Asterisk Made Easy: Learn to Install Asterisk and Linux

Learn how to install and configure Asterisk Voip PBX on Linux to make phone calls following easy video tutorials
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Instructed by Jon Manning IT & Software / Other
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  • Lectures 51
  • Length 6.5 hours
  • Skill Level Beginner Level
  • Languages English
  • Includes Lifetime access
    30 day money back guarantee!
    Available on iOS and Android
    Certificate of Completion
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About This Course

Published 2/2015 English

Course Description

Learn how to install and configure Asterisk Voip PBX to make phone calls in 1 Day

Do you want to learn how to install Asterisk version 11 or 13?

Are you confused by the Asterisk documentation?

Do you want an easy to follow tutorial on Asterisk?

Are you overwhelmed by the complexity of the Asterisk configuration?

Do you have little to no experience with Linux?

Are you trying to build an Asterisk phone system?

Many years ago I was very interested in learning Asterisk to replace my employers old phone systems. I spent a lot of time reading the documentation, using Google to search the internet and any book I could get my hands on. Even with all of this I was never able to build a working Asterisk phone system. I was very frustrated and was about to give up but then a friend of mine offered to tutor me. He showed me simple step by step training starting with how to compile Asterisk software then going all the way through to configuring a working Asterisk phone system. Within a few days I had the confidence and Asterisk skills to take any computer and turn it into a working Asterisk phone system.

You need someone to show you step by step how to work with Asterisk. Using a simple to follow video guide to walk you through how to install Asterisk, write the text based configuration files and what the network environment looks like without wasting time searching the internet or paying thousands of dollars for on site training.

You will be able to compile and install Asterisk after spending 1 hour following my step by step video tutorials. After you spend a day completing the course you will have the skill set to take any computer and turn it in to an Asterisk phone system. Once you follow the course and install a couple of Asterisk systems you will be an Asterisk expert.

Sure you can use google and other internet resources to teach yourself but how many months will something like that take? There will be a lot of good and bad information when reading different articles trying to piece together training materials. A lot of the documentation you will find is outdated and is not relevant anymore making your training obsolete. You also have the option of paying for Asterisk training which costs thousands of dollars for a single course and requires you travel to the training center incurring thousands of dollars in airfare. You will also have to your valuable vacation time.

Or you could buy this course I have created with easy video tutorials for only $297 and save your self a lot of time and hassle. Plus you can sell Asterisk phone systems to clients for well over $1000 each providing a custom built Asterisk phone system.

With more than one million Asterisk systems being used in more than 170 countries, Asterisk is an important skill to have. Asterisk is in use by almost all Fortune 1000 companies creating many job opportunities for anyone with an Asterisk skill set.

You should not take this course unless you will follow along with the videos and labs to build your own Asterisk environment. This course is for people who want to learn following along with the video labs and apply that knowledge to sell Asterisk systems or get a job working with Asterisk.

With my course I want all my students to be completely satisfied, if you have any problems or questions I want to hear from you. Let me know what I can do to fix the course so it is improved for everyone giving you no risk buying this course. You are also backed by the udemy money back guarantee.

If you want be an Asterisk expert click on Take This Course button and start configuring your first Asterisk server.

What are the requirements?

  • Asterisk source code, CentOS Linux server and a sip softphone will be needed when taking the course but we will walk through the downloading and installation step by step.
  • You do not need to have previous experience with Linux, telephony or Asterisk. Of course it would be helpful but is not required.

What am I going to get from this course?

  • Install Asterisk from Source on CentOS Linux
  • Sell Asterisk Cloud Based PBX Services
  • Sell on Premise Asterisk VoIP PBX Servers
  • Write Custom Asterisk Dialplans
  • Troubleshoot, Support and Maintain Asterisk Servers
  • Get a Job Working with Asterisk VoIP Systems
  • Troubleshoot using TCPdump and Wireshark

Who is the target audience?

  • This Asterisk course is designed for all skill levels to compile and install Asterisk from source on a CentOS Linux server. No prior Linux or Asterisk experience is required. I work in a lab environment to show you step by step how to get Asterisk running and configure it.
  • You do not need to have previous experience with Linux, telephony or Asterisk. Of course it would be helpful but is not required.

What you get with this course?

Not for you? No problem.
30 day money back guarantee.

Forever yours.
Lifetime access.

Learn on the go.
Desktop, iOS and Android.

Get rewarded.
Certificate of completion.


Section 1: Getting Started with CentOS Linux and Asterisk

Welcome to my course and I really appreciate your support. I just want to make a quick introduction before we get started.


This lecture introduce the course syllabus and what tools you will be using while going through the Linux and Asterisk labs. I have also included my contact information for students to contact me with questions related to the course.


In this lecture we talk about what Asterisk is, what you can do with it, and the various features included with Asterisk. We will talk about some opportunities to make money using Asterisk or getting a job working with Asterisk. Asterisk also has some hardware appliances students can use to configure and sell to clients as phone system.


In the first lab you will download and install VirtualBox to create your VM host machine. You will also download the CentOS 6.x minimal ISO image we will use to install Linux on your virtual machine. There are links included in the course resources to get you to the correct download page.


In this lab you will create your virtual machine for the Linux server to install Asterisk software on. We will talk about sizing your virtual machine, attach the CentOS ISO image to your VM guest, and setup your VM's network adapter.


Now we get to start the VM and begin the installation of CentOS Linux. Once the installation is complete we will edit the network interface configuration file to turn it on. Then we will run a command to restart the network interface card to get a dhcp address.


I will show you how to login remotely to your Linux server using the Mac or Linux ssh terminal application.


I will show you how to login remotely to your Linux server using the Windows putty ssh applicaiton.


You will perform an update on the newly install CentOS Linux operating system. I will show you how to install package for wget application and use it to download the Asterisk source code.


During this session you will download the Asterisk source code, extract the files using the tar command and then compile the Asterisk source code. This will requre installing some prerequisit packages on the operating system for Asterisk to compile properly. Once the source code is compiled we will run the commands to install Asterisk and sample configuration files.


During this session you will create a file to control the Asterisk service. This will give us the ability to configure Asterisk to start on reboot and run as a service. There are resources attached to this lecture which include the commands used during this video.


You will not enable Asterisk to start as a service on reboot and disable the SELinux security feature. I have attached a file in the resources for you to have a copy of the commands used during this video.


Doing a quick wrap up of what we did in the first lab. We review all of the commands used and explain what each one of these commands does. There are notes attached in the course resources to give you a list of all the commands used.

3 questions

Let's review what we did in lab 1 and make sure you captured some of the important knowledge.

Section 2: SIP Endpoints and SIP Carriers

In this lecture we look at what the network network devices were used in lab 1 and what additions will be added in lab 2. We will review the ports used by Asterisk and talk about which ones need to be forwarded through the firewall for SIP carrier services to work correctly. You will see network diagrams so you understand the relationship Asterisk has with SIP to deliver telephony.


In the first part of lab 2 you will download and install the XLite softphone client on your host computer. The softphone will be a SIP client to your Asterisk server once we configure it.


I will show you how to turn off the IPTables firewall that is built into CentOS so that does not block any of the SIP traffic. This is for a test environment which is fine but in a true production environment you will want to have the firewall running to block malicious internet traffic. Then we will add a SIP extension into the sip.conf file, see the file attached in course resources.


In the first part of this video we configure the XLite software with a SIP account to register against the Asterisk server. Then we dial 600 which is a built in application into the sample Asterisk dialplan. I will also show you what the default context of the extensions.conf file looks like and its relationship with SIP extensions.


In this session you will create a custom context in extensions.conf to show you how to create dialplan applications and how a SIP extension is able to access those applications.


Now we add another extension so we can do extension to extension dialing. Once that is complete we look at voicemail.conf to add mailbox's for these SIP extensions. This then allows us to create a new application in extensions.conf to make test phone calls to using the softphone. We go through how the extensions.conf dialplan works with variables and calling the different built in Asterisk applications.


In this lesson we are going to make changes to the extensions.conf dialplan allowing for dynamic dialing.


Follow along in the video to create a account to use for your SIP carrier services.


On the SIP carrier we create a sub account then using the sample config files from the carrier we configure our Asterisk lab system to use the SIP carrier.


This lesson will show you how to add the sample context from to allow dialing out to the pstn. We will copy the sample extensions.conf file and paste it into our Asterisk machine. Then we will do a reload on the Asterisk config files.


We discuss the importance of port forwarding for your Asterisk server and walk through the screens of a Linksys router.


In this presentation I discuss the SIP ALG setting that is built into many routers and firewalls. This is a setting that must be disabled in order for your Asterisk server to work correctly behind a firewall. I also walk through the screens of a Linksys router to show you where to turn off the setting.


In this session you will create a new application that dials the echo test on servers. This is going to get added to the prior context we created from the sample config files. After making sure the echo test works we will then be able to place an outbound phone call to a real phone number.


You will now make an outbound test phone call. If you have followed along then your call should be connected to the external phone number as long as you have added funds to your account. After that we will purchase a DID number from and configure Asterisk to route inbound phone calls to your XLite softphone.

Lab 2 Quiz Questions
3 questions
Section 3: Create a way for users to dial into their voicemail admin

In this lesson you will see how users are able to dial into their voicemail admin to configure their greeting and listen to voicemail messages.

Section 4: Public review of the course

I appreciate students leaving a review of the course as it helps me and other students.

Section 5: Adding Dahdi and Libpri to Asterisk, Recompile Asterisk Source

I will show you some of the hardware you can use with your Asterisk server to connect to the PSTN using legacy connections like analog phone lines and T1/PRI connections. Then we will review what we are going to do in the lab and the different commands.


In the first part of this lab you will install updates on your Linux server then download and compile Dahdi and Libpri. Once that is complete you will go into the exsiting source directory for Asterisk and recompile the Asterisk software so it sees the newly added libraries for Dahdi and Libpri.


In this lecture we will review how you would add a physical card to your Asterisk system and configure it to work with the software. Since our lab does not include any real cards we only talk about it and can't show you a real example. The steps to configure are the same but it would be dependant on the hardware used and how you want the card to work with your Asterisk server.

Lab 3 Quiz Questions
3 questions
Section 6: Asterisk Codec Selection and Install G729 Codec

In this lecture we will talk about the different codecs available for use with Asterisk and what SIP carriers typically use. We talk about the G729 codec then the lab deals will installing the G729 codec.


In this lab I will show you how to install the G729 codec. Keep in mind the G729 codec is licensed and costs money so if you are using this in a production enviorment you will need to pay for the channels you use.

1 question

We review the fourth lab with a quiz.

Section 7: Create a working phone system from the configuration files included with course

In this lecture I will look at the configuration files included in the course resources so you can build a working production phone system. We will talk about the settings of each of the files and how they work.


In this session we are going to create an audio recording for the auto attendant to answer inbound phone calls. There are muiltiple ways to create recordings so I will show you a couple of options.


In this session you will download the working sample configuration files I have attached in the course resources. Then you will take the files and load them into your Asterisk lab system.


We will discuss what changes you need to make in the sip.conf file for your server in order to make this work with your account or Vitelity account.


We will discuss what changes you need to make in the extensions.conf file for your server. I will also show you how to tailor this configuration for you or your clients.


We will discuss what changes you need to make in the extensions.conf file for your server. I will also show you how to tailor this configuration for you or your clients.


We will discuss what changes you need to make in the voicemail.conf file for your server. I will show you where to add your users names and email addresses for voicemail to email.


We will discuss what changes you need to make in the followme.conf file for your server. In this file you can setup your users extensions to dial out to a mobile phone number if the Asterisk extension rings and does not get answered.

Section 8: Configure your Linux server to send emails using a Gmail account

In this video I will show you how to configure your Linux system to send emails out for voicemail to email using a Gmail account.

Section 9: Configure your Asterisk server to log phone calls using CDR's

We will now enable CDR's so we can create call reports to see outbound and inbound calls. We will edit the /etc/asterisk/cdr_custom.conf file to enable logging.

Section 10: How to troubleshoot problems with your Asterisk server

In this lecuture we will discuss the different commands I have included in the course resources on how to troubleshoot problems with your Linux and Asterisk system.


We continue troubleshooting phone calls on the Asterisk system.


We will install tcpdump on your Linux server and I will show you how to use it. I have included example commands in the course resources.


Create a packet capture on your Linux server to include ULAW audio for playback from Wireshark.

Section 11: Bonus Lab: Learn to install the latest version of Asterisk 13 certified

In this lab we are going to use CentOS version 7 the 64bit version. This is a requirement for this install as I have customized the script to use 64bit linux. We are also going to use the latest version of Asterisk 13 certified with long term support.


Follow along step by step to install CentOS 7 64bit using the GUI interface. Last time we used a different interface so make sure to watch and see what is required.


Now that the virtual machine is ready we can install OS updates and prerequisite packages. After a reboot we will download pj project source and Asterisk 13 source to compile and install.

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Instructor Biography

Jon Manning, VOIP, SIP and Asterisk Expert, Technical Sales Professional

I started supporting call centers 14 years ago and at the time I had no experience with phone systems. Eventually I learned them inside out and started using Asterisk and Linux to reduce costs. I discovered how versatile and robust Asterisk was so I expanded by client base to sell business phone systems.

After 14 years I have turned to focus my efforts on training how to build and sell phone systems. My instruction is very actionable, I want students to take the knowledge they gain and start using it to make money or do what they choose to do with it. You will find my teaching style is well liked by my students and they are always looking for new and exciting courses from me. Just take a look at some of the reviews I have received below.

"I am by far loving your course. Please make more and more videos on Asterisk Server; I shall purchase all of them." - Ganesh Dhungana

"You have been brilliant sir! I didn’t even have to finish the course to become good enough to start supporting our clients on asterisk platforms." - Tokunboh

"Excellent Instructor, Jon, is great... Hopefully he will produce additional classes. I would take them immediately. Thank you JON !!" - David Jordan

"Excellent!!, Jon has done a great job in walking me through carefully and showing typical errors and gotchas along the way." - Bill Adams

"Interested in providing VOIP services? Take this course, Even if you plan to use FreePBX or another GUI interface for your VOIP service, learning Asterisk and Linux will give you a better understanding of what's going on under the hood of your PBX. Jon Manning gives great step by step instructions and clear explanations of what you need to do to provision SIP phones and legacy phone equipment." - Martin Fields

"Actionable knowledge, not just pure information, communicated with great professionalism, The lessons in this course go way beyond pure information. Jon (the instructor) knows his stuff inside out, and more significantly, he knows how to pass his knowledge clearly so that I can take action on it right away." - Michael Choke

"The best course for who want to enter to the Asterisk and Linux field, This course has been designed by professional instructor who has a lot of information so for beginner you just follow the lessons as i did, now i have a good knowledge about Asterisk runs on Linux how run the Asterisk operations. Also the instructor provide his email to contact him for any question that is really awesome, i liked this course and it worth the payment." - Saifaldeen ALBUSULTAN

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