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This FAQ is to be used for setting up a Peer SIP Trunk with your PBX, for info on how to set this up with a Registered SIP Trunk, click here.
Before you start, make sure your network cable is plugged into the LAN port of the UCM, not the WAN port. The UCM range from Grandstream is extremely powerful and you will only want to use the WAN port in a complex setup with a network engineer.
Go to My Voip > SIP Trunks and click “Add Trunk”.
Set up a new SIP trunk as follows:
To confirm your SIP trunk is connected you can navigate to “Status” on the UCM where you should see the SIP trunk showing as “Reachable”.
Now your SIP Trunk is reachable by the UCM!
To setup the PBX itself, you can add extensions by visiting PBX > Basic/Call Routes > Extensions.
To add an outbound call route, navigate to PBX > Basic/Call Routes > Outbound Routes.
To add an inbound route, navigate to PBX > Basic/Call Routes > Inbound Routes.
Finally, the most common mistake we see is that customers don’t open SIP and RTP firewall ports to allow Yay.com to talk to their PBX. Firewall’s normally only have a few default port open like Port 80 for web traffic, so this explicitly needs setting. We list the IP’s and ports required at My Voip > SIP Trunks.
You can check whether ports are open for everyone using a tool like this from you get signal, but remember you only want to allow access from our listed IP addresses, so this tool is good for getting you going when you open up to all, but won’t confirm once only locked down to just our IP’s.
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