[asterisk-bugs] [Asterisk 0016382]: SIP OPTIONS qualify message forever
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Dec 4 02:58:38 CST 2009
A NOTE has been added to this issue.
======================================================================
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: new
Asterisk Version: 1.4.27.1
JIRA:
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 02:58 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
======================================================================
----------------------------------------------------------------------
(0114722) lftsy (reporter) - 2009-12-04 02:58
https://issues.asterisk.org/view.php?id=16382#c114722
----------------------------------------------------------------------
Hello c0rnoTa, I have been monitoring the problem for a while but I did not
notice trouble with INVITE retransmission. Indeed, when I call a peer that
has just gone away, there will be retransmission because of no answer, and
after a 408 Timeout From my OpenSIPs Proxy.
Furthermore, I didn't notice your error message related to this
problem...
You might need to post your message in a new bug report
Have a nice day!
Issue History
Date Modified Username Field Change
======================================================================
2009-12-04 02:58 lftsy Note Added: 0114722
======================================================================
More information about the asterisk-bugs
mailing list