Planet VGW-402 4-Port SIP VoIP Gateway (2 FXS + 2 FXO)

Spesifikasi Planet VGW-402 4-Port SIP VoIP Gateway (2 FXS + 2 FXO) 

Hardware	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 IP / ICMP / ARP / RARP / SNTP 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) Firewall 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 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 SIP UDP Protocol Supports SIP compact Form :  Supports SIP HOLD Type: Send Only, 0.0.0.0 or inactive SIP Session Timer (RFC 4028) SIP Session Refresher: UAC or UAS SIP Encryption MD5 Digest Authentication (RFC 2069 / RFC 2617) Reliability of
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 VAD/CNG 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
Functions	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
Environments	Power Requirements : 12V DC, 1.5A Operating Temperature : 0 ~ 45 degrees C Operating Humidity : 10~90% relative humidity, non-condensing Emission : CE, FCC, RoHS Connectors : Two 10/100Base-TX RJ-45 Ethernet ports Four RJ-11 ports DC power jack

Berat dan ukuran Planet VGW-402 4-Port SIP VoIP Gateway (2 FXS + 2 FXO) 

Weight	500 g
Dimensions (W x D x H)	175 x 32 x 126 mm

Ringkasan Produk Planet VGW-402 4-Port SIP VoIP Gateway (2 FXS + 2 FXO) 

Hemat biaya, Komunikasi VoIP berkinerja tinggi
Untuk membangun komunikasi VoIP berkinerja tinggi dengan biaya rendah, PLANET sekarang memperkenalkan anggota terbaru dari keluarga gateway-nya, gateway SIP VoIP 4-port VGW-402 enterprise-class. Gateway VGW-402 memberikan fleksibilitas tambahan selama migrasi ke Unified Communications dengan mendukung perangkat analog tradisional, termasuk telepon analog, mesin faks, modem, sistem voicemail, dan speakerphone. Ini membantu perusahaan menghemat uang untuk panggilan jarak jauh; misalnya, pekerja jarak jauh dapat menghubungi melalui Sistem Komunikasi VoIP Bersatu seperti panggilan ekstensi namun tidak ada biaya panggilan jarak jauh yang akan terjadi. VGW-402 juga memungkinkan panggilan untuk ditransfer ke siapapun di lokasi manapun dalam sistem suara, yang memungkinkan perusahaan berkomunikasi lebih efektif dan sangat membantu untuk merampingkan proses bisnis.

Kepatuhan Standar
VGW-402 mendukung Session Initiation Protocol 2.0 (RFC 3261) untuk integrasi yang mudah dengan sistem voice over IP secara umum. VGW-402 dapat beroperasi secara luas dengan peralatan yang disediakan oleh penyedia infrastruktur VoIP, sehingga memungkinkan mereka untuk menyediakan layanan pertukaran multi media kepada pelanggan mereka dengan lebih baik.

Enhanced, Full-Feature Business Gateway
VGW-402 adalah bisnis SIP Gateway dengan fitur lengkap yang memenuhi kebutuhan komunikasi perusahaan. Ini menyediakan 2 jalur FXO plus gateway FXS 2-line dengan perangkat IP protokol SIP yang memungkinkan koneksi dengan 2 saluran telepon PSTN analog dan dengan telepon analog 2 jalur untuk membuat atau menerima panggilan VoIP melalui jaringan Internet atau VPN. Perangkat ini cocok untuk kantor PABX agar bisa melakukan panggilan VoIP tanpa mengubah kabel, rencana dial dan nomor ekstensi.

VGW-402 mendukung semua jenis fitur gateway berbasis SIP dan beberapa fungsi filter kontak, seperti 4 akun batang SIP, protokol IPv6 dan IPv4, rencana paket fleksibel dan fitur perencanaan rute, dan mengganti sinyal analog dan VoIP untuk membantu kedua protokol tersebut. untuk berkomunikasi

Aman, Berkualitas Tinggi Komunikasi VoIP
Ini dapat dengan mudah memberikan kualitas suara tol yang terjamin dengan memanfaatkan QoS 802.1p mutakhir (Quality of Service), penandaan VLAN 802.1Q, dan IP TOS (Jenis Layanan). Menggunakan suara dan data VLAN dapat dengan mudah memisahkan data dan suara, sehingga menjaga kualitas terbaik.

Mendukung Caller ID
Baik port FXS dan FXO dari fungsi IDer pendukung VGW-402, membantu pengguna mengidentifikasi nomor panggilan dan memverifikasi nomor dengan mudah. Ini juga membantu memblokir panggilan anonim dengan menyaring panggilan aneh. Port FXS mentransmisikan Caller ID, sedangkan port FXO menerima Caller ID. Caller ID beroperasi dengan telepon analog, jaringan telepon umum (PSTN) dan bursa efek swasta (PBXs).