Catching an unnatural disconnect

I’ve got Asterisk bridging a call between two SIP phones. When one of the SIP phones crashes, the call remains active and the other SIP phone hears dead air indefinetely.

Is there a way to catch a crash / disconnect like this when doing an Asterisk Dial?

Have a look at rtptimeout … rtptimeout

Thanks - it worked perfect.

RTPTimeout disconnects the calling party as well as the called party. Is it possible to salvage the caller channel if the called party RTPTimeouts?