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

[Dovid B]

----- 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 ?


> > -----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 ?
>
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 ? ;)