D-Link DVG-2032S-16CO – 32-Port VoIP Station Gateway with Dual 16 FXS Ports, SIP, QoS, T.38 Fax & Web/IVR Management

• The D-Link DVG-2032S is a 32-Port VoIP Station Gateway designed for business Internet telephony applications.

• It converts voice traffic into data packets for transmission over IP networks and provides a cost-saving solution for long-distance and international business calls.

• The gateway provides two high-density 16 FXS port modules, supporting up to 32 simultaneous Internet telephone connections.

• It allows businesses to keep using existing analog phones, conference speakerphones, and fax machines while migrating gradually to VoIP.

• The DVG-2032S is compatible with SIP Internet phone services and supports extensive call features such as call hold, call waiting, call forward, call transfer, caller ID, greeting message, and call detail records.

• It supports QoS, IPv4, optional IPv6 upgrade, SIP registration failover, web-based configuration, IVR configuration, auto-provisioning, and web-based firmware upgrade for easier deployment and management.

• 32-Port VoIP Station Gateway for business Internet telephony

• Two high-density 16 FXS ports

• Supports up to 32 simultaneous Internet phone connections

• Ideal for business phones and fax connection

• Compatible with SIP Internet phone services

• Cost-saving solution for long-distance and international calls

• Allows reuse of existing analog phones and fax machines

• Guaranteed toll-quality voice over busy networks

• Supports call hold, call waiting, call forward, and call transfer

• Caller ID support

• Greeting message support

• Call Detail Record support

• 10/100 Mbps Ethernet WAN port

• 10/100 Mbps Ethernet LAN port

• WAN supports Static IP, PPPoE, DHCP, PPTP, and DDNS

• Easy configuration using IVR or web-based GUI

• Web-based firmware upgrade

• Supports T.30 fax bypass and T.38 real-time fax relay

• Supports modem over IP up to V.34

• Supports SIP registration failover and up to three SIP servers

• Supports QoS, VLAN tagging, priority tagging, and rate control

• Product Type: 32-Port VoIP Station Gateway

• Model: DVG-2032S / DVG-2032S-16CO

• FXS Ports: Two high-density 16 FXS ports

• Concurrent Calls: Up to 32 channels

• WAN Port: 10/100 Mbps Ethernet WAN port

• LAN Port: 10/100 Mbps Ethernet LAN port

• WAN Connection: Static IP, PPPoE, DHCP, PPTP

• DDNS: Supported

• IPv4: Supported

• IPv6: Future upgradeable option

• Power Input: 100–240V AC, 50/60Hz

• Optional Power Input: -36V to -72V, 50/60Hz

• Power Consumption: 80W

• MTBF: 83,745 hours

• Dimensions: 445 × 330 × 45 mm

• Operating Temperature: -10°C to 40°C

• Storage Temperature: -20°C to 60°C

• Operating Humidity: 10% to 90% non-condensing

• Storage Humidity: 5% to 95% non-condensing

• Certifications: FCC Class B, CE Class A, CE LVD

• G.711 a-law 64K

• G.711 μ-law 64K

• G.723.1 5.3K / 6.3K

• G.726 32K

• G.729 8K

• Concurrent Calls: 32 channels depending on codec packet interval

• DTMF Detection and Generation

• Silence Suppression and Detection

• Comfort Noise Generation

• Voice Activity Detection

• 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

• Peer-to-peer call

• Call hold / retrieve

• Call waiting

• Call pickup

• Call park / retrieve with SIP server support

• Call forward: unconditional, busy, no answer

• Call transfer: attended and unattended

• Do Not Disturb

• Speed dialing

• Repeat dialing

• Three-way calling

• MWI RFC3842

• Hotline and warm line

• In-band DTMF

• Out-of-band DTMF Relay via RFC2833 or SIP INFO

• DTMF / Pulse Dial support

• Caller ID generation and detection

• Caller ID Types: DTMF, FSK-Bellcore Type 1 & 2, FSK-ETSI Type 1 & 2, FSK-NTT

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

• FXS Metering Pulse: Polarity Reversal, 12kHz calling tone, 16kHz calling tone

• T.30 Fax Bypass to G.711

• T.38 Real-Time Fax Relay

• FXS line test and diagnostics with visual alarm indication

• Modem over IP up to V.34

• ROH Tone Receiver Off-Hook Tone at 480Hz

• Loop Current Suppression

• By port registration

• By device registration with shared account

• Mixed mode with hunt number for inbound and port number for outbound

• Invite with Challenge

• Register by SIP Server IP address or domain name

• Supports RFC3986 SIP URI format

• Outbound Proxy support

• 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

• CDR Client

• Manual Peer Table for P2P calls

• E.164 Numbering and ENUM support

• Network Protocols: IP, TCP, UDP, TFTP, FTP, RTP, RTCP, ARP, RARP, ICMP, NTP, SNTP, SNMP v1/v2, HTTP, HTTPS, DNS, DNS SRV, Telnet, DHCP Server, DHCP Client, STUN Client, UPnP, IGMP Snooping, IGMP Proxy

• QoS WAN: DiffServ, IP Precedence, Priority Queue, Rate Control, 802.1Q VLAN Tagging, 802.1p Priority Tag

• QoS LAN: Rate Limit

• DDNS Support

• VPN PPTP Client

• Digest Authentication

• MD5 Encryption

• DoS Protection

• Web-based configuration

• Auto-provisioning via HTTP / HTTPS

• Telnet management

• IVR configuration

• FTP / TFTP / HTTP software upgrade

• Configuration backup and restore

• Reset to default button

• TR-069 / TR-104 optional

• Business Internet telephony

• Analog phone to VoIP migration

• IP-PBX analog extension integration

• Large office analog phone deployment

• Fax over IP applications

• Call center analog extension integration

• Long-distance and international call cost reduction

• Multi-line SIP service deployment

• Businesses keeping existing phones and fax machines

• 32-port FXS gateway projects

• Recommended for companies that need high-density analog extension integration with SIP-based VoIP systems.

• Suitable for businesses migrating from traditional analog telephony to VoIP while keeping existing phone and fax devices.

• The 32-port capacity makes it a strong choice for large offices, call centers, hotels, and multi-extension business environments.

• QoS, SIP failover, fax support, IVR, web management, and auto-provisioning make it suitable for professional and scalable VoIP deployments.