> -----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 ? _______________________________________________ --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
