Digium G100

Digium VoIP gateway was created on a blend of the Asterisk Open Source communications engine and state of the art embedded platform, G100 provides the best value for the Asterisk connectivity.

The G100 gateway's software has a base of the Asterisk communications engine and is controlled with the help of Digium’s intuitive point & click GUI interface, that ensures easy navigation and simple setup process. G100 has a power saving design with extremely efficient DSP taking care of media related operations.
The G100 Gateway comes with a T1/E1/PRI interface and is able to provide support for up to 30 simultaneous connections. It was build to support SIP-to-SIP, SIP-to-TDM and TDM-to-SIP application connections. When deployed in TDM-to-SIP environment the G100 gateway considerably decreases running costs by connecting a SIP trunking devices to business telephone system. While at SIP-to-TDM deployment it uses the VoIP gateway for connecting T1/E1/PRI services from legacy carriers with modern SIP communications system.

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Selective Call Routing

The system allows you to create a list of numbers, which will be redirected in case if the telephone, to which they call, is unavailable. For example, somebody calls you to the office during your absence and the number of the subscriber who is calling is in the list, accordingly this call will be redirected for instance to the phone of your car or some other indicated by you in the system. In case if at the moment of your absence you will be called from a number that is not in the list, the call will not be redirected but most likely will be sent to the voice mail. Thus, you will never miss a call from partners, who are important to you.

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