Sip.US Configuration Guide for Grandstream UCM61XX Firmware

Connecting a UCM

  1. Connect one end of an RJ-45 Ethernet cable into:
  • WAN port of UCM6102
  • LAN1 port of UCM6104
  • LAN port of UCM6108 and UCM6116
  1. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub
  2. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM61XX. Insert the main plug of the power adapter into a surge-protected power outlet
  3. Wait for UCM61XX to boot up. The LCD in the front will show its hardware information when the system boots up
  4. Once the UCM61XX is successfully connected to network, the LCD will display its IP address
  5. Open a web browser the the PC and enter the web GUI URL in the format:

http(s)://IP-Address:Port The default protocol is HTTPS and the default port number is 8089.

  1. The web GUI login page will be displayed. Default username and password for administrator are admin/admin

Create a UCM Extension 

After Logging into UCM61xx web GUI, click on PBX→ Basic/Call Routes→ Extensions, click on “Create New SIP Extension” to create a UCM extension.  This extension is used in UCM SIP trunk test.


Configure a VoIP Trunk

Click on PBX → Basic/Call Routes → VoIP Trunks, click on “Create New SIP Trunk”, enter the SIP trunk account information:

Click on Save, a register SIP trunk is created. Click on “Apply Changes” to make the change take effect.

After saving the trunk, you can then edit it, and an "Advanced Settings" tab will be available.

To create a secondary SIP trunk, enter the following information and then fill out the subsequent screen exactly the same as the first trunk replacing every instance of to


Configure an Outbound Route

Click on PBX → Basic/Call Routes → Outbound Routes, click on “Create New Outbound Rule”, enter the Calling Rule Name, Pattern etc. for an outbound route using the SIPUS_GW1 SIP Trunk:

Delete any existing pattern and input the following:





Configure an Inbound Route

Click on PBX → Basic/Call Routes → Inbound Routes, click on Create New Inbound Rule →, select the SIP.US VoIP trunk just created, enter the DID Pattern, Default Destination for an inbound route using the SIP.US Primary SIP trunk. With extension 1000 as the default destination, calls coming in from the SIP.US Trunk will ring extension 1000 directly.

An Inbound Rule “SIP Trunks -- SIPUS_GW1” is created.

You should now create a 2nd Inbound Route for the same DID using GW2:

- Select Trunks: "SIP Trunks -- SIPUS_GW2"
- Add the DID Pattern: "_1NXXNXXXXXX" where NXXNXXXXXX is the actual 10 digit DID.

Configure an Anonymous Call

Go to UCM61xx Web GUI, click on PBX->Extensions, and edit one extension account. Enter number on “CallerID Number” and enter names on “First Name” and “Last name”.

Click on VoIP Trunks, edit SIP Trunk. Disable “Keep Trunk CID”.


Under the Advanced tab, empty the "From User" field. 



Supplementary Notes:

If you are behind NAT and your Trunk is showing "Registered" at SIP.US, but it is registered to a private IP Address you will need to navigate to "PBX" ---> "SIP Settings" ---> "- NAT" and input your external IP Address in the "External Host" field. You must also put your local network address in the "Local Network Address" field.

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  • Avatar
    Hemeidy Mohamed

    we use grandstream 6204 i need to setup VOIP TRUNK using SIP.US

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