FreePBX is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server. FreePBX is licensed under the GNU General Public License (GPL), an open source license. FreePBX can be installed manually or as part of the pre-configured FreePBX Distro that includes the system OS, Asterisk, FreePBX GUI and assorted dependencies.

You'll need to have created a Credentials based connection on your Telnyx Mission Control Portal account, assigned this connection to a DID and outbound profile in order to make and receive calls.

Instructions to Configure a FreePBX PJSIP V13


FreePBX V13 is available to download from here.


You can view the installation guide here.

Video Walkthrough

Coming soon! This walkthrough will demonstrate setting up a credentials based connection with FreePBX. We'll also show you how to assign this connection to a newly purchased DID which will allow you to receive inbound calls. Then we'll walk you through how to assign the connection to an outbound profile such that you can make outbound calls!


For step by step instructions on each of the requirements on the Telnyx Mission Control Portal, please follow this guide.

Once you've configured your Telnyx account, you can now proceed to setup FreePBX V13 following the guide below.


Step by Step Guide

You are now all set on the Mission Control Portal side and are ready to configure your Telnyx trunk within your FreePBX V13 system.


Once you load the ISO onto your server or virtual machine, you'll have a few options to select for installation. We'll be doing a full install via asterisk 13.

1. Confirm your appropriate network settings.

2. Confirm your root password.

3. Wait for all the necessary packages to be installed.

4. More modules will be updated after successful internet tests.

5. Enter root and the password you created from step 2.

6. You'll now be provided with the URL you need to use in order to access the FreePBX web interface.

Chan_pjsip TrunkConfiguration: 

The default behavior of FreePBX version 13 is to use chan_pjsip for endpoints and trunks. 

Selection of either chan_pjsip or can_sip from within your distribution can be found in the Admin Web tool under Settings -> Advanced Settings ->Dialplan and Operational -> SIP Channel Driver

To configure FreePBX to work with Telnyx SIP Trunking service, you should make configuration changes in 3 areas:

  • Connectivity --> Trunks

  • Connectivity --> Outbound Routes

  • Connectivity --> Inbound Routes

Each of these is configured using the Admin Web tool provided by FreePBX.

Step 1: 

On FreePBX, go to Connectivity -> Trunks page
Click on + Add Trunk → select Add SIP (chan_pjsip) Trunk.

When adding the new trunk, many settings are available, and most have defaults already configured.

To configure a Telnyx SIP Trunking account, make modifications to the following options:

--> General Settings:

  1. Trunk Name: Telnyx_userAuth

  2. Outbound CallerID: your_Telnyx_number

  3. CID Options: Allow Any CID

--> Dialed Number Manipulation Rules:  (This entire section can be left at defaults) or
You can enter the Dial patterns Wizards.
→ Prepend = <Leave Empty>, Prefix = <Leave Empty>, Match Pattern = NXXNXXXXXX
→ Prepend = 1, Prefix = <Leave Empty>, Match Pattern = 1NXXNXXXXXX

--> PJSIP Settings:

Select the "pjsip Settings" tab and edit the settings under the "General" sub-tab. 

  1. Username : <Enter the user name which you have created in the connections tab on Telnyx Portal>

  2. Secret : <The "Secret" is the password for your trunk found under the connection → "show password" link in your Telnyx portal>

  3. Authentication : Outbound

  4. Registration : Send

  5. Language Code : English

  6. SIP Server :

  7. SIP Server Port : 5060

  8. Context : from-pstn

  9. Transport :

Select the "Advanced" sub-tab under the "pjsip Settings" tab. Look at the image below. You must edit the "From Domain" field to have “”

Select the "Codecs" sub-tab under the "pjsip Settings" tab. Here you should select ulaw, alaw, gsm, g722, g729, Opus. All other boxes should be unchecked.

Because Telnyx supports the following codecs:
G.711U (PCMU)
G.711A (PCMA)
Opus (supported for IB and OB calls, for IB calls though it's only allowed when using TLS or TCP transport)

After doing the above, please click on submit and apply config

Step 2 : 

Outbound Routes :
Now we need to configure “Outbound Routes”.  

Make your way to Connectivity -> Outbound Routes.

Select "Connectivity" then "Outbound Routes." To create a new "Outbound Route," you must first enter a distinctive "Route Name." Then select the trunk you just created as the top route in "Trunk Sequence for Matched Routes."

1. Route Name : Outbound_Telnyx
2. Route CID : <Number which you have purchased on the portal>
3. Trunk Sequence for Matched Routes : <Select the trunk which you have created>

Select the "Dial Patterns" tab. Enter dial patterns exactly like the image below. They will allow for you to dial 10 Digits (U.S. Calling), 11 Digits (North American Calling). After entering this info, and clicking "Submit" then "Apply Config". 

Step 3:
Inbound Routes :
Make your way to Connectivity -> Inbound Routes.

To direct calls from to an extension you must create an inbound route.
To start select "Inbound Routes" from the "Connectivity" menu on your FreePBX interface.

The image below demonstrates an inbound route that will send ANY call to a certain extension. To direct a specific number to a specific extension you would create a route and set the "DID Number" field to your 11 digit DID with (for instance : 12172031700).

In the Above screen shot, 1010 is my test extension created on the FreePBX. 

NOTE: By default, when creating a SIP Connection in the Telnyx Mission Control Portal, the number formats for the ANI and DNIS will be set to E.164. This means Telnyx will send the dialled number in the SIP INVITE to your FreePBX system with 11 digits. As the DID number above is in 11 digit format, the call will be accepted and routed to the extension. However, you can control the number formats as you desire and can read more about it here.

That's it, you've now completed the configuration of FreePBX PJSIP V13 Credentials Trunk and can now make and receive calls by using Telnyx as your SIP provider!

Additional Resources

Review our getting started with guide to make sure your Telnyx Mission Control Portal account is setup correctly!

Check out FreePBX's help section for community or paid support.

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