[Asterisk-Users] ChanIsAvail and SIP

[Matt Schulte]

All, I was reading over the chanisavail command in the wiki and was
wondering a couple things. 

First and foremost, what does this command do to determine if SIP is
available? All I could tell from a debug is that it simply checks to see
if the peer's port is open and doesn't run any callflows. Is this true?

Second, I understand that running Cut on SIP may be a little difficult.
Because the final destination becomes
SIP/peer-XXXX .. XXXX = random characters, because they can be letters
and numbers applying a range in Cut wouldn't be possible. Any
suggestions on how to get by this? Is there any other var manipulation
command?