Showing posts with label Telephony Engine. Show all posts
Showing posts with label Telephony Engine. Show all posts

Thursday, August 6, 2009

FreePBX v3 Developers Preview Announced.

FreePBX v3
Folks at FreePBX has announced the availability of FreePBX v3 Developer preview and invites all who those do testing to iron out the system for rest of the users.
In addition to improving already fantastic PBX system by taking in the user feedback and real world deployement information, FreePBX has fused the TCAPI project and included some contributions from the Bandwidth.com Phonebooth Project. The team has also picked picked Kohana, Doctrine and jQuery as core components to realize the systems design goals.
From the get go, the project has some great goals including but nor limited to;
  • Make FreePBX truly telephony engine independent
  • A well thought out modular design building on the past 5 years of experience
  • Use of a true MVC framework
  • Choose best in breed components and build on top of them
  • Excellent documentation to encourage developer involvement
  • Easily Skinnable
You will find documentation and examples on FreePBX wiki as well as a fully functioning CD, ISO image for download coming soon that will help all of us to get started in evaluating and contributing to FreePBX V3 project!

FreePBX V3 Announcement

Wednesday, August 6, 2008

YATE 2.0, Has Been Released.

Yate 2 has been released after going through Alpha and beta cycles testing. But it comes at a time when the Open Source telephony is going through some changes in the SIP frontier at the moment. As we reported OpenSER became Kamailio and got a new fork called OpenSIPS.

YATE 2.0's new features include analog lines, robbed bit, SS7, passive recording on E1/T1 and analogic, clustering, high availability, MGCP, and Jingle. libpri was replaced with ysig. In YateClient, jingle support was added and the interface was improved. The GTK interface was replaced with a Qt interface.

  • Support for more hardware interfaces and protocols - added SS7, analogic support, RBS, better ISDN, passive recording
  • Clustering, balancing and failover support, Linux-HA integration
  • Improved client functionality - switched from GTK to Qt
  • Easier involvement of the Yate community - swtched from CVS to SVN
  • Added MGCP and Jingle support
Yate for me looks to be next-generation telephony solution. YATE is focused on Voice over Internet Protocol (VoIP) and PSTN, at the moment but it is capable of being easily extended and Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine. Also taking into the concideration that YATE is scalble with clustering which most of my favorite Telephony Engines are still trying to achive. So Keep an eye on Yate. Your calls may travel through this system somewhere on line.
You can download, and learn more about Yate at the YATE Project.