⚙️ Open Source Telecom Stack

Telecommunications Research & Development Center.

Open-source platform. Omnichannel product bringing together WhatsApp, SMS, email, and telecom workflows. This page brings together setup guidance, open documentation direction, feature overview, MYLINEHUB demo videos, implementation context, and telecom workflow understanding in one place.

🤖 AI Calling Demo

Experience the AI telephone demo

This demo uses the ChatGPT Real-Time API and can reduce calling cost by 60–80%, to approximately ₹2 per minute. If your team can build and optimize custom models, the cost can drop even further. We also help you set up the complete telecom infrastructure in-house based on your requirements.

Call AI Demo Line
+91 7042149841
If busy, try again later.
📚 Full Documentation Roadmap

Install Mylinehub AI Communication / Autodialer - Do It Yourself Guide

On-Premise Setup Roadmap. Learn Asterisk first, configure FreePBX next, finalize router/firewall, then install MYLINEHUB.

Training focuses on implementation decisions, AI calling architecture, WhatsApp automation workflows, white-labeling, scaling, and India telecom compliance. Docs remain open so your team can execute step-by-step after training. GSM gateway or else SIP trunk connection process depends on vendor / hareware you choose.

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Step 1 — Learn the Asterisk Base Layer

Start with the telecom engine first so the rest of the stack 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
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Step 2 — 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.
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Step 3 — Add GSM Gateway or SIM-Based Calling

Use mobile-number connectivity where the business use case requires SIM-based telecom routing.

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 →
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Step 4 — Finalize Router, NAT, Firewall, and Media Flow

Fix networking cleanly so calls register correctly and audio flows properly 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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Step 5 — Install and Connect MYLINEHUB

Complete the rollout by connecting the application layer on top of the telephony foundation.

Install MYLINEHUB (Start Here)
Follow these steps in order for a clean production setup.
Start Installation
View installation checklist
  1. Install MYLINEHUB on Ubuntu Server (Complete Production Guide)
  2. Install PostgreSQL and PGVector for MYLINEHUB AI Features
  3. Configure SSL Using Let’s Encrypt for MYLINEHUB
  4. Nginx Reverse Proxy Configuration for MYLINEHUB Services
  5. Create systemd Service for MYLINEHUB VoiceBridge
  6. How to Load Initial Telecom Data Safely in MYLINEHUB (Organization, AMI, SSH, Users, IVR, Queue)
  7. VoiceBridge DB-Driven Configuration: stasis_app_config + Instructions (Safe Initial Data)
  8. Fetch & Clear Backend Service Logs (MYLINEHUB Debugging Guide)
  9. Monitor Spring Boot JMX Over SSH for MYLINEHUB
  10. Increase Linux ulimit for MYLINEHUB (Fix Too Many Open Files)
  11. Troubleshoot HTTP Connection Reset Issues in MYLINEHUB
  12. Reset Database for New Customer in MYLINEHUB
  13. Setup Gitea Git Server for MYLINEHUB Open-Source
  14. Reset Angular Setup for MYLINEHUB UI (Clean Rebuild & Fix Broken Builds)
Open All Docs →
💎 Premium Open-Source 🧩 White-label ready ⚡ Built for scale

Premium Open-Source Telecom Features Unlimited power to personalize

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
Workflows, covering all
Templates, tagging, journeys, triggers, actions.
Listen Quietly
Connect Securely + Reporting
Call control, quality, logs, analytics, scale patterns.
Clear targeting
Execution-focused
Better results
Perfect for
Scaling & save cost
  • Call centers / BPOs with in-house PBX/telecom
  • Businesses wanting WhatsApp CRM under their own brand
  • CRM resellers / agencies building white-label offerings
  • Founders / tech leads tired of vendor lock-in
  • Teams who want their own API layer + data ownership
Ownership mindset Implementation-first
Not for
Avoid mismatch
  • Total beginners with zero telecom context
  • People who want “done-for-you SaaS cloud”
  • Anyone expecting Asterisk installation training
This requires telecom expertise, not for beginners.
What you will build
A beautiful overview of the MYLINEHUB stack
Omnichannel Telecommunication Suite
Secure your data
Track call conversions
AI Call Agent
AI Email Agent
Bulk communication
Connect your frontend
Multi-user interaction
01 / 08

Product Videos - Open Source Premium CRM + Autodialer

Best Return on Investment Deal + Fabulous Features

  • 🚀 Launch Campaign
  • 📩 Customer WhatsApp Message
  • 🤖 AI-Powered Instant Reply
  • 📅 AI-Assisted Call Schedule
  • 📊 Insightful Performance Reports
  • 💡 Expert Support & Guidance

Our Solutions
🤖 TRY WHATSAPP BOT

Empower your business with smart, AI-driven solutions.

🤖 WhatsApp AI Bot

Engage customers automatically via WhatsApp.

📡 Telecom Solutions

Integrate voice, SMS, and WhatsApp seamlessly.

📊 CRM Integration

Create unified sales workflows.

⚙️ Business Automation

Automate repetitive workflows.

Company Policy Deck, Below are text documents, kindly read these carefully!

GENERAL POLICIES
PRIVACY POLICIES
REFUND POLICIES
RETURN POLICIES
SHIPPING POLICIES
TERMS & CONDITIONS