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IP Phone (UTP2000)
Product Description

5 Lines Business IP Phone with PoE: The UTP2000 is a 5-line enterprise VoIP phone compliant with SIP/IAX2 standards. This VoIP phone has a new stylish design. With the features of dual 10/100Mpbs auto-sensing Ethernet ports and integrated IEEE Power over Ethernet (802.3af) circuitry, the UTP2000 VoIP phone offers a choice of powering and cabling options to help reduce cabling expenses and cord clutter. The UTP2000 also supports automated provisioning through TFTP/TFP (by MAC address) for mass deployment, provides excellent voice clarity, as well as many more advanced features.

The UTP2000 VoIP phone features an excellent full-duplex speakerphone with advanced acoustic echo cancellation, together with 3 dynamic context-sensitive soft keys, 5 programmable keys and a 5-positiong navigation key, including keys for call transfer, call hold, mute, headset, phonebook and call history, plus the backlit graphic LCD display, make the equipment as an elegant VoIP phone solution for enhancing office productivity.

The UTP2000 VoIP phone is capable of handling peer-to-peer and SIP proxy / IP PBX registration, authenticated to interact with major IP PBX / SIP Gateway / VoIP phone in the market. Either for IP PBX system integrator or Internet Telephony Service Provider (ISTP), the UTP2000 is an ideal VoIP phone solution. 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 UTP2000
- Supports 5 SIP lines and 1 IAX line registering simultaneously
- SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
- Dual 10/100Mbps Ethernet ports (switched/routed) with integrated Power over Ethernet (802.3af)
- Support DHCP (client/server), Static IP, PPPoE for xDSL
- 3 context-sensitive soft keys, 5 programmable keys, a 5-position navigation key, volume keys and predefined keys for call transfer, call hold, mute, redial, speaker, phonebook, etc.
- 2.55mm Headset connector
- Full-duplex speakerphone with advanced acoustic echo cancellation
- Support codec: G. 711 (A-law/u-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: Caller ID display or block, conference call, call transfer (blind or attended), call hold, Call waiting, DND, Black List, Limited List, Call history, Voicemail, SMS
- Call logs: Incoming call, Outgoing call, Missed call (100 entries each); Phonebook: 500 entries
- Support comprehensive customized dial plan
- 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
- 3 lines X 16 characters backlit graphic LCD display
- Reversible base stand / wall mount


Specifications

Product
Description5-Line Enterprise IP Phone with PoE
ModelUTP2000
Hardware
WAN1 X 10/100Mpbs RJ45 port Power over Ethernet (802.3af) compliant
LAN1 X 10/100Mpbs RJ45 port
Headset Jack2.55mm headset connector
SpeakerFull-duplex Speakerphone
LCD Display3 lines X 16 characters backlit graphic LCD
MemorySDRAM: 128M
Flash Memory: 16M
Function Keys11 dedicated function keys (Hold, Transfer, Speakerphone, Redial, Mute, headset, Call Logs, Phonebook, Network, Directory, Esc)
3 context-sensitive soft keys
5 programmed keys
5 navigation keys
2 volume keys
Features & Benefits
StandardSIP 2.0 (RFC 3261) and correlative RFCs; Support IAX2 protocol
Support 5 SIP lines and 1 IAX line registering simultaneously
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 CodecG.711(A-law/ µ -law), G.723.1 (5.3k/6.3k), G .729A /B, G.726, G.722 (wideband)
Voice StandardAuto Gain Control (AGC)
G.168/165 compliant 16ms echo cancellation
Auto Echo Cancellation (AEC)
Voice Activity Detection (VAD)
Comfort Noise Generation (CNG)
In-band and Out-band DTMF relay (In-audio, RFC2833, SIP info)
Adaptive Jitter Buffer
Call FeaturesCall waiting, call forward, call transfer (blind / attended), call hold, 3-way conference call, paging and intercom, auto-answer
Customized dial plan
Caller ID display / block
DND (do not disturb), Black List, Limited List
Support Voicemail, SMS
Call Logs: Incoming call, Outgoing call, Missed call (100 entries each) Phonebook:500 entries
Network and Management
Access ModeDHCP (client/server), Static IP, PPPoE for xDSL
ManagementWeb, Keypad, Telnet
Auto-provisioning through TFTP/FTP/HTTP
Firmware upgrade through FTP, TFTP
Configuration file download/upload
Support Syslog
ProtocolsTCP/IP/UDP, DHCP, PPPoE, SNTP, STUN, MD5, DNS, RTP, RTCP, Telnet, HTTP, FTP, TFTP
ApplicationsNAT Traversal (STUN); VLAN; DMZ; QoS with diffserv; VPN (L2TP/UDP TUNNEL) , Open VPN/TLS, SRTP security protocol; SNTP Client; Firewall; DNS relay; Main DNS and secondary DNS server.
Operating Requirements
Operating Temp.0~40 degree C
Storage Temp.-25~60 degree C
Operating Humidity10~90% Non-condensing
Storage Humidity10~90% Non-condensing
Power RequirementInput 100~240V AC, Output 5V DC 1A
Power ConsumptionIdle: 2.0W Active: 2.3W
Regulatory ComplianceCE, FCC part 15 class B, RoHS
Packages Contents
UTP2000 IP Phone unit1
Power Adapter  1  
RJ45 Ethernet Cable1
CD with User Manual1
Model: VoIP Phone UTP2000
Standard: SIP, IAX2
Productivity: 10, 000pcs/month
Origin: China
Transportation: Air Freight/Courier/ Sea Freight
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