[asterisk-bugs] [Asterisk 0015149]: No audio on SIP RE-INVITE connecting with AllWorx PBX

[Asterisk Bug Tracker]

The following issue has been set as DUPLICATE OF issue 0018601. 
====================================================================== 
https://issues.asterisk.org/view.php?id=15149 
====================================================================== 
Reported By:                monettes
Assigned To:                qwell
====================================================================== 
Project:                    Asterisk
Issue ID:                   15149
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.0.9 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 suspended
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-05-19 00:51 CDT
Last Modified:              2011-01-12 08:51 CST
====================================================================== 
Summary:                    No audio on SIP RE-INVITE connecting with AllWorx
PBX
Description: 
We have a user with AllWorx registering a SIP DID with our Asterisk server.
When the submit 2x SIP RE-INVITEs, Asterisk doesn't use the new RTP Port of
the last SIP INVITE and creates a no-audio call.

The SIP DEBUG logs shows the right ports and report Asterisk decoding the
proper RTP port, but when you analyse the RTP packets, we see Asterisk
sending to the RTP port of the first SIP RE-INVITE, not the last one.

I attache the debug sip logs and the captured packets of a sample call.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
duplicate of        0018601 No RTP port update when SIP RE-INVITE i...
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-01-12 08:51 lmadsen        Relationship added       duplicate of 0018601
======================================================================