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