This FAQ document provides a concise and practical reference for Grandstream’s HT801/HT802 and HT812/HT814/HT818 V2 Analog Telephone Adapters. These ATAs enable analog phones and fax machines to operate on modern VoIP networks while offering high-quality audio, strong security, flexible provisioning, and advanced telephony features. The following sections answer the most common setup and operational questions to help users install, configure, and manage their HT8xx V2 devices efficiently.
Product Overview & Hardware Basics
These ATAs allow users to connect analog phones and fax machines to VoIP systems, this enables high-quality IP telephony for home or office environments. They convert analog signals from traditional phones into digital SIP traffic and support a wide range of VoIP features and provisioning options.
The main difference is the number of FXS ports and network interfaces.
- HT801 V2: 1 FXS, 10/100Mbps port
- HT802 V2: 2 FXS, 10/100Mbps port
- HT812 V2: 2 FXS, dual Gigabit ports
- HT814 V2: 4 FXS, dual Gigabit ports
- HT818 V2: 8 FXS, dual Gigabit ports
The HT81x models also include NAT routing functionality and more LEDs.
FXS ports provide analog line interfaces where you plug standard analog phones or fax machines using RJ11 cables. These ports allow analog devices to make and receive VoIP calls through SIP registration.
HT812/814/818 V2 include two 10/100/1000 Mbps Ethernet ports, labeled NET1 and NET2. By default, NET1 acts as WAN (DHCP client), and NET2 acts as LAN (DHCP server). Their roles can be swapped in the Web UI.
HT801/HT802 V2 models include one 10/100Mbps Ethernet port used to connect the ATA to a router or modem. They do not include routing capabilities.
Yes. All models support multiple voice codecs including G.722 and OPUS, enabling high-definition audio for clearer and more natural voice communication.
Yes. All models support T.38 Fax Relay up to 14.4 kbps, and automatically switch to G.711 Pass-Through if needed, ensuring reliable fax transmission.
They include Caller ID, call waiting, call forwarding, blind/attended transfer, 3-way conferencing, do not disturb, hold, paging, and many star-code based features. These functions work directly from the analog handset.
Security includes SRTP for media encryption, TLS/SIPS/HTTPS for signaling protection, unique security certificates, AES encrypted provisioning files, and support for TR-069 secure management.
Each model includes Power, NET1, NET2, and Phone LEDs for every FXS port. LEDs show booting, network link status, registration status, off-hook/busy, and voicemail indicators via slow blinking.
Yes. All HT801/802 and HT812/HT814/HT818 V2 devices have a reset button that must be pressed for 7 seconds to restore factory defaults, unless this function has been locked by a provider.
HT801/802 use 5V/1A adapters, while HT812/HT814/HT818 use 12V adapters with different amperage depending on the model (0.5A, 1A, or 1.5A). Using the correct power supply is required for stable operation.
Yes. HT products can work with 4-wire analog phones, but only the TIP and RING pair is actually used for operation. These are the two middle wires of the RJ-11 interface, and they are the only ones connected to the device’s hardware board. This means 4-wire phones function normally, but the ATA only uses the standard 2-wire analog signaling.
Yes. On HT812, HT814, and HT818 V2 models, the NET1 and NET2 ports can have their roles switched from LAN to WAN and vice versa. By default, NET1 operates as a WAN (DHCP client) and NET2 as a LAN (DHCP server), but the device allows changing these modes through the Web UI, giving flexibility depending on your network setup.
Installation, Ports, and LED Indicators
Each unit includes the ATA device, a power supply, an Ethernet cable, and documentation. Models with multiple FXS ports (HT812/814/818) include the appropriate power adapter and necessary cables. Users should verify all items are present before installation.
Simply plug a standard RJ11 telephone cable from your analog phone or fax machine into one of the ATA’s FXS ports. The phone LEDs will light accordingly once the device is powered and ready.
Insert an Ethernet cable into the NET port and connect the other end to your router, modem, or switch. When connected properly, the NET LED will turn on and blink during data transmission.
When using NET1, the ATA functions as a DHCP client. Connect NET1 to your router or switch. Once powered, NET1 and the FXS LEDs should turn solid, indicating readiness.
When using NET2, the ATA acts as a DHCP server. Connect a PC or switch to NET2. The device will assign local IPs (default 192.168.2.1). NET2 and FXS LEDs will light when ready.
Yes. HT81x V2 models allow switching NET1 and NET2 operating modes in the Web UI, giving flexibility in deployment either as a client device or as a small router.
Phone LEDs show the status of each FXS port:
- OFF: Unregistered
- Solid Blue: Registered
- Blinking (0.5s): Off-hook or busy
- Slow Blinking: Indicates voicemail waiting
These indicators help users quickly confirm port and registration states.
The NET LED turns solid when the adapter is connected to a network and blinks during data activity. If it is off, network connectivity is absent or the device is still booting.
The Power LED flashes during boot-up across all models. Once the boot process completes, it becomes solid. If LEDs remain off, the device may not be receiving power.
On HT801/HT802 and HT812/814/818, a slow-blinking phone LED indicates that new voicemail messages are available for that FXS line. The stutter tone will also play when picking up the handset.
Unplug the Ethernet cable, press and hold the reset button for 7 seconds, then release. The device will reboot and restore all default settings unless locked by a service provider.
Ensure the correct power adapter is being used (5V for HT80x, 12V for HT81x). Incorrect voltage may prevent startup. If still unresponsive, check the outlet or replace the adapter as required by the model specifications.
Network Configuration, IVR, and Web UI Access
Pick up the handset of a phone connected to an FXS port and dial *** to open the IVR. Then dial 02, and the device will announce its current IP address. This applies whether the unit is using DHCP, Static IP, or PPPoE mode.
When connected via NET1, the unit works as a DHCP client. Web access from WAN may be disabled by default. To enable it, enter the IVR (*** → option 12) and press 9 to toggle web access ON. After rebooting, open the displayed IP address in your browser.
Connect a PC directly to the NET2 port. NET2 uses a default gateway IP of 192.168.2.1. Enter this address into your browser, then log in with the admin password. Ensure your PC obtains an IP in the 192.168.2.x range.
There are three access levels:
- Admin (full rights) – default password printed on device
- User (basic settings only)
- Viewer (view-only access)
User and viewer levels are disabled by default and must be enabled under System Settings → Security Settings → Web Access.
HT80x and HT81x V2 support modern browsers including Chrome (58+), Firefox (53+), IE (10+), Safari (5.1.4+), and Opera (44+). You must be on the same network segment as the ATA.
Use the IVR option 01 and press 9 to toggle among DHCP, Static IP, and PPPoE. If Static IP is selected, configure IP, Subnet, Gateway, and DNS using options 02–05. PPPoE credentials must be configured in the Web UI. A reboot is required for changes to take effect.
After setting Static IP mode in menu 01, enter a 12-digit IP (e.g., 192168000026) when prompted by options 02–05 for IP, Subnet, Gateway, and DNS. The unit must be rebooted for changes to apply.
This option lets you cycle through supported voice codecs using key 9, including PCMU/A, iLBC, G726, G723, G729, OPUS, and G722. This can help optimize audio quality or bandwidth usage.
The ATA requires 12 continuous digits without decimals. If any IP section is less than three digits, add leading zeros (e.g., “26” becomes “026”). Incorrect entries prompt an error and return to the main menu.
Log into the Web UI as admin, go to System Settings → Security Settings → User Info Management. Enter and confirm the new password (8–20 characters including uppercase, lowercase, and numbers). After first login, the system forces a password change
In the Web UI, navigate to System Settings → Security Settings → Web/SSH Access. Select the Web Access Mode (HTTP or HTTPS) and assign a new port in the range 1–65535. Apply and save changes.
The ATA normally plays a dial tone, but you can configure it to play a busy tone instead by setting “Play Busy Tone When Account is Unregistered” to YES and “Outgoing Call Without Registration” to NO in the Web UI.
SIP Registration & Telephony Feature Configuration
HT80x V2 ( HT801/HT802):
Go to the FXS Port page and set Account Active = Yes. Enter the Primary SIP Server, Failover Server (optional), Outbound Proxy (optional), SIP User ID, Authenticate ID, Name, and Authenticate Password. Apply the settings, then verify registration under Status → Port Status.
HT81x ( HT812/HT814/HT818):
Go to the Profile page and set Account Active = Yes. Enter the Primary SIP Server, Failover Server (optional), and Outbound Proxy (optional).
Then go to the FXS Port page and enter the SIP User ID, Authenticate ID, Name, and Authenticate Password. Make sure the Profile ID matches the profile you activated earlier. Apply the settings, then verify registration under Status → Port Status.
In the Web UI, navigate to Status → Port Status → Registration.
- Registered means the account is active.
- Not Registered means the credentials or SIP server configuration must be corrected.
When multiple ports are registered, the device will trigger simultaneous ring with a one-second delay between each phone connected to the ATA. This ensures orderly ringing across ports.
Yes. Enter the Failover SIP Server field under the Profile/FXS Port settings. If the primary server becomes unreachable, the ATA switches automatically to the backup server to maintain service continuity.
Features include Caller ID, call waiting, blind and attended transfer, 3-way conferencing, call forwarding, do not disturb, paging, message waiting indication (MWI), flash/hook operations, and DND. Most features are accessible through star codes or analog phone flash keys.
Use the star codes:
- *51 Enable call waiting (for all calls)
- *50 Disable call waiting (for all calls)
Per-call versions also exist using *70 (disable) and *71 (enable). Call waiting must also be enabled in the service provider’s settings if required.
The ATAs allow forward types via star codes:
- *72/*73 Enable/cancel unconditional forward
- *90 / *91 Enable/cancel busy forward
- *92 / *93 Enable/cancel delayed forward
Dial the feature code + target number + #. The dial tone confirms successful activation.
For blind transfer, press FLASH, dial *87, then the destination number.
For attended transfer, press FLASH, dial the destination, speak to the party, then hang up (or press FLASH again if needed).
“Enable Call Feature” must be enabled in the Web UI.
Star codes allow quick toggling:
- *16 Enable SRTP
- *17 Disable SRTP
Per-call versions exist using *18 (enable for only this call) and *19 (disable for only this call). SRTP ensures encrypted audio streams.
Pick up the handset (you may hear a stutter tone), dial ***, then choose option 86. Enter the voicemail password if configured. FXS LEDs will slowly blink when new messages are waiting.
Yes. While on a call, press FLASH, dial the third party’s number, and once they answer, press FLASH again to join all three callers. If the third party does not answer, press FLASH to return to the original call. Configuration options also allow transferring parties if the initiator hangs up.
Call Operations, Dialing Behavior & Star Codes
Lift the handset and dial the number directly. You may either wait for the 4-second no-key timeout or press # to send the call immediately (if “Use # as Dial Key” is enabled). This works for internal extensions and external numbers based on your SIP provider’s dialing rules.
The ATA plays dial tone by default even if registration fails. You can change this behavior:
- Enable “Play Busy Tone When Account is Unregistered.”
- Set “Outgoing Call Without Registration” to NO.
This will play a busy tone instead until the account registers.
You can call another VoIP device directly via IP addressing if both devices are reachable:
- Using IVR: *** → 47, then enter the IP address.
- Using star code: Dial *47, then the IP encoded as * instead of dots (example: 192.168.0.160 → 192*168*0*160).
Press # or wait 4 seconds.
Yes. Append the port using “*” to represent “:”.
Example: 192*168*1*20*5062 for IP 192.168.1.20:5062.
This allows targeting custom SIP listening ports.
When a second call arrives, you will hear the call waiting tone. Press FLASH (or hook flash) to switch between callers. The original caller is placed on hold automatically.
Press the FLASH button (or perform a quick hook flash). The remote party is placed on hold. Press FLASH again to resume the conversation. If no flash button exists, quickly toggle the phone’s hook switch.
During a call:
- Press FLASH.
- Dial *87.
- Dial the destination number and press # or wait 4 seconds.
The original call is transferred immediately.
During a call:
- Press FLASH.
- Dial the target number and press # or wait.
- Speak with the third party; hang up or press FLASH to complete the transfer.
If the third party does not answer, FLASH returns you to the original call.
While on a call:
- Press FLASH to get a dial tone.
- Dial the third party and press # or wait.
- When they answer, press FLASH again to join all parties.
If the initiator hangs up, the conference ends unless “Transfer on Conference Hang Up” is enabled.
Lift the handset and dial *69. The ATA automatically calls the most recent incoming caller.
Dial *** then 7X, where X is the port number.
Example: To call the phone on FXS port 1, dial *** → 71.
This works even without any SIP registration.
Some commonly used feature codes include:
- *02 Force codec per call
- *16 / *17 Enable/Disable SRTP
- *30 / *31 Block/Send Caller ID
- *50 / *51 Disable/Enable call waiting
- *72 / *73 Enable/Cancel unconditional call forward
- *78 / *79 Enable/Disable Do Not Disturb
These codes give users quick control without accessing the Web UI.
Provisioning, Firmware Upgrade & Factory Reset
The devices support firmware upgrades via TFTP, HTTP, HTTPS, FTP, and FTPS. You simply configure the firmware server path in the Web UI and choose the desired protocol. Local upgrade via file upload is also supported.
Log into the Web UI, go to Maintenance → Upgrade → Firmware, enter the server path (IP or FQDN), choose the upgrade protocol, then click Save and Apply. Reboot the device to start the upgrade process. Always check the Status page afterward to confirm the firmware version has changed.
Yes. After downloading the firmware from Grandstream’s website, extract it and upload it through the Firmware section in the Web UI. Select the file, click Upload, and wait until the process completes.
Place the firmware files in the TFTP server’s root directory, ensure the TFTP server is set to Transmit Only, and connect the ATA and TFTP host on the same LAN. Point the Firmware Server Path to the TFTP server’s IP, save settings, and reboot the ATA.
Prefix and Postfix allow the ATA to download only firmware files whose filenames match a specific pattern. This enables storing multiple firmware versions in the same directory while ensuring each device retrieves only the files intended for it.
Under Automatic Upgrade, the ATA can check daily or weekly for new firmware or configuration updates at a scheduled time. Service providers can stagger upgrades by setting different check intervals using provisioning parameters to separate or delay upgrade actions so they don’t all happen at the same time and cause network congestion
Provisioning is supported via TFTP, HTTP/HTTPS, FTP/FTPS, as well as TR-069 for remote cloud management. Configuration files may be binary or XML, and the ATA automatically attempts downloading cfgMAC.xml, cfgMAC, cfgModel.xml, or cfg.xml depending on availability.
GDMS (Grandstream Device Management System) is a cloud platform that allows centralized provisioning, monitoring, firmware upgrade, and configuration of multiple HT80x/HT81x V2 devices across distributed locations. It simplifies large-scale deployment.
Yes. The ATAs support the event:resync SIP NOTIFY command, which forces the device to check its provisioning server for updates. They also support event:reset (if enabled), allowing remote factory reset initiation through SIP NOTIFY after authentication.
Unplug the Ethernet cable, press and hold the reset button for 7 seconds, then release. The ATA will reboot and return to factory defaults unless the reset function has been locked by a provider.
Dial ***, enter 99, wait for the “reset” prompt, then enter the encoded MAC address using the keypad mapping (A=22, B=222, C=2222, etc.). The device will reboot after about 15 seconds.
Under Maintenance → Restore Factory, you may choose between:
- Full Reset – restores ALL settings.
- ISP Data Reset – resets basic network settings (IP mode, PPPoE, web port).
- VOIP Data Reset – resets SIP-related and provisioning settings.
A reset will reboot the device and remove the selected configuration sets.