[asterisk-bugs] [Asterisk 0016382]: SIP OPTIONS qualify message forever
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Feb 4 03:24:48 CST 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:
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Project: Asterisk
Issue ID: 16382
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
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: 2010-02-04 03:24 CST
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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
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(0117698) lftsy (reporter) - 2010-02-04 03:24
https://issues.asterisk.org/view.php?id=16382#c117698
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* Here are some steps to start reproducing manually the bug:
-> Please refer to post above (0114758) lftsy (reporter) 2009-12-04
10:30
* When problem appears without my intervention, it appears on a peer that
has a bad xDSL connection so that it often changes it's IP/port of contact.
* The topology I use is a simple OpenSIPs 1.4.5 server in front of 2
Asterisk servers, the proxy does nothing except Keepalive and loadbalancing
on source IP. All AAA is done on Asterisk. BUT, I can reproduce the bug on
a direct IP connection to the Asterisk server!
* I have already put a 0245667911.pcap [^] (443,527 bytes) 2009-12-04
10:30 files above to see the problem. It corresponds to the manipulation to
reproduce the bug. You will see that 2 different Contact IP/ports are
PINGed from Asterisk. A sip prune realtime / or a reload of Asterisk server
doesn't stop the Flooding.
* My sip.conf is also at the beginning of the ticket
Next time it will happen, I will print you further output you have
requested here, hoping it will help us solving this bug.
Best Regards,
Marc Leurent
Issue History
Date Modified Username Field Change
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2010-02-04 03:24 lftsy Note Added: 0117698
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