[asterisk-bugs] [Asterisk 0014374]: Revision 172517 segfault after using A *2 transfer to B and B dial *2

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14374 
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Reported By:                aragon
Assigned To:                otherwiseguy
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Project:                    Asterisk
Issue ID:                   14374
Category:                   Applications/app_transfer
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.23 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!): 172517 
Request Review:              
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Date Submitted:             2009-01-30 13:40 CST
Last Modified:              2009-02-02 13:25 CST
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Summary:                    Revision 172517 segfault after using A *2 transfer
to B and B dial *2
Description: 
SIP environment
Extension 6011 dials 6010
6010 dials *26002 and ends call to complete transfer
6002 answers and dials *2 but does receive audio for transfer prompt
Asterisk segfaults after dialing *2

Backtrace attached
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 (0099243) otherwiseguy (administrator) - 2009-02-02 13:25
 http://bugs.digium.com/view.php?id=14374#c99243 
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I see the issue, since there is a masquerade happening with the transfer,
it looks like the channel is hung up and is freed.  Unfortunately,
mixmonitor_thread() doesn't know that the channel has disappeared and tries
to access it.  The good news is that my fix really doesn't have anything to
do with it (other than letting things get far enough in the process to
expose the bug), the bad news it doesn't look particularly easy to fix. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-02 13:25 otherwiseguy   Note Added: 0099243                          
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