[asterisk-bugs] [Asterisk 0014282]: conference calling crashes Asterisk

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14282 
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Reported By:                cheesegrits
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   14282
Category:                   Applications/app_meetme
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.0.3 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 169479 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-01-19 22:50 CST
Last Modified:              2009-01-22 10:54 CST
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Summary:                    conference calling crashes Asterisk
Description: 
I have a reproducible situation which crashes Asterisk. Conference calls
can be started, but as soon as anyone leaves the conference, or anything
else happens to initiate a soft hangup, it segmentation faults.

It's happening on both my test servers, running the following config:

Centos 5.2 (ISO's from centos.org, then fully yum updated)
Asterisk 1.6.0.3 AND latest SVN code (rev# 169479)
FreePBX 2.5.1
DAHDI 2.1.0.3 (installed from dahdi-linux-complete)

Everything is pretty much vanilla. A couple of extensions on each box, and
an IAX2 trunk between the two. One VoIP line on one of the boxes as the
main inbound/outbound route.

No telephony hardware, so it's using dahdi_dummy for timing. I've
confirmed with dahdi_test that this seems to be functioning correctly, and
lsmod and modprobe both look good.

I've tried doing a soft hangup from the CLI on regular calls, but that
works fine, no errors. It only seems to be conference calls which have this
issue.

This happens with any type of call in the conference, i.e. local
extension, coming in thru the IAX2 trunk, or inbound routes.

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---------------------------------------------------------------------- 
 (0098456) svnbot (reporter) - 2009-01-22 10:54
 http://bugs.digium.com/view.php?id=14282#c98456 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 170150

_U  branches/1.6.1/
U   branches/1.6.1/apps/app_meetme.c

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r170150 | file | 2009-01-22 10:54:22 -0600 (Thu, 22 Jan 2009) | 18 lines

Merged revisions 170148 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | 11 lines
  
  Merged revisions 170147 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4
lines
    
    If we are unable to request a DAHDI pseudo channel and we are using
the user introduction without review option make sure it gets unset so
other code does not blindly assume a DAHDI pseudo channel exists.
    (closes issue http://bugs.digium.com/view.php?id=14282)
    Reported by: cheesegrits
  ........
................

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http://svn.digium.com/view/asterisk?view=rev&revision=170150 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-22 10:54 svnbot         Checkin                                      
2009-01-22 10:54 svnbot         Note Added: 0098456                          
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