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VoIP & Audio Conferencing Projects

This repository contains two VoIP/Asterisk-based projects demonstrating basic calling and multi-party audio conferencing.


Project 1: Basic VoIP Call Between Zoiper and JIST

Description:
This project demonstrates a simple VoIP call setup using Asterisk PBX. A user can make a call from Zoiper softphone to a JIST client.

Installation & Setup:

  1. Install Asterisk (Linux):
    sudo apt update
    sudo apt install -y asterisk
    sudo systemctl start asterisk
    sudo systemctl enable asterisk

Install Zoiper Softphone:

Download Zoiper from https://www.zoiper.com

Install and configure with the SIP account created in Asterisk.

Install JIST Client:

Follow official JIST installation instructions.

Configure SIP account with Asterisk credentials.

Make a Call:

Open Zoiper and register using your SIP account.

Dial the SIP address or extension of the JIST client.

Ensure the call connects successfully.

Files & Configuration:

Asterisk configuration files (sip.conf, extensions.conf) for this scenario are included in the project folder.

Project 2: Audio Conferencing with Conference ID 600

Description: This project demonstrates a multi-party audio conference using Asterisk ConfBridge:

Conference ID 600: Normal users join here.

Conference ID 601: Admin joins here.

Installation & Setup:

Install Asterisk: (Same steps as Project 1)

Configure Conference in Asterisk:

Edit confbridge.conf to define rooms 600 (normal users) and 601 (admin).

Ensure proper roles and privileges are set for each room.

Join the Conference:

Normal user dials extension 600.

Admin dials extension 601.

Verify communication between users and admin.

Additional Information:

Detailed configurations, dialplans, and scripts are available in the conf_with_ID_600 folder.

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Next-gen WebRTC Contact Center with Asterisk, FreeSWITCH & Kurento

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