[asterisk-bugs] [Asterisk 0018627]: sip call fails to hang up - asterisk uses 99% resources

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18627 
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Reported By:                John Fawcett
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18627
Category:                   Channels/chan_sip/General
Reproducibility:            have not tried
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.8.2 
JIRA:                       SWP-3033 
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:             2011-01-16 11:23 CST
Last Modified:              2011-02-03 01:34 CST
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Summary:                    sip call fails to hang up - asterisk uses 99%
resources
Description: 
After upgrading to 1.8.2 (and also present in 1.8.1.1) asterisk is often
found to be consuming 99% of resources. 
This has been traced to calls which have been hangup by the remote party
but are still show as active (in BYE state) in asterisk.
It does seem to be random but happens so frequently that I can easily
reproduce this state by doing test calls until it happens. I can do any
debugging or tracing that is needed, just point me in the direction of what
is needed.
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 (0131429) John Fawcett (reporter) - 2011-02-03 01:34
 https://issues.asterisk.org/view.php?id=18627#c131429 
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I am now able to reproduce very easily. I have redone the capture of
debugging info shortly after starting asterisk so hopefully the info in the
logs is easier to read. I noticed that I had not attached the sip call
history last time so have also attached that. This time the call that
caused the issue was from extension 215 to queue 252. The problem happens
after hanging up the call.

One thing I notice is that there is always a failed lock when this
happens:

=== ---> Tried and failed to get Lock https://issues.asterisk.org/view.php?id=1
(chan_sip.c): MUTEX 3614
__sip_autodestruct p->owner 0xb2814ea0 (0)

Also the channel is left active even though the call has been closed (see
sip_show_channel_1.txt) attached. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-03 01:34 John Fawcett   Note Added: 0131429                          
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