VoIP Phone Supports 2 SIP Lines

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 IP 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