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Asterisk's CLI is where you, the administrator, control and monitor the Asterisk PBX. The Asterisk CLI provides you with real-time information about voice channels, extensions, contexts, and more. Using the CLI, you can start and stop the Asterisk server, as described earlier in the chapter. You can do status queries about the voice channels, place and hang up calls, and add or remove extensions and contexts.
To start asterisk in command line mode you should use
# asterisk -c
If you have just installed Asterisk, you can try running it for the first time in console mode with some debugging applied
with this command:
/usr/sbin/asterisk -vvvgc
To stop enter command
stop now
See Asterisk CLI for available commands.
Use the command stop now to shut down Asterisk.
If the Asterisk server is already running, you can launch the Asterisk CLI by starting another instance of Asterisk in client mode:
# asterisk -r
The first command you should probably learn is help, which displays a list of valid commands or, when used as follows, gives command-specific assistance:
pbx*CLI> help show channels
Notice the pbx*CLI> prompt. This prompt will vary depending on the Unix hostname of your machine—in this case, it's pbx.
The simplest way to control Asterisk from an external shell or application is to issue the command asterisk with the option -rx followed by the CLI command. Any CLI command may be entered from the system shell in this fashion.
ExampleSay you wanted to see the dialplan for extension 23 in the context [my-phones]; you would do this with asterisk -rx "dialplan show 23@my-phones" entered in the shell:
#asterisk -rx "dialplan show 23@my-phones"
[ Context 'my-phones' created by 'pbx_config' ]
'23' => 1. Wait(1) [pbx_config]
2. Answer() [pbx_config]
3. Playback(hello-world) [pbx_config]
4. Wait(1) [pbx_config]
5. Hangup() [pbx_config]
-= 1 extension (5 priorities) in 1 context. =-
#
There are several types of voice channels in Asterisk. There can be analog FXO (foreign exchange office) provided by POTS telephone company line and an X100P card. There can also be voice channels provided by H.323, SIP, IAX, SCCP, and MGCP (all different VoIP signaling protocols), as well as by FXS (foreign exchange station) devices like QuickNet's Phone Jack interface card and PRI interfaces, too.
You can use the Asterisk CLI to inspect each type of channel, but the details provided are different for each:
sip show channels
zap show channels
iax2 show channels
Each of these commands provides a list of channel numbers that can correspond to a numeric reference in the channel name. Take a look at a sip show channels command:
sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 0 active SIP channels
The command output shows that there is no SIP active sip channel. Useful CLI commands to check sip peers/users:
sip show peers Show all SIP peers (including friends)
sip show users Show all SIP users (including friends)
sip show registry Show status of hosts we register with
sip set debug on Show all SIP messages
module reload chan_sip.so Reload configuration file
You can dig deeper using the CLI, to the configuration of a specific channel, by running the zap show channel <channel#> command, like this:
pbx*CLI> zap show channel 1
Channel: 1
File Descriptor: 17
Span: 1
Extension:
Context: incoming
Caller ID string:
Destroy: 0
Signalling Type: FXS Kewlstart
Owner: <None>
Real: <None>
Callwait: <None>
Threeway: <None>
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: yes
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
There are many channel-specific settings, most of which are user-definable. You can see from this list that, even within the scope of a single Zaptel channel, Asterisk is extremely flexible. In our test server, this channel is an FXO channel—that is, a channel that connects to the telephone company. In order to interact correctly with the phone company's switch, the X100P (or other compatible) interface has to use FXS signaling, so that, to the phone company, it appears to be a regular telephone answering the line and not a newfangled Asterisk server (FXO/FXS signaling was introduced in Chapter 4). Other types of channels (SIP, H.323, etc.) have their own channel-specific settings.
The show dialplan CLI command can be used two ways: showing the entire system-wide dial-plan, or only showing a specific context or extension. Here's the output for a small, but complete, dial-plan:
pbx*CLI> show dialplan
[ Context 'default' created by 'pbx_config' ]
'3101' => 1. Dial(SIP/[email protected]) [pbx_config]
2. Voicemail(u4101) [pbx_config]
'3102' => 1. Dial(SIP/[email protected]) [pbx_config]
2. Voicemail(u4101) [pbx_config]
'3103' => 1. Dial(SIP/[email protected]) [pbx_config]
2. Voicemail(u4101) [pbx_config]
'8500' => 1. VoiceMailMain() [pbx_config]
'_9NXXXXXX' => 1. Dial(${PSTN}/{EXTEN:1}) [pbx_config]
2. Congestion() [pbx_config]
Ignore pattern => '9' [pbx_config]
[ Context 'incoming' created by 'pbx_config' ]
's' => 1. Answer() [pbx_config]
2. Dial(SIP/[email protected]|30) [pbx_config]
3. Voicemail(u4101) [pbx_config]
4. Hangup() [pbx_config]
In the default context, several extensions are configured to ring calls on specific SIP telephones with static IP addresses and an extension pattern that matches outbound calls. The 8500 extension is one that allows private users of the Asterisk system to access the voice mail management greeting.
There's also an incoming context that attempts to ring a certain phone with the s extension and then transfers the call on the current channel to a voice mailbox if nobody answers the phone.
To get a specific extension's dial-plan only, run the show dialplan CLI command with an argument of extension@context:
pbx*CLI> show dialplan 3102@default
[ Context 'default' created by 'pbx_config' ]
'3102' => 1. Dial(SIP/[email protected]) [pbx_config]
2. Voicemail(u4101) [pbx_config]
In this case, only the extension definition for 3102 in the default context is shown.
Earlier, we covered the primary configuration file for the dial-plan: extensions.conf. You can also alter the dial-plan using CLI commands. The difference between using the CLI and using the configuration file is this: CLI dial-plan changes, such as additions and removals of extensions, occur immediately, whereas changes to extensions.conf require a restart of Asterisk (or at least a config-reload), which introduces downtime.
Consider the following CLI command:
pbx*CLI> add extension 120,1,Dial(SIP/[email protected]:5060) into localThe command add extension is followed by the same definition string you would see in an exten directive in extensions.conf—that is, the extension number, priority, and application command for the extension. The final part, into local, tells Asterisk which context to add the extension into—in this case, local.
The CLI can also be used to drop extensions, using the remove extension command, and to re-create the extensions.conf file based on the current "live" dial-plan, using the save dialplan command.
If you prefer to use the configuration file to maintain your dial-plan, you can avoid total Asterisk restarts by issuing a reload extensions command, which re-reads the extensions.conf file and affects the dial-plan accordingly. That said, using the CLI may be the preferred method for programming the dial-plan, because it gives you immediate feedback in case of syntax errors or other common mistakes. In order to save changes made at the CLI, your extensions.conf must contain the writeprotect=no setting in its general section, and you must issue a save dialplan command at the CLI. This will cause the current dial-plan configuration to overwrite anything in your extensions.conf file (including comments).
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Last modified: March 12, 2019
