Troubleshooting a SIP Trunk over a SBC


(Ian Ford) #1

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,


#2

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.


(system) closed #3

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