[Asterisk-Users] Re: Asterisk SIP architecture question

[David Thomas]

Does asterisk have support for SIP session timers?

David

On 11/24/05, Olle E. Johansson <oej at edvina.net> 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
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