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

[Steve Langstaff]

 > -----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>
> 
> > 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 ?