Grandstream HT801v2
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| - | This guide | + | This guide explains how to configure a '''Grandstream HandyTone HT801v2''' for use with VoIP.ms services. The HT801v2 is a compact, single-port Analog Telephone Adapter (ATA) that allows users to connect a traditional analog phone or fax machine to the VoIP.ms network. Along with the two-port HT802v2 and their previous-generation counterparts, it has remained one of the [[ATA Devices#Most Popular ATA Devices|most commonly used ATA devices]] among VoIP.ms customers in recent years, based on internal network analytics. |
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=== Device details === | === Device details === | ||
Latest revision as of 20:33, 18 June 2026
This guide explains how to configure a Grandstream HandyTone HT801v2 for use with VoIP.ms services. The HT801v2 is a compact, single-port Analog Telephone Adapter (ATA) that allows users to connect a traditional analog phone or fax machine to the VoIP.ms network. Along with the two-port HT802v2 and their previous-generation counterparts, it has remained one of the most commonly used ATA devices among VoIP.ms customers in recent years, based on 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
 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
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)
- Pick up the connected phone, dial:
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)
- Username:
General Settings (FXS PORT1)
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
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
- 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:
4443for a Voice Echo Test.
