SIP/IAX2 VoIP Adapter with 1 FXS, 1 PSTN Pass-Through, 2 RJ45: The UTA2001 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.
 

Cost-effective, easy-to-install and simple-to-use, the VoIP adapter UTA2001 converts standard telephones to IP-based networks with these benefits. Equipped with 1 FXS port, the UTA2001 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 (PPTP/L2TP), 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 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 VoIP adapter UTA2001.
 

The UTA2001 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.
 

Key Features of the VoIP Adapter UTA2001

- Support 2 SIP servers registering simultaneously; Compatible with IAX2 protocol (pending)
- SIP supports SIP domain, SIP authentication (none, basic, MD5), DNS name of server, Peer to Peer/ IP call
- 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.711 (A-law/u-law), G.729A/B, G.726
- 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); Adaptive Jitter Buffer
- DTMF relay: RFC2833, SIP info
- Support customized dial peer
- Call features: Caller ID (FSK/DTMF); 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
- Answer or make PSTN calls
- Support lifeline: auto-switch PSTN mode when internet fail or power off
- Support remote auto-provisioning through TFTP/FTP server for mass deployment
- Support device configuration via built-in IVR, Web browser or central configuration file
- Support T.38 FAX
- Support PPTP/L2TP VPN