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首页 > Products > 8-Fxsvoip Gateway (HT-882) (ATA)
8-Fxsvoip Gateway (HT-882) (ATA)
单价 $120.00 / Piece对比
询价 暂无
浏览 277
发货 Chinaguangdongshenzhen
库存 10Piece
过期 长期有效
更新 2020-08-04 08:50
 
详细信息
Product Name: 8-Fxsvoip Gateway (HT-882) (ATA) Model NO.: HT-882 Application: Enterprise Ports: 8 Firewall: Firewall Transfer Rate: 10/100Mbps Type: Wired VoIP Protocal: H. 323&Sip Transportation: by Air Trademark: DBL Specification: Package: 10PCS/Carton Origin: China Product Description 1. Support ITU H. 323 V4 and IETF SIP V22. Speech quality ensured by QoS3.8Channels of voip gateway and VLAN and QoS4. NAT Transversal and Route functionsThe DBL 8-FXS Port SIP/H. 323 speech quality ensured VoIP Gateway (HT-882) is designed as a compact, high performance, and low cost VoIP Analog Terminal Adaptor (FXS Gateway). It comes with 8 FXS ports to interface with traditional analog phone sets or PBX trunk lines for VoIP communications. The HT-882 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-882 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.Key FeaturesOpen Standard VoIP Protocols (ITU H. 323 V4 and IETF SIP V2)Single or Multiple Server RegistrationsPeer-to-Peer IP CallsTwo 10/100 Ethernet for WAN / LAN connectionsSpeech 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 FeaturesEight RJ-11 FXS ports for traditional phone sets or PBX's trunk linesLEDs for Power, Ready, Status, WAN, PC, FXS portsCall Forward, Call Hold, Call TransferDial PlanCaller IDEnhanced FeaturesDynamic 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 StandardsITU: 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