Packing: 10pcs/carton
Model NO.: VoIP Phone UTP1400
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 IP Phone_VoIP Phone_UTP1400 Introduction:
Professional VoIP Phone Based on SIP/IAX2: 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 the 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 codec: 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.168/165), 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); DMZ; Firewall; DNS relay
- Support automated provisioning through TFTP/TFP/HTTP for mass deployment
- Support management via web interfaces, keypad and telnet
Specifications of the VoIP Phone UTP1400
Professional VoIP Phone Based on SIP/IAX2: 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 the 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 codec: 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.168/165), 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); DMZ; Firewall; DNS relay
- Support automated provisioning through TFTP/TFP/HTTP for mass deployment
- Support management via web interfaces, keypad and telnet
Specifications of the VoIP Phone UTP1400
| Product | |
| Description | Professional VoIP Phone Based on SIP/IAX2 |
| Model | UTP1400 |
| Hardware Specification | |
| 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 with advanced acoustic echo cancellation |
| LCD Display | 3 lines dot-matrix graphic backlit LCD |
| Memory | SDRAM: 8M Flash Memory: 2M |
| 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 |
| Features & Benefits | |
| Standard | SIP v1 (RFC2543), v2 (RFC 3261) & correlative RFCs Support 2 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: 1.5W Active: 1.8W |
| Regulatory Compliance | CE, FCC part 15 class B, RoHS |
| Packages Contents | |
| UTP1400 IP Phone unit | 1 |
| Power Adapter | 1 |
| RJ45 Ethernet Cable | 1 |
| CD with User Manual | 1 |
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