asterisk-cli

TLDR

Connect to the running Asterisk console

$ asterisk -r

Connect with verbose output

$ asterisk -rvvv

Execute a CLI command from shell

$ asterisk -rx "sip show peers"

Reload the dialplan

$ asterisk -rx "dialplan reload"

Show active calls

$ asterisk -rx "core show channels"

Show SIP registrations

$ asterisk -rx "sip show registry"

Gracefully shutdown Asterisk

$ asterisk -rx "core stop gracefully"

SYNOPSIS

asterisk -r [-v...]
asterisk -rx "command"

Asterisk CLI is the command-line interface for managing the Asterisk PBX (Private Branch Exchange) server. It provides real-time monitoring, configuration reloading, and administrative control over the telephony system.
Connect to a running Asterisk instance with asterisk -r. For scripts and automation, use asterisk -rx "command" to execute commands non-interactively. Tab completion is available for commands within the interactive console.
The CLI provides access to all Asterisk subsystems including SIP/PJSIP peers, active channels, dialplan, voicemail, queues, and the internal database. Verbosity levels (-v flags) control the amount of real-time event output displayed.
Configuration changes typically require module reloads rather than full restarts. Use dialplan reload for extensions.conf changes or module reload for specific modules.

PARAMETERS

-r

Connect to a running Asterisk instance via remote console

-R

Same as -r, but attempt reconnection if disconnected

-v

Increase verbosity (can be repeated: -vvv for more detail)

-x command

Execute a CLI command and exit

-c

Start Asterisk in console mode (foreground)

-g

Dump core on crash (for debugging)

-n

Disable ANSI color output

CLI COMMANDS

core show help [command]

Display help for CLI commands

core show channels

List active channels and calls

core show applications

List available dialplan applications

dialplan reload

Reload the dialplan configuration

sip show peers

Display configured SIP peers

sip show registry

Show SIP registration status

pjsip show endpoints

Display PJSIP endpoints (modern SIP stack)

module reload module

Reload a specific module

logger reload

Reload logging configuration

core stop gracefully

Stop Asterisk after calls complete

core stop now

Stop Asterisk immediately

database show [family]

Display AstDB database contents

CAVEATS

The Asterisk daemon must be running before connecting with -r. Some commands require specific modules to be loaded. The legacy chan_sip module uses sip commands while modern deployments use pjsip commands. Heavy verbosity can impact performance on busy systems.

HISTORY

Asterisk was created by Mark Spencer at Digium (now Sangoma) and first released in 1999. It pioneered software-based PBX systems and VoIP telephony. The CLI has evolved alongside Asterisk, with the introduction of PJSIP in Asterisk 12 (2013) providing a modern SIP stack alongside the legacy chan_sip module.