When Metro Trains Melbourne (MTM), a mass transit operator in Melbourne, Australia, decided to replace their legacy private branch exchange (PBX) with a VoIP telecom solution, Asterisk was the only option. According to Principle Software Engineer, Marcus Morrison, MTM considers Asterisk the premiere open source IP PBX solution.
Sponsored by Digium, Asterisk is a free and open source telecommunications framework used to create customized telecom applications by turning an ordinary computer into a communications server. Asterisk has become the world’s most widely used telecommunications platform for powering IP PBX systems, VoIP gateways, and conference servers. It provides ample flexibility for experienced telecom engineers to tailor applications for small and large businesses, call centers, and even government entities.
Prior to deployment, the MTM engineers had plenty of experience with Linux, telephony systems (including Asterisk), and open source technologies. With a high degree of confidence, they agreed that Asterisk could be customized to meet the needs of their business.
Transitioning from a Legacy PBX to VoIP
MTM’s network consists of over 3,000 employees and interconnects more than 40 three-car train sets with 215 railway stations along 520 miles of track across Australia’s capital city. The trains travel over 18 million miles, providing more than 228 million customer boardings each year. The trains offer over 14,000 services each week, and carry over 415,000 passengers each weekday.
MTM has over 900 telephony end-points across the network, many of which are auto-answer units designed to present public address information to customers at railway stations, while others are on-demand service information units that present service information when requested.
“While we knew we would realize cost savings with our transition to VoIP and Asterisk, our primary motivation for the change was to improve the quality of our communications infrastructure to better deliver accurate service information to our customers,” says Morrison.
Interfacing Asterisk with a Legacy PBX
The primary challenge was the interface to the existing PBX system. “As capacity had become an issue for MTM, we began a requirements-gathering exercise focused on our Customer Information System’s interactions with the legacy PBX system,” says Morrison. “We already have a high quality IP network interconnecting our stations, but we were experiencing an increasing number of line faults on the older PBX system and we needed a way to stage a migration from the legacy analog PBX to VoIP.”
MTM had endpoints at every railway station, yet it wasn’t feasible to make the migration to VoIP in what Morrison refers to as a ‘big bang’ fashion. Instead, the system needed to be capable of delivering digitized audio message to newly provisioned IP end-points and be able to route calls out to the existing PBX system for stations that did not have IP end-points installed yet.
“We did not hesitate in building our own customized Asterisk installation and found it to be quite straightforward,” says Morrison. “Due to the size of the railway, we knew Asterisk had good support for interfacing between IP endpoints and legacy PBX systems,” says Morrison. “We needed to procure phone sets and a number of Digium analog telephony adapters in order to bridge our Asterisk installation with the legacy PBX system.”
Digium analog cards utilize VoiceBus™ technology to maximize system compatibility and to prevent systems conflicts. They make it possible to connect analog phones and POTS (Plain Old Telephone System) lines with VoIP PBX hardware using Asterisk.
Implementing SIP with Asterisk
Ultimately, the new Asterisk installation delivers digitized audio to each station, along with MTM control centers and select maintenance facilities. “We are no longer capacity-bound regarding the number of lines that could be provisioned,” says Morrison. “We also have many more options available to us with regards to integrating various software packages with Asterisk. We transitioned our Customer Information System to deliver audio messages over SIP via Asterisk.”
Session Initiation Protocol (SIP) is a VoIP and signaling protocol for creating, modifying, and terminating Internet telephone calls, multimedia distribution, and multimedia conferences. It uses proxy servers to help route requests to the user’s current location. SIP is very flexible and runs on top of several different transport protocols. It allows for call or media transfer, call conferencing, call hold, and more. SIP plays a major role in the enablement of IP telephony and VoIP.
Costs Reduced by Estimated 60 Percent
“Asterisk allowed us to begin migration away from an aging PBX system,” says Morrison. “It also allows us to address significant capacity issues across our network and has increased the fidelity with which we deliver audio announcements. Furthermore, Asterisk has let us begin to explore integration with centralized emergency management and public address systems, along with digitally generated voice.”
Because the MTM engineering staff was already aware of Asterisk’s capabilities, designing and implementing the new telephony system was a smooth process. “It was surprisingly easy to implement our design and we all found it to be a rewarding experience,” he says.
Morrison says exact cost savings are hard to quantify, as MTM is still transitioning away from the old PBX installation. “Once the transition is complete and we are no longer paying line rental and maintenance for the legacy equipment, I believe we will realize a significant reduction in costs, estimated at around 60 percent.”