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.
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