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Planet VGW-1620FS 16-Port SIP VoIP Gateway (16 FXS)
MPN: VGW-1620FS
SKU: PLA-VGW-1620FS
UPC:
$497.20
Special Price : (Login for Your Price)
Availability
Usually Ships 10-14 Business Days
High Quality yet Affordable for All Businesses
PLANET VGW-1620FS enterprise-class 16-port SIP VoIP Gateway provides added flexibility during migration to Unified Communications by supporting the traditional analog devices. These devices include analog phones, fax machines, modems, voicemail systems and speakerphones.
Enhanced, Full-Featured Business Gateway
PLANET VGW-1620FS 16-port FXS SIP VoIP Gateway is a fully IETF SIP RFC 3261 standard compliant residential gateway that provides a total solution for integrating voice-data network, with built-in SIP trunk and TLS/SRTP security, up to 16 concurrent connections. Voice communications can be established from anywhere around the world, and it not only provides quality voice communications, but also offers secure, reliable Internet sharing capabilities for daily voice and Internet communications.
Distributed VoIP Network Infrastructure
PLANET VGW series is easy to use for all types of businesses. The VGW-1620FS offers quality voice communications and real-time fax data over IP networks and it does not need human resources to deploy a VoIP network. With the optimized SIP architecture, PLANET VGW-1620FS is the ideal choice for P2P/SIP proxy (IP PBX) voice chat, and ITSP cost-saving solution.
| MPN: | VGW-1620FS |
|---|---|
| Manufacturer | Planet Technology |
| Show Group Special Price | Yes |
| Hardware | |
|---|---|
| LAN | 4 x 10/100BASE-TX RJ45 port |
| Voice | 16 x RJ11 connection (32 x Foreign eXchange Station) |
| Console | 1 x RS232, 115200bps |
| Weight | 2700g |
| Dimensions (W x D x H) | 440 x 230 x 44 mm |
| Power Requirements | 100-240VAC, 50-60 Hz |
| Power Consumption | 30W |
| Protocols and Standard | |
| FXS | Dial Mode: DTMF and Pulse Pulse: 10 and 20 PPS Caller ID: DTMF/FSK CLI Presentation Max Cable Length: 3KM Reverse Polarity Programmable Call Progress Tone |
| Voice & Fax | G.711A/U law, G.723.1, G.729A/B,G.726 and iLBC Silence Suppression Comfort Noise Generation (CNG) Voice Activity Detection (VAD) Echo Cancellation (G.168), with up to 128ms Adaptive (Dynamic) Jitter Buffer Hook Flash Programmable Gain Control T.38/Pass-through Modem/POS DTMF mode: Signal/RFC 2833/INBAND VLAN 802.1P and 802.1Q Layer 3 QoS and DiffServ |
| VoIP | IETF Session Initiation Protocol (SIP) v2.0 (UDP/TCP) RFC 3261 and Session Description Protocol (SDP) RTP (RFC 2833), RFC 3262, RFC 3263, RFC 3264, RFC 3265, RFC 3515, RFC 2976 and RFC 3311 RTP/RTCP, RFC 2198 and RFC 1889 RFC 4028 Session Timer RFC 3266 IPv6 in SDP RFC 2806 TEL URI RFC 3581 NAT and rport Primary/Backup SIP Server Outbound Proxy DNS SRV/A Query/NATPR Query SIP Trunk Early Media/Early Answer NAT:STUN, Static/Dynamic NAT |
| Supplementary Service | Call Waiting Blind Transfer Attend Transfer Call Forward on Busy Call Forward on No Reply Unconditional Call Forward Warm/Immediately Hotline Call Hold Do-not-disturb 3-Way Conferencing Message Waiting Indicator |
| Software Features | Hunting Group Web ACL Telnet ACL Action URL PPPoE/IPv4/IPv6 Digitmap Bandwidth Optimization Routing Rules based Prefixes Caller/Called Number Manipulation |
| Management | SNMP v1/v2/v3 TR069 Auto Provisioning Web/Telnet Configuration Backup/Restore Firmware Upgrade via Web CDR Syslog Ping and Tracert Test Network Capture Outward Test (GR909) NTP and Daylight Saving Time IVR local Maintenance |
| Standards Conformance | |
| Emission | CE, FCC |
SIP Applications
- IETF SIP RFC3261 based on UDP/TCP/TLS
- 16-line FXS connects to analog phone set or PABX
- Fax over T.38 and Pass-through
- ITU-T G.711 A-law, G.711 μ-law, G.723.1 and G.729 voice coding
- In-band / out of band DTMF (RFC4733, RFC2833 / SIP INFO)
- Echo cancellation exceeding ITU-T G.168, up to 128ms tail length
- Supports SIP Trunk and Caller ID: DTMF/FSK CLI Presentation
Internet Features
- Supports SNMP v1/v2/v3
- Supports VLAN 802.1P and 802.1Q
- Supports Layer3 QoS and DiffServ
- Supports STUN (RFC 3489) and Outbound Proxy
- Supports TR069 and Auto Provisioning
- Supports TLS/SRTP Security
Call Features
- Call waiting, transfer (Blind transfer, Attend transfer)
- Call hold, quick pick
- Call forwarding unconditional
- Call forwarding on no reply
- Hotline, speed dial, direct IP call
- Do not disturb (DND), 3-way conferencing
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