[asterisk-bugs] [Asterisk 0014539]: Missing one way audio on one phone after the bridge

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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http://bugs.digium.com/view.php?id=14539 
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Reported By:                jeremey_g
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14539
Category:                   Core/RTP
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     acknowledged
Target Version:             1.6.0.10
Asterisk Version:           1.6.0.6-rc1 
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:             2009-02-24 09:16 CST
Last Modified:              2009-05-07 08:24 CDT
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Summary:                    Missing one way audio on one phone after the bridge
Description: 
Asterisk should not drop the new rtp stream with time stamp zero after
bridging two sip peers.

Details:
SIP phone A and B are registered to *.

A SIP call comes from a cisco pstn gateway to * on extension 2000, which
bridges  to Phone B.

After bridging, there is no audio in Phone B.

The party that sent the call through the gateway, can both receive and
send audio.