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

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Aug 18 13:52:34 CDT 2010


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:                jpeeler
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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:              
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Date Submitted:             2009-12-03 10:04 CST
Last Modified:              2010-08-18 13:52 CDT
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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
======================================================================
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...
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---------------------------------------------------------------------- 
 (0126112) zerohalo (reporter) - 2010-08-18 13:52
 https://issues.asterisk.org/view.php?id=16382#c126112 
---------------------------------------------------------------------- 
I wish I could add anything additional here, but since we've patched out
the read/write of peer data from ast_db in chan_sip, which granted, is a
very big hammer approach, we haven't run into this again. It definitely
seems like the problem is related to mysql connectivity blipping (on the
local machine), but it's been very hard to reproduce reliably. It
definitely takes decent load and patience for it to happen, generally
pruning peers and killing connections will increase the chance of it
happening, but never reliably happens when we want it to. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-18 13:52 zerohalo       Note Added: 0126112                          
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