[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. 
====================================================================== 
http://bugs.digium.com/view.php?id=14374 
====================================================================== 
Reported By:                aragon
Assigned To:                putnopvut
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2009-01-30 13:40 CST
Last Modified:              2009-02-04 11:06 CST
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 (0099431) putnopvut (administrator) - 2009-02-04 11:06
 http://bugs.digium.com/view.php?id=14374#c99431 
---------------------------------------------------------------------- 
I've started a branch to fix this issue. It may be checked out from

http://svn.digium.com/svn/asterisk/team/mmichelson/chan_fixup

In it, I have already corrected the mixmonitor issue reported here. I now
am attempting to do some similar cleanup in app_chanspy. The branch is
based off the current tip of the Asterisk 1.4 branch. I expect that once I
have completed my work in the branch, I will post a review request on
reviewboard. I will provide a link to that once I have made the review
request. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-04 11:06 putnopvut      Note Added: 0099431                          
======================================================================