### Install wscat via npm

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/Getting-Started-with-ARI

Installs the wscat command-line utility, which is used to establish a WebSocket connection for receiving ARI events. Requires Node.js and npm to be installed.

```bash
apt-get install npm
npm install -g wscat
```

--------------------------------

### AEAP Setup Response - Example 1

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-External-Application-Protocol-AEAP

An example of a 'setup' response confirming the 'ulaw' codec selection.

```json
{
 "response": "setup",
 "id": "7180e464-c022-40e6-a49c-9d2007f008a7",
 "codecs": [{ "name": "ulaw" }],
}

```

--------------------------------

### AEAP Setup Request - Example 3

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-External-Application-Protocol-AEAP

An example of an AEAP 'setup' request including optional 'params' for language and gender.

```json
{
 "request": "setup",
 "id": "b3eda99f-464f-477e-b8a3-4ebafda0b482",
 "version": "0.1.0",
 "codecs": [ { "name": "ulaw" } ],
 "params": { "language": "en", "gender": "male" },
}

```

--------------------------------

### Install MySQL Server and Start Service (Bash)

Source: https://docs.asterisk.org/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/ODBC/Getting-Asterisk-Connected-to-MySQL-via-ODBC

Installs the MySQL server package using yum and starts the mysqld service. This is the initial step for setting up MySQL.

```bash
# sudo yum install mysql-server
# sudo service mysqld start
```

--------------------------------

### Install Blink SIP Client using setup.py

Source: https://docs.asterisk.org/Deployment/Secure-Calling/Secure-Calling-Tutorial/Installing-Blink-SIP-client

Installs the Blink SIP client after downloading and extracting its source code. This process involves navigating to the source directory and executing the Python setup script.

```bash
sudo python setup.py install
```

--------------------------------

### AEAP Setup Request - Example 1

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-External-Application-Protocol-AEAP

An example of a basic AEAP 'setup' request specifying the 'ulaw' codec.

```json
{
 "request": "setup",
 "id": "9c8d7625-d70d-4225-8b4f-53063ea8e774",
 "version": "0.1.0",
 "codecs": [ { "name": "ulaw" } ],
}

```

--------------------------------

### Install Asterisk Sample Configurations

Source: https://docs.asterisk.org/Test-Suite-Documentation/Using-Python3

This command installs the sample configuration files for Asterisk. It's important to back up any existing custom configurations in `/etc/asterisk` before running this command, as it will overwrite them.

```bash
sudo make samples

```

--------------------------------

### PJSIP Configuration Wizard Examples

Source: https://docs.asterisk.org/Asterisk_22_Documentation/API_Documentation/Module_Configuration/res_pjsip_config_wizard

Examples demonstrating how to use the PJSIP configuration wizard for phone provisioning and trunk setup.

```APIDOC
## PJSIP Configuration Wizard Examples

### Phone Configuration Example (`myphone`)

This configuration snippet creates the necessary endpoint, aor, contact, auth, and phoneprov objects for a phone to receive provisioning information, register, and handle calls.

```ini
[myphone]
type = wizard
sends_auth = no
accepts_auth = yes
sends_registrations = no
accepts_registrations = yes
has_phoneprov = yes
transport = ipv4
has_hint = yes
hint_exten = 1000
inbound_auth/username = testname
inbound_auth/password = test password
endpoint/allow = ulaw
endpoint/context = default
phoneprov/MAC = 001122aa4455
phoneprov/PROFILE = profile1
```

### Trunk Configuration Example (`mytrunk`)

This configuration snippet creates the necessary endpoint, aor, contact, auth, identify, and registration objects for a trunk to another PBX or ITSP that requires registration.

```ini
[mytrunk]
type = wizard
sends_auth = yes
accepts_auth = no
sends_registrations = yes
accepts_registrations = no
transport = ipv4
remote_hosts = sip1.myitsp.com:5060,sip2.myitsp.com:5060
outbound_auth/username = testname
outbound_auth/password = test password
endpoint/allow = ulaw
endpoint/context = default
```
```

--------------------------------

### Install Asterisk Initialization Script (Bash)

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Installing-Initialization-Scripts

Installs the Asterisk initscript, which is responsible for starting, monitoring, stopping, and restarting the Asterisk process automatically when the server boots. This command is executed from the Asterisk source directory.

```bash
#!/bin/bash
# Install the Asterisk init script
make config
```

--------------------------------

### Install ASTTest using Make

Source: https://docs.asterisk.org/Test-Suite-Documentation/Using-Python3

This snippet shows the standard command-line steps to compile and install the ASTTest tool using the 'make' utility. It assumes the user has navigated to the 'asttest' directory.

```bash
cd asttest
make
sudo make install
```

--------------------------------

### Install sipp Tool

Source: https://docs.asterisk.org/Test-Suite-Documentation/Using-Python3

This command installs the compiled sipp tool to the system. It requires superuser privileges.

```bash
sudo make install

```

--------------------------------

### Build and Install sipp Tool

Source: https://docs.asterisk.org/Test-Suite-Documentation/Using-Python3

This snippet demonstrates how to download, build, and install the sipp tool from its source code. It includes steps for checking out a specific known-working version and configuring the build with necessary options like OpenSSL, PCAP, RTPStream, SCTP, and GSL. Ensure you have the required development packages installed before running this.

```bash
cd /usr/src/asterisk
git clone https://github.com/SIPp/sipp.git
cd sipp
git checkout v3.6.0 ## This is the latest version we KNOW works.
./build.sh --prefix=/usr --with-openssl --with-pcap --with-rtpstream --with-sctp --with-gsl CFLAGS=-w

```

--------------------------------

### AEAP Setup Request - Example 2

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-External-Application-Protocol-AEAP

An example of an AEAP 'setup' request with multiple codecs, including 'opus' with an attribute.

```json
{
 "request": "setup",
 "id": "7180e464-c022-40e6-a49c-9d2007f008a7",
 "version": "0.1.0",
 "codecs": [ { "name": "ulaw" }, { "name": "opus", "attributes": { "maxplaybackrate": 8000 } } ],
}

```

--------------------------------

### Install curl via apt-get

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/Getting-Started-with-ARI

Installs the curl utility, which is used to send HTTP requests to control channels within the Stasis dialplan application via ARI. This is a common method for interacting with RESTful APIs.

```bash
apt-get install curl
```

--------------------------------

### Start Tigase Server

Source: https://docs.asterisk.org/Configuration/Interfaces/Distributed-Device-State/Distributed-Device-State-with-XMPP-PubSub

This command starts the Tigase server using its startup script. It requires navigating to the Tigase installation directory and specifying the path to the configuration file.

```bash
# cd /opt/Tigase-4.3.1-b1858
# ./scripts/tigase.sh start etc/tigase.conf
```

--------------------------------

### Asterisk Error Response Example: Setup Failed

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-External-Application-Protocol-AEAP

An example of an error response for a 'setup' request, indicating a failure due to unsupported codecs. It includes the request type, ID, and a specific error message.

```json
{
 "response": "setup",
 "id": "b3eda99f-464f-477e-b8a3-4ebafda0b482",
 "error_msg": "No supported codec(s)"
}
```

--------------------------------

### Install Asterisk Sample Configuration Files

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Installing-Sample-Files

This command installs a set of sample configuration files for Asterisk. Existing modified sample files will be renamed with a .old extension before installation.

```bash
[root@server asterisk-14.X.Y]# make samples

```

--------------------------------

### Start DAHDI Service (Bash)

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Validating-Your-Installation

These commands demonstrate how to start the DAHDI service. The primary command uses the initscript. Alternative commands for Red Hat-based systems (`service`) and Debian-based systems (`invoke-rc.d`) are also provided.

```bash
[root@server asterisk-14.X.Y]# /etc/init.d/dadhi start

```

```bash
[root@server asterisk-14.X.Y]# service dahdi start

```

```bash
[root@server asterisk-14.X.Y]# invoke-rc.d dahdi start

```

--------------------------------

### Install Asterisk Log Rotation Script (Bash)

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Installing-Initialization-Scripts

Installs the logrotation script for Asterisk. This script helps manage log files by compressing and rotating them, saving disk space and improving searchability. Execute this command from the Asterisk source directory.

```bash
#!/bin/bash
# Install the Asterisk log rotation script
make install-logrotate
```

--------------------------------

### Configure ARI User in Asterisk

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/Getting-Started-with-ARI

Sets up an ARI user with credentials in ari.conf. 'enabled = yes' and 'pretty = yes' enable the ARI interface and pretty-print JSON responses. The 'asterisk' user is for demonstration; use secure credentials in production.

```ini
[general]
enabled = yes 
pretty = yes 

[asterisk]
type = user
read_only = no
password = asterisk
```

--------------------------------

### Check sipp Tool Version

Source: https://docs.asterisk.org/Test-Suite-Documentation/Using-Python3

After building and installing sipp, this command verifies the installed version to ensure it meets the required specification. A successful output indicates the tool is ready for use.

```bash
./sipp -v

```

--------------------------------

### Connecting to Asterisk ARI

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/Getting-Started-with-ARI

This section details how to establish a WebSocket connection to Asterisk ARI using wscat and subscribe to events for a specific application.

```APIDOC
## Connect to Asterisk using `wscat`

### Description
Establish a WebSocket connection to the Asterisk ARI interface to receive real-time events.

### Method
GET (via WebSocket)

### Endpoint
`ws://localhost:8088/ari/events?api_key=<key>&app=<app_name>`

### Parameters
#### Query Parameters
- **api_key** (string) - Required - The API key for authentication (e.g., `asterisk:asterisk`).
- **app** (string) - Required - The name of the Stasis application to subscribe to (e.g., `hello-world`).

### Request Example
```bash
wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk&app=hello-world"
```

### Response
Upon connection, Asterisk will confirm the WebSocket connection and the creation of the specified Stasis application.

#### Success Response (Connection Established)
```
connected (press CTRL+C to quit)
Creating Stasis app 'hello-world'
```

#### Event Example (StasisStart)
```json
{
 "application":"hello-world",
 "type":"StasisStart",
 "timestamp":"2014-05-20T13:15:27.131-0500",
 "args":[],
 "channel":{
 "id":"1400609726.3",
 "state":"Up",
 "name":"PJSIP/1000-00000001",
 "caller":{
 "name":"",
 "number":""},
 "connected":{
 "name":"",
 "number":""},
 "accountcode":"",
 "dialplan":{
 "context":"default",
 "exten":"1000",
 "priority":3},
 "creationtime":"2014-05-20T13:15:26.628-0500"}
 }
```
```

--------------------------------

### Build and Install Asterisk with Development Flags

Source: https://docs.asterisk.org/Test-Suite-Documentation/Using-Python3

This section covers building and installing Asterisk with specific development flags enabled. The `--enable-dev-mode` flag is crucial for testing, and `--disable-binary-modules` simplifies the build process by excluding external codecs. The `make install-headers` command is necessary for the testsuite.

```bash
make
sudo make install
sudo make install-headers

```

--------------------------------

### Asterisk SLA Configuration (sla.conf)

Source: https://docs.asterisk.org/Configuration/Applications/Shared-Line-Appearances-SLA/SLA-Configuration-Examples/Basic-SLA-Configuration-Example

This snippet shows the basic configuration for SLA in Asterisk's sla.conf file. It defines trunks (line1, line2) and stations (station1, station2, station3) with their respective device and context mappings. Dependencies include dahdi.conf and sip.conf for device and context setup.

```asterisk
[line1]
type=trunk 
device=DAHDI/1 
autocontext=line1 

[line2] 
type=trunk 
device=DAHDI/2 
autocontext=line2 

[station] 
type=station 
trunk=line1 
trunk=line2 
autocontext=sla_stations 

[station1](station) 
device=SIP/station1 

[station2](station) 
device=SIP/station2 

[station3](station) 
device=SIP/station3

```

--------------------------------

### Start Asterisk Service (Bash)

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Validating-Your-Installation

This command initiates the Asterisk service using its initscript. Asterisk typically runs as a background daemon, so no output is usually displayed on the command line upon successful startup.

```bash
[root@server asterisk-14.X.Y]# /etc/init.d/asterisk start
Starting asterisk: 

```

--------------------------------

### Install Sample Configuration Files

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Building-and-Installing-Asterisk

Installs the sample configuration files to the default Asterisk configuration directory (`/etc/asterisk/`). This command will overwrite any existing configuration files.

```bash
make samples

```

--------------------------------

### Download Asterisk using wget

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/What-to-Download

This snippet demonstrates how to download the Asterisk source code using the `wget` command-line utility. It retrieves a specific version (asterisk-14-current.tar.gz) from the official Asterisk download server. The output shows the download progress and confirmation.

```bash
[root@server:/usr/local/src]# wget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-14-current.tar.gz
--2017-04-28 15:45:36-- https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-14-current.tar.gz
Resolving downloads.asterisk.org (downloads.asterisk.org)... 76.164.171.238
Connecting to downloads.asterisk.org (downloads.asterisk.org)|76.164.171.238|:443... connected.
HTTP request sent, awaiting response... 200 OK
Length: 40692588 (39M) [application/x-gzip]
Saving to: ‘asterisk-14-current.tar.gz’

asterisk-14-current.tar.gz 100%[======================================================================>] 38.81M 3.32MB/s in 12s 

2017-04-28 15:45:47 (3.37 MB/s) - ‘asterisk-14-current.tar.gz’ saved [40692588/40692588]

```

--------------------------------

### Generate Development Documentation

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Building-and-Installing-Asterisk

Uses doxygen to locally generate HTML development documentation from the Asterisk source code. The documentation is created in the `doc/` subdirectory of the source code.

```bash
make progdocs

```

--------------------------------

### Setting Up Documentation Build Environment

Source: https://docs.asterisk.org/Contributing-to-the-Documentation

These commands outline the steps to set up a Python virtual environment and install dependencies required for building the Asterisk documentation locally. It includes installing virtualenv, creating the environment, activating it, and installing packages from requirements.txt.

```bash
# Install python3-virtualenv
$ sudo (apt | dnf | yum) install python3-virtualenv
$ cd asterisk-documentation
$ python3 -m venv --system-site-packages --symlinks .venv
$ source .venv/bin/activate
$ pip install -r requirements.txt

```

--------------------------------

### AEAP Get Request - Example 1

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-External-Application-Protocol-AEAP

An example of a 'get' request to retrieve the 'language' parameter.

```json
{
 "request": "get",
 "id": "f066e049-d4ef-4093-a05b-0effef1bc077",
 "params": ["language"],
}

```

--------------------------------

### AEAP Get Response - Example 1

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-External-Application-Protocol-AEAP

An example of a 'get' response returning the 'language' parameter with its value.

```json
{
 "response": "get",
 "id": "f066e049-d4ef-4093-a05b-0effef1bc077",
 "params": { "language" : "en" },
}

```

--------------------------------

### Install Asterisk to a Custom Directory

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Building-and-Installing-Asterisk

Configures the Asterisk installation to use a custom directory specified by the --prefix option. This allows for sandboxing Asterisk instances or installing to non-default locations.

```bash
[root@server asterisk-14.X.Y]# ./configure --prefix=/usr/local/my_special_folder

```

--------------------------------

### Asterisk Endpoint Configuration

Source: https://docs.asterisk.org/Development/Roadmap/Asterisk-12-Projects/New-SIP-channel-driver/New-SIP-Channel-Driver-Architecture/res_sip-design/Roadmap/Asterisk-12-Projects/New-SIP-channel-driver/New-SIP-Channel-Driver-Architecture/res_sip-design/Endpoints-and-Location-A-Match-Made-in-Heaven

Example configuration for an Asterisk endpoint. This sets up basic parameters like context, disallowed codecs, allowed codecs, and associated AORs.

```asterisk
[5000]
type=endpoint
context=default
disallow=all
allow=g722
allow=ulaw
aors=5000
```

--------------------------------

### Asterisk Error Response Example: Get Failed

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-External-Application-Protocol-AEAP

An example of an error response for a 'get' request, indicating that a requested parameter (language) is not available. It shows the response type, ID, and the error message.

```json
{
 "response": "get",
 "id": "f066e049-d4ef-4093-a05b-0effef1bc077",
 "error_msg": "No language parameter available"
}
```

--------------------------------

### Verify Asterisk Corosync Cluster Members (CLI)

Source: https://docs.asterisk.org/Configuration/Interfaces/Distributed-Device-State/Corosync

These snippets show the expected output from the Asterisk CLI command 'corosync show members' to verify the cluster status. The first example shows a single node setup, while the second shows a two-node cluster. This command is used after installing and configuring Corosync and Asterisk.

```bash
*CLI> corosync show members

=============================================================
=== Cluster members =========================================
=============================================================
===
=== Node 1
=== --> Group: asterisk
=== --> Address 1: <host #1 ip goes here>
===
=============================================================

```

```bash
*CLI> corosync show members 

=============================================================
=== Cluster members =========================================
=============================================================
===
=== Node 1
=== --> Group: asterisk
=== --> Address 1: <host #1 ip goes here>
=== Node 2
=== --> Group: asterisk
=== --> Address 1: <host #2 ip goes here>
===
=============================================================

```

--------------------------------

### Asterisk Dialing Examples

Source: https://docs.asterisk.org/Development/Roadmap/Asterisk-12-Projects/New-SIP-channel-driver/New-SIP-Channel-Driver-Architecture/res_sip-design/Roadmap/Asterisk-12-Projects/New-SIP-channel-driver/New-SIP-Channel-Driver-Architecture/res_sip-design/Endpoints-and-Location-A-Match-Made-in-Heaven

Examples demonstrating how to dial using the Dial application in Asterisk, covering endpoints, explicit AORs, and explicit SIP URIs.

```asterisk
Dial(PJSIP/5000)
```

```asterisk
Dial(PJSIP/5000/5000)
```

```asterisk
Dial(PJSIP/5000/sip:1234@test.com)
```

--------------------------------

### Build and Install pjproject

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Prerequisites/PJSIP-pjproject

Commands to build and install the pjproject library after configuration. It first ensures dependencies are met, then compiles the source code, and finally installs the compiled libraries and headers.

```bash
# make dep
# make
# make install
```

--------------------------------

### Asterisk Static Realtime HTTP GET Request Example

Source: https://docs.asterisk.org/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/cURL

This snippet demonstrates an example HTTP GET request made by Asterisk to retrieve static realtime data. It specifies the target file ('pjsip.conf') as a URL parameter. The request includes standard HTTP headers like 'User-Agent' and 'Host'.

```http
GET /astconfig/static?file=pjsip.conf HTTP/1.1
User-Agent: asterisk-libcurl-agent/1.0
Host: localhost:8000
Accept */*


```

--------------------------------

### Install Asterisk

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Building-and-Installing-Asterisk

Installs the compiled Asterisk program and modules to the system. This command should be run after a successful compilation.

```bash
[root@server asterisk-14.X.Y]# make install

```

--------------------------------

### Secure MySQL Installation (Bash)

Source: https://docs.asterisk.org/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/ODBC/Getting-Asterisk-Connected-to-MySQL-via-ODBC

Executes the mysql_secure_installation script to apply basic security measures to the MySQL installation. Recommended for production environments.

```bash
# sudo /usr/bin/mysql_secure_installation
```

--------------------------------

### Configure and Build Asterisk

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Prerequisites/PJSIP-pjproject

Standard commands to configure, build, and install Asterisk. Assumes Asterisk 15+ is being used. This process typically involves downloading and linking the pjproject dependency.

```bash
./configure
make && make install
```

--------------------------------

### Asterisk Dialplan Goto() Examples with Priority Labels

Source: https://docs.asterisk.org/Deployment/Basic-PBX-Functionality/Auto-attendant-and-IVR-Menus/Goto-Application-and-Priority-Labels

Demonstrates the usage of the Asterisk Goto() application with different parameter counts (1, 2, and 3) and the use of named priority labels (e.g., 'monkeys', 'start'). This allows for flexible navigation within and between dialplan contexts. Assumes Asterisk dialplan syntax.

```asterisk
[StartingContext]
exten => 100,1,Goto(monkeys)
 same => n,NoOp(We skip this)
 same => n(monkeys),Playback(tt-monkeys)
 same => n,Hangup()

exten => 200,1,Goto(start,1) ; play tt-weasels then tt-monkeys

exten => 300,1,Goto(start,monkeys) ; only play tt-monkeys

exten => 400,1,Goto(JumpingContext,start,1) ; play hello-world

exten => start,1,NoOp()
 same => n,Playback(tt-weasels)
 same => n(monkeys),Playback(tt-monkeys)

[JumpingContext]
exten => start,1,NoOp()
 same => n,Playback(hello-world)
 same => n,Hangup()
```

--------------------------------

### Install Blink Dependencies on Ubuntu

Source: https://docs.asterisk.org/Deployment/Secure-Calling/Secure-Calling-Tutorial/Installing-Blink-SIP-client

Installs essential Python libraries required for the Blink SIP client on Ubuntu 12.04. This command ensures all prerequisites are met before proceeding with the Blink installation.

```bash
sudo apt-get update
sudo apt-get install python-sipsimple python-gnutls python-eventlib python-application python-cjson python-eventlet python-qt4 python-twisted-core python-zope.interface
```

--------------------------------

### Start Asterisk in Console Mode with Verbosity 1

Source: https://docs.asterisk.org/Operation/Logging/Verbosity-in-Core-and-Remote-Consoles

This command starts the Asterisk application in console mode, setting the root console verbosity to level 1. This is useful for initial setup or debugging.

```bash
asterisk -cv

```

--------------------------------

### Run Blink SIP Client

Source: https://docs.asterisk.org/Deployment/Secure-Calling/Secure-Calling-Tutorial/Installing-Blink-SIP-client

Executes the Blink SIP client from the command line. This command launches the application, which should then appear in the graphical desktop environment.

```bash
blink
```

--------------------------------

### Asterisk Message Router Setup and Callback Examples

Source: https://docs.asterisk.org/Development/Roadmap/Asterisk-12-Projects/Asterisk-12-API-Improvements/Stasis-Message-Bus/Using-the-Stasis-Message-Bus

This C code illustrates the setup and usage of the Asterisk stasis_message_router. It includes example callbacks for specific message types (e.g., stasis_cache_update_type) and a default route handler for miscellaneous messages, including the final unsubscribe message. The `load_module` function demonstrates router creation and addition of routes, while `unload_module` shows how to unsubscribe.

```c
static void updates(void *data, struct stasis_subscription *sub,
 struct stasis_topic *topic, struct stasis_message *message)
{
 /* Since this came from a message router, we know the type of the
 * message. We can cast the data without checking its type.
 */
 struct stasis_cache_update *update = stasis_message_data(message);
 /* ..

}
static void default_route(void *data, struct stasis_subscription *sub,
 struct stasis_topic *topic, struct stasis_message *message)
{
 if (stasis_subscription_final_message(sub, message)) {
 /* subscription cleanu */
 }
 /* ..
}
static int load_module(void)
{
 router = stasis_message_router_create(
 stasis_caching_get_topic(ast_channel_topic_all_cached()));
 if (!router) {
 return AST_MODULE_LOAD_FAILURE;
 }
 stasis_message_router_add(router, stasis_cache_update_type(),
 updates, NULL);
 stasis_message_router_set_default(router, default_route, NULL);
 /* ..
 return AST_MODULE_LOAD_SUCCESS;
}

static int unload_module(void)
{
 /* .. */
 stasis_message_router_unsubscribe(router);
 router = NULL;
 return 0;
}

```

--------------------------------

### Asterisk Dialplan: Replace Bytes at Offset (Byte Mode)

Source: https://docs.asterisk.org/Asterisk_18_Documentation/API_Documentation/Dialplan_Functions/FILE

Replaces a specified number of bytes in a file starting from a given offset. This example replaces 1 byte starting at offset 0 with 'bar'.

```asterisk
same => n,Set(FILE(/tmp/test.txt,0,1)=bar)
```

--------------------------------

### Build and Install Asterisk

Source: https://docs.asterisk.org/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/cURL

Compiles and installs Asterisk after modules have been selected, including the cURL related modules, onto the system.

```bash
make && make install
```

--------------------------------

### Asterisk Dialplan Hint Application Example

Source: https://docs.asterisk.org/Asterisk_16_Documentation/API_Documentation/Module_Configuration/res_pjsip_config_wizard

Demonstrates how to specify valid extensions.conf application expressions for dialplan hints. This includes examples of using Dial() and Gosub() with variables.

```asterisk-dialplan
Dial(${HINT})
Gosub(stdexten,${EXTEN},1(${HINT}))
```

--------------------------------

### List All Options in Menuselect

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Using-Menuselect-to-Select-Asterisk-Options

Lists all available configuration options within the Asterisk Menuselect build system. This command is useful for understanding the full scope of customizable features before building Asterisk.

```bash
[root@server asterisk-14.X.Y]# menuselect/menuselect --list-options

```

--------------------------------

### Compile Asterisk

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Building-and-Installing-Asterisk

Compiles the Asterisk source code. This is a prerequisite for installation and may take several minutes to complete.

```bash
[root@server asterisk-14.X.Y]# make

```

--------------------------------

### Asterisk Error: No SRTP Module Loaded

Source: https://docs.asterisk.org/Deployment/Secure-Calling/Secure-Calling-Tutorial

This error occurs when SRTP is enabled in the Asterisk configuration but the necessary SRTP module is not loaded. It indicates that the libsrtp library is either not installed or Asterisk was not recompiled after its installation.

```log
[Jan 24 09:29:16] ERROR[10167]: chan_sip.c:27987 setup_srtp: No SRTP module loaded, can\'t setup SRTP session.

```

--------------------------------

### Serving Asterisk Documentation Locally

Source: https://docs.asterisk.org/Contributing-to-the-Documentation

This command starts the built-in MkDocs web server to allow local viewing of the rendered documentation. The website is typically located in the './temp/site' directory. Options can be passed via SERVE_OPTS.

```bash
$ make serve

```

--------------------------------

### Install ODBC and MariaDB Connector (Bash)

Source: https://docs.asterisk.org/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/ODBC/Getting-Asterisk-Connected-to-MySQL-via-ODBC

Installs the necessary unixODBC packages and the MariaDB ODBC connector using yum. Development packages for unixODBC are included as they are needed for building Asterisk ODBC modules.

```bash
# sudo yum install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel
```

```bash
# sudo yum install mariadb-connector-odbc
```

--------------------------------

### Get Asterisk Build Machine Type (Dialplan)

Source: https://docs.asterisk.org/Asterisk_16_Documentation/API_Documentation/Dialplan_Functions/VERSION

This snippet shows how to get the machine type on which Asterisk was configured using VERSION(BUILD_MACHINE). The output is stored in the 'buildmachine' channel variable. This requires that build information is present in the Asterisk installation.

```asterisk-dialplan
same => n,Set(buildmachine=${VERSION(BUILD_MACHINE)})

```

--------------------------------

### Combine Asterisk Debug and Verbose Levels on Startup

Source: https://docs.asterisk.org/Operation/Logging/Basic-Logging-Start-up-Options

Allows simultaneous configuration of both debug and verbose logging levels for Asterisk at startup. The combined arguments specify the desired intensity for each logging type. This requires the Asterisk executable to be present.

```bash
asterisk -dddvvv
```

--------------------------------

### Start Asterisk with Core Dump Option

Source: https://docs.asterisk.org/Development/Debugging/Getting-a-Backtrace

This command demonstrates how to start the Asterisk process with the '-g' option, which enables the generation of a core file if Asterisk crashes. This is a prerequisite for obtaining a backtrace. The safe_asterisk script automatically includes this option.

```bash
asterisk -g
```

--------------------------------

### ast_coredumper Configuration File Example

Source: https://docs.asterisk.org/Development/Debugging/Getting-a-Backtrace

An example of the ast_debug_tools.conf file used by Asterisk's debug tools, including ast_coredumper. This file specifies core dumps and search patterns. It must adhere to bash semantics as it is sourced by bash.

```bash
# This file is used by the Asterisk debug tools.
# Unlike other Asterisk config files, this one is
# "sourced" by bash and must adhere to bash semantics.
#
# A list of coredumps and/or coredump search patterns.
# Bash extended globs are enabled and any resulting files
# that aren't actually coredumps are silently ignored
# so you can be liberal with the globs.
#
# If your patterns contains spaces be sure to only quote
```

--------------------------------

### Start Asterisk with Console and Verbosity

Source: https://docs.asterisk.org/Getting-Started/Hello-World

Starts the Asterisk service with an interactive control console and sets the verbosity level to 5. This is useful for real-time monitoring and debugging during startup.

```bash
asterisk -cvvvvv
```

--------------------------------

### Install LibPRI Library (Shell)

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Prerequisites/Building-and-Installing-LibPRI

Installs the compiled LibPRI library to the appropriate system library directory. This command should be executed after a successful compilation.

```shell
[root@server libpri-1.X.Y]# make install
```

--------------------------------

### SIP INVITE Request Example for Endpoint Identification

Source: https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Asterisk-PJSIP-Troubleshooting-Guide

An example of an incoming SIP INVITE request. Asterisk uses the 'From' header's user part (e.g., 'eggowaffles') to identify the endpoint. If no matching endpoint is found, the call is rejected.

```sip
<--- Received SIP request (541 bytes) from UDP:127.0.0.1:5061 --->
INVITE sip:service@127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-27600-1-0
From: breakfast <sip:eggowaffles@127.0.0.1:5061>;tag=27600SIPpTag001
To: sut <sip:service@127.0.0.1>
Call-ID: 1-27600@127.0.0.1
CSeq: 1 INVITE
Contact: sip:eggowaffles@127.0.0.1:5061
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 163

v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20


```

--------------------------------

### Asterisk Module File Example

Source: https://docs.asterisk.org/Fundamentals/Asterisk-Architecture/Asterisk-Architecture-The-Big-Picture

Demonstrates the typical file extension for Asterisk modules and provides an example of a core module's naming convention. These modules are loaded by the Asterisk core to provide extended functionality.

```bash
# Example of a module file
*.so

# Example of a core module file
pbx_xxxxx.so
```

--------------------------------

### SIP Geolocation Header Example (By Reference)

Source: https://docs.asterisk.org/Deployment/Geolocation

This example demonstrates how a SIP Geolocation header can be used to indicate a location description document by reference. The recipient is expected to retrieve the XML document from the provided URI. Asterisk itself does not retrieve information from these URIs.

```sip
Geolocation: <<https://geoloc.example.com?location=some_location_reference>>
```

--------------------------------

### Asterisk Dialplan: Goto() Syntax Example

Source: https://docs.asterisk.org/Latest_API/API_Documentation/Dialplan_Applications/Goto

Demonstrates the basic syntax for the Goto() dialplan application. It shows how to specify context, extension, and priority, or just priority if the context and extension are the current ones.  Arguments must be provided in the correct order.

```asterisk-dialplan
Goto(context,extension,priority)
Goto(extension,priority)
Goto(priority)
```

--------------------------------

### Shell Command: Backup and Edit extensions.conf

Source: https://docs.asterisk.org/Getting-Started/Hello-World

Backs up the existing extensions.conf file by renaming it to extensions.sample and then opens a new, blank extensions.conf file for editing using the Vim editor. This prepares the dialplan configuration.

```bash
mv extensions.conf extensions.sample
vim extensions.conf

```

--------------------------------

### PJSIP Phone Provisioning Configuration Example (pjsip.conf)

Source: https://docs.asterisk.org/Asterisk_18_Documentation/API_Documentation/Module_Configuration/res_pjsip_phoneprov_provider

This snippet shows an example configuration for a user within the pjsip.conf file, demonstrating how to define phone provisioning parameters for a PJSIP endpoint. It includes required fields like MAC and PROFILE, as well as optional variables that are passed through to the phone provisioning system.

```ini
[1000]

type = phoneprovr

endpoint = ep1000

MAC = deadbeef4dad
PROFILE = grandstream2

LINEKEYS = 2
LINE = 1

OTHERVAR = othervalue
```

--------------------------------

### AEAP Get API

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-External-Application-Protocol-AEAP

Retrieves attributes, settings, and other data from a remote application after a successful setup.

```APIDOC
## POST /websites/asterisk/get

### Description
Retrieves specified attributes and settings from the remote application.

### Method
POST

### Endpoint
/websites/asterisk/get

### Parameters
#### Request Body
- **request** (string) - Required - Must be 'get'.
- **id** (string) - Required - A unique identifier for the request.
- **params** (array) - Required - A list of parameter names to retrieve.
  - (string) - The name of the parameter to retrieve.

### Request Example
```json
{
  "request": "get",
  "id": "f066e049-d4ef-4093-a05b-0effef1bc077",
  "params": [
    "language"
  ]
}
```

### Response
#### Success Response (200)
- **response** (string) - Will be 'get'.
- **id** (string) - The unique ID from the request.
- **params** (object) - Key-value pairs of requested parameters and their values.
  - **name** (string) - The name of the parameter.
  - **value** (any) - The value of the parameter (type depends on the parameter).

#### Response Example
```json
{
  "response": "get",
  "id": "f066e049-d4ef-4093-a05b-0effef1bc077",
  "params": {
    "language": "en"
  }
}
```
```

--------------------------------

### Asterisk Dialplan for Call Recording with MixMonitor

Source: https://docs.asterisk.org/Fundamentals/Key-Concepts/Frames/Audiohooks

This Asterisk dialplan snippet demonstrates the initial setup for recording calls using MixMonitor. It answers an incoming call, starts recording, and places the call into a queue. This basic setup will stop recording if an attended transfer occurs.

```asterisk
exten => 1,1,Answer()
exten => 1,n,MixMonitor(training_recording.wav)
exten => 1,n,Queue(techsupport)
```

--------------------------------

### PJSIP Phone Configuration Wizard Example

Source: https://docs.asterisk.org/Asterisk_16_Documentation/API_Documentation/Module_Configuration/res_pjsip_config_wizard

This configuration snippet demonstrates how to use the PJSIP Configuration Wizard to set up objects necessary for a phone to obtain provisioning information, register, and handle calls. It outlines parameters for authentication, registration, and hints.

```asterisk-config
[myphone]
type = wizard
sends_auth = no
accepts_auth = yes
sends_registrations = no
accepts_registrations = yes
has_phoneprov = yes
transport = ipv4
has_hint = yes
hint_exten = 1000
inbound_auth/username = testname
inbound_auth/password = test password
endpoint/allow = ulaw
endpoint/context = default
phoneprov/MAC = 001122aa4455
phoneprov/PROFILE = profile1

```

--------------------------------

### Example: Removing and Replacing Hangup Handlers - Asterisk Dialplan

Source: https://docs.asterisk.org/Configuration/Dialplan/Subroutines/Hangup-Handlers

This example shows how to remove a hangup handler (hdlr1) and replace another (hdlr2 with hdlr4) using `CHANNEL(hangup_handler_pop)`. It starts by adding three handlers and then demonstrates the removal and replacement process, including the resulting handler definitions.

```asterisk-dialplan
; Some dialplan extension
same => n,Set(CHANNEL(hangup_handler_push)=hdlr3,s,1(args));
same => n,Set(CHANNEL(hangup_handler_push)=hdlr2,s,1(args));
same => n,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1(args));
; Remove hdlr1
same => n,Set(CHANNEL(hangup_handler_pop)=)
; Replace hdlr2 with hdlr4
same => n,Set(CHANNEL(hangup_handler_pop)=hdlr4,s,1(args));

; Continuing in some dialplan extension

[hdlr1]

exten => s,1,Verbose(0, Not Executed)
same => n,Return()

[hdlr2]

exten => s,1,Verbose(0, Not Executed)
same => n,Return()

[hdlr3]

exten => s,1,Verbose(0, Executed Second)
same => n,Return()

[hdlr4]

exten => s,1,Verbose(0, Executed First)
same => n,Return()
```

--------------------------------

### Building Asterisk Documentation

Source: https://docs.asterisk.org/Contributing-to-the-Documentation

This command initiates the build process for the static Asterisk documentation. It assumes the environment is set up and dependencies are installed.

```bash
$ make

```

--------------------------------

### Handling Channel Events (StasisEnd)

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/Getting-Started-with-ARI

This section covers the `StasisEnd` event, which is received when a channel leaves the Stasis application, typically when the call is hung up.

```APIDOC
## Handling Channel Events (StasisEnd)

### Description
The `StasisEnd` event is generated by Asterisk when a channel terminates its engagement with a Stasis application. This event signals the end of the call leg within the application's context.

### Event Type
`StasisEnd`

### Event Data Structure
- **application** (string) - The name of the Stasis application.
- **type** (string) - Always `StasisEnd` for this event.
- **timestamp** (string) - The time the event occurred.
- **channel** (object) - Details about the channel that is leaving Stasis.
  - **id** (string) - The unique identifier of the channel.
  - **state** (string) - The final state of the channel (e.g., `Up`).
  - **name** (string) - The name of the channel (e.g., `PJSIP/1000-00000001`).
  - **caller** (object) - Information about the caller.
  - **connected** (object) - Information about the connected party.
  - **accountcode** (string) - The account code associated with the channel.
  - **dialplan** (object) - Dialplan context, extension, and priority.
  - **creationtime** (string) - The time the channel was created.

### Event Example
```json
{
 "application":"hello-world",
 "type":"StasisEnd",
 "timestamp":"2014-05-20T13:30:01.852-0500",
 "channel":{
 "id":"1400609726.3",
 "state":"Up",
 "name":"PJSIP/1000-00000001",
 "caller":{
 "name":"",
 "number":""},
 "connected":{
 "name":"",
 "number":""},
 "accountcode":"",
 "dialplan":{
 "context":"default",
 "exten":"1000",
 "priority":3},
 "creationtime":"2014-05-20T13:15:26.628-0500"}
 }
```
```

--------------------------------

### Navigate to LibPRI Source Directory (Shell)

Source: https://docs.asterisk.org/Getting-Started/Installing-Asterisk/Installing-Asterisk-From-Source/Prerequisites/Building-and-Installing-LibPRI

Changes the current working directory to the extracted LibPRI source code directory. This is the first step before compilation.

```shell
[root@server src]# cd libpri-1.X.Y
```

--------------------------------

### Get IAX Peer Preferred Codec by Index

Source: https://docs.asterisk.org/Certified-Asterisk_20.7_Documentation/API_Documentation/Dialplan_Functions/IAXPEER

Retrieves a specific preferred codec for an IAX2 peer based on its index (starting from '0').

```asterisk dialplan
same => n, Set(PREFERRED_CODEC=${IAXPEER(peer10,codec[0])})
```

--------------------------------

### Recompile Asterisk with ODBC Modules

Source: https://docs.asterisk.org/Configuration/Interfaces/Back-end-Database-and-Realtime-Connectivity/ODBC/Getting-Asterisk-Connected-to-MySQL-via-ODBC

These commands are used to recompile Asterisk after installing necessary ODBC and MySQL components. It ensures that Asterisk includes the required ODBC modules for database integration. The process involves navigating to the source directory, running configure, and then performing make and make install.

```bash
# cd ~/asterisk-source/
# ./configure
# make && make install
```

--------------------------------

### Get IAX Peer Specific Codec (Asterisk Dialplan)

Source: https://docs.asterisk.org/Certified-Asterisk_18.9_Documentation/API_Documentation/Dialplan_Functions/IAXPEER

Retrieves the preferred codec at a specific index for an IAX2 peer. Codecs are indexed starting from '0'.

```asterisk
IAXPEER(peername,codec[x])
```

--------------------------------

### Install and Configure Radiusclient Library

Source: https://docs.asterisk.org/Configuration/Reporting/Channel-Event-Logging-CEL/CEL-Configuration-Examples/RADIUS-CEL-Backend

Steps to download, compile, and install the Radiusclient-ng library, along with essential configuration for authentication and accounting servers, and dictionary file inclusion.

```bash
root@localhost:/usr/local/src# tar xvfz radiusclient-ng-0.5.2.tar.gz 
root@localhost:/usr/local/src# cd radiusclient-ng-0.5.2
root@localhost:/usr/local/src/radiusclient-ng-0.5.2#./configure 
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make 
root@localhost:/usr/local/src/radiusclient-ng-0.5.2# make install

# Configuration examples:
# authserver localhost
# acctserver localhost

# Add to /usr/local/etc/radiusclient-ng/dictionary:
# $INCLUDE /path/to/dictionary.digium
```

--------------------------------

### Asterisk chan_sip Template Example: Basic Options and Inheritance

Source: https://docs.asterisk.org/Fundamentals/Asterisk-Configuration/Asterisk-Configuration-Files/Templates/Using-Templates

A comprehensive example from the chan_sip configuration file, showcasing template definitions for basic options, natted phones, public phones, and codec preferences, along with inheritance.

```asterisk-config
[basic-options](!) ; a template
 dtmfmode=rfc2833
 context=from-office
 type=friend

[natted-phone](!,basic-options) ; another template inheriting basic-options
 nat=yes
 directmedia=no
 host=dynamic

[public-phone](!,basic-options) ; another template inheriting basic-options
 nat=no
 directmedia=yes

[my-codecs](!) ; a template for my preferred codecs
 disallow=all
 allow=ilbc
 allow=g729
 allow=gsm
 allow=g723
 allow=ulaw

[ulaw-phone](!) ; and another one for ulaw-only
 disallow=all
 allow=ulaw

; and finally instantiate a few phones
;
; [2133](natted-phone,my-codecs)
; secret = peekaboo
; [2134](natted-phone,ulaw-phone)
; secret = not_very_secret
; [2136](public-phone,ulaw-phone)
; secret = not_very_secret_either
```

--------------------------------

### Playback Media on a Channel

Source: https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/Getting-Started-with-ARI

This section demonstrates how to use the Asterisk ARI REST API to play a sound file to a specific channel that has entered the Stasis application.

```APIDOC
## Playback Media on a Channel

### Description
Initiates playback of a media file (e.g., a sound) to a specified channel. This is typically done after receiving a `StasisStart` event.

### Method
POST

### Endpoint
`/ari/channels/{channelId}/play`

### Parameters
#### Path Parameters
- **channelId** (string) - Required - The unique identifier of the channel to play media to (e.g., `1400609726.3`).

#### Query Parameters
- **media** (string) - Required - The media to play, specified as a URI (e.g., `sound:hello-world`).

### Request Example
```bash
curl -v -u asterisk:asterisk -X POST "http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world"
```

### Response
#### Success Response (201 Created)
Returns a JSON object representing the created Playback resource.

- **id** (string) - The unique identifier for this playback operation.
- **media_uri** (string) - The media that was queued for playback.
- **target_uri** (string) - The URI of the channel targeted for playback.
- **language** (string) - The language code for the media playback.
- **state** (string) - The current state of the playback (e.g., `queued`).

#### Response Example
```json
{
 "id":"9567ea46-440f-41be-a044-6ecc8100730a",
 "media_uri":"sound:hello-world",
 "target_uri":"channel:1400609726.3",
 "language":"en",
 "state":"queued"}
```

### Events
- **PlaybackStarted**: Sent when playback begins.
- **PlaybackFinished**: Sent when playback completes.

#### Event Example (PlaybackStarted)
```json
{
 "application":"hello-world",
 "type":"PlaybackStarted",
 "playback":{
 "id":"9567ea46-440f-41be-a044-6ecc8100730a",
 "media_uri":"sound:hello-world",
 "target_uri":"channel:1400609726.3",
 "language":"en",
 "state":"playing"}
 }
```
```