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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: 0.54s
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  • Database queries: 33
  • GZIP: Disabled
  • Server load: 0.75

FreeSwitch

PLEASE VISIT http://www.freeswitch.org for more up to date info.



What is it?

FreeSWITCH™ is an open source communications platform. FreeSWITCH™ is a library which ships with a small executable that loads the library, launches the core, and performs the various tasks that are defined by the modules. In its base form FreeSWITCH™ is a soft-switch or PBX telephony application, not completely unlike Asterisk but capable of handling thousands of simultaneous calls.

FreeSWITCH™ makes it possible to build a softphone, PBX system, soft-switch, or interface with other open source PBX systems such as OpenPBX.org, Bayonne, YATE or Asterisk. It can also be used to build a voip switching platform uniting various technologies such as SIP (using the Nokia Sofia library), H.323, IAX2, LDAP, Zeroconf, XMPP / Jingle etc.

As a library FreeSWITCH™ can be used by developers to enable switching in their custom applications. FreeSWITCH™ can be launched not only from a C application, but also via PHP, Perl, or a variety of other languages.

FreeSWITCH™ is written in C, built from the ground up (not a fork of another code base). It is designed to take advantage of as many existing software libraries as possible. It has a modular, extensible architecture, with few and necessary functionality in core ([libfreeswitch]]) with optional modules to do the rest.

FreeSWITCH runs on Windows, Mac OSX, Linux, *BSD, and other Unix flavours.

It is licensed under the MPL.

Created by admin, Last modification by James Finstrom on Mon 03 of Mar, 2008 [05:29 UTC]

Comments

Comments Filter

free canadian 514 montreal did number

by habib hayek on Saturday 23 of February, 2008 [17:05:35 UTC]
plz any one can tell me where i can find a free did number strting 514 incoming and outgoing only 514 unlimited??? i need it i am in montreal and i want to send one for my parents.



and seconde do u have any idea if i have a free phoneline did number how to use it with xlite or ata not with their softphone???

any help

by habib hayek on Thursday 10 of January, 2008 [13:48:28 UTC]
hi my name is habib and looking to this informations plz:
first i am new user of sip and voip i am in canada and i want to call our office out of canada in dubai for example if i have an internet connection or a dsl connection in my office there i ahve a ata device here in canada and dubai and i have sip software on both computer what do i need to call dubai over voip or sip and i want to connect my land line in dubai to my device or modem if my computer so i can make calls to dubai via voip or sip but i will pay local call not international because i am using my dubai land line is that posssible or i am dreaming i hope i can find something thx for u all plz send me email to hayik@hotmail.com

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