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Asterisk Configuration

These configuration files act as a skeleton for a simple Asterisk PBX deployment using a SIP trunk to provide one DID, with two internal extensions. The majority of the configuration files are documented inline and are self explanatory.

This configuration has been tested on the following Linux distributions:

Ubuntu Server 18.04 | Asterisk 13.18.3

The following SIP providers were used to construct these configuration files:

Twilio

Installation

Install the latest version of Asterisk from your distribution's repositories.

$ sudo apt install asterisk -y

Remove the distribution's configuration files from /etc/asterisk/.

$ sudo rm -rf /etc/asterisk/*

Download these configuration files to /etc/asterisk/.

$ ls /etc/asterisk/
extensions.conf  modules.conf  sip.conf  voicemail.conf

Modify the configuration to suit your needs. Refer to the configuration hints for more details.

Configuration hints

Configuration files

sip.conf

This file's primary purpose is to define how Asterisk, trunks and extensions communicate and authenticate with one another.

Two templates are provided, each containing the common properties required to create either a new trunk or extension.

extensions.conf

This configuration file details how calls should move through Asterisk.

Contexts are used to namespace internal extensions, incoming calls and outgoing calls. This prevents abuse and makes future modifications easier to integrate.

modules.conf

This file is required to load Asterisk's modules. Asterisk is able to automatically load modules, so this file shouldn't need touching.

voicemail.conf

The voicemail configuration file details how voicemails work.

Configuration modifications

Create a new trunk

In sip.conf, create a new trunk which inherits from the trunk template.

Provide at the very least, a context and a host. Optionally, provide a fromuser.

context: Incoming calls will be redirected to this context.
fromuser: This is the caller ID which will appear when placing outbound calls (N.B. Twilio requires this parameter).
host: This defines how Asterisk connects to the trunk.

If authentication is used by your SIP provider, the credentials should also be inserted into this trunk definition.

[newtrunk](trunk)
    context=+441234567890
    fromuser=+441234567890
    host=example.pstn.ie1.twilio.com

Remembering that incoming calls from the trunk will be redirected to its context, create a new context in extensions.conf of the same name.

[+441234567890]
    exten => _+X.,1,Goto(internal,0,1)

This exten line states: "For all incoming calls whose caller ID starts with a '+' and is of any length, dial extension 0 in the internal context."

To use this trunk for outbound calls, add a new context in extensions.conf and reference it within the internal context's dialplan.

[outgoing_newtrunk]
    exten => _X.,1,Dial(SIP/newtrunk/+${EXTEN})
    same => n,Hangup

This exten line states: "For all outgoing calls whose caller ID is of any length, prepend the caller ID with a '+' and then call it using the newtrunk SIP trunk."

[internal]
    ...
    exten => _+X,1,Goto(outgoing_newtrunk,${EXTEN:1},1)

See the extisting outgoing redirects for more examples of how to parse dialed numbers.

Create a new extension

In sip.conf, creat e a new extension that inherits from the internal template.

This only requires a secret to be specified.

[1234](internal)
    secret=supersecurepassword

Create a new dialplan in the internal context in extensions.conf.

[internal]
    ...
    exten => 1000,1,Dial(SIP/1000,10)
    same => n,VoiceMail(${EXTEN})
    same => n,Hangup

Configure the voicemail for this extension in voicemail.conf.

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