[Asterisk-Users] Re: call does not hangup after client quits

[Stephen J. Wilcox]

Hi Jeremy,
 thanks for this, it looks like just what i was after. i've not finished testing 
it but it seems to do what i need.

Steve

On Tue, 9 Aug 2005, Jeremy Gault wrote:

> Steve,
> 
> If I am understanding your situation correctly (i.e. you are using a SIP 
> client and then forcibly disconnecting/shutting it off during a call) 
> you may want to look at your sip.conf for a setting called rtptimeout. 
> This may do exactly what you want.
> 
> When on a SIP call, and you disconnect/shut off your client (without 
> properly hanging up first) then (obviously) * does not receive a SIP 
> message saying the call has ended. However, the RTP (audio) stream will 
> stop. The rtptimeout setting lets you define a time period that after 
> <x> seconds of no audio packets, it's assumed the SIP client has gone 
> away and the call should be terminated.
> 
> Jeremy
> 
> Stephen J. Wilcox wrote:
> 
> >Hello,
> > can anyone help with my problem below, searching doesnt show any results..
> >
> >thanks
> >Steve
> >
> >
> >On Wed, 3 Aug 2005, Stephen J. Wilcox wrote:
> >
> >  
> >
> >>Hi,
> >> I'm seeing a problem where if I place a call, then forcibly quit or turn off 
> >>the client the call stays active.
> >>
> >>The frames counters stop so its apparent the client has gone away but the call 
> >>remains active.
> >>
> >>Asterisk is CVS-HEAD 23-Jun-05
> >>
> >>What is supposed to happen in this scenario?
> >>
> >>thanks
> >>Steve
> >>
> >>
> >>    
> >>
> >
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