===== NOTE ======= Produk "Grandstream GXW4108" telah discontinue (End of Life), dan telah di gantikan ke produk "Grandstream HT881" , dapat anda temukan di halaman toko kami ============================================================================
Product Overview : The GXW FXO IP Analog Gateway series offers the small enterprise, SOHO, remote offices and multi-location enterprises a cost-effective, easy to deploy VoIP FXO solution. The GXW4108 series 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 Grandstream GXW-4108 offers an easy to manage, easy to configure IP communications solution for any small business or businesses with virtual and/or branch locations who want to leverage their broadband network and/or add new IP Technology to their current phone system. The Grandstream Enterprise Analog VoIP Gateway GXW410x series converts SIP/RTP IP calls to traditional PSTN calls. There are two models - the GXW4104 and GXW4108, which have either 4 and 8 FXO ports respectively.
Grandstream GXW-4108 Features : - 8 FXO analog port gateways - Video surveillance - 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 accounts, associated with physical line ports, each account corresponding to one of the multiple SIP profile - 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 - Channel configurable for one stage or two stage dialing, Default is 2 stage dialing. One stage dialing requires user to configure Off-Hook Auto Dial to a SIP Number. - 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