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UTP3000

3 Lines Business VoIP Phone with Large Graphic LCD

Overview

The UTP3000 is the latest enhanced business VoIP phone of Uni-Ta. With sleek and elegant design, large 128X96 high resolution graphic backlit LCD (3.2”), RJ9 and 3.5MM headset jacks, 3 SIP-line keys, 6 programmable keys, 3 dynamic content-sensitive soft keys, a 5-position navigation key, including keys for voice mail, call transfer, call hold, mute, volume control, plus an excellent full-duplex speakerphone with advanced acoustic echo cancellation, make the UTP3000 a full-feature and high quality VoIP phone for enhancing office productivity.
 
The feature of dual auto-sensing switched/routed Ethernet ports allows the VoIP phone UTP3000 to share internet connection with computers, help to reduce cable clutter and expense. Integrated with IEEE 802.3af Power over Ethernet, the VoIP phone UTP3000 can receive power through Ethernet cable, eliminating the need for external power supply and allows easy deployment with centralized powering and backup. The VoIP phone UTP3000 also supports remote, zero-touch automated provisioning and upgrade from variety severs, such as FTP/TFTP/HTTP, good for mass deployment.
 
The VoIP phone UTP3000 is capable of handling peer-to-peer and SIP proxy / IP PBX registration, authenticated to interact with major IP PBXs / SIP Gateways / VoIP phones in the market. It’s the delivery platform for IP voice services that makes benefits from the internet telephony in business communication services.
 
Key Features of VoIP Phone UTP3000
■ 3 lines indicators with individual SIP account profiles
■ SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
■ Compatible with IAX2 protocol
■ Large 128X96 high resolution graphic backlit LCD (3.2”)
■ RJ9 and 3.5MM headset jacks
■ 6 programmable keys, 3 context-sensitive soft keys, a 5-position navigation key, volume keys and predefined keys for voicemail, call transfer, call hold, mute, redial, speaker, phonebook, etc.
■ Full-duplex speakerphone with advanced acoustic echo cancellation (96ms max filter length).
■ Dual 10/100Mbps Ethernet ports (switched/routed) with integrated Power over Ethernet (802.3af)
■ Support DHCP (client/server), Static IP, PPPoE for xDSL
■ Support codecs: G.711(A-law/µ-law), G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.722
■ Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
■ Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
■ Call features: voicemail, SMS, caller ID display or block, conference call, call Forward, call Transfer (blind or attended), call hold, call waiting, paging and intercom, call park/pickup, join call, click to dial, DND, black list, limited list, call history
■ Support NAT Traversal (STUN); VLAN; DMZ; QoS with diffserv; VPN (L2TP/UDP TUNNEL), SRTP security protocol; SNTP Client; Firewall; DNS relay; Main DNS and secondary DNS server.
■ Support auto-provisioning through TFTP/TFP/HTTP for mass deployment
■ Support management via keypad, web interfaces and telnet
■ Reversible base stand / wall mount
 
Features & Benefits of VoIP Phone UTP3000
Standards
SIP v1 (RFC2543, v2 (RFC3261) & correlative RFCs; Support IAX2 protocol
Support 3 SIP lines
SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
Compatible with Asterisk, Trixbox and other SIP/IAX platforms.
Voice Codec
G.711(A-law/µ-law), G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.722 (wideband)
Voice Standard
DTMF relay: RFC2833, SIP info
Auto Gain Control (AGC)
G.168/165 compliant 16ms echo cancellation
Auto Echo Cancellation (AEC)
Voice Activity Detection (VAD)
Comfort Noise Generation (CNG)
Adaptive Jitter Buffer
Call Features
Call waiting, call forward, call transfer (blind / attended), call hold, 3-way conference call, paging and intercom, call park/pickup, join call, click to dial, hotline
Customized dial peer
Caller ID display / block
DND (do not disturb), Black List, Limited List
Support Voicemail, SMS
Peer to Peer/ IP call
Call Logs: Incoming call, Outgoing call, Missed call (100 entries each)
Phonebook:500 entries
MWI: Message Waiting Indicator
Access Mode
DHCP (client/server), Static IP, PPPoE for xDSL
Management
Web, Keypad, Telnet management
Management with different account right
Auto-provisioning through TFTP/FTP/HTTP
Firmware upgrade through TFTP/ FTP
Configuration file download/upload
Support Syslog
Protocols
TCP/IP/UDP, DHCP, PPPoE, SNTP, STUN, MD5, DNS, RTP, RTCP, Telnet, HTTP, FTP, TFTP
Applications
NAT Traversal (STUN); VLAN; DMZ; QoS with diffserv; VPN (L2TP/UDP TUNNEL), SRTP security protocol; SNTP Client; Firewall; DNS relay; Main DNS and secondary DNS server.
 
Hardware Specifications of VoIP Phone UTP3000
WAN Port
(for connecting to internet)
1 X 10/100Mpbs RJ45
Power over Ethernet (802.3af) compliant
LAN Port
(for connecting to PC)
1 X 10/100Mpbs RJ45
NAT/Router
Yes
DHCP
Client/Server
Headset Jack
RJ9 and 3.5MM headset jack
Speakerphone
Full-duplex Speakerphone
Function Keys
10 dedicated function keys (Voicemail, Phonebook, Hold, Transfer, Speakerphone, Redial, Mute, volume control, RLS )
3 SIP-line indicators
3 context-sensitive soft keys
6 programmable keys
5 navigation keys
LCD Display
128X96 high resolution graphic backlit LCD
LCD Size: 3.2 inches
 

 

 

 

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Tel: +44 775 9845710  - sales & support 24/7 hrs Fax: +44 116 2352411 
E-mail:sales@uni-ta.co.uk Tech support: support@uni-ta.co.uk
Address: 1st Floor, Halford House, Charles Street, Leicester LE1 1HA, UK

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