Integrating CUCM with Asterisk using SIP Trunk

Are you working in a scenario where you have Cisco Unified Communications Manager and you would like to install Asterisk and integrate with it?

There are various reason why you may need Asterisk: For example – You may want to do IVR setup/ Agent Call Routing/ Voicemail setup and your company don’t want to invest in purchasing Cisco Unity Connection or Cisco Unified Contact Center Express. To save cost and get your requirements done, Asterisk is one of the cheapest solutions.

Here is the step to step guide to integrate CUCM with Asterisk using SIP Trunk. Let’s get started:

We will use the following topology and number ranges to create the trunk.

10XXX Phones are connected to CUCM and 20XXX Phones are connected to Asterisk. We will create a SIP Trunk between CUCM and Asterisk to route the calls between 10XXX and 20XXX.

Configuration on Cisco Unified Communications Manager

Trunk Configuration

  • Login to Cisco Unified Communications Manager
  • Navigate to Device > Trunk > Add a New Trunk
  • Trunk Type > SIP Trunk
  • Device Protocol > SIP Trunk
  • Trunk Service Type > None (Default)
  • Click on Next
  • Device Name > Enter a name of the SIP Trunk
  • Description > Enter a short description of the SIP Trunk
  • Device Pool > Select a Device Pool for the SIP Trunk
  • Inbound Calls > Select CSS that coincides with the devices routed through this trunk.
  • SIP Information > Enter the IP Address of Asterisk Server under Destination Address
  • Destination Port > By default the port number is 5060. Enter 5060 unless you have modified the listening port in Asterisk.
  • SIP Trunk Security Profile > Select Non Secure SIP Trunk Security Profile
  • SIP Profile – Select Standard SIP Profile
  • Click on Save
  • Click on Apply
  • Click on Reset
Enter Device Name, Description, Device Pool etc.
Select Inbound CSS
Enter Destination IP Address, Port Number and Profile information

Modifying SIP Trunk Security Profile

  • Navigate to System > Security > SIP Trunk Security Profile
  • Click on Find
  • Click on Non Secure SIP Trunk Profile or the profile which you have used in your SIP Trunk as SIP Trunk Security Profile
  • Outgoing Transport Type > Select UDP from the drop down
  • Click on Save

Creating a Route Pattern

  • Navigate to Call Routing > Route/Hunt > Route Pattern
  • Click on Add New
  • Route Pattern > 20XXX
  • Route Partition > Select a partition that is accessible to the IP Phones registered to CUCM who can dial into Asterisk Server
  • Description > Enter a short description
  • Gateway/Route List > Select the Trunk which you had created in the above step.
  • Click on Save
Configuring Route Pattern

Configuration in Asterisk

Add a SIP Trunk in Asterisk

  • Login to your Asterisk PBX
  • Navigate to PBX > Trunks >
  • Click on Add a SIP Trunk
  • Trunk Name > Enter a name of the SIP Trunk
  • Locate Dialed Number Manipulation Rules and Put 10XXX in the Match Pattern Box
  • Trunk Name > Enter a name of the SIP Trunk
  • Peer Details : Enter the following and replace the IP Address with your CUCM IP Address


  • Click on Submit Changes
  • Click on Apply Config
Enter a Trunk Name and Match Pattern
Enter Trunk Name and Peer Details

Configure Outbound Routes

  • Click on Outbound Routes
  • Route Name > Enter a Route Name (this route will be towards CUCM)
  • Locate Dial Patterns that will use this Route and enter 10XXX in the Match Pattern Box
  • Locate Trunk Sequence for Matched Routes and Select the Trunk which you had created in the above step.
  • Click on Submit Changes
  • Click on Apply Config

That’s All! The configuration is complete. You can start making calls from Asterisk to CUCM and vice-versa and all the calls will work.

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