LINKSYS SPA942-EU TELEFON VoIP 2xRJ45, 4 linie, SPA942-EU

LINKSYS SPA942-EU TELEFON VoIP 2xRJ45, 4 linie
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Kod producenta:	SPA942-EU
Producent:	Linksys
Kategoria:	Telefony i bramki VoIP
Cena netto:	372,36 zł
Cena brutto:	454,28 zł
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Dodatkowe informacje na temat tego produktu:
tel. 022 812 62 36 tel. kom. 607 120 152 e-mail: biuro@eurobajt.pl
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Gwarancja: R012M
Kod Producenta: SPA942-EU
Standard Voice: SIP
Liczba portów WAN (RJ45): 2 szt.
Funkcja Routera (NAT): Tak
Zarządzanie: HTTP, HTTPS
Kodeki: G.711 (A-law and u-law)
G.726 (16/24/32/40 kbps)
G.729 A
G.723.1 (6.3 kbps, 5.3 kbps)
Zasilanie: POE (802.3af), Zewnętrzne
Wymiary (wys x szer x gr): 160 x 195 x 180 mm
Waga: 9752 g
Kolor: Czarny
Opis:

Wyświetlacz LCD, 128 x 64 z podświetleniem
Podświetlane przyciski: Audio, Słuchawki, Głośnik
Przycisk do nawigacji w menu
Podświetlenie gdy oczekuje nagrana wiadomość
Kontrola głośności
Jack2.5mm na
słuchawki

Informacje dodatkowe:

Zaawansowany telefon IP z portem PoE oraz obsługa 4 linii
Możliwość bezpośredniego połączenia z internetowym operatorem lub z centrala IP PBX
Posiada dwu portowy przełącznik, głośnik, identyfikacje dzwoniącego,, zawieszanie
rozmowy, konferencja
Łatwa instalacja oraz wysoki poziom zabezpieczeń,

Model	SPA942

Ruch sieciowy	MAC address (IEEE 802.3) IPv4 - Internet Protocol v4 (RFC 791) ARP - Address Resolution, Protocol DNS - A record (RFC 1706), SRV record (RFC 2782) DHCP Client - Dynamic Host Configuration Protocol (RFC 2131) ICMP - Internet Control Message Protocol (RFC 792) TCP - Transmission Control, Protocol (RFC793) UDP - User Datagram Protocol (RFC 768) RTP - Real Time Protocol (RFC 1889) (RFC 1890) RTCP - Real Time Control Protocol (RFC 1889) DiffServ - Differentiated Services (RFC 2475) ToS - Type of Service, (RFC 791, 1349) VLAN tagging 802.1p/Q - Layer 2 quality of service (QoS) SNTP - Simple Network Time Protocol (RFC 2030)
Bramka głosowa	SIP v2 - Session Initiation Protocol version 2 (RFC, 3261, 3262, 3263, 3264) SIP proxy redundancy - dynamic via DNS SRV, A records Reregistration with primary SIP proxy Server SIP support in NAT networks (including STUN) SIPFrag (RFC 3420) Secure (encrypted) calling via SRTP, Codec name assignment Dynamic payload support Adjustable audio frames per packet DTMF - Dual-tone multifrequency, in-band and out-of-band (RFC 2833) (SIP INFO) Flexible dial plan, support with interdigit Times IP address/URI dialing support Call progress tone generation Jitter buffer: adaptive Frame loss concealment VAD - Voice activity detection with silence suppression Attenuation/gain adjustments, MWI - Message waiting indicator tones VMWI - Voicemail waiting indicator, via NOTIFY, SUBSCRIBE Caller ID support (name and number) Third-party call control (RFC 3725)
Funkcje, Telefonu	One voice line with two call appearances Line status: active line indication, name and number Menu-driven user interface Shared line appearance* Speakerphone Call hold Music on hold* Call Whiting, Caller ID name and number Outbound caller ID bloking Call transfer: attended and blind Three-way call conferencing with local mixing Multiparty conferencing via external conference, Bridge Automatic redial of last calling and last called numbers On-hook dialing Call pickup: selective and group* Call park and unpark* Call swap Call back on busy Call blocking: anonymous and selective Call forwarding, : unconditional, no answer, on busy Hot line and warm line automatic calling Call logs (60 entries each): made, answered, and missed calls Redial from call logs Personal directory with auto-dial (100, entries) Do not disturb (callers hear line busy tone) Digits dialed with number auto-completion Anonymous caller blocking Uniform Resource Identifier (URI) (IP) dialing support (vanity numbers) On-hook default audio configuration, (speakerphone and headset) Multiple ring tones with selectable ring tone per Line Called number with directory name matching Ability to call number using name: directory matching or via caller, ID Subsequent incoming calls show calling name and number Date and time with support for intelligent daylight savings Call duration and start time stored in call logs Call timer Name and identity (text) displayed at startup, Distinctive ringing based on calling and called number 10 user-downloadable ring tones Speed dialing, eight entries Configurable dial/numbering plan support Intercom* Group paging* NAT, Traversal, including STUN support DNS SRV and multiple A records for proxy lookup and proxy redundancy Syslog, debug, report generation, and event jogging Secure call encrypted voice communication support Built-in web server for, administration and configuration with multiple security levels Automated remote provisioning, multiple methods; up to 256 bit encryption (HTTP, HTTPS, Trivial File Transfer Protocol [TFTP]) Option to require administrator, password to reset unit to factory defaults * Feature requires support by call server

www: http://www.cisco.com/en/US/pr...

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