Chatsterisk

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⚠️ Archived: This project is archived because I don't need it as a learning resource anymore, but a lot of parts are still good to reference, such as: Asterisk (setup, making video calls) and WebRTC (video chat app).

VoIP application ran using: Drash and Asterisk, to make and demonstrate voice calls using asterisk. Also acts as a training project for Drash, Sockets and Asterisk

I've decided to skip the IVR menu because I ddon't think we have a keypad for softphones int eh Telephone app? See here

Tools Used

• Drash - web server
• Wocket - web socket server
• deno bundlle - bundling client TS code
• Asterisk
• Rhum - Testing framework
• Dawn - browser testing framework (not yet added)
• Dami - ami client

Project Flow

1. Docker environment is started

2. server bundles client ts, and starts, handling any web requests

3. Asterisk starts, listening for any call requests

4. ami starts, listening on any requests to the socket

5. Asterisks AMI is configured to run (see manager.conf), on port 5038. An AMI is a part of Asterisk, and we can connect to it using telnet localhost 5038 to issue commands, then when connected, do Action: Login ^SHIFT ... (or telnet 0.0.0.0 5038 when inside the container)

Setup Phones

1. Register 2 phones, for demo-alice (pass = verysecretpassword as seen in sip.conf) and demo-bob (pass = anothersecretpassword). Domain is 0.0.0.0 (because docker). Username is also demo-alice (see sip.conf), or. it's thee extension eg 6001 or 6002
2. Make a call! We can see the dialplans in extensions.conf (note the context for our 2 phones are from-internal), so make a call to 6001 (to alice) or 6002 (to bob)

Directory Structure

• .docker - Holds docker-related files, such as dockerfiles, and asterisk configuration files

• .github - Holds our workflows

• src/socket - Our AMI socket server

• src/server - Our drash application that acts as the server for this project, to serve a client on the web

  • middleware - Direcotry to hold our custom middleware to pipe into Drash
  • public - Our client assets
  • .env - Our env file
  • app.ts - Our entrypoint file to start the server
  • denon.json - Similar to PM2, we use denon to start and watch our server (app.ts)
  • deps.ts - Holds our dependencies
  • response.ts - A unique way to override the Drash response so we can customise how the server responds to a client (NOT IN USE)

Deno Information

Best Practices / Consistencies

• Packages
  • For importing packages, a deps.ts file will hold all imports for the whole project, and all those will then be exported. Should a file require any package or module, they will use the deps.ts file to do so.
• Use mod.ts to export your module
• Use app.ts as your entry point file (eg web apps)
• Use deno fmt

Built-in Deno Utilities

See here

Tools Used

• Deno - Runtime environment
• Drash - HTTP Microframework
• Drash's Template Engine
• Asterisk - PBX System
• Denon - File watcher and server starter
• Paladin - Better secure our responses
• Docker

Resources

• Deno's Website
• Deno's GitHub
• Drash's Docs
• Asterisk Hello World
• Asterisk Docs

Help

• Resolving .ts extensions

  • As Typescript doesn't have a native way to resolve these due to how Deno is built, a different fix has been implemented. See here for some more information and the related commit here

The Set Up

Most of the steps were following using asterisks main documentation: https://wiki.asterisk.org/wiki/display/AST/Hello+World

Public Server

1. Install asterisk

   apt install asterisk

2. Setup the conf files and the step using the link above. It includes the sip/pjsip/extensions config files

3. Restart asterisk

   asterisk -rx "core restart now"

4. Start asterisk

   asterisk -cvvvvv

5. Connect using a softphone, where the domain is the IP of the server

6. Make a call to 100

Docker

1. Docker compose file content:

asterisk:
    container_name: asterisk_pbx
    image: hibou/asterisk:14
    ports:
      - "5060:5060/udp"
      - "5060:5060/tcp"
      - "16384-16394:16384-16394/udp"
      - "8088:8088"
      - "8089:8089"
    volumes:
      - "./.docker/config/asterisk/sip.conf:/etc/asterisk/sip.conf"
      - "./.docker/config/asterisk/extensions.conf:/etc/asterisk/extensions.conf"
      - "./.docker/config/asterisk/pjsip.conf:/etc/asterisk/pjsip.conf"
      - "./.docker/config/asterisk/rtp.conf:/etc/asterisk/rtp.conf"
    labels:
      - "traefik.enable=false"

2. Dockerfile content:

FROM centos:6

# Set up EPEL
RUN curl -L http://download.fedoraproject.org/pub/epel/6/x86_64/epel-release-6-8.noarch.rpm -o /tmp/epel-release-6-8.noarch.rpm && \
 rpm -ivh /tmp/epel-release-6-8.noarch.rpm && \
 rm -f /tmp/epel-release-6-8.noarch.rpm

# Update and install asterisk
RUN yum update -y && yum install -y asterisk

# Set config as a volume
VOLUME /etc/asterisk

# And when the container is started, run asterisk
ENTRYPOINT [ "/usr/sbin/asterisk", "-fcvvvv" ]

3. Setup the config files: sip/pjsip/extnesions/rtp. Follow the link above on the content needed, and for rtp.conf, use:

rtpstart=16384
rtpend=16394

4 Start docker

    docker-compose up

5. Connect using a softphone, where the domain is 0.0.0.0 for the container