---
product_id: 104920153
title: "Grandstream Hybrid ATA with FXS and FXO Ports (HT813)"
brand: "grandstream"
price: "€ 160.02"
currency: EUR
in_stock: true
reviews_count: 13
category: "Grandstream"
url: https://www.desertcart.hr/products/104920153-grandstream-hybrid-ata-with-fxs-and-fxo-ports-ht813
store_origin: HR
region: Croatia
---

# 2 GigE NAT router ports 1 FXS port for analog phone 1 FXO port for PSTN backup Grandstream Hybrid ATA with FXS and FXO Ports (HT813)

**Brand:** grandstream
**Price:** € 160.02
**Availability:** ✅ In Stock

## Summary

> 📡 Stay connected, stay ahead — the hybrid VoIP gateway for the modern pro.

## Quick Answers

- **What is this?** Grandstream Hybrid ATA with FXS and FXO Ports (HT813) by grandstream
- **How much does it cost?** € 160.02 with free shipping
- **Is it available?** Yes, in stock and ready to ship
- **Where can I buy it?** [www.desertcart.hr](https://www.desertcart.hr/products/104920153-grandstream-hybrid-ata-with-fxs-and-fxo-ports-ht813)

## 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

- • **Reliable PSTN Backup Lifeline:** 1 FXO port ensures uninterrupted calls by integrating traditional phone lines as failover.
- • **Advanced Security & Encryption:** Supports TLS and SRTP protocols to keep your communications private and protected.
- • **High-Speed Dual Gigabit Ethernet:** 2 GigE ports deliver blazing fast, secure network connectivity for smooth VoIP calls.
- • **Seamless Analog-to-VoIP Transition:** Convert your legacy phone effortlessly with 1 FXS port supporting analog devices.
- • **Unconditional Call Forwarding Magic:** Forward PSTN calls to any SIP address flawlessly, ensuring you never miss a call—wherever you are.

## Overview

The Grandstream HT813 Hybrid ATA combines 1 FXS port for analog phone support and 1 FXO port for PSTN line backup, enhanced by dual Gigabit Ethernet ports. Designed for seamless analog-to-VoIP conversion, it offers HD voice quality, robust security with TLS/SRTP encryption, and flexible call forwarding options. Ideal for professionals seeking reliable, secure, and versatile VoIP integration with legacy phone systems.

## Description

The HT813 is an analog telephone adapter that features 1 analog telephone FXS port and 1 PSTN line FXO port in order to offer backup lifeline support using a PSTN line. The integration of a FXO and FXS port enables this hybrid ATA to support remote calling to and from the PSTN line. For added flexibility, the FXS port extends VoIP service to one analog device. Users can convert their analog technology to VoIP thanks to the HT813's ultra-compact size, HD voice quality, advanced VoIP functionality, high-end security protection and multiple auto provisioning options. These advanced features also allow service providers to offer high quality IP service to customers looking to upgrade to VoIPp>

Review: Great unit, the "Unconditional Call Forward to VoIP" option is great. - I am originally from a country where phone calls from the US are over 0.5$ per minute. The local phone call rates in that country are only a couple of US cents per minute if calling a cell phone from a land line (landline to landline is even cheaper). The FXO port on the Grandstream HT813 is what makes it possible to save a lot if calling that country from the US a lot. This unit is a good tradeoff between being easy to setup and having a lot of configuration options. You may spend a bit more time configuring the unit but will have more control over its configuration. The notable features is the encryption support (TLS and SRTP secure connection) and the support of the option to use pulse mode phones on the FXS line (this cannot be done with Cisco SPA112 and SPA122). The only two alternatives to this device in this price range with FXO port worth looking at are the Obi110 and Obi212 but these units do not provide as many lower level configuration options as Grandstream HT813 and the older Obi110 does not seem to support secure connection. There is also a Cisco SPA232D-G1 but it has sound quality echo problems so I wouldn't use it. The one feature of HT813 which deserves a stand alone mention is the option called "Unconditional Call Forward to VoIP" where one can specify a SIP address to forward all incoming calls from PSTN to VoIP over the FXO port. If this option is configured, the SIP phone at the SIP address specified will start ringing after a desired number of rings on the PSTN line upon an incoming call. The SIP phone rings as if it was just another phone on the PSTN line and the person calling the PSTN number has no way of telling this is going on because there is not change in the dial tone etc. This feature is convenient because occasionally my relatives may need to call me in the US from their local cell phones so they would just call their landline in a foreign country to be transferred to the VoIP. I have configured the HT813 so that the forward to my SIP address begins after 10 rings, this way my relatives in a foreign country can still use the landline as usual without me having to answer their local phone calls (as most people wouldn't wait for 10 rings before they hang up). I don't know why but the SIP forward is only active for 3 - 4 rings after that the SIP call gets terminated and the PSTN line just keeps ringing as it used to until after the caller hangs up etc. I couldn't figure out where the 3-4 rings limitation comes from, maybe some settings but I see this more as an advantage because in most cases a call is answered within 4 rings. UPDATE Aug 23, 2022: I could never set up this unit properly so that one could dial different VIOP phone numbers from a foreign country's PSTN line though this device (that is when calling from a cell phone in a foreign country to a landline in a foreign country, said landline being connected to HT813 transferring to a VOIP line). The problem I was having is that the tone dialing would not always work correctly. I ended up setting up the unit so that it would automatically do the "Unconditional Call Forward to VoIP" and call my VOIP line as described above. Since for my application this approach was actually less confusing for my relative abroad to call me. But for some people this would be an issue as one wouldn't be able to dial different VOIP phone numbers if needed. There are a lot of settings in the menu related to how the dialing is interpreted based on the region and PSTN format but I am not familiar with those to determine if the dialing issue may be fixed by adjusting these settings. Also, the 3-4 ring limit mentioned above ".. the SIP forward is only active for 3 - 4 rings ..." is determined by a parameter in the menu and can be adjusted.
Review: Excellent product, very easy to setup and works reliably. - This is a niche product for those who works with (or dabbles) in Voice over IP. In my scenario, I have an Asterisk server setup with digital SIP trunks, but needed to bring in a good old fashioned PSTN/POTS line into the system. For this setup, one would need a device with an FXO interface. The HT813 unit comes with one FXS and one FXO ports, which is what I needed. The POTS line goes into the FXO and one can hook up a regular landline phone to the FXS if needed. The HT813 is very easy to setup for Asterisk and works reliably. The unit also has key features that makes it a suitable for different telco standards. For example, detecting a hung up line can be voltage change (US) or hang up tone (Asia). It comes with support for various international standards for caller ID so you can deploy it nearly worldwide. I would recommend this unit to those who has a need to bring in just one POTS line into their VoIP setup.

## Features

- Grandstream 1 FXS, 1 FXO, 2 GigE NAT Router HT813

## Technical Specifications

| Specification | Value |
|---------------|-------|
| ASIN | B07GY4WWP3 |
| Batteries | CR123A batteries required. |
| Best Sellers Rank | #63,357 in Office Products ( See Top 100 in Office Products ) #3 in VoIP Telephone Routers |
| Customer Reviews | 4.4 4.4 out of 5 stars (247) |
| Date First Available | August 29, 2018 |
| Is Discontinued By Manufacturer | No |
| Item Weight | 11.6 ounces |
| Item model number | HT813 |
| Language | English, English, English, English, English |
| Manufacturer | Grandstream |
| National Stock Number | 0 |
| Product Dimensions | 3.56 x 1.14 x 5.14 inches |

## Product Details

- **Brand:** Grandstream
- **Color:** black
- **Compatible Devices:** Laptop, Personal Computer, Smartphone, Tablet
- **Connectivity Technology:** RJ11
- **Frequency Band Class:** Single-Band
- **Included Components:** 1
- **Model Name:** HT813
- **Recommended Uses For Product:** bridging analog and VoIP systems
- **Special Feature:** Not_Performance_Used
- **Wireless Communication Standard:** 802.11a

## Images

![Grandstream Hybrid ATA with FXS and FXO Ports (HT813) - Image 1](https://m.media-amazon.com/images/I/61mjKGbGK5S.jpg)

## Questions & Answers

**Q: We have a poe switch and planning to buy 12 grandstream phones. can this provide 12 concurrent calls on a single line?**
A: You can only have one call active on a POTS line at any given time.

**Q: Does this product convert an analog pots line to a gateway that i can use with freepbx? we have a pots line but would like to setup ip phones.**
A: Yes,

I am using it to take my Xfinity POTS lines (Analog) and connect to my office FreePBX system.

**Q: Can this handle an fxs and an fxo with two lines (i.e. two pots phone numbers)?**
A: no only one input for both

**Q: Can I use this to make my voip service compatible with my anlaog lines in my home?**
A: Yea, as long as it's compatible with your voip provider, you'll need to check with them first

## Customer Reviews

### ⭐⭐⭐⭐⭐ Great unit, the "Unconditional Call Forward to VoIP" option is great.
*by B***T on September 3, 2021*

I am originally from a country where phone calls from the US are over 0.5$ per minute. The local phone call rates in that country are only a couple of US cents per minute if calling a cell phone from a land line (landline to landline is even cheaper). The FXO port on the Grandstream HT813 is what makes it possible to save a lot if calling that country from the US a lot. This unit is a good tradeoff between being easy to setup and having a lot of configuration options. You may spend a bit more time configuring the unit but will have more control over its configuration. The notable features is the encryption support (TLS and SRTP secure connection) and the support of the option to use pulse mode phones on the FXS line (this cannot be done with Cisco SPA112 and SPA122). The only two alternatives to this device in this price range with FXO port worth looking at are the Obi110 and Obi212 but these units do not provide as many lower level configuration options as Grandstream HT813 and the older Obi110 does not seem to support secure connection. There is also a Cisco SPA232D-G1 but it has sound quality echo problems so I wouldn't use it. The one feature of HT813 which deserves a stand alone mention is the option called "Unconditional Call Forward to VoIP" where one can specify a SIP address to forward all incoming calls from PSTN to VoIP over the FXO port. If this option is configured, the SIP phone at the SIP address specified will start ringing after a desired number of rings on the PSTN line upon an incoming call. The SIP phone rings as if it was just another phone on the PSTN line and the person calling the PSTN number has no way of telling this is going on because there is not change in the dial tone etc. This feature is convenient because occasionally my relatives may need to call me in the US from their local cell phones so they would just call their landline in a foreign country to be transferred to the VoIP. I have configured the HT813 so that the forward to my SIP address begins after 10 rings, this way my relatives in a foreign country can still use the landline as usual without me having to answer their local phone calls (as most people wouldn't wait for 10 rings before they hang up). I don't know why but the SIP forward is only active for 3 - 4 rings after that the SIP call gets terminated and the PSTN line just keeps ringing as it used to until after the caller hangs up etc. I couldn't figure out where the 3-4 rings limitation comes from, maybe some settings but I see this more as an advantage because in most cases a call is answered within 4 rings. UPDATE Aug 23, 2022: I could never set up this unit properly so that one could dial different VIOP phone numbers from a foreign country's PSTN line though this device (that is when calling from a cell phone in a foreign country to a landline in a foreign country, said landline being connected to HT813 transferring to a VOIP line). The problem I was having is that the tone dialing would not always work correctly. I ended up setting up the unit so that it would automatically do the "Unconditional Call Forward to VoIP" and call my VOIP line as described above. Since for my application this approach was actually less confusing for my relative abroad to call me. But for some people this would be an issue as one wouldn't be able to dial different VOIP phone numbers if needed. There are a lot of settings in the menu related to how the dialing is interpreted based on the region and PSTN format but I am not familiar with those to determine if the dialing issue may be fixed by adjusting these settings. Also, the 3-4 ring limit mentioned above ".. the SIP forward is only active for 3 - 4 rings ..." is determined by a parameter in the menu and can be adjusted.

### ⭐⭐⭐⭐⭐ Excellent product, very easy to setup and works reliably.
*by M***. on April 17, 2025*

This is a niche product for those who works with (or dabbles) in Voice over IP. In my scenario, I have an Asterisk server setup with digital SIP trunks, but needed to bring in a good old fashioned PSTN/POTS line into the system. For this setup, one would need a device with an FXO interface. The HT813 unit comes with one FXS and one FXO ports, which is what I needed. The POTS line goes into the FXO and one can hook up a regular landline phone to the FXS if needed. The HT813 is very easy to setup for Asterisk and works reliably. The unit also has key features that makes it a suitable for different telco standards. For example, detecting a hung up line can be voltage change (US) or hang up tone (Asia). It comes with support for various international standards for caller ID so you can deploy it nearly worldwide. I would recommend this unit to those who has a need to bring in just one POTS line into their VoIP setup.

### ⭐⭐⭐⭐⭐ FXO and FXS port for Asterisk
*by B***R on August 1, 2025*

We didn't need the FXS port, only the FXO as the device we had didn't natively terminate SIP. It's a little PITA to configure initialy, but once locked in, it works well.

## Frequently Bought Together

- Grandstream Hybrid ATA with FXS and FXO Ports (HT813)
- Grandstream HandyTone 801 Single-Port Analog Telephone Adapter (HT801)
- Grandstream 2-FXS Port Analog Telephone Adapter (HT802)

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*Product available on Desertcart Croatia*
*Store origin: HR*
*Last updated: 2026-02-27*