[Asterisk-Users] Gastman and SIP?

[Tilghman Lesher]

On Friday 26 September 2003 02:03 pm, James Sizemore wrote:
> 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.

That's also how Zap channels work.  Unfortunately, SIP channels are a
little more difficult to link to a particular icon, as SIP channels
are created and destroyed on the fly and channel numbers are not
reused (note that SIP/3846 means that this is the 3,846th SIP session
since asterisk was last restarted).

> 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?

Originate should allow a call to be started from the GUI.  The
originating channel rings, then the call is started as if the
extension entered in the GUI was actually dialled on that channel.

I'm not sure what invite should do, though.

-Tilghman