![]() Hopefully it will work later, when Deutsche Telekom. The dialplan is truly the heart of any Asterisk system, as it defines how Asterisk handles inbound and outbound calls. ![]() ![]() Digium helped me set up the TE220B, so I don’t believe there is a configuration problem there. Any call coming into the box on the PRI is receiving ‘ss-noservice’. It's the originate action that needs to be launched on the number you want to connect to call 1 and listen to the originate events of this action using the actionID. Now all works as expected, at least in the simulation I did with AsteriskNOW. Google Voice or via the Google Talk web client requires the use of Asterisk 11. Currently, any call originating on the Asterisk box is calling out across the SIP trunk just fine. Routing Incoming Calls to Queues - Asterisk Documentation Routing Incoming Calls to Queues Then in extensions.ael, you can do these things: The Main Menu At Digium, incoming callers are sent to the 'mainmenu' context, where they are greeted, and directed to the numbers they choose. To put it on hold, play the musicĪfter playing the music, we call PAMI in file 1 to connect to the AMI and launch an action. When call 1 arrives, it goes into agi-callin.php. Purpose: to have information from the 2 channels to perform a bridge action. You need the knowledge of AMI and AGI to do this. ![]()
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