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

[Asterisk Bug Tracker]

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:                
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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:              2011-01-24 08:02 CST
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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.
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Relationships       ID      Summary
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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...
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 (0130923) Bernard Merindol (reporter) - 2011-01-24 08:02
 https://issues.asterisk.org/view.php?id=18468#c130923 
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I have tested on my labs version 1.8.3-rc1 on this isue with 3 aastra
phone. The bug is same.
When I use 3 aastra phones, A call B, B tansfert (attended) A to C, after
hangup B the sip part is hang on asterisk. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-24 08:02 Bernard MerindolNote Added: 0130923                          
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