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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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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:              2010-02-04 03:24 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
----------------------------------------------------------------------
duplicate of        0016764 Sip Channels Colapse
related to          0015716 [patch] chan_sip fails to destroy chann...
related to          0015627 [patch] Asterisk runs out of sockets
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 (0117698) lftsy (reporter) - 2010-02-04 03:24
 https://issues.asterisk.org/view.php?id=16382#c117698 
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* Here are some steps to start reproducing manually the bug:
 -> Please refer to post above (0114758) lftsy (reporter) 2009-12-04
10:30

* When problem appears without my intervention, it appears on a peer that
has a bad xDSL connection so that it often changes it's IP/port of contact.


* The topology I use is a simple OpenSIPs 1.4.5 server in front of 2
Asterisk servers, the proxy does nothing except Keepalive and loadbalancing
on source IP. All AAA is done on Asterisk. BUT, I can reproduce the bug on
a direct IP connection to the Asterisk server!

* I have already put a 0245667911.pcap [^] (443,527 bytes) 2009-12-04
10:30 files above to see the problem. It corresponds to the manipulation to
reproduce the bug. You will see that 2 different Contact IP/ports are
PINGed from Asterisk. A sip prune realtime / or a reload of Asterisk server
doesn't stop the Flooding.

* My sip.conf is also at the beginning of the ticket

Next time it will happen, I will print you further output you have
requested here, hoping it will help us solving this bug.

Best Regards,
Marc Leurent 

Issue History 
Date Modified    Username       Field                    Change               
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2010-02-04 03:24 lftsy          Note Added: 0117698                          
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