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voip-info.org

Created by: system,Last modification on Tue 29 of Dec, 2009 [14:38 UTC] by spamblock

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.


Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelines before you post.


NEWS





News Resources


Getting Started


Connecting VOIP to the PSTN and Cellular Networks


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


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


VOIP Events and Conferences


Business Services


Resources



Suggestions and Questions


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222

333

by 377704497, Wednesday 23 of December, 2009 [17:15:44 UTC]
222

333Cisco 7911G Configure

by gisvpn, Sunday 22 of November, 2009 [19:13:14 UTC]
Hello,

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
222

333Designating outgoing Trunks For Individual Phones

by estesvoip, Wednesday 04 of November, 2009 [21:36:33 UTC]
Is there a way to force a single "telephone"; i.e. extension, to use a particular provider/trunk each time some one dials out on that device? Is there a setting that can be added to extensions.conf or something? I know how to route based on what was dialed but cannot figure out how to router based on the number from which they are dialing.
222

333New Media Gateway Controller Simulator

by voipemulator, Friday 16 of October, 2009 [20:08:11 UTC]
http://voipemulator.weebly.com
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.

222

333Re: I'm a new register user

by vasu1223, Sunday 04 of October, 2009 [22:44:58 UTC]
Hi every one.. I have connected the polycom ip 550 in care center. I am hearing the echo in my polycom phone. It is connected with power over ethernet.

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.


222

333Adore infotech Release New Softswitch,Mobile softphone

by tutu, Saturday 03 of October, 2009 [07:14:54 UTC]
222

333I'm a new register user

by usky.david, Monday 21 of September, 2009 [07:32:11 UTC]
Hi~everyone
I'm a new register user!
222

333Re: Flash cisco 7906 to SIP

by eazysnatch, Wednesday 09 of September, 2009 [07:53:25 UTC]
This video will help you upgrade firmware for your 7906 to use SIP
http://www.youtube.com/watch?v=PkEFlqsFp80

Greetings
222

333Re: VOIP Gateway

by wateen, Tuesday 08 of September, 2009 [13:59:13 UTC]
Great Idea!
222

333VOIP Gateway

by t4mack, Wednesday 02 of September, 2009 [18:53:45 UTC]
Hi,
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