[asterisk-dev] SIP ghost calls

[Hans Petter Selasky]

On Thursday 30 November 2006 18:03, Zoa wrote:
> Google for rtptimeout
>
> Zoa

Yes, I found "rtptimeout" and "rtpholdtimeout". But I think "qualify" is more 
interesting.

What about "outside bridges", when Asterisk does not forward the RTP stream?

On Thursday 30 November 2006 18:09, Volkov Alexei wrote:
> Hi Hans Petter!
>
> First of all you do not will pay for unconnected call.
> If someone answer your ghost call, i think it disconnect quickly for two
> reasons
>     INVITE not confirmed
>     empty RTP stream for some predefined time (3 min by default) leads
> to call disconnect.
>
> And last, but not least, try to use qualify=3000 in sip friend. It makes
> asterisk to peroidicaly ping sip endpoint to track the reachability.

Whould it be an idea of Asterisk would qualify a SIP phone also when it is 
connected, to see if it is still present. That would work in any telephone 
state, and not just when one is receiving RTP packets or the call is on hold?

Is "qualify" the equivalent to STATUS_ENQUIRY in the EuroISDN world ?

>
> > Hi,
> >
> > I'm playing around with a new SIP phone connected to Asterisk 1.2.13. I
> > am performing some tests, and one of them is the "cable unplug" test.
> >
> > During a call I unplugged the ethernet cable, and to my surprise, the
> > call was not disconnected. Even after 20 minutes. This is very bad!
> >
> > Can anyone explain why Asterisk does not ping the SIP phone regularly,
> > and if there is no reply, disconnect all associated calls?
> >
> > Just imagine what happens if two persons call eachother, using SIP phones
> > over the PSTN network, and both disconnect their phones. Who is going to
> > pay for the ghost call, which might last forever?
> >

Thanks,
--HPS