[asterisk-users] dead sip channel

[Wolfgang Pichler]

hi,

try to set the rtptimeout value in sip.conf to a resonable value - so 
asterisk will kill the channels if it does not receive rtp traffic for 
the specified time

regards,
Wolfgang

Jerry Geis schrieb:
> I have ran into a case using 1.4.22 where a SIP call to an asterisk 
> client (running a slow PC) to ALSA
> does not hangup the call when it is done. The server is using call files 
> to initiate the call, the client answers on
> the ALSA port, the server plays the message and hangs up.
>
> I found that SOMETIMES -its hard to recreate - that the slow pc keeps 
> the SIP channel active. further calls in
> are getting a busy signal and the one call is NEVER hung up.
>
> How can I detect this and hang up the channel on the slow PC. I verified 
> on the server that it thinks NO calls are active.
>
> my context looks like this:
> [mycontext]
> exten => s,1,ChanIsAvail(Console/Dsp)
> exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
> exten => s,n,Playback(beep)
> exten => s,n,Dial(Console/dsp)
> exten => s,n,Hangup
>
> [smvoice-busy]
> exten => s,1,playtones(busy)
> exten => s,1,wait(10)
> exten => s,1,Hangup
>
> Thanks,
>
> Jerry
>
>
>
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