Glossary

Asterisk Gateway Interface. Lets external scripts (PHP, Python, and others) control a call by communicating with Asterisk over stdin/stdout or TCP.

Asterisk Manager Interface. A TCP socket API for monitoring events and issuing actions in Asterisk, commonly used by click-to-dial integrations and dashboards.

Asterisk REST Interface. A modern asynchronous API exposing channels, bridges, and endpoints over HTTP and WebSocket for building custom call applications.

Call Detail Record. The per-call accounting data Asterisk writes (start, answer, and end times, duration, disposition) to files or databases.

The calling party number (and optionally name) presented on an outbound call, set from the endpoint or manipulated in the dialplan.

A single call leg passing through Asterisk. Channels represent connections to endpoints and are what dialplan applications act on.

A coder-decoder that compresses audio for transmission, such as ulaw, alaw, GSM, or Opus. Endpoints negotiate which codec a call uses.

The Asterisk conference bridge application for multi-party audio conferences, configured with user, bridge, and menu profiles in confbridge.conf.

A named section of the dialplan that groups extensions. Calls enter a specific context and can only reach extensions visible from it, making contexts the basic unit of call routing and security.

Dual-Tone Multi-Frequency, the touch-tone signals phones send for digit input during a call, used by IVRs and feature codes.

The core call-routing configuration of Asterisk, written mostly in extensions.conf as contexts, extensions, and priorities that decide how every call is handled.

In PJSIP configuration, the SIP entity Asterisk communicates with: a phone, trunk, or other user agent. Defined by an endpoint section in pjsip.conf.

A dialplan entry that matches a dialed number or pattern within a context and triggers a sequence of prioritized steps.

Inter-Asterisk eXchange protocol version 2. A lightweight Asterisk-native VoIP protocol that carries signaling and media over a single UDP port (4569).

Private Branch Exchange, a private telephone switch. Asterisk is a software PBX that routes calls between internal extensions and the outside world.

The modern SIP channel driver in Asterisk (chan_pjsip), replacing the older chan_sip. Configured in pjsip.conf using endpoints, AORs, auths, and transports.

An Asterisk call queue (app_queue) that parks incoming calls and distributes them to logged-in agents according to a chosen ring strategy.

Real-time Transport Protocol. Carries the actual audio (media) of a VoIP call after SIP signaling has set it up.

Session Initiation Protocol, the standard signaling protocol used to set up, manage, and tear down VoIP calls between Asterisk and phones or carriers.

A connection between Asterisk and another PBX or an ITSP/carrier, used to send and receive external calls.

Asterisk's built-in voice messaging system (app_voicemail), configured in voicemail.conf with mailboxes per user.