[asterisk-bugs] [Asterisk 0016868]: One way audio after placing call on hold and resuming

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16868 
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Reported By:                jordankirby
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16868
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.2 
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:             2010-02-19 09:21 CST
Last Modified:              2010-02-19 09:22 CST
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Summary:                    One way audio after placing call on hold and
resuming
Description: 
This fault occurs on 1.6.1.11, 1.6.2.0, 1.6.2.2, 1.6.2.3-RC2, SVN-247894.

sip.conf: directrtpsetup=yes, nat=yes
Both extensions: canreinvite=yes, nat=yes
Phones are on the same LAN and behind NAT.

Server is in a separate location also behind NAT. All standard internal
and external calls work fine.

Problem:

Extension A calls extension B. Extension A puts the call on hold,
extension B gets played music as expected. When extension A resumes the
call extension B can't hear extension A. 

This seems to be because Asterisk sends the external IP address of
extension B to extension A in the SDP when the call is resumed.
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 (0118278) jordankirby (reporter) - 2010-02-19 09:22
 https://issues.asterisk.org/view.php?id=16868#c118278 
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The same problem occurs when the server is not behind NAT (phones still
NATd).
Attached traces are from a server not behind NAT. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-19 09:22 jordankirby    Note Added: 0118278                          
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