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.
If you are removing content from pages, please leave a polite comment explaining the reason for the changes. Uncivil comments, user names, or content will result in the removal of the users editing privileges.
If you are removing content from pages, please leave a polite comment explaining the reason for the changes. Uncivil comments, user names, or content will result in the removal of the users editing privileges.
Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelines before you post.
NEWS
- 2009-12-18 - Free asterisk cnam (callerid) queries at www.asteriskcnam.com
- 2009-12-18 - How to Compile FreePBX for the PIKA WARP APPLIANCE
- 2009-12-18 - Polycom VVX 1500 Business Media Phone, Asterisk & OCS? An Interview with Chris Wortt, Polycom’s EMEA VoIP Sales Manager
- 2009-12-18 - Asterisk PBX on a PCI card with integrated telephony ports, hardware based echo cancellation, Ethernet interface and Compact Flash port.
- 2009-12-17 - How to sell a phone system Part 3 - Tips and advice on preparing a professional quote
- 2009-12-17 - Government of Canada supports research and innovation at PIKA Technologies
- 2009-12-16 - PBX Reseller Survey - Take this short survey and be entered to win a $250 AMEX gift card
- 2009-12-16 - WYBECOM: New TAPI Connector released, TALK (Telephony Application Library Kit) is now compliant with most manufacturers. Download now!.
- 2009-12-14 - Why the Google Nexus One is so critical for the industry?
- 2009-12-14 - FreeSWITCH 1.0.5pre9 is now available.
- 2009-12-14 - Kamailio (OpenSER) 3.0.0rc3 is now available
- 2009-12-14 - How to sell a phone system Part 2 - Designing your customer's call flow
- 2009-12-14 - Interesting interview with Phil Wolff of Skype Journal about the future of VoIP
- 2009-12-14 - Skype for Symbian beta release!
- 2009-12-14 - New open source CTI client for Asterisk.
- 2009-12-11 - How to configure fax using FreePBX on the PIKA WARP Appliance
- 2009-12-11 - OwnPages introduces Realtime Asterisk GUI
- 2009-12-10 - PBX in a Flash introduces bootable USB flash installer for Asterisk.
- 2009-12-10 - Santa Claus gives Snom phones!
- 2009-12-09 - TELES takes over Ecotel GSM gateways from Vierling
- 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
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
333 *astTECS Hosted dialer Solutions
*astTECS introducing the Hosted predictive dialer that can dial 10,00,000+ Phone numbers daily.
- JUST RS 1200/- SEAT
- NO SECURITY DEPOSIT
KEY FEATURES:
-Predictive Dialing
-Progressive dialing
-C R M
- I V R's
-Multiparty conference
- G S M Integration
- C D R
CONTACT INFO:
SUNIL VJ
PH NO : 9986494312
EMAIL : v.sunil@asttecs.com
Website : www.asttecs.com
333 *astTECS G4 GSM CARD
ast g4 - For 30 subscribers.
Monthly Savings upto 60%
ast g4 converts usual land line to mobile calls to cost saving mobile to mobile calls. This helps companies to attain the ROI within 6-8 Months.
KEY FEATURES:
- Rapid deployment.
- Customization.
- Voice mail system.
- Multiparty conference
- Real - time reporting
- Real-time statistics.
-ROI in less than 4 months.
FOR CONTACT DETAILS:
SUNIL VJ
PH NO : 9986494312
Email : v.sunil@asttecs.com
Website : www.asttecs.com
333 *astTECS Call Center Solutions
The call center version is a comprehensive customer contact platform that seamlessly integrates with your existing voice and data systems,whether VOIP,PRI,and GSM.
-ast c 5 - For 5 Subscribers.
-astc30 - For 30Subscribers.
-astc60 - For 60Subscribers.
Features:
Vicidial: is a call center solution for both inbound & Out bound call center.
KEY FEATURES:
-Predictive Dialing
-Progressive Dialing
-IVR,s
-Multiparty Conference
-C R M
-Voice Mail system
-Recording
-Customization
-G S M Integration
-V Logger
FOR CONTACT DETAILS:
SUNIL VJ
PH NO : 9986494312
EMAIL : v.sunil@asttecs.com
WEBSITE : www.asttecs.com
333
http://igbt-china.com/sitemapcategories.xml
http://igbt-china.com/sitemapproducts.xml
http://igbt-china.com/sitemapindex.xml
http://igbt-china.com/sitemap.xml
http://igbt-china.com/sitemaps.xml
http://igbt-china.com/index.php?main_page=index&cPath=1
http://igbt-china.com/index.php?main_page=site_map
http://chinaimportexport.wikispaces.com/
http://chinaimportexport.wikispaces.com/$sitemap
http://chinaimportexport.wikispaces.com/Wholesale+Power+Modules+IGBT+Manufacturer+exporting+direct+from+China
http://chinaimportexport.wikispaces.com/Wholesale+Integrated+Circuits+Manufacturer+exporting+direct+from+China
http://chinaimportexport.wikispaces.com/Wholesale+Power+Modules+IGBT+Manufacturer+exporting+direct+from+China-15
http://en.module-china.com/index.php?main_page=sitemapxml
http://en.module-china.com/sitemapcategories.xml
http://en.module-china.com/sitemapproducts.xml
http://en.module-china.com/sitemapindex.xml
http://en.module-china.com/sitemap.xml
http://en.module-china.com/sitemaps.xml
http://en.module-china.com/index.php?main_page=index&cPath=1
http://en.module-china.com/index.php?main_page=site_map
http://igbt-module.wikispaces.com/
http://igbt-module.wikispaces.com/$sitemap
http://igbt-module.wikispaces.com/Wholesale+Power+Modules+IGBT+Manufacturer+exporting+direct+from+China
http://igbt-module.wikispaces.com/Wholesale+Integrated+Circuits+Manufacturer+exporting+direct+from+China
http://igbt-module.wikispaces.com/Wholesale+Power+Modules+IGBT+Manufacturer+exporting+direct+from+China-15
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!