voip-info.org
Created by: system,Last modification on Thu 16 of Oct, 2008 [02:07 UTC] by cosmicwombat
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
- 2008-10-16 - Interview-Philippe Lindheimer talks about AsteriskNow 1.5 w/FreePBX
- 2008-10-15 - TRex - Asterisk- based Carrier Grade Softswitch with up to 40 E1 per chassis
- 2008-10-15 - Brekeke Software Receives a 2008 INTERNET TELEPHONY Excellence Award
- 2008-10-09 - TELES VoIP and Mobile Gateways receive Siemens HiPath ready certificate
- 2008-10-09 - Squire VOIP SS7 Media Gateway provides UK operator Stour Marine with interconnect solution
- 2008-10-07 - Massive, 900 port Asterisk deployment case study released by Avanzada7 & Redfone (PDF-alert)
- 2008-10-06 - First iPhone VOIP Application in iTunes App Store
- 2008-10-04 - Microsoft restores VOIP to Microsoft Live Messenger
- 2008-10-03 - Vertu Signature S Design luxury dual-mode handset has been released
- 2008-10-01 - Bandwidth.com Launches Sonus To Deliver Its Next Generation SIP Network
- 2008-10-01 - NOVACOM becomes Be IP and will continue to distribute innovative Asterisk based solutions
News Resources
- Software Releases - Check here for recent VoIP-related software releases.
- VoIP Services - Check here for news on Voip Services.
- Training and Conferences - Check here for news on 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.
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
- 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
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
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?
333dialplan settings-Linksys 3102
Can anybody help me with configuring dialplan for this parameters? Unhappily I don´t know US (NY) standard dial plan string character. In VoIP (SIP) calls, I would like to use Czech dial plan, with czech call signalls.
Should it be: (*xx|346911<:@gw0>|0<:@gw0>|00<:@gw0>|2-9xxxxxx<:@gw0>|1xxx2-9xxxxxxS0<:@gw0>|xxxxxxxxxxxx.<:@gw0>|<#,:>2-9xxxxxxxxS0) ??? Outside Dial Tone: 425@-10;30(0.33/0.33/1,0.66/0.66/1) (VoIP account)
I will be glad, if someone trying it.
Thanks! :-)
333Looking for TAPI to SIP Bridging Software
m_a_naqvi@yahoo.com
333Im Looking for Automated VOIP voice broadcasting
whereby the cost for calls are actually free.
Is there a system out there? Also can a system be made and at what price?
Karl Jackson
GSM
540-465-4432
karl@shentel.net
333Nevemind - wrong forum
333with skype
I've seen some things online that show you hi.