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  • About VoIP
    • What is VoIP and what it can do for you
    • Introduction to VoIP (video)
    • Why should you switch to VoIP services?
    • Analog Telephony
    • Digital Telephony
    • What is SIP?
    • How to start with VoIP telephony
    • Web based VoIP
    • How to choose a right VoIP provider?
    • Wi-Fi network and VoIP
    • VoIP Codecs
    • Free sip account
    • Confidential calls
    • VPN: UDP or TCP?
    • Mobile VoIP
    • VoIP on your mobile
  • Asterisk IP-PBX
    • All about Asterisk
    • About Mark Spencer
    • Asterisk SIP Media NAT
    • VoIP Codecs
    • Cisco ATA186 + Asterisk Config
  • Who we are?
  • How to start
  • Free SIP account
  • Configs
    • Grandstream Budgetone
ubuntu-linux-phone

Ubuntu (Linux) on your phone? Yes! And now officially!

By Nik On January 4, 2013 · In Open source, Ubuntu

News from Canonical: Ubuntu is coming to the phone

When we began developing Unity a few years ago, the aim was to create a single family of interfaces that work the same way on different devices. This means that unlike most of our rivals, we are able to use a single underlying OS across all [...]

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Ekiga_in_a_Call3

A fully functional VoIP Client (SIP) finally released (free download)

By Nik On November 29, 2012 · In Open source, Softphone VoIP

Great news for today, guys! The legendary Linux softphone is back for more (available for Windows also)!

Three years after the 3.2 release, Ekiga 4.0 aka “The Victory Release” is finally available.

This is a major release with many major improvements.

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webrtc_2

WebRTC from Google: making real-time communication free to implement

By Nik On July 12, 2012 · In Android, Google Voice, Open source

After Microsoft bought Skype for US$8.5 billion, Google has released a developer preview of WebRTC – an open framework enabling implementation of voice and video Real Time Communications in the browser with the use of HTML 5 and JavaScript APIs.

It’s possible that the Skype acquisition could mean restricting the client from technologies and devices [...]

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compass_25547_voip_course

Introduction to Voice over IP (VoIP)

By Nik On July 11, 2012 · In Asterisk, Mobile VoIP, Open source, SIP protocol

One important step into adopting VoIP is to choose a VoIP service, which will allow you to make and receive cheap or free local and international phone calls. It is important to choose the right type of VoIP service. Your needs and the way you will communicate should help you decide which type [...]

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free phone calls

How to configure Internet tel. and SIP settings on Nokia phone (E52)

By Nik On March 28, 2012 · In Google Voice, Mobile VoIP, Open source, SIP protocol, Symbian, VoIP calls quality

In this article we will talk about Nokia Internet telephony and SIP settings since many people asking me about that. I have check all configuration myself on my old Nokia E52 as well as E71 buddy and everything works fine. Therefore you can use the same setup procedure on any E-series with Symbian.

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Easy setup how to Asterisk with Google talk and Google voice

Google voice / Google talk and Asterisk configuration

By Nik On December 8, 2011 · In Asterisk, Google Voice, Open source

This article describes the configuration process of Asterisk with Google voice.

Asterisk communicates with Google Voice and Google Talk using the chan_gtalk Channel Driver and the res_jabber Resource module. Before proceeding, please ensure that both are compiled and part of your installation. Compilation of res_jabber and chan_gtalk for use with Google Talk / Voice are dependant [...]

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Asterisk 10.0.0-rc1 Now Available!

By Nik On November 14, 2011 · In Asterisk, Open source, SIP protocol

Digium releases Asterisk 10.

As you may already know Asterisk is a communications platform that allows developers to create powerful business phone systems and unified communications solutions. Since its introduction 12 years ago, Asterisk has been used, free of charge, in nearly every country of the world to power telephone and other [...]

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SIP VoIP calls NAT

SIP based VoIP behind NAT

By Nik On November 10, 2011 · In Asterisk, Google Voice, History, Open source, SIP protocol

For all the technology behind Voice over IP (VoIP), you’d expect that it would work on every network, but this unfortunately isn’t the case. Network Address Translation (NAT) is a common practice used in networks, and it doesn’t play well with VoIP. Solving this problem requires an understanding of NAT, VoIP [...]

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VoIP Quick start guide

By Nik On November 9, 2011 · In Open source, QoS - Quality of Service, SIP protocol, VoIP calls quality, VoIP over VPN

Hi there. Today I found a quick start guide on VoIP written for Cisco 3600 series modular routers- click here to download PDF. I think it could be useful even if you just a newbie and what to learn how VoIP works, so I decided to share it. Please let me know if you think it’s [...]

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Asterisk and Google Voice

By Nik On June 21, 2011 · In Asterisk, Google Voice, Mobile VoIP, Open source, SIP protocol

Asterisk: How to make calls using Google Prerequisites

Asterisk can communicate with Google Voice and Google Talk  using the chan_gtalk Channel Driver and the res_jabber  Resource module. Please ensure that both are compiled and part  of your installation prior following the configuration guide  below. Compilation of res_jabber and chan_gtalk for use with  Google [...]

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  • Recent articles

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    • US fixed VoIP market to see steady growth
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    • What is VoIP termination
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    • TCP vs UDP – you must know this
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    • Asterisk 10.0.0-rc1 Now Available!
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    • Think on solutions: VoIP phone system
    • Asterisk and Google Voice
    • VoIP Codec: Payload size
    • Nokia SIP settings
    • The QoS Dilemma
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    • VoIP for Facebook!
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    • Digital Telephony
    • Analog Telephony
    • Mobile VoIP – the future of mobile communications
    • Sip on Android
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  • More to read on VoIP

    • About
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    • How to start with VoIP telephony
    • Multiplexing RTP Data and Control Packets on a Single Port
    • On-line payments
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    • What is VoIP and what it can do for you
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    • Asterisk™: The Definitive Guide (new window) “Asterisk has been emblematic of the way that open source software has changed business—and changed the world”
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    • Ubuntu how-to www.ubuntuka.com Miscellaneous Ubuntu Tips, Tricks and Hints
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voip-sip.org – Internet VoIP calls, International SIP termination, VoIP hardware

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