| Packing: | 10pcs/carton |
|---|---|
| Model NO.: | VoIP Phone UTP3000 |
| Standard: | SIP/IAX2 |
| Productivity: | 10, 000 pcs/month |
| Shipment Terms: | Air freight/Courier/Sea Freight |
| Trademark: | Uni-Ta |
| Origin: | China |
| Usage: | Telephone |
| Type: | VoIP Products |
| Style: | Wired |
| Export Markets: | North America, South America, Eastern Europe, Southeast Asia, Africa, Oceania, Mid East, Eastern Asia, Western Europe |
Product Description Introduction of the VoIP Phone UTP3000
3 Lines Business VoIP Phone with Large LCD: 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 inch), 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 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 the 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 jack
- 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 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.168/165), and AGC (Automatic Gain Control)
- DTMF relay: RFC2833, SIP info
- 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
- Call Logs: Incoming call, Outgoing call, Missed call (100 entries each); Phonebook: 500 entries
- Support NAT Traversal (STUN); VLAN (voice VLAN / data VLAN); QoS with diffserv; VPN (L2TP); SRTP security protocol; SNTP Client; DMZ; 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
Specifications of the VoIP Phone UTP3000
3 Lines Business VoIP Phone with Large LCD: 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 inch), 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 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 the 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 jack
- 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 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.168/165), and AGC (Automatic Gain Control)
- DTMF relay: RFC2833, SIP info
- 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
- Call Logs: Incoming call, Outgoing call, Missed call (100 entries each); Phonebook: 500 entries
- Support NAT Traversal (STUN); VLAN (voice VLAN / data VLAN); QoS with diffserv; VPN (L2TP); SRTP security protocol; SNTP Client; DMZ; 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
Specifications of the VoIP Phone UTP3000
| Product | |
| Description | 3 Lines Business VoIP Phone with Large LCD |
| Model | UTP3000 |
| Hardware Specification | |
| WAN Port (for connecting to Internet) | 1 X 10/100Mpbs RJ45 port Power over Ethernet (802.3af) compliant |
| LAN Port (for connecting to PC) | 1 X 10/100Mpbs RJ45 port |
| Headset Jack | RJ9 and 3.5MM headset jack |
| Speaker | Full-duplex Speakerphone advanced acoustic echo cancellation |
| LCD Display | 128X96 high resolution graphic backlit LCD LCD Size: 3.2" |
| Memory | SDRAM: 16M Flash Memory: 4M |
| 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 |
| Features & Benefits | |
| Standard | SIP v1 (FRC2543), v2 (RFC3261) Support 3 SIP lines Support IAX2 protocol 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 line echo cancellation(LEC) Acoustic 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, auto-answer, join call, click to dial Customized dial peer 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 MWI: Message Waiting Indicator |
| Network and Management | |
| 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; QoS with diffserv; VPN (L2TP) ; SRTP security protocol; SNTP Client; DMZ; Firewall; DNS relay; Main DNS and secondary DNS server. |
| Operating Requirements | |
| Operating Temp. | 0~40 degree C |
| Storage Temp. | -25~60 degree C |
| Operating Humidity | 10~90% Non-condensing |
| Storage Humidity | 10~90% Non-condensing |
| Power Requirement | Input 100~240V AC, Output 5V DC 1A |
| Power Consumption | Idle: 2.5W Active: 2.8W |
| Regulatory Compliance | CE, FCC part 15 class B, RoHS |
| Packages Contents | |
| UTP3000 IP Phone unit | 1 |
| Power Adapter | 1 |
| RJ45 Ethernet Cable | 1 |
| CD with User Manual | 1 |
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