After some search in wiki I was able to do what I wanted. Here is how it is,
The .call file should appear something like this and it has to be placed
in /var/spool/asterisk/outgoing of asterisk-1,
Channel: local/[EMAIL PROTECTED] ; Any extension can be called using
local/extension@context
Hi,
I would like to connect two linux machines running asterisk and then
originate SIP calls from one asterisk and terminate it on the other
asterisk. Terminating the call is not a problem because I can give the
call handle to say AGI application on the terminating asterisk. How do i
This is the how long is a piece of string question.
It all depends on the hardware Asterisk sits on, the codecs in use, the
dialtone provider (SIP vs IAX vs T1/E1) etc.
Do a wiki search and you'll find some examples of what folks have found.
As for originate on one and terminat on another;
Shawn,
Thanks for info that would solve the problem of manually making calls
and connecting the phones at the either ends. But my requirement is
slightly different. I've the following .call file in the
/var/spool/asterisk/outgoing directory of asterisk-1
asterisk-1 - SIP -
