This FAQ document provides clear and structured answers to the most common questions about the Grandstream HT813 Analog Telephone Adapter. It covers product functionality, installation, SIP configuration, PSTN/FXO behavior, call features, firmware management, and maintenance. Use this guide to quickly understand and operate the HT813 while maximizing its VoIP–PSTN integration capabilities.
Product Overview & Hardware Basics
The HT813 is an analog telephone adapter that combines one FXS port, one FXO PSTN line port, and an IP interface into a single device. This design allows the unit to bridge VoIP networks with analog telephones and the PSTN. As a result, users can originate or terminate calls from VoIP to PSTN, use an analog phone as a VoIP extension, and maintain PSTN access if Internet service fails.
The device features one RJ11 FXS port for connecting an analog phone or fax machine, and one RJ11 FXO port for connecting a PSTN line. The FXS port extends a SIP account to an analog phone, while the FXO port allows the ATA to access or route calls through the PSTN line.
It offers two 10/100 Mbps RJ45 network ports:
- The WAN port connects the HT813 to an existing router or modem.
- The LAN port allows the HT813 to act as a router, and provides DHCP services to connected devices.
Lifeline support ensures phone continuity during failures. If the HT813 loses power or loses SIP registration, it will automatically hard-relay the FXS port to the FXO port, which then allows the connected analog phone to directly use the PSTN line without configuration.
The device has LEDs for POWER, WAN, LAN, FXS, and FXO.
- POWER: solid when on, flashing when booting.
- WAN/LAN: solid when network link is active.
- FXS/FXO: solid when registered, blinking for off-hook/busy, slow blinking for voicemail indication.
The unit uses a 12V/0.5A DC power adapter and supports a wide AC input range (100–240V at 50/60Hz), making it compatible with global power standards.
Yes. Although it has only one FXS and one FXO port, each port supports its own independent SIP profile, this results in a two separate SIP accounts to be registered simultaneously.
Supported codecs include G.711 µ/a, G.726, G.723.1, G.729A/B, iLBC, and OPUS. This wide range ensures compatibility with various VoIP providers and optimizes performance for both high-bandwidth and low-bandwidth environments.
Yes. The device fully supports T.38 Fax-over-IP, which is the preferred method, and it can also fall back to G.711 fax pass-through for providers that do not support T.38. This makes it suitable for environments where faxing is still required.
Security features include TLS, SIPS, HTTPS, and SRTP for encrypted signaling and media streams. Additionally, the device supports AES-encrypted provisioning, ensuring that configuration files and firmware downloads remain secure.
Installation, LEDs & Basic Network Setup
The package includes the HT813 unit, a power adapter, an Ethernet cable, and a quick installation guide. Before installation, users should verify that all components are present; if anything is missing, they must contact their system administrator or vendor.
To connect an analog phone, simply plug an RJ11 telephone cable into the FXS port on the HT813, then plug the other end into your standard analog phone or fax machine. No additional configuration is required for basic connectivity.
The WAN port should be used when the HT813 is connected behind a router, modem, or switch. In this setup, the HT813 operates as a DHCP client, receiving an IP address automatically from the existing network.
The LAN port is used when you want the HT813 to act as a router. In this mode, the HT813 assigns DHCP addresses to devices connected to its LAN port, which allows it to operate as a small NAT router for one or more devices.
The HT813 is ready for use when the POWER, FXS, and either WAN or LAN LEDs are solidly lit (not flashing). Flashing LEDs indicate booting, network negotiation, or port activity, depending on the LED type.
When the FXS LED is flashing, it means the line is connected and ready to make/receive calls if the SIP settings are configured correctly.
• Fast flashing FXS LED → the phone is off-hook or busy
• Slow flashing FXS LED → there is voicemail waiting
- Solid Blue: Port is active and registered.
- Fast Blinking (every 500 ms): Off-hook or busy line.
- Slow Blinking: New voicemail message available.
- Off: Port is not registered.
Pick up the handset and dial *** to enter the IVR menu.
Then dial 02, and the device announces its current IP address assigned via DHCP, static mode, or PPPoE.
The HT813 supports:
- DHCP Mode: Automatically obtains network settings.
- Static IP Mode: Allows manual entry of IP, subnet mask, gateway, and DNS.
- PPPoE Mode: Used when connecting directly to an ISP requiring PPPoE credentials (set via Web UI).
Any changes to IP address, subnet, gateway, or DNS require the HT813 to be rebooted for the new settings to take effect. This can be done through IVR option 99 or the Web UI.
When the LAN port is used, the HT813’s role switches to a router that must provide DHCP and NAT functions. Therefore, NAT Router must be enabled under Basic Settings → Device Mode to ensure proper network routing behavior, such as assigning IP addresses to devices under the LAN.
Web Interface & Management
To access the Web GUI via WAN, first retrieve the HT813’s IP address using the IVR (*** → 02). Then open a browser on a PC connected to the same network and enter that IP address. If the device is reachable, the login page will appear prompting for the administrator password.
When connecting through the LAN port, the HT813 always uses the default IP 192.168.2.1. Your computer must be set to an IP within the 192.168.2.x range to access the Web UI in this mode.
- Admin Level:
admin(full access) - End User Level:
123(limited access to status and basic settings) - Viewer Level:
viewer(read-only access)
These passwords are case-sensitive and should be changed after first login.
- Administrator: Full access to all pages, configuration, provisioning, and reboot/reset options.
- End User: Can view status pages and modify certain basic settings.
- Viewer: Read-only mode with no ability to modify configuration.
End User and Viewer roles are disabled by default and must be enabled in Advanced Settings.
Log in as Admin → go to Advanced Settings → New Admin Password → enter the new password → confirm → click Apply. This secures the device and prevents unauthorized access.
Clicking Update saves changes but does not activate them. Clicking Apply activates the settings immediately. If a reboot is needed, the Web GUI will display a prompt with a reboot button.
Yes. Under Basic Settings → HTTP(S) Web Port, you can change the access port to any value between 1 and 65535, provided “Web Access Mode” is set to HTTP(S). This is useful for avoiding port conflicts or improving security.
At the bottom of any configuration page, there is a Reboot button. Clicking it triggers a remote reboot and displays a confirmation message. After approximately 30 seconds, you can log back in.
Supported browsers include Internet Explorer 10+, Chrome 58+, Firefox 53+, Safari 5.1.4+, and Opera 44+, ensuring proper rendering and functionality of all Web UI elements.
When enabled, “Lock Keypad Update” prevents factory reset operations via physical reset button or IVR commands. Pressing the physical reset button will only reboot the device, not restore factory defaults.
SIP Accounts, Call Features & Call Handling
The HT813 can register two independent SIP accounts, one for the FXS port and one for the FXO port. Each port holds its own SIP configuration, allowing separate phone numbers or services for VoIP calling and PSTN bridging.
To register a SIP account, you must set:
- Account Active → Yes
- Primary SIP Server (IP or FQDN)
- Failover SIP Server (optional)
- SIP User ID (usually a phone number/extension)
- Authenticate ID
- Authenticate Password
- Name (label for the account)
These settings must be applied before the account attempts registration.
Go to Status → Port Status on the Web UI. If the registration shows Registered, the SIP account is active. If it displays Not Registered, the settings need review or the server may be unreachable.
Pick up the handset, dial the destination number, and either press # (if configured as the send key) or simply wait 4 seconds, which is the default “No Key Entry Timeout.”
Dial *02 followed by the codec code before the phone number.
Examples:
- PCMU → *027110
- PCMA → *027111
- G723 → *02723
This overrides the preferred codec list for that call only.
- Block caller ID for all calls: *30
- Re-enable caller ID for all calls: *31
These settings apply until changed again.
- Enable call waiting: *51
- Disable call waiting: *50
- For per-call control:
- Disable: *70 + number
- Enable: *71 + number
Call waiting must also be enabled in the SIP account settings.
During a call:
- Press FLASH to get a new dial tone.
- Dial *87, then the destination number.
- Press # or wait 4 seconds.
Once the tone plays, hang up to complete the blind transfer.
- Press FLASH to get dial tone.
- Dial the destination number + # or wait 4 seconds.
- Speak with the third party.
- Hang up to complete the transfer, or FLASH back to rejoin the original caller if the transfer fails.
- During a call, press FLASH.
- Dial the third party’s number + # or wait 4 seconds.
- When they answer, press FLASH again to merge all participants.
- You may drop the third party by pressing FLASH again.
Dialing *69 initiates a Call Return, automatically dialing the last incoming caller’s number received on the FXS port.
- Pick up the handset (stutter tone will play if voicemail exists).
- Dial ***.
- Enter 86 for voicemail access.
- Enter voicemail password if configured.
The FXS LED will blink slowly when new messages are present.
Two indicators are provided:
- Slow blinking FXS LED
- Stutter tone heard for a few seconds after going off-hook
These remain active until the mailbox is cleared.
Two methods are available:
- IVR method: *** → 47 → enter IP (use * instead of dots).
- Star code method: Dial *47, then the encoded IP.
Example: 192.168.0.160 → 192*168*0*160
If port is needed: add *port at the end.
Random SIP/RTP ports must be set to No.
Direct IP calling requires predictable port usage so both devices can communicate directly. If random ports are used, the receiving device may not know which SIP/RTP ports to expect, and this prevents the direct call from establishing.
FXO/PSTN Operation & Advanced Routing
PSTN pass-through allows the analog phone connected to the FXS port to directly use the PSTN line connected to the FXO port. The HT813 automatically activates lifeline pass-through during power loss or when SIP registration is lost, this can ensure that users can still make and receive PSTN calls without Internet or VoIP service.
To place a PSTN call:
- Pick up the analog phone.
- Dial the PSTN access code (default is *00, but can be changed in the Web UI).
- Wait for the PSTN dial tone.
- Dial the destination number normally.
This allows the FXS phone to seize the PSTN line.
A VoIP user calls the SIP account registered on the FXO port. After the call connects:
- The caller hears either a dial tone or a continuous tone (if PIN is enabled).
- If PIN is configured, the caller must enter it correctly (up to 3 tries).
- Once validated, the caller receives PSTN dial tone.
- The user enters the PSTN number to complete the call.
This feature bridges IP calls to the local PSTN line.
The HT813 allows three attempts to enter a valid PIN. If all three attempts fail, the device automatically terminates the call to prevent unauthorized PSTN usage.
When someone calls the PSTN line connected to the FXO port:
- The analog phone rings by default for 4 rings (configurable).
- If unanswered, the caller hears a dial tone or continuous tone (if PIN is required).
- After entering the correct PIN (up to 3 tries), the caller is connected to VoIP mode.
- The caller can then dial a VoIP number via the HT813’s FXO SIP account.
This allows remote users to make VoIP calls through the HT813.
Certificate verification occurs within restricted factory-access functions. These areas require factory-level password credentials and should only be accessed when instructed by the manufacturer. Users are warned not to enter this menu without explicit guidance.
If “Forward to PSTN” is enabled for the FXO port, any VoIP call that dials the FXO SIP account number will be automatically forwarded to the programmed PSTN number. In this case, second-stage dialing is skipped entirely.
Under the FXO port settings, users can configure the “Forward to VoIP” field. If the incoming PSTN call is not answered within the configured number of rings (default is 4), the HT813 forwards the call to the specified VoIP number on the SIP server.
“Route Call to PSTN” allows specific dialed patterns to bypass VoIP entirely. By defining rules in the FXS dial plan (e.g., {L: 626x+}), calls matching that pattern are automatically sent to the PSTN line. This is useful for:
- Emergency numbers
- Local PSTN-only services
- Special prefixes required to stay on the PSTN
One-stage dialing eliminates second-stage input for VoIP-to-PSTN calls. If enabled in the FXO settings and supported by the SIP server, the PSTN number is included directly in the Request-URI. The HT813 then automatically seizes the PSTN line and dials the number without waiting for further input.
By default, the HT813 allows 4 rings before switching the incoming PSTN call to VoIP mode or forwarding it (if configured). This value is adjustable under the FXO port configuration page.
Firmware, Provisioning, Reset & Maintenance
The HT813 supports TFTP, FTP, FTPS, HTTP, and HTTPS for firmware downloads. This flexibility allows service providers and administrators to use the protocol that best matches their provisioning environment, security level, and network restrictions. HTTPS is the default method for secure upgrades.
Go to Advanced Settings → Firmware Upgrade and Provisioning, enter the desired Firmware Server Path (IP or FQDN), choose the appropriate upgrade protocol, and enable “Always Check for New Firmware.” Then click Apply, and finally reboot the ATA to trigger the upgrade.
When this option is set to Yes, the HT813 will automatically check the configured server for new firmware every time it boots, ensuring the device stays updated without requiring manual intervention.
First, download and unzip the firmware from the Grandstream website. In the Web UI, navigate to Advanced Settings, browse your computer for the HT813fw.bin file, and click Upload Firmware. The device will install the firmware locally without needing a network server.
Firmware prefix and postfix fields allow administrators to store multiple firmware versions in the same directory. The HT813 will only download firmware whose filename begins or ends with the specified prefix/postfix. This enables more granular control over which devices upgrade to which version.
Upon booting, the device automatically attempts to download configuration files in this order:
- cfg000b82xxxxxx.xml (device-specific XML)
- cfg000b82xxxxxx (device-specific binary)
- cfght813.xml (model-specific XML)
- cfg.xml (global configuration)
The first file found is used.
A SIP NOTIFY message with the check-sync event triggers the HT813 to immediately re-synchronize with its provisioning server. This forces an instant configuration or firmware refresh instead of waiting for the next scheduled check.
Unplug the Ethernet cable, insert a pin into the reset hole on the back of the unit, hold it for about 7 seconds, then release. The device will reboot and return to full factory default settings, provided keypad locking is not enabled.
Pick up the phone and dial *** to access the IVR, then enter 99. The device will prompt for the encoded MAC address. After entering it correctly, the HT813 will reboot and restore factory defaults automatically.
Under Basic Settings → Reset Type, you can select:
- Full Reset: Restores all device settings to factory defaults.
- ISP Data Reset: Resets network-related settings (IP mode, PPPoE, web port).
- VOIP Data Reset: Clears SIP settings, provisioning data, and service provider information.