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.
SPECIFICATION
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:
DTMF
FSK-Bellcore Type 1 & 2
FSK-ETSI Type 1 & 2
FSK-NTT
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
indication
Inward self-test:
Loopback - codec
Loopback - analog
SLIC DC power voltage
Tip / Ring DC feed
Ringer
Outward Test (GR909 Standard) :
REN
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:
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 Support:
WAN: DiffServ, IP Precedence, Priority Queue,
Rate Control, 802.1Q (VLAN Tagging), 802.1p (Priority
Tag)
LAN: Rate Limit
DDNS Support
Network Security Specifications
VPN PPTP Client
DIGEST Authentication
MD5 Encryption
DoS Protection
Management
Web-based Configuration
Auto-provisioning (HTTP / HTTPS)
Telnet
IVR
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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