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Product Details
VoIP Phone with Dual Ethernet Ports
| Packing: | 10PCS/CTN |
|---|---|
| Model NO.: | VoIP Phone UTP1200 |
| Standard: | SIP |
| Productivity: | 10, 000pcs/month |
| Trademark: | Uni-Ta |
| Origin: | China |
| Transportation: | Air Freight/Courier/ Sea Freight |
Entry Level VoIP Phone with 2 SIP Lines: Stylish and functional in design, the VoIP Phone UTP1200 is ideal for a residence or business using a hosted IP telephony service, an IP PBX, or a large scale IP centrex deployment. Based on open standards, the VoIP Phone UTP1200 is broadly interoperable with SIP platforms and VoIP hardware from major third party vendors. The remote automated provisioning feature also saves the hassle and expenses of managing, preloading and re-configuring customer premise equipment for mass deployment.
By utilizing the cutting-edge quality of service, echo cancellation, comfort noisy generation and voice compensation technology, the VoIP Phone UTP1200 can effortlessly provides the excellent voice quality. Meanwhile, the dual 10M/100Mbps auto-sensing Ethernet ports on the VoIP Phone allow users to install in an existing network location without interfering with desktop PC network connections.
The VoIP Phone UTP1200 features a full-duplex quality speakerphone with excellent voice delivery, with an ease to use on/off button, as well as buttons for MWI, conference call, call transfer, call hold, call history, redial, mute, menu, volume control etc., plus the 3 line backlit LCD display, dual 10M/100Mbps auto-sensing Ethernet ports (switched/routed), DHCP (client/server), and the utilizing of cutting-edge Digital Signal Processing, make the UTP1200 a very cost-effective, yet feature-rich VoIP Phone choice for any business.
Key Features of the VoIP Phone UTP1200
- Support SIP v1 (RFC2543), v2 (RFC3261)
- 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)
- Support DHCP (client/server), Static IP, PPPoE for xDSL
- 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
- DTMF relay: RFC2833, SIP info
- Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
- Call features: Voicemail, caller ID display or block, 3-way conferencing, call transfer (blind/attended), Call forward, Call hold, Call waiting, DND, Black List, Limited List, Call history, phonebook (500 entries), MWI
- Support comprehensive customized dial peer
- Support NAT Traversal (STUN), VLAN (voice VLAN / data VLAN); QoS with diffserv; SNTP Client; Firewall; Main DNS and secondary DNS server.
- Support auto-provisioning through TFTP/TFP/HTTP for mass deployment
- Support management via web interfaces, keypad and telnet
- 3 lines 7-segment backlit LCD display
Specifications of the VoIP Phone UTP1200
| Product | |
| Description | Entry Level IP Phone with 2 SIP Lines |
| Model | UTP1200 |
| Hardware | |
| WAN Port (for connecting to Internet) | 1 X 10/100Mpbs RJ45 port |
| LAN Port (for connecting to PC) | 1 X 10/100Mpbs RJ45 port |
| Speaker | Full-duplex Speakerphone |
| LCD Display | 3 lines 7-segment backlit LCD |
| Memory | SDRAM: 8M Flash Memory: 2M |
| Function Keys | 14 dedicated function keys (MWI, Phonebook, Conference, Hold, Transfer, Speakerphone, Redial, Mute, Call history, Menu, volume control, etc.) 4 navigation keys |
| Features & Benefits | |
| Standard | SIP v1 (FRC2543), v2 (RFC3261) Support 2 SIP Lines SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call |
| Voice Codec | G.711(A-law/ µ -law), G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.722 |
| 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, auto-answer, join call, click to dial Customized dial peer Caller ID display/block DND (do not disturb), Black List, Limited List 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 (voice VLAN / data VLAN); QoS with diffserv; SNTP Client; DMZ; Firewall; 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: 1.5W Active: 1.8W |
| Packages Contents | |
| UTP1200 IP Phone unit | 1 |
| Power Adapter | 1 |
| RJ45 Ethernet Cable | 1 |
| CD with User Manuals | 1 |
