The UTA2011 is an Analog Telephone Adapter based on SIP & IAX2 standards, offers 1 FXS port to connect existing analog telephone or fax machine to IP-based data networks and 1 PSTN Pass-Through port, for toggling operations between SIP and PSTN networks.

 

Cost-effective, easy-to-install and simple-to-use, the UTA2011 converts standard telephones to IP-based networks with these benefits. Equipped with 1 FXS port, the UTA2011 can save installation cost and extend your past investment in telephones, video conferencing and speaker. With the features of dual 10/100Mbps auto-sensing Ethernet ports, DHCP (client/server), T.38 fax transmission over IP network, auto-provisioning through TFTP/FTP/HTTP server, compatible with VPN (L2TP/UDP Tunnel), VLAN (voice VLAN/data VLAN), QoS with diffserv, customized dial plan, caller ID display, call hold, call waiting, call transfer, call forward, DND, IP telephony service providers and enterprise users can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet or Local Area Network (LAN) by the UTA2011.

 

The UTA2011 is an ideal VoIP telephone adapter solution that will help VoIP service providers or resellers with all of the required attributes to grow and retain customers by delivering a robust and consistent VoIP solution

 

■ 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