FreeSWITCH

Here is a little bit of information on how some companies are starting to use FreeSWITCH in production.

ClueCon was last week and we are all pretty tired. =D Thanks to everyone who attended and we look forward to seeing you in Aug 2008! A lot of great information was released including the new OpenZAP library that provides zaptel and sangoma ... [More] support for FreeSWITCH on both analog and ISDN. The project is still new so we need to do a lot more development and testing but we already have basic functionality underway. Visit our IRC channel #freeswitch on irc.freenode.net or see http://fisheye.freeswitch.org/browse/OpenZAP for details. Moshe Yudkowsky also provided this informative report on his blog: http://www.oreillynet.com/etel/blog/2007/06/cluecon.html   [Less]

I wanted to remind everyone that ClueCon is in a few weeks and if you plan to attend you should reserve your place now. Sangoma will be giving away a few FREE T1 and analog cards to some lucky attendees Zap Micro will be giving a FREE 4 port ... [More] analog card with 1 FXO and 1 FXS module to all attendees. Rhino will also be giving away a FREE Cards to some lucky attendees.  [Less]

A while ago I made this conference application for Asterisk 1.2.  Since I don't use it much these days, I thought I'd share it with everyone so download it. You can also just build it right from the net with the astxs utility I created (included in ... [More] the asterisk distribution). What? Isn't this the FreeSWITCH homepage? you ask. Well, I did my fair share of Asterisk development before I decided to write FreeSWITCH. In fact, I'm still the #3 most decorated developer in thier Karma Hall of Fame even though I have been busy for almost a year and a half doing development here. It supports a bunch of features like: silence supression, playing files, and a bunch more things you can do with the FreeSWITCH conference (but not all of them =D) "lock", "unlock", "mute", "unmute", "kick", "mark", "list", "killsound", "play", "dial", "admin", "unadmin", "vol", "silence", "verbose", "dtmf" To install it right from the net follow this simple instruction. From the 1.2 source tree, where you normally type make, execute this command: export ASTSRC=`pwd` perl ./contrib/scripts/astxs -install http://www.freeswitch.org/asterisk_stuff/app_confcall.c Also get the config from this url: http://www.freeswitch.org/asterisk_stuff/confcall.conf [Less]

FreeSWITCH's Open Source Softswitch Controls TelcoBridges' Carrier-Grade Telecom Platform to Route PSTN Calls to Truphone WiFi Subscribers Montreal, Quebec, Canada (PRWEB) June 5, 2007 -- TelcoBridges and ... [More] FreeSWITCH announce that Truphone has selected FreeSWITCH and TelcoBridges' proven, reliable and highly redundant carrier-grade telephony platform to enable VoIP calls on mobile phones. Truphone is a mobile internet network operator that brings VoIP to mobile phones via WiFi using SIP. The three companies have collaborated closely to port FreeSWITCH's open source telephony application code to TelcoBridges' hardware platform and have adapted it successfully to fulfill Truphone's requirements for a media gateway, bridging calls between the Internet, using VoIP, and the PSTN. [Read Entire Story] "Truphone selected TelcoBridges' enabling technologies to build the carrier-grade PSTN media gateway they need. After careful study of the available options, it was TelcoBridges' innovative technologies and carrier-grade architecture, coupled with FreeSWITCH's open source telephony applications, that were chosen to realize this key portion of their network," said Gaetan Campeau, President, Founder and EVP Sales, TelcoBridges Inc. "Truphone is an innovative mobile VoIP company requiring carrier-grade solutions to deliver free or VoIP-rate calls from mobile phones. Truphone works wherever there is access to the Internet via WiFi," said James Body, Network Director of Truphone: "Scalability, reliability and support for the AMR voice codec were the key decision criteria for Truphone when selecting our media gateway solution. We found all this with TelcoBridges' platform." The PSTN media gateway is a key element for the bridging of PSTN calls to Truphone's WiFi subscribers. Calls originating from the PSTN must be transcoded from G.711 to the AMR voice codec and bridged to IP. The media gateway uses SIP signaling to establish and tear down the VoIP connections on the IP network and uses SS7 signaling to initiate and release calls on the TDM side. "We designed FreeSWITCH from the beginning to be able to withstand the burden of media and transcoding but with the foresight to be able to control devices that can do it for you. TelcoBridges is the perfect compliment to our software and a prime example of how to use it in a high-density environment," said Anthony Minessale II, creator of FreeSWITCH. "The selection of FreeSWITCH and TelcoBridges to build this gateway adheres perfectly to Truphone's approach to building carrier-grade networks using open source and off-the-shelf technologies," added Body of Truphone. About TelcoBridges TelcoBridges is a disruptive innovator paving the way for solutions providers and OEMs to secure a significant share of the growing market for enhanced voice and video services. TelcoBridges' customers win more business by building on the unrivaled scalability, flexibility and reliability of our telecom platform. TelcoBridges' carrier-grade technologies are deployed in over 30 countries with some of the largest operators in the world. About Truphone In mid-2006, Truphone was first to introduce true internet telephony [VoIP for mass market mobile phones, enabling free and low-cost calls via Wi-Fi and the Internet. In December 2006 Truphone was selected as a Technology Pioneer for 2007 by the World Economic Forum and, at the end of 2006, received the biggest Series-A venture capital investment of the year for a European technology company. In March 2007 the company was listed in the Red Herring 100 Europe, which recognises Europe's 100 most promising companies driving the future of technology. For more information visit www.truphonepressoffice.com and www.truphone.com. About FreeSWITCH FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice, chat and other communications driven products scaling from a softswitch down to a softphone. It can be used as a simple switching engine, a media gateway or a media server to host IVR applications using many scripting languages to control the callflow. For more information visit www.freeswitch.org or join us at ClueCon '07 Telephony Developer Conference June 26-28 in Chicago, IL - www.cluecon.com . For further information, please contact: TELCOBRIDGES Inc Marc St-Onge Director, Marketing TelcoBridges Inc. Tel: 1 450 655 8993 x135 Fax: 1 450 655 9511 Website: www.telcobridges.com    Jim Tirjan Sales Director TelcoBridges Inc. Tel: 1 408 374 1004 Fax: 1 408 374 1519 Website: www.telcobridges.com FREESWITCH Michael Jerris FreeSWITCH Tel: 1 360 227 4889 Website: www.freeswitch.org [Less]

FreeSWITCH's Open Source Softswitch Controls TelcoBridges' Carrier-Grade Telecom Platform to Route PSTN Calls to Truphone WiFi Subscribers Montreal, Quebec, Canada (PRWEB) June 5, 2007 -- TelcoBridges and FreeSWITCH announce that Truphone has ... [More] selected FreeSWITCH and TelcoBridges' proven, reliable and highly redundant carrier-grade telephony platform to enable VoIP calls on mobile phones. Truphone is a mobile internet network operator that brings VoIP to mobile phones via WiFi using SIP. The three companies have collaborated closely to port FreeSWITCH's open source telephony application code to TelcoBridges' hardware platform and have adapted it successfully to fulfill Truphone's requirements for a media gateway, bridging calls between the Internet, using VoIP, and the PSTN. "Truphone selected TelcoBridges' enabling technologies to build the carrier-grade PSTN media gateway they need. After careful study of the available options, it was TelcoBridges' innovative technologies and carrier-grade architecture, coupled with FreeSWITCH's open source telephony applications, that were chosen to realize this key portion of their network," said Gaetan Campeau, President, Founder and EVP Sales, TelcoBridges Inc. "Truphone is an innovative mobile VoIP company requiring carrier-grade solutions to deliver free or VoIP-rate calls from mobile phones. Truphone works wherever there is access to the Internet via WiFi," said James Body, Network Director of Truphone: "Scalability, reliability and support for the AMR voice codec were the key decision criteria for Truphone when selecting our media gateway solution. We found all this with TelcoBridges' platform." The PSTN media gateway is a key element for the bridging of PSTN calls to Truphone's WiFi subscribers. Calls originating from the PSTN must be transcoded from G.711 to the AMR voice codec and bridged to IP. The media gateway uses SIP signaling to establish and tear down the VoIP connections on the IP network and uses SS7 signaling to initiate and release calls on the TDM side. "We designed FreeSWITCH from the beginning to be able to withstand the burden of media and transcoding but with the foresight to be able to control devices that can do it for you. TelcoBridges is the perfect compliment to our software and a prime example of how to use it in a high-density environment," said Anthony Minessale II, creator of FreeSWITCH. "The selection of FreeSWITCH and TelcoBridges to build this gateway adheres perfectly to Truphone's approach to building carrier-grade networks using open source and off-the-shelf technologies," added Body of Truphone. About TelcoBridges TelcoBridges is a disruptive innovator paving the way for solutions providers and OEMs to secure a significant share of the growing market for enhanced voice and video services. TelcoBridges' customers win more business by building on the unrivaled scalability, flexibility and reliability of our telecom platform. TelcoBridges' carrier-grade technologies are deployed in over 30 countries with some of the largest operators in the world. About Truphone In mid-2006, Truphone was first to introduce true internet telephony [VoIP for mass market mobile phones, enabling free and low-cost calls via Wi-Fi and the Internet. In December 2006 Truphone was selected as a Technology Pioneer for 2007 by the World Economic Forum and, at the end of 2006, received the biggest Series-A venture capital investment of the year for a European technology company. In March 2007 the company was listed in the Red Herring 100 Europe, which recognises Europe's 100 most promising companies driving the future of technology. For more information visit www.truphonepressoffice.com and www.truphone.com. About FreeSWITCH FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice, chat and other communications driven products scaling from a softswitch down to a softphone. It can be used as a simple switching engine, a media gateway or a media server to host IVR applications using many scripting languages to control the callflow. For more information visit www.freeswitch.org or join us at ClueCon '07 Telephony Developer Conference June 26-28 in Chicago, IL - www.cluecon.com . For further information, please contact: TELCOBRIDGES Inc Marc St-Onge Director, Marketing TelcoBridges Inc. Tel: 1 450 655 8993 x135 Fax: 1 450 655 9511 Website: www.telcobridges.com Jim Tirjan Sales Director TelcoBridges Inc. Tel: 1 408 374 1004 Fax: 1 408 374 1519 Website: www.telcobridges.com FREESWITCH Michael Jerris FreeSWITCH Tel: 1 360 227 4889 Website: www.freeswitch.org [Less]

I’ve been working on FreeSWITCH for nearly 2 years and on the dawn of our first release I wanted to take some time to share the story behind the software project and provide a glimpse of what’s to come. This story will also appear in the first issue ... [More] of OST Magazine so get a copy, it's FREE!   [Read The Entire Article] The idea to write FreeSWITCH was first conceived in the spring of 2005 during a weekly Asterisk developer conference call. At the time, I was heavily contributing code and features to the pending 1.2 release of Asterisk and we were brainstorming on what to do for the future. We all agreed that there were serious limitations with the current Asterisk code base and we had to face the issue that fixing some of the problems would sacrifice existing features and may take a great deal of time. One idea was to branch the code so there was a gutted version of the code alongside the one everybody was used to. This new branch could be worked on without the pressure of disrupting the vast Asterisk user base. The problem was that the developers did not want to split their efforts on two versions of the same code and have to move bug fixes and tweaks back and forth between the two branches. My suggestion was to start “Asterisk 2.0” from scratch in a separate repository and spring it on the public when it was ready. I think that idea intrigued some of the developers, but ultimately the reality set in of taking that much time to start over and they eventually lost interest. I, however, did not. It was apparent that I was the only one who was serious about this idea and I spent the next several days daydreaming about how I could design a new telephony application. After a few days, I could no longer resist the urge to make a new directory on my PC and start editing a blank choir.c. Yes, Choir! That was the first name I chose because I envisioned the various communication components working together in unison like a choir singing in perfect harmony. I spent the next 5 days taking any ideas that came to mind and organizing them into a handful of files. That first effort hardly resembled a telephone switch. I knew that I needed to create a stable core that I wanted it to be cross-platform so it was easy to come up with a basic subsystem based on Apache’s APR library that could load shared modules and sit idly on the screen waiting for the shutdown command. This was the way the code would stay for the next 6 months. The lesson I learned over those 5 days was that it’s not exactly easy to write an application of this scale. My original goal was to pick up where Asterisk left off and improve upon the idea, but the more I thought about it, the more I realized that what I wanted to improve were pure fundamentals in design and functionality. I came to the conclusion that Asterisk simply was not meant to do some of the things that I wanted it to do because it was not the right type of software for my demands. This is when I knew that I was not going to write another PBX. I was in fact going to write an entirely different class of application. I spent the next several months leading up to the first annual ClueCon conference debating how to design Choir properly. I wanted to make sure that I had the right plan before I started writing any more code. At this point I was still doing a fair share of Asterisk development and I put some of my ideas to the test with some of my 3rd party modules. I also asked many of my collogues for input and spent countless hours debating how to do things “right” which is often in the Telephony world, a matter of perspective. At ClueCon that year, I had a chance to meet some of the most influential developers in the Telephony industry and I returned home with a great deal of inspiration. At the same time, tension was breaking out in the Asterisk camp because several of the developers had decided to fork the project. This new project dubbed OpenPBX (now Call Weaver) caused somewhat of a rift in the Asterisk community. Initially OpenPBX adopted many of my 3rd party Asterisk modules into their new code base and consulted me for tips on what to change to improve stability. At one point they even approached me to see how I was doing with my new project with the idea that maybe we could rekindle the old plan of a side-by-side rewrite. That idea didn’t pan out but I think the monumental moment came when someone asked me “How long would it take to get it making calls?” I didn’t know, so I decided to find out. The short answer was “1 week”. Making a call was a small victory in comparison to what I needed to accomplish. There was a lot to be done and my unimpressive feat was hardly enough to attract much attention. The good news was that the project was finally underway! I worked on the code in solitude for 3 straight months trying to get something presentable for the public eye. During that time the name changed to Pandora and eventually to its ultimate name, FreeSWITCH. In January of 2006 we were open for business with a public SVN repository and a mailing list. At that time we had a hand full of modules and a very small feature list but we had a working core that compiled and ran on several UNIX varieties including Mac OS X, Linux and BSD it also worked on Microsoft Windows as a console application. As time progressed we picked up some momentum, taking a break or two along the way to make a few mistakes and start over on a few modules. In all we had started four different SIP endpoint modules before finally deciding on Sofia SIP. We had an alarming five different RTP stacks before finally writing our own. In that time I also developed mod_dingaling to interface with GoogleTalk, a feature-rich conference bridge module. For the first year I mainly focused on making the core as stable as possible, providing several external interfaces to facilitate development such as embedded javascript for IVR, an XML-RPC interface and a TCP based socket interface for remote control and event monitoring. At the second annual ClueCon, on a day that seemed like an eternity from my daydreaming days, I presented FreeSWITCH to my fellow developers for the first time. Nearly nine months after that presentation, 2 years since that first idea, and a month before the 3rd annual ClueCon, we are at a BETA phase of development and soon will be releasing FreeSWITCH 1.0. We have attracted some brave developers along the way who are already using it in production and providing us with the critical feedback we need to make our first release a success. The final piece of the puzzle before the release is an interface to my new open source TDM abstraction library called OpenZAP. The new mod_openzap, driven by this BSD licensed code, will replace the current Sangoma specific mod_wanpipe and provide support for Sangoma cards as well as several other varieties of TDM hardware as new modules are developed. OpenZAP will also provide a simple interface to analog and ISDN signaling. The Idea behind OpenZAP is that it will be possible for applications to use a single API interface to control any TDM hardware supported by the library. OpenZAP provides a way to normalize all the various features so if one card lacks a particular feature, it can be implemented in software either within the main OpenZAP library or within the vendor-specific interface between the library and the native hardware API. It almost seems impossible that we managed to accomplish so much in such a short time, but I think in the end, the old adage rings true “Necessity is the mother of invention”. There is still a long way to go but I would like to take this opportunity along the way to thank all of the people who helped get us this far. Below is a list of all of the people currently in our growing AUTHORS file. Anthony Minessale II (That’s me!) Michael Jerris (Our Invaluable build master and cross-platformologist. [Yeah, I made that word up]) Brian K. West (Our devoted MAC guru. We would be nowhere without his help.) Joshua Colp (Helped make the first SIP module which is now on the cutting room floor.) Michal "cypromis" Bielicki (He’s been around since day 1. Thanks for believing!) James Martelletti (He integrated mono into FreeSWITCH.) Johny Kadarisman (Helped us get the python module working.) Yossi Neiman (Wrote mod_cdr to gather call details.) Stefan Knoblich (Helped us with our journey through SIP.) Justin Unger (Found a good many bugs we didn’t want to have.) Paul D. Tinsley (SIP presence and other great ideas.) Ken Rice (How did his name get here?, JK A huge help with testing and patches.) Neal Horman (A humongous help with the conference module.) Michael Murdock (Our friend from CopperCom with tons of feedback and patches.) Matt Klein (Tons of SIP help and help making sure we work on FreeBSD.) Justin Cassidy (The techie behind the curtain making sure stuff works.) Bret McDanel (The bravest of them all trying most things before anyone else and finding all the hidden bugs, I mean easter eggs!) [Less]

I would like to announce that FreeSWITCH will be entering into a BETA status within the week so we can produce a series of release candidates which will ultimately produce a formal release by the end of the summer or sooner if possible. Our software ... [More] is growing rapidly and we've come a long way from our modest initial public release in January of 2006. I hope everyone enjoys the opportunity to participate in the development process which is one of the best benefits of open source software in my opinion. Not only do we have a few finishing touches to put on the code, we also have to institute a version policy, make sure the WIKI is accurate and, of course, find all the bugs so we can focus our energy on forward development and stay away from nasty unresolved bugs. In order to make the debugging process successful we ask that everyone use our jira tracker for all bug reports http://jira.freeswitch.org, feature requests, or feature contributions. We are glad to help but it's beginning to be more than we can handle in real-time so we really need to document issues on the tracker so we won't forget! We also ask anyone who receives help to please pay it forward and document it on the WIKI http://wiki.freeswitch.org. There are plenty of logistical and clerical necessities to make FreeSWITCH a success so anyone who is interested in helping out by being a bug marshal or managing the svn for one of the modules or any way you think you can help, please mail the dev list http://lists.freeswitch.org or visit us on IRC. Please also feel free to start a new page on the wiki with your irc nick and your paypal address or wishlist urls so when you help someone they can show their gratitude properly. Thank you all for participating and making FreeSWITCH a fun project! [Less]

There is an update to the pre-release of the N800 version of FreeSWITCH at http://www.freeswitch.org/downloads/n800/ The new version includes up to date code and mod_speex built just for arm and the new jitterbuffer to be used with mod_alsa (a clone ... [More] of portaudio that uses libasound just for the N800) There is also a mini-webserver running on port 8080 so http://127.0.0.1:8080/api/alsa with l/p freeswitch/works will give you a small web based softphone of sorts (enter 888 and press dial to call our developers conference) [Less]

This week we added an optional jitter buffer to the RTP stack that you can turn on with a channel variable from your dialplan. On an inbound call for use on the inbound channel: (setting this before the call is answered is mandatory) <action ... [More] application="set" data="jitterbuffer_msec=180"/> <action application="answer"/> Or to set it on the subsequent outbound call: export sets a variable on both the current channel and on any channels it creates, the 'nolocal:' disables setting it on the current channel and only sets it on the subsequent outbound channels <action application="export" data="nolocal:jitterbuffer_msec=180"/> <action application="bridge" data="sofia/default/[email protected]"/> [Less]