[asterisk-users] dead sip channel

[Jerry Geis]

>
> 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
>   
I uncommeted the rtptimeout=60 value in sip.conf and did a reload.
It still hasnt dropped the dead call after a couple minutes now...

Do I have to stop and start again? Was hoping it would just drop the 
call and continue on.

Jerry