Welcome To FreeSWITCH
The World's First Cross-Platform Scalable FREE Multi-Protocol Soft Switch
FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.
FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX. The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.
We support various communication technologies such as Skype, SIP, H.323 and GoogleTalk making it easy to interface with other open source PBX systems such as sipXecs, Call Weaver, Bayonne, YATE or Asterisk.
FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.
FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 12, 16, 24, 32 or 48 kilohertz and can bridge channels of different rates. The G.729 codec is also available under a commercial license.
FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.
FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two.
Our developers are heavily involved in open source and have donated code and other resources to other telephony projects including openSER, sipXecs, The Asterisk Open Source PBX and Call Weaver.
a Spec Sheet is available on our Wiki.
Cal Leeming (foxx on IRC) was kind enough to join our weekly conference call yesterday to discuss a very interesting issue that apparently has at least one phone manufacturer in a bit of a panic. We're withholding details for now to give that manufacturer time to react. Expect a more detailed story here in a couple of days.
The FreeSWITCH Team is proud to announce the release of FreeSWITCH 1.2.14!
Available today via git, http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2, and the deb and yum repos.
This is a maintenance release to address several bugs that have been identified since the last release.
Also dont forget ClueCon Weekly Conference Call! Every Wed at 1PM EST! For more information on how to join see: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions
Bluebox-ng is an open-source VoIP/UC vulnerability scanner. It has been written in CoffeeScript using Node.js powers.
- RFC compliant
- TLS and IPv6 support
- SIP over websockets (and WSS) support (draft-ietf-sipcore-sip-websocket-08)
- SHODAN, exploitsearch.net and Google Dorks
- SIP common security tools (scan, extension/password bruteforce, etc.)
- REGISTER, OPTIONS, INVITE, MESSAGE, SUBSCRIBE, PUBLISH, OK, ACK, CANCEL, BYE and Ringing requests support
- Authentication through different types of requests
- SIP denial of service (DoS) testing
- SRV and NAPTR discovery
- Dumb fuzzing
- Common VoIP servers web management panels discovery
- Automatic exploit searching (Exploit DB, PacketStorm, Metasploit)
- Automatic vulnerability searching (CVE, OSVDB, NVD)
- Colored output
- Command completion
- It runs in GNU/Linux, Mac OS X and Windows
So this is yet another tool in the toolbox you can use to help test the security of your UC/VoIP installations.
As some of you may have heard by now, our very own Brian K West
(a.k.a. bkw_) is going to be marrying the love of his life, Gregory A
The ceremony will be held on October 7th in NYC. Please join with us all in congratulating him and his spouse to be.
For those of you who would like to help out with the costs of the
planning and ceremony, or would just like to send them a gift, his
paypal address is firstname.lastname@example.org. No gift is too small (or too large
If you're in, or around, NYC on the 6th or 7th, we'd entertain the idea
of having a FreeSWITCH Users' dinner somewhere in the area on one of
those two nights if there's enough interest. Feel free to email me
privately (at email@example.com) if you'll be in the area
and interested in congratulating them, in person, over dinner.
This handy tip just came by my desk. It's geared toward Asterisk, however the principles apply to FreeSWITCH - or any other software that employs the use of audio files. This blog post is useful to anyone wanting to learn more about these handy tools.
Welcome to the last week of July!
The big news from last week was announced on the weekly conference call: FreeSWITCH 1.2.12 was officially released. This is now the recommended production release of FreeSWITCH. We also had a simple FreeSWITCH programming how-to on using channel variables with examples from mod_valet_parking and mod_voicemail.
On this week's conference call we are glad to have with us long-time FreeSWITCH community member and code contributor Chris Rienzo. Chris has lots of experience with handling media for various applications, such as you'd use with mod_unimrcp. This week Chris will be discussing the theme "mod_spandsp - it's not just for faxing!" Mod_spandsp makes use of the powerful SpanDSP library and Chris will show us some of the interesting things that this makes available for FreeSWITCH users.
For those of you coming to ClueCon 2013: we'll see you next week! We are really looking forward to seeing everyone again. If you haven't registered for ClueCon or booked your room yet then we highly recommend that you do so right away. We hope to see you all at the can't-miss event of the season!
Talk to you next week - from Chicago!
We are pleased to announce that FreeSWITCH version 1.2.12 has been released! Ken's mailing list post details some of the changes and contains any follow up conversations regarding this release.
What's new in this release? Lots and lots of bug fixes. 1.2.12 is now the recommended FreeSWITCH version for production environments. As usual, test it with your configuration prior to migrating a production server.
As of this writing the CDN has been purged so you can download the latest source tarballs and RPMs. Naturally the git repo is tagged already. The DEB files are coming soon.
Thanks for all your help in making FreeSWITCH get better and better.
Welcome to July!
We've had lot's of great news lately with FreeSWITCH and now there's even more. Our friends at Yealink have reported on a very large VoIP installation in Spain that utilizes Yealink phones and FreeSWITCH servers. The complete writeup can be found here. We look forward to reporting on more success stories in the future.
With the recent announcement about FreeSWITCH 1.4 beta and WebRTC support there has been a lot of interest in just how to make this all work. On this week's conference call we will be taking a closer look at the code and showing you how to setup both sides of a WebRTC connection with FreeSWITCH. Please join usthis Wednesday at 1PM Eastern, 10AM Pacific for this interesting discussion.
We also had an interesting discussion on last week's conference call. Lorenzo Magani and Alexandr Dubovikov from the HOMER project gave us a tour of their new version 3.5 release. The HOMER SIP capture system is an excellent - and free - tool that can be put into production with any FreeSWITCH server. HOMER is an invaluable tool used for SIP diagnostics and troubleshooting and we highly recommend it for anyone using FreeSWITCH and SIP in a production environment.
Have a great week and we'll talk to you on Wednesday.
Flowroute issued a press release today that highlights their role in connecting FreeSWITCH's WebRTC platform to the PSTN. You can read the original press release here at the Wall Street Journal.
Try the Online FreeSWITCH WebRTC demo at https://webrtc.freeswitch.org
Thanks to Flowroute for supporting FreeSWITCH and ClueCon!