D-Link DVG-2032S/16CO/C1A 16-ports FXS modular Gateway


    162,425.00 Kshs 162,425.00 Kshs 162425.0 KES


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    16-ports FXS modular Gateway, 1 10/100M LAN, 1 10/100M WAN, 1 open slot, QoS, DHCP Server, NAT, Dynamic DNS, Support Call Control Protocol SIP, Call features support, IP Routing, RIP v1, RIP v2, MAC Filtering, IP Filtering, Web-configuration, Telnet, CLI, TFTP, SNMP support, 2 cables for TELCO-50 ports

    Part Number: DVG-2032S/16CO/C1A

    The DVG-2032S VoIP Station Gateway presents an ideal Internet telephone solution for business use. This gateway converts voice traffic into data packets for transmission over the Internet. It combines the industry’s latest Voice over IP (VoIP) network technology with advanced communication features and is fully compatible with SIP Internet phone services. High port densities allow it to provide a low cost of ownership, convenience, and great savings for companies needing to place frequent long-distance and international business calls.


    Voice Features

    G.711 a-law 64K

    Packet Interval: 20/30/40 ms

    Concurrent Calls: 32 ch @ 20 ms

    G.711 μ-law 64K

    Packet Interval: 20/30/40 ms

    Concurrent Calls: 32 ch @ 20 ms

    G.723.1 5.3K/6.3K

    Packet Interval: 30/60/90 ms

    Concurrent Calls: 32 ch @ 30 ms

    G.726 32K

    Packet Interval: 20/30/40 ms

    Concurrent Calls: 32 ch @ 20 ms

    G.729 8K

    Packet Interval: 20/30/40 ms

    Concurrent Calls: 32 ch @ 20 ms

    DTMF Detection and Generation

    Silence Suppression & Detection

    Comfort Noise Generation (CNG)

    Voice Activity Detection (VAD)

    Echo Cancellation (G.165/G.168)

    Adaptive (Dynamic) Jitter Buffer

    Call Progress Tone Generation

    Auto or 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 Relay (RFC2833 or SIP INFO)

    DTMF / PULSE Dial Support

    Caller ID Generation / Detection:


    FSK-Bellcore Type 1 & 2

    FSK-ETSI Type 1 & 2


    FSK: Calling Name, Number, Date and Time, VMWI

    FXS Metering Pulse:

    Polarity Reversal

    12 kHz calling tone

    16 kHz calling tone

    T.30 FAX Bypass to G.711, T.38 Real-Time FAX Relay

    FXS Line test and diagnostics with visual alarm


    Inward self-test:

    Loopback - codec

    Loopback - analog

    SLIC DC power voltage

    Tip / Ring DC feed


    Outward Test (GR909 Standard) :


    Phone Line disconnected

    H.F. DC Voltage (Hazardous and foreign DC Voltage)

    H.F. AC Voltage (Hazardous and foreign AC Voltage)

    Tip / Ring Short

    Modem over IP up to V.34

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

    Loop Current Suppression

    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 Call Management

    Support Outbound Proxy

    Register up to three SIP servers

    SIP Registration Failover Mechanism

    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 Support

    CDR Client

    Manual Peer Table (for P2P calls)

    E.164 Numbering, ENUM support

    IP Network Specifications

    Support IPv4, IPv6 future upgradable (Option)

    WAN: Static IP, PPPoE, DHCP, PPTP

    Network Protocol Support:



    DNS SRV, Telnet, DHCP Server, DHCP Client,

    STUN Client, UPnP, IGMP snooping, IGMP proxy

    QoS Support:

    WAN: DiffServ, IP Precedence, Priority Queue,

    Rate Control, 802.1Q (VLAN Tagging), 802.1p (Priority


    LAN: Rate Limit

    DDNS Support

    Network Security Specifications

    VPN PPTP Client

    DIGEST Authentication

    MD5 Encryption

    DoS Protection


    Web-based Configuration

    Auto-provisioning (HTTP / HTTPS)



    FTP / TFTP / HTTP Software Upgrade

    Configuration Backup and Restore

    Reset to Default Button

    TR-069/104 (Option)

    SIP, Voice and FAX Related Standard

    RFC1889 RTP: A Transport Protocol for Real-Time Applications.

    RFC2543 SIP: Session Initiation Protocol

    RFC2833 RTP Payload for DTMF Digits, Telephony

    Tones and Telephony Signals

    RFC2880 Internet Fax T.30 Feature Mapping

    RFC2976 The SIP INFO Method

    RFC3261 SIP: Session Initiation Protocol

    RFC3262 Reliability of Provisional Responses in

    Session Initiation Protocol (SIP)

    RFC3263 Session Initiation Protocol (SIP): Locating SIP Servers

    RFC3264 An Offer/Answer Model with Session Description Protocol (SDP)

    RFC3265 Session Initiation Protocol (SIP) - Specific Event Notification

    RFC3311 The Session Initiation Protocol (SIP) UPDATE Method

    RFC3323 A Privacy Mechanism for the Session Initiation Protocol (SIP)

    RFC3325 Private Extensions to the Session Initiation

    Protocol (SIP) for Asserted Identity within Trusted Networks

    RFC3362 Real-time Facsimile (T.38) - Image/t38 MIME Sub-type Registration

    RFC3515 The Session Initiation Protocol (SIP) Refer Method

    RFC3550 RTP: A Transport Protocol for Real-Time

    Applications. July 2003

    RFC3665 Session Initiation Protocol (SIP) Basic Call

    Flow Examples

    RFC3824 Using E.164 numbers with the Session

    Initiation Protocol (SIP)

    RFC3842 A Message Summary and Message Waiting

    Indication Event Package for the Session Initiation

    Protocol (SIP)

    RFC3891 The Session Initiation Protocol (SIP)

    “Replaces” Header

    RFC3892 The Session Initiation Protocol (SIP) Referred-By Mechanism

    RFC3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)

    RFC3986 Uniform Resource Identifier (URI): Generic Syntax

    RFC4028 Session Timers in the Session Initiation Protocol (SIP)

    Draft-IETF-sipping-service-examples-08 for call features

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