Grandstream HT802v2
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| - | + | 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, the HT801v2—along with their "v1" predecessors—have been the [[ATA Devices#Most Popular ATA Devices|most widely-used ATAs]] by our customers in recent years, according to internal network analytics. | |
| - | = | + | == Background == |
| - | + | [[wikipedia:Grandstream Networks|Grandstream]] announced the release of a new lineup of VoIP [[ATA Devices|Analog Telephone Adapters (ATAs)]] and hardware gateways on October 30, 2024 that would arrive to market at the start of 2025; included in the announcement were details about a replacement for the venerable [[Grandstream HandyTone 802 - HT802|HandyTone 802]] ATA named the HT802v2.<sup>{[[#CITEREF1|1]]|</sup><span id="REFCITE1" style="display: hidden;"></span> The clear intention was to refresh both product lines with upgraded hardware while avoiding any disruption to the widespread popularity the existing models already enjoyed, and so the new replacements generally targeted their predecessors' feature sets and form factors with exacting precision. The bulk of the announced changes were located "under the hood," such as being built atop modern SoCs that operated at faster clock speeds, had more onboard volatile memory and conformed to an expanded set of compliance standards. The physical designs and even the earlier model names/numbers were largely reused in their entirety (though retail SKUs and internal product numbers were, of course, changed), and for most the only visible change was a <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">v2</code> appended to their name (for "version 2"). | |
| - | + | ||
| - | - | + | |
| - | + | ||
| - | = | + | === Device details === |
| - | :* Grandstream HT802 V2 | + | The HT802v2 is an entry-level model intended for residential and light [[wikipedia:Small office/home office|SOHO (Small Office/Home Office)]] installations that is capable of leveraging a single network link into two simultaneously operable analog phone ports. Its form factor is very compact, measuring 10×10×2.95cm and weighing 114g, even utilizing one of the small, ubiquitous [[wikipedia:USB hardware#Micro connectors|USB Micro-B ports]] for the AC power input (expecting 5VDC/1A), which allows it to be powered by all but the most feeble of mobile phone chargers manufactured in the last decade. Yet despite its diminutive footprint, it nevertheless manages to provide a range of features that is both deep and broad, including: |
| - | :* Latest firmware installed ([https://www.grandstream.com/support/firmware|check Grandstream site]) ''(We | + | * 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 |
| + | * Incoming and outgoing [[Caller ID]] with support for custom formatting and ringtones | ||
| + | * A [[wikipedia:Fast Ethernet#Copper|10/100Mbps Ethernet over twisted pair]] port | ||
| + | * Optional [[Call Encryption - TLS/SRTP|TLS and SRTP encryption protocols]] to secure calls and accounts | ||
| + | * [[Audio Conferencing|Three-way call conferencing]] | ||
| + | * Call transfers, [[Call Forwarding|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 | ||
| + | * [[Virtual Fax|Fax over IP capability]] (using either T.38 or G.711 passthrough) | ||
| + | |||
| + | <gallery caption="Grandstream HandyTone HT802v2 photos (click to expand)" heights="224" perrow="3" widths="224" style="margin: 1.5em auto; text-align: center; width: fit-content;"> | ||
| + | File:Grandstream HandyTone HT802v2 - Front View.png|Front view | ||
| + | File:Grandstream HandyTone HT802v2 - Top View.png|Top view, showing status LEDs for power, network connection, and in-use indicator for both analog phone ports | ||
| + | File:Grandstream HandyTone HT802v2 - Rear View.png|Rear view with Ethernet and dual phone ports, USB Micro-B power input and pinhole reset | ||
| + | </gallery> | ||
| + | |||
| + | == Configuring your device == | ||
| + | [[File:HT802v2 FXS1 General Settings AccountRegistration.png|thumb|right|512px|FXS Port 1 General Settings > Account Registration]] | ||
| + | |||
| + | === Prerequisites === | ||
| + | * Grandstream HT802 V2 | ||
| + | :* Latest firmware installed ([https://www.grandstream.com/support/firmware|check Grandstream site]) ''(We recommend that users update their device firmware to at least version 1.0.5.7.)'' | ||
:* VoIP.ms account with a configured DID and a sub-account | :* VoIP.ms account with a configured DID and a sub-account | ||
:* Internet access via Ethernet | :* Internet access via Ethernet | ||
:* Phone or fax device connected to FXS Port 1 | :* Phone or fax device connected to FXS Port 1 | ||
| - | =Logging into the | + | === Logging into the web interface === |
: Connect the HT802 to your network and power it on. | : Connect the HT802 to your network and power it on. | ||
| - | ::* Pick up the connected phone, dial: | + | ::* Pick up the connected phone, dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">* * * 0 2</code> (Star Star Star 0 2) |
| - | ::* ** | + | |
The ATA will read its IP address. | The ATA will read its IP address. | ||
Open a browser and go to that IP (e.g., http://192.168.1.50). | Open a browser and go to that IP (e.g., http://192.168.1.50). | ||
: Login | : Login | ||
| - | :* Username: admin | + | :* Username: <code>admin</code> |
:* Password: found on device label (or as set by you) | :* Password: found on device label (or as set by you) | ||
| - | =General Settings (FXS PORT1)= | + | === General Settings (FXS PORT1) === |
| + | [[File:HT802v2 FXS1 General Settings NetworkSettings.png|thumb|right|512px|FXS Port 1 General Settings > Network Settings]] | ||
| + | |||
| + | ==== Account Registration ==== | ||
:* '''Account Active:''' ✅ Checked | :* '''Account Active:''' ✅ Checked | ||
| - | :* '''Primary SIP Server''': popserver.voip.ms (or your nearest VoIP.ms POP) | + | :* '''Primary SIP Server''': popserver.voip.ms (or your nearest VoIP.ms POP) ''eg: montreal1.voip.ms'' |
| - | + | ||
:* '''Prefer Primary SIP Server''' Yes | :* '''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 User ID''' Your VoIP.ms sub-account username | ||
:* '''SIP Authenticate ID''' Same as SIP User ID | :* '''SIP Authenticate ID''' Same as SIP User ID | ||
:* '''SIP Authentication Password''' Your sub-account password | :* '''SIP Authentication Password''' Your sub-account password | ||
| - | :* '''NAT Traversal''' Keep-Alive | + | |
| + | ==== Network Settings ==== | ||
| + | :* '''DNS Mode''': A Record | ||
| + | :* '''NAT Traversal''' Keep-Alive | ||
: Click '''Save and Apply'''. | : Click '''Save and Apply'''. | ||
| - | =SIP Settings Tab= | + | === SIP Settings Tab === |
| - | :* '''Register Expiration''': | + | [[File:FXS Port1 Advanced Settings.png|thumb|right|512px|FXS Port 1 Advanced Settings > Security Settings]] |
| + | |||
| + | ==== SIP Basic settings ==== | ||
| + | :* '''SIP Registration''': Yes | ||
| + | :* '''Register Expiration''': 2 (minutes) | ||
:* '''Enable SIP OPTIONS Keep Alive''': Yes | :* '''Enable SIP OPTIONS Keep Alive''': Yes | ||
:* Local SIP Port: 5060 (change if multiple ATAs) | :* Local SIP Port: 5060 (change if multiple ATAs) | ||
| Line 47: | Line 75: | ||
:* '''P-Emergency-Info''' → No | :* '''P-Emergency-Info''' → No | ||
| - | =Codec Settings Tab= | + | ==== Codec Settings Tab ==== |
: '''Preferred Vocoder''' (in all sessions): G722, (G.711 μ-law) then set G711 for the rest. | : '''Preferred Vocoder''' (in all sessions): G722, (G.711 μ-law) then set G711 for the rest. | ||
| - | =Advanced Settings= | + | ==== Advanced Settings ==== |
: '''Check SIP User ID for Incoming INVITE''': Enabled | : '''Check SIP User ID for Incoming INVITE''': Enabled | ||
: '''Allow incoming SIP messages from SIP proxy only''': Enabled | : '''Allow incoming SIP messages from SIP proxy only''': Enabled | ||
| - | |||
| - | =Faxing= | + | ==== Faxing ==== |
| + | [[File:FXS Port1 Call Freatures Settings.png|thumb|right|512px|FXS Port 1 Call Features Settings]] | ||
: Ensure options like Enable‑Call‑Waiting, Caller‑ID, and Tone are unchecked — avoiding disruptions during fax sessions. | : 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. | : Security Settings: Enable Allow incoming packets from SIP proxy only to restrict access to VoIP.ms servers. | ||
| Line 67: | Line 95: | ||
: Under '''Call Features Settings''': | : Under '''Call Features Settings''': | ||
:* Set ''' Enable Local Call Features''' '''No''' to prevent unintended feature code usage via the fax machine. | :* Set ''' Enable Local Call Features''' '''No''' to prevent unintended feature code usage via the fax machine. | ||
| - | |||
: Under '''Codec Settings''': | : Under '''Codec Settings''': | ||
| Line 77: | Line 104: | ||
::* '''Enable Line Echo Canceller (LEC)''' → Check | ::* '''Enable Line Echo Canceller (LEC)''' → Check | ||
| - | =Testing= | + | == Testing == |
: Reboot the ATA after applying settings. | : Reboot the ATA after applying settings. | ||
: Check '''Status tab''' → Registration should be '''Yes'''. | : Check '''Status tab''' → Registration should be '''Yes'''. | ||
| - | : Dial: 4443 for a Voice Echo Test. | + | : Dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">4443</code> for a Voice Echo Test. |
| + | |||
| + | == References == | ||
| + | # [[#REFCITE1|^]] <span class="plainlinks" id="CITEREF1">Bowers, Phil. (October 30, 2024) "[https://blog.grandstream.com/press-releases/grandstream-releases-new-atas-and-voip-gateways Grandstream Releases New ATA's and VoIP Gateways]." ''[https://blog.grandstream.com/ Grandstream Networks, Inc.]'' [https://web.archive.org/web/20241101213744if_/https://blog.grandstream.com/press-releases/grandstream-releases-new-atas-and-voip-gateways Archived] from the original on Nov. 1, 2024. Last accessed on Feb. 23, 2026.</span> | ||
| + | |||
| + | == External links == | ||
| + | * [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht802v2 Official product page] | ||
| + | * [https://www.grandstream.com/hubfs/Product_Documentation/HT802%20v2/Datasheet_HT802_V2_English.pdf Product datasheet and specifications] (PDF) | ||
| + | * [https://documentation.grandstream.com/article-categories/ht8xx-series Grandstream Documentation Center] | ||
| + | ** [https://documentation.grandstream.com/knowledge-base/ht802-v2-quick-installation-guide/ HT802v2 Quick Installation Guide] | ||
| + | ** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-administration-guide/ HT802v2 Administration Guide] | ||
| + | ** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-user-guide/ HT802v2 User Guide] | ||
| + | * [https://www.grandstream.com/support/firmware Firmware download site] | ||
| + | |||
| + | |||
| + | [[Category:Analog Telephone Adapters]] | ||
| + | [[Category:Grandstream]] | ||
Latest revision as of 20:21, 14 April 2026
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, the 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 on October 30, 2024 that would arrive to market at the start of 2025; included in the announcement were details about a replacement for the venerable HandyTone 802 ATA named the HT802v2.{1| The clear intention was to refresh both product lines with upgraded hardware while avoiding any disruption to the widespread popularity the existing models already enjoyed, and so the new replacements generally targeted their predecessors' feature sets and form factors with exacting precision. The bulk of the announced changes were located "under the hood," such as being built atop modern SoCs that operated at faster clock speeds, had more onboard volatile memory and conformed to an expanded set of compliance standards. The physical designs and even the earlier model names/numbers were largely reused in their entirety (though retail SKUs and internal product numbers were, of course, changed), and for most the only visible change was a v2 appended to their name (for "version 2").
Device details
The HT802v2 is an entry-level model intended for residential and light SOHO (Small Office/Home Office) installations that is capable of leveraging a single network link into two simultaneously operable analog phone ports. Its form factor is very compact, measuring 10×10×2.95cm and weighing 114g, even utilizing one of the small, ubiquitous USB Micro-B ports for the AC power input (expecting 5VDC/1A), which allows it to be powered by all but the most feeble of mobile phone chargers manufactured in the last decade. Yet despite its diminutive footprint, it nevertheless manages to provide a range of features that is both deep and broad, 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 10/100Mbps Ethernet over twisted pair 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 recommend that users update their device firmware to at least version 1.0.5.7.)
- 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.
