----- Original Message ----- From: "Steve Langstaff" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@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: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B >> >> >> From: "Jared Smith" <[EMAIL PROTECTED]> >> >> > 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 ? ;) _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
