VOIP and Ruby. The Convergence of Web and Voice Applications using Open Source Software. Justin Grammens Localtone Interactive

Size: px
Start display at page:

Download "VOIP and Ruby. The Convergence of Web and Voice Applications using Open Source Software. Justin Grammens Localtone Interactive justin@localtone."

Transcription

1 VOIP and Ruby The Convergence of Web and Voice Applications using Open Source Software Justin Grammens Localtone Interactive

2 VOIP is NOT About Cheap Phone Calls Other companies are already doing this cheaper and better.

3 VOIP Applications! It's about the applications that we can build!

4 What We Will Cover Why do VOIP now? Asterisk Adhearsion Telegraph Demos

5 Why Now? Only recently has good Open Source Software been developed ( Rails / Asterisk ) Telecoms are slow to react Few applications merge voice and web Cell phone are everywhere!

6 Why Now? 2.7 Billion mobile phones. 1.4 Billion fixed-lines. 1/3 of Internet Users access the internet from their mobile phone. iphone has shown consumers why they need the interactive internet on their phone. Others will follow. Most phone users can be identified by a standardized numerical system.

7 VOIP / Web Analogy Technology Web VOIP Protocol HTTP FTP RTP SIP Industry Standard IAX Asterisk Specific H.323 Obsolete Jingle Gtalk Skype - Proprietary Codec gzip, jpg, gif, mp3, ogg, wma, flv, mpeg, avi g.711 high bandwidth gsm medium bandwidth g.729 low bandwidth Server Apache / Lighttpd Asterisk, Freeswitch Interactivity CGI AGI Asterisk Gateway Interface AMI Asterisk Manager Interface

8 Asterisk + Open Source Private Branch Exchange (PBX) + Very powerful and flexible + Relatively Stable - Messy to deal with in terms of extending functionality. +++ Free!

9 Asterisk : Terminology Channel A channel is what can setup and receive calls. Dialplan Script of what to do with a call. Written in the asterisk macro language. AGI Stdin/out TCP method allowing external applications to dynamically write dialplans. AMI Allows sending of commands and listen for stateful events.

10 Typical Voice System VOIP Clients SIP Rails PSTN Network Origination/ Termination Server SIP / IAX Asterisk Server AGI / AMI / Adhearsion / Telegraph PSTN Network Analog Interface Card Zaptel / Other

11 Asterisk Dialplan Language [demo] ; Sample from Asterisk configuration extensions.conf file ; ; We start with what to do when a call first comes in. ; exten => s,1,wait(1) ; Wait a second, just for fun exten => s,n,answer ; Answer the line exten => s,n,set(timeout(digit)=5) ; Set Digit Timeout to 5 seconds exten => s,n,set(timeout(response)=10) ; Set Response Timeout to 10 seconds exten => s,n(restart),background(demo-congrats) ; Play a congratulatory message exten => s,n(instruct),background(demo-instruct); Play some instructions exten => s,n,waitexten ; Wait for an extension to be dialed. exten => 2,1,BackGround(demo-moreinfo) exten => 2,n,Goto(s,instruct) ; Give some more information. exten => 3,1,Set(LANGUAGE()=fr) exten => 3,n,Goto(s,restart) ; Set language to french ; Start with the congratulations exten => 1000,1,Goto(default,s,1)

12 Difficulties in Asterisk Conditional Loops Error Handling Complex Data Structure Date and time handling Database / LDAP Integration RegEx Pattern Matching Extending the language Portability - Asterisk v.s Freeswitch, etc. Variables Object Oriented Design

13 Ruby / Asterisk Integration Tools RAGI Just for AGI. Not integrated with Rails. No longer active. RAMI Just for Management Interface. No Rails Integration. Not Active. Adhearsion Active. Good for writing pure voice applications. Not tied with Rails (but can be without too much effort). Telegraph Active. Tightly integrated with Rails. Embraces the Voice/Web Analogy.

14 Adhearsion Standalone server that talks with Asterisk Developed by Jay Phillips of Codemecca Open Source Current version is Development on 0.8 is nearly complete. Lots of new changes.

15 Adhearsion Put the line below in extensions.conf Tells Asterisk to process all calls by our Adhearsion server exten => _X.,1,Agi(agi:// ) or... when extension 888 is dialed. exten => 888,1,Agi(agi:// )

16 Adhearsion - dialplan.rb adhearsion { play %w(press-1 for minneapolis press-2 for chicago or press-3 for dallas weather otherwise-press 4) selection = input() w = new_weather case selection when '1' then play w.weather_report("minneapolis, MN") when '2' then play w.weather_report("chicago, IL") when '3' then play w.weather_report("dallas, TX") else simon = new_simon_game simon.start end }

17 Adhearsion - Demos SIP Phone XLite Asterisk extensions.conf [ adhearsion ] exten => 8000,1, Agi(agi://...) Adhearsion dialplan.rb adhearsion { code.. code.. }

18 Adhearsion - Demo Notes: Start up Asterisk : sudo asterisk Show asterisk CLI. Start up Adhearsion 0.8 Server : ~/development/adhersion/trunk/bin/ahn start. in the rumadhearsion directory Point Xlite Phone to Localhost dial extension 8000

19 Adhearsion Weather Demo Demo #1 - Weather - Parses data from Yahoo RSS feed <yweather:forecast day="mon" date="31 Dec 2007" low="6" high="19" text="flurries" code="13" /> rep = %W(weather is-currently #{w.current.temp} degrees today high #{today.high} low #{today.low}) + w.current.desc

20 Adhearsion Simon Says Demo #2 - Play Simon Says Game def verify_attempt if attempt_correct? call_context.play 'good' else call_context.play %W(#{number.size - 1} times wrong-try-again-smarty) reset end end

21 Adhearsion Write Ruby in our dial plans! Ability to use any Ruby gems we need (Active Record, etc.) Test and debug our application in isolation. Bring OO practices to VOIP development

22 Adhearsion It's abstracted and portable across other PBXes It's simple It's extensible It's readable It's maintainable It's fun!

23 Adhearsion Where's the Rails? Not directly integrated with Rails by choice. Written to stand on it's own, but you can link in your models using ActiveRecord. Looking for VOIP in the MVC framework? Look no further than Telegraph...

24 Telegraph Written by a company named Idapted. Extracted from production application ( Idapted's distributed voice system for English language learning EnglishQuad ) Started with RAGI / RAMI Tightly Integrated with the Rails/Web Interface They claim it embraces the Voice/Web analogy

25 Telegraph Installs into any Rails project as a plugin script/plugin install svn://rubyforge.org/var/svn/telegraph/trunk start up the server script/agi_server Interfaces with the gateway (incoming calls) script/ami_server Interfaces with the Asterisk manager

26 Telegraph Banking Demo SIP Phone XLite Rails Application Asterisk AGI Server

27 Telegraph Add this to your extensions.conf exten => s, n, AGI(agi://localhost/account) respond_to do wants #r index.html wants.html { render } # Telegraph allows render_voice # which uses the index.voice file wants.voice{ render_voice } end

28 Banking Demo index.voice: voice.play "welcome-to-demo voice.link_to_dtmf 'banking-main-menu' do link 1, :action=>'new' link 2, :action=>'list' link :default, :action=>'index' end

29 Telegraph - Demo Start up telegraph server : telegraph/banking_demo ruby script/asterisk_server ruby script/server Visit Dial Extension 9000

30 Real World Application estara Offers a service where a user browsing a site can enter their phone number. The system will dial their number, ask the person to hold and then dial customer service. We'll do this.

31 Demo Topology Cell Phone PSTN Origination/ Termination Server Internet Asterisk Rails Application AGI Demos: 1. Using the browser to initiate phone call. 2. Who Wants To Be A Billionaire game. AMI

32 Demo Using PSTN Telegraph Demos - Use the browser to initiate a wakeup call. Use the browser to initiate phone calls to 10 digit phone numbers and bridge the calls Use the browser to initiate a call and verify correct code was entered. Adhearsion Demo - Adhearsion My Who Wants To Be A Billionaire application.

33 Resources

VOIP and Ruby. The Convergence of Web and Voice Applications using Open Source Software. Justin Grammens Localtone Interactive justin@localtone.

VOIP and Ruby. The Convergence of Web and Voice Applications using Open Source Software. Justin Grammens Localtone Interactive justin@localtone. VOIP and Ruby The Convergence of Web and Voice Applications using Open Source Software Justin Grammens Localtone Interactive justin@localtone.com VOIP is NOT About Cheap Phone Calls Other companies are

More information

Open Source Telephony Projects as an Application Development Platform. Frederic Dickey (fdickey@sangoma.com) Director Product Management

Open Source Telephony Projects as an Application Development Platform. Frederic Dickey (fdickey@sangoma.com) Director Product Management Open Source Telephony Projects as an Application Development Platform Frederic Dickey (fdickey@sangoma.com) Director Product Management About this presentation For newcomers to Asterisk For long time CTI

More information

and Voice Applications Eyal Wirsansky, Verso Technologies JaxJUG

and Voice Applications Eyal Wirsansky, Verso Technologies JaxJUG Voice Over IP, and Voice Applications Eyal Wirsansky, Verso Technologies JaxJUG Analog Telephony Mr. W AG Bell X What the *!@# is aa Switch?? Moving to Digital Voice (TDM) Separation of Voice and Signaling

More information

Overview of Asterisk (*) Jeff Gunther

Overview of Asterisk (*) Jeff Gunther Overview of Asterisk (*) Jeff Gunther Agenda Background Introduction to Asterisk and review the core components of it s architecture. Exploration of Asterisk s telephony and call features. Review some

More information

Micronet VoIP Solution with Asterisk

Micronet VoIP Solution with Asterisk Application Note Micronet VoIP Solution with Asterisk 1. Introduction This is the document for the applications between Micronet units and Asterisk IP PBX. It will show you some basic configurations in

More information

Introduction. What is DUNDi? Configuring Asterisk for use with DUNDi

Introduction. What is DUNDi? Configuring Asterisk for use with DUNDi Introduction This paper will explore how to configure and setup the DUNDi directory service on your Asterisk PBX system. DUNDi is not very hard to configure in Asterisk, however at the time of this writing,

More information

Mediatrix 3000 with Asterisk June 22, 2011

Mediatrix 3000 with Asterisk June 22, 2011 Mediatrix 3000 with Asterisk June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Network Topology... 3 Equipment Detail... 3 Configuration of the Fax Extension... 4 Configuration

More information

VoIP and FreeBSD. The daemon meets the phone. May 15th, 2008 University of Ottawa,, Ottawa, Canada Massimiliano Stucchi stucchi@briantel.

VoIP and FreeBSD. The daemon meets the phone. May 15th, 2008 University of Ottawa,, Ottawa, Canada Massimiliano Stucchi stucchi@briantel. VoIP and FreeBSD The daemon meets the phone May 15th, 2008 University of Ottawa,, Ottawa, Canada Massimiliano Stucchi stucchi@briantel.com Agenda Introduction Terms Introduction to Asterisk key concepts

More information

Asterisk & ENUM. Extending the Open Source PBX. Michael Haberler, IPA Otmar Lendl, nic.at

Asterisk & ENUM. Extending the Open Source PBX. Michael Haberler, IPA Otmar Lendl, nic.at Asterisk & ENUM Extending the Open Source PBX Michael Haberler, IPA Otmar Lendl, nic.at Why a ENUM-enable a PBX? your PBX doubles as an IP/PSTN gateway for your existing numbers becomes a dual contact

More information

Introduction to VOIP. Stephen Okay Abdus Salam Int l Center for Theoretical Physics Trieste, Italy, February 21, 2007

Introduction to VOIP. Stephen Okay Abdus Salam Int l Center for Theoretical Physics Trieste, Italy, February 21, 2007 Introduction to VOIP Stephen Okay Abdus Salam Int l Center for Theoretical Physics Trieste, Italy, February 21, 2007 Intro to VOIP Classic Telephony Data Networks(Review) VOIP What it is Protocols Hardware

More information

Open source VoIP Networks

Open source VoIP Networks Open source VoIP Networks Standard PC hardware inexpensive add-in vs. embedded designs Ing. Bruno Impens Overview History Comparison PC - Embedded More on VoIP VoIP Hardware VoIP more than talk More...

More information

Basic configuration of the GXW410x with Asterisk

Basic configuration of the GXW410x with Asterisk Basic configuration of the GXW410x with Asterisk Please note that due to the customizable nature of both the GXW410x and Asterisk and the vast deployment possibilities, these instructions should be taken

More information

Asterisk. http://www.asterisk.org. http://www.kismetwireless.net/presentations.shtml. Michael Kershaw <dragorn@kismetwireless.net>

Asterisk. http://www.asterisk.org. http://www.kismetwireless.net/presentations.shtml. Michael Kershaw <dragorn@kismetwireless.net> Asterisk * http://www.asterisk.org What Asterisk Can Do Voice Over IP (VOIP) Physical phone switch (PBX) Software phone switch Answering machine Call trees (Press 1 to...) VOIP Voice Over IP: Make telephone

More information

Setup Guide: on the MyNetFone Service. Revision History

Setup Guide: on the MyNetFone Service. Revision History Setup Guide: on the MyNetFone Service Revision History Version Author Revision Description Release Date 1.0 Sampson So Initial Draft 02/01/2008 2.0 Sampson So Update 27/09/2011 1 Table of Contents Introduction...

More information

Asterisk Overview. Berkeley In Munich Tech Talks 17.01.2007. prepared by. Emil Stoyanov emosto@web.de stoyanov@kiax.org

Asterisk Overview. Berkeley In Munich Tech Talks 17.01.2007. prepared by. Emil Stoyanov emosto@web.de stoyanov@kiax.org Asterisk Overview Berkeley In Munich Tech Talks 17.01.2007 prepared by Emil Stoyanov emosto@web.de stoyanov@kiax.org Contents What is Asterisk? Usage Scenarios Characteristics & Capabilities Architecture

More information

VOIP with Asterisk & Perl

VOIP with Asterisk & Perl VOIP with Asterisk & Perl By: Mike Frager 11/2011 The Elements of PSTN - Public Switched Telephone Network, the pre-internet phone system: land-lines & cell-phones. DID - Direct

More information

How To Use An Asterisk Server For A Phone Or Internet Communication

How To Use An Asterisk Server For A Phone Or Internet Communication Innovative Applications Using Asterisk Open Source PBX By: Mikhail Torres, Managing Director, EACOMM Corporation Introduction Asterisk provides a unique and very powerful platform to enable converged applications.

More information

VoIP Workshop PacNOG3

VoIP Workshop PacNOG3 VoIP Workshop PacNOG3 Rarotonga, Cook Islands June 2007 Labs 1-4, Asterisk Lab 5, INOC-DBA Lab 6-7, Cisco Voice Gateways Lab 8, CODECS Page 1 of 13 Lab Summary Server logins are as you have set up in previous

More information

Introduction p. 7 About This Book p. 1 Conventions Used in This Book p. 2 What You Don't Have to Read p. 2 Foolish Assumptions p. 2 How This Book Is

Introduction p. 7 About This Book p. 1 Conventions Used in This Book p. 2 What You Don't Have to Read p. 2 Foolish Assumptions p. 2 How This Book Is Foreword p. xxi Introduction p. 7 About This Book p. 1 Conventions Used in This Book p. 2 What You Don't Have to Read p. 2 Foolish Assumptions p. 2 How This Book Is Organized p. 3 Introducing Asterisk!

More information

IP Telephony with Asterisk. Sunday A. Folayan

IP Telephony with Asterisk. Sunday A. Folayan IP Telephony with Asterisk Sunday A. Folayan There lived the PSTN. A few years ago, everyone struggled to convert data (IP) into sound, and move it over the Public Switched Telephone Network (PSTN) infrastructure

More information

Wildix Management System (WMS) White Paper

Wildix Management System (WMS) White Paper Wildix Management System (WMS) White Paper February 2007 Author: Giuseppe Innamorato Wildix Management System White Paper Status: Draft 0.1 Page 1 Index: 1. Management Summary...3 2. Document purpose...3

More information

Fig. Setting up of a VoIP call. Fig. Experimental setup

Fig. Setting up of a VoIP call. Fig. Experimental setup Volume 5, Issue 6, June 2015 ISSN: 2277 128X International Journal of Advanced Research in Computer Science and Software Engineering Research Paper Available online at: www.ijarcsse.com Asterisk VoIP Private

More information

Chapter 1 - Introduction

Chapter 1 - Introduction Chapter 1 - Introduction Asterisk is revolutionary, reliable, scalable, open source, free software that makes possible powerful enterprise telephone systems. Asterisk systems are in use world-wide, reliably

More information

So, you need to deploy a Private Branch exchange

So, you need to deploy a Private Branch exchange Asterisk Open-Source PBX System Use one system to manage voice over IP and conventional phone lines, manage voice mail and run CGI-like applications for phone users. BY BRETT SCHWARZ So, you need to deploy

More information

VoIP-PSTN Interoperability by Asterisk and SS7 Signalling

VoIP-PSTN Interoperability by Asterisk and SS7 Signalling VoIP-PSTN Interoperability by Asterisk and SS7 Signalling Jan Rudinsky CESNET, z. s. p. o. Zikova 4, 160 00 Praha 6, Czech Republic rudinsky@cesnet.cz Abstract. PSTN, the world's circuit-switched network,

More information

Internet Technology Voice over IP

Internet Technology Voice over IP Internet Technology Voice over IP Peter Gradwell BT Advert from 1980s Page 2 http://www.youtube.com/v/o0h65_pag04 Welcome to Gradwell Gradwell provides technology for every line on your business card Every

More information

Configuring a Pure-IP SIP Trunk in Lync 2013

Configuring a Pure-IP SIP Trunk in Lync 2013 Configuring a Pure-IP SIP Trunk in Lync 2013 Contents Configuring a Pure-IP SIP Trunk in Lync 2013... 1 Introduction - Product version: Microsoft Lync Server 2013... 2 Pure-IP SIP Trunk configuration tasks...

More information

Practical Guide. How to setup VoIP Infrastructure using AsteriskNOW

Practical Guide. How to setup VoIP Infrastructure using AsteriskNOW Practical Guide How to setup VoIP Infrastructure using AsteriskNOW Table of Contents 1. Background...1 2. The VoIP scenarios...2 3. Before getting started...3 3.1 Training Kits...3 3.2 Software requirements...3

More information

Specialty Answering Service. All rights reserved.

Specialty Answering Service. All rights reserved. 0 Contents 1 Introduction... 3 2 Features... 4 2.1 Hardware Requirement... 4 2.2 Protocol Support... 4 2.3 Configuration... 4 2.4 Applications... 5 2.5 Graphical User Interfaces... 5 3 History and Evolution

More information

Open Source VoiceXML Interpreter over Asterisk for Use in IVR Applications

Open Source VoiceXML Interpreter over Asterisk for Use in IVR Applications Open Source VoiceXML Interpreter over Asterisk for Use in IVR Applications Lerato Lerato, Maletšabisa Molapo and Lehlohonolo Khoase Dept. of Maths and Computer Science, National University of Lesotho Roma

More information

IP-PBX Quick Start Guide

IP-PBX Quick Start Guide IP-PBX Quick Start Guide Introduce... 3 Configure and set up the IP-PBX... 4 How to change the IP address... 7 Set up extensions and make internal calls... 8 How to make calls via the FXO port... 10 How

More information

Avaya IP Office 8.1 Configuration Guide

Avaya IP Office 8.1 Configuration Guide Avaya IP Office 8.1 Configuration Guide Performed By tekvizion PVS, Inc. Contact: 214-242-5900 www.tekvizion.com Revision: 1.1 Date: 10/14/2013 Copyright 2013 by tekvizion PVS, Inc. All Rights Reserved.

More information

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment Application Note May 2009 Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment 2009 Cisco Systems, Inc. All rights reserved. Page 1 of 20 Contents Introduction 3 Audience 3 Scope

More information

IBM WebSphere Application Server Communications Enabled Applications Setup guide

IBM WebSphere Application Server Communications Enabled Applications Setup guide Copyright IBM Corporation 2009, 2011 All rights reserved IBM WebSphere Application Server Communications Enabled Applications Setup guide What this exercise is about... 1 Lab requirements... 2 What you

More information

Asterisk. the general purpose Open Source Telephony platform. a Advanced Scenarios. Klaus Peter Junghanns (kapejod) 2004 Junghanns.

Asterisk. the general purpose Open Source Telephony platform. a Advanced Scenarios. Klaus Peter Junghanns (kapejod) 2004 Junghanns. Asterisk the general purpose Open Source Telephony platform a Advanced Scenarios Klaus Peter Junghanns (kapejod) Junghanns.NET GmbH http://www.junghanns.net/asterisk/ Traditional Setup of a 3 location

More information

Application Note. Using a Dialogic Media Gateway Series as a PSTN Gateway with an Asterisk IP-PBX Server

Application Note. Using a Dialogic Media Gateway Series as a PSTN Gateway with an Asterisk IP-PBX Server Using a Dialogic Media Gateway Series as a PSTN Gateway with an Asterisk IP-PBX Server Using a Dialogic Media Gateway Series as a PSTN Gateway with an Asterisk IP-PBX Server Executive Summary This application

More information

OpenVox GSM Gateway Function Manual

OpenVox GSM Gateway Function Manual Rev: 1.0 Date: April 15, 2014 From: OpenVox support group Contact info: support@openvox.cn OpenVox GSM Gateway Function Manual OpenVox VoxStack GSM Gateway is a feature-rich, highly available and flexible

More information

Asterisk: A Non-Technical Overview

Asterisk: A Non-Technical Overview Asterisk: A Non-Technical Overview Nasser K. Manesh nasser@millenigence.com Millenigence, Inc. 5000 Birch St., Suite 8100 Newport Beach, CA 92660 June 2004, Revised December 2004 Executive Summary Asterisk

More information

VoIP and IP Telephony @ IT Tralee

VoIP and IP Telephony @ IT Tralee VoIP and IP Telephony @ IT Tralee chris.bradshaw@staff.ittralee.ie Presentation outline: Basic overview of IP telephony and technology Detailed overview of VoIP @ IT Tralee deployment How IPT has benefited

More information

A sysadmin s view of VoIP

A sysadmin s view of VoIP A sysadmin s view of VoIP Ewen McNeill Naos Ltd 2006/01/23 Sysadmin Miniconf, Linux.Conf.Au 2006 Ewen McNeill (Naos Ltd) A sysadmin s view of VoIP Sysadmin Miniconf LCA2006 1 / 40 Outline

More information

Leveraging Asterisk to Deliver Large Scale VoIP Services in a Carrier Environment. JR Richardson

Leveraging Asterisk to Deliver Large Scale VoIP Services in a Carrier Environment. JR Richardson Leveraging Asterisk to Deliver Large Scale VoIP Services in a Carrier Environment JR Richardson Early VoIP Environment Telecom Act of 1996, mass competition, Telco's needed value add features and capabilities,

More information

NetVanta 7100 Exercise Service Provider SIP Trunk

NetVanta 7100 Exercise Service Provider SIP Trunk NetVanta 7100 Exercise Service Provider SIP Trunk PSTN NetVanta 7100 FXS 0/1 x2001 SIP Eth 0/0 x2004 SIP Server 172.23.102.87 Hosted by x2003 www.voxitas.com In this exercise, you will create a SIP trunk

More information

CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005

CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 From the Cisco CVOICE 642-432 Exam Topics Voice over IP Technologies Describe the similarities and differences between PSTN and VoIP including

More information

Connecting Your Enterprise With Asterisk: IAX to Carriers. Dayton Turner Voxter Communications

Connecting Your Enterprise With Asterisk: IAX to Carriers. Dayton Turner Voxter Communications Connecting Your Enterprise With Asterisk: IAX to Carriers Dayton Turner Voxter Communications What is IAX? Inter Asterisk exchange Developed by Digium and the Open Source Community Alternative to SIP,

More information

VOIP (Voice Over Internet Protocol) Hacking-Fake Calling

VOIP (Voice Over Internet Protocol) Hacking-Fake Calling VOIP (Voice Over Internet Protocol) Hacking-Fake Calling Author: Avinash Singh Co-Author: Akash Shukla Avinash Singh Corporate Trainer (Virscent Technologies Pvt. Ltd.) Appin Certified Ethical Hacker (ACEH)

More information

640-460 - Implementing Cisco IOS Unified Communications (IIUC)

640-460 - Implementing Cisco IOS Unified Communications (IIUC) 640-460 - Implementing Cisco IOS Unified Communications (IIUC) Course Introduction Course Introduction Module 1 - Cisco Unified Communications System Introduction Cisco Unified Communications System Introduction

More information

Unicorn60x0 IP ANALOG GATEWAY ASTERISK CONFIGURATION

Unicorn60x0 IP ANALOG GATEWAY ASTERISK CONFIGURATION Unicorn60x0 IP ANALOG GATEWAY ASTERISK CONFIGURATION BASIC CONFIGURATION OF THE Unicorn60x0 WITH ASTERISK Due to the various deployment possibilities of the Unicorn60x0 and Asterisk, this configuration

More information

Configuring Positron s V114 as a VoIP gateway for a 3cx system

Configuring Positron s V114 as a VoIP gateway for a 3cx system Assumptions: Configuring Positron s V114 as a VoIP gateway for a 3cx system The IP address of the V114 is 192.168.1.2 The IP address of the 3CX PBX System is 192.168.1.110 3CX already has some IP phones

More information

DUNDi, So Easy A Caveman Could Do It!

DUNDi, So Easy A Caveman Could Do It! DUNDi, So Easy A Caveman Could Do It! General Description JR Richardson Engineering for the Masses hubguru@gmail.com DUNDi is a peer-to-peer system for locating Internet gateways to telephony services.

More information

TEL 500 WRITE UP WEEK 8 FREE PBX SIP LAB SUBMITTED TO: PROF. RONNY BULL BY: ANUSHA ALIGAPALLY

TEL 500 WRITE UP WEEK 8 FREE PBX SIP LAB SUBMITTED TO: PROF. RONNY BULL BY: ANUSHA ALIGAPALLY TEL 500 WRITE UP WEEK 8 FREE PBX SIP LAB SUBMITTED TO: PROF. RONNY BULL BY: ANUSHA ALIGAPALLY DATE: 11/05/2014 ABSTRACT: Private Branch Exchange has multiple phones connected to it which are in the same

More information

An introduction to PHP & AGI

An introduction to PHP & AGI February 12 th - 13 th 2007 PHP Phone Home An introduction to PHP & AGI This talk is not...... about REST... about Web Services... by Paul Reinheimer In fact, I am not Paul Reinheimer The REST talk is

More information

Quick Installation Guide

Quick Installation Guide Quick Installation Guide MegaPBX Version 2.1 Quick Installation Guide v2.1 www.allo.com 2 Table of Contents Initial Setup of MegaPBX... 4 Notification LEDs (On the Front Panel of the Gateway)... 5 Create

More information

IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week

IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week Course Title: No. of Hours: IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week 1 Course Duration: 3 Months (12weeks) No. Of Hours: 7 Hrs./Day- 5 days/week.

More information

IP Based Voice Server Application With PBX Using Free SWITCH ISSN 2319-9725

IP Based Voice Server Application With PBX Using Free SWITCH ISSN 2319-9725 IP Based Voice Server Application With PBX Using Free SWITCH ISSN 2319-9725 Sachin Mallikarjun Hunur Shreyas S M Abstract: In today s world, individuals are able to build phone systems that outperform

More information

Secure, Multi-lateral Peering with Asterisk TM V1.2 22 November 2005

Secure, Multi-lateral Peering with Asterisk TM V1.2 22 November 2005 Secure, Multi-lateral Peering with Asterisk TM V1.2 22 November 2005 Contents Multi-lateral Peering: Why... 1 Current Deployments... 1 Distributed Architecture... 1 Centralized Architecture... 2 Multi-lateral

More information

VoIP and IP Telephony

VoIP and IP Telephony VoIP and IP Telephony Reach Out and Ping Someone ISAC Spring School 2006 21 March 2006 Anthony Kava, Sr. Network Admin Pottawattamie County IT Definition VoIP Voice over Internet Protocol Voice Transport

More information

Department of Communications and Networking. S-38.2131/3133 Networking Technology, laboratory course A/B

Department of Communications and Networking. S-38.2131/3133 Networking Technology, laboratory course A/B Department of Communications and Networking S-38.2131/3133 Networking Technology, laboratory course A/B Work Number 29: VoIP Student Edition Preliminary Exercises and Laboratory Assignments Original document

More information

How to Configure the Allworx 6x, 24x and 48x for use with Integra Telecom SIP Solutions

How to Configure the Allworx 6x, 24x and 48x for use with Integra Telecom SIP Solutions How to Configure the Allworx 6x, 24x and 48x for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the Allworx 6x IP PBX to connect to Integra Telecom

More information

Enterprise Voice and Online Services with Microsoft Lync Server 2013

Enterprise Voice and Online Services with Microsoft Lync Server 2013 Course 20337B: Enterprise Voice and Online Services with Microsoft Lync Server 2013 Course Details Course Outline Module 1: Voice Architecture This module introduce Enterprise Voice features of Lync Server

More information

Applications between Asotel VoIP and Asterisk

Applications between Asotel VoIP and Asterisk Applications between Asotel VoIP and Asterisk This document is describing the configuring manner of registering and communicating with Asterisk only. Please visit the official WEB of Asterisk http://www.asterisk,

More information

Shared Components PSTN gateways PSTN gateways New IP/PSTN Gateway Define New IP/PSTN Gateway Define the PSTN Gateway FQDN FQDN Next

Shared Components PSTN gateways PSTN gateways New IP/PSTN Gateway Define New IP/PSTN Gateway Define the PSTN Gateway FQDN FQDN Next Microsoft Lync 2013 Integration with VoIP.co.uk SAFEgateway In order to integrate Microsoft Lync 2013 with the VoIP.co.uk SAFEgateway you must configure both the Microsoft Lync server and the VoIP.co.uk

More information

Stack Num IP Username Password

Stack Num IP Username Password This document applies to OpenVox GSM Gateway WGW1002G,VS-GW1202-4/8G and VS-GW1600 series. There are two RJ45 Network ports, ETH1 and ETH2. If you choose ETH1, you can access Board 1 only, and access other

More information

QUICK START GUIDE RELEASE 7

QUICK START GUIDE RELEASE 7 QUICK START GUIDE RELEASE 7 VOICENT AUTOREMINDER TM VOICENT BROADCASTBYPHONE TM VOICENT AGENTDIALER TM VOICENT FLEX PBX TM VOICENT IVR STUDIO TM VOICENT TELEPHONY CRM TM VOICENT GATEWAY TM TABLE OF CONTENT

More information

NOC Workshop VoIP in the NOC labs SANOG10

NOC Workshop VoIP in the NOC labs SANOG10 NOC Workshop VoIP in the NOC labs SANOG10 New Delhi, India August 29 - September 2, 2007 Page 1 of 10 Lab Summary NOC Workshop, SANOG10 - VoIP in the NOC We only have limited time for this portion of the

More information

MS 20337A: Enterprise Voice and Online Services with Microsoft Lync 2013

MS 20337A: Enterprise Voice and Online Services with Microsoft Lync 2013 MS 20337A: Enterprise Voice and Online Services with Microsoft Lync 2013 Description: This five-day instructor-led course teaches how to design and configure Enterprise Voice and Online Services in Microsoft

More information

2N OfficeRoute. 2N OfficeRoute & Siemens HiPath (series 3000) connected via SIP trunk. Quick guide. www.2n.cz. Version 1.00

2N OfficeRoute. 2N OfficeRoute & Siemens HiPath (series 3000) connected via SIP trunk. Quick guide. www.2n.cz. Version 1.00 2N OfficeRoute 2N OfficeRoute & Siemens HiPath (series 3000) connected via SIP trunk Quick guide Version 1.00 www.2n.cz 1 2N OfficeRoute has these parameters: IP address 192.168.1.120 Incoming port: 5060

More information

2- Technical Training (9 weeks) 3- Applied Project (3 weeks) 4- On Job Training (OJT) (4 weeks)

2- Technical Training (9 weeks) 3- Applied Project (3 weeks) 4- On Job Training (OJT) (4 weeks) Course Title: Prerequisites: Training Program (5 months) IP Implementation in Private Branch Exchanges Must fresh graduates Communication/Electronics Engineers" 1- Soft Skills Training (4 weeks) 1. Communication

More information

Ryan Brown October 9, 2004 The Burgh Live, LLC. Voice over IP using Asterisk (*)

Ryan Brown October 9, 2004 The Burgh Live, LLC. Voice over IP using Asterisk (*) Ryan Brown October 9, 2004 The Burgh Live, LLC Voice over IP using Asterisk (*) What is Asterisk? * (http://www.asterisk.org www.asterisk.org) ) is an Open Source Private Branch Exchange (PBX) and Interactive

More information

How to use IP-0x to connect to Skype

How to use IP-0x to connect to Skype How to use IP-0x to connect to Skype Product Guide Version: 1.0 2010-10-14 Content CONTACT ATCOM... 2 HOW TO USE IP-0X TO CONNECT TO SKYPE... 2 YOU NEED TO SIGN UP A SKYPE MANAGER ACCOUNT.... 2 REGISTER

More information

A Guide to Connecting to FreePBX

A Guide to Connecting to FreePBX A Guide to Connecting to FreePBX FreePBX is a basic web Graphical User Interface that manages Asterisk PBX. It includes many features available in other PBX systems such as voice mail, conference calling,

More information

Configuring an Etherspeak SIP Trunk in Microsoft Lync 2013

Configuring an Etherspeak SIP Trunk in Microsoft Lync 2013 Configuring an Etherspeak SIP Trunk in Microsoft Lync 2013 This is to cover the steps needed for basic functionality to communicate with Etherspeak s SIP trunking service. Many environments are different

More information

Asterisk Business Edition TM Digium Partner Certification

Asterisk Business Edition TM Digium Partner Certification Asterisk Business Edition TM Digium Partner Certification Cyberdata VoIPSpeaker Interoperability Report April 2007 Digium, Inc. 150 West Park Loop, Suite 100 Huntsville, AL 35806 Main Number: 256.428.6000

More information

EarthLink Business SIP Trunking. Asterisk 11.2 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. Asterisk 11.2 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking Asterisk 11.2 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0

More information

ilanga: A Next Generation VoIP-based, TDMenabled

ilanga: A Next Generation VoIP-based, TDMenabled ilanga: A Next Generation VoIP-based, TDMenabled PBX J. Penton, A. Terzoli Computer Science Department Rhodes University Grahamstown, 6140 Email: j.penton@ru.ac.za Tel: (046) 603 8640; Fax: (046) 636 1915

More information

TEL-500 Project Report. Auto-Dialler System. Voice Communications. Done By: - AKASH ANANTHANARAYANAN SANJEEVAKUMAR DEVARAJA

TEL-500 Project Report. Auto-Dialler System. Voice Communications. Done By: - AKASH ANANTHANARAYANAN SANJEEVAKUMAR DEVARAJA TEL-500 Project Report Auto-Dialler System Voice Communications Done By: - AKASH ANANTHANARAYANAN SANJEEVAKUMAR DEVARAJA 1 Index Page Contents Page Number Abstract 3 Introduction 4 Flow Chart 5 Resources

More information

Crash Course in Asterisk

Crash Course in Asterisk Crash Course in Asterisk Despite its name, Asterisk is no mere footnote to the IP-PBX market. The open source product is one of the most disruptive technologies in the industry. Here s what you need to

More information

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents First Steps... 3 Identifying your MAC Address... 3 Identifying your Dynamic IP Address...

More information

Vulnerability Scan. January 6, 2015

Vulnerability Scan. January 6, 2015 Vulnerability Scan January 6, 2015 Results of Vulnerability Security Scan The results of your Ethos Info Vulnerability Security Scan are detailed below. The scan ran from Sat Dec 27 07:07:00 2014 UTC until

More information

This manual contains product information for the GSM Series cards. The manual is organized in the following manner:

This manual contains product information for the GSM Series cards. The manual is organized in the following manner: Allo.com. 2012 All rights reserved. No part of this publication may be copied, distributed, transmitted, transcribed, stored in a retrieval system, or translated into any human or computer

More information

Personalizing Your Individual Phone Line Setup For assistance, please call 1-800-453-2251 ext. 102.

Personalizing Your Individual Phone Line Setup For assistance, please call 1-800-453-2251 ext. 102. Personalizing Your Individual Phone Line Setup For assistance, please call 1-800-453-2251 ext. 102. With these instructions, you will: 1. Record your greeting. 2. Configure your 911 setting. 3. Learn how

More information

http://webrtcbook.com

http://webrtcbook.com ! This is a sample chapter of WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web by Alan B. Johnston and Daniel C. Burnett, Second Edition. For more information or to buy the paperback or ebook

More information

Contents. Specialty Answering Service. All rights reserved.

Contents. Specialty Answering Service. All rights reserved. Contents 1 Introduction... 2 2 PBX... 3 3 IP PBX... 4 3.1 How It Works... 4 3.2 Functions of IP PBX... 5 3.3 Benefits of IP PBX... 5 4 Evolution of IP PBX... 6 4.1 Fuelling Factors... 6 4.1.1 Demands from

More information

Grandstream Networks, Inc. How to Integrate UCM6100 with Microsoft Lync Server

Grandstream Networks, Inc. How to Integrate UCM6100 with Microsoft Lync Server Grandstream Networks, Inc. How to Integrate UCM6100 with Microsoft Lync Server Index Table of Contents OVERVIEW... 3 UCM6100 CONFIGURATION... 4 STEP 1: CREATE SIP PEER TRUNK... 4 STEP 2: CONFIGURE OUTBOUND

More information

How to Configure the NEC SV8100 for use with Integra Telecom SIP Solutions

How to Configure the NEC SV8100 for use with Integra Telecom SIP Solutions How to Configure the NEC SV8100 for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the NEC SV8100 IP PBX to connect to Integra Telecom SIP trunks.

More information

3CX IP PBX Phone System Technical Training Deerfield.com

3CX IP PBX Phone System Technical Training Deerfield.com 3CX IP PBX Phone System Technical Training Deerfield.com 1 Copyright 2002 ACNielsen Agenda High level overview of SIP & RTP Call set-up, ports needed and codecs SIP phone configuration VOIP Gateway configuration

More information

Integrating Asterisk FreePBX with Lync Server 2010

Integrating Asterisk FreePBX with Lync Server 2010 1 Integrating Asterisk FreePBX with Lync Server 2010 Author: Baaskar R 1 www.baaskarcharles.com 2 Integrating Asterisk FreePBX with Lync Server 2010... 1 AsteriskNow package Source... 3 Installing AsteriskNow...

More information

Configuration Notes 290

Configuration Notes 290 Configuring Mediatrix 41xx FXS Gateway with the Asterisk IP PBX System June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 About Mediatrix 41xx Series FXS Gateways...

More information

How to Configure the Toshiba Strata CIX for use with Integra Telecom SIP Solutions

How to Configure the Toshiba Strata CIX for use with Integra Telecom SIP Solutions How to Configure the Toshiba Strata CIX for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the Toshiba Strata CIX IP PBX to connect to Integra

More information

Avaya Aura SIP Trunking Training

Avaya Aura SIP Trunking Training Avaya Aura SIP Trunking Training 5 Day Course Lecture & Demo WHO NEEDS TO ATTEND This class is suited to those who are new to administering Avaya systems, need to know how to send calls through an Avaya

More information

Internet Telephony PBX System

Internet Telephony PBX System Internet Telephony PBX System GSM Gateway PSTN call busy forward to GSM Configuration Copyright PLANET Technology Corporation. All rights reserved. Case 32: PSTN call busy forward to GSM Configuration

More information

Softswitch & Asterisk Billing System

Softswitch & Asterisk Billing System Softswitch & Asterisk Billing System IP Telephony Process and architecture is known as Softswitch. Softswitch is used to bridge traditional PSTN and VoIP by linking PSTN to IP networks and managing traffic

More information

An Introduction to VoIP Protocols

An Introduction to VoIP Protocols An Introduction to VoIP Protocols www.netqos.com Voice over IP (VoIP) offers the vision of a converged network carrying multiple types of traffic (voice, video, and data, to name a few). To carry out this

More information

TEL 500. Voice Communications. Week 1 Write Up. Session Initiation Protocol Lab. Submitted To: Prof Ronny Bull. By: Sai Sharan Korvi

TEL 500. Voice Communications. Week 1 Write Up. Session Initiation Protocol Lab. Submitted To: Prof Ronny Bull. By: Sai Sharan Korvi TEL 500 Voice Communications Week 1 Write Up Session Initiation Protocol Lab Submitted To: Prof Ronny Bull By: Sai Sharan Korvi Date: 09/10/2014 ABSTRACT: Softphone is usually a software which can be used

More information

Building an Asterisk Based Call Center. presented by Matt Florell

Building an Asterisk Based Call Center. presented by Matt Florell Building an Asterisk Based Call Center presented by Matt Florell Inbound Only Call Center Base Asterisk Proprietary options Open-Source Inbound/Outbound options Base Asterisk Inbound Only Base Asterisk

More information

Integration of GSM Module with PC Mother Board (GSM Trunking) WHITE/Technical PAPER. Author: Srinivasa Rao Bommana (srinivasrao.bommana@wipro.

Integration of GSM Module with PC Mother Board (GSM Trunking) WHITE/Technical PAPER. Author: Srinivasa Rao Bommana (srinivasrao.bommana@wipro. (GSM Trunking) WHITE/Technical PAPER Author: Srinivasa Rao Bommana (srinivasrao.bommana@wipro.com) Table of Contents 1. ABSTRACT... 3 2. INTRODUCTION... 3 3. PROPOSED SYSTEM... 4 4. SOLUTION DESCRIPTION...

More information

Running Asterisk in a Corporate Environment: a Beginner s Tale

Running Asterisk in a Corporate Environment: a Beginner s Tale Running Asterisk in a Corporate Environment: a Beginner s Tale Stephen Uhler Sun Microsystems Laboratories 2005, Sun Microsystems Stephen Uhler Astricon 2005 Sun Labs (1/19) How I Got into the Phone Business

More information

Application Notes Rev. 1.0 Last Updated: January 9, 2015

Application Notes Rev. 1.0 Last Updated: January 9, 2015 SBC 1000/2000 Series Configuration Guide with Cisco Unified Call Manager v9.1 for Level 3 Voice Complete SM SIP Trunk Deployments Application Notes Rev. 1.0 Last Updated: January 9, 2015 Contents 1 Document

More information

Enterprise Voice and Online Services with Microsoft Lync Server 2013

Enterprise Voice and Online Services with Microsoft Lync Server 2013 CÔNG TY CỔ PHẦN TRƯỜNG CNTT TÂN ĐỨC TAN DUC INFORMATION TECHNOLOGY SCHOOL JSC LEARN MORE WITH LESS! Enterprise Voice and Online Services with Microsoft Lync Server 2013 Course 20337B: Five days; Instructor-Led

More information

Quick Start Guide CREATING A NEW SITE

Quick Start Guide CREATING A NEW SITE IVY is our complete control panel for managing you or your customers SIP trunks and hosted PBX settings. This guide will help you get up and running with IVY as quickly as possible. First thing we need

More information

Merging Old and New Telephony with Asterisk

Merging Old and New Telephony with Asterisk Merging Old and New Telephony with Asterisk Greg Vance Digium, Inc. gvance@digium.com Asterisk a Global Phenomenon Digium Confidential Digium Confidential What is Asterisk? platform The world s from which

More information