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voip-info.org

Created by: system,Last modification on Wed 10 of Mar, 2010 [07:21 UTC] by fahham

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

333Re: How i can fix sip with ip address in setting.conf file??

by umeshgaire, Wednesday 03 of March, 2010 [03:36:26 UTC]
I have also got a same problem with my sip device Linksys pap2 this device is going to damage it's processor, so if you have gurantee of this device you need to change it, or if it's working please disable the DHCP and put a static ip always.. I used to put Static IP it's very good. Because we wouldn't get confused.
http://findcheapestvoip.blogspot.com
222

333Hosted VoIP for Business

by penkelpens, Saturday 27 of February, 2010 [11:00:29 UTC]
Checkout:

www.cytracom.com
222

333Great VoIP service provider: AudeoCom.net

by rweinstein, Wednesday 17 of February, 2010 [04:37:38 UTC]
Several months ago I switched to Audeo Communications for my small business phone service. This VoIP service provider has been great. I love the find-me-follow-me and auto-attendant features and that I get my voice-mail in my e-mail (on my Blackberry). I love being able to listen to my voice-mail without having to dial-in; I just download it and listen to it.
The voice quality is great and it's downright cheap.
222

333Re: VOIP Guide - http://www.thevoipguide.org

by thevoipguide, Wednesday 17 of February, 2010 [03:20:02 UTC]
222

333VOIP Guide - http://www.thevoipguide.org

by thevoipguide, Sunday 14 of February, 2010 [01:13:14 UTC]
http://www.thevoipguide.org

Check it out!
222

333http://igbt-china.com/

by igbt-module, Saturday 30 of January, 2010 [18:53:06 UTC]
222

333http://en.module-china.com/index.php?main_page=sitemapxml

by igbt-module, Saturday 30 of January, 2010 [18:41:50 UTC]
222

333VIMS-integrated solution office network

by vu_quocviet, Friday 29 of January, 2010 [07:39:10 UTC]
VIMS Server provides organizations and business solutions to call, Fax Free 100% in the system between the headquarters away from each other (not limited to geographic distance) via an IP network like the Internet, Leasedline / MegaWAN / xDSL ... with support to connect through VoIP Gateway allows users to use the device, usually with a fax using the traditional dial-up service. Software is built on SIP technology platform running on CentOS 5 operating system with Web interface allows users to easily exploited and the system administrator. With the ability to support most standard encryption device (CODEC) current as G711, GSM, G729 ... allows users to comfortably select terminals rich as SIP VoIP phone, VoIP gateway, VoIP adapter, or use software on your computer dial (Softphone) to use the service.
222

333Adhearsion with Asterisk 1.4.28

by mukteshwar, Friday 22 of January, 2010 [11:46:06 UTC]
When I am specifying files to play using input (e.g input 3, :play =>
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
222

333How i can fix sip with ip address in setting.conf file??

by hitendra, Wednesday 20 of January, 2010 [08:43:19 UTC]
Hello,

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.