SIP trunking and remote worker support (far-end and near-end NAT traversal) with support for an optionally redundant media relay for media anchoring
New voice conferencing solution based on the FreeSWITCH media server. Thanks to Michal Bielicki and Pawel Pierscionek from VoiceWorks and of course the FreeSWITCH team (Anthony Minessale, Michael Jerries and Brian West). Conferencing solution supports HD Voice and offers dynamic Web based conference controls
sipXecs cluster management now allows creating a fully distributed system that is centrally managed
New IVR / Auto-attendant also based on the FreeSWITCH media server
Support for 64 bit CPU architectures for both Interl/AMD as well as PPC
A new integrated reporting solution based on Jasper reports
Source routing that allows call routing based on where the call originates (i.e. in a branch office)
Lot's of updated phone and gateway profiles with plug & play management (Linksys, Cisco (thanks to Sen Heng), Grandstream, Aastra, Polycom, Audiocodes, and others)
Improved operations including certificate management, log and snapshot management, patch and upgrade management, DNS/DHCP configuration assistance, backup/restore to FTP, time and DST management
Bulk provisioning of Counterpath Bria Professional phone
Click-to-call from the user portal
In addition, the release includes a lot of fixes of previously broken things, including improvements to the RLS based presence server, performance improvements in the proxy, and probably a few things I forgot. Thanks to everyone who participated and a special thanks to our test team led by Raghu.