attendant console installed on a few different receptionists PCs
and it doesn't seem to matter if it is in the online or offline state.
Both ways it seems to have the same functionality.  What am I missing
with this?  Is there a setting in call manager that I am missing?

 

Jeff

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders
Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
hereby notified that disclosing, copying, distributing, or taking any
action in reliance on the contents of this information is strictly
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From cpoper at hamilton.ca  Wed Jun 21 10:08:50 2006
From: cpoper at hamilton.ca (Poper, Cathy)
Date: Wed, 21 Jun 2006 10:08:50 -0400
Subject: [cisco-voip] IPCCx ???'s
Message-ID: <46262932FF67F74CA6D537C31A69698701242272@endeavour.InfoNetwork>

Hi Scott
 
What OS version were you running before you applied this service
release?
This service release only applied to OS version 2000.4.2 release train.
(We're running OS version 2000.2.7 (IPCC express 3.0(3a), CCM 3.3.(3)
sr4a - so this service release doesn't apply, and won't be installed.)
 
Taken from release notes, see the Caution notes:
  

Operating System Upgrade Service Release 2000.4.2sr7a
(win-OS-Upgrade-K9.2000-4-2sr7a.exe)

 

Release date: 09 June 2006

Document Revision 2

Microsoft provides monthly releases of security hotfixes on the 2nd
Tuesday of each month.  Cisco's monthly OS Service Release is scheduled
to post on the 3rd Tuesday of each month. 

Cisco will continue to test and release Microsoft hotfixes that meet our
criteria for Critical hotfixes in 1 business day.  Any applicable
critical hotfix released by Microsoft will be added to the Cisco IP
Telephony Operating System, SQL Server, Security Updates
<http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/osbios.h
tm>  document with and explanation about whether or not it is critical
for Cisco IP Telephony servers and when it will be released by Cisco.

Caution:  Do not apply this service release with OS version 2000.2.7,
2000.4.1 or previous OS release trains.  This service release is only
compatible with the OS 2000.4.2 release train.  You should apply this
service release to all servers in your cluster.  This installation
causes call-processing interruptions and requires a reboot.  Close all
programs before proceeding including Internet Explorer to avoid
conflicts with the software being installed and/or upgraded.

 
 

Cathy Poper 
City of Hamilton/IT Services 
Network Support Analyst 
55 York Blvd., 6th floor 
Hamilton, Ontario 
Canada 
L8L 3K7 
Tel: 905-546-2424 x5481 
Fax: 905-546-2333 
cpoper at hamilton.ca 

	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	Sent: Monday, June 19, 2006 12:43 PM
	To: Voll, Scott; cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] IPCCx ???'s
	
	
	I've referred to this document in the past. It seems to be the
same file name all the time since everytime I go back, it's always
updated. This one seems to be updated June 12, 2006. sr7a does not seem
to be on the list. :(
	 
	 
	
http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1225/c
cmigration_09186a00803d82f5.pdf
	

Cisco CRS 4.0(1) SR1 Solution Set (Sheet 1 of 2)CallManager

OS Release

Hardware

IP Phones for Cisco Agent Desktop

IP Phones for Cisco IP Phone Agent

4.1(2) - JTAPI 2.1(2.3)*

   SR1 - JTAPI 2.1(2.5)

4.1(3) - JTAPI 2.1(3.3)*- JTAPI 2.1(3.9)* SR2 - JTAPI 2.1(3.11)* SR3a -
JTAPI 2.1(3.14)

4.2(1) - JTAPI 2.2(1.5) SR1 - JTAPI 2.2(1.5)

OS 2000.2.5 SR6   SR7 SR8

OS 2000.2.6 SR1 SR2 SR3   SR4   SR5   SR6

OS 2000.2.7* SR1*   SR2*   SR3   SR6

   SR7* 

SR8*

OS 2000.4.11 

SR11 

SR21 SR31 SR41 SR51*OS 2000.4.1a1

OS 2000.4.2*

   SR1* SR2* SR3* SR4 SR5a 

SR6 SR7

	 
	
------------------------------------------------------------------------
--------
	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 <mailto:Scott.Voll at wesd.org>  
		To: cisco-voip at puck.nether.net 
		Sent: Monday, June 19, 2006 12:35 PM
		Subject: [cisco-voip] IPCCx ???'s


		I have IPCCx 4.0.1sr1build 55.  I upgraded my CM's and
IPCCx to OS ver 4.2.sr7a.

		 

		This morning when I came in the agents could not log in.
Opened a TAC case and they had me move the CRS 3.5 license files out and
then agents could log in. (the Licensing and resource manager service
failed type error)

		 

		Now I'm dealing with the supervisors can select a team.
Any idea how to fix it?  The Supervisors were all removed from the teams
so I added them back and clicked update, but I'm still having the same
problem.  This CM and IPCCx is integrated with AD 2003.

		 

		TAC says that 4.2.sr7a is not supported on IPCCx.  I'm
trying to find anywhere it says this other then the metrix.  No note of
this is in the release notes I could find.

		 

		Thanks

		 

		Scott

		
________________________________


		

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

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From Scott.Voll at wesd.org  Wed Jun 21 10:43:21 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 21 Jun 2006 07:43:21 -0700
Subject: [cisco-voip] IPCCx ???'s
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED4E6B@Apollo.wesd.org>

I was on 4.2.sr5 -> 4.2.sr7a

 

See my answer from earlier
https://puck.nether.net/pipermail/cisco-voip/2006-June/012960.html

 

Scott

 

________________________________

From: Poper, Cathy [mailto:cpoper at hamilton.ca] 
Sent: Wednesday, June 21, 2006 7:09 AM
To: Lelio Fulgenzi; Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] IPCCx ???'s

 

Hi Scott

 

What OS version were you running before you applied this service
release?

This service release only applied to OS version 2000.4.2 release train.

(We're running OS version 2000.2.7 (IPCC express 3.0(3a), CCM 3.3.(3)
sr4a - so this service release doesn't apply, and won't be installed.)

 

Taken from release notes, see the Caution notes:

  

Operating System Upgrade Service Release 2000.4.2sr7a
(win-OS-Upgrade-K9.2000-4-2sr7a.exe)

 

Release date: 09 June 2006

Document Revision 2

Microsoft provides monthly releases of security hotfixes on the 2nd
Tuesday of each month.  Cisco's monthly OS Service Release is scheduled
to post on the 3rd Tuesday of each month. 

Cisco will continue to test and release Microsoft hotfixes that meet our
criteria for Critical hotfixes in 1 business day.  Any applicable
critical hotfix released by Microsoft will be added to the Cisco IP
Telephony Operating System, SQL Server, Security Updates
<http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/osbios.h
tm>  document with and explanation about whether or not it is critical
for Cisco IP Telephony servers and when it will be released by Cisco.

Caution:  Do not apply this service release with OS version 2000.2.7,
2000.4.1 or previous OS release trains.  This service release is only
compatible with the OS 2000.4.2 release train.  You should apply this
service release to all servers in your cluster.  This installation
causes call-processing interruptions and requires a reboot.  Close all
programs before proceeding including Internet Explorer to avoid
conflicts with the software being installed and/or upgraded.

 

 

Cathy Poper 
City of Hamilton/IT Services 
Network Support Analyst 
55 York Blvd., 6th floor 
Hamilton, Ontario 
Canada 
L8L 3K7 
Tel: 905-546-2424 x5481 
Fax: 905-546-2333 
cpoper at hamilton.ca 

	-----Original Message-----
	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
	Sent: Monday, June 19, 2006 12:43 PM
	To: Voll, Scott; cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] IPCCx ???'s

	I've referred to this document in the past. It seems to be the
same file name all the time since everytime I go back, it's always
updated. This one seems to be updated June 12, 2006. sr7a does not seem
to be on the list. :(

	 

	 

	
http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1225/c
cmigration_09186a00803d82f5.pdf

Cisco CRS 4.0(1) SR1 Solution Set (Sheet 1 of 2)CallManager

OS Release

Hardware

IP Phones for Cisco Agent Desktop

IP Phones for Cisco IP Phone Agent

4.1(2) - JTAPI 2.1(2.3)*

   SR1 - JTAPI 2.1(2.5)

4.1(3) - JTAPI 2.1(3.3)*- JTAPI 2.1(3.9)* SR2 - JTAPI 2.1(3.11)* SR3a -
JTAPI 2.1(3.14)

4.2(1) - JTAPI 2.2(1.5) SR1 - JTAPI 2.2(1.5)

OS 2000.2.5 SR6   SR7 SR8

OS 2000.2.6 SR1 SR2 SR3   SR4   SR5   SR6

OS 2000.2.7* SR1*   SR2*   SR3   SR6

   SR7* 

SR8*

OS 2000.4.11 

SR11 

SR21 SR31 SR41 SR51*OS 2000.4.1a1

OS 2000.4.2*

   SR1* SR2* SR3* SR4 SR5a 

SR6 SR7

	 

	
------------------------------------------------------------------------
--------
	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 <mailto:Scott.Voll at wesd.org>  

		To: cisco-voip at puck.nether.net 

		Sent: Monday, June 19, 2006 12:35 PM

		Subject: [cisco-voip] IPCCx ???'s

		 

		I have IPCCx 4.0.1sr1build 55.  I upgraded my CM's and
IPCCx to OS ver 4.2.sr7a.

		 

		This morning when I came in the agents could not log in.
Opened a TAC case and they had me move the CRS 3.5 license files out and
then agents could log in. (the Licensing and resource manager service
failed type error)

		 

		Now I'm dealing with the supervisors can select a team.
Any idea how to fix it?  The Supervisors were all removed from the teams
so I added them back and clicked update, but I'm still having the same
problem.  This CM and IPCCx is integrated with AD 2003.

		 

		TAC says that 4.2.sr7a is not supported on IPCCx.  I'm
trying to find anywhere it says this other then the metrix.  No note of
this is in the release notes I could find.

		 

		Thanks

		 

		Scott

		
________________________________


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

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From sodonnell at CCSINET.com  Wed Jun 21 11:14:22 2006
From: sodonnell at CCSINET.com (Scott O'Donnell)
Date: Wed, 21 Jun 2006 11:14:22 -0400
Subject: [cisco-voip] Looking for IP Video Surveillance
Message-ID: <17A170DC8DD5CC44BA56B24B366EB60B01EDFD81@E3K-VS1.ccsi.local>

I know it doesn't fall directly in line with IPT but I figured many who
support IPT would support IP Video Surveillance design.
 
I'm interested in finding out who are the big names in IP Video
Surveillance.
I need a scalable solution with centralized management.
 
Any opinions are appreciated.
 
Scott
 
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From jburton at NETechCorp.com  Wed Jun 21 11:29:43 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Wed, 21 Jun 2006 11:29:43 -0400
Subject: [cisco-voip] PRI line option question-
In-Reply-To: <8A9E98B05F39B14E82E546ECF9A81969012181A1@exchange.sgh.com>
References: <050944EE48FF384F9C663F7048C47DA704143122@webmail.asheville.k12.nc.us>, 
	<F3F50B61-A22B-444A-AD2F-CCCF9EB5FC54@cisco.com>
	<54FE69CA-C2C3-4C0F-AB5B-39C05C26F3A8@mimectl>,
	<8A9E98B05F39B14E82E546ECF9A81969012181A1@exchange.sgh.com>
Message-ID: <18F7195F-3531-4895-938F-5331AE9B2C2C@mimectl>

True, IF and a big IF you can get the telco to send it down the display ie.   



From: Ben D. Kusa
Sent: Tue 6/20/2006 9:05 PM
To: Jason Burton; Ryan Ratliff; Tim Reimers
Cc: Kelly Looney; cisco-voip at puck.nether.net; Alan Surrette
Subject: RE: [cisco-voip] PRI line option question-


I think h323 does support naming if it comes via display ie. Maybe telco can change it, but I have never had luck with that.



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burton
Sent: Tuesday, June 20, 2006 8:37 PM
To: Ryan Ratliff; Tim Reimers
Cc: Kelly Looney; cisco-voip at puck.nether.net; Alan Surrette
Subject: Re: [cisco-voip] PRI line option question-

H.323 doesn't support calling name for PRI.  If you do a debug you can see the calling name come in the facility ie, but CM doesn't support this.




From: Ryan Ratliff
Sent: Tue 6/20/2006 10:52 AM
To: Tim Reimers
Cc: Kelly Looney; cisco-voip at puck.nether.net; Alan Surrette
Subject: Re: [cisco-voip] PRI line option question-
You need to get a 'debug isdn q931' for an inbound call to see where  
the name is appearing and why CM isn't showing it.
 
-Ryan
 
On Jun 20, 2006, at 10:42 AM, Tim Reimers wrote:
 
 
Hi everyone-
 
I wanted to ask a question -
 
 
Our ISDN PRI provider has started billing us (belatedly) for an  
option they  have provisioned on our line some time ago, and not told  
us.
 
It's a caller ID option called "Calling party number to name"
 
Before I decide that I don't want this, and they should take it off--
 
Does Cisco CCM support this?
 
Here's the details:
 
Cisco 2651XM running c2600-ipvoice-mz.123-8.T3
with two PRIs attached, 30 out of 47 channels live.
The gateway is in H.323 mode, with the appropriate (working fine)  
config in CCM.
 
when you get an outside call, you get just numbers, not names..
 
So Bell is providing us names with our Caller ID, but I guess our  
equipment is either incapable of showing that, or it's ignoring it.
 
 
Any thoughts?
 
Can I configure this to work, so that I can see what info -is-  
available from various people?
 
 
thanks, Tim
 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
 
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cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip
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From JOlson at mlnusa.com  Wed Jun 21 11:30:52 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Wed, 21 Jun 2006 11:30:52 -0400
Subject: [cisco-voip] Attendant Console online/offline
Message-ID: <FE2D092F8B1516488B99342551AACFD41CEE9894@mlnusa-exch1.mlnusa.com>

With an AC pilot where do you specify for a call to go to voicemail if
no attendants are logged in?

-----Original Message-----
From: Ryan Ratliff
Sent: Tuesday, June 20, 2006 9:17 AM
To: Olson, Jeffrey
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant Console online/offline

How do you have your AC hunt group configured?  If you set it to go  
to UserA-Line1, etc then the attendant must be logged in and online  
to receive calls.  If you set it to go to a dn/partition specifically  
then the AC console status will have no bearing on how calls are  
extended.

-Ryan

On Jun 19, 2006, at 4:46 PM, Olson, Jeffrey wrote:

I have attendant console installed on a few different receptionists  
PCs and it doesn't seem to matter if it is in the online or offline  
state.  Both ways it seems to have the same functionality.  What am I  
missing with this?  Is there a setting in call manager that I am  
missing?



Jeff

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders  
Network USA, Inc. which is confidential and legally privileged. The  
information is intended only for the use of the individual or entity  
named on this E-mail. If you are not the intended recipient, you are  
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_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip



From jburton at NETechCorp.com  Wed Jun 21 11:31:33 2006
From: jburton at NETechCorp.com (Jason Burton)
Date: Wed, 21 Jun 2006 11:31:33 -0400
Subject: [cisco-voip] Echo Problems
Message-ID: <BF46DA14-97F3-4A71-9C2A-BEB130F01368@mimectl>

Has anyone experienced any problems with echo after moving to 12.4(6)T2?  I had to move to this version to fix another issue, but now I'm getting complaints of echo on a couple of PRIs. 
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From rleetun at co.boulder.co.us  Wed Jun 21 11:32:51 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Wed, 21 Jun 2006 09:32:51 -0600
Subject: [cisco-voip] Meeting Place Express
Message-ID: <963CE94787005149A52C17F33A028E12051D48CD@EVS1B.boco.co.boulder.co.us>

Good morning,
 
Has anyone installed MeetingPlace Express?  I have a few questions for
whomever if possible.  Thanks so much!
 
Rob
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From Joe_Bell at adp.com  Wed Jun 21 11:49:05 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Wed, 21 Jun 2006 10:49:05 -0500
Subject: [cisco-voip] Attendant Console online/offline
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BC891EE@EXCHANGEDS01.ds.ad.adp.com>

Y0u can use the "Always Route Member" option in older AC or a second
line group in the newer.  Both options should point toward a route point
with call forward all to vm to get the call to a Unity handler.

Joe

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Olson, Jeffrey
Sent: Wednesday, June 21, 2006 8:31 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant Console online/offline

With an AC pilot where do you specify for a call to go to voicemail if
no attendants are logged in?

-----Original Message-----
From: Ryan Ratliff
Sent: Tuesday, June 20, 2006 9:17 AM
To: Olson, Jeffrey
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Attendant Console online/offline

How do you have your AC hunt group configured?  If you set it to go  
to UserA-Line1, etc then the attendant must be logged in and online  
to receive calls.  If you set it to go to a dn/partition specifically  
then the AC console status will have no bearing on how calls are  
extended.

-Ryan

On Jun 19, 2006, at 4:46 PM, Olson, Jeffrey wrote:

I have attendant console installed on a few different receptionists  
PCs and it doesn't seem to matter if it is in the online or offline  
state.  Both ways it seems to have the same functionality.  What am I  
missing with this?  Is there a setting in call manager that I am  
missing?



Jeff

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders  
Network USA, Inc. which is confidential and legally privileged. The  
information is intended only for the use of the individual or entity  
named on this E-mail. If you are not the intended recipient, you are  
hereby notified that disclosing, copying, distributing, or taking any  
action in reliance on the contents of this information is strictly  
prohibited, and that the E-mail should be returned to this company  
immediately. If you have received this E-mail in error, please  
contact the sender immediately.


_______________________________________________
cisco-voip mailing list
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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 message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.


From janderson at enventis.com  Wed Jun 21 11:52:48 2006
From: janderson at enventis.com (Jeff Anderson)
Date: Wed, 21 Jun 2006 10:52:48 -0500
Subject: [cisco-voip] T1 Clocking on 3825 with First Generation VWIC-2MFT
Message-ID: <20060621105249114.00000000172@janderson-3948>

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From jason.aarons at us.didata.com  Wed Jun 21 12:01:30 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 21 Jun 2006 12:01:30 -0400
Subject: [cisco-voip] T1 Clocking on 3825 with First Generation VWIC-2MFT
References: <20060621105249114.00000000172@janderson-3948>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E76D@USNAEXCH.na.didata.local>

You need a VWIC2-2MFT-T1 "Second Generation MFT" which has dual PLLs to clock data independently.  What does show diag say you have ?

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
Sent: Wed 6/21/2006 11:52 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] T1 Clocking on 3825 with First Generation VWIC-2MFT


I have the following configuration on a router:
 
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-select 1 T1 0/0/0
network-clock-select 2 T1 0/1/0
 
Controller T1 0/0/0 is configured for clock source line
Controller T1 0/0/1 in not in use
 
Controller T1 0/1/0 is configured for clock source line
Controller T1 0/1/1 is not in use
 
The T1 on controller 0/0/0 is not showing any slip errors. The controller on T1 0/1/0 is slipping badly, about 2400 slips seconds in a 24 hours period.
 
I need to get this router configured for 2 independent clock sources, both using line. Controller 0/0/0 is a tie line to a PBX. Controller 0/1/0 is a T1 CAS to the carrier.
 
I think all I need to do is:
 
no network-clock-participate wic 0
no network-clock-select 1 T1 0/0/0
 
Can someone please confirm that my changes will correct the problem or offer a solution that will? Will someone also explain what would happen to calls in progress while the clock changes are being made?
 
Thanks,
Jeff 


 



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From janderson at enventis.com  Wed Jun 21 12:10:28 2006
From: janderson at enventis.com (Jeff Anderson)
Date: Wed, 21 Jun 2006 11:10:28 -0500
Subject: [cisco-voip] T1 Clocking on 3825 with First Generation VWIC-2MFT
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F013E76D@USNAEXCH.na.didata.local>
Message-ID: <20060621111028548.00000000172@janderson-3948>

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From MLinsemier at apcapital.com  Wed Jun 21 12:11:42 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Wed, 21 Jun 2006 12:11:42 -0400
Subject: [cisco-voip] Meeting Place Express
Message-ID: <AB0A35F2486F81468971AA68558BE719040375E5@smieln01winm01.micoanet.com>

I have installed it in pilot, what questions are you looking to have
answered?

 

Matt

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Leetun, Rob
Sent: Wednesday, June 21, 2006 11:33 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Meeting Place Express

 

Good morning,

 

Has anyone installed MeetingPlace Express?  I have a few questions for
whomever if possible.  Thanks so much!

 

Rob


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This communication and any attachments are CONFIDENTIAL and may be 
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From Scott.Voll at wesd.org  Wed Jun 21 12:58:15 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 21 Jun 2006 09:58:15 -0700
Subject: [cisco-voip] Latancy / Jitter
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED4EBF@Apollo.wesd.org>

I'm looking at QoS stats from CM for phone calls.  Is this out of
recommended specs?

 

Jitter < 15

Lantancy < 175

 

Thanks

 

Scott

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From Joe_Bell at adp.com  Wed Jun 21 13:14:23 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Wed, 21 Jun 2006 12:14:23 -0500
Subject: [cisco-voip] Latancy / Jitter
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BCB11CF@EXCHANGEDS01.ds.ad.adp.com>

According to the Enterprise QoS SRND page 1-13
(http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmi
gration_09186a008049b062.pdf), your spec's are:

Voice (Bearer Traffic)

A summary of the key QoS requirements and recommendations for Voice
(bearer traffic) are:

* Voice traffic should be marked to DSCP EF per the QoS Baseline and RFC
3246.

* Loss should be no more than 1 %.

* One-way Latency (mouth-to-ear) should be no more than 150 ms.

* Average one-way Jitter should be targeted under 30 ms.

* 21-320 kbps of guaranteed priority bandwidth is required per call
(depending on the sampling

rate, VoIP codec and Layer 2 media overhead).

So, your latency is a tad bit high, but humans don't really notice it
until it is consistently above 200 ms.

Joe

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, June 21, 2006 9:58 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Latancy / Jitter

 

I'm looking at QoS stats from CM for phone calls.  Is this out of
recommended specs?

 

Jitter < 15

Lantancy < 175

 

Thanks

 

Scott



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 reports at cmreports.com  Wed Jun 21 13:29:29 2006
From: reports at cmreports.com (Reports@CMReports.com)
Date: Wed, 21 Jun 2006 13:29:29 -0400
Subject: [cisco-voip] Serial Number from Call Manager (Frazee, Timothy)
References: <s48e8617.043@freedom7.flemingc.on.ca>
	<399D7AD8-B7BA-44BA-9D2D-92E4E4E2D5D6@cisco.com>
Message-ID: <017d01c69558$42798a40$5100ae0c@HP8240>

Today, CMReports.com released an application designed to gather exactly this 
information.  It is on sale until July 1st for $69.

http://www.cmreports.com/shop/index.php?main_page=index&cPath=5&zenid=ccc78358dfc71c255c7d9adf5b796708

Thanks!
John Staples


----- Original Message ----- 
From: "Ryan Ratliff" <rratliff at cisco.com>
To: "Robert Harvey" <ROHARVEY at flemingc.on.ca>
Cc: <cisco-voip at puck.nether.net>
Sent: Tuesday, June 13, 2006 9:45 AM
Subject: Re: [cisco-voip] Serial Number from Call Manager (Frazee, Timothy)


> Are you getting the mac address or serial number?
>
> Serial numbers are not stored in the database.
>
> -Ryan
>
> On Jun 13, 2006, at 9:31 AM, Robert Harvey wrote:
>
> Hi Eby,
>
> I'm not sure about CCM5.x, but with all previous versions I have just
> queried the dbo.device table via an ODBC connection to the publisher.
> This is incredibly easy, flexible and the price is right :)  Email me if
> you want further instruction on the process.
>
> Robert.
>
>
> Hi,
>   Is it possible to get the phone's serial number from call manager
> using BAT tool? I know how to get it one by one but it will take a
> long
> time to go to each phone and copying it. Any Ideas?
>
> Thanks,
>
> Eby
> _______________________________________________
> 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 BurwellJ at firstcharter.com  Wed Jun 21 13:41:08 2006
From: BurwellJ at firstcharter.com (Jason Burwell)
Date: Wed, 21 Jun 2006 13:41:08 -0400
Subject: [cisco-voip] Video Call Bandwidth
Message-ID: <697161D0107BAD40AFE9A86B06509AC90643E790@FCEXCLUSTER.FirstCharter.com>

Just curious to hear what the lowest bandwidth setting anyone has had
luck with for video calls. I am looking to deploy some 7985s to branch
sites but want to keep a reasonable video quality with the lowest
possible bandwidth. Is there a "drop-off" point? I really think the 384k
is more than I'd like to use since it is equivalent to 16 G729 calls and
most branch sites only have one full T1. 
 
Any advise?
 
Thanks!
Jason
 
Jason T. Burwell
Infrastructure/Telecommunications
First Charter Bank
10200 David Taylor Drive
Charlotte, NC 28262
ph-704-688-4664
fax-704-688-4901
www.firstcharter.com
 


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addressed and may contain confidential and/or privileged material. Any review,
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this information by persons or entities other than the intended recipient is prohibited.
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From jim.koniecki at us.didata.com  Wed Jun 21 13:50:59 2006
From: jim.koniecki at us.didata.com (Jim Koniecki (US))
Date: Wed, 21 Jun 2006 13:50:59 -0400
Subject: [cisco-voip] Video Call Bandwidth
Message-ID: <C65033A817B5734392E55CB9DF446104057E589C@USNAEXCH.na.didata.local>

Jason, 

 

I currently have my VTA set to 128K for me it is acceptable. @ 128K any
motion or movement will cause it to become choppy and noticeable.

 
@ 384K you will have a much better experience.

 

 

 

I would set key units to 384K and the less concerned to 128K. 

 

My 2 cents.

 

 

/Jim K 

 

Jim Koniecki 

267-671-8500

Y-jkoniecki

 

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Burwell
Sent: Wednesday, June 21, 2006 1:41 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Video Call Bandwidth

 

Just curious to hear what the lowest bandwidth setting anyone has had luck
with for video calls. I am looking to deploy some 7985s to branch sites but
want to keep a reasonable video quality with the lowest possible bandwidth.
Is there a "drop-off" point? I really think the 384k is more than I'd like
to use since it is equivalent to 16 G729 calls and most branch sites only
have one full T1. 

 

Any advise?

 

Thanks!

Jason

 

Jason T. Burwell

Infrastructure/Telecommunications

First Charter Bank

10200 David Taylor Drive

Charlotte, NC 28262

ph-704-688-4664

fax-704-688-4901

www.firstcharter.com

 



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reliance upon,
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From CMStaffin at ra.rockwell.com  Wed Jun 21 13:56:45 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Wed, 21 Jun 2006 12:56:45 -0500
Subject: [cisco-voip] Video Call Bandwidth
In-Reply-To: <697161D0107BAD40AFE9A86B06509AC90643E790@FCEXCLUSTER.FirstCharter.com>
Message-ID: <OF10A4ABCD.5E8C4734-ON86257194.0062679F-86257194.0062823B@ra.rockwell.com>

Jason currently I have 7985's in the lab and the lowest I would go would be
256k.  Anything lower and it starts to get pretty blocky.

Also take a look at Sony Ipella as they make very good products that are
the same price and far better as far as usability is concerned.

Craig




                                                                           
             "Jason Burwell"                                               
             <BurwellJ at firstch                                             
             arter.com>                                                 To 
             Sent by:                  <cisco-voip at puck.nether.net>        
             cisco-voip-bounce                                          cc 
             s at puck.nether.net                                             
                                                                   Subject 
                                       [cisco-voip] Video Call Bandwidth   
             06/21/2006 12:41                                              
             PM                                                            
                                                                           
                                                                           
                                                                           
                                                                           




Just curious to hear what the lowest bandwidth setting anyone has had luck
with for video calls. I am looking to deploy some 7985s to branch sites but
want to keep a reasonable video quality with the lowest possible bandwidth.
Is there a "drop-off" point? I really think the 384k is more than I'd like
to use since it is equivalent to 16 G729 calls and most branch sites only
have one full T1.

Any advise?

Thanks!
Jason

Jason T. Burwell
Infrastructure/Telecommunications
First Charter Bank
10200 David Taylor Drive
Charlotte, NC 28262
ph-704-688-4664
fax-704-688-4901
www.firstcharter.com



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reliance upon,
this information by persons or entities other than the intended recipient
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**********************************************************************
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From bcarter at sentinel.com  Wed Jun 21 14:10:54 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Wed, 21 Jun 2006 13:10:54 -0500
Subject: [cisco-voip] what the best way to create a back door for VM
Message-ID: <6A0744639B86DD459433108DBF101D26FC6546@MAIL01.sentinel.com>

We just call our main number and hit * and get "please enter your ID" 

-----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, June 15, 2006 2:03 PM
To: James Grace
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] what the best way to create a back door for VM

Just take a spare DID, route it through a CTI RP that is CFA into vm.
Then in Unity make a call routing rule that sends callers to subscriber
sign in.

-Ryan

On Jun 15, 2006, at 2:46 PM, James Grace wrote:

Ive seen site the has a back door number config on there system.   
When you call it,  it goes direct to "please enter your ID"



James D. Grace

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation



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


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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  Wed Jun 21 14:14:26 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Wed, 21 Jun 2006 13:14:26 -0500
Subject: [cisco-voip] CME - IP Phone "?" button
Message-ID: <6A0744639B86DD459433108DBF101D26FC6547@MAIL01.sentinel.com>

 
I just completed my first CME installation.  On the IP Phones, when a
user hits the ? they don't get any help files.  It is like there is no
help files on CME.  Is this normal??


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 
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From keith.klevenski at rig.net  Wed Jun 21 14:22:04 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Wed, 21 Jun 2006 13:22:04 -0500
Subject: [cisco-voip] Changing Windows Local Administrator Account Affects
	MLA?
Message-ID: <7F0CE0A1A0961E498A41329535342266A10776@HOUEXH02.rignet.org>

Hi all,

 

I have a 4.1.3 cluster that I need to change the local administrator
account to.  No problem on a couple of 3.3.5 clusters that are not
running MLA, but when I changed the local administrator account password
on the 4.1.3 publisher when I went to log in go ccmadmin with my MLA
account access is denied.  Same thing when logging in with
ccmadministrator.  Fortunately I can still get into the subscriber (2
server cluster), but I'm not sure what the problem is.  I see these
errors in the event log when attempting to log in to ccmadmin with my
MLA account:

 

The server was unable to logon the Windows NT account 'keith' due to the
following error: Logon failure: unknown user name or bad password.  The
data is the error code. 

For additional information specific to this message please visit the
Microsoft Online Support site located at:
http://www.microsoft.com/contentredirect.asp.

 

I'm using the standard procedure for resetting the local administrator
account:

 

To Change Windows Administrator Password:

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

Please make sure that the Local Administrator password is SAME on all
servers in the

cluster.

 

To change the password, perform the following steps:

 

1. Right click My Computer.

2. Select Manage.

3. Go to Local Users and Groups.

4. Go to Users.

5. Right click Administrator.

6. Select Set Password.

7. Type in the password and confirm.

 

I am also restarting the WWW service after making the change.  I've
rebooted the publisher as well, but I must be missing something.  I
can't find any special instructions on changing local admin account with
MLA running.  Seems like it shouldn't have any affect on MLA.  I've
changed the password back to the original and I'm still having the same
problem.  I was thinking about disabling MLA from the subscriber and
seeing if I can log in with local admin account to ccmadmin, but I don't
want to hose myself. 

 

Any ideas?  TIA!!

 

Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077         

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

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

 

"Extending the power of your knowledge"

 

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From rratliff at cisco.com  Wed Jun 21 14:36:45 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 21 Jun 2006 14:36:45 -0400
Subject: [cisco-voip] Changing Windows Local Administrator Account
	Affects MLA?
In-Reply-To: <7F0CE0A1A0961E498A41329535342266A10776@HOUEXH02.rignet.org>
References: <7F0CE0A1A0961E498A41329535342266A10776@HOUEXH02.rignet.org>
Message-ID: <C0B38B85-0726-4DCC-8F79-C6E15850F746@cisco.com>

You can disable MLA just by going into the MLAParameters table in the  
CCM030X database on the publisher.  The parameter name is similar to  
MLAEnabled but should be obvious.  Then simply restart IIS (not just  
WWW) and you should be good to go.

If you only restarted WWW and not IISAdmin I'd try bouncing both  
before going into SQL to disable MLA.

-Ryan

On Jun 21, 2006, at 2:22 PM, Keith Klevenski wrote:

Hi all,



I have a 4.1.3 cluster that I need to change the local administrator  
account to.  No problem on a couple of 3.3.5 clusters that are not  
running MLA, but when I changed the local administrator account  
password on the 4.1.3 publisher when I went to log in go ccmadmin  
with my MLA account access is denied.  Same thing when logging in  
with ccmadministrator.  Fortunately I can still get into the  
subscriber (2 server cluster), but I?m not sure what the problem is.   
I see these errors in the event log when attempting to log in to  
ccmadmin with my MLA account:



The server was unable to logon the Windows NT account 'keith' due to  
the following error: Logon failure: unknown user name or bad  
password.  The data is the error code.

For additional information specific to this message please visit the  
Microsoft Online Support site located at: http://www.microsoft.com/ 
contentredirect.asp.



I?m using the standard procedure for resetting the local  
administrator account:



To Change Windows Administrator Password:

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

Please make sure that the Local Administrator password is SAME on all  
servers in the

cluster.



To change the password, perform the following steps:



1. Right click My Computer.

2. Select Manage.

3. Go to Local Users and Groups.

4. Go to Users.

5. Right click Administrator.

6. Select Set Password.

7. Type in the password and confirm.



I am also restarting the WWW service after making the change.  I?ve  
rebooted the publisher as well, but I must be missing something.  I  
can?t find any special instructions on changing local admin account  
with MLA running.  Seems like it shouldn?t have any affect on MLA.   
I?ve changed the password back to the original and I?m still having  
the same problem.  I was thinking about disabling MLA from the  
subscriber and seeing if I can log in with local admin account to  
ccmadmin, but I don?t want to hose myself.



Any ideas?  TIA!!



Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

http://www.rig.net



"Extending the power of your knowledge"



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



From keith.klevenski at rig.net  Wed Jun 21 14:51:03 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Wed, 21 Jun 2006 13:51:03 -0500
Subject: [cisco-voip] Changing Windows Local Administrator Account
	Affects MLA?
Message-ID: <7F0CE0A1A0961E498A41329535342266A10778@HOUEXH02.rignet.org>

Gotcha, I only restarted WWW.  Since I rebooted the server obviously IIS
was restarted and I did restart it again, but I still cannot get to
ccmadmin with any MLA account from the publisher.  I guess that is my
main question.  Should you disable MLA before changing the local admin
account?  Are there any dependencies between the local admin account and
MLA account?  I haven't seen any specific considerations regarding this.
It worked fine with 3.3.5 (with no MLA) and I only restarted WWW on
those servers.

Any information about the dependencies between the local admin account
and MLA would be welcome if anyone has any info.  I guess at this point
I don't have much choice but to disable MLA from SQL, restart IIS and
see if I can get in with the local admin account.  Then reenable MLA and
hope for the best.  I'm just afraid of locking myself out of ccmadmin as
right now I still have accesses through the sub.  Is there any risk of
locking yourself out of ccmadmin?  Can you disable MLA from the sub?  I
can get to ccmadmin from there and to the MLA parameters.

Thanks!

Keith


-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Wednesday, June 21, 2006 1:37 PM
To: Keith Klevenski
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing Windows Local Administrator Account
Affects MLA?

You can disable MLA just by going into the MLAParameters table in the  
CCM030X database on the publisher.  The parameter name is similar to  
MLAEnabled but should be obvious.  Then simply restart IIS (not just  
WWW) and you should be good to go.

If you only restarted WWW and not IISAdmin I'd try bouncing both  
before going into SQL to disable MLA.

-Ryan

On Jun 21, 2006, at 2:22 PM, Keith Klevenski wrote:

Hi all,



I have a 4.1.3 cluster that I need to change the local administrator  
account to.  No problem on a couple of 3.3.5 clusters that are not  
running MLA, but when I changed the local administrator account  
password on the 4.1.3 publisher when I went to log in go ccmadmin  
with my MLA account access is denied.  Same thing when logging in  
with ccmadministrator.  Fortunately I can still get into the  
subscriber (2 server cluster), but I'm not sure what the problem is.   
I see these errors in the event log when attempting to log in to  
ccmadmin with my MLA account:



The server was unable to logon the Windows NT account 'keith' due to  
the following error: Logon failure: unknown user name or bad  
password.  The data is the error code.

For additional information specific to this message please visit the  
Microsoft Online Support site located at: http://www.microsoft.com/ 
contentredirect.asp.



I'm using the standard procedure for resetting the local  
administrator account:



To Change Windows Administrator Password:

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

Please make sure that the Local Administrator password is SAME on all  
servers in the

cluster.



To change the password, perform the following steps:



1. Right click My Computer.

2. Select Manage.

3. Go to Local Users and Groups.

4. Go to Users.

5. Right click Administrator.

6. Select Set Password.

7. Type in the password and confirm.



I am also restarting the WWW service after making the change.  I've  
rebooted the publisher as well, but I must be missing something.  I  
can't find any special instructions on changing local admin account  
with MLA running.  Seems like it shouldn't have any affect on MLA.   
I've changed the password back to the original and I'm still having  
the same problem.  I was thinking about disabling MLA from the  
subscriber and seeing if I can log in with local admin account to  
ccmadmin, but I don't want to hose myself.



Any ideas?  TIA!!



Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

http://www.rig.net



"Extending the power of your knowledge"



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



From SCASPER at mandtbank.com  Wed Jun 21 14:54:11 2006
From: SCASPER at mandtbank.com (STEVEN CASPER)
Date: Wed, 21 Jun 2006 14:54:11 -0400
Subject: [cisco-voip] Latancy / Jitter
Message-ID: <s4995d9c.040@comrcm01intgwia.mandtbank.com>

How about Max Jitter - we sometimes see as much as 50-60 ms though Avg
Jitter is low.

Steve

>>> "Bell, Joe" <Joe_Bell at adp.com> 06/21/06 1:14 PM >>>
According to the Enterprise QoS SRND page 1-13
(http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmi
gration_09186a008049b062.pdf), your spec's are:

Voice (Bearer Traffic)

A summary of the key QoS requirements and recommendations for Voice
(bearer traffic) are:

* Voice traffic should be marked to DSCP EF per the QoS Baseline and
RFC
3246.

* Loss should be no more than 1 %.

* One-way Latency (mouth-to-ear) should be no more than 150 ms.

* Average one-way Jitter should be targeted under 30 ms.

* 21-320 kbps of guaranteed priority bandwidth is required per call
(depending on the sampling

rate, VoIP codec and Layer 2 media overhead).

So, your latency is a tad bit high, but humans don't really notice it
until it is consistently above 200 ms.

Joe

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, June 21, 2006 9:58 AM
To: cisco-voip at puck.nether.net 
Subject: [cisco-voip] Latancy / Jitter

 

I'm looking at QoS stats from CM for phone calls.  Is this out of
recommended specs?

 

Jitter < 15

Lantancy < 175

 

Thanks

 

Scott



This message and any attachments are intended only for the use of the
addressee and may contain information that is privileged and
confidential. If the reader of the message is not the intended recipient
or an authorized representative of the intended recipient, you are
hereby notified that any dissemination of this communication is strictly
prohibited. If you have received this communication in error, please
notify us immediately by e-mail and delete the message and any
attachments from your system.


From CMStaffin at ra.rockwell.com  Wed Jun 21 15:14:33 2006
From: CMStaffin at ra.rockwell.com (Craig M Staffin)
Date: Wed, 21 Jun 2006 14:14:33 -0500
Subject: [cisco-voip] Latancy / Jitter
In-Reply-To: <s4995d9c.040@comrcm01intgwia.mandtbank.com>
Message-ID: <OF139F40AA.C595C75E-ON86257194.00699621-86257194.0069A1D9@ra.rockwell.com>

Steven,

Take a look at Packeteer.  I have tested in the lab about 5% packet loss as
well as 200ms jitter and it still comes across crystal clear.  Now
obviously this not cisco supported but if it works thats all that matters.

Craig




                                                                           
             "STEVEN CASPER"                                               
             <SCASPER at mandtban                                             
             k.com>                                                     To 
             Sent by:                  Joe_Bell at adp.com,                   
             cisco-voip-bounce         cisco-voip at puck.nether.net          
             s at puck.nether.net                                          cc 
                                                                           
                                                                   Subject 
             06/21/2006 01:54          Re: [cisco-voip] Latancy / Jitter   
             PM                                                            
                                                                           
                                                                           
                                                                           
                                                                           
                                                                           




How about Max Jitter - we sometimes see as much as 50-60 ms though Avg
Jitter is low.

Steve

>>> "Bell, Joe" <Joe_Bell at adp.com> 06/21/06 1:14 PM >>>
According to the Enterprise QoS SRND page 1-13
(http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmi
gration_09186a008049b062.pdf), your spec's are:

Voice (Bearer Traffic)

A summary of the key QoS requirements and recommendations for Voice
(bearer traffic) are:

* Voice traffic should be marked to DSCP EF per the QoS Baseline and
RFC
3246.

* Loss should be no more than 1 %.

* One-way Latency (mouth-to-ear) should be no more than 150 ms.

* Average one-way Jitter should be targeted under 30 ms.

* 21-320 kbps of guaranteed priority bandwidth is required per call
(depending on the sampling

rate, VoIP codec and Layer 2 media overhead).

So, your latency is a tad bit high, but humans don't really notice it
until it is consistently above 200 ms.

Joe







________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, June 21, 2006 9:58 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Latancy / Jitter



I'm looking at QoS stats from CM for phone calls.  Is this out of
recommended specs?



Jitter < 15

Lantancy < 175



Thanks



Scott



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
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From Joe_Bell at adp.com  Wed Jun 21 16:00:35 2006
From: Joe_Bell at adp.com (Bell, Joe)
Date: Wed, 21 Jun 2006 15:00:35 -0500
Subject: [cisco-voip] Latancy / Jitter
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BCB1443@EXCHANGEDS01.ds.ad.adp.com>

Its all about the average as the human ear won't pick up anomalies.
That kind of points to the nebulous idea of a MOS (Mean Opinion Score).
We need to engineer to these numbers listed because MOS has been
determined to be high when within these spec's.

A consistently high amount of jitter is a problem, but an anomaly is a
part of the data network.

Joe


-----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, June 21, 2006 11:54 AM
To: Bell, Joe; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Latancy / Jitter

How about Max Jitter - we sometimes see as much as 50-60 ms though Avg
Jitter is low.

Steve

>>> "Bell, Joe" <Joe_Bell at adp.com> 06/21/06 1:14 PM >>>
According to the Enterprise QoS SRND page 1-13
(http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmi
gration_09186a008049b062.pdf), your spec's are:

Voice (Bearer Traffic)

A summary of the key QoS requirements and recommendations for Voice
(bearer traffic) are:

* Voice traffic should be marked to DSCP EF per the QoS Baseline and
RFC
3246.

* Loss should be no more than 1 %.

* One-way Latency (mouth-to-ear) should be no more than 150 ms.

* Average one-way Jitter should be targeted under 30 ms.

* 21-320 kbps of guaranteed priority bandwidth is required per call
(depending on the sampling

rate, VoIP codec and Layer 2 media overhead).

So, your latency is a tad bit high, but humans don't really notice it
until it is consistently above 200 ms.

Joe

 

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net 
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, June 21, 2006 9:58 AM
To: cisco-voip at puck.nether.net 
Subject: [cisco-voip] Latancy / Jitter

 

I'm looking at QoS stats from CM for phone calls.  Is this out of
recommended specs?

 

Jitter < 15

Lantancy < 175

 

Thanks

 

Scott



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
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_______________________________________________
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This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.


From jkienzle at westerngas.com  Wed Jun 21 16:17:32 2006
From: jkienzle at westerngas.com (John Kienzle)
Date: Wed, 21 Jun 2006 14:17:32 -0600
Subject: [cisco-voip] Problems Changing Exchange Partner server in Unity
	4.0.4 SR1
Message-ID: <FAFB6347F8C8D248A8D31D47F2C2BDDE28F1@wgntgilex1.wgr.com>

We have installed a new Exchange 2003 SP2 server (SRV2) on Windows 2003
R2. Unity currently points at our old Exchange 2000 (SRV1). Utilizing
the steps in this document:
http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_upgrade_g
uides_chapter09186a0080205a8b.html#wp1359133

We've run the Dbwalker, Dirt, Permissions wizard and followed the steps
to specify a different Exchange server as the Unity partner. All
accounts have the correct permissions(verified several times). Pilot
accounts moved to the new exchange server are able to get voicemail. The
problem is when we ran the Cisco Unity Message Store Configuration
Wizard, it was going smoothly until we got to the step to "Select
Mailbox Location". You make your selection to the new Exchange, click
next and the wizard completely disappears and doesn't appear to finish.
The partner exchange server in the Unity registry still points at the
old SRV1. We followed the same procedure at our other locations with no
problems. Reboots on the new Exchange and Unity have been done per the
document.

Any thoughts as to what is happening would be great.

John Kienzle
Western Gas Resources
Field Support Analyst, Gillette
Phone (307) 685-5870
Cell (307) 680-1746
jkienzle at westerngas.com





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From rratliff at cisco.com  Wed Jun 21 16:29:52 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 21 Jun 2006 16:29:52 -0400
Subject: [cisco-voip] Changing Windows Local Administrator Account
	Affects MLA?
In-Reply-To: <7F0CE0A1A0961E498A41329535342266A10778@HOUEXH02.rignet.org>
References: <7F0CE0A1A0961E498A41329535342266A10778@HOUEXH02.rignet.org>
Message-ID: <4932BB0C-45A6-426B-80D8-2B41BD67DB90@cisco.com>

In CM 4.x the MLA parameters are clusterwide since they are stored in  
the database.  In previous versions we actually pointed the CCMAdmin  
IIS virtual directory to a new location.  Now it's the behind-the- 
scenes authentication that is changed.

As long as you don't restart IIS on the sub no changes to MLA will be  
picked up there.

-Ryan

On Jun 21, 2006, at 2:51 PM, Keith Klevenski wrote:

Gotcha, I only restarted WWW.  Since I rebooted the server obviously IIS
was restarted and I did restart it again, but I still cannot get to
ccmadmin with any MLA account from the publisher.  I guess that is my
main question.  Should you disable MLA before changing the local admin
account?  Are there any dependencies between the local admin account and
MLA account?  I haven't seen any specific considerations regarding this.
It worked fine with 3.3.5 (with no MLA) and I only restarted WWW on
those servers.

Any information about the dependencies between the local admin account
and MLA would be welcome if anyone has any info.  I guess at this point
I don't have much choice but to disable MLA from SQL, restart IIS and
see if I can get in with the local admin account.  Then reenable MLA and
hope for the best.  I'm just afraid of locking myself out of ccmadmin as
right now I still have accesses through the sub.  Is there any risk of
locking yourself out of ccmadmin?  Can you disable MLA from the sub?  I
can get to ccmadmin from there and to the MLA parameters.

Thanks!

Keith


-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Wednesday, June 21, 2006 1:37 PM
To: Keith Klevenski
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing Windows Local Administrator Account
Affects MLA?

You can disable MLA just by going into the MLAParameters table in the
CCM030X database on the publisher.  The parameter name is similar to
MLAEnabled but should be obvious.  Then simply restart IIS (not just
WWW) and you should be good to go.

If you only restarted WWW and not IISAdmin I'd try bouncing both
before going into SQL to disable MLA.

-Ryan

On Jun 21, 2006, at 2:22 PM, Keith Klevenski wrote:

Hi all,



I have a 4.1.3 cluster that I need to change the local administrator
account to.  No problem on a couple of 3.3.5 clusters that are not
running MLA, but when I changed the local administrator account
password on the 4.1.3 publisher when I went to log in go ccmadmin
with my MLA account access is denied.  Same thing when logging in
with ccmadministrator.  Fortunately I can still get into the
subscriber (2 server cluster), but I'm not sure what the problem is.
I see these errors in the event log when attempting to log in to
ccmadmin with my MLA account:



The server was unable to logon the Windows NT account 'keith' due to
the following error: Logon failure: unknown user name or bad
password.  The data is the error code.

For additional information specific to this message please visit the
Microsoft Online Support site located at: http://www.microsoft.com/
contentredirect.asp.



I'm using the standard procedure for resetting the local
administrator account:



To Change Windows Administrator Password:

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

Please make sure that the Local Administrator password is SAME on all
servers in the

cluster.



To change the password, perform the following steps:



1. Right click My Computer.

2. Select Manage.

3. Go to Local Users and Groups.

4. Go to Users.

5. Right click Administrator.

6. Select Set Password.

7. Type in the password and confirm.



I am also restarting the WWW service after making the change.  I've
rebooted the publisher as well, but I must be missing something.  I
can't find any special instructions on changing local admin account
with MLA running.  Seems like it shouldn't have any affect on MLA.
I've changed the password back to the original and I'm still having
the same problem.  I was thinking about disabling MLA from the
subscriber and seeing if I can log in with local admin account to
ccmadmin, but I don't want to hose myself.



Any ideas?  TIA!!



Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

http://www.rig.net



"Extending the power of your knowledge"



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


From bcarter at sentinel.com  Wed Jun 21 16:41:29 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Wed, 21 Jun 2006 15:41:29 -0500
Subject: [cisco-voip] SMARTnet Price&Partnumber IPC Bundle
Message-ID: <6A0744639B86DD459433108DBF101D26FC6560@MAIL01.sentinel.com>

I can't find SMARTnet pricing for the IP Communication "Applications for
All" bundle.

IPC-COM-4X-BUN	IPC Commercial, 2 MCS svrs, CCM, Unity Con, VPT + apps.

I am looking for Essential 24X7X4 pricing.  Thanks.


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 sales at doretel.com  Wed Jun 21 16:48:21 2006
From: sales at doretel.com (Sales@Doretel)
Date: Wed, 21 Jun 2006 16:48:21 -0400
Subject: [cisco-voip] SMARTnet Price&Partnumber IPC Bundle
References: <6A0744639B86DD459433108DBF101D26FC6560@MAIL01.sentinel.com>
Message-ID: <101401c69574$08892b30$6501a8c0@ShaneCompaq>

Bill,

You probably have to break each part down and SMARTnet it that way. If you 
can send me a list of part numbers I will try my best to look it all up for 
you and send pricing over!

Shane Breen
Doretel Communications, Inc.
Director of Sales & Marketing
Cisco Registered Partner
Office: 404.755.5721
Fax: 404.529.4617
sbreen at doretel.com
AIM: shanebreen2003
www.doretel.com


----- Original Message ----- 
From: "Carter, Bill" <bcarter at sentinel.com>
To: <cisco-voip at puck.nether.net>
Sent: Wednesday, June 21, 2006 4:41 PM
Subject: [cisco-voip] SMARTnet Price&Partnumber IPC Bundle


>I can't find SMARTnet pricing for the IP Communication "Applications for
> All" bundle.
>
> IPC-COM-4X-BUN IPC Commercial, 2 MCS svrs, CCM, Unity Con, VPT + apps.
>
> I am looking for Essential 24X7X4 pricing.  Thanks.
>
>
> 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 


From Scott.Voll at wesd.org  Wed Jun 21 16:48:13 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 21 Jun 2006 13:48:13 -0700
Subject: [cisco-voip] What causes a Err-disable?
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED4F3C@Apollo.wesd.org>

What causes a port on a 3560 to go into err-disable?  I know a shut no
shut fixes it but how can we fix it to not do it at all?

 

 

IE>

 

FastEthernet0/4 is down, line protocol is down (err-disabled)

  Hardware is Fast Ethernet, address is 0017.9468.0506 (bia
0017.9468.0506)

  MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec, 

     reliability 255/255, txload 1/255, rxload 1/255

  Encapsulation ARPA, loopback not set

  Keepalive set (10 sec)

  Auto-duplex, Auto-speed, media type is 10/100BaseTX

  input flow-control is off, output flow-control is unsupported 

  ARP type: ARPA, ARP Timeout 04:00:00

  Last input never, output 2w6d, output hang never

  Last clearing of "show interface" counters never

  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0

  Queueing strategy: fifo

  Output queue: 0/40 (size/max)

  5 minute input rate 0 bits/sec, 0 packets/sec

  5 minute output rate 0 bits/sec, 0 packets/sec

     167934 packets input, 27324253 bytes, 0 no buffer

     Received 64162 broadcasts (0 multicast)

     0 runts, 1 giants, 0 throttles

     1 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored

     0 watchdog, 60442 multicast, 0 pause input

     0 input packets with dribble condition detected

     629225 packets output, 130100556 bytes, 0 underruns

     1 output errors, 1440 collisions, 1 interface resets

     0 babbles, 0 late collision, 0 deferred

     0 lost carrier, 0 no carrier, 0 PAUSE output

     0 output buffer failures, 0 output buffers swapped out

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From Brian.Henry at apptis.com  Wed Jun 21 16:52:22 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Wed, 21 Jun 2006 16:52:22 -0400
Subject: [cisco-voip] What causes a Err-disable?
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC05B883@CH-EXCH-02.intelltech.apptis.inc>

Do you have any thresholds put into place on the port or use port security.

Brian
--------------------------
Sent from my BlackBerry Wireless Handheld
 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
To: cisco-voip at puck.nether.net
Sent: Wed Jun 21 16:48:13 2006
Subject: [cisco-voip] What causes a Err-disable?

What causes a port on a 3560 to go into err-disable?  I know a shut no shut fixes it but how can we fix it to not do it at all?

 

 

IE>

 

FastEthernet0/4 is down, line protocol is down (err-disabled)

  Hardware is Fast Ethernet, address is 0017.9468.0506 (bia 0017.9468.0506)

  MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec, 

     reliability 255/255, txload 1/255, rxload 1/255

  Encapsulation ARPA, loopback not set

  Keepalive set (10 sec)

  Auto-duplex, Auto-speed, media type is 10/100BaseTX

  input flow-control is off, output flow-control is unsupported 

  ARP type: ARPA, ARP Timeout 04:00:00

  Last input never, output 2w6d, output hang never

  Last clearing of "show interface" counters never

  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0

  Queueing strategy: fifo

  Output queue: 0/40 (size/max)

  5 minute input rate 0 bits/sec, 0 packets/sec

  5 minute output rate 0 bits/sec, 0 packets/sec

     167934 packets input, 27324253 bytes, 0 no buffer

     Received 64162 broadcasts (0 multicast)

     0 runts, 1 giants, 0 throttles

     1 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored

     0 watchdog, 60442 multicast, 0 pause input

     0 input packets with dribble condition detected

     629225 packets output, 130100556 bytes, 0 underruns

     1 output errors, 1440 collisions, 1 interface resets

     0 babbles, 0 late collision, 0 deferred

     0 lost carrier, 0 no carrier, 0 PAUSE output

     0 output buffer failures, 0 output buffers swapped out

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From tednugent69 at yahoo.com  Wed Jun 21 16:56:56 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 21 Jun 2006 13:56:56 -0700 (PDT)
Subject: [cisco-voip] What causes a Err-disable?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED4F3C@Apollo.wesd.org>
Message-ID: <20060621205656.7966.qmail@web36103.mail.mud.yahoo.com>

Speed, DUP and Native VLAN mismatches can all cause
err disable..

--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> What causes a port on a 3560 to go into err-disable?
>  I know a shut no
> shut fixes it but how can we fix it to not do it at
> all?
> 
>  
> 
>  
> 
> IE>
> 
>  
> 
> FastEthernet0/4 is down, line protocol is down
> (err-disabled)
> 
>   Hardware is Fast Ethernet, address is
> 0017.9468.0506 (bia
> 0017.9468.0506)
> 
>   MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec, 
> 
>      reliability 255/255, txload 1/255, rxload 1/255
> 
>   Encapsulation ARPA, loopback not set
> 
>   Keepalive set (10 sec)
> 
>   Auto-duplex, Auto-speed, media type is
> 10/100BaseTX
> 
>   input flow-control is off, output flow-control is
> unsupported 
> 
>   ARP type: ARPA, ARP Timeout 04:00:00
> 
>   Last input never, output 2w6d, output hang never
> 
>   Last clearing of "show interface" counters never
> 
>   Input queue: 0/75/0/0 (size/max/drops/flushes);
> Total output drops: 0
> 
>   Queueing strategy: fifo
> 
>   Output queue: 0/40 (size/max)
> 
>   5 minute input rate 0 bits/sec, 0 packets/sec
> 
>   5 minute output rate 0 bits/sec, 0 packets/sec
> 
>      167934 packets input, 27324253 bytes, 0 no
> buffer
> 
>      Received 64162 broadcasts (0 multicast)
> 
>      0 runts, 1 giants, 0 throttles
> 
>      1 input errors, 0 CRC, 0 frame, 0 overrun, 0
> ignored
> 
>      0 watchdog, 60442 multicast, 0 pause input
> 
>      0 input packets with dribble condition detected
> 
>      629225 packets output, 130100556 bytes, 0
> underruns
> 
>      1 output errors, 1440 collisions, 1 interface
> resets
> 
>      0 babbles, 0 late collision, 0 deferred
> 
>      0 lost carrier, 0 no carrier, 0 PAUSE output
> 
>      0 output buffer failures, 0 output buffers
> swapped out
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From Scott.Voll at wesd.org  Wed Jun 21 16:57:17 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Wed, 21 Jun 2006 13:57:17 -0700
Subject: [cisco-voip] What causes a Err-disable?
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED4F40@Apollo.wesd.org>

Pretty much a normal Auto QoS IP Phone switch with a couple vlans.
Nothing for port security added.

 

________________________________

From: Brian Henry [mailto:Brian.Henry at apptis.com] 
Sent: Wednesday, June 21, 2006 1:52 PM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] What causes a Err-disable?

 

Do you have any thresholds put into place on the port or use port
security.

Brian
--------------------------
Sent from my BlackBerry Wireless Handheld


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
To: cisco-voip at puck.nether.net
Sent: Wed Jun 21 16:48:13 2006
Subject: [cisco-voip] What causes a Err-disable?

What causes a port on a 3560 to go into err-disable?  I know a shut no
shut fixes it but how can we fix it to not do it at all?





IE>



FastEthernet0/4 is down, line protocol is down (err-disabled)

  Hardware is Fast Ethernet, address is 0017.9468.0506 (bia
0017.9468.0506)

  MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec,

     reliability 255/255, txload 1/255, rxload 1/255

  Encapsulation ARPA, loopback not set

  Keepalive set (10 sec)

  Auto-duplex, Auto-speed, media type is 10/100BaseTX

  input flow-control is off, output flow-control is unsupported

  ARP type: ARPA, ARP Timeout 04:00:00

  Last input never, output 2w6d, output hang never

  Last clearing of "show interface" counters never

  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0

  Queueing strategy: fifo

  Output queue: 0/40 (size/max)

  5 minute input rate 0 bits/sec, 0 packets/sec

  5 minute output rate 0 bits/sec, 0 packets/sec

     167934 packets input, 27324253 bytes, 0 no buffer

     Received 64162 broadcasts (0 multicast)

     0 runts, 1 giants, 0 throttles

     1 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored

     0 watchdog, 60442 multicast, 0 pause input

     0 input packets with dribble condition detected

     629225 packets output, 130100556 bytes, 0 underruns

     1 output errors, 1440 collisions, 1 interface resets

     0 babbles, 0 late collision, 0 deferred

     0 lost carrier, 0 no carrier, 0 PAUSE output

     0 output buffer failures, 0 output buffers swapped out

-------------- next part --------------
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From tednugent69 at yahoo.com  Wed Jun 21 17:01:16 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 21 Jun 2006 14:01:16 -0700 (PDT)
Subject: [cisco-voip] What causes a Err-disable?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED4F3C@Apollo.wesd.org>
Message-ID: <20060621210116.50811.qmail@web36107.mail.mud.yahoo.com>

Also and more likely

Bad or non-Cisco-approved SFP. 
 Remove the SFP module from the switch, and replace it
with a Cisco-approved module. Use the errdisable
recovery cause gbic-invalid global configuration
command to verify the port status, and enter a time
interval to recover from the error-disable state. 

Refer to the switch command reference guide for
information on the errdisable recovery command. 
 

--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> What causes a port on a 3560 to go into err-disable?
>  I know a shut no
> shut fixes it but how can we fix it to not do it at
> all?
> 
>  
> 
>  
> 
> IE>
> 
>  
> 
> FastEthernet0/4 is down, line protocol is down
> (err-disabled)
> 
>   Hardware is Fast Ethernet, address is
> 0017.9468.0506 (bia
> 0017.9468.0506)
> 
>   MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec, 
> 
>      reliability 255/255, txload 1/255, rxload 1/255
> 
>   Encapsulation ARPA, loopback not set
> 
>   Keepalive set (10 sec)
> 
>   Auto-duplex, Auto-speed, media type is
> 10/100BaseTX
> 
>   input flow-control is off, output flow-control is
> unsupported 
> 
>   ARP type: ARPA, ARP Timeout 04:00:00
> 
>   Last input never, output 2w6d, output hang never
> 
>   Last clearing of "show interface" counters never
> 
>   Input queue: 0/75/0/0 (size/max/drops/flushes);
> Total output drops: 0
> 
>   Queueing strategy: fifo
> 
>   Output queue: 0/40 (size/max)
> 
>   5 minute input rate 0 bits/sec, 0 packets/sec
> 
>   5 minute output rate 0 bits/sec, 0 packets/sec
> 
>      167934 packets input, 27324253 bytes, 0 no
> buffer
> 
>      Received 64162 broadcasts (0 multicast)
> 
>      0 runts, 1 giants, 0 throttles
> 
>      1 input errors, 0 CRC, 0 frame, 0 overrun, 0
> ignored
> 
>      0 watchdog, 60442 multicast, 0 pause input
> 
>      0 input packets with dribble condition detected
> 
>      629225 packets output, 130100556 bytes, 0
> underruns
> 
>      1 output errors, 1440 collisions, 1 interface
> resets
> 
>      0 babbles, 0 late collision, 0 deferred
> 
>      0 lost carrier, 0 no carrier, 0 PAUSE output
> 
>      0 output buffer failures, 0 output buffers
> swapped out
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From tednugent69 at yahoo.com  Wed Jun 21 17:14:14 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 21 Jun 2006 14:14:14 -0700 (PDT)
Subject: [cisco-voip] What causes a Err-disable?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED4F3C@Apollo.wesd.org>
Message-ID: <20060621211414.67064.qmail@web36110.mail.mud.yahoo.com>

Also seen bad cabling cause Error disable, crossed
pair or likewise. Usually this is pretty immediate
unlike the others I've listed which can take some time
to "build up".


--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> What causes a port on a 3560 to go into err-disable?
>  I know a shut no
> shut fixes it but how can we fix it to not do it at
> all?
> 
>  
> 
>  
> 
> IE>
> 
>  
> 
> FastEthernet0/4 is down, line protocol is down
> (err-disabled)
> 
>   Hardware is Fast Ethernet, address is
> 0017.9468.0506 (bia
> 0017.9468.0506)
> 
>   MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec, 
> 
>      reliability 255/255, txload 1/255, rxload 1/255
> 
>   Encapsulation ARPA, loopback not set
> 
>   Keepalive set (10 sec)
> 
>   Auto-duplex, Auto-speed, media type is
> 10/100BaseTX
> 
>   input flow-control is off, output flow-control is
> unsupported 
> 
>   ARP type: ARPA, ARP Timeout 04:00:00
> 
>   Last input never, output 2w6d, output hang never
> 
>   Last clearing of "show interface" counters never
> 
>   Input queue: 0/75/0/0 (size/max/drops/flushes);
> Total output drops: 0
> 
>   Queueing strategy: fifo
> 
>   Output queue: 0/40 (size/max)
> 
>   5 minute input rate 0 bits/sec, 0 packets/sec
> 
>   5 minute output rate 0 bits/sec, 0 packets/sec
> 
>      167934 packets input, 27324253 bytes, 0 no
> buffer
> 
>      Received 64162 broadcasts (0 multicast)
> 
>      0 runts, 1 giants, 0 throttles
> 
>      1 input errors, 0 CRC, 0 frame, 0 overrun, 0
> ignored
> 
>      0 watchdog, 60442 multicast, 0 pause input
> 
>      0 input packets with dribble condition detected
> 
>      629225 packets output, 130100556 bytes, 0
> underruns
> 
>      1 output errors, 1440 collisions, 1 interface
> resets
> 
>      0 babbles, 0 late collision, 0 deferred
> 
>      0 lost carrier, 0 no carrier, 0 PAUSE output
> 
>      0 output buffer failures, 0 output buffers
> swapped out
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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From keith.klevenski at rig.net  Wed Jun 21 17:26:43 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Wed, 21 Jun 2006 16:26:43 -0500
Subject: [cisco-voip] Changing Windows Local Administrator Account
	Affects MLA?
Message-ID: <7F0CE0A1A0961E498A41329535342266A10798@HOUEXH02.rignet.org>

Thanks for the info Ryan.

Well, I went ahead and disabled MLA on the pub in SQL and I could log
into ccmadmin with the local password no problem.  While I was at that
point I went ahead and changed the local admin password again and
restarted IIS.  I could still log in to ccmadmin just fine.  But when I
reendabled MLA (through SQL) I still could not log in with any MLA
accounts, even ccmadministrator.  Same authentication error.  There
shouldn't be any difference enabling MLA in SQL than from the GUI (I
don't think) so I just reenabled it from SQL since I had EM open.  I
went ahead and reenabled MLA from the GUI, reset the ccmadministrator
password, restarted IIS on both servers in the cluster and still I
cannot log in with ccmadministrator or any MLA account.

So all I did (apparently) to cause MLA logins to fail authentication was
to change the local administrator password and restart the WWW service
on the publisher (but the failure was only on the publisher until I
changed the local admin account on the sub and now it is failing there
too).  

How could changing the local admin password cause MLA to fail
authentication?  Now I'm stuck without MLA until I figure out what the
issue is.  I've restarted IIS over and over, enabled/reenabled MLA from
SQL/GUI with no luck.  At least I can get in with MLA disabled, but this
is not desirable for many obvious reasons and I will need to get this
fixed quickly.

Any words of wisdom are welcome.  ;)

Keith

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com] 
Sent: Wednesday, June 21, 2006 3:30 PM
To: Keith Klevenski
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing Windows Local Administrator Account
Affects MLA?

In CM 4.x the MLA parameters are clusterwide since they are stored in  
the database.  In previous versions we actually pointed the CCMAdmin  
IIS virtual directory to a new location.  Now it's the behind-the- 
scenes authentication that is changed.

As long as you don't restart IIS on the sub no changes to MLA will be  
picked up there.

-Ryan

On Jun 21, 2006, at 2:51 PM, Keith Klevenski wrote:

Gotcha, I only restarted WWW.  Since I rebooted the server obviously IIS
was restarted and I did restart it again, but I still cannot get to
ccmadmin with any MLA account from the publisher.  I guess that is my
main question.  Should you disable MLA before changing the local admin
account?  Are there any dependencies between the local admin account and
MLA account?  I haven't seen any specific considerations regarding this.
It worked fine with 3.3.5 (with no MLA) and I only restarted WWW on
those servers.

Any information about the dependencies between the local admin account
and MLA would be welcome if anyone has any info.  I guess at this point
I don't have much choice but to disable MLA from SQL, restart IIS and
see if I can get in with the local admin account.  Then reenable MLA and
hope for the best.  I'm just afraid of locking myself out of ccmadmin as
right now I still have accesses through the sub.  Is there any risk of
locking yourself out of ccmadmin?  Can you disable MLA from the sub?  I
can get to ccmadmin from there and to the MLA parameters.

Thanks!

Keith


-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Wednesday, June 21, 2006 1:37 PM
To: Keith Klevenski
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing Windows Local Administrator Account
Affects MLA?

You can disable MLA just by going into the MLAParameters table in the
CCM030X database on the publisher.  The parameter name is similar to
MLAEnabled but should be obvious.  Then simply restart IIS (not just
WWW) and you should be good to go.

If you only restarted WWW and not IISAdmin I'd try bouncing both
before going into SQL to disable MLA.

-Ryan

On Jun 21, 2006, at 2:22 PM, Keith Klevenski wrote:

Hi all,



I have a 4.1.3 cluster that I need to change the local administrator
account to.  No problem on a couple of 3.3.5 clusters that are not
running MLA, but when I changed the local administrator account
password on the 4.1.3 publisher when I went to log in go ccmadmin
with my MLA account access is denied.  Same thing when logging in
with ccmadministrator.  Fortunately I can still get into the
subscriber (2 server cluster), but I'm not sure what the problem is.
I see these errors in the event log when attempting to log in to
ccmadmin with my MLA account:



The server was unable to logon the Windows NT account 'keith' due to
the following error: Logon failure: unknown user name or bad
password.  The data is the error code.

For additional information specific to this message please visit the
Microsoft Online Support site located at: http://www.microsoft.com/
contentredirect.asp.



I'm using the standard procedure for resetting the local
administrator account:



To Change Windows Administrator Password:

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

Please make sure that the Local Administrator password is SAME on all
servers in the

cluster.



To change the password, perform the following steps:



1. Right click My Computer.

2. Select Manage.

3. Go to Local Users and Groups.

4. Go to Users.

5. Right click Administrator.

6. Select Set Password.

7. Type in the password and confirm.



I am also restarting the WWW service after making the change.  I've
rebooted the publisher as well, but I must be missing something.  I
can't find any special instructions on changing local admin account
with MLA running.  Seems like it shouldn't have any affect on MLA.
I've changed the password back to the original and I'm still having
the same problem.  I was thinking about disabling MLA from the
subscriber and seeing if I can log in with local admin account to
ccmadmin, but I don't want to hose myself.



Any ideas?  TIA!!



Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

http://www.rig.net



"Extending the power of your knowledge"



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



From dphillips at sapc.edu  Wed Jun 21 17:28:45 2006
From: dphillips at sapc.edu (Phillips, Danny)
Date: Wed, 21 Jun 2006 17:28:45 -0400
Subject: [cisco-voip] What causes a Err-disable?
Message-ID: <18F0B5EFCCE1A449B07A3A9DF61DC07308ACDA@xchange.sapc.edu>

BPDU Guard will also cause this.


Danny 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ted Nugent
Sent: Wednesday, June 21, 2006 5:14 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] What causes a Err-disable?

Also seen bad cabling cause Error disable, crossed
pair or likewise. Usually this is pretty immediate
unlike the others I've listed which can take some time
to "build up".


--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> What causes a port on a 3560 to go into err-disable?
>  I know a shut no
> shut fixes it but how can we fix it to not do it at
> all?
> 
>  
> 
>  
> 
> IE>
> 
>  
> 
> FastEthernet0/4 is down, line protocol is down
> (err-disabled)
> 
>   Hardware is Fast Ethernet, address is
> 0017.9468.0506 (bia
> 0017.9468.0506)
> 
>   MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec, 
> 
>      reliability 255/255, txload 1/255, rxload 1/255
> 
>   Encapsulation ARPA, loopback not set
> 
>   Keepalive set (10 sec)
> 
>   Auto-duplex, Auto-speed, media type is
> 10/100BaseTX
> 
>   input flow-control is off, output flow-control is
> unsupported 
> 
>   ARP type: ARPA, ARP Timeout 04:00:00
> 
>   Last input never, output 2w6d, output hang never
> 
>   Last clearing of "show interface" counters never
> 
>   Input queue: 0/75/0/0 (size/max/drops/flushes);
> Total output drops: 0
> 
>   Queueing strategy: fifo
> 
>   Output queue: 0/40 (size/max)
> 
>   5 minute input rate 0 bits/sec, 0 packets/sec
> 
>   5 minute output rate 0 bits/sec, 0 packets/sec
> 
>      167934 packets input, 27324253 bytes, 0 no
> buffer
> 
>      Received 64162 broadcasts (0 multicast)
> 
>      0 runts, 1 giants, 0 throttles
> 
>      1 input errors, 0 CRC, 0 frame, 0 overrun, 0
> ignored
> 
>      0 watchdog, 60442 multicast, 0 pause input
> 
>      0 input packets with dribble condition detected
> 
>      629225 packets output, 130100556 bytes, 0
> underruns
> 
>      1 output errors, 1440 collisions, 1 interface
> resets
> 
>      0 babbles, 0 late collision, 0 deferred
> 
>      0 lost carrier, 0 no carrier, 0 PAUSE output
> 
>      0 output buffer failures, 0 output buffers
> swapped out
> 
> > _______________________________________________
> 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  Wed Jun 21 18:19:39 2006
From: it at cimgroup.com (IT)
Date: Wed, 21 Jun 2006 15:19:39 -0700
Subject: [cisco-voip] Nokia E61 CCM Client
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0340F462@exchange.cimgroup.com>

Hello,

I have the Nokia E61 (amazing little device), and I am dying to try the
CallManager Wifi integration. Does anyone know when Cisco will be
releasing the client? Or is this a message better suited for the Nokia
mailing list?

Here is the Nokia page:
http://www.businesssoftware.nokia.com/cisco_systems.php

Here is the Cisco page:
http://www.cisco.com/iam/unified/ipcc1/IP_Communications.htm


Anyone know if I can beta-test this?

Thanks,
Avidan


From bcarter at sentinel.com  Wed Jun 21 18:21:25 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Wed, 21 Jun 2006 17:21:25 -0500
Subject: [cisco-voip] What causes a Err-disable?
Message-ID: <6A0744639B86DD459433108DBF101D26FC656E@MAIL01.sentinel.com>

I always add...
 
errdisable recovery cause all
errdisable recovery interval 30
 
It can be difficult to install and configure new switches when
errdisable keeps shutting down ports.
 


________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
	Sent: Wednesday, June 21, 2006 3:48 PM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] What causes a Err-disable?
	
	

	What causes a port on a 3560 to go into err-disable?  I know a
shut no shut fixes it but how can we fix it to not do it at all?

	 

	 

	IE>

	 

	FastEthernet0/4 is down, line protocol is down (err-disabled)

	  Hardware is Fast Ethernet, address is 0017.9468.0506 (bia
0017.9468.0506)

	  MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec, 

	     reliability 255/255, txload 1/255, rxload 1/255

	  Encapsulation ARPA, loopback not set

	  Keepalive set (10 sec)

	  Auto-duplex, Auto-speed, media type is 10/100BaseTX

	  input flow-control is off, output flow-control is unsupported 

	  ARP type: ARPA, ARP Timeout 04:00:00

	  Last input never, output 2w6d, output hang never

	  Last clearing of "show interface" counters never

	  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output
drops: 0

	  Queueing strategy: fifo

	  Output queue: 0/40 (size/max)

	  5 minute input rate 0 bits/sec, 0 packets/sec

	  5 minute output rate 0 bits/sec, 0 packets/sec

	     167934 packets input, 27324253 bytes, 0 no buffer

	     Received 64162 broadcasts (0 multicast)

	     0 runts, 1 giants, 0 throttles

	     1 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored

	     0 watchdog, 60442 multicast, 0 pause input

	     0 input packets with dribble condition detected

	     629225 packets output, 130100556 bytes, 0 underruns

	     1 output errors, 1440 collisions, 1 interface resets

	     0 babbles, 0 late collision, 0 deferred

	     0 lost carrier, 0 no carrier, 0 PAUSE output

	     0 output buffer failures, 0 output buffers swapped out



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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 recipient, please contact the sender and 
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From bcarter at sentinel.com  Wed Jun 21 18:25:27 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Wed, 21 Jun 2006 17:25:27 -0500
Subject: [cisco-voip] Nokia E61 CCM Client
Message-ID: <6A0744639B86DD459433108DBF101D26FC656F@MAIL01.sentinel.com>

I think I saw you need to configure the phone in CallManager as a 7960. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, June 21, 2006 5:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Nokia E61 CCM Client

Hello,

I have the Nokia E61 (amazing little device), and I am dying to try the
CallManager Wifi integration. Does anyone know when Cisco will be
releasing the client? Or is this a message better suited for the Nokia
mailing list?

Here is the Nokia page:
http://www.businesssoftware.nokia.com/cisco_systems.php

Here is the Cisco page:
http://www.cisco.com/iam/unified/ipcc1/IP_Communications.htm


Anyone know if I can beta-test this?

Thanks,
Avidan

_______________________________________________
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 d_abelski at inbox.ru  Wed Jun 21 23:19:11 2006
From: d_abelski at inbox.ru (d_abelski at inbox.ru)
Date: Thu, 22 Jun 2006 09:19:11 +0600
Subject: [cisco-voip] Looking for IP Video Surveillance
In-Reply-To: <17A170DC8DD5CC44BA56B24B366EB60B01EDFD81@E3K-VS1.ccsi.local>
References: <17A170DC8DD5CC44BA56B24B366EB60B01EDFD81@E3K-VS1.ccsi.local>
Message-ID: <1322697129.20060622091911@inbox.ru>

Hi. Not really dealing with that, but.
We did IP Video Surveillance using Axis hardware and Software and
Sony. Try to look at them. Can`t say anything bad about both.

Regards,
Daniel Abelski

SOD> I know it doesn't fall directly in line with IPT but I
SOD> figured many who support IPT would support IP Video Surveillance
SOD> design.
SOD> ?
SOD> I'm interested in finding out who are the big names in IP Video Surveillance.
SOD> I need a scalable solution with centralized management.
SOD> ?
SOD> Any opinions are appreciated.
SOD> ?
SOD> Scott
SOD> ?


SOD> __________ ?????????? NOD32 1.1613 (20060621) __________

SOD> ??? ????????? ????????? ???????????? ???????? NOD32.
SOD> http://www.eset.com





-- 
? ?????????,
 d                          mailto:d_abelski at inbox.ru


From alfred.thyri at mii.at  Thu Jun 22 02:45:56 2006
From: alfred.thyri at mii.at (Thyri Alfred)
Date: Thu, 22 Jun 2006 08:45:56 +0200
Subject: [cisco-voip] CME - IP Phone "?" button
References: <6A0744639B86DD459433108DBF101D26FC6547@MAIL01.sentinel.com>
Message-ID: <18576071F1E2B047A6850107573475D48F8FF1@smiivie0123.at-work.ent>

Yes this is normal. Although it is described in the official
administration guide
(http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps5012/products
_feature_guide_chapter09186a00801812e1.html#wp2175456), it is not
supported, because there is no XML support on a router...

If you also have an installed CCM than you could point the link to the
CCM.

Alfred.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Mittwoch, 21. Juni 2006 20:14
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CME - IP Phone "?" button

 
I just completed my first CME installation.  On the IP Phones, when a
user hits the ? they don't get any help files.  It is like there is no
help files on CME.  Is this normal??


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.

_______________________________________________
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From roharvey at flemingc.on.ca  Thu Jun 22 08:50:55 2006
From: roharvey at flemingc.on.ca (Robert Harvey)
Date: Thu, 22 Jun 2006 08:50:55 -0400
Subject: [cisco-voip] Looking for IP Video Surveillance
Message-ID: <s49a59fb.045@freedom7.flemingc.on.ca>

Hi Scott,

We have been using Axis IP Cameras (http://www.axis.com) connected to a
Milestone XProtect DVR Server (http://www.milestonesys.com) with great
results.  We particularly enjoy that the entire system is IP and
therefore we are able to extend security cameras wherever we have
ethernet connectivity.  The reason we went with Milestone's DVR server
was that it works extremely well with Axis cameras and support PTZ
cameras for motion detection recording.  I hope this helps.

Regards,

Robert.

 

----------------------------------------------------------------
Robert Harvey
Network Support Analyst
Fleming College
Peterborough, ON, Canada
Office (705) 749-5513
Fax (705) 749-5540
experience.success
----------------------------------------------------------------

>>>>>>>>>>>
I know it doesn't fall directly in line with IPT but I figured many
who
support IPT would support IP Video Surveillance design.
 
I'm interested in finding out who are the big names in IP Video
Surveillance.
I need a scalable solution with centralized management.
 
Any opinions are appreciated.
 
Scott
>>>>>>>>>>>




From rvillagran at coasin.cl  Thu Jun 22 09:39:12 2006
From: rvillagran at coasin.cl (Rodrigo Villagran)
Date: Thu, 22 Jun 2006 09:39:12 -0400
Subject: [cisco-voip] What causes a Err-disable?
In-Reply-To: <6A0744639B86DD459433108DBF101D26FC656E@MAIL01.sentinel.com>
Message-ID: <200606221338.k5MDchsQ001175@mail.coasin.cl>

Please send a sh logg to see if there some message of the motive of the
err-disable, also send a sh errdisable detect to see the events that send
the port to the err-disable

 

Rodrigo 

 

  _____  

De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de Carter, Bill
Enviado el: mi?rcoles, 21 de junio de 2006 18:21
Para: Voll, Scott; cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] What causes a Err-disable?

 

I always add...

 

errdisable recovery cause all

errdisable recovery interval 30

 

It can be difficult to install and configure new switches when errdisable
keeps shutting down ports.

 

 


  _____  


From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Wednesday, June 21, 2006 3:48 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] What causes a Err-disable?

What causes a port on a 3560 to go into err-disable?  I know a shut no shut
fixes it but how can we fix it to not do it at all?

 

 

IE>

 

FastEthernet0/4 is down, line protocol is down (err-disabled)

  Hardware is Fast Ethernet, address is 0017.9468.0506 (bia 0017.9468.0506)

  MTU 1500 bytes, BW 10000 Kbit, DLY 1000 usec, 

     reliability 255/255, txload 1/255, rxload 1/255

  Encapsulation ARPA, loopback not set

  Keepalive set (10 sec)

  Auto-duplex, Auto-speed, media type is 10/100BaseTX

  input flow-control is off, output flow-control is unsupported 

  ARP type: ARPA, ARP Timeout 04:00:00

  Last input never, output 2w6d, output hang never

  Last clearing of "show interface" counters never

  Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0

  Queueing strategy: fifo

  Output queue: 0/40 (size/max)

  5 minute input rate 0 bits/sec, 0 packets/sec

  5 minute output rate 0 bits/sec, 0 packets/sec

     167934 packets input, 27324253 bytes, 0 no buffer

     Received 64162 broadcasts (0 multicast)

     0 runts, 1 giants, 0 throttles

     1 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored

     0 watchdog, 60442 multicast, 0 pause input

     0 input packets with dribble condition detected

     629225 packets output, 130100556 bytes, 0 underruns

     1 output errors, 1440 collisions, 1 interface resets

     0 babbles, 0 late collision, 0 deferred

     0 lost carrier, 0 no carrier, 0 PAUSE output

     0 output buffer failures, 0 output buffers swapped out

 
 
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Any review, retransmission, dissemination, or other use of this information
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From rratliff at cisco.com  Thu Jun 22 09:51:24 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 22 Jun 2006 09:51:24 -0400
Subject: [cisco-voip] Changing Windows Local Administrator Account
	Affects MLA?
In-Reply-To: <7F0CE0A1A0961E498A41329535342266A10798@HOUEXH02.rignet.org>
References: <7F0CE0A1A0961E498A41329535342266A10798@HOUEXH02.rignet.org>
Message-ID: <FAB393A5-DC68-4C76-A0B9-ADE8BA6DCFE1@cisco.com>

Make sure MLA debugs are enabled, gather them and open a TAC SR.

-Ryan

On Jun 21, 2006, at 5:26 PM, Keith Klevenski wrote:

Thanks for the info Ryan.

Well, I went ahead and disabled MLA on the pub in SQL and I could log
into ccmadmin with the local password no problem.  While I was at that
point I went ahead and changed the local admin password again and
restarted IIS.  I could still log in to ccmadmin just fine.  But when I
reendabled MLA (through SQL) I still could not log in with any MLA
accounts, even ccmadministrator.  Same authentication error.  There
shouldn't be any difference enabling MLA in SQL than from the GUI (I
don't think) so I just reenabled it from SQL since I had EM open.  I
went ahead and reenabled MLA from the GUI, reset the ccmadministrator
password, restarted IIS on both servers in the cluster and still I
cannot log in with ccmadministrator or any MLA account.

So all I did (apparently) to cause MLA logins to fail authentication was
to change the local administrator password and restart the WWW service
on the publisher (but the failure was only on the publisher until I
changed the local admin account on the sub and now it is failing there
too).

How could changing the local admin password cause MLA to fail
authentication?  Now I'm stuck without MLA until I figure out what the
issue is.  I've restarted IIS over and over, enabled/reenabled MLA from
SQL/GUI with no luck.  At least I can get in with MLA disabled, but this
is not desirable for many obvious reasons and I will need to get this
fixed quickly.

Any words of wisdom are welcome.  ;)

Keith

-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Wednesday, June 21, 2006 3:30 PM
To: Keith Klevenski
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing Windows Local Administrator Account
Affects MLA?

In CM 4.x the MLA parameters are clusterwide since they are stored in
the database.  In previous versions we actually pointed the CCMAdmin
IIS virtual directory to a new location.  Now it's the behind-the-
scenes authentication that is changed.

As long as you don't restart IIS on the sub no changes to MLA will be
picked up there.

-Ryan

On Jun 21, 2006, at 2:51 PM, Keith Klevenski wrote:

Gotcha, I only restarted WWW.  Since I rebooted the server obviously IIS
was restarted and I did restart it again, but I still cannot get to
ccmadmin with any MLA account from the publisher.  I guess that is my
main question.  Should you disable MLA before changing the local admin
account?  Are there any dependencies between the local admin account and
MLA account?  I haven't seen any specific considerations regarding this.
It worked fine with 3.3.5 (with no MLA) and I only restarted WWW on
those servers.

Any information about the dependencies between the local admin account
and MLA would be welcome if anyone has any info.  I guess at this point
I don't have much choice but to disable MLA from SQL, restart IIS and
see if I can get in with the local admin account.  Then reenable MLA and
hope for the best.  I'm just afraid of locking myself out of ccmadmin as
right now I still have accesses through the sub.  Is there any risk of
locking yourself out of ccmadmin?  Can you disable MLA from the sub?  I
can get to ccmadmin from there and to the MLA parameters.

Thanks!

Keith


-----Original Message-----
From: Ryan Ratliff [mailto:rratliff at cisco.com]
Sent: Wednesday, June 21, 2006 1:37 PM
To: Keith Klevenski
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Changing Windows Local Administrator Account
Affects MLA?

You can disable MLA just by going into the MLAParameters table in the
CCM030X database on the publisher.  The parameter name is similar to
MLAEnabled but should be obvious.  Then simply restart IIS (not just
WWW) and you should be good to go.

If you only restarted WWW and not IISAdmin I'd try bouncing both
before going into SQL to disable MLA.

-Ryan

On Jun 21, 2006, at 2:22 PM, Keith Klevenski wrote:

Hi all,



I have a 4.1.3 cluster that I need to change the local administrator
account to.  No problem on a couple of 3.3.5 clusters that are not
running MLA, but when I changed the local administrator account
password on the 4.1.3 publisher when I went to log in go ccmadmin
with my MLA account access is denied.  Same thing when logging in
with ccmadministrator.  Fortunately I can still get into the
subscriber (2 server cluster), but I'm not sure what the problem is.
I see these errors in the event log when attempting to log in to
ccmadmin with my MLA account:



The server was unable to logon the Windows NT account 'keith' due to
the following error: Logon failure: unknown user name or bad
password.  The data is the error code.

For additional information specific to this message please visit the
Microsoft Online Support site located at: http://www.microsoft.com/
contentredirect.asp.



I'm using the standard procedure for resetting the local
administrator account:



To Change Windows Administrator Password:

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

Please make sure that the Local Administrator password is SAME on all
servers in the

cluster.



To change the password, perform the following steps:



1. Right click My Computer.

2. Select Manage.

3. Go to Local Users and Groups.

4. Go to Users.

5. Right click Administrator.

6. Select Set Password.

7. Type in the password and confirm.



I am also restarting the WWW service after making the change.  I've
rebooted the publisher as well, but I must be missing something.  I
can't find any special instructions on changing local admin account
with MLA running.  Seems like it shouldn't have any affect on MLA.
I've changed the password back to the original and I'm still having
the same problem.  I was thinking about disabling MLA from the
subscriber and seeing if I can log in with local admin account to
ccmadmin, but I don't want to hose myself.



Any ideas?  TIA!!



Keith Klevenski
IP Telephony Engineer

RigNet Inc.
1880 S. Dairy Ashford

Suite 300
Houston, TX 77077

Office:   +1 281.674.0702
Mobile:  +1 713.677.3925

http://www.rig.net



"Extending the power of your knowledge"



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


From LSOCARRAS at broward.org  Thu Jun 22 10:00:56 2006
From: LSOCARRAS at broward.org (Socarras, Luis)
Date: Thu, 22 Jun 2006 10:00:56 -0400
Subject: [cisco-voip] Editing Missed Calls
Message-ID: <5BFD4B266FCBED44938AE3BB35ACD0AA014FF197@UVEXVS01.bc.broward.cty>

Does anybody know if there is any way to have a 9 automatically added to
an entry in the Missed Call directory as opposed to pressing EditDial
and then 9. 

 

Thanks,

Luis

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From MLinsemier at apcapital.com  Thu Jun 22 10:16:54 2006
From: MLinsemier at apcapital.com (Linsemier, Matthew)
Date: Thu, 22 Jun 2006 10:16:54 -0400
Subject: [cisco-voip] Editing Missed Calls
Message-ID: <AB0A35F2486F81468971AA68558BE7190403783A@smieln01winm01.micoanet.com>

Luis,

 

If you are MGCP, you can do the following:

With the release of Cisco CallManager version 3.3(3)SR1, there is an
alternative solution available for incoming calls that arrive from a
Q.931 gateway. Four new parameters are added to Cisco CallManager under
the Clusterwide Parameters (Device - PRI and MGCP Gateway) section, in
the service parameters for Cisco CallManager. They are:

*         National Number Prefix-This parameter defines the number that
is prefixed to an incoming national number and provides a means to help
identify national numbers, if necessary. This parameter allows you to
prefix specified digits to the calling number of an inbound call based
on the Type of Number field in an inbound offered call (ISDN SETUP
message). For example, a national call from 4085551212 can be assigned
the prefix 91. The maximum length is eight digits, and the allowed
values are 0 through 9, #, and *.

*         International Number Prefix-This parameter defines the number
that is prefixed to an incoming international number and provides a
means to help identify international numbers, if necessary. This
parameter allows you to prefix specified digits to the calling number of
an inbound call based on the Type of Number field in an inbound offered
call (ISDN SETUP message). For example, an international call from
456789 can be assigned the prefix 9 011. The maximum length is eight
digits, and the allowed values are 0 through 9, #, and *.

*         Subscriber Number Prefix-This parameter defines the number
that is prefixed to an incoming subscriber number, and provides a means
to help identify subscriber numbers, if necessary. This parameter allows
you to prefix specified digits to the calling number of an inbound call
based on the Type of Number field in an inbound offered call (ISDN setup
message). For example, a subscriber call from 1212 can be assigned the
prefix 9. The maximum length is eight digits, and the allowed values are
0 through 9, #, and *.

*         Unknown Number Prefix-This parameter defines the number that
is prefixed to an incoming unknown number and provides a means to help
identify unknown number types, if necessary. This parameter allows you
to prefix specified digits to the calling number of an inbound call
based on the Type of Number field in an inbound offered call (ISDN SETUP
message). For example, an unknown call from 5550123 can be assigned the
prefix 7. The maximum length is eight digits, and the allowed values are
0 through 9, #, and *.

Note:  These prefix values are added to numbers in missed or received
incoming calls based on the inbound Q.931 call type value. This method
only applies to Q.931 gateways. As for all other Cisco CallManager
versions, the National Number Prefix can be set to 91.

If you are H.323, you can do the following:

 

Create a translation rule similar to the following:

 

voice translation-rule 1

 rule 1 /^517/ /9517/

!

voice translation-profile ParseInbound

 translate calling 1

 

dial-peer voice 100 pots

 translation-profile incoming ParseInbound

 destination-pattern 9T

 incoming called-number .

 direct-inward-dial

 port 0/2/0:23

 

Granted, this will only translate numbers from the 517 area code, but
you get the idea.  I don't have a H.323 gateway in front of my so I
can't test it, but this should work.

 

-Matt

 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Socarras, Luis
Sent: Thursday, June 22, 2006 10:01 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Editing Missed Calls

 

Does anybody know if there is any way to have a 9 automatically added to
an entry in the Missed Call directory as opposed to pressing EditDial
and then 9. 

 

Thanks,

Luis


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From jsteinberg at gmail.com  Thu Jun 22 10:19:22 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 22 Jun 2006 10:19:22 -0400
Subject: [cisco-voip] Editing Missed Calls
In-Reply-To: <5BFD4B266FCBED44938AE3BB35ACD0AA014FF197@UVEXVS01.bc.broward.cty>
References: <5BFD4B266FCBED44938AE3BB35ACD0AA014FF197@UVEXVS01.bc.broward.cty>
Message-ID: <f1b65f880606220719l7085aac7x90be9194b9afea41@mail.gmail.com>

Here you go...

http://www.ciscotaccc.com/voice/showcase?case=K12586600

You may have to manipulate it a bit for your CCM environment but this
details the basic concept.

Justin

On 6/22/06, Socarras, Luis <LSOCARRAS at broward.org> wrote:
>
>
>
> Does anybody know if there is any way to have a 9 automatically added to an
> entry in the Missed Call directory as opposed to pressing EditDial and then
> 9.
>
>
>
> Thanks,
>
> Luis
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From zephy316 at gmail.com  Thu Jun 22 11:29:55 2006
From: zephy316 at gmail.com (Israel Lang)
Date: Thu, 22 Jun 2006 11:29:55 -0400
Subject: [cisco-voip] Echo Problems
In-Reply-To: <BF46DA14-97F3-4A71-9C2A-BEB130F01368@mimectl>
References: <BF46DA14-97F3-4A71-9C2A-BEB130F01368@mimectl>
Message-ID: <91e236d30606220829g3222e2d2i72238deec7599a84@mail.gmail.com>

I can't remember the exact version of 12.4 but I did experience problems. I
think it might be more the DSP firmware that may be the problem. I
downgraded my IOS to a 12.3 version that was working fine and still had the
issue. Got a firmware update for the DSPs and that seems to have resolved
the issue.

On 6/21/06, Jason Burton <jburton at netechcorp.com> wrote:
>
>  Has anyone experienced any problems with echo after moving to 12.4(6)T2?
> I had to move to this version to fix another issue, but now I'm getting
> complaints of echo on a couple of PRIs.
>
> _______________________________________________
> 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 SPackett at fenwick.com  Thu Jun 22 12:36:34 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Thu, 22 Jun 2006 09:36:34 -0700
Subject: [cisco-voip] Unified Operations Manager 1.1 - Anyone installed it
	yet?
Message-ID: <965C0255FD5828438E34F9A68ECA76E7778DAD@svcexch02.firm.fenwick.llp>

I just got the CDs for Unified Operations Manager 1.1, but have not yet
installed it as I have to decommission and rebuild another server for
it.  Has anyone installed it yet?  What kind of load does it put on the
server?

Thanks in advance.
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From BillS at tns.its.psu.edu  Thu Jun 22 12:59:29 2006
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Thu, 22 Jun 2006 12:59:29 -0400
Subject: [cisco-voip] OS 2000.4.2 sr7 upgrade
Message-ID: <07A14D6987A5F647B2415D72EDF63B560597F988@tnsex01.tns.its.psu.edu>

Yesterday I attempted an upgrade on our cluster from 2000.2.7sr4 to 2000.4.2
then applied sr7.

CallManager version is 4.1(2)sr1.  Cluster hardware all around is HP DL-380
G3 (like MCS-7835 or 7845)

The short story is that after the upgrade and a few hours of frustrating
poor or non-existent phone service, we bailed out to a set of mirrors taken
the day before.  The main symptom of a sick CM cluster was that on the
subscribers where phones registered, CPU loads upon boot went to 100% and
stayed there.  CCM.EXE was the process using most of the CPU.  The high load
caused failed registrations, incredibly slow response times on phones, VM
and gateway ports registering and then going unregistered, and other
weirdness.

After an hour and a half talking to TAC on a P1 case we were unable to wait
any longer and had to bail out.

The case is still open and we're hoping to get to the bottom of it but I'm
putting this out there in case any of you have seen something like this,
have feedback/thoughts, etc.

---
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/


From rratliff at cisco.com  Thu Jun 22 13:04:10 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 22 Jun 2006 13:04:10 -0400
Subject: [cisco-voip] OS 2000.4.2 sr7 upgrade
In-Reply-To: <07A14D6987A5F647B2415D72EDF63B560597F988@tnsex01.tns.its.psu.edu>
References: <07A14D6987A5F647B2415D72EDF63B560597F988@tnsex01.tns.its.psu.edu>
Message-ID: <B17FEC5E-0A02-4316-A8C0-EE9E3183F28A@cisco.com>

This is a known bug and SR7 should no longer be on CCO because of it.

CSCse47327 is the bug.

Please unicast me the SR number and I'll check it out.

-Ryan

On Jun 22, 2006, at 12:59 PM, Simon, Bill wrote:

Yesterday I attempted an upgrade on our cluster from 2000.2.7sr4 to  
2000.4.2
then applied sr7.

CallManager version is 4.1(2)sr1.  Cluster hardware all around is HP  
DL-380
G3 (like MCS-7835 or 7845)

The short story is that after the upgrade and a few hours of frustrating
poor or non-existent phone service, we bailed out to a set of mirrors  
taken
the day before.  The main symptom of a sick CM cluster was that on the
subscribers where phones registered, CPU loads upon boot went to 100%  
and
stayed there.  CCM.EXE was the process using most of the CPU.  The  
high load
caused failed registrations, incredibly slow response times on  
phones, VM
and gateway ports registering and then going unregistered, and other
weirdness.

After an hour and a half talking to TAC on a P1 case we were unable  
to wait
any longer and had to bail out.

The case is still open and we're hoping to get to the bottom of it  
but I'm
putting this out there in case any of you have seen something like this,
have feedback/thoughts, etc.

---
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 cj.alan at gmail.com  Thu Jun 22 13:24:37 2006
From: cj.alan at gmail.com (Alan Su)
Date: Fri, 23 Jun 2006 01:24:37 +0800
Subject: [cisco-voip] CallManager Handle Call Routing with PBX
	interoperability
Message-ID: <8f9f19f00606221024r634f68c4o45d37e8d14de3133@mail.gmail.com>

Dear All,

I have a question about the PBX interoperability with CCM. What can I do to
make CCM to handle all the call routing even for the PBX end-phone?
One of the solution I heard about is to use SIP trunk connecting with CCM &
PBX, but I'm not quite sure how to do that. Any suggestion or comment would
be very grateful. Thanks.


Regards,

Alan
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From rratliff at cisco.com  Thu Jun 22 13:33:39 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 22 Jun 2006 13:33:39 -0400
Subject: [cisco-voip] CallManager Handle Call Routing with PBX
	interoperability
In-Reply-To: <8f9f19f00606221024r634f68c4o45d37e8d14de3133@mail.gmail.com>
References: <8f9f19f00606221024r634f68c4o45d37e8d14de3133@mail.gmail.com>
Message-ID: <DF84CA1D-C978-4B9D-9043-210F65D4DEC2@cisco.com>

Basically you'll need some type of voice gateway to receive calls  
from the PBX (via PRI or other physical link) and send them to the  
CM.     Once the call hits the CM it gets routed normally to a phone,  
route pattern, etc.

-Ryan

On Jun 22, 2006, at 1:24 PM, Alan Su wrote:

Dear All,

I have a question about the PBX interoperability with CCM. What can I  
do to make CCM to handle all the call routing even for the PBX end- 
phone?
One of the solution I heard about is to use SIP trunk connecting with  
CCM & PBX, but I'm not quite sure how to do that. Any suggestion or  
comment would be very grateful. Thanks.


Regards,

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


From CORTIZ at broward.org  Thu Jun 22 13:34:50 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Thu, 22 Jun 2006 13:34:50 -0400
Subject: [cisco-voip] OS 2000.4.2 sr7 upgrade
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC3528@MREXVS01.bc.broward.cty>

That's weird.  I upgraded to this about 3 weeks ago and I have no issue.
Why would that be???

-----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, June 22, 2006 1:04 PM
To: Simon, Bill
Cc: 'cisco-voip at puck.nether.net'
Subject: Re: [cisco-voip] OS 2000.4.2 sr7 upgrade

This is a known bug and SR7 should no longer be on CCO because of it.

CSCse47327 is the bug.

Please unicast me the SR number and I'll check it out.

-Ryan

On Jun 22, 2006, at 12:59 PM, Simon, Bill wrote:

Yesterday I attempted an upgrade on our cluster from 2000.2.7sr4 to  
2000.4.2
then applied sr7.

CallManager version is 4.1(2)sr1.  Cluster hardware all around is HP  
DL-380
G3 (like MCS-7835 or 7845)

The short story is that after the upgrade and a few hours of frustrating
poor or non-existent phone service, we bailed out to a set of mirrors  
taken
the day before.  The main symptom of a sick CM cluster was that on the
subscribers where phones registered, CPU loads upon boot went to 100%  
and
stayed there.  CCM.EXE was the process using most of the CPU.  The  
high load
caused failed registrations, incredibly slow response times on  
phones, VM
and gateway ports registering and then going unregistered, and other
weirdness.

After an hour and a half talking to TAC on a P1 case we were unable  
to wait
any longer and had to bail out.

The case is still open and we're hoping to get to the bottom of it  
but I'm
putting this out there in case any of you have seen something like this,
have feedback/thoughts, etc.

---
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


From rratliff at cisco.com  Thu Jun 22 13:35:46 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 22 Jun 2006 13:35:46 -0400
Subject: [cisco-voip] OS 2000.4.2 sr7 upgrade
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC3528@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC3528@MREXVS01.bc.broward.cty>
Message-ID: <E961622B-19C7-4A8A-B273-A44785991195@cisco.com>

Lucky?  Check out the bug details to see the registry change that  
caused the problem.  You can then check your server.

-Ryan

On Jun 22, 2006, at 1:34 PM, Ortiz, Carlos wrote:

That's weird.  I upgraded to this about 3 weeks ago and I have no issue.
Why would that be???

-----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, June 22, 2006 1:04 PM
To: Simon, Bill
Cc: 'cisco-voip at puck.nether.net'
Subject: Re: [cisco-voip] OS 2000.4.2 sr7 upgrade

This is a known bug and SR7 should no longer be on CCO because of it.

CSCse47327 is the bug.

Please unicast me the SR number and I'll check it out.

-Ryan

On Jun 22, 2006, at 12:59 PM, Simon, Bill wrote:

Yesterday I attempted an upgrade on our cluster from 2000.2.7sr4 to
2000.4.2
then applied sr7.

CallManager version is 4.1(2)sr1.  Cluster hardware all around is HP
DL-380
G3 (like MCS-7835 or 7845)

The short story is that after the upgrade and a few hours of frustrating
poor or non-existent phone service, we bailed out to a set of mirrors
taken
the day before.  The main symptom of a sick CM cluster was that on the
subscribers where phones registered, CPU loads upon boot went to 100%
and
stayed there.  CCM.EXE was the process using most of the CPU.  The
high load
caused failed registrations, incredibly slow response times on
phones, VM
and gateway ports registering and then going unregistered, and other
weirdness.

After an hour and a half talking to TAC on a P1 case we were unable
to wait
any longer and had to bail out.

The case is still open and we're hoping to get to the bottom of it
but I'm
putting this out there in case any of you have seen something like this,
have feedback/thoughts, etc.

---
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


From jsteinberg at gmail.com  Thu Jun 22 13:47:02 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 22 Jun 2006 13:47:02 -0400
Subject: [cisco-voip] OS 2000.4.2 sr7 upgrade
In-Reply-To: <E961622B-19C7-4A8A-B273-A44785991195@cisco.com>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC3528@MREXVS01.bc.broward.cty>
	<E961622B-19C7-4A8A-B273-A44785991195@cisco.com>
Message-ID: <f1b65f880606221047o7e62fc2dx2106a7df645f67b0@mail.gmail.com>

The information in the bug id states that no problems have been
observed with uniprocessor servers.

So does this mean that even the QA lab doesn't get new hardware? :)

justin




On 6/22/06, Ryan Ratliff <rratliff at cisco.com> wrote:
> Lucky?  Check out the bug details to see the registry change that
> caused the problem.  You can then check your server.
>
> -Ryan
>
> On Jun 22, 2006, at 1:34 PM, Ortiz, Carlos wrote:
>
> That's weird.  I upgraded to this about 3 weeks ago and I have no issue.
> Why would that be???
>
> -----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, June 22, 2006 1:04 PM
> To: Simon, Bill
> Cc: 'cisco-voip at puck.nether.net'
> Subject: Re: [cisco-voip] OS 2000.4.2 sr7 upgrade
>
> This is a known bug and SR7 should no longer be on CCO because of it.
>
> CSCse47327 is the bug.
>
> Please unicast me the SR number and I'll check it out.
>
> -Ryan
>
> On Jun 22, 2006, at 12:59 PM, Simon, Bill wrote:
>
> Yesterday I attempted an upgrade on our cluster from 2000.2.7sr4 to
> 2000.4.2
> then applied sr7.
>
> CallManager version is 4.1(2)sr1.  Cluster hardware all around is HP
> DL-380
> G3 (like MCS-7835 or 7845)
>
> The short story is that after the upgrade and a few hours of frustrating
> poor or non-existent phone service, we bailed out to a set of mirrors
> taken
> the day before.  The main symptom of a sick CM cluster was that on the
> subscribers where phones registered, CPU loads upon boot went to 100%
> and
> stayed there.  CCM.EXE was the process using most of the CPU.  The
> high load
> caused failed registrations, incredibly slow response times on
> phones, VM
> and gateway ports registering and then going unregistered, and other
> weirdness.
>
> After an hour and a half talking to TAC on a P1 case we were unable
> to wait
> any longer and had to bail out.
>
> The case is still open and we're hoping to get to the bottom of it
> but I'm
> putting this out there in case any of you have seen something like this,
> have feedback/thoughts, etc.
>
> ---
> 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
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From cj.alan at gmail.com  Thu Jun 22 13:48:45 2006
From: cj.alan at gmail.com (Alan Su)
Date: Fri, 23 Jun 2006 01:48:45 +0800
Subject: [cisco-voip] CallManager Handle Call Routing with PBX
	interoperability
In-Reply-To: <DF84CA1D-C978-4B9D-9043-210F65D4DEC2@cisco.com>
References: <8f9f19f00606221024r634f68c4o45d37e8d14de3133@mail.gmail.com>
	<DF84CA1D-C978-4B9D-9043-210F65D4DEC2@cisco.com>
Message-ID: <8f9f19f00606221048o68c769fev564fbb242f141c22@mail.gmail.com>

Hi Ryan,

Yes, I've tried to use IOS MGCP gateway and it worked. But it will occupy
the channel between GW-PBX. For example, extension 1315 & 1316 are PBX
analog phone, if I transfer all calls from PBX to GW and then to CCM for
call routing, even those 2 extensions will occupy the channel for in/out
calls. Unless I configure all the analog extensions in PBX and forward
others to GW. However, it'll cause other issues such as when I dialing
1315xxxx for off-net call, the PBX will treat it as 1315 extension. Because
the PBX can be upgraded to IP-based, so I was wondering is there any
solution like SIP-trunk can solve the issue. Thanks a lot ~


Regards,

Alan


2006/6/23, Ryan Ratliff <rratliff at cisco.com>:
>
> Basically you'll need some type of voice gateway to receive calls
> from the PBX (via PRI or other physical link) and send them to the
> CM.     Once the call hits the CM it gets routed normally to a phone,
> route pattern, etc.
>
> -Ryan
>
> On Jun 22, 2006, at 1:24 PM, Alan Su wrote:
>
> Dear All,
>
> I have a question about the PBX interoperability with CCM. What can I
> do to make CCM to handle all the call routing even for the PBX end-
> phone?
> One of the solution I heard about is to use SIP trunk connecting with
> CCM & PBX, but I'm not quite sure how to do that. Any suggestion or
> comment would be very grateful. Thanks.
>
>
> Regards,
>
> Alan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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From rratliff at cisco.com  Thu Jun 22 13:56:12 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 22 Jun 2006 13:56:12 -0400
Subject: [cisco-voip] CallManager Handle Call Routing with PBX
	interoperability
In-Reply-To: <8f9f19f00606221048o68c769fev564fbb242f141c22@mail.gmail.com>
References: <8f9f19f00606221024r634f68c4o45d37e8d14de3133@mail.gmail.com>
	<DF84CA1D-C978-4B9D-9043-210F65D4DEC2@cisco.com>
	<8f9f19f00606221048o68c769fev564fbb242f141c22@mail.gmail.com>
Message-ID: <48D167BB-C1B2-4404-BBA2-F4B4F2343B82@cisco.com>

I've no experience with a sip trunk to a PBX I'm afraid.

I have seen some PBXs that could act as an H.323 gateway to CM.  IIRC  
the only trick was to make sure "MTP Required" was set due to H.323  
version issues.

-Ryan

On Jun 22, 2006, at 1:48 PM, Alan Su wrote:

Hi Ryan,

Yes, I've tried to use IOS MGCP gateway and it worked. But it will  
occupy the channel between GW-PBX. For example, extension 1315 & 1316  
are PBX analog phone, if I transfer all calls from PBX to GW and then  
to CCM for call routing, even those 2 extensions will occupy the  
channel for in/out calls. Unless I configure all the analog  
extensions in PBX and forward others to GW. However, it'll cause  
other issues such as when I dialing 1315xxxx for off-net call, the  
PBX will treat it as 1315 extension. Because the PBX can be upgraded  
to IP-based, so I was wondering is there any solution like SIP-trunk  
can solve the issue. Thanks a lot ~


Regards,

Alan


2006/6/23, Ryan Ratliff <rratliff at cisco.com>: Basically you'll need  
some type of voice gateway to receive calls
from the PBX (via PRI or other physical link) and send them to the
CM.     Once the call hits the CM it gets routed normally to a phone,
route pattern, etc.

-Ryan

On Jun 22, 2006, at 1:24 PM, Alan Su wrote:

Dear All,

I have a question about the PBX interoperability with CCM. What can I
do to make CCM to handle all the call routing even for the PBX end-
phone?
One of the solution I heard about is to use SIP trunk connecting with
CCM & PBX, but I'm not quite sure how to do that. Any suggestion or
comment would be very grateful. Thanks.


Regards,

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




From CORTIZ at broward.org  Thu Jun 22 14:15:13 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Thu, 22 Jun 2006 14:15:13 -0400
Subject: [cisco-voip] OS 2000.4.2 sr7 upgrade
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC3529@MREXVS01.bc.broward.cty>

All of my servers are dual processor 7835's and 7845's.  I guess I will
be upgrading to 7a ASAP....

-----Original Message-----
From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Thursday, June 22, 2006 1:47 PM
To: Ortiz, Carlos; cisco-voip at puck.nether.net; Simon, Bill
Subject: Re: [cisco-voip] OS 2000.4.2 sr7 upgrade

The information in the bug id states that no problems have been
observed with uniprocessor servers.

So does this mean that even the QA lab doesn't get new hardware? :)

justin




On 6/22/06, Ryan Ratliff <rratliff at cisco.com> wrote:
> Lucky?  Check out the bug details to see the registry change that
> caused the problem.  You can then check your server.
>
> -Ryan
>
> On Jun 22, 2006, at 1:34 PM, Ortiz, Carlos wrote:
>
> That's weird.  I upgraded to this about 3 weeks ago and I have no
issue.
> Why would that be???
>
> -----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, June 22, 2006 1:04 PM
> To: Simon, Bill
> Cc: 'cisco-voip at puck.nether.net'
> Subject: Re: [cisco-voip] OS 2000.4.2 sr7 upgrade
>
> This is a known bug and SR7 should no longer be on CCO because of it.
>
> CSCse47327 is the bug.
>
> Please unicast me the SR number and I'll check it out.
>
> -Ryan
>
> On Jun 22, 2006, at 12:59 PM, Simon, Bill wrote:
>
> Yesterday I attempted an upgrade on our cluster from 2000.2.7sr4 to
> 2000.4.2
> then applied sr7.
>
> CallManager version is 4.1(2)sr1.  Cluster hardware all around is HP
> DL-380
> G3 (like MCS-7835 or 7845)
>
> The short story is that after the upgrade and a few hours of
frustrating
> poor or non-existent phone service, we bailed out to a set of mirrors
> taken
> the day before.  The main symptom of a sick CM cluster was that on the
> subscribers where phones registered, CPU loads upon boot went to 100%
> and
> stayed there.  CCM.EXE was the process using most of the CPU.  The
> high load
> caused failed registrations, incredibly slow response times on
> phones, VM
> and gateway ports registering and then going unregistered, and other
> weirdness.
>
> After an hour and a half talking to TAC on a P1 case we were unable
> to wait
> any longer and had to bail out.
>
> The case is still open and we're hoping to get to the bottom of it
> but I'm
> putting this out there in case any of you have seen something like
this,
> have feedback/thoughts, etc.
>
> ---
> 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
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From sean at northrock.bm  Thu Jun 22 14:45:58 2006
From: sean at northrock.bm (Sean Watkins)
Date: Thu, 22 Jun 2006 15:45:58 -0300
Subject: [cisco-voip] Strip name in FROM in invite?
Message-ID: <6B02498FEBBA0C489FA4CFDB2CAFB53D054E21@nrc-exch1.NORTHROCK.LOCAL>


Hi,

Is there any easy way to strip the name from a SIP invite leaving a Cisco
GW?

Ie the "unknown" junk below? 

Sean




Jun 22 15:16:39 ADT: Sent: 
INVITE sip:1990X at X:5060 SIP/2.0
Via: SIP/2.0/UDP  X:5060;x-route-tag="tgrp:PRIs"
From: "unknown" <sip:X>;tag=AA3CD44-E68  <----------------------------
To: <sip:1990X at X>
Date: Thu, 22 Jun 2006 18:16:39 GMT
Call-ID: 15374E06-15211DB-8043CCE5-2DF7505B at X
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 355866039-22155739-2151795941-771182683
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 1151000199
Contact: <sip:X:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 274
--
Sean Watkins
North Rock Communications
Phone: 441-540-4102
 

From sebastian.averbuj at ifxnw.com.ar  Thu Jun 22 15:32:53 2006
From: sebastian.averbuj at ifxnw.com.ar (Sebastian Averbuj)
Date: Thu, 22 Jun 2006 16:32:53 -0300
Subject: [cisco-voip] Program to convert some .wav file to g729
In-Reply-To: <48D167BB-C1B2-4404-BBA2-F4B4F2343B82@cisco.com>
Message-ID: <014801c69632$a87cf310$0401a8c0@sebas>

Dear List,

	Did u know some program to convert .wav file to .g729?

Tks a lot 

Regards.

--
Sebastian  Averbuj



From rratliff at cisco.com  Thu Jun 22 15:39:55 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 22 Jun 2006 15:39:55 -0400
Subject: [cisco-voip] Program to convert some .wav file to g729
In-Reply-To: <014801c69632$a87cf310$0401a8c0@sebas>
References: <014801c69632$a87cf310$0401a8c0@sebas>
Message-ID: <46280719-2E52-4833-9E18-1CADD152470A@cisco.com>

The CallManager MOH Audio Translator service will do it for you if  
you have g.729 MOH enabled.

Be aware that when it does the converting it spikes the cpu so don't  
do it on call processing servers during business hours.

-Ryan

On Jun 22, 2006, at 3:32 PM, Sebastian Averbuj wrote:

Dear List,

	Did u know some program to convert .wav file to .g729?

Tks a lot

Regards.

--
Sebastian  Averbuj


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


From lassekim at gmail.com  Thu Jun 22 15:52:29 2006
From: lassekim at gmail.com (Lasse Kim Christiansen)
Date: Thu, 22 Jun 2006 21:52:29 +0200
Subject: [cisco-voip] Temp Fail IP Phone Message
In-Reply-To: <f513ea484a8678c674b507432b3efbb7@cisco.com>
References: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAANu8Kq6czuUuapNUwKTVli+KAAAAQAAAAgfQWkl/o2kamcUlzD6fnbwEAAAAA@inxi.com>
	<FB78DCD6-22BA-4CAB-B183-627806B029C2@mac.com>
	<f513ea484a8678c674b507432b3efbb7@cisco.com>
Message-ID: <98DD19F3-9F31-4DC3-99F2-E0A9A0C8D26B@gmail.com>

Did you ever find the solution ?

Because i'm in exactly the same situation now and are in dire need  
for a "quick fix" :-)

Lasse

On 26/05/2006, at 14.55, Kevin Thorngren wrote:

> Temp fail can happen if the MGCP GW unregisters from CCM.  I suspect
> this might be the problem here.  For example, the GW unregisters, CCM
> signals temp fail to the phone but does not disconnect the call so the
> audio stays up.  But since the GW is unregistered and CCM relays DTMF
> this stops working although the audio path is still up.
>
> I would recommend gathering detailed CCM traces from the primary CCM
> for the MGCP GW, that corresponds to the temp fail messages, to see if
> the GW is unregistering at that time.  You can also take a quick look
> in the Application Event log for the unregister events.
>
> Kevin
> On May 26, 2006, at 1:09 AM, vemail at mac.com wrote:
>
>>
>>
>> The "Temp Fail" message is a failure of the audio path connection
>> usually due to DSP errors or codec issues.  There can be an issue
>> with the DSP of the IP Phone (if it's only happening to one phone) or
>> the DSP resources on a Voice Gateway.  If the error is occurring with
>> multiple phones, before disconnecting the troubled calls, identify
>> the gateway endpoint at the IP phone and check the log of the gateway
>> endpoint for any DSP errors (assuming debug logging is active).  You
>> can also try to query the DSP to identify which DSP is not  
>> responding.
>>
>> Also, the connection failure could be caused by a codec negotiation
>> issue such as (for instance) the endpoint does not have G.729
>> resources available however calls to the location whereupon the
>> gateway is located requires the use of the G.729 codec.
>>
>> The issue can be hardware or software related.
>>
>>
>> On May 25, 2006, at 11:21 PM, Michael L. Windon wrote:
>>
>>> All, any help you could you could provide would be greatly
>>> appreciated.
>>>
>>>
>>>
>>> Environment:
>>>
>>>
>>>
>>> CallManager 4.1.3 SR3a
>>>
>>> 3745 14.4.6.T2
>>>
>>> 8 T1 ? Pri MGCP Connected to a MLS-100 using Pri-NI2 ISDN type
>>>
>>>
>>>
>>> Situation:
>>>
>>>
>>>
>>> During a circuit switch bound call, the cisco IP phone which are
>>> 7911 and 7960 display a Temp Fail message on the bottom of the
>>> screen. Once this happens the call continues but the distant-end
>>> does not hear any buttons pushed. This is not good because users
>>> can?t enter in information into systems like voice mail or any
>>> other prompt and collect applications.
>>>
>>>
>>>
>>> What I have tried:
>>>
>>> Upgrading IOS 3745
>>>
>>> Using the latest device pack April 2006
>>>
>>> Verified routing between ip phone and call manager
>>>
>>> Search through detail traces
>>>
>>> Called tac and send traces, and isdn q931 debugs, mgcp packet  
>>> debugs.
>>>
>>>
>>>
>>> I have many people tell me that this message is caused when the IP
>>> phone stop receiving ack of its keep alive messages to or from the
>>> Call Manager, but I assure you the call manager is not down. I can
>>> ping and traceroute directly to the phone from the callmanager
>>> during a call. The phone is also registered with the call manager
>>> and is not dropping into SRST mode. I don?t know where to go from
>>> here. Does anyone know of another way to watch or detect this event
>>> on the call manager other then trace logs, as I have not found
>>> anything of interested in them or don?t know how to read them. Like
>>> I said before any help would be greatly appreciated.
>>>
>>>
>>>
>>> Mike
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip



From tednugent69 at yahoo.com  Thu Jun 22 17:15:47 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 22 Jun 2006 14:15:47 -0700 (PDT)
Subject: [cisco-voip] TOS marking of SCCP on CM 3.3 to CS3
In-Reply-To: <98DD19F3-9F31-4DC3-99F2-E0A9A0C8D26B@gmail.com>
Message-ID: <20060622211547.33719.qmail@web36112.mail.mud.yahoo.com>

This has been driving me nuts.
I'm running CM3.3(4)SR2 and I've changed the TOS Mask
(IP TOS BitMask for Comm. between CCM and Device*) to
CS3 (0x60). As the param implies it marks the traffic
correctly from CM to my remote site endpoint however
the traffic from the endpoint (IP Phone) towards CM is
still at AF31? Short of remarking the control traffic
at the switch or upgrading to 4.1 can anyone let me
know how this could be accomplished?
Thanks Ted


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

From jrybak at annese.com  Thu Jun 22 17:44:04 2006
From: jrybak at annese.com (jake rybak)
Date: Thu, 22 Jun 2006 17:44:04 -0400
Subject: [cisco-voip] TOS marking of SCCP on CM 3.3 to CS3
In-Reply-To: <20060622211547.33719.qmail@web36112.mail.mud.yahoo.com>
References: <20060622211547.33719.qmail@web36112.mail.mud.yahoo.com>
Message-ID: <449B0F24.1030001@annese.com>

I would guess it would be your mls qos map cos-dscp statement on the switch.

Ted Nugent wrote:
> This has been driving me nuts.
> I'm running CM3.3(4)SR2 and I've changed the TOS Mask
> (IP TOS BitMask for Comm. between CCM and Device*) to
> CS3 (0x60). As the param implies it marks the traffic
> correctly from CM to my remote site endpoint however
> the traffic from the endpoint (IP Phone) towards CM is
> still at AF31? Short of remarking the control traffic
> at the switch or upgrading to 4.1 can anyone let me
> know how this could be accomplished?
> Thanks Ted
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around 
> http://mail.yahoo.com 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>   


From tednugent69 at yahoo.com  Thu Jun 22 18:10:22 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 22 Jun 2006 15:10:22 -0700 (PDT)
Subject: [cisco-voip] TOS marking of SCCP on CM 3.3 to CS3
In-Reply-To: <449B0F24.1030001@annese.com>
Message-ID: <20060622221023.17791.qmail@web36108.mail.mud.yahoo.com>

Thats exactly what I thought too at first... so I set
the port to trust DSCP and just for grins I set the
COS-DSCP map back to 24.. same dealio..

also the trunk to the gateway is set to trust DSCP...



--- jake rybak <jrybak at annese.com> wrote:

> I would guess it would be your mls qos map cos-dscp
> statement on the switch.
> 
> Ted Nugent wrote:
> > This has been driving me nuts.
> > I'm running CM3.3(4)SR2 and I've changed the TOS
> Mask
> > (IP TOS BitMask for Comm. between CCM and Device*)
> to
> > CS3 (0x60). As the param implies it marks the
> traffic
> > correctly from CM to my remote site endpoint
> however
> > the traffic from the endpoint (IP Phone) towards
> CM is
> > still at AF31? Short of remarking the control
> traffic
> > at the switch or upgrading to 4.1 can anyone let
> me
> > know how this could be accomplished?
> > Thanks Ted
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> > http://mail.yahoo.com 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >   
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From dphillips at sapc.edu  Thu Jun 22 18:16:41 2006
From: dphillips at sapc.edu (Phillips, Danny)
Date: Thu, 22 Jun 2006 18:16:41 -0400
Subject: [cisco-voip] Background Logo
Message-ID: <18F0B5EFCCE1A449B07A3A9DF61DC07308ACE3@xchange.sapc.edu>

We want to add our college's logo to the 7912, 7940, and 7960 IP phones
running CM 4.1(3), could someone point me in the right direction.

 

 

Thanks 

Danny,

 

 

 

 

 

`````````````````````````````````````````````````

Danny Phillips

Network Administrator

St. Andrews Presbyterian College

910-277-5079

dphillips at sapc.edu

 

`````````````````````````````````````````````````

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From pdavidson at tdktca.com  Thu Jun 22 18:32:01 2006
From: pdavidson at tdktca.com (pdavidson at tdktca.com)
Date: Thu, 22 Jun 2006 17:32:01 -0500
Subject: [cisco-voip] Background Logo
In-Reply-To: <mailman.10297.1151014484.1478.cisco-voip@puck.nether.net>
Message-ID: <OF9971A336.487933A5-ON86257195.007AF095-86257195.007BC857@tca.us.intdk>

Danny-

There are only a few ways to do this- and unfortunately, under CCM4.X, I'm
not happy with any of them.

The by the book answer is to use the Idle Screen functionality.  You'll see
a setting for the URL to set for idle, as well as the idle timeout timer,
on the Device Settings page under CCM Administrator.  The URL must return a
valid CMXML object, containing the information you want to display.  This
functionality works- I run it regularly.

However- under CCM4, the definition of 'idle' is a little bit funky.  I've
had it kick in while I'm dialing before, replacing what I'm dialing and
stopping me from dialing more digits.  You also almost lose the onhook
dialahead functionality- you have to exit from the idle screen first, then
onhook dial- a retraining step for the users.

I have seen this done with custom authored phone loads, which involve BTXML
authoring- as a university, you might be able to get some help from Cisco
to do this.  I know that the 'managed' IP-PBX offering from Covad does this
on 7960's- I've seen them running- but that's using an MGCP load talking to
something other than Callmanager.

If you run the SIP image on the phone, it's MUCH simpler- just tell the
phone the name of a .bmp file on your TFTP server, and you're done- with no
loss of functionality.  But, CCM4 doesn't make SIP phone administration
simple- it will work, but it will be a very manual process for you.  CCM5
changes that, is my understanding- but the CCM5 SIP Phone load has other
oddities- your mileage may vary.

Hope this helps.

Paul Davidson, Manager, MIS
TDK Corporation of America
http://www.component.tdk.com



                                                                           
             cisco-voip-reques                                             
             t at puck.nether.net                                             
             Sent by:                                                   To 
             cisco-voip-bounce         cisco-voip at puck.nether.net          
             s at puck.nether.net                                          cc 
                                                                           
                                                                   Subject 
             06/22/2006 05:14          cisco-voip Digest, Vol 40, Issue    
             PM                        119                                 
                                                                           
                                                                           
             Please respond to                                             
             cisco-voip at puck.n                                             
                 ether.net                                                 
                                                                           
                                                                           





Date: Thu, 22 Jun 2006 18:16:41 -0400
From: "Phillips, Danny" <dphillips at sapc.edu>
Subject: [cisco-voip] Background Logo
To: <cisco-voip at puck.nether.net>
Message-ID: <18F0B5EFCCE1A449B07A3A9DF61DC07308ACE3 at xchange.sapc.edu>
Content-Type: text/plain; charset="us-ascii"

We want to add our college's logo to the 7912, 7940, and 7960 IP phones
running CM 4.1(3), could someone point me in the right direction.





Thanks

Danny,











`````````````````````````````````````````````````

Danny Phillips

Network Administrator

St. Andrews Presbyterian College

910-277-5079

dphillips at sapc.edu



`````````````````````````````````````````````````




From Scott.Voll at wesd.org  Thu Jun 22 19:02:29 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu, 22 Jun 2006 16:02:29 -0700
Subject: [cisco-voip] OS 2000.4.2 sr7 upgrade
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED509D@Apollo.wesd.org>

Go to sr8.  has the VNC update also.

scott

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: Thursday, June 22, 2006 11:15 AM
To: Justin Steinberg; cisco-voip at puck.nether.net; Simon, Bill
Subject: Re: [cisco-voip] OS 2000.4.2 sr7 upgrade

All of my servers are dual processor 7835's and 7845's.  I guess I will
be upgrading to 7a ASAP....

-----Original Message-----
From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Thursday, June 22, 2006 1:47 PM
To: Ortiz, Carlos; cisco-voip at puck.nether.net; Simon, Bill
Subject: Re: [cisco-voip] OS 2000.4.2 sr7 upgrade

The information in the bug id states that no problems have been
observed with uniprocessor servers.

So does this mean that even the QA lab doesn't get new hardware? :)

justin




On 6/22/06, Ryan Ratliff <rratliff at cisco.com> wrote:
> Lucky?  Check out the bug details to see the registry change that
> caused the problem.  You can then check your server.
>
> -Ryan
>
> On Jun 22, 2006, at 1:34 PM, Ortiz, Carlos wrote:
>
> That's weird.  I upgraded to this about 3 weeks ago and I have no
issue.
> Why would that be???
>
> -----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, June 22, 2006 1:04 PM
> To: Simon, Bill
> Cc: 'cisco-voip at puck.nether.net'
> Subject: Re: [cisco-voip] OS 2000.4.2 sr7 upgrade
>
> This is a known bug and SR7 should no longer be on CCO because of it.
>
> CSCse47327 is the bug.
>
> Please unicast me the SR number and I'll check it out.
>
> -Ryan
>
> On Jun 22, 2006, at 12:59 PM, Simon, Bill wrote:
>
> Yesterday I attempted an upgrade on our cluster from 2000.2.7sr4 to
> 2000.4.2
> then applied sr7.
>
> CallManager version is 4.1(2)sr1.  Cluster hardware all around is HP
> DL-380
> G3 (like MCS-7835 or 7845)
>
> The short story is that after the upgrade and a few hours of
frustrating
> poor or non-existent phone service, we bailed out to a set of mirrors
> taken
> the day before.  The main symptom of a sick CM cluster was that on the
> subscribers where phones registered, CPU loads upon boot went to 100%
> and
> stayed there.  CCM.EXE was the process using most of the CPU.  The
> high load
> caused failed registrations, incredibly slow response times on
> phones, VM
> and gateway ports registering and then going unregistered, and other
> weirdness.
>
> After an hour and a half talking to TAC on a P1 case we were unable
> to wait
> any longer and had to bail out.
>
> The case is still open and we're hoping to get to the bottom of it
> but I'm
> putting this out there in case any of you have seen something like
this,
> have feedback/thoughts, etc.
>
> ---
> 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
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

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


From jonvoip at gmail.com  Thu Jun 22 19:36:06 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 22 Jun 2006 18:36:06 -0500
Subject: [cisco-voip] IPCC Express 3.5(3) install error messgae connecting
	to LDAP
Message-ID: <5d093f9a0606221636v60c10aaftfe11a787e7619094@mail.gmail.com>

`Installing IPCC 3.5(3) on a fresh box.

I am connected thru AD on CCM.

I run setup, enter the info, create a profile, use the default
repository profile, and when I click next, I get:

LDAP Configuration - Error

There has been some failure while updating the LDAP. Please contact
Administrator.


I have verified the information is correct, I have connectivity to the
LDAP server by host name. What is wrong?




Jonathan

From tednugent69 at yahoo.com  Thu Jun 22 19:44:36 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 22 Jun 2006 16:44:36 -0700 (PDT)
Subject: [cisco-voip] TOS marking of SCCP on CM 3.3 to CS3
In-Reply-To: <449B1C84.2020301@annese.com>
Message-ID: <20060622234436.12625.qmail@web36106.mail.mud.yahoo.com>

Well at least I'm not smoking crack! Any other ideas?


Internet Protocol, Src: 192.168.8.50 (192.168.8.50),
Dst: 10.83.136.111 (10.83.136.111)
    Version: 4
    Header length: 20 bytes
    Differentiated Services Field: 0x68 (DSCP 0x1a:
Assured Forwarding 31; ECN: 0x00)
        0110 10.. = Differentiated Services Codepoint:
Assured Forwarding 31 (0x1a)
        .... ..0. = ECN-Capable Transport (ECT): 0
        .... ...0 = ECN-CE: 0
    Total Length: 40
    Identification: 0x7e35 (32309)
    Flags: 0x00
        0... = Reserved bit: Not set
        .0.. = Don't fragment: Not set
        ..0. = More fragments: Not set
    Fragment offset: 0
    Time to live: 64
    Protocol: TCP (0x06)
    Header checksum: 0xa096 [correct]
        Good: True
        Bad : False
    Source: 192.168.8.50 (192.168.8.50)
    Destination: 10.83.136.111 (10.83.136.111)



--- jake rybak <jrybak at annese.com> wrote:

> I also been caught not changing mgcp signaling and
> seeing the wrong 
> traffic hit a policy.map at af31 thinking it was
> sccp.
> 
> Ted Nugent wrote:
> > 3.3(4) doesn't list the DSCP on the phone like
> 3.3(5)+
> > but I'm definitely seeing the class-map increment
> AF31
> > when I hit any keys while a call is active... 
> > I guess I'll need to pull some sniffer traces to
> > confirm this since all config appears to be
> correct.
> > Probably need to reload the switch just confirm
> it's
> > reading QOS correctly.
> > I was thinking it might be a bug and I guess a
> sniffer
> > will confirm it. I welcome any other input and
> I'll
> > post back what the capture shows me.
> >
> >
> > --- jake rybak <jrybak at annese.com> wrote:
> >
> >   
> >> I had the same problem or at least I thought. The
> >> phone would say 
> >> AF31(bug) but the packet captures said cs3. I
> would
> >> see them match a 
> >> class-map at af31 on my wan router until I
> changed
> >> the cos-dscp mapping.
> >>
> >> It was a while ago so I do not recall the exact
> >> scenario.
> >>
> >> Ted Nugent wrote:
> >>     
> >>> Thats exactly what I thought too at first... so
> I
> >>>       
> >> set
> >>     
> >>> the port to trust DSCP and just for grins I set
> >>>       
> >> the
> >>     
> >>> COS-DSCP map back to 24.. same dealio..
> >>>
> >>> also the trunk to the gateway is set to trust
> >>>       
> >> DSCP...
> >>     
> >>>
> >>> --- jake rybak <jrybak at annese.com> wrote:
> >>>
> >>>   
> >>>       
> >>>> I would guess it would be your mls qos map
> >>>>         
> >> cos-dscp
> >>     
> >>>> statement on the switch.
> >>>>
> >>>> Ted Nugent wrote:
> >>>>     
> >>>>         
> >>>>> This has been driving me nuts.
> >>>>> I'm running CM3.3(4)SR2 and I've changed the
> TOS
> >>>>>       
> >>>>>           
> >>>> Mask
> >>>>     
> >>>>         
> >>>>> (IP TOS BitMask for Comm. between CCM and
> >>>>>           
> >> Device*)
> >>     
> >>>>>       
> >>>>>           
> >>>> to
> >>>>     
> >>>>         
> >>>>> CS3 (0x60). As the param implies it marks the
> >>>>>       
> >>>>>           
> >>>> traffic
> >>>>     
> >>>>         
> >>>>> correctly from CM to my remote site endpoint
> >>>>>       
> >>>>>           
> >>>> however
> >>>>     
> >>>>         
> >>>>> the traffic from the endpoint (IP Phone)
> towards
> >>>>>       
> >>>>>           
> >>>> CM is
> >>>>     
> >>>>         
> >>>>> still at AF31? Short of remarking the control
> >>>>>       
> >>>>>           
> >>>> traffic
> >>>>     
> >>>>         
> >>>>> at the switch or upgrading to 4.1 can anyone
> let
> >>>>>       
> >>>>>           
> >>>> me
> >>>>     
> >>>>         
> >>>>> know how this could be accomplished?
> >>>>> Thanks Ted
> >>>>>
> >>>>>
> >>>>>
> >>>>>           
> >>
> __________________________________________________
> >>     
> >>>>> Do You Yahoo!?
> >>>>> Tired of spam?  Yahoo! Mail has the best spam
> >>>>>       
> >>>>>           
> >>>> protection around 
> >>>>     
> >>>>         
> >>>>> http://mail.yahoo.com 
> >>>>>
> _______________________________________________
> >>>>> cisco-voip mailing list
> >>>>> cisco-voip at puck.nether.net
> >>>>>
> >>>>>       
> >>>>>           
> >>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>     
> >>>>     
> >>>>         
> >>>>>   
> >>>>>       
> >>>>>           
> >>>> _______________________________________________
> >>>> cisco-voip mailing list
> >>>> cisco-voip at puck.nether.net
> >>>>
> >>>>         
> >>
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>     
> >>>>     
> >>>>         
> >>>
> __________________________________________________
> >>> Do You Yahoo!?
> >>> Tired of spam?  Yahoo! Mail has the best spam
> >>>       
> >> protection around 
> >>     
> >>> http://mail.yahoo.com 
> >>>
> >>>   
> >>>       
> >>     
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> > http://mail.yahoo.com 
> >
> >   
> 
> 


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

From tednugent69 at yahoo.com  Thu Jun 22 19:47:43 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 22 Jun 2006 16:47:43 -0700 (PDT)
Subject: [cisco-voip] IPCC Express 3.5(3) install error messgae
	connecting to LDAP
In-Reply-To: <5d093f9a0606221636v60c10aaftfe11a787e7619094@mail.gmail.com>
Message-ID: <20060622234743.80214.qmail@web36110.mail.mud.yahoo.com>

What do you mean "I am connected thru AD on CCM"?? Are
you using the CM AD Plugin?


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

> `Installing IPCC 3.5(3) on a fresh box.
> 
> I am connected thru AD on CCM.
> 
> I run setup, enter the info, create a profile, use
> the default
> repository profile, and when I click next, I get:
> 
> LDAP Configuration - Error
> 
> There has been some failure while updating the LDAP.
> Please contact
> Administrator.
> 
> 
> I have verified the information is correct, I have
> connectivity to the
> LDAP server by host name. What is wrong?
> 
> 
> 
> 
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jonvoip at gmail.com  Thu Jun 22 19:49:29 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Thu, 22 Jun 2006 18:49:29 -0500
Subject: [cisco-voip] IPCC Express 3.5(3) install error messgae
	connecting to LDAP
In-Reply-To: <20060622234743.80214.qmail@web36110.mail.mud.yahoo.com>
References: <5d093f9a0606221636v60c10aaftfe11a787e7619094@mail.gmail.com>
	<20060622234743.80214.qmail@web36110.mail.mud.yahoo.com>
Message-ID: <5d093f9a0606221649t6aad0d78od6ded62f572398bb@mail.gmail.com>

Yes, I am using the AD plugin on CM4.1(3)

On 6/22/06, Ted Nugent <tednugent69 at yahoo.com> wrote:
> What do you mean "I am connected thru AD on CCM"?? Are
> you using the CM AD Plugin?
>
>
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
>
> > `Installing IPCC 3.5(3) on a fresh box.
> >
> > I am connected thru AD on CCM.
> >
> > I run setup, enter the info, create a profile, use
> > the default
> > repository profile, and when I click next, I get:
> >
> > LDAP Configuration - Error
> >
> > There has been some failure while updating the LDAP.
> > Please contact
> > Administrator.
> >
> >
> > I have verified the information is correct, I have
> > connectivity to the
> > LDAP server by host name. What is wrong?
> >
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>

From tednugent69 at yahoo.com  Thu Jun 22 19:53:13 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Thu, 22 Jun 2006 16:53:13 -0700 (PDT)
Subject: [cisco-voip] IPCC Express 3.5(3) install error messgae
	connecting to LDAP
In-Reply-To: <5d093f9a0606221649t6aad0d78od6ded62f572398bb@mail.gmail.com>
Message-ID: <20060622235313.9681.qmail@web36102.mail.mud.yahoo.com>

Can you provide your LDAP info... obviously change it
to protect the inocent...



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

> Yes, I am using the AD plugin on CM4.1(3)
> 
> On 6/22/06, Ted Nugent <tednugent69 at yahoo.com>
> wrote:
> > What do you mean "I am connected thru AD on CCM"??
> Are
> > you using the CM AD Plugin?
> >
> >
> > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > > `Installing IPCC 3.5(3) on a fresh box.
> > >
> > > I am connected thru AD on CCM.
> > >
> > > I run setup, enter the info, create a profile,
> use
> > > the default
> > > repository profile, and when I click next, I
> get:
> > >
> > > LDAP Configuration - Error
> > >
> > > There has been some failure while updating the
> LDAP.
> > > Please contact
> > > Administrator.
> > >
> > >
> > > I have verified the information is correct, I
> have
> > > connectivity to the
> > > LDAP server by host name. What is wrong?
> > >
> > >
> > >
> > >
> > > Jonathan
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around
> > http://mail.yahoo.com
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jason.aarons at us.didata.com  Thu Jun 22 21:16:32 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Thu, 22 Jun 2006 21:16:32 -0400
Subject: [cisco-voip] VG224 / SCCP / Modem
References: <20060620115006658.00000003156@janderson-3948><C1FE15183DA37645BC0633BC604E44F013E76A@USNAEXCH.na.didata.local>
	<5973.24.234.204.56.1150855270.squirrel@www.dignans.com>
Message-ID: <C1FE15183DA37645BC0633BC604E44F013E787@USNAEXCH.na.didata.local>

This is inaccurate.  
 
Using 4.1(3) with VG224 under 12.4T you can use modem/faxes with SCCP.
 
I've had a VG224 with stcapp (SCCP) running for 2 months with no issues.  My VG24 is co-located on a lan with the T1 CAS gateway.  
 
Clock slips/errorrs via show controller t1 will kill modems/faxes.  Check your clocking, etc.

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Andrew Dignan
Sent: Tue 6/20/2006 10:01 PM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VG224 / SCCP / Modem



The VG224 using the SCCP image does not support fax/modems:

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/prod_release_note09186a008041df56.html

------------------------------------------------
*Fax and modems are not supported on FXS ports with SCCP enabled. FXS
ports with either fax or modems attached should be configured in H.323 or
SIP mode.
-------------------------------------------------

andy dignan

> You can plug a cheap analog phone into that VG224 and after the same x
> minutes get disconnected? Verify you can reproduce the problem.
>
> Open a TAC case, sounds like you need some debugs, SDL traces, takes
> -jason
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
> Sent: Tue 6/20/2006 12:50 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] VG224 / SCCP / Modem
>
>
> Question:
>
> I have a client who is using a VG224 that is configured to use SCCP on CCM
> 4.1. The majority of the ports are used for modems but there are devices
> that need skinny features as well. The problem is the modem calls fail
> after a random amount of time. It could be anywhere from 5 - 30 minutes.
> That same modem plugged directly into a 1FB will have no reliability
> issues.
>
> I have searched Cisco's website and their forums looking for some
> configuration guidelines for modem usage and found nothing. Does anyone
> have any advice to offer?
>
> Thanks,
> Jeff Anderson
>
>
>
>
>
>
> -----------------------------------------
> 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 cj.alan at gmail.com  Thu Jun 22 21:23:20 2006
From: cj.alan at gmail.com (Alan Su)
Date: Fri, 23 Jun 2006 09:23:20 +0800
Subject: [cisco-voip] CallManager Handle Call Routing with PBX
	interoperability
In-Reply-To: <48D167BB-C1B2-4404-BBA2-F4B4F2343B82@cisco.com>
References: <8f9f19f00606221024r634f68c4o45d37e8d14de3133@mail.gmail.com>
	<DF84CA1D-C978-4B9D-9043-210F65D4DEC2@cisco.com>
	<8f9f19f00606221048o68c769fev564fbb242f141c22@mail.gmail.com>
	<48D167BB-C1B2-4404-BBA2-F4B4F2343B82@cisco.com>
Message-ID: <8f9f19f00606221823o57de7c13o927fb45fe55ac2a0@mail.gmail.com>

Hi Ryan,

Many thanks for your sharing, I'll check if the PBX can act as H.323 GW or
not. ^^


Regards/Alan


2006/6/23, Ryan Ratliff <rratliff at cisco.com>:
>
> I've no experience with a sip trunk to a PBX I'm afraid.
>
> I have seen some PBXs that could act as an H.323 gateway to CM.  IIRC
> the only trick was to make sure "MTP Required" was set due to H.323
> version issues.
>
> -Ryan
>
> On Jun 22, 2006, at 1:48 PM, Alan Su wrote:
>
> Hi Ryan,
>
> Yes, I've tried to use IOS MGCP gateway and it worked. But it will
> occupy the channel between GW-PBX. For example, extension 1315 & 1316
> are PBX analog phone, if I transfer all calls from PBX to GW and then
> to CCM for call routing, even those 2 extensions will occupy the
> channel for in/out calls. Unless I configure all the analog
> extensions in PBX and forward others to GW. However, it'll cause
> other issues such as when I dialing 1315xxxx for off-net call, the
> PBX will treat it as 1315 extension. Because the PBX can be upgraded
> to IP-based, so I was wondering is there any solution like SIP-trunk
> can solve the issue. Thanks a lot ~
>
>
> Regards,
>
> Alan
>
>
> 2006/6/23, Ryan Ratliff <rratliff at cisco.com>: Basically you'll need
> some type of voice gateway to receive calls
> from the PBX (via PRI or other physical link) and send them to the
> CM.     Once the call hits the CM it gets routed normally to a phone,
> route pattern, etc.
>
> -Ryan
>
> On Jun 22, 2006, at 1:24 PM, Alan Su wrote:
>
> Dear All,
>
> I have a question about the PBX interoperability with CCM. What can I
> do to make CCM to handle all the call routing even for the PBX end-
> phone?
> One of the solution I heard about is to use SIP trunk connecting with
> CCM & PBX, but I'm not quite sure how to do that. Any suggestion or
> comment would be very grateful. Thanks.
>
>
> Regards,
>
> Alan
> _______________________________________________
> 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  Fri Jun 23 00:33:13 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 23 Jun 2006 00:33:13 -0400
Subject: [cisco-voip] Uninstalling AD connector and going back to DC
	Directory...
Message-ID: <5d093f9a0606222133k729ec700n14cc2022130cb69@mail.gmail.com>

We accidentally integrated our CallManager 4.1(3)sr3b publisher with
Active Directory.

Since then, nothing but problems and we want to back out.

We followed the instructions here:

http://www.cisco.com/warp/public/788/AVVID/change2_dcdirectory.html

But they don't work.

The document includes this line:

"The proper way to back out of this configuration and revert to using
the embedded DC Directory is to re-run the plugin installation and
choose the option to uninstall the plugin."

However, there is no such option on the installer and it is not listed
in the add/remove programs on the PUB.

We need this reverted, it is preventing us from installing IPCC
Express... I am trying to avoid opening a TAC case.



Jonathan

From cisco at dignans.com  Fri Jun 23 02:21:01 2006
From: cisco at dignans.com (Andrew Dignan)
Date: Fri, 23 Jun 2006 01:21:01 -0500 (CDT)
Subject: [cisco-voip] VG224 / SCCP / Modem
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F013E787@USNAEXCH.na.didata.local>
References: <20060620115006658.00000003156@janderson-3948><C1FE15183DA37645BC0633BC604E44F013E76A@USNAEXCH.na.didata.local>
	<5973.24.234.204.56.1150855270.squirrel@www.dignans.com>
	<C1FE15183DA37645BC0633BC604E44F013E787@USNAEXCH.na.didata.local>
Message-ID: <2104.24.234.204.227.1151043661.squirrel@www.dignans.com>

I am having trouble finding this in the 12.4 release notes.  Do you have a
link that states this?


> This is inaccurate.
>
> Using 4.1(3) with VG224 under 12.4T you can use modem/faxes with SCCP.
>
> I've had a VG224 with stcapp (SCCP) running for 2 months with no issues.
> My VG24 is co-located on a lan with the T1 CAS gateway.
>
> Clock slips/errorrs via show controller t1 will kill modems/faxes.  Check
> your clocking, etc.
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net on behalf of Andrew Dignan
> Sent: Tue 6/20/2006 10:01 PM
> To: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] VG224 / SCCP / Modem
>
>
>
> The VG224 using the SCCP image does not support fax/modems:
>
> http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/prod_release_note09186a008041df56.html
>
> ------------------------------------------------
> *Fax and modems are not supported on FXS ports with SCCP enabled. FXS
> ports with either fax or modems attached should be configured in H.323 or
> SIP mode.
> -------------------------------------------------
>
> andy dignan
>
>> You can plug a cheap analog phone into that VG224 and after the same x
>> minutes get disconnected? Verify you can reproduce the problem.
>>
>> Open a TAC case, sounds like you need some debugs, SDL traces, takes
>> -jason
>>
>> ________________________________
>>
>> From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
>> Sent: Tue 6/20/2006 12:50 PM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] VG224 / SCCP / Modem
>>
>>
>> Question:
>>
>> I have a client who is using a VG224 that is configured to use SCCP on
>> CCM
>> 4.1. The majority of the ports are used for modems but there are
>> devices
>> that need skinny features as well. The problem is the modem calls fail
>> after a random amount of time. It could be anywhere from 5 - 30
>> minutes.
>> That same modem plugged directly into a 1FB will have no reliability
>> issues.
>>
>> I have searched Cisco's website and their forums looking for some
>> configuration guidelines for modem usage and found nothing. Does anyone
>> have any advice to offer?
>>
>> Thanks,
>> Jeff Anderson
>>
>>
>>
>>
>>
>>
>> -----------------------------------------
>> Disclaimer:
>>
>> This e-mail communication and any attachments may contain
>> confidential and privileged information and is for use by the
>> designated addressee(s) named above only.  If you are not the
>> intended addressee, you are hereby notified that you have received
>> this communication in error and that any use or reproduction of
>> this email or its contents is strictly prohibited and may be
>> unlawful.  If you have received this communication in error, please
>> notify us immediately by replying to this message and deleting it
>> from your computer. Thank you.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>


From aman.chugh at gmail.com  Fri Jun 23 02:27:15 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Fri, 23 Jun 2006 01:27:15 -0500
Subject: [cisco-voip] Uninstalling AD connector and going back to DC
	Directory...
In-Reply-To: <5d093f9a0606222133k729ec700n14cc2022130cb69@mail.gmail.com>
References: <5d093f9a0606222133k729ec700n14cc2022130cb69@mail.gmail.com>
Message-ID: <d4c697ad0606222327n1ac6b2b7m4d0750e89767c56@mail.gmail.com>

if the plugin was installed then it should be listed in ADD/REMOVE
programmes.You can uninstall it by running it again,Also if you need to
verfiy your directory configuration check the following registery key

HKEY_LOCAL_MACHINE\SOFTWARE\Cisco Systems, Inc.\Directory Configuration

- Aman

On 6/22/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> We accidentally integrated our CallManager 4.1(3)sr3b publisher with
> Active Directory.
>
> Since then, nothing but problems and we want to back out.
>
> We followed the instructions here:
>
> http://www.cisco.com/warp/public/788/AVVID/change2_dcdirectory.html
>
> But they don't work.
>
> The document includes this line:
>
> "The proper way to back out of this configuration and revert to using
> the embedded DC Directory is to re-run the plugin installation and
> choose the option to uninstall the plugin."
>
> However, there is no such option on the installer and it is not listed
> in the add/remove programs on the PUB.
>
> We need this reverted, it is preventing us from installing IPCC
> Express... I am trying to avoid opening a TAC case.
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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From ARUNAS at lt.ibm.com  Fri Jun 23 02:46:53 2006
From: ARUNAS at lt.ibm.com (Arunas Sutkevicius)
Date: Fri, 23 Jun 2006 09:46:53 +0300
Subject: [cisco-voip] Uninstalling AD connector and going back to
	DC	Directory...
In-Reply-To: <5d093f9a0606222133k729ec700n14cc2022130cb69@mail.gmail.com>
Message-ID: <OF2D6D2485.74DD141B-ONC2257196.00254136-C2257196.0025C6EB@lt.ibm.com>

Probably you are hitting the CSCsb20774 bug. So solution of this bug 
should help solve installation problems, without reverting to DC 
Directory.

Regards,
Arunas




"Jonathan Charles" <jonvoip at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net
2006.06.23 07:33

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

Subject
[cisco-voip] Uninstalling AD connector and going back to DC Directory...






We accidentally integrated our CallManager 4.1(3)sr3b publisher with
Active Directory.

Since then, nothing but problems and we want to back out.

We followed the instructions here:

http://www.cisco.com/warp/public/788/AVVID/change2_dcdirectory.html

But they don't work.

The document includes this line:

"The proper way to back out of this configuration and revert to using
the embedded DC Directory is to re-run the plugin installation and
choose the option to uninstall the plugin."

However, there is no such option on the installer and it is not listed
in the add/remove programs on the PUB.

We need this reverted, it is preventing us from installing IPCC
Express... I am trying to avoid opening a TAC case.



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

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From b.turnbow at twt.it  Fri Jun 23 03:25:48 2006
From: b.turnbow at twt.it (Brian Turnbow)
Date: Fri, 23 Jun 2006 09:25:48 +0200
Subject: [cisco-voip] Strip name in FROM in invite?
In-Reply-To: <6B02498FEBBA0C489FA4CFDB2CAFB53D054E21@nrc-exch1.NORTHROCK.LOCAL>
Message-ID: <52964DBFB59FA34F98E96921BE21C90601465A5E@twt-exch.milano.twt>

You cannot strip the from field but you can change it.
Take a look at 
http://www.cisco.com/en/US/products/ps6441/prod_troubleshooting_guide09186a00801d3d3a.html#wp1385050

Regards
Brian 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sean Watkins
Sent: gioved? 22 giugno 2006 20.46
To: 'ciscovoip'
Subject: [cisco-voip] Strip name in FROM in invite?


Hi,

Is there any easy way to strip the name from a SIP invite leaving a Cisco
GW?

Ie the "unknown" junk below? 

Sean




Jun 22 15:16:39 ADT: Sent: 
INVITE sip:1990X at X:5060 SIP/2.0
Via: SIP/2.0/UDP  X:5060;x-route-tag="tgrp:PRIs"
From: "unknown" <sip:X>;tag=AA3CD44-E68  <----------------------------
To: <sip:1990X at X>
Date: Thu, 22 Jun 2006 18:16:39 GMT
Call-ID: 15374E06-15211DB-8043CCE5-2DF7505B at X
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 355866039-22155739-2151795941-771182683
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 1151000199
Contact: <sip:X:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 274
--
Sean Watkins
North Rock Communications
Phone: 441-540-4102
 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From CORTIZ at broward.org  Fri Jun 23 08:28:46 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 23 Jun 2006 08:28:46 -0400
Subject: [cisco-voip] Stable Call Mgr Version
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC352B@MREXVS01.bc.broward.cty>

Can anyone suggest a stable CM version and IPCC Express that they have
successfully running?

 

I am currently running 4.02a(sr2b) with IPCC 3.5.2 and Unity 4.05.

 

I would like to move to Call Mgr version 4.1.3 or 4.2 (if recommended).
I also realize that IPCC Express will need to be upgraded as part of
this.

 

Thanks,

 

Carlos

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From tednugent69 at yahoo.com  Fri Jun 23 10:39:45 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 23 Jun 2006 07:39:45 -0700 (PDT)
Subject: [cisco-voip] Stable Call Mgr Version
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC352B@MREXVS01.bc.broward.cty>
Message-ID: <20060623143945.50074.qmail@web36109.mail.mud.yahoo.com>

Well you're running on probably the most stable
version of CM that 4.x has to offer... Other then that
I would probably run with CM4.1(3)SR2... I've seen
some issues with SR3 and thier still working them
out... sr2b just came out the other day. I've only
deployed Unity 4.1 a few times and each went very
smoothly with very few followups.. (but 4.0(5) wasn't
a bad release either... unless there are features
you're looking for I would stick with 4.0(5))

I personally would not touch CM/Unity 4.2 yet...
they're both fairly new... I would probably give them
a couple month. 

IPCC 3.5.3 SR3 seemed to be pretty stable as well...
You could not pay me enough to support IPCCX 4.x
yet... they are getting closer with newer releases but
right now.. no way.. no how! 
You can also reference "golden bridge" at the link
below.

http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/ipcmtrix.htm


--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> Can anyone suggest a stable CM version and IPCC
> Express that they have
> successfully running?
> 
>  
> 
> I am currently running 4.02a(sr2b) with IPCC 3.5.2
> and Unity 4.05.
> 
>  
> 
> I would like to move to Call Mgr version 4.1.3 or
> 4.2 (if recommended).
> I also realize that IPCC Express will need to be
> upgraded as part of
> this.
> 
>  
> 
> Thanks,
> 
>  
> 
> Carlos
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From tednugent69 at yahoo.com  Fri Jun 23 10:54:42 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 23 Jun 2006 07:54:42 -0700 (PDT)
Subject: [cisco-voip] TOS marking of SCCP on CM 3.3 to CS3
In-Reply-To: <001201c696b6$783c7480$68b55d80$@com>
Message-ID: <20060623145442.5649.qmail@web36114.mail.mud.yahoo.com>

Thanks and Yes you're right.. that does the trick.
However.. Thats cheating! :-P 
The phone is still marking AF31 and we're just
remapping it on the switch. Sniffer trace don't lie.
:-/

Internet Protocol, Src: 192.168.8.50 (192.168.8.50),
Dst: 10.83.136.111 (10.83.136.111)
    Version: 4
    Header length: 20 bytes
    Differentiated Services Field: 0x68 (DSCP 0x1a:
Assured Forwarding 31; ECN: 0x00)
    Total Length: 40
    Identification: 0xbcb5 (48309)
    Flags: 0x00
    Fragment offset: 0
    Time to live: 64
    Protocol: TCP (0x06)
    Header checksum: 0x6216 [correct]
    Source: 192.168.8.50 (192.168.8.50)
    Destination: 10.83.136.111 (10.83.136.111)

--- Jake Rybak <jrybak at annese.com> wrote:

> Do not trust DSCP. Trust COS and it will work. Make
> sure the mls qos map
> cos-dscp 0 8 16 24 34 46 48 56
> 
> 
> -----Original Message-----
> From: Ted Nugent [mailto:tednugent69 at yahoo.com] 
> Sent: Thursday, June 22, 2006 19:45
> To: jake rybak
> Cc: ciscovoip Voip
> Subject: Re: [cisco-voip] TOS marking of SCCP on CM
> 3.3 to CS3
> 
> Well at least I'm not smoking crack! Any other
> ideas?
> 
> 
> Internet Protocol, Src: 192.168.8.50 (192.168.8.50),
> Dst: 10.83.136.111 (10.83.136.111)
>     Version: 4
>     Header length: 20 bytes
>     Differentiated Services Field: 0x68 (DSCP 0x1a:
> Assured Forwarding 31; ECN: 0x00)
>         0110 10.. = Differentiated Services
> Codepoint:
> Assured Forwarding 31 (0x1a)
>         .... ..0. = ECN-Capable Transport (ECT): 0
>         .... ...0 = ECN-CE: 0
>     Total Length: 40
>     Identification: 0x7e35 (32309)
>     Flags: 0x00
>         0... = Reserved bit: Not set
>         .0.. = Don't fragment: Not set
>         ..0. = More fragments: Not set
>     Fragment offset: 0
>     Time to live: 64
>     Protocol: TCP (0x06)
>     Header checksum: 0xa096 [correct]
>         Good: True
>         Bad : False
>     Source: 192.168.8.50 (192.168.8.50)
>     Destination: 10.83.136.111 (10.83.136.111)
> 
> 
> 
> --- jake rybak <jrybak at annese.com> wrote:
> 
> > I also been caught not changing mgcp signaling and
> seeing the wrong 
> > traffic hit a policy.map at af31 thinking it was
> sccp.
> > 
> > Ted Nugent wrote:
> > > 3.3(4) doesn't list the DSCP on the phone like
> > 3.3(5)+
> > > but I'm definitely seeing the class-map
> increment
> > AF31
> > > when I hit any keys while a call is active... 
> > > I guess I'll need to pull some sniffer traces to
> confirm this since 
> > > all config appears to be
> > correct.
> > > Probably need to reload the switch just confirm
> > it's
> > > reading QOS correctly.
> > > I was thinking it might be a bug and I guess a
> > sniffer
> > > will confirm it. I welcome any other input and
> > I'll
> > > post back what the capture shows me.
> > >
> > >
> > > --- jake rybak <jrybak at annese.com> wrote:
> > >
> > >   
> > >> I had the same problem or at least I thought.
> The phone would say
> > >> AF31(bug) but the packet captures said cs3. I
> > would
> > >> see them match a
> > >> class-map at af31 on my wan router until I
> > changed
> > >> the cos-dscp mapping.
> > >>
> > >> It was a while ago so I do not recall the exact
> scenario.
> > >>
> > >> Ted Nugent wrote:
> > >>     
> > >>> Thats exactly what I thought too at first...
> so
> > I
> > >>>       
> > >> set
> > >>     
> > >>> the port to trust DSCP and just for grins I
> set
> > >>>       
> > >> the
> > >>     
> > >>> COS-DSCP map back to 24.. same dealio..
> > >>>
> > >>> also the trunk to the gateway is set to trust
> > >>>       
> > >> DSCP...
> > >>     
> > >>>
> > >>> --- jake rybak <jrybak at annese.com> wrote:
> > >>>
> > >>>   
> > >>>       
> > >>>> I would guess it would be your mls qos map
> > >>>>         
> > >> cos-dscp
> > >>     
> > >>>> statement on the switch.
> > >>>>
> > >>>> Ted Nugent wrote:
> > >>>>     
> > >>>>         
> > >>>>> This has been driving me nuts.
> > >>>>> I'm running CM3.3(4)SR2 and I've changed the
> > TOS
> > >>>>>       
> > >>>>>           
> > >>>> Mask
> > >>>>     
> > >>>>         
> > >>>>> (IP TOS BitMask for Comm. between CCM and
> > >>>>>           
> > >> Device*)
> > >>     
> > >>>>>       
> > >>>>>           
> > >>>> to
> > >>>>     
> > >>>>         
> > >>>>> CS3 (0x60). As the param implies it marks
> the
> > >>>>>       
> > >>>>>           
> > >>>> traffic
> > >>>>     
> > >>>>         
> > >>>>> correctly from CM to my remote site endpoint
> > >>>>>       
> > >>>>>           
> > >>>> however
> > >>>>     
> > >>>>         
> > >>>>> the traffic from the endpoint (IP Phone)
> > towards
> > >>>>>       
> > >>>>>           
> > >>>> CM is
> > >>>>     
> > >>>>         
> > >>>>> still at AF31? Short of remarking the
> control
> > >>>>>       
> > >>>>>           
> > >>>> traffic
> > >>>>     
> > >>>>         
> > >>>>> at the switch or upgrading to 4.1 can anyone
> > let
> > >>>>>       
> > >>>>>           
> > >>>> me
> > >>>>     
> > >>>>         
> > >>>>> know how this could be accomplished?
> > >>>>> Thanks Ted
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>           
> > >>
> > __________________________________________________
> > >>     
> > >>>>> Do You Yahoo!?
> > >>>>> Tired of spam?  Yahoo! Mail has the best
> spam
> > >>>>>       
> > >>>>>           
> > >>>> protection around
> > >>>>     
> > >>>>         
> > >>>>> http://mail.yahoo.com
> > >>>>>
> > _______________________________________________
> > >>>>> cisco-voip mailing list
> > >>>>> cisco-voip at puck.nether.net
> > >>>>>
> > >>>>>       
> > >>>>>           
> > >>
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > >>     
> 
=== message truncated ===


__________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
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From erickbe at yahoo.com  Fri Jun 23 10:58:04 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Fri, 23 Jun 2006 07:58:04 -0700 (PDT)
Subject: [cisco-voip] VG248 and 6608,
	Garbage in terminal connections with modem dialup
Message-ID: <20060623145804.95903.qmail@web37215.mail.mud.yahoo.com>

Anyone have any ideas on how to tackle issues where on console/terminal dialup modem connections at lower baud rates (2400, 9600, etc) connect fine and work but there is intermintent garbage characters that pop up on the display and occassionally the connection disconnects after a few minutes. 

Many modem models were tried, and if a regular POTS line is used theres no problems calling the same equipment. 

CCM 4.0(2a)SR2b, Vg248 was on 1.21 but updated to 1.31 with same symptons. 6608 firmware is D004040000023. 

The VG248 is plugged into the 6500 with 6608 blade and CCM. No errors on the VG248 port or 6608 T1 ports. FAX's work fine and only these modem dialup connections are affected.

Have fax relay disabled on 6608 T1 port and on VG248 side (both in CCM and on Vg248 itself). Have tried adjusting the input gains on VG248 port and no change.

Thanks.


From tednugent69 at yahoo.com  Fri Jun 23 12:34:11 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 23 Jun 2006 09:34:11 -0700 (PDT)
Subject: [cisco-voip] Stable Call Mgr Version
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F013E78D@USNAEXCH.na.didata.local>
Message-ID: <20060623163411.20592.qmail@web36112.mail.mud.yahoo.com>

Yep.. thanks for the catch


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

> I think you had a typo sr2b doesn't exist, there is
> a ciscocm.4-1-3-sr3b.exe on CCO that came out this
> week -jason
> 
> ________________________________
> 
> From: cisco-voip-bounces at puck.nether.net on behalf
> of Ted Nugent
> Sent: Fri 6/23/2006 10:39 AM
> To: Ortiz, Carlos; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Stable Call Mgr Version
> 
> 
> 
> Well you're running on probably the most stable
> version of CM that 4.x has to offer... Other then
> that
> I would probably run with CM4.1(3)SR2... I've seen
> some issues with SR3 and thier still working them
> out... sr2b just came out the other day. I've only
> deployed Unity 4.1 a few times and each went very
> smoothly with very few followups.. (but 4.0(5)
> wasn't
> a bad release either... unless there are features
> you're looking for I would stick with 4.0(5))
> 
> I personally would not touch CM/Unity 4.2 yet...
> they're both fairly new... I would probably give
> them
> a couple month.
> 
> IPCC 3.5.3 SR3 seemed to be pretty stable as well...
> You could not pay me enough to support IPCCX 4.x
> yet... they are getting closer with newer releases
> but
> right now.. no way.. no how!
> You can also reference "golden bridge" at the link
> below.
> 
>
http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/ipcmtrix.htm
> 
> 
> --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
> 
> > Can anyone suggest a stable CM version and IPCC
> > Express that they have
> > successfully running?
> >
> > 
> >
> > I am currently running 4.02a(sr2b) with IPCC 3.5.2
> > and Unity 4.05.
> >
> > 
> >
> > I would like to move to Call Mgr version 4.1.3 or
> > 4.2 (if recommended).
> > I also realize that IPCC Express will need to be
> > upgraded as part of
> > this.
> >
> > 
> >
> > Thanks,
> >
> > 
> >
> > Carlos
> >
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> 
> 
> 
> 
> -----------------------------------------
> 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 rvillagran at coasin.cl  Fri Jun 23 12:37:22 2006
From: rvillagran at coasin.cl (Rodrigo Villagran)
Date: Fri, 23 Jun 2006 12:37:22 -0400
Subject: [cisco-voip] Background Logo
In-Reply-To: <OF9971A336.487933A5-ON86257195.007AF095-86257195.007BC857@tca.us.intdk>
Message-ID: <200606231636.k5NGashB028631@mail.coasin.cl>

That's true for the 7940/60 but for the 7912 you must need to do this:

7940/60:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186
a00801c0764.shtml

7912:
http://www.cisco.com/en/US/products/hw/phones/ps379/products_administration_
guide_chapter09186a008066c7c2.html


Greetings
Rodrigo V

-----Mensaje original-----
De: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] En nombre de
pdavidson at tdktca.com
Enviado el: jueves, 22 de junio de 2006 18:32
Para: cisco-voip at puck.nether.net
Asunto: Re: [cisco-voip] Background Logo

Danny-

There are only a few ways to do this- and unfortunately, under CCM4.X, I'm
not happy with any of them.

The by the book answer is to use the Idle Screen functionality.  You'll see
a setting for the URL to set for idle, as well as the idle timeout timer,
on the Device Settings page under CCM Administrator.  The URL must return a
valid CMXML object, containing the information you want to display.  This
functionality works- I run it regularly.

However- under CCM4, the definition of 'idle' is a little bit funky.  I've
had it kick in while I'm dialing before, replacing what I'm dialing and
stopping me from dialing more digits.  You also almost lose the onhook
dialahead functionality- you have to exit from the idle screen first, then
onhook dial- a retraining step for the users.

I have seen this done with custom authored phone loads, which involve BTXML
authoring- as a university, you might be able to get some help from Cisco
to do this.  I know that the 'managed' IP-PBX offering from Covad does this
on 7960's- I've seen them running- but that's using an MGCP load talking to
something other than Callmanager.

If you run the SIP image on the phone, it's MUCH simpler- just tell the
phone the name of a .bmp file on your TFTP server, and you're done- with no
loss of functionality.  But, CCM4 doesn't make SIP phone administration
simple- it will work, but it will be a very manual process for you.  CCM5
changes that, is my understanding- but the CCM5 SIP Phone load has other
oddities- your mileage may vary.

Hope this helps.

Paul Davidson, Manager, MIS
TDK Corporation of America
http://www.component.tdk.com



                                                                           
             cisco-voip-reques                                             
             t at puck.nether.net                                             
             Sent by:                                                   To 
             cisco-voip-bounce         cisco-voip at puck.nether.net          
             s at puck.nether.net                                          cc 
                                                                           
                                                                   Subject 
             06/22/2006 05:14          cisco-voip Digest, Vol 40, Issue    
             PM                        119                                 
                                                                           
                                                                           
             Please respond to                                             
             cisco-voip at puck.n                                             
                 ether.net                                                 
                                                                           
                                                                           





Date: Thu, 22 Jun 2006 18:16:41 -0400
From: "Phillips, Danny" <dphillips at sapc.edu>
Subject: [cisco-voip] Background Logo
To: <cisco-voip at puck.nether.net>
Message-ID: <18F0B5EFCCE1A449B07A3A9DF61DC07308ACE3 at xchange.sapc.edu>
Content-Type: text/plain; charset="us-ascii"

We want to add our college's logo to the 7912, 7940, and 7960 IP phones
running CM 4.1(3), could someone point me in the right direction.





Thanks

Danny,











`````````````````````````````````````````````````

Danny Phillips

Network Administrator

St. Andrews Presbyterian College

910-277-5079

dphillips at sapc.edu



`````````````````````````````````````````````````



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


From CORTIZ at broward.org  Fri Jun 23 13:41:47 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Fri, 23 Jun 2006 13:41:47 -0400
Subject: [cisco-voip] Stable Call Mgr Version
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC3537@MREXVS01.bc.broward.cty>

So according to this matrix CM 4.1.3 SR1 is supported with IPC 3.5.3 but
if you apply SR2 to CM then you have to upgrade IPCC to 4.x....does that
make any sense?

 

-----Original Message-----
From: Ted Nugent [mailto:tednugent69 at yahoo.com] 
Sent: Friday, June 23, 2006 10:40 AM
To: Ortiz, Carlos; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stable Call Mgr Version

Well you're running on probably the most stable
version of CM that 4.x has to offer... Other then that
I would probably run with CM4.1(3)SR2... I've seen
some issues with SR3 and thier still working them
out... sr2b just came out the other day. I've only
deployed Unity 4.1 a few times and each went very
smoothly with very few followups.. (but 4.0(5) wasn't
a bad release either... unless there are features
you're looking for I would stick with 4.0(5))

I personally would not touch CM/Unity 4.2 yet...
they're both fairly new... I would probably give them
a couple month. 

IPCC 3.5.3 SR3 seemed to be pretty stable as well...
You could not pay me enough to support IPCCX 4.x
yet... they are getting closer with newer releases but
right now.. no way.. no how! 
You can also reference "golden bridge" at the link
below.

http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/ipc
mtrix.htm


--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> Can anyone suggest a stable CM version and IPCC
> Express that they have
> successfully running?
> 
>  
> 
> I am currently running 4.02a(sr2b) with IPCC 3.5.2
> and Unity 4.05.
> 
>  
> 
> I would like to move to Call Mgr version 4.1.3 or
> 4.2 (if recommended).
> I also realize that IPCC Express will need to be
> upgraded as part of
> this.
> 
>  
> 
> Thanks,
> 
>  
> 
> Carlos
> 
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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


From tednugent69 at yahoo.com  Fri Jun 23 13:58:23 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 23 Jun 2006 10:58:23 -0700 (PDT)
Subject: [cisco-voip] Stable Call Mgr Version
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC3537@MREXVS01.bc.broward.cty>
Message-ID: <20060623175823.41815.qmail@web36109.mail.mud.yahoo.com>

Ahhh yes.. This is mistake that all of my SEs have
made at one time or another... I apologize I should
have clarified... this is NOT the CM compatibility
matrix.. This is Cisco IPT test release "aka Golden
Bridge". Cisco takes groupings of HW/SW and tests them
all together to try and find the best matches at a
given life cycle. They then document the bugs and
problems with interoperability. In a nutshell that is
the purpose of the GB. It's not perfect but it
provides an outline to build off of and which in a
Cisco Partner environment gives us a leg to stand on
when things go south. We try and follow GB for any new
releases that we bring to a customer (and intermix
that with field experience). 

The CM compatibility matrix is here
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/ccmcomp.htm

And the CRS/IPCCX matrix is here
http://www.cisco.com/univercd/cc/td/doc/product/voice/sw_ap_to/crscomtx.pdf



--- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:

> So according to this matrix CM 4.1.3 SR1 is
> supported with IPC 3.5.3 but
> if you apply SR2 to CM then you have to upgrade IPCC
> to 4.x....does that
> make any sense?
> 
>  
> 
> -----Original Message-----
> From: Ted Nugent [mailto:tednugent69 at yahoo.com] 
> Sent: Friday, June 23, 2006 10:40 AM
> To: Ortiz, Carlos; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Stable Call Mgr Version
> 
> Well you're running on probably the most stable
> version of CM that 4.x has to offer... Other then
> that
> I would probably run with CM4.1(3)SR2... I've seen
> some issues with SR3 and thier still working them
> out... sr2b just came out the other day. I've only
> deployed Unity 4.1 a few times and each went very
> smoothly with very few followups.. (but 4.0(5)
> wasn't
> a bad release either... unless there are features
> you're looking for I would stick with 4.0(5))
> 
> I personally would not touch CM/Unity 4.2 yet...
> they're both fairly new... I would probably give
> them
> a couple month. 
> 
> IPCC 3.5.3 SR3 seemed to be pretty stable as well...
> You could not pay me enough to support IPCCX 4.x
> yet... they are getting closer with newer releases
> but
> right now.. no way.. no how! 
> You can also reference "golden bridge" at the link
> below.
> 
>
http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/ipc
> mtrix.htm
> 
> 
> --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
> 
> > Can anyone suggest a stable CM version and IPCC
> > Express that they have
> > successfully running?
> > 
> >  
> > 
> > I am currently running 4.02a(sr2b) with IPCC 3.5.2
> > and Unity 4.05.
> > 
> >  
> > 
> > I would like to move to Call Mgr version 4.1.3 or
> > 4.2 (if recommended).
> > I also realize that IPCC Express will need to be
> > upgraded as part of
> > this.
> > 
> >  
> > 
> > Thanks,
> > 
> >  
> > 
> > Carlos
> > 
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From roharvey at flemingc.on.ca  Fri Jun 23 15:24:00 2006
From: roharvey at flemingc.on.ca (Robert Harvey)
Date: Fri, 23 Jun 2006 15:24:00 -0400
Subject: [cisco-voip] VG224 / SCCP / Modem
Message-ID: <s49c079b.000@freedom7.flemingc.on.ca>

Hi Jason,

I guess there has been a minor modification to fax/modem support under
SCCP in that they now state fax/modem pass-through is supported;
however, the recommendation remains to use alternate protocols for FXS
ports intended for fax/modem use.

R.

"Modem and fax-relay are not supported on FXS ports with SCCP enabled;
only modem and fax pass-through are supported. Configure FXS ports with
either fax or modems attached in H.323 or SIP mode."

from:
http://www.cisco.com/en/US/products/ps6441/prod_release_note09186a00804a19ae.html



Date: Fri, 23 Jun 2006 01:21:01 -0500 (CDT)
From: "Andrew Dignan" <cisco at dignans.com>
Subject: Re: [cisco-voip] VG224 / SCCP / Modem
To: "Jason Aarons (US)" <jason.aarons at us.didata.com>
Cc: cisco-voip at puck.nether.net
Message-ID: <2104.24.234.204.227.1151043661.squirrel at www.dignans.com>
Content-Type: text/plain;charset=iso-8859-1

I am having trouble finding this in the 12.4 release notes.  Do you
have a
link that states this?


> This is inaccurate.
>
> Using 4.1(3) with VG224 under 12.4T you can use modem/faxes with
SCCP.
>
> I've had a VG224 with stcapp (SCCP) running for 2 months with no
issues.
> My VG24 is co-located on a lan with the T1 CAS gateway.
>
> Clock slips/errorrs via show controller t1 will kill modems/faxes. 
Check
> your clocking, etc.
>
> ________________________________
>
> From: cisco-voip-bounces at puck.nether.net on behalf of Andrew Dignan
> Sent: Tue 6/20/2006 10:01 PM
> To: cisco-voip at puck.nether.net 
> Subject: Re: [cisco-voip] VG224 / SCCP / Modem
>
>
>
> The VG224 using the SCCP image does not support fax/modems:
>
>
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/prod_release_note09186a008041df56.html

>
> ------------------------------------------------
> *Fax and modems are not supported on FXS ports with SCCP enabled.
FXS
> ports with either fax or modems attached should be configured in
H.323 or
> SIP mode.
> -------------------------------------------------
>
> andy dignan
>
>> You can plug a cheap analog phone into that VG224 and after the same
x
>> minutes get disconnected? Verify you can reproduce the problem.
>>
>> Open a TAC case, sounds like you need some debugs, SDL traces,
takes
>> -jason
>>
>> ________________________________
>>
>> From: cisco-voip-bounces at puck.nether.net on behalf of Jeff Anderson
>> Sent: Tue 6/20/2006 12:50 PM
>> To: cisco-voip at puck.nether.net 
>> Subject: [cisco-voip] VG224 / SCCP / Modem
>>
>>
>> Question:
>>
>> I have a client who is using a VG224 that is configured to use SCCP
on
>> CCM
>> 4.1. The majority of the ports are used for modems but there are
>> devices
>> that need skinny features as well. The problem is the modem calls
fail
>> after a random amount of time. It could be anywhere from 5 - 30
>> minutes.
>> That same modem plugged directly into a 1FB will have no
reliability
>> issues.
>>
>> I have searched Cisco's website and their forums looking for some
>> configuration guidelines for modem usage and found nothing. Does
anyone
>> have any advice to offer?
>>
>> Thanks,
>> Jeff Anderson
>>
>>
>>
>>
>>
>>
>> -----------------------------------------
>> Disclaimer:
>>
>> This e-mail communication and any attachments may contain
>> confidential and privileged information and is for use by the
>> designated addressee(s) named above only.  If you are not the
>> intended addressee, you are hereby notified that you have received
>> this communication in error and that any use or reproduction of
>> this email or its contents is strictly prohibited and may be
>> unlawful.  If you have received this communication in error, please
>> notify us immediately by replying to this message and deleting it
>> from your computer. Thank you.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net 
>> https://puck.nether.net/mailman/listinfo/cisco-voip 
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net 
> https://puck.nether.net/mailman/listinfo/cisco-voip 
>
>
>
>
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>



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

Message: 4
Date: Fri, 23 Jun 2006 01:27:15 -0500
From: "Aman Chugh" <aman.chugh at gmail.com>
Subject: Re: [cisco-voip] Uninstalling AD connector and going back to
	DC	Directory...
To: "Jonathan Charles" <jonvoip at gmail.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Message-ID:
	<d4c697ad0606222327n1ac6b2b7m4d0750e89767c56 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

if the plugin was installed then it should be listed in ADD/REMOVE
programmes.You can uninstall it by running it again,Also if you need
to
verfiy your directory configuration check the following registery key

HKEY_LOCAL_MACHINE\SOFTWARE\Cisco Systems, Inc.\Directory
Configuration

- Aman

On 6/22/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> We accidentally integrated our CallManager 4.1(3)sr3b publisher with
> Active Directory.
>
> Since then, nothing but problems and we want to back out.
>
> We followed the instructions here:
>
> http://www.cisco.com/warp/public/788/AVVID/change2_dcdirectory.html 
>
> But they don't work.
>
> The document includes this line:
>
> "The proper way to back out of this configuration and revert to
using
> the embedded DC Directory is to re-run the plugin installation and
> choose the option to uninstall the plugin."
>
> However, there is no such option on the installer and it is not
listed
> in the add/remove programs on the PUB.
>
> We need this reverted, it is preventing us from installing IPCC
> Express... I am trying to avoid opening a TAC case.
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net 
> https://puck.nether.net/mailman/listinfo/cisco-voip 
>
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------------------------------

Message: 5
Date: Fri, 23 Jun 2006 09:46:53 +0300
From: Arunas Sutkevicius <ARUNAS at lt.ibm.com>
Subject: Re: [cisco-voip] Uninstalling AD connector and going back to
	DC	Directory...
To: "Jonathan Charles" <jonvoip at gmail.com>
Cc: ciscovoip <cisco-voip at puck.nether.net>
Message-ID:
	<OF2D6D2485.74DD141B-ONC2257196.00254136-C2257196.0025C6EB at lt.ibm.com>
Content-Type: text/plain; charset="us-ascii"

Probably you are hitting the CSCsb20774 bug. So solution of this bug 
should help solve installation problems, without reverting to DC 
Directory.

Regards,
Arunas




"Jonathan Charles" <jonvoip at gmail.com> 
Sent by: cisco-voip-bounces at puck.nether.net 
2006.06.23 07:33

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

Subject
[cisco-voip] Uninstalling AD connector and going back to DC
Directory...






We accidentally integrated our CallManager 4.1(3)sr3b publisher with
Active Directory.

Since then, nothing but problems and we want to back out.

We followed the instructions here:

http://www.cisco.com/warp/public/788/AVVID/change2_dcdirectory.html 

But they don't work.

The document includes this line:

"The proper way to back out of this configuration and revert to using
the embedded DC Directory is to re-run the plugin installation and
choose the option to uninstall the plugin."

However, there is no such option on the installer and it is not listed
in the add/remove programs on the PUB.

We need this reverted, it is preventing us from installing IPCC
Express... I am trying to avoid opening a TAC case.



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

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

Message: 6
Date: Fri, 23 Jun 2006 09:25:48 +0200
From: "Brian Turnbow" <b.turnbow at twt.it>
Subject: Re: [cisco-voip] Strip name in FROM in invite?
To: "Sean Watkins" <sean at northrock.bm>,	"ciscovoip"
	<cisco-voip at puck.nether.net>
Message-ID:
	<52964DBFB59FA34F98E96921BE21C90601465A5E at twt-exch.milano.twt>
Content-Type: text/plain;	charset="iso-8859-1"

You cannot strip the from field but you can change it.
Take a look at 
http://www.cisco.com/en/US/products/ps6441/prod_troubleshooting_guide09186a00801d3d3a.html#wp1385050


Regards
Brian 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sean
Watkins
Sent: gioved? 22 giugno 2006 20.46
To: 'ciscovoip'
Subject: [cisco-voip] Strip name in FROM in invite?


Hi,

Is there any easy way to strip the name from a SIP invite leaving a
Cisco
GW?

Ie the "unknown" junk below? 

Sean




Jun 22 15:16:39 ADT: Sent: 
INVITE sip:1990X at X:5060 SIP/2.0
Via: SIP/2.0/UDP  X:5060;x-route-tag="tgrp:PRIs"
From: "unknown" <sip:X>;tag=AA3CD44-E68  <----------------------------
To: <sip:1990X at X>
Date: Thu, 22 Jun 2006 18:16:39 GMT
Call-ID: 15374E06-15211DB-8043CCE5-2DF7505B at X
Supported: timer,100rel
Min-SE:  1800
Cisco-Guid: 355866039-22155739-2151795941-771182683
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE,
NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 1151000199
Contact: <sip:X:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 274
--
Sean Watkins
North Rock Communications
Phone: 441-540-4102
 
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net 
https://puck.nether.net/mailman/listinfo/cisco-voip 



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

Message: 7
Date: Fri, 23 Jun 2006 08:28:46 -0400
From: "Ortiz, Carlos" <CORTIZ at broward.org>
Subject: [cisco-voip] Stable Call Mgr Version
To: <cisco-voip at puck.nether.net>
Message-ID:
	<26B357EF9FB7D04A989C9E82A9E5436E01BC352B at MREXVS01.bc.broward.cty>
Content-Type: text/plain; charset="us-ascii"

Can anyone suggest a stable CM version and IPCC Express that they have
successfully running?

 

I am currently running 4.02a(sr2b) with IPCC 3.5.2 and Unity 4.05.

 

I would like to move to Call Mgr version 4.1.3 or 4.2 (if
recommended).
I also realize that IPCC Express will need to be upgraded as part of
this.

 

Thanks,

 

Carlos

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End of cisco-voip Digest, Vol 40, Issue 121
*******************************************

----------------------------------------------------------------
Robert Harvey
Network Support Analyst
Fleming College
Peterborough, ON, Canada
Office (705) 749-5513
Fax (705) 749-5540
experience.success
----------------------------------------------------------------

From jonvoip at gmail.com  Fri Jun 23 18:08:51 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 23 Jun 2006 17:08:51 -0500
Subject: [cisco-voip] IPCC 4.0(4) upgrading from 3.5(3)
Message-ID: <5d093f9a0606231508w60b126f2j39656c450fb78ac@mail.gmail.com>

Simple off-box IPCC Express 3.5 upgrading to 4.0(4).... I am starting
the install, I hit the 'backup' button to back up the 3.5, and it asks
for license files after doing some checks...

It comes up and gives me a warning that my IPCC 4 license doesn't have
the same ports as the 3.5 license (less), which is ok, and when I hit
ok,

It does some stuff, says, "Database requested MSDE" and it give me this error:

"Caught Exception accessing Spanlink LCC LDAP data. Please check the log file."

And the installation terminates.

Is there any way to bypass this backup thing (There isn't really
anything there...)

Or fix this error?

There is no log file, I checked the system for all files and nothing
from the time of the error (related to this anyway).




Jonathan

From jonvoip at gmail.com  Fri Jun 23 22:25:01 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 23 Jun 2006 22:25:01 -0400
Subject: [cisco-voip] Uninstalling AD connector and going back to DC
	Directory...
In-Reply-To: <OF2D6D2485.74DD141B-ONC2257196.00254136-C2257196.0025C6EB@lt.ibm.com>
References: <5d093f9a0606222133k729ec700n14cc2022130cb69@mail.gmail.com>
	<OF2D6D2485.74DD141B-ONC2257196.00254136-C2257196.0025C6EB@lt.ibm.com>
Message-ID: <5d093f9a0606231925v69e91ea3n378bc11ffdddaec7@mail.gmail.com>

OK, I found the problem...

This doc,
http://www.cisco.com/warp/public/788/AVVID/change2_dcdirectory.html is
wrong.

It leaves out quite a bit of what you need to actually do.

First off, you need to edit the ini file in the c:\dcdsrvr directory (
Directorymanager.ini or something...) the same way you did the registry.

Second, after you finish this, you need to rerun the AD integration plugin,
which now has an option to uninstall the plugin. And then do this all over
again on each and every subscriber.

After that... we were golden and completely AD free...



Jonathan

On 6/23/06, Arunas Sutkevicius <ARUNAS at lt.ibm.com> wrote:
>
>
> Probably you are hitting the CSCsb20774 bug. So solution of this bug
> should help solve installation problems, without reverting to DC Directory.
>
> Regards,
> Arunas
>
>
>
>  *"Jonathan Charles" <jonvoip at gmail.com>*
> Sent by: cisco-voip-bounces at puck.nether.net
>
> 2006.06.23 07:33
>   To
> ciscovoip <cisco-voip at puck.nether.net>  cc
>
>  Subject
> [cisco-voip] Uninstalling AD connector and going back to DC
>  Directory...
>
>
>
>
>
>
> We accidentally integrated our CallManager 4.1(3)sr3b publisher with
> Active Directory.
>
> Since then, nothing but problems and we want to back out.
>
> We followed the instructions here:
>
> http://www.cisco.com/warp/public/788/AVVID/change2_dcdirectory.html
>
> But they don't work.
>
> The document includes this line:
>
> "The proper way to back out of this configuration and revert to using
> the embedded DC Directory is to re-run the plugin installation and
> choose the option to uninstall the plugin."
>
> However, there is no such option on the installer and it is not listed
> in the add/remove programs on the PUB.
>
> We need this reverted, it is preventing us from installing IPCC
> Express... I am trying to avoid opening a TAC case.
>
>
>
> Jonathan
> _______________________________________________
> 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  Fri Jun 23 22:27:59 2006
From: it at cimgroup.com (IT)
Date: Fri, 23 Jun 2006 19:27:59 -0700
Subject: [cisco-voip] Nokia E61 CCM Client
Message-ID: <B487D4C1FE6D04439061018C6EBC109B0345CBAF@exchange.cimgroup.com>

Where did you see this? Do I need to load specific software on the
phone, or is the built-in SIP client enough?


-----Original Message-----
From: Carter, Bill [mailto:bcarter at sentinel.com] 
Sent: Wednesday, June 21, 2006 3:25 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Nokia E61 CCM Client

I think I saw you need to configure the phone in CallManager as a 7960. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, June 21, 2006 5:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Nokia E61 CCM Client

Hello,

I have the Nokia E61 (amazing little device), and I am dying to try the
CallManager Wifi integration. Does anyone know when Cisco will be
releasing the client? Or is this a message better suited for the Nokia
mailing list?

Here is the Nokia page:
http://www.businesssoftware.nokia.com/cisco_systems.php

Here is the Cisco page:
http://www.cisco.com/iam/unified/ipcc1/IP_Communications.htm


Anyone know if I can beta-test this?

Thanks,
Avidan

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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 jgrace at digitelusa.net  Fri Jun 23 22:15:32 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri, 23 Jun 2006 22:15:32 -0400
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
Message-ID: <200606232227250.SM01452@3729434770>

We have a 3750 with dhcp scope config on it.  We have all type of phone and
wireless ap (cisco) that are pulling from this scope.  But for some reason
the 7940s and 7960s are grabing ip from the data remote dhcp server.  The
switch is not recognizing these phones

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From jonvoip at gmail.com  Fri Jun 23 22:38:46 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 23 Jun 2006 22:38:46 -0400
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
In-Reply-To: <200606232227250.SM01452@3729434770>
References: <200606232227250.SM01452@3729434770>
Message-ID: <5d093f9a0606231938l506486eft36f53e774835f063@mail.gmail.com>

Do they show up in a 'show cdp neighbors'? If not, you may want to
enable CDP so the phones can see the voice vlan.


Jonathan

On 6/23/06, James Grace <jgrace at digitelusa.net> wrote:
>
>
>
>
> We have a 3750 with dhcp scope config on it.  We have all type of phone and
> wireless ap (cisco) that are pulling from this scope.  But for some reason
> the 7940s and 7960s are grabing ip from the data remote dhcp server.  The
> switch is not recognizing these phones
>
>
>
> James D. Grace
>
> CCNP CCNA MCSE MCDBA
>
> Sr. System Engineer / Professional Svc.
>
> Digitel Corporation
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From jonvoip at gmail.com  Fri Jun 23 22:39:47 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 23 Jun 2006 22:39:47 -0400
Subject: [cisco-voip] Stable Call Mgr Version
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC352B@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC352B@MREXVS01.bc.broward.cty>
Message-ID: <5d093f9a0606231939k39aab90bxd3dd16cb34e810b7@mail.gmail.com>

By far, the most stable and fastest version of CallManager is
3.3(5)sr2b... none of that having to wait for some app while
configuring the system... and it is rock stable.



Jonathan

On 6/23/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
>
>
>
>
> Can anyone suggest a stable CM version and IPCC Express that they have
> successfully running?
>
>
>
> I am currently running 4.02a(sr2b) with IPCC 3.5.2 and Unity 4.05.
>
>
>
> I would like to move to Call Mgr version 4.1.3 or 4.2 (if recommended).  I
> also realize that IPCC Express will need to be upgraded as part of this.
>
>
>
> Thanks,
>
>
>
> Carlos
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From aman.chugh at gmail.com  Fri Jun 23 22:41:39 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Fri, 23 Jun 2006 21:41:39 -0500
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
In-Reply-To: <200606232227250.SM01452@3729434770>
References: <200606232227250.SM01452@3729434770>
Message-ID: <d4c697ad0606231941s6cf78fc6te5fe689cab81dd83@mail.gmail.com>

What Phone load you using on the phones ,please provide the configuration of
a port on which you have phone connected which is doing this.

Thanks
Aman


On 6/23/06, James Grace <jgrace at digitelusa.net> wrote:
>
>   We have a 3750 with dhcp scope config on it.  We have all type of phone
> and wireless ap (cisco) that are pulling from this scope.  But for some
> reason the 7940s and 7960s are grabing ip from the data remote dhcp server.
> The switch is not recognizing these phones
>
>
>
> James D. Grace
>
> *CCNP CCNA MCSE MCDBA*
>
> Sr. System Engineer / Professional Svc.
>
> *Digitel Corporation*
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From tednugent69 at yahoo.com  Fri Jun 23 22:42:32 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 23 Jun 2006 19:42:32 -0700 (PDT)
Subject: [cisco-voip] Nokia E61 CCM Client
In-Reply-To: <B487D4C1FE6D04439061018C6EBC109B0345CBAF@exchange.cimgroup.com>
Message-ID: <20060624024232.83333.qmail@web36112.mail.mud.yahoo.com>

I think this is what you're looking for... still not
sure where you get it but at least it's a start..

http://www.europe.nokia.com/nokia/0,0,85515,0.html#software


--- IT <it at cimgroup.com> wrote:

> Where did you see this? Do I need to load specific
> software on the
> phone, or is the built-in SIP client enough?
> 
> 
> -----Original Message-----
> From: Carter, Bill [mailto:bcarter at sentinel.com] 
> Sent: Wednesday, June 21, 2006 3:25 PM
> To: IT; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Nokia E61 CCM Client
> 
> I think I saw you need to configure the phone in
> CallManager as a 7960. 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of IT
> Sent: Wednesday, June 21, 2006 5:20 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Nokia E61 CCM Client
> 
> Hello,
> 
> I have the Nokia E61 (amazing little device), and I
> am dying to try the
> CallManager Wifi integration. Does anyone know when
> Cisco will be
> releasing the client? Or is this a message better
> suited for the Nokia
> mailing list?
> 
> Here is the Nokia page:
>
http://www.businesssoftware.nokia.com/cisco_systems.php
> 
> Here is the Cisco page:
>
http://www.cisco.com/iam/unified/ipcc1/IP_Communications.htm
> 
> 
> Anyone know if I can beta-test this?
> 
> Thanks,
> Avidan
> 
> _______________________________________________
> 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.
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jonvoip at gmail.com  Fri Jun 23 22:45:05 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 23 Jun 2006 22:45:05 -0400
Subject: [cisco-voip] Stable Call Mgr Version
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC3537@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC3537@MREXVS01.bc.broward.cty>
Message-ID: <5d093f9a0606231945o2dbc624v568aeba727bc3c8f@mail.gmail.com>

Are you looking at this:

http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1225/ccmigration_09186a00803d82f5.pdf

Cuz, I am running 4.1(3)sr3b with IPCC 3.5(2)sr3 and page 48 of that
compatibility guide shows 3.5(3)sr3 being supported in 4.1(3)sr3a (and
I have it running with 4.1(3)sr3b (which came out last week).

In fact, that shows 3.5(4) supporting CCM4.2... So, I am not sure what
you are looking at...



Jonathan

On 6/23/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
> So according to this matrix CM 4.1.3 SR1 is supported with IPC 3.5.3 but
> if you apply SR2 to CM then you have to upgrade IPCC to 4.x....does that
> make any sense?
>
>
>
> -----Original Message-----
> From: Ted Nugent [mailto:tednugent69 at yahoo.com]
> Sent: Friday, June 23, 2006 10:40 AM
> To: Ortiz, Carlos; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Stable Call Mgr Version
>
> Well you're running on probably the most stable
> version of CM that 4.x has to offer... Other then that
> I would probably run with CM4.1(3)SR2... I've seen
> some issues with SR3 and thier still working them
> out... sr2b just came out the other day. I've only
> deployed Unity 4.1 a few times and each went very
> smoothly with very few followups.. (but 4.0(5) wasn't
> a bad release either... unless there are features
> you're looking for I would stick with 4.0(5))
>
> I personally would not touch CM/Unity 4.2 yet...
> they're both fairly new... I would probably give them
> a couple month.
>
> IPCC 3.5.3 SR3 seemed to be pretty stable as well...
> You could not pay me enough to support IPCCX 4.x
> yet... they are getting closer with newer releases but
> right now.. no way.. no how!
> You can also reference "golden bridge" at the link
> below.
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/ipc
> mtrix.htm
>
>
> --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
>
> > Can anyone suggest a stable CM version and IPCC
> > Express that they have
> > successfully running?
> >
> >
> >
> > I am currently running 4.02a(sr2b) with IPCC 3.5.2
> > and Unity 4.05.
> >
> >
> >
> > I would like to move to Call Mgr version 4.1.3 or
> > 4.2 (if recommended).
> > I also realize that IPCC Express will need to be
> > upgraded as part of
> > this.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Carlos
> >
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jwydra at Burwood.com  Fri Jun 23 22:50:31 2006
From: jwydra at Burwood.com (Wydra, Jason)
Date: Fri, 23 Jun 2006 21:50:31 -0500
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
Message-ID: <97F472A9F637E44B954D7FBABC800430C148C3@bur-msg-02.Burwood.com>

Are you doing true trunking (encap dot1q) on these ports or are you
doing an access vlan and voice vlan? 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Friday, June 23, 2006 9:39 PM
To: James Grace
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch

Do they show up in a 'show cdp neighbors'? If not, you may want to
enable CDP so the phones can see the voice vlan.


Jonathan

On 6/23/06, James Grace <jgrace at digitelusa.net> wrote:
>
>
>
>
> We have a 3750 with dhcp scope config on it.  We have all type of
phone and
> wireless ap (cisco) that are pulling from this scope.  But for some
reason
> the 7940s and 7960s are grabing ip from the data remote dhcp server.
The
> switch is not recognizing these phones
>
>
>
> James D. Grace
>
> CCNP CCNA MCSE MCDBA
>
> Sr. System Engineer / Professional Svc.
>
> Digitel Corporation
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip



From tednugent69 at yahoo.com  Fri Jun 23 23:02:44 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Fri, 23 Jun 2006 20:02:44 -0700 (PDT)
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
In-Reply-To: <97F472A9F637E44B954D7FBABC800430C148C3@bur-msg-02.Burwood.com>
Message-ID: <20060624030244.78319.qmail@web36105.mail.mud.yahoo.com>

Another Problem I've seen that could be contributing
to this is if you set the access vlan and voice vlan
the same. (I think it's a bug but have not
investigated it).
sw acces vlan 4
sw voice vlan 4
Changing to a different access vlan typically fixes
this. Also mokeying with the DHCP properties have
caused the same problem and simply changing the access
vlan to to something else and changing it back
resolves the problem. You may need to down the sw port
to get this to take.
 

--- "Wydra, Jason" <jwydra at Burwood.com> wrote:

> Are you doing true trunking (encap dot1q) on these
> ports or are you
> doing an access vlan and voice vlan? 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Jonathan
> Charles
> Sent: Friday, June 23, 2006 9:39 PM
> To: James Grace
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] 7940 7960 wont grab ip
> from dhcp off switch
> 
> Do they show up in a 'show cdp neighbors'? If not,
> you may want to
> enable CDP so the phones can see the voice vlan.
> 
> 
> Jonathan
> 
> On 6/23/06, James Grace <jgrace at digitelusa.net>
> wrote:
> >
> >
> >
> >
> > We have a 3750 with dhcp scope config on it.  We
> have all type of
> phone and
> > wireless ap (cisco) that are pulling from this
> scope.  But for some
> reason
> > the 7940s and 7960s are grabing ip from the data
> remote dhcp server.
> The
> > switch is not recognizing these phones
> >
> >
> >
> > James D. Grace
> >
> > CCNP CCNA MCSE MCDBA
> >
> > Sr. System Engineer / Professional Svc.
> >
> > Digitel Corporation
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> _______________________________________________
> 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
> 


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

From jonvoip at gmail.com  Fri Jun 23 23:05:20 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 23 Jun 2006 23:05:20 -0400
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
In-Reply-To: <20060624030244.78319.qmail@web36105.mail.mud.yahoo.com>
References: <97F472A9F637E44B954D7FBABC800430C148C3@bur-msg-02.Burwood.com>
	<20060624030244.78319.qmail@web36105.mail.mud.yahoo.com>
Message-ID: <5d093f9a0606232005k2df0286du322cd9059c5039ca@mail.gmail.com>

Another fun thing is if there is a rogue VLAN sitting in the vlan database...

Do a show vlan and see if there are VLANs on the switch that don't
make sense to you... if so, go into VLAN database and delete them...

Also, note the fact that you CANNOT configure a 3550, 3560 or 3570 to
have a voice vlan and configured as a trunk port... the voice vlan is
ignored...



Jonathan

On 6/23/06, Ted Nugent <tednugent69 at yahoo.com> wrote:
> Another Problem I've seen that could be contributing
> to this is if you set the access vlan and voice vlan
> the same. (I think it's a bug but have not
> investigated it).
> sw acces vlan 4
> sw voice vlan 4
> Changing to a different access vlan typically fixes
> this. Also mokeying with the DHCP properties have
> caused the same problem and simply changing the access
> vlan to to something else and changing it back
> resolves the problem. You may need to down the sw port
> to get this to take.
>
>
> --- "Wydra, Jason" <jwydra at Burwood.com> wrote:
>
> > Are you doing true trunking (encap dot1q) on these
> > ports or are you
> > doing an access vlan and voice vlan?
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Jonathan
> > Charles
> > Sent: Friday, June 23, 2006 9:39 PM
> > To: James Grace
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] 7940 7960 wont grab ip
> > from dhcp off switch
> >
> > Do they show up in a 'show cdp neighbors'? If not,
> > you may want to
> > enable CDP so the phones can see the voice vlan.
> >
> >
> > Jonathan
> >
> > On 6/23/06, James Grace <jgrace at digitelusa.net>
> > wrote:
> > >
> > >
> > >
> > >
> > > We have a 3750 with dhcp scope config on it.  We
> > have all type of
> > phone and
> > > wireless ap (cisco) that are pulling from this
> > scope.  But for some
> > reason
> > > the 7940s and 7960s are grabing ip from the data
> > remote dhcp server.
> > The
> > > switch is not recognizing these phones
> > >
> > >
> > >
> > > James D. Grace
> > >
> > > CCNP CCNA MCSE MCDBA
> > >
> > > Sr. System Engineer / Professional Svc.
> > >
> > > Digitel Corporation
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >
> > _______________________________________________
> > 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
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>

From bababooey at cox.net  Sat Jun 24 01:22:58 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Sat, 24 Jun 2006 01:22:58 -0400
Subject: [cisco-voip] Operator setup
References: <mailman.10674.1151090674.1478.cisco-voip@puck.nether.net>
Message-ID: <000601c6974e$41973670$0f0110ac@corporation2>

In a call manager system I have my operator attached through a pbx, can I 
make it so users can just dial "0"
and convert this to a seven digit number to dial thru to the pbx?

Also, if I have a range of extensions from 1001 - 1010, I would like anyone 
who dial 1002-1010 to be directed to 1001?
Is this possbile in CM 4.1.3?  All extensions should have full dial out 
capability

Thanks
Bob



From bababooey at cox.net  Sat Jun 24 01:27:27 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Sat, 24 Jun 2006 01:27:27 -0400
Subject: [cisco-voip] Publisher as dhcp server
References: <mailman.10776.1151116959.1478.cisco-voip@puck.nether.net>
Message-ID: <000d01c6974e$e20ec5a0$0f0110ac@corporation2>

does it go against best practice to utilize the ccm publisher as a dhcp 
server ccm 4.1.3?



From ealeatherman at gmail.com  Sat Jun 24 02:10:28 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Sat, 24 Jun 2006 02:10:28 -0400
Subject: [cisco-voip] Stable Call Mgr Version
In-Reply-To: <26B357EF9FB7D04A989C9E82A9E5436E01BC352B@MREXVS01.bc.broward.cty>
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC352B@MREXVS01.bc.broward.cty>
Message-ID: <94a1afde0606232310g5d88a730r81d460c4bafaf02a@mail.gmail.com>

CCM 4.1.3 has been pretty solid for us.
We just deployed IPCC Express 4.0(x) and are already running into multiple
bugs with it.. I think maybe if you can hold off on the 3.X versions of that
you might want to for now.. just based on our experience so far and what i'm
reading here.

We're looking at upgrading Unity from 4.0.4, either to 4.1 or 4.2.. fwiw I
had a engineer from cisco tell me this week that he recommended 4.2.

On 6/23/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
>
>  Can anyone suggest a stable CM version and IPCC Express that they have
> successfully running?
>
>
>
> I am currently running 4.02a(sr2b) with IPCC 3.5.2 and Unity 4.05.
>
>
>
> I would like to move to Call Mgr version 4.1.3 or 4.2 (if recommended).  I
> also realize that IPCC Express will need to be upgraded as part of this.
>
>
>
> Thanks,
>
>
>
> Carlos
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Sat Jun 24 02:17:29 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Sat, 24 Jun 2006 02:17:29 -0400
Subject: [cisco-voip] Unified Operations Manager 1.1 - Anyone installed
	it yet?
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E7778DAD@svcexch02.firm.fenwick.llp>
References: <965C0255FD5828438E34F9A68ECA76E7778DAD@svcexch02.firm.fenwick.llp>
Message-ID: <94a1afde0606232317n103776bft731d5b8ee75310c7@mail.gmail.com>

I'm interested also :)

On 6/22/06, Stu Packett <SPackett at fenwick.com> wrote:
>
>  I just got the CDs for Unified Operations Manager 1.1, but have not yet
> installed it as I have to decommission and rebuild another server for it.
> Has anyone installed it yet?  What kind of load does it put on the server?
>
> Thanks in advance.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Sat Jun 24 02:27:33 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Sat, 24 Jun 2006 02:27:33 -0400
Subject: [cisco-voip] Latancy / Jitter
In-Reply-To: <s4995d9c.040@comrcm01intgwia.mandtbank.com>
References: <s4995d9c.040@comrcm01intgwia.mandtbank.com>
Message-ID: <94a1afde0606232327w53b0df28i136e45ff3dafcee6@mail.gmail.com>

It seems to me when I actually look at it there is some jitter right at the
start of the call and then it goes down to virtually nothing for the rest of
the call.. i wonder if there is just some sort of path setup in the
switching on the network (or i'm just making it up cause it sounds good ;))

Also tend to see more jitter when talking to Unity.


On 6/21/06, STEVEN CASPER <SCASPER at mandtbank.com> wrote:
>
> How about Max Jitter - we sometimes see as much as 50-60 ms though Avg
> Jitter is low.
>
> Steve
>
>
>
-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From ric at bonbon.net  Sat Jun 24 04:29:57 2006
From: ric at bonbon.net (Ric)
Date: Sat, 24 Jun 2006 16:29:57 +0800
Subject: [cisco-voip] Unregistered remote handset in CM -> forward call
	out PSTN
In-Reply-To: <f1b65f880606191419t12613159j572e98abb86be0e2@mail.gmail.com>
References: <mailman.9011.1150743082.1478.cisco-voip@puck.nether.net>
	<24DFD2417491E547AC80AE140706C485039D9F93@EX001.homesbyavi.local>
	<f1b65f880606191419t12613159j572e98abb86be0e2@mail.gmail.com>
Message-ID: <82BB5AA1-1215-4FB3-8828-0A839C2E1BAB@bonbon.net>

This feature is only available in CM 4.2 today - no CM 5.0 release  
has it nor any 4.1 or earlier.

Ric

On 20 Jun 2006, at 5:19 AM, Justin Steinberg wrote:

> With CallManager 4.2(1) you have new fields on the DN for:
> Forward Unregistered Internal & Forward Unregistered External
>
> I'm not sure what flavors of CCM 5.x have those fields or whether this
> can be accomplished in 4.1 or below via some type of HG config.
>
> Justin
>
>
>
> On 6/19/06, Corbett Enders <cenders at homesbyavi.com> wrote:
>> Have the situation where we have a number of remote sites using 2811s
>> and SRST (using FXO boards for analog telco lines).
>>
>> If the WAN gets dropped, any calls outoing leave via the analog  
>> line. If
>> we call the unpublished analog number, calls ring in on the primary
>> phone.
>>
>> However, we'd like anyone trying to dial the extension from head  
>> office
>> to leave head office via the PRI and ring through to the unpublished
>> analog number on the FXO port.  Doesn't seem to be a field for  
>> this. The
>> field "Forward No Coverage" seems to only apply to CTI route  
>> points...
>>
>> 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 kthorngr at cisco.com  Sat Jun 24 09:15:45 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Sat, 24 Jun 2006 09:15:45 -0400
Subject: [cisco-voip] Publisher as dhcp server
In-Reply-To: <000d01c6974e$e20ec5a0$0f0110ac@corporation2>
References: <mailman.10776.1151116959.1478.cisco-voip@puck.nether.net>
	<000d01c6974e$e20ec5a0$0f0110ac@corporation2>
Message-ID: <8384b155d0c29fa40bc29d4cab19fc51@cisco.com>

This chapter of the CCM 4.x SRND discusses placing the DHCP server on  
the Publisher:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
products_implementation_design_guide_chapter09186a0080447513.html

"DHCP, Standalone versus Coresident Server

Typically, DHCP should be configured to run on a standalone server  
because a large number of devices requesting DHCP configuration could  
cause high CPU and memory usage, which can impact server performance.  
Therefore, you should not run DHCP on a Cisco CallManager server. In  
some instances, given a small Cisco CallManager deployment with no more  
than 1000 devices registering to the cluster, you may run DHCP on a  
Cisco CallManager server; however, if the server experiences high CPU  
load, you should move DHCP to a standalone server. If more than 1000  
devices are registered to the cluster, DHCP must be run on a standalone  
server. "

Kevin
On Jun 24, 2006, at 1:27 AM, Bob A. Bowie wrote:

> does it go against best practice to utilize the ccm publisher as a dhcp
> server ccm 4.1.3?
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From kthorngr at cisco.com  Sat Jun 24 09:18:12 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Sat, 24 Jun 2006 09:18:12 -0400
Subject: [cisco-voip] Operator setup
In-Reply-To: <000601c6974e$41973670$0f0110ac@corporation2>
References: <mailman.10674.1151090674.1478.cisco-voip@puck.nether.net>
	<000601c6974e$41973670$0f0110ac@corporation2>
Message-ID: <815b27f415b809aff53b582de9d8b807@cisco.com>

You can use Translation Patterns to solve both of these issues.  Create 
a Translation Pattern for 0 and translate the called number to the 7 
digit number.  Make sure to set the CSS appropriately.

For the second you can create two Translation Patterns; one pattern of 
100[2-9] and the other of 1010.  Translate the called number to 1001 
and set the CSS.

Kevin
On Jun 24, 2006, at 1:22 AM, Bob A. Bowie wrote:

> In a call manager system I have my operator attached through a pbx, 
> can I
> make it so users can just dial "0"
> and convert this to a seven digit number to dial thru to the pbx?
>
> Also, if I have a range of extensions from 1001 - 1010, I would like 
> anyone
> who dial 1002-1010 to be directed to 1001?
> Is this possbile in CM 4.1.3?  All extensions should have full dial out
> capability
>
> Thanks
> Bob
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


From tednugent69 at yahoo.com  Sat Jun 24 09:21:47 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sat, 24 Jun 2006 06:21:47 -0700 (PDT)
Subject: [cisco-voip] Publisher as dhcp server
In-Reply-To: <000d01c6974e$e20ec5a0$0f0110ac@corporation2>
Message-ID: <20060624132147.83273.qmail@web36101.mail.mud.yahoo.com>

I don't know that it's "AGAINST" best practice however
I personally would recommend against. I've seen tons
of customer run dhcp off the PUB without a hitch, well
at least until I went in and upgraded thier servers
and forgot to reconfigure DHCP (whoops).
On the other hand there are other who have had nothing
but problem with M$ DHCP, I'm not sure if this boils
down to versioning or what? I think it's more personal
preference but I alway like to use an IOS device for
my Phones DHCP and I've yet to have any problem
running on IOS.

--- "Bob A. Bowie" <bababooey at cox.net> wrote:

> does it go against best practice to utilize the ccm
> publisher as a dhcp 
> server ccm 4.1.3?
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From bababooey at cox.net  Sat Jun 24 10:51:18 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Sat, 24 Jun 2006 10:51:18 -0400
Subject: [cisco-voip] Operator setup
References: <mailman.10674.1151090674.1478.cisco-voip@puck.nether.net>
	<000601c6974e$41973670$0f0110ac@corporation2>
	<815b27f415b809aff53b582de9d8b807@cisco.com>
Message-ID: <001101c6979d$a6cb0360$0f0110ac@corporation2>

Kevin, thanks for the answers, I am not sure what needs to be set in the CSS 
appropratiely,
I currently have one CSS for all phones, will i need to adjust this?

Thanks
Bob


----- Original Message ----- 
From: "Kevin Thorngren" <kthorngr at cisco.com>
To: "Bob A. Bowie" <bababooey at cox.net>
Cc: <cisco-voip at puck.nether.net>
Sent: Saturday, June 24, 2006 9:18 AM
Subject: Re: [cisco-voip] Operator setup


> You can use Translation Patterns to solve both of these issues.  Create a 
> Translation Pattern for 0 and translate the called number to the 7 digit 
> number.  Make sure to set the CSS appropriately.
>
> For the second you can create two Translation Patterns; one pattern of 
> 100[2-9] and the other of 1010.  Translate the called number to 1001 and 
> set the CSS.
>
> Kevin
> On Jun 24, 2006, at 1:22 AM, Bob A. Bowie wrote:
>
>> In a call manager system I have my operator attached through a pbx, can I
>> make it so users can just dial "0"
>> and convert this to a seven digit number to dial thru to the pbx?
>>
>> Also, if I have a range of extensions from 1001 - 1010, I would like 
>> anyone
>> who dial 1002-1010 to be directed to 1001?
>> Is this possbile in CM 4.1.3?  All extensions should have full dial out
>> capability
>>
>> Thanks
>> Bob
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
> 



From jgrace at digitelusa.net  Sat Jun 24 11:21:23 2006
From: jgrace at digitelusa.net (James Grace)
Date: Sat, 24 Jun 2006 11:21:23 -0400
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
In-Reply-To: <5d093f9a0606232005k2df0286du322cd9059c5039ca@mail.gmail.com>
Message-ID: <200606241133187.SM01452@3729434770>

What I notice also, is the these phone are getting address from the data
remote dhcp server.  But when you look at the setting on the phone it has
the correct vlan id 

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, June 23, 2006 11:05 PM
To: Ted Nugent
Cc: Wydra, Jason; James Grace; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch

Another fun thing is if there is a rogue VLAN sitting in the vlan
database...

Do a show vlan and see if there are VLANs on the switch that don't
make sense to you... if so, go into VLAN database and delete them...

Also, note the fact that you CANNOT configure a 3550, 3560 or 3570 to
have a voice vlan and configured as a trunk port... the voice vlan is
ignored...



Jonathan

On 6/23/06, Ted Nugent <tednugent69 at yahoo.com> wrote:
> Another Problem I've seen that could be contributing
> to this is if you set the access vlan and voice vlan
> the same. (I think it's a bug but have not
> investigated it).
> sw acces vlan 4
> sw voice vlan 4
> Changing to a different access vlan typically fixes
> this. Also mokeying with the DHCP properties have
> caused the same problem and simply changing the access
> vlan to to something else and changing it back
> resolves the problem. You may need to down the sw port
> to get this to take.
>
>
> --- "Wydra, Jason" <jwydra at Burwood.com> wrote:
>
> > Are you doing true trunking (encap dot1q) on these
> > ports or are you
> > doing an access vlan and voice vlan?
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Jonathan
> > Charles
> > Sent: Friday, June 23, 2006 9:39 PM
> > To: James Grace
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] 7940 7960 wont grab ip
> > from dhcp off switch
> >
> > Do they show up in a 'show cdp neighbors'? If not,
> > you may want to
> > enable CDP so the phones can see the voice vlan.
> >
> >
> > Jonathan
> >
> > On 6/23/06, James Grace <jgrace at digitelusa.net>
> > wrote:
> > >
> > >
> > >
> > >
> > > We have a 3750 with dhcp scope config on it.  We
> > have all type of
> > phone and
> > > wireless ap (cisco) that are pulling from this
> > scope.  But for some
> > reason
> > > the 7940s and 7960s are grabing ip from the data
> > remote dhcp server.
> > The
> > > switch is not recognizing these phones
> > >
> > >
> > >
> > > James D. Grace
> > >
> > > CCNP CCNA MCSE MCDBA
> > >
> > > Sr. System Engineer / Professional Svc.
> > >
> > > Digitel Corporation
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >
> > _______________________________________________
> > 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
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
---



---



From commo_ssg_31f at yahoo.com  Sat Jun 24 12:04:59 2006
From: commo_ssg_31f at yahoo.com (ash AD)
Date: Sat, 24 Jun 2006 09:04:59 -0700 (PDT)
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
In-Reply-To: <200606241133187.SM01452@3729434770>
Message-ID: <20060624160459.25313.qmail@web50402.mail.yahoo.com>

If you are dot1q trunking to the phone, try setting the native VLAN on the trunk to the phone that you would like it to DHCP from. This worked to fix bugs on older 3524XL switches. I am sure you have tried this but, have you erased the phone configurationson the phones (option 33)? Good luck.
   
  Peter

James Grace <jgrace at digitelusa.net> wrote:
  What I notice also, is the these phone are getting address from the data
remote dhcp server. But when you look at the setting on the phone it has
the correct vlan id 

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, June 23, 2006 11:05 PM
To: Ted Nugent
Cc: Wydra, Jason; James Grace; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch

Another fun thing is if there is a rogue VLAN sitting in the vlan
database...

Do a show vlan and see if there are VLANs on the switch that don't
make sense to you... if so, go into VLAN database and delete them...

Also, note the fact that you CANNOT configure a 3550, 3560 or 3570 to
have a voice vlan and configured as a trunk port... the voice vlan is
ignored...



Jonathan

On 6/23/06, Ted Nugent wrote:
> Another Problem I've seen that could be contributing
> to this is if you set the access vlan and voice vlan
> the same. (I think it's a bug but have not
> investigated it).
> sw acces vlan 4
> sw voice vlan 4
> Changing to a different access vlan typically fixes
> this. Also mokeying with the DHCP properties have
> caused the same problem and simply changing the access
> vlan to to something else and changing it back
> resolves the problem. You may need to down the sw port
> to get this to take.
>
>
> --- "Wydra, Jason" wrote:
>
> > Are you doing true trunking (encap dot1q) on these
> > ports or are you
> > doing an access vlan and voice vlan?
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Jonathan
> > Charles
> > Sent: Friday, June 23, 2006 9:39 PM
> > To: James Grace
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] 7940 7960 wont grab ip
> > from dhcp off switch
> >
> > Do they show up in a 'show cdp neighbors'? If not,
> > you may want to
> > enable CDP so the phones can see the voice vlan.
> >
> >
> > Jonathan
> >
> > On 6/23/06, James Grace 
> > wrote:
> > >
> > >
> > >
> > >
> > > We have a 3750 with dhcp scope config on it. We
> > have all type of
> > phone and
> > > wireless ap (cisco) that are pulling from this
> > scope. But for some
> > reason
> > > the 7940s and 7960s are grabing ip from the data
> > remote dhcp server.
> > The
> > > switch is not recognizing these phones
> > >
> > >
> > >
> > > James D. Grace
> > >
> > > CCNP CCNA MCSE MCDBA
> > >
> > > Sr. System Engineer / Professional Svc.
> > >
> > > Digitel Corporation
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >
> > _______________________________________________
> > 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
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
---



---


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


				
---------------------------------
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From zthoran at yahoo.com  Sat Jun 24 12:08:19 2006
From: zthoran at yahoo.com (Zevier Thoran)
Date: Sat, 24 Jun 2006 09:08:19 -0700 (PDT)
Subject: [cisco-voip] IP Phones Serial Numbers?
Message-ID: <20060624160820.11599.qmail@web37106.mail.mud.yahoo.com>

Hi,
I want to grab the serial numbers of all the Cisco IP Phones on my network. Does anyone know of a tool that can do this job?
Thanks
-Zevier

 __________________________________________________
Do You Yahoo!?
Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 
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From tednugent69 at yahoo.com  Sat Jun 24 12:09:25 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sat, 24 Jun 2006 09:09:25 -0700 (PDT)
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
In-Reply-To: <200606241133187.SM01452@3729434770>
Message-ID: <20060624160925.26750.qmail@web36106.mail.mud.yahoo.com>

Can you provide a switchport config from one of the
phone ports off the 3750 so we can get a better idea
of what we're looking at? Also, can you check if there
is a helper address on whatever is providing routing
for that VLAN?

Another weird problem I ran into was that most newer
servers these days have the ability to trunk on their
NICs which can cause issues for DHCP if the
circumstance are right. I?m having a hard time
thinking of an example right now but I would check if
the sw port for this DHCP server is trunking and if so
disable it or remove the unwanted VLAN from the trunk.


--- James Grace <jgrace at digitelusa.net> wrote:

> What I notice also, is the these phone are getting
> address from the data
> remote dhcp server.  But when you look at the
> setting on the phone it has
> the correct vlan id 
> 
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com] 
> Sent: Friday, June 23, 2006 11:05 PM
> To: Ted Nugent
> Cc: Wydra, Jason; James Grace;
> cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] 7940 7960 wont grab ip
> from dhcp off switch
> 
> Another fun thing is if there is a rogue VLAN
> sitting in the vlan
> database...
> 
> Do a show vlan and see if there are VLANs on the
> switch that don't
> make sense to you... if so, go into VLAN database
> and delete them...
> 
> Also, note the fact that you CANNOT configure a
> 3550, 3560 or 3570 to
> have a voice vlan and configured as a trunk port...
> the voice vlan is
> ignored...
> 
> 
> 
> Jonathan
> 
> On 6/23/06, Ted Nugent <tednugent69 at yahoo.com>
> wrote:
> > Another Problem I've seen that could be
> contributing
> > to this is if you set the access vlan and voice
> vlan
> > the same. (I think it's a bug but have not
> > investigated it).
> > sw acces vlan 4
> > sw voice vlan 4
> > Changing to a different access vlan typically
> fixes
> > this. Also mokeying with the DHCP properties have
> > caused the same problem and simply changing the
> access
> > vlan to to something else and changing it back
> > resolves the problem. You may need to down the sw
> port
> > to get this to take.
> >
> >
> > --- "Wydra, Jason" <jwydra at Burwood.com> wrote:
> >
> > > Are you doing true trunking (encap dot1q) on
> these
> > > ports or are you
> > > doing an access vlan and voice vlan?
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On
> > > Behalf Of Jonathan
> > > Charles
> > > Sent: Friday, June 23, 2006 9:39 PM
> > > To: James Grace
> > > Cc: cisco-voip at puck.nether.net
> > > Subject: Re: [cisco-voip] 7940 7960 wont grab ip
> > > from dhcp off switch
> > >
> > > Do they show up in a 'show cdp neighbors'? If
> not,
> > > you may want to
> > > enable CDP so the phones can see the voice vlan.
> > >
> > >
> > > Jonathan
> > >
> > > On 6/23/06, James Grace <jgrace at digitelusa.net>
> > > wrote:
> > > >
> > > >
> > > >
> > > >
> > > > We have a 3750 with dhcp scope config on it. 
> We
> > > have all type of
> > > phone and
> > > > wireless ap (cisco) that are pulling from this
> > > scope.  But for some
> > > reason
> > > > the 7940s and 7960s are grabing ip from the
> data
> > > remote dhcp server.
> > > The
> > > > switch is not recognizing these phones
> > > >
> > > >
> > > >
> > > > James D. Grace
> > > >
> > > > CCNP CCNA MCSE MCDBA
> > > >
> > > > Sr. System Engineer / Professional Svc.
> > > >
> > > > Digitel Corporation
> > > >
> > > >
> > > >
> _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> > > >
> > > _______________________________________________
> > > 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
> > >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around
> > http://mail.yahoo.com
> >
> ---
> 
> 
> 
> ---
> 
> 
> 


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

From tednugent69 at yahoo.com  Sat Jun 24 12:13:00 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sat, 24 Jun 2006 09:13:00 -0700 (PDT)
Subject: [cisco-voip] IP Phones Serial Numbers?
In-Reply-To: <20060624160820.11599.qmail@web37106.mail.mud.yahoo.com>
Message-ID: <20060624161300.86759.qmail@web36105.mail.mud.yahoo.com>

I just saw this the otherday, I've not tried it but
this has been a question I've heard dozens of time and
if it works it will be a life saver for many customers

http://www.cmreports.com/shop/index.php?main_page=index&cPath=5&zenid=ccc78358dfc71c255c7d9adf5b796708




--- Zevier Thoran <zthoran at yahoo.com> wrote:

> Hi,
> I want to grab the serial numbers of all the Cisco
> IP Phones on my network. Does anyone know of a tool
> that can do this job?
> Thanks
> -Zevier
> 
>  __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com >
_______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jsteinberg at gmail.com  Sat Jun 24 12:13:38 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Sat, 24 Jun 2006 12:13:38 -0400
Subject: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
In-Reply-To: <200606241133187.SM01452@3729434770>
References: <5d093f9a0606232005k2df0286du322cd9059c5039ca@mail.gmail.com>
	<200606241133187.SM01452@3729434770>
Message-ID: <f1b65f880606240913n33c93741n6360028c0ba4f367@mail.gmail.com>

James,

Make sure you remove the switchport voice vlan command from the
interface that connects to your data vlan DHCP server.

The server handling your data vlan DHCP service could be responding to
the DHCP request faster than the 3750.

Justin



On 6/24/06, James Grace <jgrace at digitelusa.net> wrote:
> What I notice also, is the these phone are getting address from the data
> remote dhcp server.  But when you look at the setting on the phone it has
> the correct vlan id
>
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Friday, June 23, 2006 11:05 PM
> To: Ted Nugent
> Cc: Wydra, Jason; James Grace; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] 7940 7960 wont grab ip from dhcp off switch
>
> Another fun thing is if there is a rogue VLAN sitting in the vlan
> database...
>
> Do a show vlan and see if there are VLANs on the switch that don't
> make sense to you... if so, go into VLAN database and delete them...
>
> Also, note the fact that you CANNOT configure a 3550, 3560 or 3570 to
> have a voice vlan and configured as a trunk port... the voice vlan is
> ignored...
>
>
>
> Jonathan
>
> On 6/23/06, Ted Nugent <tednugent69 at yahoo.com> wrote:
> > Another Problem I've seen that could be contributing
> > to this is if you set the access vlan and voice vlan
> > the same. (I think it's a bug but have not
> > investigated it).
> > sw acces vlan 4
> > sw voice vlan 4
> > Changing to a different access vlan typically fixes
> > this. Also mokeying with the DHCP properties have
> > caused the same problem and simply changing the access
> > vlan to to something else and changing it back
> > resolves the problem. You may need to down the sw port
> > to get this to take.
> >
> >
> > --- "Wydra, Jason" <jwydra at Burwood.com> wrote:
> >
> > > Are you doing true trunking (encap dot1q) on these
> > > ports or are you
> > > doing an access vlan and voice vlan?
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On
> > > Behalf Of Jonathan
> > > Charles
> > > Sent: Friday, June 23, 2006 9:39 PM
> > > To: James Grace
> > > Cc: cisco-voip at puck.nether.net
> > > Subject: Re: [cisco-voip] 7940 7960 wont grab ip
> > > from dhcp off switch
> > >
> > > Do they show up in a 'show cdp neighbors'? If not,
> > > you may want to
> > > enable CDP so the phones can see the voice vlan.
> > >
> > >
> > > Jonathan
> > >
> > > On 6/23/06, James Grace <jgrace at digitelusa.net>
> > > wrote:
> > > >
> > > >
> > > >
> > > >
> > > > We have a 3750 with dhcp scope config on it.  We
> > > have all type of
> > > phone and
> > > > wireless ap (cisco) that are pulling from this
> > > scope.  But for some
> > > reason
> > > > the 7940s and 7960s are grabing ip from the data
> > > remote dhcp server.
> > > The
> > > > switch is not recognizing these phones
> > > >
> > > >
> > > >
> > > > James D. Grace
> > > >
> > > > CCNP CCNA MCSE MCDBA
> > > >
> > > > Sr. System Engineer / Professional Svc.
> > > >
> > > > Digitel Corporation
> > > >
> > > >
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > >
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> > > >
> > > _______________________________________________
> > > 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
> > >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
> >
> ---
>
>
>
> ---
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From bababooey at cox.net  Sat Jun 24 12:28:20 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Sat, 24 Jun 2006 12:28:20 -0400
Subject: [cisco-voip] IP Communicator headset
References: <mailman.3.1151164806.86094.cisco-voip@puck.nether.net>
Message-ID: <000301c697ab$352978f0$0f0110ac@corporation2>

What are the best usb headsets for IP Communicator?
I am looking for quality and price in my comparisons.

Thanks,
Bob



From kthorngr at cisco.com  Sat Jun 24 13:55:43 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Sat, 24 Jun 2006 13:55:43 -0400
Subject: [cisco-voip] Operator setup
In-Reply-To: <001101c6979d$a6cb0360$0f0110ac@corporation2>
References: <mailman.10674.1151090674.1478.cisco-voip@puck.nether.net>
	<000601c6974e$41973670$0f0110ac@corporation2>
	<815b27f415b809aff53b582de9d8b807@cisco.com>
	<001101c6979d$a6cb0360$0f0110ac@corporation2>
Message-ID: <257bf7991606a9485efacdc3bf1ecbda@cisco.com>

The CSS will need to contain the Partition(s) that you want the numbers 
translated to.  In your case it sounds like you will use the one CSS 
for all phones.

Kevin
On Jun 24, 2006, at 10:51 AM, Bob A. Bowie wrote:

> Kevin, thanks for the answers, I am not sure what needs to be set in 
> the CSS appropratiely,
> I currently have one CSS for all phones, will i need to adjust this?
>
> Thanks
> Bob
>
>
> ----- Original Message ----- From: "Kevin Thorngren" 
> <kthorngr at cisco.com>
> To: "Bob A. Bowie" <bababooey at cox.net>
> Cc: <cisco-voip at puck.nether.net>
> Sent: Saturday, June 24, 2006 9:18 AM
> Subject: Re: [cisco-voip] Operator setup
>
>
>> You can use Translation Patterns to solve both of these issues.  
>> Create a Translation Pattern for 0 and translate the called number to 
>> the 7 digit number.  Make sure to set the CSS appropriately.
>>
>> For the second you can create two Translation Patterns; one pattern 
>> of 100[2-9] and the other of 1010.  Translate the called number to 
>> 1001 and set the CSS.
>>
>> Kevin
>> On Jun 24, 2006, at 1:22 AM, Bob A. Bowie wrote:
>>
>>> In a call manager system I have my operator attached through a pbx, 
>>> can I
>>> make it so users can just dial "0"
>>> and convert this to a seven digit number to dial thru to the pbx?
>>>
>>> Also, if I have a range of extensions from 1001 - 1010, I would like 
>>> anyone
>>> who dial 1002-1010 to be directed to 1001?
>>> Is this possbile in CM 4.1.3?  All extensions should have full dial 
>>> out
>>> capability
>>>
>>> Thanks
>>> Bob
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>


From tednugent69 at yahoo.com  Sat Jun 24 14:04:12 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sat, 24 Jun 2006 11:04:12 -0700 (PDT)
Subject: [cisco-voip] IP Communicator headset
In-Reply-To: <000301c697ab$352978f0$0f0110ac@corporation2>
Message-ID: <20060624180412.70631.qmail@web36114.mail.mud.yahoo.com>

IMHO Plantronics makes the best headsets for this use.
On thier site they have a link to "Find my headset"
which will help direct you and then it's just a mater
of $, providing they have built in DSPs they should
all get the job done.

http://www.plantronics.com/north_america/en_US/index.jhtml

And another good site for purchasing and searching is
http://www.headsetzone.com/index.html

and from Cisco's docs

For your reference, other headset and handset vendors
include: 

? Actiontec - www.actiontec.com 

? Clarisys - www.clarisys.net 

? CommuniTech - www.communitech.com 

? Danacom - www.danacomusa.com 

? Eutectics - www.eutecticsinc.com 

? GNNetcom - www.gnnetcom.com 

? Plantronics - www.plantronics.com 

? Technology Link Corporation (for Japan only) -
www.link-corp.co.jp 

? VXI - www.vxicorp.com 



--- "Bob A. Bowie" <bababooey at cox.net> wrote:

> What are the best usb headsets for IP Communicator?
> I am looking for quality and price in my
> comparisons.
> 
> Thanks,
> Bob
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jonvoip at gmail.com  Sat Jun 24 14:58:56 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 24 Jun 2006 13:58:56 -0500
Subject: [cisco-voip] Cisco Desktop Administrator, errors galore...
Message-ID: <5d093f9a0606241158w49959f1fs589d24cbeccf1cf4@mail.gmail.com>

So, I have IPCC 3.5(3) installed, configuring it now...

I can log into appadmin and the JTAPI and ICD subsystems are in
service... So, I am definitely connecting to the directory...

When I open Cisco Desktop Administrator (to configure agents and
stuff), I get the following error:

"Disconnected from primary Directory Service, will try to connect to
backup services. This process may take time and further update
attempts may fail. Continue with the reconneciton?"

I hit yes... and eventually, I get this:

"Failed to bind to LDAP server. Can't contact LDAP server."

The MIVR log shows the following:

883: Jun 24 11:54:02.656 CDT %MIVR-SS_ENT_SRV-3-ERROR_LOAD_LDAP:Fail
to read LDAP INFO: Exception=Exception caught:
javax.naming.CommunicationException: 10.0.0.112:8404 [Root exception
is java.net.ConnectException: Operation timed out: connect]

Which would be fine, but the DC Directory isn't at 10.0.0.112... I
have it by host name and it resolves to 10.0.0.121 (where it actually
is).

So, I find in the registry, this:

HKLM\SOFTWARE\SPANLINK\Site Setup

I change LDAP_HOSTA the IP to the right one... but the password is
wrong and now I get:

877: Jun 24 12:07:42.171 CDT %MIVR-SS_ENT_SRV-3-ERROR_LOAD_LDAP:Fail
to read LDAP INFO: Exception=Exception caught:
javax.naming.AuthenticationException: [LDAP: error code 49 - Invalid
credentials]

The password is in the same registry location in LDAP_PWD,
encrypted... So, I copy the encrypted password from the ccndir.ini and
paste it in there... and no luck, same error, invalid credentials...

So, I reboot.

No change.

I delete the repository and reset it up, no change...

How do I fix this? It looks like appadmin (CRA itself) can connect to
DC Directory fine, but Desktop Admin is using other stuff and can't...
How do I reinitialize this?



Jonathan

From lelio at uoguelph.ca  Sat Jun 24 15:47:35 2006
From: lelio at uoguelph.ca (lelio at uoguelph.ca)
Date: Sat, 24 Jun 2006 15:47:35 -0400
Subject: [cisco-voip] CIPTUG Registration Open
Message-ID: <1151178455.449d96d77d3e0@webmail.uoguelph.ca>

Just a brief reminder to those on the list that annual CIPTUG meeting is coming
up soon. You have to be a CIPTUG member to join but even with the membership
fee and the conference fee together, I think it's a good deal. Take a look at
www.ciptug.org for more details.

A few of us on the list got together, albeit briefly, early on in the
conference, which made it even better. Of course Scott, et al, might disagree
that meeting me face to face could be categorized as 'better'. ;)

I also believe that they are continuing the practice that if you present a
session, your conference fees are waived.

Whether you're a beginner or a seasoned veteran, it's an excellent conference to
attend.

Lelio



From bcarter at sentinel.com  Sat Jun 24 16:23:49 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Sat, 24 Jun 2006 15:23:49 -0500
Subject: [cisco-voip] Unified Operations Manager 1.1 - Anyone
	installedit yet?
Message-ID: <6A0744639B86DD459433108DBF101D26FC6648@MAIL01.sentinel.com>

I will be installing one the end of August.  I will keep the list
posted.


________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
	Sent: Saturday, June 24, 2006 1:17 AM
	To: Stu Packett
	Cc: cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Unified Operations Manager 1.1 -
Anyone installedit yet?
	
	
	I'm interested also :)
	
	
	On 6/22/06, Stu Packett <SPackett at fenwick.com> wrote: 

		I just got the CDs for Unified Operations Manager 1.1,
but have not yet installed it as I have to decommission and rebuild
another server for it.  Has anyone installed it yet?  What kind of load
does it put on the server?

		Thanks in advance. 


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




	-- 
	Ed Leatherman
	IP Telephony Coordinator
	West Virginia University
	Telecommunications and Network Operations 



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.
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From CORTIZ at broward.org  Sat Jun 24 20:35:37 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Sat, 24 Jun 2006 20:35:37 -0400
Subject: [cisco-voip] Stable Call Mgr Version
References: <26B357EF9FB7D04A989C9E82A9E5436E01BC3537@MREXVS01.bc.broward.cty>
	<5d093f9a0606231945o2dbc624v568aeba727bc3c8f@mail.gmail.com>
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E03048E0D@MREXVS01.bc.broward.cty>

I need to look again, but I think Ted explained that the Golden Bridge DOC is really just a subset of the compat matrix. In any case if you are running 4.1.3 sr3b and IPCC 3.5(2) sr3 then great....that seems like a good combo for me as it gets me to a CM version I want and doesn't affect the IPCC install that much (just a new sr for me).  Any glaring issues when you upgraded to these versions?  
 
Carlos

________________________________

From: Jonathan Charles [mailto:jonvoip at gmail.com]
Sent: Fri 6/23/2006 10:45 PM
To: Ortiz, Carlos
Cc: Ted Nugent; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stable Call Mgr Version



Are you looking at this:

http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1225/ccmigration_09186a00803d82f5.pdf

Cuz, I am running 4.1(3)sr3b with IPCC 3.5(2)sr3 and page 48 of that
compatibility guide shows 3.5(3)sr3 being supported in 4.1(3)sr3a (and
I have it running with 4.1(3)sr3b (which came out last week).

In fact, that shows 3.5(4) supporting CCM4.2... So, I am not sure what
you are looking at...



Jonathan

On 6/23/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
> So according to this matrix CM 4.1.3 SR1 is supported with IPC 3.5.3 but
> if you apply SR2 to CM then you have to upgrade IPCC to 4.x....does that
> make any sense?
>
>
>
> -----Original Message-----
> From: Ted Nugent [mailto:tednugent69 at yahoo.com]
> Sent: Friday, June 23, 2006 10:40 AM
> To: Ortiz, Carlos; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Stable Call Mgr Version
>
> Well you're running on probably the most stable
> version of CM that 4.x has to offer... Other then that
> I would probably run with CM4.1(3)SR2... I've seen
> some issues with SR3 and thier still working them
> out... sr2b just came out the other day. I've only
> deployed Unity 4.1 a few times and each went very
> smoothly with very few followups.. (but 4.0(5) wasn't
> a bad release either... unless there are features
> you're looking for I would stick with 4.0(5))
>
> I personally would not touch CM/Unity 4.2 yet...
> they're both fairly new... I would probably give them
> a couple month.
>
> IPCC 3.5.3 SR3 seemed to be pretty stable as well...
> You could not pay me enough to support IPCCX 4.x
> yet... they are getting closer with newer releases but
> right now.. no way.. no how!
> You can also reference "golden bridge" at the link
> below.
>
> http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/ipc
> mtrix.htm
>
>
> --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
>
> > Can anyone suggest a stable CM version and IPCC
> > Express that they have
> > successfully running?
> >
> >
> >
> > I am currently running 4.02a(sr2b) with IPCC 3.5.2
> > and Unity 4.05.
> >
> >
> >
> > I would like to move to Call Mgr version 4.1.3 or
> > 4.2 (if recommended).
> > I also realize that IPCC Express will need to be
> > upgraded as part of
> > this.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Carlos
> >
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


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From jonvoip at gmail.com  Sat Jun 24 22:46:10 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 24 Jun 2006 21:46:10 -0500
Subject: [cisco-voip] Unified Operations Manager 1.1 - Anyone
	installedit yet?
In-Reply-To: <6A0744639B86DD459433108DBF101D26FC6648@MAIL01.sentinel.com>
References: <AcaXVe5qoElKHcE+RRu84GIgbH2U7gAdhyJg>
	<6A0744639B86DD459433108DBF101D26FC6648@MAIL01.sentinel.com>
Message-ID: <5d093f9a0606241946k7baabf62mb46d933f5d7d5adf@mail.gmail.com>

OK, this is going to sound dumb, but what is it?



Jonathan

On 6/24/06, Carter, Bill <bcarter at sentinel.com> wrote:
>
>
> I will be installing one the end of August.  I will keep the list posted.
>
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed
> Leatherman
> Sent: Saturday, June 24, 2006 1:17 AM
> To: Stu Packett
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Unified Operations Manager 1.1 - Anyone
> installedit yet?
>
>
> I'm interested also :)
>
>
> On 6/22/06, Stu Packett <SPackett at fenwick.com> wrote:
> >
> >
> >
> >
> > I just got the CDs for Unified Operations Manager 1.1, but have not yet
> installed it as I have to decommission and rebuild another server for it.
> Has anyone installed it yet?  What kind of load does it put on the server?
> >
> > Thanks in advance.
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>
> 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
>
>
>

From jonvoip at gmail.com  Sat Jun 24 23:37:30 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 24 Jun 2006 22:37:30 -0500
Subject: [cisco-voip] Cisco Desktop Administrator, errors galore...
In-Reply-To: <5d093f9a0606241158w49959f1fs589d24cbeccf1cf4@mail.gmail.com>
References: <5d093f9a0606241158w49959f1fs589d24cbeccf1cf4@mail.gmail.com>
Message-ID: <5d093f9a0606242037x21eb5364r89cd92729c398db9@mail.gmail.com>

OK, an update...

I checked the DirectoryConfiguration.ini file on the CCM, and the
encrypted password matches the Spanlink registry settings on the IPCC.

Somewhere in the IPCC, an incorrect password is being stored... but where?



Jonathan

On 6/24/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> So, I have IPCC 3.5(3) installed, configuring it now...
>
> I can log into appadmin and the JTAPI and ICD subsystems are in
> service... So, I am definitely connecting to the directory...
>
> When I open Cisco Desktop Administrator (to configure agents and
> stuff), I get the following error:
>
> "Disconnected from primary Directory Service, will try to connect to
> backup services. This process may take time and further update
> attempts may fail. Continue with the reconneciton?"
>
> I hit yes... and eventually, I get this:
>
> "Failed to bind to LDAP server. Can't contact LDAP server."
>
> The MIVR log shows the following:
>
> 883: Jun 24 11:54:02.656 CDT %MIVR-SS_ENT_SRV-3-ERROR_LOAD_LDAP:Fail
> to read LDAP INFO: Exception=Exception caught:
> javax.naming.CommunicationException: 10.0.0.112:8404 [Root exception
> is java.net.ConnectException: Operation timed out: connect]
>
> Which would be fine, but the DC Directory isn't at 10.0.0.112... I
> have it by host name and it resolves to 10.0.0.121 (where it actually
> is).
>
> So, I find in the registry, this:
>
> HKLM\SOFTWARE\SPANLINK\Site Setup
>
> I change LDAP_HOSTA the IP to the right one... but the password is
> wrong and now I get:
>
> 877: Jun 24 12:07:42.171 CDT %MIVR-SS_ENT_SRV-3-ERROR_LOAD_LDAP:Fail
> to read LDAP INFO: Exception=Exception caught:
> javax.naming.AuthenticationException: [LDAP: error code 49 - Invalid
> credentials]
>
> The password is in the same registry location in LDAP_PWD,
> encrypted... So, I copy the encrypted password from the ccndir.ini and
> paste it in there... and no luck, same error, invalid credentials...
>
> So, I reboot.
>
> No change.
>
> I delete the repository and reset it up, no change...
>
> How do I fix this? It looks like appadmin (CRA itself) can connect to
> DC Directory fine, but Desktop Admin is using other stuff and can't...
> How do I reinitialize this?
>
>
>
> Jonathan
>

From ealeatherman at gmail.com  Sun Jun 25 00:13:27 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Sun, 25 Jun 2006 00:13:27 -0400
Subject: [cisco-voip] Unified Operations Manager 1.1 - Anyone
	installedit yet?
In-Reply-To: <5d093f9a0606241946k7baabf62mb46d933f5d7d5adf@mail.gmail.com>
References: <6A0744639B86DD459433108DBF101D26FC6648@MAIL01.sentinel.com>
	<5d093f9a0606241946k7baabf62mb46d933f5d7d5adf@mail.gmail.com>
Message-ID: <94a1afde0606242113x519a0f64v2f33c5a2fb3bd67c@mail.gmail.com>

It's a hopefully better replacement for Ciscoworks ITEM

http://www.cisco.com/en/US/customer/products/ps6535/index.html

On 6/24/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> OK, this is going to sound dumb, but what is it?
>
>
>
> Jonathan
>
> On 6/24/06, Carter, Bill <bcarter at sentinel.com> wrote:
> >
> >
> > I will be installing one the end of August.  I will keep the list
> posted.
> >
> >
> >  ________________________________
> >  From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed
> > Leatherman
> > Sent: Saturday, June 24, 2006 1:17 AM
> > To: Stu Packett
> > Cc: cisco-voip at puck.nether.net
> > Subject: Re: [cisco-voip] Unified Operations Manager 1.1 - Anyone
> > installedit yet?
> >
> >
> > I'm interested also :)
> >
> >
> > On 6/22/06, Stu Packett <SPackett at fenwick.com> wrote:
> > >
> > >
> > >
> > >
> > > I just got the CDs for Unified Operations Manager 1.1, but have not
> yet
> > installed it as I have to decommission and rebuild another server for
> it.
> > Has anyone installed it yet?  What kind of load does it put on the
> server?
> > >
> > > Thanks in advance.
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >
> >
> >
> >
> > --
> > Ed Leatherman
> > IP Telephony Coordinator
> > West Virginia University
> > Telecommunications and Network Operations
> >
> > 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
> >
> >
> >
>



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From jonvoip at gmail.com  Sun Jun 25 00:19:17 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sat, 24 Jun 2006 23:19:17 -0500
Subject: [cisco-voip] Unified Operations Manager 1.1 - Anyone
	installedit yet?
In-Reply-To: <94a1afde0606242113x519a0f64v2f33c5a2fb3bd67c@mail.gmail.com>
References: <6A0744639B86DD459433108DBF101D26FC6648@MAIL01.sentinel.com>
	<5d093f9a0606241946k7baabf62mb46d933f5d7d5adf@mail.gmail.com>
	<94a1afde0606242113x519a0f64v2f33c5a2fb3bd67c@mail.gmail.com>
Message-ID: <5d093f9a0606242119n42933c1v47f4a8677d86472b@mail.gmail.com>

sounds kinda cool... Better than that stupid syslog server we are using...


Jonathan

On 6/24/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
> It's a hopefully better replacement for Ciscoworks ITEM
>
> http://www.cisco.com/en/US/customer/products/ps6535/index.html
>
>
> On 6/24/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > OK, this is going to sound dumb, but what is it?
> >
> >
> >
> > Jonathan
> >
> > On 6/24/06, Carter, Bill <bcarter at sentinel.com> wrote:
> > >
> > >
> > > I will be installing one the end of August.  I will keep the list
> posted.
> > >
> > >
> > >  ________________________________
> > >  From: cisco-voip-bounces at puck.nether.net
> > > [mailto: cisco-voip-bounces at puck.nether.net] On Behalf
> Of Ed
> > > Leatherman
> > > Sent: Saturday, June 24, 2006 1:17 AM
> > > To: Stu Packett
> > > Cc: cisco-voip at puck.nether.net
> > > Subject: Re: [cisco-voip] Unified Operations Manager 1.1 - Anyone
> > > installedit yet?
> > >
> > >
> > > I'm interested also :)
> > >
> > >
> > > On 6/22/06, Stu Packett < SPackett at fenwick.com> wrote:
> > > >
> > > >
> > > >
> > > >
> > > > I just got the CDs for Unified Operations Manager 1.1, but have not
> yet
> > > installed it as I have to decommission and rebuild another server for
> it.
> > > Has anyone installed it yet?  What kind of load does it put on the
> server?
> > > >
> > > > Thanks in advance.
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> > > >
> > >
> > >
> > >
> > > --
> > > Ed Leatherman
> > > IP Telephony Coordinator
> > > West Virginia University
> > > Telecommunications and Network Operations
> > >
> > > 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
> > >
> > >
> > >
> >
>
>
>
> --
>
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations

From lelio at uoguelph.ca  Sun Jun 25 00:47:57 2006
From: lelio at uoguelph.ca (lelio at uoguelph.ca)
Date: Sun, 25 Jun 2006 00:47:57 -0400
Subject: [cisco-voip] Unified Operations Manager 1.1 -
	Anyone	installedit yet?
In-Reply-To: <94a1afde0606242113x519a0f64v2f33c5a2fb3bd67c@mail.gmail.com>
References: <6A0744639B86DD459433108DBF101D26FC6648@MAIL01.sentinel.com>
	<5d093f9a0606241946k7baabf62mb46d933f5d7d5adf@mail.gmail.com>
	<94a1afde0606242113x519a0f64v2f33c5a2fb3bd67c@mail.gmail.com>
Message-ID: <1151210877.449e157db1039@webmail.uoguelph.ca>

you mean we'll actually get alerts less than 5 days later? ;)

Quoting Ed Leatherman <ealeatherman at gmail.com>:

> It's a hopefully better replacement for Ciscoworks ITEM
>
> http://www.cisco.com/en/US/customer/products/ps6535/index.html
>
> On 6/24/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > OK, this is going to sound dumb, but what is it?
> >
> >
> >
> > Jonathan
> >
> > On 6/24/06, Carter, Bill <bcarter at sentinel.com> wrote:
> > >
> > >
> > > I will be installing one the end of August.  I will keep the list
> > posted.
> > >
> > >
> > >  ________________________________
> > >  From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed
> > > Leatherman
> > > Sent: Saturday, June 24, 2006 1:17 AM
> > > To: Stu Packett
> > > Cc: cisco-voip at puck.nether.net
> > > Subject: Re: [cisco-voip] Unified Operations Manager 1.1 - Anyone
> > > installedit yet?
> > >
> > >
> > > I'm interested also :)
> > >
> > >
> > > On 6/22/06, Stu Packett <SPackett at fenwick.com> wrote:
> > > >
> > > >
> > > >
> > > >
> > > > I just got the CDs for Unified Operations Manager 1.1, but have not
> > yet
> > > installed it as I have to decommission and rebuild another server for
> > it.
> > > Has anyone installed it yet?  What kind of load does it put on the
> > server?
> > > >
> > > > Thanks in advance.
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> > > >
> > >
> > >
> > >
> > > --
> > > Ed Leatherman
> > > IP Telephony Coordinator
> > > West Virginia University
> > > Telecommunications and Network Operations
> > >
> > > 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
> > >
> > >
> > >
> >
>
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>



From lelio at uoguelph.ca  Sun Jun 25 01:09:23 2006
From: lelio at uoguelph.ca (lelio at uoguelph.ca)
Date: Sun, 25 Jun 2006 01:09:23 -0400
Subject: [cisco-voip] Background Logo
In-Reply-To: <OF9971A336.487933A5-ON86257195.007AF095-86257195.007BC857@tca.us.intdk>
References: <OF9971A336.487933A5-ON86257195.007AF095-86257195.007BC857@tca.us.intdk>
Message-ID: <1151212162.449e1a8303f8b@webmail.uoguelph.ca>

I believe this is ultimately true for the 79[46]0, but the 7912 handles screen
images quite well (considering what you're working with I guess).

If you search for 7912 logo on the Cisco forums, you'll get tonnes of hits with
updated and corrected instructions. It took me a few times to get it right, but
eventually I did.

Something to look out for is that I found that the image you create is actually
a reverse of what you send, so if you would want an completely filled out
screen, your original image would be blank. I end up creating the image,
reversing it, and then using the reverse to create the screen logo.

The other thing is, is that the index number is what is used to trigger the
download of a new logo to the phone.

I think those were the only two biggies. Oh, and of course, unless you have some
graphics art background and a good graphics program, don't be to upset with what
appears on the phone screen. A pixel on the phone is much bigger than what is on
your screen and hence, WYSIWYG(A) does not really apply. ;)

I ended up enlarging the image on my screen to the size it would be on the
phone, and that gave me a great idea of what it looked like. I had to hold a
ruler to the screen at the time to make sure I was getting the right size, but
in the end it all worked out.



Quoting pdavidson at tdktca.com:

> Danny-
>
> There are only a few ways to do this- and unfortunately, under CCM4.X, I'm
> not happy with any of them.
>
> The by the book answer is to use the Idle Screen functionality.  You'll see
> a setting for the URL to set for idle, as well as the idle timeout timer,
> on the Device Settings page under CCM Administrator.  The URL must return a
> valid CMXML object, containing the information you want to display.  This
> functionality works- I run it regularly.
>
> However- under CCM4, the definition of 'idle' is a little bit funky.  I've
> had it kick in while I'm dialing before, replacing what I'm dialing and
> stopping me from dialing more digits.  You also almost lose the onhook
> dialahead functionality- you have to exit from the idle screen first, then
> onhook dial- a retraining step for the users.
>
> I have seen this done with custom authored phone loads, which involve BTXML
> authoring- as a university, you might be able to get some help from Cisco
> to do this.  I know that the 'managed' IP-PBX offering from Covad does this
> on 7960's- I've seen them running- but that's using an MGCP load talking to
> something other than Callmanager.
>
> If you run the SIP image on the phone, it's MUCH simpler- just tell the
> phone the name of a .bmp file on your TFTP server, and you're done- with no
> loss of functionality.  But, CCM4 doesn't make SIP phone administration
> simple- it will work, but it will be a very manual process for you.  CCM5
> changes that, is my understanding- but the CCM5 SIP Phone load has other
> oddities- your mileage may vary.
>
> Hope this helps.
>
> Paul Davidson, Manager, MIS
> TDK Corporation of America
> http://www.component.tdk.com
>
>
>
>
>              cisco-voip-reques
>              t at puck.nether.net
>              Sent by:                                                   To
>              cisco-voip-bounce         cisco-voip at puck.nether.net
>              s at puck.nether.net                                          cc
>
>                                                                    Subject
>              06/22/2006 05:14          cisco-voip Digest, Vol 40, Issue
>              PM                        119
>
>
>              Please respond to
>              cisco-voip at puck.n
>                  ether.net
>
>
>
>
>
>
>
> Date: Thu, 22 Jun 2006 18:16:41 -0400
> From: "Phillips, Danny" <dphillips at sapc.edu>
> Subject: [cisco-voip] Background Logo
> To: <cisco-voip at puck.nether.net>
> Message-ID: <18F0B5EFCCE1A449B07A3A9DF61DC07308ACE3 at xchange.sapc.edu>
> Content-Type: text/plain; charset="us-ascii"
>
> We want to add our college's logo to the 7912, 7940, and 7960 IP phones
> running CM 4.1(3), could someone point me in the right direction.
>
>
>
>
>
> Thanks
>
> Danny,
>
>
>
>
>
>
>
>
>
>
>
> `````````````````````````````````````````````````
>
> Danny Phillips
>
> Network Administrator
>
> St. Andrews Presbyterian College
>
> 910-277-5079
>
> dphillips at sapc.edu
>
>
>
> `````````````````````````````````````````````````
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



From jonvoip at gmail.com  Sun Jun 25 04:53:44 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 25 Jun 2006 03:53:44 -0500
Subject: [cisco-voip] 7914s not displaying line state on shared lines
Message-ID: <5d093f9a0606250153r3c85153t96e455dd86f080f4@mail.gmail.com>

We have CallManager 4.1(3)sr3b with OS4.2sr8...

Receptionist has a 7971 with two 7914 expansion modules.

Created shared lines of the executives on the 7914s.

We see them ring when you call them, but we don't see any line state
info on the 7914s (it always shows the phones on hook, no matter what
they are doing, unless they are ringing).

We are running the latest DevPack, version 28-2 and the 7914s have
picked up the latest load.

What am I missing?



Jonathan

From aman.chugh at gmail.com  Sun Jun 25 04:58:15 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Sun, 25 Jun 2006 14:28:15 +0530
Subject: [cisco-voip] 7914s not displaying line state on shared lines
In-Reply-To: <5d093f9a0606250153r3c85153t96e455dd86f080f4@mail.gmail.com>
References: <5d093f9a0606250153r3c85153t96e455dd86f080f4@mail.gmail.com>
Message-ID: <d4c697ad0606250158i110c9c09y48a49955e561e44f@mail.gmail.com>

What is privacy set to .check the privacy settings

-- Aman


On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> We have CallManager 4.1(3)sr3b with OS4.2sr8...
>
> Receptionist has a 7971 with two 7914 expansion modules.
>
> Created shared lines of the executives on the 7914s.
>
> We see them ring when you call them, but we don't see any line state
> info on the 7914s (it always shows the phones on hook, no matter what
> they are doing, unless they are ringing).
>
> We are running the latest DevPack, version 28-2 and the 7914s have
> picked up the latest load.
>
> What am I missing?
>
>
>
> Jonathan
> _______________________________________________
> 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  Sun Jun 25 05:37:00 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 25 Jun 2006 04:37:00 -0500
Subject: [cisco-voip] CallManager 4.2 - Anyone running it in production?
Message-ID: <5d093f9a0606250237o1a952d63q91c62deaaf6163fc@mail.gmail.com>

I am running 4.2(1)sr1 in a lab, and I like some of the features with
it... Anyone playing with it in the real world yet? Any major issues,
any 'oh my god, why oh why did I put this in production?!?!?!?'

Just curious, cuz that CFNC would be pretty cool for my 7920s and that
directed Call Park thing is also pretty cool.





Jonathan

From Scott.Voll at wesd.org  Sun Jun 25 17:18:06 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Sun, 25 Jun 2006 14:18:06 -0700
Subject: [cisco-voip] Operator setup
References: <mailman.10674.1151090674.1478.cisco-voip@puck.nether.net>
	<000601c6974e$41973670$0f0110ac@corporation2>
Message-ID: <407055A92CECCB499C922A2D35FC19A605D75A1F@Apollo.wesd.org>

simple translation pattern should work for this.
 
Scott

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Bob A. Bowie
Sent: Fri 6/23/2006 10:22 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Operator setup



In a call manager system I have my operator attached through a pbx, can I
make it so users can just dial "0"
and convert this to a seven digit number to dial thru to the pbx?

Also, if I have a range of extensions from 1001 - 1010, I would like anyone
who dial 1002-1010 to be directed to 1001?
Is this possbile in CM 4.1.3?  All extensions should have full dial out
capability

Thanks
Bob


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


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From tednugent69 at yahoo.com  Sun Jun 25 18:21:28 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sun, 25 Jun 2006 15:21:28 -0700 (PDT)
Subject: [cisco-voip] IPCC Upgrade issues to 4.0??
Message-ID: <20060625222128.80642.qmail@web36109.mail.mud.yahoo.com>

I searched through my emails but I can't seem to find
the post where someone was having issues upgrading to
IPCC 4.0 but it was failing with the backup piece? I
remember they said that they were trying to by pass
this since they didn't have much on the server anyway?
Well I just had the same issue and i was able to
bypass all that nonsense by just uninstalling the old
version of IPCC and reinstalling 4.0. Seems pretty
simple but at first glance you might think it's like
CM where you can't uninstall. Anyway... if this helps
good... if not send it to the bit bucket

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

From jonvoip at gmail.com  Sun Jun 25 18:52:06 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 25 Jun 2006 17:52:06 -0500
Subject: [cisco-voip] IPCC Upgrade issues to 4.0??
In-Reply-To: <20060625222128.80642.qmail@web36109.mail.mud.yahoo.com>
References: <20060625222128.80642.qmail@web36109.mail.mud.yahoo.com>
Message-ID: <5d093f9a0606251552i2f291109md7a424de44f2bfea@mail.gmail.com>

That was me....

However, given the hostile response I received from this group, we
have decided to stay at 3.5(3)sr3 for the time being.




Jonathan

On 6/25/06, Ted Nugent <tednugent69 at yahoo.com> wrote:
> I searched through my emails but I can't seem to find
> the post where someone was having issues upgrading to
> IPCC 4.0 but it was failing with the backup piece? I
> remember they said that they were trying to by pass
> this since they didn't have much on the server anyway?
> Well I just had the same issue and i was able to
> bypass all that nonsense by just uninstalling the old
> version of IPCC and reinstalling 4.0. Seems pretty
> simple but at first glance you might think it's like
> CM where you can't uninstall. Anyway... if this helps
> good... if not send it to the bit bucket
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From tednugent69 at yahoo.com  Sun Jun 25 19:01:10 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Sun, 25 Jun 2006 16:01:10 -0700 (PDT)
Subject: [cisco-voip] IPCC Upgrade issues to 4.0??
In-Reply-To: <5d093f9a0606251552i2f291109md7a424de44f2bfea@mail.gmail.com>
Message-ID: <20060625230110.20707.qmail@web36101.mail.mud.yahoo.com>

OK... I thought it was you but could not find the
post... I just set this up in the lab for studying
purposes. I honestly don't think it's production ready
yet.


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

> That was me....
> 
> However, given the hostile response I received from
> this group, we
> have decided to stay at 3.5(3)sr3 for the time
> being.
> 
> 
> 
> 
> Jonathan
> 
> On 6/25/06, Ted Nugent <tednugent69 at yahoo.com>
> wrote:
> > I searched through my emails but I can't seem to
> find
> > the post where someone was having issues upgrading
> to
> > IPCC 4.0 but it was failing with the backup piece?
> I
> > remember they said that they were trying to by
> pass
> > this since they didn't have much on the server
> anyway?
> > Well I just had the same issue and i was able to
> > bypass all that nonsense by just uninstalling the
> old
> > version of IPCC and reinstalling 4.0. Seems pretty
> > simple but at first glance you might think it's
> like
> > CM where you can't uninstall. Anyway... if this
> helps
> > good... if not send it to the bit bucket
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> protection around
> > http://mail.yahoo.com
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> 


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

From jonvoip at gmail.com  Sun Jun 25 19:11:27 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Sun, 25 Jun 2006 18:11:27 -0500
Subject: [cisco-voip] IPCC Upgrade issues to 4.0??
In-Reply-To: <20060625230110.20707.qmail@web36101.mail.mud.yahoo.com>
References: <5d093f9a0606251552i2f291109md7a424de44f2bfea@mail.gmail.com>
	<20060625230110.20707.qmail@web36101.mail.mud.yahoo.com>
Message-ID: <5d093f9a0606251611u20d12f4bsdbb7c1f0c6f28a83@mail.gmail.com>

Well, after I saw the posts here, I have to agree...



Jonathan

On 6/25/06, Ted Nugent <tednugent69 at yahoo.com> wrote:
> OK... I thought it was you but could not find the
> post... I just set this up in the lab for studying
> purposes. I honestly don't think it's production ready
> yet.
>
>
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
>
> > That was me....
> >
> > However, given the hostile response I received from
> > this group, we
> > have decided to stay at 3.5(3)sr3 for the time
> > being.
> >
> >
> >
> >
> > Jonathan
> >
> > On 6/25/06, Ted Nugent <tednugent69 at yahoo.com>
> > wrote:
> > > I searched through my emails but I can't seem to
> > find
> > > the post where someone was having issues upgrading
> > to
> > > IPCC 4.0 but it was failing with the backup piece?
> > I
> > > remember they said that they were trying to by
> > pass
> > > this since they didn't have much on the server
> > anyway?
> > > Well I just had the same issue and i was able to
> > > bypass all that nonsense by just uninstalling the
> > old
> > > version of IPCC and reinstalling 4.0. Seems pretty
> > > simple but at first glance you might think it's
> > like
> > > CM where you can't uninstall. Anyway... if this
> > helps
> > > good... if not send it to the bit bucket
> > >
> > > __________________________________________________
> > > Do You Yahoo!?
> > > Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> > > http://mail.yahoo.com
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > >
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>

From Nick.Kassel at Charles-Stanley.co.uk  Mon Jun 26 08:52:39 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 26 Jun 2006 13:52:39 +0100
Subject: [cisco-voip] MGCP on CCM 4.2
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648445@CSEXCH01.charles-stanley.co.uk>

I'm having a little trouble getting a MGCP gateway working with CCM 4.2,
does anyone know if I have to use a minimum IOS, I currently trying with
12.3 (11)

I'm unable to download the configuration.

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From jonvoip at gmail.com  Mon Jun 26 09:26:07 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 26 Jun 2006 08:26:07 -0500
Subject: [cisco-voip] IPCC Not answering calls
Message-ID: <5d093f9a0606260626t19b151adhfa7649dfe79fa5f1@mail.gmail.com>

I dial in to the CTI RP @ 1501, I get silence for 5 seconds, and then
a busy signal.

Getting this error in the MIVR log:

1149: Jun 26 08:25:07.687 CDT %MIVR-SS_TEL-5-CTIPG_ROUTE_EVENT:CTI
Port Group route event: Route Session=[1501/(P1-rmjtapi)
GCID=(3,62)->INVALID]->DISCONNECTED,Route Address=1501,Failure
reason=CTI accept timer expires after 4016 ms, end route
connection,Exception=java.lang.InterruptedException

Now, if I remove, the Primary Dialog Group (Cisco Media Group #0) from
the JTAPI trigger, I see the phone transfer from 1501 to one of my CTI
ports and then disconnect... and I get:

1433: Jun 26 08:29:24.484 CDT
%MIVR-APP_MGR-6-ABORTING_CONTACT:Aborting contact:
Application=ICD,Task id=18,000,000,019,Contact id=24,Contact
implementation id=70/3,Contact
Class=com.cisco.call.CallContact,Contact Type=Cisco JTAPI
Call,Exception=com.cisco.channel.ChannelUnsupportedException:
com.cisco.dialog.PlayPromptDialog is not supported

1434: Jun 26 08:29:24.484 CDT
%MIVR-SS_TEL-5-AUDIO_TREATMENT_FAILED:Audio treatment failed: All Call
ids=CallID:24 MediaId:70/3 Task:18000000019,Audio
treatment=UNRECOV_ERROR,Exception=com.cisco.channel.ChannelUnsupportedException:
com.cisco.dialog.PlayPromptDialog is not supported


What am I missing?


Jonathan

From jonvoip at gmail.com  Mon Jun 26 09:26:57 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 26 Jun 2006 08:26:57 -0500
Subject: [cisco-voip] 7914s not displaying line state on shared lines
In-Reply-To: <d4c697ad0606250158i110c9c09y48a49955e561e44f@mail.gmail.com>
References: <5d093f9a0606250153r3c85153t96e455dd86f080f4@mail.gmail.com>
	<d4c697ad0606250158i110c9c09y48a49955e561e44f@mail.gmail.com>
Message-ID: <5d093f9a0606260626m75c575ecpa6acd94405816ebe@mail.gmail.com>

I checked and globally Privacy is set to false, also, I tried it on
some other phones and NONE of them show the shared line, line state...


Jonathan

On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
>
> What is privacy set to .check the privacy settings
>
> -- Aman
>
>
> On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> We have CallManager 4.1(3)sr3b with OS4.2sr8...
>
> Receptionist has a 7971 with two 7914 expansion modules.
>
> Created shared lines of the executives on the 7914s.
>
> We see them ring when you call them, but we don't see any line state
> info on the 7914s (it always shows the phones on hook, no matter what
> they are doing, unless they are ringing).
>
> We are running the latest DevPack, version 28-2 and the 7914s have
> picked up the latest load.
>
> What am I missing?
>
>
>
> Jonathan
> _______________________________________________
> 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  Mon Jun 26 09:36:11 2006
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Mon, 26 Jun 2006 09:36:11 -0400
Subject: [cisco-voip] Unified Operations Manager 1.1 - Anyone
	installe	dit yet?
Message-ID: <07A14D6987A5F647B2415D72EDF63B560597FB7E@tnsex01.tns.its.psu.edu>

Not sure what beefs others have with ITM but we are huge fans of it.  It
needs a lot of configuration but once we got it set up it has alerted us
promptly and accurately.  (At first it alerted us about everything and
anything, mostly uninteresting things.)  The tools for gathering info from
phones are not as useful, in my opinion, as the tools for alerting, but we
use the info-gathering tools at times too.

---
Bill Simon - bills at tns.its.psu.edu - (814) 865-2270
http://tns.its.psu.edu/ <http://tns.its.psu.edu/> 



 


  _____  

From: Ed Leatherman [mailto:ealeatherman at gmail.com] 
Sent: Sunday, June 25, 2006 12:13 AM
To: Jonathan Charles
Cc: cisco-voip at puck.nether.net; Stu Packett
Subject: Re: [cisco-voip] Unified Operations Manager 1.1 - Anyone
installedit yet?


It's a hopefully better replacement for Ciscoworks ITEM

http://www.cisco.com/en/US/customer/products/ps6535/index.html
<http://www.cisco.com/en/US/customer/products/ps6535/index.html> 


On 6/24/06, Jonathan Charles <jonvoip at gmail.com <mailto:jonvoip at gmail.com> >
wrote: 

OK, this is going to sound dumb, but what is it? 



Jonathan

On 6/24/06, Carter, Bill <bcarter at sentinel.com <mailto:bcarter at sentinel.com>
> wrote:
>
>
> I will be installing one the end of August.  I will keep the list posted. 
>
>
>  ________________________________
>  From: cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> 
> [mailto:  <mailto:cisco-voip-bounces at puck.nether.net>
cisco-voip-bounces at puck.nether.net] On Behalf Of Ed
> Leatherman
> Sent: Saturday, June 24, 2006 1:17 AM
> To: Stu Packett
> Cc: cisco-voip at puck.nether.net  <mailto:cisco-voip at puck.nether.net> 
> Subject: Re: [cisco-voip] Unified Operations Manager 1.1 - Anyone
> installedit yet?
>
>
> I'm interested also :)
>
>
> On 6/22/06, Stu Packett < SPackett at fenwick.com
<mailto:SPackett at fenwick.com> > wrote:
> >
> >
> >
> >
> > I just got the CDs for Unified Operations Manager 1.1, but have not yet
> installed it as I have to decommission and rebuild another server for it. 
> Has anyone installed it yet?  What kind of load does it put on the server?
> >
> > Thanks in advance.
> > _______________________________________________
> > cisco-voip mailing list 
> > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> 
> > https://puck.nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 
> >
> >
> >
>
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> West Virginia University
> Telecommunications and Network Operations
>
> 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  <mailto:cisco-voip at puck.nether.net> 
> https://puck.nether.net/mailman/listinfo/cisco-voip
<https://puck.nether.net/mailman/listinfo/cisco-voip> 
>
>
>





-- 
Ed Leatherman 
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations 

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From lelio at uoguelph.ca  Mon Jun 26 09:38:44 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 26 Jun 2006 09:38:44 -0400
Subject: [cisco-voip] 7914s not displaying line state on shared lines
References: <5d093f9a0606250153r3c85153t96e455dd86f080f4@mail.gmail.com><d4c697ad0606250158i110c9c09y48a49955e561e44f@mail.gmail.com>
	<5d093f9a0606260626m75c575ecpa6acd94405816ebe@mail.gmail.com>
Message-ID: <004601c69925$d8366140$47196883@cfs.uoguelph.ca>

I'm pretty sure that the Builtin Bridge Enable parameter has to be enabled, either in the system parameters or on the phone configuration itself. After any of these changes (privacy setting included) the phone has to be reset/restarted or something like that.

--------------------------------------------------------------------------------
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: Jonathan Charles 
  To: Aman Chugh 
  Cc: ciscovoip 
  Sent: Monday, June 26, 2006 9:26 AM
  Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines


  I checked and globally Privacy is set to false, also, I tried it on
  some other phones and NONE of them show the shared line, line state...


  Jonathan

  On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
  >
  > What is privacy set to .check the privacy settings
  >
  > -- Aman
  >
  >
  > On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
  > >
  > We have CallManager 4.1(3)sr3b with OS4.2sr8...
  >
  > Receptionist has a 7971 with two 7914 expansion modules.
  >
  > Created shared lines of the executives on the 7914s.
  >
  > We see them ring when you call them, but we don't see any line state
  > info on the 7914s (it always shows the phones on hook, no matter what
  > they are doing, unless they are ringing).
  >
  > We are running the latest DevPack, version 28-2 and the 7914s have
  > picked up the latest load.
  >
  > What am I missing?
  >
  >
  >
  > Jonathan
  > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  >
  >
  _______________________________________________
  cisco-voip mailing list
  cisco-voip at puck.nether.net
  https://puck.nether.net/mailman/listinfo/cisco-voip
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From jonvoip at gmail.com  Mon Jun 26 09:46:05 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 26 Jun 2006 08:46:05 -0500
Subject: [cisco-voip] 7914s not displaying line state on shared lines
In-Reply-To: <004601c69925$d8366140$47196883@cfs.uoguelph.ca>
References: <5d093f9a0606250153r3c85153t96e455dd86f080f4@mail.gmail.com>
	<d4c697ad0606250158i110c9c09y48a49955e561e44f@mail.gmail.com>
	<5d093f9a0606260626m75c575ecpa6acd94405816ebe@mail.gmail.com>
	<004601c69925$d8366140$47196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0606260646m1abf4cadla2692e9b50b8b820@mail.gmail.com>

The bridge is enabled globally and manually on both devices


Jonathan

On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
> I'm pretty sure that the Builtin Bridge Enable parameter has to be enabled,
> either in the system parameters or on the phone configuration itself. After
> any of these changes (privacy setting included) the phone has to be
> reset/restarted or something like that.
>
> --------------------------------------------------------------------------------
> 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: Jonathan Charles
> To: Aman Chugh
> Cc: ciscovoip
> Sent: Monday, June 26, 2006 9:26 AM
> Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
>
> I checked and globally Privacy is set to false, also, I tried it on
> some other phones and NONE of them show the shared line, line state...
>
>
> Jonathan
>
> On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> >
> > What is privacy set to .check the privacy settings
> >
> > -- Aman
> >
> >
> > On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > We have CallManager 4.1(3)sr3b with OS4.2sr8...
> >
> > Receptionist has a 7971 with two 7914 expansion modules.
> >
> > Created shared lines of the executives on the 7914s.
> >
> > We see them ring when you call them, but we don't see any line state
> > info on the 7914s (it always shows the phones on hook, no matter what
> > they are doing, unless they are ringing).
> >
> > We are running the latest DevPack, version 28-2 and the 7914s have
> > picked up the latest load.
> >
> > What am I missing?
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Mon Jun 26 09:54:02 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 26 Jun 2006 08:54:02 -0500
Subject: [cisco-voip] 7914s not displaying line state on shared lines
In-Reply-To: <007a01c69927$1ae9c4e0$47196883@cfs.uoguelph.ca>
References: <5d093f9a0606250153r3c85153t96e455dd86f080f4@mail.gmail.com>
	<d4c697ad0606250158i110c9c09y48a49955e561e44f@mail.gmail.com>
	<5d093f9a0606260626m75c575ecpa6acd94405816ebe@mail.gmail.com>
	<004601c69925$d8366140$47196883@cfs.uoguelph.ca>
	<5d093f9a0606260646m1abf4cadla2692e9b50b8b820@mail.gmail.com>
	<007a01c69927$1ae9c4e0$47196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0606260654v531abb9eua07fcc52bca334f3@mail.gmail.com>

Dozens of times at this point...

I am checking the traces now...



Jonathan

On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
> ..and you've restarted the phones?
>
> that's just bizzare.
>
> --------------------------------------------------------------------------------
> 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: Jonathan Charles
> To: Lelio Fulgenzi
> Cc: Aman Chugh ; ciscovoip
> Sent: Monday, June 26, 2006 9:46 AM
> Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
>
> The bridge is enabled globally and manually on both devices
>
>
> Jonathan
>
> On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> >
> > I'm pretty sure that the Builtin Bridge Enable parameter has to be
> enabled,
> > either in the system parameters or on the phone configuration itself.
> After
> > any of these changes (privacy setting included) the phone has to be
> > reset/restarted or something like that.
> >
> >
> --------------------------------------------------------------------------------
> > 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: Jonathan Charles
> > To: Aman Chugh
> > Cc: ciscovoip
> > Sent: Monday, June 26, 2006 9:26 AM
> > Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
> >
> > I checked and globally Privacy is set to false, also, I tried it on
> > some other phones and NONE of them show the shared line, line state...
> >
> >
> > Jonathan
> >
> > On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> > >
> > > What is privacy set to .check the privacy settings
> > >
> > > -- Aman
> > >
> > >
> > > On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > >
> > > We have CallManager 4.1(3)sr3b with OS4.2sr8...
> > >
> > > Receptionist has a 7971 with two 7914 expansion modules.
> > >
> > > Created shared lines of the executives on the 7914s.
> > >
> > > We see them ring when you call them, but we don't see any line state
> > > info on the 7914s (it always shows the phones on hook, no matter what
> > > they are doing, unless they are ringing).
> > >
> > > We are running the latest DevPack, version 28-2 and the 7914s have
> > > picked up the latest load.
> > >
> > > What am I missing?
> > >
> > >
> > >
> > > Jonathan
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>

From chris.ellington at nsi1.com  Mon Jun 26 10:04:14 2006
From: chris.ellington at nsi1.com (Chris Ellington)
Date: Mon, 26 Jun 2006 10:04:14 -0400
Subject: [cisco-voip] cisco-voip Digest, Vol 40, Issue 128
In-Reply-To: <mailman.11050.1151203572.1478.cisco-voip@puck.nether.net>
Message-ID: <C0C5619E.EC1E%chris.ellington@nsi1.com>


>    4. Re: Unified Operations Manager 1.1 - Anyone installedit yet?
>       (Carter, Bill)

> ------------------------------
> 
> Message: 4
> Date: Sat, 24 Jun 2006 15:23:49 -0500
> From: "Carter, Bill" <bcarter at sentinel.com>
> Subject: Re: [cisco-voip] Unified Operations Manager 1.1 - Anyone
> installedit yet?
> To: "Ed Leatherman" <ealeatherman at gmail.com>, "Stu Packett"
> <SPackett at fenwick.com>
> Cc: cisco-voip at puck.nether.net
> Message-ID:
> <6A0744639B86DD459433108DBF101D26FC6648 at MAIL01.sentinel.com>
> Content-Type: text/plain; charset="us-ascii"
> 
> I will be installing one the end of August.  I will keep the list
> posted.
> 
> 

I just completed a CUOM install and it's very very very nice.  Make sure you
have LOTS of RAM - 2GB is minimum, 4GB is better.  I've got it running on a
DL320 monitoring 1 cluster of 3 CCMs, 2 Unitys and 2 IPCC 4.0 servers as
well as a 25+ node WAN (and the associated LANs).


VERY VERY VERY nice, particularly for an operations staff to monitor the
network or for a small shop that is overworked already.

chris
-- 
Christopher S. Ellington
CCIE #6814
Network Solutions, Inc.
(317) 566 8897
Chris.Ellington at nsi1.com



From Nick.Kassel at Charles-Stanley.co.uk  Mon Jun 26 10:05:29 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 26 Jun 2006 15:05:29 +0100
Subject: [cisco-voip] MGCP on CCM 4.2
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648446@CSEXCH01.charles-stanley.co.uk>

The ISDN status shows the layer 1 and 2 as up and the MGCP shows as
registered on the gateway but I am unable to get it registered on the
CCM, can anyone help me out with this?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 1:53 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP on CCM 4.2

 

I'm having a little trouble getting a MGCP gateway working with CCM 4.2,
does anyone know if I have to use a minimum IOS, I currently trying with
12.3 (11)

I'm unable to download the configuration.

 


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Registered in England No. 1903304

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The information contained in this e-mail is strictly confidential, some or all
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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
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From jonvoip at gmail.com  Mon Jun 26 10:09:25 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 26 Jun 2006 09:09:25 -0500
Subject: [cisco-voip] 7914s not displaying line state on shared lines
In-Reply-To: <009601c69928$b99ea000$47196883@cfs.uoguelph.ca>
References: <5d093f9a0606250153r3c85153t96e455dd86f080f4@mail.gmail.com>
	<d4c697ad0606250158i110c9c09y48a49955e561e44f@mail.gmail.com>
	<5d093f9a0606260626m75c575ecpa6acd94405816ebe@mail.gmail.com>
	<004601c69925$d8366140$47196883@cfs.uoguelph.ca>
	<5d093f9a0606260646m1abf4cadla2692e9b50b8b820@mail.gmail.com>
	<007a01c69927$1ae9c4e0$47196883@cfs.uoguelph.ca>
	<5d093f9a0606260654v531abb9eua07fcc52bca334f3@mail.gmail.com>
	<009601c69928$b99ea000$47196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0606260709o7972a50aye09ec06fbfbd7743@mail.gmail.com>

The thing is, I have downloaded and installed the latest firmware and
have confirmed the 7914s are running it.

Here's the worst part... the 7971s don't even show correct line state
on shared lines... and they are running the latest code...

I don't get it... I am going to try a different firmware first...

Jonathan

On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
> I think this definately requires a TAC case.
>
> --------------------------------------------------------------------------------
> 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: Jonathan Charles
> To: Lelio Fulgenzi
> Cc: ciscovoip
> Sent: Monday, June 26, 2006 9:54 AM
> Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
>
> Dozens of times at this point...
>
> I am checking the traces now...
>
>
>
> Jonathan
>
> On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> >
> > ..and you've restarted the phones?
> >
> > that's just bizzare.
> >
> >
> --------------------------------------------------------------------------------
> > 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: Jonathan Charles
> > To: Lelio Fulgenzi
> > Cc: Aman Chugh ; ciscovoip
> > Sent: Monday, June 26, 2006 9:46 AM
> > Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
> >
> > The bridge is enabled globally and manually on both devices
> >
> >
> > Jonathan
> >
> > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > >
> > > I'm pretty sure that the Builtin Bridge Enable parameter has to be
> > enabled,
> > > either in the system parameters or on the phone configuration itself.
> > After
> > > any of these changes (privacy setting included) the phone has to be
> > > reset/restarted or something like that.
> > >
> > >
> >
> --------------------------------------------------------------------------------
> > > 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: Jonathan Charles
> > > To: Aman Chugh
> > > Cc: ciscovoip
> > > Sent: Monday, June 26, 2006 9:26 AM
> > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
> lines
> > >
> > > I checked and globally Privacy is set to false, also, I tried it on
> > > some other phones and NONE of them show the shared line, line state...
> > >
> > >
> > > Jonathan
> > >
> > > On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> > > >
> > > > What is privacy set to .check the privacy settings
> > > >
> > > > -- Aman
> > > >
> > > >
> > > > On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > > >
> > > > We have CallManager 4.1(3)sr3b with OS4.2sr8...
> > > >
> > > > Receptionist has a 7971 with two 7914 expansion modules.
> > > >
> > > > Created shared lines of the executives on the 7914s.
> > > >
> > > > We see them ring when you call them, but we don't see any line state
> > > > info on the 7914s (it always shows the phones on hook, no matter what
> > > > they are doing, unless they are ringing).
> > > >
> > > > We are running the latest DevPack, version 28-2 and the 7914s have
> > > > picked up the latest load.
> > > >
> > > > What am I missing?
> > > >
> > > >
> > > >
> > > > Jonathan
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
>
>

From jonvoip at gmail.com  Mon Jun 26 10:09:56 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 26 Jun 2006 09:09:56 -0500
Subject: [cisco-voip] IPCC Not answering calls
In-Reply-To: <20060626135802.7623.qmail@web36115.mail.mud.yahoo.com>
References: <5d093f9a0606260626t19b151adhfa7649dfe79fa5f1@mail.gmail.com>
	<20060626135802.7623.qmail@web36115.mail.mud.yahoo.com>
Message-ID: <5d093f9a0606260709w3f55802m7718a6bfa2d09519@mail.gmail.com>

Yep, first thing I checked...

On 6/26/06, Ted Nugent <tednugent69 at yahoo.com> wrote:
> Is Jtapi the same version on IPCC and CM?
>
> --- Jonathan Charles <jonvoip at gmail.com> wrote:
>
> > I dial in to the CTI RP @ 1501, I get silence for 5
> > seconds, and then
> > a busy signal.
> >
> > Getting this error in the MIVR log:
> >
> > 1149: Jun 26 08:25:07.687 CDT
> > %MIVR-SS_TEL-5-CTIPG_ROUTE_EVENT:CTI
> > Port Group route event: Route
> > Session=[1501/(P1-rmjtapi)
> > GCID=(3,62)->INVALID]->DISCONNECTED,Route
> > Address=1501,Failure
> > reason=CTI accept timer expires after 4016 ms, end
> > route
> > connection,Exception=java.lang.InterruptedException
> >
> > Now, if I remove, the Primary Dialog Group (Cisco
> > Media Group #0) from
> > the JTAPI trigger, I see the phone transfer from
> > 1501 to one of my CTI
> > ports and then disconnect... and I get:
> >
> > 1433: Jun 26 08:29:24.484 CDT
> > %MIVR-APP_MGR-6-ABORTING_CONTACT:Aborting contact:
> > Application=ICD,Task id=18,000,000,019,Contact
> > id=24,Contact
> > implementation id=70/3,Contact
> > Class=com.cisco.call.CallContact,Contact Type=Cisco
> > JTAPI
> >
> Call,Exception=com.cisco.channel.ChannelUnsupportedException:
> > com.cisco.dialog.PlayPromptDialog is not supported
> >
> > 1434: Jun 26 08:29:24.484 CDT
> > %MIVR-SS_TEL-5-AUDIO_TREATMENT_FAILED:Audio
> > treatment failed: All Call
> > ids=CallID:24 MediaId:70/3 Task:18000000019,Audio
> >
> treatment=UNRECOV_ERROR,Exception=com.cisco.channel.ChannelUnsupportedException:
> > com.cisco.dialog.PlayPromptDialog is not supported
> >
> >
> > What am I missing?
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>

From aman.chugh at gmail.com  Mon Jun 26 10:11:00 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Mon, 26 Jun 2006 09:11:00 -0500
Subject: [cisco-voip] MGCP on CCM 4.2
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC648446@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC648446@CSEXCH01.charles-stanley.co.uk>
Message-ID: <d4c697ad0606260711u2d4b2123m60d1a13fceb99c62@mail.gmail.com>

do a debug mgcp packet on the router and make sure the Domain Name of the
callmanager matches the hostname for the router.

On 6/26/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote:
>
>   The ISDN status shows the layer 1 and 2 as up and the MGCP shows as
> registered on the gateway but I am unable to get it registered on the CCM,
> can anyone help me out with this?
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Nick Kassel
> *Sent:* Monday, June 26, 2006 1:53 PM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] MGCP on CCM 4.2
>
>
>
> I'm having a little trouble getting a MGCP gateway working with CCM 4.2,
> does anyone know if I have to use a minimum IOS, I currently trying with
> 12.3 (11)
>
> I'm unable to download the configuration.
>
>
>
>
>
> ***********************************************************************************************
>
>
> 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.
>
> _______________________________________________
> 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  Mon Jun 26 10:17:55 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Mon, 26 Jun 2006 09:17:55 -0500
Subject: [cisco-voip] IPCC Not answering calls
In-Reply-To: <5d093f9a0606260709w3f55802m7718a6bfa2d09519@mail.gmail.com>
References: <5d093f9a0606260626t19b151adhfa7649dfe79fa5f1@mail.gmail.com>
	<20060626135802.7623.qmail@web36115.mail.mud.yahoo.com>
	<5d093f9a0606260709w3f55802m7718a6bfa2d09519@mail.gmail.com>
Message-ID: <d4c697ad0606260717q62b8191dhc026ff60b770e7b@mail.gmail.com>

Is the cti rp which is controlled by ipcc is that registered.Are the ipcc
subsystems in service.

On 6/26/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Yep, first thing I checked...
>
> On 6/26/06, Ted Nugent <tednugent69 at yahoo.com> wrote:
> > Is Jtapi the same version on IPCC and CM?
> >
> > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> >
> > > I dial in to the CTI RP @ 1501, I get silence for 5
> > > seconds, and then
> > > a busy signal.
> > >
> > > Getting this error in the MIVR log:
> > >
> > > 1149: Jun 26 08:25:07.687 CDT
> > > %MIVR-SS_TEL-5-CTIPG_ROUTE_EVENT:CTI
> > > Port Group route event: Route
> > > Session=[1501/(P1-rmjtapi)
> > > GCID=(3,62)->INVALID]->DISCONNECTED,Route
> > > Address=1501,Failure
> > > reason=CTI accept timer expires after 4016 ms, end
> > > route
> > > connection,Exception=java.lang.InterruptedException
> > >
> > > Now, if I remove, the Primary Dialog Group (Cisco
> > > Media Group #0) from
> > > the JTAPI trigger, I see the phone transfer from
> > > 1501 to one of my CTI
> > > ports and then disconnect... and I get:
> > >
> > > 1433: Jun 26 08:29:24.484 CDT
> > > %MIVR-APP_MGR-6-ABORTING_CONTACT:Aborting contact:
> > > Application=ICD,Task id=18,000,000,019,Contact
> > > id=24,Contact
> > > implementation id=70/3,Contact
> > > Class=com.cisco.call.CallContact,Contact Type=Cisco
> > > JTAPI
> > >
> > Call,Exception=com.cisco.channel.ChannelUnsupportedException:
> > > com.cisco.dialog.PlayPromptDialog is not supported
> > >
> > > 1434: Jun 26 08:29:24.484 CDT
> > > %MIVR-SS_TEL-5-AUDIO_TREATMENT_FAILED:Audio
> > > treatment failed: All Call
> > > ids=CallID:24 MediaId:70/3 Task:18000000019,Audio
> > >
> > treatment=UNRECOV_ERROR,Exception=
> com.cisco.channel.ChannelUnsupportedException:
> > > com.cisco.dialog.PlayPromptDialog is not supported
> > >
> > >
> > > What am I missing?
> > >
> > >
> > > Jonathan
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
> >
> _______________________________________________
> 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  Mon Jun 26 10:20:18 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 26 Jun 2006 15:20:18 +0100
Subject: [cisco-voip] MGCP on CCM 4.2
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648447@CSEXCH01.charles-stanley.co.uk>

The hostname has no periods and is also identical to the domain name on
the MGCP gateway configuration on CCM. I have also made sure that no IP
domain name has been configured, this is for BRI, does this make any
difference?

 

 

 

________________________________

From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Monday, June 26, 2006 3:16 PM
To: Nick Kassel
Subject: Re: [cisco-voip] MGCP on CCM 4.2

 

no periods in the hostname for the router with mgcp.

 



 

On 6/26/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote: 

The ISDN status shows the layer 1 and 2 as up and the MGCP shows as
registered on the gateway but I am unable to get it registered on the
CCM, can anyone help me out with this? 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 1:53 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP on CCM 4.2

 

I'm having a little trouble getting a MGCP gateway working with CCM 4.2,
does anyone know if I have to use a minimum IOS, I currently trying with
12.3 (11)

I'm unable to download the configuration.

 


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

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.


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



 

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

The information contained in this e-mail is strictly confidential, some or all
of which may be legally privileged. It is for the intended recipient only.
Access to this e-mail by any other person is prohibited. If you are not the
intended recipient, any use, disclosure, copying, printing, distribution of,
replying to or any action taken or omitted to be taken in reliance on this
e-mail, is prohibited and may be unlawful. Please contact the sender immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that there
may be a delay in processing the instructions received from e-mails after
Charles Stanley has received them. You are advised that urgent, time sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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

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

Charles Stanley Sutherlands and 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.
-------------- next part --------------
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From wcdixo at aurora.lib.il.us  Mon Jun 26 10:24:11 2006
From: wcdixo at aurora.lib.il.us (Dixon, Wayne)
Date: Mon, 26 Jun 2006 09:24:11 -0500
Subject: [cisco-voip] Basic VOIP Setup Needs
Message-ID: <6D8BEE91433474478D294A8867FFDC6EE3F819@WMAIN3.aurorapubliclibrary.org>

Hi all, 
 
I'm wondering what basics are needed for allowing IP Phones access on a
Cisco Switch network...   We are looking at implementing some IP Phones.

 
I'm assuming that a Dedicated VLAN for the IP Phones, use of Native VLAN
for non-tagged packets and a DHCP server are needed... but I'm not sure
what else is needed.  
 
If anybody has a good source for some basic information on basic VOIP
setup would be greatly appreciated.
 
Thanks
 
Wayne
 
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From jonvoip at gmail.com  Mon Jun 26 10:24:11 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 26 Jun 2006 09:24:11 -0500
Subject: [cisco-voip] IPCC Not answering calls
In-Reply-To: <d4c697ad0606260717q62b8191dhc026ff60b770e7b@mail.gmail.com>
References: <5d093f9a0606260626t19b151adhfa7649dfe79fa5f1@mail.gmail.com>
	<20060626135802.7623.qmail@web36115.mail.mud.yahoo.com>
	<5d093f9a0606260709w3f55802m7718a6bfa2d09519@mail.gmail.com>
	<d4c697ad0606260717q62b8191dhc026ff60b770e7b@mail.gmail.com>
Message-ID: <5d093f9a0606260724o50862592n300ad3709e1b7d63@mail.gmail.com>

All subsystems are in service and the CTI RP and CTI Ports are registered.

RM-CM Subsystem IN_SERVICE
CMT Subsystem IN_SERVICE
Core Reporting Subsystem IN_SERVICE
JTAPI Subsystem IN_SERVICE
HTTP Subsystem IN_SERVICE
Application Subsystem IN_SERVICE
Enterprise Server Data Subsystem IN_SERVICE

Database and email are out of service, but I am not using them...


Jonathan


On 6/26/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> Is the cti rp which is controlled by ipcc is that registered.Are the ipcc
> subsystems in service.
>
>
> On 6/26/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > Yep, first thing I checked...
> >
> > On 6/26/06, Ted Nugent <tednugent69 at yahoo.com > wrote:
> > > Is Jtapi the same version on IPCC and CM?
> > >
> > > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> > >
> > > > I dial in to the CTI RP @ 1501, I get silence for 5
> > > > seconds, and then
> > > > a busy signal.
> > > >
> > > > Getting this error in the MIVR log:
> > > >
> > > > 1149: Jun 26 08:25:07.687 CDT
> > > > %MIVR-SS_TEL-5-CTIPG_ROUTE_EVENT:CTI
> > > > Port Group route event: Route
> > > > Session=[1501/(P1-rmjtapi)
> > > > GCID=(3,62)->INVALID]->DISCONNECTED,Route
> > > > Address=1501,Failure
> > > > reason=CTI accept timer expires after 4016 ms, end
> > > > route
> > > > connection,Exception=java.lang.InterruptedException
> > > >
> > > > Now, if I remove, the Primary Dialog Group (Cisco
> > > > Media Group #0) from
> > > > the JTAPI trigger, I see the phone transfer from
> > > > 1501 to one of my CTI
> > > > ports and then disconnect... and I get:
> > > >
> > > > 1433: Jun 26 08:29:24.484 CDT
> > > > %MIVR-APP_MGR-6-ABORTING_CONTACT:Aborting contact:
> > > > Application=ICD,Task id=18,000,000,019,Contact
> > > > id=24,Contact
> > > > implementation id=70/3,Contact
> > > > Class=com.cisco.call.CallContact,Contact Type=Cisco
> > > > JTAPI
> > > >
> > >
> Call,Exception=com.cisco.channel.ChannelUnsupportedException
> :
> > > > com.cisco.dialog.PlayPromptDialog is not supported
> > > >
> > > > 1434: Jun 26 08:29:24.484 CDT
> > > > %MIVR-SS_TEL-5-AUDIO_TREATMENT_FAILED:Audio
> > > > treatment failed: All Call
> > > > ids=CallID:24 MediaId:70/3 Task:18000000019,Audio
> > > >
> > >
> treatment=UNRECOV_ERROR,Exception=com.cisco.channel.ChannelUnsupportedException:
> > > > com.cisco.dialog.PlayPromptDialog is not supported
> > > >
> > > >
> > > > What am I missing?
> > > >
> > > >
> > > > Jonathan
> > > > _______________________________________________
> > > > cisco-voip mailing list
> > > > cisco-voip at puck.nether.net
> > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > >
> > >
> > >
> > > __________________________________________________
> > > Do You Yahoo!?
> > > Tired of spam?  Yahoo! Mail has the best spam protection around
> > > http://mail.yahoo.com
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>

From cenders at homesbyavi.com  Mon Jun 26 10:32:06 2006
From: cenders at homesbyavi.com (Corbett Enders)
Date: Mon, 26 Jun 2006 08:32:06 -0600
Subject: [cisco-voip] CallManager 4.2 - Anyone running it in
References: <mailman.3.1151251203.9300.cisco-voip@puck.nether.net>
Message-ID: <24DFD2417491E547AC80AE140706C48503946455@EX001.homesbyavi.local>

We're upgrading our production environment tonight...  we don't have a
lab/dev environment to test in.  We need the "call forward unregistered"
feature...


From Scott.Voll at wesd.org  Mon Jun 26 10:39:25 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 26 Jun 2006 07:39:25 -0700
Subject: [cisco-voip] MGCP on CCM 4.2
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED5167@Apollo.wesd.org>

Do you have all the ccm-manager and mgcp commands?  Sounds like you
might be missing something.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 7:20 AM
To: Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2

 

The hostname has no periods and is also identical to the domain name on
the MGCP gateway configuration on CCM. I have also made sure that no IP
domain name has been configured, this is for BRI, does this make any
difference?

 

 

 

________________________________

From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Monday, June 26, 2006 3:16 PM
To: Nick Kassel
Subject: Re: [cisco-voip] MGCP on CCM 4.2

 

no periods in the hostname for the router with mgcp.

 



 

On 6/26/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote: 

The ISDN status shows the layer 1 and 2 as up and the MGCP shows as
registered on the gateway but I am unable to get it registered on the
CCM, can anyone help me out with this? 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 1:53 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP on CCM 4.2

 

I'm having a little trouble getting a MGCP gateway working with CCM 4.2,
does anyone know if I have to use a minimum IOS, I currently trying with
12.3 (11)

I'm unable to download the configuration.

 


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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
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The London International Financial Futures &
Options Exchange. 

This footnote also confirms that this email message has been swept by
McAfee
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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.
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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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_______________________________________________
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https://puck.nether.net/mailman/listinfo/cisco-voip




 

 


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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
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From Nick.Kassel at Charles-Stanley.co.uk  Mon Jun 26 10:43:57 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Mon, 26 Jun 2006 15:43:57 +0100
Subject: [cisco-voip] MGCP on CCM 4.2
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648448@CSEXCH01.charles-stanley.co.uk>

Yes, all the standard MGCP commands are in as the gateway states it's
registered when I do a sho ccm-manager, also the isdn status is up so
the config had been downloaded. Just unable to get it registered on the
CCM.  

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, June 26, 2006 3:39 PM
To: Nick Kassel; Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MGCP on CCM 4.2

 

Do you have all the ccm-manager and mgcp commands?  Sounds like you
might be missing something.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 7:20 AM
To: Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2

 

The hostname has no periods and is also identical to the domain name on
the MGCP gateway configuration on CCM. I have also made sure that no IP
domain name has been configured, this is for BRI, does this make any
difference?

 

 

 

________________________________

From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Monday, June 26, 2006 3:16 PM
To: Nick Kassel
Subject: Re: [cisco-voip] MGCP on CCM 4.2

 

no periods in the hostname for the router with mgcp.

 



 

On 6/26/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote: 

The ISDN status shows the layer 1 and 2 as up and the MGCP shows as
registered on the gateway but I am unable to get it registered on the
CCM, can anyone help me out with this? 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 1:53 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP on CCM 4.2

 

I'm having a little trouble getting a MGCP gateway working with CCM 4.2,
does anyone know if I have to use a minimum IOS, I currently trying with
12.3 (11)

I'm unable to download the configuration.

 


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



 

 


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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
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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
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From Scott.Voll at wesd.org  Mon Jun 26 10:49:57 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 26 Jun 2006 07:49:57 -0700
Subject: [cisco-voip] Basic VOIP Setup Needs
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED516C@Apollo.wesd.org>

  

What kind of Switch?

 

Do you already have CM and VGW's setup?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dixon, Wayne
Sent: Monday, June 26, 2006 7:24 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Basic VOIP Setup Needs

 

Hi all, 

 

I'm wondering what basics are needed for allowing IP Phones access on a
Cisco Switch network...   We are looking at implementing some IP Phones.


 

I'm assuming that a Dedicated VLAN for the IP Phones, use of Native VLAN
for non-tagged packets and a DHCP server are needed... but I'm not sure
what else is needed.  

 

If anybody has a good source for some basic information on basic VOIP
setup would be greatly appreciated.

 

Thanks

 

Wayne

 

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From Joe.Pollere at us.didata.com  Mon Jun 26 10:59:26 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Mon, 26 Jun 2006 10:59:26 -0400
Subject: [cisco-voip] MGCP on CCM 4.2
Message-ID: <C65033A817B5734392E55CB9DF446104058DB55D@USNAEXCH.na.didata.local>

Do you have the GW Hostname in your HOSTS file on the CCM server, could
be a DNS issue.

 

Joe

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 10:44 AM
To: Voll, Scott; Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2

 

Yes, all the standard MGCP commands are in as the gateway states it's
registered when I do a sho ccm-manager, also the isdn status is up so
the config had been downloaded. Just unable to get it registered on the
CCM.  

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, June 26, 2006 3:39 PM
To: Nick Kassel; Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MGCP on CCM 4.2

 

Do you have all the ccm-manager and mgcp commands?  Sounds like you
might be missing something.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 7:20 AM
To: Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2

 

The hostname has no periods and is also identical to the domain name on
the MGCP gateway configuration on CCM. I have also made sure that no IP
domain name has been configured, this is for BRI, does this make any
difference?

 

 

 

________________________________

From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Monday, June 26, 2006 3:16 PM
To: Nick Kassel
Subject: Re: [cisco-voip] MGCP on CCM 4.2

 

no periods in the hostname for the router with mgcp.

 



 

On 6/26/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote: 

The ISDN status shows the layer 1 and 2 as up and the MGCP shows as
registered on the gateway but I am unable to get it registered on the
CCM, can anyone help me out with this? 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 1:53 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP on CCM 4.2

 

I'm having a little trouble getting a MGCP gateway working with CCM 4.2,
does anyone know if I have to use a minimum IOS, I currently trying with
12.3 (11)

I'm unable to download the configuration.

 


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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
McAfee
VirusScan and SurfControl Email Filter software.

 

 


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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
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VirusScan and SurfControl Email Filter software.


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

 

 


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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.

 

 


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The information contained in this e-mail is strictly confidential, some
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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 ealeatherman at gmail.com  Mon Jun 26 11:12:24 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 26 Jun 2006 11:12:24 -0400
Subject: [cisco-voip] Basic VOIP Setup Needs
In-Reply-To: <6D8BEE91433474478D294A8867FFDC6EE3F819@WMAIN3.aurorapubliclibrary.org>
References: <6D8BEE91433474478D294A8867FFDC6EE3F819@WMAIN3.aurorapubliclibrary.org>
Message-ID: <94a1afde0606260812u71e1639cnce07f2eaf1c77cb0@mail.gmail.com>

Wayne,

a good resource would be the SRND for callmanager,
www.cisco.com/go/srnd
Its under the Unified Communications section, just pick the one that matches
your callmanager version. Lots of good info in there.

Ed

On 6/26/06, Dixon, Wayne <wcdixo at aurora.lib.il.us> wrote:
>
>  Hi all,
>
> I'm wondering what basics are needed for allowing IP Phones access on a
> Cisco Switch network...   We are looking at implementing some IP Phones.
>
> I'm assuming that a Dedicated VLAN for the IP Phones, use of Native VLAN
> for non-tagged packets and a DHCP server are needed... but I'm not sure what
> else is needed.
>
> If anybody has a good source for some basic information on basic VOIP
> setup would be greatly appreciated.
>
> Thanks
>
>  Wayne
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From wcdixo at aurora.lib.il.us  Mon Jun 26 11:19:33 2006
From: wcdixo at aurora.lib.il.us (Dixon, Wayne)
Date: Mon, 26 Jun 2006 10:19:33 -0500
Subject: [cisco-voip] Basic VOIP Setup Needs
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED516C@Apollo.wesd.org>
Message-ID: <6D8BEE91433474478D294A8867FFDC6EE3F81A@WMAIN3.aurorapubliclibrary.org>

The Switches are 3500XLs... It is an Avaya system, and call managing
will be handled by that and forgive my ignorance, but what is A VGW?
Voice Gateway?
 
If it is a Voice Gateway, then we're still awaiting that to come in, we
just ordered that last week.. but I was asked to see what would be
needed to get the network ready for IP Phones
 
Wayne
 

	-----Original Message-----
	From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
	Sent: Monday, June 26, 2006 9:50 AM
	To: Dixon, Wayne; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] Basic VOIP Setup Needs
	
	
	  

	What kind of Switch?

	 

	Do you already have CM and VGW's setup?

	 

	Scott

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dixon, Wayne
	Sent: Monday, June 26, 2006 7:24 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Basic VOIP Setup Needs

	 

	Hi all, 

	 

	I'm wondering what basics are needed for allowing IP Phones
access on a Cisco Switch network...   We are looking at implementing
some IP Phones.  

	 

	I'm assuming that a Dedicated VLAN for the IP Phones, use of
Native VLAN for non-tagged packets and a DHCP server are needed... but
I'm not sure what else is needed.  

	 

	If anybody has a good source for some basic information on basic
VOIP setup would be greatly appreciated.

	 

	Thanks

	 

	Wayne

	 

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From jonvoip at gmail.com  Mon Jun 26 11:20:06 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 26 Jun 2006 10:20:06 -0500
Subject: [cisco-voip] IPCC Not answering calls
In-Reply-To: <B498EEDDD483A3498AE8E9ED5E80D3120A062D@hulk.ns1.brad-v.com>
References: <5d093f9a0606260724o50862592n300ad3709e1b7d63@mail.gmail.com>
	<B498EEDDD483A3498AE8E9ED5E80D3120A062D@hulk.ns1.brad-v.com>
Message-ID: <5d093f9a0606260820i647c13fand1aae972597ad7f2@mail.gmail.com>

Fixed it.

The JTAPI and ICD subsystems were set to login to the CCM with the
same user... I changed them to different users, registred JTAPI to the
CTI stuff, and ICD to the agents.

Of course, now I am having DC Directory replication problems... so,
item 461 is fixed, items 462 through 1131 need to be addressed...




Jonathan

On 6/26/06, Brad Vonarx <brad.vonarx at voipintegration.com> wrote:
> Jonathan,  I would check the call manager application event log for MTP
> errors, check to see if it lists out of MTP resources.
>
> You may be trying to transcode from another codec to G711 on the IPCC
> server.
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Monday, June 26, 2006 7:24 AM
> To: Aman Chugh
> Cc: ciscovoip; Ted Nugent
> Subject: Re: [cisco-voip] IPCC Not answering calls
>
> All subsystems are in service and the CTI RP and CTI Ports are
> registered.
>
> RM-CM Subsystem IN_SERVICE
> CMT Subsystem IN_SERVICE
> Core Reporting Subsystem IN_SERVICE
> JTAPI Subsystem IN_SERVICE
> HTTP Subsystem IN_SERVICE
> Application Subsystem IN_SERVICE
> Enterprise Server Data Subsystem IN_SERVICE
>
> Database and email are out of service, but I am not using them...
>
>
> Jonathan
>
>
> On 6/26/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> > Is the cti rp which is controlled by ipcc is that registered.Are the
> > ipcc subsystems in service.
> >
> >
> > On 6/26/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > Yep, first thing I checked...
> > >
> > > On 6/26/06, Ted Nugent <tednugent69 at yahoo.com > wrote:
> > > > Is Jtapi the same version on IPCC and CM?
> > > >
> > > > --- Jonathan Charles <jonvoip at gmail.com> wrote:
> > > >
> > > > > I dial in to the CTI RP @ 1501, I get silence for 5 seconds, and
>
> > > > > then a busy signal.
> > > > >
> > > > > Getting this error in the MIVR log:
> > > > >
> > > > > 1149: Jun 26 08:25:07.687 CDT
> > > > > %MIVR-SS_TEL-5-CTIPG_ROUTE_EVENT:CTI
> > > > > Port Group route event: Route
> > > > > Session=[1501/(P1-rmjtapi)
> > > > > GCID=(3,62)->INVALID]->DISCONNECTED,Route
> > > > > Address=1501,Failure
> > > > > reason=CTI accept timer expires after 4016 ms, end route
> > > > > connection,Exception=java.lang.InterruptedException
> > > > >
> > > > > Now, if I remove, the Primary Dialog Group (Cisco Media Group
> > > > > #0) from the JTAPI trigger, I see the phone transfer from
> > > > > 1501 to one of my CTI
> > > > > ports and then disconnect... and I get:
> > > > >
> > > > > 1433: Jun 26 08:29:24.484 CDT
> > > > > %MIVR-APP_MGR-6-ABORTING_CONTACT:Aborting contact:
> > > > > Application=ICD,Task id=18,000,000,019,Contact id=24,Contact
> > > > > implementation id=70/3,Contact
> > > > > Class=com.cisco.call.CallContact,Contact Type=Cisco JTAPI
> > > > >
> > > >
> > Call,Exception=com.cisco.channel.ChannelUnsupportedException
> > :
> > > > > com.cisco.dialog.PlayPromptDialog is not supported
> > > > >
> > > > > 1434: Jun 26 08:29:24.484 CDT
> > > > > %MIVR-SS_TEL-5-AUDIO_TREATMENT_FAILED:Audio
> > > > > treatment failed: All Call
> > > > > ids=CallID:24 MediaId:70/3 Task:18000000019,Audio
> > > > >
> > > >
> >
> treatment=UNRECOV_ERROR,Exception=com.cisco.channel.ChannelUnsupportedEx
> ception:
> > > > > com.cisco.dialog.PlayPromptDialog is not supported
> > > > >
> > > > >
> > > > > What am I missing?
> > > > >
> > > > >
> > > > > Jonathan
> > > > > _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > >
> > > >
> > > > __________________________________________________
> > > > Do You Yahoo!?
> > > > Tired of spam?  Yahoo! Mail has the best spam protection around
> > > > http://mail.yahoo.com
> > > >
> > > _______________________________________________
> > > 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 Scott.Voll at wesd.org  Mon Jun 26 11:26:15 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 26 Jun 2006 08:26:15 -0700
Subject: [cisco-voip] Basic VOIP Setup Needs
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED518E@Apollo.wesd.org>

  

Wayne-

 

Here is a good link for the 3500xl.

 

http://www.cisco.com/warp/public/788/voip/config_troubleshoot_its.html

 

yes VGW is Voice Gateway.  

 

How does Avaya work with support since your using another vendors
network?  I know Cisco doesn't like supporting other vendors.

 

Scott

 

 

________________________________

From: Dixon, Wayne [mailto:wcdixo at aurora.lib.il.us] 
Sent: Monday, June 26, 2006 8:20 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Basic VOIP Setup Needs

 

The Switches are 3500XLs... It is an Avaya system, and call managing
will be handled by that and forgive my ignorance, but what is A VGW?
Voice Gateway?

 

If it is a Voice Gateway, then we're still awaiting that to come in, we
just ordered that last week.. but I was asked to see what would be
needed to get the network ready for IP Phones

 

Wayne

 

	-----Original Message-----
	From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
	Sent: Monday, June 26, 2006 9:50 AM
	To: Dixon, Wayne; cisco-voip at puck.nether.net
	Subject: RE: [cisco-voip] Basic VOIP Setup Needs

	What kind of Switch?

	 

	Do you already have CM and VGW's setup?

	 

	Scott

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dixon, Wayne
	Sent: Monday, June 26, 2006 7:24 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Basic VOIP Setup Needs

	 

	Hi all, 

	 

	I'm wondering what basics are needed for allowing IP Phones
access on a Cisco Switch network...   We are looking at implementing
some IP Phones.  

	 

	I'm assuming that a Dedicated VLAN for the IP Phones, use of
Native VLAN for non-tagged packets and a DHCP server are needed... but
I'm not sure what else is needed.  

	 

	If anybody has a good source for some basic information on basic
VOIP setup would be greatly appreciated.

	 

	Thanks

	 

	Wayne

	 

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From ealeatherman at gmail.com  Mon Jun 26 11:30:16 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 26 Jun 2006 11:30:16 -0400
Subject: [cisco-voip] Basic VOIP Setup Needs
In-Reply-To: <6D8BEE91433474478D294A8867FFDC6EE3F81A@WMAIN3.aurorapubliclibrary.org>
References: <407055A92CECCB499C922A2D35FC19A605ED516C@Apollo.wesd.org>
	<6D8BEE91433474478D294A8867FFDC6EE3F81A@WMAIN3.aurorapubliclibrary.org>
Message-ID: <94a1afde0606260830q6a59947ehbcc87316b8ec9cfa@mail.gmail.com>

Ah!~ scratch my suggestion about the SRND for callmanager then :) It might
still have some tidbits on QoS and switches in there but its mostly relating
to callmanager.

On 6/26/06, Dixon, Wayne <wcdixo at aurora.lib.il.us> wrote:
>
>  The Switches are 3500XLs... It is an Avaya system, and call managing will
> be handled by that and forgive my ignorance, but what is A VGW?  Voice
> Gateway?
>
> If it is a Voice Gateway, then we're still awaiting that to come in, we
> just ordered that last week.. but I was asked to see what would be needed to
> get the network ready for IP Phones
>
>  Wayne
>
>
>  -----Original Message-----
> *From:* Voll, Scott [mailto:Scott.Voll at wesd.org]
> *Sent:* Monday, June 26, 2006 9:50 AM
> *To:* Dixon, Wayne; cisco-voip at puck.nether.net
> *Subject:* RE: [cisco-voip] Basic VOIP Setup Needs
>
>  What kind of Switch?
>
>
>
> Do you already have CM and VGW's setup?
>
>
>
> Scott
>
>
>  ------------------------------
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Dixon, Wayne
> *Sent:* Monday, June 26, 2006 7:24 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* [cisco-voip] Basic VOIP Setup Needs
>
>
>
> Hi all,
>
>
>
> I'm wondering what basics are needed for allowing IP Phones access on a
> Cisco Switch network...   We are looking at implementing some IP Phones.
>
>
>
> I'm assuming that a Dedicated VLAN for the IP Phones, use of Native VLAN
> for non-tagged packets and a DHCP server are needed... but I'm not sure what
> else is needed.
>
>
>
> If anybody has a good source for some basic information on basic VOIP
> setup would be greatly appreciated.
>
>
>
> Thanks
>
>
>
> Wayne
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From CarlosL at mrs.es  Mon Jun 26 11:30:39 2006
From: CarlosL at mrs.es (Carlos Lopez Fernandez)
Date: Mon, 26 Jun 2006 17:30:39 +0200
Subject: [cisco-voip] - CRA
Message-ID: <687262A118FB1146BFF85FB25047267914AAD8@mrsalow3s001.mrs.local>

thanks for your reply, 
What can Ido If I want to use my own Script?
 
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From CarlosL at mrs.es  Mon Jun 26 11:30:59 2006
From: CarlosL at mrs.es (Carlos Lopez Fernandez)
Date: Mon, 26 Jun 2006 17:30:59 +0200
Subject: [cisco-voip] Voip application
Message-ID: <687262A118FB1146BFF85FB25047267914AAD9@mrsalow3s001.mrs.local>


> Hi all,
> Here its my scenario:
> 
> I receive a call via E1 in a 3640 router with an E1 card,  then via dial-peer voice XXX voip it goes to another cisco router (2801) who runs a Vxml application, but the call ends ans nothing happens. Via debug, I saw that the application starts but > ...> ..
> 
> I try the same vxml in another router directly via a dial-peer voice XXX POTS and run perfectly.
> 
> Any ideas??
> 
> There are the configuration in the routers
> 
> Cisco 3640:
> 
> Dial-peer voice 123 voip
> Destination-pattern 123
> Session target ipv4:10.10.10.1
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
> 
> 
> Cisco 2801
> 
> interface Loopback0
>  ip address 10.10.11.1 255.255.255.0
> !
> dial-peer voice 123 voip
>  service acme
>  destination-pattern 123
>  session target ipv4:10.10.11.1
>  incoming called-number 123
>  dtmf-relay h245-alphanumeric
>  codec g711ulaw
>  no vad
> 
> 
> 
> Thanks in advance
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From CarlosL at mrs.es  Mon Jun 26 11:31:25 2006
From: CarlosL at mrs.es (Carlos Lopez Fernandez)
Date: Mon, 26 Jun 2006 17:31:25 +0200
Subject: [cisco-voip]  7912 Firmware
Message-ID: <687262A118FB1146BFF85FB25047267914AADA@mrsalow3s001.mrs.local>

> Hi all, 
> I have a call manager Express 3.3 in a 12.3(14)T5 IOS, and I?m notice delay in voice, and after analize my network, these are the results:
> 
> 60ms in a ping to any cisco ephone 7912 (10 devices) but 1ms in a ping toone cisco ephone 7960
> 
> All devices are in the same VLAN (in the same switch cat 3560Poe).
> 
> 
> Any ideas of that elevated times ????
> 
> I?m thinking that could be the 7912 firmware (CP7912060000SCCP050124A.sbin) but I don?t find any information about that
> 
> Thanks in advance
> 
> 
> Carlos J.
> 
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From BRiley at jackhenry.com  Mon Jun 26 11:35:20 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Mon, 26 Jun 2006 10:35:20 -0500
Subject: [cisco-voip] - CRA
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002B7EAFB@smoxchg.jhacorp.com>

Buy IPCC. I have heard you can get to some of the hidden sites on CRA
and upload a script but I have only heard of people doing this for TAPS.
I don't know if it would work for anything else. 

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carlos Lopez
Fernandez
Sent: Monday, June 26, 2006 10:31 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] - CRA

 

thanks for your reply, 

What can Ido If I want to use my own Script?

 

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 CORTIZ at broward.org  Mon Jun 26 11:44:32 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon, 26 Jun 2006 11:44:32 -0400
Subject: [cisco-voip] CallManager 4.2 - Anyone running it in
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC354C@MREXVS01.bc.broward.cty>

What version are you upgrading from?  Please let us know how it
goes.......

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Corbett Enders
Sent: Monday, June 26, 2006 10:32 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager 4.2 - Anyone running it in

We're upgrading our production environment tonight...  we don't have a
lab/dev environment to test in.  We need the "call forward unregistered"
feature...

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


From bcarter at sentinel.com  Mon Jun 26 12:26:23 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 26 Jun 2006 11:26:23 -0500
Subject: [cisco-voip] CME Direct Transfer to Voicemail
Message-ID: <6A0744639B86DD459433108DBF101D26FC66A7@MAIL01.sentinel.com>

 
 
In CME how can a user transfer directly to a voicemail box?


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 pwalenta at wi.rr.com  Mon Jun 26 12:32:09 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Mon, 26 Jun 2006 11:32:09 -0500
Subject: [cisco-voip] CME Direct Transfer to Voicemail
In-Reply-To: <6A0744639B86DD459433108DBF101D26FC66A7@MAIL01.sentinel.com>
Message-ID: <00fb01c6993e$12d15a90$0232a8c0@walentawk1>

This should cover what you're looking for:

http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_tech_note0918
6a00802ab979.shtml 

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
> Sent: Monday, June 26, 2006 11:26 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] CME Direct Transfer to Voicemail
> 
>  
>  
> In CME how can a user transfer directly to a voicemail box?
> 
> 
> 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


From mcaudill at cisco.com  Mon Jun 26 12:48:10 2006
From: mcaudill at cisco.com (mcaudill at cisco.com)
Date: Mon, 26 Jun 2006 17:48:10 +0100
Subject: [cisco-voip] (no subject)
Message-ID: <200606261639.k5QGdGFF068578@puck.nether.net>

The message contains Unicode characters and has been sent as a binary attachment.

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From CORTIZ at broward.org  Mon Jun 26 13:26:54 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon, 26 Jun 2006 13:26:54 -0400
Subject: [cisco-voip] Stable Call Mgr Version
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC3550@MREXVS01.bc.broward.cty>

Jonathan,

Following up on my question from last week about CM 4.1.3 and IPCC
versions. I saw that you have been having some problems with IPCC.  When
did this start? After any of the upgrades?

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, June 23, 2006 10:45 PM
To: Ortiz, Carlos
Cc: Ted Nugent; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Stable Call Mgr Version

Are you looking at this:

http://www.cisco.com/application/pdf/en/us/guest/products/ps1846/c1225/c
cmigration_09186a00803d82f5.pdf

Cuz, I am running 4.1(3)sr3b with IPCC 3.5(2)sr3 and page 48 of that
compatibility guide shows 3.5(3)sr3 being supported in 4.1(3)sr3a (and
I have it running with 4.1(3)sr3b (which came out last week).

In fact, that shows 3.5(4) supporting CCM4.2... So, I am not sure what
you are looking at...



Jonathan

On 6/23/06, Ortiz, Carlos <CORTIZ at broward.org> wrote:
> So according to this matrix CM 4.1.3 SR1 is supported with IPC 3.5.3
but
> if you apply SR2 to CM then you have to upgrade IPCC to 4.x....does
that
> make any sense?
>
>
>
> -----Original Message-----
> From: Ted Nugent [mailto:tednugent69 at yahoo.com]
> Sent: Friday, June 23, 2006 10:40 AM
> To: Ortiz, Carlos; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Stable Call Mgr Version
>
> Well you're running on probably the most stable
> version of CM that 4.x has to offer... Other then that
> I would probably run with CM4.1(3)SR2... I've seen
> some issues with SR3 and thier still working them
> out... sr2b just came out the other day. I've only
> deployed Unity 4.1 a few times and each went very
> smoothly with very few followups.. (but 4.0(5) wasn't
> a bad release either... unless there are features
> you're looking for I would stick with 4.0(5))
>
> I personally would not touch CM/Unity 4.2 yet...
> they're both fairly new... I would probably give them
> a couple month.
>
> IPCC 3.5.3 SR3 seemed to be pretty stable as well...
> You could not pay me enough to support IPCCX 4.x
> yet... they are getting closer with newer releases but
> right now.. no way.. no how!
> You can also reference "golden bridge" at the link
> below.
>
>
http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/gblink/ipc
> mtrix.htm
>
>
> --- "Ortiz, Carlos" <CORTIZ at broward.org> wrote:
>
> > Can anyone suggest a stable CM version and IPCC
> > Express that they have
> > successfully running?
> >
> >
> >
> > I am currently running 4.02a(sr2b) with IPCC 3.5.2
> > and Unity 4.05.
> >
> >
> >
> > I would like to move to Call Mgr version 4.1.3 or
> > 4.2 (if recommended).
> > I also realize that IPCC Express will need to be
> > upgraded as part of
> > this.
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Carlos
> >
> > > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
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> Tired of spam?  Yahoo! Mail has the best spam protection around
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>


From bcarter at sentinel.com  Mon Jun 26 13:56:27 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 26 Jun 2006 12:56:27 -0500
Subject: [cisco-voip] CallManager DR scenario
Message-ID: <6A0744639B86DD459433108DBF101D26FC66BC@MAIL01.sentinel.com>

 
I am implementing a CallManager cluster with servers located at both HQ
and a DR site.  The Publisher will be located at HQ, with 2 subscribers
at HQ and the 2 subscribers at the DR site.  

If HQ is offline for an extended period of time (building blows up), the
publisher will not be available.  In this situation, CallManager
configuration changes will probably have to be made, but the publisher
will be down.  

* Can changes be made to the system without an active publisher?

* I know subscriber won't replicated database to other subscribers.
What happens when the publisher comes back online?  Are changes
replicated from Subscriber to Publisher?

* With 2 subscribers located at the DR site, will I have to make
configuration on both subscriber CallManagers (see previous question)? 

* In a period of extended outage, I assume it would be best to restore
the Publisher to one of the Subscribers at DR, correct?

It seems like a Catch-22 for CallManager DR design.  Do you locate the
Publisher at HQ or DR?  I guess it depends on which site will go
down?!?!?  HaHaHa!!


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 Scott.Voll at wesd.org  Mon Jun 26 14:02:35 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 26 Jun 2006 11:02:35 -0700
Subject: [cisco-voip] CallManager DR scenario
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED51FF@Apollo.wesd.org>

To my understanding you can not make changes to the Cluster without the
Pub.  The sub's have a read only DB.

With that said.... all other questions are null and void.

Like you said, the question is where can you put the pub so it is always
alive.


Scott


-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Monday, June 26, 2006 10:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager DR scenario

 
I am implementing a CallManager cluster with servers located at both HQ
and a DR site.  The Publisher will be located at HQ, with 2 subscribers
at HQ and the 2 subscribers at the DR site.  

If HQ is offline for an extended period of time (building blows up), the
publisher will not be available.  In this situation, CallManager
configuration changes will probably have to be made, but the publisher
will be down.  

* Can changes be made to the system without an active publisher?

* I know subscriber won't replicated database to other subscribers.
What happens when the publisher comes back online?  Are changes
replicated from Subscriber to Publisher?

* With 2 subscribers located at the DR site, will I have to make
configuration on both subscriber CallManagers (see previous question)? 

* In a period of extended outage, I assume it would be best to restore
the Publisher to one of the Subscribers at DR, correct?

It seems like a Catch-22 for CallManager DR design.  Do you locate the
Publisher at HQ or DR?  I guess it depends on which site will go
down?!?!?  HaHaHa!!


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


From tednugent69 at yahoo.com  Mon Jun 26 14:16:05 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Mon, 26 Jun 2006 11:16:05 -0700 (PDT)
Subject: [cisco-voip] CallManager DR scenario
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED51FF@Apollo.wesd.org>
Message-ID: <20060626181605.91731.qmail@web36111.mail.mud.yahoo.com>

Correct SUBs are read only and the SUB pull down thier
subscription from the PUB, there is no replication if
the PUB is gone. If you require the PUB at the HQ site
then the only alternative that I see if HQ goes away
is to have a cold standby MCS at the DR site and
perform a BARs recovery onto the the NEW PUB and or
rotate a single mirrored drive out to the DR site.

--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> To my understanding you can not make changes to the
> Cluster without the
> Pub.  The sub's have a read only DB.
> 
> With that said.... all other questions are null and
> void.
> 
> Like you said, the question is where can you put the
> pub so it is always
> alive.
> 
> 
> Scott
> 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Carter, Bill
> Sent: Monday, June 26, 2006 10:56 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] CallManager DR scenario
> 
>  
> I am implementing a CallManager cluster with servers
> located at both HQ
> and a DR site.  The Publisher will be located at HQ,
> with 2 subscribers
> at HQ and the 2 subscribers at the DR site.  
> 
> If HQ is offline for an extended period of time
> (building blows up), the
> publisher will not be available.  In this situation,
> CallManager
> configuration changes will probably have to be made,
> but the publisher
> will be down.  
> 
> * Can changes be made to the system without an
> active publisher?
> 
> * I know subscriber won't replicated database to
> other subscribers.
> What happens when the publisher comes back online? 
> Are changes
> replicated from Subscriber to Publisher?
> 
> * With 2 subscribers located at the DR site, will I
> have to make
> configuration on both subscriber CallManagers (see
> previous question)? 
> 
> * In a period of extended outage, I assume it would
> be best to restore
> the Publisher to one of the Subscribers at DR,
> correct?
> 
> It seems like a Catch-22 for CallManager DR design. 
> Do you locate the
> Publisher at HQ or DR?  I guess it depends on which
> site will go
> down?!?!?  HaHaHa!!
> 
> 
> 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
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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From Scott.Voll at wesd.org  Mon Jun 26 14:30:48 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Mon, 26 Jun 2006 11:30:48 -0700
Subject: [cisco-voip] CallManager DR scenario
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED5211@Apollo.wesd.org>

Then have fun with all the IP changes, etc.

-----Original Message-----
From: Ted Nugent [mailto:tednugent69 at yahoo.com] 
Sent: Monday, June 26, 2006 11:16 AM
To: Voll, Scott; Carter, Bill; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager DR scenario

Correct SUBs are read only and the SUB pull down thier
subscription from the PUB, there is no replication if
the PUB is gone. If you require the PUB at the HQ site
then the only alternative that I see if HQ goes away
is to have a cold standby MCS at the DR site and
perform a BARs recovery onto the the NEW PUB and or
rotate a single mirrored drive out to the DR site.

--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> To my understanding you can not make changes to the
> Cluster without the
> Pub.  The sub's have a read only DB.
> 
> With that said.... all other questions are null and
> void.
> 
> Like you said, the question is where can you put the
> pub so it is always
> alive.
> 
> 
> Scott
> 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Carter, Bill
> Sent: Monday, June 26, 2006 10:56 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] CallManager DR scenario
> 
>  
> I am implementing a CallManager cluster with servers
> located at both HQ
> and a DR site.  The Publisher will be located at HQ,
> with 2 subscribers
> at HQ and the 2 subscribers at the DR site.  
> 
> If HQ is offline for an extended period of time
> (building blows up), the
> publisher will not be available.  In this situation,
> CallManager
> configuration changes will probably have to be made,
> but the publisher
> will be down.  
> 
> * Can changes be made to the system without an
> active publisher?
> 
> * I know subscriber won't replicated database to
> other subscribers.
> What happens when the publisher comes back online? 
> Are changes
> replicated from Subscriber to Publisher?
> 
> * With 2 subscribers located at the DR site, will I
> have to make
> configuration on both subscriber CallManagers (see
> previous question)? 
> 
> * In a period of extended outage, I assume it would
> be best to restore
> the Publisher to one of the Subscribers at DR,
> correct?
> 
> It seems like a Catch-22 for CallManager DR design. 
> Do you locate the
> Publisher at HQ or DR?  I guess it depends on which
> site will go
> down?!?!?  HaHaHa!!
> 
> 
> 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
> 
> _______________________________________________
> 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  Mon Jun 26 14:33:18 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Mon, 26 Jun 2006 14:33:18 -0400
Subject: [cisco-voip] CallManager DR scenario
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC355B@MREXVS01.bc.broward.cty>

Hey it's a disaster.......Move that subnet...... :)

-----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, June 26, 2006 2:31 PM
To: Ted Nugent; Carter, Bill; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager DR scenario

Then have fun with all the IP changes, etc.

-----Original Message-----
From: Ted Nugent [mailto:tednugent69 at yahoo.com] 
Sent: Monday, June 26, 2006 11:16 AM
To: Voll, Scott; Carter, Bill; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager DR scenario

Correct SUBs are read only and the SUB pull down thier
subscription from the PUB, there is no replication if
the PUB is gone. If you require the PUB at the HQ site
then the only alternative that I see if HQ goes away
is to have a cold standby MCS at the DR site and
perform a BARs recovery onto the the NEW PUB and or
rotate a single mirrored drive out to the DR site.

--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> To my understanding you can not make changes to the
> Cluster without the
> Pub.  The sub's have a read only DB.
> 
> With that said.... all other questions are null and
> void.
> 
> Like you said, the question is where can you put the
> pub so it is always
> alive.
> 
> 
> Scott
> 
> 
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On
> Behalf Of Carter, Bill
> Sent: Monday, June 26, 2006 10:56 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] CallManager DR scenario
> 
>  
> I am implementing a CallManager cluster with servers
> located at both HQ
> and a DR site.  The Publisher will be located at HQ,
> with 2 subscribers
> at HQ and the 2 subscribers at the DR site.  
> 
> If HQ is offline for an extended period of time
> (building blows up), the
> publisher will not be available.  In this situation,
> CallManager
> configuration changes will probably have to be made,
> but the publisher
> will be down.  
> 
> * Can changes be made to the system without an
> active publisher?
> 
> * I know subscriber won't replicated database to
> other subscribers.
> What happens when the publisher comes back online? 
> Are changes
> replicated from Subscriber to Publisher?
> 
> * With 2 subscribers located at the DR site, will I
> have to make
> configuration on both subscriber CallManagers (see
> previous question)? 
> 
> * In a period of extended outage, I assume it would
> be best to restore
> the Publisher to one of the Subscribers at DR,
> correct?
> 
> It seems like a Catch-22 for CallManager DR design. 
> Do you locate the
> Publisher at HQ or DR?  I guess it depends on which
> site will go
> down?!?!?  HaHaHa!!
> 
> 
> 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
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

_______________________________________________
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From tednugent69 at yahoo.com  Mon Jun 26 14:56:46 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Mon, 26 Jun 2006 11:56:46 -0700 (PDT)
Subject: [cisco-voip] CallManager DR scenario
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED5211@Apollo.wesd.org>
Message-ID: <20060626185646.35599.qmail@web36103.mail.mud.yahoo.com>

And by that I'm assuming that you have a better idea
that you're looking to share... :-)


--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> Then have fun with all the IP changes, etc.
> 
> -----Original Message-----
> From: Ted Nugent [mailto:tednugent69 at yahoo.com] 
> Sent: Monday, June 26, 2006 11:16 AM
> To: Voll, Scott; Carter, Bill;
> cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CallManager DR scenario
> 
> Correct SUBs are read only and the SUB pull down
> thier
> subscription from the PUB, there is no replication
> if
> the PUB is gone. If you require the PUB at the HQ
> site
> then the only alternative that I see if HQ goes away
> is to have a cold standby MCS at the DR site and
> perform a BARs recovery onto the the NEW PUB and or
> rotate a single mirrored drive out to the DR site.
> 
> --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
> 
> > To my understanding you can not make changes to
> the
> > Cluster without the
> > Pub.  The sub's have a read only DB.
> > 
> > With that said.... all other questions are null
> and
> > void.
> > 
> > Like you said, the question is where can you put
> the
> > pub so it is always
> > alive.
> > 
> > 
> > Scott
> > 
> > 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Carter, Bill
> > Sent: Monday, June 26, 2006 10:56 AM
> > To: cisco-voip at puck.nether.net
> > Subject: [cisco-voip] CallManager DR scenario
> > 
> >  
> > I am implementing a CallManager cluster with
> servers
> > located at both HQ
> > and a DR site.  The Publisher will be located at
> HQ,
> > with 2 subscribers
> > at HQ and the 2 subscribers at the DR site.  
> > 
> > If HQ is offline for an extended period of time
> > (building blows up), the
> > publisher will not be available.  In this
> situation,
> > CallManager
> > configuration changes will probably have to be
> made,
> > but the publisher
> > will be down.  
> > 
> > * Can changes be made to the system without an
> > active publisher?
> > 
> > * I know subscriber won't replicated database to
> > other subscribers.
> > What happens when the publisher comes back online?
> 
> > Are changes
> > replicated from Subscriber to Publisher?
> > 
> > * With 2 subscribers located at the DR site, will
> I
> > have to make
> > configuration on both subscriber CallManagers (see
> > previous question)? 
> > 
> > * In a period of extended outage, I assume it
> would
> > be best to restore
> > the Publisher to one of the Subscribers at DR,
> > correct?
> > 
> > It seems like a Catch-22 for CallManager DR
> design. 
> > Do you locate the
> > Publisher at HQ or DR?  I guess it depends on
> which
> > site will go
> > down?!?!?  HaHaHa!!
> > 
> > 
> > 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
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> 


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From ealeatherman at gmail.com  Mon Jun 26 15:04:42 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 26 Jun 2006 15:04:42 -0400
Subject: [cisco-voip] CallManager DR scenario
In-Reply-To: <20060626181605.91731.qmail@web36111.mail.mud.yahoo.com>
References: <407055A92CECCB499C922A2D35FC19A605ED51FF@Apollo.wesd.org>
	<20060626181605.91731.qmail@web36111.mail.mud.yahoo.com>
Message-ID: <94a1afde0606261204s3d09ed2bh3c5b8675d31dcddf@mail.gmail.com>

We have a very similar setup (pub+2 subs at one site, 2subs across town at
another site). I have a spare MCS of the same model sitting at the secondary
site for this purpose, just rebuild the pub from an offsite BARS backup and
move the subnet to there. can't do any changes till publisher is back up
though.

basically it sounds like you need a strategy to rebuild a publisher quickly
so you can continue doing changes.

On 6/26/06, Ted Nugent <tednugent69 at yahoo.com> wrote:
>
> Correct SUBs are read only and the SUB pull down thier
> subscription from the PUB, there is no replication if
> the PUB is gone. If you require the PUB at the HQ site
> then the only alternative that I see if HQ goes away
> is to have a cold standby MCS at the DR site and
> perform a BARs recovery onto the the NEW PUB and or
> rotate a single mirrored drive out to the DR site.
>
> --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
>
> > To my understanding you can not make changes to the
> > Cluster without the
> > Pub.  The sub's have a read only DB.
> >
> > With that said.... all other questions are null and
> > void.
> >
> > Like you said, the question is where can you put the
> > pub so it is always
> > alive.
> >
> >
> > Scott
> >
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On
> > Behalf Of Carter, Bill
> > Sent: Monday, June 26, 2006 10:56 AM
> > To: cisco-voip at puck.nether.net
> > Subject: [cisco-voip] CallManager DR scenario
> >
> >
> > I am implementing a CallManager cluster with servers
> > located at both HQ
> > and a DR site.  The Publisher will be located at HQ,
> > with 2 subscribers
> > at HQ and the 2 subscribers at the DR site.
> >
> > If HQ is offline for an extended period of time
> > (building blows up), the
> > publisher will not be available.  In this situation,
> > CallManager
> > configuration changes will probably have to be made,
> > but the publisher
> > will be down.
> >
> > * Can changes be made to the system without an
> > active publisher?
> >
> > * I know subscriber won't replicated database to
> > other subscribers.
> > What happens when the publisher comes back online?
> > Are changes
> > replicated from Subscriber to Publisher?
> >
> > * With 2 subscribers located at the DR site, will I
> > have to make
> > configuration on both subscriber CallManagers (see
> > previous question)?
> >
> > * In a period of extended outage, I assume it would
> > be best to restore
> > the Publisher to one of the Subscribers at DR,
> > correct?
> >
> > It seems like a Catch-22 for CallManager DR design.
> > Do you locate the
> > Publisher at HQ or DR?  I guess it depends on which
> > site will go
> > down?!?!?  HaHaHa!!
> >
> >
> > 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
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> __________________________________________________
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> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
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>



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From ealeatherman at gmail.com  Mon Jun 26 15:07:50 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 26 Jun 2006 15:07:50 -0400
Subject: [cisco-voip] Crystal Reports
Message-ID: <94a1afde0606261207o65465374x3700e92d215147e9@mail.gmail.com>

Anyone good with crystal reports that could suggest how to get started with
it? specifically we want to grab reports out of IPCC Express 4.x and send
out those reports to various manager's email addresses. I've never used
crystal reports before so i'm basically at square one with this. Does anyone
else use it for this?

Thanks!

-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From Timothy_Frazee at adp.com  Mon Jun 26 15:25:48 2006
From: Timothy_Frazee at adp.com (Frazee, Timothy)
Date: Mon, 26 Jun 2006 14:25:48 -0500
Subject: [cisco-voip] MGCP on CCM 4.2
In-Reply-To: <C65033A817B5734392E55CB9DF446104058DB55D@USNAEXCH.na.didata.local>
References: <C65033A817B5734392E55CB9DF446104058DB55D@USNAEXCH.na.didata.local>
Message-ID: <970C4A9D-4F24-4746-AC60-697CA2367343@mimectl>

are you binding mgcp to a loopback interface?

with that, do you have all the correct commands to ensure proper operation? (config)

can ccm ping the loopback interface? (routing)






From: Joe Pollere (US)
Sent: Mon 6/26/2006 9:59 AM
To: Nick Kassel; Voll, Scott; Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2


Do you have the GW Hostname in your HOSTS file on the CCM server, could be a DNS issue.
 
Joe
 



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 10:44 AM
To: Voll, Scott; Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2
 
Yes, all the standard MGCP commands are in as the gateway states it's registered when I do a sho ccm-manager, also the isdn status is up so the config had been downloaded. Just unable to get it registered on the CCM.  
 



From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, June 26, 2006 3:39 PM
To: Nick Kassel; Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MGCP on CCM 4.2
 
Do you have all the ccm-manager and mgcp commands?  Sounds like you might be missing something.
 
Scott
 



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 7:20 AM
To: Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2
 
The hostname has no periods and is also identical to the domain name on the MGCP gateway configuration on CCM. I have also made sure that no IP domain name has been configured, this is for BRI, does this make any difference?
 
 
 



From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Monday, June 26, 2006 3:16 PM
To: Nick Kassel
Subject: Re: [cisco-voip] MGCP on CCM 4.2
 
no periods in the hostname for the router with mgcp.
 


 
On 6/26/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote: 
The ISDN status shows the layer 1 and 2 as up and the MGCP shows as registered on the gateway but I am unable to get it registered on the CCM, can anyone help me out with this? 
 



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 1:53 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP on CCM 4.2
 
I'm having a little trouble getting a MGCP gateway working with CCM 4.2, does anyone know if I have to use a minimum IOS, I currently trying with 12.3 (11)
I'm unable to download the configuration.
 

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The information contained in this e-mail is strictly confidential, some or all
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recipient(s) accesses them and that delivery receipt does not constitute
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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 
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
mailto: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,
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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
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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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
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may be a delay in processing the instructions received from e-mails after
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and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************************************


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

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

Charles Stanley Sutherlands and 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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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 Greg_Hinson at adp.com  Mon Jun 26 15:45:47 2006
From: Greg_Hinson at adp.com (Hinson, Greg)
Date: Mon, 26 Jun 2006 14:45:47 -0500
Subject: [cisco-voip] MGCP on CCM 4.2
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BCD8CFE@EXCHANGEDS01.ds.ad.adp.com>

In the earlier part of this string, I saw that the voice gateway was
unable to download tftp configuration. I ran into a similar problem with
4.2 callmanager, where the gateway never registered with CCM. Check
ccm-manager status on the gateway (show ccm-manager) and ensure the tftp
configurations show "downloaded successfully". If not, Make sure you
have the following two commands in the running config:
 
ccm-manager config
ccm-manager config server xxx.xxx.xxx.xxx (Ip of the callmanager's TFTP
server address)
 
If TFTP is sourced from the callmanager, make sure the TFTP service is
started.
 
 
-Greg
 
________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Frazee, Timothy
Sent: Monday, June 26, 2006 12:26 PM
To: Joe Pollere (US); Nick Kassel; Voll, Scott; Justin Steinberg; Aman
Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2
 
are you binding mgcp to a loopback interface?
 
with that, do you have all the correct commands to ensure proper
operation? (config)
 
can ccm ping the loopback interface? (routing)
 
 
 
 
________________________________

From: Joe Pollere (US)
Sent: Mon 6/26/2006 9:59 AM
To: Nick Kassel; Voll, Scott; Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2
Do you have the GW Hostname in your HOSTS file on the CCM server, could
be a DNS issue.
 
Joe
 
________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 10:44 AM
To: Voll, Scott; Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2
 
Yes, all the standard MGCP commands are in as the gateway states it's
registered when I do a sho ccm-manager, also the isdn status is up so
the config had been downloaded. Just unable to get it registered on the
CCM.  
 
________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, June 26, 2006 3:39 PM
To: Nick Kassel; Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] MGCP on CCM 4.2
 
Do you have all the ccm-manager and mgcp commands?  Sounds like you
might be missing something.
 
Scott
 
________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 7:20 AM
To: Justin Steinberg; Aman Chugh
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] MGCP on CCM 4.2
 
The hostname has no periods and is also identical to the domain name on
the MGCP gateway configuration on CCM. I have also made sure that no IP
domain name has been configured, this is for BRI, does this make any
difference?
 
 
 
________________________________

From: Justin Steinberg [mailto:jsteinberg at gmail.com] 
Sent: Monday, June 26, 2006 3:16 PM
To: Nick Kassel
Subject: Re: [cisco-voip] MGCP on CCM 4.2
 
no periods in the hostname for the router with mgcp.
 


 
On 6/26/06, Nick Kassel <Nick.Kassel at charles-stanley.co.uk> wrote: 
The ISDN status shows the layer 1 and 2 as up and the MGCP shows as
registered on the gateway but I am unable to get it registered on the
CCM, can anyone help me out with this? 
 
________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Monday, June 26, 2006 1:53 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] MGCP on CCM 4.2
 
I'm having a little trouble getting a MGCP gateway working with CCM 4.2,
does anyone know if I have to use a minimum IOS, I currently trying with
12.3 (11)
I'm unable to download the configuration.
 

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

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.

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

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

The information contained in this e-mail is strictly confidential, some
or all
of which may be legally privileged. It is for the intended recipient
only.
Access to this e-mail by any other person is prohibited. If you are not
the
intended recipient, any use, disclosure, copying, printing, distribution
of,
replying to or any action taken or omitted to be taken in reliance on
this
e-mail, is prohibited and may be unlawful. Please contact the sender
immediately
should this e-mail have been incorrectly addressed or transmitted.

You accept that any instructions are deemed to have been given at the
time the
recipient(s) accesses them and that delivery receipt does not constitute
acknowledgement or receipt by the intended recipient(s). You accept that
there
may be a delay in processing the instructions received from e-mails
after
Charles Stanley has received them. You are advised that urgent, time
sensitive
and confidential communications should not be sent by e-mail. 

You acknowledge that e-mails are not secure and you accept the risk of
malfunction, viruses, unauthorised interference, mis-delivery or delay.
************************************************************************
************************


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

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

Charles Stanley Sutherlands and 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.
________________________________

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. 


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 SPackett at fenwick.com  Mon Jun 26 17:04:57 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Mon, 26 Jun 2006 14:04:57 -0700
Subject: [cisco-voip] CallManager DR scenario
Message-ID: <965C0255FD5828438E34F9A68ECA76E7778DDA@svcexch02.firm.fenwick.llp>

I'm thinking about moving my Pub to our DR site as well.  I'm just
concerned about network traffic between the sites.  I know there's a
rule to not have phones registered to the Pub because it does many
roles.  Does anyone know what roles that might be?  I have a DS3 between
my HQ and DR site and I think I have enough bandwidth to handle the
traffic.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Monday, June 26, 2006 10:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager DR scenario

 
I am implementing a CallManager cluster with servers located at both HQ
and a DR site.  The Publisher will be located at HQ, with 2 subscribers
at HQ and the 2 subscribers at the DR site.  

If HQ is offline for an extended period of time (building blows up), the
publisher will not be available.  In this situation, CallManager
configuration changes will probably have to be made, but the publisher
will be down.  

* Can changes be made to the system without an active publisher?

* I know subscriber won't replicated database to other subscribers.
What happens when the publisher comes back online?  Are changes
replicated from Subscriber to Publisher?

* With 2 subscribers located at the DR site, will I have to make
configuration on both subscriber CallManagers (see previous question)? 

* In a period of extended outage, I assume it would be best to restore
the Publisher to one of the Subscribers at DR, correct?

It seems like a Catch-22 for CallManager DR design.  Do you locate the
Publisher at HQ or DR?  I guess it depends on which site will go
down?!?!?  HaHaHa!!


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From bcarter at sentinel.com  Mon Jun 26 17:07:33 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 26 Jun 2006 16:07:33 -0500
Subject: [cisco-voip] CallManager DR scenario
Message-ID: <6A0744639B86DD459433108DBF101D26FC66F5@MAIL01.sentinel.com>

Actually it isn't a bad idea to have drives available for cold standby.
IPCC Express cold standby servers will be located at the DR site.  The
subnet will be extended to the DR site via metro fiber.  When the rotate
drives for IPCC they can have standby drives for the DR site.  

I need to think about server placement and groups.  I was thinking of
the Publisher being a "Glass House" server.  CM groups would be
configured with primary and secondary as HQCM02 and DRCM04, or HQCM03
and DRCM05.

Maybe I could do this:
Group A - HQCM02, DRCM04, HQCM01
Group B - HQCM03, DRCM05, HQCM01

In this scenario, during a prolonged outage, I could put HQCM01 drives
into the DRCM05 server.  Then devices would register to either DRCM04 or
HQCM01 server.

-----Original Message-----
From: Ted Nugent [mailto:tednugent69 at yahoo.com] 
Sent: Monday, June 26, 2006 1:57 PM
To: Voll, Scott; Carter, Bill; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] CallManager DR scenario

And by that I'm assuming that you have a better idea that you're looking
to share... :-)


--- "Voll, Scott" <Scott.Voll at wesd.org> wrote:

> Then have fun with all the IP changes, etc.
> 
> -----Original Message-----
> From: Ted Nugent [mailto:tednugent69 at yahoo.com]
> Sent: Monday, June 26, 2006 11:16 AM
> To: Voll, Scott; Carter, Bill;
> cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CallManager DR scenario
> 
> Correct SUBs are read only and the SUB pull down thier subscription 
> from the PUB, there is no replication if the PUB is gone. If you 
> require the PUB at the HQ site then the only alternative that I see if

> HQ goes away is to have a cold standby MCS at the DR site and perform 
> a BARs recovery onto the the NEW PUB and or rotate a single mirrored 
> drive out to the DR site.
> 
> --- "Voll, Scott" <Scott.Voll at wesd.org> wrote:
> 
> > To my understanding you can not make changes to
> the
> > Cluster without the
> > Pub.  The sub's have a read only DB.
> > 
> > With that said.... all other questions are null
> and
> > void.
> > 
> > Like you said, the question is where can you put
> the
> > pub so it is always
> > alive.
> > 
> > 
> > Scott
> > 
> > 
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, 
> > Bill
> > Sent: Monday, June 26, 2006 10:56 AM
> > To: cisco-voip at puck.nether.net
> > Subject: [cisco-voip] CallManager DR scenario
> > 
> >  
> > I am implementing a CallManager cluster with
> servers
> > located at both HQ
> > and a DR site.  The Publisher will be located at
> HQ,
> > with 2 subscribers
> > at HQ and the 2 subscribers at the DR site.  
> > 
> > If HQ is offline for an extended period of time (building blows up),

> > the publisher will not be available.  In this
> situation,
> > CallManager
> > configuration changes will probably have to be
> made,
> > but the publisher
> > will be down.  
> > 
> > * Can changes be made to the system without an active publisher?
> > 
> > * I know subscriber won't replicated database to other subscribers.
> > What happens when the publisher comes back online?
> 
> > Are changes
> > replicated from Subscriber to Publisher?
> > 
> > * With 2 subscribers located at the DR site, will
> I
> > have to make
> > configuration on both subscriber CallManagers (see previous 
> > question)?
> > 
> > * In a period of extended outage, I assume it
> would
> > be best to restore
> > the Publisher to one of the Subscribers at DR, correct?
> > 
> > It seems like a Catch-22 for CallManager DR
> design. 
> > Do you locate the
> > Publisher at HQ or DR?  I guess it depends on
> which
> > site will go
> > down?!?!?  HaHaHa!!
> > 
> > 
> > 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.
> > 
> > _______________________________________________
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> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
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> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
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From keith.klevenski at rig.net  Mon Jun 26 17:11:15 2006
From: keith.klevenski at rig.net (Keith Klevenski)
Date: Mon, 26 Jun 2006 16:11:15 -0500
Subject: [cisco-voip] Billing Software Tested with CCM 5.0
Message-ID: <7F0CE0A1A0961E498A41329535342266A10A6A@HOUEXH02.rignet.org>

Anyone know of any billing software that has been tested or has planned
support for CCM 5.0?  We plan on moving to 5.0 this fall/winter and we
are in the process of possibly going to a different vendor for billing
for various reasons.  Any info is welcome!

Keith Klevenski

 

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From Matt.Slaga at us.didata.com  Mon Jun 26 17:20:24 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Mon, 26 Jun 2006 17:20:24 -0400
Subject: [cisco-voip] CallManager 4.2 - Anyone running it in
Message-ID: <C1FE15183DA37645BC0633BC604E44F0024AC476@USNAEXCH.na.didata.local>

Something to keep in mind with 4.2, there is no upgrade path from 4.2 to
5.0.  If you want to upgrade to version 5, you will have to wait until
5.1 (or possibly later) to upgrade.
 



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: Monday, June 26, 2006 11:45 AM
To: Corbett Enders; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager 4.2 - Anyone running it in

What version are you upgrading from?  Please let us know how it
goes.......

Carlos

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Corbett Enders
Sent: Monday, June 26, 2006 10:32 AM
To: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CallManager 4.2 - Anyone running it in

We're upgrading our production environment tonight...  we don't have a
lab/dev environment to test in.  We need the "call forward unregistered"
feature...

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

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From pwalenta at wi.rr.com  Mon Jun 26 17:24:03 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Mon, 26 Jun 2006 16:24:03 -0500
Subject: [cisco-voip] CallManager DR scenario
In-Reply-To: <6A0744639B86DD459433108DBF101D26FC66BC@MAIL01.sentinel.com>
Message-ID: <012201c69966$d9a3ea70$0232a8c0@walentawk1>

All the other comments to this have been great.

I'll add one thing about "moving the subnet".

Almost all of my recent designs now put the publisher and subscribers in
their own small subnets (/29 or /30).  This makes them far more portable in
the event of problems. 

> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net 
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
> Sent: Monday, June 26, 2006 12:56 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] CallManager DR scenario
> 
>  
> I am implementing a CallManager cluster with servers located 
> at both HQ and a DR site.  The Publisher will be located at 
> HQ, with 2 subscribers at HQ and the 2 subscribers at the DR site.  
> 
> If HQ is offline for an extended period of time (building 
> blows up), the publisher will not be available.  In this 
> situation, CallManager configuration changes will probably 
> have to be made, but the publisher will be down.  
> 
> * Can changes be made to the system without an active publisher?
> 
> * I know subscriber won't replicated database to other subscribers.
> What happens when the publisher comes back online?  Are 
> changes replicated from Subscriber to Publisher?
> 
> * With 2 subscribers located at the DR site, will I have to 
> make configuration on both subscriber CallManagers (see 
> previous question)? 
> 
> * In a period of extended outage, I assume it would be best 
> to restore the Publisher to one of the Subscribers at DR, correct?
> 
> It seems like a Catch-22 for CallManager DR design.  Do you 
> locate the Publisher at HQ or DR?  I guess it depends on 
> which site will go down?!?!?  HaHaHa!!
> 
> 
> 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


From bprentiss at gmail.com  Mon Jun 26 17:56:11 2006
From: bprentiss at gmail.com (Brian Prentiss)
Date: Mon, 26 Jun 2006 14:56:11 -0700
Subject: [cisco-voip] Basic VOIP Setup Needs
In-Reply-To: <94a1afde0606260830q6a59947ehbcc87316b8ec9cfa@mail.gmail.com>
References: <407055A92CECCB499C922A2D35FC19A605ED516C@Apollo.wesd.org>
	<6D8BEE91433474478D294A8867FFDC6EE3F81A@WMAIN3.aurorapubliclibrary.org>
	<94a1afde0606260830q6a59947ehbcc87316b8ec9cfa@mail.gmail.com>
Message-ID: <7e82b2130606261456l72b17b0em7e6b50c56118c321@mail.gmail.com>

Ed,

I think you were close.  Have a look at the Enterprise QoS Solution
Reference Network Design Guide Version
3.3</application/pdf/en/us/guest/netsol/ns432/c649/ccmigration_09186a008049b062.pdf>
.

It is on http://www.cisco.com/go/srnd page.

It might again be more call manager specific, but that doc was a good read
the last time I went through it.

I hope that helps.
Brian

On 6/26/06, Ed Leatherman <ealeatherman at gmail.com> wrote:
>
> Ah!~ scratch my suggestion about the SRND for callmanager then :) It might
> still have some tidbits on QoS and switches in there but its mostly relating
> to callmanager.
>
> On 6/26/06, Dixon, Wayne <wcdixo at aurora.lib.il.us> wrote:
>
> >  The Switches are 3500XLs... It is an Avaya system, and call managing
> > will be handled by that and forgive my ignorance, but what is A VGW?  Voice
> > Gateway?
> >
> > If it is a Voice Gateway, then we're still awaiting that to come in, we
> > just ordered that last week.. but I was asked to see what would be needed to
> > get the network ready for IP Phones
> >
> >  Wayne
> >
> >
> >  -----Original Message-----
> > *From:* Voll, Scott [mailto:Scott.Voll at wesd.org]
> > *Sent:* Monday, June 26, 2006 9:50 AM
> > *To:* Dixon, Wayne; cisco-voip at puck.nether.net
> > *Subject:* RE: [cisco-voip] Basic VOIP Setup Needs
> >
> >  What kind of Switch?
> >
> >
> >
> > Do you already have CM and VGW's setup?
> >
> >
> >
> > Scott
> >
> >
> >  ------------------------------
> >
> > *From:* cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net]
> > *On Behalf Of *Dixon, Wayne
> > *Sent:* Monday, June 26, 2006 7:24 AM
> > *To:* cisco-voip at puck.nether.net
> > *Subject:* [cisco-voip] Basic VOIP Setup Needs
> >
> >
> >
> > Hi all,
> >
> >
> >
> > I'm wondering what basics are needed for allowing IP Phones access on a
> > Cisco Switch network...   We are looking at implementing some IP Phones.
> >
> >
> >
> > I'm assuming that a Dedicated VLAN for the IP Phones, use of Native VLAN
> > for non-tagged packets and a DHCP server are needed... but I'm not sure what
> > else is needed.
> >
> >
> >
> > If anybody has a good source for some basic information on basic VOIP
> > setup would be greatly appreciated.
> >
> >
> >
> > Thanks
> >
> >
> >
> > Wayne
> >
> >
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
>
>
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
>
> --
> Ed Leatherman
> IP Telephony Coordinator
> 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 cshaffer at gmail.com  Mon Jun 26 18:15:27 2006
From: cshaffer at gmail.com (Curt Shaffer)
Date: Mon, 26 Jun 2006 18:15:27 -0400
Subject: [cisco-voip] CCM download
Message-ID: <002b01c6996e$07caa090$181b10ac@betsy>

I heard a while back that you can download Cisco Call Manger for free but I
can't find it. I really dislike the navigation on their site sometimes. Can
anyone point me in the right direction?

 

 

Thanks

 

Curt

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From ealeatherman at gmail.com  Mon Jun 26 19:00:56 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Mon, 26 Jun 2006 19:00:56 -0400
Subject: [cisco-voip] CallManager DR scenario
In-Reply-To: <6A0744639B86DD459433108DBF101D26FC66F5@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26FC66F5@MAIL01.sentinel.com>
Message-ID: <94a1afde0606261600v16cc050fqcbeb3acf63e3dd3d@mail.gmail.com>

The way I configured my device pools was something like:

pub at main site.

sub A, sub B at main site
Sub C, sub D at secondary site

I split campus up into two areas, phones in one area go to the main site and
in the other go to the secondary.
CCM Groups: ABC, BAD, CDA, DCB

then based on the primary line of a phone:
ABC = even numbered DN's in main area
BAD = odd numbered DN's in main area
CDA = even numbered DN's in second area
DCB = odd numbered DN's in second area

this way either site can cover if the other goes offline, and the loads are
relatively well balanced on each server. have a seperate TFTP Server at the
main site that could turn into a subscriber if needed, and a spare server at
the secondary site that could be pub or sub if needed. Has worked out well
so far, havent had to rebuild any servers yet though (jinx!)

On 6/26/06, Carter, Bill <bcarter at sentinel.com> wrote:
>
> Actually it isn't a bad idea to have drives available for cold standby.
> IPCC Express cold standby servers will be located at the DR site.  The
> subnet will be extended to the DR site via metro fiber.  When the rotate
> drives for IPCC they can have standby drives for the DR site.
>
> I need to think about server placement and groups.  I was thinking of
> the Publisher being a "Glass House" server.  CM groups would be
> configured with primary and secondary as HQCM02 and DRCM04, or HQCM03
> and DRCM05.
>
> Maybe I could do this:
> Group A - HQCM02, DRCM04, HQCM01
> Group B - HQCM03, DRCM05, HQCM01
>
> In this scenario, during a prolonged outage, I could put HQCM01 drives
> into the DRCM05 server.  Then devices would register to either DRCM04 or
> HQCM01 server.
>
>
-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From commo_ssg_31f at yahoo.com  Mon Jun 26 19:03:06 2006
From: commo_ssg_31f at yahoo.com (ash AD)
Date: Mon, 26 Jun 2006 16:03:06 -0700 (PDT)
Subject: [cisco-voip] Basic VOIP Setup Needs
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED518E@Apollo.wesd.org>
Message-ID: <20060626230306.56333.qmail@web50410.mail.yahoo.com>

Don't forget the cisco switches may have some issues detecting that they are phones, since you are probably not using Cisco IP phones. You will probably have to set the VLANs for the phones manually as access vlans on the ports assigned to non-cisco IP phones. Good Luck. 

"Voll, Scott" <Scott.Voll at wesd.org> wrote:        v\:* {behavior:url(#default#VML);}  o\:* {behavior:url(#default#VML);}  w\:* {behavior:url(#default#VML);}  .shape {behavior:url(#default#VML);}        st1\:*{behavior:url(#default#ieooui) }                 Wayne?
   
  Here is a good link for the 3500xl.
   
  http://www.cisco.com/warp/public/788/voip/config_troubleshoot_its.html
   
  yes VGW is Voice Gateway.  
   
  How does Avaya work with support since your using another vendors network?  I know Cisco doesn?t like supporting other vendors.
   
  Scott
   
   
      
---------------------------------
  
  From: Dixon, Wayne [mailto:wcdixo at aurora.lib.il.us] 
Sent: Monday, June 26, 2006 8:20 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Basic VOIP Setup Needs

   
    The Switches are 3500XLs... It is an Avaya system, and call managing will be handled by that and forgive my ignorance, but what is A VGW?  Voice Gateway?

     

    If it is a Voice Gateway, then we're still awaiting that to come in, we just ordered that last week.. but I was asked to see what would be needed to get the network ready for IP Phones

     

  Wayne
   
    -----Original Message-----
From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Monday, June 26, 2006 9:50 AM
To: Dixon, Wayne; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Basic VOIP Setup Needs
  What kind of Switch?
   
  Do you already have CM and VGW?s setup?
   
  Scott
   
      
---------------------------------
  
  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Dixon, Wayne
Sent: Monday, June 26, 2006 7:24 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Basic VOIP Setup Needs

   
    Hi all, 

     

    I'm wondering what basics are needed for allowing IP Phones access on a Cisco Switch network...   We are looking at implementing some IP Phones.  

     

    I'm assuming that a Dedicated VLAN for the IP Phones, use of Native VLAN for non-tagged packets and a DHCP server are needed... but I'm not sure what else is needed.  

     

    If anybody has a good source for some basic information on basic VOIP setup would be greatly appreciated.

     

    Thanks

     

  Wayne
   

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From jrlyon at Trinsic.com  Mon Jun 26 19:04:31 2006
From: jrlyon at Trinsic.com (Lyon, John  R.)
Date: Mon, 26 Jun 2006 19:04:31 -0400
Subject: [cisco-voip] Area code blocking in BTS
Message-ID: <11C0A2D166B2D045A9429C4B23177A5E014CBF80@tpa-pzb1-mbx-01.ztel.com>

Hi all,

 

I've got a customer requesting that outdials to certain area codes be
blocked from his multiple office locations.  There is a CME at each site
with a trunk coming back to a BTS 10200 softswitch in our office.  Is
there a way I can block the outdials in the BTS without doing it for all
of our customers, or is this something that will need to be done at the
CME level?

 

Thanks,

John

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From commo_ssg_31f at yahoo.com  Mon Jun 26 19:17:08 2006
From: commo_ssg_31f at yahoo.com (ash AD)
Date: Mon, 26 Jun 2006 16:17:08 -0700 (PDT)
Subject: [cisco-voip] mcaudill@cisco.com
In-Reply-To: <200606261639.k5QGdGFF068578@puck.nether.net>
Message-ID: <20060626231708.9368.qmail@web50407.mail.yahoo.com>

Your message contained a vrirus. Please inform your administrator that you are infected with the W32.Mytob.AF at mm virus. Please no one open the email from mcaudill at cisco.com with no subject. Good luck.
   
  ~Peter

mcaudill at cisco.com wrote:
  The message contains Unicode characters and has been sent as a binary attachment.

_______________________________________________
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cisco-voip at puck.nether.net
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From michael.windon at gmail.com  Mon Jun 26 19:33:20 2006
From: michael.windon at gmail.com (Michael L. Windon)
Date: Mon, 26 Jun 2006 18:33:20 -0500
Subject: [cisco-voip] Selections Unavailable list.xml
Message-ID: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAANu8Kq6czuUuapNUwKTVli+KAAAAQAAAAUQWFDF4pikmPW4N94YtPmQEAAAAA@inxi.com>

 

All, what am I doing wrong here!!?!?! I used these instructions to configure
Background Images for a 7911:

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


http://www.cisco.com/en/US/products/hw/phones/ps379/products_administration_
guide_chapter09186a00805adae5.html#wp1088046Creating Custom Background
Images 


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

 

I created the List.xml file here it is listed below:

 

<CiscoIPPhoneImageList> 
 
<ImageItem Image="TFTP:Desktops/95x34x1/TN-911.png"
URL="TFTP:Desktops/95x34x1/911.png" /> 
 
</CiscoIPPhoneImageList>
 
----------------------------------------------------------------------------
--

 

I also created the Desktops/95x34x1 directory structure and ensure the
CCMService user has full access to the directory. I have also restarted the
TFTP service and restarted the IP Phone. Not matter what I do when I try to
modify the background "Selections Unavailable" is displayed at the bottom of
the display. What am I missing?

 

Mike

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From jonvoip at gmail.com  Mon Jun 26 20:19:44 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Mon, 26 Jun 2006 19:19:44 -0500
Subject: [cisco-voip] Selections Unavailable list.xml
In-Reply-To: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAANu8Kq6czuUuapNUwKTVli+KAAAAQAAAAUQWFDF4pikmPW4N94YtPmQEAAAAA@inxi.com>
References: <!~!UENERkVCMDkAAQACAAAAAAAAAAAAAAAAABgAAAAAAAAANu8Kq6czuUuapNUwKTVli+KAAAAQAAAAUQWFDF4pikmPW4N94YtPmQEAAAAA@inxi.com>
Message-ID: <5d093f9a0606261719l29f1ba0dp6823213e6b3ef035@mail.gmail.com>

I did this today and it worked great

First off, make sure the files are really PMG files (MSPAINT works
great for converting stuff)

Second, make sure that you follow this format:

<ImageItem Image="TFTP:Desktops/320x212x12/USFLAG-tn.PNG"
URL="TFTP:Desktops/320x212x12/USFLAG.PNG"/>

on two lines (to follow the doc).

Third, put it in this directory under TFTPPATH:

Desktops\320x212x12\

Does it not even give you the default background?


Jonathan


On 6/26/06, Michael L. Windon <michael.windon at gmail.com> wrote:
>
>
>
>
>
>
> All, what am I doing wrong here!!?!?! I used these instructions to configure
> Background Images for a 7911:
>
> -------------------------------------------------------------------------------------------------------------------------------------------------------
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_administration_guide_chapter09186a00805adae5.html#wp1088046Creating
> Custom Background Images
>
> -------------------------------------------------------------------------------------------------------------------------------------------------------
>
>
>
> I created the List.xml file here it is listed below:
>
>   <CiscoIPPhoneImageList>
>
> <ImageItem Image="TFTP:Desktops/95x34x1/TN-911.png"
> URL="TFTP:Desktops/95x34x1/911.png" />
>
> </CiscoIPPhoneImageList>
>
> ------------------------------------------------------------------------------
>
>
>
>
> I also created the Desktops/95x34x1 directory structure and ensure the
> CCMService user has full access to the directory. I have also restarted the
> TFTP service and restarted the IP Phone. Not matter what I do when I try to
> modify the background "Selections Unavailable" is displayed at the bottom of
> the display. What am I missing?
>
>
>
> Mike
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From cshaffer at gmail.com  Mon Jun 26 20:59:27 2006
From: cshaffer at gmail.com (Curt Shaffer)
Date: Mon, 26 Jun 2006 20:59:27 -0400
Subject: [cisco-voip] Lab requirements
Message-ID: <009901c69984$f1027240$181b10ac@betsy>

I am looking into setting up a lab for VoIP testing and practice for the
CCVP. I found what I think is a good deal on a 1720 with 32/16 memory and a
2610 with 64/16 memory. Can anyone out there tell me if this will suffice
for a small test environment? I am planning on getting some VICs as well for
practice on configuring those interfaces as well. I just want to make sure I
buy the right tool for the job. Also with all of this in mind would I be
better off with Call Manager Express to put on the routers (if they will
even run them) or would it be wise to just get the full thing. I guess what
would be best is just some suggestions from everyone out there who has labs
on what they are using and what works/worked well for them.

 

Thanks

 

Curt

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From bcarter at sentinel.com  Mon Jun 26 21:19:30 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 26 Jun 2006 20:19:30 -0500
Subject: [cisco-voip] Basic VOIP Setup Needs
Message-ID: <6A0744639B86DD459433108DBF101D26FC6709@MAIL01.sentinel.com>

Sorry about the Avaya thing.  Is this related to Messenger Library?



________________________________

	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
	Sent: Monday, June 26, 2006 10:30 AM
	To: Dixon, Wayne
	Cc: Voll, Scott; cisco-voip at puck.nether.net
	Subject: Re: [cisco-voip] Basic VOIP Setup Needs
	
	
	Ah!~ scratch my suggestion about the SRND for callmanager then
:) It might still have some tidbits on QoS and switches in there but its
mostly relating to callmanager.
	
	
	On 6/26/06, Dixon, Wayne <wcdixo at aurora.lib.il.us> wrote: 

		The Switches are 3500XLs... It is an Avaya system, and
call managing will be handled by that and forgive my ignorance, but what
is A VGW?  Voice Gateway?
		 
		If it is a Voice Gateway, then we're still awaiting that
to come in, we just ordered that last week.. but I was asked to see what
would be needed to get the network ready for IP Phones
		
		 
		Wayne
		 
		

			-----Original Message-----
			From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
			Sent: Monday, June 26, 2006 9:50 AM
			To: Dixon, Wayne; cisco-voip at puck.nether.net
			Subject: RE: [cisco-voip] Basic VOIP Setup Needs
			
			
			  

			What kind of Switch?

			 

			Do you already have CM and VGW's setup?

			 

			Scott

			 

			
________________________________


			From: cisco-voip-bounces at puck.nether.net
[mailto: cisco-voip-bounces at puck.nether.net
<mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Dixon, Wayne
			Sent: Monday, June 26, 2006 7:24 AM
			To: cisco-voip at puck.nether.net
			Subject: [cisco-voip] Basic VOIP Setup Needs

			 

			Hi all, 

			 

			I'm wondering what basics are needed for
allowing IP Phones access on a Cisco Switch network...   We are looking
at implementing some IP Phones.  

			 

			I'm assuming that a Dedicated VLAN for the IP
Phones, use of Native VLAN for non-tagged packets and a DHCP server are
needed... but I'm not sure what else is needed.  

			 

			If anybody has a good source for some basic
information on basic VOIP setup would be greatly appreciated.

			 

			Thanks

			 

			Wayne

			 


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




	-- 
	Ed Leatherman
	IP Telephony Coordinator
	West Virginia University
	Telecommunications and Network Operations 



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From nathan.a.reeves at gmail.com  Mon Jun 26 21:22:47 2006
From: nathan.a.reeves at gmail.com (Nathan Reeves)
Date: Tue, 27 Jun 2006 09:22:47 +0800
Subject: [cisco-voip] MGCP on CCM 4.2
Message-ID: <62833d8a0606261822r18de266bm42142eb14d39c22a@mail.gmail.com>

Do you have a loopback plug / active circuit on the BRI?  I know I have a
router here which I was setting up and I found the gateway wouldnt register
with CCM until the BRI was active.  Either an E1 loopback or the actual live
circuit connected will allow the gateway to register.

Cheers

Nathan
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From bcarter at sentinel.com  Mon Jun 26 21:23:37 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Mon, 26 Jun 2006 20:23:37 -0500
Subject: [cisco-voip] Nokia E61 CCM Client
Message-ID: <6A0744639B86DD459433108DBF101D26FC670A@MAIL01.sentinel.com>

I think I saw this mentioned here.  Configure CCM with a 7960 with the
MAC of your phone.

-----Original Message-----
From: IT [mailto:it at cimgroup.com] 
Sent: Friday, June 23, 2006 9:28 PM
To: Carter, Bill; IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Nokia E61 CCM Client

Where did you see this? Do I need to load specific software on the
phone, or is the built-in SIP client enough?


-----Original Message-----
From: Carter, Bill [mailto:bcarter at sentinel.com]
Sent: Wednesday, June 21, 2006 3:25 PM
To: IT; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Nokia E61 CCM Client

I think I saw you need to configure the phone in CallManager as a 7960. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of IT
Sent: Wednesday, June 21, 2006 5:20 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Nokia E61 CCM Client

Hello,

I have the Nokia E61 (amazing little device), and I am dying to try the
CallManager Wifi integration. Does anyone know when Cisco will be
releasing the client? Or is this a message better suited for the Nokia
mailing list?

Here is the Nokia page:
http://www.businesssoftware.nokia.com/cisco_systems.php

Here is the Cisco page:
http://www.cisco.com/iam/unified/ipcc1/IP_Communications.htm


Anyone know if I can beta-test this?

Thanks,
Avidan

_______________________________________________
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.


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 
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From nathan.a.reeves at gmail.com  Mon Jun 26 22:03:43 2006
From: nathan.a.reeves at gmail.com (Nathan Reeves)
Date: Tue, 27 Jun 2006 10:03:43 +0800
Subject: [cisco-voip] Weird behaviour after a reboot
Message-ID: <62833d8a0606261903q760c5fb4r85af3305567fedad@mail.gmail.com>

Saw an issues with our CCM yesterday which I'd be interested to hear if
anyone else has seen similar.

Had a bugcheck reboot on the SUB early Sunday morning.  Everything failed
back to the SUB once it came back up, but most of the AC Pilot Points were
not working.  Was seeing CPU spikes on both the SUB and PUB.  Rebooted the
PUB and things went back to normal on that server.   Was still seeing CPU
spikes on the SUB (mostly caused by CTIManager.exe and CCM.exe).

Tried to get the AC Pilot Points working during the day with no luck.
Completed a SUB reboot in the evening and when the server came back up, none
of the devices would fail back.  Saw errors in the event log to do with
CTIManager, Cisco Extended Functions and RIS Data Collector all terminating
unexpectedly.  Completed a second reboot and this time things came up
normally, and devices failed back to the SUB.

I'm trying to work out why I would see services fail like this on one
restart, yet work normally on the next restart.  Our only recent change has
been the installation of 4.1.3 Sr3a from 4.1.3 Sr2.  Anyone ever seen this
kind of issue with restarts causing CCM to not come up cleanly?

Thanks

Nathan
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From tednugent69 at yahoo.com  Mon Jun 26 22:18:56 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Mon, 26 Jun 2006 19:18:56 -0700 (PDT)
Subject: [cisco-voip] Weird behaviour after a reboot
In-Reply-To: <62833d8a0606261903q760c5fb4r85af3305567fedad@mail.gmail.com>
Message-ID: <20060627021856.10215.qmail@web36111.mail.mud.yahoo.com>

I've seen similar issues in the past with 4.1(3) at
multiple customer sites, both with OS patches and CM
patches. I was actually able to reproduce it in the
lab on both 4.13sr1 and sr2. When the problem occured
I was able to see in DBLHelper that replication was
actually broken at the time. For whatever reason
simply rebooting the PUB resolved the problem in all
cases. We investigated this with TAC and got no
solution except "well if that works then after patches
reboot the PUB" outsourced TAC at its best and low
BINGO. At any rate we added this to our upgrade
procedure and it appears to be the key.
Patch PUB (reboot)
Patch SUBs and other server per the upgrade guides
(reboot)
and lastly reboot the PUB again.




--- Nathan Reeves <nathan.a.reeves at gmail.com> wrote:

> Saw an issues with our CCM yesterday which I'd be
> interested to hear if
> anyone else has seen similar.
> 
> Had a bugcheck reboot on the SUB early Sunday
> morning.  Everything failed
> back to the SUB once it came back up, but most of
> the AC Pilot Points were
> not working.  Was seeing CPU spikes on both the SUB
> and PUB.  Rebooted the
> PUB and things went back to normal on that server.  
> Was still seeing CPU
> spikes on the SUB (mostly caused by CTIManager.exe
> and CCM.exe).
> 
> Tried to get the AC Pilot Points working during the
> day with no luck.
> Completed a SUB reboot in the evening and when the
> server came back up, none
> of the devices would fail back.  Saw errors in the
> event log to do with
> CTIManager, Cisco Extended Functions and RIS Data
> Collector all terminating
> unexpectedly.  Completed a second reboot and this
> time things came up
> normally, and devices failed back to the SUB.
> 
> I'm trying to work out why I would see services fail
> like this on one
> restart, yet work normally on the next restart.  Our
> only recent change has
> been the installation of 4.1.3 Sr3a from 4.1.3 Sr2. 
> Anyone ever seen this
> kind of issue with restarts causing CCM to not come
> up cleanly?
> 
> Thanks
> 
> Nathan
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From lelio at uoguelph.ca  Mon Jun 26 22:21:16 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Mon, 26 Jun 2006 22:21:16 -0400
Subject: [cisco-voip] Weird behaviour after a reboot
References: <20060627021856.10215.qmail@web36111.mail.mud.yahoo.com>
Message-ID: <008301c69990$60496180$ca01a8c0@dellectable>

We've got into the habbit of a cluster reboot after any patches. Takes a 
while, but I'm sure it's saved us from calls to the TAC which say 'reboot 
please'.

--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Network 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: "Ted Nugent" <tednugent69 at yahoo.com>
To: "Nathan Reeves" <nathan.a.reeves at gmail.com>; 
<cisco-voip at puck.nether.net>
Sent: Monday, June 26, 2006 10:18 PM
Subject: Re: [cisco-voip] Weird behaviour after a reboot


> I've seen similar issues in the past with 4.1(3) at
> multiple customer sites, both with OS patches and CM
> patches. I was actually able to reproduce it in the
> lab on both 4.13sr1 and sr2. When the problem occured
> I was able to see in DBLHelper that replication was
> actually broken at the time. For whatever reason
> simply rebooting the PUB resolved the problem in all
> cases. We investigated this with TAC and got no
> solution except "well if that works then after patches
> reboot the PUB" outsourced TAC at its best and low
> BINGO. At any rate we added this to our upgrade
> procedure and it appears to be the key.
> Patch PUB (reboot)
> Patch SUBs and other server per the upgrade guides
> (reboot)
> and lastly reboot the PUB again.
>
>
>
>
> --- Nathan Reeves <nathan.a.reeves at gmail.com> wrote:
>
>> Saw an issues with our CCM yesterday which I'd be
>> interested to hear if
>> anyone else has seen similar.
>>
>> Had a bugcheck reboot on the SUB early Sunday
>> morning.  Everything failed
>> back to the SUB once it came back up, but most of
>> the AC Pilot Points were
>> not working.  Was seeing CPU spikes on both the SUB
>> and PUB.  Rebooted the
>> PUB and things went back to normal on that server.
>> Was still seeing CPU
>> spikes on the SUB (mostly caused by CTIManager.exe
>> and CCM.exe).
>>
>> Tried to get the AC Pilot Points working during the
>> day with no luck.
>> Completed a SUB reboot in the evening and when the
>> server came back up, none
>> of the devices would fail back.  Saw errors in the
>> event log to do with
>> CTIManager, Cisco Extended Functions and RIS Data
>> Collector all terminating
>> unexpectedly.  Completed a second reboot and this
>> time things came up
>> normally, and devices failed back to the SUB.
>>
>> I'm trying to work out why I would see services fail
>> like this on one
>> restart, yet work normally on the next restart.  Our
>> only recent change has
>> been the installation of 4.1.3 Sr3a from 4.1.3 Sr2.
>> Anyone ever seen this
>> kind of issue with restarts causing CCM to not come
>> up cleanly?
>>
>> Thanks
>>
>> Nathan
>> > _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From BillS at tns.its.psu.edu  Mon Jun 26 23:12:59 2006
From: BillS at tns.its.psu.edu (Simon, Bill)
Date: Mon, 26 Jun 2006 23:12:59 -0400
Subject: [cisco-voip] Weird behaviour after a reboot
Message-ID: <07A14D6987A5F647B2415D72EDF63B56037A8227@tnsex01.tns.its.psu.edu>

I hate this kind of stuff.  It's "the Microsoft solution."  Something not
working?  "Reboot."  

I'll accept that--what choice do I have?--but I'd like to know why.  


-----Original Message-----
From: Lelio Fulgenzi
To: Ted Nugent; Nathan Reeves; cisco-voip at puck.nether.net
Sent: 6/26/2006 10:21 PM
Subject: Re: [cisco-voip] Weird behaviour after a reboot

We've got into the habbit of a cluster reboot after any patches. Takes a

while, but I'm sure it's saved us from calls to the TAC which say
'reboot 
please'.

------------------------------------------------------------------------
--------
Lelio Fulgenzi, B.A.
Network 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: "Ted Nugent" <tednugent69 at yahoo.com>
To: "Nathan Reeves" <nathan.a.reeves at gmail.com>; 
<cisco-voip at puck.nether.net>
Sent: Monday, June 26, 2006 10:18 PM
Subject: Re: [cisco-voip] Weird behaviour after a reboot


> I've seen similar issues in the past with 4.1(3) at
> multiple customer sites, both with OS patches and CM
> patches. I was actually able to reproduce it in the
> lab on both 4.13sr1 and sr2. When the problem occured
> I was able to see in DBLHelper that replication was
> actually broken at the time. For whatever reason
> simply rebooting the PUB resolved the problem in all
> cases. We investigated this with TAC and got no
> solution except "well if that works then after patches
> reboot the PUB" outsourced TAC at its best and low
> BINGO. At any rate we added this to our upgrade
> procedure and it appears to be the key.
> Patch PUB (reboot)
> Patch SUBs and other server per the upgrade guides
> (reboot)
> and lastly reboot the PUB again.
>
>
>
>
> --- Nathan Reeves <nathan.a.reeves at gmail.com> wrote:
>
>> Saw an issues with our CCM yesterday which I'd be
>> interested to hear if
>> anyone else has seen similar.
>>
>> Had a bugcheck reboot on the SUB early Sunday
>> morning.  Everything failed
>> back to the SUB once it came back up, but most of
>> the AC Pilot Points were
>> not working.  Was seeing CPU spikes on both the SUB
>> and PUB.  Rebooted the
>> PUB and things went back to normal on that server.
>> Was still seeing CPU
>> spikes on the SUB (mostly caused by CTIManager.exe
>> and CCM.exe).
>>
>> Tried to get the AC Pilot Points working during the
>> day with no luck.
>> Completed a SUB reboot in the evening and when the
>> server came back up, none
>> of the devices would fail back.  Saw errors in the
>> event log to do with
>> CTIManager, Cisco Extended Functions and RIS Data
>> Collector all terminating
>> unexpectedly.  Completed a second reboot and this
>> time things came up
>> normally, and devices failed back to the SUB.
>>
>> I'm trying to work out why I would see services fail
>> like this on one
>> restart, yet work normally on the next restart.  Our
>> only recent change has
>> been the installation of 4.1.3 Sr3a from 4.1.3 Sr2.
>> Anyone ever seen this
>> kind of issue with restarts causing CCM to not come
>> up cleanly?
>>
>> Thanks
>>
>> Nathan
>> > _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

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

From sodonnell at CCSINET.com  Mon Jun 26 23:41:51 2006
From: sodonnell at CCSINET.com (Scott O'Donnell)
Date: Mon, 26 Jun 2006 23:41:51 -0400
Subject: [cisco-voip] Interfacing to Bogen Paging system.
Message-ID: <17A170DC8DD5CC44BA56B24B366EB60B020045E2@E3K-VS1.ccsi.local>

I'm attempting to connect a voice gateway to a Bogen Paging system.
The unit I'm wiring to is a Bogen WMT-1a
(http://www.twacomm.com/catalog/model_WMT1A.htm)
 
I've tried to connecting it via an FXO and FXS interface without
success.
When it's wired to the FXS, the voice port shows "off-hook" as soon as I
jack it into the gateway.
 
Can anyone help me out with this? I know I'm missing a fundamental
understanding of how this needs to work and would very much like to
know.
 
Thanks
Scott
 
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From sodonnell at CCSINET.com  Mon Jun 26 23:48:36 2006
From: sodonnell at CCSINET.com (Scott O'Donnell)
Date: Mon, 26 Jun 2006 23:48:36 -0400
Subject: [cisco-voip] Help with CUE Editor.
Message-ID: <17A170DC8DD5CC44BA56B24B366EB60B020045E3@E3K-VS1.ccsi.local>

I need to create a CUE script that provides an autoattendant.
The prompt needs to 6 dept options (1-6) but also allow the caller to
input the extension of a person directly if they know it.
The extension numbers start with 1xxx and 2xxx.
If I use a simple menu option in the script, I can't provide a way to
allow the user to dial a full extension.
I'm attempting to use the get string method and then use a switch to
figure out if the caller pressed a dept number or extension but I can't
get it to work.
Does anyone have a script I can look at that does this?
 
 
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From SPackett at fenwick.com  Tue Jun 27 04:14:24 2006
From: SPackett at fenwick.com (Stu Packett)
Date: Tue, 27 Jun 2006 01:14:24 -0700
Subject: [cisco-voip] CIPTUG Registration Open
Message-ID: <965C0255FD5828438E34F9A68ECA76E7778DE8@svcexch02.firm.fenwick.llp>

I attended my first CIPTUG last year and it was great.  Like I told the
others on this list, I wish I had attended these before.  The workshops
were excellent and a few agendas on product roadmaps if you're
interested in finding out what's on the plate from Cisco.

There was a lot of vendor representation and I'm sure it's going to be
bigger this year as attendance has grown every year.  I highly suggest
you attend if you're on this list.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
lelio at uoguelph.ca
Sent: Saturday, June 24, 2006 12:48 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CIPTUG Registration Open

Just a brief reminder to those on the list that annual CIPTUG meeting is
coming
up soon. You have to be a CIPTUG member to join but even with the
membership
fee and the conference fee together, I think it's a good deal. Take a
look at
www.ciptug.org for more details.

A few of us on the list got together, albeit briefly, early on in the
conference, which made it even better. Of course Scott, et al, might
disagree
that meeting me face to face could be categorized as 'better'. ;)

I also believe that they are continuing the practice that if you present
a
session, your conference fees are waived.

Whether you're a beginner or a seasoned veteran, it's an excellent
conference to
attend.

Lelio


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


From 2005xj at gmail.com  Tue Jun 27 04:33:02 2006
From: 2005xj at gmail.com (jun xie)
Date: Tue, 27 Jun 2006 16:33:02 +0800
Subject: [cisco-voip] tcl2.0 script
Message-ID: <9d859ea60606270133q7c72bbf3hcbc061d08a4986bb@mail.gmail.com>

hi,all
when i write the tcl scrip on my AS5350 whose ios version is 12.3(11)T8 ,i
want to judge if is the public phone access,that i want to get the
information tag 'Leg_oli'.but failed,the TCL IVR 2.0 programing
guide.pdfsaid  this tag(leg_oli) have been removed since
12.3.100.
so how should i do then,do they have the replacment tag to get the ani II
digits?
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From PEKOU at gr.ibm.com  Tue Jun 27 04:51:40 2006
From: PEKOU at gr.ibm.com (Evi Pekou)
Date: Tue, 27 Jun 2006 11:51:40 +0300
Subject: [cisco-voip] VG224 questions
Message-ID: <OF987C3137.20490186-ONC225719A.002F2FD1-C225719A.00307292@gr.ibm.com>

Hello,

I have the following setup PSTN->2821->VG224->Fax Machines with VG224 
configured for MGCP (IOS 12.4-2.T3, CCM 4.1.3)
and fax pass-through.
Users are constantly complaining about problems sending and receiving 
faxes.
I have noticed that the FXS ports on the VG224 are off-hook, while the fax 
 machines connected to these port are idle
and the RTMT shows that the FXS ports are idle. With a port reset the 
problem is solved.

Can anyone please suggest an IOS version to use for VG224?
Is SCCP configuration preffered over MGCP? What about fax pass-through 
versus fax-relay?

Any comments would be appreciated.

Regards,
Evi
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From isinkaye at yahoo.com  Tue Jun 27 06:17:48 2006
From: isinkaye at yahoo.com (olubunmi)
Date: Tue, 27 Jun 2006 03:17:48 -0700 (PDT)
Subject: [cisco-voip] best books/ resources  for  Cisco IP services
In-Reply-To: <mailman.11722.1151380243.1478.cisco-voip@puck.nether.net>
Message-ID: <20060627101748.6234.qmail@web30508.mail.mud.yahoo.com>

folks ..
  looking for resources to help develop ip services application in -house. I understand Citrix has a solution ..has any one with experiene on these ?
   
  Thanks 
   
  Bunmi

  

 		
---------------------------------
Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates.
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From pwalenta at wi.rr.com  Tue Jun 27 07:21:29 2006
From: pwalenta at wi.rr.com (pwalenta at wi.rr.com)
Date: Tue, 27 Jun 2006 06:21:29 -0500
Subject: [cisco-voip] Interfacing to Bogen Paging system.
In-Reply-To: <17A170DC8DD5CC44BA56B24B366EB60B020045E2@E3K-VS1.ccsi.local>
References: <17A170DC8DD5CC44BA56B24B366EB60B020045E2@E3K-VS1.ccsi.local>
Message-ID: <c6b4d65c1daeb.1daebc6b4d65c@rdc-kc.rr.com>

Cisco FXO/FXS ports cannot interface directly with a pure 600ohm audio
connection.

You need a Bogen TAMB module (I've used them many times).  It takes an
analog phone port and converts it to a 600ohm output for the device you
already have.

http://www.bogen.com/products/telephonepaging/

----- Original Message -----
From: Scott O'Donnell <sodonnell at CCSINET.com>
Date: Monday, June 26, 2006 10:44 pm
Subject: [cisco-voip] Interfacing to Bogen Paging system.
To: cisco-voip at puck.nether.net

> I'm attempting to connect a voice gateway to a Bogen Paging system.
> The unit I'm wiring to is a Bogen WMT-1a
> (http://www.twacomm.com/catalog/model_WMT1A.htm)
> 
> I've tried to connecting it via an FXO and FXS interface without
> success.
> When it's wired to the FXS, the voice port shows "off-hook" as soon 
> as I
> jack it into the gateway.
> 
> Can anyone help me out with this? I know I'm missing a fundamental
> understanding of how this needs to work and would very much like to
> know.
> 
> Thanks
> Scott
> 
> 

From pwalenta at wi.rr.com  Tue Jun 27 07:23:31 2006
From: pwalenta at wi.rr.com (pwalenta at wi.rr.com)
Date: Tue, 27 Jun 2006 06:23:31 -0500
Subject: [cisco-voip] best books/ resources  for  Cisco IP services
In-Reply-To: <20060627101748.6234.qmail@web30508.mail.mud.yahoo.com>
References: <mailman.11722.1151380243.1478.cisco-voip@puck.nether.net>
	<20060627101748.6234.qmail@web30508.mail.mud.yahoo.com>
Message-ID: <c5d5e2831bac5.1bac5c5d5e283@rdc-kc.rr.com>

If you're referring to IP Phone services, this is a reasonable book to read:

http://www.ciscopress.com/title/1587050609

----- Original Message -----
From: olubunmi <isinkaye at yahoo.com>
Date: Tuesday, June 27, 2006 5:20 am
Subject: [cisco-voip] best books/ resources  for  Cisco IP services
To: cisco-voip at puck.nether.net

> folks ..
>  looking for resources to help develop ip services application in -
> house. I understand Citrix has a solution ..has any one with 
> experiene on these ?
>   
>  Thanks 
>   
>  Bunmi
> 
>  
> 
>         	
> ---------------------------------
> Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates.

From pwalenta at wi.rr.com  Tue Jun 27 07:38:25 2006
From: pwalenta at wi.rr.com (pwalenta at wi.rr.com)
Date: Tue, 27 Jun 2006 06:38:25 -0500
Subject: [cisco-voip] Basic VOIP Setup Needs
In-Reply-To: <6A0744639B86DD459433108DBF101D26FC6709@MAIL01.sentinel.com>
References: <6A0744639B86DD459433108DBF101D26FC6709@MAIL01.sentinel.com>
Message-ID: <c5aeb7ab1ca2d.1ca2dc5aeb7ab@rdc-kc.rr.com>

It's been a while since I've done Avaya, but I do remember a few things.

The phones support 802.1q trunking/VLAN setup.

They require some configuration in DHCP to work.

Complete documentation on how to setup an Avaya phone can be found here:

http://support.avaya.com/elmodocs2/ip_phones/r2_4/output/index.html

That link can answer all of your questions in detail.  Avaya used to
have a configuration paper on how to do this, but I cannot seem to
locate it now.

----- Original Message -----
From: "Carter, Bill" <bcarter at sentinel.com>
Date: Monday, June 26, 2006 8:20 pm
Subject: Re: [cisco-voip] Basic VOIP Setup Needs
To: "Dixon, Wayne" <wcdixo at aurora.lib.il.us>
Cc: cisco-voip at puck.nether.net

> Sorry about the Avaya thing.  Is this related to Messenger Library?
> 
> 
> 
> ________________________________
> 
> 	From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ed Leatherman
> 	Sent: Monday, June 26, 2006 10:30 AM
> 	To: Dixon, Wayne
> 	Cc: Voll, Scott; cisco-voip at puck.nether.net
> 	Subject: Re: [cisco-voip] Basic VOIP Setup Needs
> 	
> 	
> 	Ah!~ scratch my suggestion about the SRND for callmanager then
> :) It might still have some tidbits on QoS and switches in there 
> but its
> mostly relating to callmanager.
> 	
> 	
> 	On 6/26/06, Dixon, Wayne <wcdixo at aurora.lib.il.us> wrote: 
> 
>        	The Switches are 3500XLs... It is an Avaya system, and
> call managing will be handled by that and forgive my ignorance, but 
> whatis A VGW?  Voice Gateway?
>                 
>        	If it is a Voice Gateway, then we're still awaiting that
> to come in, we just ordered that last week.. but I was asked to see 
> whatwould be needed to get the network ready for IP Phones
>        	
>                 
>        	Wayne
>                 
>        	
> 
>                	-----Original Message-----
>                	From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
>                	Sent: Monday, June 26, 2006 9:50 AM
>                	To: Dixon, Wayne; cisco-voip at puck.nether.net
>                	Subject: RE: [cisco-voip] Basic VOIP Setup Needs
>                	
>                	
>                          
> 
>                	What kind of Switch?
> 
>                         
> 
>                	Do you already have CM and VGW's setup?
> 
>                         
> 
>                	Scott
> 
>                         
> 
>                	
> ________________________________
> 
> 
>                	From: cisco-voip-bounces at puck.nether.net
> [mailto: cisco-voip-bounces at puck.nether.net
> <mailto:cisco-voip-bounces at puck.nether.net> ] On Behalf Of Dixon, 
> Wayne                	Sent: Monday, June 26, 2006 7:24 AM
>                	To: cisco-voip at puck.nether.net
>                	Subject: [cisco-voip] Basic VOIP Setup Needs
> 
>                         
> 
>                	Hi all, 
> 
>                         
> 
>                	I'm wondering what basics are needed for
> allowing IP Phones access on a Cisco Switch network...   We are 
> lookingat implementing some IP Phones.  
> 
>                         
> 
>                	I'm assuming that a Dedicated VLAN for the IP
> Phones, use of Native VLAN for non-tagged packets and a DHCP server 
> areneeded... but I'm not sure what else is needed.  
> 
>                         
> 
>                	If anybody has a good source for some basic
> information on basic VOIP setup would be greatly appreciated.
> 
>                         
> 
>                	Thanks
> 
>                         
> 
>                	Wayne
> 
>                         
> 
> 
>        	_______________________________________________
>        	cisco-voip mailing list
>        	cisco-voip at puck.nether.net
>        	https://puck.nether.net/mailman/listinfo/cisco-voip
>        	
>        	
>        	
> 
> 
> 
> 
> 	-- 
> 	Ed Leatherman
> 	IP Telephony Coordinator
> 	West Virginia University
> 	Telecommunications and Network Operations 
> 
> 
> 
> 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 Matt.Slaga at us.didata.com  Tue Jun 27 08:43:48 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Tue, 27 Jun 2006 08:43:48 -0400
Subject: [cisco-voip] best books/ resources  for  Cisco IP services
Message-ID: <C1FE15183DA37645BC0633BC604E44F0024FAC3E@USNAEXCH.na.didata.local>

If you are looking to create simple services for your company, you
should look into Aptigen.  They offer a drag-and-drop IP phone service
development environment.  You can create and deliver an application to
your users in less than 4 hours.  They have a few flavors, and
definitely worth checking out. 

Metreos offers something similar as well, except they offer a much
deeper integration with jtapi, whereas Aptigen offers text/database
manipulation.

Thanks,
Matt

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
pwalenta at wi.rr.com
Sent: Tuesday, June 27, 2006 7:24 AM
To: olubunmi
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] best books/ resources for Cisco IP services

If you're referring to IP Phone services, this is a reasonable book to
read:

http://www.ciscopress.com/title/1587050609

----- Original Message -----
From: olubunmi <isinkaye at yahoo.com>
Date: Tuesday, June 27, 2006 5:20 am
Subject: [cisco-voip] best books/ resources  for  Cisco IP services
To: cisco-voip at puck.nether.net

> folks ..
>  looking for resources to help develop ip services application in - 
> house. I understand Citrix has a solution ..has any one with experiene

> on these ?
>   
>  Thanks
>   
>  Bunmi
> 
>  
> 
>         	
> ---------------------------------
> Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low
rates.
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

-----------------------------------------
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From jgrace at digitelusa.net  Tue Jun 27 09:37:03 2006
From: jgrace at digitelusa.net (James Grace)
Date: Tue, 27 Jun 2006 09:37:03 -0400
Subject: [cisco-voip] how to modify the layout in crs 4.0.3
Message-ID: <200606270949640.SM01452@3729434770>

Below is how I configured crs 3.5.3 to display on the default layout pop
screen, the name of the Que that's was called

 

Add a field  called display and give it a index number 

Then goto the layout (default) and add the field to the layout

Go down to the workflow 

And make sure the show enterprise and allow agent to edit enterprise

On the script 

After the select resource set 

Change it to select and add the set data type

In the set data type tie the enterprise index and variable together

The variable is a string with the string in ""  "test"

Add the connect step under the set data type 

 

This does not work in 4.0.3.  can some one walk me thru some steps to make
this happen.

Right now on my pop screen for ip phone agents.   I get all the standard
info anis dnis.  I  also have the field name showing up on the layout but
the var string is not getting into the layout 

 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From akoul at dataways.gr  Tue Jun 27 10:25:49 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Tue, 27 Jun 2006 17:25:49 +0300
Subject: [cisco-voip] Link to primary CCM fails while logged in with
	Extension Mobility
Message-ID: <44A13FED.8000001@dataways.gr>

Hi all,
if a user is logged in with extension mobility, and the link to the 
primary ccm fails what happens with that phone?



From SechabaM at kss.co.za  Tue Jun 27 10:32:20 2006
From: SechabaM at kss.co.za (Sechaba Mokoena)
Date: Tue, 27 Jun 2006 16:32:20 +0200
Subject: [cisco-voip]  Failed Cellphone calls across MPLS network
Message-ID: <3E110434E2419D47883D119DD226ACA105C50A@kss-jhb-exch-01.kss.co.za>

Hi,
 
I'm having trouble dialing cellphone numbers from an IP phone at a remote branch. Basically I have three sites which I will refer to as JHB,DBN, and CPT. My CallManager is at the JHB site with a 2821 router using an E1 for PSTN access. All sites are connected to each other through an MPLS network. The phones at the DBN and CPT sites register with the CallManager at the JHB site. All landline calls made from the DBN and CPT sites breakout via BRIs on 2821 routers located at each site. My client wants all cellphone calls made at the DBN and CPT sites to break out on the JHB 2821 router to make use of Leased Cost Routing. I have created specific route patterns for this requirement directing all cellphone calls to the gateway in JHB. When placing a call to a cellphone, it rings once and the call is dropped. I ran isdn q931 debugs on the JHB router and found this: Cause i = 0x80AF - Resource unavailable, unspecified. Cellphone calls made from JHB go through without any problems, does anyone have any suggestions?
 
Regards,
 
Sechaba Mokoena

This message, including any attachments is intended only for
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contain privileged or confidential information. Retransmission, 
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From jonvoip at gmail.com  Tue Jun 27 10:39:35 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 27 Jun 2006 09:39:35 -0500
Subject: [cisco-voip] 7914s not displaying line state on shared lines
In-Reply-To: <5d093f9a0606261530u319ac552s620af09dffda2775@mail.gmail.com>
References: <5d093f9a0606260709o7972a50aye09ec06fbfbd7743@mail.gmail.com>
	<20060626210340.75328.qmail@web37206.mail.mud.yahoo.com>
	<5d093f9a0606261530u319ac552s620af09dffda2775@mail.gmail.com>
Message-ID: <5d093f9a0606270739v2dad77d5xe50ef44cb35af7d6@mail.gmail.com>

OK, no phones on the network are showing line state... I have set
privacy to false on both PUB and SUB, enabled the bridge, installed
extended services, updated the firmware to 8.0(3)...

I am completely out of ideas...


Jonathan

On 6/26/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> Bite me
>
> So, when you quitting?
>
> On 6/26/06, Erick Bergquist <erickbe at yahoo.com> wrote:
> > IW
> >
> > ----- Original Message ----
> > From: Jonathan Charles <jonvoip at gmail.com>
> > To: Lelio Fulgenzi <lelio at uoguelph.ca>
> > Cc: ciscovoip <cisco-voip at puck.nether.net>
> > Sent: Monday, June 26, 2006 9:09:25 AM
> > Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
> >
> > The thing is, I have downloaded and installed the latest firmware and
> > have confirmed the 7914s are running it.
> >
> > Here's the worst part... the 7971s don't even show correct line state
> > on shared lines... and they are running the latest code...
> >
> > I don't get it... I am going to try a different firmware first...
> >
> > Jonathan
> >
> > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > >
> > > I think this definately requires a TAC case.
> > >
> > > --------------------------------------------------------------------------------
> > > 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: Jonathan Charles
> > > To: Lelio Fulgenzi
> > > Cc: ciscovoip
> > > Sent: Monday, June 26, 2006 9:54 AM
> > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
> > >
> > > Dozens of times at this point...
> > >
> > > I am checking the traces now...
> > >
> > >
> > >
> > > Jonathan
> > >
> > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > >
> > > > ..and you've restarted the phones?
> > > >
> > > > that's just bizzare.
> > > >
> > > >
> > > --------------------------------------------------------------------------------
> > > > 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: Jonathan Charles
> > > > To: Lelio Fulgenzi
> > > > Cc: Aman Chugh ; ciscovoip
> > > > Sent: Monday, June 26, 2006 9:46 AM
> > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
> > > >
> > > > The bridge is enabled globally and manually on both devices
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > > >
> > > > > I'm pretty sure that the Builtin Bridge Enable parameter has to be
> > > > enabled,
> > > > > either in the system parameters or on the phone configuration itself.
> > > > After
> > > > > any of these changes (privacy setting included) the phone has to be
> > > > > reset/restarted or something like that.
> > > > >
> > > > >
> > > >
> > > --------------------------------------------------------------------------------
> > > > > 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: Jonathan Charles
> > > > > To: Aman Chugh
> > > > > Cc: ciscovoip
> > > > > Sent: Monday, June 26, 2006 9:26 AM
> > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
> > > lines
> > > > >
> > > > > I checked and globally Privacy is set to false, also, I tried it on
> > > > > some other phones and NONE of them show the shared line, line state...
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > > > On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> > > > > >
> > > > > > What is privacy set to .check the privacy settings
> > > > > >
> > > > > > -- Aman
> > > > > >
> > > > > >
> > > > > > On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > > > > >
> > > > > > We have CallManager 4.1(3)sr3b with OS4.2sr8...
> > > > > >
> > > > > > Receptionist has a 7971 with two 7914 expansion modules.
> > > > > >
> > > > > > Created shared lines of the executives on the 7914s.
> > > > > >
> > > > > > We see them ring when you call them, but we don't see any line state
> > > > > > info on the 7914s (it always shows the phones on hook, no matter what
> > > > > > they are doing, unless they are ringing).
> > > > > >
> > > > > > We are running the latest DevPack, version 28-2 and the 7914s have
> > > > > > picked up the latest load.
> > > > > >
> > > > > > What am I missing?
> > > > > >
> > > > > >
> > > > > >
> > > > > > Jonathan
> > > > > > _______________________________________________
> > > > > > cisco-voip mailing list
> > > > > > cisco-voip at puck.nether.net
> > > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > >
> > > > > >
> > > > > _______________________________________________
> > > > > cisco-voip mailing list
> > > > > cisco-voip at puck.nether.net
> > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > >
> > > >
> > > >
> > >
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> >
> >
>

From jsteinberg at gmail.com  Tue Jun 27 10:47:36 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Tue, 27 Jun 2006 10:47:36 -0400
Subject: [cisco-voip] Failed Cellphone calls across MPLS network
In-Reply-To: <3E110434E2419D47883D119DD226ACA105C50A@kss-jhb-exch-01.kss.co.za>
References: <3E110434E2419D47883D119DD226ACA105C50A@kss-jhb-exch-01.kss.co.za>
Message-ID: <f1b65f880606270747k2ccd110erdcc37ac313eb2a10@mail.gmail.com>

Sechaba,

It is possible that your E1 provider at JHB is rejecting the calls
from DBN & CPT because those calls are sending ANI's from DBN & CPT
that are not part of your E1 trunk group.

To test, on your RPs that send cell calls from DBN & CPT to JHB you
could have a calling party transform mask of a DID at JHB to see if
this is the cause.

Justin



On 6/27/06, Sechaba Mokoena <SechabaM at kss.co.za> wrote:
>
> Hi,
>
> I'm having trouble dialing cellphone numbers from an IP phone at a remote
> branch. Basically I have three sites which I will refer to as JHB,DBN, and
> CPT. My CallManager is at the JHB site with a 2821 router using an E1 for
> PSTN access. All sites are connected to each other through an MPLS network.
> The phones at the DBN and CPT sites register with the CallManager at the JHB
> site. All landline calls made from the DBN and CPT sites breakout via BRIs
> on 2821 routers located at each site. My client wants all cellphone calls
> made at the DBN and CPT sites to break out on the JHB 2821 router to make
> use of Leased Cost Routing. I have created specific route patterns for this
> requirement directing all cellphone calls to the gateway in JHB. When
> placing a call to a cellphone, it rings once and the call is dropped. I ran
> isdn q931 debugs on the JHB router and found this: Cause i = 0x80AF -
> Resource unavailable, unspecified. Cellphone calls made from JHB go through
> without any problems, does anyone have any suggestions?
>
> Regards,
>
> Sechaba Mokoena
> ________________________________
> This message, including any attachments is intended only for the persons or
> entity to which it is addressed, and may contain privileged or confidential
> information. Retransmission, dissemination, copying, distribution,
> modification and / or publication of this message by a person other than the
> intended recipient is strictly prohibited by the sender. If you have
> received this message in error, or you feel it is unacceptable
> communication, a chain letter or offensive material of any nature, please
> delete the material and any copies from all systems and report it to:
> info at kss.co.za
> ________________________________
>
> _______________________________________________
> 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 Jun 27 10:55:08 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 27 Jun 2006 07:55:08 -0700
Subject: [cisco-voip] CCM download
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED52E3@Apollo.wesd.org>

Call Manager is not on the CCO to download.  Only updates.  You will
need to order. Or PUT it.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Curt Shaffer
Sent: Monday, June 26, 2006 3:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CCM download

 

I heard a while back that you can download Cisco Call Manger for free
but I can't find it. I really dislike the navigation on their site
sometimes. Can anyone point me in the right direction?

 

 

Thanks

 

Curt

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From lelio at uoguelph.ca  Tue Jun 27 10:56:45 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 27 Jun 2006 10:56:45 -0400
Subject: [cisco-voip] 7914s not displaying line state on shared lines
References: <5d093f9a0606260709o7972a50aye09ec06fbfbd7743@mail.gmail.com><20060626210340.75328.qmail@web37206.mail.mud.yahoo.com><5d093f9a0606261530u319ac552s620af09dffda2775@mail.gmail.com>
	<5d093f9a0606270739v2dad77d5xe50ef44cb35af7d6@mail.gmail.com>
Message-ID: <004201c699f9$ea121d80$47196883@cfs.uoguelph.ca>

And we've made sure the lines are in the same partition? ;)

Try creating a couple of new phones from scratch with a shared line, other than that, my next step would be to restart the cluster.

Then TAC it.

--------------------------------------------------------------------------------
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: Jonathan Charles 
  To: Erick Bergquist 
  Cc: ciscovoip 
  Sent: Tuesday, June 27, 2006 10:39 AM
  Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines


  OK, no phones on the network are showing line state... I have set
  privacy to false on both PUB and SUB, enabled the bridge, installed
  extended services, updated the firmware to 8.0(3)...

  I am completely out of ideas...


  Jonathan

  On 6/26/06, Jonathan Charles <jonvoip at gmail.com> wrote:
  > Bite me
  >
  > So, when you quitting?
  >
  > On 6/26/06, Erick Bergquist <erickbe at yahoo.com> wrote:
  > > IW
  > >
  > > ----- Original Message ----
  > > From: Jonathan Charles <jonvoip at gmail.com>
  > > To: Lelio Fulgenzi <lelio at uoguelph.ca>
  > > Cc: ciscovoip <cisco-voip at puck.nether.net>
  > > Sent: Monday, June 26, 2006 9:09:25 AM
  > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
  > >
  > > The thing is, I have downloaded and installed the latest firmware and
  > > have confirmed the 7914s are running it.
  > >
  > > Here's the worst part... the 7971s don't even show correct line state
  > > on shared lines... and they are running the latest code...
  > >
  > > I don't get it... I am going to try a different firmware first...
  > >
  > > Jonathan
  > >
  > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  > > >
  > > > I think this definately requires a TAC case.
  > > >
  > > > --------------------------------------------------------------------------------
  > > > 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: Jonathan Charles
  > > > To: Lelio Fulgenzi
  > > > Cc: ciscovoip
  > > > Sent: Monday, June 26, 2006 9:54 AM
  > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
  > > >
  > > > Dozens of times at this point...
  > > >
  > > > I am checking the traces now...
  > > >
  > > >
  > > >
  > > > Jonathan
  > > >
  > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  > > > >
  > > > > ..and you've restarted the phones?
  > > > >
  > > > > that's just bizzare.
  > > > >
  > > > >
  > > > --------------------------------------------------------------------------------
  > > > > 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: Jonathan Charles
  > > > > To: Lelio Fulgenzi
  > > > > Cc: Aman Chugh ; ciscovoip
  > > > > Sent: Monday, June 26, 2006 9:46 AM
  > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
  > > > >
  > > > > The bridge is enabled globally and manually on both devices
  > > > >
  > > > >
  > > > > Jonathan
  > > > >
  > > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  > > > > >
  > > > > > I'm pretty sure that the Builtin Bridge Enable parameter has to be
  > > > > enabled,
  > > > > > either in the system parameters or on the phone configuration itself.
  > > > > After
  > > > > > any of these changes (privacy setting included) the phone has to be
  > > > > > reset/restarted or something like that.
  > > > > >
  > > > > >
  > > > >
  > > > --------------------------------------------------------------------------------
  > > > > > 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: Jonathan Charles
  > > > > > To: Aman Chugh
  > > > > > Cc: ciscovoip
  > > > > > Sent: Monday, June 26, 2006 9:26 AM
  > > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
  > > > lines
  > > > > >
  > > > > > I checked and globally Privacy is set to false, also, I tried it on
  > > > > > some other phones and NONE of them show the shared line, line state...
  > > > > >
  > > > > >
  > > > > > Jonathan
  > > > > >
  > > > > > On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
  > > > > > >
  > > > > > > What is privacy set to .check the privacy settings
  > > > > > >
  > > > > > > -- Aman
  > > > > > >
  > > > > > >
  > > > > > > On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
  > > > > > > >
  > > > > > > We have CallManager 4.1(3)sr3b with OS4.2sr8...
  > > > > > >
  > > > > > > Receptionist has a 7971 with two 7914 expansion modules.
  > > > > > >
  > > > > > > Created shared lines of the executives on the 7914s.
  > > > > > >
  > > > > > > We see them ring when you call them, but we don't see any line state
  > > > > > > info on the 7914s (it always shows the phones on hook, no matter what
  > > > > > > they are doing, unless they are ringing).
  > > > > > >
  > > > > > > We are running the latest DevPack, version 28-2 and the 7914s have
  > > > > > > picked up the latest load.
  > > > > > >
  > > > > > > What am I missing?
  > > > > > >
  > > > > > >
  > > > > > >
  > > > > > > Jonathan
  > > > > > > _______________________________________________
  > > > > > > cisco-voip mailing list
  > > > > > > cisco-voip at puck.nether.net
  > > > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
  > > > > > >
  > > > > > >
  > > > > > _______________________________________________
  > > > > > cisco-voip mailing list
  > > > > > cisco-voip at puck.nether.net
  > > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
  > > > > >
  > > > >
  > > > >
  > > >
  > > >
  > > _______________________________________________
  > > 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 aman.chugh at gmail.com  Tue Jun 27 10:59:09 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Tue, 27 Jun 2006 20:29:09 +0530
Subject: [cisco-voip] VG224 questions
In-Reply-To: <OF987C3137.20490186-ONC225719A.002F2FD1-C225719A.00307292@gr.ibm.com>
References: <OF987C3137.20490186-ONC225719A.002F2FD1-C225719A.00307292@gr.ibm.com>
Message-ID: <d4c697ad0606270759q759f8b45p9946dc24c9752945@mail.gmail.com>

Check out CSCse13526 ,it gave us a lot of pain to get this bug openned with
Cisco.but this is definetly something which you could be hitting in
4.1.3,fix was rolled out in es
58.Also do you have FXS ports unregistering a lot.

-- Aman


On 6/27/06, Evi Pekou <PEKOU at gr.ibm.com> wrote:
>
>
> Hello,
>
> I have the following setup PSTN->2821->VG224->Fax Machines with VG224
> configured for MGCP (IOS 12.4-2.T3, CCM 4.1.3)
> and fax pass-through.
> Users are constantly complaining about problems sending and receiving
> faxes.
> I have noticed that the FXS ports on the VG224 are off-hook, while the fax
>  machines connected to these port are idle
> and the RTMT shows that the FXS ports are idle. With a port reset the
> problem is solved.
>
> Can anyone please suggest an IOS version to use for VG224?
> Is SCCP configuration preffered over MGCP? What about fax pass-through
> versus fax-relay?
>
> Any comments would be appreciated.
>
> Regards,
>
> Evi
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From Scott.Voll at wesd.org  Tue Jun 27 11:01:48 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 27 Jun 2006 08:01:48 -0700
Subject: [cisco-voip] Lab requirements
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED52E6@Apollo.wesd.org>

If you can get a little VMWare server to run CM I think you would be
better off for the CCVP.  CIPT test is very in depth into the CM
application.  If you are just doing a test lab with a couple FXO ports
those routers should work fine I think.  You will need IP Plus IOS and
DSP resources because neither come stock with them.  Don't for get a
little switch and phones.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Curt Shaffer
Sent: Monday, June 26, 2006 5:59 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Lab requirements

 

I am looking into setting up a lab for VoIP testing and practice for the
CCVP. I found what I think is a good deal on a 1720 with 32/16 memory
and a 2610 with 64/16 memory. Can anyone out there tell me if this will
suffice for a small test environment? I am planning on getting some VICs
as well for practice on configuring those interfaces as well. I just
want to make sure I buy the right tool for the job. Also with all of
this in mind would I be better off with Call Manager Express to put on
the routers (if they will even run them) or would it be wise to just get
the full thing. I guess what would be best is just some suggestions from
everyone out there who has labs on what they are using and what
works/worked well for them.

 

Thanks

 

Curt

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From Scott.Voll at wesd.org  Tue Jun 27 11:05:12 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 27 Jun 2006 08:05:12 -0700
Subject: [cisco-voip] Weird behaviour after a reboot
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED52E7@Apollo.wesd.org>

What OS?  4.2sr7 had a bug where CCM caused 100% CPU utilization.  SR7a
fixed it.

 

scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nathan Reeves
Sent: Monday, June 26, 2006 7:04 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Weird behaviour after a reboot

 

Saw an issues with our CCM yesterday which I'd be interested to hear if
anyone else has seen similar.

 

Had a bugcheck reboot on the SUB early Sunday morning.  Everything
failed back to the SUB once it came back up, but most of the AC Pilot
Points were not working.  Was seeing CPU spikes on both the SUB and PUB.
Rebooted the PUB and things went back to normal on that server.   Was
still seeing CPU spikes on the SUB (mostly caused by CTIManager.exe and
CCM.exe).

 

Tried to get the AC Pilot Points working during the day with no luck.
Completed a SUB reboot in the evening and when the server came back up,
none of the devices would fail back.  Saw errors in the event log to do
with CTIManager, Cisco Extended Functions and RIS Data Collector all
terminating unexpectedly.  Completed a second reboot and this time
things came up normally, and devices failed back to the SUB. 

 

I'm trying to work out why I would see services fail like this on one
restart, yet work normally on the next restart.  Our only recent change
has been the installation of 4.1.3 Sr3a from 4.1.3 Sr2.  Anyone ever
seen this kind of issue with restarts causing CCM to not come up
cleanly? 

 

Thanks

 

Nathan 

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From rratliff at cisco.com  Tue Jun 27 11:04:54 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 27 Jun 2006 11:04:54 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in with
	Extension Mobility
In-Reply-To: <44A13FED.8000001@dataways.gr>
References: <44A13FED.8000001@dataways.gr>
Message-ID: <23DB8A63-22DE-41CC-B9E5-5ED326B68DE5@cisco.com>

The phone will act just like it would if you were not logged in with  
EM, ie it will register with the backup CallManager server.

-Ryan

On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou wrote:

Hi all,
if a user is logged in with extension mobility, and the link to the
primary ccm fails what happens with that phone?


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


From aman.chugh at gmail.com  Tue Jun 27 11:06:42 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Tue, 27 Jun 2006 20:36:42 +0530
Subject: [cisco-voip] 7914s not displaying line state on shared lines
In-Reply-To: <5d093f9a0606270739v2dad77d5xe50ef44cb35af7d6@mail.gmail.com>
References: <5d093f9a0606260709o7972a50aye09ec06fbfbd7743@mail.gmail.com>
	<20060626210340.75328.qmail@web37206.mail.mud.yahoo.com>
	<5d093f9a0606261530u319ac552s620af09dffda2775@mail.gmail.com>
	<5d093f9a0606270739v2dad77d5xe50ef44cb35af7d6@mail.gmail.com>
Message-ID: <d4c697ad0606270806h4515b0ccw9d06ba9792d6798d@mail.gmail.com>

lets a get a detailed ccm trace from the callmanager server with which
phones are registered.

On 6/27/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> OK, no phones on the network are showing line state... I have set
> privacy to false on both PUB and SUB, enabled the bridge, installed
> extended services, updated the firmware to 8.0(3)...
>
> I am completely out of ideas...
>
>
> Jonathan
>
> On 6/26/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > Bite me
> >
> > So, when you quitting?
> >
> > On 6/26/06, Erick Bergquist <erickbe at yahoo.com> wrote:
> > > IW
> > >
> > > ----- Original Message ----
> > > From: Jonathan Charles <jonvoip at gmail.com>
> > > To: Lelio Fulgenzi <lelio at uoguelph.ca>
> > > Cc: ciscovoip <cisco-voip at puck.nether.net>
> > > Sent: Monday, June 26, 2006 9:09:25 AM
> > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
> lines
> > >
> > > The thing is, I have downloaded and installed the latest firmware and
> > > have confirmed the 7914s are running it.
> > >
> > > Here's the worst part... the 7971s don't even show correct line state
> > > on shared lines... and they are running the latest code...
> > >
> > > I don't get it... I am going to try a different firmware first...
> > >
> > > Jonathan
> > >
> > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > >
> > > > I think this definately requires a TAC case.
> > > >
> > > >
> --------------------------------------------------------------------------------
> > > > 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: Jonathan Charles
> > > > To: Lelio Fulgenzi
> > > > Cc: ciscovoip
> > > > Sent: Monday, June 26, 2006 9:54 AM
> > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
> lines
> > > >
> > > > Dozens of times at this point...
> > > >
> > > > I am checking the traces now...
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > > >
> > > > > ..and you've restarted the phones?
> > > > >
> > > > > that's just bizzare.
> > > > >
> > > > >
> > > >
> --------------------------------------------------------------------------------
> > > > > 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: Jonathan Charles
> > > > > To: Lelio Fulgenzi
> > > > > Cc: Aman Chugh ; ciscovoip
> > > > > Sent: Monday, June 26, 2006 9:46 AM
> > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on
> shared lines
> > > > >
> > > > > The bridge is enabled globally and manually on both devices
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > > > >
> > > > > > I'm pretty sure that the Builtin Bridge Enable parameter has to
> be
> > > > > enabled,
> > > > > > either in the system parameters or on the phone configuration
> itself.
> > > > > After
> > > > > > any of these changes (privacy setting included) the phone has to
> be
> > > > > > reset/restarted or something like that.
> > > > > >
> > > > > >
> > > > >
> > > >
> --------------------------------------------------------------------------------
> > > > > > 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: Jonathan Charles
> > > > > > To: Aman Chugh
> > > > > > Cc: ciscovoip
> > > > > > Sent: Monday, June 26, 2006 9:26 AM
> > > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on
> shared
> > > > lines
> > > > > >
> > > > > > I checked and globally Privacy is set to false, also, I tried it
> on
> > > > > > some other phones and NONE of them show the shared line, line
> state...
> > > > > >
> > > > > >
> > > > > > Jonathan
> > > > > >
> > > > > > On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> > > > > > >
> > > > > > > What is privacy set to .check the privacy settings
> > > > > > >
> > > > > > > -- Aman
> > > > > > >
> > > > > > >
> > > > > > > On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > > > > > >
> > > > > > > We have CallManager 4.1(3)sr3b with OS4.2sr8...
> > > > > > >
> > > > > > > Receptionist has a 7971 with two 7914 expansion modules.
> > > > > > >
> > > > > > > Created shared lines of the executives on the 7914s.
> > > > > > >
> > > > > > > We see them ring when you call them, but we don't see any line
> state
> > > > > > > info on the 7914s (it always shows the phones on hook, no
> matter what
> > > > > > > they are doing, unless they are ringing).
> > > > > > >
> > > > > > > We are running the latest DevPack, version 28-2 and the 7914s
> have
> > > > > > > picked up the latest load.
> > > > > > >
> > > > > > > What am I missing?
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > Jonathan
> > > > > > > _______________________________________________
> > > > > > > cisco-voip mailing list
> > > > > > > cisco-voip at puck.nether.net
> > > > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > >
> > > > > > >
> > > > > > _______________________________________________
> > > > > > cisco-voip mailing list
> > > > > > cisco-voip at puck.nether.net
> > > > > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > >
> > > > >
> > > > >
> > > >
> > > >
> > > _______________________________________________
> > > 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  Tue Jun 27 11:10:29 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 27 Jun 2006 08:10:29 -0700
Subject: [cisco-voip] Failed Cellphone calls across MPLS network
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED52EC@Apollo.wesd.org>

Could it be a resource (transcoding) / or codec problem?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Sechaba Mokoena
Sent: Tuesday, June 27, 2006 7:32 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Failed Cellphone calls across MPLS network

 

Hi,

 

I'm having trouble dialing cellphone numbers from an IP phone at a
remote branch. Basically I have three sites which I will refer to as
JHB,DBN, and CPT. My CallManager is at the JHB site with a 2821 router
using an E1 for PSTN access. All sites are connected to each other
through an MPLS network. The phones at the DBN and CPT sites register
with the CallManager at the JHB site. All landline calls made from the
DBN and CPT sites breakout via BRIs on 2821 routers located at each
site. My client wants all cellphone calls made at the DBN and CPT sites
to break out on the JHB 2821 router to make use of Leased Cost Routing.
I have created specific route patterns for this requirement directing
all cellphone calls to the gateway in JHB. When placing a call to a
cellphone, it rings once and the call is dropped. I ran isdn q931 debugs
on the JHB router and found this: Cause i = 0x80AF - Resource
unavailable, unspecified. Cellphone calls made from JHB go through
without any problems, does anyone have any suggestions?

 

Regards,

 

Sechaba Mokoena

________________________________

This message, including any attachments is intended only for the persons
or entity to which it is addressed, and may contain privileged or
confidential information. Retransmission, dissemination, copying,
distribution, modification and / or publication of this message by a
person other than the intended recipient is strictly prohibited by the
sender. If you have received this message in error, or you feel it is
unacceptable communication, a chain letter or offensive material of any
nature, please delete the material and any copies from all systems and
report it to:

info at kss.co.za

________________________________

 

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From jonvoip at gmail.com  Tue Jun 27 11:11:14 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Tue, 27 Jun 2006 10:11:14 -0500
Subject: [cisco-voip] 7914s not displaying line state on shared lines
In-Reply-To: <004201c699f9$ea121d80$47196883@cfs.uoguelph.ca>
References: <5d093f9a0606260709o7972a50aye09ec06fbfbd7743@mail.gmail.com>
	<20060626210340.75328.qmail@web37206.mail.mud.yahoo.com>
	<5d093f9a0606261530u319ac552s620af09dffda2775@mail.gmail.com>
	<5d093f9a0606270739v2dad77d5xe50ef44cb35af7d6@mail.gmail.com>
	<004201c699f9$ea121d80$47196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0606270811v43a39f9dlbb73a049453755ef@mail.gmail.com>

Yeah, I just figured that out...

Apparently, I didn't even think to check that...

I made an inbound call, and I noticed that both lines didn't ring. Now
they do....

This is fixed.

Thanks to all for the help.



Jonathan

On 6/27/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>
> And we've made sure the lines are in the same partition? ;)
>
> Try creating a couple of new phones from scratch with a shared line, other
> than that, my next step would be to restart the cluster.
>
> Then TAC it.
>
>
> --------------------------------------------------------------------------------
> 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: Jonathan Charles
>
> To: Erick Bergquist
> Cc: ciscovoip
> Sent: Tuesday, June 27, 2006 10:39 AM
> Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
>
> OK, no phones on the network are showing line state... I have set
> privacy to false on both PUB and SUB, enabled the bridge, installed
> extended services, updated the firmware to 8.0(3)...
>
> I am completely out of ideas...
>
>
> Jonathan
>
> On 6/26/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > Bite me
> >
> > So, when you quitting?
> >
> > On 6/26/06, Erick Bergquist <erickbe at yahoo.com> wrote:
> > > IW
> > >
> > > ----- Original Message ----
> > > From: Jonathan Charles <jonvoip at gmail.com>
> > > To: Lelio Fulgenzi <lelio at uoguelph.ca>
> > > Cc: ciscovoip <cisco-voip at puck.nether.net>
> > > Sent: Monday, June 26, 2006 9:09:25 AM
> > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
> lines
> > >
> > > The thing is, I have downloaded and installed the latest firmware and
> > > have confirmed the 7914s are running it.
> > >
> > > Here's the worst part... the 7971s don't even show correct line state
> > > on shared lines... and they are running the latest code...
> > >
> > > I don't get it... I am going to try a different firmware first...
> > >
> > > Jonathan
> > >
> > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > >
> > > > I think this definately requires a TAC case.
> > > >
> > > >
> --------------------------------------------------------------------------------
> > > > 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: Jonathan Charles
> > > > To: Lelio Fulgenzi
> > > > Cc: ciscovoip
> > > > Sent: Monday, June 26, 2006 9:54 AM
> > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
> lines
> > > >
> > > > Dozens of times at this point...
> > > >
> > > > I am checking the traces now...
> > > >
> > > >
> > > >
> > > > Jonathan
> > > >
> > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > > >
> > > > > ..and you've restarted the phones?
> > > > >
> > > > > that's just bizzare.
> > > > >
> > > > >
> > > >
> --------------------------------------------------------------------------------
> > > > > 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: Jonathan Charles
> > > > > To: Lelio Fulgenzi
> > > > > Cc: Aman Chugh ; ciscovoip
> > > > > Sent: Monday, June 26, 2006 9:46 AM
> > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
> lines
> > > > >
> > > > > The bridge is enabled globally and manually on both devices
> > > > >
> > > > >
> > > > > Jonathan
> > > > >
> > > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > > > > >
> > > > > > I'm pretty sure that the Builtin Bridge Enable parameter has to be
> > > > > enabled,
> > > > > > either in the system parameters or on the phone configuration
> itself.
> > > > > After
> > > > > > any of these changes (privacy setting included) the phone has to
> be
> > > > > > reset/restarted or something like that.
> > > > > >
> > > > > >
> > > > >
> > > >
> --------------------------------------------------------------------------------
> > > > > > 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: Jonathan Charles
> > > > > > To: Aman Chugh
> > > > > > Cc: ciscovoip
> > > > > > Sent: Monday, June 26, 2006 9:26 AM
> > > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on
> shared
> > > > lines
> > > > > >
> > > > > > I checked and globally Privacy is set to false, also, I tried it
> on
> > > > > > some other phones and NONE of them show the shared line, line
> state...
> > > > > >
> > > > > >
> > > > > > Jonathan
> > > > > >
> > > > > > On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
> > > > > > >
> > > > > > > What is privacy set to .check the privacy settings
> > > > > > >
> > > > > > > -- Aman
> > > > > > >
> > > > > > >
> > > > > > > On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > > > > > > >
> > > > > > > We have CallManager 4.1(3)sr3b with OS4.2sr8...
> > > > > > >
> > > > > > > Receptionist has a 7971 with two 7914 expansion modules.
> > > > > > >
> > > > > > > Created shared lines of the executives on the 7914s.
> > > > > > >
> > > > > > > We see them ring when you call them, but we don't see any line
> state
> > > > > > > info on the 7914s (it always shows the phones on hook, no matter
> what
> > > > > > > they are doing, unless they are ringing).
> > > > > > >
> > > > > > > We are running the latest DevPack, version 28-2 and the 7914s
> have
> > > > > > > picked up the latest load.
> > > > > > >
> > > > > > > What am I missing?
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > > Jonathan
> > > > > > > _______________________________________________
> > > > > > > cisco-voip mailing list
> > > > > > > cisco-voip at puck.nether.net
> > > > > > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > > >
> > > > > > >
> > > > > > _______________________________________________
> > > > > > cisco-voip mailing list
> > > > > > cisco-voip at puck.nether.net
> > > > > >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > > > > >
> > > > >
> > > > >
> > > >
> > > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >
> > >
> > >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From lelio at uoguelph.ca  Tue Jun 27 11:14:51 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 27 Jun 2006 11:14:51 -0400
Subject: [cisco-voip] 7914s not displaying line state on shared lines
References: <5d093f9a0606260709o7972a50aye09ec06fbfbd7743@mail.gmail.com>
	<20060626210340.75328.qmail@web37206.mail.mud.yahoo.com>
	<5d093f9a0606261530u319ac552s620af09dffda2775@mail.gmail.com>
	<5d093f9a0606270739v2dad77d5xe50ef44cb35af7d6@mail.gmail.com>
	<004201c699f9$ea121d80$47196883@cfs.uoguelph.ca>
	<5d093f9a0606270811v43a39f9dlbb73a049453755ef@mail.gmail.com>
Message-ID: <00a701c699fc$7016f980$47196883@cfs.uoguelph.ca>

...and the case of the mysterious shared line is solved!

--------------------------------------------------------------------------------
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: Jonathan Charles 
  To: Lelio Fulgenzi 
  Cc: Erick Bergquist ; ciscovoip 
  Sent: Tuesday, June 27, 2006 11:11 AM
  Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines


  Yeah, I just figured that out...

  Apparently, I didn't even think to check that...

  I made an inbound call, and I noticed that both lines didn't ring. Now
  they do....

  This is fixed.

  Thanks to all for the help.



  Jonathan

  On 6/27/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  >
  >
  > And we've made sure the lines are in the same partition? ;)
  >
  > Try creating a couple of new phones from scratch with a shared line, other
  > than that, my next step would be to restart the cluster.
  >
  > Then TAC it.
  >
  >
  > --------------------------------------------------------------------------------
  > 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: Jonathan Charles
  >
  > To: Erick Bergquist
  > Cc: ciscovoip
  > Sent: Tuesday, June 27, 2006 10:39 AM
  > Subject: Re: [cisco-voip] 7914s not displaying line state on shared lines
  >
  > OK, no phones on the network are showing line state... I have set
  > privacy to false on both PUB and SUB, enabled the bridge, installed
  > extended services, updated the firmware to 8.0(3)...
  >
  > I am completely out of ideas...
  >
  >
  > Jonathan
  >
  > On 6/26/06, Jonathan Charles <jonvoip at gmail.com> wrote:
  > > Bite me
  > >
  > > So, when you quitting?
  > >
  > > On 6/26/06, Erick Bergquist <erickbe at yahoo.com> wrote:
  > > > IW
  > > >
  > > > ----- Original Message ----
  > > > From: Jonathan Charles <jonvoip at gmail.com>
  > > > To: Lelio Fulgenzi <lelio at uoguelph.ca>
  > > > Cc: ciscovoip <cisco-voip at puck.nether.net>
  > > > Sent: Monday, June 26, 2006 9:09:25 AM
  > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
  > lines
  > > >
  > > > The thing is, I have downloaded and installed the latest firmware and
  > > > have confirmed the 7914s are running it.
  > > >
  > > > Here's the worst part... the 7971s don't even show correct line state
  > > > on shared lines... and they are running the latest code...
  > > >
  > > > I don't get it... I am going to try a different firmware first...
  > > >
  > > > Jonathan
  > > >
  > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  > > > >
  > > > > I think this definately requires a TAC case.
  > > > >
  > > > >
  > --------------------------------------------------------------------------------
  > > > > 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: Jonathan Charles
  > > > > To: Lelio Fulgenzi
  > > > > Cc: ciscovoip
  > > > > Sent: Monday, June 26, 2006 9:54 AM
  > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
  > lines
  > > > >
  > > > > Dozens of times at this point...
  > > > >
  > > > > I am checking the traces now...
  > > > >
  > > > >
  > > > >
  > > > > Jonathan
  > > > >
  > > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  > > > > >
  > > > > > ..and you've restarted the phones?
  > > > > >
  > > > > > that's just bizzare.
  > > > > >
  > > > > >
  > > > >
  > --------------------------------------------------------------------------------
  > > > > > 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: Jonathan Charles
  > > > > > To: Lelio Fulgenzi
  > > > > > Cc: Aman Chugh ; ciscovoip
  > > > > > Sent: Monday, June 26, 2006 9:46 AM
  > > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on shared
  > lines
  > > > > >
  > > > > > The bridge is enabled globally and manually on both devices
  > > > > >
  > > > > >
  > > > > > Jonathan
  > > > > >
  > > > > > On 6/26/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  > > > > > >
  > > > > > > I'm pretty sure that the Builtin Bridge Enable parameter has to be
  > > > > > enabled,
  > > > > > > either in the system parameters or on the phone configuration
  > itself.
  > > > > > After
  > > > > > > any of these changes (privacy setting included) the phone has to
  > be
  > > > > > > reset/restarted or something like that.
  > > > > > >
  > > > > > >
  > > > > >
  > > > >
  > --------------------------------------------------------------------------------
  > > > > > > 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: Jonathan Charles
  > > > > > > To: Aman Chugh
  > > > > > > Cc: ciscovoip
  > > > > > > Sent: Monday, June 26, 2006 9:26 AM
  > > > > > > Subject: Re: [cisco-voip] 7914s not displaying line state on
  > shared
  > > > > lines
  > > > > > >
  > > > > > > I checked and globally Privacy is set to false, also, I tried it
  > on
  > > > > > > some other phones and NONE of them show the shared line, line
  > state...
  > > > > > >
  > > > > > >
  > > > > > > Jonathan
  > > > > > >
  > > > > > > On 6/25/06, Aman Chugh <aman.chugh at gmail.com> wrote:
  > > > > > > >
  > > > > > > > What is privacy set to .check the privacy settings
  > > > > > > >
  > > > > > > > -- Aman
  > > > > > > >
  > > > > > > >
  > > > > > > > On 6/25/06, Jonathan Charles <jonvoip at gmail.com> wrote:
  > > > > > > > >
  > > > > > > > We have CallManager 4.1(3)sr3b with OS4.2sr8...
  > > > > > > >
  > > > > > > > Receptionist has a 7971 with two 7914 expansion modules.
  > > > > > > >
  > > > > > > > Created shared lines of the executives on the 7914s.
  > > > > > > >
  > > > > > > > We see them ring when you call them, but we don't see any line
  > state
  > > > > > > > info on the 7914s (it always shows the phones on hook, no matter
  > what
  > > > > > > > they are doing, unless they are ringing).
  > > > > > > >
  > > > > > > > We are running the latest DevPack, version 28-2 and the 7914s
  > have
  > > > > > > > picked up the latest load.
  > > > > > > >
  > > > > > > > What am I missing?
  > > > > > > >
  > > > > > > >
  > > > > > > >
  > > > > > > > Jonathan
  > > > > > > > _______________________________________________
  > > > > > > > cisco-voip mailing list
  > > > > > > > cisco-voip at puck.nether.net
  > > > > > > >
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  > > > > > > >
  > > > > > > >
  > > > > > > _______________________________________________
  > > > > > > cisco-voip mailing list
  > > > > > > cisco-voip at puck.nether.net
  > > > > > >
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  > > > > > >
  > > > > >
  > > > > >
  > > > >
  > > > >
  > > > _______________________________________________
  > > > cisco-voip mailing list
  > > > cisco-voip at puck.nether.net
  > > > https://puck.nether.net/mailman/listinfo/cisco-voip
  > > >
  > > >
  > > >
  > > >
  > > >
  > >
  > _______________________________________________
  > cisco-voip mailing list
  > cisco-voip at puck.nether.net
  > https://puck.nether.net/mailman/listinfo/cisco-voip
  >
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From rratliff at cisco.com  Tue Jun 27 11:33:15 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 27 Jun 2006 11:33:15 -0400
Subject: [cisco-voip] Failed Cellphone calls across MPLS network
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED52EC@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A605ED52EC@Apollo.wesd.org>
Message-ID: <31962C31-8A52-475E-B629-44B76C1C5F03@cisco.com>

Depends on who is sending the disconnect with cause 0x80AF.  If CM is  
sending this then it is going to be a codec mismatch.    The fact  
that it rings once before disconnecting probably means that it fails  
when the cell phone answers the call.

-Ryan

On Jun 27, 2006, at 11:10 AM, Voll, Scott wrote:

Could it be a resource (transcoding) / or codec problem?



From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
bounces at puck.nether.net] On Behalf Of Sechaba Mokoena
Sent: Tuesday, June 27, 2006 7:32 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Failed Cellphone calls across MPLS network



Hi,



I'm having trouble dialing cellphone numbers from an IP phone at a  
remote branch. Basically I have three sites which I will refer to as  
JHB,DBN, and CPT. My CallManager is at the JHB site with a 2821  
router using an E1 for PSTN access. All sites are connected to each  
other through an MPLS network. The phones at the DBN and CPT sites  
register with the CallManager at the JHB site. All landline calls  
made from the DBN and CPT sites breakout via BRIs on 2821 routers  
located at each site. My client wants all cellphone calls made at the  
DBN and CPT sites to break out on the JHB 2821 router to make use of  
Leased Cost Routing. I have created specific route patterns for this  
requirement directing all cellphone calls to the gateway in JHB. When  
placing a call to a cellphone, it rings once and the call is dropped.  
I ran isdn q931 debugs on the JHB router and found this: Cause i =  
0x80AF - Resource unavailable, unspecified. Cellphone calls made from  
JHB go through without any problems, does anyone have any suggestions?



Regards,



Sechaba Mokoena

This message, including any attachments is intended only for the  
persons or entity to which it is addressed, and may contain  
privileged or confidential information. Retransmission,  
dissemination, copying, distribution, modification and / or  
publication of this message by a person other than the intended  
recipient is strictly prohibited by the sender. If you have received  
this message in error, or you feel it is unacceptable communication,  
a chain letter or offensive material of any nature, please delete the  
material and any copies from all systems and report it to:

info at kss.co.za



_______________________________________________
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  Tue Jun 27 11:47:25 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 27 Jun 2006 11:47:25 -0400
Subject: [cisco-voip] 7941/7961 Tone Generator
Message-ID: <C1FE15183DA37645BC0633BC604E44F0024FB0FB@USNAEXCH.na.didata.local>

**3 turns on "Debug Mode" on a 7941, however I don't get a Tone
Generator softkey when placing a call.  I've done this before on
7940/7960s with no issues.

 

Anyone tried the 41/61/70 ? These phones run a Java applet...

 

 

http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09
186a00800945bd.shtml#testtone




-----------------------------------------
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 Andrius-conf at elsis.lt  Tue Jun 27 11:49:06 2006
From: Andrius-conf at elsis.lt (Andrius Kislas)
Date: Tue, 27 Jun 2006 18:49:06 +0300
Subject: [cisco-voip] Failed Cellphone calls across MPLS network
In-Reply-To: <31962C31-8A52-475E-B629-44B76C1C5F03@cisco.com>
References: <407055A92CECCB499C922A2D35FC19A605ED52EC@Apollo.wesd.org>
	<31962C31-8A52-475E-B629-44B76C1C5F03@cisco.com>
Message-ID: <44A15372.5000208@elsis.lt>

I had a situation when it rings once and disconnects when VT Advantage
was on. I had to overwrite bearer channel capability to "Speach" on
voice-port.

Andrius

Ryan Ratliff wrote:
> Depends on who is sending the disconnect with cause 0x80AF.  If CM is  
> sending this then it is going to be a codec mismatch.    The fact  
> that it rings once before disconnecting probably means that it fails  
> when the cell phone answers the call.
> 
> -Ryan
> 
> On Jun 27, 2006, at 11:10 AM, Voll, Scott wrote:
> 
> Could it be a resource (transcoding) / or codec problem?
> 
> 
> 
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip- 
> bounces at puck.nether.net] On Behalf Of Sechaba Mokoena
> Sent: Tuesday, June 27, 2006 7:32 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Failed Cellphone calls across MPLS network
> 
> 
> 
> Hi,
> 
> 
> 
> I'm having trouble dialing cellphone numbers from an IP phone at a  
> remote branch. Basically I have three sites which I will refer to as  
> JHB,DBN, and CPT. My CallManager is at the JHB site with a 2821  
> router using an E1 for PSTN access. All sites are connected to each  
> other through an MPLS network. The phones at the DBN and CPT sites  
> register with the CallManager at the JHB site. All landline calls  
> made from the DBN and CPT sites breakout via BRIs on 2821 routers  
> located at each site. My client wants all cellphone calls made at the  
> DBN and CPT sites to break out on the JHB 2821 router to make use of  
> Leased Cost Routing. I have created specific route patterns for this  
> requirement directing all cellphone calls to the gateway in JHB. When  
> placing a call to a cellphone, it rings once and the call is dropped.  
> I ran isdn q931 debugs on the JHB router and found this: Cause i =  
> 0x80AF - Resource unavailable, unspecified. Cellphone calls made from  
> JHB go through without any problems, does anyone have any suggestions?
> 
> 
> 
> Regards,
> 
> 
> 
> Sechaba Mokoena
> 
> This message, including any attachments is intended only for the  
> persons or entity to which it is addressed, and may contain  
> privileged or confidential information. Retransmission,  
> dissemination, copying, distribution, modification and / or  
> publication of this message by a person other than the intended  
> recipient is strictly prohibited by the sender. If you have received  
> this message in error, or you feel it is unacceptable communication,  
> a chain letter or offensive material of any nature, please delete the  
> material and any copies from all systems and report it to:
> 
> info at kss.co.za
> 
> 
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


From rratliff at cisco.com  Tue Jun 27 11:50:20 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 27 Jun 2006 11:50:20 -0400
Subject: [cisco-voip] Failed Cellphone calls across MPLS network
In-Reply-To: <44A15372.5000208@elsis.lt>
References: <407055A92CECCB499C922A2D35FC19A605ED52EC@Apollo.wesd.org>
	<31962C31-8A52-475E-B629-44B76C1C5F03@cisco.com>
	<44A15372.5000208@elsis.lt>
Message-ID: <16DAC03A-6203-45FE-861A-C30406E6F300@cisco.com>

That is a very common problem with people just rolling out VTA  
unfortunately.  I belive the disconnect in that case though will come  
from the Telco with a cause of "bearer-cap not implemented".

-Ryan

On Jun 27, 2006, at 11:49 AM, Andrius Kislas wrote:

I had a situation when it rings once and disconnects when VT Advantage
was on. I had to overwrite bearer channel capability to "Speach" on
voice-port.

Andrius

Ryan Ratliff wrote:
> Depends on who is sending the disconnect with cause 0x80AF.  If CM is
> sending this then it is going to be a codec mismatch.    The fact
> that it rings once before disconnecting probably means that it fails
> when the cell phone answers the call.
>
> -Ryan
>
> On Jun 27, 2006, at 11:10 AM, Voll, Scott wrote:
>
> Could it be a resource (transcoding) / or codec problem?
>
>
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Sechaba Mokoena
> Sent: Tuesday, June 27, 2006 7:32 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Failed Cellphone calls across MPLS network
>
>
>
> Hi,
>
>
>
> I'm having trouble dialing cellphone numbers from an IP phone at a
> remote branch. Basically I have three sites which I will refer to as
> JHB,DBN, and CPT. My CallManager is at the JHB site with a 2821
> router using an E1 for PSTN access. All sites are connected to each
> other through an MPLS network. The phones at the DBN and CPT sites
> register with the CallManager at the JHB site. All landline calls
> made from the DBN and CPT sites breakout via BRIs on 2821 routers
> located at each site. My client wants all cellphone calls made at the
> DBN and CPT sites to break out on the JHB 2821 router to make use of
> Leased Cost Routing. I have created specific route patterns for this
> requirement directing all cellphone calls to the gateway in JHB. When
> placing a call to a cellphone, it rings once and the call is dropped.
> I ran isdn q931 debugs on the JHB router and found this: Cause i =
> 0x80AF - Resource unavailable, unspecified. Cellphone calls made from
> JHB go through without any problems, does anyone have any suggestions?
>
>
>
> Regards,
>
>
>
> Sechaba Mokoena
>
> This message, including any attachments is intended only for the
> persons or entity to which it is addressed, and may contain
> privileged or confidential information. Retransmission,
> dissemination, copying, distribution, modification and / or
> publication of this message by a person other than the intended
> recipient is strictly prohibited by the sender. If you have received
> this message in error, or you feel it is unacceptable communication,
> a chain letter or offensive material of any nature, please delete the
> material and any copies from all systems and report it to:
>
> info at kss.co.za
>
>
>
> _______________________________________________
> 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 jason.aarons at us.didata.com  Tue Jun 27 12:03:58 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 27 Jun 2006 12:03:58 -0400
Subject: [cisco-voip] Failed Cellphone calls across MPLS network
Message-ID: <C1FE15183DA37645BC0633BC604E44F0024FB145@USNAEXCH.na.didata.local>

I noticed the bear cap had to be set to speak this with 7985s and
certain IOS.

Another workaround is setting you Gateway's Location to 0 Video causing
retry as Audio on the phone.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Tuesday, June 27, 2006 8:50 AM
To: Andrius Kislas
Cc: Sechaba Mokoena; Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Failed Cellphone calls across MPLS network

That is a very common problem with people just rolling out VTA  
unfortunately.  I belive the disconnect in that case though will come  
from the Telco with a cause of "bearer-cap not implemented".

-Ryan

On Jun 27, 2006, at 11:49 AM, Andrius Kislas wrote:

I had a situation when it rings once and disconnects when VT Advantage
was on. I had to overwrite bearer channel capability to "Speach" on
voice-port.

Andrius

Ryan Ratliff wrote:
> Depends on who is sending the disconnect with cause 0x80AF.  If CM is
> sending this then it is going to be a codec mismatch.    The fact
> that it rings once before disconnecting probably means that it fails
> when the cell phone answers the call.
>
> -Ryan
>
> On Jun 27, 2006, at 11:10 AM, Voll, Scott wrote:
>
> Could it be a resource (transcoding) / or codec problem?
>
>
>
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-
> bounces at puck.nether.net] On Behalf Of Sechaba Mokoena
> Sent: Tuesday, June 27, 2006 7:32 AM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] Failed Cellphone calls across MPLS network
>
>
>
> Hi,
>
>
>
> I'm having trouble dialing cellphone numbers from an IP phone at a
> remote branch. Basically I have three sites which I will refer to as
> JHB,DBN, and CPT. My CallManager is at the JHB site with a 2821
> router using an E1 for PSTN access. All sites are connected to each
> other through an MPLS network. The phones at the DBN and CPT sites
> register with the CallManager at the JHB site. All landline calls
> made from the DBN and CPT sites breakout via BRIs on 2821 routers
> located at each site. My client wants all cellphone calls made at the
> DBN and CPT sites to break out on the JHB 2821 router to make use of
> Leased Cost Routing. I have created specific route patterns for this
> requirement directing all cellphone calls to the gateway in JHB. When
> placing a call to a cellphone, it rings once and the call is dropped.
> I ran isdn q931 debugs on the JHB router and found this: Cause i =
> 0x80AF - Resource unavailable, unspecified. Cellphone calls made from
> JHB go through without any problems, does anyone have any suggestions?
>
>
>
> Regards,
>
>
>
> Sechaba Mokoena
>
> This message, including any attachments is intended only for the
> persons or entity to which it is addressed, and may contain
> privileged or confidential information. Retransmission,
> dissemination, copying, distribution, modification and / or
> publication of this message by a person other than the intended
> recipient is strictly prohibited by the sender. If you have received
> this message in error, or you feel it is unacceptable communication,
> a chain letter or offensive material of any nature, please delete the
> material and any copies from all systems and report it to:
>
> info at kss.co.za
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

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

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

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



From Scott.Voll at wesd.org  Tue Jun 27 12:07:17 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 27 Jun 2006 09:07:17 -0700
Subject: [cisco-voip] 7941/7961 Tone Generator
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED5302@Apollo.wesd.org>

I can't even get the tone generator working on my 40's.  What CM version
and firmware are you running?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons
(US)
Sent: Tuesday, June 27, 2006 8:47 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] 7941/7961 Tone Generator

 

**3 turns on "Debug Mode" on a 7941, however I don't get a Tone
Generator softkey when placing a call.  I've done this before on
7940/7960s with no issues.

 

Anyone tried the 41/61/70 ? These phones run a Java applet...

 

 

http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09
186a00800945bd.shtml#testtone

________________________________



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 lelio at uoguelph.ca  Tue Jun 27 12:09:26 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Tue, 27 Jun 2006 12:09:26 -0400
Subject: [cisco-voip] 7941/7961 Tone Generator
References: <407055A92CECCB499C922A2D35FC19A605ED5302@Apollo.wesd.org>
Message-ID: <015801c69a04$101ab410$47196883@cfs.uoguelph.ca>

ditto here. can't get that softkey enabled at all. i'm running 4.1.3sr3a with 7.2(4.0) on my 7960.
--------------------------------------------------------------------------------
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: Jason Aarons (US) ; cisco-voip at puck.nether.net 
  Sent: Tuesday, June 27, 2006 12:07 PM
  Subject: Re: [cisco-voip] 7941/7961 Tone Generator


  I can't even get the tone generator working on my 40's.  What CM version and firmware are you running?

   

  Scott

   


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

  From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jason Aarons (US)
  Sent: Tuesday, June 27, 2006 8:47 AM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] 7941/7961 Tone Generator

   

  **3 turns on "Debug Mode" on a 7941, however I don't get a Tone Generator softkey when placing a call.  I've done this before on 7940/7960s with no issues.

   

  Anyone tried the 41/61/70 ? These phones run a Java applet.

   

   

  http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800945bd.shtml#testtone



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





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From CORTIZ at broward.org  Tue Jun 27 12:25:42 2006
From: CORTIZ at broward.org (Ortiz, Carlos)
Date: Tue, 27 Jun 2006 12:25:42 -0400
Subject: [cisco-voip] VG224 questions
Message-ID: <26B357EF9FB7D04A989C9E82A9E5436E01BC3572@MREXVS01.bc.broward.cty>

I had some intermittent fax issues with a MGCP controlled  VG224 also. 

 

The command "no ccm-manager fax protocol cisco" seemed to helped with
the intermittent receiving fax issues

 

We upgraded the mem on the VG224 so that we could run a new IOS and
address a TACACS bug. 

 

Currently have IOS vg224-i6s-mz.124-8.bin.

 

Here is part of the config that seems to be working ok for us now.  

Any suggestions for improvements are welcome: 

 

 

version 12.4

no service pad

service timestamps debug datetime msec localtime show-timezone

service timestamps log datetime msec localtime show-timezone

service password-encryption

!

hostname !

boot-start-marker

boot system slot0:/vg224-i6s-mz.124-8.bin

boot-end-marker

!

logging buffered 4096 debugging

!

clock timezone est -5

clock summer-time edt recurring

ip cef

ip domain list 

!

voice-card 0

!

voice-port 2/0

!

voice-port 2/1

!

voice-port 2/2

!

voice-port 2/3

!

voice-port 2/4

!

voice-port 2/5

!

voice-port 2/6

!

voice-port 2/7

!

voice-port 2/8

!

voice-port 2/9

!

voice-port 2/10

!

voice-port 2/11

!

voice-port 2/12

!

voice-port 2/13

!

voice-port 2/14

!

voice-port 2/15

!

voice-port 2/16

!

voice-port 2/17

!

voice-port 2/18

!

voice-port 2/19

!

voice-port 2/20

!

voice-port 2/21

!

voice-port 2/22

!

voice-port 2/23

!

ccm-manager redundant-host 

ccm-manager mgcp

no ccm-manager fax protocol cisco

ccm-manager music-on-hold

ccm-manager config server 

ccm-manager config

!

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

no mgcp package-capability res-package

mgcp package-capability sst-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

!

mgcp profile default

!

!

!

dial-peer voice 99920 pots

 service mgcpapp

 port 2/0

!

dial-peer voice 99921 pots

 service mgcpapp

 port 2/1

!

dial-peer voice 99922 pots

 service mgcpapp

 port 2/2

!

dial-peer voice 99923 pots

 service mgcpapp

 port 2/3

!

dial-peer voice 99924 pots

 service mgcpapp

 port 2/4

!

dial-peer voice 99925 pots

 service mgcpapp

 port 2/5

!

dial-peer voice 99926 pots

 service mgcpapp

 port 2/6

!

dial-peer voice 99927 pots

 service mgcpapp

 port 2/7

!

dial-peer voice 99928 pots

 service mgcpapp

 port 2/8

!

dial-peer voice 99929 pots

 service mgcpapp

 port 2/9

!

dial-peer voice 999210 pots

 service mgcpapp

 port 2/10

!

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Aman Chugh
Sent: Tuesday, June 27, 2006 10:59 AM
To: Evi Pekou
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VG224 questions

 

Check out CSCse13526 ,it gave us a lot of pain to get this bug openned
with Cisco.but this is definetly something which you could be hitting in
4.1.3 ,fix was rolled out in es 58.Also do you have FXS ports
unregistering a lot. 

 

-- Aman

 

On 6/27/06, Evi Pekou <PEKOU at gr.ibm.com> wrote: 


Hello, 

I have the following setup PSTN->2821->VG224->Fax Machines with VG224
configured for MGCP (IOS 12.4-2.T3, CCM 4.1.3) 
and fax pass-through. 
Users are constantly complaining about problems sending and receiving
faxes. 
I have noticed that the FXS ports on the VG224 are off-hook, while the
fax  machines connected to these port are idle 
and the RTMT shows that the FXS ports are idle. With a port reset the
problem is solved. 

Can anyone please suggest an IOS version to use for VG224? 
Is SCCP configuration preffered over MGCP? What about fax pass-through
versus fax-relay? 

Any comments would be appreciated. 

Regards, 
 

Evi 


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



 

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From cenders at homesbyavi.com  Tue Jun 27 12:28:51 2006
From: cenders at homesbyavi.com (Corbett Enders)
Date: Tue, 27 Jun 2006 10:28:51 -0600
Subject: [cisco-voip] CallManager 4.2 - Anyone running it in production
References: <mailman.3.1151424005.34772.cisco-voip@puck.nether.net>
Message-ID: <24DFD2417491E547AC80AE140706C48503946490@EX001.homesbyavi.local>

The upgrade from 4.1 to 4.2 went fine last night.  Took about 4 hours to
upgrade a 2 node CM cluster (running on  fairly new and fast HP DL380s).
Had to update the OS as well which accounted for some of the time.

As for not having an upgrade path to 5.0, that is OK because from what
I've heard I see no reason to run 5.0 in production...

Apparently we were the first in Western Canada to upgrade from 4.1 to
4.2...

Corbett Enders
Business Development
Homes by Avi Inc.
Tel: (403) 536-7170
Fax: (403) 536-7171
Email: cenders at homesbyavi.com 
www.homesbyavi.com


From rleetun at co.boulder.co.us  Tue Jun 27 12:46:09 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Tue, 27 Jun 2006 10:46:09 -0600
Subject: [cisco-voip] VG224 questions
Message-ID: <963CE94787005149A52C17F33A028E12051D4921@EVS1B.boco.co.boulder.co.us>

Here is a great fax troubleshooting doc on the Cisco site:
 
http://www.cisco.com/en/US/customer/tech/tk652/tk777/technologies_tech_n
ote09186a0080114565.shtml
 
Rob

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ortiz, Carlos
Sent: Tuesday, June 27, 2006 10:26 AM
To: Aman Chugh; Evi Pekou
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VG224 questions



I had some intermittent fax issues with a MGCP controlled  VG224 also. 

 

The command "no ccm-manager fax protocol cisco" seemed to helped with
the intermittent receiving fax issues

 

We upgraded the mem on the VG224 so that we could run a new IOS and
address a TACACS bug. 

 

Currently have IOS vg224-i6s-mz.124-8.bin.

 

Here is part of the config that seems to be working ok for us now.  

Any suggestions for improvements are welcome: 

 

 

version 12.4

no service pad

service timestamps debug datetime msec localtime show-timezone

service timestamps log datetime msec localtime show-timezone

service password-encryption

!

hostname !

boot-start-marker

boot system slot0:/vg224-i6s-mz.124-8.bin

boot-end-marker

!

logging buffered 4096 debugging

!

clock timezone est -5

clock summer-time edt recurring

ip cef

ip domain list 

!

voice-card 0

!

voice-port 2/0

!

voice-port 2/1

!

voice-port 2/2

!

voice-port 2/3

!

voice-port 2/4

!

voice-port 2/5

!

voice-port 2/6

!

voice-port 2/7

!

voice-port 2/8

!

voice-port 2/9

!

voice-port 2/10

!

voice-port 2/11

!

voice-port 2/12

!

voice-port 2/13

!

voice-port 2/14

!

voice-port 2/15

!

voice-port 2/16

!

voice-port 2/17

!

voice-port 2/18

!

voice-port 2/19

!

voice-port 2/20

!

voice-port 2/21

!

voice-port 2/22

!

voice-port 2/23

!

ccm-manager redundant-host 

ccm-manager mgcp

no ccm-manager fax protocol cisco

ccm-manager music-on-hold

ccm-manager config server 

ccm-manager config

!

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

no mgcp package-capability res-package

mgcp package-capability sst-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static

!

mgcp profile default

!

!

!

dial-peer voice 99920 pots

 service mgcpapp

 port 2/0

!

dial-peer voice 99921 pots

 service mgcpapp

 port 2/1

!

dial-peer voice 99922 pots

 service mgcpapp

 port 2/2

!

dial-peer voice 99923 pots

 service mgcpapp

 port 2/3

!

dial-peer voice 99924 pots

 service mgcpapp

 port 2/4

!

dial-peer voice 99925 pots

 service mgcpapp

 port 2/5

!

dial-peer voice 99926 pots

 service mgcpapp

 port 2/6

!

dial-peer voice 99927 pots

 service mgcpapp

 port 2/7

!

dial-peer voice 99928 pots

 service mgcpapp

 port 2/8

!

dial-peer voice 99929 pots

 service mgcpapp

 port 2/9

!

dial-peer voice 999210 pots

 service mgcpapp

 port 2/10

!

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Aman Chugh
Sent: Tuesday, June 27, 2006 10:59 AM
To: Evi Pekou
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] VG224 questions

 

Check out CSCse13526 ,it gave us a lot of pain to get this bug openned
with Cisco.but this is definetly something which you could be hitting in
4.1.3 ,fix was rolled out in es 58.Also do you have FXS ports
unregistering a lot. 

 

-- Aman

 

On 6/27/06, Evi Pekou <PEKOU at gr.ibm.com> wrote: 


Hello, 

I have the following setup PSTN->2821->VG224->Fax Machines with VG224
configured for MGCP (IOS 12.4-2.T3, CCM 4.1.3) 
and fax pass-through. 
Users are constantly complaining about problems sending and receiving
faxes. 
I have noticed that the FXS ports on the VG224 are off-hook, while the
fax  machines connected to these port are idle 
and the RTMT shows that the FXS ports are idle. With a port reset the
problem is solved. 

Can anyone please suggest an IOS version to use for VG224? 
Is SCCP configuration preffered over MGCP? What about fax pass-through
versus fax-relay? 

Any comments would be appreciated. 

Regards, 
 

Evi 


_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
<http://portal.mxlogic.com/redir/?atXKnsvshKrjLtYQsFIcTpdAVPmEBCbdSaY3iv
NU6U9GX33VkDa3JsJaBGBPdpb6XZuZXFFCzCX39KVKQGmGncRAIn8lrxrW0FpKNnwqj-f0QS
MyUzsSyy-yMC--ed79JBcS2_id41FrJaBGBPdpb6BQQg1A99FI6y0mr-AVlxgTfM-u0USyrh
dFTpopvjuvhdVXcO88yu>  



 

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From cisco-voip at ibh.net  Tue Jun 27 13:00:05 2006
From: cisco-voip at ibh.net (Andre Beck)
Date: Tue, 27 Jun 2006 19:00:05 +0200
Subject: [cisco-voip] What causes a Err-disable?
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED4F3C@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A605ED4F3C@Apollo.wesd.org>
Message-ID: <20060627170005.GA13130@ibh.de>

Scott,

On Wed, Jun 21, 2006 at 01:48:13PM -0700, Voll, Scott wrote:
> What causes a port on a 3560 to go into err-disable?  I know a shut no
> shut fixes it but how can we fix it to not do it at all?

Just to add to the list of possible reasons for a port to go err-disabled:
Cisco IP Phones, when left unpowered and plugged into a non-PoE Switch,
tend to get the port into loopback err-disable instantly. This is probably
due to the old proprietary PoE circuitry beeing interpreted as a loop by
the switch.

-- 
                  The _S_anta _C_laus _O_peration
  or "how to turn a complete illusion into a neverending money source"

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From wsisk at cisco.com  Tue Jun 27 13:08:18 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 27 Jun 2006 13:08:18 -0400
Subject: [cisco-voip] CallManager DR scenario
In-Reply-To: <965C0255FD5828438E34F9A68ECA76E7778DDA@svcexch02.firm.fenwick.llp>
References: <965C0255FD5828438E34F9A68ECA76E7778DDA@svcexch02.firm.fenwick.llp>
Message-ID: <263E3D5F-C410-4EA1-B3A1-3F82BD92FB4F@cisco.com>

it publishes the SQL database and Directory information (if using DC  
Directory) to all nodes in the cluster.

Note that all processes within the cluster read from the publisher  
database by default.  ccm.exe on sub1 only reads the SQL database on  
sub1 if it cannot contact SQL on the publisher.

/Wes

On Jun 26, 2006, at 5:04 PM, Stu Packett wrote:

I'm thinking about moving my Pub to our DR site as well.  I'm just
concerned about network traffic between the sites.  I know there's a
rule to not have phones registered to the Pub because it does many
roles.  Does anyone know what roles that might be?  I have a DS3 between
my HQ and DR site and I think I have enough bandwidth to handle the
traffic.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Monday, June 26, 2006 10:56 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CallManager DR scenario


I am implementing a CallManager cluster with servers located at both HQ
and a DR site.  The Publisher will be located at HQ, with 2 subscribers
at HQ and the 2 subscribers at the DR site.

If HQ is offline for an extended period of time (building blows up), the
publisher will not be available.  In this situation, CallManager
configuration changes will probably have to be made, but the publisher
will be down.

* Can changes be made to the system without an active publisher?

* I know subscriber won't replicated database to other subscribers.
What happens when the publisher comes back online?  Are changes
replicated from Subscriber to Publisher?

* With 2 subscribers located at the DR site, will I have to make
configuration on both subscriber CallManagers (see previous question)?

* In a period of extended outage, I assume it would be best to restore
the Publisher to one of the Subscribers at DR, correct?

It seems like a Catch-22 for CallManager DR design.  Do you locate the
Publisher at HQ or DR?  I guess it depends on which site will go
down?!?!?  HaHaHa!!


This email may contain proprietary and confidential information for the
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From cisco-voip at ibh.net  Tue Jun 27 13:14:36 2006
From: cisco-voip at ibh.net (Andre Beck)
Date: Tue, 27 Jun 2006 19:14:36 +0200
Subject: [cisco-voip] CME Questions for Rings.
In-Reply-To: <D2BBAE709B4D874DAC556C7EB9D938FC15E986@exchange.TGASolutions.local>
References: <D2BBAE709B4D874DAC556C7EB9D938FC15E986@exchange.TGASolutions.local>
Message-ID: <20060627171436.GB13130@ibh.de>

On Mon, Jun 19, 2006 at 08:42:55PM -0400, Jim McBurnett wrote:
> The distintive ring tone selection exists in CME 4.. but I have not
> played with it..

AFAIK you don't *require* 4.x for distinctive rings, just the relevant
DistinctiveRingList.xml on the TFTP and a sufficiently new phone firm-
ware. I'm adding this to CMEs since phones started to complain that it's
missing, but I admit I haven't tested whether calls to lines configured
with different rings actually play them correctly...

-- 
                  The _S_anta _C_laus _O_peration
  or "how to turn a complete illusion into a neverending money source"

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From wsisk at cisco.com  Tue Jun 27 13:20:33 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 27 Jun 2006 13:20:33 -0400
Subject: [cisco-voip] Weird behaviour after a reboot
In-Reply-To: <62833d8a0606261903q760c5fb4r85af3305567fedad@mail.gmail.com>
References: <62833d8a0606261903q760c5fb4r85af3305567fedad@mail.gmail.com>
Message-ID: <610F4C7D-4968-4A77-B286-893E04974E77@cisco.com>

1. bugcheck on the sub - what OS version? CSCsd53918,
2. rebooted sub, services did not come back - what os version?  
CSCsd18255
3. CTIManager spiking the CPU - CSCsc26241 and the followup  
CSCsd94511 which is not yet resolved.  Both of these would interfere  
with your AC Pilot Points.  Both would additionally be triggered by a  
CTIManager with existing device handle going out of service (server  
BSOD or reboot)

/Wes


On Jun 26, 2006, at 10:03 PM, Nathan Reeves wrote:

Saw an issues with our CCM yesterday which I'd be interested to hear  
if anyone else has seen similar.

Had a bugcheck reboot on the SUB early Sunday morning.  Everything  
failed back to the SUB once it came back up, but most of the AC Pilot  
Points were not working.  Was seeing CPU spikes on both the SUB and  
PUB.  Rebooted the PUB and things went back to normal on that  
server.   Was still seeing CPU spikes on the SUB (mostly caused by  
CTIManager.exe and CCM.exe).

Tried to get the AC Pilot Points working during the day with no  
luck.  Completed a SUB reboot in the evening and when the server came  
back up, none of the devices would fail back.  Saw errors in the  
event log to do with CTIManager, Cisco Extended Functions and RIS  
Data Collector all terminating unexpectedly.  Completed a second  
reboot and this time things came up normally, and devices failed back  
to the SUB.

I'm trying to work out why I would see services fail like this on one  
restart, yet work normally on the next restart.  Our only recent  
change has been the installation of 4.1.3 Sr3a from 4.1.3 Sr2.   
Anyone ever seen this kind of issue with restarts causing CCM to not  
come up cleanly?

Thanks

Nathan
_______________________________________________
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From cisco-voip at ibh.net  Tue Jun 27 13:21:48 2006
From: cisco-voip at ibh.net (Andre Beck)
Date: Tue, 27 Jun 2006 19:21:48 +0200
Subject: [cisco-voip] Different rings on the same phone set?
In-Reply-To: <025601c693d1$82584a30$47196883@cfs.uoguelph.ca>
References: <FF5F9DB1ADC7BB42BFC2C6C1F6638F05E81339@HQEXCHVS1.DC.howrey.net>
	<025601c693d1$82584a30$47196883@cfs.uoguelph.ca>
Message-ID: <20060627172148.GC13130@ibh.de>

On Mon, Jun 19, 2006 at 02:52:26PM -0400, Lelio Fulgenzi wrote:
> I'm hoping to clear up (unless someone already has and is willing to share) whether or not I can simply copy the ringlist.xml file to the alternate ringlist.xml file and have it supported by TAC. This way, you can use all ringtypes on all lines.

Ugh. I'm building a grand unified Ringlist by merging Ringlist.xml and
DistinctiveRinglist.xml from the very day I found out what the Distinctive
one is all about. Never had a problem with it. I didn't even remotely
think about it would be TAC unsupported, as ringlists are clearly stated
to be user-editable everywhere. Is DistinctiveRinglist.xml something
special? One could think so, given the fact that it just contains some
very short and useless ringers, but I attributed this to sloppyness on
the side of the developers...

-- 
                  The _S_anta _C_laus _O_peration
  or "how to turn a complete illusion into a neverending money source"

-> Andre Beck    +++ ABP-RIPE +++    IBH Prof. Dr. Horn GmbH, Dresden <-

From rratliff at cisco.com  Tue Jun 27 13:23:52 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Tue, 27 Jun 2006 13:23:52 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in with
	Extension Mobility
In-Reply-To: <23DB8A63-22DE-41CC-B9E5-5ED326B68DE5@cisco.com>
References: <44A13FED.8000001@dataways.gr>
	<23DB8A63-22DE-41CC-B9E5-5ED326B68DE5@cisco.com>
Message-ID: <CD1C9D30-D73C-4853-92F4-F815082CF9EE@cisco.com>

Just to clarify when you login with EM it actually changes the  
database so no matter which server the phone registers with as long  
as that EM profile is logged in the phone will get the EM lines.  The  
user won't get automatically logged out or anything just because the  
phone fails over to another server.

-Ryan

On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:

The phone will act just like it would if you were not logged in with
EM, ie it will register with the backup CallManager server.

-Ryan

On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou wrote:

Hi all,
if a user is logged in with extension mobility, and the link to the
primary ccm fails what happens with that phone?


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From wsisk at cisco.com  Tue Jun 27 13:26:12 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Tue, 27 Jun 2006 13:26:12 -0400
Subject: [cisco-voip] 7941/7961 Tone Generator
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0024FB0FB@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0024FB0FB@USNAEXCH.na.didata.local>
Message-ID: <10D91E4F-1A73-4DFE-8962-E886305CB533@cisco.com>

CSCee44462 - Need ability to generate inband test tone on 7970 like  
7960 does

/Wes

On Jun 27, 2006, at 11:47 AM, Jason Aarons ((US)) wrote:

**3 turns on ?Debug Mode? on a 7941, however I don?t get a Tone  
Generator softkey when placing a call.  I?ve done this before on  
7940/7960s with no issues.



Anyone tried the 41/61/70 ? These phones run a Java applet?





http://www.cisco.com/en/US/products/hw/phones/ps379/ 
products_tech_note09186a00800945bd.shtml#testtone





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From Scott.Voll at wesd.org  Tue Jun 27 13:42:46 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Tue, 27 Jun 2006 10:42:46 -0700
Subject: [cisco-voip] 7941/7961 Tone Generator
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED5353@Apollo.wesd.org>

Any reason it doesn't work on the 40/60 On CM 4.1?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Wes Sisk
Sent: Tuesday, June 27, 2006 10:26 AM
To: Jason Aarons ((US))
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7941/7961 Tone Generator

 

CSCee44462 - Need ability to generate inband test tone on 7970 like 7960
does

 

/Wes

 

On Jun 27, 2006, at 11:47 AM, Jason Aarons ((US)) wrote:

 

**3 turns on "Debug Mode" on a 7941, however I don't get a Tone
Generator softkey when placing a call.  I've done this before on
7940/7960s with no issues.

 

Anyone tried the 41/61/70 ? These phones run a Java applet...

 

 

http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09
186a00800945bd.shtml#testtone

 

________________________________

 

 


Disclaimer:

This e-mail communication and any attachments may contain
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From commo_ssg_31f at yahoo.com  Tue Jun 27 18:24:21 2006
From: commo_ssg_31f at yahoo.com (ash AD)
Date: Tue, 27 Jun 2006 15:24:21 -0700 (PDT)
Subject: [cisco-voip] Selections Unavailable list.xml
In-Reply-To: <5d093f9a0606261719l29f1ba0dp6823213e6b3ef035@mail.gmail.com>
Message-ID: <20060627222421.9271.qmail@web50406.mail.yahoo.com>

What's up Mike? Also, don't forget to backup the folder prior to upgrades. CCM upgrades will often overwrite this file.
   
  Peter
  

Jonathan Charles <jonvoip at gmail.com> wrote:
  I did this today and it worked great

First off, make sure the files are really PMG files (MSPAINT works
great for converting stuff)

Second, make sure that you follow this format:

URL="TFTP:Desktops/320x212x12/USFLAG.PNG"/>

on two lines (to follow the doc).

Third, put it in this directory under TFTPPATH:

Desktops\320x212x12\

Does it not even give you the default background?


Jonathan


On 6/26/06, Michael L. Windon wrote:
>
>
>
>
>
>
> All, what am I doing wrong here!!?!?! I used these instructions to configure
> Background Images for a 7911:
>
> -------------------------------------------------------------------------------------------------------------------------------------------------------
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_administration_guide_chapter09186a00805adae5.html#wp1088046Creating
> Custom Background Images
>
> -------------------------------------------------------------------------------------------------------------------------------------------------------
>
>
>
> I created the List.xml file here it is listed below:
>
> 
>
> > URL="TFTP:Desktops/95x34x1/911.png" />
>
> 
>
> ------------------------------------------------------------------------------
>
>
>
>
> I also created the Desktops/95x34x1 directory structure and ensure the
> CCMService user has full access to the directory. I have also restarted the
> TFTP service and restarted the IP Phone. Not matter what I do when I try to
> modify the background "Selections Unavailable" is displayed at the bottom of
> the display. What am I missing?
>
>
>
> Mike
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


 				
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From bababooey at cox.net  Tue Jun 27 21:01:45 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Tue, 27 Jun 2006 21:01:45 -0400
Subject: [cisco-voip] 7970 display question
References: <mailman.12025.1151430174.1478.cisco-voip@puck.nether.net>
Message-ID: <000301c69a4e$6deb46a0$0f0110ac@corporation2>

I have some numbers that when dialed I force the user to enter '#' to 
complete the dialed number.
I am wondering if there is a way to have the display indicate to the user 
"hit # to complete call", either on the splash
screen the dialed number is displayed on or the background.

thanks
bob



From nathan.a.reeves at gmail.com  Tue Jun 27 22:20:58 2006
From: nathan.a.reeves at gmail.com (Nathan Reeves)
Date: Wed, 28 Jun 2006 10:20:58 +0800
Subject: [cisco-voip] Weird behaviour after a reboot
In-Reply-To: <610F4C7D-4968-4A77-B286-893E04974E77@cisco.com>
References: <62833d8a0606261903q760c5fb4r85af3305567fedad@mail.gmail.com>
	<610F4C7D-4968-4A77-B286-893E04974E77@cisco.com>
Message-ID: <62833d8a0606271920k16263f3dt706b47237bd69c0b@mail.gmail.com>

Thanks for the pointers there.  Sounds pretty much like what we saw.  For
reference we're running 2000.2.7sr8.

The services in Q were CTIManager, RIS Data Collector and the Cisco Extended
Functions.  CCM and others started okay, but the three mentioned services
all terminated upon startup.

Time for me to start searching the Bug reports a bit more often.

Much appreciated.


On 6/28/06, Wes Sisk <wsisk at cisco.com> wrote:
>
> 1. bugcheck on the sub - what OS version? CSCsd53918,
> 2. rebooted sub, services did not come back - what os version?
> CSCsd18255
> 3. CTIManager spiking the CPU - CSCsc26241 and the followup
> CSCsd94511 which is not yet resolved.  Both of these would interfere
> with your AC Pilot Points.  Both would additionally be triggered by a
> CTIManager with existing device handle going out of service (server
> BSOD or reboot)
>
> /Wes
>
>
>
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From jason.aarons at us.didata.com  Tue Jun 27 22:50:08 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Tue, 27 Jun 2006 22:50:08 -0400
Subject: [cisco-voip] 7941/7961 Tone Generator
Message-ID: <C1FE15183DA37645BC0633BC604E44F0024FB9D0@USNAEXCH.na.didata.local>

Is there another instrument/test tool that would generate 1004 Hz tone
at -15 dB besides the phone to test ERL?

 

Is there a workaround for measuring ELR if the phone won't generate a
test tone.

 

Would a T-BERD or similar measure ERL in a T1 CAS environment?

 

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Tuesday, June 27, 2006 10:26 AM
To: Jason Aarons (US)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] 7941/7961 Tone Generator

 

CSCee44462 - Need ability to generate inband test tone on 7970 like 7960
does

 

/Wes

 

On Jun 27, 2006, at 11:47 AM, Jason Aarons ((US)) wrote:

 

**3 turns on "Debug Mode" on a 7941, however I don't get a Tone
Generator softkey when placing a call.  I've done this before on
7940/7960s with no issues.

 

Anyone tried the 41/61/70 ? These phones run a Java applet...

 

 

http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09
186a00800945bd.shtml#testtone

 

________________________________

 

 


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 your computer. Thank you.

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

 




-----------------------------------------
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From netsecuredata at gmail.com  Wed Jun 28 01:25:09 2006
From: netsecuredata at gmail.com (Jorge Evangelista)
Date: Wed, 28 Jun 2006 00:25:09 -0500
Subject: [cisco-voip] CCM download
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED52E3@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A605ED52E3@Apollo.wesd.org>
Message-ID: <de85c96f0606272225m235e7916v1ca91b6cb9566a70@mail.gmail.com>

There was a version beta for CCM 5.0 that you could download from
Cisco with your account CCO. But Cisco retired CCM beta months later.
I have heard in a forum people that said configure CCM 5.0 on VMware.
Also, I am looking for CCM for Linux.
Is there a certain link in Internet I can download CCM beta?


On 6/27/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>
>
> Call Manager is not on the CCO to download.  Only updates.  You will need to
> order. Or PUT it.
>
>
>
> Scott
>
>
> ________________________________
>
>
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> Curt Shaffer
> Sent: Monday, June 26, 2006 3:15 PM
> To: cisco-voip at puck.nether.net
> Subject: [cisco-voip] CCM download
>
>
>
>
> I heard a while back that you can download Cisco Call Manger for free but I
> can't find it. I really dislike the navigation on their site sometimes. Can
> anyone point me in the right direction?
>
>
>
>
>
> Thanks
>
>
>
> Curt
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 
"The network is the computer"

From akoul at dataways.gr  Wed Jun 28 02:47:22 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Wed, 28 Jun 2006 09:47:22 +0300
Subject: [cisco-voip] Link to primary CCM fails while logged in with
 Extension Mobility
In-Reply-To: <CD1C9D30-D73C-4853-92F4-F815082CF9EE@cisco.com>
References: <44A13FED.8000001@dataways.gr>
	<23DB8A63-22DE-41CC-B9E5-5ED326B68DE5@cisco.com>
	<CD1C9D30-D73C-4853-92F4-F815082CF9EE@cisco.com>
Message-ID: <44A225FA.4050103@dataways.gr>

Hi Ryan,
i am sorry that i didn't clarify it but i was refering to the situation 
where there is an SRST involved:
there is a CCM cluster and an SRST configured at the Device Pool to the 
remote site.
At the remote site a user logs in with EM but at the USER device profile 
there is no reference about the device pool.
What happens to that phone while it is logged in with EM and the remote 
site loses connectivity with all the CCM cluster?

Cheers
Anthony

Ryan Ratliff wrote:
> Just to clarify when you login with EM it actually changes the 
> database so no matter which server the phone registers with as long as 
> that EM profile is logged in the phone will get the EM lines.  The 
> user won't get automatically logged out or anything just because the 
> phone fails over to another server.
>
> -Ryan
>
> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
>
> The phone will act just like it would if you were not logged in with
> EM, ie it will register with the backup CallManager server.
>
> -Ryan
>
> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou wrote:
>
> Hi all,
> if a user is logged in with extension mobility, and the link to the
> primary ccm fails what happens with that phone?
>
>
> _______________________________________________
> 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 CarlosL at mrs.es  Wed Jun 28 05:34:53 2006
From: CarlosL at mrs.es (Carlos Lopez Fernandez)
Date: Wed, 28 Jun 2006 11:34:53 +0200
Subject: [cisco-voip] 7912 delay
Message-ID: <687262A118FB1146BFF85FB25047267914AAFF@mrsalow3s001.mrs.local>

Hi all

Ihave 5 cisco 7912 taht respond to a ping with 65ms and in the same lan
I have a 7960 that respond with 1ms 
Any ideas??

Thanks in advance
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From tednugent69 at yahoo.com  Wed Jun 28 08:16:06 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 28 Jun 2006 05:16:06 -0700 (PDT)
Subject: [cisco-voip] 7912 delay
In-Reply-To: <687262A118FB1146BFF85FB25047267914AAFF@mrsalow3s001.mrs.local>
Message-ID: <20060628121606.96124.qmail@web36104.mail.mud.yahoo.com>

Speed and Dup configured as Auto on the switch?

--- Carlos Lopez Fernandez <CarlosL at mrs.es> wrote:

> Hi all
> 
> Ihave 5 cisco 7912 taht respond to a ping with 65ms
> and in the same lan
> I have a 7960 that respond with 1ms 
> Any ideas??
> 
> Thanks in advance
> > _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From tednugent69 at yahoo.com  Wed Jun 28 08:19:07 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 28 Jun 2006 05:19:07 -0700 (PDT)
Subject: [cisco-voip] Link to primary CCM fails while logged in with
	Extension Mobility
In-Reply-To: <44A225FA.4050103@dataways.gr>
Message-ID: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>

The phone resets and goes to it's default profile. EM
will not work in SRST since the EM service runs on CCM
which is not longer in the picture

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

> Hi Ryan,
> i am sorry that i didn't clarify it but i was
> refering to the situation 
> where there is an SRST involved:
> there is a CCM cluster and an SRST configured at the
> Device Pool to the 
> remote site.
> At the remote site a user logs in with EM but at the
> USER device profile 
> there is no reference about the device pool.
> What happens to that phone while it is logged in
> with EM and the remote 
> site loses connectivity with all the CCM cluster?
> 
> Cheers
> Anthony
> 
> Ryan Ratliff wrote:
> > Just to clarify when you login with EM it actually
> changes the 
> > database so no matter which server the phone
> registers with as long as 
> > that EM profile is logged in the phone will get
> the EM lines.  The 
> > user won't get automatically logged out or
> anything just because the 
> > phone fails over to another server.
> >
> > -Ryan
> >
> > On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
> >
> > The phone will act just like it would if you were
> not logged in with
> > EM, ie it will register with the backup
> CallManager server.
> >
> > -Ryan
> >
> > On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
> wrote:
> >
> > Hi all,
> > if a user is logged in with extension mobility,
> and the link to the
> > primary ccm fails what happens with that phone?
> >
> >
> > _______________________________________________
> > 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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Tired of spam?  Yahoo! Mail has the best spam protection around 
http://mail.yahoo.com 

From ealeatherman at gmail.com  Wed Jun 28 08:55:55 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 28 Jun 2006 08:55:55 -0400
Subject: [cisco-voip] CME Feature Question
Message-ID: <94a1afde0606280555r48961944od139900a9b257120@mail.gmail.com>

For those running CME (I guess 4.0 now?)

I'm trying to evaluate some pro's and con's to putting a CME router in a
building here on campus. Specifically, the president of the university has
decided that we will provide his house with VOIP service. With this comes a
long list of 'gotta have' features. I would prefer to put CME in to segment
that situation from the rest of the campus for management purposes, but i'm
concerned that we'll be missing some things -

Specifically, they need:
Call Barge
Shared Line Privacy Toggle
Group Paging - I heard CME can do this
Fast Dials/Personal Address Book (could I point a service on their phone
back to the main campus cluster for this?)
Intercom between any 2 of the twenty phones in the house. This I think is
going to be an ugly PLAR setup to do it how they want, as they want to be
able to see line state of each phone on the other phones.

I can set them up on the campus callmanager system instead and provide most
of these functions, but then I need to figure out how to do paging - looking
at some options here with some third parties. Will probably need to put in
an SRST router anyway, so the CME isnt going to be that much more if we that
route.

Pestering my SE about this also, but i'm interested in other opinions from
folks familiar with CME.

Thanks!

-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From rleetun at co.boulder.co.us  Wed Jun 28 09:17:40 2006
From: rleetun at co.boulder.co.us (Leetun, Rob)
Date: Wed, 28 Jun 2006 07:17:40 -0600
Subject: [cisco-voip] Have you seen this article?
Message-ID: <963CE94787005149A52C17F33A028E12051D4940@EVS1B.boco.co.boulder.co.us>

 Cisco Call Manager Flaw Could Invite Hackers 

Vulnerabilities in Cisco's Call Manager software could open the door for
hackers to reconfigure VoIP settings and gain access to individual
users' account information, according to researchers at Kansas City,
Mo.-based solution provider FishNet Security. 

By Kevin McLaughlin, CRN
<http://www.crn.com/;jsessionid=AOR2TDNJYUNU0QSNDLPSKHSCJUNN2JVN> 
Jun 19, 2006 
URL:http://www.ddj.com/dept/security/189500728
<outbind://214/dept/security/189500728> 


Vulnerabilities in Cisco's Call Manager software could open the door for
hackers to reconfigure VoIP
<http://www.channelweb.com/encyclopedia/defineterm.jhtml?term=VoIP&x=&y=
>  settings and gain access to individual users' account information,
according to researchers at Kansas City, Mo.-based solution provider
FishNet Security. 

In a report issued Monday, Jake Reynolds, senior security engineer at
FishNet, said the vulnerability affects versions 3.1 and higher of Call
Manager, which handles call routing and call signaling functions in
Cisco VoIP systems. A lack of input validation and output encoding in
the Web administration interface for Call Manager could allow hackers to
execute cross-site scripting attacks, Reynolds wrote. 

Cross site scripting attacks usually involve tricking users with access
privileges into clicking on a URL
<http://www.channelweb.com/encyclopedia/defineterm.jhtml?term=URL&x=&y=>
in an email or Web page. 

In the Call Manager scenario, attackers would send a request to the Call
Manager Web interface that causes malicious JavaScript to be included.
If the administrator could be tricked into submitting this tainted
request, the malicious code would execute in the victim's Web browser
and potentially give attackers the ability to delete or reconfigure
system components and gain access to confidential user information,
according to the report. 

In a statement, Cisco's Product Security Incident Response Team (PSRIT)
recommended that users verify link destinations before clicking on URLs.


Although there are no workarounds for the issue, Cisco has fixed the
vulnerability and fixes will be incorporated in all supported
CallManager trains in versions 4.3(1), 4.2(3), 4.1(3)SR4 and 3.3(5)SR3,
according to the statement. 

To guard against attacks, FishNet recommends that companies limit
network connectivity to Call Manager wherever possible to prevent
hackers from discovering public Web interfaces. 

"Simple Google queries are all an attacker needs in this case to obtain
the target Call Manager address. There are few compelling reasons one
could present that would justify public access to Call Manager web
interfaces," according to the report. 

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From rratliff at cisco.com  Wed Jun 28 09:28:44 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 28 Jun 2006 09:28:44 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in with
	Extension Mobility
In-Reply-To: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
Message-ID: <69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>

I just tested with a 7970 and when the phone drops into SRST mode it  
keeps the EM profile dn(s).  Since you don't configure ephone/dns in  
SRST the phone keeps whatever dns it had before it unregistered with CM.

-Ryan

On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:

The phone resets and goes to it's default profile. EM
will not work in SRST since the EM service runs on CCM
which is not longer in the picture

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

> Hi Ryan,
> i am sorry that i didn't clarify it but i was
> refering to the situation
> where there is an SRST involved:
> there is a CCM cluster and an SRST configured at the
> Device Pool to the
> remote site.
> At the remote site a user logs in with EM but at the
> USER device profile
> there is no reference about the device pool.
> What happens to that phone while it is logged in
> with EM and the remote
> site loses connectivity with all the CCM cluster?
>
> Cheers
> Anthony
>
> Ryan Ratliff wrote:
>> Just to clarify when you login with EM it actually
> changes the
>> database so no matter which server the phone
> registers with as long as
>> that EM profile is logged in the phone will get
> the EM lines.  The
>> user won't get automatically logged out or
> anything just because the
>> phone fails over to another server.
>>
>> -Ryan
>>
>> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
>>
>> The phone will act just like it would if you were
> not logged in with
>> EM, ie it will register with the backup
> CallManager server.
>>
>> -Ryan
>>
>> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
> wrote:
>>
>> Hi all,
>> if a user is logged in with extension mobility,
> and the link to the
>> primary ccm fails what happens with that phone?
>>
>>
>> _______________________________________________
>> 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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Tired of spam?  Yahoo! Mail has the best spam protection around
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From lelio at uoguelph.ca  Wed Jun 28 09:32:44 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 28 Jun 2006 09:32:44 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
Message-ID: <002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>

I would expect the same to happen if the PUB went down and access to the SUBs was still available.

Can I assume that during this time, no logins or logouts would be 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: Ryan Ratliff 
  To: Ted Nugent 
  Cc: cisco-voip at puck-nether.net 
  Sent: Wednesday, June 28, 2006 9:28 AM
  Subject: Re: [cisco-voip] Link to primary CCM fails while logged in withExtension Mobility


  I just tested with a 7970 and when the phone drops into SRST mode it  
  keeps the EM profile dn(s).  Since you don't configure ephone/dns in  
  SRST the phone keeps whatever dns it had before it unregistered with CM.

  -Ryan

  On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:

  The phone resets and goes to it's default profile. EM
  will not work in SRST since the EM service runs on CCM
  which is not longer in the picture

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

  > Hi Ryan,
  > i am sorry that i didn't clarify it but i was
  > refering to the situation
  > where there is an SRST involved:
  > there is a CCM cluster and an SRST configured at the
  > Device Pool to the
  > remote site.
  > At the remote site a user logs in with EM but at the
  > USER device profile
  > there is no reference about the device pool.
  > What happens to that phone while it is logged in
  > with EM and the remote
  > site loses connectivity with all the CCM cluster?
  >
  > Cheers
  > Anthony
  >
  > Ryan Ratliff wrote:
  >> Just to clarify when you login with EM it actually
  > changes the
  >> database so no matter which server the phone
  > registers with as long as
  >> that EM profile is logged in the phone will get
  > the EM lines.  The
  >> user won't get automatically logged out or
  > anything just because the
  >> phone fails over to another server.
  >>
  >> -Ryan
  >>
  >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
  >>
  >> The phone will act just like it would if you were
  > not logged in with
  >> EM, ie it will register with the backup
  > CallManager server.
  >>
  >> -Ryan
  >>
  >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
  > wrote:
  >>
  >> Hi all,
  >> if a user is logged in with extension mobility,
  > and the link to the
  >> primary ccm fails what happens with that phone?
  >>
  >>
  >> _______________________________________________
  >> 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
  >


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

  _______________________________________________
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From rratliff at cisco.com  Wed Jun 28 09:34:15 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 28 Jun 2006 09:34:15 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
In-Reply-To: <002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
Message-ID: <7CE665D3-BB34-4FB4-8EA6-0F3803E6F3E9@cisco.com>

If the server where your EM ip phone service is down then no logins  
or logouts would be possible.  If it's the pub that's down you also  
could not force a logout via the CCMAdmin.

And it goes without saying that in SRST mode you can't logout.

-Ryan

On Jun 28, 2006, at 9:32 AM, Lelio Fulgenzi wrote:

I would expect the same to happen if the PUB went down and access to  
the SUBs was still available.

Can I assume that during this time, no logins or logouts would be  
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: Ryan Ratliff
To: Ted Nugent
Cc: cisco-voip at puck-nether.net
Sent: Wednesday, June 28, 2006 9:28 AM
Subject: Re: [cisco-voip] Link to primary CCM fails while logged in  
withExtension Mobility

I just tested with a 7970 and when the phone drops into SRST mode it
keeps the EM profile dn(s).  Since you don't configure ephone/dns in
SRST the phone keeps whatever dns it had before it unregistered with CM.

-Ryan

On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:

The phone resets and goes to it's default profile. EM
will not work in SRST since the EM service runs on CCM
which is not longer in the picture

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

 > Hi Ryan,
 > i am sorry that i didn't clarify it but i was
 > refering to the situation
 > where there is an SRST involved:
 > there is a CCM cluster and an SRST configured at the
 > Device Pool to the
 > remote site.
 > At the remote site a user logs in with EM but at the
 > USER device profile
 > there is no reference about the device pool.
 > What happens to that phone while it is logged in
 > with EM and the remote
 > site loses connectivity with all the CCM cluster?
 >
 > Cheers
 > Anthony
 >
 > Ryan Ratliff wrote:
 >> Just to clarify when you login with EM it actually
 > changes the
 >> database so no matter which server the phone
 > registers with as long as
 >> that EM profile is logged in the phone will get
 > the EM lines.  The
 >> user won't get automatically logged out or
 > anything just because the
 >> phone fails over to another server.
 >>
 >> -Ryan
 >>
 >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
 >>
 >> The phone will act just like it would if you were
 > not logged in with
 >> EM, ie it will register with the backup
 > CallManager server.
 >>
 >> -Ryan
 >>
 >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
 > wrote:
 >>
 >> Hi all,
 >> if a user is logged in with extension mobility,
 > and the link to the
 >> primary ccm fails what happens with that phone?
 >>
 >>
 >> _______________________________________________
 >> 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
 >


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

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



From rratliff at cisco.com  Wed Jun 28 09:36:40 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 28 Jun 2006 09:36:40 -0400
Subject: [cisco-voip] Have you seen this article?
In-Reply-To: <963CE94787005149A52C17F33A028E12051D4940@EVS1B.boco.co.boulder.co.us>
References: <963CE94787005149A52C17F33A028E12051D4940@EVS1B.boco.co.boulder.co.us>
Message-ID: <6F5E2DDC-AA71-46FB-9FE3-6A4AB74CE20A@cisco.com>

http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
tsd_products_security_response09186a00806c0846.html

-Ryan

On Jun 28, 2006, at 9:17 AM, Leetun, Rob wrote:

  Cisco Call Manager Flaw Could Invite Hackers

Vulnerabilities in Cisco's Call Manager software could open the door  
for hackers to reconfigure VoIP settings and gain access to  
individual users' account information, according to researchers at  
Kansas City, Mo.-based solution provider FishNet Security.

By Kevin McLaughlin, CRN
Jun 19, 2006
URL:http://www.ddj.com/dept/security/189500728

Vulnerabilities in Cisco's Call Manager software could open the door  
for hackers to reconfigure VoIP settings and gain access to  
individual users' account information, according to researchers at  
Kansas City, Mo.-based solution provider FishNet Security.

In a report issued Monday, Jake Reynolds, senior security engineer at  
FishNet, said the vulnerability affects versions 3.1 and higher of  
Call Manager, which handles call routing and call signaling functions  
in Cisco VoIP systems. A lack of input validation and output encoding  
in the Web administration interface for Call Manager could allow  
hackers to execute cross-site scripting attacks, Reynolds wrote.

Cross site scripting attacks usually involve tricking users with  
access privileges into clicking on a URL in an email or Web page.

In the Call Manager scenario, attackers would send a request to the  
Call Manager Web interface that causes malicious JavaScript to be  
included. If the administrator could be tricked into submitting this  
tainted request, the malicious code would execute in the victim's Web  
browser and potentially give attackers the ability to delete or  
reconfigure system components and gain access to confidential user  
information, according to the report.

In a statement, Cisco's Product Security Incident Response Team  
(PSRIT) recommended that users verify link destinations before  
clicking on URLs.

Although there are no workarounds for the issue, Cisco has fixed the  
vulnerability and fixes will be incorporated in all supported  
CallManager trains in versions 4.3(1), 4.2(3), 4.1(3)SR4 and 3.3(5) 
SR3, according to the statement.

To guard against attacks, FishNet recommends that companies limit  
network connectivity to Call Manager wherever possible to prevent  
hackers from discovering public Web interfaces.

"Simple Google queries are all an attacker needs in this case to  
obtain the target Call Manager address. There are few compelling  
reasons one could present that would justify public access to Call  
Manager web interfaces," according to the report.

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


From jsteinberg at gmail.com  Wed Jun 28 09:38:30 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 28 Jun 2006 09:38:30 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
In-Reply-To: <002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
Message-ID: <f1b65f880606280638o5e24f889x3fa9341364cb56d3@mail.gmail.com>

This isn't specific to EM.   But what happens if you are operating in
SRST mode and you lose power to all the phones and the SRST router.
Do the phones or router keep record of their last DN in flash or would
you be completely down after power came back with access to the CM
cluster still unavailable.

Justin

On 6/28/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
> I would expect the same to happen if the PUB went down and access to the
> SUBs was still available.
>
> Can I assume that during this time, no logins or logouts would be 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: Ryan Ratliff
> To: Ted Nugent
> Cc: cisco-voip at puck-nether.net
> Sent: Wednesday, June 28, 2006 9:28 AM
> Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
> withExtension Mobility
>
> I just tested with a 7970 and when the phone drops into SRST mode it
> keeps the EM profile dn(s).  Since you don't configure ephone/dns in
> SRST the phone keeps whatever dns it had before it unregistered with CM.
>
> -Ryan
>
> On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
>
> The phone resets and goes to it's default profile. EM
> will not work in SRST since the EM service runs on CCM
> which is not longer in the picture
>
> --- Anthony Kouloglou <akoul at dataways.gr> wrote:
>
> > Hi Ryan,
> > i am sorry that i didn't clarify it but i was
> > refering to the situation
> > where there is an SRST involved:
> > there is a CCM cluster and an SRST configured at the
> > Device Pool to the
> > remote site.
> > At the remote site a user logs in with EM but at the
> > USER device profile
> > there is no reference about the device pool.
> > What happens to that phone while it is logged in
> > with EM and the remote
> > site loses connectivity with all the CCM cluster?
> >
> > Cheers
> > Anthony
> >
> > Ryan Ratliff wrote:
> >> Just to clarify when you login with EM it actually
> > changes the
> >> database so no matter which server the phone
> > registers with as long as
> >> that EM profile is logged in the phone will get
> > the EM lines.  The
> >> user won't get automatically logged out or
> > anything just because the
> >> phone fails over to another server.
> >>
> >> -Ryan
> >>
> >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
> >>
> >> The phone will act just like it would if you were
> > not logged in with
> >> EM, ie it will register with the backup
> > CallManager server.
> >>
> >> -Ryan
> >>
> >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
> > wrote:
> >>
> >> Hi all,
> >> if a user is logged in with extension mobility,
> > and the link to the
> >> primary ccm fails what happens with that phone?
> >>
> >>
> >> _______________________________________________
> >> 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
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From lelio at uoguelph.ca  Wed Jun 28 09:39:01 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 28 Jun 2006 09:39:01 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
	<7CE665D3-BB34-4FB4-8EA6-0F3803E6F3E9@cisco.com>
Message-ID: <003601c69ab8$36eb5450$47196883@cfs.uoguelph.ca>

So, if I run the EM service on a server other than the pub and that server remains up while the pub is down, EM logins and logouts can continue?

--------------------------------------------------------------------------------
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: Ryan Ratliff 
  To: Lelio Fulgenzi 
  Cc: Ted Nugent ; cisco-voip at puck-nether.net 
  Sent: Wednesday, June 28, 2006 9:34 AM
  Subject: Re: [cisco-voip] Link to primary CCM fails while logged in withExtension Mobility


  If the server where your EM ip phone service is down then no logins  
  or logouts would be possible.  If it's the pub that's down you also  
  could not force a logout via the CCMAdmin.

  And it goes without saying that in SRST mode you can't logout.

  -Ryan

  On Jun 28, 2006, at 9:32 AM, Lelio Fulgenzi wrote:

  I would expect the same to happen if the PUB went down and access to  
  the SUBs was still available.

  Can I assume that during this time, no logins or logouts would be  
  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: Ryan Ratliff
  To: Ted Nugent
  Cc: cisco-voip at puck-nether.net
  Sent: Wednesday, June 28, 2006 9:28 AM
  Subject: Re: [cisco-voip] Link to primary CCM fails while logged in  
  withExtension Mobility

  I just tested with a 7970 and when the phone drops into SRST mode it
  keeps the EM profile dn(s).  Since you don't configure ephone/dns in
  SRST the phone keeps whatever dns it had before it unregistered with CM.

  -Ryan

  On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:

  The phone resets and goes to it's default profile. EM
  will not work in SRST since the EM service runs on CCM
  which is not longer in the picture

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

   > Hi Ryan,
   > i am sorry that i didn't clarify it but i was
   > refering to the situation
   > where there is an SRST involved:
   > there is a CCM cluster and an SRST configured at the
   > Device Pool to the
   > remote site.
   > At the remote site a user logs in with EM but at the
   > USER device profile
   > there is no reference about the device pool.
   > What happens to that phone while it is logged in
   > with EM and the remote
   > site loses connectivity with all the CCM cluster?
   >
   > Cheers
   > Anthony
   >
   > Ryan Ratliff wrote:
   >> Just to clarify when you login with EM it actually
   > changes the
   >> database so no matter which server the phone
   > registers with as long as
   >> that EM profile is logged in the phone will get
   > the EM lines.  The
   >> user won't get automatically logged out or
   > anything just because the
   >> phone fails over to another server.
   >>
   >> -Ryan
   >>
   >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
   >>
   >> The phone will act just like it would if you were
   > not logged in with
   >> EM, ie it will register with the backup
   > CallManager server.
   >>
   >> -Ryan
   >>
   >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
   > wrote:
   >>
   >> Hi all,
   >> if a user is logged in with extension mobility,
   > and the link to the
   >> primary ccm fails what happens with that phone?
   >>
   >>
   >> _______________________________________________
   >> 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
   >


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

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


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From rratliff at cisco.com  Wed Jun 28 09:40:16 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 28 Jun 2006 09:40:16 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
In-Reply-To: <003601c69ab8$36eb5450$47196883@cfs.uoguelph.ca>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
	<7CE665D3-BB34-4FB4-8EA6-0F3803E6F3E9@cisco.com>
	<003601c69ab8$36eb5450$47196883@cfs.uoguelph.ca>
Message-ID: <C5AC4F1A-726A-467F-8429-A597A3C8FE5E@cisco.com>

Nope.  Remember an EM login requires a database change.  No pub, no  
database changes.

However if your pub remains up but your server hosting the EM login/ 
logout service goes down you can force a logout via CCMAdmin.

-Ryan

On Jun 28, 2006, at 9:39 AM, Lelio Fulgenzi wrote:

So, if I run the EM service on a server other than the pub and that  
server remains up while the pub is down, EM logins and logouts can  
continue?

------------------------------------------------------------------------ 
--------
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: Ryan Ratliff
To: Lelio Fulgenzi
Cc: Ted Nugent ; cisco-voip at puck-nether.net
Sent: Wednesday, June 28, 2006 9:34 AM
Subject: Re: [cisco-voip] Link to primary CCM fails while logged in  
withExtension Mobility

If the server where your EM ip phone service is down then no logins
or logouts would be possible.  If it's the pub that's down you also
could not force a logout via the CCMAdmin.

And it goes without saying that in SRST mode you can't logout.

-Ryan

On Jun 28, 2006, at 9:32 AM, Lelio Fulgenzi wrote:

I would expect the same to happen if the PUB went down and access to
the SUBs was still available.

Can I assume that during this time, no logins or logouts would be
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: Ryan Ratliff
To: Ted Nugent
Cc: cisco-voip at puck-nether.net
Sent: Wednesday, June 28, 2006 9:28 AM
Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
withExtension Mobility

I just tested with a 7970 and when the phone drops into SRST mode it
keeps the EM profile dn(s).  Since you don't configure ephone/dns in
SRST the phone keeps whatever dns it had before it unregistered with CM.

-Ryan

On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:

The phone resets and goes to it's default profile. EM
will not work in SRST since the EM service runs on CCM
which is not longer in the picture

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

  > Hi Ryan,
  > i am sorry that i didn't clarify it but i was
  > refering to the situation
  > where there is an SRST involved:
  > there is a CCM cluster and an SRST configured at the
  > Device Pool to the
  > remote site.
  > At the remote site a user logs in with EM but at the
  > USER device profile
  > there is no reference about the device pool.
  > What happens to that phone while it is logged in
  > with EM and the remote
  > site loses connectivity with all the CCM cluster?
  >
  > Cheers
  > Anthony
  >
  > Ryan Ratliff wrote:
  >> Just to clarify when you login with EM it actually
  > changes the
  >> database so no matter which server the phone
  > registers with as long as
  >> that EM profile is logged in the phone will get
  > the EM lines.  The
  >> user won't get automatically logged out or
  > anything just because the
  >> phone fails over to another server.
  >>
  >> -Ryan
  >>
  >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
  >>
  >> The phone will act just like it would if you were
  > not logged in with
  >> EM, ie it will register with the backup
  > CallManager server.
  >>
  >> -Ryan
  >>
  >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
  > wrote:
  >>
  >> Hi all,
  >> if a user is logged in with extension mobility,
  > and the link to the
  >> primary ccm fails what happens with that phone?
  >>
  >>
  >> _______________________________________________
  >> 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
  >


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

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





From rratliff at cisco.com  Wed Jun 28 09:40:44 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 28 Jun 2006 09:40:44 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
In-Reply-To: <f1b65f880606280638o5e24f889x3fa9341364cb56d3@mail.gmail.com>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
	<f1b65f880606280638o5e24f889x3fa9341364cb56d3@mail.gmail.com>
Message-ID: <92535326-43F1-445E-822F-509089040E09@cisco.com>

My 7970 kept the EM dn through a power cycle while in SRST mode.

-Ryan

On Jun 28, 2006, at 9:38 AM, Justin Steinberg wrote:

This isn't specific to EM.   But what happens if you are operating in
SRST mode and you lose power to all the phones and the SRST router.
Do the phones or router keep record of their last DN in flash or would
you be completely down after power came back with access to the CM
cluster still unavailable.

Justin

On 6/28/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
> I would expect the same to happen if the PUB went down and access  
> to the
> SUBs was still available.
>
> Can I assume that during this time, no logins or logouts would be  
> 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: Ryan Ratliff
> To: Ted Nugent
> Cc: cisco-voip at puck-nether.net
> Sent: Wednesday, June 28, 2006 9:28 AM
> Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
> withExtension Mobility
>
> I just tested with a 7970 and when the phone drops into SRST mode it
> keeps the EM profile dn(s).  Since you don't configure ephone/dns in
> SRST the phone keeps whatever dns it had before it unregistered  
> with CM.
>
> -Ryan
>
> On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
>
> The phone resets and goes to it's default profile. EM
> will not work in SRST since the EM service runs on CCM
> which is not longer in the picture
>
> --- Anthony Kouloglou <akoul at dataways.gr> wrote:
>
> > Hi Ryan,
> > i am sorry that i didn't clarify it but i was
> > refering to the situation
> > where there is an SRST involved:
> > there is a CCM cluster and an SRST configured at the
> > Device Pool to the
> > remote site.
> > At the remote site a user logs in with EM but at the
> > USER device profile
> > there is no reference about the device pool.
> > What happens to that phone while it is logged in
> > with EM and the remote
> > site loses connectivity with all the CCM cluster?
> >
> > Cheers
> > Anthony
> >
> > Ryan Ratliff wrote:
> >> Just to clarify when you login with EM it actually
> > changes the
> >> database so no matter which server the phone
> > registers with as long as
> >> that EM profile is logged in the phone will get
> > the EM lines.  The
> >> user won't get automatically logged out or
> > anything just because the
> >> phone fails over to another server.
> >>
> >> -Ryan
> >>
> >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
> >>
> >> The phone will act just like it would if you were
> > not logged in with
> >> EM, ie it will register with the backup
> > CallManager server.
> >>
> >> -Ryan
> >>
> >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
> > wrote:
> >>
> >> Hi all,
> >> if a user is logged in with extension mobility,
> > and the link to the
> >> primary ccm fails what happens with that phone?
> >>
> >>
> >> _______________________________________________
> >> 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
> >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


From lelio at uoguelph.ca  Wed Jun 28 09:45:40 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 28 Jun 2006 09:45:40 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
	<7CE665D3-BB34-4FB4-8EA6-0F3803E6F3E9@cisco.com>
	<003601c69ab8$36eb5450$47196883@cfs.uoguelph.ca>
	<C5AC4F1A-726A-467F-8429-A597A3C8FE5E@cisco.com>
Message-ID: <005301c69ab9$24b9c720$47196883@cfs.uoguelph.ca>

AH - I see what you were getting at. Thanks!

--------------------------------------------------------------------------------
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: Ryan Ratliff 
  To: Lelio Fulgenzi 
  Cc: Ted Nugent ; cisco-voip at puck-nether.net 
  Sent: Wednesday, June 28, 2006 9:40 AM
  Subject: Re: [cisco-voip] Link to primary CCM fails while logged in withExtension Mobility


  Nope.  Remember an EM login requires a database change.  No pub, no  
  database changes.

  However if your pub remains up but your server hosting the EM login/ 
  logout service goes down you can force a logout via CCMAdmin.

  -Ryan

  On Jun 28, 2006, at 9:39 AM, Lelio Fulgenzi wrote:

  So, if I run the EM service on a server other than the pub and that  
  server remains up while the pub is down, EM logins and logouts can  
  continue?

  ------------------------------------------------------------------------ 
  --------
  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: Ryan Ratliff
  To: Lelio Fulgenzi
  Cc: Ted Nugent ; cisco-voip at puck-nether.net
  Sent: Wednesday, June 28, 2006 9:34 AM
  Subject: Re: [cisco-voip] Link to primary CCM fails while logged in  
  withExtension Mobility

  If the server where your EM ip phone service is down then no logins
  or logouts would be possible.  If it's the pub that's down you also
  could not force a logout via the CCMAdmin.

  And it goes without saying that in SRST mode you can't logout.

  -Ryan

  On Jun 28, 2006, at 9:32 AM, Lelio Fulgenzi wrote:

  I would expect the same to happen if the PUB went down and access to
  the SUBs was still available.

  Can I assume that during this time, no logins or logouts would be
  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: Ryan Ratliff
  To: Ted Nugent
  Cc: cisco-voip at puck-nether.net
  Sent: Wednesday, June 28, 2006 9:28 AM
  Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
  withExtension Mobility

  I just tested with a 7970 and when the phone drops into SRST mode it
  keeps the EM profile dn(s).  Since you don't configure ephone/dns in
  SRST the phone keeps whatever dns it had before it unregistered with CM.

  -Ryan

  On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:

  The phone resets and goes to it's default profile. EM
  will not work in SRST since the EM service runs on CCM
  which is not longer in the picture

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

    > Hi Ryan,
    > i am sorry that i didn't clarify it but i was
    > refering to the situation
    > where there is an SRST involved:
    > there is a CCM cluster and an SRST configured at the
    > Device Pool to the
    > remote site.
    > At the remote site a user logs in with EM but at the
    > USER device profile
    > there is no reference about the device pool.
    > What happens to that phone while it is logged in
    > with EM and the remote
    > site loses connectivity with all the CCM cluster?
    >
    > Cheers
    > Anthony
    >
    > Ryan Ratliff wrote:
    >> Just to clarify when you login with EM it actually
    > changes the
    >> database so no matter which server the phone
    > registers with as long as
    >> that EM profile is logged in the phone will get
    > the EM lines.  The
    >> user won't get automatically logged out or
    > anything just because the
    >> phone fails over to another server.
    >>
    >> -Ryan
    >>
    >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
    >>
    >> The phone will act just like it would if you were
    > not logged in with
    >> EM, ie it will register with the backup
    > CallManager server.
    >>
    >> -Ryan
    >>
    >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
    > wrote:
    >>
    >> Hi all,
    >> if a user is logged in with extension mobility,
    > and the link to the
    >> primary ccm fails what happens with that phone?
    >>
    >>
    >> _______________________________________________
    >> 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
    >


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

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




-------------- next part --------------
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URL: https://puck.nether.net/pipermail/cisco-voip/attachments/20060628/929e6113/attachment.html 

From Eric.A.Fretz at L-3Com.com  Wed Jun 28 09:53:11 2006
From: Eric.A.Fretz at L-3Com.com (Fretz, EA Eric @ IS)
Date: Wed, 28 Jun 2006 08:53:11 -0500
Subject: [cisco-voip] Have you seen this article?
Message-ID: <F2E442781002EA41B3DB013749748D3033A1EF@gvlexch02.is.l-3com.com>

Any network admin that leaves his/her CCM web server facing the internet
needs a good flogging.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, June 28, 2006 8:37 AM
To: Leetun, Rob
Cc: ciscovoip
Subject: Re: [cisco-voip] Have you seen this article?


http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
tsd_products_security_response09186a00806c0846.html

-Ryan

On Jun 28, 2006, at 9:17 AM, Leetun, Rob wrote:

  Cisco Call Manager Flaw Could Invite Hackers

Vulnerabilities in Cisco's Call Manager software could open the door  
for hackers to reconfigure VoIP settings and gain access to  
individual users' account information, according to researchers at  
Kansas City, Mo.-based solution provider FishNet Security.

By Kevin McLaughlin, CRN
Jun 19, 2006
URL:http://www.ddj.com/dept/security/189500728

Vulnerabilities in Cisco's Call Manager software could open the door  
for hackers to reconfigure VoIP settings and gain access to  
individual users' account information, according to researchers at  
Kansas City, Mo.-based solution provider FishNet Security.

In a report issued Monday, Jake Reynolds, senior security engineer at  
FishNet, said the vulnerability affects versions 3.1 and higher of  
Call Manager, which handles call routing and call signaling functions  
in Cisco VoIP systems. A lack of input validation and output encoding  
in the Web administration interface for Call Manager could allow  
hackers to execute cross-site scripting attacks, Reynolds wrote.

Cross site scripting attacks usually involve tricking users with  
access privileges into clicking on a URL in an email or Web page.

In the Call Manager scenario, attackers would send a request to the  
Call Manager Web interface that causes malicious JavaScript to be  
included. If the administrator could be tricked into submitting this  
tainted request, the malicious code would execute in the victim's Web  
browser and potentially give attackers the ability to delete or  
reconfigure system components and gain access to confidential user  
information, according to the report.

In a statement, Cisco's Product Security Incident Response Team  
(PSRIT) recommended that users verify link destinations before  
clicking on URLs.

Although there are no workarounds for the issue, Cisco has fixed the  
vulnerability and fixes will be incorporated in all supported  
CallManager trains in versions 4.3(1), 4.2(3), 4.1(3)SR4 and 3.3(5) 
SR3, according to the statement.

To guard against attacks, FishNet recommends that companies limit  
network connectivity to Call Manager wherever possible to prevent  
hackers from discovering public Web interfaces.

"Simple Google queries are all an attacker needs in this case to  
obtain the target Call Manager address. There are few compelling  
reasons one could present that would justify public access to Call  
Manager web interfaces," according to the report.

_______________________________________________
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 akoul at dataways.gr  Wed Jun 28 09:52:59 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Wed, 28 Jun 2006 16:52:59 +0300
Subject: [cisco-voip] Link to primary CCM fails while logged
 in	withExtension Mobility
In-Reply-To: <C5AC4F1A-726A-467F-8429-A597A3C8FE5E@cisco.com>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>	<7CE665D3-BB34-4FB4-8EA6-0F3803E6F3E9@cisco.com>	<003601c69ab8$36eb5450$47196883@cfs.uoguelph.ca>
	<C5AC4F1A-726A-467F-8429-A597A3C8FE5E@cisco.com>
Message-ID: <44A289BB.5030609@dataways.gr>

Hi all,
but when a phone is in EM mode, how does it now who is SRST? How does it 
keep track of it?
There is no device pool setting in the User Device Profile.

Cheers
Akoul

Ryan Ratliff wrote:
> Nope.  Remember an EM login requires a database change.  No pub, no  
> database changes.
>
> However if your pub remains up but your server hosting the EM login/ 
> logout service goes down you can force a logout via CCMAdmin.
>
> -Ryan
>
> On Jun 28, 2006, at 9:39 AM, Lelio Fulgenzi wrote:
>
> So, if I run the EM service on a server other than the pub and that  
> server remains up while the pub is down, EM logins and logouts can  
> continue?
>
> ------------------------------------------------------------------------ 
> --------
> 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: Ryan Ratliff
> To: Lelio Fulgenzi
> Cc: Ted Nugent ; cisco-voip at puck-nether.net
> Sent: Wednesday, June 28, 2006 9:34 AM
> Subject: Re: [cisco-voip] Link to primary CCM fails while logged in  
> withExtension Mobility
>
> If the server where your EM ip phone service is down then no logins
> or logouts would be possible.  If it's the pub that's down you also
> could not force a logout via the CCMAdmin.
>
> And it goes without saying that in SRST mode you can't logout.
>
> -Ryan
>
> On Jun 28, 2006, at 9:32 AM, Lelio Fulgenzi wrote:
>
> I would expect the same to happen if the PUB went down and access to
> the SUBs was still available.
>
> Can I assume that during this time, no logins or logouts would be
> 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: Ryan Ratliff
> To: Ted Nugent
> Cc: cisco-voip at puck-nether.net
> Sent: Wednesday, June 28, 2006 9:28 AM
> Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
> withExtension Mobility
>
> I just tested with a 7970 and when the phone drops into SRST mode it
> keeps the EM profile dn(s).  Since you don't configure ephone/dns in
> SRST the phone keeps whatever dns it had before it unregistered with CM.
>
> -Ryan
>
> On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
>
> The phone resets and goes to it's default profile. EM
> will not work in SRST since the EM service runs on CCM
> which is not longer in the picture
>
> --- Anthony Kouloglou <akoul at dataways.gr> wrote:
>
>   > Hi Ryan,
>   > i am sorry that i didn't clarify it but i was
>   > refering to the situation
>   > where there is an SRST involved:
>   > there is a CCM cluster and an SRST configured at the
>   > Device Pool to the
>   > remote site.
>   > At the remote site a user logs in with EM but at the
>   > USER device profile
>   > there is no reference about the device pool.
>   > What happens to that phone while it is logged in
>   > with EM and the remote
>   > site loses connectivity with all the CCM cluster?
>   >
>   > Cheers
>   > Anthony
>   >
>   > Ryan Ratliff wrote:
>   >> Just to clarify when you login with EM it actually
>   > changes the
>   >> database so no matter which server the phone
>   > registers with as long as
>   >> that EM profile is logged in the phone will get
>   > the EM lines.  The
>   >> user won't get automatically logged out or
>   > anything just because the
>   >> phone fails over to another server.
>   >>
>   >> -Ryan
>   >>
>   >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
>   >>
>   >> The phone will act just like it would if you were
>   > not logged in with
>   >> EM, ie it will register with the backup
>   > CallManager server.
>   >>
>   >> -Ryan
>   >>
>   >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
>   > wrote:
>   >>
>   >> Hi all,
>   >> if a user is logged in with extension mobility,
>   > and the link to the
>   >> primary ccm fails what happens with that phone?
>   >>
>   >>
>   >> _______________________________________________
>   >> 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
>   >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>   

From jonvoip at gmail.com  Wed Jun 28 09:58:03 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 28 Jun 2006 08:58:03 -0500
Subject: [cisco-voip] 7970 display question
In-Reply-To: <000301c69a4e$6deb46a0$0f0110ac@corporation2>
References: <mailman.12025.1151430174.1478.cisco-voip@puck.nether.net>
	<000301c69a4e$6deb46a0$0f0110ac@corporation2>
Message-ID: <5d093f9a0606280658k7ce5b6d2r2751e1e13cd5e304@mail.gmail.com>

Include a Route Pattern without the #, block it, with a custom msg to add the #


Jonathan

On 6/27/06, Bob A. Bowie <bababooey at cox.net> wrote:
> I have some numbers that when dialed I force the user to enter '#' to
> complete the dialed number.
> I am wondering if there is a way to have the display indicate to the user
> "hit # to complete call", either on the splash
> screen the dialed number is displayed on or the background.
>
> thanks
> bob
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From jonvoip at gmail.com  Wed Jun 28 09:58:53 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 28 Jun 2006 08:58:53 -0500
Subject: [cisco-voip] CCM download
In-Reply-To: <de85c96f0606272225m235e7916v1ca91b6cb9566a70@mail.gmail.com>
References: <407055A92CECCB499C922A2D35FC19A605ED52E3@Apollo.wesd.org>
	<de85c96f0606272225m235e7916v1ca91b6cb9566a70@mail.gmail.com>
Message-ID: <5d093f9a0606280658r5a1f3f67n15760c7f253f2b78@mail.gmail.com>

Basically, you need to get a copy of 3.3 or better, then you can
download upgrades all the way to 4.2


Jonathan

On 6/28/06, Jorge Evangelista <netsecuredata at gmail.com> wrote:
> There was a version beta for CCM 5.0 that you could download from
> Cisco with your account CCO. But Cisco retired CCM beta months later.
> I have heard in a forum people that said configure CCM 5.0 on VMware.
> Also, I am looking for CCM for Linux.
> Is there a certain link in Internet I can download CCM beta?
>
>
> On 6/27/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
> >
> >
> >
> > Call Manager is not on the CCO to download.  Only updates.  You will need to
> > order. Or PUT it.
> >
> >
> >
> > Scott
> >
> >
> > ________________________________
> >
> >
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> > Curt Shaffer
> > Sent: Monday, June 26, 2006 3:15 PM
> > To: cisco-voip at puck.nether.net
> > Subject: [cisco-voip] CCM download
> >
> >
> >
> >
> > I heard a while back that you can download Cisco Call Manger for free but I
> > can't find it. I really dislike the navigation on their site sometimes. Can
> > anyone point me in the right direction?
> >
> >
> >
> >
> >
> > Thanks
> >
> >
> >
> > Curt
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>
>
> --
> "The network is the computer"
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From lelio at uoguelph.ca  Wed Jun 28 09:59:55 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 28 Jun 2006 09:59:55 -0400
Subject: [cisco-voip] Have you seen this article?
References: <963CE94787005149A52C17F33A028E12051D4940@EVS1B.boco.co.boulder.co.us>
	<6F5E2DDC-AA71-46FB-9FE3-6A4AB74CE20A@cisco.com>
Message-ID: <008601c69abb$227bfa30$47196883@cfs.uoguelph.ca>

There's a CallManager 4.3(1)? WAH?????????????????????

--------------------------------------------------------------------------------
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: Ryan Ratliff 
  To: Leetun, Rob 
  Cc: ciscovoip 
  Sent: Wednesday, June 28, 2006 9:36 AM
  Subject: Re: [cisco-voip] Have you seen this article?


  http://www.cisco.com/en/US/products/sw/voicesw/ps556/ 
  tsd_products_security_response09186a00806c0846.html

  -Ryan

  On Jun 28, 2006, at 9:17 AM, Leetun, Rob wrote:

    Cisco Call Manager Flaw Could Invite Hackers

  Vulnerabilities in Cisco's Call Manager software could open the door  
  for hackers to reconfigure VoIP settings and gain access to  
  individual users' account information, according to researchers at  
  Kansas City, Mo.-based solution provider FishNet Security.

  By Kevin McLaughlin, CRN
  Jun 19, 2006
  URL:http://www.ddj.com/dept/security/189500728

  Vulnerabilities in Cisco's Call Manager software could open the door  
  for hackers to reconfigure VoIP settings and gain access to  
  individual users' account information, according to researchers at  
  Kansas City, Mo.-based solution provider FishNet Security.

  In a report issued Monday, Jake Reynolds, senior security engineer at  
  FishNet, said the vulnerability affects versions 3.1 and higher of  
  Call Manager, which handles call routing and call signaling functions  
  in Cisco VoIP systems. A lack of input validation and output encoding  
  in the Web administration interface for Call Manager could allow  
  hackers to execute cross-site scripting attacks, Reynolds wrote.

  Cross site scripting attacks usually involve tricking users with  
  access privileges into clicking on a URL in an email or Web page.

  In the Call Manager scenario, attackers would send a request to the  
  Call Manager Web interface that causes malicious JavaScript to be  
  included. If the administrator could be tricked into submitting this  
  tainted request, the malicious code would execute in the victim's Web  
  browser and potentially give attackers the ability to delete or  
  reconfigure system components and gain access to confidential user  
  information, according to the report.

  In a statement, Cisco's Product Security Incident Response Team  
  (PSRIT) recommended that users verify link destinations before  
  clicking on URLs.

  Although there are no workarounds for the issue, Cisco has fixed the  
  vulnerability and fixes will be incorporated in all supported  
  CallManager trains in versions 4.3(1), 4.2(3), 4.1(3)SR4 and 3.3(5) 
  SR3, according to the statement.

  To guard against attacks, FishNet recommends that companies limit  
  network connectivity to Call Manager wherever possible to prevent  
  hackers from discovering public Web interfaces.

  "Simple Google queries are all an attacker needs in this case to  
  obtain the target Call Manager address. There are few compelling  
  reasons one could present that would justify public access to Call  
  Manager web interfaces," according to the report.

  _______________________________________________
  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 JOlson at mlnusa.com  Wed Jun 28 10:00:26 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Wed, 28 Jun 2006 10:00:26 -0400
Subject: [cisco-voip] Agent state
Message-ID: <FE2D092F8B1516488B99342551AACFD41D7BE593@mlnusa-exch1.mlnusa.com>

I have 1 agent who has not been able to put herself in the ready state.
She is able to login but when she tried to put herself in the ready
state she receives an error saying  

"Request to change agent state failed.  Unable to change agent state."

I have removed her from the IPCC deleting her user account and recreated
it.  I have stopped and started the engine.  Still nothing.  

I can't figure out why this is only happening with this one person and
not anyone else.

 

Any ideas?

 

 

Jeff


********************************************************************
CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

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From jonvoip at gmail.com  Wed Jun 28 10:00:55 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 28 Jun 2006 09:00:55 -0500
Subject: [cisco-voip] Have you seen this article?
In-Reply-To: <F2E442781002EA41B3DB013749748D3033A1EF@gvlexch02.is.l-3com.com>
References: <F2E442781002EA41B3DB013749748D3033A1EF@gvlexch02.is.l-3com.com>
Message-ID: <5d093f9a0606280700i2feb5284v28779598a25a7082@mail.gmail.com>

When I worked at my last company, we had a cust with their CCMs
available via RDP over the internet (directly)... the username and
password was administrator cisco...

Eeek



Jonathan

On 6/28/06, Fretz, EA Eric @ IS <Eric.A.Fretz at l-3com.com> wrote:
> Any network admin that leaves his/her CCM web server facing the internet
> needs a good flogging.
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
> Sent: Wednesday, June 28, 2006 8:37 AM
> To: Leetun, Rob
> Cc: ciscovoip
> Subject: Re: [cisco-voip] Have you seen this article?
>
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/
> tsd_products_security_response09186a00806c0846.html
>
> -Ryan
>
> On Jun 28, 2006, at 9:17 AM, Leetun, Rob wrote:
>
>   Cisco Call Manager Flaw Could Invite Hackers
>
> Vulnerabilities in Cisco's Call Manager software could open the door
> for hackers to reconfigure VoIP settings and gain access to
> individual users' account information, according to researchers at
> Kansas City, Mo.-based solution provider FishNet Security.
>
> By Kevin McLaughlin, CRN
> Jun 19, 2006
> URL:http://www.ddj.com/dept/security/189500728
>
> Vulnerabilities in Cisco's Call Manager software could open the door
> for hackers to reconfigure VoIP settings and gain access to
> individual users' account information, according to researchers at
> Kansas City, Mo.-based solution provider FishNet Security.
>
> In a report issued Monday, Jake Reynolds, senior security engineer at
> FishNet, said the vulnerability affects versions 3.1 and higher of
> Call Manager, which handles call routing and call signaling functions
> in Cisco VoIP systems. A lack of input validation and output encoding
> in the Web administration interface for Call Manager could allow
> hackers to execute cross-site scripting attacks, Reynolds wrote.
>
> Cross site scripting attacks usually involve tricking users with
> access privileges into clicking on a URL in an email or Web page.
>
> In the Call Manager scenario, attackers would send a request to the
> Call Manager Web interface that causes malicious JavaScript to be
> included. If the administrator could be tricked into submitting this
> tainted request, the malicious code would execute in the victim's Web
> browser and potentially give attackers the ability to delete or
> reconfigure system components and gain access to confidential user
> information, according to the report.
>
> In a statement, Cisco's Product Security Incident Response Team
> (PSRIT) recommended that users verify link destinations before
> clicking on URLs.
>
> Although there are no workarounds for the issue, Cisco has fixed the
> vulnerability and fixes will be incorporated in all supported
> CallManager trains in versions 4.3(1), 4.2(3), 4.1(3)SR4 and 3.3(5)
> SR3, according to the statement.
>
> To guard against attacks, FishNet recommends that companies limit
> network connectivity to Call Manager wherever possible to prevent
> hackers from discovering public Web interfaces.
>
> "Simple Google queries are all an attacker needs in this case to
> obtain the target Call Manager address. There are few compelling
> reasons one could present that would justify public access to Call
> Manager web interfaces," according to the report.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From rratliff at cisco.com  Wed Jun 28 10:14:16 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 28 Jun 2006 10:14:16 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged
	in	withExtension Mobility
In-Reply-To: <44A289BB.5030609@dataways.gr>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>	<7CE665D3-BB34-4FB4-8EA6-0F3803E6F3E9@cisco.com>	<003601c69ab8$36eb5450$47196883@cfs.uoguelph.ca>
	<C5AC4F1A-726A-467F-8429-A597A3C8FE5E@cisco.com>
	<44A289BB.5030609@dataways.gr>
Message-ID: <36B6A363-128D-4E3A-9E2A-19428674D628@cisco.com>

The device pool is set on the physical phone itself and this will not  
change when you login with EM.

Basically if you can't configure it on the UDP then it takes that  
setting from the phone config itself.

-Ryan

On Jun 28, 2006, at 9:52 AM, Anthony Kouloglou wrote:

Hi all,
but when a phone is in EM mode, how does it now who is SRST? How does  
it keep track of it?
There is no device pool setting in the User Device Profile.

Cheers
Akoul

Ryan Ratliff wrote:
> Nope.  Remember an EM login requires a database change.  No pub,  
> no  database changes.
>
> However if your pub remains up but your server hosting the EM  
> login/ logout service goes down you can force a logout via CCMAdmin.
>
> -Ryan
>
> On Jun 28, 2006, at 9:39 AM, Lelio Fulgenzi wrote:
>
> So, if I run the EM service on a server other than the pub and  
> that  server remains up while the pub is down, EM logins and  
> logouts can  continue?
>
> ---------------------------------------------------------------------- 
> -- --------
> 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: Ryan Ratliff
> To: Lelio Fulgenzi
> Cc: Ted Nugent ; cisco-voip at puck-nether.net
> Sent: Wednesday, June 28, 2006 9:34 AM
> Subject: Re: [cisco-voip] Link to primary CCM fails while logged  
> in  withExtension Mobility
>
> If the server where your EM ip phone service is down then no logins
> or logouts would be possible.  If it's the pub that's down you also
> could not force a logout via the CCMAdmin.
>
> And it goes without saying that in SRST mode you can't logout.
>
> -Ryan
>
> On Jun 28, 2006, at 9:32 AM, Lelio Fulgenzi wrote:
>
> I would expect the same to happen if the PUB went down and access to
> the SUBs was still available.
>
> Can I assume that during this time, no logins or logouts would be
> 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: Ryan Ratliff
> To: Ted Nugent
> Cc: cisco-voip at puck-nether.net
> Sent: Wednesday, June 28, 2006 9:28 AM
> Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
> withExtension Mobility
>
> I just tested with a 7970 and when the phone drops into SRST mode it
> keeps the EM profile dn(s).  Since you don't configure ephone/dns in
> SRST the phone keeps whatever dns it had before it unregistered  
> with CM.
>
> -Ryan
>
> On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
>
> The phone resets and goes to it's default profile. EM
> will not work in SRST since the EM service runs on CCM
> which is not longer in the picture
>
> --- Anthony Kouloglou <akoul at dataways.gr> wrote:
>
>   > Hi Ryan,
>   > i am sorry that i didn't clarify it but i was
>   > refering to the situation
>   > where there is an SRST involved:
>   > there is a CCM cluster and an SRST configured at the
>   > Device Pool to the
>   > remote site.
>   > At the remote site a user logs in with EM but at the
>   > USER device profile
>   > there is no reference about the device pool.
>   > What happens to that phone while it is logged in
>   > with EM and the remote
>   > site loses connectivity with all the CCM cluster?
>   >
>   > Cheers
>   > Anthony
>   >
>   > Ryan Ratliff wrote:
>   >> Just to clarify when you login with EM it actually
>   > changes the
>   >> database so no matter which server the phone
>   > registers with as long as
>   >> that EM profile is logged in the phone will get
>   > the EM lines.  The
>   >> user won't get automatically logged out or
>   > anything just because the
>   >> phone fails over to another server.
>   >>
>   >> -Ryan
>   >>
>   >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
>   >>
>   >> The phone will act just like it would if you were
>   > not logged in with
>   >> EM, ie it will register with the backup
>   > CallManager server.
>   >>
>   >> -Ryan
>   >>
>   >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
>   > wrote:
>   >>
>   >> Hi all,
>   >> if a user is logged in with extension mobility,
>   > and the link to the
>   >> primary ccm fails what happens with that phone?
>   >>
>   >>
>   >> _______________________________________________
>   >> 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
>   >
>
>
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam protection around
> http://mail.yahoo.com
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


From CarlosL at mrs.es  Wed Jun 28 10:20:44 2006
From: CarlosL at mrs.es (Carlos Lopez Fernandez)
Date: Wed, 28 Jun 2006 16:20:44 +0200
Subject: [cisco-voip] 7912 delay
Message-ID: <687262A118FB1146BFF85FB25047267914AB04@mrsalow3s001.mrs.local>

Yes all ports are configure as auto
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From jcc at unc.edu  Wed Jun 28 10:20:33 2006
From: jcc at unc.edu (Justin Church)
Date: Wed, 28 Jun 2006 10:20:33 -0400
Subject: [cisco-voip] IOS GW PRI Registration to separate CallManagers
Message-ID: <44A29031.8040904@unc.edu>

We're running 2821 IOS gateways using MGCP backhaul to CCM 4.1(3).  I 
have several spare PRI interfaces and would like to register a different 
PRI on the same GW with a different CM (specifically my CM 5.0(3)).  Is 
that possible?

-jc

From bababooey at cox.net  Wed Jun 28 10:21:23 2006
From: bababooey at cox.net (bababooey at cox.net)
Date: Wed, 28 Jun 2006 10:21:23 -0400
Subject: [cisco-voip] 7970 display question
Message-ID: <12916582.1151504483936.JavaMail.root@eastrmwml08.mgt.cox.net>

where do I set the custom msg?
thanks
---- Jonathan Charles <jonvoip at gmail.com> wrote: 
> Include a Route Pattern without the #, block it, with a custom msg to add the #
> 
> 
> Jonathan
> 
> On 6/27/06, Bob A. Bowie <bababooey at cox.net> wrote:
> > I have some numbers that when dialed I force the user to enter '#' to
> > complete the dialed number.
> > I am wondering if there is a way to have the display indicate to the user
> > "hit # to complete call", either on the splash
> > screen the dialed number is displayed on or the background.
> >
> > thanks
> > bob
> >
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >


From rratliff at cisco.com  Wed Jun 28 10:25:13 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 28 Jun 2006 10:25:13 -0400
Subject: [cisco-voip] IOS GW PRI Registration to separate CallManagers
In-Reply-To: <44A29031.8040904@unc.edu>
References: <44A29031.8040904@unc.edu>
Message-ID: <5CCE596C-7F79-40BC-906C-2AA54CC41E0F@cisco.com>

I don't think you can run two mgcp processes on a router.

-Ryan

On Jun 28, 2006, at 10:20 AM, Justin Church wrote:

We're running 2821 IOS gateways using MGCP backhaul to CCM 4.1(3).  I
have several spare PRI interfaces and would like to register a different
PRI on the same GW with a different CM (specifically my CM 5.0(3)).  Is
that possible?

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


From jcc at unc.edu  Wed Jun 28 10:29:24 2006
From: jcc at unc.edu (Justin Church)
Date: Wed, 28 Jun 2006 10:29:24 -0400
Subject: [cisco-voip] IOS GW PRI Registration to separate CallManagers
In-Reply-To: <5CCE596C-7F79-40BC-906C-2AA54CC41E0F@cisco.com>
References: <44A29031.8040904@unc.edu>
	<5CCE596C-7F79-40BC-906C-2AA54CC41E0F@cisco.com>
Message-ID: <44A29244.2020403@unc.edu>

They don't both have to be MGCP.  I could use SIP for the PRI registered 
to CM 5.0.  Any reason that wouldn't work?

-jc

Ryan Ratliff wrote:
> I don't think you can run two mgcp processes on a router.
> 
> -Ryan
> 
> On Jun 28, 2006, at 10:20 AM, Justin Church wrote:
> 
> We're running 2821 IOS gateways using MGCP backhaul to CCM 4.1(3).  I
> have several spare PRI interfaces and would like to register a different
> PRI on the same GW with a different CM (specifically my CM 5.0(3)).  Is
> that possible?
> 
> -jc
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 

From rratliff at cisco.com  Wed Jun 28 10:31:34 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 28 Jun 2006 10:31:34 -0400
Subject: [cisco-voip] IOS GW PRI Registration to separate CallManagers
In-Reply-To: <44A29244.2020403@unc.edu>
References: <44A29031.8040904@unc.edu>
	<5CCE596C-7F79-40BC-906C-2AA54CC41E0F@cisco.com>
	<44A29244.2020403@unc.edu>
Message-ID: <FED076D4-0254-4E4C-9874-9F1F8A81B8E6@cisco.com>

Don't see why not.

When you define the second PRI just don't include the 'service mgcp'  
and you can then do whatever you want with it via dial-peers.

-Ryan

On Jun 28, 2006, at 10:29 AM, Justin Church wrote:

They don't both have to be MGCP.  I could use SIP for the PRI  
registered to CM 5.0.  Any reason that wouldn't work?

-jc

Ryan Ratliff wrote:
> I don't think you can run two mgcp processes on a router.
> -Ryan
> On Jun 28, 2006, at 10:20 AM, Justin Church wrote:
> We're running 2821 IOS gateways using MGCP backhaul to CCM 4.1(3).  I
> have several spare PRI interfaces and would like to register a  
> different
> PRI on the same GW with a different CM (specifically my CM 5.0 
> (3)).  Is
> that possible?
> -jc
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip


From Greg_Hinson at adp.com  Wed Jun 28 10:41:28 2006
From: Greg_Hinson at adp.com (Hinson, Greg)
Date: Wed, 28 Jun 2006 09:41:28 -0500
Subject: [cisco-voip] IOS GW PRI Registration to separate CallManagers
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BD074B3@EXCHANGEDS01.ds.ad.adp.com>

I believe you can run multiple MGCP processes, going to different
callmanagers. there is a command that is labeled 
"MGCP profile default" that is part of the running config parameters for
MGCP. You should be able to create multiple MGCP profiles, assigning
each their own mgcp call-agent, but I haven't tested that.

- Greg Hinson

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
Sent: Wednesday, June 28, 2006 7:25 AM
To: Justin Church
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] IOS GW PRI Registration to separate
CallManagers

I don't think you can run two mgcp processes on a router.

-Ryan

On Jun 28, 2006, at 10:20 AM, Justin Church wrote:

We're running 2821 IOS gateways using MGCP backhaul to CCM 4.1(3).  I
have several spare PRI interfaces and would like to register a different
PRI on the same GW with a different CM (specifically my CM 5.0(3)).  Is
that possible?

-jc
_______________________________________________
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 Jun 28 10:55:02 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 28 Jun 2006 09:55:02 -0500
Subject: [cisco-voip] Have you seen this article?
In-Reply-To: <008601c69abb$227bfa30$47196883@cfs.uoguelph.ca>
References: <963CE94787005149A52C17F33A028E12051D4940@EVS1B.boco.co.boulder.co.us>
	<6F5E2DDC-AA71-46FB-9FE3-6A4AB74CE20A@cisco.com>
	<008601c69abb$227bfa30$47196883@cfs.uoguelph.ca>
Message-ID: <5d093f9a0606280755p59110f6ehfbb6bf721a8b651f@mail.gmail.com>

CCM 4.3 isn't supposed to come out until end of the year...

It is supposed to be on Windows 2003...

I just checked Cisco's site, I saw no mention of it yet...



Jonathan

On 6/28/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
>
> There's a CallManager 4.3(1)? WAH?????????????????????
>
> --------------------------------------------------------------------------------
> 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: Ryan Ratliff
> To: Leetun, Rob
> Cc: ciscovoip
>
> Sent: Wednesday, June 28, 2006 9:36 AM
> Subject: Re: [cisco-voip] Have you seen this article?
>
> http://www.cisco.com/en/US/products/sw/voicesw/ps556/
> tsd_products_security_response09186a00806c0846.html
>
> -Ryan
>
> On Jun 28, 2006, at 9:17 AM, Leetun, Rob wrote:
>
>   Cisco Call Manager Flaw Could Invite Hackers
>
> Vulnerabilities in Cisco's Call Manager software could open the door
> for hackers to reconfigure VoIP settings and gain access to
> individual users' account information, according to researchers at
> Kansas City, Mo.-based solution provider FishNet Security.
>
> By Kevin McLaughlin, CRN
> Jun 19, 2006
> URL:http://www.ddj.com/dept/security/189500728
>
> Vulnerabilities in Cisco's Call Manager software could open the door
> for hackers to reconfigure VoIP settings and gain access to
> individual users' account information, according to researchers at
> Kansas City, Mo.-based solution provider FishNet Security.
>
> In a report issued Monday, Jake Reynolds, senior security engineer at
> FishNet, said the vulnerability affects versions 3.1 and higher of
> Call Manager, which handles call routing and call signaling functions
> in Cisco VoIP systems. A lack of input validation and output encoding
> in the Web administration interface for Call Manager could allow
> hackers to execute cross-site scripting attacks, Reynolds wrote.
>
> Cross site scripting attacks usually involve tricking users with
> access privileges into clicking on a URL in an email or Web page.
>
> In the Call Manager scenario, attackers would send a request to the
> Call Manager Web interface that causes malicious JavaScript to be
> included. If the administrator could be tricked into submitting this
> tainted request, the malicious code would execute in the victim's Web
> browser and potentially give attackers the ability to delete or
> reconfigure system components and gain access to confidential user
> information, according to the report.
>
> In a statement, Cisco's Product Security Incident Response Team
> (PSRIT) recommended that users verify link destinations before
> clicking on URLs.
>
> Although there are no workarounds for the issue, Cisco has fixed the
> vulnerability and fixes will be incorporated in all supported
> CallManager trains in versions 4.3(1), 4.2(3), 4.1(3)SR4 and 3.3(5)
> SR3, according to the statement.
>
> To guard against attacks, FishNet recommends that companies limit
> network connectivity to Call Manager wherever possible to prevent
> hackers from discovering public Web interfaces.
>
> "Simple Google queries are all an attacker needs in this case to
> obtain the target Call Manager address. There are few compelling
> reasons one could present that would justify public access to Call
> Manager web interfaces," according to the report.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>

From jonvoip at gmail.com  Wed Jun 28 11:06:03 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 28 Jun 2006 10:06:03 -0500
Subject: [cisco-voip] 7970 display question
In-Reply-To: <12916582.1151504483936.JavaMail.root@eastrmwml08.mgt.cox.net>
References: <12916582.1151504483936.JavaMail.root@eastrmwml08.mgt.cox.net>
Message-ID: <5d093f9a0606280806g3022b827mfea7979188d0d058@mail.gmail.com>

Good question...

I thought you could do it in an ini file (I checked the enterprise and
service parameters) but I just checked all of them and nothing... You
might have to edit the SQL directly, however, that could prove to be
incredibly dangerous.

You could just select 'invalid number format' and notify the users
that this means enter the # sign...



Jonathan

On 6/28/06, bababooey at cox.net <bababooey at cox.net> wrote:
> where do I set the custom msg?
> thanks
> ---- Jonathan Charles <jonvoip at gmail.com> wrote:
> > Include a Route Pattern without the #, block it, with a custom msg to add the #
> >
> >
> > Jonathan
> >
> > On 6/27/06, Bob A. Bowie <bababooey at cox.net> wrote:
> > > I have some numbers that when dialed I force the user to enter '#' to
> > > complete the dialed number.
> > > I am wondering if there is a way to have the display indicate to the user
> > > "hit # to complete call", either on the splash
> > > screen the dialed number is displayed on or the background.
> > >
> > > thanks
> > > bob
> > >
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
>
>

From robert.bell1 at comcast.net  Wed Jun 28 11:17:02 2006
From: robert.bell1 at comcast.net (robert.bell1 at comcast.net)
Date: Wed, 28 Jun 2006 15:17:02 +0000
Subject: [cisco-voip] Cisco & Avaya integration
Message-ID: <062820061517.18768.44A29D6E0008E289000049502200751150CE04040A0DD29B9D0A0D019D@comcast.net>

Maybe I'm crazy, but we are upgrading out layer 2 infrastructure to Cisco 3560 switches and the question was raised, what would that do to our Avaya VoIP solution (Please don't flame me, I have no choice over VoIP solutions).  I'm much more familiar with Cisco CallManager and CCME.  
 
I got this crazy idea though:  Why can't I make Avaya phones work with our CCME routers?  I'm not really sure how they operate, but I assume they are similar to Cisco phones.  I think I found some firmware I have to push to them, but I have a few questions:
1.  Has anyone successfully done what I propose (Avaya IP phones working on a Cisco VoIP network?  And if so can I see your config files relevant to making that work.  
2.  Does anyone know the requirements for the Avaya IP phones?  For instance, I know the Cisco phones require the firmware to be downloaded and pulls the config from the router.  It looks like the Avaya phones also need to pull the config as a file from a tftp server, but I really don't know anything about Avaya VoIP.  If anyone has a link to the basic requirements / operation I'd appreciate it. Oh, and FYI - I've got an Avaya 4606 phone to play with.
 
My basic goal was to make sure the phones were compatible with the PoE provided by the Cisco 3560 switches (which it is) and to make sure that it would continue to work with the existing Avaya VoIP solution (which it will). I just thought it would be neat to see them work integrated and possibly use SRST for remote survivability of our outsites using VoIP.  
 
B.
-------------- next part --------------
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From nmarus at gmail.com  Wed Jun 28 11:30:31 2006
From: nmarus at gmail.com (Nick Marus)
Date: Wed, 28 Jun 2006 11:30:31 -0400
Subject: [cisco-voip] Have you seen this article?
In-Reply-To: <5d093f9a0606280700i2feb5284v28779598a25a7082@mail.gmail.com>
References: <F2E442781002EA41B3DB013749748D3033A1EF@gvlexch02.is.l-3com.com>
	<5d093f9a0606280700i2feb5284v28779598a25a7082@mail.gmail.com>
Message-ID: <1a9f6d3f0606280830o1d22d326x44a3570f6010ccf8@mail.gmail.com>

What is their ip? heheheheh ;)

On 6/28/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> When I worked at my last company, we had a cust with their CCMs
> available via RDP over the internet (directly)... the username and
> password was administrator cisco...
>
> Eeek
>
>
>
> Jonathan
>
> On 6/28/06, Fretz, EA Eric @ IS <Eric.A.Fretz at l-3com.com> wrote:
> > Any network admin that leaves his/her CCM web server facing the internet
> > needs a good flogging.
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
> > Sent: Wednesday, June 28, 2006 8:37 AM
> > To: Leetun, Rob
> > Cc: ciscovoip
> > Subject: Re: [cisco-voip] Have you seen this article?
> >
> >
> > http://www.cisco.com/en/US/products/sw/voicesw/ps556/
> > tsd_products_security_response09186a00806c0846.html
> >
> > -Ryan
> >
> > On Jun 28, 2006, at 9:17 AM, Leetun, Rob wrote:
> >
> >   Cisco Call Manager Flaw Could Invite Hackers
> >
> > Vulnerabilities in Cisco's Call Manager software could open the door
> > for hackers to reconfigure VoIP settings and gain access to
> > individual users' account information, according to researchers at
> > Kansas City, Mo.-based solution provider FishNet Security.
> >
> > By Kevin McLaughlin, CRN
> > Jun 19, 2006
> > URL:http://www.ddj.com/dept/security/189500728
> >
> > Vulnerabilities in Cisco's Call Manager software could open the door
> > for hackers to reconfigure VoIP settings and gain access to
> > individual users' account information, according to researchers at
> > Kansas City, Mo.-based solution provider FishNet Security.
> >
> > In a report issued Monday, Jake Reynolds, senior security engineer at
> > FishNet, said the vulnerability affects versions 3.1 and higher of
> > Call Manager, which handles call routing and call signaling functions
> > in Cisco VoIP systems. A lack of input validation and output encoding
> > in the Web administration interface for Call Manager could allow
> > hackers to execute cross-site scripting attacks, Reynolds wrote.
> >
> > Cross site scripting attacks usually involve tricking users with
> > access privileges into clicking on a URL in an email or Web page.
> >
> > In the Call Manager scenario, attackers would send a request to the
> > Call Manager Web interface that causes malicious JavaScript to be
> > included. If the administrator could be tricked into submitting this
> > tainted request, the malicious code would execute in the victim's Web
> > browser and potentially give attackers the ability to delete or
> > reconfigure system components and gain access to confidential user
> > information, according to the report.
> >
> > In a statement, Cisco's Product Security Incident Response Team
> > (PSRIT) recommended that users verify link destinations before
> > clicking on URLs.
> >
> > Although there are no workarounds for the issue, Cisco has fixed the
> > vulnerability and fixes will be incorporated in all supported
> > CallManager trains in versions 4.3(1), 4.2(3), 4.1(3)SR4 and 3.3(5)
> > SR3, according to the statement.
> >
> > To guard against attacks, FishNet recommends that companies limit
> > network connectivity to Call Manager wherever possible to prevent
> > hackers from discovering public Web interfaces.
> >
> > "Simple Google queries are all an attacker needs in this case to
> > obtain the target Call Manager address. There are few compelling
> > reasons one could present that would justify public access to Call
> > Manager web interfaces," according to the report.
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


-- 

Nick Marus
nmarus at gmail.com
homepage: http://www.nicholasmarus.com

From nmarus at gmail.com  Wed Jun 28 11:52:03 2006
From: nmarus at gmail.com (Nick Marus)
Date: Wed, 28 Jun 2006 11:52:03 -0400
Subject: [cisco-voip] Cisco & Avaya integration
In-Reply-To: <062820061517.18768.44A29D6E0008E289000049502200751150CE04040A0DD29B9D0A0D019D@comcast.net>
References: <062820061517.18768.44A29D6E0008E289000049502200751150CE04040A0DD29B9D0A0D019D@comcast.net>
Message-ID: <1a9f6d3f0606280852l7bad85cu8a4b7873a410a331@mail.gmail.com>

I work for a company that deploys both avaya and cisco voip and we
have been asked to similar stuff. I have done the opposite, that is
getting a cisco phone with a sip image working on an avaya. For what
you want, as far as i know, this would require the Avaya phones to be
running the avaya sip image and you would need cm5 for sip phone set
support. I am wanting to lab up something similar myself using the
4610/4620's. The 4606 is an older avaya phone and may not support sip,
and I dont think it is being sold anymore. I cant recall tho.

For #2, Cisco Phones use tftp just as avaya does. the server can be
anything that can do host tftp. Avaya's files are slightly differnt in
format, but essentially the same. A config file and a firmware load.
Avaya does require differnt dhcp options to tell the phone where the
tftp server is tho. Cisco =150, avaya is 176 and uses a string that
includes several required options.

Everything between avaya and Cisco can integrate (both directions) if
you use h323 on your cisco gateways. i have a callmanager that gets
trunks directly from an avaya s8700 with no external cisco gateway and
I have worked on some other sites that have avaya trunking to a cisco
h323 gateway.

Nick

On 6/28/06, robert.bell1 at comcast.net <robert.bell1 at comcast.net> wrote:
>
>
>
> Maybe I'm crazy, but we are upgrading out layer 2 infrastructure to Cisco
> 3560 switches and the question was raised, what would that do to our Avaya
> VoIP solution (Please don't flame me, I have no choice over VoIP solutions).
>  I'm much more familiar with Cisco CallManager and CCME.
>
>
>
> I got this crazy idea though:  Why can't I make Avaya phones work with our
> CCME routers?  I'm not really sure how they operate, but I assume they are
> similar to Cisco phones.  I think I found some firmware I have to push to
> them, but I have a few questions:
>
> 1.  Has anyone successfully done what I propose (Avaya IP phones working on
> a Cisco VoIP network?  And if so can I see your config files relevant to
> making that work.
>
> 2.  Does anyone know the requirements for the Avaya IP phones?  For
> instance, I know the Cisco phones require the firmware to be downloaded and
> pulls the config from the router.  It looks like the Avaya phones also need
> to pull the config as a file from a tftp server, but I really don't know
> anything about Avaya VoIP.  If anyone has a link to the basic requirements /
> operation I'd appreciate it. Oh, and FYI - I've got an Avaya 4606 phone to
> play with.
>
>
>
> My basic goal was to make sure the phones were compatible with the PoE
> provided by the Cisco 3560 switches (which it is) and to make sure that it
> would continue to work with the existing Avaya VoIP solution (which it
> will). I just thought it would be neat to see them work integrated and
> possibly use SRST for remote survivability of our outsites using VoIP.
>
>
>
> B.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 

Nick Marus
nmarus at gmail.com
homepage: http://www.nicholasmarus.com

From akoul at dataways.gr  Wed Jun 28 12:03:55 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Wed, 28 Jun 2006 19:03:55 +0300
Subject: [cisco-voip] clid on fxs ports
Message-ID: <44A2A86B.4030700@dataways.gr>

Hi all,
on a 2851 with one E1 and 4 FXS ports i have configured the voice port 
as follows:

voice-port 0/3/2
 cptone GB
 description xxx
 station-id name yyy
 station-id number 1234
 caller-id enable type 1


there is also these:

voice-port 0/2/0:15
 translation-profile incoming TP01
 bearer-cap Speech
interface Serial0/2/0:15
 no ip address
 isdn switch-type primary-net5
 isdn incoming-voice voice
 no cdp enable


dial-peer voice 10 pots
 destination-pattern 9T
 port 0/2/0:15


when i call from the analog phone to an ipphone extension internally i 
have clid (both name and number) to the ipphone

but when i make calls through the pri, i get "uknown number" to the 
external phone that i have called.
Is this ficed somehow?

Cheers
Anthony

From rratliff at cisco.com  Wed Jun 28 12:07:39 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Wed, 28 Jun 2006 12:07:39 -0400
Subject: [cisco-voip] clid on fxs ports
In-Reply-To: <44A2A86B.4030700@dataways.gr>
References: <44A2A86B.4030700@dataways.gr>
Message-ID: <7747FB40-03EA-472C-B947-95F8336708A9@cisco.com>

Do you have a q931 debug showing the outbound call?

-Ryan

On Jun 28, 2006, at 12:03 PM, Anthony Kouloglou wrote:

Hi all,
on a 2851 with one E1 and 4 FXS ports i have configured the voice port
as follows:

voice-port 0/3/2
  cptone GB
  description xxx
  station-id name yyy
  station-id number 1234
  caller-id enable type 1


there is also these:

voice-port 0/2/0:15
  translation-profile incoming TP01
  bearer-cap Speech
interface Serial0/2/0:15
  no ip address
  isdn switch-type primary-net5
  isdn incoming-voice voice
  no cdp enable


dial-peer voice 10 pots
  destination-pattern 9T
  port 0/2/0:15


when i call from the analog phone to an ipphone extension internally i
have clid (both name and number) to the ipphone

but when i make calls through the pri, i get "uknown number" to the
external phone that i have called.
Is this ficed somehow?

Cheers
Anthony
_______________________________________________
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  Wed Jun 28 13:26:16 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 28 Jun 2006 13:26:16 -0400
Subject: [cisco-voip] Installing IPCC Express 4.5
Message-ID: <C1FE15183DA37645BC0633BC604E44F00253EA0C@USNAEXCH.na.didata.local>

IPCC 4.5 is for use with CallManager 5.0.  No upgrade path to IPCC 4.5,
new deployments only.  There is expected a IPCC 4.6 for upgrading IPCC
for use with CCM 5.X and 4.3 (Win2k3)

 

Changes to using SOAP AXL for Communication with CallManager.

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Hansel Gideon
Gunawan
Sent: Tuesday, June 20, 2006 2:26 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Installing IPCC Express 4.5

 

Hi all,

 

I am wondering if we can install IPCC Express 4.5 in a generic server.
Will the registry tweak for previous version of IPCCX works? I have
searched the archive and cant find anything related to this version of
IPCC Express.

 

 

Regards,

 

Hansel Gideon Gunawan




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From tednugent69 at yahoo.com  Wed Jun 28 14:03:54 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 28 Jun 2006 11:03:54 -0700 (PDT)
Subject: [cisco-voip] Link to primary CCM fails while logged in with
	Extension Mobility
In-Reply-To: <69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
Message-ID: <20060628180354.7937.qmail@web36106.mail.mud.yahoo.com>

I stand corrected... thanks

--- Ryan Ratliff <rratliff at cisco.com> wrote:

> I just tested with a 7970 and when the phone drops
> into SRST mode it  
> keeps the EM profile dn(s).  Since you don't
> configure ephone/dns in  
> SRST the phone keeps whatever dns it had before it
> unregistered with CM.
> 
> -Ryan
> 
> On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
> 
> The phone resets and goes to it's default profile.
> EM
> will not work in SRST since the EM service runs on
> CCM
> which is not longer in the picture
> 
> --- Anthony Kouloglou <akoul at dataways.gr> wrote:
> 
> > Hi Ryan,
> > i am sorry that i didn't clarify it but i was
> > refering to the situation
> > where there is an SRST involved:
> > there is a CCM cluster and an SRST configured at
> the
> > Device Pool to the
> > remote site.
> > At the remote site a user logs in with EM but at
> the
> > USER device profile
> > there is no reference about the device pool.
> > What happens to that phone while it is logged in
> > with EM and the remote
> > site loses connectivity with all the CCM cluster?
> >
> > Cheers
> > Anthony
> >
> > Ryan Ratliff wrote:
> >> Just to clarify when you login with EM it
> actually
> > changes the
> >> database so no matter which server the phone
> > registers with as long as
> >> that EM profile is logged in the phone will get
> > the EM lines.  The
> >> user won't get automatically logged out or
> > anything just because the
> >> phone fails over to another server.
> >>
> >> -Ryan
> >>
> >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
> >>
> >> The phone will act just like it would if you were
> > not logged in with
> >> EM, ie it will register with the backup
> > CallManager server.
> >>
> >> -Ryan
> >>
> >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
> > wrote:
> >>
> >> Hi all,
> >> if a user is logged in with extension mobility,
> > and the link to the
> >> primary ccm fails what happens with that phone?
> >>
> >>
> >> _______________________________________________
> >> 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
> >
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around
> http://mail.yahoo.com
> 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From jonvoip at gmail.com  Wed Jun 28 14:46:43 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Wed, 28 Jun 2006 13:46:43 -0500
Subject: [cisco-voip] Have you seen this article?
In-Reply-To: <1a9f6d3f0606280830o1d22d326x44a3570f6010ccf8@mail.gmail.com>
References: <F2E442781002EA41B3DB013749748D3033A1EF@gvlexch02.is.l-3com.com>
	<5d093f9a0606280700i2feb5284v28779598a25a7082@mail.gmail.com>
	<1a9f6d3f0606280830o1d22d326x44a3570f6010ccf8@mail.gmail.com>
Message-ID: <5d093f9a0606281146g564333d1jaeb693761889648f@mail.gmail.com>

I have blotted everything else about them from my memory.


J

On 6/28/06, Nick Marus <nmarus at gmail.com> wrote:
> What is their ip? heheheheh ;)
>
> On 6/28/06, Jonathan Charles <jonvoip at gmail.com> wrote:
> > When I worked at my last company, we had a cust with their CCMs
> > available via RDP over the internet (directly)... the username and
> > password was administrator cisco...
> >
> > Eeek
> >
> >
> >
> > Jonathan
> >
> > On 6/28/06, Fretz, EA Eric @ IS <Eric.A.Fretz at l-3com.com> wrote:
> > > Any network admin that leaves his/her CCM web server facing the internet
> > > needs a good flogging.
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Ryan Ratliff
> > > Sent: Wednesday, June 28, 2006 8:37 AM
> > > To: Leetun, Rob
> > > Cc: ciscovoip
> > > Subject: Re: [cisco-voip] Have you seen this article?
> > >
> > >
> > > http://www.cisco.com/en/US/products/sw/voicesw/ps556/
> > > tsd_products_security_response09186a00806c0846.html
> > >
> > > -Ryan
> > >
> > > On Jun 28, 2006, at 9:17 AM, Leetun, Rob wrote:
> > >
> > >   Cisco Call Manager Flaw Could Invite Hackers
> > >
> > > Vulnerabilities in Cisco's Call Manager software could open the door
> > > for hackers to reconfigure VoIP settings and gain access to
> > > individual users' account information, according to researchers at
> > > Kansas City, Mo.-based solution provider FishNet Security.
> > >
> > > By Kevin McLaughlin, CRN
> > > Jun 19, 2006
> > > URL:http://www.ddj.com/dept/security/189500728
> > >
> > > Vulnerabilities in Cisco's Call Manager software could open the door
> > > for hackers to reconfigure VoIP settings and gain access to
> > > individual users' account information, according to researchers at
> > > Kansas City, Mo.-based solution provider FishNet Security.
> > >
> > > In a report issued Monday, Jake Reynolds, senior security engineer at
> > > FishNet, said the vulnerability affects versions 3.1 and higher of
> > > Call Manager, which handles call routing and call signaling functions
> > > in Cisco VoIP systems. A lack of input validation and output encoding
> > > in the Web administration interface for Call Manager could allow
> > > hackers to execute cross-site scripting attacks, Reynolds wrote.
> > >
> > > Cross site scripting attacks usually involve tricking users with
> > > access privileges into clicking on a URL in an email or Web page.
> > >
> > > In the Call Manager scenario, attackers would send a request to the
> > > Call Manager Web interface that causes malicious JavaScript to be
> > > included. If the administrator could be tricked into submitting this
> > > tainted request, the malicious code would execute in the victim's Web
> > > browser and potentially give attackers the ability to delete or
> > > reconfigure system components and gain access to confidential user
> > > information, according to the report.
> > >
> > > In a statement, Cisco's Product Security Incident Response Team
> > > (PSRIT) recommended that users verify link destinations before
> > > clicking on URLs.
> > >
> > > Although there are no workarounds for the issue, Cisco has fixed the
> > > vulnerability and fixes will be incorporated in all supported
> > > CallManager trains in versions 4.3(1), 4.2(3), 4.1(3)SR4 and 3.3(5)
> > > SR3, according to the statement.
> > >
> > > To guard against attacks, FishNet recommends that companies limit
> > > network connectivity to Call Manager wherever possible to prevent
> > > hackers from discovering public Web interfaces.
> > >
> > > "Simple Google queries are all an attacker needs in this case to
> > > obtain the target Call Manager address. There are few compelling
> > > reasons one could present that would justify public access to Call
> > > Manager web interfaces," according to the report.
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
>
> --
>
> Nick Marus
> nmarus at gmail.com
> homepage: http://www.nicholasmarus.com
>

From tednugent69 at yahoo.com  Wed Jun 28 14:49:25 2006
From: tednugent69 at yahoo.com (Ted Nugent)
Date: Wed, 28 Jun 2006 11:49:25 -0700 (PDT)
Subject: [cisco-voip] Link to primary CCM fails while logged in with
	Extension Mobility
In-Reply-To: <20060628180354.7937.qmail@web36106.mail.mud.yahoo.com>
Message-ID: <20060628184925.11271.qmail@web36111.mail.mud.yahoo.com>

FYI, and this may not be relevant because of the
vintage however you should check the results on the
particular versions you are running in production
because it obviously varies by version. I would have
expected the results that Ryan outlined however when I
just tested this in the lab with (CM3.3(4) SRST 3.1)
the phone reset and registered with SRST using the
phones default profile as I remember it doing. I'm
guessing that this is either a bug in the versions I'm
running or maintaining the logged in profile was added
in newer releases?


--- Ted Nugent <tednugent69 at yahoo.com> wrote:

> I stand corrected... thanks
> 
> --- Ryan Ratliff <rratliff at cisco.com> wrote:
> 
> > I just tested with a 7970 and when the phone drops
> > into SRST mode it  
> > keeps the EM profile dn(s).  Since you don't
> > configure ephone/dns in  
> > SRST the phone keeps whatever dns it had before it
> > unregistered with CM.
> > 
> > -Ryan
> > 
> > On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
> > 
> > The phone resets and goes to it's default profile.
> > EM
> > will not work in SRST since the EM service runs on
> > CCM
> > which is not longer in the picture
> > 
> > --- Anthony Kouloglou <akoul at dataways.gr> wrote:
> > 
> > > Hi Ryan,
> > > i am sorry that i didn't clarify it but i was
> > > refering to the situation
> > > where there is an SRST involved:
> > > there is a CCM cluster and an SRST configured at
> > the
> > > Device Pool to the
> > > remote site.
> > > At the remote site a user logs in with EM but at
> > the
> > > USER device profile
> > > there is no reference about the device pool.
> > > What happens to that phone while it is logged in
> > > with EM and the remote
> > > site loses connectivity with all the CCM
> cluster?
> > >
> > > Cheers
> > > Anthony
> > >
> > > Ryan Ratliff wrote:
> > >> Just to clarify when you login with EM it
> > actually
> > > changes the
> > >> database so no matter which server the phone
> > > registers with as long as
> > >> that EM profile is logged in the phone will get
> > > the EM lines.  The
> > >> user won't get automatically logged out or
> > > anything just because the
> > >> phone fails over to another server.
> > >>
> > >> -Ryan
> > >>
> > >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff
> wrote:
> > >>
> > >> The phone will act just like it would if you
> were
> > > not logged in with
> > >> EM, ie it will register with the backup
> > > CallManager server.
> > >>
> > >> -Ryan
> > >>
> > >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
> > > wrote:
> > >>
> > >> Hi all,
> > >> if a user is logged in with extension mobility,
> > > and the link to the
> > >> primary ccm fails what happens with that phone?
> > >>
> > >>
> > >> _______________________________________________
> > >> 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
> > >
> > 
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam
> > protection around
> > http://mail.yahoo.com
> > 
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> >
> https://puck.nether.net/mailman/listinfo/cisco-voip
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> http://mail.yahoo.com 
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
> 


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

From ealeatherman at gmail.com  Wed Jun 28 15:20:34 2006
From: ealeatherman at gmail.com (Ed Leatherman)
Date: Wed, 28 Jun 2006 15:20:34 -0400
Subject: [cisco-voip] CCM download
In-Reply-To: <5d093f9a0606280658r5a1f3f67n15760c7f253f2b78@mail.gmail.com>
References: <407055A92CECCB499C922A2D35FC19A605ED52E3@Apollo.wesd.org>
	<de85c96f0606272225m235e7916v1ca91b6cb9566a70@mail.gmail.com>
	<5d093f9a0606280658r5a1f3f67n15760c7f253f2b78@mail.gmail.com>
Message-ID: <94a1afde0606281220i47f50f83n640a05876f67c588@mail.gmail.com>

Won't he still need an upgrade kit to get from 3x to 4x? ie the discs rather
than just a patch file? been too long since we did that.

On 6/28/06, Jonathan Charles <jonvoip at gmail.com> wrote:
>
> Basically, you need to get a copy of 3.3 or better, then you can
> download upgrades all the way to 4.2
>
>
> Jonathan
>
> On 6/28/06, Jorge Evangelista <netsecuredata at gmail.com> wrote:
> > There was a version beta for CCM 5.0 that you could download from
> > Cisco with your account CCO. But Cisco retired CCM beta months later.
> > I have heard in a forum people that said configure CCM 5.0 on VMware.
> > Also, I am looking for CCM for Linux.
> > Is there a certain link in Internet I can download CCM beta?
> >
> >
> > On 6/27/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
> > >
> > >
> > >
> > > Call Manager is not on the CCO to download.  Only updates.  You will
> need to
> > > order. Or PUT it.
> > >
> > >
> > >
> > > Scott
> > >
> > >
> > > ________________________________
> > >
> > >
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of
> > > Curt Shaffer
> > > Sent: Monday, June 26, 2006 3:15 PM
> > > To: cisco-voip at puck.nether.net
> > > Subject: [cisco-voip] CCM download
> > >
> > >
> > >
> > >
> > > I heard a while back that you can download Cisco Call Manger for free
> but I
> > > can't find it. I really dislike the navigation on their site
> sometimes. Can
> > > anyone point me in the right direction?
> > >
> > >
> > >
> > >
> > >
> > > Thanks
> > >
> > >
> > >
> > > Curt
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >
> >
> >
> > --
> > "The network is the computer"
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



-- 
Ed Leatherman
IP Telephony Coordinator
West Virginia University
Telecommunications and Network Operations
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From jsteinberg at gmail.com  Wed Jun 28 16:13:09 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 28 Jun 2006 16:13:09 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
In-Reply-To: <92535326-43F1-445E-822F-509089040E09@cisco.com>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
	<f1b65f880606280638o5e24f889x3fa9341364cb56d3@mail.gmail.com>
	<92535326-43F1-445E-822F-509089040E09@cisco.com>
Message-ID: <f1b65f880606281313k7b45ce1ehced66b043a8252e9@mail.gmail.com>

I did a little afternoon testing on this power cycle question I had.

I registered a 7960 running FW 7.2(2) to CCM in my lab and then
stopped the CCM service.  The phone successfully failed over to my
SRST 3.3 router.

Then I power cycled the SRST router and when the router came back
online the phone registered back to SRST with the correct DN.
Rebooted the phone and it still reregistered with the correct DN.

It even registered back to SRST with the correct DN after power
cycling the phone and SRST router at the same time.

I then power cycled the router and did a factory reset procedure on
the 7960, choosing to save the network config during the process.
After the phone reset it registered back to the SRST router but
without any DN.

So the phone must save the most recent DN somewhere in flash and
retains this even through power failure.

I did not test this with EM.

Justin




On 6/28/06, Ryan Ratliff <rratliff at cisco.com> wrote:
> My 7970 kept the EM dn through a power cycle while in SRST mode.
>
> -Ryan
>
> On Jun 28, 2006, at 9:38 AM, Justin Steinberg wrote:
>
> This isn't specific to EM.   But what happens if you are operating in
> SRST mode and you lose power to all the phones and the SRST router.
> Do the phones or router keep record of their last DN in flash or would
> you be completely down after power came back with access to the CM
> cluster still unavailable.
>
> Justin
>
> On 6/28/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> >
> > I would expect the same to happen if the PUB went down and access
> > to the
> > SUBs was still available.
> >
> > Can I assume that during this time, no logins or logouts would be
> > 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: Ryan Ratliff
> > To: Ted Nugent
> > Cc: cisco-voip at puck-nether.net
> > Sent: Wednesday, June 28, 2006 9:28 AM
> > Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
> > withExtension Mobility
> >
> > I just tested with a 7970 and when the phone drops into SRST mode it
> > keeps the EM profile dn(s).  Since you don't configure ephone/dns in
> > SRST the phone keeps whatever dns it had before it unregistered
> > with CM.
> >
> > -Ryan
> >
> > On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
> >
> > The phone resets and goes to it's default profile. EM
> > will not work in SRST since the EM service runs on CCM
> > which is not longer in the picture
> >
> > --- Anthony Kouloglou <akoul at dataways.gr> wrote:
> >
> > > Hi Ryan,
> > > i am sorry that i didn't clarify it but i was
> > > refering to the situation
> > > where there is an SRST involved:
> > > there is a CCM cluster and an SRST configured at the
> > > Device Pool to the
> > > remote site.
> > > At the remote site a user logs in with EM but at the
> > > USER device profile
> > > there is no reference about the device pool.
> > > What happens to that phone while it is logged in
> > > with EM and the remote
> > > site loses connectivity with all the CCM cluster?
> > >
> > > Cheers
> > > Anthony
> > >
> > > Ryan Ratliff wrote:
> > >> Just to clarify when you login with EM it actually
> > > changes the
> > >> database so no matter which server the phone
> > > registers with as long as
> > >> that EM profile is logged in the phone will get
> > > the EM lines.  The
> > >> user won't get automatically logged out or
> > > anything just because the
> > >> phone fails over to another server.
> > >>
> > >> -Ryan
> > >>
> > >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
> > >>
> > >> The phone will act just like it would if you were
> > > not logged in with
> > >> EM, ie it will register with the backup
> > > CallManager server.
> > >>
> > >> -Ryan
> > >>
> > >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
> > > wrote:
> > >>
> > >> Hi all,
> > >> if a user is logged in with extension mobility,
> > > and the link to the
> > >> primary ccm fails what happens with that phone?
> > >>
> > >>
> > >> _______________________________________________
> > >> 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
> > >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>
>

From lelio at uoguelph.ca  Wed Jun 28 16:17:46 2006
From: lelio at uoguelph.ca (Lelio Fulgenzi)
Date: Wed, 28 Jun 2006 16:17:46 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
	<f1b65f880606280638o5e24f889x3fa9341364cb56d3@mail.gmail.com>
	<92535326-43F1-445E-822F-509089040E09@cisco.com>
	<f1b65f880606281313k7b45ce1ehced66b043a8252e9@mail.gmail.com>
Message-ID: <022401c69aef$ebab4850$47196883@cfs.uoguelph.ca>

I'm wondering if stopping the CCM service is enough of a test. Could there perhaps be another service that the SRST router is connecting to?

I don't see how the SRST router and phones can continue to operate after a reboot if connectivity to the cluster is lost.

--------------------------------------------------------------------------------
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: Justin Steinberg 
  To: Ryan Ratliff 
  Cc: Lelio Fulgenzi ; Ted Nugent ; cisco-voip at puck-nether.net 
  Sent: Wednesday, June 28, 2006 4:13 PM
  Subject: Re: [cisco-voip] Link to primary CCM fails while logged in withExtension Mobility


  I did a little afternoon testing on this power cycle question I had.

  I registered a 7960 running FW 7.2(2) to CCM in my lab and then
  stopped the CCM service.  The phone successfully failed over to my
  SRST 3.3 router.

  Then I power cycled the SRST router and when the router came back
  online the phone registered back to SRST with the correct DN.
  Rebooted the phone and it still reregistered with the correct DN.

  It even registered back to SRST with the correct DN after power
  cycling the phone and SRST router at the same time.

  I then power cycled the router and did a factory reset procedure on
  the 7960, choosing to save the network config during the process.
  After the phone reset it registered back to the SRST router but
  without any DN.

  So the phone must save the most recent DN somewhere in flash and
  retains this even through power failure.

  I did not test this with EM.

  Justin




  On 6/28/06, Ryan Ratliff <rratliff at cisco.com> wrote:
  > My 7970 kept the EM dn through a power cycle while in SRST mode.
  >
  > -Ryan
  >
  > On Jun 28, 2006, at 9:38 AM, Justin Steinberg wrote:
  >
  > This isn't specific to EM.   But what happens if you are operating in
  > SRST mode and you lose power to all the phones and the SRST router.
  > Do the phones or router keep record of their last DN in flash or would
  > you be completely down after power came back with access to the CM
  > cluster still unavailable.
  >
  > Justin
  >
  > On 6/28/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
  > >
  > > I would expect the same to happen if the PUB went down and access
  > > to the
  > > SUBs was still available.
  > >
  > > Can I assume that during this time, no logins or logouts would be
  > > 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: Ryan Ratliff
  > > To: Ted Nugent
  > > Cc: cisco-voip at puck-nether.net
  > > Sent: Wednesday, June 28, 2006 9:28 AM
  > > Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
  > > withExtension Mobility
  > >
  > > I just tested with a 7970 and when the phone drops into SRST mode it
  > > keeps the EM profile dn(s).  Since you don't configure ephone/dns in
  > > SRST the phone keeps whatever dns it had before it unregistered
  > > with CM.
  > >
  > > -Ryan
  > >
  > > On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
  > >
  > > The phone resets and goes to it's default profile. EM
  > > will not work in SRST since the EM service runs on CCM
  > > which is not longer in the picture
  > >
  > > --- Anthony Kouloglou <akoul at dataways.gr> wrote:
  > >
  > > > Hi Ryan,
  > > > i am sorry that i didn't clarify it but i was
  > > > refering to the situation
  > > > where there is an SRST involved:
  > > > there is a CCM cluster and an SRST configured at the
  > > > Device Pool to the
  > > > remote site.
  > > > At the remote site a user logs in with EM but at the
  > > > USER device profile
  > > > there is no reference about the device pool.
  > > > What happens to that phone while it is logged in
  > > > with EM and the remote
  > > > site loses connectivity with all the CCM cluster?
  > > >
  > > > Cheers
  > > > Anthony
  > > >
  > > > Ryan Ratliff wrote:
  > > >> Just to clarify when you login with EM it actually
  > > > changes the
  > > >> database so no matter which server the phone
  > > > registers with as long as
  > > >> that EM profile is logged in the phone will get
  > > > the EM lines.  The
  > > >> user won't get automatically logged out or
  > > > anything just because the
  > > >> phone fails over to another server.
  > > >>
  > > >> -Ryan
  > > >>
  > > >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
  > > >>
  > > >> The phone will act just like it would if you were
  > > > not logged in with
  > > >> EM, ie it will register with the backup
  > > > CallManager server.
  > > >>
  > > >> -Ryan
  > > >>
  > > >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
  > > > wrote:
  > > >>
  > > >> Hi all,
  > > >> if a user is logged in with extension mobility,
  > > > and the link to the
  > > >> primary ccm fails what happens with that phone?
  > > >>
  > > >>
  > > >> _______________________________________________
  > > >> 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
  > > >
  > >
  > >
  > > __________________________________________________
  > > Do You Yahoo!?
  > > Tired of spam?  Yahoo! Mail has the best spam protection around
  > > http://mail.yahoo.com
  > >
  > > _______________________________________________
  > > cisco-voip mailing list
  > > cisco-voip at puck.nether.net
  > > https://puck.nether.net/mailman/listinfo/cisco-voip
  > >
  > > _______________________________________________
  > > cisco-voip mailing list
  > > cisco-voip at puck.nether.net
  > > https://puck.nether.net/mailman/listinfo/cisco-voip
  > >
  > >
  > >
  >
  >
-------------- next part --------------
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From wsisk at cisco.com  Wed Jun 28 16:25:59 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 28 Jun 2006 16:25:59 -0400
Subject: [cisco-voip] 7941/7961 Tone Generator
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0024FB9D0@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0024FB9D0@USNAEXCH.na.didata.local>
Message-ID: <44A2E5D7.1060901@cisco.com>

There are test tones available in IOS:

http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps5207/products_command_reference_chapter09186a00801a7f3e.html#wp1644472

/Wes

Jason Aarons (US) wrote:
>
> Is there another instrument/test tool that would generate 1004 Hz tone 
> at -15 dB besides the phone to test ERL?
>
> Is there a workaround for measuring ELR if the phone won?t generate a 
> test tone.
>
> Would a T-BERD or similar measure ERL in a T1 CAS environment?
>
> ------------------------------------------------------------------------
>
> *From:* Wes Sisk [mailto:wsisk at cisco.com]
> *Sent:* Tuesday, June 27, 2006 10:26 AM
> *To:* Jason Aarons (US)
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] 7941/7961 Tone Generator
>
> CSCee44462 - Need ability to generate inband test tone on 7970 like 
> 7960 does
>
> /Wes
>
> On Jun 27, 2006, at 11:47 AM, Jason Aarons ((US)) wrote:
>
> **3 turns on ?Debug Mode? on a 7941, however I don?t get a Tone 
> Generator softkey when placing a call. I?ve done this before on 
> 7940/7960s with no issues.
>
> Anyone tried the 41/61/70 ? These phones run a Java applet?
>
> http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a00800945bd.shtml#testtone
>
> ------------------------------------------------------------------------
>
> *
> **Disclaimer:**
>
> **This e-mail communication and any attachments may contain**
> **confidential and privileged information and is for use by the**
> **designated addressee(s) named above only. If you are not the**
> **intended addressee, you are hereby notified that you have received**
> **this communication in error and that any use or reproduction of**
> **this email or its contents is strictly prohibited and may be**
> **unlawful. If you have received this communication in error, please**
> **notify us immediately by replying to this message and deleting it**
> **from your computer. Thank you.***
>
> _______________________________________________
>
> cisco-voip mailing list
>
> cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
>
> https://puck..nether.net/mailman/listinfo/cisco-voip 
> <https://puck.nether.net/mailman/listinfo/cisco-voip>
>
> ------------------------------------------------------------------------
>
> *
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only. If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful. If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
> *
>

From jsteinberg at gmail.com  Wed Jun 28 16:38:08 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Wed, 28 Jun 2006 16:38:08 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged in
	withExtension Mobility
In-Reply-To: <022401c69aef$ebab4850$47196883@cfs.uoguelph.ca>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>
	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>
	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>
	<f1b65f880606280638o5e24f889x3fa9341364cb56d3@mail.gmail.com>
	<92535326-43F1-445E-822F-509089040E09@cisco.com>
	<f1b65f880606281313k7b45ce1ehced66b043a8252e9@mail.gmail.com>
	<022401c69aef$ebab4850$47196883@cfs.uoguelph.ca>
Message-ID: <f1b65f880606281338s6a3c0a22la398210346e300f2@mail.gmail.com>

I'm going to retract my earlier statements.

After shutting down the CallManagers completely, the 7960 (FW 7.22)
phone does not register back to SRST router after power cycling the
phone.

Perhaps the phone needs to TFTP from the CCM before it can register to
the SRST router.

This seems like a limitation of SRST.  I can see a bad storm knocking
out a T1 line and commercial power.

Is there something I'm missing or is the answer to have a PoE switch
and a SRST router on a UPS?

Justin




On 6/28/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
> I'm wondering if stopping the CCM service is enough of a test. Could there
> perhaps be another service that the SRST router is connecting to?
>
> I don't see how the SRST router and phones can continue to operate after a
> reboot if connectivity to the cluster is lost.
>
> --------------------------------------------------------------------------------
>
> 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: Justin Steinberg
> To: Ryan Ratliff
> Cc: Lelio Fulgenzi ; Ted Nugent ; cisco-voip at puck-nether.net
> Sent: Wednesday, June 28, 2006 4:13 PM
> Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
> withExtension Mobility
>
> I did a little afternoon testing on this power cycle question I had.
>
> I registered a 7960 running FW 7.2(2) to CCM in my lab and then
> stopped the CCM service.  The phone successfully failed over to my
> SRST 3.3 router.
>
> Then I power cycled the SRST router and when the router came back
> online the phone registered back to SRST with the correct DN.
> Rebooted the phone and it still reregistered with the correct DN.
>
> It even registered back to SRST with the correct DN after power
> cycling the phone and SRST router at the same time.
>
> I then power cycled the router and did a factory reset procedure on
> the 7960, choosing to save the network config during the process.
> After the phone reset it registered back to the SRST router but
> without any DN.
>
> So the phone must save the most recent DN somewhere in flash and
> retains this even through power failure.
>
> I did not test this with EM.
>
> Justin
>
>
>
>
> On 6/28/06, Ryan Ratliff <rratliff at cisco.com> wrote:
> > My 7970 kept the EM dn through a power cycle while in SRST mode.
> >
> > -Ryan
> >
> > On Jun 28, 2006, at 9:38 AM, Justin Steinberg wrote:
> >
> > This isn't specific to EM.   But what happens if you are operating in
> > SRST mode and you lose power to all the phones and the SRST router.
> > Do the phones or router keep record of their last DN in flash or would
> > you be completely down after power came back with access to the CM
> > cluster still unavailable.
> >
> > Justin
> >
> > On 6/28/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > >
> > > I would expect the same to happen if the PUB went down and access
> > > to the
> > > SUBs was still available.
> > >
> > > Can I assume that during this time, no logins or logouts would be
> > > 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: Ryan Ratliff
> > > To: Ted Nugent
> > > Cc: cisco-voip at puck-nether.net
> > > Sent: Wednesday, June 28, 2006 9:28 AM
> > > Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
> > > withExtension Mobility
> > >
> > > I just tested with a 7970 and when the phone drops into SRST mode it
> > > keeps the EM profile dn(s).  Since you don't configure ephone/dns in
> > > SRST the phone keeps whatever dns it had before it unregistered
> > > with CM.
> > >
> > > -Ryan
> > >
> > > On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
> > >
> > > The phone resets and goes to it's default profile. EM
> > > will not work in SRST since the EM service runs on CCM
> > > which is not longer in the picture
> > >
> > > --- Anthony Kouloglou <akoul at dataways.gr> wrote:
> > >
> > > > Hi Ryan,
> > > > i am sorry that i didn't clarify it but i was
> > > > refering to the situation
> > > > where there is an SRST involved:
> > > > there is a CCM cluster and an SRST configured at the
> > > > Device Pool to the
> > > > remote site.
> > > > At the remote site a user logs in with EM but at the
> > > > USER device profile
> > > > there is no reference about the device pool.
> > > > What happens to that phone while it is logged in
> > > > with EM and the remote
> > > > site loses connectivity with all the CCM cluster?
> > > >
> > > > Cheers
> > > > Anthony
> > > >
> > > > Ryan Ratliff wrote:
> > > >> Just to clarify when you login with EM it actually
> > > > changes the
> > > >> database so no matter which server the phone
> > > > registers with as long as
> > > >> that EM profile is logged in the phone will get
> > > > the EM lines.  The
> > > >> user won't get automatically logged out or
> > > > anything just because the
> > > >> phone fails over to another server.
> > > >>
> > > >> -Ryan
> > > >>
> > > >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
> > > >>
> > > >> The phone will act just like it would if you were
> > > > not logged in with
> > > >> EM, ie it will register with the backup
> > > > CallManager server.
> > > >>
> > > >> -Ryan
> > > >>
> > > >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
> > > > wrote:
> > > >>
> > > >> Hi all,
> > > >> if a user is logged in with extension mobility,
> > > > and the link to the
> > > >> primary ccm fails what happens with that phone?
> > > >>
> > > >>
> > > >> _______________________________________________
> > > >> 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
> > > >
> > >
> > >
> > > __________________________________________________
> > > Do You Yahoo!?
> > > Tired of spam?  Yahoo! Mail has the best spam protection around
> > > http://mail.yahoo.com
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >
> >
> >
>

From wsisk at cisco.com  Wed Jun 28 16:44:05 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 28 Jun 2006 16:44:05 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged
 in	withExtension Mobility
In-Reply-To: <f1b65f880606281338s6a3c0a22la398210346e300f2@mail.gmail.com>
References: <20060628121907.87466.qmail@web36109.mail.mud.yahoo.com>	<69B7B0A4-6C32-423B-BBA7-BC96A973B874@cisco.com>	<002901c69ab7$5e200fd0$47196883@cfs.uoguelph.ca>	<f1b65f880606280638o5e24f889x3fa9341364cb56d3@mail.gmail.com>	<92535326-43F1-445E-822F-509089040E09@cisco.com>	<f1b65f880606281313k7b45ce1ehced66b043a8252e9@mail.gmail.com>	<022401c69aef$ebab4850$47196883@cfs.uoguelph.ca>
	<f1b65f880606281338s6a3c0a22la398210346e300f2@mail.gmail.com>
Message-ID: <44A2EA15.6070401@cisco.com>

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From jason.aarons at us.didata.com  Wed Jun 28 17:47:29 2006
From: jason.aarons at us.didata.com (Jason Aarons (US))
Date: Wed, 28 Jun 2006 17:47:29 -0400
Subject: [cisco-voip] Link to primary CCM fails while logged
	inwithExtension Mobility
Message-ID: <C1FE15183DA37645BC0633BC604E44F00253EF30@USNAEXCH.na.didata.local>

SRND states the goal of SRST is WAN link failure.  

I've seen this debated about whether it works or not, phones in parts of
Africa lose the T1 and the power, and don't register under SRST when
power is restored.

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Justin
Steinberg
Sent: Wednesday, June 28, 2006 4:38 PM
To: Lelio Fulgenzi
Cc: cisco-voip at puck-nether.net; Ted Nugent
Subject: Re: [cisco-voip] Link to primary CCM fails while logged
inwithExtension Mobility

I'm going to retract my earlier statements.

After shutting down the CallManagers completely, the 7960 (FW 7.22)
phone does not register back to SRST router after power cycling the
phone.

Perhaps the phone needs to TFTP from the CCM before it can register to
the SRST router.

This seems like a limitation of SRST.  I can see a bad storm knocking
out a T1 line and commercial power.

Is there something I'm missing or is the answer to have a PoE switch
and a SRST router on a UPS?

Justin




On 6/28/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>
> I'm wondering if stopping the CCM service is enough of a test. Could
there
> perhaps be another service that the SRST router is connecting to?
>
> I don't see how the SRST router and phones can continue to operate
after a
> reboot if connectivity to the cluster is lost.
>
>
------------------------------------------------------------------------
--------
>
> 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: Justin Steinberg
> To: Ryan Ratliff
> Cc: Lelio Fulgenzi ; Ted Nugent ; cisco-voip at puck-nether.net
> Sent: Wednesday, June 28, 2006 4:13 PM
> Subject: Re: [cisco-voip] Link to primary CCM fails while logged in
> withExtension Mobility
>
> I did a little afternoon testing on this power cycle question I had.
>
> I registered a 7960 running FW 7.2(2) to CCM in my lab and then
> stopped the CCM service.  The phone successfully failed over to my
> SRST 3.3 router.
>
> Then I power cycled the SRST router and when the router came back
> online the phone registered back to SRST with the correct DN.
> Rebooted the phone and it still reregistered with the correct DN.
>
> It even registered back to SRST with the correct DN after power
> cycling the phone and SRST router at the same time.
>
> I then power cycled the router and did a factory reset procedure on
> the 7960, choosing to save the network config during the process.
> After the phone reset it registered back to the SRST router but
> without any DN.
>
> So the phone must save the most recent DN somewhere in flash and
> retains this even through power failure.
>
> I did not test this with EM.
>
> Justin
>
>
>
>
> On 6/28/06, Ryan Ratliff <rratliff at cisco.com> wrote:
> > My 7970 kept the EM dn through a power cycle while in SRST mode.
> >
> > -Ryan
> >
> > On Jun 28, 2006, at 9:38 AM, Justin Steinberg wrote:
> >
> > This isn't specific to EM.   But what happens if you are operating
in
> > SRST mode and you lose power to all the phones and the SRST router.
> > Do the phones or router keep record of their last DN in flash or
would
> > you be completely down after power came back with access to the CM
> > cluster still unavailable.
> >
> > Justin
> >
> > On 6/28/06, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> > >
> > > I would expect the same to happen if the PUB went down and access
> > > to the
> > > SUBs was still available.
> > >
> > > Can I assume that during this time, no logins or logouts would be
> > > 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: Ryan Ratliff
> > > To: Ted Nugent
> > > Cc: cisco-voip at puck-nether.net
> > > Sent: Wednesday, June 28, 2006 9:28 AM
> > > Subject: Re: [cisco-voip] Link to primary CCM fails while logged
in
> > > withExtension Mobility
> > >
> > > I just tested with a 7970 and when the phone drops into SRST mode
it
> > > keeps the EM profile dn(s).  Since you don't configure ephone/dns
in
> > > SRST the phone keeps whatever dns it had before it unregistered
> > > with CM.
> > >
> > > -Ryan
> > >
> > > On Jun 28, 2006, at 8:19 AM, Ted Nugent wrote:
> > >
> > > The phone resets and goes to it's default profile. EM
> > > will not work in SRST since the EM service runs on CCM
> > > which is not longer in the picture
> > >
> > > --- Anthony Kouloglou <akoul at dataways.gr> wrote:
> > >
> > > > Hi Ryan,
> > > > i am sorry that i didn't clarify it but i was
> > > > refering to the situation
> > > > where there is an SRST involved:
> > > > there is a CCM cluster and an SRST configured at the
> > > > Device Pool to the
> > > > remote site.
> > > > At the remote site a user logs in with EM but at the
> > > > USER device profile
> > > > there is no reference about the device pool.
> > > > What happens to that phone while it is logged in
> > > > with EM and the remote
> > > > site loses connectivity with all the CCM cluster?
> > > >
> > > > Cheers
> > > > Anthony
> > > >
> > > > Ryan Ratliff wrote:
> > > >> Just to clarify when you login with EM it actually
> > > > changes the
> > > >> database so no matter which server the phone
> > > > registers with as long as
> > > >> that EM profile is logged in the phone will get
> > > > the EM lines.  The
> > > >> user won't get automatically logged out or
> > > > anything just because the
> > > >> phone fails over to another server.
> > > >>
> > > >> -Ryan
> > > >>
> > > >> On Jun 27, 2006, at 11:04 AM, Ryan Ratliff wrote:
> > > >>
> > > >> The phone will act just like it would if you were
> > > > not logged in with
> > > >> EM, ie it will register with the backup
> > > > CallManager server.
> > > >>
> > > >> -Ryan
> > > >>
> > > >> On Jun 27, 2006, at 10:25 AM, Anthony Kouloglou
> > > > wrote:
> > > >>
> > > >> Hi all,
> > > >> if a user is logged in with extension mobility,
> > > > and the link to the
> > > >> primary ccm fails what happens with that phone?
> > > >>
> > > >>
> > > >> _______________________________________________
> > > >> 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
> > > >
> > >
> > >
> > > __________________________________________________
> > > Do You Yahoo!?
> > > Tired of spam?  Yahoo! Mail has the best spam protection around
> > > http://mail.yahoo.com
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > >
> > >
> >
> >
>
_______________________________________________
cisco-voip mailing list
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https://puck.nether.net/mailman/listinfo/cisco-voip

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From Jon_Pritchard at adp.com  Wed Jun 28 18:55:12 2006
From: Jon_Pritchard at adp.com (Pritchard, Jon)
Date: Wed, 28 Jun 2006 17:55:12 -0500
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BD07B17@EXCHANGEDS01.ds.ad.adp.com>

Does anyone know what parked calls and calls on hold being dropped at exactly 50 seconds is a telltale sign of?

In a cluster with 7 sites there is one site which is experience this problem.  All the other sites work fine.  No service parameters are even close to 50 seconds (or 5000 milliseconds).  Both internal and external calls are affected.  ?SQL replication appears to be fine.  Rebooting CallManagers does not change behavior.  Never seen anything like this in 5 years.  Any insight would be appreciated.

Jon Pritchard


This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.


From Jon_Pritchard at adp.com  Wed Jun 28 19:16:30 2006
From: Jon_Pritchard at adp.com (Pritchard, Jon)
Date: Wed, 28 Jun 2006 18:16:30 -0500
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds
References: <C1FE15183DA37645BC0633BC604E44F00253EF7F@USNAEXCH.na.didata.local>
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3B5568C4@EXCHANGEDS01.ds.ad.adp.com>

Yes...but if you don't pick up the call in 10 seconds (50 seconds total) it gets dropped. 
 
Jon

________________________________

From: Jason Aarons (US) [mailto:jason.aarons at us.didata.com]
Sent: Wed 6/28/2006 6:06 PM
To: Pritchard, Jon
Subject: RE: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds



If you lower your Park Reversion timeout to 40 seconds do you get calls back?

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Pritchard, Jon
Sent: Wednesday, June 28, 2006 6:55 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds

Does anyone know what parked calls and calls on hold being dropped at exactly 50 seconds is a telltale sign of?

In a cluster with 7 sites there is one site which is experience this problem.  All the other sites work fine.  No service parameters are even close to 50 seconds (or 5000 milliseconds).  Both internal and external calls are affected.   SQL replication appears to be fine.  Rebooting CallManagers does not change behavior.  Never seen anything like this in 5 years.  Any insight would be appreciated.

Jon Pritchard


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

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unlawful.  If you have received this communication in error, please
notify us immediately by replying to this message and deleting it
from your computer. Thank you.




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From kthorngr at cisco.com  Wed Jun 28 19:28:33 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Wed, 28 Jun 2006 19:28:33 -0400
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds
In-Reply-To: <472D0CBC7B31C1499B77C1DAA3308B3BD07B17@EXCHANGEDS01.ds.ad.adp.com>
References: <472D0CBC7B31C1499B77C1DAA3308B3BD07B17@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <50949191d8d776d156babe1bac631710@cisco.com>

Does this happen with IP Phone to IP Phone calls or with calls through 
a GW or both?

If you are experiencing the problem with an MGCP GW try the command "no 
mgcp timer receive-rtcp".  If this doesn't help please describe more 
about the call flow and the devices involved when the failures happen.

Kevin
On Jun 28, 2006, at 6:55 PM, Pritchard, Jon wrote:

> Does anyone know what parked calls and calls on hold being dropped at 
> exactly 50 seconds is a telltale sign of?
>
> In a cluster with 7 sites there is one site which is experience this 
> problem.  All the other sites work fine.  No service parameters are 
> even close to 50 seconds (or 5000 milliseconds).  Both internal and 
> external calls are affected.  ?SQL replication appears to be fine.  
> Rebooting CallManagers does not change behavior.  Never seen anything 
> like this in 5 years.  Any insight would be appreciated.
>
> Jon Pritchard
>
>
> This message and any attachments are intended only for the use of the 
> addressee and may contain information that is privileged and 
> confidential. If the reader of the message is not the intended 
> recipient or an authorized representative of the intended recipient, 
> you are hereby notified that any dissemination of this communication 
> is strictly prohibited. If you have received this communication in 
> error, please notify us immediately by e-mail and delete the message 
> and any attachments from your system.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



From Jon_Pritchard at adp.com  Wed Jun 28 20:11:36 2006
From: Jon_Pritchard at adp.com (Pritchard, Jon)
Date: Wed, 28 Jun 2006 19:11:36 -0500
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds
References: <472D0CBC7B31C1499B77C1DAA3308B3BD07B17@EXCHANGEDS01.ds.ad.adp.com>
	<50949191d8d776d156babe1bac631710@cisco.com>
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3B5568C6@EXCHANGEDS01.ds.ad.adp.com>

GW inbound and outbound.  We've now tried routing calls from another gateway over the WAN and those calls work fine.  We are looking for discrepencies in the mgcp config.
 
Jon

________________________________

From: Kevin Thorngren [mailto:kthorngr at cisco.com]
Sent: Wed 6/28/2006 6:28 PM
To: Pritchard, Jon
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds



Does this happen with IP Phone to IP Phone calls or with calls through
a GW or both?

If you are experiencing the problem with an MGCP GW try the command "no
mgcp timer receive-rtcp".  If this doesn't help please describe more
about the call flow and the devices involved when the failures happen.

Kevin
On Jun 28, 2006, at 6:55 PM, Pritchard, Jon wrote:

> Does anyone know what parked calls and calls on hold being dropped at
> exactly 50 seconds is a telltale sign of?
>
> In a cluster with 7 sites there is one site which is experience this
> problem.  All the other sites work fine.  No service parameters are
> even close to 50 seconds (or 5000 milliseconds).  Both internal and
> external calls are affected.   SQL replication appears to be fine. 
> Rebooting CallManagers does not change behavior.  Never seen anything
> like this in 5 years.  Any insight would be appreciated.
>
> Jon Pritchard
>
>
> This message and any attachments are intended only for the use of the
> addressee and may contain information that is privileged and
> confidential. If the reader of the message is not the intended
> recipient or an authorized representative of the intended recipient,
> you are hereby notified that any dissemination of this communication
> is strictly prohibited. If you have received this communication in
> error, please notify us immediately by e-mail and delete the message
> and any attachments from your system.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>




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From jrlyon at Trinsic.com  Wed Jun 28 20:56:54 2006
From: jrlyon at Trinsic.com (Lyon, John  R.)
Date: Wed, 28 Jun 2006 20:56:54 -0400
Subject: [cisco-voip] Paging on a 7936
Message-ID: <11C0A2D166B2D045A9429C4B23177A5E014CC052@tpa-pzb1-mbx-01.ztel.com>

I have a stubborn 7936 conference station that won't pick up the pages
from a dn that is sending the pages to a multicast address.  This is
using a CME setup.  The phones at the site and the conference station
are on the same switch and the phones pick up the pages, so it doesn't
appear to be a multicasting issue.  I've added the paging-dn line to the
7936's ephone config exactly as it is on all the other phones in the
office, yet they work and it doesn't.  Everything else on the 7936 seems
to be ok... inside and outside phone calls, registration with the CME,
etc.  Just to be sure it wasn't hardware, we've swapped out the 7936,
but still no joy in Mudville.  Any ideas as to what might be amiss would
be greatly appreciated.

 

Thanks,

John

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From Jon_Pritchard at adp.com  Wed Jun 28 20:58:48 2006
From: Jon_Pritchard at adp.com (Pritchard, Jon)
Date: Wed, 28 Jun 2006 19:58:48 -0500
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds
Message-ID: <472D0CBC7B31C1499B77C1DAA3308B3BD07B61@EXCHANGEDS01.ds.ad.adp.com>

Lack of 'no mgcp timer receive-rtcp' was the problem.  A combination of
the IP connectivity (i.e. call on hold) and, in this case, using
multicast moh_live locally on the gateway so there was no unicast stream
arriving from an external source for the specific B channel.

 

Jon

 

________________________________

From: Pritchard, Jon 
Sent: Wednesday, June 28, 2006 5:12 PM
To: Kevin Thorngren
Cc: cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds

 

GW inbound and outbound.  We've now tried routing calls from another
gateway over the WAN and those calls work fine.  We are looking for
discrepencies in the mgcp config.

 

Jon

 

________________________________

From: Kevin Thorngren [mailto:kthorngr at cisco.com]
Sent: Wed 6/28/2006 6:28 PM
To: Pritchard, Jon
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds

Does this happen with IP Phone to IP Phone calls or with calls through
a GW or both?

If you are experiencing the problem with an MGCP GW try the command "no
mgcp timer receive-rtcp".  If this doesn't help please describe more
about the call flow and the devices involved when the failures happen.

Kevin
On Jun 28, 2006, at 6:55 PM, Pritchard, Jon wrote:

> Does anyone know what parked calls and calls on hold being dropped at
> exactly 50 seconds is a telltale sign of?
>
> In a cluster with 7 sites there is one site which is experience this
> problem.  All the other sites work fine.  No service parameters are
> even close to 50 seconds (or 5000 milliseconds).  Both internal and
> external calls are affected.   SQL replication appears to be fine. 
> Rebooting CallManagers does not change behavior.  Never seen anything
> like this in 5 years.  Any insight would be appreciated.
>
> Jon Pritchard
>
>
> This message and any attachments are intended only for the use of the
> addressee and may contain information that is privileged and
> confidential. If the reader of the message is not the intended
> recipient or an authorized representative of the intended recipient,
> you are hereby notified that any dissemination of this communication
> is strictly prohibited. If you have received this communication in
> error, please notify us immediately by e-mail and delete the message
> and any attachments from your system.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>



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 wsisk at cisco.com  Wed Jun 28 21:07:25 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Wed, 28 Jun 2006 21:07:25 -0400
Subject: [cisco-voip] Paging on a 7936
In-Reply-To: <11C0A2D166B2D045A9429C4B23177A5E014CC052@tpa-pzb1-mbx-01.ztel.com>
References: <11C0A2D166B2D045A9429C4B23177A5E014CC052@tpa-pzb1-mbx-01.ztel.com>
Message-ID: <FFB8765E-92D9-4F61-BA36-D4FCCDD54630@cisco.com>

what phone load on the 7936?
does it show as being offhook and receiving mcast on the LCD?
are you trunking to the phone or using hardcoding access vlan = voice  
vlan?
in a packet capture do you see the 7936 send out an mcast join?

/Wes

On Jun 28, 2006, at 8:56 PM, Lyon, John R. wrote:

I have a stubborn 7936 conference station that won?t pick up the  
pages from a dn that is sending the pages to a multicast address.   
This is using a CME setup.  The phones at the site and the conference  
station are on the same switch and the phones pick up the pages, so  
it doesn?t appear to be a multicasting issue.  I?ve added the paging- 
dn line to the 7936?s ephone config exactly as it is on all the other  
phones in the office, yet they work and it doesn?t.  Everything else  
on the 7936 seems to be ok? inside and outside phone calls,  
registration with the CME, etc.  Just to be sure it wasn?t hardware,  
we?ve swapped out the 7936, but still no joy in Mudville.  Any ideas  
as to what might be amiss would be greatly appreciated.



Thanks,

John

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

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From jrlyon at Trinsic.com  Wed Jun 28 21:30:33 2006
From: jrlyon at Trinsic.com (Lyon, John  R.)
Date: Wed, 28 Jun 2006 21:30:33 -0400
Subject: [cisco-voip] Paging on a 7936
Message-ID: <11C0A2D166B2D045A9429C4B23177A5E014CC053@tpa-pzb1-mbx-01.ztel.com>

Phone load is cmterm_7936.3-3-5-0

LCD on the phone doesn't change when a page is sent from another phone.

Switchport config matches all the other phone ports:

 

interface FastEthernet0/19

 switchport trunk native vlan 12

 switchport mode trunk

 switchport voice vlan 13

 mls qos trust cos

 spanning-tree portfast

 

Don't really have the means to do a packet capture at the site.  Is
there a debug on the switch or CME that I can run to get the multicast
info?

 

--John

________________________________

From: Wes Sisk [mailto:wsisk at cisco.com] 
Sent: Wednesday, June 28, 2006 9:07 PM
To: Lyon, John R.
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Paging on a 7936

 

what phone load on the 7936?

does it show as being offhook and receiving mcast on the LCD?

are you trunking to the phone or using hardcoding access vlan = voice
vlan?

in a packet capture do you see the 7936 send out an mcast join?

 

/Wes

 

On Jun 28, 2006, at 8:56 PM, Lyon, John R. wrote:

 

I have a stubborn 7936 conference station that won't pick up the pages
from a dn that is sending the pages to a multicast address.  This is
using a CME setup.  The phones at the site and the conference station
are on the same switch and the phones pick up the pages, so it doesn't
appear to be a multicasting issue.  I've added the paging-dn line to the
7936's ephone config exactly as it is on all the other phones in the
office, yet they work and it doesn't.  Everything else on the 7936 seems
to be ok... inside and outside phone calls, registration with the CME,
etc.  Just to be sure it wasn't hardware, we've swapped out the 7936,
but still no joy in Mudville.  Any ideas as to what might be amiss would
be greatly appreciated.

 

Thanks,

John

_______________________________________________

cisco-voip mailing list

cisco-voip at puck.nether.net

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

 

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From erickbe at yahoo.com  Wed Jun 28 22:19:30 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Wed, 28 Jun 2006 19:19:30 -0700 (PDT)
Subject: [cisco-voip] IOS GW PRI Registration to separate CallManagers
In-Reply-To: <5CCE596C-7F79-40BC-906C-2AA54CC41E0F@cisco.com>
Message-ID: <20060629021930.27987.qmail@web37201.mail.mud.yahoo.com>

Maybe by using mgcp profiles on the router. I haven't used them yet so can't comment on how well they work or not, etc but looks like a neat feature. 

----- Original Message ----
From: Ryan Ratliff <rratliff at cisco.com>
To: Justin Church <jcc at unc.edu>
Cc: cisco-voip at puck.nether.net
Sent: Wednesday, June 28, 2006 9:25:13 AM
Subject: Re: [cisco-voip] IOS GW PRI Registration to separate CallManagers

I don't think you can run two mgcp processes on a router.

-Ryan

On Jun 28, 2006, at 10:20 AM, Justin Church wrote:

We're running 2821 IOS gateways using MGCP backhaul to CCM 4.1(3).  I
have several spare PRI interfaces and would like to register a different
PRI on the same GW with a different CM (specifically my CM 5.0(3)).  Is
that possible?

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

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From akoul at dataways.gr  Thu Jun 29 06:19:09 2006
From: akoul at dataways.gr (Anthony Kouloglou)
Date: Thu, 29 Jun 2006 13:19:09 +0300
Subject: [cisco-voip] clid on fxs ports
In-Reply-To: <7747FB40-03EA-472C-B947-95F8336708A9@cisco.com>
References: <44A2A86B.4030700@dataways.gr>
	<7747FB40-03EA-472C-B947-95F8336708A9@cisco.com>
Message-ID: <44A3A91D.5030803@dataways.gr>

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From Brian.Henry at apptis.com  Thu Jun 29 07:49:12 2006
From: Brian.Henry at apptis.com (Brian Henry)
Date: Thu, 29 Jun 2006 07:49:12 -0400
Subject: [cisco-voip] clid on fxs ports
Message-ID: <2F09B2162CCDEA4EA05C089B24FBC2BC02838242@CH-EXCH-02.intelltech.apptis.inc>

First talk to your provider and let them know that you wish to control the ANI and not them with the Billing Number that is associated with the PRI.
 
Brian

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Anthony Kouloglou
Sent: Thu 6/29/2006 6:19 AM
To: Ryan Ratliff
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] clid on fxs ports


Ryan
here is a csim start command from that router

Lon2851#csim start 90030xxxxxxxxxx
csim: called number = 90030xxxxxxxxxx, loop count = 1 ping count = 0

*Jun 29 11:11:08.959: ISDN Se0/2/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x0, Called num 0030xxxxxxxxxx
*Jun 29 11:11:08.959: ISDN Se0/2/0:15 Q931: TX -> SETUP pd = 8  callref = 0x0164 
        Bearer Capability i = 0x8090A3 
                Standard = CCITT 
                Transfer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA9839E 
                Exclusive, Channel 30 
        Called Party Number i = 0x80, '0030xxxxxxxxxx' 
                Plan:Unknown, Type:Unknown
*Jun 29 11:11:09.103: ISDN Se0/2/0:15 Q931: RX <- SETUP_ACK pd = 8  callref = 0x8164 
        Channel ID i = 0xA9839E 
                Exclusive, Channel 30 
        Progress Ind i = 0x8288 - In-band info or appropriate now available 
*Jun 29 11:11:15.555: ISDN Se0/2/0:15 Q931: RX <- ALERTING pd = 8  callref = 0x8164
*Jun 29 11:11:17.839: ISDN Se0/2/0:15 Q931: RX <- CONNECT pd = 8  callref = 0x8164
*Jun 29 11:11:17.839: %ISDN-6-CONNECT: Interface Serial0/2/0:29 is now connected to xxxxxxxxxx N/A
*Jun 29 11:11:17.839: ISDN Se0/2/0:15 Q931: TX -> CONNECT_ACK pd = 8  callref = 0x0164
*Jun 29 11:11:18.391: ISDN Se0/2/0:15 Q931: RX <- DISCONNECT pd = 8  callref = 0x8164 
        Cause i = 0x8090 - Normal call clearing 
        Progress Ind i = 0x8288 - In-band info or appropriate now available 
*Jun 29 11:11:18.395: ISDN Se0/2/0:15 Q931: call_disc: PI received in disconnect; Postpone sending RELEASE for callid 0x8164.
csim: loop = 1, failed = 0  
csim: call attempted = 1, setup failed = 0, tone failed = 1

Lon2851#

I guess there is no Calling party in the setup message.
When there is a call from an ip phone, there is calling party (the extension of the ccm). Of course the POTS provider just shows the number of the PRI to the called party and not the internal extension of the CCM.
But when i make calls from the router ( with csim start) or the fxs (a modem attached there), there is no calling party on the q931setup message and so the provider does not shows anything to the called.
I have tried configuring isdn calling-number etc on the serial 0/2/0:15 but nothing changes. Must i put something there or at the dial-peer pots?
The customer wants this feature desperately!!!

Cheers,
Anthony
Ryan Ratliff wrote: 

	Do you have a q931 debug showing the outbound call? 
	
	-Ryan 
	
	On Jun 28, 2006, at 12:03 PM, Anthony Kouloglou wrote: 
	
	Hi all, 
	on a 2851 with one E1 and 4 FXS ports i have configured the voice port 
	as follows: 
	
	voice-port 0/3/2 
	 cptone GB 
	 description xxx 
	 station-id name yyy 
	 station-id number 1234 
	 caller-id enable type 1 
	
	
	there is also these: 
	
	voice-port 0/2/0:15 
	 translation-profile incoming TP01 
	 bearer-cap Speech 
	interface Serial0/2/0:15 
	 no ip address 
	 isdn switch-type primary-net5 
	 isdn incoming-voice voice 
	 no cdp enable 
	
	
	dial-peer voice 10 pots 
	 destination-pattern 9T 
	 port 0/2/0:15 
	
	
	when i call from the analog phone to an ipphone extension internally i 
	have clid (both name and number) to the ipphone 
	
	but when i make calls through the pri, i get "uknown number" to the 
	external phone that i have called. 
	Is this ficed somehow? 
	
	Cheers 
	Anthony 
	_______________________________________________ 
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	cisco-voip at puck.nether.net 
	https://puck.nether.net/mailman/listinfo/cisco-voip 
	
	

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From Nick.Kassel at Charles-Stanley.co.uk  Thu Jun 29 08:06:37 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 29 Jun 2006 13:06:37 +0100
Subject: [cisco-voip] Dialpeer mismatch
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648457@CSEXCH01.charles-stanley.co.uk>

We had an issue with a new installation of a remote site today, they
have a PRI E1 on a 1760 router with a range of DDI numbers.

 

When I added the following dial peers to allow 4 digit dialling when the
WAN is down across SRST we came across a problem. When in SRST the 4
digit dialling worked fine.

 

dial-peer voice 2000 pots

destination-pattern 2...

prefix 02079532

port 0/0:15

 

dial-peer voice 6000 pots

destination-pattern 6...

prefix 02071496

port 0/0:15

 

The issue we had was that now any inbound calls originating from 207 XXX
XXXX fail, the calls came in to the router and there were no errors on
the debug then you received a dialtone after hanging up the debug would
show the call clearing down as you would expect. Any other calls from
other area codes were fine.

 

Obviously the 2 prefix being presented was mismatching with the
destination-pattern 2... in the dial-peer voice 2000 pots dial peer. 

 

Is there anyway I can get round as we would very much like to have these
dial-peers in.

 

 

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You acknowledge that e-mails are not secure and you accept the risk of
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Registered in England No. 1903304

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

Authorised and Regulated by the Financial Services Authority, Member of the
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From JOlson at mlnusa.com  Thu Jun 29 08:13:23 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Thu, 29 Jun 2006 08:13:23 -0400
Subject: [cisco-voip] Request to change agent state failed. Unable to change
	agent state.
Message-ID: <FE2D092F8B1516488B99342551AACFD41D7BE891@mlnusa-exch1.mlnusa.com>

I have 1 agent who has not been able to put herself in the ready state.
She is able to login but when she tried to put herself in the ready
state she receives an error saying  

"Request to change agent state failed.  Unable to change agent state."

I have removed her from the IPCC deleting her user account and recreated
it.  I have stopped and started the engine.  Still nothing.  

 

Any ideas?

 

 

 


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CONFIDENTIALITY NOTICE:
This E-mail transmission contains information from Mortgage Lenders Network USA, Inc. which is confidential and legally privileged. The information is intended only for the use of the individual or entity named on this E-mail. If you are not the intended recipient, you are hereby notified that disclosing, copying, distributing, or taking any action in reliance on the contents of this information is strictly prohibited, and that the E-mail should be returned to this company immediately. If you have received this E-mail in error, please contact the sender immediately.

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From pwalenta at wi.rr.com  Thu Jun 29 08:22:13 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu, 29 Jun 2006 07:22:13 -0500
Subject: [cisco-voip] Dialpeer mismatch
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC648457@CSEXCH01.charles-stanley.co.uk>
Message-ID: <007b01c69b76$a983a120$0232a8c0@walentawk1>

You mentioned that these are DDI numbers for you?  I'd recommend putting
"direct-inward-dial" on those peers.
 
I've seen this behavior before, where some peers have DID on them, and some
don't, and incoming calls seem to prefer non-did enabled dial-peers.  My
general rule of thumb is that when I have DID on a line, all POTS peers get
direct-inward-dial on them just to prevent problems (or to prevent someone
from getting dial-tone accidentally).


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
Sent: Thursday, June 29, 2006 7:07 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Dialpeer mismatch



We had an issue with a new installation of a remote site today, they have a
PRI E1 on a 1760 router with a range of DDI numbers.

 

When I added the following dial peers to allow 4 digit dialling when the WAN
is down across SRST we came across a problem. When in SRST the 4 digit
dialling worked fine.

 

dial-peer voice 2000 pots

destination-pattern 2.

prefix 02079532

port 0/0:15

 

dial-peer voice 6000 pots

destination-pattern 6.

prefix 02071496

port 0/0:15

 

The issue we had was that now any inbound calls originating from 207 XXX
XXXX fail, the calls came in to the router and there were no errors on the
debug then you received a dialtone after hanging up the debug would show the
call clearing down as you would expect. Any other calls from other area
codes were fine.

 

Obviously the 2 prefix being presented was mismatching with the
destination-pattern 2. in the dial-peer voice 2000 pots dial peer. 

 

Is there anyway I can get round as we would very much like to have these
dial-peers in.

 

 




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

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
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  Thu Jun 29 08:37:59 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 29 Jun 2006 13:37:59 +0100
Subject: [cisco-voip] Dialpeer mismatch
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC648459@CSEXCH01.charles-stanley.co.uk>

Thanks Phil, sorry the two dial-peers I listed are outgoing dial-peers
only, they are so the remote site can call HQ using 4 digits when the
WAN is down, these are the remaining dial-peers in the config, I already
have the direct-inward-dial on the dial-peer voice 9 pots the only
inbound pots dial-peer I have and the DDI numbers I was referring to
should use dial-peer voice 1 voip for inbound calls. 

 

dial-peer voice 9 pots

 destination-pattern 9T

 progress_ind alert enable 8

 direct-inward-dial

 port 0/0:15

!

dial-peer voice 100 voip

 incoming called-number .

 codec g711ulaw

 no vad

!

dial-peer voice 1 voip

 destination-pattern 5537..

 progress_ind setup enable 3

 translate-outgoing calling 1

 voice-class h323 1

 session target ipv4:............

 dtmf-relay h245-alphanumeric

 codec g711ulaw

 no vad

!

dial-peer voice 2 voip

 preference 1

 destination-pattern 5537..

 progress_ind setup enable 3

 translate-outgoing calling 1

 voice-class h323 1

 session target ipv4:............

 dtmf-relay h245-alphanumeric

 codec g711ulaw

 no vad

!

 

________________________________

From: Philip Walenta [mailto:pwalenta at wi.rr.com] 
Sent: Thursday, June 29, 2006 1:22 PM
To: Nick Kassel; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Dialpeer mismatch

 

You mentioned that these are DDI numbers for you?  I'd recommend putting
"direct-inward-dial" on those peers.

 

I've seen this behavior before, where some peers have DID on them, and
some don't, and incoming calls seem to prefer non-did enabled
dial-peers.  My general rule of thumb is that when I have DID on a line,
all POTS peers get direct-inward-dial on them just to prevent problems
(or to prevent someone from getting dial-tone accidentally).

	 

	
________________________________


	From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
	Sent: Thursday, June 29, 2006 7:07 AM
	To: cisco-voip at puck.nether.net
	Subject: [cisco-voip] Dialpeer mismatch

	We had an issue with a new installation of a remote site today,
they have a PRI E1 on a 1760 router with a range of DDI numbers.

	 

	When I added the following dial peers to allow 4 digit dialling
when the WAN is down across SRST we came across a problem. When in SRST
the 4 digit dialling worked fine.

	 

	dial-peer voice 2000 pots

	destination-pattern 2...

	prefix 02079532

	port 0/0:15

	 

	dial-peer voice 6000 pots

	destination-pattern 6...

	prefix 02071496

	port 0/0:15

	 

	The issue we had was that now any inbound calls originating from
207 XXX XXXX fail, the calls came in to the router and there were no
errors on the debug then you received a dialtone after hanging up the
debug would show the call clearing down as you would expect. Any other
calls from other area codes were fine.

	 

	Obviously the 2 prefix being presented was mismatching with the
destination-pattern 2... in the dial-peer voice 2000 pots dial peer. 

	 

	Is there anyway I can get round as we would very much like to
have these dial-peers in.

	 

	 

	 


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

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Access to this e-mail by any other person is prohibited. If you are not
the
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of,
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You accept that any instructions are deemed to have been given at the
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Charles Stanley has received them. You are advised that urgent, time
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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
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
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From jgrace at digitelusa.net  Thu Jun 29 09:14:39 2006
From: jgrace at digitelusa.net (James Grace)
Date: Thu, 29 Jun 2006 09:14:39 -0400
Subject: [cisco-voip] realtime reports do not reset itself after 24 hrs
Message-ID: <200606290927312.SM01452@3729434770>

We are running crs and cm co resident.

Cm 4.2  

Crs 4.0.4

 For some reason when we look at the real time reports, they don't reset
itself.  We have to manually do this every night.  This was like from the
beginning.   Any body seen this 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From wsisk at cisco.com  Thu Jun 29 09:36:28 2006
From: wsisk at cisco.com (Wes Sisk)
Date: Thu, 29 Jun 2006 09:36:28 -0400
Subject: [cisco-voip] Paging on a 7936
In-Reply-To: <11C0A2D166B2D045A9429C4B23177A5E014CC053@tpa-pzb1-mbx-01.ztel.com>
References: <11C0A2D166B2D045A9429C4B23177A5E014CC053@tpa-pzb1-mbx-01.ztel.com>
Message-ID: <4BE28DA4-179C-45C7-9F9F-1AB495BBFD44@cisco.com>

and these are the same vlans other phones are using?
can you force it to access only port and set the access vlan to be  
the voice vlan just to confirm?

does this phone have a shared line to the line that is initiating the  
page?

/Wes
On Jun 28, 2006, at 9:30 PM, Lyon, John R. wrote:

Phone load is cmterm_7936.3-3-5-0

LCD on the phone doesn?t change when a page is sent from another phone.

Switchport config matches all the other phone ports:



interface FastEthernet0/19

  switchport trunk native vlan 12

  switchport mode trunk

  switchport voice vlan 13

  mls qos trust cos

  spanning-tree portfast



Don?t really have the means to do a packet capture at the site.  Is  
there a debug on the switch or CME that I can run to get the  
multicast info?



--John

From: Wes Sisk [mailto:wsisk at cisco.com]
Sent: Wednesday, June 28, 2006 9:07 PM
To: Lyon, John R.
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Paging on a 7936



what phone load on the 7936?

does it show as being offhook and receiving mcast on the LCD?

are you trunking to the phone or using hardcoding access vlan = voice  
vlan?

in a packet capture do you see the 7936 send out an mcast join?



/Wes



On Jun 28, 2006, at 8:56 PM, Lyon, John R. wrote:



I have a stubborn 7936 conference station that won?t pick up the  
pages from a dn that is sending the pages to a multicast address.   
This is using a CME setup.  The phones at the site and the conference  
station are on the same switch and the phones pick up the pages, so  
it doesn?t appear to be a multicasting issue.  I?ve added the paging- 
dn line to the 7936?s ephone config exactly as it is on all the other  
phones in the office, yet they work and it doesn?t.  Everything else  
on the 7936 seems to be ok? inside and outside phone calls,  
registration with the CME, etc.  Just to be sure it wasn?t hardware,  
we?ve swapped out the 7936, but still no joy in Mudville.  Any ideas  
as to what might be amiss would be greatly appreciated.



Thanks,

John

_______________________________________________

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From rsingleton at novateck.com  Thu Jun 29 09:48:36 2006
From: rsingleton at novateck.com (Robert Singleton)
Date: Thu, 29 Jun 2006 08:48:36 -0500
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds
In-Reply-To: <472D0CBC7B31C1499B77C1DAA3308B3BD07B17@EXCHANGEDS01.ds.ad.adp.com>
References: <472D0CBC7B31C1499B77C1DAA3308B3BD07B17@EXCHANGEDS01.ds.ad.adp.com>
Message-ID: <200606290848593.SM01876@mail.leasenet.net>

On Wed, 2006-06-28 at 17:55 -0500, Pritchard, Jon wrote:
> Does anyone know what parked calls and calls on hold being dropped at exactly 50 seconds is a telltale sign of?

<snip>

> No service parameters are even close to 50 seconds (or 5000 milliseconds).

The timer parameter clue may invalidate what I'm about to say, but....

In our setup, we had a problem with seemingly random calls being dropped
from Call Park, but only at one location. After working for a long time
to duplicate the issue, it was discovered that the caller was dropped at
the expiration of the Call Park Reversion Timer. In the affected branch,
incoming calls were directed to an attendant console hunt group. The
lines that were members of this hunt group were on a 7960. The mode of
failure was tricky to catch, but here's what was happening: Let's say a
call came in and the huntgroup directed it to the first member of the
huntgroup, DN 2007001. The call was placed on Park. Before the Call Park
Reversion Timer expires, another call has come in and is on 2007001.
When the Call Park Reversion Timer expires, the parked call drops rather
than ringing back. The same thing happens with any line; if the specific
line the call was parked from is busy when the timer expires, the call
is dropped. Note that this only happened to calls parked by the
Attendant Console at this location. Calls parked by individuals
functioned correctly.

After troubleshooting with TAC, it was revealed as a known issue. The
workaround is simple enough; simulate a circular huntgroup with
forwarding on each of the affected lines; 2007001 fwd bu/na to 2007002,
2007002 fwd bu/na to 2007003, et al and finally 2007006 fwd bu/na
2007001. Apparently, Call Park Reversion subsumes hunt groups, but not
the forwarding, so when the line that parked the call is not available,
forwarding will send it to the next line, and so on until it finds a
free line or comes full circle. Even when all the lines are busy, at
least the caller gets reorder and drops the call themselves rather than
being unceremoniously disconnected.

I mentioned that this was only happening in one location. That location
happened to be rolled out with Attendant Console in place. In that
branch, the 7960 was programmed with the Attendant Console Hunt Group
members from the start. In the other locations with Attendant Consoles,
Attendant Console was added after calls were already being handled with
a pseudo-huntgroup done by forwarding each line to the next. The
forwarding was not implicitly removed from teh lines when the Attendant
Console was installed, so the workaround was already in place,
preventing us from ever having the problem in the other locations.

Interestingly enough, it was also discovered that at the problem
location, if the attendant pressed the Park button on the phone itself,
the call would not drop. It would only drop if parked from the Attendant
Console application and only when the other conditions were met.

Robert



From kthorngr at cisco.com  Thu Jun 29 10:10:25 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu, 29 Jun 2006 10:10:25 -0400
Subject: [cisco-voip] Dialpeer mismatch
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC648459@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC648459@CSEXCH01.charles-stanley.co.uk>
Message-ID: <cf0b0bc925e5880927cf91eea3937507@cisco.com>

For inbound dial peers "incoming called-number" has priority over  
"destination-pattern".  Try adding "incoming called-number ." to dial  
peer 9 to force all incoming calls to use this dial peer.

Kevin
On Jun 29, 2006, at 8:37 AM, Nick Kassel wrote:

> Thanks Phil, sorry the two dial-peers I listed are outgoing dial-peers  
> only, they are so the remote site can call HQ using 4 digits when the  
> WAN is down, these are the remaining dial-peers in the config, I  
> already have the direct-inward-dial on the dial-peer voice 9 pots the  
> only inbound pots dial-peer I have and the DDI numbers I was referring  
> to should use dial-peer voice 1 voip for inbound calls.
> ?
> dial-peer voice 9 pots
> ?destination-pattern 9T
> ?progress_ind alert enable 8
> ?direct-inward-dial
> ?port 0/0:15
> !
> dial-peer voice 100 voip
> ?incoming called-number .
> ?codec g711ulaw
> ?no vad
> !
> dial-peer voice 1 voip
> ?destination-pattern 5537..
> ?progress_ind setup enable 3
> ?translate-outgoing calling 1
> ?voice-class h323 1
> ?session target ipv4:????
> ?dtmf-relay h245-alphanumeric
> ?codec g711ulaw
> ?no vad
> !
> dial-peer voice 2 voip
> ?preference 1
> ?destination-pattern 5537..
> ?progress_ind setup enable 3
> ?translate-outgoing calling 1
> ?voice-class h323 1
> ?session target ipv4:????
> ?dtmf-relay h245-alphanumeric
> ?codec g711ulaw
> ?no vad
> !
> ?
>
> From: Philip Walenta [mailto:pwalenta at wi.rr.com]
> Sent: Thursday, June 29, 2006 1:22 PM
> To: Nick Kassel; cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] Dialpeer mismatch
> ?
> You mentioned that these are DDI numbers for you?? I'd recommend  
> putting "direct-inward-dial" on those peers.
> ?
> I've seen this behavior before, where some peers have DID on them, and  
> some don't, and incoming calls seem to prefer non-did enabled  
> dial-peers.? My general rule of thumb is that when I have DID on a  
> line, all POTS peers get direct-inward-dial on them just to prevent  
> problems (or?to prevent?someone from getting dial-tone accidentally).
>> ?
>>
>>
>> From: cisco-voip-bounces at puck.nether.net  
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
>> Sent: Thursday, June 29, 2006 7:07 AM
>> To: cisco-voip at puck.nether.net
>> Subject: [cisco-voip] Dialpeer mismatch
>> We had an issue with a new installation of a remote site today, they  
>> have a PRI E1 on a 1760 router with a range of DDI numbers.
>> ?
>> When I added the following dial peers to allow 4 digit dialling when  
>> the WAN is down across SRST we came across a problem. When in SRST  
>> the 4 digit dialling worked fine.
>> ?
>> dial-peer voice 2000 pots
>> destination-pattern 2?
>> prefix 02079532
>> port 0/0:15
>> ?
>> dial-peer voice 6000 pots
>> destination-pattern 6?
>> prefix 02071496
>> port 0/0:15
>> ?
>> The issue we had was that now any inbound calls originating from 207  
>> XXX XXXX fail, the calls came in to the router and there were no  
>> errors on the debug then you received a dialtone after hanging up the  
>> debug would show the call clearing down as you would expect. Any  
>> other calls from other area codes were fine.
>> ?
>> Obviously the 2 prefix being presented was mismatching with the  
>> destination-pattern 2? in the dial-peer voice 2000 pots dial peer.
>> ?
>> Is there anyway I can get round as we would very much like to have  
>> these dial-peers in.
>> ?
>> ?
>>
>> ?
>>
>>   
>> ********************************************************************** 
>> *************************
>>
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>> The?information?contained?in?this?e- 
>> mail?is?strictly?confidential,?some?or?all
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>> mail?by?any?other?person?is?prohibited.?If?you?are?not?the
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>> intended?recipient,?any?use,?disclosure,?copying,?printing,?distributi 
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>>   
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>> recipient(s)?accesses?them?and?that?delivery?receipt?does?not?constitu 
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>> acknowledgement?or?receipt?by?the?intended?recipient(s).?You?accept?th 
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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?divisio 
>> ns?of?Charles?Stanley?&?Co.?Ltd
>>
>>   
>> Authorised?and?Regulated?by?the?Financial?Services?Authority,?Member?o 
>> f?the
>>   
>> London?Stock?Exchange,?International?Securities?Markets?Association,?a 
>> nd?The?London?International?Financial?Futures?&
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>>
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>> 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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>
>
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> *********************************************************************** 
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>
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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?on 
> ly.
>   
> Access?to?this?e- 
> mail?by?any?other?person?is?prohibited.?If?you?are?not?the
>   
> intended?recipient,?any?use,?disclosure,?copying,?printing,?distributio 
> n?of,
>   
> replying?to?or?any?action?taken?or?omitted?to?be?taken?in?reliance?on?t 
> his
>   
> e- 
> mail,?is?prohibited?and?may?be?unlawful.?Please?contact?the?sender?imme 
> diately
>  should?this?e-mail?have?been?incorrectly?addressed?or?transmitted.
>
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> You?accept?that?any?instructions?are?deemed?to?have?been?given?at?the?t 
> ime?the
>   
> recipient(s)?accesses?them?and?that?delivery?receipt?does?not?constitut 
> e
>   
> acknowledgement?or?receipt?by?the?intended?recipient(s).?You?accept?tha 
> t?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?se 
> nsitive
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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?division 
> s?of?Charles?Stanley?&?Co.?Ltd
>
>   
> Authorised?and?Regulated?by?the?Financial?Services?Authority,?Member?of 
> ?the
>   
> London?Stock?Exchange,?International?Securities?Markets?Association,?an 
> d?The?London?International?Financial?Futures?&
>  Options?Exchange.
>
>   
> This?footnote?also?confirms?that?this?email?message?has?been?swept?by?M 
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>  VirusScan?and?SurfControl?Email?Filter?software.
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From Nick.Kassel at Charles-Stanley.co.uk  Thu Jun 29 10:13:01 2006
From: Nick.Kassel at Charles-Stanley.co.uk (Nick Kassel)
Date: Thu, 29 Jun 2006 15:13:01 +0100
Subject: [cisco-voip] Dialpeer mismatch
Message-ID: <253FAE7A06B11B49A3A29770B2B8BAFC64845A@CSEXCH01.charles-stanley.co.uk>

 Ok, so If I add the incoming called-number. to dial-peer 9, do I still
need to keep dial-peer 100

 

________________________________

From: Kevin Thorngren [mailto:kthorngr at cisco.com] 
Sent: Thursday, June 29, 2006 3:10 PM
To: Nick Kassel
Cc: ciscovoip Voip
Subject: Re: [cisco-voip] Dialpeer mismatch

 

For inbound dial peers "incoming called-number" has priority over
"destination-pattern". Try adding "incoming called-number ." to dial
peer 9 to force all incoming calls to use this dial peer. 

 

Kevin 

On Jun 29, 2006, at 8:37 AM, Nick Kassel wrote: 

	 

	Thanks Phil, sorry the two dial-peers I listed are outgoing
dial-peers only, they are so the remote site can call HQ using 4 digits
when the WAN is down, these are the remaining dial-peers in the config,
I already have the direct-inward-dial on the dial-peer voice 9 pots the
only inbound pots dial-peer I have and the DDI numbers I was referring
to should use dial-peer voice 1 voip for inbound calls. 

	  

	dial-peer voice 9 pots 

	 destination-pattern 9T 

	 progress_ind alert enable 8 

	 direct-inward-dial 

	 port 0/0:15 

	! 

	dial-peer voice 100 voip 

	 incoming called-number . 

	 codec g711ulaw 

	 no vad 

	! 

	dial-peer voice 1 voip 

	 destination-pattern 5537.. 

	 progress_ind setup enable 3 

	 translate-outgoing calling 1 

	 voice-class h323 1 

	 session target ipv4:............ 

	 dtmf-relay h245-alphanumeric 

	 codec g711ulaw 

	 no vad 

	! 

	dial-peer voice 2 voip 

	 preference 1 

	 destination-pattern 5537.. 

	 progress_ind setup enable 3 

	 translate-outgoing calling 1 

	 voice-class h323 1 

	 session target ipv4:............ 

	 dtmf-relay h245-alphanumeric 

	 codec g711ulaw 

	 no vad 

	! 

	  

	 

	From: Philip Walenta [mailto:pwalenta at wi.rr.com] 

	Sent: Thursday, June 29, 2006 1:22 PM 

	To: Nick Kassel; cisco-voip at puck.nether.net 

	Subject: RE: [cisco-voip] Dialpeer mismatch 

	  

	You mentioned that these are DDI numbers for you?  I'd recommend
putting "direct-inward-dial" on those peers. 

	  

	I've seen this behavior before, where some peers have DID on
them, and some don't, and incoming calls seem to prefer non-did enabled
dial-peers.  My general rule of thumb is that when I have DID on a line,
all POTS peers get direct-inward-dial on them just to prevent problems
(or to prevent someone from getting dial-tone accidentally). 

		  

		 

		From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel 

		Sent: Thursday, June 29, 2006 7:07 AM 

		To: cisco-voip at puck.nether.net 

		Subject: [cisco-voip] Dialpeer mismatch 

		We had an issue with a new installation of a remote site
today, they have a PRI E1 on a 1760 router with a range of DDI numbers. 

		  

		When I added the following dial peers to allow 4 digit
dialling when the WAN is down across SRST we came across a problem. When
in SRST the 4 digit dialling worked fine. 

		  

		dial-peer voice 2000 pots 

		destination-pattern 2... 

		prefix 02079532 

		port 0/0:15 

		  

		dial-peer voice 6000 pots 

		destination-pattern 6... 

		prefix 02071496 

		port 0/0:15 

		  

		The issue we had was that now any inbound calls
originating from 207 XXX XXXX fail, the calls came in to the router and
there were no errors on the debug then you received a dialtone after
hanging up the debug would show the call clearing down as you would
expect. Any other calls from other area codes were fine. 

		  

		Obviously the 2 prefix being presented was mismatching
with the destination-pattern 2... in the dial-peer voice 2000 pots dial
peer. 

		  

		Is there anyway I can get round as we would very much
like to have these dial-peers in. 

		  

		  

		 

		  

		 

	
************************************************************************
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		The information contained in this e-mail is strictly
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		Access to this e-mail by any other person is prohibited.
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		You accept that any instructions are deemed to have been
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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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Association, and The London International Financial Futures & 

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		This footnote also confirms that this email message has
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	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
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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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Association, and The London International Financial Futures & 

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

	_______________________________________________ 

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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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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
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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
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From jsteinberg at gmail.com  Thu Jun 29 10:14:32 2006
From: jsteinberg at gmail.com (Justin Steinberg)
Date: Thu, 29 Jun 2006 10:14:32 -0400
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds
In-Reply-To: <200606290848593.SM01876@mail.leasenet.net>
References: <472D0CBC7B31C1499B77C1DAA3308B3BD07B17@EXCHANGEDS01.ds.ad.adp.com>
	<200606290848593.SM01876@mail.leasenet.net>
Message-ID: <f1b65f880606290714v2bec3002jd395b383cac8a4db@mail.gmail.com>

What version of CallManager?
Do you have a bug id?
Is it fixed in newer versions?

AC admin guide specifically states not to enable any forwarding on member DN's.

Justin

On 6/29/06, Robert Singleton <rsingleton at novateck.com> wrote:
> On Wed, 2006-06-28 at 17:55 -0500, Pritchard, Jon wrote:
> > Does anyone know what parked calls and calls on hold being dropped at exactly 50 seconds is a telltale sign of?
>
> <snip>
>
> > No service parameters are even close to 50 seconds (or 5000 milliseconds).
>
> The timer parameter clue may invalidate what I'm about to say, but....
>
> In our setup, we had a problem with seemingly random calls being dropped
> from Call Park, but only at one location. After working for a long time
> to duplicate the issue, it was discovered that the caller was dropped at
> the expiration of the Call Park Reversion Timer. In the affected branch,
> incoming calls were directed to an attendant console hunt group. The
> lines that were members of this hunt group were on a 7960. The mode of
> failure was tricky to catch, but here's what was happening: Let's say a
> call came in and the huntgroup directed it to the first member of the
> huntgroup, DN 2007001. The call was placed on Park. Before the Call Park
> Reversion Timer expires, another call has come in and is on 2007001.
> When the Call Park Reversion Timer expires, the parked call drops rather
> than ringing back. The same thing happens with any line; if the specific
> line the call was parked from is busy when the timer expires, the call
> is dropped. Note that this only happened to calls parked by the
> Attendant Console at this location. Calls parked by individuals
> functioned correctly.
>
> After troubleshooting with TAC, it was revealed as a known issue. The
> workaround is simple enough; simulate a circular huntgroup with
> forwarding on each of the affected lines; 2007001 fwd bu/na to 2007002,
> 2007002 fwd bu/na to 2007003, et al and finally 2007006 fwd bu/na
> 2007001. Apparently, Call Park Reversion subsumes hunt groups, but not
> the forwarding, so when the line that parked the call is not available,
> forwarding will send it to the next line, and so on until it finds a
> free line or comes full circle. Even when all the lines are busy, at
> least the caller gets reorder and drops the call themselves rather than
> being unceremoniously disconnected.
>
> I mentioned that this was only happening in one location. That location
> happened to be rolled out with Attendant Console in place. In that
> branch, the 7960 was programmed with the Attendant Console Hunt Group
> members from the start. In the other locations with Attendant Consoles,
> Attendant Console was added after calls were already being handled with
> a pseudo-huntgroup done by forwarding each line to the next. The
> forwarding was not implicitly removed from teh lines when the Attendant
> Console was installed, so the workaround was already in place,
> preventing us from ever having the problem in the other locations.
>
> Interestingly enough, it was also discovered that at the problem
> location, if the attendant pressed the Park button on the phone itself,
> the call would not drop. It would only drop if parked from the Attendant
> Console application and only when the other conditions were met.
>
> Robert
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

From kthorngr at cisco.com  Thu Jun 29 10:32:14 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu, 29 Jun 2006 10:32:14 -0400
Subject: [cisco-voip] Dialpeer mismatch
In-Reply-To: <253FAE7A06B11B49A3A29770B2B8BAFC64845A@CSEXCH01.charles-stanley.co.uk>
References: <253FAE7A06B11B49A3A29770B2B8BAFC64845A@CSEXCH01.charles-stanley.co.uk>
Message-ID: <55394112b86bfc743ab44ada695ba10f@cisco.com>

That depends :-)  Since DP 100 is a voip peer it does not have any  
affect on the pots peer.  You may want to keep this to force a known  
call configuration for inbound voip calls.  This URL should help:

http://www.cisco.com/en/US/tech/tk652/tk90/ 
technologies_tech_note09186a008010fed1.shtml

Kevin
On Jun 29, 2006, at 10:13 AM, Nick Kassel wrote:

> ?Ok, so If I add the incoming called-number. to dial-peer 9, do I  
> still need to keep dial-peer 100
> ?
>
> From: Kevin Thorngren [mailto:kthorngr at cisco.com]
> Sent: Thursday, June 29, 2006 3:10 PM
> To: Nick Kassel
> Cc: ciscovoip Voip
> Subject: Re: [cisco-voip] Dialpeer mismatch
> ?
> For inbound dial peers "incoming called-number" has priority over  
> "destination-pattern". Try adding "incoming called-number ." to dial  
> peer 9 to force all incoming calls to use this dial peer.
> ?
> Kevin
> On Jun 29, 2006, at 8:37 AM, Nick Kassel wrote:
>> ?
>> Thanks Phil, sorry the two dial-peers I listed are outgoing  
>> dial-peers only, they are so the remote site can call HQ using 4  
>> digits when the WAN is down, these are the remaining dial-peers in  
>> the config, I already have the direct-inward-dial on the dial-peer  
>> voice 9 pots the only inbound pots dial-peer I have and the DDI  
>> numbers I was referring to should use dial-peer voice 1 voip for  
>> inbound calls.
>> ?
>> dial-peer voice 9 pots
>> ?destination-pattern 9T
>> ?progress_ind alert enable 8
>> ?direct-inward-dial
>> ?port 0/0:15
>> !
>> dial-peer voice 100 voip
>> ?incoming called-number .
>> ?codec g711ulaw
>> ?no vad
>> !
>> dial-peer voice 1 voip
>> ?destination-pattern 5537..
>> ?progress_ind setup enable 3
>> ?translate-outgoing calling 1
>> ?voice-class h323 1
>> ?session target ipv4:????
>> ?dtmf-relay h245-alphanumeric
>> ?codec g711ulaw
>> ?no vad
>> !
>> dial-peer voice 2 voip
>> ?preference 1
>> ?destination-pattern 5537..
>> ?progress_ind setup enable 3
>> ?translate-outgoing calling 1
>> ?voice-class h323 1
>> ?session target ipv4:????
>> ?dtmf-relay h245-alphanumeric
>> ?codec g711ulaw
>> ?no vad
>> !
>> ?
>> ?
>> From: Philip Walenta [mailto:pwalenta at wi.rr.com]
>> Sent: Thursday, June 29, 2006 1:22 PM
>> To: Nick Kassel; cisco-voip at puck.nether.net
>> Subject: RE: [cisco-voip] Dialpeer mismatch
>> ?
>> You mentioned that these are DDI numbers for you?? I'd recommend  
>> putting "direct-inward-dial" on those peers.
>> ?
>> I've seen this behavior before, where some peers have DID on them,  
>> and some don't, and incoming calls seem to prefer non-did enabled  
>> dial-peers.? My general rule of thumb is that when I have DID on a  
>> line, all POTS peers get direct-inward-dial on them just to prevent  
>> problems (or?to prevent?someone from getting dial-tone accidentally).
>>> ?
>>>
>>> ?
>>> From: cisco-voip-bounces at puck.nether.net  
>>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Kassel
>>> Sent: Thursday, June 29, 2006 7:07 AM
>>> To: cisco-voip at puck.nether.net
>>> Subject: [cisco-voip] Dialpeer mismatch
>>> We had an issue with a new installation of a remote site today, they  
>>> have a PRI E1 on a 1760 router with a range of DDI numbers.
>>> ?
>>> When I added the following dial peers to allow 4 digit dialling when  
>>> the WAN is down across SRST we came across a problem. When in SRST  
>>> the 4 digit dialling worked fine.
>>> ?
>>> dial-peer voice 2000 pots
>>> destination-pattern 2?
>>> prefix 02079532
>>> port 0/0:15
>>> ?
>>> dial-peer voice 6000 pots
>>> destination-pattern 6?
>>> prefix 02071496
>>> port 0/0:15
>>> ?
>>> The issue we had was that now any inbound calls originating from 207  
>>> XXX XXXX fail, the calls came in to the router and there were no  
>>> errors on the debug then you received a dialtone after hanging up  
>>> the debug would show the call clearing down as you would expect. Any  
>>> other calls from other area codes were fine.
>>> ?
>>> Obviously the 2 prefix being presented was mismatching with the  
>>> destination-pattern 2? in the dial-peer voice 2000 pots dial peer.
>>> ?
>>> Is there anyway I can get round as we would very much like to have  
>>> these dial-peers in.
>>> ?
>>> ?
>>> ?
>>> ?
>>> ?
>>> ********************************************************************* 
>>> **************************
>>> ?
>>> 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,?distribut 
>>> ion?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?im 
>>> mediately
>>> 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?constit 
>>> ute
>>> acknowledgement?or?receipt?by?the?intended?recipient(s).?You?accept?t 
>>> hat?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?divisi 
>>> ons?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?o 
>> nly.
>> ?
>> Access?to?this?e- 
>> mail?by?any?other?person?is?prohibited.?If?you?are?not?the
>> ?
>> intended?recipient,?any?use,?disclosure,?copying,?printing,?distributi 
>> on?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?imm 
>> ediately
>> 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?constitu 
>> te
>> ?
>> acknowledgement?or?receipt?by?the?intended?recipient(s).?You?accept?th 
>> at?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?s 
>> ensitive
>> 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?divisio 
>> ns?of?Charles?Stanley?&?Co.?Ltd
>>
>> ?
>> Authorised?and?Regulated?by?the?Financial?Services?Authority,?Member?o 
>> f?the
>> ?
>> London?Stock?Exchange,?International?Securities?Markets?Association,?a 
>> nd?The?London?International?Financial?Futures?&
>> Options?Exchange.
>>
>> ?
>> This?footnote?also?confirms?that?this?email?message?has?been?swept?by? 
>> McAfee
>> VirusScan?and?SurfControl?Email?Filter?software.
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> *********************************************************************** 
> ************************
>
>   
> The?information?contained?in?this?e- 
> mail?is?strictly?confidential,?some?or?all
>   
> of?which?may?be?legally?privileged.?It?is?for?the?intended?recipient?on 
> ly.
>   
> Access?to?this?e- 
> mail?by?any?other?person?is?prohibited.?If?you?are?not?the
>   
> intended?recipient,?any?use,?disclosure,?copying,?printing,?distributio 
> n?of,
>   
> replying?to?or?any?action?taken?or?omitted?to?be?taken?in?reliance?on?t 
> his
>   
> e- 
> mail,?is?prohibited?and?may?be?unlawful.?Please?contact?the?sender?imme 
> diately
>  should?this?e-mail?have?been?incorrectly?addressed?or?transmitted.
>
>   
> You?accept?that?any?instructions?are?deemed?to?have?been?given?at?the?t 
> ime?the
>   
> recipient(s)?accesses?them?and?that?delivery?receipt?does?not?constitut 
> e
>   
> acknowledgement?or?receipt?by?the?intended?recipient(s).?You?accept?tha 
> t?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?se 
> nsitive
>  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?division 
> s?of?Charles?Stanley?&?Co.?Ltd
>
>   
> Authorised?and?Regulated?by?the?Financial?Services?Authority,?Member?of 
> ?the
>   
> London?Stock?Exchange,?International?Securities?Markets?Association,?an 
> d?The?London?International?Financial?Futures?&
>  Options?Exchange.
>
>   
> This?footnote?also?confirms?that?this?email?message?has?been?swept?by?M 
> cAfee
>  VirusScan?and?SurfControl?Email?Filter?software.
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From BRiley at jackhenry.com  Thu Jun 29 11:40:09 2006
From: BRiley at jackhenry.com (Bill Riley III)
Date: Thu, 29 Jun 2006 10:40:09 -0500
Subject: [cisco-voip] Unity 4.0.4
Message-ID: <13591CB4B2359F4D96E2606DD6B840A002BB1066@smoxchg.jhacorp.com>

 

Hi,i have Unity 4.0(4) Unified Messaging (integrated with Exchange 2003)
on Windows 2003. 



When a certain extension tries to leave a voicemail to other people
internally we get this in the event viewer.

 

Source: CiscoUnity_ConvMsg 
Category: Network 
EventID: 10015 
Description: 
IAvDohMessage::Submit returned[0x800404dc] on line 683 of file
e:\views\cs_ue4.0.3.159\un_Core2\ConversationEng\AvStateSvr\AvSMsgSend.c
pp

 

They never get the voicemail that is left by the person. This person can
still receive VM but can not send.  

 

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
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From rsingleton at novateck.com  Thu Jun 29 12:59:31 2006
From: rsingleton at novateck.com (Robert Singleton)
Date: Thu, 29 Jun 2006 11:59:31 -0500
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds
In-Reply-To: <f1b65f880606290714v2bec3002jd395b383cac8a4db@mail.gmail.com>
References: <472D0CBC7B31C1499B77C1DAA3308B3BD07B17@EXCHANGEDS01.ds.ad.adp.com>
	<200606290848593.SM01876@mail.leasenet.net>
	<f1b65f880606290714v2bec3002jd395b383cac8a4db@mail.gmail.com>
Message-ID: <200606291159109.SM01876@mail.leasenet.net>

On Thu, 2006-06-29 at 10:14 -0400, Justin Steinberg wrote:

> What version of CallManager?

At the time (July 2004), we were running 4.0(1)sr2a

> Do you have a bug id?

There were two TAC cases associated with this problem, each about a year
apart. The first engineer gave this bug report: CSCdz37627

The second gave us this one, which is what fixed the problem: CSCea74550

> Is it fixed in newer versions?

We are currently running 4.0(2a)sr2, but the Attendant Consoles are
still configured with call forwarding on the member lines, so I don't
know for sure if it has changed since then.

> AC admin guide specifically states not to enable any forwarding on member DN's.

The first response we got from TAC, once we were able to duplicate the
issue, was that it was actually working as designed. Here is an excerpt
from the case notes:

--------------
"When the park timer expires, the parked call attempts to return to the
attendant. But in this case, the attendant is busy with another call.
So, the attempt to return the call to attendant fails with Busy error
which is as per design intent.

Today you and I duplicated this several times and if we parked the call to an   
IP Phone we were able to retrive it, but if we call into AC, she parks the      
calls and the second person calls the same receptionist, it failed every time.

You can always submit a feature request through your sales partner so that call 
can go to a differetn AC member if the person initiated the park is busy."
--------------

The case was closed with the following resolution:
--------------
this is  only happening if the receptionist gets an ohter call in between and   
park call has no place to go back after expired time.

working as design.  suggested cust to submit a feature request
--------------

We apparently accepted this behavior for about a year. It is possible, 
however, that during this time, we may have reconfigured the telephones, 
accidentally applying the workaround.

In May of 2005, we had an unrelated problem that resulted in dumping the 
CallManagers and reinstalling from scratch with an unusable backup. This 
resulted in the "whitepaper" configuration of the Attendant Consoles (not
to mention a LOOONG weekend of clicking and typing in our entire database
of about 500 phones at that time) and the subsequent recurrence of the 
original complaint. We opened a new TAC case.

This engineer found the 2nd bug report quickly and referred us to it.
The verbiage of the workaround in the bug report does say that the user
is "forced" to use call forwarding (or call waiting), but that is the
closest it comes to mentioning anything counter to the manual.

Robert



From Scott.Voll at wesd.org  Thu Jun 29 15:09:15 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Thu, 29 Jun 2006 12:09:15 -0700
Subject: [cisco-voip] Agent state
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED55F1@Apollo.wesd.org>

Phone assoc. with the rmuser?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Olson, Jeffrey
Sent: Wednesday, June 28, 2006 7:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Agent state

 

I have 1 agent who has not been able to put herself in the ready state.
She is able to login but when she tried to put herself in the ready
state she receives an error saying  

"Request to change agent state failed.  Unable to change agent state."

I have removed her from the IPCC deleting her user account and recreated
it.  I have stopped and started the engine.  Still nothing.  

I can't figure out why this is only happening with this one person and
not anyone else.

 

Any ideas?

 

 

Jeff

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

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Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
hereby notified that disclosing, copying, distributing, or taking any
action in reliance on the contents of this information is strictly
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From erickbe at yahoo.com  Thu Jun 29 15:20:03 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu, 29 Jun 2006 12:20:03 -0700 (PDT)
Subject: [cisco-voip] 7920s crashing CCM 3.3(3)sr4a
Message-ID: <20060629192003.8751.qmail@web37205.mail.mud.yahoo.com>

Anyone ever see when a 7920 goes out of range and unregisters, then comes back the CCM service terminates?

7920s w/2.01 firmware, CCM 3.3(3)sr4a


From rratliff at cisco.com  Thu Jun 29 15:23:26 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 29 Jun 2006 15:23:26 -0400
Subject: [cisco-voip] 7920s crashing CCM 3.3(3)sr4a
In-Reply-To: <20060629192003.8751.qmail@web37205.mail.mud.yahoo.com>
References: <20060629192003.8751.qmail@web37205.mail.mud.yahoo.com>
Message-ID: <047F3E3B-BCDC-47EE-9C40-833FD434A30C@cisco.com>

Yup, it's a certain message the phone sends twice during registration.

CSCuk51513 fixed in 3.3(3)ES53, 3.3(4)ES14 and later.

You need to get the last ES for 3.3(3) since SR4a does not have the fix.

-Ryan

On Jun 29, 2006, at 3:20 PM, Erick Bergquist wrote:

Anyone ever see when a 7920 goes out of range and unregisters, then  
comes back the CCM service terminates?

7920s w/2.01 firmware, CCM 3.3(3)sr4a

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


From erickbe at yahoo.com  Thu Jun 29 15:30:45 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu, 29 Jun 2006 12:30:45 -0700 (PDT)
Subject: [cisco-voip] 7920s crashing CCM 3.3(3)sr4a
In-Reply-To: <047F3E3B-BCDC-47EE-9C40-833FD434A30C@cisco.com>
Message-ID: <20060629193045.55953.qmail@web37202.mail.mud.yahoo.com>

Yea, I just found that myself. Thanks for the quick reply.  Is 3.3.3 ES72 the latest ES for 3.3.3 ?

----- 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: Thursday, June 29, 2006 2:23:26 PM
Subject: Re: [cisco-voip] 7920s crashing CCM 3.3(3)sr4a

Yup, it's a certain message the phone sends twice during registration.

CSCuk51513 fixed in 3.3(3)ES53, 3.3(4)ES14 and later.

You need to get the last ES for 3.3(3) since SR4a does not have the fix.

-Ryan

On Jun 29, 2006, at 3:20 PM, Erick Bergquist wrote:

Anyone ever see when a 7920 goes out of range and unregisters, then  
comes back the CCM service terminates?

7920s w/2.01 firmware, CCM 3.3(3)sr4a

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






From rratliff at cisco.com  Thu Jun 29 15:31:03 2006
From: rratliff at cisco.com (Ryan Ratliff)
Date: Thu, 29 Jun 2006 15:31:03 -0400
Subject: [cisco-voip] 7920s crashing CCM 3.3(3)sr4a
In-Reply-To: <20060629193045.55953.qmail@web37202.mail.mud.yahoo.com>
References: <20060629193045.55953.qmail@web37202.mail.mud.yahoo.com>
Message-ID: <A19DA501-E2C6-4051-AD92-E4E6B5858247@cisco.com>

Close enough, 76 is the absolute latest.

-Ryan

On Jun 29, 2006, at 3:30 PM, Erick Bergquist wrote:

Yea, I just found that myself. Thanks for the quick reply.  Is 3.3.3  
ES72 the latest ES for 3.3.3 ?

----- 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: Thursday, June 29, 2006 2:23:26 PM
Subject: Re: [cisco-voip] 7920s crashing CCM 3.3(3)sr4a

Yup, it's a certain message the phone sends twice during registration.

CSCuk51513 fixed in 3.3(3)ES53, 3.3(4)ES14 and later.

You need to get the last ES for 3.3(3) since SR4a does not have the fix.

-Ryan

On Jun 29, 2006, at 3:20 PM, Erick Bergquist wrote:

Anyone ever see when a 7920 goes out of range and unregisters, then
comes back the CCM service terminates?

7920s w/2.01 firmware, CCM 3.3(3)sr4a

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




From erickbe at yahoo.com  Thu Jun 29 15:51:36 2006
From: erickbe at yahoo.com (Erick Bergquist)
Date: Thu, 29 Jun 2006 12:51:36 -0700 (PDT)
Subject: [cisco-voip] 7920s crashing CCM 3.3(3)sr4a
In-Reply-To: <A19DA501-E2C6-4051-AD92-E4E6B5858247@cisco.com>
Message-ID: <20060629195136.72670.qmail@web37211.mail.mud.yahoo.com>

Thanks... 

----- 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: Thursday, June 29, 2006 2:31:03 PM
Subject: Re: [cisco-voip] 7920s crashing CCM 3.3(3)sr4a

Close enough, 76 is the absolute latest.

-Ryan

On Jun 29, 2006, at 3:30 PM, Erick Bergquist wrote:

Yea, I just found that myself. Thanks for the quick reply.  Is 3.3.3  
ES72 the latest ES for 3.3.3 ?

----- 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: Thursday, June 29, 2006 2:23:26 PM
Subject: Re: [cisco-voip] 7920s crashing CCM 3.3(3)sr4a

Yup, it's a certain message the phone sends twice during registration.

CSCuk51513 fixed in 3.3(3)ES53, 3.3(4)ES14 and later.

You need to get the last ES for 3.3(3) since SR4a does not have the fix.

-Ryan

On Jun 29, 2006, at 3:20 PM, Erick Bergquist wrote:

Anyone ever see when a 7920 goes out of range and unregisters, then
comes back the CCM service terminates?

7920s w/2.01 firmware, CCM 3.3(3)sr4a

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








From kfedor at amt.ru  Thu Jun 29 16:53:45 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Fri, 30 Jun 2006 00:53:45 +0400
Subject: [cisco-voip] Parked and Hold Calls Dropped at 50 Seconds
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC70648ECF2@amthqex01.amt.ru>




--------------
working as design.  suggested cust to submit a feature request
--------------

Very familiar answer :), it is easy do nothing then do something.

The question is , if design is bad ?

Actually I think that there is no design - it works how this was
programmed and no body think how it must work. 

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




From kfedor at amt.ru  Thu Jun 29 17:09:23 2006
From: kfedor at amt.ru (Fedorov, Konstantin)
Date: Fri, 30 Jun 2006 01:09:23 +0400
Subject: [cisco-voip] clid on fxs ports
Message-ID: <8B5577ED1D5D2F4AA0E4651EC41C6DC70648ECF3@amthqex01.amt.ru>

Hi,

 

Csim start - how I remember is not supported command.

 

Can you provide debug when you call from actual phone attached to fxs ?

-----------------------
Sincerely Yours,
Konstantin Fedorov

  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony
Kouloglou
Sent: Thursday, June 29, 2006 2:19 PM
To: Ryan Ratliff
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] clid on fxs ports

 

Ryan
here is a csim start command from that router

Lon2851#csim start 90030xxxxxxxxxx
csim: called number = 90030xxxxxxxxxx, loop count = 1 ping count = 0

*Jun 29 11:11:08.959: ISDN Se0/2/0:15 Q931: Applying typeplan for
sw-type 0x12 is 0x0 0x0, Called num 0030xxxxxxxxxx
*Jun 29 11:11:08.959: ISDN Se0/2/0:15 Q931: TX -> SETUP pd = 8  callref
= 0x0164 
        Bearer Capability i = 0x8090A3 
                Standard = CCITT 
                Transfer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA9839E 
                Exclusive, Channel 30 
        Called Party Number i = 0x80, '0030xxxxxxxxxx' 
                Plan:Unknown, Type:Unknown
*Jun 29 11:11:09.103: ISDN Se0/2/0:15 Q931: RX <- SETUP_ACK pd = 8
callref = 0x8164 
        Channel ID i = 0xA9839E 
                Exclusive, Channel 30 
        Progress Ind i = 0x8288 - In-band info or appropriate now
available 
*Jun 29 11:11:15.555: ISDN Se0/2/0:15 Q931: RX <- ALERTING pd = 8
callref = 0x8164
*Jun 29 11:11:17.839: ISDN Se0/2/0:15 Q931: RX <- CONNECT pd = 8
callref = 0x8164
*Jun 29 11:11:17.839: %ISDN-6-CONNECT: Interface Serial0/2/0:29 is now
connected to xxxxxxxxxx N/A
*Jun 29 11:11:17.839: ISDN Se0/2/0:15 Q931: TX -> CONNECT_ACK pd = 8
callref = 0x0164
*Jun 29 11:11:18.391: ISDN Se0/2/0:15 Q931: RX <- DISCONNECT pd = 8
callref = 0x8164 
        Cause i = 0x8090 - Normal call clearing 
        Progress Ind i = 0x8288 - In-band info or appropriate now
available 
*Jun 29 11:11:18.395: ISDN Se0/2/0:15 Q931: call_disc: PI received in
disconnect; Postpone sending RELEASE for callid 0x8164.
csim: loop = 1, failed = 0  
csim: call attempted = 1, setup failed = 0, tone failed = 1

Lon2851#

I guess there is no Calling party in the setup message.
When there is a call from an ip phone, there is calling party (the
extension of the ccm). Of course the POTS provider just shows the number
of the PRI to the called party and not the internal extension of the
CCM.
But when i make calls from the router ( with csim start) or the fxs (a
modem attached there), there is no calling party on the q931setup
message and so the provider does not shows anything to the called.
I have tried configuring isdn calling-number etc on the serial 0/2/0:15
but nothing changes. Must i put something there or at the dial-peer
pots?
The customer wants this feature desperately!!!

Cheers,
Anthony
Ryan Ratliff wrote: 

Do you have a q931 debug showing the outbound call? 

-Ryan 

On Jun 28, 2006, at 12:03 PM, Anthony Kouloglou wrote: 

Hi all, 
on a 2851 with one E1 and 4 FXS ports i have configured the voice port 
as follows: 

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From duncanw at otc.edu  Thu Jun 29 17:45:10 2006
From: duncanw at otc.edu (Brian Duncan)
Date: Thu, 29 Jun 2006 16:45:10 -0500
Subject: [cisco-voip] CER Administrative Password Problem
Message-ID: <03e301c69bc5$4bcce6a0$6d0110ac@otc.edu>

Perhaps we are all just missing something, but we found today that ANY of
the users on our network can log into any of the Cisco Emergency Responder
web pages.  Only our IP Telephony group is listed in any of the four groups
controlling admission for these pages.  Only our admin group can log onto
the server itself, but the administrative web sites are wide open.  We are
running CER 1.3(1a).  Has anyone else experienced this?    

 

Thanks,

Brian Duncan

Assistant Coordinator of Telecommunications

Ozarks Technical Community College

 

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From ml at nesys.it  Thu Jun 29 20:20:49 2006
From: ml at nesys.it (Andrea Riela)
Date: Fri, 30 Jun 2006 02:20:49 +0200
Subject: [cisco-voip] about cme 4.x sip softphone and g729
Message-ID: <45b508c7a511049cb2466073fb20dae0@nesys.it>

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


Hi folks,

it's normal that a x-lite softphone (mac os x version) registered on 
CME 4.x system:

- - calls correctly all internal numbers (ip phones Cisco)

- - the client supports only g711a/u, GSM, iLBC and SPX ... then it uses 
g711a/u ... but all dial-peer voip use g729, then when I try to call a 
voip number, cme try to connect that with dial-peer pots, and it 
doesn't work ... no way to do that? I could only:
1- change dial-peer voip codec to g711
2- change softphone with g729 support
???

any advice will be appreciated
Regards
Andrea
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (Darwin)

iD8DBQFEpG5jMakHrsrHP9wRAiKpAJ9ntMJZMwhqfNaBf33yXZ4ptp55eQCgv2uK
yFsuB+qGTJjER2FXwHcp35E=
=iHgg
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From jim at tgasolutions.com  Thu Jun 29 20:49:24 2006
From: jim at tgasolutions.com (Jim McBurnett)
Date: Thu, 29 Jun 2006 20:49:24 -0400
Subject: [cisco-voip] Cisco & Avaya integration
Message-ID: <D2BBAE709B4D874DAC556C7EB9D938FC15EEAE@exchange.TGASolutions.local>


But does every Avaya phone support the SIP load?


SIP - CME 3.4 or 4.0

SIP trunks too, if Avaya supports it!

 Thanks,
jim

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Marus
Sent: Wednesday, June 28, 2006 11:52 AM
To: robert.bell1 at comcast.net
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Cisco & Avaya integration

I work for a company that deploys both avaya and cisco voip and we have
been asked to similar stuff. I have done the opposite, that is getting a
cisco phone with a sip image working on an avaya. For what you want, as
far as i know, this would require the Avaya phones to be running the
avaya sip image and you would need cm5 for sip phone set support. I am
wanting to lab up something similar myself using the 4610/4620's. The
4606 is an older avaya phone and may not support sip, and I dont think
it is being sold anymore. I cant recall tho.

For #2, Cisco Phones use tftp just as avaya does. the server can be
anything that can do host tftp. Avaya's files are slightly differnt in
format, but essentially the same. A config file and a firmware load.
Avaya does require differnt dhcp options to tell the phone where the
tftp server is tho. Cisco =150, avaya is 176 and uses a string that
includes several required options.

Everything between avaya and Cisco can integrate (both directions) if
you use h323 on your cisco gateways. i have a callmanager that gets
trunks directly from an avaya s8700 with no external cisco gateway and I
have worked on some other sites that have avaya trunking to a cisco
h323 gateway.

Nick

On 6/28/06, robert.bell1 at comcast.net <robert.bell1 at comcast.net> wrote:
>
>
>
> Maybe I'm crazy, but we are upgrading out layer 2 infrastructure to 
> Cisco 3560 switches and the question was raised, what would that do to

> our Avaya VoIP solution (Please don't flame me, I have no choice over
VoIP solutions).
>  I'm much more familiar with Cisco CallManager and CCME.
>
>
>
> I got this crazy idea though:  Why can't I make Avaya phones work with

> our CCME routers?  I'm not really sure how they operate, but I assume 
> they are similar to Cisco phones.  I think I found some firmware I 
> have to push to them, but I have a few questions:
>
> 1.  Has anyone successfully done what I propose (Avaya IP phones 
> working on a Cisco VoIP network?  And if so can I see your config 
> files relevant to making that work.
>
> 2.  Does anyone know the requirements for the Avaya IP phones?  For 
> instance, I know the Cisco phones require the firmware to be 
> downloaded and pulls the config from the router.  It looks like the 
> Avaya phones also need to pull the config as a file from a tftp 
> server, but I really don't know anything about Avaya VoIP.  If anyone 
> has a link to the basic requirements / operation I'd appreciate it. 
> Oh, and FYI - I've got an Avaya 4606 phone to play with.
>
>
>
> My basic goal was to make sure the phones were compatible with the PoE

> provided by the Cisco 3560 switches (which it is) and to make sure 
> that it would continue to work with the existing Avaya VoIP solution 
> (which it will). I just thought it would be neat to see them work 
> integrated and possibly use SRST for remote survivability of our
outsites using VoIP.
>
>
>
> B.
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>


-- 

Nick Marus
nmarus at gmail.com
homepage: http://www.nicholasmarus.com
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


From kthorngr at cisco.com  Thu Jun 29 20:46:49 2006
From: kthorngr at cisco.com (Kevin Thorngren)
Date: Thu, 29 Jun 2006 20:46:49 -0400
Subject: [cisco-voip] CER Administrative Password Problem
In-Reply-To: <03e301c69bc5$4bcce6a0$6d0110ac@otc.edu>
References: <03e301c69bc5$4bcce6a0$6d0110ac@otc.edu>
Message-ID: <f8ba0b29dda87e6e25bb48c07d657abb@cisco.com>

Do you have more than one CER server?

If so then.....
"You should include the same people (user names and passwords) in each  
group on both the primary and standby servers within a single Cisco ER  
group. However, access is allowed based on the union of the user groups  
defined on the two servers. Thus, if a user is only defined in the  
group on the primary server, that user can still log into the backup  
server."
http://www.cisco.com/univercd/cc/td/doc/product/voice/respond/res13/ 
admin13/e911conf.htm#wp1034982

HTH,
Kevin
On Jun 29, 2006, at 5:45 PM, Brian Duncan wrote:

>
> Perhaps we are all just missing something, but we found today that ANY  
> of the users on our network can log into any of the Cisco Emergency  
> Responder web pages.? Only our IP Telephony group is listed in any of  
> the four groups controlling admission for these pages.? Only our admin  
> group can log onto the server itself, but the administrative web sites  
> are wide open.? We are running CER 1.3(1a).? Has anyone else  
> experienced this?? ??
> ?
> Thanks,
> Brian Duncan
> Assistant Coordinator of Telecommunications
> Ozarks Technical Community College
> ?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
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From bcarter at sentinel.com  Thu Jun 29 22:15:25 2006
From: bcarter at sentinel.com (Carter, Bill)
Date: Thu, 29 Jun 2006 21:15:25 -0500
Subject: [cisco-voip] CDR insert
Message-ID: <6A0744639B86DD459433108DBF101D26FC6886@MAIL01.sentinel.com>

Brain fade here....Do I need CDR insert enabled on all CallManagers or
just the Publisher?



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 
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From pwalenta at wi.rr.com  Thu Jun 29 22:23:44 2006
From: pwalenta at wi.rr.com (Philip Walenta)
Date: Thu, 29 Jun 2006 21:23:44 -0500
Subject: [cisco-voip] CDR insert
In-Reply-To: <6A0744639B86DD459433108DBF101D26FC6886@MAIL01.sentinel.com>
Message-ID: <001001c69bec$36db32c0$0232a8c0@walentawk1>

Just the publisher.


  _____  

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Carter, Bill
Sent: Thursday, June 29, 2006 9:15 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] CDR insert



Brain fade here..Do I need CDR insert enabled on all CallManagers or just
the Publisher?



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,
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From jgrace at digitelusa.net  Fri Jun 30 00:20:57 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri, 30 Jun 2006 00:20:57 -0400
Subject: [cisco-voip] how do you get a copy of the CME aa script
Message-ID: <200606300033187.SM01452@3729434770>

I want to modify the cue aa script 2.2.2   but I need a copy of it so that I
can make the changes without having to do it all from the beginning.  Is
there one out there 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From nmarus at gmail.com  Fri Jun 30 09:40:26 2006
From: nmarus at gmail.com (Nick Marus)
Date: Fri, 30 Jun 2006 09:40:26 -0400
Subject: [cisco-voip] Cisco & Avaya integration
In-Reply-To: <D2BBAE709B4D874DAC556C7EB9D938FC15EEAE@exchange.TGASolutions.local>
References: <D2BBAE709B4D874DAC556C7EB9D938FC15EEAE@exchange.TGASolutions.local>
Message-ID: <1a9f6d3f0606300640t70b77e4jd36d448c8a09219b@mail.gmail.com>

Yes. You can get firmware to convert the phone to sip from Avaya.

On 6/29/06, Jim McBurnett <jim at tgasolutions.com> wrote:
>
> But does every Avaya phone support the SIP load?
>
>
> SIP - CME 3.4 or 4.0
>
> SIP trunks too, if Avaya supports it!
>
>  Thanks,
> jim
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Nick Marus
> Sent: Wednesday, June 28, 2006 11:52 AM
> To: robert.bell1 at comcast.net
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Cisco & Avaya integration
>
> I work for a company that deploys both avaya and cisco voip and we have
> been asked to similar stuff. I have done the opposite, that is getting a
> cisco phone with a sip image working on an avaya. For what you want, as
> far as i know, this would require the Avaya phones to be running the
> avaya sip image and you would need cm5 for sip phone set support. I am
> wanting to lab up something similar myself using the 4610/4620's. The
> 4606 is an older avaya phone and may not support sip, and I dont think
> it is being sold anymore. I cant recall tho.
>
> For #2, Cisco Phones use tftp just as avaya does. the server can be
> anything that can do host tftp. Avaya's files are slightly differnt in
> format, but essentially the same. A config file and a firmware load.
> Avaya does require differnt dhcp options to tell the phone where the
> tftp server is tho. Cisco =150, avaya is 176 and uses a string that
> includes several required options.
>
> Everything between avaya and Cisco can integrate (both directions) if
> you use h323 on your cisco gateways. i have a callmanager that gets
> trunks directly from an avaya s8700 with no external cisco gateway and I
> have worked on some other sites that have avaya trunking to a cisco
> h323 gateway.
>
> Nick
>
> On 6/28/06, robert.bell1 at comcast.net <robert.bell1 at comcast.net> wrote:
> >
> >
> >
> > Maybe I'm crazy, but we are upgrading out layer 2 infrastructure to
> > Cisco 3560 switches and the question was raised, what would that do to
>
> > our Avaya VoIP solution (Please don't flame me, I have no choice over
> VoIP solutions).
> >  I'm much more familiar with Cisco CallManager and CCME.
> >
> >
> >
> > I got this crazy idea though:  Why can't I make Avaya phones work with
>
> > our CCME routers?  I'm not really sure how they operate, but I assume
> > they are similar to Cisco phones.  I think I found some firmware I
> > have to push to them, but I have a few questions:
> >
> > 1.  Has anyone successfully done what I propose (Avaya IP phones
> > working on a Cisco VoIP network?  And if so can I see your config
> > files relevant to making that work.
> >
> > 2.  Does anyone know the requirements for the Avaya IP phones?  For
> > instance, I know the Cisco phones require the firmware to be
> > downloaded and pulls the config from the router.  It looks like the
> > Avaya phones also need to pull the config as a file from a tftp
> > server, but I really don't know anything about Avaya VoIP.  If anyone
> > has a link to the basic requirements / operation I'd appreciate it.
> > Oh, and FYI - I've got an Avaya 4606 phone to play with.
> >
> >
> >
> > My basic goal was to make sure the phones were compatible with the PoE
>
> > provided by the Cisco 3560 switches (which it is) and to make sure
> > that it would continue to work with the existing Avaya VoIP solution
> > (which it will). I just thought it would be neat to see them work
> > integrated and possibly use SRST for remote survivability of our
> outsites using VoIP.
> >
> >
> >
> > B.
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
>
>
> --
>
> Nick Marus
> nmarus at gmail.com
> homepage: http://www.nicholasmarus.com
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>


-- 

Nick Marus
nmarus at gmail.com
homepage: http://www.nicholasmarus.com

From JOlson at mlnusa.com  Fri Jun 30 10:46:58 2006
From: JOlson at mlnusa.com (Olson, Jeffrey)
Date: Fri, 30 Jun 2006 10:46:58 -0400
Subject: [cisco-voip] Agent state
Message-ID: <FE2D092F8B1516488B99342551AACFD41D7BED3A@mlnusa-exch1.mlnusa.com>

Yes I verified that the phone is associated with rmuser.  The user is
able to login and logout.  It's putting the phone in the ready state
where we run into the issue.

 

Jeff

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org] 
Sent: Thursday, June 29, 2006 3:09 PM
To: Olson, Jeffrey; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Agent state

 

Phone assoc. with the rmuser?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Olson, Jeffrey
Sent: Wednesday, June 28, 2006 7:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Agent state

 

I have 1 agent who has not been able to put herself in the ready state.
She is able to login but when she tried to put herself in the ready
state she receives an error saying  

"Request to change agent state failed.  Unable to change agent state."

I have removed her from the IPCC deleting her user account and recreated
it.  I have stopped and started the engine.  Still nothing.  

I can't figure out why this is only happening with this one person and
not anyone else.

 

Any ideas?

 

 

Jeff

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

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Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
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From david.strzemienski at digiconasp.com  Fri Jun 30 11:16:08 2006
From: david.strzemienski at digiconasp.com (Dave Strzemienski)
Date: Fri, 30 Jun 2006 11:16:08 -0400
Subject: [cisco-voip] Agent state
In-Reply-To: <FE2D092F8B1516488B99342551AACFD41D7BED3A@mlnusa-exch1.mlnusa.com>
Message-ID: <AACDDB871EEE6B45A43740896323B5B5099C7C@MX.digiconasp.com>

Last time I ran into this error I cold booted the user's phone and it
fixed the problem.

Could be worth a shot...

-Dave

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Olson, Jeffrey
Sent: Friday, June 30, 2006 10:47 AM
To: Voll, Scott; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Agent state

Yes I verified that the phone is associated with rmuser.  The user is
able to login and logout.  It's putting the phone in the ready state
where we run into the issue.

 

Jeff

 

________________________________

From: Voll, Scott [mailto:Scott.Voll at wesd.org]
Sent: Thursday, June 29, 2006 3:09 PM
To: Olson, Jeffrey; cisco-voip at puck.nether.net
Subject: RE: [cisco-voip] Agent state

 

Phone assoc. with the rmuser?

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Olson, Jeffrey
Sent: Wednesday, June 28, 2006 7:00 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Agent state

 

I have 1 agent who has not been able to put herself in the ready state.
She is able to login but when she tried to put herself in the ready
state she receives an error saying  

"Request to change agent state failed.  Unable to change agent state."

I have removed her from the IPCC deleting her user account and recreated
it.  I have stopped and started the engine.  Still nothing.  

I can't figure out why this is only happening with this one person and
not anyone else.

 

Any ideas?

 

 

Jeff

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

CONFIDENTIALITY NOTICE:

This E-mail transmission contains information from Mortgage Lenders
Network USA, Inc. which is confidential and legally privileged. The
information is intended only for the use of the individual or entity
named on this E-mail. If you are not the intended recipient, you are
hereby notified that disclosing, copying, distributing, or taking any
action in reliance on the contents of this information is strictly
prohibited, and that the E-mail should be returned to this company
immediately. If you have received this E-mail in error, please contact
the sender immediately.

 



From cshaffer at gmail.com  Fri Jun 30 11:16:35 2006
From: cshaffer at gmail.com (Curt Shaffer)
Date: Fri, 30 Jun 2006 11:16:35 -0400
Subject: [cisco-voip] Voice IOS oddity
Message-ID: <002501c69c58$2d6b0000$181b10ac@betsy>

I know this is not router support forum but I am wondering if any of you
have seen this. I uploaded c1700-sv3y-mz.121-27b.bin to my 1750. The
procedure went fine and if I do a dir flash: it shows a good checksum. Now
if I boot in normal manner it says it can't find the filename
c1700-sv3y-mz.121-27b.bin, but if I boot from rommon it loads with no
problem. Any clues?

 

Thanks

 

Curt

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From Scott.Voll at wesd.org  Fri Jun 30 11:21:04 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 30 Jun 2006 08:21:04 -0700
Subject: [cisco-voip] Voice IOS oddity
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED56AB@Apollo.wesd.org>

Add the statements Boot system flash:blah to your config and make sure
your configreg is set to x2102.  see if that does anything for you.  

 

Scott

 

PS.  Why are you running such in old IOS?

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Curt Shaffer
Sent: Friday, June 30, 2006 8:17 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Voice IOS oddity

 

I know this is not router support forum but I am wondering if any of you
have seen this. I uploaded c1700-sv3y-mz.121-27b.bin to my 1750. The
procedure went fine and if I do a dir flash: it shows a good checksum.
Now if I boot in normal manner it says it can't find the filename
c1700-sv3y-mz.121-27b.bin, but if I boot from rommon it loads with no
problem. Any clues?

 

Thanks

 

Curt

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From Scott.Voll at wesd.org  Fri Jun 30 11:27:02 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 30 Jun 2006 08:27:02 -0700
Subject: [cisco-voip] PBX integration
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED56AF@Apollo.wesd.org>

Has anyone ever done a Unity integration with a Nitsuko?  If so, what
kind of integration was it?

 

Thanks

 

Scott

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From aman.chugh at gmail.com  Fri Jun 30 13:19:50 2006
From: aman.chugh at gmail.com (Aman Chugh)
Date: Fri, 30 Jun 2006 22:49:50 +0530
Subject: [cisco-voip] PBX integration
In-Reply-To: <407055A92CECCB499C922A2D35FC19A605ED56AF@Apollo.wesd.org>
References: <407055A92CECCB499C922A2D35FC19A605ED56AF@Apollo.wesd.org>
Message-ID: <d4c697ad0606301019x7268ec89ja7cb19c27fe95249@mail.gmail.com>

PBX supported by Unity are listed on cco.Whatt is this PBX ,is it IP or
TDM.If ip ,Unity does support integration with SIP phone systems.

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_system_requirements_hardware09186a00806cd9d2.html#wp465631

-- Aman

On 6/30/06, Voll, Scott <Scott.Voll at wesd.org> wrote:
>
>   Has anyone ever done a Unity integration with a Nitsuko?  If so, what
> kind of integration was it?
>
>
>
> Thanks
>
>
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From jonvoip at gmail.com  Fri Jun 30 15:39:18 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 30 Jun 2006 14:39:18 -0500
Subject: [cisco-voip] Removing users from the Corporate Directory (on the
	phones)
Message-ID: <5d093f9a0606301239s42991e12x47cd3f243a6496eb@mail.gmail.com>

So, I have CCM4.1(3), but I am seeing my admin accounts (JTAPI,
RMJTAPI, appadmin, etc...) showing up in the directory on the phone...

How do I remove them?




Jonathan

From Joe.Pollere at us.didata.com  Fri Jun 30 16:16:41 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Fri, 30 Jun 2006 16:16:41 -0400
Subject: [cisco-voip] Removing users from the Corporate Directory (on
	thephones)
Message-ID: <C65033A817B5734392E55CB9DF44610405A027ED@USNAEXCH.na.didata.local>

Jonathan,

Add CiscoPrivateUser to the description field.

>From page 14-17 of the CM 4.1 SRND:

You can prevent system and service accounts from appearing in Cisco
CallManager Administration by adding the string CiscoPrivateUser to the
user's Description field. The CCM Administrator, CCMSysUser, and IPMA
SysUser accounts have this field set by default, but you can safely add
the
description to the CCM Directory Manager account as well. Use Microsoft
ADSIEdit (Active Directory Service Interfaces, available as a part of
the Windows 2000 Support Tools) or any other LDAP tool to
update the Description field.

Regards,

Joe

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Friday, June 30, 2006 3:39 PM
To: ciscovoip
Subject: [cisco-voip] Removing users from the Corporate Directory (on
thephones)

So, I have CCM4.1(3), but I am seeing my admin accounts (JTAPI,
RMJTAPI, appadmin, etc...) showing up in the directory on the phone...

How do I remove them?




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

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

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



From jonvoip at gmail.com  Fri Jun 30 16:20:54 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 30 Jun 2006 15:20:54 -0500
Subject: [cisco-voip] Removing users from the Corporate Directory (on
	thephones)
In-Reply-To: <C65033A817B5734392E55CB9DF44610405A027ED@USNAEXCH.na.didata.local>
References: <C65033A817B5734392E55CB9DF44610405A027ED@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0606301320k365a22fak56490a71ec383258@mail.gmail.com>

Sounds good, but there I still want to be able to work with these
accounts. I just want them to not show up on the phone, directory,
corporate directory...



Jonathan

On 6/30/06, Joe Pollere (US) <Joe.Pollere at us.didata.com> wrote:
> Jonathan,
>
> Add CiscoPrivateUser to the description field.
>
> From page 14-17 of the CM 4.1 SRND:
>
> You can prevent system and service accounts from appearing in Cisco
> CallManager Administration by adding the string CiscoPrivateUser to the
> user's Description field. The CCM Administrator, CCMSysUser, and IPMA
> SysUser accounts have this field set by default, but you can safely add
> the
> description to the CCM Directory Manager account as well. Use Microsoft
> ADSIEdit (Active Directory Service Interfaces, available as a part of
> the Windows 2000 Support Tools) or any other LDAP tool to
> update the Description field.
>
> Regards,
>
> Joe
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Friday, June 30, 2006 3:39 PM
> To: ciscovoip
> Subject: [cisco-voip] Removing users from the Corporate Directory (on
> thephones)
>
> So, I have CCM4.1(3), but I am seeing my admin accounts (JTAPI,
> RMJTAPI, appadmin, etc...) showing up in the directory on the phone...
>
> How do I remove them?
>
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
>

From Matt.Slaga at us.didata.com  Fri Jun 30 16:37:43 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 30 Jun 2006 16:37:43 -0400
Subject: [cisco-voip] Removing users from the Corporate Directory
	(onthephones)
Message-ID: <C1FE15183DA37645BC0633BC604E44F0025B6437@USNAEXCH.na.didata.local>

They took it out in 4.1, but you used to be able to leave out the first
name to omit from the directory.

Now it is a little more involved:

1.  Open Notepad on the publisher server.

2.  Cut and paste these four lines and save the file as a text file on
the C: drive as hideuser.ldif.

Note:  The spaces should appear as this output shows. Ensure that there
are no extra spaces in the LDIF file. Otherwise, the script fails when
you run it.

dn: cn=[userid], ou=users, o=cisco.com
changeType:modify
replace:Description
Description:CiscoPrivateUser

3.  Set the [userid] to be the user you want to hide. For example: 

dn: cn=ctiuser, ou=users, o=cisco.com
changeType:modify
replace:Description
Description:CiscoPrivateUser

4.  Select Start > Run > cmd to open a command prompt on the publisher
server and run this script in order to set the description field in DC
Directory: 

ldapmodify -h <server name> -p 8404 -D "cn=Directory
Manager,o=cisco.com" -w <DCDAdmin Password> -c -f hideuser.ldif.txt


This output shows a successful run:

C:\>ldapmodify -h CallManagerPub -p 8404 -D 
"cn=Directory Manager, o=cisco.com" -w Cisco -c -f hideuser.ldif.txt
modifying entry cn=ctiuser, ou=users, o=cisco.com



-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
Charles
Sent: Friday, June 30, 2006 4:21 PM
To: Joe Pollere (US)
Cc: ciscovoip
Subject: Re: [cisco-voip] Removing users from the Corporate Directory
(onthephones)

Sounds good, but there I still want to be able to work with these
accounts. I just want them to not show up on the phone, directory,
corporate directory...



Jonathan

On 6/30/06, Joe Pollere (US) <Joe.Pollere at us.didata.com> wrote:
> Jonathan,
>
> Add CiscoPrivateUser to the description field.
>
> From page 14-17 of the CM 4.1 SRND:
>
> You can prevent system and service accounts from appearing in Cisco 
> CallManager Administration by adding the string CiscoPrivateUser to 
> the user's Description field. The CCM Administrator, CCMSysUser, and 
> IPMA SysUser accounts have this field set by default, but you can 
> safely add the description to the CCM Directory Manager account as 
> well. Use Microsoft ADSIEdit (Active Directory Service Interfaces, 
> available as a part of the Windows 2000 Support Tools) or any other 
> LDAP tool to update the Description field.
>
> Regards,
>
> Joe
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan 
> Charles
> Sent: Friday, June 30, 2006 3:39 PM
> To: ciscovoip
> Subject: [cisco-voip] Removing users from the Corporate Directory (on
> thephones)
>
> So, I have CCM4.1(3), but I am seeing my admin accounts (JTAPI, 
> RMJTAPI, appadmin, etc...) showing up in the directory on the phone...
>
> How do I remove them?
>
>
>
>
> Jonathan
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain confidential

> and privileged information and is for use by the designated 
> addressee(s) named above only.  If you are not the intended addressee,

> you are hereby notified that you have received this communication in 
> error and that any use or reproduction of this email or its contents 
> is strictly prohibited and may be unlawful.  If you have received this

> communication in error, please notify us immediately by replying to 
> this message and deleting it from your computer. Thank you.
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

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

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



From jonvoip at gmail.com  Fri Jun 30 16:40:48 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 30 Jun 2006 15:40:48 -0500
Subject: [cisco-voip] Removing users from the Corporate Directory
	(onthephones)
In-Reply-To: <C1FE15183DA37645BC0633BC604E44F0025B6437@USNAEXCH.na.didata.local>
References: <C1FE15183DA37645BC0633BC604E44F0025B6437@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0606301340l6c6f8099k7084a74afd8c0d5@mail.gmail.com>

That looks really unpleasant... I will try it...



Jonathan

On 6/30/06, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> They took it out in 4.1, but you used to be able to leave out the first
> name to omit from the directory.
>
> Now it is a little more involved:
>
> 1.  Open Notepad on the publisher server.
>
> 2.  Cut and paste these four lines and save the file as a text file on
> the C: drive as hideuser.ldif.
>
> Note:  The spaces should appear as this output shows. Ensure that there
> are no extra spaces in the LDIF file. Otherwise, the script fails when
> you run it.
>
> dn: cn=[userid], ou=users, o=cisco.com
> changeType:modify
> replace:Description
> Description:CiscoPrivateUser
>
> 3.  Set the [userid] to be the user you want to hide. For example:
>
> dn: cn=ctiuser, ou=users, o=cisco.com
> changeType:modify
> replace:Description
> Description:CiscoPrivateUser
>
> 4.  Select Start > Run > cmd to open a command prompt on the publisher
> server and run this script in order to set the description field in DC
> Directory:
>
> ldapmodify -h <server name> -p 8404 -D "cn=Directory
> Manager,o=cisco.com" -w <DCDAdmin Password> -c -f hideuser.ldif.txt
>
>
> This output shows a successful run:
>
> C:\>ldapmodify -h CallManagerPub -p 8404 -D
> "cn=Directory Manager, o=cisco.com" -w Cisco -c -f hideuser.ldif.txt
> modifying entry cn=ctiuser, ou=users, o=cisco.com
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> Charles
> Sent: Friday, June 30, 2006 4:21 PM
> To: Joe Pollere (US)
> Cc: ciscovoip
> Subject: Re: [cisco-voip] Removing users from the Corporate Directory
> (onthephones)
>
> Sounds good, but there I still want to be able to work with these
> accounts. I just want them to not show up on the phone, directory,
> corporate directory...
>
>
>
> Jonathan
>
> On 6/30/06, Joe Pollere (US) <Joe.Pollere at us.didata.com> wrote:
> > Jonathan,
> >
> > Add CiscoPrivateUser to the description field.
> >
> > From page 14-17 of the CM 4.1 SRND:
> >
> > You can prevent system and service accounts from appearing in Cisco
> > CallManager Administration by adding the string CiscoPrivateUser to
> > the user's Description field. The CCM Administrator, CCMSysUser, and
> > IPMA SysUser accounts have this field set by default, but you can
> > safely add the description to the CCM Directory Manager account as
> > well. Use Microsoft ADSIEdit (Active Directory Service Interfaces,
> > available as a part of the Windows 2000 Support Tools) or any other
> > LDAP tool to update the Description field.
> >
> > Regards,
> >
> > Joe
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > Charles
> > Sent: Friday, June 30, 2006 3:39 PM
> > To: ciscovoip
> > Subject: [cisco-voip] Removing users from the Corporate Directory (on
> > thephones)
> >
> > So, I have CCM4.1(3), but I am seeing my admin accounts (JTAPI,
> > RMJTAPI, appadmin, etc...) showing up in the directory on the phone...
> >
> > How do I remove them?
> >
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain confidential
>
> > and privileged information and is for use by the designated
> > addressee(s) named above only.  If you are not the intended addressee,
>
> > you are hereby notified that you have received this communication in
> > error and that any use or reproduction of this email or its contents
> > is strictly prohibited and may be unlawful.  If you have received this
>
> > communication in error, please notify us immediately by replying to
> > this message and deleting it from your computer. Thank you.
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
>

From Matt.Slaga at us.didata.com  Fri Jun 30 16:45:39 2006
From: Matt.Slaga at us.didata.com (Matt Slaga (US))
Date: Fri, 30 Jun 2006 16:45:39 -0400
Subject: [cisco-voip] Removing users from the Corporate Directory
	(onthephones)
Message-ID: <C1FE15183DA37645BC0633BC604E44F0025B6451@USNAEXCH.na.didata.local>

Link is below...  Forgot to add it to the last post.

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_note09186a00804d2087.shtml#topic1
 

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, June 30, 2006 4:41 PM
To: Matt Slaga (US)
Cc: Joe Pollere (US); ciscovoip
Subject: Re: [cisco-voip] Removing users from the Corporate Directory
(onthephones)

That looks really unpleasant... I will try it...



Jonathan

On 6/30/06, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> They took it out in 4.1, but you used to be able to leave out the 
> first name to omit from the directory.
>
> Now it is a little more involved:
>
> 1.  Open Notepad on the publisher server.
>
> 2.  Cut and paste these four lines and save the file as a text file on

> the C: drive as hideuser.ldif.
>
> Note:  The spaces should appear as this output shows. Ensure that 
> there are no extra spaces in the LDIF file. Otherwise, the script 
> fails when you run it.
>
> dn: cn=[userid], ou=users, o=cisco.com changeType:modify 
> replace:Description Description:CiscoPrivateUser
>
> 3.  Set the [userid] to be the user you want to hide. For example:
>
> dn: cn=ctiuser, ou=users, o=cisco.com
> changeType:modify
> replace:Description
> Description:CiscoPrivateUser
>
> 4.  Select Start > Run > cmd to open a command prompt on the publisher

> server and run this script in order to set the description field in DC
> Directory:
>
> ldapmodify -h <server name> -p 8404 -D "cn=Directory 
> Manager,o=cisco.com" -w <DCDAdmin Password> -c -f hideuser.ldif.txt
>
>
> This output shows a successful run:
>
> C:\>ldapmodify -h CallManagerPub -p 8404 -D "cn=Directory Manager, 
> o=cisco.com" -w Cisco -c -f hideuser.ldif.txt modifying entry 
> cn=ctiuser, ou=users, o=cisco.com
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan 
> Charles
> Sent: Friday, June 30, 2006 4:21 PM
> To: Joe Pollere (US)
> Cc: ciscovoip
> Subject: Re: [cisco-voip] Removing users from the Corporate Directory
> (onthephones)
>
> Sounds good, but there I still want to be able to work with these 
> accounts. I just want them to not show up on the phone, directory, 
> corporate directory...
>
>
>
> Jonathan
>
> On 6/30/06, Joe Pollere (US) <Joe.Pollere at us.didata.com> wrote:
> > Jonathan,
> >
> > Add CiscoPrivateUser to the description field.
> >
> > From page 14-17 of the CM 4.1 SRND:
> >
> > You can prevent system and service accounts from appearing in Cisco 
> > CallManager Administration by adding the string CiscoPrivateUser to 
> > the user's Description field. The CCM Administrator, CCMSysUser, and

> > IPMA SysUser accounts have this field set by default, but you can 
> > safely add the description to the CCM Directory Manager account as 
> > well. Use Microsoft ADSIEdit (Active Directory Service Interfaces, 
> > available as a part of the Windows 2000 Support Tools) or any other 
> > LDAP tool to update the Description field.
> >
> > Regards,
> >
> > Joe
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan 
> > Charles
> > Sent: Friday, June 30, 2006 3:39 PM
> > To: ciscovoip
> > Subject: [cisco-voip] Removing users from the Corporate Directory 
> > (on
> > thephones)
> >
> > So, I have CCM4.1(3), but I am seeing my admin accounts (JTAPI, 
> > RMJTAPI, appadmin, etc...) showing up in the directory on the
phone...
> >
> > How do I remove them?
> >
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain 
> > confidential
>
> > and privileged information and is for use by the designated
> > addressee(s) named above only.  If you are not the intended 
> > addressee,
>
> > you are hereby notified that you have received this communication in

> > error and that any use or reproduction of this email or its contents

> > is strictly prohibited and may be unlawful.  If you have received 
> > this
>
> > communication in error, please notify us immediately by replying to 
> > this message and deleting it from your computer. Thank you.
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain confidential

> and privileged information and is for use by the designated 
> addressee(s) named above only.  If you are not the intended addressee,

> you are hereby notified that you have received this communication in 
> error and that any use or reproduction of this email or its contents 
> is strictly prohibited and may be unlawful.  If you have received this

> communication in error, please notify us immediately by replying to 
> this message and deleting it from your computer. Thank you.
>
>

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

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



From toddnh65 at gmail.com  Fri Jun 30 17:16:31 2006
From: toddnh65 at gmail.com (Todd Franklin)
Date: Fri, 30 Jun 2006 17:16:31 -0400
Subject: [cisco-voip] First Syllable Being Chopped
Message-ID: <6101bda50606301416n571510fn8d312fc9e4b883ac@mail.gmail.com>

Our Police station has 12 POTS lines which we have routed into a 2801 thru
FXOs.
They continue to say that when someone from outside calls in, the first
syllable of each sentence is chopped off by the person calling in.  It makes
it quite difficult for the officers to figure out what someone is saying
sometimes.  They also complained about a "pulsing" noise today, though I
didn't witness that firsthand.

The 2801 is configured as H323, if that matters.

Todd
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From Joe.Pollere at us.didata.com  Fri Jun 30 17:16:59 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Fri, 30 Jun 2006 17:16:59 -0400
Subject: [cisco-voip] Removing users from the Corporate Directory
	(onthephones)
Message-ID: <C65033A817B5734392E55CB9DF44610405A028FD@USNAEXCH.na.didata.local>

That's weird...When I last did this I was using the AD plugin with CM
4.1.3 as I remember I didn't have to execute ldifde -i -f hideuser.ldif.
All I did was add CiscoPrivateUser to the Description Field and it hid
those accounts from appearing in the Corporate Dir.


Joe


-----Original Message-----
From: Matt Slaga (US) 
Sent: Friday, June 30, 2006 4:46 PM
To: Jonathan Charles
Cc: Joe Pollere (US); ciscovoip
Subject: RE: [cisco-voip] Removing users from the Corporate Directory
(onthephones)

Link is below...  Forgot to add it to the last post.

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
ch_note09186a00804d2087.shtml#topic1
 

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, June 30, 2006 4:41 PM
To: Matt Slaga (US)
Cc: Joe Pollere (US); ciscovoip
Subject: Re: [cisco-voip] Removing users from the Corporate Directory
(onthephones)

That looks really unpleasant... I will try it...



Jonathan

On 6/30/06, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> They took it out in 4.1, but you used to be able to leave out the 
> first name to omit from the directory.
>
> Now it is a little more involved:
>
> 1.  Open Notepad on the publisher server.
>
> 2.  Cut and paste these four lines and save the file as a text file on

> the C: drive as hideuser.ldif.
>
> Note:  The spaces should appear as this output shows. Ensure that 
> there are no extra spaces in the LDIF file. Otherwise, the script 
> fails when you run it.
>
> dn: cn=[userid], ou=users, o=cisco.com changeType:modify 
> replace:Description Description:CiscoPrivateUser
>
> 3.  Set the [userid] to be the user you want to hide. For example:
>
> dn: cn=ctiuser, ou=users, o=cisco.com
> changeType:modify
> replace:Description
> Description:CiscoPrivateUser
>
> 4.  Select Start > Run > cmd to open a command prompt on the publisher

> server and run this script in order to set the description field in DC
> Directory:
>
> ldapmodify -h <server name> -p 8404 -D "cn=Directory 
> Manager,o=cisco.com" -w <DCDAdmin Password> -c -f hideuser.ldif.txt
>
>
> This output shows a successful run:
>
> C:\>ldapmodify -h CallManagerPub -p 8404 -D "cn=Directory Manager, 
> o=cisco.com" -w Cisco -c -f hideuser.ldif.txt modifying entry 
> cn=ctiuser, ou=users, o=cisco.com
>
>
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net
> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan 
> Charles
> Sent: Friday, June 30, 2006 4:21 PM
> To: Joe Pollere (US)
> Cc: ciscovoip
> Subject: Re: [cisco-voip] Removing users from the Corporate Directory
> (onthephones)
>
> Sounds good, but there I still want to be able to work with these 
> accounts. I just want them to not show up on the phone, directory, 
> corporate directory...
>
>
>
> Jonathan
>
> On 6/30/06, Joe Pollere (US) <Joe.Pollere at us.didata.com> wrote:
> > Jonathan,
> >
> > Add CiscoPrivateUser to the description field.
> >
> > From page 14-17 of the CM 4.1 SRND:
> >
> > You can prevent system and service accounts from appearing in Cisco 
> > CallManager Administration by adding the string CiscoPrivateUser to 
> > the user's Description field. The CCM Administrator, CCMSysUser, and

> > IPMA SysUser accounts have this field set by default, but you can 
> > safely add the description to the CCM Directory Manager account as 
> > well. Use Microsoft ADSIEdit (Active Directory Service Interfaces, 
> > available as a part of the Windows 2000 Support Tools) or any other 
> > LDAP tool to update the Description field.
> >
> > Regards,
> >
> > Joe
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan 
> > Charles
> > Sent: Friday, June 30, 2006 3:39 PM
> > To: ciscovoip
> > Subject: [cisco-voip] Removing users from the Corporate Directory 
> > (on
> > thephones)
> >
> > So, I have CCM4.1(3), but I am seeing my admin accounts (JTAPI, 
> > RMJTAPI, appadmin, etc...) showing up in the directory on the
phone...
> >
> > How do I remove them?
> >
> >
> >
> >
> > Jonathan
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain 
> > confidential
>
> > and privileged information and is for use by the designated
> > addressee(s) named above only.  If you are not the intended 
> > addressee,
>
> > you are hereby notified that you have received this communication in

> > error and that any use or reproduction of this email or its contents

> > is strictly prohibited and may be unlawful.  If you have received 
> > this
>
> > communication in error, please notify us immediately by replying to 
> > this message and deleting it from your computer. Thank you.
> >
> >
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain confidential

> and privileged information and is for use by the designated 
> addressee(s) named above only.  If you are not the intended addressee,

> you are hereby notified that you have received this communication in 
> error and that any use or reproduction of this email or its contents 
> is strictly prohibited and may be unlawful.  If you have received this

> communication in error, please notify us immediately by replying to 
> this message and deleting it from your computer. Thank you.
>
>

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

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



From Scott.Voll at wesd.org  Fri Jun 30 17:22:32 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 30 Jun 2006 14:22:32 -0700
Subject: [cisco-voip] First Syllable Being Chopped
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED575E@Apollo.wesd.org>

Turn off VAD.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Friday, June 30, 2006 2:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] First Syllable Being Chopped

 

Our Police station has 12 POTS lines which we have routed into a 2801
thru FXOs.
They continue to say that when someone from outside calls in, the first
syllable of each sentence is chopped off by the person calling in.  It
makes it quite difficult for the officers to figure out what someone is
saying sometimes.  They also complained about a "pulsing" noise today,
though I didn't witness that firsthand. 

The 2801 is configured as H323, if that matters.

Todd

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From Scott.Voll at wesd.org  Fri Jun 30 17:24:03 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 30 Jun 2006 14:24:03 -0700
Subject: [cisco-voip] First Syllable Being Chopped
Message-ID: <407055A92CECCB499C922A2D35FC19A605ED575F@Apollo.wesd.org>

dial-peer voice 6001 voip

 preference 2

 destination-pattern 6331

 voice-class codec 1

 session target ipv4:x.x.x.x

 dtmf-relay h245-alphanumeric

 no vad

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, June 30, 2006 2:23 PM
To: Todd Franklin; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] First Syllable Being Chopped

 

Turn off VAD.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Friday, June 30, 2006 2:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] First Syllable Being Chopped

 

Our Police station has 12 POTS lines which we have routed into a 2801
thru FXOs.
They continue to say that when someone from outside calls in, the first
syllable of each sentence is chopped off by the person calling in.  It
makes it quite difficult for the officers to figure out what someone is
saying sometimes.  They also complained about a "pulsing" noise today,
though I didn't witness that firsthand. 

The 2801 is configured as H323, if that matters.

Todd

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From scott.coner at us.didata.com  Fri Jun 30 17:25:59 2006
From: scott.coner at us.didata.com (Scott Coner (US))
Date: Fri, 30 Jun 2006 17:25:59 -0400
Subject: [cisco-voip] First Syllable Being Chopped
References: <407055A92CECCB499C922A2D35FC19A605ED575F@Apollo.wesd.org>
Message-ID: <C1FE15183DA37645BC0633BC604E44F00C91D3@USNAEXCH.na.didata.local>

What IOS are you running?  I had a similar issue with 12.3(14)T4.  I could not reproduce the issue after converting the gateway to MGCP.
 
Scott

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Voll, Scott
Sent: Fri 6/30/2006 5:24 PM
To: Todd Franklin; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] First Syllable Being Chopped



dial-peer voice 6001 voip

 preference 2

 destination-pattern 6331

 voice-class codec 1

 session target ipv4:x.x.x.x

 dtmf-relay h245-alphanumeric

 no vad

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Friday, June 30, 2006 2:23 PM
To: Todd Franklin; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] First Syllable Being Chopped

 

Turn off VAD.

 

Scott

 

________________________________

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Todd Franklin
Sent: Friday, June 30, 2006 2:17 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] First Syllable Being Chopped

 

Our Police station has 12 POTS lines which we have routed into a 2801 thru FXOs.
They continue to say that when someone from outside calls in, the first syllable of each sentence is chopped off by the person calling in.  It makes it quite difficult for the officers to figure out what someone is saying sometimes.  They also complained about a "pulsing" noise today, though I didn't witness that firsthand. 

The 2801 is configured as H323, if that matters.

Todd




-----------------------------------------
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 jonvoip at gmail.com  Fri Jun 30 17:29:48 2006
From: jonvoip at gmail.com (Jonathan Charles)
Date: Fri, 30 Jun 2006 16:29:48 -0500
Subject: [cisco-voip] Removing users from the Corporate Directory
	(onthephones)
In-Reply-To: <C65033A817B5734392E55CB9DF44610405A028FD@USNAEXCH.na.didata.local>
References: <C65033A817B5734392E55CB9DF44610405A028FD@USNAEXCH.na.didata.local>
Message-ID: <5d093f9a0606301429v7f6b9edakc85ffd4f7fc8c9a3@mail.gmail.com>

Description, where?



Jonathan

On 6/30/06, Joe Pollere (US) <Joe.Pollere at us.didata.com> wrote:
> That's weird...When I last did this I was using the AD plugin with CM
> 4.1.3 as I remember I didn't have to execute ldifde -i -f hideuser.ldif.
> All I did was add CiscoPrivateUser to the Description Field and it hid
> those accounts from appearing in the Corporate Dir.
>
>
> Joe
>
>
> -----Original Message-----
> From: Matt Slaga (US)
> Sent: Friday, June 30, 2006 4:46 PM
> To: Jonathan Charles
> Cc: Joe Pollere (US); ciscovoip
> Subject: RE: [cisco-voip] Removing users from the Corporate Directory
> (onthephones)
>
> Link is below...  Forgot to add it to the last post.
>
> http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
> ch_note09186a00804d2087.shtml#topic1
>
>
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Friday, June 30, 2006 4:41 PM
> To: Matt Slaga (US)
> Cc: Joe Pollere (US); ciscovoip
> Subject: Re: [cisco-voip] Removing users from the Corporate Directory
> (onthephones)
>
> That looks really unpleasant... I will try it...
>
>
>
> Jonathan
>
> On 6/30/06, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> > They took it out in 4.1, but you used to be able to leave out the
> > first name to omit from the directory.
> >
> > Now it is a little more involved:
> >
> > 1.  Open Notepad on the publisher server.
> >
> > 2.  Cut and paste these four lines and save the file as a text file on
>
> > the C: drive as hideuser.ldif.
> >
> > Note:  The spaces should appear as this output shows. Ensure that
> > there are no extra spaces in the LDIF file. Otherwise, the script
> > fails when you run it.
> >
> > dn: cn=[userid], ou=users, o=cisco.com changeType:modify
> > replace:Description Description:CiscoPrivateUser
> >
> > 3.  Set the [userid] to be the user you want to hide. For example:
> >
> > dn: cn=ctiuser, ou=users, o=cisco.com
> > changeType:modify
> > replace:Description
> > Description:CiscoPrivateUser
> >
> > 4.  Select Start > Run > cmd to open a command prompt on the publisher
>
> > server and run this script in order to set the description field in DC
> > Directory:
> >
> > ldapmodify -h <server name> -p 8404 -D "cn=Directory
> > Manager,o=cisco.com" -w <DCDAdmin Password> -c -f hideuser.ldif.txt
> >
> >
> > This output shows a successful run:
> >
> > C:\>ldapmodify -h CallManagerPub -p 8404 -D "cn=Directory Manager,
> > o=cisco.com" -w Cisco -c -f hideuser.ldif.txt modifying entry
> > cn=ctiuser, ou=users, o=cisco.com
> >
> >
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > Charles
> > Sent: Friday, June 30, 2006 4:21 PM
> > To: Joe Pollere (US)
> > Cc: ciscovoip
> > Subject: Re: [cisco-voip] Removing users from the Corporate Directory
> > (onthephones)
> >
> > Sounds good, but there I still want to be able to work with these
> > accounts. I just want them to not show up on the phone, directory,
> > corporate directory...
> >
> >
> >
> > Jonathan
> >
> > On 6/30/06, Joe Pollere (US) <Joe.Pollere at us.didata.com> wrote:
> > > Jonathan,
> > >
> > > Add CiscoPrivateUser to the description field.
> > >
> > > From page 14-17 of the CM 4.1 SRND:
> > >
> > > You can prevent system and service accounts from appearing in Cisco
> > > CallManager Administration by adding the string CiscoPrivateUser to
> > > the user's Description field. The CCM Administrator, CCMSysUser, and
>
> > > IPMA SysUser accounts have this field set by default, but you can
> > > safely add the description to the CCM Directory Manager account as
> > > well. Use Microsoft ADSIEdit (Active Directory Service Interfaces,
> > > available as a part of the Windows 2000 Support Tools) or any other
> > > LDAP tool to update the Description field.
> > >
> > > Regards,
> > >
> > > Joe
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > > Charles
> > > Sent: Friday, June 30, 2006 3:39 PM
> > > To: ciscovoip
> > > Subject: [cisco-voip] Removing users from the Corporate Directory
> > > (on
> > > thephones)
> > >
> > > So, I have CCM4.1(3), but I am seeing my admin accounts (JTAPI,
> > > RMJTAPI, appadmin, etc...) showing up in the directory on the
> phone...
> > >
> > > How do I remove them?
> > >
> > >
> > >
> > >
> > > Jonathan
> > > _______________________________________________
> > > cisco-voip mailing list
> > > cisco-voip at puck.nether.net
> > > https://puck.nether.net/mailman/listinfo/cisco-voip
> > >
> > > -----------------------------------------
> > > Disclaimer:
> > >
> > > This e-mail communication and any attachments may contain
> > > confidential
> >
> > > and privileged information and is for use by the designated
> > > addressee(s) named above only.  If you are not the intended
> > > addressee,
> >
> > > you are hereby notified that you have received this communication in
>
> > > error and that any use or reproduction of this email or its contents
>
> > > is strictly prohibited and may be unlawful.  If you have received
> > > this
> >
> > > communication in error, please notify us immediately by replying to
> > > this message and deleting it from your computer. Thank you.
> > >
> > >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> > -----------------------------------------
> > Disclaimer:
> >
> > This e-mail communication and any attachments may contain confidential
>
> > and privileged information and is for use by the designated
> > addressee(s) named above only.  If you are not the intended addressee,
>
> > you are hereby notified that you have received this communication in
> > error and that any use or reproduction of this email or its contents
> > is strictly prohibited and may be unlawful.  If you have received this
>
> > communication in error, please notify us immediately by replying to
> > this message and deleting it from your computer. Thank you.
> >
> >
>
> -----------------------------------------
> Disclaimer:
>
> This e-mail communication and any attachments may contain
> confidential and privileged information and is for use by the
> designated addressee(s) named above only.  If you are not the
> intended addressee, you are hereby notified that you have received
> this communication in error and that any use or reproduction of
> this email or its contents is strictly prohibited and may be
> unlawful.  If you have received this communication in error, please
> notify us immediately by replying to this message and deleting it
> from your computer. Thank you.
>
>

From jgrace at digitelusa.net  Fri Jun 30 17:39:26 2006
From: jgrace at digitelusa.net (James Grace)
Date: Fri, 30 Jun 2006 17:39:26 -0400
Subject: [cisco-voip] how to install the ip server  Lcd Clock
Message-ID: <200606301752515.SM01452@3729434770>

Ive seen the big lcd time clock on ip phones, and wanted to know what
service is it.  Anybody know 

 

James D. Grace 

CCNP CCNA MCSE MCDBA

Sr. System Engineer / Professional Svc.

Digitel Corporation

 

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From Joe.Pollere at us.didata.com  Fri Jun 30 17:52:29 2006
From: Joe.Pollere at us.didata.com (Joe Pollere (US))
Date: Fri, 30 Jun 2006 17:52:29 -0400
Subject: [cisco-voip] Removing users from the Corporate Directory
	(onthephones)
Message-ID: <C65033A817B5734392E55CB9DF44610405A0294F@USNAEXCH.na.didata.local>

Under AD Users and Computers > account properties > general tab.

-----Original Message-----
From: Jonathan Charles [mailto:jonvoip at gmail.com] 
Sent: Friday, June 30, 2006 5:30 PM
To: Joe Pollere (US)
Cc: Matt Slaga (US); ciscovoip
Subject: Re: [cisco-voip] Removing users from the Corporate Directory
(onthephones)

Description, where?



Jonathan

On 6/30/06, Joe Pollere (US) <Joe.Pollere at us.didata.com> wrote:
> That's weird...When I last did this I was using the AD plugin with CM
> 4.1.3 as I remember I didn't have to execute ldifde -i -f
hideuser.ldif.
> All I did was add CiscoPrivateUser to the Description Field and it hid
> those accounts from appearing in the Corporate Dir.
>
>
> Joe
>
>
> -----Original Message-----
> From: Matt Slaga (US)
> Sent: Friday, June 30, 2006 4:46 PM
> To: Jonathan Charles
> Cc: Joe Pollere (US); ciscovoip
> Subject: RE: [cisco-voip] Removing users from the Corporate Directory
> (onthephones)
>
> Link is below...  Forgot to add it to the last post.
>
>
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_te
> ch_note09186a00804d2087.shtml#topic1
>
>
> -----Original Message-----
> From: Jonathan Charles [mailto:jonvoip at gmail.com]
> Sent: Friday, June 30, 2006 4:41 PM
> To: Matt Slaga (US)
> Cc: Joe Pollere (US); ciscovoip
> Subject: Re: [cisco-voip] Removing users from the Corporate Directory
> (onthephones)
>
> That looks really unpleasant... I will try it...
>
>
>
> Jonathan
>
> On 6/30/06, Matt Slaga (US) <Matt.Slaga at us.didata.com> wrote:
> > They took it out in 4.1, but you used to be able to leave out the
> > first name to omit from the directory.
> >
> > Now it is a little more involved:
> >
> > 1.  Open Notepad on the publisher server.
> >
> > 2.  Cut and paste these four lines and save the file as a text file
on
>
> > the C: drive as hideuser.ldif.
> >
> > Note:  The spaces should appear as this output shows. Ensure that
> > there are no extra spaces in the LDIF file. Otherwise, the script
> > fails when you run it.
> >
> > dn: cn=[userid], ou=users, o=cisco.com changeType:modify
> > replace:Description Description:CiscoPrivateUser
> >
> > 3.  Set the [userid] to be the user you want to hide. For example:
> >
> > dn: cn=ctiuser, ou=users, o=cisco.com
> > changeType:modify
> > replace:Description
> > Description:CiscoPrivateUser
> >
> > 4.  Select Start > Run > cmd to open a command prompt on the
publisher
>
> > server and run this script in order to set the description field in
DC
> > Directory:
> >
> > ldapmodify -h <server name> -p 8404 -D "cn=Directory
> > Manager,o=cisco.com" -w <DCDAdmin Password> -c -f hideuser.ldif.txt
> >
> >
> > This output shows a successful run:
> >
> > C:\>ldapmodify -h CallManagerPub -p 8404 -D "cn=Directory Manager,
> > o=cisco.com" -w Cisco -c -f hideuser.ldif.txt modifying entry
> > cn=ctiuser, ou=users, o=cisco.com
> >
> >
> >
> > -----Original Message-----
> > From: cisco-voip-bounces at puck.nether.net
> > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > Charles
> > Sent: Friday, June 30, 2006 4:21 PM
> > To: Joe Pollere (US)
> > Cc: ciscovoip
> > Subject: Re: [cisco-voip] Removing users from the Corporate
Directory
> > (onthephones)
> >
> > Sounds good, but there I still want to be able to work with these
> > accounts. I just want them to not show up on the phone, directory,
> > corporate directory...
> >
> >
> >
> > Jonathan
> >
> > On 6/30/06, Joe Pollere (US) <Joe.Pollere at us.didata.com> wrote:
> > > Jonathan,
> > >
> > > Add CiscoPrivateUser to the description field.
> > >
> > > From page 14-17 of the CM 4.1 SRND:
> > >
> > > You can prevent system and service accounts from appearing in
Cisco
> > > CallManager Administration by adding the string CiscoPrivateUser
to
> > > the user's Description field. The CCM Administrator, CCMSysUser,
and
>
> > > IPMA SysUser accounts have this field set by default, but you can
> > > safely add the description to the CCM Directory Manager account as
> > > well. Use Microsoft ADSIEdit (Active Directory Service Interfaces,
> > > available as a part of the Windows 2000 Support Tools) or any
other
> > > LDAP tool to update the Description field.
> > >
> > > Regards,
> > >
> > > Joe
> > >
> > > -----Original Message-----
> > > From: cisco-voip-bounces at puck.nether.net
> > > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Jonathan
> > > Charles
> > > Sent: Friday, June 30, 2006 3:39 PM
> > > To: ciscovoip
> > > Subject: [cisco-voip] Removing users from the Corporate Directory
> > > (on
> > > thephones)
> > >
> > > So, I have CCM4.1(3), but I am seeing my admin accounts (JTAPI,
> > > RMJTAPI, appadmin, etc...) showing up in the directory on the
> phone...
> > >
> > > How do I remove them?
> > >
> > >
> > >
> > >
> > > Jonathan
> > > _______________________________________________
> > > 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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From voipermike at gmail.com  Fri Jun 30 20:29:16 2006
From: voipermike at gmail.com (Mike Mike)
Date: Fri, 30 Jun 2006 17:29:16 -0700
Subject: [cisco-voip] how to install the ip server Lcd Clock
In-Reply-To: <200606301752515.SM01452@3729434770>
References: <200606301752515.SM01452@3729434770>
Message-ID: <8bbfc1f40606301729t29519b00ic5b3f097a46b2706@mail.gmail.com>

It is just called clock.  Check out the SDK, it's part of that:

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_programming_usage_guide09186a008062ff25.html


On 6/30/06, James Grace <jgrace at digitelusa.net> wrote:
>
>   Ive seen the big lcd time clock on ip phones, and wanted to know what
> service is it.  Anybody know
>
>
>
> James D. Grace
>
> *CCNP CCNA MCSE MCDBA*
>
> Sr. System Engineer / Professional Svc.
>
> *Digitel Corporation*
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
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From bababooey at cox.net  Fri Jun 30 21:32:10 2006
From: bababooey at cox.net (Bob A. Bowie)
Date: Fri, 30 Jun 2006 21:32:10 -0400
Subject: [cisco-voip] Status of phones in ccm 4.1.3
References: <mailman.887.1151702873.46310.cisco-voip@puck.nether.net>
Message-ID: <000e01c69cae$2c70a460$0f0110ac@corporation2>

I am needing a way to actively monitor phone calls on the network.
Someone told me about web attendant but I cannot seem to find any 
documentation for it in 4.1.

Is this still an active product?  any other solutions to monitor phone usage 
real time?  preferable no cost but need any solutins.

thanx
bob



From Scott.Voll at wesd.org  Fri Jun 30 23:01:48 2006
From: Scott.Voll at wesd.org (Voll, Scott)
Date: Fri, 30 Jun 2006 20:01:48 -0700
Subject: [cisco-voip] Status of phones in ccm 4.1.3
References: <mailman.887.1151702873.46310.cisco-voip@puck.nether.net>
	<000e01c69cae$2c70a460$0f0110ac@corporation2>
Message-ID: <407055A92CECCB499C922A2D35FC19A605D75A21@Apollo.wesd.org>

it was called that back in 3.x.  it's call attendent console in 4.x.
 
Scott

________________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Bob A. Bowie
Sent: Fri 6/30/2006 6:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Status of phones in ccm 4.1.3



I am needing a way to actively monitor phone calls on the network.
Someone told me about web attendant but I cannot seem to find any
documentation for it in 4.1.

Is this still an active product?  any other solutions to monitor phone usage
real time?  preferable no cost but need any solutins.

thanx
bob


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


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