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


135,700.00 KSh 135700.0 KES 150,775.00 KSh

150,775.00 KSh

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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)


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


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

Product Datasheet