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Sat 22 of Nov, 2008 [02:33 UTC]

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voip-info.org

Created by: system,Last modification on Thu 20 of Nov, 2008 [12:16 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.

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

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.
222

333Why is Diana removing Kamailio and adding OpenSIPs ?

by matsk, Sunday 05 of October, 2008 [18:23:52 UTC]
Please stop doing that !
222

333Siemens HiPath - can it pass 14 digit ANI

by svick, Thursday 02 of October, 2008 [21:52:43 UTC]
I need to know if Siemens can pass a 14 digit ANI? I have a special need for this capability, so if anyone knows, please let me know. There is nothing on their site that tells me one way of the other and I don't want to call sales people just yet . . . not ready for that monkey on my back.
222

333Soyo g668 phone problem

by yorweb, Saturday 27 of September, 2008 [04:45:18 UTC]
When setting up one of my phones I input PPPoE in error instead of DHCP. Now when I boot up the phone it locks up trying to connect and I can't get in to reset it. I also can not connect to it with internet explorer. Anyone have any idea how to get in to reconfigure it other than just throwing it away and starting fresh? Any help is appreciated.
222

333comment modifier asterisk en langue français?

by keepora, Friday 12 of September, 2008 [09:03:33 UTC]
Qu'est-ce-qu'on doit faire après qu'on ai mis le son en français dans /var/lib/asterisk/sounds ? Faut-il modifier le fichier extension.conf? comment?
222

333dialplan settings-Linksys 3102

by vaca, Tuesday 09 of September, 2008 [14:19:19 UTC]
Hello. I would like to configure my Linksys SPA 3102 voip gate to implicite call through SPTN (classic telephone line), and after pressing "#", following call will be redirect to the internet (SIP account).

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! :-)