[asterisk-bugs] [Asterisk 0010665]: [patch] SIP Session-Timers Support in Asterisk
noreply at bugs.digium.com
noreply at bugs.digium.com
Thu Oct 11 00:53:12 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=10665
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Reported By: rjain
Assigned To:
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Project: Asterisk
Issue ID: 10665
Category: Channels/chan_sip/General
Reproducibility: always
Severity: feature
Priority: normal
Status: ready for testing
Asterisk Version: 1.4.11
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 81013
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 09-07-2007 03:43 CDT
Last Modified: 10-11-2007 00:53 CDT
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Summary: [patch] SIP Session-Timers Support in Asterisk
Description:
The Asterisk SIP stack currently does not support SIP Session-Timers (RFC
4028). This leads to defunct SIP sessions in Asterisk when calls do not
clear through normal signaling procedures due to network or end-point
failures.
John Todd recently discussed this concept on asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2007-July/028574.html
John Todd, JR Richardson and Kevin Fleming have expressed interest in
seeing this feature supported in Asterisk.
A software design document for this feature and code changes (unified diff
of chan_sip.c) are attached to this report. Digium has my code submission
agreement on file.
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rjain - 10-11-07 00:53
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loloski: Following are some sample parameters that you can put in your
sip.conf in both your Asterisk boxes:
session-timers=originate
session-expires=120
session-minse=90
session-refresher=uas
These will cause SIP sessions to/from Asterisk to be refreshed every 60
seconds. You can disconnect ethernet (as JT suggested) or kill one or more
components abruptly while calls are active and notice that calls are
released when the next session refresh happens.
Issue History
Date Modified Username Field Change
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10-11-07 00:53 rjain Note Added: 0071798
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