WebRTC Softphone module now available for FreePBX!

Schmooze Com, Inc. announces first public release of WebRTC Softphone module for FreePBX.


Neenah WI, – January 27, 2014 – Today, Schmooze announces the availability of the BETA release of The FreePBX WebRTC Softphone. This is the first public release of an officially supported WebRTC module for the world’s most popular Open Source PBX platform, FreePBX.


WebRTC stands for “Web Real-Time Communications,” a technology focused on embedding real-time communications, such as voice, directly within web browsers.


The new WebRTC add-on module allows FreePBX users to enable real-time communications from a web browser directly with their FreePBX system. System administrators will enable an additional WebRTC device in their end users User Control Panel, thereby allowing end users to make and take calls directly from a supported web browser. Anytime a user’s regular extension rings, the WebRTC Softphone will also ring if they are logged into the phone, allowing them to take calls directly from a browser.


“This Free WebRTC add-on to FreePBX will enable end users of FreePBX to stay in touch, as well as provide exciting new options for contact centers and businesses wanting to utilize FreePBX within their environments.” commented Preston McNair, Vice President, Sales and Marketing at Schmooze Com, Inc./FreePBX. “In my personal testing I have even used it directly with the browser on my Android Tablet; this is just the start of where this technology will eventually take us.”


FreePBX system administrators can download the WebRTC module from within the FreePBX Module Admin, or check out this video that shows off some of the new features enabled by the WebRTC module.

For the latest FreePBX news, updates and information follow FreePBX and Schmooze Com, Inc. on Twitter at @freepbx @schmoozecom


On Behalf of the Schmooze Com/FreePBX Team,


Preston McNair, VP of Sales and Marketing, FreePBX/Schmooze Com Inc 


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Copyright © 2013, Schmooze Com Inc FreePBX is a Registered Trademark of Schmooze Com Inc All Rights Reserved.

25 thoughts on “WebRTC Softphone module now available for FreePBX!

  1. You need to have at least Asterisk 11.5, the module won’t show up in Module Admin until next week as we are doing limited structured releases of this module for distros we can control so we don’t have to deal with a large amount of bug requests. It will be released on other systems shortly.

    Can you explain what you mean by “Manual Install Fail”

  2. Yeah, tried that. I got a weird response.

    [root@freepbx ~]# yum update asterisk
    Loaded plugins: fastestmirror, kmod
    Loading mirror speeds from cached hostfile
    Setting up Update Process
    No Match for argument: asterisk
    No package asterisk available.
    No Packages marked for Update
    [root@freepbx ~]#

    I mean I obviously have asterisk installed. Not sure what I’m missing.

  3. That because you operation system does’t know where locate packages. You must:
    1. go to link http://packages.asterisk.org
    2. select your OS
    3. copy *.repo file with version asterisk 11 to: /etc/apt/sources.list on Debian like OS;
    and /etc/yum.repos.d/ RHEL like OS.
    4. Be sure set enabled=1
    5. yum update asterisk
  4. Last issue (hopefully). I’m getting a dependency error it seems.

    --> Finished Dependency Resolution
    Error: Package: wanpipe-7.0.2-kernel. (@pbx)
    Requires: dahdi-linux = 2.6.1
    Removing: dahdi-linux-2.6.1-13_centos6.x86_64 (@anaconda-CentOS-201207061011.x86_64/6.3)
    dahdi-linux = 2.6.1-13_centos6
    Updated By: dahdi-linux- (asterisk-current)
    dahdi-linux =
    Available: dahdi-linux-2.6.1-7_centos6.x86_64 (pbx)
    dahdi-linux = 2.6.1-7_centos6
    Available: dahdi-linux-2.6.2-1_centos6.x86_64 (asterisk-current)
    dahdi-linux = 2.6.2-1_centos6
    Available: dahdi-linux-2.7.0-1_centos6.x86_64 (asterisk-current)
    dahdi-linux = 2.7.0-1_centos6
    You could try using --skip-broken to work around the problem
    You could try running: rpm -Va --nofiles --nodigest
    [root@freepbx ~]#

    Any ideas?

  5. My webRTC with Chrome is working only for outbound calls, inbound calls goes to voice mail. no registration appear in peer status.
    I opened my firewall http:tcp 8088 for my PC.
    BTW my PBX is on the Cloud in a VPS
    Thanks great apps
  6. Thanks TM1000. My question was about the ports and protocols that need to be opened/forwarded in our firewall/router (Or in the FreePBX box if exposed to public internet). Can you clarify? I have installed the WebRTC module and activated one of the extensions and I can’t make/receive calls.
  7. Our remote WebRTC extension is working! We established an SSH session tunneling ports 80 and 8088 to our test server on the cloud and it worked! So I assume these 2 ports need to be opened in the FreePBX server (Or like in our case, tunneled via encrypted SSH via PUTTY for a safer connection).

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