Digital VoIP Gateways
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G800 Digital Gateways
Built on a powerful combination of the Asterisk open source communications engine and a state-of- the-art embedded platform, the G800 digital gateways provide the best value for connecting traditional telephony (T1/E1/PRI) to IP (SIP).
Sangoma’s gateway software is based on the Asterisk communications engine and is managed through an intuitive point-and-click GUI, which allows for easy navigation and effortless setup. The digital gateways feature a power-saving embedded design with a highly efficient digital signal processor (DSP) handling all media-related operations.
The G800 VoIP gateway includes eight software-selectable T1/E1/PRI interfaces and supports up to 240 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, G800 gateways are proven to support most applications.
Gateway Applications
SIP Trunking
Businesses with legacy phone systems can install a gateway to seamlessly connect to VoIP services or a SIP trunk and eliminate the PSTN connection.
PSTN Trunking
Businesses already IP at the core can install gateways to connect to the PSTN for failover.
Remote Connectivity
Connect legacy Phone Systems to remote sites over private VoIP links.
Virtual Connectivity
Connect virtualized systems to legacy TMD services.
Gateway Features
Simplified Deployment
All G800 gateways have a Quick Wizard to quickly and easily deploy, eliminating the need of IT admin and complicated setup process.
Built-In Local Survivability
In the event of a VoIP failure, the gateway will allow internal extensions to continue calling each other and automatically route to a failover PSTN connection if available.
Highly Interoperable
All G800 gateways support the standard VoIP/SIP protocol and have built-in transcoding so you can trust that your phone calls and faxes will work even in challenging environments.
Flexible Routing
Advanced flexible call routing with automatic failover and bypass routing.
Advanced Troubleshooting
Built-in call tracing and debugging utilities.
Field Upgradable
Firmware updates can be performed remotely through a web-based GUI.
Fax
Support for T.38 faxing.
Gateway Specifications
VoIP Interface
- SIP
- Fax support – up to G3 FAX, using T.38
- Modem support – up to V.90, using G.711
- VoIP channel support: 8
- Audio Codecs:
- G.711 (a-law/µ-law) (64 kbps)
- G.729a (8kbps)
- G.723.1 (5.3/6.4 kbps)
- G.726
Telephony Interface
- 8 T1/E1/PRI w/ RJ-45
LAN Interface
- 1 x RJ-45, GigaBit, full / half duplex
Models
Appliances
- 1G800F - Octal Span Digital T1/E1/PRI to VoIP Gateway Appliance, North American Power Supply
- 1G801F - Octal Span Digital T1/E1/PRI to VoIP Gateway Appliance, Europe Power Supply
- 1G802F - Octal Span Digital T1/E1/PRI to VoIP Gateway Appliance, United Kingdom Power Supply
- 1G803F - Octal Span Digital T1/E1/PRI to VoIP Gateway Appliance, Australia Power Supply
Brackets
- 3244-00042 - Digium Gateway Appliance Single Unit 19" Rack Mount Ear