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Sat 17 of May, 2008 [00:55 UTC]

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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
  • Execution time: 0.44s
  • Memory usage: 2.19MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 0.61

YATE

YATE - Yet Another Telephony Engine

Yate it's a softswitch with PBX features, which can be disabled. Due to fact that is very flexibile it can be integrated with other services like Web. It runs under Linux, BSD and Windows.

Page Contents

Official website


Mailing list


Latest stable release



CVS repository


Latest News


  • 4 Feb 2008 - Yate 2.0.0 alpha 2 released. New routing module allows sending ENUM routed or forked calls to numbers of registered phones.

  • 3 September 2007 - Yate 1.3 released. Minor fixes and improvements mainly in client and SIP.

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

About Yate version 2


   * Support for more operating systems and hardware architectures
   * Better integration in the target operating systems
   * Easier interoperation with database schemas
   * Support for more hardware interfaces and protocols
   * Clustering, balancing and failover support, Linux-HA integration
   * Improved client functionality
   * Easier involvement of the Yate community 
   * MGCP for client-server and gateway control support added.
   * SS7 support added
   * ISDN new stack with passive recording support
   * RBS and analogic cards support 

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




Supported telephony hardware

  • Sangoma
  • Digium
  • OpenVox
  • Rhino Equipment
  • ZapMicro


Downloads


Support



Application Examples



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 diana on Tue 25 of Mar, 2008 [13:03 UTC]

Comments

Comments Filter

yate source code needed for windows

by SAQIB IRSHAD on Saturday 19 of January, 2008 [20:52:12 UTC]
Hi,
I need yate source code for windows. Can any body help me.

Regards,
Saqib

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