voip-info.org
Created by: system,Last modification on Tue 09 of Feb, 2010 [19:12 UTC] by alanlindsay
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
- 2010-02-09 - Great business VoIP buying checklist
- 2010-02-09 - First Asterisk visual voicemail application for iPhone.
- 2010-02-09 - Squire Technologies Launch SMSC and SMS Gateway at MWC 2010
- 2010-02-09 - OwnPages releases real time patch for AsteriskGUI 2.0 Mysql or postgres - better manageability, easier integration.
- 2010-02-09 - The SIP Schoolâ„¢ at UCEXPO London in March SIP Trunking Seminar - "Getting it right the 1st time"
- 2010-02-09 - Rho releases version 3.3 of AsteriskC2D and AsteriskC2DPro (iPhone click to call for Asterisk), new version has native Thirdlane support
- 2010-02-08 - US Patent Office Agrees to review C2/Acceris controversial VoIP patent
- 2010-02-08 - How to configure a Patton 4114 FXO Gateway with 3CX video tutorial
- 2010-02-06 - Anveo unveils Voice 2.0 communication suite with powerful Visual Call Flow technology. Visually create Voice IVR applications without any special skills.
- 2010-02-05 - Unified Recording release Open Unified Recording OUR under GPLv3. Open Source Sip Call recording and web interface.
- 2010-02-05 - How to Install FreePBX on the PIKA WARP Appliance without PADS
- 2010-02-04 - Howler Technologies WolfPack G.729 Transcoding SDK Available Now Makes Windows IP PBX G.729 Transcoding Affordable
- 2010-02-04 - Sprint Tango connects Sprint Mobile To Avaya and Cisco PBXs
- 2010-02-04 - 8x8 launches Facebook VoIP app
- 2010-02-04 - TELES strengthens ties with Du Pont
- 2010-02-04 - MCS, First succesful calls between Microsoft OCS and AS55x gateway using SIP/TCP
- 2010-02-04 - Wybecom, NEW TALK (Telephony Application Library Kit) Gadget! Get all TALK features in a windows vista / 7 gadget, demo here
- 2010-02-02 - 3CX and Beronet announce strategic partnership for complete interoperability
- 2010-02-02 - http://www.bulkcnam.com/ is offering FREE CNAM Queries for Asterisk, FreePBX, Trixbox etc...
- 2010-02-02 - Netfors and OpenVox Partner to Deliver SS7 Products and Professional Chan_SS7 Support
- 2010-02-01 - Patton Gateways Technincal Webinar March 2, 2pm EST - Learn configuration, advanced options, and troubleshooting
- 2010-02-01 - http://voipemulator.weebly.com VoIPEmulator - Simulating (H.248, MEGACO) Softswitch / Media Gateway Controller.
- 2010-02-01 - Redfone Branches Out: An Interview with Mark Warren, President of Redfone Communications
- 2010-02-01 - Outbound telephone research centre based on Asterisk.
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
333http://igbt-china.com/
http://flotsam47.igbt-china.com/
http://audiosector.igbt-china.com/
http://shop.igbt-china.com/
http://store.igbt-china.com/
http://en.module-china.com/
http://igbt-china.com/
http://chinaglobaltrader.com/
http://module-china.com/
333http://en.module-china.com/index.php?main_page=sitemapxml
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://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
333VIMS-integrated solution office network
333Adhearsion with Asterisk 1.4.28
files_to_play, :timeout => 5), playback of the sequence of files is
not stopped immediately when I press # or * keys, however keys 0-9
working fine. I am using Asterisk 1.4.28 and Adhearsion 0.8.3. I have
also tried Adhearsion 0.8.2 with Asterisk 1.4.28 with no luck.
It was working great on asterisk 1.4.12.1 with Adhearsion 0.8.2
Please help me if anybody can.
Thanks,
Mukteshwar
mukteshwarp@gmail.com
333How i can fix sip with ip address in setting.conf file??
we can manually assign the ip address in xlite properties but some times same sip we assign to other system that time sip easily accepted to other ip address so on this case we required to fix it sip with ip address in setting.conf so how could we do it.
333Hello world
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.