[Asterisk-Users] Gastman and SIP?

[James Sizemore]

I have been testing Gastman and Astman with SIP calls. As I have no Zap 
phones, so I have a few question on what is normal behavior? When a call 
comes in and I have created extensions for all phones (example: Channel 
= "SIP\3846") Should the little lines connect between the pre-made 
extension or should they pop up temporary icons with no connection to 
the hand made extensions?  The Green light does light up.

What should "Invite" and "Originate" do, right now they just  ring a 
phone once and hangup.
Anyone know of any other programs that I can be tested for call status 
and redirection?