VOIP USING ASTERISK

Voip is the technology that converts your voice into a digital signal allow you to make a call directly from a computer,a voip phone or other data devices. VoIP (voice over IP) is the transmission of voice and multimedia content over Internet Protocol (IP) networks. VoIP historical

Project Title

VOIP USING ASTERISK

Project Area of Specialization

Internet of Things

Project Summary

Voip is the technology that converts your voice into a digital signal allow you to make a call directly from a computer,a voip phone or other data devices.

VoIP (voice over IP) is the transmission of voice and multimedia content over Internet Protocol (IP) networks. VoIP historically referred to using IP to connect private branch exchanges (PBXs), but the term is now used interchangeably with IP telephony.

Asterisk is technology and protocol agnostic, which means that you can connect it to the outside world using VoIP or traditional telephony technologies. It also means that you can use virtually any standards-based IP phone.

Project Objectives

This project provides an optimal solution for those who want to switch from analog to VoIP service. With this solution users can integrate their Legacy PBX system with VOIP with using their current PBX system and all legacy equipment i.e. (Analog Phones) etc. They will be able to keep their Traditional PBX and use much less expensive VOIP lines. It is used by independent ventures, considerable associations, call centers, transporters, and government workplace around the world.

• Establish VOIP infrastructure between two branches.
• Fault tolerance with redundant GSM link .

Project Implementation Method

After installing and basic setup of the Asterisk server, now we can configure the server according to our requirements, because Asterisk is an open source software based solution of PBX so we can design desired custom dial plans and rules. So now we are going to configure IP PBX system as per our requirements to achieve our objective.In order to communicate between all IP phones or Soft phones to each other SIP accounts are need to be created on IP PBX system and needs to be registered on the desired phone (IP phone or Soft phone). All the extension registration information is defined in the file SIP.CONF and the file location in Linux directory from root is /etc/asterisk/sip.conf.

Benefits of the Project

• Ubiquitous access to the asterisk PBX.
• Automated attendant and call monitoring.
• Easier to manage.
• Cost effective .
• Flexibility.
• Support multi VOIP protocols.
• After completion of project we will be able to do Asterisk Certification (dCAA, dCAP).

Technical Details of Final Deliverable

We integrated the Analog PBX system with Asterisk VoIP PBX system, both IP phones and analog phones can communicate end to end. Configured GSM gateway for the fallback support and are attached to Asterisk on local network. GSM over IP router enable to provide basic call-handling support for IP phones when they lose connection to the server in case of ISP connection is down. Setup an IVR menu like call centers, so when the call comes in the system IVR menu is played and guided the caller to follow the menu or dial their desired extension.

Final Deliverable of the Project

HW/SW integrated system

Type of Industry

Telecommunication

Technologies

Internet of Things (IoT)

Sustainable Development Goals

Industry, Innovation and Infrastructure

Required Resources

If you need this project, please contact me on contact@adikhanofficial.com