[asterisk-users] FXO ports locking up

[Alec Davis]

> 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