----- Original Message ----- From: "snacktime" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com>
Sent: Monday, April 11, 2005 1:32 PM
Subject: [Asterisk-Users] Callback application
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/[EMAIL PROTECTED],30,r)
[from-livevoip] exten => 800xxxxxxx,1,Ringing exten => 800xxxxxxx,2,Wait(1) exten => 800xxxxxxx,3,Answer exten => 800xxxxxxx,4,DISA(no-password|outgoing) _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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