[asterisk-bugs] [Asterisk 0016382]: SIP OPTIONS qualify message forever

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16382 
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Reported By:                lftsy
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16382
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                       SWP-478 
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:             2009-12-03 10:04 CST
Last Modified:              2010-02-03 10:27 CST
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Summary:                    SIP OPTIONS qualify message forever
Description: 
Hello, I have a trouble with different Asterisk versions (1.4.26, 1.4.27,
1.4.27.1). When I use the steps below, Asterisk starts to send SIP OPTIONS
to the previous IP/port used by a SIP realtime peer (that has been pruned)
and will keep trying to send SIP OPTIONS pings forever, event if the peer
is connected with a new IP/port address.

I have just checked with my old Asterisk 1.2.27 with the same sip.conf and
I do not have the problem, the SIP OPTIONS stops once register timer has
expired.

During my experience to reproduce the bug, I have been able to have 10
IP/port currently pinged by the Asterisk server for one single peer.
And the only way to stop it is to restart Asterisk...

Thank you for your attention!
Marc Leurent
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Relationships       ID      Summary
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related to          0015716 [patch] chan_sip fails to destroy chann...
related to          0015627 [patch] Asterisk runs out of sockets
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 (0117616) lmadsen (administrator) - 2010-02-03 10:27
 https://issues.asterisk.org/view.php?id=16382#c117616 
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Actually, I'll go ahead and request some addtional information:

* console output demonstrating this issue, with sip history and debug
level logging enabled (probably best done from logger.conf by enabling
'full' and then doing a 'logger reload' at the CLI, followed by 'core set
verbose 10' and 'core set debug 10')

* SIP console output from Asterisk with SIP history enabled from sip.conf

* sip.conf demonstrating this issue, and what topology is being used, and
how best to reproduce this.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-03 10:27 lmadsen        Note Added: 0117616                          
2010-02-03 10:27 lmadsen        Status                   acknowledged =>
feedback
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