Asterisk Guide

The only Asterisk book written for the Asterisk you actually run today. Every other guide on the shelf teaches chan_sip, a module that no longer exists. This second edition is rebuilt from the ground up for Asterisk 22 LTS, where PJSIP is the one and only SIP channel, and every configuration and command in the book was verified against a reproducible Asterisk 22 Docker lab. If an example is in here, it ran.

Starting from a clean Linux box, you will install Asterisk, build a working PJSIP PBX, and grow it into a real system: SIP and PJSIP internals, WebRTC browser phones over WSS/DTLS-SRTP, SIP trunking with DIDs and the PSTN, a complete dialplan, queues and call-center features, voicemail, conferencing with ConfBridge, CDR/CEL accounting, AMI/AGI/ARI integration, PJSIP Realtime, hardening against toll fraud, and deployment, monitoring and scaling for production.

Written by Flavio E. Goncalves, Asterisk dCAP, CEO of SipPulse, and author of several telephony titles, with end-of-chapter quizzes and a companion hands-on course and labs at VoIP School Blackbelt. Whether you are new to Asterisk or migrating an aging chan_sip system to PJSIP, this is the current, lab-proven reference for Asterisk 22.