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Rp4.997.000

Grandstream GXW4108 - 8FXO VoIP Analog Gateway

Rp4.997.000
  • Kondisi: Baru
  • Waktu Preorder: 30 Hari
  • Min. Pemesanan: 1 Buah
  • Etalase: ANALOG VoIP GATEWAYs - FXO and FXS VoIP GATEWAYs
===== NOTE =======
Produk "Grandstream GXW4108" telah discontinue (End of Life), dan telah di gantikan ke produk "Grandstream HT881" , dapat anda temukan di halaman toko kami
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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
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