login | register
Tue 09 of Feb, 2010 [21:31 UTC]

voip-info.org

History

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.


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


Hit counter

This page has been viewed 13935686 times since being created on Sat 01 of Oct, 2005 [16:47 UTC]

RSS Feeds

  • Image Page Changes
  • Image Comments

Comments

Comments Filter
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.


222

333Hello world

by pvassalli, Monday 11 of January, 2010 [14:27:56 UTC]
I'm a new user!!!
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.