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Summary:ASTERISK-13529: Issue with Initial call setup when Asterisk 1.6.1-rc1 integrated with Microsoft OCS 2007
Reporter:TheOldSaint (theoldsaint)Labels:
Date Opened:2009-02-06 05:03:45.000-0600Date Closed:2011-06-07 14:08:07
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 14421-3.diff
( 1) OCtoSNOM_Not_Working_Traces.log
( 2) OCtoSNOM_Working_logs.txt
( 3) SNOMtoOC.log
Description:Hi,

I have installed Asterisk 1.6.1-rc1 version. And Configured Asterisk as the Gateway at the Mediation Server (OCS).

And I have tried the below scenario - call from SNOM to OC as mentioned in the below topology.


SNOM (On TLS) --> Asterisk 1.6.1-rc1 --> MedSrv --> OCS --> OC

I have created SIP_TRUNK configuration at the asterisk to forward calls to the Mediation Server of OCS 2007.

When SNOM dialed OC then Asterisk is forwarding this Initial Dialog request i.e INVITE to the Mediation Server. But Mediation Server is responding with 400 bad/Invalid contact information for the INVITE message to the Asterisk. Then Asterisk is sending 603 declined response to the SNOM.

But what ever the contact header,which SNOM sent in the INVITE message has got valid syntax.

Do I need to change anything at the Asterisk to make the call successful.

Could you please help me in resolving these issue.


OC - Microsoft Office Communicator
OCS 2007 - Microsoft Office Communication Server 2007
MedSrv - Mediation Server

Thanks in advance.
Comments:By: Paul Belanger (pabelanger) 2009-02-06 09:31:47.000-0600

Please refer to http://www.asterisk.org/support for user support.

If you believe this is a bug, refer to http://www.asterisk.org/developers/bug-guidelines

By: Tilghman Lesher (tilghman) 2009-02-06 10:21:34.000-0600

You'll need to generate the output of "sip set debug" for this set of calls and upload the ENTIRE output as a file to this issue for us to pursue this.

By: TheOldSaint (theoldsaint) 2009-02-09 01:15:40.000-0600

I have uploaded three log files

1. OCtoSNOM_Working_logs.txt - This logs belongs to the below topology and also when BYE initiated from OC itself to terminate the call. This is working fine.

OC --> OCS 2007 --> MedSrv --> Asterisk 1.6.1-rc1 --> SNOM (On TLS)

2. OCtoSNOM_Not_Working_Traces.log - This logs belongs to the below topology. Here I am facing an issue with Asterisk 1.6.1-rc1. When BYE initiated from SNOM to terminate the call, then something is happening (I guess, TLS connection getting closed) and BYE didn't reach at the Asterisk. Asterisk is under impression that the call leg between the Asterisk & SNOM is still active. Once the OC initiates the BYE then Asteris is forwarding BYE to the SNOM, for which SNOM is responding with 481 call leg doesn't exist.

OC --> OCS 2007 --> MedSrv --> Asterisk 1.6.1-rc1 --> SNOM (On TLS)

3. SNOMtoOC.log - This log belongs to the below topology. I have created SIP_TRUNK configuration at the asterisk to forward calls to the Mediation Server of OCS 2007.When SNOM dialed OC then Asterisk is forwarding this Initial Dialog request i.e INVITE to the Mediation Server. But Mediation Server is responding with 400 bad/Invalid contact information for the INVITE message to the Asterisk. Then Asterisk is sending 603 declined response to the SNOM.But what ever the contact header,which SNOM sent in the INVITE message has got valid syntax.

SNOM (On TLS) --> Asterisk 1.6.1-rc1 --> MedSrv --> OCS --> OC


It looks like "Point 2 & 3" got some issue with Asterisk 1.6.1-rc1. Could you please help me in resolving these issues.

Thanks

By: Joshua C. Colp (jcolp) 2009-03-10 15:51:19

I have attached a patch which should hopefully fix the 3rd issue. Please give it a try.

By: Leif Madsen (lmadsen) 2009-05-22 14:32:35

Pinging original reporter for testing feedback.

By: Leif Madsen (lmadsen) 2009-05-26 09:54:40

I'm closing this issue as there has been no response from the reporter. Due to the relative obscurity of the Microsoft OCS server, this issue could sit here forever without input from the original reporter.

Please reopen when you are able to provide the necessary information and testing. Thanks!