FreePBX is the world most popular and widely adopted open source IP telephony software. The core VoIP communication is based on Asterisk 13 - The most powerful IP telephony platform. This course is designed for the newbies, small & medium business that like to use the IP telephony - PBX or even the solution providers that like to gear up for telephony services to the end users.
The course starts with initial telephony concepts and terms used in phone service industry. No prior technical knowledge about telecommunication required to take this course.
The course has project depicting the real world scenario. This helps the students to have activity based learning and can apply learned knowledge to the real world. The course clearly explained the phone system features with lab practice for configurations.
Proprietary systems invites a license fee for most Business-critical features and UC Features like mobility and third party integration. To avoid the expensive add-ons go the open source way.
Use a single system to manage multiple tools and platforms, thereby having “greater employee productivity, reduced costs and a means to improve customer engagement.
The course covers the latest release of FreePBX 14.
In this video, I will train you on VoIP & Traditional telephony. All the related protocols and concepts.
We will learn about traditional telephony trunk interfaces i.e FXO & FXS and will show you the required hardware to connect the telephone lines into the FreePBX server.
In this lecutre, You will learn basic details and information about digital PSTN - T1/E1 circuits and the hardware use to connect the T1/E1 into the FreePBX.
In this lecture, You will learn about the FreePBX server dimension like what level of server specifications are required for the server to install and use FreePBX.
In this Lecture, You will learn to download the latest version of FreePBX 14 from its official website.
In this lecture, You will learn to install the FreePBX 14 from its distro ISO Image on Virtual box machine.
In this lecture, I will show you to perform the initial setup on FreePBX 14 and activate the FreePBX server to be able to use the official Sangoma Support and Commercial modules installation.
In this lecture, I will show you to perform the general Asterisk SIP setting in FreePBX 14.
You will learn to set up the, DNS, Static IP address Network Interface and Hostname in FreePBX 14.
In this lecture, You will learn to setup the email alert & notification in FreePBX 14 that will be triggered in case if any crash or intrusion detection happen.
In this lecture, You will learn to setup the time zone setting in FreePBX 14. This is mandatory for CDR - Call Detail Report to show you accurate report.
In this lecture, you will learn to setup an email alert to monitor your FreePBX 14 storage. In situation where your hard drive disk is full or have abnormal disk functioning. The FreePBX 14 will trigger an email alert to you.
In this lecture, We will have quick review of Module 3 topics.
Listing the extensions that we will create in our FreePBX.
You will learn to create PJSIP extensions in this video lecture.
After the extensions are successfully created. Next these extension needs to be assigned to the end point devices that include desk phones and soft phone. We will use Digium and Yealink for desk phones. Zoiper and Xlite for the soft phone. These configurations remain same for any SIP device available in market.
In this lecture, You will learn to configure & assign the PJSIP extension to the soft phone SIP client i.e Zoiper & Xlite and will make your first test call.
IP Desk Phone Overview.
In this lecture, You will learn to configure the Yealink T19P Phone. This procedure applies to the most of the IP Phones available as of today.
In this lecture, You will learn to configure Digium D70 IP Phone using web GUI. This method is generic and applies on most of the IP Phones these days.
In this lecture, I will demonstrate you the internal calls between the extensions that we created like call from Zoiper softphone to the Yealink Desk Phone and vice versa.
In this lecture, You will learn the configurations related to PSTN (Public Switch Telephone Network) using DAHDI Channel driver in FreePBX for both analog & digtial.
in this lecture, You will learn to create the analog FXS extension. The FXS extension use the DAHDI channel driver.
In this lecture, We will perform the FXS extension testing. We will call from Zoiper soft phone to our FXS analog extension.
Quick Recap of Module 4.
Once your phones are working internally. Its time to go for outside connectivity. In this lecture, I will show you the trunk outside connectivity with FreePBX.
In this lecture, You will learn about Dial Pattern. Dial Pattern allow us to control Outbound/Inbound routes.
In this lecture, You will learn to setup the DAHDI PSTN FXO trunk set up.
In this lecture, We will setup the outbound route for FXO trunk.
In this lecture, You will learn to setup the inbound route for FXO trunk.
In this lecture, You learn to configure and setup the T1/E1 Digital PSTN hardware in FreePBX.
In this lecture, You will learn about SIP as protocol and trunking details between two FreePBX in different locations.
In this lecture, You will learn to setup the outbound route for the SIP trunk created between two FreePBX servers that are located on two different sites.
In this lecture, You will learn to setup the inbound route for the SIP trunk created between two FreePBX servers that are located on two different sites.
In this lecture, You will test the sip trunking between two FreePBX servers by dialing internal DIDs.
In this lecture, You will learn to restrict the user's extensions from dialing local or long distance calls.
Quick Recap of Module 5.
in this lecture, You will learn to add Voicemail feature to you extension and setup your voicemail box.
In this lecture, You will learn to configure the Find Me/Follow Me feature of FreePBX.
You will learn to configure the working & non-working hours of your business in FreePBX.
You will learn to upload the correct audio format of voice prompts into the FreePBX.
In this lecture, You will learn to create the announcement in FreePBX.
Creating Extensions Ring Groups.
Creating IVR - Interactive Voice Response - in FreePBX.
In this lecture, We will test our complete Inbound Call Flow like Time Conditions, Non-Working Hours Announcement and Ring Group by using the IVR.
Quick Recap of Module 6.
FreePBX Features Codes & Extensions Printing List.
Users Management in FreePBX.
UCP - Users Control Panel.
In this lecture, You will learn to enable the call recording features in FreePBX in various ways.
CDR - Call Details Report is the FreePBX reporting statistics that help you to analyze the PBX usage and simple billing usage.
CEL - Call Events Logging - The advance form of CDR where call records are traced based on call events. Its used for complex billing.
In this lecture, You will be able to perform the Backup and Restore features in FreePBX.
in this lecture, You will be able to perform the updates on FreePBX.
In case you have good experience on Asterisk CLI - then FreePBX offers you to have a Asterisk info report that help you to analyze your Asterisk server using CLI commands.
This is the final and course conclusion lecture by the Instructor.
FreePBX official Wiki.
NEF system is the leading e-learning provider for VoIP related training and services. It offers wide array of training and services related to VoIP telephony, contact center and CRM. It has expert and certified team to conduct training both onsite and video based. The training portfolio includes Linux, Networking and Open Source IP Telephony and Call Center Solution.