The WP810, WP822, and WP825 are portable Wi-Fi IP phones designed for reliable, high-quality communication with dual-band Wi-Fi, roaming support, HD voice, and long battery life. The WP825 also offers an IP67 rugged design for more demanding environments.
This document presents a collection of frequently asked questions (FAQs) to help users understand, configure, and operate these devices effectively.
General Information
The three models share the same core features, dual-band Wi-Fi, 2 SIP accounts, HD voice, and roaming, but differ mainly in battery performance and durability.
WP810 offers 6-hour talk time and 120-hour standby, making it suitable for lighter usage.
WP822 and WP825 offer 8-hour talk time and 200-hour standby, ideal for heavier daily use.
WP825 also has a ruggedized IP67-rated design, meaning it is water-resistant, dust-resistant, and drop-safe from up to 2.5 meters, making it suitable for harsher environments.
All three handsets support 2 SIP accounts and 2 simultaneous lines, allowing users to manage multiple identities or services from the same device without additional hardware.
The handsets support 802.11 a/b/g/n/ac on both 2.4 GHz and 5 GHz bands. Roaming is fully supported, allowing the handset to switch between access points as the user moves through a building.
Each phone includes:
- The handset
- 5V universal power supply
- Charging cradle
- Belt clip
- Rechargeable Li-ion battery.
- Quick Installation Guide
Yes. All three models support HAC, meaning they are designed to work effectively with hearing aids for clearer audio.
The WP825 features a fully ruggedized IP67-rated design, meaning it is waterproof, dustproof, resistant to cleaning chemicals, and drop-safe up to 2.5 meters. This gives it a higher level of protection compared to the WP810 and WP822.
The devices support G.711μ/a, G.729A/B, G.722 (wideband), iLBC, Opus, and both in-band and out-of-band DTMF modes. They also include VAD, CNG, AEC, PLC, AJB, AGC, and ANS for better voice quality.
Security options include SRTP for encrypted audio, TLS for encrypted signaling, 802.1x authentication, strong password protection, and 256-bit AES-encrypted configuration files. SHA-256 and SHA-512 authentication are supported per RFC8760.
- WP810: 1.8-inch (128×160) TFT color LCD
- WP822 & WP825: 2.4-inch (240×320) TFT color LCD
The larger displays provide better clarity and easier navigation.
Each handset includes PTT key, Volume keys, Navigation keys, Dial key/Off-hook key, Hands-free/Speaker key, Alphanumeric keypad, Softkeys, Contacts, Menu/OK key, Power/On-hook key, */Mute key, and a dedicated #/ Lock key. The proximity sensor assists with screen behavior when near the user’s ear.
Setup & Network Configuration
The Home Screen displays key information such as the SIP account registration status, and users can press the UP key to quickly view the phone’s IP address and MAC address. It also includes additional pages for settings shortcuts and notifications.
Slide off the battery cover, place the battery with the electrodes aligned in the bottom-left corner, and close the cover firmly. The guides also warn to avoid short-circuiting the battery and to never use a damaged one to prevent potential harm.
Yes. We recommend fully charging the battery before initial use to ensure best long-term performance and calibration.
Long-press the # key to lock the keypad. To unlock it, press the Unlock softkey, then press # again. This helps prevent accidental key presses when carrying the device.
Navigate to Menu → Settings → Network → Wi-Fi Networks. The phone will list available networks. Select your network, enter the password, and the Wi-Fi icon will appear when successfully connected.
The handset automatically prefers 5 GHz because it typically provides higher throughput. However, if the 5 GHz signal becomes weak, it will intelligently switch to 2.4 GHz for better stability.
Yes. These phones can connect to Wi-Fi networks that require additional login or sign-up steps before allowing full Internet access.
Go to Status → Network Status. This menu displays IP information, DHCP vs. static assignment, and both IPv4 and IPv6 details.
You can view SSID, BSSID, signal strength, transmit/receive packets, error rates, drop counts, connection speed, channel, and frequency. These details help diagnose Wi-Fi performance issues.
They support a wide range of standards including SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, DNS, DHCP, SSH, TFTP, NTP, STUN, SIMPLE, IPv6, and 802.1x.
The handsets support 802.11e/WMM for Wi-Fi Quality of Service and Layer 3 QoS settings such as ToS, DiffServ, and MPLS, helping prioritize voice traffic.
To turn the handset ON, long-press the Power key until the screen lights up.
To turn it OFF, long-press the Power key again while the handset is idle, then select Power Off or Reboot from the softkey options.
You can pre-configure Wi-Fi settings from the Web GUI under Network → Wi-Fi Settings, allowing the handset to automatically connect to known networks, up to 20 saved SSIDs. Even hidden SSIDs can connect automatically when their credentials are preloaded.
Calling & Telephony Features
All three models support hold, transfer, call forward, call waiting, 3-way audio conferencing, auto answer, off-hook auto dial, call logs, and flexible dial plans. They also support personalized ringtones, server redundancy, and push-to-talk.
Access the phone’s IP address from a browser, then go to Account → Account 1/2. Enable the account, fill in SIP Server, SIP User ID, Authentication ID, and Authentication Password. After saving and applying, the phone attempts to register immediately.
The phones store up to 100 call records, including incoming, outgoing, missed, and all-calls categories.
Open Settings → Call Features → Speed Dial, choose a key (0–9), assign a SIP account and phone number, and save. Long-pressing that digit then dials the number automatically.
Call the first party, press Options → Conference, and dial the second party. The conference will bridge automatically once the second call is established.
- Unconditional forward: *72 + number
- Cancel unconditional: *73
- Busy forward: *90 + number
- Cancel busy: *91
- No-answer forward: *92 + number
- Cancel no-answer: *93
These codes are dialed off-hook.
Yes. Each handset includes a dedicated PTT button. When configured, pressing the button instantly switches the device from receiving mode to transmitting voice, similar to a two-way radio.
Yes. In the call log, the Add to Contacts option lets you instantly create a new entry using the selected number.
Yes, provided the SIP server supports SMS. You can create, view, delete, and reply to messages from the built-in SMS application.
The phones support a downloadable XML phonebook with up to 500 entries, allowing centralized management of contacts.
You can place calls using earpiece mode, speakerphone mode, or headset mode. Press the Dial key to call through the earpiece, press the Speaker key to call through the loudspeaker, or use the Dial key while a headset is connected to call through the headset.
After entering the phone number, press the “Account” softkey to toggle between your registered SIP accounts. The selected account will appear on the screen, and the call will be placed using that specific line.
From Menu → Call History, select any log and press the Dial key to call it.
From Menu → Contacts (or the Contacts key), choose a contact and press the Dial key. The handset will automatically dial using the default account assigned to that contact.
To make a direct IP call, dial *47 followed by the target IP address (e.g., *47192*168*1*10).
You can append a port number using #port (e.g., *47192*168*1*10#5062).
To receive IP calls, Use Random Port must be set to No in the Web UI.
From the handset, go to Menu → Settings → Call Features → Auto Answer, choose the account, set Auto Answer to Enabled, and select the mode.
From the Web UI, go to Account → Call Settings and set Auto Answer to Yes.
A blind transfer sends the caller directly to the destination number without speaking to the receiving party first.
An attended transfer lets you call the target number first, consult with them, then complete the transfer. The handset provides softkeys for both transfer types during active calls.
To create a conference, call the first party, press Options → Conference, enter the second number, and press Join Conf. The handset automatically bridges all parties.
To join a UCM conference room, simply dial the conference extension, and enter the password if required.
To send a group page, unlock the handset and press the Talk softkey to open the paging interface. Set the default mode to Group Paging, scroll to choose the paging group, then press and hold the Talk softkey while speaking. All devices subscribed to that group will receive the broadcast.
Enable PTT from Menu → Settings → Advanced Settings → PTT and Paging → PTT Settings. Once enabled, ensure the multi-function button is set to Push-to-Talk. Press the PTT key once to access the PTT/Paging page, then press and hold the PTT key to transmit audio to the selected channel. Devices on the same channel will receive the PTT message.
Web UI, Provisioning & Security
Ensure the phone and computer are on the same network. Retrieve the phone’s IP from Status → Network Status, enter it in your browser, and log in using the admin or user credentials.
The guides recommend Google Chrome or Mozilla Firefox. Older Internet Explorer versions and some Safari/Edge players have limited support.
- Admin: admin / random password (printed on sticker)
- User: user / 123
If admin restricts User Web Access, only admin login is allowed.
Go to Settings → Advanced Settings → System Upgrade → Firmware. Enter the TFTP/HTTP/HTTPS server URL and apply settings. A reboot may be required to start the upgrade.
The phones support TFTP, HTTP, HTTPS, FTP, and FTPS for both upgrading firmware and downloading configuration files.
During boot, the phone automatically requests configuration files named cfgxxxxxxxxxxxx or cfgxxxxxxxxxxxx.xml, based on its MAC address. If these are unavailable, it attempts to download a generic cfg.xml file. Provisioning can be done over TFTP or HTTP(S).
They include SRTP and TLS for encrypted voice and signaling, 802.1x authentication, secure user/admin passwords, and AES-encrypted configuration files. An on-screen icon indicates when both signaling and media encryption are active.
Under Maintenance → Security Settings, admins can:
- Restrict what users can change from the keypad (unrestricted, basic-only, constrained, or locked).
- Require passwords for factory resets.
- Validate server certificates for secure connections.
- Upload custom TLS certificates or SSH public keys.
- Set allowed TLS versions and disable weak cipher suites.
- Control web access mode, allowed ports, session timeout, and login attempt limits.
GDS37xx Integration
In Peering Mode, the WP810/22/25 and the GDS37xx communicate directly over the LAN using their IP addresses. The GDS37xx is configured with the handset’s IP address and SIP port (5060 for Account 1, 5062 for Account 2). When visitors press the doorbell, the GDS37xx places a direct video call to the handset, and the user can enter the Remote PIN to open the door during the active call.
To enable Direct IP calls, log into the GDS37xx Web GUI, go to SIP Settings → SIP Advanced Settings, and enable Direct IP Call. Then, in Door System Settings, set Call Doorbell Number and enter the handset’s IP address with the appropriate SIP port (e.g., 192.168.5.124:5060) so the GDS can contact the WP810/22/25 directly.
On the WP810/22/25 Web UI, go to Account → General Settings and set Account Active = Yes.
Then go to Account → SIP Settings and confirm that Local SIP Port matches the port configured on the GDS37xx side. Once aligned, the handset will accept direct calls from the door system.
When using a SIP server, both the GDS37xx and the WP810/22/25 must register their SIP accounts to the PBX. The GDS is set to Call Doorbell Number, which would be the handset’s extension (e.g., 1001). When visitors press the doorbell, the PBX routes the INVITE to the WP810/22/25, and the user can answer the call and remotely open the door using the configured Remote PIN.
On the WP810/22/25 Web GUI, navigate to Value-added Service → Door System, click Add, and enter:
- Door System Name (e.g., GDS3710)
- System Number (the GDS SIP extension, e.g., 1000)
- Access Password (must match the GDS’s “Remote PIN to Open Door”).
- During a live call from the GDS37xx, the handset will display an “Open” button. Pressing it sends the SIP command that triggers the door relay.
Maintenance & Diagnostics
Under Menu → Diagnostics, you can test the LCD, LED, keypad, audio loopback, proximity sensor, vibration motor, and certificate verification. These help identify hardware or configuration issues.
Navigate to Settings → Advanced Settings → Factory Reset and confirm. This restores all settings to default and removes any custom configuration.
Do not power off or unplug the phone during upgrading. If the wrong firmware is used, the phone will display an error message to indicate the failure.
The Maintenance → Syslog page allows you to choose the protocol (UDP or TLS), syslog server, and logging level (DEBUG, INFO, WARNING, or ERROR). You can also filter logs by keyword or include SIP messages in the logs.
For deeper debugging, Maintenance → Packet Capture lets you capture traffic internally or to USB, choose timed or continuous capture, and even capture by Wi-Fi channel using Monitor Mode.
You can diagnose Wi-Fi or connectivity problems by navigating to Menu → Status → Network Statistics. This screen displays the connected SSID, BSSID, signal strength, link speed, frequency, IPv4/IPv6 details, and packet statistics such as TX/RX errors and drop rates. Reviewing these values helps identify weak signal, interference, or access point issues.
Go to Menu → Status → Account Status to check whether each account shows Registered or Unregistered. If unregistered, verify the SIP server address, username, authentication ID, and password under Settings → Account (LCD) or Account → General Settings (Web UI). Also ensure the correct Local SIP Port is configured and confirm that the network settings (IP, DNS, gateway) are correct under Status → Network Status.
You can view all packet statistics, including TX/RX packets, error packets, error rate, drop packets, and drop rate, by navigating on the handset to:
Menu → Status → Network Statistics.
This screen also shows the connected SSID, BSSID, signal strength, link speed, channel, and frequency, which helps diagnose Wi-Fi and performance issues.