[asterisk-bugs] [Asterisk 0018468]: SIP crash on transfer

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Feb 7 14:41:26 CST 2011


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18468 
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Reported By:                cchantep
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18468
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-12-13 12:52 CST
Last Modified:              2011-02-07 14:41 CST
====================================================================== 
Summary:                    SIP crash on transfer
Description: 
When trying to transfer a call from an Aastra SIP phone (6755i) --
A<-->B(Aastra)<-->C -- nonetheless call failed, but then Asterisk no longer
manager any SIP function.

Main Asterisk process still works, "sip show users" still displays users,
but phone are no longer registered, hard/soft phone cannot perform any new
registeration, no call can be done.

Even if I try to send a REGISTER UDP message through netcat from Asterisk
machine, I don't see anything in a Asterisk console (with sip debug
enabled).

It seems that a major part a SIP handling is then broken. Moreover I can't
reload sip module from Asterisk console. I need to kill it before
restarting service.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0018204 deadlock on 1.8.0-rc2 and crash on 1.8....
related to          0018403 [patch] Deadlock on SIP blind transfer ...
related to          0018455 Problem on TRANSFER using SNOM transfer...
related to          0018734 Combination dtmfmode=info, directmedia=...
====================================================================== 

---------------------------------------------------------------------- 
 (0131617) lmadsen (administrator) - 2011-02-07 14:41
 https://issues.asterisk.org/view.php?id=18468#c131617 
---------------------------------------------------------------------- 
num = <value optimized out>
        ch = <value optimized out>



That typically means asterisk wasn't compiled with DONT_OPTIMIZE in the
Compiler Flags section of menuselect. Please install Asterisk with that
option enabled and submit the backtrace as previous. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-07 14:41 lmadsen        Note Added: 0131617                          
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