SIP2SIP connection via CALL

Today I started with SIP and Asterisk. I new to this technique. (So hopefully my question is not to easy! :wink: )

I need a simple CALL-file to put in the OUTGOING map of Asterisk. The idea is simple.
“CALL SIP/6000 + CALL SIP/6001 and connect them!”

( Like a normal call but then with a CALL-file.)
Just like the working example below, stored in “”-file:

Channel: SIP/6000
Application: Playback
Data: hello-world


Use Dial as the application (or use an extension).

Today I did some hours of tests. At this moment I think I don’t see how the idea of the programming works. And it should be simple. Still I didn’t get is working. So hereby a post again.

If I use:
Channel: SIP/6000
Application: Dail

The SIP 6000 is called and then disconnected automaticly.
I tried many options that the SIP 6000 isn’t disconnected but I didn’t succeed.

Can somebody give me an extra hint?

The idea is still the same. Via a CALL-file is the OUTGOING map in Asterisk.
“CALL SIP/6000 + CALL SIP/6001 and connect them!”
(or call 6000 and when connected vaal 6001)


At this moment I have the following CALL file:

Appplication: Originate
Channel: SIP/6000
Exten: 6001
Priority: 1
Timeout: 60000
Context: default

The SIP6000 phone is ringing. When I answer the call the connection is lost.

The Idea is that, when answer the call 6000, there will be a call made to 6001. and put them together as one call (When 6001 answers the phone). This does not happen.

Can somebody give a tip?Idea?

The call file now looks correct, which is confirmed by the fact that the phone rings. You now have a problem with the SIP call, for which you will need SIP debugging.

I’m assuming that extension 6001 dials device SIP/6001.

In your first attempt you mis-spelled Dial and didn’t provide it with any data.