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 SIPTRUNK.com Trunk even if you are behind a NAT. A new guide for that firmware will be forthcoming.
Grandstream UCM61xx IPPBX firmware version 1.0.5.14 has passed SIP Trunk interoperability testing with SIPTRUNK.COM on April 16th, 2014. This document introduces major configuration steps performed for interoperability testing between SIPTRUNK.COM 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:http://www.grandstream.com/products/ucm_series/ucm61xx/documents/ucm61xx_usermanual_english.pdf .
Connecting a UCM
- Connect one end of an RJ-45 Ethernet cable into:
- WAN port of UCM6102
- LAN1 port of UCM6104
- LAN port of UCM6108 and UCM6116
- Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
- 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.
- Wait for UCM61xx to boot up. The LCD in the front will show its hardware information when the system boots up.
- Once the UCM61xx is successfully connected to network, the LCD will display its IP address.
- 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.
- 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:
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 gw1.siptrunk.com to gw2.siptrunk.com:
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 SIPTRUNKCOM-GW1 SIP Trunk:
Delete any existing pattern and input the following:
_1NXXNXXXXXX
_N11
_N33
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 "SIPTRUNKCOM-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 -- SIPTRUNKCOM-GW1"
Strip: "0"
Prepend: "1"
Use Failover Trunk: "SIP Trunks -- SIPTRUNKCOM-GW2"
Strip: "0"
Prepend: "1"
Sequence 3 Outbound Rule:
Calling Rule Name: "SIPTRUNK_3"
Pattern: "_011."
Privilege Level: "Internal"
Use Trunk: "SIP Trunks -- SIPTRUNKCOM-GW1"
Strip: "3"
Use Failover Trunk: "SIP Trunks -- SIPTRUNKCOM-GW2"
Strip: "3"
Configure an Inbound Route
Click on PBX → Basic/Call Routes → Inbound Routes, click on Create New Inbound Rule →, select the SIPTRUNK.COM VoIP trunk just created, enter the DID Pattern, Default Destination for an inbound route using the SIPTRUNKCOMPrimary SIP trunk. With extension 1001 as the default destination, calls coming in from the SIPTRUNK.COM Trunk will ring extension 1001 directly.
***NOTE: We have found the pattern _!NXXNXXXXXX to be the best match for most use cases.***
An Inbound Rule “SIP Trunks -- SIPTRUNKCOM-GW1” is created.
You should now create a 2nd Inbound Route for the same DID using GW2:
- Select Trunks: "SIP Trunks -- SIPTRUNKCOM-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”.
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.
Thanks for this work perfectly.
How about GAC 2500 configuration do have the instruction as well?
Hoping you can provide it also.
thanks in advance
FYI, the UCM UI has been updated and does not look like the images in this article anymore. I think an update is in order. thanks