Witam,
Mam do sprzedania telefon VoIP Linksys IP Phone SPA922-EU. Sprzęt jest w 100% sprawny.

Kod Producenta: SPA922-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)
Zasilanie: 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
Jack 2.5mm na słuchawki

Brak zasilacza w koplecie

Informacje dodatkowe:

Zaawansowany telefon IP z portem PoE oraz obsługa 1 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 SPA922

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/prod/collateral/voicesw/ps6788/phones/ps10033/ps10036/data_sheet_c78-[zasłonięte]365-00.html

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