Grandstream HT801v2
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Revision as of 16:00, 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.
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Device details
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:
- Support for two SIP profiles, compatible with both VoIP.ms DIDs and sub-accounts, each mapped to a dedicated FXS (RJ11) phone port
- Incoming and outgoing Caller ID with support for custom formatting and ringtones
- A T-Base10/100 Ethernet (RJ45) port
- Optional TLS and SRTP encryption protocols to secure calls and accounts
- Three-way call conferencing
- Call transfers, forwarding, hold and do-not-disturb
- SIP server failover to ensure your device stays connected to the service even if the default server goes offline
- Fax over IP capability (using either T.38 or G.711 passthrough)
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.