[asterisk-bugs] [Asterisk 0015545]: Not passing audio on a sip call in and out on the same peer

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=15545 
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Reported By:                kobaz
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15545
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0.10 
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:             2009-07-21 10:07 CDT
Last Modified:              2009-09-18 07:48 CDT
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Summary:                    Not passing audio on a sip call in and out on the
same peer
Description: 
This worked in 1.4.x, so I'm assuming this is a bug.

Call comes in from an itsp via sip.  We then proceed to dial out that same
itsp (ie: call forwarding).  The remote side answers the call, but no audio
is passed.

This happens on 1.6.0.10, but it's not available as a product version.

rtp packets are zero during the call.  The asterisk box is also not behind
nat.
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 (0110934) lmadsen (administrator) - 2009-09-18 07:48
 https://issues.asterisk.org/view.php?id=15545#c110934 
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Are you still getting this issue with 1.6.0.16-rc1?

If so, could you provide the same output, but with the 'sip history' also
enabled, along with any console output, and 'core set debug 10' enabled
after turning it on in logger.conf (console => notice,warning,error,debug),
followed by a 'logger reload'

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2009-09-18 07:48 lmadsen        Note Added: 0110934                          
2009-09-18 07:48 lmadsen        Status                   new => feedback     
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