asterisk users - Sep 2005 - how to connect two SIP channels

Hello,

 I have two extensions 100 and 200 and I need in "h" context dial
first
extension by asterisk and when the extensions
100 is answered, then to dial exten 200 and then bridge them together..

 How can I do that?

 I need it to "call back" when called party is busy.

 thanks

Hi, I am using AsteriskJava and I have some problems, I will appreciate 
any help...

My system has the following architecture (in the server side):

- An app server (connected to the asterisk console)
- An AGI Server (developed with AsteriskJava)
- An AGI Script (executed by the above AGI Server)

In the client side (Agents answering call center calls):

- A softphone
- A client program (used to search and register call details)

Here is the thing:

- From AGI Server I detect that a call is coming from PSTN and launch 
the AGI Script
- From AGI Script I put the call in the queue and I loose the control of 
the call (here is my first confusion)
- The agent answer the call (using his/her softphone) and I get the 
event from the Asterisk Console with my App Server.

Now, I need to play something (TTS, wav, etc) to the caller based on the 
client application wich is connected to my App Server. What I want you 
to know is that the information to be played to the caller comes from an 
external source.

So, my two big questions/confusions are:

- How can I get the entire control of the call depending on the status 
of the call, for example, if the call is in the queue and I need to play 
or do something with it, where and how I have the control? until now, 
when I put the call in the queue I loss the control until the caller or 
the agent hangs the call.

- Once the call is answered by the Agent, how can I unlink the two 
channels (releasing the agent) to let the caller hear the text that the 
agent sent.


Thanks in advance,

Sebas


-- 
Sebastian Silva
G R U P O  G A U S S
Depto. Sistemas
Av. Libertador 6250 4 piso
Tl.: 4 706-2222 (int. 121)
ssilva@gaussar.com