The "Free" Stands for Freedom

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.

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Before continuing with this guide, please review our Asterisk Design Guide for considerations that affect all Asterisk-based deployments.

We have included a short configuration guide below to use your account with FreePBX. This guide assumes that you have a working FreePBX installation, and that you have command line access to that system. This guide also assumes that you're using an Analog Telephony Adapter (ATA) to connect your fax machine to FreePBX. We used a Cisco SPA112 in this example, though any of our Certified ATA's should work much the same way. There are several ways to configure FreePBX to use your account, this guide will focus on what we've found is the simplest method to do so. We are always open to improving this documentation, if you've found an easier way, please feel free to open a ticket through our support portal and let us know!

Requirements: FreePBX 12.0.74+ with Asterisk 13.4+

  • add_circle Chan_SIP:

    Create Your Trunk

    Navigate to Connectivity -> Trunks and create a new Chan_SIP trunk.


    When done, your configuration should resemble the screenshot below:


    You should remove all of the content in the USER Details section.

    Then, under Incoming -> Register String, add this line:


    Remember to Submit Changes & Applyyour configuration!

    Create Your Extension

    Next, we're going to create an extension for your ATA so it can register to Asterisk and receive/make calls on behalf of your fax machine. Navigate to Applications -> Extensionsand click the drop down to create a new, Generic Chan_SIP device. For now, put your T38fax DID in the "User Extension" field, give your extension a friendly display name, and Submit Your Changes. After you submit your changes, you can make any adjustments required to the extension itself, such as NAT or other requirements of your network.


    Be sure to grab the "Secret" as this is the password you'll need to add to the ATA (along with your DID) to register your ATA to this extension.

    Create Your Outbound Route

    Now, navigate to Connectivity -> Outbound Routesand create a new Outbound route. Give it a name, and duplicate the dial patterns found in the screenshot below. make sure to add your t38fax trunk to the "Trunk Sequence for Matched Routes" section, Submit your changes, and Apply your configuration.

    When done, your configuration should resemble the screenshot below:


    Create Your Inbound Route

    Now, navigate to Connectivity -> Inbound Routesand create a new Inbound route. For "DID Number" be sure to use the DID you received from, and set the "Destination" to the extension you created earlier.

    Adjust Your SIP Settings

    Next, you'll need to make a small adjustment to your SIP settings. Navigate to Settings -> Asterisk SIP Settings from the upper right hand menu. Under the General SIP Settings tab, find the Audio Codecs section, and configure T38 Pass-Through: Yes with Redundancy. Submit your changes and Apply your Configuration.


    Configure udptl_custom.conf

    Add the following to udptl_custom.conf:


    Make sure to save those files, and then run the following command:

    amportal reload

    You should now add your extension's credentials to your ATA so it can successfully register, and now you should be able to successfully fax using your SIP account! Need to test it out? Send a one-page fax to: 1-215-825-8792 and we'll send you a fax back that includes a small picture of your original document, and verify that ECM is enabled on your device! If you have any trouble with this, please feel free to open a ticket with our support team and someone can assist you!

  •  add_circle Chan_PJSIP: 

    Coming Soon!

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Enter your SIP Username here. (This should be a DID assigned to your account that is available for registration. IE: 18043027000)
 Still Confused? Learn More! 

Enter the SIP Password for the SIP Username you entered above.
 Still Confused? Learn More! 

Create & Enter a new, secure password for your ATA.
 Still Confused? Learn More! 

Enter your first DID here. Ensure that the "Dial/DIDX1" string points to the correct context.
 Still Confused? Learn More! 

Enter your second DID here. Ensure that the "Dial/DIDX2" string points to the correct context.
 Still Confused? Learn More!