Knowledge Hub

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

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How Can MYLO Reduce the Cost and Complexity of Running Business Telephony?
2026-09-30 • 8 min
Can My CRM Make Calls Through MYLO?
2026-09-30 • 8 min
Does MYLO Make AI Voice Calls, or Can I Add an AI Calling Agent Later?
2026-09-30 • 7 min
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Showing 9 of 396 results
VoiceBridge • 12 min
VoiceBridge DB-Driven Runtime Configuration Explained
How VoiceBridge loads runtime configuration from the database: call routing, RTP params, AI instructions, and environment-specific overrides without redeploys.
2026-02-19 • voicebridge, postgresql, runtime-config
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Telecom Architecture • 12 min
Why Asterisk Alone Struggles Beyond 200 Concurrent Calls (And How to Design a Scalable Architecture)
Understand real scalability limits of standalone Asterisk and FreePBX, why databases and SIP handling become bottlenecks, and how modern telecom architectures handle hundreds of concurrent calls reliably.
2026-02-17 • asterisk, concurrent-calls, scalable-telephony
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Deployment • 15 min
How to Scale Asterisk to Hundreds of Concurrent Calls Using Kamailio (Complete Install & Integration Guide)
Step-by-step Kamailio installation, SIP proxy basics, shared database routing, and full integration with Asterisk to support large-scale concurrent call handling.
2026-02-17 • kamailio, asterisk, sip-proxy
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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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