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
- 2009-12-09 - PCPhoneSoft New "Magicfeatures" Software Plugin Adds Cool Features To magicJack
- 2009-12-09 - Choosing a VoIP Billing Solution; Bundled or Stand-Alone. An article by Dean Hansen of DTH Software, Inc
- 2009-12-09 - FreeSWITCH 1.0.5pre8 is now available.
- 2009-12-09 - How to sell a phone system part 1 - The Sales Meeting
- 2009-12-09 - Xorcom Twinstar: VoIPon Interviews Ruth Bridger, VP Marketing, Xorcom
- 2009-12-09 - wav2g729: Free tools to convert wav PCM file to G.729 file
- 2009-12-09 - IndustryDynamics enters into a definitive distribution agreement with First Telecom of Greece to Skype enable businesses in the Balkan region
- 2009-12-08 - VoIP Mobile Tools Collection from FonoSIP.com
- 2009-12-08 - Yealink announces the complete interoperability test with Elastix
- 2009-12-08 - Nerd Vittles Tutorial: Free Google Voice Dialer for Windows = Unlimited Free Calls to the U.S. & Canada
- 2009-12-08 - STARFACE wins award IP-PBX Product of the year
- 2009-12-08 - Skype for SIP open! - beta
- 2009-12-07 - Why Apple wants to control the iPhone platform? - The real reason...
- 2009-12-07 - FCC seeking public comment on converting PSTN to all IP (VoIP)
- 2009-12-07 - VoipOperator A new FREE Asterisk dialer and call notification application with phonebook for Windows.
- 2008-12-07 - Beginning OpenVPN 2.0.9 - A new book by Markus Feilner to build and integrate Virtual Private Networks using OpenVPN. Read more
- 2009-12-07 - How to effectively compare PBX systems
- 2009-12-07 - Asterisk vs OpenSIPS
- 2009-12-06 - New VoIPEmulator release (ver 2.923)
- 2009-12-04 - Win Free Polycom HD Voice Phones this Holiday Season - No Strings Attached
- 2009-12-04 - How to add a custom string to PIKA WARP LCD IP screen in FreePBX
- 2009-12-04 - Redfone Launches new low cost Single T1/E1 foneBRIDGE2
- 2009-12-03 - PIKA Integrates FreePBX onto WARP Appliance
- 2009-12-03 - iLBC vs g729 - Which codec is right for me?
- 2009-12-01 - Sangoma: What’s Next? Interview with Serge Forest, VP Marketing, Sangoma
- 2009-12-01 - FREE Guide to (some) IETF working groups from The SIP School™
- 2009-12-01 - Version 1.2 of SafiServer and SafiWorkshop brings easy web service integration to the Asterisk world
- 2009-11-30 - Exclusive interview with Mark Spencer, creator of Asterisk.
- 2009-11-25 - Why VoIP Paging Instead of Analogue + ATA? Interview with Bill Majerzak, Cyberdata
- 2009-11-24 - Tips to speed up your deployment time article written by Kerry Garrison
- 2009-11-24 - Developing a deployment test plan to help your installations go smoother
- 2009-11-23 - WebSpeedTechnology now offers FREE Trixbox & AsteriskNOW Virtual Appliances!!!
- 2009-11-20 - aitendant launches free order, CRM, billing & customer self-service platform ( PHONE+)
- 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!
- 2009-11-17 - 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.
- 2009-11-07 - A2Billing New released version v.1.4.2.1 with Automated Install - Install A2Billing in less than 1 minute.
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
- http://www.asteriskgeekz.com - VOIP for small 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
Hit counter
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Page Changes
- Comments
Comments
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
333Flash cisco 7906 to SIP
I can't flash my 7906 to SIP i use 76xx how to but when phone get my SEPXXXXXX.cfg from my TFTP I get auth failed i try different images no luck i try passwords like "cisco" and "1234" i see somewhere this is default but no luck can some1 give me little help.
Greetings