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, but please read the How to add information to this wiki page and the Posting Guidelines before you post.
NEWS
- 2008-12-04 - Asterikast Releases Episode 5 Building a Menuing System.
- 2008-12-03 - Sangoma introduces New Line of Analog Voice Cards.
- 2008-12-01 - HD VoIP interview series with Polycom Co-Founder Jeffery Rodman
- 2008-11-29 - Nokia 6260 slide is a new Nokia Series 40 VoIP capable handset which was released earlier this week.
- 2008-11-25 - Salesforce and STARFACE announce new telephony connector for asterisk & salesforce
- 2008-11-25 - The Edgepbx just released new 8 ports IPPBX - FX08A, with LCD/15G CF/WIFI supported
- 2008-11-25 - SIP SIMPLE gains traction through a reference open source implementation
- 2008-11-24 - RockBochs Inc. releases the new PhoneBochs Mini for smaller installations.
- 2008-11-23 - Singapore-Israel R&D Fund to invest in Xorcom and Voiceroute
- 2008-11-17 - FreePBX announces Corporate Sponsor
- 2008-11-13 - OpenSIPS (Open SIP Server) starts the public discussions on the new design
- 2008-11-11 - Xorcom Announces Disk-on-Key based IP-PBX Recovery Device
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.
- 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 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
Please post new/other servers here, because they will be removed.- Asterisk: Open Source PBX
- Bayonne: 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)
- 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...
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.
- Misc. VoIP Software: Software that doesn't fit into other categories.
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
- 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 Websites: Other VOIP websites on the Internet
Suggestions and Questions
- How to add information to this wiki
- Suggestions: Put your requests and suggestions here


Comments
333Re: comment modifier asterisk en langue français?
333query extensions for caller ID of current call
We're currently using Asterisk 1.2.
333how can i dial a number while play the sound to the caller at the same time
Is it possible to use the asterisk dial the callee and play the sound to the caller at the same time?
i have trie the dial(),but its failed.it always play the sound over and then dial the next number.
hi,guys,how can i do to realise my thought?
regards!!
333Asterisk server on internet
Are there any blocking issues I should know about it?
333Q: PRovider detect the extensions
I've been trying to find out if there is a way for a voip provider to detect what extensions I'm sending before forwarding it to it's Destination.
I am trying to make a call to a destination no. play a short message and hung up but the provider is detecting my message and will not forward the call if the message is a certain length.
any ideas?
thanks in advance
333How do you upload files
333Why is Diana removing Kamailio and adding OpenSIPs ?
333Siemens HiPath - can it pass 14 digit ANI
333Soyo g668 phone problem
333comment modifier asterisk en langue français?