D-Link DVG-6004S/E VoIP Gateway with built-in 4 FXS

    19,103.80 KSh 19,103.80 KSh 19103.8 KES

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    VoIP Gateway with built-in 4 FXO, 1 10/100Mbps WAN & 4-port 10/100Mbps Switch

    Part Number: DVG-6004S/E

    The D-Link DVG-6004S VoIP trunk gateway connects an IP network to POTS (plain old telephone service) analog lines to send IP voice and data to the conventional phone sets and fax machines. Designed as a cost-effective VoIP solution for a small to medium-sized business and the branch office, these devices provide 4 FXO (Foreign eXchange Office) ports for 4 phone line connections. IP network connection can be through any of the 2 LAN ports or the WAN port (all 10/100/1000Base-T Ethernet interfaces).

    The DVG-6004S VoIP gateway can be deployed to interface VoIP infrastructure with a Public Switched Telephone Network (PSTN)

    They allow analog phone callers to place calls through the long-distance Internet trunk line. The gateways accept dial tone and ring current from the connected equipment, and facilitate the use of conventional phone sets and fax machines on a VoIP network

    General features

    Interfaces

    • WAN RJ-45 10/100/1000Base-T ports

    • 2 LAN RJ-45 10/100/1000Base-T ports

    • 4 RJ-11 FXO ports

    Diagnostic LEDs

    • Power

    • Alarm

    • Register

    • Provisioning

    • WAN

    • LAN 1~2

    • Line 1~4

    Buttons

    • Reset button

    • Power on/off button

    Voice Features

    • Codecs: G.722, G.711 a/Ој-law, G.729, G.726, G.723.1, GSM 6.10 Full Rate

    • DTMF Detection and Generation

    • Silence Suppression & Detection

    • CNG (Comfort Noise Generation)

    • VAD (Voice Activity Detection)

    • G.165/G.168 Echo Cancellation

    • Adaptive (Dynamic) Jitter Buffer

    • Call progress tone detection (FXO) and generation (FXS)

    • Programmable Gain Control

    • Built-in Local Mixer

    • ITU-T V.152 Voice-band Data over IP Networks

    SIP Call Features

    • Peer to Peer Call

    • Call Hold / Retrieve

    • Call Waiting

    • Call Pick Up

    • Call Park / Retrieve (SIP Server Required)

    • Call Forward - unconditional, busy, no answer

    • Call Transfer - attended, unattended

    • Do Not Disturb

    • Speed Dialing

    • Repeat Dialing

    • Three-way Calling

    • MWI (RFC-3842)

    • Hot Line and Warm Line

    Telephony Specifications

    • In-Band DTMF, Out-of-Band DTMF, DTMF Relay (RFC2833 or SIP INFO)

    • DTMF / PULSE Dial

    • Caller ID Generation (FXS) and Detection (FXO): DTMF, FSK-Bellcore Type 1 and 2, FSK-ETSI Type 1 and 2, FSK-NTT

    • FSK: РёРјСЏ звонящего, номер, дата Рё время, VMWI

    • FXS metering pulse: Polarity Reversal, 12 kHz/16 kHz calling tone

    • Polarity Reversal Detection (FXO) and Generation (FXS)

    • T.30 FAX Bypass, T.38 Real Time FAX Relay

    • Recordable Greeting Message (FXO)

    • FAX and Modem over IP (up to 14,400bps)

    • ROH Tone (Receiver Off-Hook Tone @ 480 Hz)

    • Loop Current Suppression

    IP Network Specifications

    • WAN: Static IP, PPPoE, DHCP, PPTP

    • Network protocols: IP, TCP, UDP, TFTP, FTP, RTP, RTCP, ARP, RARP, ICMP, NTP, SNTP, HTTP, HTTPS, DNS, DNS SRV, Telnet, DHCP Server, DHCP Client, STUN Client, UPnP, IGMP, IGMP snooping, IGMP proxy, RTSP ALG

    • NAT functions

    - Support up to 255 Clients

    - Port Forwarding (Virtual Servers)

    - DMZ

    - Port Triggering

    - IPv4, IPv6

    • QoS support

    - WAN: Diffserv, IP Precedence Priority Queue

    - Rate Control

    - 802.1Q (VLAN Tagging), 802.1p (Priority Tag)

    - LAN: Rate Limit

    • DDNS support

    • Dyndns.org (Dynamic and Custom)

    SIP Account Management

    • By port registration

    • By device registration (share account)

    • Mixed mode (Hunt number for inbound, by port number for outbound)

    • Invite with Challenge

    • Register by SIP Server IP Address or Domain Name

    • Support RFC3986 SIP URI format

    SIP Method Support

    • ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PING, PRACK, PUBLISH, REFER, REGISTER, SUBSCRIBE, UPDATE

    SIP Call Management

    • Outbound Proxy support

    • Register up to 3 SIP servers

    • SIP Registration Failover

    • Group Hunting

    • Privacy mechanism/Private extensions to SIP

    • Session Timers (Update/Re-invite)

    • DNS SRV support

    • Call Types: Voice / Modem / FAX

    • Call Routing by Prefix Number

    • User Programmable Dial Plan

    • Toll-Free support (FXO)

    • Automatic Calling Number Manipulation (VoIP & FXO)

    • CDR Client

    • Manual Peer Table (for P2P calls)

    • E.164 Numbering, ENUM support

    Network Security Specifications

    • PPTP client

    • DIGEST Authentication

    • MD5 Encryption

    • DoS Protection

    Management

    • Web based configuration

    • Auto-provisioning (HTTP/HTTPS/TFTP)

    • Telnet

    • IVR

    • FTP/TFTP/HTTP software upgrade

    • Configuration Backup and Restore

    • Rest to Default button

    • TR-069/104 (optional)

    • SNMP v1/v2c/v3dvg

    Physical Parameters

    Power Input

    • Power adapter: DC 12V / 2 A

    Operating temperature

    • 0В°C ~ +45В°C

    Storage temperature

    • -25В°C ~ +75В°C

    Operating humidity

    • Up to 90% (non-condensing)

    Weight

    • 450 g

    Dimensions

    • 226 x 155 x 80 mm

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