Himanshu Pal

VoIP Capacity Planning: Sizing Servers, Trunks and Bandwidth

How to work out how many concurrent calls you actually need, what limits an Asterisk server in practice, and why transcoding is usually the thing that breaks first.

WebRTC with Asterisk: Browser Calling That Actually Works

What webrtc=yes really turns on, why WSS and DTLS are mandatory rather than optional, and the configuration that gets a browser making calls through Asterisk.

Asterisk ARI: Building Call Control Applications with Stasis

How the Asterisk REST Interface works, what Stasis actually does, the WebSocket event model, and the bridge subscription gotcha that catches everyone once.

Hardening an Asterisk Server: A Practical Security Checklist

A working checklist for securing an Asterisk PBX — network exposure, SIP credentials, dialplan permissions, management interfaces, and the monitoring that catches what prevention misses.

Asterisk Music on Hold: Classes, Formats and Streaming Sources

How music on hold classes work, why you should convert MoH files to a native format, per-queue and per-channel music, and how to disable it cleanly.

Asterisk Voicemail: Configuration, Email Delivery and Common Failures

How voicemail.conf is structured, the VoiceMail application options that matter, how to get voicemail-to-email working, and why messages sometimes go to the wrong mailbox.

What a Session Border Controller Does (and When You Actually Need One)

What an SBC is for, the five jobs it performs at the edge of a VoIP network, and an honest assessment of when a PBX with a good firewall is enough.

VoIP Toll Fraud: How Attackers Get In and How to Stop Them

How toll fraud actually works, why the bill lands on you, and the layered controls that limit the damage when an extension is compromised.

Asterisk Manager Interface (AMI): A Practical Guide with Examples

How AMI works, what the read and write permission classes actually control, a worked login-and-originate session, and the security mistakes that expose a PBX.

Encrypting VoIP: SIP over TLS and SRTP, Configured Properly

Why unencrypted SIP is readable by anyone on the path, how TLS and SRTP each protect a different half of the call, and how to configure both in Asterisk without breaking audio.