Grandstream GXW4108 Analog FXO IP Gateway – 8 Port
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.
$299.97
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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.
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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
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