[asterisk-bugs] [Asterisk 0016115]: Lockup in chan_sip
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Dec 20 16:36:08 UTC 2010
The following issue has been CLOSED
======================================================================
https://issues.asterisk.org/view.php?id=16115
======================================================================
Reported By: fmarie
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 16115
Category: Channels/chan_sip/General
Reproducibility: random
Severity: major
Priority: normal
Status: closed
Asterisk Version: 1.6.2.0-rc3
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: open
Fixed in Version:
======================================================================
Date Submitted: 2009-10-22 21:20 CDT
Last Modified: 2010-12-20 10:36 CST
======================================================================
Summary: Lockup in chan_sip
Description:
Hello.
We are using Asterisk 1.6.2.0-rc3 on openSUSE 11.1 mainly for call
recording, fax and IVR. Asterisk is connected to a softswitch as a SIP
peer, the softswitch is connected to the PSTN via SS7 and to MGCP and SIP
phones.
Asterisk is locking up randomly. The console works, but no more SIP
traffic is accepted. This started when we switched (because of fax
problems) from 1.6.1.6 to 1.6.2.0-rc2. At first, it happened about once a
week but has progressed to once-twice per day. No improvement after
upgrading to 1.6.2.0-rc3. Restarting Asterisk via /etc/init.d/asterisk
script helps (until the next lockup).
The last message in the log (set to verbose) before the lockup is
something like
chan_sip.c: Maximum retries exceeded on transmission
23150-AQ-002ecd38-506161c24 at localdomain.com for seqno 2554278 (Critical
Response) -- See doc/sip-retransmit.txt.
However, we have also seen this message without a lockup following.
I attached the output of "core show locks". I will also attach sip debug
output as soon as I get it. What else should I do to help debug this
problem?
Also, until a better solution, is there a way to monitor Asterisk for this
kind of lockup and restart it?
Thanks,
Marie Fischer
======================================================================
Issue History
Date Modified Username Field Change
======================================================================
2010-12-20 10:36 lmadsen Status acknowledged => closed
======================================================================
More information about the asterisk-bugs
mailing list