[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:                putnopvut
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Project:                    Asterisk
Issue ID:                   14374
Category:                   Applications/app_transfer
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     ready for testing
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:              2009-02-05 08:42 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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 (0099494) aragon (reporter) - 2009-02-05 08:42
 http://bugs.digium.com/view.php?id=14374#c99494 
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While I was looking at my own posted CLI capture I noticed I failed to
remove a hotdesk script for my tests.

I cleaned up my mess and re-tested both patches.

Both patches are good.
*1 and *2 and all SIP transfers are working as expected without segfaults
and can be re-used with mixmonitor in dial plan.

This bug can be closed out as a result of both patches.

Thanks, and sorry about my goof. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-02-05 08:42 aragon         Note Added: 0099494                          
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