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  • Samuel, Thu 03 of Jul, 2008 [13:41 UTC]: ok thank you
  • Mats Karlsson, Thu 03 of Jul, 2008 [13:37 UTC]: Nice Samuel, will look forward to rad it.
  • bwl_fernstudent, Thu 03 of Jul, 2008 [09:08 UTC]: Your blog shows some usefull code
  • Samuel, Thu 03 of Jul, 2008 [08:04 UTC]: I'll translate it, for sure
  • Mats Karlsson, Wed 02 of Jul, 2008 [20:46 UTC]: LOL, in french! Translate it to English and I will read it.
  • Samuel, Wed 02 of Jul, 2008 [08:07 UTC]: Hello, i wrote a blog about Asterisk, speaking about installation,programming and more http://sambranche.blogspot.com/
  • Nick Barnes, Tue 01 of Jul, 2008 [17:46 UTC]: Steve - Asterisk doesn't 'fit into linux' - it's an application which runs on top of Linux.
  • Steve, Mon 30 of Jun, 2008 [18:07 UTC]: anyone know where I can find a block diagram of how asterisk fits into linux. my f'ing bosses want me to draw something up.. ugh.
  • akbar, Fri 27 of Jun, 2008 [10:37 UTC]: marley_boyz@yahoo.com how to configure call forward, call back, call pick up using TDM and asterisk 1.2.13... please help me.. thx...
  • Matthew Williams, Tue 24 of Jun, 2008 [22:37 UTC]: We are looking for Tier II VoIP Support Technicians in St Louis. Send resumes to mwilliams AT voxitas DOT com.
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voip-info.org

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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Created by System, Last modification by spamblock on Fri 04 of Jul, 2008 [05:05 UTC]

Comments

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Multiple SIP 183 and SDP

by Shawn on 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


Multiple SIP 183 and SDP

by Shawn on Tuesday 01 of July, 2008 [21:57:06 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


by Dean Vesuvio on Saturday 28 of June, 2008 [08:06:47 UTC]

Geting CALL STATUS withing agi

by sandip Dutta on 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.

Re: New to the scene

by John Atkins on 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.

International Calls Zero Cost - Save A Lot Of Money

by Mahmoud on Friday 20 of June, 2008 [19:44:34 UTC]
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Re: Urgent Routes Needed

by Alberto Sagredo on Friday 20 of June, 2008 [11:00:51 UTC]
Spam on Comments. Please remove them..

Urgent Routes Needed

by arsal on Friday 06 of June, 2008 [04:49:46 UTC]

We are in urgent Need Quality Providers of " Mexico, Cuba , Dominican Republic, Philippines, Romania Mobile and Hait"


Details :

Payement terms we prefer Post pay 7/3 and in some cases can do Ve10.

We are currently seeking:

Cuba------------------ with 5 min ACD, ASR 45%---------------------$0.49-$0.63---(Available traffic: 10k-15k)
Haiti------------------- Mobile-All Rectal,Haitel,Digicel,Others--------$0.1410-------(Available Traffic 17k-21k)
Philippines------------ Proper, Manila, Breakout,PLDT----------------$0.082-$0.084-(Available Traffic 24k-29k)
Philippines------------ Globe,Smart 70% ASR, 8+ min ACD----------$0107-$0.110--(Available Traffic 17k-19k)
Dominican Republic-- Mobile, Verizon, Orange-----------------------$0.61-$0.062--(Available Traffic 44k)
Dominican Republic-- Tricom and Others-----------------------------$0.048--------(Available Traffic 9k)
Mexico---------------- Mobile-------------------------------------------$0.074--------(Available Traffic 26k)
Brazil------------------ Mobile------------------------------------------$0.087
Honduras------------- Mobile Open, Proper---------------------------$0.077
Romania-------------- Mobile Orange---------------------------------$0.041
Romania ------------- Mobile VodaPhone-----------------------------$0.073

(Traffic up to 31k-37500)

For our overflow traffic, we are seeking quality and reliable suppliers to bid on this Route

Please include your contact details, time to call on US CST, Route quality and rates information.

Our Company Introduction:

Company: US

System: Nextone

Member Arbinet: since last 3 years

Terms: Flexible

Client: Major Tier 1

Accounting: Professional/CPA Level

Proud participant of ITW and GTM.


Looking forward to hear from you.

Msn : axal87@live.com

Email me :arsal987@gmail.com

Regards,

Axal

REDIAL / RECALL on Asterisk

by Marco Sambo on 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

BRAND NEW T-MOBILE SIDEKICK LX JUST FOR $150

by cwaysales on 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

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