Summary:ASTERISK-14160: No audio on SIP RE-INVITE connecting with AllWorx PBX
Reporter:Stephan Monette (monettes)Labels:
Date Opened:2009-05-19 00:51:23Date Closed:2011-06-07 14:00:46
Versions:Frequency of
Environment:Attachments:( 0) debugsipreinvite.doc
( 1) valencia.pcap
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.
Comments:By: Joshua C. Colp (jcolp) 2009-05-21 08:36:57

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.

By: Jason Parker (jparker) 2009-08-20 15:56:36

Closing due to lack of response from reporter.

Please reopen when you can provide the information that was requested.