[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.
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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/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:
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Date Submitted: 2009-12-03 10:04 CST
Last Modified: 2009-12-04 08:11 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
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related to 0015716 [patch] chan_sip fails to destroy chann...
related to 0015627 [patch] Asterisk runs out of sockets
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(0114731) lmadsen (administrator) - 2009-12-04 08:11
https://issues.asterisk.org/view.php?id=16382#c114731
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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
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2009-12-04 08:11 lmadsen Note Added: 0114731
2009-12-04 08:11 lmadsen Status new => feedback
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