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 Posted: Thu Jan 03, 2008 4:49 am Post subject: [asterisk-users] How to automaticaly close callswhenAsterisk |
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| Quote: | -----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dovid B
From: "Jared Smith" <jsmith at digium.com>
| Quote: | There is a SIP timers patch in the bug tracker (see
http://bugs.digium.com/view.php?id=10665) that currently implements
this, and it's being tested in the team/group/sip_session_timers/
branch in SVN. Please test this out and help provide feedback, so
that we can get this put into Asterisk in time for the next
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major release.
Jared,
I would think of using rtptimeout. There is a reason why you
did not mention it and I am curious as to why.
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Does rtptimeout help if you are using canreinvite=yes ? |
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