[sopel1000]

So, I have a problem. I have a running asterisk system, where extension "pg" takes care of all the outgoing traffic. There is an AGI script written (I do not have access to that, it is compiled). I have a trunk to one of the wholesale voip opertators.

I want to connect two endpoints using a) Manager login b) .call filles copied to outgoing folder.

What do I do? The following:
(let's consider manager first)

Action: Login
Username: XXX
Secret: YYY

Action: Originate
Channel: SIP/trunkname/1234567234 (called number no.1)
Context: pg
Exten: 7275551212 (the other number)
Priority: 1
Callerid: Operator <1000>

Action: Logoff

This should, according to me, work, but I may be wrong. The idea is to call first party, let her hear the ringing tone, then dial the second number and connect these two parties. The first telefone rings (called by SIP/trunkname/....) but after it is picked up, it is disconnected. A php script will be written which will "talk" to the Asterisk as a manager and make him connect two endpoints.

Where is the problem?

[sopel1000]

HI, so I managed to make it not disconnect on a clean Asterisk. So, it rings one phone, then it rings the other one, but I cannot hear anything between them, like they were not bridged. No RTP. What is wrong?

[sopel1000]

Does anyone EVER reply here? It's unbelieveable, I've posted already 6 posts, some were less demanding, some more and got no reply, why? I sincerely doubt that noone knows the answer...