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Help CenterVoIP FXO Gateway With 2 Channels HT-322
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Brand: Etross or OEM Brand
Module: HT-322
Description:
The HT-322 is designed as a compact, high performance, and low cost FXO Gateway. The FXO detection is optimized to avoid the hold up of the PSTN line when the other party is disconnected. This has been one of the key issue in the design of FXO gateway. The incoming PSTN Caller ID is also transmitted to the VoIP user for more user friendly operation. The HT-322 is a full featured FXO gateway and is designed for easy installation and configuration. It is an ideal solution for VoIP to PSTN termination in both SME and SOHO environment.
Key Features:
1.Open Standard VoIP Protocols (ITU H.323 V4 and IETF SIP V2)
2.Single or Multiple Server Registrations
3.Two 10/100 Ethernet circuits connect to the LAN and an additional device
4.Two FXO ports for PSTN terminations
5.Speech quality ensured by QoS at the Ethernet and IP layers and comprehensive jitter buffer
6.Line Echo Cancellation
7.VLAN and QoS support
8.NAT Transversal and Router functions
9.Voice prompts, HTTP Web, Auto Provision support for configuration and updates
Highly stable embedded Linux operating system in high performance ARM 9 Processor
Supported Standards:
1.ITU: H.323 V4, H.225, H.235, H.245, H.450
2.RFC 1889 - RTP/RTCP
3.RFC 2327 – SDP
4.RFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
5.RFC 2976 – SIP INFO Method
6.RFC 3261 – SIP
7.RFC 3264 – Offer/Answer model with SDP
8.RFC 3515 – SIP REFER Method
9.RFC 3842 – A Message Summary and Message Waiting Indicator
10.RFC 3489 – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
11.RFC 3891 – SIP “Replaces” Header
12.RFC 3892 – SIP Referred-By Mechanism
13.draft-ietf-sipping-cc-transfer-04 – Session Initiation Protocol Call Control - Transfer
14.Codec: G.711 (A/µ law), G.729A/B, G.723.1
15.DTMF: RFC 2833, In-band DTMF, SIP INFO
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