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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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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 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
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:              2010-06-06 15:50 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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 (0123035) kobaz (reporter) - 2010-06-06 15:50
 https://issues.asterisk.org/view.php?id=15545#c123035 
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I'll test with 1.6.2 

Issue History 
Date Modified    Username       Field                    Change               
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2010-06-06 15:50 kobaz          Note Added: 0123035                          
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