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

[Jeremy Gault]

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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