---
product_id: 155703752
title: "HT812 2 FXS Analog Telephone Adapter VoIP FAX Gateway WAN LAN"
brand: "grandstream"
price: "835.10 DT"
currency: TND
in_stock: true
reviews_count: 6
category: "Grandstream"
url: https://www.desertcart.tn/products/155703752-ht812-2-fxs-analog-telephone-adapter-voip-fax-gateway-wan
store_origin: TN
region: Tunisia
---

# Dual Gigabit WAN/LAN ports 2 FXS ports for analog devices TLS & SRTP call encryption HT812 2 FXS Analog Telephone Adapter VoIP FAX Gateway WAN LAN

**Brand:** grandstream
**Price:** 835.10 DT
**Availability:** ✅ In Stock

## Summary

> 📡 Future-proof your office calls with rock-solid VoIP power!

## Quick Answers

- **What is this?** HT812 2 FXS Analog Telephone Adapter VoIP FAX Gateway WAN LAN by grandstream
- **How much does it cost?** 835.10 DT with free shipping
- **Is it available?** Yes, in stock and ready to ship
- **Where can I buy it?** [www.desertcart.tn](https://www.desertcart.tn/products/155703752-ht812-2-fxs-analog-telephone-adapter-voip-fax-gateway-wan)

## Best For

- grandstream enthusiasts

## Why This Product

- Trusted grandstream brand quality
- Free international shipping included
- Worldwide delivery with tracking
- 15-day hassle-free returns

## Key Features

- • **Rock-Solid Security:** Advanced TLS and SRTP encryption protect your calls and credentials from eavesdropping and hacking.
- • **Reliable Failover & Fax Support:** Automatic SIP server failover keeps calls flowing, plus T.38 Fax-over-IP ensures legacy fax compatibility.
- • **Blazing Fast Network Performance:** Dual Gigabit ports and built-in NAT router deliver up to 100Mbps routing speeds for uninterrupted connectivity.
- • **Effortless Large-Scale Deployment:** Automated provisioning via TR-069 and XML config files means zero-hassle setup and management at scale.
- • **Seamless Analog-to-VoIP Integration:** Connect your legacy phones effortlessly to modern VoIP networks with 2 dedicated FXS ports.

## Overview

The Grandstream HT812 is a professional-grade analog telephone adapter and VoIP gateway featuring 2 FXS ports, dual Gigabit WAN/LAN interfaces, and a built-in NAT router capable of 100Mbps routing speeds. It supports secure TLS and SRTP encryption, automated provisioning for easy deployment, and reliable failover SIP server switching. Ideal for seamlessly integrating traditional analog phones into modern VoIP networks with exceptional voice quality and robust security.

## Description

The HT812 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built using Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments. The HT812 comes with 2 easy-to-use FXS ports, an integrated Gigabit NAT router, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

Review: Not worthy - After few hours in working, not getting incoming calls
Review: Compatible with Zen's UK "BT digital voice" switchover - Bought as I'm a zen.co.uk customer, and have UniFi throughout the house (I don't use the Fritzbox). With the BT/UK "digital voice" switchover in full-swing, I wanted a means to keep our old land line alive. This worked perfectly. Configuration hasn't been utterly straight-forward, but I've finally found the following settings work (for Zen customers): - Ensure your NAT/firewall is forwarding the required ports through to the Grandstream (5060/UDP from 212.23.7.228, and 20000-30000/UDP from 62.3.88.0/28 and 62.3.88.16/28) - Primary SIP server: voip.zen.co.uk - Secondary SIP server: voip2.zen.co.uk - Prefer Primary SIP server: No - Outbound proxy: Blank/None - Allow DHCP Option 120: No - SIP Transport: UDP - NAT traversal: No - DNS Mode: A Record - DNS SRV: No - Fallback Timer: 60 - Use Request Routing ID in SIP INVITE Header: No - Sip Registration: Yes - Outgoing Call without Registration: Yes - Register Expiration: 5 - Enable SIP OPTIONS/NOTIFY keep alive: No - Local SIP port: 5060 - Local RTP port: 5004 - Random SIP port: No - Random RTP port: No - Enable RTCP: Yes - Hold Target Before Refer: Yes - Refer-To Use Target: No - Remove OBP: No - Support SIP Instance ID: Yes - Validate Incoming SIP Message: No - Use Privacy Header: No - Use P-Preferred-Identity Header: Yes - Use P-Access-Network-Info Header: Yes - Use P-Emergency-Info Header: Yes - Use P-Asserted-Identity Header: No - SIP REGISTER Contact Header Uses: LAN - Caller ID Fetch Order: Auto - DTMF: RFC2833 - SLIC: UK - Caller ID Scheme: SIN 227 - Dial Plan: (You may want to customise) ... { 999 | 10[015] | 11[129] | 11[68]xx | 1[45]7[1-2] | 8001111 | 845464x | [1235789]xxxxxxxxx | 23456789]xxxxx | 1410[1235789]xxxxxxxxx | x+ }

## Features

- Supports 2 SIP profiles through 2 FXS ports and dual Gigabit ports
- Includes a built-in NAT router which can handle routing speeds up to 100MBps
- TLS and SRTP security encryption technology to protect calls and accounts
- Automated provisioning options include TR-069 and XML config files

## Technical Specifications

| Specification | Value |
|---------------|-------|
| ASIN | B01M2ZJQAF |
| Batteries | 1 AAAA batteries required. |
| Batteries Included | No |
| Batteries Required | No |
| Best Sellers Rank | #324,790 in Electronics ( See Top 100 in Electronics ) #1 in VoIP Routers |
| Brand | Grandstream |
| Colour Screen | No |
| Compatible Devices | Laptop, Personal Computer, Smart Television, Smartphone, Tablet |
| Connector Type | Ethernet |
| Country of Origin | China |
| Customer Reviews | 4.3 4.3 out of 5 stars (470) |
| Data Transfer Rate | 100 Megabits Per Second |
| Date First Available | 17 November 2016 |
| Does it contain liquid? | No |
| Has Auto Focus | No |
| Imported By | Pure Life office system |
| Includes Rechargeable Battery | No |
| Includes remote | No |
| Item Dimensions LxWxH | 22.9 x 33 x 7.1 Centimeters |
| Item Weight | 290 g |
| Item model number | HT-812 |
| Manufacturer | Grandstream |
| Model | HT-812 |
| Model Year | 2016 |
| Mounting Hardware | Analog telephone adaptor, network cable, power adaptor and stand |
| Net Quantity | 1.00 Count |
| Number of Ports | 2 |
| Operating System | RouterOS |
| Product Dimensions | 22.86 x 33.02 x 7.11 cm; 290 g |
| Programmable Buttons | No |
| Special Features | Automated provisioning options include TR-069 and XML config files, Failover SIP server automatically switches to secondary server if main server loses connection, Includes a built-in NAT router which can handle routing speeds up to 100MBps, Supports 2 SIP profiles through 2 FXS ports and dual Gigabit ports, Supports 3-way voice conferencing, Supports T.38 Fax for creating Fax-over-IP, Supports a … |
| Standing screen display size | 5 Inches |
| Voltage | 28 Volts |

## Product Details

- **Brand:** Grandstream
- **Model Name:** HT812
- **Special Feature:** WPS
- **Compatible Devices:** Personal Computer
- **Recommended Uses For Product:** Security
- **Included Components:** 1
- **Connectivity Technology:** Ethernet
- **Item Weight:** 0.29 Kilograms
- **Number of Ports:** 2
- **Control Method:** Remote

## Images

![HT812 2 FXS Analog Telephone Adapter VoIP FAX Gateway WAN LAN - Image 1](https://m.media-amazon.com/images/I/61IbQl+Lp+L.jpg)
![HT812 2 FXS Analog Telephone Adapter VoIP FAX Gateway WAN LAN - Image 2](https://m.media-amazon.com/images/I/61YTEysgAuL.jpg)
![HT812 2 FXS Analog Telephone Adapter VoIP FAX Gateway WAN LAN - Image 3](https://m.media-amazon.com/images/I/61tYGZjAhIL.jpg)

## Questions & Answers

**Q: Can i use this as a sip server to have my land line as a trunk and use smartphones with sip client to make calls??**
A: You need an IP PBX for connecting your PSTN line, HT812 is a 2 port FXS Gateway which is used for connecting analog Phones & work analog phone into IP Platform.

**Q: Can this product be used for connecting pstn (pots) to an asterisk server for integrating ivrs?**
A: Yes, you can

## Customer Reviews

### ⭐⭐ Not worthy
*by A***R on 29 December 2018*

After few hours in working, not getting incoming calls

### ⭐⭐⭐⭐⭐ Compatible with Zen's UK "BT digital voice" switchover
*by T***E on 30 May 2023*

Bought as I'm a zen.co.uk customer, and have UniFi throughout the house (I don't use the Fritzbox). With the BT/UK "digital voice" switchover in full-swing, I wanted a means to keep our old land line alive. This worked perfectly. Configuration hasn't been utterly straight-forward, but I've finally found the following settings work (for Zen customers): - Ensure your NAT/firewall is forwarding the required ports through to the Grandstream (5060/UDP from 212.23.7.228, and 20000-30000/UDP from 62.3.88.0/28 and 62.3.88.16/28) - Primary SIP server: voip.zen.co.uk - Secondary SIP server: voip2.zen.co.uk - Prefer Primary SIP server: No - Outbound proxy: Blank/None - Allow DHCP Option 120: No - SIP Transport: UDP - NAT traversal: No - DNS Mode: A Record - DNS SRV: No - Fallback Timer: 60 - Use Request Routing ID in SIP INVITE Header: No - Sip Registration: Yes - Outgoing Call without Registration: Yes - Register Expiration: 5 - Enable SIP OPTIONS/NOTIFY keep alive: No - Local SIP port: 5060 - Local RTP port: 5004 - Random SIP port: No - Random RTP port: No - Enable RTCP: Yes - Hold Target Before Refer: Yes - Refer-To Use Target: No - Remove OBP: No - Support SIP Instance ID: Yes - Validate Incoming SIP Message: No - Use Privacy Header: No - Use P-Preferred-Identity Header: Yes - Use P-Access-Network-Info Header: Yes - Use P-Emergency-Info Header: Yes - Use P-Asserted-Identity Header: No - SIP REGISTER Contact Header Uses: LAN - Caller ID Fetch Order: Auto - DTMF: RFC2833 - SLIC: UK - Caller ID Scheme: SIN 227 - Dial Plan: (You may want to customise) ... { 999 | 10[015] | 11[129] | 11[68]xx | 1[45]7[1-2] | <0=44>8001111 | <0=44>845464x | <0=44>[1235789]xxxxxxxxx | <=441638>23456789]xxxxx | 1410[1235789]xxxxxxxxx | x+ }

### ⭐⭐⭐⭐⭐ Very impressive- lots of setup details though
*by L***X on 23 October 2025*

We switched from Ooma to a SIP provider (VOIP.ms, but there are lots of others). We got a few more features and are saving a little money. This was a little tricky to setup (it has lots of settings). Most of the settings are fine if left at default, but it's capable of encrypting traffic and a whole bunch of other stuff and there are settings for every possible feature. In fact, there are five or six options to backup/restore the configuration (seems like any of them will work, though).

## Frequently Bought Together

- Grandstream Powerful 2-Port ATA with Gigabit NAT Router
- Grandstream GS-HT814 4 Port Ata with 4 Fxs Ports and Gigabit NAT Router Voip Phone and Device, Black
- Grandstream 2-FXS Port Analog Telephone Adapter (HT802)

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*Product available on Desertcart Tunisia*
*Store origin: TN*
*Last updated: 2026-06-23*