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

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=15149 
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Reported By:                monettes
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15149
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0.9 
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:             2009-05-19 00:51 CDT
Last Modified:              2009-05-21 08:36 CDT
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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.
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 (0105186) file (administrator) - 2009-05-21 08:36
 https://issues.asterisk.org/view.php?id=15149#c105186 
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Please attach a complete console log with debug set to go to console in
logger.conf and "core set debug 4" executed in the CLI. As well a new sip
debug with also rtp debug is needed. Please also make sure you are using
the latest version. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-05-21 08:36 file           Note Added: 0105186                          
2009-05-21 08:36 file           Status                   new => feedback     
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