Here’s the strange problem. I’ve set up Asterisk to connect to Broadvoice as the trunk, and then I have a SIP phone here on my desk which connects to Asterisk.
I’ve set up a dialplan on Asterisk so that my SIP phone here goes into a context called
staff. From within that context, it can dial out through the Broadvoice SIP line. It works fine. I can dial out and get good quality calls. I’m impressed.
Incoming calls are another matter. I’ve set it up so that callers can dial in and “press 1 for sales”, etc. For simplicity, I only have one extension going: extension 1. I want to get that working.
The user presses “1”, and my SIP phone on my desk rings, and I can answer it, but when I do, there’s no sound transmission in either direction.
Any ideas on this? How can I debug this strange problem? The server machine has multiple interfaces, and the SIP phone is connecting over an OpenVPN tunnel. Could there be some Asterisk routing problem, or??
Thanks for any help. I’m 95% of the way there, and very impressed with the system. I just need to fix this last, difficult 5%.