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UTP1400

Professional IP Phone Based on SIP/IAX2

Overview
Stylish and functional in design, the VoIP phone UTP1400, broadly interoperable with SIP/IAX2 platforms and VoIP hardware from major third party vendors, is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large scale IP centrex deployment.

The VoIP phone UTP1400 features a full-duplex speakerphone with advanced acoustic echo cancellation, dot-matrix graphic backlit LCD, additional features including 3-way conferencing, call transfer (blind/attended), call forward, call waiting, DND, Voicemail, SMS, customized dial peer, 3 soft keys, as well as DHCP (client/server), NAT traversal (STUN), VLAN (voice VLAN/data VLAN), QoS with diffserv, VPN (L2TP).
 
By utilizing the cutting-edge quality of service, echo cancellation, comfort noisy generation and voice compensation technology, the VoIP phone UTP1400 can effortlessly provides the excellent voice quality. Meanwhile, the dual 10M/100Mbps auto-sensing Ethernet ports on the IP Phone allow users to install in an existing network location without interfering with desktop PC network connections. The VoIP phone UTP1400 also provides easy configuration thru manual operation (phone keypad and web interfaces) or personalized automated provisioning via central configuration file for mass deployment.
 
Key Features of VoIP Phone UTP1400
■ Support 2 SIP lines
■ SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
■ Compatible with IAX2 protocol
■ 3-line dot-matrix graphic backlit LCD
■ Dual 10/100Mbps Ethernet ports (switched/routed)
■ DHCP (client/server), Static IP, PPPoE for xDSL
■ Full-duplex speakerphone with advanced acoustic echo cancellation
■ Support codecs: G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.711(A-law/µ-law), 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 comprehensive customized dial peer
■ Support NAT Traversal (STUN); VLAN (voice VLAN/ data VLAN); QoS with diffserv; VPN (L2TP/UDP tunnel); DMZ; Firewall; DNS relay
■ Support automated provisioning through TFTP/TFP/HTTP for mass deployment
■ Support management via web interfaces, keypad and telnet
 
Features & Benefits of VoIP Phone UTP1400
Standards
SIP v1 (RFC2543, v2 (RFC3261) & correlative RFCs; Support IAX2 protocol
Support 2 SIP lines
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 UTP1400
WAN Port
(for connecting to internet)
1 X 10/100Mpbs RJ45
LAN Port
(for connecting to PC)
1 X 10/100Mpbs RJ45
NAT/Router
Yes
DHCP
Client/Server
Headset Jack
No
Speaker
Full-duplex Speakerphone
Function Keys
14 dedicated function keys (MWI, Phonebook, Hold, Transfer, Speakerphone, Redial, Mute, Call history, Menu, volume control, etc.)
3 soft keys
4 navigation keys
LCD Display
3-line dot-matrix graphic backlit LCD

 

 

 

 

Copyright © Uni-Ta Technology Co. (Uk), Ltd.
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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