Troubleshooting a SIP Trunk over a SBC

[Ianford]

Hi, New to SIP trunking… thought this looked simple enough, but…

I am behind a Session Boarder Controller, I was given a test DID, the PBX IP, the Gateway and Subnet. I followed everything suggested here… https://wiki.freepbx.org/display/SBC/FreePBX-PBXact+Trunking+Configuration. Since it’s a Peer to Peer, I can’t verify the peer connection in the asterisk info page.

I call the DID number, I get a busy signal, and no record of it in any logs I’m aware of. I make an external call from the phones, I get “All circuits are busy now…” and an ‘restrictedroute …’ in the call details.

Not sure what I’m doing wrong here, but I could use a win. Any thoughts?

I just need to know ways to diagnose the problem.

Thanks in advance,

[penguinpbx (C. Maj)]

Try connecting to the Asterisk CLI and watching there what happens when you make a call. “core set verbose” command should let you see anything.