[Asterisk-Users] Callback application

[snacktime]

I wasn't sure how else to label this thread because I'm not sure on
the correct terminology to use when decribing what I'm trying to do...

I am using livevoip and have a DID with them also, both using SIP. 
THe big picture is that I'm making a callback application.  Right now
I'm testing out a couple of things just using DISA.

What I'm trying to do is setup a two legged call using * and DISA,
with both legs going to/from livevoip, and set the call up in a way
where the voice traffic goes straight between livevoip/livevoip once
both legs are established.  What I don't know is how to tell if I have
succeeded in this.

Using the following I get both legs up and * say's it's created a
native bridge between the two legs.  However a 'sip show channels'
still shows both channels in *.   How do I tell if the voice data is
not going through * anymore?

Basically once the legs are joined, with one originating from livevoip
and one terminating to livevoip, I want my * box out of the picture as
far as the voice data stream goes.

 

[outgoing]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@livevoip-out,30,r)

[from-livevoip]
exten => 800xxxxxxx,1,Ringing
exten => 800xxxxxxx,2,Wait(1)
exten => 800xxxxxxx,3,Answer
exten => 800xxxxxxx,4,DISA(no-password|outgoing)