SIP-T42S

Yealink SIP-T42S IP je komunikacijski alat za rad u dinamičnoj okolini uz koji ćete uživati u vrhunskoj kvaliteti zvuka i brojnim naprednim funkcionalnostima. U odnosu na prethodnika, model SIP-T42G, nudi fluidnije korisničko sučelje i općenito brži rad. Do 12 SIP korisničkih računa i dovoljan broj programabilnih tipki svakako će dodatno poboljšati produktivnost svakog korisnika ovog telefona. Ugrađena Yealink Optima HD tehnologija i širokopojasni Opus codec brinu za kristalno čistu komunikaciju, a dvostruki Gigabit Ethernet priključak će se pobrinuti za brz prijenos podataka ako kroz telefon spojite i osobno računalo. Novost u odnosu na prethodnika je i dodatak USB priključka, čime postaje moguće telefon dodatno proširiti spajanjem Bluetooth uređaja, Wi-Fi mrežom ili snimanjem razgovora na USB memoriju.

957,00 kn Cijena bez PDV. Kontaktirajte nas za partnerske cijene.
  • 2.7" 192x64 zaslon s pozadinskim osvjetljenjem
  • Podrška za Opus codec
  • USB 2.0 (uz dodatne module - spajanje na Bluetooth, Wi-Fi i snimanje razgovora)
  • Unificirani T4S Auto-P template
  • Unificirani T4S firmware
  • Do 12 SIP računa
  • Dva Gigabit Ethernet porta
  • PoE
  • Dizajn bez korištenja papirnih oznaka
  • Podrška za EHS i naglavne slušalice
  • Postolje s dva kuta nagiba
  • Mogućnost montaže na zid (uz dodatni nosač)
Phone Features

12 VoIP accounts
One-touch speed dial, redial
Call forward, call waiting
Call transfer, call hold
Call return, group listening
Mute, auto answer, DND
3-way conference call
Direct IP call without SIP proxy
Ring tone selection/import/delete
Hotline, emergency call
Set date time manually or automatically
Dial plan, XML Browser, Action URL/URI
RTCP-XR (RFC3611), VQ-RTCPXR (RFC6035)
USB port (2.0 compliant):
Bluetooth earphone through BT40,
Wi-Fi through WF40,
USB call recording through USB flash drive
Enhanced DSS key

Audio Features

HD voice: HD handset, HD speaker
Hearing aid compatible (HAC) handset
Wideband codec: Opus*, G.722
Narrowband codec: Opus*, G.711(A/µ), G.723.1, G.729AB, G.726, iLBC
DTMF: In-band, Out-of-band(RFC 2833) and SIP INFO
Full-duplex hands-free speakerphone with AEC
VAD, CNG, AEC, PLC, AJB, AGC

IP-PBX Features

Busy Lamp Field (BLF)
Bridged Line Apperance (BLA)
Anonymous call, anonymous call rejection
Hot-desking, voice mail
Flexible seating
Call park, call pickup
Executive and Assistant
Centralized call recording
Visual voice mail
Call recording

Directory

Local phonebook up to 1000 entries
Black list
XML/LDAP remote phonebook
Smart dialing
Phonebook search/import/export
Call history: dialed/received/missed/forwarded

Management

Configuration: browser/phone/auto-provision
Auto provision via FTP/TFTP/HTTP/HTTPS for mass deploy
Auto-provision with PnP
Zero-sp-touch, TR-069
Phone lock for personal privacy protection
Reset to factory, reboot
Package tracing export, system log

Display and Indicator

2.7" 192x64-pixel graphical LCD with backlight
LED for call and message waiting indication
Dual-color (red or green) illuminated LEDs for line status information
Intuitive user interface with icons and soft keys
Multilingual user interface
Caller ID with name and number
Power saving

Feature keys

6 line keys with LED
6 line keys can be programmed up to 15 paperless DSS keys (3-page view)
5 features keys: message, headset, mute, redial, hands-free speakerphone
4 context-sensitive “soft” keys
6 navigation keys
2 volume control keys
Illuminated mute key
Illuminated headset key
Illuminated hands-free speakerphone key

Interface

Dual-port Gigabit Ethernet
Power over Ethernet (IEEE 802.3af), Class 2
1 x USB port (2.0 compliant)
1 x RJ9 (4P4C) handset port
1 x RJ9 (4P4C) headset port
1 x RJ12 (6P6C) EHS port

Other Physical Features

Stand with 2 adjustable angles
Wall mountable
External Yealink AC adapter (optional):AC 100~240V input and DC 5V/1.2A output
Power consumption (PSU): 1.69-5.0W
Power consumption (PoE): 2.47-12.5W
Dimension (W*D*H*T): 212mm*189mm*175mm*54mm
Operating humidity: 10~95%
Operating temperature: -10~50°C

Network and Security

SIP v1 (RFC2543), v2 (RFC3261)
Call server redundancy supported
NAT traversal: STUN mode
Proxy mode and peer-to-peer SIP link mode
IP assignment: static/DHCP
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 DSCP
SRTP for voice
Transport Layer Security (TLS)
HTTPS certificate manager
AES encryption for configuration file
Digest authentication using MD5/MD5-sess
OpenVPN, IEEE802.1X
IPv6
LLDP/CDP/DHCP VLAN
ICE

Slični proizvodi

SIP-T46S

Zamjena za model SIP-T46G s podrškom za Opus codec i brojnim poboljšanjima

1.485,00 kn

VP-T49G

Revolucionarni telefon za video kolaboraciju

4.718,00 kn

SIP-T48S

Zamjena za model SIP-T48G s podrškom za Opus codec i brojnim poboljšanjima

2.013,00 kn