I’ve booked 3 VoIP numbers and i’ve embedded them into my asterisk configuration (sip.conf) in this way:
register => user1:firstname.lastname@example.org:number1
register => user2:email@example.com:number2
register => user3:firstname.lastname@example.org:number3
For my system scope, i work with outbound call files like this:
I want to produce a “dynamic” call file: in other words, if the number1 is busy, i want that the call file will be like this:
My idea is to make a channel scan (something similar to sip show channels), and select the free trunk (associated to number2 in my example) but i don’t know if it’s the best solution and how to implement it in the best way.
Can you suggest me a better way to proceed?
Thanks in advance!