[Asterisk-Users] Callback application

[Rod Bacon]

I don't know if what you're trying to do is possible, but the easiest way to 
check would be to take a look at the raw packets on the ethernet interface 
of your * server once a call is in progress. If indeed the RTP can be handed 
off to the 2 endpoints, you should only see SIP traffic at your server. 
TCPDUMP is your friend.


----- Original Message ----- 
From: "snacktime" <snacktime at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at 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/${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)
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