i’m tryng to do a click to dial service in order to let people call two PSTN phone numbers.
I’ve set asterisk dialplan like this:
i’ve wrote a sip client (using twisted) that sends an invite to number A through asterisk, then when user picks up the phone, sends a Refer to asterisk with Refer-to header: numberB@mysipprovider.
Asterisk accepts the Refer method, so it starts sending notify to my client and an invite to number B.
When user at number B picks up phone my client sends a BYE to asterisk…
In such configuration, no audio is estabilished between A and B…i’m guessing why
can anybody help me? I’m going crazy…it’s from february that i try to create this service…
(used manager, file .call but always the same problem: no audio!)
p.s i’ve asterisk 1.12.18