SIP.US Configuration Guide for Grandstream UCM6100 Series PBX

3/24/16 NOTE: The newest firmware supplied by Grandstream has an additional feature on the trunks for "NAT." Please make sure that box is NOT CHECKED on your SIP.US Trunk even if you are behind a NAT. A new guide for that firmware will be forthcoming.

UCM61xx_SIP_Trunk Configuration

Grandstream UCM61xx IPPBX firmware version has passed SIP Trunk interoperability testing with SIP.US on April 16th, 2014. This document introduces major configuration steps performed for interoperability testing between SIP.US and  Grandstream UCM61xx_IPPBX.  For detailed information to configure SIP Extensions, PSTN lines, SIP trunks and all the other system settings via web GUI, please download the UCM61xx user manual from Grandstream web site: .

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

2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.

3. 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.

4. Wait for UCM61xx to boot up. The LCD in the front will show its hardware information when the system boots up.

5. Once the UCM61xx is successfully connected to network, the LCD will display its IP address.

6. Open a web browser on 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.

7. 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 à Basic/Call Routes à Extensions, click on “Create New User” 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/IAX Trunk”, enter the SIP trunk account information:


On newer firmware the screen above is split and there is an "Advanced" screen which may not appear until after the trunk is initially created. Certain settings on the advanced screen need to be filled out like the image below.

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


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:




An Outbound Rule is created. You should also do the following during this step:

- In the "Use Failover Trunk on your first rule:" section select the "SIPUS-GW2 Trunk" and set the "Strip:" field to "0"

- Create 2 more New Outbound Rules:

Sequence 2 Outbound Rule:
Calling Rule Name: "SIPTRUNK_2"
Pattern: "_NXXNXXXXXX"
Privilege Level: "Internal"
Use Trunk: "SIP Trunks -- SIPUS-GW1"
Strip: "0"
Prepend: "1"
Use Failover Trunk: "SIP Trunks -- SIPUS-GW2"
Strip: "0"
Prepend: "1"

Sequence 3 Outbound Rule:
Calling Rule Name: "SIPTRUNK_3"
Pattern: "_011."
Privilege Level: "Internal"
Use Trunk: "SIP Trunks -- SIPUS-GW1"
Strip: "3"
Use Failover Trunk: "SIP Trunks -- SIPUS-GW2"
Strip: "3"

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 SIPUSPrimary SIP trunk. With extension 1001 as the default destination, calls coming in from the SIP.US Trunk will ring extension 1001 directly.

***NOTE: Certain versions of the firmware require a "+" symbol before the digits. Example _+15557778888

If you are seeing "407 Unauthorized" in the logs when you try and call your Grandstream, this is usually the cause***


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”, and empty the option of “From User”.

Go SIP.US account Web Page, empty the “CallerID Override” and select on for “Contact Override”.


 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 IP Address" field. You must also put your local network address in the "Local Network Address" field.

Some people prefer to set a single caller ID (CID) for the entire PBX rather than setting it on each extension. Note that the number you wish to call goes in the "Global OutBound CID:" field, which is circled in the image below. Please place 10 digits in that field and no text. If you place text in that field certain carriers will reject your calls.

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