[asterisk-bugs] [Asterisk 0010571]: SIP hairpin invokes Local within app_dial to produce a crash.

noreply at bugs.digium.com noreply at bugs.digium.com
Wed Sep 19 11:03:27 CDT 2007


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10571 
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Reported By:                dtyoo
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10571
Category:                   Applications/app_dial
Reproducibility:            unable to reproduce
Severity:                   crash
Priority:                   normal
Status:                     new
Asterisk Version:            1.4.9  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             08-27-2007 08:43 CDT
Last Modified:              09-19-2007 11:03 CDT
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Summary:                    SIP hairpin invokes Local within app_dial to produce
a crash.
Description: 
We are getting crashes in app_voicemail on a fairly regular basis.  I'm
still working on steps to re-produce, but I thought I would post the
backtraces here in case someone could glean anything from them.  I will
update if I can figure out the steps to re-produce.
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---------------------------------------------------------------------- 
 dtyoo - 09-19-07 11:03  
---------------------------------------------------------------------- 
Corydon76, qwell-

We did get another crash that looks very similar but is not exactly the
same to the ones we originally reported here.  This is on one of the
upgraded servers that is running 1.4.11 with Corydon76's local channel
de-optimization patch.  This crash points to chan_sip, but it definitely
seems related.  This is the last line of the messages file:

[Sep 19 09:17:29] ERROR[15843]
/usr/src/asterisk-test/1.4.11/asterisk-1.4.11/include/asterisk/lock.h:
chan_sip.c line 15175 (sipsock
_read): mutex '&p->owner->lock' freed more times than we've locked!

What was happening at the time of the crash is a call being delivered to a
large ring group with 32 sip peers being dialed with
Dial(SIP/PEER1&SIP/PEER2...).

I've also uploaded a new bt/btfull from this core file. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
09-19-07 11:03  dtyoo          Note Added: 0070798                          
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