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

NEWS

• 2008-05-09 - Watch videos on Building High-Performance VoIP Clusters with Asterisk on YouTube.
• 2008-05-09 - ITS Telecom's Flagship Product, Connecto, Awarded by AT&T in the Fast-Pitch Contest held in CTIA exhibition, Vegas US
• 2008-05-09 - Very Cool HD Commercial from Voice Peering Forum: Features Digium, Thomas Howe, France Telecom, Sonus and many more!
• 2008-05-09 - AIM Call Out offers free calling worldwide in more than 200 Countries. Their voice platform supports any SIP compatible device to make calls.
• 2008-05-08 - TigerCube.net goes live major new support features and online shop... get ready for TigerCube v2.0
• 2008-05-08 - The Free Telephony Project demonstrates extreme low power Asterisk - Asterisk with 4 FXS ports on just 3 Watts
• 2008-05-07 - ALLWIN TECHIs Web Call Server, VoIP Gateway and Router 3 in 1 possible?
• 2008-05-06 - Rhino Equipment unveils the new look of their Ceros Appliance
• 2008-05-06 - Japan VoIP and Asterisk Lounge Azabu Juban Tokyo
• 2008-05-05 - AIM Call Out now works with Asterisk as a SIP Termination Gateway to 200 countries.
• 2008-05-05 - Build a High Performance Telephony System with New Book on OpenSER

News Resources

• Software Releases - Check here for recent VoIP-related software releases.
• VoIP Services - Check here for news on Voip Services.
• Training and Conferences - Check here for news on 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.
• 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.

Connecting Phones to VOIP

• IP Phones: VoIP phones both hardware and software
• Analog Telephone Adapters: VoIP analog telephone adapters ATA - see Cheapest ATAs and Service
  • See also VOIP Routers
  • See also Asterisk hardware home analog: includes some comparison of external ATA and PCI card
• Digital Telephone Adapters: VoIP Digital/TDM telephone adapters
• Dial Pulse to Touchtone DTMF Converters - connect that old rotary phone to DTMF VOIP equipment
• VOIP Paging and Intercom
• VOIP Payphones
• VOIP and TTY VOIP and hearing impaired TTY terminals
• VOIP Paging Equipment - paging with VOIP
• Free VoIP Networks - list of Free VoIP Providers
• Wireless VOIP: Cut the wires! Roam free with wireless VOIP

Connecting VOIP to the PSTN and Cellular Networks

• Cheapest ATAs and Service - For Calling PSTN Numbers, or receiving phone calls from PSTN Numbers
• Configuring GSM VoIP gateways with Cisco Call Manager - Step by step guide
• 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

PBX and Servers - VOIP PBX and Servers

• Asterisk: Open Source PBX
• Bayonne: Open source PBX
• FreeSwitch: Open source cross platform (*nix, Win, OSX) multiprotocol (SIP, IAX, Googletalk, Jabber, XMPP, WOOMERA/H.323, PRI/T1/E1) softswitch.
• CallWeaver: Vendor independent, cross-platform open source IP-PBX, originally derived from Asterisk.
• OpenSER: Flexible and powerful open source GPL SIP ( RFC3261 ) server with integrated TLS
• Sip Express Router: High-performance, configurable, free SIP ( RFC3261 ) server
• sipX Solution Summary: SIPfoundry - The SIP PBX for Linux (L-GPL)
• YATE - Open Source Linux/Windows GPL Telephony Server and Client (has support for SIP, H.323, IAX2, E1/T1, voicemail), H.323 - SIP translator.
• more...

VOIP Misc.

• VOIP Websites: Other VOIP websites on the Internet
• Policy and Regulatory: VOIP legal and regulatory information
• VOIP Jobs: Finding a VOIP Job
• VOIP Providers For Sale: Buy or Sell infrastructure
• Silicon Chips specifically designed to support VOIP
• Telecom Fraud
• Special Purpose Phones: For those with different needs.

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

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

• Training and Conferences - Check here for recent Training and Conferences
• Asterisk-Tag.org Annual German conference on Asterisk and related topics
• Astricon 2005/2006/2007
• 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

VOIP Websites: Other VOIP websites on the Internet

Suggestions and Questions

• How to add information to this wiki
• Suggestions: Put your requests and suggestions here

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