Knowledge Hub
Practical guides and playbooks for WhatsApp automation, IVR, CRM, AI hiring, and performance marketing.
FreeSWITCH • 11 min
Create a first FreeSWITCH user, configure a softphone, reload XML, confirm registration in fs_cli, and test a simple extension-to-extension call.
2026-09-02 • FreeSWITCH extension, softphone, SIP registration
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FreeSWITCH • 13 min
A practical bridge for Asterisk engineers moving to FreeSWITCH: CLI, Sofia profiles, XML dialplan, Event Socket, queues, and a learning path that reuses what you already know.
2026-09-01 • FreeSWITCH, Asterisk, Sofia SIP
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FreeSWITCH • 14 min
Plan a clean FreeSWITCH deployment on Linux, choose package or source installation, verify services and ports, and complete the checks that should happen before the first SIP device connects.
2026-09-01 • FreeSWITCH installation, Linux, Debian
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WebRTC • 18 min
Complete practical guide to build a public website WebRTC call button using Janus, coturn, Asterisk, FreePBX, and ARI/Stasis so anonymous concurrent users can call your bot from the browser without creating SIP extensions per user.
2026-03-15 • janus, webrtc, website-button
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WebRTC • 18 min
Complete practical guide to set up a WebRTC browser phone on Asterisk and FreePBX, fix the case where presence NOTIFY arrives but the call does not ring, and correctly configure Max Contacts, Remove Existing, DTLS-SRTP, ICE, NAT, and multi-browser ringing.
2026-03-14 • webrtc-phone, asterisk, freepbx
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VoiceBridge • 10 min
Understand what Stasis is in Asterisk ARI, how a call enters a Stasis application, and why it is the key handoff point for custom voice bots, media apps, and real-time telecom control.
2026-03-12 • stasis, asterisk, ari
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WebRTC • 10 min
Learn what an ICE candidate is in WebRTC, the difference between host, server reflexive, and relay candidates, and how candidate selection decides whether calls connect directly or through TURN.
2026-03-12 • ice-candidate, webrtc, stun
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Telecom • 9 min
Learn what RTCP is, how it works alongside RTP, and how it helps monitor jitter, packet loss, latency, and call quality in real-time voice systems.
2026-03-11 • rtcp, rtp, voip
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WebRTC • 11 min
Understand what DTLS-SRTP is in WebRTC, why browsers require it, how keys are negotiated, and how secure audio flows between browser, media server, and backend systems.
2026-03-11 • dtls-srtp, webrtc, srtp
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