[asterisk-bugs] [Asterisk 0018204]: deadlock on 1.8.0-rc2 and crash on 1.8.0 with multipule sip channels
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Nov 2 09:20:52 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18204
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Reported By: knkcn
Assigned To:
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Project: Asterisk
Issue ID: 18204
Category: Core/Channels
Reproducibility: sometimes
Severity: crash
Priority: normal
Status: new
Asterisk Version: 1.8.0
JIRA:
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: 2010-10-25 02:30 CDT
Last Modified: 2010-11-02 09:20 CDT
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Summary: deadlock on 1.8.0-rc2 and crash on 1.8.0 with
multipule sip channels
Description:
Asterisk 1.8.0-rc2 as a call center server run well with 1 or 2 concurrent
lines. We did about 50 concurrent lines lab test with sipp and it run well
too. But under a more than 20 lines field trial, the Asterisk deadlocked.
After deadlocked I found many rtp channels and udp5060 had a large number
on Recv-Q. At the same time "core show channels" showed nothing and many
cli commands hang random. I collected some information using "gdb
/usr/sbin/asterisk <pid>" but I have not enough time to debug what locked
it before restarted it.
After that I upgraded the server to 1.8.0 with "DONT_OPTIMIZE" and
"DEBUG_THREADS", then I did the test again. But this time it crashed when
10 phones were calling in.
BTW, after the "DEBUG_THREADS" is enabled, every time the "bridge"
application is executed the following error will be reported.
[Oct 25 15:03:04] ERROR[25661] lock.c: features.c line 5551 (bridge_exec):
mutex 'current_dest_chan' freed more times than we've loc
ked!
[Oct 25 15:03:04] ERROR[25661] lock.c: features.c line 5551 (bridge_exec):
Error releasing mutex: Operation not permitted
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(0128552) lmadsen (administrator) - 2010-11-02 09:20
https://issues.asterisk.org/view.php?id=18204#c128552
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knkcn: I understand you're having issues, but the issue tracker is a best
effort system and you will not be able to receive immediate attention until
the issue has been assigned to a developer.
If you have an immediate need to have this resolved I'd suggest hiring a
consultant if you are unable to resolve the issue. If you have some
development skills you may be able to get some help on the asterisk-dev
mailing list (or #asterisk-dev IRC channel) if you're having an issue with
a particular section of the code.
Beyond that, you can use the asterisk-users mailing list (or #asterisk IRC
channel), or you'll have to be patient until this issue has been assigned
to a developer.
Issue History
Date Modified Username Field Change
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2010-11-02 09:20 lmadsen Note Added: 0128552
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