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Wed 01 of Aug, 2007 [09:12 UTC]

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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.39s
  • Memory usage: 2.18MB
  • Database queries: 31
  • GZIP: Disabled
  • Server load: 2.63

YATE

YATE - Yet Another Telephony Engine

Yate is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses.

Official website


Mailing list


Latest stable release


CVS repository


Latest News

  • 16 Apr 2007 - Yate 1.2.0 released. Added Jingle and XML support, PBX improved.

About Yate version 1

Yate version 1 is a direct result of the work on the Yate09 development versions.
We added features, made lots of improvments and fixed many problems.

The following notable features are available:
  • H.323 - using OpenH323 stack
  • IAX - using Yate's IAX stack
  • SIP - using Yate's SIP stack
  • Jingle - using Yate's XMPP and Jingle stacks (from version 1.2.0, works as another server's external component)
  • RTP - using Yate's RTP stack, works with the H.323, SIP and Jingle protocols
  • hardware support for Sangoma and Digium boards - only digital lines (ISDN) - using libpri
  • analog fax send or receive file in Linux (only from version 1.1.0)
  • audio codecs - G.711, GSM, iLBC, many other in pass-through mode
  • databases support - mysql and postgresql (all the other by using an external language)
  • routing from a file using regexroute
  • routing and authentication
    • from a database using register
    • from a file using regfile
    • from a RADIUS server
  • call forking and fallbacks
  • fallback routing from a database (starting with version 1.1.0)
  • accounting and, or billing
    • in a file using cdrfile
    • in a database using register
    • to a RADIUS server
  • conferencing - the number of participants is limited only by the server's hardware performance
  • customizable PBX for switching between calls, putting them on hold and initiating transfers and conferences
  • a skinnable, Gtk-2 based graphical client interface supporting many lines and accounts at once

Supported operating systems

  • FreeBSD
  • GNU/Linux
  • Windows
  • ucLinux

Supported telephony hardware

  • Sangoma
  • Digium
  • OpenVox

Downloads


Support


Licensing

Yate is licensed under the GNU General Public License (GPL) with an exception to allow linking with OpenH323 and PWlib, which are both licensed under MPL.

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Created by Florian Overkamp, Last modification by Balwinder S Dheeman on Sat 30 of Jun, 2007 [05:42 UTC]

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