3 Lines Business IP Phone with Large Graphic LCD

slsam
Joined: Wed 13 of Jan, 2010

3 Lines Business IP Phone with Large Graphic LCD

Posted:Wed 13 of Jan, 2010 (08:07 UTC)
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 & ATAs. Click http://www.uni-ta.com to learn more on our company and products now.


VoIP Phone UTP3000: 3 Lines Business VoIP Phone with Large graphic LCD

Key Features
- 3 lines indicators with individual SIP account profiles
- SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
- Compatible with IAX2 protocol
- Large 128X96 high resolution graphic backlit LCD (3.2”)
- RJ9 and 3.5MM headset jack
- 6 programmable keys, 3 context-sensitive soft keys, a 5-position navigation key, volume keys and predefined keys for voicemail, call transfer, call hold, mute, redial, speaker, phonebook, etc.
- Full-duplex speakerphone with advanced acoustic echo cancellation (96ms max filter length).
- Dual 10/100Mbps Ethernet ports (switched/routed) with integrated Power over Ethernet (802.3af)
- Support DHCP (client/server), Static IP, PPPoE for xDSL
- 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
- Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168/165), and AGC (Automatic Gain Control)
- DTMF relay: RFC2833, SIP info
- 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
- Call Logs: Incoming call, Outgoing call, Missed call (100 entries each); Phonebook: 500 entries
- Support NAT Traversal (STUN); VLAN (voice VLAN / data VLAN); QoS with diffserv; VPN (L2TP); SRTP security protocol; SNTP Client; DMZ; Firewall; DNS relay; Main DNS and secondary DNS server.
- Support auto-provisioning through TFTP/TFP/HTTP for mass deployment
- Support management via keypad, web interfaces and telnet
- Reversible base stand / wall mount

VoIP Phone UT1400: Professional VoIP Phone Based on SIP/IAX2

Key Features
- 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

Univox S800: Asterisk based Embedded IP PBX

- Equipped with 8 FXO/FXS ports
- Powered by Asterisk (open source)
- Intuitive Web-based GUI
- CPU: 400MHz Blackfin 533 Chip
- Fully support SIP & IAX2 protocols
- Unlimited SIP/IAX2 user extensions and 30 concurrent calls
- Integrated PSTN trunks, analog phone/FAX ports & SIP/IAX trunk options
- Flexible dial plan
- Excellent OSLEC (Open Source Line Echo Cancellation)
- Up to 8 FXO/FXS channels
- Support for a combination of FXO and FXS modules
- Built-in router ideal for small offices
- MMC/SD slot for external storage
- RS-232 port for console configuration
- Voice codec: G.729, G.711 a-law/ µ-law, G.726, GSM, Speex
- Rich IP PBX features: Voice Mail, Auto Attendant & IVR, Conference Bridge, Caller ID, Call Waiting/Park/Hold/Retrieve/Pickup/Forward/Transfer, Call Logs, Call Queues, Hunt/Ring Group, Music on Hold



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