FreeSWITCH

The first round of improvements are in from our 1.0 line! There are several new features including: Open Source ASR with mod_pocketsphinx Open Source TTS with mod_flite Improved ACL with user matching. Enhanced SNOM support. And many more!

blogs.zdnet.com — When the time came for a new PBX, Brian Snipes chose to do something a bit unconventional. The IT manager at law firm Hare, Wynn, Newell, and Newton LLP didn’t purchase a commercial PBX, nor did he settle on the open source market ... [More] leader, Asterisk. No, Snipes chose to become the first enterprise to deploy the new open source IP PBX, FreeSwitch. read more at digg.com [Less]

I wanted to let everyone know that ClueCon 2008 is just under 3 weeks away! Everyone who is planning to attend: Please make sure you are registered and have your rooms booked. We have an overflowing schedule of speakers but we could still probably ... [More] squeeze in a few more lightening talks if anyone wants to speak. This year is shaping up to be the best yet with developers or reps from most of the popular Open Source VoIP applications represented including: FreeSWITCH(tm) (naturally) ;) OpenSER OPAL sipX/SIPfoundry Asterisk Yate Bayonne SofaSWITCH We also have reps or developers from several hardware vendors including: Sangoma PIKA Digium Dialogic SNOM OCTASIC We also have 2 professional voice firms attending including: GM Voices (creator of the famous Cepstral Callie and all the FreeSWITCH(tm) prompts) Allison Smith (The voice of Asterisk(tm)) http://www.cluecon.com [Less]

Sip Foundry announced today that they are using FreeSWITCH in their next release.   sipXecs adds a fully featured conferencing server. That conferencing server is based on FreeSWITCH and over the last couple of months a very good cooperation ... [More] emerged between the two respective developer communities. There are lot's of synergies between FreeSWITCH and sipXecs where FreeSWITCH provides a very powerful state-of-the-art media framework. sipXecs adds ease-of-use Web based management, cluster management, and all the other services provided by sipXecs already. see sipfoundry.org for details: http://www.sipfoundry.org/component/option,com_ezine/Itemid,82/task,read...     [Less]

I was reading this story over on Slashdot about how SPIT (spam over internet telephony) will be worse than SPAM. I find it hard to believe that will be the case. If you use software like FreeSWITCH to frontend all your phone calls it can give you ... [More] the ability to knock out* the people you don't want to talk to in a heart beat.   What is your take on this?   * No you can't literally knock people out via telephone... yet!   [Less]

I'll be going on vacation till June 21st so our production may lag just a teenie bit. Don't worry, we should be releasing 1.0.1 while I'm gone and with the slow down it gives our community time to prepare for the upcomming ClueCon 2008 http://www.cluecon.com    

I just wanted to take the time to defend our use of XML if I can. =D Originally, FreeSWITCH used the same .ini format that Asterisk uses in all of it's config. Actually the interface to parse .ini still exists and a module writer is able to use it ... [More] if he pleases. The reasoning for XMLizing what we chose to XMLize becomes more clear when you begin to scale the system. FreeSWITCH parses it's XML registry when it first starts and keeps it in memory. This is one big entity that can be navigated similar to a file system. There are top level major sections: configuration, dialplan, directory and phrases. All of the bits and pieces of these sections are exploded out onto the disk in the default arrangement so you can edit the portion of the document you need and it also allows you to insert small XML representations of a single entity such as a SIP UA, a user on the system, the configuration for what modules you want to load. All of these files will be concatenated in the end into 1 big XML document that the entire core and its modules can access with a common API that gives you the entities as a tangible object that can be extended without more code. Now for the interesting part. It's possible to bind callback functions to certain sections of the XML registry so that when code tries to access a particular section your handler can conjure the XML on-demand any way it wants and deliver it back in place of whatever is in the default document. For instance, you can bind your dialplan and user directory sections to mod_xml_curl, A callback that translates XML lookups to and from a nearby HTTP server. When the call hits the XML dialplan the cURL handler kicks in and posts a request to a custom CGI. The CGI gets post data that is identical to a web form being posted with a few dozen facts about the channel that you can use to determine where the call was intended to go and who's calling etc. The CGI then creates a small snippet of XML that will satisfy the request and returns it to FreeSWITCH who will parse it as if it was part of the static config. The same can be done when someone registers to SIP or tries to check their voicemail or when the config is called up during load. Trust me. I am far from XML's biggest fan, but I do feel it's actual uses are lost in a sea of contrived forced solutions that give it a bad name. To serialize complex objects to and from a text format and to make markup that is easily parsed and generated are the real strengths of XML. The best used XML is XML nobody ever realizes is even there. Certainly that is not true completely with FreeSWITCH today but as we scale and more GUI's and config apps are made, the curve will be in our favor and most people besides only the truly advanced users will ever have to see the XML again. Caveat: We do have mod_dialplan_asterisk in tree that will allow your dialplan to look and feel more like the asterisk extensions.conf (using .ini) for a sort of cross-over feel. [Less]

It's been almost 3 years since the idea was hashed to write a new telephony platform. Today we have reached the milestone of 1.0.0. Thank you to everyone who helped make this release a success! Let today mark the rebirth of Open Source Telephony as Phoenix is born from the ashes of past failures!

It looks like Zaptel is changing it's name to DAHDI. http://digg.com/tech_news/Zaptel_Project_Renamed_to_DAHDI I wonder how that will change the landscape of the Zaptel interface going forward for legacy support of existing devices. We may need a ... [More] sugar DAHDI to help us port the changes. As we all know, Zaptel is the grand DAHDI of all hardware telephony interfaces and it will be a strange new world without it. [Less]

This will most likely be the last release before the upcomming 1.0 release next Monday! See the Download link above for details.