Hi everyone!
I need to make two “channel originate SIP/XXXXX@XXX.XXX.XXX.XXX …” commands, play a sound and communicate to each other.
I’m using asterisk as a SIP phone in another Asterisk server, so, my asterisk phone has no authentication info of the other phones. (I can’t use “Dial” command)
You can create some dialplan that you originate the channels into that will play the sound and then drop both parties into a conference bridge so they can talk to each other.
channel originate SIP/XXXXX@XXX.XXX.XXX.XXX extension s@mycontext
channel originate SIP/YYYY@YYY.YYY.YYY.YYY extension s@mycontext
[mycontext]
exten => s,1,NoOP()
same => n,Answer()
same => n,Playback(tt-monkeys)
same => n,ConfBridge(my-bridge)
same => n,Hangup()
Great!!! Thanks Bro!!
One more question, I’m using the extensions.conf preinstalled file.
Where should I add a new context?
I don’t understand where it should be, im using de “demo” context.
Thank you very much! Saludos from México!
(Sorry if bad english)
You can add your new context anywhere in extensions.conf
I’d recommend adding it at the bottom for ease of finding it.
Thank you very much, that solved my problem.
High five!
I don’t understand why you cannot use Dial().
Because the phones aren’t registered on the asterisk I’m using as a phone. They are registered in the Asterisk server.
The Dial() Application can call the phones via their SIP URI just like Originate does.