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 Posted: Thu Jan 03, 2008 4:19 pm Post subject: [asterisk-users] How to automaticaly closecallswhenAsterisk |
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----- Original Message -----
From: "Steve Langstaff" <steve.langstaff at citel.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Thursday, January 03, 2008 11:49 AM
Subject: Re: [asterisk-users] How to automaticaly closecallswhenAsterisk
didn't receive the bye request ?
| Quote: | | 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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Nope which Jared just explained to me. I am so used to not allowing invites
that this one just went right over my head.
Zoooooooooooooooooooooooooooooooooom. What was that ?  |
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