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voip-info.org

Created by: system,Last modification on Sat 19 of Jul, 2008 [17:32 UTC] by waquip

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
  • FreeSwitch: Open source cross platform (*nix, Win, OSX) multiprotocol (SIP, IAX, Googletalk, Jabber, XMPP, WOOMERA/H.323, PRI/T1/E1) softswitch.
  • CallWeaver: Vendor independent, cross-platform open source IP-PBX, originally derived from Asterisk.
  • OpenSER: Flexible and powerful open source GPL SIP ( RFC3261 ) server with integrated TLS
  • Sip Express Router: High-performance, configurable, free SIP ( RFC3261 ) server
  • sipX Solution Summary: SIPfoundry - The SIP PBX for Linux (L-GPL)
  • YATE - Open Source Linux/Windows GPL Telephony Server and Client (has support for SIP, H.323, IAX2, E1/T1, voicemail), H.323 - SIP translator.
  • 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

333What about the other projects?

by anthm, Friday 18 of July, 2008 [18:44:34 UTC]
Please add links to the menu on the left for the other projects, Asterisk is not the only Open Source VoIP app.
FreeSWITCH href=http://www.freeswitch.org
sipX http://sipx-wiki.calivia.com/index.php/SipX#sipX_-_The_SIP_PBX_for_Linux
CallWeaver http://www.callweaver.org
YATE http://yate.null.ro/pmwiki/
Bayonne http://www.gnu.org/software/bayonne/
OpenSER http://www.openser.org/
222

333how to configure the sflphone account?

by zwh, Friday 18 of July, 2008 [07:29:47 UTC]
i have been install a sflphone client,but i cannot use it ,how i configure the sflphone account?and how i use
the sflphone to call somebody esle?please tell me the way of using it.thanks very much!
222

333Help for Call Recording

by seamaster, Wednesday 16 of July, 2008 [14:23:54 UTC]
Hi,

our customer need to record some non-crypto (rtp) and crypto (srtp) phone call in a VoIP asterisk network; Is there a solution for this issue?

Thanks
222

333Help with Polycom 430 SIP and Switchvox

by marshalr, Thursday 10 of July, 2008 [21:36:45 UTC]
New to game. Need guidance on how to reconfig Polycom and set up as different extension in Switchvox.
222

333Multiple SIP 183 and SDP

by ShawninCO, Tuesday 01 of July, 2008 [21:57:39 UTC]
1. "A" places a call to B. A 183 with SDP is received. In band info is heard and all is good.
2. Before the call is answered , B sends another 183 with SDP. The following fields differ in this second 183:

 Session ID,
 Session Version,
 Owner Address,
 Media Port.

All other SDP and all SIP info (e.g. Call-ID) is the same between the two 183's.

The rtp from B is being sent to the right port on A (same as step 1 above), but this new in-band info is not heard. I would expect this. Here's the problem. I've tested this with a couple of different boxes acting as "A". On some it works - and some it doesn't. Does anyone know if the scenario outlined above is legal? -Thanks


222

333Geting CALL STATUS withing agi

by bantisandy, Wednesday 25 of June, 2008 [10:46:49 UTC]
Hi I am new to asterisk. I have written a C AGI to make call from my sip phone to another sip/zap phone. I can make call use EXEC function from within my AGI but I want to get the call status, whether it matured/failed. The following is my code.

main()
{
       char line100="";
int len=0,x=0,y=0,pos=0;
      setlinebuf(stdout);
      setlinebuf(stderr);

     while(1)
     {
        fgets(line,100,stdin);
        if(strlen(line)<=1) break;
     }

      printf("EXEC DIAL SIP/1000|30|g \n");

//Call status here to do something
}


Now I want to get the status of the call in the agi how can i. Plz help me out, thnx in advance.
222

333Re: New to the scene

by Jatkins77, Monday 23 of June, 2008 [19:04:43 UTC]
I was able to work this issue out. Turned out to be a bad card. Had to get a replacement.
222

333REDIAL / RECALL on Asterisk

by zipge, Wednesday 04 of June, 2008 [12:54:46 UTC]
Hi all!
I've a problem with my Asterisk. In my configuration, to make an external call, I've to put 0 before the number that I want to call. It's work! But if I recieved an answered/unaswered call and I want to recall it, Asterisk don't recognize the number without initially 0 to route it outside.
Can someone help me?

I'm sorry for my english, I hope u understand it!
Thank you all

Zipge
222

333BRAND NEW T-MOBILE SIDEKICK LX JUST FOR $150

by cwaysales, Wednesday 04 of June, 2008 [12:13:48 UTC]
DESCRIPTION
.................

Kick it to the next level. The new Sidekick LX is sleeker and slimmer, including a large screen that incorporates high-definition LCD technology, a camera with flash, and mood lights which lets users set specific settings for various communication alerts. The LX is also Bluetooth capable, and has signature swivel screen, amazing keyboard, plus the killer MySpace experience, you can't miss.

Features
Text messaging
1.3 Megapixel camera
MegaTones, Wallpaper, HiFi Ringers, & Games
Games
Music player
Bluetooth wireless technology
E-mail
Full QWERTY keyboard
Picture messaging
Swivel Screen
External caller ID
Personal Information Mgr
Micro SD memory slot
Calendar
Phone book
Speed dial
Size: 4.6 x 2.4 x 0.7 in.
Weight: 5.30 oz
Battery: 1130mAh Li-ion
Talk Time: up to 5 hours
Standby Time: up to 3 days
Band (frequency): 850 MHz;900 MHz;1800 MHz;1900 MHz


PACKAGE CONTENTS

Original T-Mobile Retail Box
New Sidekick LX (blue)
Original Battery & Cover
Original Home Charger
Original Carrying Case
Original Stereo Handsfree Headset
128MB Memory Card
Manual


eMAIL : c-waylimited1@hotmail.com
222

333BRAND NEW T-MOBILE SIDEKICK LX JUST FOR $150

by cwaysales, Wednesday 04 of June, 2008 [12:09:17 UTC]
DESCRIPTION

Kick it to the next level. The new Sidekick LX is sleeker and slimmer, including a large screen that incorporates high-definition LCD technology, a camera with flash, and mood lights which lets users set specific settings for various communication alerts. The LX is also Bluetooth capable, and has signature swivel screen, amazing keyboard, plus the killer MySpace experience, you can't miss.

Features
Text messaging
1.3 Megapixel camera
MegaTones, Wallpaper, HiFi Ringers, & Games
Games
Music player
Bluetooth wireless technology
E-mail
Full QWERTY keyboard
Picture messaging
Swivel Screen
External caller ID
Personal Information Mgr
Micro SD memory slot
Calendar
Phone book
Speed dial
Size: 4.6 x 2.4 x 0.7 in.
Weight: 5.30 oz
Battery: 1130mAh Li-ion
Talk Time: up to 5 hours
Standby Time: up to 3 days
Band (frequency): 850 MHz;900 MHz;1800 MHz;1900 MHz


PACKAGE CONTENTS

Original T-Mobile Retail Box
New Sidekick LX (blue)
Original Battery & Cover
Original Home Charger
Original Carrying Case
Original Stereo Handsfree Headset
128MB Memory Card
Manual