[asterisk-bugs] [Asterisk 0015149]: No audio on SIP RE-INVITE connecting with AllWorx PBX
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Aug 20 15:56:37 CDT 2009
The following issue has been UPDATED.
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https://issues.asterisk.org/view.php?id=15149
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Reported By: monettes
Assigned To: qwell
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Project: Asterisk
Issue ID: 15149
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: major
Priority: normal
Status: closed
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:
Resolution: suspended
Fixed in Version:
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Date Submitted: 2009-05-19 00:51 CDT
Last Modified: 2009-08-20 15:56 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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(0109404) qwell (administrator) - 2009-08-20 15:56
https://issues.asterisk.org/view.php?id=15149#c109404
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Closing due to lack of response from reporter.
Please reopen when you can provide the information that was requested.
Issue History
Date Modified Username Field Change
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2009-08-20 15:56 qwell Note Added: 0109404
2009-08-20 15:56 qwell Status feedback => resolved
2009-08-20 15:56 qwell Resolution open => suspended
2009-08-20 15:56 qwell Assigned To => qwell
2009-08-20 15:56 qwell Status resolved => closed
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