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Grandstream - GXW4104 - 4FXO VoIP Analog Gateway

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Rp4.204.845,00

Sinopsis

"4-FXO analog VoIP gateways for up to 4 PSTN line"
The GXW410x series of FXO IP Gateways series offers the small enterprise, SOHO, remote offices and multi-location enterprises a cost-effective, easy to deploy VoIP FXO solution. The GXW4104 series allows any business to seamlessly connect devices within an office or multiple offices with up to 4 PSTN lines, to an IP PBX system, or with an existing traditional phone system.

Grandstream - GXW4104 - 4FXO VoIP Analog Gateway

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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 GXW4104 series allows any business to seamlessly connect multiple locations with up to 4 PSTN lines, to an IP PBX system, or with an existing traditional phone system.

The Grandstream GXW-4104 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-4104 Features

  • 4 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.
  • PSTN to VoIP 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

The installation is the same for either model. A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-4104 series. In this environment, the SIP server handles SIP registration and call control and the GXW-4104 processes media conversion between IP and PSTN calls. By design, the system supports the North American call progress tones and signaling standards on PSTN sides.

Grandstream GXW-4104 Technical Specification

  • Configurable channel dialing to improve dial-out reliability
    - digit length: default 100ms
    - digit volume: gain [-31,0]dB, default -11dB
    - dial pause between digits: default 100ms
    - wait for dial-tone: yes/no, default yes (1 for Yes, 2 for No)
    - one-stage ( use 1 ) or 2 stage (use 2) dialing: default of 2 stage dialing
    - Syntax: ch (or chan or channel) x-y: val; ch
  • Configurable PSTN Termination
    - Enable current disconnect: default of disabled. Some special PBXs and CO lines use line power drop to indicate PSTN hang-up. When this is the configuration, please consult your PSTN line service provider for the correct PSTN disconnect method.
    - AC termination impedance: default North America. This impedance works with parameters of Busy/Re-order tone in Call Progress Table. Users have to set BUSY/REORDER tone values to enable this parameter.
    - Busy or re-order tones: following busy or reorder tone of call progress tones is used to teardown regular PSTN call if detected
  • Configurable call progress/termination tones via pattern matching
    - Dial-tone: f1/f2(350/440), v1/v2( -11/ -11), on1/off1(0/0), on2/off2(0/0)
    - Ring back tone: f1/f2(default 440/480), on/off(default 2s/4s)
    - Busy tone: f1/f2(480/620), on/off(0.5/0.5s), duration (8s)
    - Re-order tone: f1/f2( 480/620 ), on/off(25/25), duration (default 8s)
    - Confirmation tone: f1/f2(350/440), on/off(0.1/0.1s), duration (default 8s)
    - Usage Syntax:
    - ch x-y: f1(or freq1 or frequency1)=val1@vol1, f2 (or freq2 or frequency2) = val2@vol2, c (or cad or cadence) = on1/off1-on2/off2-on3/off3; ch3:
    - x,y - 0-9 digit.
    - Configure Channel voice settings,
    - Voice volume: gain control, [-31, 31], default 1 dB
    - Audio input gain: [-31, 31], default 0 dB
    - Silence Suppression: 1 - enabled, 2 - disabled, default is 1
    - Line echo cancellation: 1 - enabled, 2 - disabled; default is 1

Configure other channel settings, PSTN Silence Timeout, default 60 sec. This serves as a last measure to address PSTN run-away calls. It is not supposed to replace above regular PSTN disconnect methods.

DTMF Method via : default value is in-audio

1 - in-audio
2 - RFC2833
3 - in-audio and RFC2833
4 - SIP Info
5 - in-audio and RFC2833

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