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Wed 25 of Nov, 2009 [05:44 UTC]

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voip-info.org

Created by: system,Last modification on Wed 25 of Nov, 2009 [02:02 UTC] by redpepper

Welcome to the VOIP Wiki - a reference guide to all things VOIP

This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips & tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. However, the Wiki is primarily for information, not for advertising.

Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelines before you post.



NEWS





News Resources


Getting Started


Connecting VOIP to the PSTN and Cellular Networks


PBX and Servers - VoIP PBX and Servers

Popular choices - please do not alter this list, add new entries here
  • Asterisk: Open Source PBX
  • CallWeaver: Vendor independent, cross-platform open source IP-PBX, originally derived from Asterisk.
  • FreeSWITCH: Open source cross platform (*nix, Win, OSX) multiprotocol (SIP, IAX, Googletalk, Jabber, XMPP, WOOMERA/H.323, PRI/T1/E1) softswitch.
  • Kamailio: Flexible and powerful open source GPL SIP ( RFC3261 ) server with integrated TLS (former OpenSer)
  • Sip Express Router: High-performance, configurable, free SIP ( RFC3261 ) server
  • sipX Solution Summary: SIPfoundry - The SIP PBX for Linux (L-GPL)
  • 3CX Phone System: Windows PBX with free and commercial versions
  • Yate - open source softswitch/PBX, with E1/T1, ISDN, SS7, RBS, analogics, IAX, H.323, SIP, MGCP, Jingle (GoogleTalk), for Windows, Linux and BSD's
  • more...

Protocols - the language of VOIP


Markup Languages

  • Basic call routing and rules for UA's or VOIP serversCPL
  • IVR Presentation and dialog management: VoiceXML, CallXML
  • Call control / conferencing / call routing: CCXML
  • IVR / Speech recognition definition: SRGS
  • IVR / Speech synthesis definition: SSML
  • IVR / prompting / recording / conferencing / DTMF / Voice: CallXML

Traditional Telephone Network



VOIP Events and Conferences


Business Services


Resources


Suggestions and Questions


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by 377704497, Monday 23 of November, 2009 [17:22:00 UTC]
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333Cisco 7911G Configure

by gisvpn, Sunday 22 of November, 2009 [19:13:14 UTC]
Hello,

I have a Cisco 7911 with SIP11.8-3-1S firmware installed. I can access the Information page by simply typing in the IP Address of the phone into my web brower. I would like to configure the SIP settings on the phone - can this be done via a web interface. If so what is the URL what I would need to type in ?

Many thanks in advance.

Regards,

GISVPN
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by 377704497, Sunday 22 of November, 2009 [17:17:49 UTC]
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by 377704497, Sunday 22 of November, 2009 [17:15:07 UTC]
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by 377704497, Sunday 22 of November, 2009 [17:11:48 UTC]
222

333http://igbt-module.wikispaces.com/

by 377704497, Friday 20 of November, 2009 [10:32:31 UTC]
222

333Designating outgoing Trunks For Individual Phones

by estesvoip, Wednesday 04 of November, 2009 [21:36:33 UTC]
Is there a way to force a single "telephone"; i.e. extension, to use a particular provider/trunk each time some one dials out on that device? Is there a setting that can be added to extensions.conf or something? I know how to route based on what was dialed but cannot figure out how to router based on the number from which they are dialing.
222

333WYBECOM TALK : the first free CTI Solution

by wybecom, Wednesday 04 of November, 2009 [10:18:34 UTC]
TALK (Telephony Application Library Kit) is an open source solution providing all the elements to facilitate computer telephony integration using standard technologies. Embellished with an AJAX web based phone control, this solution can provide all of your users all the telephony features they want through a simple browser. From presence to directories via the call controlling, TALK is an effective way to link your phone system to your business applications.
Both modular and scalable, TALK adapts to your environment without technical constraints and access to source code guarantee the longevity of your infrastructure.


Download it now!

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333http://chinaimportexport.wikispaces.com/

by 377704497, Monday 19 of October, 2009 [17:36:45 UTC]
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333New Media Gateway Controller Simulator

by voipemulator, Friday 16 of October, 2009 [20:08:11 UTC]
http://voipemulator.weebly.com
VoipEmulator is a MEGACO signaling testing tool, provide developers and QA test engineers with the ability to perform sophisticated MEGACO (H.248) signaling functionality testing (Fax, T.38, 3WayCalling, Basic call...).

With VoipEmulator, you can easily emulate any Media Gateway Controller (Soft Switch) behavior, thereby increase interoperability with a large scale of VoIP implementations.