> Is there a way to detect that there is no longer really an
> active call happening and force a hangup or reset the
> channel? It'd be great if this could happen automatically.
> Or as a temporary fix , is there a way to setup and extension
> that the SIP phone could dial which would clear any active
> calls associated with it? Right now if this happens, I need
> to login to the Asterisk CLI and issue a hangup command. If
> I don't, the channel appears to be in-use forever.
This may be the answer
sip.conf:
;--------------------------- RTP timers
----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP
timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or
RTCP activity
; on the audio channel
; when we're not on hold. This is to be able
to hangup
; a call in the case of a phone disappearing
from the net,
; like a powerloss or grandma tripping over
a cable.
Alec Davis
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