[asterisk-bugs] [Asterisk 0012143]: Asterisk stops responding after quick hangup between asterisk boxes

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12143 
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Reported By:                kactus
Assigned To:                
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Project:                    Asterisk
Issue ID:                   12143
Category:                   Channels/chan_sip/General
Reproducibility:            sometimes
Severity:                   block
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.0 
SVN Revision (number only!): 104866 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             03-05-2008 04:11 CST
Last Modified:              03-11-2008 22:14 CDT
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Summary:                    Asterisk stops responding after quick hangup between
asterisk boxes
Description: 
Hello I've been playing around with a two server dynamic configuration and
have been able to repeatedly lockup asterisk with several 1.6 builds beta 4
and svn 104866 (latest at time of testing) so that it does not even accept
any further commands at the console or from sip devices. The only way to
make asterisk respond after this is by killing the process or restarting
the service but even this throws an error before being restarted by
safe_asterisk.

What we are doing is probably unconventional but either way it is unlikely
that a lockup of asterisk should result. I will attach backtrace
information for you as well, but please let me know if you need more. I had
collected more but a machine restart over the weekend meant I had to
recreate even the information I am posting now.

We have two servers running asterisk 1.6 (voip01 & voip02) and two snom
360 handsets (0040 and  0041) with version 6+ firmware each registering to
a different server, natting between the handsets and the servers while the
servers sit on a local lan.

0040 ->nat-> SIP -> voip01 -> IAX -> voip02 -> SIP -> nat -> 0041

Normal calls work from 0040 to 0041 when dialing 141 and voice works both
ways without issue. 

However if you hang up right as the 0041 starts ringing it will continue
to ring, when you pick up the line will be dead and from that point onwards
asterisk no longer accepts sip requests, calls or even commands such as
iax2 show channels until it is killed. Logs on the snom phones after
reregister show multiple registration requests but no answers and there are
no more screen output to the cli after 'extension is ringing'. You cannot
even run shutdown now from the cli and have to kill the process. For the
back trace we have done a kill -11 to force a segmentation fault to
hopefully give you more information.

This only happens when storing the sip information in the database, we can
not replicate this when only using flat files or when running extensions
from the database but sip from flat files. This might be due to a timing
delay in hanging up on time however as if you hang up too soon or too late
the phone behaves normally with out issue. In our test setup if you hang up
on the first indial ring on the second phone it causes this issue 99% of
the time (about 1 second after initiating the call).

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---------------------------------------------------------------------- 
 kactus - 03-11-08 22:14  
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Hello sorry it took me so long to get this info to you I was not able to
get to this until now.

Hopefully this is the information that you are after.

Please let me know if you need any more information or if you need me to
test a different svn, I have not updated since 104866 to prevent adding
additional layers of complexity.

Cheers 

Issue History 
Date Modified   Username       Field                    Change               
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03-11-08 22:14  kactus         Note Added: 0083792                          
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