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

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Dec 4 08:11:13 CST 2009


The following issue requires your FEEDBACK. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16382 
====================================================================== 
Reported By:                lftsy
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16382
Category:                   Channels/chan_sip/DatabaseSupport
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.27.1 
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:              2009-12-04 08:11 CST
====================================================================== 
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
----------------------------------------------------------------------
related to          0015716 [patch] chan_sip fails to destroy chann...
related to          0015627 [patch] Asterisk runs out of sockets
====================================================================== 

---------------------------------------------------------------------- 
 (0114731) lmadsen (administrator) - 2009-12-04 08:11
 https://issues.asterisk.org/view.php?id=16382#c114731 
---------------------------------------------------------------------- 
Could you also provide a console trace while this is happening, along with
SIP debug and SIP history enabled? Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-12-04 08:11 lmadsen        Note Added: 0114731                          
2009-12-04 08:11 lmadsen        Status                   new => feedback     
======================================================================




More information about the asterisk-bugs mailing list