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voip-info.org

Created by: system,Last modification on Mon 05 of Jan, 2009 [20:03 UTC] by jgg

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




News Resources


Getting Started


Connecting Phones to VOIP


Connecting VOIP to the PSTN and Cellular Networks


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.


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


VOIP Websites: Other VOIP websites on the Internet


Suggestions and Questions


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222

333voice silent issue

by me_sanjay1986, Tuesday 30 of December, 2008 [22:45:47 UTC]
Hi,

I have 20 users does calls trough my asterisk server. what when sometimes the voice become silent for few second with all the users. but other end users are still connected after 10-15 sec they again resume their conversation. Please help immediately.


Thanks

Sanjay
222

333Re: Asterisk server on internet

by chauhan_delhi, Wednesday 17 of December, 2008 [20:53:58 UTC]
Yes u can install in asterisk in saperate network.......
222

333Re: how can i dial a number while play the sound to the caller at the same time

by chauhan_delhi, Wednesday 17 of December, 2008 [20:51:56 UTC]
HI,


Background()
Dial()

try with this..
222

333Unity 5.0

by brogers, Wednesday 17 of December, 2008 [19:19:49 UTC]
Does anyone know about call handlers I can play or record under my login. Do I need a higher access over an administrator or go directly to Unity Server.
222

333Re: comment modifier asterisk en langue français?

by jsr123, Tuesday 25 of November, 2008 [14:08:39 UTC]
Je crois que vous aviez deux choix: 1 choisir langue francais au moment d'installer le systeme de nouvelle, ou sinon, 2 mettez vos changes dans extensions_custom.conf (seulement). Il vaudrait mieux ne pas changer extensions.conf ou extensions_additional.conf. Bonnes chances
222

333query extensions for caller ID of current call

by mrsenoj, Monday 17 of November, 2008 [17:34:57 UTC]
I need a way to query Asterisk for the caller IDs of calls to specific extensions. My eventual goal is to integrate this ability into an existing application to allow users to quickly retrieve information based on the caller ID. The existing application was developed in C#, so the ability to do this easily from within C# would be a plus.

We're currently using Asterisk 1.2.
222

333how can i dial a number while play the sound to the caller at the same time

by data212, Monday 10 of November, 2008 [08:11:26 UTC]
hi guys:
 Is it possible to use the asterisk dial the callee and play the sound to the caller at the same time?

i have trie the dial(),but its failed.it always play the sound over and then dial the next number.

hi,guys,how can i do to realise my thought?
regards!!

222

333Asterisk server on internet

by pcababie, Friday 31 of October, 2008 [14:11:29 UTC]
Is it possible to have the asterisk server in a different network than a voIP gateway?
Are there any blocking issues I should know about it?
222

333Q: PRovider detect the extensions

by yaniby, Friday 24 of October, 2008 [20:58:39 UTC]
Hi Everybody,
I've been trying to find out if there is a way for a voip provider to detect what extensions I'm sending before forwarding it to it's Destination.
I am trying to make a call to a destination no. play a short message and hung up but the provider is detecting my message and will not forward the call if the message is a certain length.
any ideas?
thanks in advance
222

333How do you upload files

by patlavery, Thursday 09 of October, 2008 [17:56:35 UTC]
Hi, I was just wondering how to upload files to my account. There doesn't seem to be any kind of interface, thanks.