Deep Telephony Learning Path

Prepare yourself for Deep Telephony Tech.

Telephony is built from several layers, technologies and sources of information. You do not need to learn everything at once. The steps below guide you through the stack in a practical order and show you where to go deeper.

It is not just us. The learning path brings together open-source projects, documentation, telecom platforms and practical implementation experience. Follow each source for deeper information.
01
Understand the telecom base Asterisk and PBX fundamentals first.
02
Connect the infrastructure Trunks, GSM, routing, NAT and media.
03
Add the application layer Bring MYLINEHUB into the working stack.
AI Calling Demo

Experience AI calling on your own infrastructure.

Request an on-premise AI calling demo configured around your requirement. We can prepare the telephony and AI calling flow on your infrastructure so your team can experience how it works in its own environment.

Approx. ₹10,000 setup charges

Final charges depend on the required call flow, infrastructure, integrations and configuration. More involved requirements may exceed ₹10,000.

Request On-Premise Demo →
Your Learning Path

Learn the stack. Then build it yourself.

Start with Asterisk, move through PBX configuration and telecom connectivity, understand networking and media flow, then connect MYLINEHUB on top.

01

Learn the Asterisk Base Layer

Start with the telecom engine so everything above it makes practical sense.

Start with Asterisk Tutorials (Recommended For Beginners)

Many beginner tutorials online are based on legacy SIP (chan_sip). MYLINEHUB documentation is aligned with modern PJSIP architecture, helping you move from basic PBX learning to real production telecom deployments.

View Asterisk series checklist
02

Configure FreePBX Properly

Move into extensions, trunks, IVR, queues and business-ready PBX configuration.

FreePBX Series (UI-Based Asterisk Control)

If you don’t want to write dialplan or manage raw Asterisk configs, FreePBX lets you control your PBX through a UI. These tutorials are written for FreePBX 16/17 with PJSIP (modern SIP), and every guide includes step-by-step screenshots so you can configure safely.

Recommended routing order: Trunks → Inbound Routes → Outbound Routes (so calls can enter, route correctly, and exit).

FreePBX Setup →
View FreePBX series checklist (15 guides)
  1. FreePBX SysAdmin Module: Updates, Backups, and Security • 10 min
  2. FreePBX Certificate Management: HTTPS, PEM, and Renewals • 7 min
  3. FreePBX Advanced Settings (Asterisk 20): Secure and Correct Defaults • 10 min
  4. FreePBX Bulk Handler / Admin: Handling Bulk Updates Safely • 6 min
  5. FreePBX SIP Settings: NAT, RTP Ports, and Codec Choices • 10 min
  6. Reset Network for SIP in FreePBX: Fix NAT, RTP, and One-Way Audio • 6 min
  7. FreePBX Firewall: Trusted Networks, Ports, and ARI/AMI Access • 10 min
  8. FreePBX Trunks: SIP Provider Setup, Registration, and Identify/Match • 9 min
  9. FreePBX Inbound Routes: DID Mapping, Time Conditions, and Destinations • 11 min
  10. FreePBX Outbound Routes: Patterns, Trunks, and CallerID • 7 min
  11. FreePBX Extensions: PJSIP Extensions and Common Pitfalls • 10 min
  12. FreePBX IVR: Menus, Prompts, and Clean Customer Journeys • 10 min
  13. FreePBX Queues: Strategies, Agents, and Monitoring • 10 min
  14. FreePBX Conference: MeetMe/ConfBridge Setup Basics • 6 min
  15. FreePBX Dynamic Routes: Build Smart Routing Logic • 7 min
Tip: Do SysAdmin + Certificates + Advanced defaults first, then fix SIP/NAT + Firewall, and only then configure Trunks.
03

Connect GSM or SIP Calling

Connect the telecom provider layer according to the calling method and infrastructure available to you.

GSM Gateway Setup Using Mobile Number for Telecom in India

Explore how a GSM gateway bridges SIM-based mobile calling with IP telephony systems. This guide covers SIM status, SIP / VoIP endpoint setup, client, server, and IP-based registration modes, routing rules, and practical integration with FreePBX, Asterisk, and similar telecom platforms.

Gateway Guide →
04

Understand Router, NAT, Firewall & Media

Get the network path right so registration, signaling and audio work correctly in real deployments.

Router Config (SIP + NAT + Port Forwarding)

Before blaming trunks or codecs, fix networking first. This guide explains NAT, port forwarding, RTP ranges, and the common reasons for one-way audio and registration failures.

Router Guide →
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Explore real product features and implementation patterns — dialer, CRM workflows, AI calling, automation, reporting, roles, and white-label UI.

Core stack
Dialer + CRM foundation
Calling flows, pipelines, roles, and dashboards.
Automation
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Templates, tagging, journeys, triggers, actions.
Listen Quietly
Connect Securely + Reporting
Call control, quality, logs, analytics, scale patterns.
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Better results
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Ownership mindset Implementation-first
Not for
Avoid mismatch
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  • People who want ā€œdone-for-you SaaS cloudā€
  • Anyone expecting Asterisk installation training
This requires telecom expertise, not for beginners.

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āš™ļø Business Automation

Automate repetitive workflows.