VoIP Forum


Subject: full range of comsumer & enterprise ip phones
unitatech
Newbie
Rank: 1



UID 2044
Digest Posts 0
Credits 0
Posts 13
Reading Access 10
Status Offline
Post at Aug-29,2008 06:48 Profile | P.M.
full range of comsumer & enterprise ip phones

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 http://www.uni-ta.com.cn 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 http://www.uni-ta.com.cn/Products.asp?Id=33

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 http://www.uni-ta.com.cn/Products.asp?id=48&opid=1

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 http://www.uni-ta.com.cn/Products.asp?id=51&opid=1

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 http://www.uni-ta.com.cn/Products.asp?id=53&opid=9

More product information is available at http://www.uni-ta.com.cn/Prclass.asp


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



Top


All times are GMT, the time now is Nov-20,2008 14:38

    Powered by © 2001-2006 VoIP Forum.
Processed in 0.246779 second(s), 6 queries

Clear Cookies - Contact Us - VoIP Forum - Archiver - WAP - Multi-Directory