Knowledge Hub

Practical guides and playbooks for WhatsApp automation, IVR, CRM, AI hiring, and performance marketing.

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Using Janus to Create WebRTC Button for Website (Allows Anonymous Concurrent Users)
2026-03-15 • 18 min
Setup WebRTC Phone in Asterisk / FreePBX and Resolve Call Not Coming
2026-03-14 • 18 min
What Is Stasis in Asterisk ARI? How Dialplan Hands Control to Your App
2026-03-12 • 10 min
All (186) AI (1) AI in Telecom (1) Business Growth (2) Business Metrics (1) Business Strategy (1) Business Telecom (7) CRM (2) Call Center (9) Call Center Management (1) Call Center Operations (2) Call Center Performance (2) Call Center Setup (1) Database (2) Deployment (13) DevOps (1) FreePBX (15) Frontend (1) MYLINEHUB (1) Networking (2) Operations (4) Sales Operations (1) Security (1) Telecom (63) Telecom Architecture (4) Telecom Innovation (1) Troubleshooting (13) VoiceBridge (23) WebRTC (7) WebRTC Architecture (1) WebRTC Integration (1) WebRTC Scaling (1) WebRTC Troubleshooting (1)
Showing 9 of 186 results
Deployment • 15 min
Build a Scalable VoIP Core with OpenSIPS and Asterisk (Installation, Routing Logic, and Shared DB Explained)
Learn how OpenSIPS enables distributed SIP routing, shared user databases, and large-scale inbound call handling when combined with Asterisk media servers.
2026-02-17 • opensips, asterisk, sip-router
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Troubleshooting • 12 min
AMI Connection Issue Resolution in Asterisk: Login Failures, Permissions, and Network Fixes
Step-by-step troubleshooting for Asterisk AMI connection problems including authentication errors, bind address issues, and firewall blocking.
2026-02-17 • ami, asterisk, connection
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Telecom • 13 min
Protocols Used for Your Call Legs Using Asterisk (SIP, SDP, RTP, SRTP, TLS, WebRTC)
Understand the real protocols behind phone calls in Asterisk—SIP signaling, SDP negotiation, RTP media flow, and secure WebRTC layers.
2026-02-16 • asterisk, sip, rtp
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Deployment • 12 min
Generate Absolutely Free SSL Certificate for Your Setup (Let’s Encrypt on Ubuntu) with Auto-Renew Script
Step-by-step Let’s Encrypt SSL generation on Ubuntu plus automated renewal script and verification for long-term secure deployments.
2026-02-16 • ssl, lets-encrypt, certbot
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Troubleshooting • 14 min
Wireshark Live Monitoring for SIP & RTP: Capture Calls, Decode Audio, and Detect One-Way Voice
Complete Wireshark troubleshooting workflow for VoIP—live SIP capture, RTP stream decoding, NAT detection, and one-way audio diagnosis.
2026-02-16 • wireshark, sip, rtp
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Troubleshooting • 11 min
Solving Initial FreePBX Disk Space Problem: Logs, Recordings, and Safe Cleanup
Fix disk-full errors in FreePBX by locating log growth, call recordings, voicemail storage, and applying safe cleanup plus prevention steps.
2026-02-16 • freepbx, disk-space, logs
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Telecom • 12 min
Ports Required for FreePBX + Asterisk: Complete Firewall Guide for SIP, RTP, WebRTC, ARI, and AMI
A full production-ready firewall reference for FreePBX and Asterisk covering SIP, RTP, HTTPS, AMI, ARI, and WebRTC security exposure.
2026-02-15 • freepbx, asterisk, ports
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Deployment • 11 min
How to Setup NGINX Reverse Proxy for Telecom Applications (HTTPS, WebSockets, Security)
Production-safe NGINX reverse proxy configuration for telecom dashboards and FreePBX-style apps with HTTPS, headers, and WebSocket support.
2026-02-15 • nginx, reverse-proxy, https
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Networking • 10 min
How to Setup Two Ethernet Cables on One Machine (Ubuntu + Debian) for SIP & LAN Separation
Configure two Ethernet interfaces on one Linux server safely for telecom use—static IPs, routing priorities, and avoiding SIP/RTP instability.
2026-02-15 • ubuntu, debian, dual-nic
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