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IT & Software Network & Security Asterisk

Complete Asterisk Training

Learn how to install and configure an Asterisk PBX, covering version 16 | Instructor dCAP since 2006
Bestseller
Rating: 4.4 out of 54.4 (446 ratings)
2,250 students
Created by Flavio E. Goncalves
Last updated 6/2020
English
English [Auto]
30-Day Money-Back Guarantee

What you'll learn

  • Install and configure a complete fully functional PBX
  • Make, Receive, Capture, Record and Transfer Calls
  • Build an IVR (Interactive Voice Response)
  • Configure an audio conference room
  • Configure a Voice Mail
  • Generate CDRs
  • Implement the required basic security
  • Run clients behind NAT
  • Run Asterisk in a cloud such as AWS

Requirements

  • At least two years of experience with Linux
  • At least two years of experience with Local Area Networks
  • Familiarity with virtualized environments such as VirtualBox

Description

This training will teach you how to install Asterisk in an Ubuntu Server, build a complete, fully functional PBX with basic and advanced features. 

Asterisk is a great opportunity for thousands of developers, resellers, system integrators, ITSPs, contact centers and small to large companies. You will have the freedom to deliver your own solutions. No more time spent discussing complicated licensing schemes. In our region 9 out of 10 Contact Centers use Asterisk do deliver its services. 

A large amount of students who took our classes are now providing services or founded companies to work with Asterisk, Many of them developed dialers, call centers and other applications. Other students grew their businesses by  leasing Asterisk boxes as a service, many in the cloud.  There are many opportunities for professionals with Asterisk knowledge and experience. From 2005 to 2018 we have taught more than 2500 students at V.Office.

Asterisk is about generating real life results, the cost per T1/E1 port is less than 20% of the incumbents. You won't pay a penny for features such as Interactive Voice Response, VoiceMail, Fax, Queues and Computer Telephony Integration.

Asterisk is much easier than you think, any person with a small knowledge in Linux is capable to build a complete PBX.

The reason I jumped at Asterisk in 2004 was that I was frustrated with the cost of proprietary IP PBXs. I was upset by our vendors forcing us to invest thousands of dollars to sell their products controlling the market. I wanted to deliver for my customers, the best, the simplest and be the real trusted advisor, not only the sales pitcher. Sorry,  I can't sell what I don't believe.   I decided to investigate Asterisk and from the moment I installed at the first time, I knew it would lead to a revolution in telephony. Asterisk is not only a PBX, it is a sophisticated phone system. With Asterisk you can build PBXs, Voicemail servers, ITSP providers, Contact Centers and Application Servers. 

I decided to write a book and it was published in 2005, named "Configuration Guide for Asterisk PBX", translated to Portuguese and Spanish. After this experience, I got my dCAP (Digium Certified Asterisk Professional) in 2006, easily passing the test in the first attempt. Since then, Asterisk has become an important source of work and income. We have installed hundreds of systems. I have actually created a new company called SipPulse Routing and Billing Solutions for SIP based on the experience with Asterisk and OpenSIPS.

This training covers some of the most recent developments of Asterisk such as the version 15 and  chan_pjsip.

What is the promise of this training:

By the end of this training you will be able to:

  • Install an Asterisk box from scratch compiling the source code

  • Connect your Asterisk to ITSPs and phone companies using SIP trunks

  • Build a complete PBX with IVRs, Voicemail, Follow Me and Conference Rooms

  • Activate music on hold

  • Transfer, Capture and Park calls

  • Use Asterisk behind NAT in a cloud such as AWS

  • Use clients behind NAT

  • Learn to develop advanced dialplans

  • Generate a CDR in a database

  • Deploy the required security to avoid being hacked in the first week


Who this course is for:

  • IP PBX value added resellers tech support and development professionals
  • Internet Telephony Service Providers tech support and development professionals
  • Call Center tech support and development professionals
  • IT consultants
  • Telephony consultants

Featured review

Robert Rivera
Robert Rivera
65 courses
18 reviews
Rating: 5.0 out of 511 months ago
The only reason I bought this course is for the security lecture. that alone is worth 5 stars. no other voip course on Udemy teaches security for voip that I have seen. What good is a PBX without security.

Course content

11 sections • 90 lectures • 8h 37m total length

  • Preview03:37
  • Udemy and Github Notes
    04:19
  • Preview13:24
  • Asterisk Architecture
    06:39
  • Use Cases
    14:08

  • Notes on Asterisk 16
    05:15
  • Dimensioning Hardware for Asterisk
    14:27
  • Asterisk 16 Installation
    07:35
  • Lab 1 Install an Asterisk Server
    1 question

  • Section Overview
    04:46
  • Building a PBX part I
    04:35
  • Lab 3.1 Part I, devices in sip.conf
    05:22
  • Lab 3.1 Part II Softphone Configurations
    03:56
  • IMPORTANT: Disable STUN in the softphones
    02:19
  • Building a PBX part II
    05:28
  • Lab 3.1 Part III SIP Trunk Configuration
    06:17
  • Lab 3.1 Building a PBX Part I and II
    1 question
  • Building a PBX Part III-1
    14:43
  • Building a PBX Part III-2
    04:20
  • Lab 3.2 Dialplan
    10:52
  • Lab 3.2 Building a PBX part III
    1 question
  • Building a PBX part IV
    14:58
  • Lab 3.3 Part I, AutoAttendant
    06:40
  • Lab 3.3 Part II, Voice Menu
    04:02
  • Lab 3.3 Part III Voicemail
    09:32
  • Lab 3.3 Part IV Conference Room
    03:02
  • Lab 3.3
    3 questions

  • PBX Features Overview
    10:37
  • Lab 4 - Music On Hold
    07:48
  • Lab 4 - Call Transfer
    08:33
  • Lab 4 - Call Parking
    03:05
  • Lab 4 - Call Pickup
    04:10
  • Lab 4 - Follow Me
    05:43
  • Lab 4 - PBX Features
    1 question

  • Advanced Topics on SIP section overview
    02:48
  • Introduction to SIP
    12:23
  • SIP headers and body
    09:30
  • SIP media encoding
    13:13
  • SIP Part I
    03:55
  • SIP Part II - SIP Channel Naming
    01:17
  • SIP Part III - CLI commands
    04:47
  • SIP Part IV - RTP Timeout
    01:34
  • SIP Part V - QoS tagging
    03:32
  • PJSIP Part I - Overview and Extensions
    05:14
  • PJSIP Part II - SIP Trunks
    01:42
  • PJSIP Part III - Converting sip.conf to pjsip.conf
    02:23
  • PJSIP Part IV PJSIP Console Command Tour
    02:52
  • PJSIP Part V PJSIP History
    04:01
  • PJSIP Part VI - Channel Naming , ACLs and SIPxPJSIP comparison
    03:05
  • Lab 5 - Creating Extensions with PJSIP
    09:34
  • Lab 5 - Sending and receiving calls using siptrunk in PJSIP
    07:08
  • Lab 5 - Creating extensions and trunks with PJSIP
    1 question
  • SIP Nat Traversal
    17:18
  • Client behind NAT using chan_sip
    04:23
  • Client behind NAT using PJSIP
    02:16
  • Asterisk behind NAT using chan_sip
    03:04
  • Asterisk behind NAT using pjsip
    00:54
  • Application Layer Gateway
    04:17
  • SIP Section Summary
    02:57

  • Section overview of advanced topics in dialplan
    03:24
  • Dialplan authorization using context inclusion
    02:20
  • Dialplan Processing Order
    01:25
  • Call routing based on time
    02:49
  • General section of extensions.conf
    03:02
  • The Dial application
    07:06
  • Answer, Progress,Playback and Hangup
    03:36
  • Demo Call Progress and Hangup causes
    04:19
  • Background, Read and WaitExten
    03:22
  • Asterisk DB
    04:04
  • Application Extensions Language
    03:02
  • Grouping and counting calls
    01:45
  • Dialplan Section Summary
    00:50
  • Lab 6 - Dialplan authorization using contexts
    04:33
  • Lab 6 - Dialplan security with multiple contexts
    1 question

  • CDR overview
    04:39
  • Demo Call Detail Records
    08:43
  • Demo Call Detail Record in an ODBC database
    12:31

  • Troubleshooting section overview
    01:27
  • TroubleShooting Methodology
    08:43
  • How to troubleshoot when Asterisk does not start
    03:00
  • Logging in the dialplan using NoOP() and Verbose()
    05:36
  • Logger.conf
    05:27
  • Getting Help
    06:45
  • NGREP DEMO
    05:37
  • SNGREP Demo
    05:28
  • WIRESHARK DEMO
    08:27

  • Security for Asterisk
    12:02
  • Lab 7 Part I, Installing IPTABLES for Asterisk
    06:20
  • Lab 7 Part II, Installing Fail2Ban for Asterisk
    06:25
  • Lab 7 - Configuration of IPTABLEs and Fail2Ban
    1 question

  • Downloading and Installing Ubuntu 18.04 for Asterisk 16
    04:27
  • Preview10:28
  • Downloading and Installing Ubuntu 16.04 for Asterisk 15
    07:49
  • Using a permanent IP in the Server
    05:41
  • Installing Zoiper Free
    02:33
  • Installing X-Lite Free
    01:13
  • Installing Blink Free
    02:16

Instructor

Flavio E. Goncalves
CTO at SipPulse, dCAP, CCNP, CCSP, 20+ years of experience
Flavio E. Goncalves
  • 4.3 Instructor Rating
  • 1,731 Reviews
  • 6,153 Students
  • 5 Courses

Flavio E. Goncalves was born in 1966. Having a strong interest in computers, he got his first personal computer in 1983, and since then, it has been almost an addiction. He received his degree in engineering in 1989 with a focus on computer-aided design and manufacturing.

He is also the CTO of SipPulse Routing and Billing Solutions, a company dedicated to implementing phone companies based on SIP, VoIP providers, and large-scale new generation telephony systems. 

He is an instructor since 1993 when he received its Novell Certified Novell Instructor certificate and managed a NAEC.  Since then he passed more than 50 tests being certified as Novell MCNE/MCNI, Microsoft MCSE/MCT, Cisco CCSP/CCNP/CCDP, Asterisk dCAP, and some others.

He is the author of three books. "Configuration Guide for Asterisk", Building Telephpny Systems for OpenSER and Building Telephony Systems for OpenSIPS

His experience with networks, protocol analyzers, and IP telephony combined with his teaching experience gave him an edge to write technical books and online courses.

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