analog-to-voip – Ringwood Digital Store

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

WooSheets Integration — Thu, 28 May 2026 17:52:57 +0000

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

Proudly sold by the Ringwood Digital Store.

Grandstream HT801 SIP ATA (1x WAN, 1x FXS)

WooSheets Integration — Thu, 28 May 2026 17:52:55 +0000

The Simplest Path from Analogue Phone to VoIP

The HT801 is a single 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 HT801 is the most streamlined and cost-effective way to bring a single analogue handset or fax machine onto a modern SIP-based VoIP platform.

Key Highlights

1× RJ11 FXS port for connecting a single analogue phone or fax machine
1 SIP profile supported via a single 10/100 Mbps auto-sensing 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 HT801 delivers exceptional voice quality across a wide range of application environments, with wideband codec support ensuring compatibility with virtually any VoIP platform or SIP trunk provider. Dynamic jitter buffer and advanced line echo cancellation ensure consistently clear audio regardless of network conditions.

G.711 (Annex I/II), G.723.1, G.729A/B, G.726, iLBC, and Opus codec support with dynamic jitter buffer and advanced line echo cancellation
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
Layer 2 (802.1Q VLAN) and Layer 3 (ToS, DiffServ, MPLS) QoS support for prioritised voice traffic

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 complete feature set business users expect, delivered through an analogue handset they already have.

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

Security, Provisioning & UCM Integration

TLS and SRTP security encryption protect calls and accounts, with a unique security certificate per unit providing device-level protection out of the box. Automated provisioning options include TR-069 and XML config files for streamlined mass deployment across large sites. Zero Configuration provisioning is available when used with Grandstream’s UCM series of IP PBXs, and full GDMS compatibility allows centralised management alongside other Grandstream devices at no additional cost.

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

Proudly sold by the Ringwood Digital Store.

Grandstream HT814 SIP ATA (1x LAN, 1x WAN, 4x FXS)

WooSheets Integration — Thu, 28 May 2026 17:52:53 +0000

Connect Up to 4 Analogue Phones to Any VoIP Network

The HT814 is an advanced 4-port VoIP gateway with 4 FXS ports and an integrated Gigabit NAT router, built upon Grandstream’s market-leading SIP ATA/gateway technology with millions of units successfully deployed worldwide. It’s the practical middle ground between the single-port HT813 and the 8-port HT818 — ideal for small offices or branch locations with up to four analogue desk phones or fax machines that need to remain operational on a VoIP platform. Each of the 4 FXS ports is colour-coded with LED status indicators, making it easy to identify port activity at a glance.

Key Highlights

4× RJ11 FXS ports for connecting analogue phones or fax machines
2 SIP profiles supported across all 4 FXS ports, with dual Gigabit Ethernet ports
Integrated Gigabit NAT router handling routing speeds up to 100 Mbps
Colour-coded FXS ports with LED status indicators for quick port identification
Built upon Grandstream’s market-leading SIP ATA/gateway technology

HD Voice Quality & Broad Codec Support

The HT814 features exceptional voice quality in various application environments, with wideband HD audio codec support delivering noticeably clearer calls than standard narrowband telephony. A broad codec range ensures compatibility across different VoIP platforms, SIP trunk providers, and IP PBX systems.

G.711 (Annex I/II), G.723.1, G.729A/B, G.726, iLBC, and Opus codec support with dynamic jitter buffer and advanced line echo cancellation
Wideband HD voice quality for superior call clarity
T.38 compliant Fax-over-IP up to 14.4kbps with automatic G.711 fax pass-through fallback
Layer 2 (802.1Q VLAN) and Layer 3 (ToS, DiffServ, MPLS) QoS support for prioritised voice traffic

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 — covering everything business users expect from a desk phone, delivered seamlessly through their existing analogue handsets.

Caller ID, call waiting, call transfer, call forward, hold, and DND
3-way voice conferencing per port
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

Security, Provisioning & UCM Integration

TLS and SRTP security encryption protect calls and accounts, with a unique security certificate per unit providing an additional layer of device-level protection. Automated provisioning options include TR-069 and XML config files for volume deployment. Zero Configuration provisioning is available when used with Grandstream’s UCM series of IP PBXs, making it especially quick to deploy in Grandstream UCM environments.

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 for standalone configuration

Proudly sold by the Ringwood Digital Store.

Grandstream HT818 SIP ATA (1x LAN, 1x WAN, 8x FXS)

WooSheets Integration — Thu, 28 May 2026 17:50:02 +0000

Connect Up to 8 Analogue Phones to Any VoIP Network

The HT818 is a powerful 8-port VoIP gateway with 8 FXS ports and an integrated Gigabit NAT router, built for users looking for a strong analogue-to-VoIP converter. Where FXO gateways bring PSTN lines into a VoIP system, the HT818 does the reverse — it connects existing analogue desk phones and fax machines to a SIP-based IP PBX or hosted VoIP platform, allowing businesses to preserve their investment in analogue handsets while moving to a modern telephony infrastructure.

Key Highlights

8× RJ11 FXS ports — connect up to 8 analogue phones or fax machines
2× 10/100/1000 Mbps Gigabit Ethernet ports with integrated Gigabit NAT router
2 SIP profiles supported across all 8 FXS ports
Grandstream’s market-leading SIP ATA/gateway technology with millions of units deployed worldwide

HD Voice Quality & Broad Codec Support

The HT818 carries exceptional voice quality in various application environments, with wideband HD codec support delivering noticeably clearer audio than standard narrowband telephony. A broad codec selection ensures compatibility across different VoIP platforms and network conditions.

G.711 Annex I/II, G.723.1, G.729A/B, G.726, iLBC, and Opus codec support
Wideband HD audio for superior voice clarity
T.38 compliant Fax-over-IP for reliable fax transmission across VoIP links
Layer 2 (802.1p/Q) and Layer 3 (ToS, DiffServ, MPLS) QoS support for prioritised voice traffic

Rich Telephony Features

Advanced telephony features include call transfer, call forward, call waiting, do not disturb, message waiting indication, multi-language prompts, and flexible dial plan support — covering the full feature set that business users expect from their desk phones, delivered seamlessly through the analogue handsets they already have.

Call transfer, call forward, call waiting, DND, and message waiting indication
3-way voice conferencing per port for in-office collaboration
Flexible dial plans for advanced call routing
Caller ID display and block support

Enterprise-Grade Security & Easy Management

Strong encryption with a unique security certificate per unit and automated provisioning for volume deployment and device management make the HT818 equally suitable for single-site and large multi-site deployments. Provisioning options include HTTP, HTTPS, SSH, TFTP, TR-069, and automated provisioning with AES encryption and syslog — integrating naturally into existing device management workflows.

AES encryption with unique per-unit security certificate
TLS and SRTP for secure call transmission
TR-069, HTTP/HTTPS, TFTP, and SSH provisioning
GDMS compatible for free centralised cloud management
Web UI for standalone configuration and management

Proudly sold by the Ringwood Digital Store.