asterisk
Open-source PBX and telephony server
TLDR
Connect to running server
$ asterisk -r -vvv
Run single command$ asterisk -r -x "command"
Show SIP peers$ asterisk -r -x "sip show peers"
Show active calls$ asterisk -r -x "core show channels"
Show voicemail users$ asterisk -r -x "voicemail show users"
Hangup a channel$ asterisk -r -x "hangup request channel_ID"
Reload SIP config$ asterisk -r -x "sip reload"
SYNOPSIS
asterisk [OPTIONS]
DESCRIPTION
asterisk runs and manages the Asterisk PBX (Private Branch Exchange) telephony server. It handles VoIP calls, traditional phone lines, and provides features like voicemail, conferencing, and interactive voice response (IVR).
PARAMETERS
-r
Reconnect to a running Asterisk instance-R
Same as -r but attempt reconnection if disconnected-x command
Execute a CLI command and exit-v
Increase verbosity (can be used multiple times)-c
Start Asterisk in console mode (foreground)-g
Dump core on crash-n
Disable ANSI color in console
CONFIGURATION
/etc/asterisk/asterisk.conf
Main configuration file controlling global settings, directory paths, and module loading./etc/asterisk/extensions.conf
Dialplan configuration defining call routing, IVR menus, and application logic./etc/asterisk/sip.conf
Legacy SIP channel driver configuration for peers, trunks, and registrations./etc/asterisk/pjsip.conf
Modern PJSIP stack configuration for endpoints, transports, and authentication./etc/asterisk/voicemail.conf
Voicemail system configuration including mailbox definitions and notification settings./etc/asterisk/modules.conf
Controls which Asterisk modules are loaded at startup.
CAVEATS
Requires proper configuration of SIP/PJSIP, dialplans, and extensions. Telephony requires understanding of VoIP protocols, codecs, and networking. Configuration is extensive and typically stored in /etc/asterisk/.
HISTORY
Asterisk is an open-source PBX and telephony toolkit created by Digium (now Sangoma). It powers communication solutions from small offices to large call centers.
SEE ALSO
asterisk-cli(8), sip.conf(5)