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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: 2.19MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 2.63

sipX Solution Summary

Image

sipX – The open source SIP PBX for Linux

Original Website: http://www.sipfoundry.org
SIPfoundry Wiki: http://sipx-wiki.calivia.com
Project updates: freshmeat

The sipX solution is an open source (LGPL license) Enterprise SIP PBX complete with voice mail and auto-attendant. Natively built on standard SIP, sipX is a full SIP proxy with all the architectural advantages that entails. In it's latest release the sipX proxy (2.7) is capable of processing 50.000 BHCC (busy hour call completions) on a single CPU P4 server, allows Web-based and remote management of large populations of phones and users, and fully embraces a distributed architecture. RTP media traffic does not go through the sipX server but is routed directly between end points. Also, sipX is not burdened with transcoding and other codec related operations as with SIP this is negotiated between phones and gateways directly.

Therefore and with all these attributes in mind, sipX is not only deployable as a full PBX but can also be used as a high performance Enterprise toll-bypass SIP router. It combines all common calling features, XML-based SIP call routing, forking, voice mail and auto-attendant, Web-based configuration, as well as integrated management and configuration of the PBX and attached phones and gateways.

sipX is a modular server based solution that runs on standard Linux. sipX does not require any additional hardware as it interoperates with any SIP compliant gateway, phone or application.

sipX is a native SIP communications solution strictly following and implementing all the relevant SIP IETF standards.

Summary of Available sipX Resources

Overview


sipXphone resources, downloads and documentation

sipXezPhone ("sipX easy phone") resources, downloads and documentation

Applications and Features

Documentation



Test reports

Supported Platforms

SIPfoundry and sipX in the news




Created by fm77c, Last modification by Eric Chamberlain on Tue 27 of Feb, 2007 [20:12 UTC]

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