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.
Setting up the Yay.com side of the SIP Trunk
Go to VoIP > SIP Trunks and click “Add Trunk”
Set up a new SIP trunk as follows:
- Address Nickname: Anything you like
- SIP Trunk Username: Anything you like
- Domain / IP Address: The domain name or IP Address of the PBX we are connecting to
- Port / Transport: These need to match your PBX setup. The default is normally correct at Port 5060 and Transport UDP
- Outgoing Security: This is where you choose a username and a password for when your trunk connects to us to make calls.
Setting up the Grandstream UCM side of the SIP Trunk
- Navigate to your UCM in a web browser and log in with the default username of admin and password of admin.
- Navigate to Extension / Trunk > VoIP Trunks and click “Create New SIP Trunk”.
- Choose a provider name of your choice
- At Host Name, enter “talk.yay.com"
- At Username, enter the User you created above for your SIP Trunk in the Yay.com Dashboard.
- At Password, enter the User you created above for your SIP Trunk in the Yay.com Dashboard and click “Save”
- Now click the little pencil icon to edit the trunk you’ve just created and you will see an “Advanced” tab
- Adjust the Codecs to only select G.722, then PCMA, then PCMU
- Change the DTMF Mode to RFC2833 to ensure touch tones work
- Ensure Enable Quality or Heartbeat is ticked
- Click “Save” and then make sure you click “Apply Change” at the very top of your browser to apply them to the UCM
To confirm your SIP trunk is connected you can navigate to “Status” on the UCM where you should see the SIP trunk showing as “Registered”.
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 ports open like Port 80 for web traffic, so this explicitly needs setting. We list the IP’s and ports required at VoIP > SIP Trunks.
You can check whether ports are open for everyone using a port tool 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.