Asterisk Made Easy: Learn to Install Asterisk and Linux

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.

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.

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 voip.ms account to use for your SIP carrier services.

On the voip.ms 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 voip.ms 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 voip.ms servers. This is going to get added to the prior context we created from the voip.ms 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 voip.ms account. After that we will purchase a DID number from voip.ms and configure Asterisk to route inbound phone calls to your XLite softphone.

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.

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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.

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.

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 voip.ms 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.

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.

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.