I’m creating application which allows to dial and answer calls from multiple SIP lines simultanously. My goal is to have for ex. 2 headsets and two different calls from them.
Let’s say I have four SIP numbers:
My IP addres 184.108.40.206
I registered 2 SIP numbers from my app (500 and 501) and 502 and 503 are registered from SIP hardphones.
I’ve set up connection (invite):
From field: “500” firstname.lastname@example.org
To: “502” email@example.com
And another connection in the same time:
From field: “501” firstname.lastname@example.org
To: “503” email@example.com
It seemed working until I checked which extension was used. Both calls used SIP/500. I guess it is because asterisk checks IP address of incoming connection and runs extension which fits that number. Can I change this behavior?
I don’t know what else I can change from my application, it feels like normal behavior when PBX checks FROM field of invite packet when it allows to register multiple numbers.