Configuring FreePBX to connect with Zentrunk

Overview

Zentrunk is a SIP Trunking service from Plivo that allows you to connect with fixed and mobile phones in over 200 countries. Connect your cloud or on-premise communication infrastructure to Plivo’s Zentrunk SIP Trunking service to connect to your customers easily.

This documentation provides a basic configuration to get FreePBX up and running with Plivo as the external SIP gateway.

To get started with Zentrunk using FreePBX you would need to do the following:

  1. Install FreePBX on your environment.
  2. Create a Trunk on Zentrunk using Plivo Console.
  3. Configure an Outbound Trunk.
  4. Configure the Inbound Trunk.

Installation of FreePBX

To start using Zentrunk, you would need to install FreePBX on your environment. If you already have a FreePBX instance running, you may ignore this step.

For more information on installing FreePBX, check the following official installation guides:

  • Follow the instructions here to install FreePBX on Debian.
  • Follow the instructions here to install FreePBX on CentOS.
  • Follow the instructions here to install FreePBX on Ubuntu.

Create a Trunk on Zentrunk

You can create a trunk using Plivo Console. For more information on creating a Trunk on Plivo Console, see Getting Started with Zentrunk.

Configuring an Outbound Trunk

Configuring your Outbound Trunk involves the following steps:

  1. Adding a Trunk
  2. Adding Outbound Routes
  3. Configuring an Extension
  4. Configuring X-Lite Note: There are many softphones that you can use (for example, X-Lite, Blink for Linux, etc). In this tutorial, we will be using the X-Lite Softphone.

To add a trunk

  1. From your FreePBX dashboard, hover over the Connectivity menu, and then click on Trunks.
  2. On the Trunks page, click Add Trunk, and then click Add SIP (chan_sip) Trunk.
  3. On the Add Trunk page, enter the following details in the General tab:
    • Trunk Name: A friendly name for the trunk (for example, demo-trunk)
    • Outbound Caller ID: Caller ID for the calls placed using this trunk. This has to be in the following SIP format: 
       "demo" <phone-number>

      Add Trunk Details

  4. Next, enter the following details in the sip Settings tab:
    • Host - Termination SIP Domain of your Plivo Trunk
    • Username - Username for TestAuthGroup
    • Password - Password for TestAuthGroup
    • Peer - Creates a peer connection SIP Settings
  5. Add dial pattern, you can choose to match any specific digits or use “.” as a wildcard to match one or more characters. Dial Pattern
  6. Enter a friendly name for the Trunk in the Trunk Name field, and then click Submit. A trunk will be created. You must now configure an outbound route to this trunk.

To add an outbound route

  1. From the Connectivity menu, click Outbound Routes.
  2. On the Outbound Routes page, click Add Outbound Route.
  3. In the Outbound Route - Add Route page, enter a friendly name for the route in the Route Name field.
  4. From the list of Trunk Sequence for Matched Routes, select the trunk you created in the previous section (demo-outbound). Select the Trunk
  5. Next, on the Dial Patterns tab, enter the dial pattern that matches your system. Note: You can choose to match any specific digits or use “.” as a wildcard to match one or more characters. Select Dial pattern
  6. Once you have entered your dial pattern, click Submit. The outbound route will be configured to your trunk. You must now configure an extension.

To add an extension

  1. From the Applications menu, click Extensions.
  2. On the Extensions page, click Add New Chan_SIP Extension from the Add Extension list. The Add SIP Extension page appears.
  3. In the General tab, enter the following details:
    • User Extension - The extension number to dial to reach the user
    • Display name - The callerID name for calls from this user will be set to this name
    • Secret - Password configured for the device Extension Details
  4. Once done, click Submit. Your extension is created.
  5. On the All Extensions tab, in the Actions column, click edit for your extension.
  6. In the Advanced tab, under the Edit Extension section, change the configuration for NAT Mode to Yes - (force_rport,comedia). Modify Extension
  7. Click Submit. Your extension will be created.

Asterisk SIP Settings

  1. From Setting menu, click Asterisk SIP Settings.
  2. Next, enter the following details in the General SIP Settings tab:
    • External Address - Click on ‘Detect Network Settings’ and copy the value for future reference Asterisk SIP settings
  3. Next, enter the following details in the Chan SIP Settings tab in Advance General Settings section :
    • Bind port - 5060
    • TLS Bind Port - 5061 Port Details
  4. Click Apply Config on the top navigation bar. All your configurations are complete. You can now configure X-Lite to use your Trunk. Note: In case you get an error while configuring your softphone, make sure you have clicked Apply Config on FreePBX.

Configuring an Inbound Trunk

Configuring your Inbound Trunk involves the following steps:

  1. Configuring an Extension
  2. Adding Inbound Routes
  3. Configuring X-Lite Note: There are many softphones that you can use (for example, X-Lite, Blink for Linux, etc). In this tutorial, we will be using the X-Lite Softphone.

To add an extension

  1. From the Applications menu, click Extensions.
  2. On the Extensions page, click Add New Chan_SIP Extension from the Add Extension list. The Add SIP Extension page appears.
  3. In the General tab, enter the following details:
    • User Extension - The extension number to dial to reach the user
    • Display name - The callerID name for calls from this user will be set to this name
    • Secret - Password configured for the device Extension Details
  4. Once done, click Submit. Your extension is created.
  5. On the All Extensions tab, in the Actions column, click edit for your extension.
  6. In the Advanced tab, under the Edit Extension section, change the configuration for NAT Mode to Yes - (force_rport,comedia).
  7. Addionally,check if rest of the values are configured in the below format:
    • DTMF Signaling - RFC 2833
    • Connection Type - friend
    • Port - 5060 Other Details
  8. Click Submit. Your extension will be created.

To add an inbound route

  1. From the Connectivity menu, click Inbound Routes.
  2. On the Inbound Routes page, click Add Inbound Route.
  3. In the General tab, enter the following details:
    • Description - Provide a meaningful description of what this incoming route is
    • DID Number - Define the expected DID Number if your trunk passes DID on incoming calls or leave it blank to match calls with any or no DID info.
    • Set Destination - Select Extentions from the options. Later select the extention which was created earlier. Inbound route details
  4. Click Submit. Your Inbound route will be created.

Asterisk SIP Settings

  1. From Setting menu, click Asterisk SIP Settings.
  2. Next, enter the following details in the General SIP Settings tab:
    • External Address - Click on ‘Detect Network Settings’ and copy the value for future reference Asterisk SIP Settings
  3. Next, enter the following details in the Chan SIP Settings tab in Advance General Settings section :
    • Bind port - 5060
    • TLS Bind Port - 5061 Port details
  4. Click Apply Config on the top navigation bar. All your configurations are complete. You can now configure X-Lite to use your Trunk. Note: In case you get an error while configuring your softphone, make sure you have clicked Apply Config on FreePBX.