[asterisk-bugs] [Asterisk 0011233]: SIP channel crashes on new call

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11233 
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Reported By:                jtodd
Assigned To:                murf
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Project:                    Asterisk
Issue ID:                   11233
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 89242 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             11-13-2007 11:07 CST
Last Modified:              11-13-2007 12:11 CST
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Summary:                    SIP channel crashes on new call
Description: 

Attempting a call into a SIP channel (with an invalid extension) causes
Asterisk to crash.  Not sure if this is something that is specific to SIP,
but I have no other channel types to test with at the moment.
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---------------------------------------------------------------------- 
 murf - 11-13-07 12:11  
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According to the bt, It was in the act of building a trie when it died.

Could you please attach the "from-jt" context's contents to this bug, so I
can reproduce the sequence of events that lead to this crash? You can
warp/remove any proprietary info; but I need the extension names/patterns
to be preserved intact.
It looks like it was working on _XXXXXXXXXXXXX*. when it died. 

Issue History 
Date Modified   Username       Field                    Change               
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11-13-07 12:11  murf           Note Added: 0073580                          
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