asterisk/README.md

# asterisk

## Installing Dependencies
Before installing Asterisk, you need to update your package lists and install the necessary dependencies. Open your terminal and execute the following commands:

`sudo apt update`
`sudo apt upgrade`
`sudo apt install -y wget build-essential subversion`

## Downloading and Installing Asterisk
With the dependencies in place, you can now download and compile Asterisk:

`cd /usr/src`
`sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz`
`sudo tar xvf asterisk-18-current.tar.gz`

Change into the extracted directory and install additional required packages:

`cd asterisk-18*`
`sudo contrib/scripts/install_prereq install`

Now, proceed to compile and install Asterisk:

`sudo ./configure`
`sudo make menuselect`
`sudo make`
`sudo make install`
`sudo make samples`
`sudo make config`
`sudo ldconfig`

With Asterisk installed, you can now set up its user and group permissions:

`sudo adduser --system --group --no-create-home asterisk`
`sudo chown -R asterisk:asterisk /etc/asterisk`
`sudo chown -R asterisk:asterisk /var/{lib,log,spool}/asterisk`
`sudo chown -R asterisk:asterisk /usr/lib/asterisk`

## Configuring Asterisk
Editing the main configuration file is the next step. Open it using your favorite text editor:

`sudo vim /etc/asterisk/asterisk.conf`

Make the necessary changes according to your setup, then start and enable Asterisk at boot:

`sudo systemctl start asterisk`
`sudo systemctl enable asterisk`

Ensure that Asterisk is running without issues:

`sudo asterisk -vvvr`

## Setting Up PJSIP Accounts
To handle VoIP calls, you need to set up PJSIP accounts. Edit the pjsip.conf and extensions.conf files:

`sudo vim /etc/asterisk/pjsip.conf`
`sudo vim /etc/asterisk/extensions.conf`

Add your SIP users in pjsip.conf and dial plan in extensions.conf.

## Google TTS script for Asterisk

You'll need to install the following dependencies:

`sudo apt install perl libwww-perl liblwp-protocol-https-perl sox mpg123`

Then download the script from the Google TTS repository and place it in the appropriate directory:

https://github.com/zaf/asterisk-googletts

## Voice-rt

Asterisk VoiceBot Realtime is a Node.js application that utilizes Asterisk for real-time audio transmission and processing. This application leverages Google Cloud Speech-to-Text, Google Cloud Text-to-Speech, and OpenAI for transcription, speech synthesis, and intelligent response generation.

Asterisk VoiceBot Realtime connects to a TCP server and handles audio data packets sent by Asterisk through the AudioSocket module. Upon receiving audio data, the application transcribes it using Google Cloud Speech-to-Text, sends the transcription to OpenAI for generating a response, and finally synthesizes the response into audio using Google Cloud Text-to-Speech. The generated audio is then sent back to the client through the socket.

https://hub.docker.com/r/gcareri/asterisk-voicebot-rt