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-# AsteriskConfiguration
+# 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](https://www.twilio.com/sip-trunking)
+
+## Installation
+
+1. Install the latest version of Asterisk from your distribution's repositories.
+
+```bash
+$ sudo apt install asterisk -y
+```
+
+2. Remove the distribution's configuration files from `/etc/asterisk/`.
+
+```bash
+$ sudo rm -rf /etc/asterisk/*
+```
+
+3. Download these configuration files to `/etc/asterisk/`.
+
+```bash
+$ ls /etc/asterisk/
+extensions.conf  modules.conf  sip.conf  voicemail.conf
+```
+
+4. 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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