[asterisk-bugs] [Asterisk 0014421]: Issue with Initial call setup when Asterisk 1.6.1-rc1 integrated with Microsoft OCS 2007

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14421 
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Reported By:                TheOldSaint
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14421
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1-beta1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-02-06 05:03 CST
Last Modified:              2009-02-09 01:15 CST
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Summary:                    Issue with Initial call setup when Asterisk
1.6.1-rc1 integrated with Microsoft OCS 2007
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. 
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Relationships       ID      Summary
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has duplicate       0014420 Issue with Asterisk 1.6.1-rc1 with Micr...
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 (0099679) TheOldSaint (reporter) - 2009-02-09 01:15
 http://bugs.digium.com/view.php?id=14421#c99679 
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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 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-09 01:15 TheOldSaint    Note Added: 0099679                          
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