Gateway VoIP 4 Porturi, Planet, 2 x FXO + 2 x FXS, SIP 2.0, IPv6 management, caller ID, NAT - VGW-402

Cod: VGW-402

Cost-effective, High-performance VoIP Communication
To build high-performance VoIP communications at a low cost, PLANET now introduces the latest member of its gateway family, the VGW-402 enterprise-class 4-port SIP VoIP Gateway. The VGW-402 gateway provides added flexibility during migration to Unified Communications by supporting the traditional analog devices, which include analog phones, fax machines, modems, voicemail systems, and speakerphones. It helps the company to save money on long-distance calls; for example, the remote workers can dial in through a Unified VoIP Communication System just like an extension call but no long-distance call charge would occur. The VGW-402 also allows call to be transferred to anyone at any location within the voice system, which enables the enterprise to communicate more effectively and is helpful to streamline business processes.

Standard Compliance
The VGW-402 supports Session Initiation Protocol 2.0 (RFC 3261) for easy integration with general voice over IP system. The VGW-402 is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better multi-media exchange services.

Enhanced, Full-Featured Business Gateway
The VGW-402 is a full-featured enhanced business SIP Gateway that addresses the communication needs of the enterprises. It provides the 2-line FXO plus 2-line FXS gateway with SIP protocol IP device which allows connection with 2 analog PSTN telephone lines and with 2-line analog telephone set to make or receive VoIP call over Internet or VPN network. This device is suitable for office PABX to enable to have VoIP call without changing cabling, dial plan and extension number.

The VGW-402 supports all kinds of SIP-based gateway features and multiple contact filter functions, such as 4 SIP trunk accounts, both IPv6 and IPv4 protocols, flexible dial plan and route plan features, and switch analog and VoIP signal to help both protocols to communicate.

Secure, High-Quality VoIP Communication
It can effortlessly deliver secured toll voice quality by utilizing cutting-edge 802.1p QoS (Quality of Service), 802.1Q VLAN tagging, and IP TOS (Type of Service) technology. Using voice and data VLAN can easily separate the data and voice, thus maintaining the best quality.

Supporting Caller ID
Both the FXS and FXO ports of the VGW-402 support caller ID function, helping users identify calling number and verify number easily. It also helps to block anonymous call by filtering strange calls. The FXS port transmits Caller ID, while the FXO port receives Caller ID. The Caller ID interoperates with analog phones, public switched telephone networks (PSTN) and private branch exchanges (PBXs).

4-port SIP Gateway (VGW-402)

WAN 1 x 10/100Mbps RJ-45 port
LAN 1 x 10/100Mbps RJ-45 port
Voice 4 x RJ-11 connection (2 x FXS, 2 x FXO)
Protocols and Standard
Data Networking IPv4 (RFC 791) and IPv6
IPv6 auto configuration (RFC 4862)
IPv6 only, IPv4 only or dual stack
MAC address (IEEE 802.3)
MAC clone setting
Vendor Class ID
Static IP
DHCP Client (RFC 2131), WAN port
DHCP Server, LAN port
NAT Server (RFC 1631)
PPPoE Client / DNS Client / TFTP Client
DDNS (Planet DDNS, Easy DDNS, DynDNS)
URL / IP / MAC / Port Filter
Application Program Filter
Port Forwarding (TCP, UDP or both)
Bandwidth control (download and upload), maximum bandwidth priority setting
UPnP Server at LAN port
Behind NAT, use DMZ for NAT traversal
SNTP with time zone and Daylight Saving
TCP/UDP (RFC 793/768), RTP/RTCP (RFC 1889/1890), IPV4 ICMP (RFC 792)
VoIP VLAN Support 802.1Q, 802.1P
VLAN ID Range: 2 to 4094
VLAN Priority: 0 to 7 (Highest Priority)
QoS: DiffServ (RFC 2475), TOS (RFC 791, 1394)
Voice Gateway RFC 3261 compliance
Supports up to 4 SIP Trunks to Register
SIP UDP Protocol
Supports SIP compact Form
Supports SIP HOLD Type: Send Only, or inactive
SIP Session Timer (RFC 4028)
SIP Session Refresher: UAC or UAS
SIP Encryption
MD5 Digest Authentication (RFC 2069 / RFC 2617)
Reliability of provision response PRACK (RFC 3262)
Early/Delay Media support
Offer/Answer (RFC 3264)
Message Waiting Indication (RFC 3842)
Event Notification (RFC 3265)
REFER (RFC 3515)
Supports Outbound Proxy
Supports Primary and Backup SIP Server
Supports STUN NAT Traversal
Supports “rport” parameter (RFC 3581)
Configure SIP local Port
SIP QoS Type: DiffServe or QoS
Accept Proxy Only : Yes or No
Audio Codec G.711 A-law/μ-law, G.729A, G.723.1 (6.3K, 5.3K)
Select voice codec priority : Local or Remote
Voice Payload size (ms) configuration
Silence Suppression
LEC : Line Echo Canceller
Max Echo Tail Length (G.168): 32, 64 and 128ms
Packet Loss Compensation
Automatic Gain Control
In-band/out of band DTMF (RFC4733, RFC2833 / SIP INFO)
Adaptive/Configurable Jitter Buffer
G.168 Acoustic Echo Cancellation
Configure RTP basic Port
RTP QoS Type : DiffServ or TOS
Phone Book (50 records) for peer to peer calls
Dialing Plan with drop, replace, Insert dialing digits
Selects first digit and inter digit timeout duration (Sec)
Selectable Call Progress Tone
Supports Specified Line Calling
Call Functions Supports Peer to Peer dialing
2-line FXO connects to PSTN Line
2-line FXS connects to analog phone set or PABX
Caller ID recognition DTMF (before/after 1st ring) and FSK (before 1st ring), ETSI and Bellcore
DTMF Caller ID start and stop BIT configurable
Current Drop Detection to release FXO port
Disconnect tone recognition to release FXO port
Tone Generation: Ring Back, Dial, Busy, call waiting, ROH, Warning, Holding, Stutter dial tone and disconnect tone
Configure Tone Frequency, Cadence, Level and Cycle
Select Tone specification by Country name List
Global Country Based Tone Specification
NAT Traversal support STUN, UPNP and Behind NAT
Out-Band DTMF with RFC 2833 and SIP Info
RFC 2833 Payload type: 101 or 96
DTMF send out ON and OFF Time configure
DTMF incoming recognition Minimum ON and OFF time
DTMF Relay Volume configuration
T.38 FAX Volume configuration
Flash Time transmit via SIP Info (Enable or Disable)
Message Waiting Indication (Stutter Tone Notice)
Blocks Anonymous Call
Call Hold, Call Transfer
FXO/FXS Line Configuration Activates or deactivates : Line ID, Line Phone number
Polarity Reversal detection or generation for call establish and Billing
HOT Line to desired phone number
Plays voice file to incoming call
Repeats playing voice file counts
Self-recorded voice files to upload
Generates FLASH TIME to PSTN network
T.38 or FAX Relay Type
Incoming and outgoing dB value configurable
Dialing Answer Delay time to establish call path
Answers PSTN incoming call after how many ring cycles
Caller ID detection mode by Country selection
VoIP dial to FXO/PSTN Line: 1 stage dialing and 2 stage dialing
Outgoing SIP Caller ID Selection
Supports 4 SIP Trunk
Accepts desired SIP Proxy incoming calls Only
Flexible Routing Plan Prefix Match and Length
Priority Ring
Cyclic Ring
Simultaneous Ring
Programmable Hunting Cycle
Backup Routes with Digit Manipulation
Default Routes
Flexible Dial Plans Retrieves transfer call from 3rd party by dial code (default: *#)
Inter digit time out setting
First digit dial out delay time setting
End of dial keypad number
Dial Rule : Match dial prefix and maximum digits length (1-15)
Phone Book can be exported or imported
FXS Analog 2-wire interface Flash Time Detection: range from 80 to 800 ms
ON-HOOK Voltage -48Vdc
Configure Ring Cadence, Frequency and Voltage
Supports Polarity reversal for Billing
Service Up to 1 Kilo-meter distance to analog telephone set
Generate Current Drop Time (Open Loop Disconnect time)
FXO Analog 2-wire interface Incoming Ring frequency recognition range: 10 to 70 Hz
Incoming Ring ON time recognition range: 0 to 8000ms
Incoming Ring OFF time recognition range: 0 to 8000ms
Incoming Ring Level recognition range: 10 to 95Vrms
Flash Time Detection: range from 80 to 800 ms
Configure Ring Cadence, Frequency and Voltage
Management Administrative Telnet CLI and HTTP, HTTPS
HTTP provision through MAC address
Multilingual Web User Interface
3 Levels of User Access Right with Password protection with different Web Language (Administrator, Supervisor and User)
HTTP/HTTPS Service Access limitation from WAN port
Configure Service ports at HTTP, HTTPS and telnet Services
Phone Debug Module: Device Control, Call Control, DB, Verbose
SIP Debug Module: Register, Call, SIP Message, Others
SNTP Debug Module
Device Debug Module
DSP Debug
Provides System Status Logs
Connect to external SYSLOG Server
Status display: Network, Line, SIP Trunk status
Diagnostics (debug through Syslog Event Notice)
Debug in real time by Telnet
Auto Provision via HTTP Server
SNMP v2 / Trap
Configuration Backup/Restore
Dual Firmware Image Backup
Reset to factory Default
Power Requirements 12V DC, 1.5A
Operating Temperature 0 ~ 45 degrees C
Operating Humidity 10~90% relative humidity, non-condensing
Weight 500 g
Dimensions (W x D x H) 175 x 32 x 126 mm
Emission CE, FCC, RoHS
Connectors Two 10/100Base-TX RJ-45 Ethernet ports
Four RJ-11 ports
DC power jack
Data Tip Model Versiune Descriere Download
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