---
product_id: 50395851
title: "Grandstream HT-812 2FXS Gateway, Black"
brand: "grandstream"
price: "S$117"
currency: SGD
in_stock: false
reviews_count: 8
url: https://www.desertcart.sg/products/50395851-grandstream-ht-812-2fxs-gateway-black
store_origin: SG
region: Singapore
---

# Dual Gigabit Ethernet ports 2 FXS ports for analog phones TLS & SRTP call encryption Grandstream HT-812 2FXS Gateway, Black

**Brand:** grandstream
**Price:** S$117
**Availability:** ❌ Out of Stock

## Summary

> 📞 Stay connected, secure, and ahead with Grandstream HT-812 — your analog gateway to the future!

## Quick Answers

- **What is this?** Grandstream HT-812 2FXS Gateway, Black by grandstream
- **How much does it cost?** S$117 with free shipping
- **Is it available?** Currently out of stock
- **Where can I buy it?** [www.desertcart.sg](https://www.desertcart.sg/products/50395851-grandstream-ht-812-2fxs-gateway-black)

## 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 Call Security:** Advanced TLS and SRTP encryption protect your calls and credentials from eavesdropping.
- • **Trusted Global Reliability:** Millions of units deployed worldwide prove its unmatched voice quality and durability.
- • **Built-in NAT Router Convenience:** Simplify your setup with an integrated NAT router, eliminating extra hardware clutter.
- • **Blazing Fast Network Performance:** Dual Gigabit ports ensure ultra-reliable, high-speed data routing up to 100Mbps.
- • **Seamless Analog-to-VoIP Integration:** Connect your legacy phones effortlessly to modern VoIP networks with 2 FXS ports.

## Overview

The Grandstream HT-812 2FXS Gateway is a professional-grade VoIP adapter designed to bridge traditional analog phones with modern IP networks. Featuring 2 FXS ports, dual Gigabit Ethernet ports, and a built-in NAT router capable of routing speeds up to 100Mbps, it delivers exceptional voice quality and network performance. Enhanced with TLS and SRTP encryption, it ensures secure communications, making it ideal for home and office environments transitioning to VoIP technology.

## Description

Product Description The HT812 provides powerful VoIP technology and routing capabilities for home and office environments, enabling users to successfully connect their analog devices with a manageable and powerful VoIP network. This powerful ATA is based on Grandstream market-leading SIP ATA/gateway technology and has been successfully used with millions of units worldwide. It offers exceptional voice quality in various applications and environments. The HT812 features 2 user-friendly FXS ports, an integrated gigabit nat-router, state-of-the-art encryption with a security certificate per unit. Box Contains Grandstream HT812 2FXS Gateway

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+ }
Review: Setting up takes some work. Call quality is fine. - Bought this HT512 to replace a Linksys pap2t which had died. The Grandstream shipped from USA and arrived with a USA power supply so first thing needed was a UK to USA mains adapter. Also included was a network cable. There was no user guide included (but this can be downloaded from the Grandstream website) and no phone tails to convert RJ11 to UK telephone socket. Luckily I could use the old Linksys tails. The HT512 has a WAN port and a LAN port which do different things. The WAN port is what you normally expect and is used to connect to an existing LAN (like on the |Linksys). The LAN port is a built in router/dhcp server etc. I was replacing an existing device so the WAN port is what I needed (not LAN). I used the web interface to set up the HT512. For setting up the 2 telephones, you need to get the settings from your SIP provider. Mine (Sipgate) had generic instructions for the Grandstream HT series. These showed all the settings on a single config screen, however this HT512 had the settings spread over 3 different screens. However, after finding where everything should go, I plugged the phone in and fired it it with a test call. It worked first time. In summary, the HS512 had a few shortcomings (no manual, USA power supply, no UK tails) and required a fair bit of detective work to set up but as a replacement for the Linksys pap2t, it is ideal. Even if the built in router/dchp server (LAN port) is not required.

## 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

## Technical Specifications

| Specification | Value |
|---------------|-------|
| ASIN | B01M2ZJQAF |
| Best Sellers Rank | 84,890 in Computers & Accessories ( See Top 100 in Computers & Accessories ) 637 in Routers |
| Guaranteed software updates until | unknown |
| Item model number | HT-812 |
| Language | Dutch |
| Manufacturer | Grandstream |
| Product Dimensions | 22.86 x 33.02 x 7.11 cm; 290 g |

## 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

![Grandstream HT-812 2FXS Gateway, Black - Image 1](https://m.media-amazon.com/images/I/61IbQl+Lp+L.jpg)
![Grandstream HT-812 2FXS Gateway, Black - Image 2](https://m.media-amazon.com/images/I/61YTEysgAuL.jpg)
![Grandstream HT-812 2FXS Gateway, Black - Image 3](https://m.media-amazon.com/images/I/61tYGZjAhIL.jpg)

## Questions & Answers

**Q: Are you able to supply plugs/adaptors for use in the eu and/or us with this item?**
A: Either, I believe.

**Q: Can this work with Cisco phone system?**
A: Not clear what you mean by "Cisco phone system". The Grandstream HT-812 is an ATA (Analogue Telephone Adapter). You connect it to your internet router via ethernet cable. You plug in two phones or similar (eg CO ports on a PABX). It provides Voice over IP service from the likes of SIPgate. It's configured via a web interface.

**Q: Can you connect dialup modems to this? i would like to simulate a dialup isp for an old computer.**
A: I am not sure that anyone would have attempted to use a dial-up modem over VOIP but there has to be a first time however unlikely. The HT-812 supports fax so I can see no reason why it would not work with an acoustic modem so I think I would go ahead and try. There is a single channel version which is cheaper so would provide a cheaper option. If you do try, do post the answer in a review because I would be interested in the outcome!

**Q: What is the difference between the ht812 & the ht802?**
A: The HT812 has an integrated Gigabit NAT router and the HT802 does not. You don't need to use the integrated Gigabit NAT router, it can de disabled and it will work in the exact same way as the HT802.

## Customer Reviews

### ⭐⭐⭐⭐⭐ 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+ }

### ⭐⭐⭐⭐ Setting up takes some work. Call quality is fine.
*by A***N on 8 September 2023*

Bought this HT512 to replace a Linksys pap2t which had died. The Grandstream shipped from USA and arrived with a USA power supply so first thing needed was a UK to USA mains adapter. Also included was a network cable. There was no user guide included (but this can be downloaded from the Grandstream website) and no phone tails to convert RJ11 to UK telephone socket. Luckily I could use the old Linksys tails. The HT512 has a WAN port and a LAN port which do different things. The WAN port is what you normally expect and is used to connect to an existing LAN (like on the |Linksys). The LAN port is a built in router/dhcp server etc. I was replacing an existing device so the WAN port is what I needed (not LAN). I used the web interface to set up the HT512. For setting up the 2 telephones, you need to get the settings from your SIP provider. Mine (Sipgate) had generic instructions for the Grandstream HT series. These showed all the settings on a single config screen, however this HT512 had the settings spread over 3 different screens. However, after finding where everything should go, I plugged the phone in and fired it it with a test call. It worked first time. In summary, the HS512 had a few shortcomings (no manual, USA power supply, no UK tails) and required a fair bit of detective work to set up but as a replacement for the Linksys pap2t, it is ideal. Even if the built in router/dchp server (LAN port) is not required.

### ⭐⭐⭐⭐⭐ Provided introduction to VOIP
*by C***S on 29 January 2022*

The imminent arrival of FTTP prompted the investigation of VOIP to allow the ditching of the increasingly expensive landline which is now only used by ageing relatives and neighbours who have not moved to mobile. Since I do not intend to make outgoing calls, Sipgate provides a low cost (free), local number to which I can later migrate my BT number if it works well (at additional cost). The HT-812 allows the use of my existing analogue phones to assess the performance/reliability of VOIP as an alternative to the existing copper landline so I can use the money saved to pay for fast FTTP. Configuration is somewhat daunting due to the huge number of settings in the configuration GUI but only a few need changing. After updating the firmware to the latest version (see online manual) there are even more options available but most VOIP providers do supply settings for this series of adapters as well as the SIP codes, passwords etc. I also made these additional changes to make my phones sound like BT phones rater than US ones. In Advance Settings:- System ring cadence: c=400/200-400/2000; Dial tone: f1=350@-19,f2=450@-22,c=0/0; You will need one (or two) RJ11 (male) to BT (female) adapters in order to plug in your existing phones. For traditional phones these need to be masters including the ring capacitor. For modern DECT base stations the capacitor does not seem to be required. For my experiment I had to resort to adding some old ADSL filters which contain the ring capacitor, to my adapter cables. So to the results. The HT812 works perfectly. Audio quality is better than the landline but otherwise I can see no difference. I will have to see what the reliability of the VOIP service is but for the very low number of calls it will probably not matter. I will be migrating my BT number and then will enjoy not having to pay the line rental costs which will soon recoup the cost of the HT812! Do remember that migrating your number will cease the existing copper line and all services including ADSL/FTTC. If this appears too complex a process then it might be better to pay for Digital Voice (the same thing but configured by your ISP), but for my use case the VOIP gateway is an attractive option.

## 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)

---

## Why Shop on Desertcart?

- 🛒 **Trusted by 1.3+ Million Shoppers** — Serving international shoppers since 2016
- 🌍 **Shop Globally** — Access 737+ million products across 21 categories
- 💰 **No Hidden Fees** — All customs, duties, and taxes included in the price
- 🔄 **15-Day Free Returns** — Hassle-free returns (30 days for PRO members)
- 🔒 **Secure Payments** — Trusted payment options with buyer protection
- ⭐ **TrustPilot Rated 4.5/5** — Based on 8,000+ happy customer reviews

**Shop now:** [https://www.desertcart.sg/products/50395851-grandstream-ht-812-2fxs-gateway-black](https://www.desertcart.sg/products/50395851-grandstream-ht-812-2fxs-gateway-black)

---

*Product available on Desertcart Singapore*
*Store origin: SG*
*Last updated: 2026-03-12*