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

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Nov 10 17:04:10 CST 2010


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:             1.4.38
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:              2010-11-10 17:04 CST
====================================================================== 
Summary:                    [patch] 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...
related to          0017779 [patch] tcptls.c:350 Unable to connect ...
====================================================================== 

---------------------------------------------------------------------- 
 (0128761) zerohalo (reporter) - 2010-11-10 17:04
 https://issues.asterisk.org/view.php?id=16382#c128761 
---------------------------------------------------------------------- 
jpeeler - I've spent a lot of time trying to reproduce this (which was hard
to reproduce in the first place) and I'm happy to report that I cannot
reproduce the OPTIONS flood w/ your v3 patch. There are some curious issues
with Polycom mwi resubscriptions and rtautoclear=no (another bug which I'll
keep working on figuring out - seem related to various other bugs and
similar connectivity issues that caused the floods in the first place), but
other than that, the rtautoclear setting seems to have no effect in either
direction when it comes to helping produce the issue. Thanks for all your
time and help on this! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-10 17:04 zerohalo       Note Added: 0128761                          
======================================================================




More information about the asterisk-bugs mailing list