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.
Conference phones are meant for holding conferences between three and a
bigger amount of participants. This allows all participants listening and
speaking simultaneously. Usually the conference is begun by the participant
having “Conference Phone”. Other participants call to a special number for a
conference which connects them with “Conference Bridge” that joins all lines
together.
Dean Connect
Dean Connect Delivers High quality SIP termination, number porting, Dutch area codes, SIP trunks, T38 FAX support and hosted PBX in retail and white-label model.
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