Grandstream HT802 SIP ATA (1x WAN, 2x FXS)

Description

The Go-To Entry-Level ATA for Analogue-to-VoIP Conversion

The HT802 is a compact 2-port analogue telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection, and auto provisioning options enable users to take advantage of VoIP on analogue phones. Built on Grandstream’s market-leading SIP ATA/gateway technology with millions of units successfully deployed worldwide, the HT802 is the most cost-effective way to bring one or two analogue handsets or fax machines into a modern SIP-based VoIP environment.

Key Highlights

• 2× RJ11 FXS ports for connecting analogue phones or fax machines
• 2 SIP profiles supported across both FXS ports via a single 10/100 Mbps Ethernet port
• Ultra-compact desktop form factor — minimal desk footprint
• Ideal for both individual use and large-scale commercial IP voice deployments

HD Voice Quality & Broad Codec Support

The HT802 delivers exceptional voice quality across diverse application environments, with wideband HD audio ensuring noticeably clearer calls than standard narrowband telephony. A comprehensive codec selection guarantees compatibility across different VoIP platforms and SIP trunk providers.

• G.711 (Annex I/II), G.722, G.723.1, G.729A/B, G.726-32, iLBC, and Opus codec support with dynamic jitter buffer and advanced line echo cancellation
• Wideband HD voice (G.722) for superior call clarity
• T.38 compliant Fax-over-IP up to 14.4 kbps with automatic G.711 fax pass-through fallback
• Caller ID: Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID

Full Telephony Feature Set

Telephony features include caller ID display and block, call waiting, flash, blind and attended transfer, call forward, hold, do not disturb, and 3-way conferencing — the full feature set business users expect, delivered through analogue handsets already in use. Inter-port calling between the two FXS ports is also supported without a SIP server, making it useful as a standalone two-extension solution in simple environments.

• Caller ID, call waiting, call transfer, call forward, hold, and DND
• 3-way voice conferencing
• Message waiting indication and multi-language prompts
• Flexible dial plans for call routing control
• Failover SIP server automatically switches to a secondary server if the primary loses connection
• Inter-port calling between FXS ports without a SIP server

Security, Provisioning & UCM Integration

TLS and SRTP security encryption protect calls and accounts, with a unique security certificate per unit providing device-level protection. Automated provisioning options include TR-069 and XML config files for streamlined mass deployment. Zero Configuration provisioning is available when used with Grandstream’s UCM series of IP PBXs, making deployment in UCM environments particularly fast. The HT802 is fully compatible with GDMS for free centralised cloud management across multi-site deployments.

• AES encryption with unique per-unit security certificate
• TLS and SRTP for secure call transmission
• TR-069 and XML config file provisioning for automated mass deployment
• Zero Configuration provisioning with Grandstream UCM IP PBX series
• GDMS compatible for free centralised cloud management
• Web UI, SSH, and Telnet for local management

 

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