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Grandstream GXW4108 Analog FXO IP Gateway – 8 Port

Original price was: ₦520,000.00.Current price is: ₦485,154.40.

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.

SKU: GXW4108 Category: Tag: Brand:

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

  • 8FXO ports
  • 2 10/100 Mbps network ports
  • Comprehensive codec support, caller ID, flexible dial plans and security protection
  • Advanced security protection with SRTP

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-4108 series. In this environment, the SIP server handles SIP registration and call control and the GXW4108 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.

Additional information

Spec.

Grandstream GXW-4108 Technical Specification

FXO Ports
8 FXO ports
Ethernet Ports
2 RJ45 10/100Mbps (LAN/WAN)
Video input
Supports H.264 video codec (up to 30fps and CIF resolution, Hardware Version below 2.0 only)
Audio Codecs
G711u/a
G723
G729
GSM
T.38 compliant
Configuration
Web Based
Remote TFTP/HTTP
Power Input
12 Vdc
1.25Amp
Compliance
FCC/CE/C-Trick
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 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)
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
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