2 FXO VOIP gateway

2FXO PORTS VOIP GATWAY HT-322 Key FeaturesOpen Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)Single or Multiple Server RegistrationsTwo 10/100 Ethernet circuits connect to the LAN and an additional deviceTwo FXO ports for PSTN terminationsSpeech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter bufferLine Echo CancellationVLAN and QoS supportNAT Transversal and Router functionsVoice prompts, HTTP Web, Auto Provision support for configuration and updatesHighly stable embedded Linux operating system in high performance ARM 9 ProcessorBasic Features 2 RJ-11 FXO ports for PSTN lines or PBX's extensionsLEDs for Power, Ready, Status, WAN, PC, FXO ports Call forward from PSTN to VoIP and VoIP to PSTN Dial in mode or dial out mode onlyDial PlanPassword protection for both PSTN dial in or dial outRetransmit PSTN Caller ID to VoIP terminalEnhanced Features Dynamic selection of codecAdvanced jitter bufferAutomatic traversal of NAT and firewallVLAN / QosRouterEcho cancellation for SpeakerphoneComfort noise generation (CNG)Voice activity detection (VAD)Auto provisioning (requires auto provisioning server)On line firmware upgradeMulti-language support: English and ChineseSupported Standards ITU: H.323 V4, H.225, H.235, H.245, H.450RFC 1889 - RTP/RTCPRFC 2327 – SDPRFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony SignalsRFC 2976 – SIP INFO MethodRFC 3261 – SIPRFC 3264 – Offer/Answer model with SDPRFC 3515 – SIP REFER MethodRFC 3842 – A Message Summary and Message Waiting IndicatorRFC 3489 – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)RFC 3891 – SIP “Replaces” HeaderRFC 3892 – SIP Referred-By Mechanismdraft-ietf-sipping-cc-transfer-04 – Session Initiation Protocol Call Control - TransferCodec: G.711 (A/µ law), G.729A/B, G.723.1DTMF: RFC 2833, In-band DTMF, SIP INFO