[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:                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 14:20 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
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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
related to          0017643 [patch] dialplan reload deadlocks in as...
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 (0126114) zerohalo (reporter) - 2010-08-18 14:20
 https://issues.asterisk.org/view.php?id=16382#c126114 
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jpeeler, I can't say with confidence, as our logs have been well rotated by
now - but it does sound familiar. I'll see if I can find a sacrificial
production server and revert the patches - I'm never able to reproduce this
in a lab environment. In our case, it's never just a peer or two that goes
offline, but rather a bunch at once, as when a circuit goes down and the
peers jump over to a backup circuit. 'sip show channels' will show unending
numbers of OPTIONS dialogs, and as mentioned we essentially DoS the
connection. Also, typically, the backup asterisk server will be the one
that this manifests on. (connection A goes down, phones fail-over to
connection B, pop up on asterisk server B for a split second, then
immediately go back to server A, then server B DoSs connection A). 

Issue History 
Date Modified    Username       Field                    Change               
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2010-08-18 14:20 zerohalo       Note Added: 0126114                          
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