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.Notice: Editing pages to eliminate SPAM and inappropriate content is appreciated, when there are disagreements about what is appropriate content please contact: support@voip-info.org.
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NEWS
- 2009-12-28 - Online Asterisk Training, in spanish for 2010!
- 2009-12-27 - VoIP named fastest growing industry of the last decade
- 2009-12-26 - Realtone Technology - How to configure Asterisk and VoIP Gateways for Office IPPBX;
- 2009-12-24 - A2billing 1.4.4.1 Automated Install Script released today
- 2009-12-23 - A2Billing 1.4.4 is now available
- 2009-12-22 - Why the Google Phone will run into trouble.
- 2009-12-22 - IP PBX Reseller Survey Results - Interesting results and commentary to help improve your business
- 2009-12-22 - Full PBX solution based on Asterisk with CTI daemon and CTI client!
- 2009-12-21 - PCPhoneSoft MagicJack Users Get Call Recording On Demand
- 2009-12-21 - http://www.plugpbx.org Project Starts. Run Asterisk and FreePBX on a SD card on the SheevaPlug Embedded Arm computer. Flash and go! Open source!
- 2009-12-21 - Nerd Vittles Tutorial: Introducing Phone Genie for Asterisk - Control your Asterisk server using simple email messages
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
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Comments
333
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333Cisco 7911G Configure
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
333Designating outgoing Trunks For Individual Phones
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.
333Re: I'm a new register user
How to stop the echo in my phone.
could u pls help me out in solving this problem.
Can it be solved by changing the voice codec settings of the polycom phone.
333Adore infotech Release New Softswitch,Mobile softphone
333I'm a new register user
I'm a new register user!
333Re: Flash cisco 7906 to SIP
http://www.youtube.com/watch?v=PkEFlqsFp80
Greetings
333Re: VOIP Gateway
333VOIP Gateway
I'm new to VOIP and wondering what equipment is needed to do so. What I have proposed so far is that we have a VOIP capable switch with Poe, PoE VOIP phones, a for a gateway I'm going to be building one. I was thinking for the gateway I could build a machine with a quad core processor with about 8 gigs of memory, and a netboarder A101 card. On the box I was planing on installing netboarder express and Yate. Before we get into the configuration please let me know if there is anything that I left out that I may need. Please excuse me if I'm going in the wrong direction. I haven't don't have much experience with networking a VOIP system.
Thanks