Does asterisk have support for SIP session timers? David
On 11/24/05, Olle E. Johansson <[EMAIL PROTECTED]> wrote: > Matt Riddell wrote: > > Kevin P. Fleming wrote: > > > >>Matt Riddell wrote: > >> > >> > >>>So how does Asterisk know that the media stream has been disconnected > >>>between > >>>the two remote hosts? > >> > >>It doesn't... nor does any other SIP softswitch. See my other reply for > >>a possible solution. > > > > > > I agree that you could code a fix, but saying my advice is bogus because > you > > could code a fix for Asterisk to avoid it is slightly wrong. > > > > The fact remains, if you need *very* accurate cdr's then you either don't > do > > canreinvite=yes for the peer or you code something so that Asterisk > notices > > that the rtp has stopped. The fact remains that without these, the most > > accurate CDR is going to come from the provider. > > > > If the audio goes through asterisk without re-invites, you could use the > rtptimeouts to detect a dead phone. > > /O > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users