2 FXS Ports VoIP-Gateway

2 FXS ports VOIP Gateway ,SIP Phone A VoIP FXS Gateway / Analog Telephone Adapter (ATA) is a telephone extension to the IP network. It offers a traditional telephone line (PSTN) interface to an analog telephone, PBX line extension, or a fax machine. Its WAN port interface allows access to the IP network in order to offer voice and fax services. It is a great way for turning a traditional PBX to access the low cost VoIP services and for deploying VoIP service by a ISP. An additional Ethernet allows broadband connection by the existing PC without buying additional network equipment. HT-922 (referred as “the device”) offers the features mentioned above and is an ideal low cost solution for SME and SOHO IP telephony application. Product overview2 FXS ports VOIP Gateway . Support dual protocols: ITU-H.323 V4 and IETF SIP V2 . Provide 2 RJ-11 ports for a POTS phone or a PBX's trunk line . Provide 2 RJ-45 Ethernet Ports . Support call forward/transfer/hold, phone book . Support G.711 A/μ law, G.729A/B, G.723.1 Codecs. Support QoS, NAT transversal and router function . Support VAD, CNG, ECHighlights Built-in H.323 V4 and SIP V2 Protocols Low cost Hight performance router Caller ID Auto provisioning and firmware updates Encryption transversal Multiple languages Key Features Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2) Two 10/100 Ethernet for WAN / LAN connections Peer-to-Peer IP Calls Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer Line Echo Cancellation VLAN and QoS support NAT Transversal and Router functions Voice prompts, HTTP Web, Auto Provision support for configuration and updates Highly stable embedded Linux operating system in high performance ARM 9 Processor Basic Feature 2 RJ-11 FXS ports for traditional phone set or PBX's trunk line LEDs for Power, Ready, Status, WAN, PC, FXS Call Forward, Call Hold, Call Transfer Dial Plan Caller IDEnhanced Features Dynamic selection of codec Advanced jitter buffer Automatic traversal of NAT and firewall VLAN / Qos Router Echo cancellation for Speakerphone Comfort noise generation (CNG) Voice activity detection (VAD) Auto provisioning (requires auto provisioning server) On line firmware upgrade Multi-language support: English and Chinese Supported Standards ITU: H.323 V4, H.225, H.235, H.245, H.450 RFC 1889 - RTP/RTCP RFC 2327 – SDP RFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals RFC 2976 – SIP INFO Method RFC 3261 – SIP RFC 3264 – Offer/Answer model with SDP RFC 3515 – SIP REFER Method RFC 3842 – A Message Summary and Message Waiting Indicator RFC 3489 – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) RFC 3891 – SIP “Replaces” Header RFC 3892 – SIP Referred-By Mechanism draft-ietf-sipping-cc-transfer-04 – Session Initiation Protocol Call Control - Transfer Codec: G.711 (A/μ law), G.729A/B, G.723.1 DTMF: RFC 2833, In-band DTMF, SIP INFO Install the device as shown in the connection diagram below.