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Grandstream HT801v2

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=== Device details ===
=== Device details ===
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The HT801v2 is an entry-level model intended for residential and light SOHO (Small Office/Home Office) installations, capable of leveraging a single network link into two simultaneously usable analog phone ports. The device itself is very compact, measuring 10×10×2.95cm and weighing 114g, and utilizes a ubiquitous USB Micro-B port for AC power input (expecting 5VDC/1A)—allowing it to be powered by all but the most feeble of mobile phone chargers manufactured in the last decade. Despite its small form factor it provides a broad range of features, including:
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The HT801v2 is an easy-to-use, single-port Analog Telephone Adapter (ATA) designed for residential users, small offices, and large-scale commercial IP voice deployments.  
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* Support for two SIP profiles, compatible with both [[Order a DID Number|VoIP.ms DIDs]] and [[Sub Accounts|sub-accounts]], each mapped to a dedicated FXS (RJ11) phone port
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Featuring an ultra-compact form factor measuring 10×10×2.95cm and weighing 102g, the device provides a simple and cost-effective way to integrate a traditional analog telephone or fax machine into a modern VoIP environment. Despite its small size, the HT801v2 offers a comprehensive feature set, including:
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* Incoming and outgoing [[Caller ID]] with support for custom formatting and ringtones
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* A T-Base10/100 Ethernet (RJ45) port
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* Support for a single SIP profile through a dedicated FXS (RJ11) port, compatible with both [[Order a DID Number|VoIP.ms DIDs]] and [[Sub Accounts|sub-accounts]]
 +
* Incoming and outgoing [[Caller ID]] with support for a wide range of caller ID formats
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* A 10/100 Mbps auto-sensing Ethernet (RJ45) network port
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* Automatic SIP server failover to maintain service continuity should the primary server become unavailable
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* Advanced telephony features including call transfer, [[Call Forwarding|call forwarding]], call waiting, hold, do-not-disturb, message waiting indication, multilingual prompts, and flexible dialplans
* Optional [[Call Encryption - TLS/SRTP|TLS and SRTP encryption protocols]] to secure calls and accounts
* Optional [[Call Encryption - TLS/SRTP|TLS and SRTP encryption protocols]] to secure calls and accounts
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* [[Audio Conferencing|Three-way call conferencing]]
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* [[Virtual Fax|Fax over IP capability]] using T.38 fax relay with automatic fallback to G.711 pass-through
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* Call transfers, [[Call Forwarding|forwarding]], hold and do-not-disturb
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* SIP server failover to ensure your device stays connected to the service even if the default server goes offline
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* [[Virtual Fax|Fax over IP capability]] (using either T.38 or G.711 passthrough)
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<gallery caption="Grandstream HandyTone HT801v2 photos (click to expand)" heights="224" perrow="3" widths="224" style="margin: 1.5em auto; text-align: center; width: fit-content;">
<gallery caption="Grandstream HandyTone HT801v2 photos (click to expand)" heights="224" perrow="3" widths="224" style="margin: 1.5em auto; text-align: center; width: fit-content;">

Revision as of 19:52, 18 June 2026

This guide is intended to assist with the process of configuring a Grandstream HandyTone HT801v2 for use with VoIP.ms services. This device and its single-line counterpart (HT801v2)—along with their "v1" predecessors—have been the most widely-used ATAs by our customers in recent years, according to internal network analytics.


Contents

Device details

The HT801v2 is an easy-to-use, single-port Analog Telephone Adapter (ATA) designed for residential users, small offices, and large-scale commercial IP voice deployments. Featuring an ultra-compact form factor measuring 10×10×2.95cm and weighing 102g, the device provides a simple and cost-effective way to integrate a traditional analog telephone or fax machine into a modern VoIP environment. Despite its small size, the HT801v2 offers a comprehensive feature set, including:

  • Support for a single SIP profile through a dedicated FXS (RJ11) port, compatible with both VoIP.ms DIDs and sub-accounts
  • Incoming and outgoing Caller ID with support for a wide range of caller ID formats
  • A 10/100 Mbps auto-sensing Ethernet (RJ45) network port
  • Automatic SIP server failover to maintain service continuity should the primary server become unavailable
  • Advanced telephony features including call transfer, call forwarding, call waiting, hold, do-not-disturb, message waiting indication, multilingual prompts, and flexible dialplans
  • Optional TLS and SRTP encryption protocols to secure calls and accounts
  • Fax over IP capability using T.38 fax relay with automatic fallback to G.711 pass-through
Grandstream HandyTone HT801v2 photos (click to expand)

Front view

Top view, showing status LEDs for power, network connection, and in-use indicator for both analog phone ports

Rear view with Ethernet and dual phone ports, USB Micro-B power input and pinhole reset

Configuring your device

FXS Port 1 General Settings > Account Registration

Prerequisites

  • Grandstream HT801v2
  • Latest firmware installed (Grandstream site) (We you use firmware version 1.0.11.4 or later.)
  • VoIP.ms account with a configured DID and a sub-account
  • Internet access via Ethernet
  • Phone or fax device connected to FXS Port 1

Logging into the web interface

Connect the HT801v2 to your network and power it on.
  • Pick up the connected phone, dial: * * * 0 2 (Star Star Star 0 2)

The ATA will read its IP address. Open a browser and go to that IP (e.g., http://192.168.1.50).

Login
  • Username: admin
  • Password: found on device label (or as set by you)

General Settings (FXS PORT1)

FXS Port 1 General Settings > Network Settings

Account Registration

  • Account Active: ✅ Checked
  • Primary SIP Server: popserver.voip.ms (or your nearest VoIP.ms POP) eg: montreal1.voip.ms
  • Prefer Primary SIP Server Yes
  • Outbound Proxy popserver.voip.ms (Same as the Primary SIP Server)
  • Backup Outbound Proxy This is optional you can enter a different VoIP.ms PoP Server location. eg: toronto2.voip.ms
  • Prefer Primary Outbound Proxy: Check the box, only if you entered a backup Outbound Proxy
  • SIP User ID Your VoIP.ms sub-account username
  • SIP Authenticate ID Same as SIP User ID
  • SIP Authentication Password Your sub-account password

Network Settings

  • DNS Mode: A Record
  • NAT Traversal Keep-Alive
Click Save and Apply.

SIP Settings Tab

FXS Port 1 Advanced Settings > Security Settings

SIP Basic settings

  • SIP Registration: Yes
  • Register Expiration: 2 (minutes)
  • Enable SIP OPTIONS Keep Alive: Yes
  • Local SIP Port: 5060 (change if multiple ATAs)
Disable extra headers:
  • Privacy Header → No
  • P-Preferred-Identity Header → No
  • P-Access-Network-Info → No
  • P-Emergency-Info → No

Codec Settings Tab

Preferred Vocoder (in all sessions): G722, (G.711 μ-law) then set G711 for the rest.

Advanced Settings

Check SIP User ID for Incoming INVITE: Enabled
Allow incoming SIP messages from SIP proxy only: Enabled

Faxing

FXS Port 1 Call Features Settings
Ensure options like Enable‑Call‑Waiting, Caller‑ID, and Tone are unchecked — avoiding disruptions during fax sessions.
Security Settings: Enable Allow incoming packets from SIP proxy only to restrict access to VoIP.ms servers.
Under Call Settings
Disable Local Call Features: Yes
  • Enable-Call-Waiting: uncheck
  • Enable-Call-Waiting-Caller-ID: uncheck
  • Enable-Call-Waiting-Tone: uncheck
Under Call Features Settings:
  • Set Enable Local Call Features No to prevent unintended feature code usage via the fax machine.
Under Codec Settings:
  • Fax Mode: T.38 (if you are using it for faxing with a VoIP.ms FAX POP server)
  • T.38 Max Bit Rate: 14400 bps
  • Jitter Buffer: Fixed, High
Under Analog Signal Line Configuration
  • Enable Line Echo Canceller (LEC) → Check

Testing

Reboot the ATA after applying settings.
Check Status tab → Registration should be Yes.
Dial: 4443 for a Voice Echo Test.

External links

Retrieved from "https://wiki.voip.ms/article/Grandstream_HT801v2"
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