How to dial from the asterisk CLI

There seem to be many, many articles and forum posts on the internet about how to trigger a call directly from the Asterisk command line.

Mostly they reference the “dial” command, which on Asterisk 17.4 doesn’t seem to exist.

Then there is voip-info which references the “originate” command but doesn’t give any context or examples. If you just type in originate into the command line it doesn’t recognise it.

You need to
channel originate PJSIP/02082446556@voiceflex-endpoint extension 01309655655@Voiceflex-Incoming

This command would dial on an outbound PJSIP extension at the VoIP provider. And would link it to extension 01309655655 in the context Voiceflex-Incoming.

(This article is mainly for my own reference at a later date).

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9 Responses to How to dial from the asterisk CLI

  1. Silhent says:

    No such command ‘channel originate’ (type ‘core show help channel originate’ for other possible commands)

  2. What version of Asterisk are you on?
    server*CLI> core show version
    Asterisk 14.3.0 built by root @ server on a x86_64 running Linux on 2017-02-26 14:48:13 UTC

    server*CLI> channel originate
    There are two ways to use this command. A call can be originated between a
    channel and a specific application, or between a channel and an extension in
    the dialplan. This is similar to call files or the manager originate action.
    Calls originated with this command are given a timeout of 30 seconds.

    Usage1: channel originate application [appdata]
    This will originate a call between the specified channel tech/data and the
    given application. Arguments to the application are optional. If the given
    arguments to the application include spaces, all of the arguments to the
    application need to be placed in quotation marks.

    Usage2: channel originate extension [exten@][context]
    This will originate a call between the specified channel tech/data and the
    given extension. If no context is specified, the ‘default’ context will be
    used. If no extension is given, the ‘s’ extension will be used.

  3. Silhent says:

    hey! this one is Asterisk 18.1.1, damn bored with this lame pjsip as everything works and was clear with sip.conf, all my call back scripts for php is not working now, not clear how to dialout without registration, as before Dial(SIP/123@sip.456.com:5060,123). Incoming calls also hell non logic way to terminate, no such endpoint if not specifiend fone number, so temporary make all anonymous calls to sip phone. Unknown or disables wrong options just block all sections to ignore without any error no matter what debug level you set.. old configs gone, this is why from scratch setuping now pjsip.

  4. Silhent says:

    channel request hangup all – works well.

  5. Edit /etc/asterisk/modules.conf
    set
    autoload = yes
    Then reload asterisk and see if the channel originate command appears?

    If not – check you have
    /usr/lib/asterisk/modules/res_clioriginate.so

    The migration to PJSIP is painful! The config is so different to sip.conf but it is a good change once done, much more reliable. Remember that your dialplan or originate commands need PJSIP/ not SIP/

  6. Silhent says:

    hold on, need to compile it

    [modules]
    autoload=yes
    noload => chan_rtp.so
    noload => chan_alsa.so
    noload => chan_console.so
    noload => res_hep.so
    noload => res_hep_pjsip.so
    noload => res_hep_rtcp.so
    noload => chan_sip.so
    noload => app_voicemail_odbc.so
    noload => app_voicemail_imap.so

  7. Silhent says:

    INVITE sip
    From: “Anonymous” ;tag
    SIP/2.0 100 Trying
    SIP/2.0 180 Ringing

    cool, now is working.. calling itself if not metion second end,

    channel originate PJSIP/123 extension 456@blabla
    channel originate PJSIP/123 application DISA(

  8. The first leg will have to answer before it then makes the call to the other side (the extension or application)

  9. Silhent says:

    right. and that is what i call 100 times try with this digium before get the way how it can go through(and probably just until the next version). no correct manuals or info.. ..expecting to work:
    channel originate PJSIP/123&PJSIP/234 extension 222@blabla
    but its not, and for disa(000) says No such application ‘DISA(000)’, but without arguments:

    WARNING[19761]: app_disa.c:168 disa_exec: DISA requires an argument (passcode/passcode file)
    heh.. from cli should be working outdial:
    channel originate PJSIP/sip:12125551234@sip.something.com extension 123@blabla

    Unable to create PJSIP channel – endpoint ‘sip.something.com’ was not found

    not clear if i want to make each call with different servers or to siphones without any registrations, need to put every single termination point in config?!

    got email? or freenode?

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