[asterisk-users] Check sip availability

[criptos]

Hello everybody.

I have 100+ sip extensions over 100+ different geographical places.

Many times, the phone is intentionally left off hook, or the user change the 
phone from the sip line to the land line, simply because they want to call and 
the sip phone is the "new" one... other times, the sip line, which are 
linksys' pap, tell me that is in congestion or "circuit busy", I would like to 
test, actively for the availability of the phone... 
 
I was thinking for the dial comand inside the asterisk console for a start, 
but how do u hangup a channel?

Is the an app to do this? 

Any ideas are welcome...