[asterisk-bugs] [Asterisk 0017526]: Asterisk seg faults after incoming OOH323 call fails due to congesstion
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Jun 20 13:10:54 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=17526
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Reported By: vmikhelson
Assigned To: may213
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Project: Asterisk
Issue ID: 17526
Category: Addons/chan_ooh323
Reproducibility: always
Severity: crash
Priority: normal
Status: assigned
Asterisk Version: 1.6.2.8
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: 2010-06-17 21:48 CDT
Last Modified: 2010-06-20 13:10 CDT
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Summary: Asterisk seg faults after incoming OOH323 call fails
due to congesstion
Description:
A call is placed to a SIP extension with no VoiceMail. If the call goes
through there is no problem.
If the call gets congested Asterisk experiences the following in about 10
seconds after the call ends:
[root at pbx ~]# /usr/sbin/safe_asterisk: line 145: 24682 Segmentation fault
(core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
The same SIP extension can be called from a SIP or a DAHDI extension with
no adverse effects.
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(0123643) may213 (manager) - 2010-06-20 13:10
https://issues.asterisk.org/view.php?id=17526#c123643
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Vladimir,
Are you try latest svn version of 1.6.2 addons? If no i think you can
update to this and will have no problem.
Problem is like to 0017227 and there is patch to solve it.
Issue History
Date Modified Username Field Change
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2010-06-20 13:10 may213 Note Added: 0123643
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