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How to set up a SIP trunk on a Grandstream UCM Hardware PBX using an IP address

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.

 

Setting up the Yay.com side of the SIP Trunk

Go to My 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 Public IP Address of the PBX we are connecting to. Make sure this is the public IP, and not the individual private IP address. You can find this by visiting sites such as whatismyipaddress.com
  • Port / Transport: These need to match your PBX setup. The default is normally correct at Port 5060 and Transport UDP.
  • Outgoing IP Address: The Public IP Address of the PBX we are connecting to.

 

Setting up the Grandstream UCM side of the SIP Trunk

  • Navigate to your UCM in a web browser (using its IP Address, noted on the back of the PBX),and login with the default username of admin and password of admin.
  • Navigate to PBX > Basic/Call Routes > VoIP Trunks and click “Create New SIP Trunk”.
  • Choose ‘Peer SIP Trunk’ as your type.
  • Choose a provider name of your choice.
  • At Host Name, enter “talk.yay.com”.

 

 

  • 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 “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.

  • Make sure the permission you assign to the Extension is compatible with the permission you assign to your Call Routes.
  • Password can be anything you like, but make sure it isn’t too easy!

 

 

To add an outbound call route, navigate to PBX > Basic/Call Routes > Outbound Routes.

  • The Calling Rule Name can be anything you like.
  • The Pattern should be configured as needed. The pattern shown is acceptable to show that it works, however this is very unsafe, as it allows for calls to be made to any number.
  • The Privilege level should be set so that extensions with the correct permission will be able to use it.

 

 

To add an inbound route,  navigate to PBX > Basic/Call Routes > Inbound Routes.

  • Select your SIP Trunk from the drop-down menu in ‘Trunks’.
  • The Pattern should be configured as needed. The pattern shown is acceptable to show that it works, however it allows for calls to be received from any number.
  • Assign the Default mode to your desired destination. In this case, the extension we set up earlier is the only thing assigned to receive calls.

 

 

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 http://www.yougetsignal.com/tools/open-ports/ 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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