Asterisk CLI Commands Cheat Sheet: PJSIP, Calls, Dialplan, Logs

Short answer: Connect with asterisk -rvvv (or run one command with asterisk -rx "command"). The commands you will use most on a modern PJSIP system are pjsip show endpoints, pjsip show contacts, pjsip show registrations, core show channels, dialplan show and pjsip set logger on. The old sip show peers family is gone because chan_sip was removed in Asterisk 21.

We ran these commands on our lab server (Debian 12, FreePBX 17.0.33, Asterisk 22.11) on 6 October 2026. Every command below appears in that server’s core show help list. Commands that change state (reload, restart, hangup) were checked with core show help only, not run on a live call.

Connect to the Asterisk CLI

Asterisk runs as a background daemon. You attach a console to it with -r (remote). Each extra v raises the verbosity of that console only.

asterisk -rvvv                              # interactive console, verbose level 3
asterisk -rx "core show version"            # run one command and exit (good for scripts)
asterisk -rx "pjsip show endpoints" | less  # pipe long output

Inside the console, exit or quit detaches without stopping Asterisk. Never type core stop now to leave the console: it stops the PBX and drops every call.

To find a command you half remember, use the built-in help. It accepts partial commands:

core show help pjsip show
core show help dialplan

On our lab, core show help listed 412 lines. On FreePBX the CLI is the same Asterisk CLI; FreePBX adds its own tool, fwconsole, for the GUI side (reload, module admin, backups).

System status: version, uptime, calls and channels

Command	What it shows
core show version	Asterisk version and build host
core show uptime	System uptime and time since the last reload
core show calls	Active calls and total calls processed since start
core show channels	Active channels; add concise, verbose or count
core show channel <name>	Full detail for one channel (variables, codecs, bridge)
core show sysinfo	RAM, swap, process count as Asterisk sees them
core show settings	Core settings, including directories and max calls
core show taskprocessors	Queue depth per task processor; useful when the PBX feels slow
module show like pjsip	Which PJSIP modules are loaded and running

Example from our lab (masked output, idle system):

Asterisk 22.11.0 built by jenkins @ 80c01f94bd66 on a x86_64 running Linux on 2026-08-31 08:55:03 UTC

System uptime: 3 hours, 2 minutes, 42 seconds
Last reload: 3 hours, 30 seconds

core show channels count is the quickest way to feed a monitoring check. core show calls uptime seconds adds the uptime in seconds, which is handy for scripts.

PJSIP endpoints, contacts, registrations and transports

PJSIP splits a phone or trunk into several objects: an endpoint (call settings), an aor (where to reach it), an auth (credentials), a contact (the live address a phone registered from), an identify (match by source IP) and a registration (Asterisk registering out to a provider). Each has its own show command.

Command	Use it to
pjsip show endpoints	List every endpoint with its state, contacts and channels
pjsip show endpoint 201	Full settings of one endpoint (codecs, context, NAT options)
pjsip show contacts	Registered contacts with status (Avail, Unavail, NonQual, Unknown) and round-trip time
pjsip show aors	AORs and their contacts; pjsip show aor 201 for one
pjsip show auths	Auth objects used by endpoints and registrations
pjsip show registrations	Outbound registrations to providers: Registered, Rejected, Unregistered
pjsip show identifies	IP-based matching rules for trunks
pjsip show transports	Bound transports (UDP, TCP, TLS, WSS) and addresses
pjsip show channels	Active PJSIP channels
pjsip show channelstats	Live RTP stats per channel: packet loss, jitter, RTT
pjsip show settings	Global and system options
pjsip show unidentified_requests	Sources sending requests that matched no endpoint (scanner traffic)
pjsip list endpoints	Compact one-line-per-object list (also list contacts, list aors)

Most list commands take a filter. Our lab confirmed pjsip show contacts like <pattern> and pjsip show channelstats like <pattern>, where the pattern is a regular expression:

pjsip show contacts like ^20
pjsip show channelstats like trunk

Transport output from our lab (a FreePBX 17 install with the default UDP transport):

Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress....................>
==========================================================================================

Transport:  0.0.0.0-udp               udp      3     96  0.0.0.0:5060

Objects found: 1

Qualify, register and maintenance actions

Command	What it does
pjsip qualify 201	Send a SIP OPTIONS request to all contacts on the endpoint now
pjsip send register <registration>	Unregister then re-register one outbound registration; *all for every one
pjsip send unregister <registration>	Unregister and stop future attempts (until reload)
pjsip show qualify endpoint 201	Current qualify options for the endpoint’s AORs
pjsip set maintenance	Put an endpoint into maintenance mode (Asterisk 22 command; see core show help pjsip set maintenance)
module reload res_pjsip.so	Reload PJSIP configuration

Transport changes (bind address, protocol, TLS files) are not applied by a reload. The Asterisk documentation says transport changes take effect only after a full restart. Use core restart when convenient so active calls finish first.

SIP and RTP debugging

These are the commands to reach for when calls fail or audio is missing. Turn them off when you are done; they are noisy on a busy server.

Command	What it does
pjsip set logger on	Print every SIP message in and out
pjsip set logger host 203.0.113.10	Only SIP to and from one host or subnet
pjsip set logger method INVITE	Only one request method (also REGISTER, OPTIONS, BYE and others)
pjsip set logger pcap /tmp/sip.pcap	Write SIP to a pcap file for Wireshark or sngrep
pjsip set logger off	Stop SIP logging
pjsip set history on	Keep a history buffer of SIP messages; read with pjsip show history
rtp set debug ip 203.0.113.10	Show RTP packets to and from one host (optionally host:port)
rtp set debug off	Stop RTP debugging
rtp show settings	RTP port range, strict RTP, ICE and STUN settings
rtcp set stats on	Print RTCP statistics

The full syntax, taken from the Asterisk 22 source, is pjsip set logger {on|off|host <name/subnet>|add <name/subnet>|method <method>|methodadd <method>|verbose <on/off>|pcap <filename>}. add and methodadd extend an existing filter instead of replacing it.

Our lab’s rtp show settings on a stock FreePBX 17 install:

General Settings:
----------------
  Port start:      10000
  Port end:        20000
  Checksums:       Yes
  DTMF Timeout:    1200
  Strict RTP:      Yes
  Probation:       4 frames
  Replay Protect:  Yes
  ICE support:     Yes
  STUN:            disabled

For a friendlier view of SIP ladders, our sngrep guide shows how to read the same traffic, and the SIP Trace Analyzer flags NAT and codec problems in a pasted log.

Dialplan, applications and functions

Command	What it does
dialplan show	Whole dialplan (long on FreePBX)
dialplan show from-internal	One context
dialplan show 2125551234@from-internal	Which extensions in that context match a number, in priority order
dialplan reload	Reload extensions.conf only
core show application Dial	Syntax and options of an application
core show function CALLERID	Syntax of a dialplan function
core show hints	BLF hints and their current state
dialplan show globals	Global variables
database show	AstDB contents; database show <family> to narrow

dialplan show <number>@<context> is the fastest way to answer “which route will this number take?”. We used it on lab3 to test patterns; see our dialplan pattern matching guide for real output.

On FreePBX, fwconsole context from-internal prints a context too. Edits to FreePBX-generated files are overwritten; put custom dialplan in extensions_custom.conf.

Logging, reloads and restarts

Command	What it does
core set verbose 5	Raise verbosity on this console (core set verbose off to silence)
core set debug 3 res_pjsip.so	Debug messages for one module (core set debug off)
logger show channels	Which log files exist and which levels they receive
logger rotate	Rotate and reopen log files
logger reload	Reopen log files after editing logger.conf
module reload	Reload all modules that support reload (or name one, e.g. module reload res_pjsip.so)
core reload	Global reload
core restart when convenient	Restart once all calls have ended
core restart gracefully	Stop accepting new calls, restart when current calls end
core stop gracefully	Same, but stop instead of restart

logger show channels on our lab, after we added a fail2ban log channel:

Channel                             Type     Formatter  Status    Configuration
-------                             ----     ---------  ------    -------------
/var/log/asterisk/fail2ban          File     default    Enabled    - NOTICE SECURITY
/var/log/asterisk/full              File     default    Enabled    - DEBUG NOTICE WARNING ERROR VERBOSE

On FreePBX, prefer fwconsole reload after GUI changes and fwconsole restart for a full restart. They regenerate the config files before Asterisk reloads them.

Calls, recordings and queues

Command	What it does
channel request hangup PJSIP/201-00000012	Hang up one channel; accepts all, globs like PJSIP/* or a /regex/
channel originate PJSIP/201 extension 600@from-internal	Place a test call from a phone to an extension
mixmonitor start <channel>	Start recording a live channel (stop and list also exist)
queue show	Queues, members and waiting callers
queue {pause|unpause} member	Pause an agent
confbridge list	Active conferences
voicemail show users	Mailboxes
cdr show status	Whether CDR logging is on and in which mode

channel request hangup all drops every call on the server. Run core show channels first and hang up by name.

chan_sip commands and their PJSIP replacements

Older guides still show chan_sip commands. They do not exist on Asterisk 21 or later. Use this map:

Old chan_sip command	PJSIP equivalent
sip show peers	pjsip show endpoints or pjsip show contacts
sip show peer 201	pjsip show endpoint 201
sip show registry	pjsip show registrations
sip set debug on	pjsip set logger on
sip set debug ip 203.0.113.10	pjsip set logger host 203.0.113.10
sip reload	module reload res_pjsip.so
sip qualify peer 201	pjsip qualify 201
sip show channels	pjsip show channels

Still migrating? Our chan_sip to PJSIP migration guide covers the converter script and option mapping.

Official documentation: Asterisk CLI documentation · Asterisk PJSIP configuration · Asterisk versions

Related: Troubleshoot SIP Calls with sngrep and the Asterisk PJSIP Logger · SIP Trace Analyzer: Find NAT, Codec and Dropped-Call Problems in a SIP Log · PJSIP Behind NAT: Asterisk and FreePBX Settings for Two-Way Audio · SIP Response Codes: Lookup for Every SIP Error, With Causes and Fixes · Install Asterisk 22 LTS on Debian 13: Step-by-Step Guide

See also: chan_sip to PJSIP Migration: sip_to_pjsip.py, Option Mapping, Tests · sip.conf to pjsip.conf Converter for Asterisk · PJSIP Endpoint Unreachable or Unavailable: Fix Qualify and NAT · FreePBX PJSIP Trunk Setup: Registration, IP Auth, Routes, NAT

See also: Asterisk pjsip_wizard.conf: Endpoints and Trunks in a Few Lines · Asterisk AMI Originate a Call with Python

Frequently asked questions