Grandstream HT802v2
From VoIP.ms Wiki
This guide is intended to assist with the process of configuring a Grandstream HandyTone HT802v2 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 |
Background
Grandstream announced the release of a new lineup of VoIP Analog Telephone Adapters (ATAs) and hardware gateways intended to replace their original products of both types on October 30, 2024, intending them to be available for purchase through normal channels at the start of 2025; included in the announcement were details about the new HandyTone HT802v2 ATA.{1| Hoping to utilize the popularity of their existing crop of devices, the new lineup consisted of models that targeted the feature sets and form factors of each of the earlier models that were powered by modern SoCs operating at faster clock speeds and with more onboard volatile memory. Though retail SKUs and internal product numbers were of course changed, the new devices all bore the same names and model numbers as their predecessors, but with v2 appended to each (for "version 2").
Device details
The HT802v2 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)
 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 HT802 V2
- Latest firmware installed (Grandstream site) (We you use firmware version 1.0.5.7 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 HT802 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.
References
- ^ Bowers, Phil. (October 30, 2024) "Grandstream Releases New ATA's and VoIP Gateways." Grandstream Networks, Inc. Archived from the original on Nov. 1, 2024. Last accessed on Feb. 23, 2026.
