[asterisk-bugs] [Asterisk 0015966]: Asterisk generates BYE at EXACTLY 900 seconds (15 min) and terminates call
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Sep 28 08:57:04 CDT 2009
The following issue requires your FEEDBACK.
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https://issues.asterisk.org/view.php?id=15966
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Reported By: riksta
Assigned To:
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Project: Asterisk
Issue ID: 15966
Category: Channels/chan_sip/General
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.6.1.5
JIRA:
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-09-25 13:57 CDT
Last Modified: 2009-09-28 08:57 CDT
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Summary: Asterisk generates BYE at EXACTLY 900 seconds (15
min) and terminates call
Description:
I have an incoming SIP call, which then dials out to another SIP trunk and
the calls are bridged via asterisk.
After exactly 900 seconds there is a BYE generated and the call completely
drops.
I have canreinvite=no specified in both the sip.conf general and for the
actual trunk stanza
http://office.encompassmedia.co.uk/dump.tgz has a full SIP/RTP media pcap
dump for both legs of the call which you can merge within wireshark.
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(0111422) lmadsen (administrator) - 2009-09-28 08:57
https://issues.asterisk.org/view.php?id=15966#c111422
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The problem with the wireshark trace alone is that we don't have the
information about what Asterisk is doing and what it is seeing. We do need
the SIP information per the bug guidelines when dealing with SIP issues.
If you have too much traffic, then you may need to reproduce in a lab
situation where you can control the traffic more easily. Otherwise there
might not be much the developers can do to move this issue forward, or it
might take much longer for a developer to resolve the issue.
Issue History
Date Modified Username Field Change
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2009-09-28 08:57 lmadsen Note Added: 0111422
2009-09-28 08:57 lmadsen Status new => feedback
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