Telefoni

Yealink SIP-W52P i SIP-W52H

 

Bežični SIP telefon

Yealink SIP-W52P je bežični SIP telefonski sistem dizajniran za mala i srednje velika, kao i kućna okruženja koja traže štedljivo i proširivo rešenje za SIP mobilnu komunikaciju.

Kombinovanjem pogodnosti bežične komunikacije i bogatstvo poslovnih funkcionalnosti koje nudi IP telefonija, korisniku je omogućena sloboda kretanja, glasovna komunikacija visokog kvaliteta, uz istovremeno obavljanje više zadataka. Profesionalne mogućnosti kao što su intercom, transfer poziva, preusmeravanje poziva, konferencija sa tri sagovornika, PoE napajanje, automatsko javljanje itd. su samo neke od mogućnosti koje novi sistem nudi.

Ovaj sistem je testiran i izvrsno sarađuje sa velikim brojem VoIP platformi kao što su Broadsoft, Asterisk, 3CX, i mnoge druge, a podržava brzu i jednostavnu konfiguraciju.

DECT tehnologija

CAT-iq2.0 se fokusira na visok kvalitet zvuka putem VoIPa (wideband), na manjem bandwidth-u, uz DECT GAP kompatibilnost.

Fleksibilno VoIP rešenje za svako okruženje
  • Visok HD kvalitet zvuka uz korišćenje wideband tehnologije
  • Do 4 istovremena izlazna poziva
  • Do 5 DECT slušalica po bazi
  • Do 5 VoIP korisničkih računa
  • 1.8" displej u boji sa intuitivnim korisničkim interfejsom
  • vreme razgovora: 11 sati, do 120 sati na čekanju (standby)
  • Integrisan PoE (Class 1)
  • Montaža na sto ili na zid
 
Video Features

Video codec: H.264 and H.263
Image codec: JPEG, GIF, PNG, BMP
Video capacity: up to D1 (720x480)@30fps
Video call format: CIF/QCIF
Bandwidth selection: 128kbps~1Mbps
Frame rate selection: 10~30fps
Adaptive bandwidth adjustment
I-frame adjustable
Picture-in-Picture (PIP)
Full screen for remote side
Video control of local side
TV output with PAL/NTSC format
Door phone and IP camera application

Audio Features

HD voice: HD codec, HD handset, HD speaker
Wideband codec: G.722
Narrowband codec: G.711(A/μ), G.723.1, G.729AB
DTMF: In-band, Out-of-band(RFC 2833) and SIP INFO
Full-duplex hands-free speakerphone with AEC
Voice activity detection
Comfort noise generation
Adaptive jitter buffers
Packet loss concealment

Phone Features

4 VoIP accounts, Video/Voice call
20 one-touch soft DSS keys, Speed dial
Dial/answer call type selection (video or voice)
Call forward, Call waiting, Call transfer, Call hold
Call manager, Mute, Redial, Auto answer
DND, Caller ID display, Call history, Call statistics
Voice mail
Local 3-way audio conference
Direct IP call without SIP proxy
Phonebook with contact picture,
Group manager, Black list
Picture dial, XML/LDAP remote phonebook
New message and missed call notification
Volume control, Ring tone selection
Wall paper, Video/Photos screen saver
Video/photo/wallpaper/ringtone manager
Set date time manually or automatically
National language selection
Backlight time selection
Icon-driven menu

IP-PBX and BroadSoft Features

Busy lamp field (BLF), BLF list
Message waiting indicator (MWI)
Intercom, Music on hold
Call park, Call pickup
Anonymous call, Anonymous call rejection
DND & forward synchronization
Dial Plan, Dial-now

Management

Configuration: browser/phone/auto-provision
Auto provision via FTP/TFTP/HTTP/HTTPS
for mass deploy
Auto-provision with PnP
TR069 Protocol optional
BroadSoft device management
Zero-sp-touch
Reset to factory, Reboot
Package tracing export, System log

Physical Features

TI DaVinci dual-core chipset, Resistive touch screen
7" digital TFT-LCD with 800x480 pixels resolution
Rotatable CMOS sensor camera with 2M pixels
128MB flash and 256MB DDR2 memory
27 keys including 4 soft keys
6 feature keys: Mute/Camera/Phonebook/
Transfer/Redial/Hands-free
2xLEDs for power and status indication
2xRJ45 Ethernet 10/100M ports
1xUSB2.0 port, 1xSD card slot
PSTN optional, 2.5mm headset port
A/V out: RCA style stereo audio and
composite video output
Power adapter: AC 100~240V input and
DC 5V/3A output
Power over Ethernet (PoE) optional:
IEEE 802.3af, Class 0
Power consumption: 4~10W
Net weight: 1.2Kg
Dimension: 286x89x45mm
Operating humidity: 10~95%
Storage temperature: up to 60°C

Network and Security Features

SIP v1 (RFC2543), v2 (RFC3261)
NAT transverse: STUN mode
Proxy mode and peer-to-peer SIP link mode
IP assignment: static/DHCP/PPPoE
HTTP/HTTPS web server
Time and date synchronization using SNTP
UDP/TCP/DNS-SRV(RFC 3263)
QoS: 802.1p/Q tagging (VLAN),Layer 3 ToS, and DSCP
SRTP for voice and video
Transport Layer Security (TLS)
HTTPS certificate manager
AES encryption for configuration file
Digest authentication using MD5/MD5-sess
Admin/user mode