Training | Development | Placement

VoIP Network Engineer Specialist

VoIP Network Engineer Specialist

Secure a high-value career in the telecommunications field with OneAccess Technologies. Our certified VoIP Technical Training program provides hands-on experience to configure, deploy, and troubleshoot modern Voice over IP (VoIP) and Private Branch Exchange (PBX) systems in Sri Lanka

Colombo
04 Month
RS 25,000.00
Starts: Apr 10, 2026
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Course Description

The way businesses communicate has shifted from traditional phone lines (PSTN) to Voice over IP (VoIP) systems, which are faster, more flexible, and cost-effective. Companies of all sizes—whether a local office in Sri Lanka or an international call center—depend on VoIP technology. This growing demand has created a need for skilled professionals who can configure, manage, and troubleshoot VoIP and PBX systems.

The VoIP and PBX Technical Training by OneAccess Technologies is designed to give you hands-on experience in all aspects of VoIP deployment. You will learn the basics of telecom systems, understand key protocols like SIP, and practice configuring call routing, extensions, and lab PBX systems. Our course emphasizes real-world troubleshooting, teaching you how to solve common call quality and network issues. By the end of the program, you’ll have the skills to confidently work as a VoIP/PBX Engineer, ready for careers in Sri Lanka or international companies.

Why Take This Course:

  • Learn the shift from traditional phone systems (PSTN) to modern VoIP technology
  • Gain skills in configuring, managing, and troubleshooting VoIP and PBX systems.
  • Build expertise that is relevant for both Sri Lankan and international companies.

What You Will Learn:

  • Telecom fundamentals: Understanding how voice communication works over networks.
  • Key protocols: Hands-on experience with SIP and other VoIP protocols.
  • PBX configuration: Setting up call routing, extensions, and lab PBX systems.
  • Troubleshooting skills: Diagnose and resolve common call quality and connectivity issues.

Course Benefits:

  • Hands-on practical training from planning to deployment.
  • Prepare for a career as a VoIP/PBX Engineer.
  • Gain skills that employers worldwide demand.
  • Open doors to high-value technical careers locally and globally.

Content Covered

01. Overview of VoIP and its advantages
02. VoIP vs traditional telephony (PSTN)
03. Analogue and IP Phones
04. Key components: IP phones, gateways, softphones, PBX
05. overview of VoIP Brands and Models
01. Session Initiation Protocol (SIP)
02. H.323 Protocol
03. Real-Time Transport Protocol (RTP)
04. Media Gateway Control Protocol (MGCP)
05. WebRTC basics for browser-based voice communication
01. IP addressing and subnetting for VoIP
02. VLANs and QoS (Quality of Service)
03. NAT traversal and firewall considerations
04. Bandwidth requirements and call quality
01. Setting up FreePBX / Grandstream
02. Configuring extensions, trunks, and dial plans
03. Softphones and IP phone setup
04. Call routing, voicemail, and auto-attendant
05. IVR Setup, Call Queue
06. Cloud-based VoIP platforms (3CX)
01. VoIP threats: eavesdropping, toll fraud, DoS attacks
02. Securing SIP endpoints and servers
03. Encryption with SRTP and TLS
04. Security considerations for cloud VoIP deployments
01. Call flow analysis and logging
02. Common issues and resolutions
03. Monitoring tools for VoIP networks
04. AI-powered call analytics for performance monitoring
01. Introduction to Asterisk
02. How to Set Up VoIP System with Asterisk
03 . Step-by-Step Guide to deploy own PBX system with Asterisk
01. Configuring a complete VoIP system (on-premise and cloud)
02. Simulating calls between IP phones and softphones
03. QoS and bandwidth testing
04. Security testing and troubleshooting
Exam Tasks (Step-by-Step)
01. Basic Network Setup - Configure IP addressing of PBX server, Softphone, IP phones Ensure connectivity (ping test)
02. PBX Installation & Setup - Install or access PBX (FreePBX / Grandstream / 3CX) 2–3 extensions (e.g., 1001, 1002), Register softphones
03. SIP Configuration - Configure SIP accounts, Verify registration status, Make internal calls between extensions
04. Call Routing (Dial Plan) - Configure Internal calling, Outbound dial rule (e.g., 9 + number), inbound route & Outbound Route
05. VoIP Testing - Make test calls: Extension ↔ Extension, Extension → Outside

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