Re: [asterisk-users] How to automaticaly close callswhenAsterisk didn't receive the bye request ?

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dovid B
>
>
> From: "Jared Smith" <jsmith@digium.com>
>
> > 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
> 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.

Does rtptimeout help if you are using canreinvite=yes ?

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