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
- 2009-11-24 - MagicJack Users Able To Call FreeConferenceCall.com Again With Skype
- 2009-11-24 - Developing a deployment test plan to help your installations go smoother
- 2009-11-24 - 3CX Product Overview Webinar - Tue, Dec 1, 2009 1:00 PM - 2:30 PM EST
- 2009-11-24 - Jerasoft Development signs sales representative agreement with Intalisan UAE
- 2009-11-24 - TELES unveils environmentally friendly Green PBX
- 2009-11-23 - WebSpeedTechnology.com now offers FREE Trixbox & AsteriskNOW Virtual Appliances!!! http://www.webspeedtechnology.com
- 2009-11-20 - The Facts about FAX - Best practices for successful FAX deployments
- 2009-11-20 - asterCC release 0.13 and asterCC BOX 0.13, start callcenter from one CD
- 2009-11-19 - Nerd Vittles: Ever wanted to roll your own Asterisk ISO? Well, today's your lucky day!
- 2009-11-19 - VOIP Today Magazine launch the most challenging Competition in VoIP,Asterisk and security knowledge!.
- 2009-11-19 - Free G.729 acm codec driver for windows
- 2009-11-18 - OpenBTS: Open source GSM air interface application for Asterisk PBX!
- 2009-11-07 - A2Billing New released version v.1.4.2.1 with Automated Install - Install A2Billing in less than 1 minute.
- 2009-11-17 - Jerasoft Development will be participating with new version of Billbery VoIP billing software at the ITExpo East 2010
- 2009-11-17 - Latest Phone System Comparison Chart for Fall 2009 Released by CompareBusinessProducts.com
- 2009-11-17 - Junction Networks Offers Alternative to Blocking Numbers in Response to "Traffic Pumping" Practices
- 2009-11-17 - A True Off-the-Shelf Phone System? Interview introducing plug n’ go from Foncordiax
- 2009-11-17 - Asterisk security webinar with participants from the FBI, now available in video.
- 2009-11-16 - PCPhoneSoft - magicJack + X-Lite/SIP Dual Mode Calling Plugin Released.
- 2009-11-15 - Free White Paper "Battlefield Open Source PBX and Proprietary PBX" from Eastern Management Group now available
- 2009-11-15 - Google has acquired Gizmo5
- 2009-11-14 - FIVN releases Asterisk Management 2.0 using AJAX technology for routers.
- 2009-11-13 - Elastix Blogs Lounched
- 2009-11-13 - FreeSWITCH adds support for recently released Broadvoice codecs.
- 2009-11-13 - Telesoft - Telesoft released a new call center application -XDialer .
- 2009-11-13 - Telecomax introduced Java4Call - java application for mobile phones which uses Callback and allowing call abroad with low rates.
- 2009-11-13 - New research released on performance of a soft switch on IBM's cell processor
- 2009-11-13 - Leif Madsen has put together some pretty cool docs for creating Asterisk Queues
- 2009-11-13 - Counterpaths X-Lite and Eyebeam soft phones support BV codec (The Open Source Codec! with HD Quality)
- 2009-11-12 - Free Webinar About Open Source PBX Market Trends Nov. 18, co-sponsored by John Malone (EMG) and Xorcom
- 2009-11-12 - AstriCon Videos and Presentations First batch is online
- 2009-11-12 - New Open Source Codec! with HD Quality
News Resources
- Software Releases - Check here for recent VoIP-related software releases.
- VoIP Services - Check here for news on Voip Services.
- VOIP Event Calendar - Check here for news on VOIP Events, Tradeshows, Training and Conferences
- Older News - Check out older news articles here
Getting Started
- What is VOIP? The very basics.
- Free VOIP Publications: Magazines and Newsletters free to qualified subscribers about VOIP related products
- Training: Seminars, tutorials, on-line classes. Check here for recent.
- VOIP Consultants: Finding help.
- VOIP Service Providers - VOIP service providers.
- DID Service Providers - VOIP call origination service providers.
- VoIP Practical Guide - VoIP Resources for Your Home and Business.
- How to start a VOIP Business
Connecting VOIP to the PSTN and Cellular Networks
- Cheapest ATAs and Service - For Calling PSTN Numbers, or receiving phone calls from PSTN Numbers
- ENUM - Translating E164 numbers to VoIP addresses
- Failover Switches - Automatically or Manually Switch PSTN Interfaces on failure for reundancy
- FXS-FXO Converters - Convert an FXS interface to an FXO interface
- Phone Numbers - Dozens of test and information numbers you can call for free.
- Routing calls using a free international calling service - How to use two-step dialing to save on costs.
- PSTN Gateways - VOIP to PSTN gateways (also known as: Media Gateways)
- VOIP GSM Gateways - VOIP to GSM gateways
- Cheapest ATAs and Service
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
- IP Protocols COPS, ENUM, H.323, IAX, IMS, LTP, Megaco, MGCP, PINT, RTP, SCCP, SCTP, SIMPLE, SIP, STUN, T.37, T.38, TRIP,TURN,SDP
- ITU protocols SS7, ISUP
- OSP, PacketCable MRCP
- Encryption Protocols ZRTP
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
- Analog Telephone Information
- PBX features
- PSTN Interface Hardware
- Telecom Dictionary
- Telco Engineering Information
- Telephone History
- RESPORG: Toll Free 800 Number Programming
VOIP Events and Conferences
- VOIP Event Calendar — List of upcoming VOIP related events, Conferences, Trade Shows, Training, etc.
- Training and Conferences - Check here for recent Training and Conferences
- Asterisk-Tag.org Annual German conference on Asterisk and related topics
- Astricon
- ClueCon Annual conference on open source telephony development
- VoiceCon Annual conference on IP Voice Communication.
- Voice Peering Forum on routing, interconnection and peering of Web2.0 & VoIP networks
- Global VoIP and Telephony-related events
- VoIP business related events
- AstriEurop The Asterisk European Exhibition - April 14-15-16 2010 | Paris
Business Services
Resources
- VOIP Websites: Other VOIP websites on the Internet
- Twitter VoipUser Directory: Twitter VoipUser Directory
Suggestions and Questions
- How to add information to this wiki
- Suggestions: Put your requests and suggestions here


Comments
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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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333New Media Gateway Controller Simulator
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.