[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:              2011-02-02 13:40 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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 (0131410) John Fawcett (reporter) - 2011-02-02 13:40
 https://issues.asterisk.org/view.php?id=18627#c131410 
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I haven't specifically set rtptimeout or rtpholdtimeout.

at asterisk cli, doing "sip show settings" I see

..
Global Signalling Settings:
---------------------------
..
 RTP Timeout:            0 (Disabled)
 RTP Hold Timeout:       0 (Disabled)
.. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-02 13:40 John Fawcett   Note Added: 0131410                          
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