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
Flavio E. Goncalves has graduated as an Engineer in 1988. What best defines him, is an entrepreneur with strong technical skills.
22+ years of experience in Voice over IP systems, 6500+ students on Udemy
He started as a Network and Database Administrator in 1989 and in the technical carrier got the following certificates
- Novell Master CNE in 1993
- Microsoft Certified Systems Engineer in 1995
- Cisco CCNP, CCDP, CCSP and CCIE written
- Asterisk dCAP in 2006
- OpenSIPS Instructor and creator of the OpenSIPS Bootcamp and OpenSIPS Certified Professional
He has founded four companies:
- V.Office, Network Systems Integrator, Since 1996
- SipPulse, Communication Systems, Since 2010
- Api4Com, APIs for communication, Since 2018
- WeHostVoIP, Hosting VoIP Application, Since 2020
In the VoIP School the objective is to build a community of people related to VoIP, Instructors and Students in the same place. With the teachable interface, the teaching model can be a lot better. We hope to offer more web-conference trainings and coaching.
StartSection Overview (4:46)
StartBuilding a PBX part I (4:35)
StartLab 3.1 Part I, devices in sip.conf (20:20)
StartLab 3.1 Part II Softphone Configurations (3:56)
StartIMPORTANT: Disable STUN in the softphones (2:19)
StartBuilding a PBX part II (5:28)
StartLab 3.1 Part III SIP Trunk Configuration (6:17)
StartAssignment 2: Lab 3.1Building a PBX Part I and II
StartBuilding a PBX Part III-1 (14:43)
StartBuilding a PBX Part III-2 (4:20)
StartLab 3.2 Dialplan (10:52)
StartAssignment: Lab 3.2 Building a PBX part III
StartBuilding a PBX part IV (14:58)
StartLab 3.3 Part I, AutoAttendant (6:40)
StartLab 3.3 Part II, Voice Menu (4:02)
StartLab 3.3 Part III Voicemail (9:32)
StartLab 3.3 Part IV Conference Room (3:02)
StartAssignment 4: Lab 3.3