Asterisk company Digium has released a pair of new VoIP gateways designed to simplify the process of deploying converged media networks. As with all Digium products, the G100 and G200 are built on the Asterisk open source communications engine and are managed through Digium’s point-and-click GUI interface.

The gateways support both TDM-to-SIP deployments – where the gateway connects legacy business phone systems with dynamic SIP trunking services – and SIP-to-TDM applications, according to Digium. The latter is often used to connect a modern SIP communications system with T1/E1/PRI service from legacy carriers.

“We’ve been a leader in the telephony gateway business indirectly for the past 10 years,” said Mark Amick, Digium’s director of product management. “We’ve sold over five million telephony ports in the form of interface cards which are frequently used to build gateways from standard PC hardware. We’re pleased to introduce another connectivity option for our customers with this new family of standalone gateways.”

The two new solutions will enable Asterisk users to migrate applications to virtualized, hosted or cloud environments by converting the media and signaling from TDM to SIP on a dedicated external device. This will help eliminate one of the most paramount challenges associated with running Asterisk applications in virtualized environments.

The single-span G100 sports a single software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls. The G200 model doubles the capacity of the G100, providing two T1/E1/PRI interfaces and support for up to 60 concurrent calls.

Each gateway features a power-saving embedded design, a highly efficient digital signal processor (DSP) and a user interface that "outperforms the dated designs in the market today," the company said. The G100 and the G200 also provide integrated echo cancellation, a small footprint (1U, half-width, half-depth) and no failure-prone moving parts.