IP Analog Gateways Grandstream GXW4108 Analog FXO Gateway

VoIP-FXO analog gateway to support up to 8 PSTN lines
Video surveillance port to enhance premise security
Dual 10/100 Mpbs ports (RJ45) with integrated router
Multiple SIP profiles, max of 3 profiles per system
1 year warranty against hardware defects
14 day money back guarantee
30 day email support for technical installation issues

Product Overview

The Grandstream GXW4108 FXO IP Analog Gateway offers the small enterprise, SOHO, remote offices and multi-location enterprises a cost-effective, easy to deploy VoIP FXO solution. The GXW4108 allows any business to seamlessly connect multiple locations with up to 8 PSTN lines, to an IP PBX system, or with an existing traditional phone system. The GXW4108 has 8 FXO ports. Designed and tested for full interoperability with leading IP-PBXs, soft-switches and SIP-based environments, the GXW technology ensures manageability, a simple configuration, superb voice and video quality and feature rich functionality. The GXW4108 FXO IP Analog Gateway is based on open industry standards.

Technical Specification

VoIP-FXO analog gateway to support up to 8 PSTN lines
Video surveillance port to enhance premise security
Two RJ-45 ports (switched or routed)
TFTP and HTTP firmware upgrade support
Supports Audio Codecs: G711, G723, G729 and GSM
T.38 compliant
Web management for easy configuration and installation
TFTP and HTTP firmware upgrade support
Multiple SIP profiles, max of 3 profiles per system. Each profile hosts 0 to multiple number of SIP accounts, depending on user need
One stage and two stage dialing
Two stage dialing means when after dialing the number to the GXW, be it from VoIP to GXW or from PSTN to GXW, a second dial-tone prompts users to input the final destination number to finish final dialing.
One stage dialing means user only hear dial-tone once and input a final destination number along with a pre-fix. One stage dialing need SIP server to support SIP call forward via a dial-plan
VoIP to PSTN call setup and teardown
PSTN to VoIP call setup and teardown
Support: G711, G723, G729, and GSM
Line echo canceller g.168 support
Flexible DTMF transmission method User Interface of In-audio, RFC2833, and SIP Info
Round-robin port scheduling to ensure available lines to access PSTN networks
Configurable channel dialing to improve dial-out reliability
Configurable PSTN Termination
Configurable call progress/termination tones via pattern matching

£255.00

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