4 порта FXS адаптера VOIP

HT-842R 4 FXS ports VOIP AdapterProduct DescriptionThe HT-842R is designed as a compact, high performance, and low cost Analog Terminal Adaptor (FXS Gateway). It comes with 4 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications. The HT-842R is a full featured FXS gateway and is designed for easy installation and configuration. It supports the two most widely used Open VoIP Standards (SIP and H. 323). This allows the HT-842R to interoperate seamlessly with softswitches or IP PBXs made by various vendors. Its high performance offers toll quality voice, flexible networking, and feature-rich call functions. It is an ideal low cost solution for SME environment where multiple lines are required. Hardware SpecificationARM9E Processor for high performanceDSP for voice codec and voice processingTwo 100M Based Ethernet ports in comply with IEEE 802.3 for both LAN and PCconnection.LEDs for Ethernet port statusFour FXS portsEthernet BridgeSoftware SpecificationLINUX OSBuilt-in HTTP for accessing internal parametersPPPoE dial-upNAT Broadband Router functionsDHCP ClientDHCP ServerFirmware On-line upgradePhone BookMemory DialCaller IDMultiple Language SupportWith Accounting Function Key Features Open Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2) Single or Multiple Server Registrations Peer-to-Peer IP Calls Two 10/100 Ethernet for WAN / LAN connections 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 ProcessorBasic Features Four RJ-11 FXS ports for traditional phone sets or PBX's trunk lines LEDs for Power, Ready, Status, WAN, PC, FXS ports 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 ChineseSupported 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