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Asterisk is the world’s leading open source telephony engine and tool kit. Offering flexibility unheard of in the world of proprietary communications, Asterisk empowers developers and integrators to create advanced communication solutions...for free. Asterisk® is released as open source under the GNU General Public License (GPL), and it is available for download free of charge. Asterisk® is the most popular open source software available, with the Asterisk Community being the top influencer in VoIP.
Asterisk as a gateway
It can also be built out as the heart of a media gateway, bridging the legacy PSTN to the expanding world of IP telephony. Asterisk’s modular architecture allows it to convert between a wide range ofcommunications protocols and media codecs.
Asterisk as a feature/media server
Need an IVR? Asterisk’s got you covered. How about a conference bridge? Yep. It’s in there. Whatabout an automated attendant? Asterisk does that too. How about a replacement for your aging legacy voicemail system? Can do. Unified messaging? No problem. Need a telephony interface for your web site? Ok.
Supported hardware
Asterisk® needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk® supports a number of hardware devices, most notably all of the hardware manufactured by Digium®, the creator of Asterisk®.
Supported protocols
Asterisk® supports a wide range of protocols for the handling and transmission of voice over traditional telephony interfaces including H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP).
Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol Asterisk® merges voice and data traffic seamlessly across disparate networks. The use of Packet Voice allows Asterisk® to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information.
Asterisk® provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.
DIGIUM DIGITAL CARDS
Digium's super-reliable digital line cards connect Asterisk-based communication systems to T1, E1, J1 and ISDN-BRI interfaces

DIGIUM GATEWAY
Digium has engineered gateways for easy configuration and reliable operation. The gateways draw on Digium's wealth of experience building interconnect hardware. With Asterisk under the hood and a simple, intuitive web interface for setup, the gateways set a new standard for PSTN integration.

YEALINK IP PHONES
The Yealink series is intended for discerning users with very high expectations of IP phones. It has been designed specifically for people who take great satisfaction in experiencing excellence actually being delivered. Revolutionary in appearance and advanced technical design, it is not only pleasurable and practical to use, with performance
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DIGIUM G100 VoIP GATEWAY
Built on a powerful combination of the Asterisk Open Source communications engine and a state-of- the-art embedded platform, Digium VoIP Gateways provide the best value for connecting disparate topologies of traditional telephony (T1/E1/PRI) to IP (SIP)
The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI, which allows for easy navigation and effortless setup. VoIP gateways feature a power-saving embedded design with a highly efficient digital signal processor (DSP) handling all media-related operations.
The Digium G100 VoIP Gateway includes One (1) software-selectable T1/E1/PRI interface and supports up to 30 concurrent calls. With a robust feature set including the ability to configure calling rules for connecting many combinations of telephony providers (traditional and VoIP) and PBX’s (legacy and VoIP), failover routing to ensure calls won’t fail, licensing for the maximum number calls the appliance supports, SIP to SIP transcoding and a proven design of no moving parts to fit any environment, Digium Gateways are proven to support most applications.
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DIGIUM G200 VoIP GATEWAY
Built on a powerful combination of the Asterisk Open Source communications engine and a state-of- the-art embedded platform, Digium VoIP Gateways provide the best value for connecting disparate topologies of traditional telephony (T1/E1/PRI) to IP (SIP).
The gateway software is based on the Asterisk communications engine and is managed through Digium’s intuitive point-and-click GUI, which allows for easy navigation and effortless setup. VoIP gateways feature a power-saving embedded design with a highly efficient digital signal processor (DSP) handling all media-related operations.
The Digium G200 VoIP Gateway includes two (2) software-selectable T1/E1/PRI interfaces and supports up to 60 concurrent calls. With a robust feature set including the ability to configure calling rules for connecting many combinations of telephony providers (traditional and VoIP) and PBX’s (legacy and VoIP), failover routing to ensure calls won’t fail, licensing for the maximum number calls the appliance supports, SIP to SIP transcoding and a proven design of no moving parts to fit any environment, Digium Gateways are proven to support most applications.
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