PBX System Commissioning and Integration with ASTERISK, PSTN and GSM

PBX System

Asterisk is the software-based PBX framework, it is an open-source trade worker. It was created by Mark Spencer in 1999. A reference bullet can transform a standard PC into a correspondence worker. It is utilized by each kind of business whether it is large or little, in any part like client communities’ instructive organizations, government authorities around the world.


These days VoIP communication framework is quickly turning into the standard for business correspondences around the world. As enterprises are relocating from a traditional phone system to the cost-saving and rich-highlighted VoIP environment as it is more advanced and very cost-effective as compared to the legacy system.

But to switch from Legacy PBX to a new VOIP PBX system can be costly as we have to change all the infrastructure, hardware and equipment i.e. (Traditional/Analog phones to IP Phones), etc. and all the legacy PBX equipment and hardware will be removed which will be the waste of resources and cost. By supporting a blend of conventional and VoIP communication administrations, Asterisk permits to fabricate of new phone frameworks productively, or bit by bit relocate existing frameworks to new advancements.

 A few destinations are utilizing Asterisk servers to supplant maturing exclusive PBX’s, others to give extra highlights, (for example, voice message or telephone menus), or to cut expenses via conveying long-distance calls over the Internet.

Why AsteriskNOW?

We have chosen AsteriskNOW because it is open source, its source code is available on the web, it is free of cost and easily available to download from the web, it is intended to permit diverse sorts of IP communication equipment, middleware, and programming to interface with each other reliably. AsteriskNOW makes it simple to make custom communication arrangements via consequently introducing the fundamental prerequisites like DAHDI and libpri. It’s an entire Linux dispersion with Asterisk, the DAHDI driver structure, and, the FreePBX managerial GUI. A significant part of the unpredictability of Asterisk and Linux is taken care of by the installer, the yum bundle administration utility, and the managerial GUI. With the help of GUI, it is easily configurable in a user-friendly

environment, it is not required to type long codes in the command-line interface (CLI)

because all the configuration can be done at the graphical user interface (GUI). It has rich features like voicemail, voice to Email, conference calling, call directing, ring groups, IVR menus, music on hold, and a user control panel where each user can access to their records of calling and voicemails from their desktop on a web browser by simply logging into the ID and password provided by the admin.

What is FreePBX?

FreePBX is the graphical user interface for managing and administrating the Asterisk server. It is an open-source web-based GUI that is authorized under General public license (GPL). It can be implemented physically or as a feature of the distro which is pre-arranged, which incorporates the Asterisk OS.

Traditional PBX vs Asterisk (IP PBX):

Asterisk PBX (IP PBX) is very cost-effective as compared to the Traditional PBX systems which require hardware to be deployed and a high infrastructure cost and maintenance cost, also a technician is required to manage and resolve any problem on a hardware basis, it is difficult to manage because most of them are configured with complex CLI commands which are the propriety of different service providers/vendors while IP PBX can be deployed on a simple PC with a completely software-based solution, and extensions can be configured on softphones on PC which reduces the hardware cost. GUI makes it easier for administrators to manage IP PBX systems also each user can be provided login IDs for their own preferences. Also, Asterisk is rich in features as compared to the Analog PBX’s. Asterisk provides features like voicemail, call recording, IVR menu, direct inward system access (DISA), parking, and paging which requires separate hardware in the case of Analog PBX systems.

On the off chance that IP Telephony has been deployed in various branches (in various areas) of a similar organization, it is conceivable to utilize MPLS Networks/Internet Leased Lines (With Unlimited Usage designs) to transmit voice brings over the WAN IP Network. Along these lines, the between branch calls would not bring about extra expenses.

The objective of our project is “PBX system commissioning and integration with

Asterisk, PSTN and GSM”. The main task of our project is to integrate Asterisk PBX

system with Analog PBX system for which we are using an Analog telephone adapter

(ATA device) as a gateway between IP PBX and Analog PBX. To achieve this task, we are installing the Asterisk server on two different sites and connect them with the help of SIP trunking. Then we are using the GSM link for fault tolerance, if the internet goes down the calls will be received and made with GSM service. Further, we are configuring different features of the Asterisk PBX system to explore more features.

Tools and Components:

the tools and components used in this project and complete Network topology of the project with its main features:

The following tools and components are used in this project.

  • Software:
  • VMware
  • Zoiper and X-lite Softphones
  • Hardware:
  • Analog PBX system
  • Operator Console
  • Analog Phones
  • ATA Gateway
  • IP Phones
  • GSM Modem
  • 4-port Switches
  • TP-link routers


The following are the features of this project.

  • Customized Dial plan
  • Managing Extensions
  • Ring Groups
  • Time Conditions
  • Time Groups
  • Voicemail
  • Voice to Email
  • Call Recording
  • Call forwarding
  • Call Hold
  • Automated Attendant
  • Interactive Voice Response (IVR)
  • Failover via GSM link
  • Scalability
  • Latency
  • Cost
  • Call Detail Records
  • Call monitoring
  • Complexity
  • Easy Integration


The world is adopting it and switching from the Legacy system to new VoIP technology PBX’s. 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 workplaces around the world.

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