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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16382 
====================================================================== 
Reported By:                lftsy
Assigned To:                jpeeler
====================================================================== 
Project:                    Asterisk
Issue ID:                   16382
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Target Version:             1.4.36
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:              
====================================================================== 
Date Submitted:             2009-12-03 10:04 CST
Last Modified:              2010-08-18 14:58 CDT
====================================================================== 
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
======================================================================
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
related to          0017643 [patch] dialplan reload deadlocks in as...
====================================================================== 

---------------------------------------------------------------------- 
 (0126117) lftsy (reporter) - 2010-08-18 14:58
 https://issues.asterisk.org/view.php?id=16382#c126117 
---------------------------------------------------------------------- 
Hello everybody,
We have spend some time with jpeeler (who I thank for its time) on one of
our lab server to try finding a short procedure to reproduce the bug, but
we failed to find a quick procedure without having to wait (which is going
to be useful to test patches)...

The best procedure I had, was:
* Doing note above n°114758 until pinging deprecated IP/ports starts
* And wait for uptime to increase
* Then resuming doing note n°114758 above until flooding appears!

We/I really have to take some time to find a quick and 100% procedure that
reproduces the bug!

Best Regards, 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-18 14:58 lftsy          Note Added: 0126117                          
======================================================================