Skype gateway with FreeSWITCH’s mod_skypopen

There has been a lot of talk about Skype connectivity in the VoIP blogs lately. Digium announced that they will no longer be selling their Skype channel driver, and this news rekindled interest in free/DIY methods for connecting Skype to Asterisk. Nerd Vittles has a good writeup, specific to the PBX-in-a-Flash/Incredible PBX implementation. His method relies on the SipToSis gateway application (read the SipToSis how-to for more generic setup instructions).

If you are only wanting to receive Skype calls, you can transfer them to your PBX via Tropo developer account, free.

I decided to add Skype to my home PBX last weekend, and chose FreeSWITCH’s mod_skypopen as the connector.


  • The PBX is Asterisk 1.4, and I’m holding steady on this version until 1.8 becomes a bit more stable. Thus, because I want to use Google Voice,
  • FreeSWITCH is already in place, on the same machine, as a gateway to Google Voice. It has been working well in this role.
  • mod_skypopen requires only the FreeSWITCH module and Skype. No extra connector software is required.
  • It works and was easy to install!

Asterisk 1.4 continues to be a rock-solid choice and is supported (security patches will be provided) for another year. For me, the only killer feature that 1.8 provides is Google Voice connectivity. There are bunch of other new features too, but I don’t need them. Stick with that which works, until you need the features or the support is gone.

Back to mod_skypopen. There are basically two ways to install this to FreeSWITCH. The easy way, and the hard way.

Today, the easy way. If you visit the FreeSWITCH wiki page on mod_skypopen, you’ll find some notes on building and installing the module. If you’re running a fairly standard install of FreeSWITCH that utilizes the autoload_configs structure, you can follow through the wiki instructions to build the module and the fake audio driver and then run the Perl installer found in the source directory to automatically build out your configs, configure the Skype client(s) and set up a shell script to get everything going at startup.

If, like me, you have set up FreeSWITCH only to act as a feature gateway for Asterisk, and have aggressively minimized the configuration, you’ll want to avoid the installer script and perform the necessary steps by hand. More on that in the next posting: “the hard way.” (Not really that hard.)

6 thoughts on “Skype gateway with FreeSWITCH’s mod_skypopen”

  1. Hi,
    I would like to thank you for your how to on this topic. Very helpful.

    I have one question if you will be able to answer. I have Skype calling from FreeSwitch and it works great. The issue I am running into is when the User the server is trying to call is offline. Under normal circumstances, the call would just fail and busy tone would come up (Already tried that), but when the user has Voicemail service on their account, FreeSwitch server keeps ringing. After you hang up, the called user gets a blank voicemail. You don’t hear the voicemail system, it just keeps ringing.

    Any help would be appreciated.


  2. Thanks for the update and the general idea. I will wait for your more comprehensive notes. In the meantime, I will also try my luck on a spare machine based on your comments.

  3. I’m embarrassed to say it, and it was bound to happen, but I can’t find my notes on it. I’ll have to redo the thing from scratch to figure it out. At this point I can give you the general idea though. I followed the instructions for setting up mod_skypopen on, except when I got to the script I simply broke it down and ran individual components from it. The reason: my FreeSWITCH config is barebones just for Google Voice and Skype, so the typical FreeSWITCH conf directory structure isn’t there. When I got to the part about the XML config files I added the sections by hand to my freeswitch.xml file.

    Everything worked out OK that I remember; you just have to go through each step and verify along the way.

    I will find a block of time to get some more comprehensive notes written up before too long.

  4. Thanks indeed. When I saw there was not a comment for so long, I had a feeling that you would have come to that conclusion. I have been using Skype in my Asterisk boxes with SipToSis and have plenty of problems. I was actually looking out for a solution like this.

  5. Thanks for the compliment, Ranga, and sorry I let this one go. No one commented and I thought maybe no one was interested in this topic right now. I’ll gather my notes and get an entry posted.

  6. Hi Bill: I follow your blog religiously and use it in my Asterisk PBXes in two continents. I am eagerly awaiting “the hard way”. Freeswitch as a gateway to Google Voice is running great on these servers. Thanks and looking forward. Ranga

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