Uni-Ta Technology is a leading manufacturer of VoIP equipments in China. We design, manufacture, deliver and deploy optimizing Internet Telephony equipments, especially IP phones, analog phone adapters and PCI cards for Asterisk. Click VoIP Phone,Asterisk IP PBX,VoIP Gateway,Analog Phone Adapter|China Manufacturer to learn more on our company and products now.

Model UTP-2000: 5 Lines Enterprise IP Phone with PoE



Key Features
- Supports 5 SIP lines and 1 IAX line registering simultaneously
- Dual 10/100Mbps Ethernet ports (switched/routed) with integrated Power over Ethernet (802.3af)
- 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 codecs: G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.711(A-law/µ-law)
- 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
- 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 (by MAC address) 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
For more specs on this model, click VoIP Phone,Asterisk IP PBX,VoIP Gateway,Analog Phone Adapter|China Manufacturer

Model UTP-1600: Two Lines SIP/IAX IP Phone with Dual RJ45 Ports



Key Features
- Supports 2 SIP lines registering simultaneously; Compatible with IAX2 protocol
- Dual 10/100Mbps Ethernet ports (switched/routed)
- DHCP (client/server), Static IP, PPPoE for xDSL
- Full-duplex speakerphone with advanced acoustic echo cancellation
- Support codecs: G.723.1 (5.3k/6.3k), G.729A/B, G.726, G.711(A-law/µ-law)
- 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
- 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 (by MAC address) for mass deployment
- Support management via web interfaces, keypad and telnet
- 2 X 16 characters dot-matrix graphic LCD display
For more specs on this model, click VoIP Phone,Asterisk IP PBX,VoIP Gateway,Analog Phone Adapter|China Manufacturer

Model UTP-1300: Single Line SIP/IAX IP Phone with 2 RJ45 Ports



Key Features
- Support SIP RFC3261, RFC2543 and IAX2 protocol
- Dual 10/100Mbps Ethernet ports (switched/routed)
- DHCP (client/server), Static IP, PPPoE for xDSL
- Full-duplex speakerphone with advanced acoustic echo cancellation
- Support codecs: G.711(A-law/µ-law), G.729A/B
- 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: Voicemail (with indicator), 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)
- Support comprehensive customized dial plan
- Support NAT Traversal (STUN)
- Support automated provisioning through TFTP/TFP/HTTP (by MAC address) for mass deployment
- Support management via web interfaces, keypad and telnet
- 3 lines backlit semi-graphic LCD display
For more specs on this model, click VoIP Phone,Asterisk IP PBX,VoIP Gateway,Analog Phone Adapter|China Manufacturer

Model UTA-2011: SIP/IAX Analog Phone Adapter with 1 FXS, 1 PSTN Pass-Through, 2 RJ45



Key Features
- Support 2 SIP accounts registering simultaneously; Compatible with IAX2 protocol
- Dual 10/100Mbps Ethernet ports (switched/routed)
- DHCP (client/server), Static IP, PPPoE (for ADSL, Cable modem)
- Single FXS Port (for phone/fax machine connection)
- 1 PSTN Pass-Through Port (for PSTN connection)
- Support codec: G.729A/B, G.726, G.711(A-law/µ-law), iLBC
- 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)
- In-band, out-of-band DTMF relay, RFC2833, SIP info; Adaptive Jitter Buffer
- Support customized dial plan
- Caller ID with name/number
- Call features: Call Hold, Call Waiting, Call Forwarding (No answer/Busy/All), Call Transfer (blind/attended), Conference Call, Black List & Limited List, Do Not Disturb
- Hotline calling
- Support remote auto-provisioning through TFTP/FTP server (by MAC address)
- Support device configuration via built-in IVR, Web browser or central configuration file
- Support NAT Traversal (STUN); VLAN (voice VLAN/ data VLAN); QoS with diffserv; VPN (L2TP/UDP tunnel); DMZ; Firewall; DNS relay
For more specs on this model, click VoIP Phone,Asterisk IP PBX,VoIP Gateway,Analog Phone Adapter|China Manufacturer

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We focus on the manufacturing of VoIP equipments.
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E-mail: sales@uni-ta.com.cn