Processor

2.2GHz Octa-core 64-bit processor

Operating System

Android 13 , Google Mobile Services certified

Memory

RAM: 4GB ROM: 64GB

Interface

Type-C 2.0 OTG;

Support USB Type-C headset

Protocol/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS,ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, SSH,TFTP, NTP, STUN, SIMPLE, LDAP, TR-069, 802.1x, TLS, SRTP, IPv6

Wi-Fi

Dual-band (2.4GHz and 5GHz) Wi-Fi 6 (IEEE 802.11a/b/g/n/ac/ax)

2.4G supports 20/40MHz bandwidth, 5G supports 20/40/80/160MHz bandwidth

Voice Codecs and Capabilities

Support for G.711μ /a, G.729A/B, G.722 (wide-band), iLBC, Opus, in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO),VAD, CNG, AEC, PLC,AJB,AGC, ANS

Video Codecs and Capabilities

H.264 BP/MP/HP, video resolution up to 1080p, frame rate up to 30 FPS, bit rate up to 20Mbps, 3-way video conference (1080p at 30 FPS), anti-flicker capability, auto focus and auto exposure

Customizable Function Keys

Customizable button for scan, push-to-talk, panic call, and other related functions

Bluetooth

Yes, Bluetooth 5.2, supports Bluetooth Low Energy (BLE)

CMOS (≥5 mil)

Scanner

1D: Code128, Code 49, Code 16K, (GS1128) UCC/EAN-128, AIM-128, EAN-8, EAN-13, UPC-E, UPC-A, ITF, ITF 6,

ITF 14, Matrix 2 of 5, Industrial 25, Standard 2 of 5, Code39, ISSN, ISBN, CodaBar, Code93, Code 11, Plessey, MSI Plessey, RSS

2D: Aztec, Composite, CS Code, Maxicode, Micro PDF, Micro QR, PDF 417, QR Code, Data Matrix, DotCode.

Near-Field Communication

(NFC)

13.56MHz RFID

ISO14443A/B, ISO15693,ISO18000-3, MIFARE, FeliCa RF, NFC Forum Type 1-4 Tag

Telephony Features

Hold, transfer, forward, 6-line audio conference, downloadable phonebook (XML, LDAP, up to 1000 items), call waiting, call log (up to 1000 records), auto answer, click-to-dial, flexible dial plan, hot desking, personalized music ring-tones and music on hold, server redundancy and fail-over, push-to-talk

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES based secure configuration file, SRTP, TLS, HTTPS, 802.1x media access control

HD Audio

Yes, both on handset and speakerphone, supports wideband audio

QoS

802.11e and Layer 3 (ToS, DiffServ, MPLS) QoS

Multi-language

English, Arabic, Chinese, Czech, Dutch, German, French, Hebrew, Italian, Japanese, Polish, Portuguese, Russian, Spanish, Turkish, and more

Upgrade/Provisioning

Firmware upgrade via HTTP/HTTPS, mass provisioning using TR-069 or encrypted XML configuration file, manual upload

Display

5.5 inch (1440×720) multi-point capacitive touch screen; Up to 450 NIT (brightness), display backlight adjusts automatically

Camera

Rear camera: 13MP, auto focus, with LED flashlight Front camera: 5MP, fixed focus

GPS

Supports GPS, Galilieo, and Beidou

SD Card

Supports Micro SD cards (up to 256GB)

AC Adapter

Quick charger (Output: 5V-3A/9V-2A/12V-1.5A Input: AC 100~240V, 50/60Hz)

Peripherals

Volume button(+,-), Push-To-Talk Button, Left/Right Scan Button, Power button, Vibration motor, Multi-color LED

Battery

5000mAh 3.85V rechargeable battery; 200 hours standby time and 20 hours talk time (the battery can be removed and replaced)

Microphone and Speaker

Dual microphones, HD speaker and speakerphone (1W)

Sensors

Light & Proximity sensor, accelerometer sensor, gyroscope sensor

Physical

Headset dimensions: 161x 74 x 15mm Handset weight: 0.28KG

Temperature and Humidity

Operating Temperature: -20 °C to 50 °C (-4°F to 122°F); Charging Temperature: 5-40 °C (41°F to 104°F); Storage Temperature: -20 °C to 60 °C (-4°F to 140°F);

Package Contents

Handset unit, power adapter, quick start guide, hand rope, protective shell, 3x plug adapter, Lithium battery, USB cable

Ruggedization and Protection

Dropsafe up to 1.5m height when dropped on to concrete Casing: IP65-rated waterproof and dustrproof

Static Discharge: ±15 kV (air discharge), ±8 kV (direct discharge)

Compliance

FCC, CE, EAC, IC, RCM

WP856

Key

Description

1

TF card

Slot for inserting a TF (TransFlash) or microSD card for additional storage.

2

PTT Button

Programmable key for quick access to specific functions or applications.

3

Left scan key

Button for initiating barcode or QR code scanning.

4

Dock connector

Interface for connecting the device to a docking station for charging or data transfer.

5

Charging port

Port for connecting a charger to power the device.

6

Front camera

Camera located on the front for taking selfies or video calls.

7

Earpiece

Speaker for listening to phone calls.

8

LED indicator

Light to signal notifications, charging status, or other alerts.

9

Camera

Rear camera for taking photos and videos.

10

Flashlight

LED light for illumination, used with the rear camera.

11

Volume keys

Buttons to increase or decrease the audio volume.

12

Power buttom

Button to power on/off the device or wake it from sleep mode.

13

Right scan key

Button for initiating barcode or QR code scanning.

14

Scan window

The area where the device scans barcodes or QR codes.

15

Speaker

Speaker for audio output, such as media playback or speakerphone calls.

16

Battery back cover latch

Mechanism to release the back cover to access the battery.

User Level

Username

Password

Web Pages Allowed

End User Level

user

123

Only Status, Phone Settings, Network Settings, System Settings, Application  settings, and Maintenance settings with limited access

Administrator Level

admin

All pages

Account

Displays the list of accounts supported by the WP856, the device supports up to 6 accounts

SIP User ID

Displays the registered SIP user ID.

SIP Server

Displays the SIP server address

Status

Shows whether the device is registered or not.

MAC Address

Displays the global unique ID of device, in HEX format. The MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the back of the device

NAT Type

Displays the type of NAT connection used by the device.

IPv4 Address Type

Displays The configured address type: DHCP, Static IP or PPPoE.

IPv4 Address

Displays The IP address of the device.

Subnet Mask

Displays Subnet mask of the device.

Gateway

Displays Default gateway of the device.

DNS Server 1

Displays DNS Server 1 of the device.

DNS Server 2

Displays DNS Server 2 of the device.

IPv6 Address Type

Displays The configured address type: DHCP, Static IP.

IPv6 Address

Displays The IPv6 address obtained on the device.

IPv6 Gateway

Displays IPv6 gateway of the device.

IPv6 DNS Server 1

Displays IPv6 DNS Server 1 of the device.

IPv6 DNS Server 2

Displays IPv6 DNS Server 2 of the device.

Product Model

Displays the Product model of the device.

Hardware Version

Displays the Hardware version number.

Part Number

Displays the Product part number.

Serial Number

Displays the Product Serial number.

System Version

Displays the Firmware version. This is the main software release version.

Boot Version

Displays the Booting code version.

Kernel Version

Displays the Kernel version

CPE Version

Displays the CPE Version

Certificate Type

Displays the certificate type of the device.

System Up Time

Displays the total running time of the device since last reboot.

Account x → General Settings

Account Registration

Account Active

Enables/Disables the SIP Account.

Account Name

The name associated with each account to be displayed on the LCD. (e.g., MyCompany)

SIP Server

The URL or IP address, and port of the SIP server. This is provided by your VoIP service provider (e.g., sip.mycompany.com, or IP address)

Secondary SIP Server

The URL or IP address, and port of the SIP server. This will be used when the primary SIP server fails

SIP User ID

User account information, provided by your VoIP service provider.

SIP Authentication ID

SIP service subscriber’s Authenticate ID used for authentication. It can be identical to or different from the SIP User ID.

SIP Authentication Password

The account password required for the phone to authenticate with the SIP server before the account can be registered.

After it is saved, this will appear as hidden for security purpose.

Display Name

The SIP server subscriber’s name (optional) that will be used for Caller ID display (e.g., John Doe).

TEL URI

If the phone has an assigned PSTN telephone number, this field should be set to “user=phone”. A “user=phone” parameter will be attached to the Request-URI and “To” header in the SIP request to indicate the E.164 number. If set to “Enable”, “tel:” will be used instead of “sip:” in the SIP request.

Voice Mail Access Number

Allows users to access voice messages by pressing the MESSAGE button on the phone. This value is usually the VM portal access number.

Network Settings

Outbound Proxy

The IP address or domain name of the main outbound proxy, media gateway, or session border controller. This information is utilized by the device to navigate Firewall or NAT obstacles in various network settings. In the event of detecting a symmetric NAT, STUN becomes ineffective, leaving only an outbound proxy capable of resolving the issue.

Secondary Outbound Proxy

The IP address or domain name of the Secondary Outbound Proxy, Media Gateway, or Session Border Controller. This secondary outbound proxy comes into play when the primary one encounters a failure.

DNS Mode

This parameter controls how the Search Appliance looks up IP addresses for hostnames.

Max Number Of Sip Request Retries

Sets the maximum number of retries for the device to send requests to the server. In DNS SRV configuration, if the destination address does not respond, all request messages are resent to the same address according to the configured retry times. Valid range: 1-10.

DNS SRV Failover Mode

Configures the preferred IP mode for DNS SRV. If set to “default”, the first IP from the query result will be applied. If set to “Saved one until DNS TTL”, previous IP will be applied before DNS timeout is reached. If set to “Saved one until no response”, previous IP will be applied even after DNS timeout until it cannot respond.

If the option is set with “default”, it will again try to send register messages to one IP at a time, and the process repeats.

If the option is set with “Saved one until DNS TTL”, it will send register messages to the previously registered IP first. If no response, it will try to send one at a time for each IP. This behavior lasts if DNS TTL (time-to-live) is up.

If the option is set with “Saved one until no responses”, it will send registered messages to the previously registered IP first, but this behavior will persist until the registered server does not respond.

 If “Failback follows failback expiration timer” is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until the failback timer expires.

Failback Expiration (m)

Specifies the duration (in minutes) since failover to the current SIP server or Outbound Proxy before making failback attempts to the primary SIP server or Outbound Proxy.

Register Before DNS SRV Failover

Configures whether to send REGISTER requests to the failover SIP server or Outbound Proxy before sending INVITE requests in the event of a DNS SRV failover.

NAT Traversal

Configures whether NAT traversal mechanism is activated. Please refer to user manual for more details.

If set to “STUN” and STUN server is configured, the phone will route according to the STUN server. If NAT type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone will try to use public IP addresses and port number in all the SIP&SDP messages.

The phone will send empty SDP packet to the SIP server periodically to keep the NAT port open if it is configured to be “Keep-alive”. Configure this to be “No” if an outbound proxy is used. “STUN” cannot be used if the detected NAT is symmetric NAT. Set this to “VPN” if OpenVPN is used.

Proxy-Require

A SIP Extension to notify the SIP server that the phone is behind a NAT/Firewall.

Account x → SIP Settings

Basic Settings

SIP Registration

Selects whether the phone will send SIP Register messages to the proxy/server. The default setting is “Enabled”.

Unregister Before New Registration

REGISTER Expiration (m)

Specifies the frequency (in minutes) in which the phone refreshes its registration with the specified registrar.

The maximum value is 64800 minutes (about 45 days). The default value is 60 minutes.

SUBSCRIBE Expiration (m)

Specifies the frequency (in minutes) in which the phone refreshes its subscription with the specified registrar.

The maximum value is 64800 minutes (about 45 days). The default value is 60 minutes.

Re-Register before Expiration (s)

Specifies the time frequency (in seconds) that the phone sends re-registration request before the Register Expiration. The default value is 0.

Registration Retry Wait Time (s)

Specifies the interval to retry registration if the process is failed. The valid range is 1 to 3600. The default value is 20 seconds.

Add Auth Header on Initial REGISTER

If enabled, the phone will add Authorization header in initial REGISTER request.

Default is “Disabled”.

Enable SIP OPTIONS Keep Alive

Configures whether to enable SIP OPTIONS to track account registration status. If enabled, the phone will send periodic OPTIONS messages to server to track the connection status with the server.

Default is “Disabled”.

SIP OPTIONS Keep Alive Interval (s)

Configures the time interval the phone sends OPTIONS message to the server. If set to 30 seconds, it means the phone will send an OPTIONS message to the server every 30 seconds.

SIP OPTIONS Keep Alive Maximum Tries

Configures the maximum number of times the phone will try to send OPTIONS message consistently to server without receiving a response. If set to “3”, the phone will send OPTIONS message 3 times. If no response from the server, the phone will re-register.

SUBSCRIBE for MWI

When set to “Yes”, a SUBSCRIBE for Message Waiting Indication will be sent periodically.

The default setting is “No”.

Use Privacy Header

Configures whether the “Privacy Header” is present in the SIP INVITE message.

The default setting is “default”.

Use P-Preferred- Identity Header

Configures whether the “P-Preferred-Identity Header” is present in the SIP INVITE message.

Use P-Access-Network-Info Header

Configures to use P-Access-Network-Info header in SIP request.

Default setting is “Yes”.

Use P-Emergency-Info Header

Configures to use P-Emergency-Info header in SIP request.

Default setting is “Yes”.

Use P-Asserted-Identity Header

Includes P-Asserted-Identity header in SIP request.
Disabled By Default.

Use MAC Header

The default setting is “No”.

Add MAC in User-Agent

The default setting is “No”.

SIP Transport

Selects the network protocol used for the SIP transport.

The default setting is “UDP”.

Local SIP Port

Configures the local SIP port used to listen and transmit.

SIP URI Scheme when using TLS

Specifies if “sip” or “sips” will be used when TLS/TCP is selected for SIP Transport. The default setting is “sips”.

Use Actual Ephemeral Port in Contact with TCP/TLS

Configures whether the actual ephemeral port in contact with TCP/TLS will be used when TLS/TCP is selected for SIP Transport.

The default setting is “No”.

Support SIP Instance ID

Configures whether SIP Instance ID is supported or not.

The default setting is “Yes”.

SIP T1 Timeout

SIP T1 Timeout is an estimate of the round-trip time of transactions between a client and server. If no response is received the timeout is increased and request re-transmit retries would continue until a maximum amount of time define by T2. The default setting is 0.5 seconds.

SIP T2 Timeout

SIP T2 Timeout is the maximum retransmit time of any SIP request messages (excluding the INVITE message). The re-transmitting and doubling of T1 continues until it reaches the T2 value. Default is 4 seconds.

SIP Timer D Interval

Sets the time interval of SIP Timer D. This timer specifies the wait time of response retransmissions when the client receives 3xx ~ 6xx response to an INVITE. The valid range is 0-64 seconds. If set to 0, the parameter will not take effect. The true time interval is equal to T1*64.

Remove OBP From Route

Configures to remove outbound proxy from route. If set to “Enabled”, the SIP account will notify the server to remove the proxy in NAT/Firewall environment. If set to “Always”, the SIP account will notify the server to remove the proxy unconditionally.

Enable 100rel

The use of the PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional responses (1xx series). This is very important in order to support PSTN internetworking. To invoke a reliable provisional response, the 100rel tag is appended to the value of the required header of the initial signaling messages.
Disabled by Default

SIP Registration Failure Retry Wait Time upon 403 Forbidden 

This parameter sets the wait time before an IP phone retries SIP registration after receiving a 403 Forbidden error, preventing rapid reattempts.
In seconds. Between 0-3600, default is 1200. 0 means stop retry registration upon 403 response.

Session Timer

Enable Session Timer

Configures whether to enable session timer function. It enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE). If there is no refresh via an UPDATE or re-INVITE message, the session will be terminated once the session interval expires. If set to “Yes”, the phone will use the related parameters when sending session timer according to “Session Expiration”. If set to “No”, session timer will be disabled.

The default setting is “No”.

Session Expiration (s)

Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand.

The default setting is 180. The valid range is from 90 to 64800.

Min-SE (s)

The minimum session expiration (in seconds). The default value is 90 seconds. The valid range is from 90 to 64800.

UAC Specify Refresher

As a caller, select UAC to use the phone as the refresher, or select UAS to use the callee or proxy server as the refresher. When set to “Omit”, the refresh object is not specified.

The default setting is “UAC”.

UAS Specify Refresher

As a callee, select UAC to use caller or proxy server as the refresher, or select UAS to use the phone as the refresher.

The default setting is “UAC”.

Caller Request Timer

If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it makes outbound calls.

The default setting is “No”.

Callee Request Timer

If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it receives inbound calls.

The default setting is “No”.

Force Timer

If set to “Yes”, the phone will use the Session Timer even if the remote party does not support this feature. Otherwise, Session Timer is enabled only when the remote party supports it.

The default setting is “No”.

Force INVITE

Select “Yes” to force using the INVITE method to refresh the session timer.

The default setting is “No”.

Account x → Codec Settings

Preferred Vocoder

Preferred Vocoder

(Choice 1 – 8)

Multiple vocoder types are supported on the phone, the vocoders in the list is a higher preference. Users can configure vocoders in a preference list that is included with the same preference order in SDP message.
The vocoders supported are:

Codec Negotiation Priority

Configures the phone to use which codec sequence to negotiate as the callee. When set to “Caller”, the phone negotiates by SDP codec sequence from received SIP Invite. When set to “Callee”, the phone negotiates by audio codec sequence on the phone. The default setting is “Callee”.

Use First Matching Vocoder in 200OK SDP

When set to “Yes”, the device will use the first matching vocoder in the received 200OK SDP as the codec. The default setting is “No”.

iLBC Frame Size

Selects iLBC packet frame size. Users can choose from 20ms and 30ms. The default setting is “30ms”.

G726-32 ITU Payload Type

Payload type for G726-32 in ITU packing mode. Payload 2 remains static, while payload dynamic varies dynamically.

G.726-32 Dynamic Payload Type

Specifies G726-32 payload type. Valid range is 96 to 126.

Opus Payload Type

Specifies Opus payload type. Valid range is 96 to 127. It cannot be the same as iLBC or DTMF Payload Type. Default value is 123.

DTMF

Specifies the mechanism to transmit DTMF digits. There are 3 supported modes:

Default setting is “RFC2833”.

DTMF Payload Type

Configures the payload type for DTMF using RFC2833. Cannot be the same as iLBC or OPUS payload type.

Enable Audio RED with FEC

If set to “Yes”, FEC will be enabled for audio call.

Audio FEC Payload Type

Configures audio FEC payload type. The valid range is from 96 to 126.

The default value is 121.

Audio RED Payload Type

Configures audio RED payload type. The valid range is from 96 to 126.

The default value is 124.

Silence Suppression

If set to “Yes”, when silence is detected, a small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. For codec G.723 and G.729 only.

Default setting is “No”.

Voice Frames Per TX

Configures the number of voice frames transmitted per packet. It is recommended that the IS limit value of Ethernet packet is 1500 bytes or 120 kbps. When configuring this, it should be noted that the “ptime” value for the SDP will change with different configurations here. This value is related to the codec used in the codec table or negotiate the payload type during the actual call. For example, if set to 2 and the first code is G.729, G.711 or G.726, the “ptime” value in the SDP datagram of the INVITE request is 20 ms. If the “Voice Frame/TX” setting exceeds the maximum allowed value, the phone will use and save the maximum allowed value for the selected first codec. It is recommended to use the default setting provided, and incorrect setting may affect voice quality.

The default setting is 2.

Preferred Video Codec

Preferred Video Codec

This parameter allows user to select preferred video codec from the “available” list. The device supports H.264.

Enable Video FEC

If set to “Yes,” FEC (Forward Error Correction) will be activated for the video call.

Enable RFC5168 Support

If set to “Yes”, RFC5168 support will be enabled for video call.

FEC Payload Type

Configures FEC payload type. The valid range is from 96 to 126.

Packetization Mode

Sets the video packetization mode. If set to “Single NAL Unit Mode”, the packetization mode will be negotiated as single NAL unit mode for video calls and used for video encoding regardles if the other party support the negotiation. If set to “Non-Interleaved Mode”, the packetization mode will be negotiated as Non-interleaved mode for video calls and used for video encoding regardless if the other party supports the negotiation. If set to “Prefer Non-Interleaved Mode”, the packetization mode will prioritize Non-interleaved mode to be negotiated for video calls but if the other party does not support this，the device negotiate “Single NAL Unit Mode”.

H.264 Image Size

Select the H.264 image size from “720P” , “4CIF”, “VGA”, “CIF” , “QVGA” or “QCIF”.

Use H.264 Constrained Profiles

Configures whether to use H.264 CBP to establish video call with WebRTC. The function takes effect when H.264 profile setting includes BP type. It is recommended to set to “Yes” when establish video call with WebRTC.

H.264 Profile Type

Select the H.264 profile type from Baseline Profile, Main Profile, High Profile, or BP/MP/HP.

Lower-level profiles are easier to decode, while higher-level profiles provide better compression ratio. Typically, High Profile is recommended for achieving higher video compression ratio. For devices with lower CPU performance, choose Baseline Profile for smoother playback.

In high-quality video conferencing scenarios, BP/MP/HP is often used, where all three profiles are negotiated during video decoding to achieve optimal video performance.

Video Bit Rate

The video bit rate can be adjusted based on the network environment. Increasing the video bit rate may improve video quality if the bandwidth is permitted. If the bandwidth is not permitted than the video quality will decrease due to packet loss.

SDP Bandwidth Attribute

Select the SDP bandwidth attribute from “Standard” , “Media Level” , “Session Level” or “None”.
Standard: Use AS at the session level and TIAS at the media level.
Media Level: Use AS at the media level.
Session Level: Use AS at the session level.
None: Do not change the format.
The default setting is “Media Level”. Please do not change the format or it may cause decode failure if unclear about what format the server supports.

H.264 Payload Type

Enter H.264 codec payload type. The valid range is from 96 to 127.

Packet Retransmission

When the function is enabled, signaling will carry RTX information, if the final negotiation is successful, the related media RTX function will realize packet loss retransmission purpose. When this function is disabled. then packet loss retransmission cannot be used.
It is set to NACK+RTX(SSRC-GROUP) by default.

RTP Settings

SRTP Mode

Enable SRTP mode based on your selection from the drop-down menu.

The default setting is “Disabled”.

SRTP Key Exchange 

Selects SRTP key exchange method, the options are:

By Default, the SRTP Key Exchange is set to SDES

SRTP Key Length

Allows users to specify the length of the SRTP calls. Available options are:

Default setting is AES 128&256 bit

Enable SRTP Key Life Time

Enable or disable the crypto lifetime when using SRTP. If users set to disable this option, phone does not add the crypto lifetime to SRTP header. The default setting is “Yes”.

RTCP Keep-Alive Method

Configures the RTCP channel keep-alive packet type.

RTP Keep-Alive Method

Configures the RTP channel keep-alive packet type.

RTCP Destination

Configures the server address. When there is a call, the RTCP package sent from the device will also be sent to this address. Note: The address should contain port number.

Symmetric RTP

Configures whether Symmetric RTP is used or not. Symmetric RTP means that the UA uses the same socket/port for sending and receiving the RTP stream. The default setting is “No”.

RTP IP Filter

Configures whether to filter the received RTP. If set to “Disabled”, the device will receive RTP from any address. If set to “IP Only”, the device will receive RTP from certain IP address in SDP with no port limited. If set to “IP and Port”, the device will only receive RTP from IP address & port in SDP.

RTP Timeout (s)

Configures the RTP timeout of the phone. If the phone does not receive the RTP packet within the specified RTP time, the call will be automatically disconnected. The default range is 0 and 6-600. If set to 0, the phone will not hang up the call automatically.

Account x → Call Settings

Call Features

Start Video Automatically 

Configure whether to enable video automatically when auto answering video calls. If set to “No”, video needs to be turned on manually.
Enabled by default.

Remote Video Request

Configures the preference of video request handling during an audio call. Users could select “prompt”, “accept” or “deny”.
Set to prompt by default.

Auto-answer

If set to “Yes”, the device will automatically answer incoming calls. If set to “Intercom/Paging Only”, it will answer the call based on the SIP Call-Info or Alert-Info header sent from the server/proxy.

Play Warning Tone for Auto Answer Intercom

Plays Warning Tone for Auto Answer Intercom

Intercom Barging

Configures whether to auto answer incoming intercom call when there is already an active call on the phone. When “Intercom Barging” is enabled and the current active call is an intercom call, then the incoming intercom call will be automatically rejected. If the current active call is not an intercom call, then the current active call will be put on hold and the incoming intercom call will be automatically answered. When “Intercom Barging” is disabled, a prompt will be used to indicate the incoming intercom call without interrupting the current active call.
This option is disabled by default.

Auto Preview

Configures whether to turn on video to preview the video of the caller. If set to “Yes”, the user can view the video and hear the caller on the incoming page when there is an incoming call. If set to “Yes with Ringing”, the caller can view the video of the caller and hear the ringtone on the incoming page, but cannot hear the caller. Note: If Auto Answer function has been enabled, this function does not take effect.
This option si disabled by default.

Send Anonymous

If set to “Yes”, the “From” header in outgoing INVITE messages will be set to anonymous. Default is “No”.

Reject Anonymous Call

If set to “Yes”, anonymous calls will be rejected.

The default setting is “No”.

Call Log

Configures Call Log setting on the phone.

The default setting is “Log All”.

Enable Call Features

If set to “Yes”, call features (including anonymous call, DND and etc) will be supported locally instead of using the feature code supported on SIP server/proxy. Please refer to user manual for more details.

Mute on Answer Intercom Call

When enabled, device will mute the incoming intercom call by Call-Info/Alert-Info.

Transfer On 3-Way Conference Hangup

Defines whether or not the local conference call is transferred to the other parties if the initiator of the conference hangs up.
Disabled by Default

Use # as Dial Key

Allows users to configure the “#” key as the “Send” key. If set to “Yes”, the “#” key will immediately dial out the input digits. In this case, this key is essentially equivalent to the “Send” key. If set to “No”, the “#” key is treated as part of the dialed string.
Enabled by Default.

Use # as Redial Key

Allows users to configure the “#” key as the “Redial” key. If set to “Yes”, the “#” key will immediately redial the last call. In this case, this key is essentially equivalent to the “Redial” key. If set to “No”, the “#” key is treated as part of the dialed string.

DND Call Feature On

Configuring the DND feature code. When the DND turned on, the feature code will be sent to server, then the server synchronously enables the DND function.

DND Call Feature Off

Configuring the DND feature code. When the DND turned off, the feature code will be sent to server, then the server synchronously disables the DND function.

No Key Entry Timeout (s)

This is used to set the time length before dialing the entered digits automatically when no key operation is detected.
The Default value is 4 seconds

Ring Timeout (s)

Defines the timeout (in seconds) for the rings on no answer.
The default value is 60 seconds

Refer-to Use Target Contact

When this feature is enabled, the phone uses the contact address of the target you are transferring the call to, instead of the address provided in the Refer-To header during the call transfer process.
This option is disabled by default.

RFC2543 Hold

If yes, c=0.0.0.0 will be used in INVITE SDP for hold.

Call Forwarding

Call Forwarding

Specifies the Call Forward Type. Select “Disabled” to disable call forward feature. Select “Unconditional” to forward all calls to a particular number. Select “Time based” to set a time range for the call to be forwarded. Or select “Others” to set Call Forward On No Answer and Call Forward On Busy.
Disabled by default

Dial plan

Dial Plan Prefix

Configures a prefix added to all numbers when making outbound calls.

Disable Dialplan

Defines whether to disable dial plan of the Dial Page, Contacts, Incoming Call History, Outgoing Call History, Programmable Key & Click2Dial functions. If set to “Yes”, the corresponding dial plan of the function will be disabled.

Dial Plan

Configures the dial plan rule. For syntax and examples, please refer to user manual for more details.

Dial Plan Rules:

Allow 311, 611, and 911 or any 11 digit numbers with leading digits 1617;

Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit numbers;

Allows any number with leading digit 1 followed by a 3-digit number, followed by any number between 2 and 9, followed by any 7-digit number OR Allows any length of numbers with leading digit 2, replacing the 2 with 011 when dialed.

Explanation of example rule (reading from left to right):

Note: In some cases, where the user wishes to dial strings such as *123 to activate voice mail or other applications provided by their service provider, the * should be predefined inside the dial plan feature. An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of numbers.

Max length of dial plan is up to 1024 characters.
 

Caller IDs

Caller ID Display

When set to “Auto”, the phone will look for the caller ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and From Header in the incoming SIP INVITE. When set to “Disabled”, all incoming calls are displayed with “Unavailable”.

Callee ID Display

When set to “Auto”, the phone will update the callee ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and To Header in the 180 Ringing. When set to “Disabled”, callee id will be displayed as “Unavailable”. When set to “To Header”, caller id will not be updated and displayed as To Header.

Ringtone

Account RingTone

Allows users to configure the ringtone for the account. Users can choose from different ringtones from the dropdown menu.

Note: User can also choose silent ring tone or doorbell.

Ignore Alert-Info header

Configures to play default ringtone by ignoring Alert-Info header.

The default setting is “No”.

Match Incoming Caller ID

Specifies matching rules with number, pattern, or Alert Info text (up to 10 matching rules). When the incoming caller ID or Alert Info matches the rule, the phone will ring with selected distinctive ringtone. Matching rules:

xx+ : at least 2-digit number.

xx : only 2-digit number.

[345]xx: 3-digit number with the leading digit of 3, 4 or 5.

[6-9]xx: 3-digit number with the leading digit from 6 to 9.

Users could configure the matching rule as certain text (e.g., priority) and select the custom ring tone mapped to it. The custom ring tone will be used if the phone receives SIP INVITE with Alert-Info header in the following format: Alert-Info: <http://127.0.0.1>; info=priority  When the incoming caller ID or Alert Info matches one of the 10 rules, the phone will ring with the associated ringtone.

Note: Beginning with firmware version 1.0.3.98, a new feature was introduced that enables the use of a ringtone stream via a remote URL. The functionality of this feature works as follows: the following audio file named test.wav is uploaded onto an HTTP server and the remote URL is “http://192.168.5.165:8080/test.wav;info=ring3”, the IP phone then attempts to use the provided URL first to play the ringtone. If the URL is not functional for some reason, it will then use the info=ring3 parameter, as the default ringtone.

Account x → Advanced Settings

Security Settings

Check Domain Certificates

Configures whether the domain certificates will be checked when TLS/TCP is used for SIP Transport. The default setting is “No”.
Note: The minimum supported TLS version has been updated to 1.2 or higher.

Trusted Domain Name List

Fill in the list of trusted domain names, which supports filling in the SAN list used only for domain name verification in TLS to obtain certificates. If it matches any item in the trusted domain name list, the certificate is trusted. By default, the remote proxy domain name and SIP server domain name are trusted.Supports filling in numbers/letters/-/./*, Support setting wildcard domain names, such as “*.grandstream.com”, and trust any domains that end with “.gradstream.com”.

Validate Certificate Chain

Validates CA certificate when TLS/TCP is configured for SIP.
Disabled by default

SIP CA Certificate

Select the CA certificate for server verification.

SIP User Certificate

Select the user certificate to access SIP TLS authentication content required by some specific servers. If the private key is included, upload it with the user certificate.

Validate Incoming SIP Messages

Defines whether the incoming SIP messages will be validated or not.

Allow Unsolicited REFER

 The “Allow Unsolicited REFER” option allows the handset to accept, deny, or require authentification for external REFER requests coming from the SIP server, these requests automatically transfer calls to other parties without user intervention, the available options are:

The default value is Disabled.

Accept Incoming SIP from Proxy Only 

When set to “Yes”, the SIP address of the request URL in the incoming SIP message will be checked. If it doesn’t match the SIP server address of the account, the request will be rejected.

Check SIP User ID for Incoming INVITE

If set to “Yes”, SIP User ID will be checked in the Request URI of the incoming INVITE. If it doesn’t match the device’s SIP User ID, the call will be rejected.

Allow SIP Reset

Allow SIP Notification message to perform factory reset.

The default setting is “No”.

Authenticate Incoming INVITE

If set to “Yes”, the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response.

The default setting is “No”.

SIP Realm Used for Challenge INVITE & NOTIFY

Configure this option to set the SIP server used to validate incoming INVITE or NOTIFY (for check-sync, resync, reboot). Only takes effect when Authenticate Incoming INVITE or SIP NOTIFY Authentication is enabled.

MOH

Upload Local MOH Audio File

Configures to play reminder tone when the call is on hold.

Click BROWSE to upload audio file from PC. The MOH audio file should be “.wav” or “.mp3” format. It may take a long time uploading and processing the file, please be patient.

Enable Local MOH

If set to “Yes” , the local MOH will be enabled. Users need to upload local MOH audio file. Once enabled, users could play the file when holding the call.

Advanced Features

Special Feature

Different soft switch vendors have special requirements. Therefore users may need select special features to meet these requirements. Users can choose from Standard, CBCOM, RNK, China Mobile, ZTE IMS, Mobotix, ZTE NGN, or Huawei IMS depending on the server type.

Feature Key Synchronization

This feature is used for BroadSoft or Metaswitch call feature synchronization. When set to “BroadSoft/Metaswitch”, DND and Call Forward features can be synchronized with the server. The local call forward function are disabled when this feature is active.

Conference URI

Refers to the Uniform Resource Identifier used for initiating or joining a conference call or a multi-party call session. This URI is predefined by the system or the conference server and is used by the WP856 to route conference call requests to the appropriate conferencing server.

Allow Sync Phonebook Via SIP Notify

If set to “Yes”, the device will allow SIP NOTIFY messages to sync local phonebook.
Enabled by default.

SCA

Enable SCA (Shared Call Appearance)

If enabled, Shared Call Appearance (BroadSoft standard) feature will be applied to the registered account.

Enable Barge-in

If enabled, users can barge into an active call on a shared line.

Line-seize Timeout (s)

Configures the interval (in seconds) when the line seize is considered timed out when Shared Line feature is used.

The valid range is: 15-60.

The default value is 15 seconds.

Call Park

BroadSoft Call Park

If enabled, phone will send SUBSCRIBE to BroadSoft server to obtain Call Park notifications.

Call Park Feature Code

Configures the feature access code to park current call on a parking lot or another extension.

Auto-filling Call Park Retrieve Feature Code

If enabled, the Call Park Retrieve feature code will be automatically filled in the dialing interface during call pickup. Default value is “Enable”.

Call Park Retrieve Feature Code

Configure the feature access code for retrieving the parked call on the account or BLF keys.

Basic Settings

Local RTP Port

Defines the Local RTP port used.
Audio RTP port：Port_Value+10*N
Audio RTCP port: Port_Value+10*N+1.
Default Port is 50040

Use Random Port

When set to “Yes”, this parameter will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple devices are behind the same full cone NAT. (This parameter must be set to “No” for Direct IP incoming calls, but IP outgoing calls are not affected)
Disabled by Default.

Enable In-call DTMF Display

Enables DTMF display when in-call.
Enabled by Default.

Enable LDAP Timeout Auto Search

Configures whether to display the matched content automatically in LDAP search when timeout. If set to “No”, users need to click the Search button to search the matched contacts mentioned above.
Enabled by Default

Keep-alive Interval (s)

Specifies how often the device sends a blank UDP packet to the SIP server in order to keep the “pin hole” on the NAT router to open.
Default value is 20 seconds

STUN Server

The IP address or domain name of the STUN server. STUN resolution results are displayed in the STATUS page of the device Web GUI. Only non-symmetric NAT routers work with STUN.

Use NAT IP

The NAT IP address in SIP/SDP messages. This field is blank by default settings. You should ONLY use it when required by your ITSP.

Guest Features

Guest Login

This feature allows the user to register SIP account as a guest. The guest can enter the user ID, password, and PIN, to ensure the validation and account information.

Enable/disable guest login. Users should configure SIP domain name when using this function. Users should input SIP user name and password on LCD after rebooting the phone.

Guest Login PIN Code

When set “Guest Login” option to “Yes” and “Guest Login Timeout” to “Never”, users should input the PIN code to login.

Server Alias

The server alias is used to customize the server name.

Server List

This is the list of SIP domain names under the selected severs. The registration priority of addresses in the list is from top to bottom. Up to 2 SIP domains are supported.

Call Features

Enable Video Call

Enables the video call feature.
Enabled by Default

Enable Direct IP Call Mode

Configures enable/disable direct IP call mode of the device. If set to “Yes”, the feature of direct IP call will be enabled.
Disabled by default

Enable Paging Call Mode 

Configures enable/disable paging call mode of the device. If set to “Yes”, the feature of paging call will be enabled.
Disabled by default

Enable Call Waiting

Enables the call waiting feature.
Enabled by Default

Enable Call Waiting Tone

Enables the call waiting tone when call waiting is on.
Enabled by default

Enable Transfer

Enables the Transfer function.
Enabled by default

Hold Call Before Completing Transfer

If you choose “Yes”, the transfer call status will be kept as hold status after Attended transfer operation.
Enabled by default

Enable Transfer Via Non-Transfer Programmable Key 

Programmable Key with type Speed dial, BLF and Speed dial via active account will perform as transfer programmable keys under active call. The transfer mode during the call depends on the “Default Transfer Mode” mentioned above, and can also be selected on the transfer interface, but the transfer mode for incoming calls only supports blind transfer.
Disabled by default

Auto Unhold on Line Key Press

Configures when there are multiple lines, whether to resume the line automatically when clicking the line on hold, and hold the line in the primary call. Note: if this is a manually hold operation rather than hold from switching lines, the call will not be resumed automatically.
Disabled by default

Record Mode

Configures recording mode. If set to “Record locally”, the device will use the local recorder for call recording, and the audio file will be saved according to the recorder setup. If set to “Record on UCM”, the device will send the recording feature code to the UCM server to request for recording, and the recording function will be executed by the server.

Enable Auto Record When Call Established

Configures whether to auto record when a call is established. If set to “Yes”, the call recording will start automatically when the call is established.
Disabled by default

Enable Noise Block

Configures whether to enable noise block feature. If enabled, all the background noise will be effectively blocked during the call.
Disabled by default.

Enable Local In-call DTMF Tone in Speaker Mode

When using the speaker channel in call, enable the playback effect of the local DTMF sound.
Disabled by default

Enable Headset Noise Shield

Select the appropriate shield level based on the TX loudness of the earphone. The higher the selected shield level, the more environmental noise will be reduced in the transmitted audio, but the more likely it will affect sound quality.

The default setting is set to Disabled.

Enable Handset Noise Shield 2.0

When enabled, the remote party will hear less environmental noise during a call. If set to “High Shield”, most of the environmental noise can be shielded. If set to “Moderate Shield”, some comfortable environmental noise will remain for the remote party.

The default setting is set to High Shield.

Noise Suppression Mode

Configure the noise suppression mode for the TX direction. Classic noise suppression can shield the smooth noise in the environment, while AI noise suppression can shield the smooth noise in the environment and eliminate sudden noise at the same time.

Incoming/Outgoing Call

Auto Mute on Entry

Configures whether to mute the call on entry automatically. If set to “Disabled”, then do not use auto mute function. If set to “Auto Mute on Outgoing Call”, then mute automatically when the other party answers the outgoing call. If set to “Auto Mute on Incoming Call”, then mute automatically when answering the incoming call. If set to “Mute on Incoming & Outgoing Call”, then mute automatically when the call is established.

Note: This function only takes effect when the device is changed from the idle status to call status. Users could click the Mute button on call interface to cancel the current mute status.

Rejected Call Notification

Specify whether to enable rejected call notification. Once enabled, a missed call will prompt on LCD when reject the incoming call.
Disabled by default

Special Function for Incoming Call

Defines the function for the incoming call.
If set to “None”, this function will be disabled.
If set to “Preview”, users could tap on the PREVIEW button in the call interface to check video caller without answering the incoming video call.
If set to “Call Transfer”, the phone system will pop up the “TRANSFER” key on the LCD screen when there is an incoming call, and users could tap on it to show the dialer without answering the incoming call, then, users could transfer this incoming call to others.
Set to “None” by default

Enable Conference

Enables the Conference function.
The option is enabled by default.

Auto Conference

If enabled and the user clicks “Conference ” or “Invite” (in conference) button and there exists more than 1 line but less than 6 lines, the phone will put all lines into the conference room for multi-party conference call; If the current lines are more than 6, the phone will access line selection interface, the user needs to manually choose the line before making a conference call.
This option is disabled by default.

Hold Call Before Adding Conferee

Configures whether to place the current call on hold before adding new member(s) to a conference. If enabled, the current call will be put on hold when the host presses Conference or Add key to invite new member(s). When an invited member answers the call and agrees to attend the conference, the host needs to manually resume the conference with the new member added. If disabled, the current call will not be put on hold and the invited member will join the meeting automatically after answering the call.

Return Code When Refusing Incoming Call

When refusing the incoming call, the phone will send the selected type of SIP message of the call.
The options are:

DND Settings

DND Mode

Enables/Disables DND mode on the phone.

Enable DND Reminder Ring

If enabled, the phone will play a sound when it is set to DND mode. Disabled by default.

Return Code When Enable DND

When DND is enabled, the device will send the selected type of SIP message.

Advanced Settings

Filter Characters

Filter Characters are used to filter the specific separator characters for Click2Dial or contacts imported from other devices.

These specific characters are not part of the actual phone number and needed to filter out. Users could set up multiple characters. For example, if set to “[()-]”, when dialing (0571)-8800-8888, the character “()-” will be automatically filtered and dial 057188008888 directly. Initiate calls from other places except dial screen, such as call history and contacts, will automatically filter the characters. Dialing out from Dial screen will not filter any characters.

Escape ‘#’ as %23 in SIP URI

Replaces ‘#’ by ‘%23’ in some special situations.
Enabled by default.

Auto Config CPT by Region

If set to “Yes”, the phone will configure CPT (Call Progress Tone) according to different regions automatically. If set to “No”, you can manually configure CPT parameters.

Ringback Tone

Configures tone frequencies based on parameters from the local telecom provider. By default, they are set to the North American standard. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds.
Syntax: f1=val, f2=val [, c=on1/off1[-on2/off2[-on3/off3]]];(Frequencies are in Hz and cadence on and off are in 10ms)ON is the period of ringing (‘On time’ in ‘ms’) while OFF is the period of silence. In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern. Up to three cadences are supported.

Busy Tone

Reorder Tone

Call Waiting Tone

Call Waiting Tone Gain

This configures the call waiting tone gain. By default, they are set to the North American standard. Users could adjust the tone frequencies based on parameters from the local telecom provider. Users could select “Low”, “Medium” or “High”.

Default Ring Cadence

The ringing pattern heard by the dialer before the called party picks up the call.

Video Frame Rate

The video frame rate is adjustable based on network condition. Increasing the frame rate will significantly increase the amount of data transmitted, therefore consuming more bandwidth. The video quality will deteriorate due to packet loss if extra bandwidth is not allocated.

Video Display Mode

Set the video display mode to “Original proportion”, “Equal proportional cutting” or “Proportional add black edge”. If set to “Original proportion”, the device displays video in its original proportion that received from remote party, if the remote video display proportion is different from the device, the device will stretch or compress video to display it; If set to “Equal proportional cutting”, the device will cut video to meet its own display proportion; If set to “Proportional add black edge”, the device will display video in its original proportion, if still exists spare space, the device will add black edge on it.

Enable Frame Skipping in Video Decoder

If set to default setting “Yes”, the device will skip the P frame in lost video packet to decode the I frame in the next video packet. This setting helps to reduce video distortion.

Default Video Stream

Select the default video stream enabled during video calls.

General Settings

Allow PTT/Paging When Phone Is Locked

If set to “Yes”, the device can initiate an PTT/Paging in the lock screen status.

Set to “No” by default

IGMP Keep-Alive Interval (s)

Specifies how often the phone report IGMP when PTT/ Paging function is turned on. IGMP reporter help to keep PTT/ Paging receivable in dormant state. The interval may take some effect to standby time.

Disabled value is 30 seconds

PTT/Group Paging

PTT/Group Paging Address

Set the PTT/Group Paging address.

Port

Set port number for PTT/Group paging Address.

Emergency Channel Volume

Set default volume for PTT/paging when emergency channel/group is used. the range is from 1-7

PTT Config

PTT

Configures to enable or disable PTT.

Default Channel

Set default channel for PTT. When presing and holding the PTT button, PTT will be initiated using the default channel.

Priority Channel

Set priority channel for PTT. PTT received on priority channel will take precedence over active PTT on normal channel.

Emergency Channel

Set emergency channel for PTT. Emergency channel has the highest priority. PTT using emergency channel will take precedence over PTT on priority or normal channel. Please note PTT to emergency channel will not be rejected even when device has enabled DND.

Accept While Busy

Configures whether to accept PTT while device is in active call. If set to “No”, device will ignore PTT while in active call. If set to “Yes”, while in active PTT talk, device will accept PTT if it has the same priority; If device is in active SIP call, device will accept PTT and put the SIP call on hold.

Disabled by default

Caller ID

Set Caller ID displayed on the call interface during a PTT call.

PTime (ms)

Set payload size for PTT.

Audio Codec

Set audio codec for PTT.

Channel

Set PTT channel. Users can configure whether to use the channel for transmit, subscribe, join PTT and its label. Only available and joined channel will be displayed in PTT channel list. If users need send or receive PTT, “Transmit” and “Subscribe” must be enabled for this channel.

Paging Config

Group Paging

Configures to enable or disable group paging.

Default Group

Set default paging group. When pressing and holding the PTT button, paging will be initiated using the default group.

Priority Group

Configures priority paging group. Paging received on priority group will take precedence over active paging on normal group.

Emergency Group

Set emergency group for paging. Emergency group has the highest priority. Paging using emergency group will take precedence over paging on priority or normal group.

Accept While Busy

Configures whether to accept paging while device is in active call. If set to “No”, device will ignore paging while in active call. If set to “Yes”, while in active paging call, the device will accept other paging calls if it has the same priority. If device is in an active SIP call, device will accept paging and hang up the SIP call.

Caller ID

Set Caller ID displayed on the call interface during paging.

Default is “channel(*)”

PTime (ms)

Set payload size for paging.

Audio Codec

Set audio codec for paging.

Group

Configures paging group. Users can configure whether to use the group to accept and join group, and its label. Only available and joined group will be displayed in paging group list. If users need receive paging, “Subscribe” must be enabled for this group.

Multicast Paging

Paging Priority Active

If enabled, during a multicast page if another multicast is received with higher priority (1 being the highest) that one will be played instead.

Enabled by Default.

Paging Barge

During an active call if incoming multicast page has higher priority (1 being the highest) than this value, the call will be held and multicast page will be played.

Multicast Paging Codec

The codec for sending multicast pages.

Multicast Paging Function

Enable or disable multicast paging.

Disabled by Default

Multicast Paging Address

Configure the listening address and channel name of multicast paging.

You can configure up to 10 Listening Addresses

Device Name

Display name when paired with other devices.

Bluetooth

Enables/Disables the Bluetooth function.

Bluetooth Devices

Scan available Bluetooth devices for pairing. After enabling the Bluetooth function, the device will automatically scan for available Bluetooth devices.

Wi-Fi Basics

IP Mode

Selects which Internet protocol to use. When both IPv4 and IPv6 are enabled, device attempts to use preferred protocol first and switches to the other choice if it fails.

Wi-Fi Signal Warning

When the Wi-Fi signal strength is lower than the threshold set by this level, the device will display a warning.

Disabled by default

Wi-Fi Function

This parameter enables/disables the Wi-Fi function. The default setting is set to “No”.

Wi-Fi Band

Sets the type of WiFi Band. The default setting is 2.4G&5G.

ESSID

This parameter sets the ESSID for the Wireless network. Press “Scan” to scan for the available wireless network.

Add Network

ESSID

Enter the name of hidden ESSID.

Security Mode for Hidden SSID

This parameter defines the security mode used for the wireless network when the SSID is hidden, the options are: None, WEP, or WPA/WPA2 PSK

Password

Configures the hidden ESSID password.

Advanced Settings

Host Name (option 12)

Specifies the name of the client. This field is optional but may be required by some Internet Service Providers.

Vendor Class ID (option 60)

Used by clients and servers to exchange vendor class ID.

WiFi Roaming

WiFi Roaming Mode

Set the WiFi roaming mode. The default mode is suitable for regular scenarios and has a strong battery life; Boost mode is suitable for scenarios where you need to move around a lot; Fast roaming mode supports 802.11k, 802.11v, 802.11r; Custom mode requires you to manually enter the value yourself.

Signal Threshold

Sets the WiFi signal threshold. When the WiFi signal strength of the device drops below this value, the device will scan for a hotspot above the threshold value and connect to it.

Roaming Gain (dB)

Sets the minimum signal gap of roaming between different APs. When the signal strength difference between the scanned AP and the current is greater than this value, roaming will be triggered.

Good Signal Scanning Interval (s)

Sets the time interval for signal scanning when the WiFi signal strength is higher than the signal threshold.

Poor Signal Scanning Interval (s)

Sets the time interval for signal scanning when the WiFi signal strength is lower than the signal threshold and there is no hotspot which is higher than the current signal strength.

802.11r

Enables faster handoff between Wi-Fi access points by pre-authenticating, reducing interruptions during roaming, ideal for VoIP and video calls.

802.11v

Improves Wi-Fi efficiency by helping devices select better access points and manage power usage, optimizing performance and battery life.

Web/SSH Access

Enable SSH

When set to “Yes”, this option allows remote access to the device via SSH (Secure Shell) from any IP address. SSH access provides command-line control of the device for administrative tasks.

SSH Port

Specifies the port number used for SSH access to the device.

The port used is port 22 and is open by default.

Access Method

Allows users to select HTTP or HTTPS for Web Access.

Web Port

By default, HTTP uses port 80 and HTTPS uses port 443. This field is for customizing the web port.

Enable User Web Access

Administrators can disable or enable the user web access.

It is disabled by default.

Web Server User Certificate

Selects the user certificate as the web server certificate to encrypt web access.

User Login Timeout

Sets login timeout (in minutes) for user. If there is no activity within the specified amount of time, the user will be logged out, and the system will jump to the login page automatically. If set to 0, the user will not be logged out automatically.

Default value is 15 minutes.

User Info Management

Current Admin Password

Enter current login user password. This field is case sensitive.

New Admin Password

Allows the user to change the admin password. The password field is purposely blank after clicking the “Save” button for security purpose. This field is case sensitive with a maximum length of 32 characters.

Confirm New Admin Password

Enter the new Admin password again to confirm.

New User Password

Allows the administrator to set the password for user-level web GUI access. This field is case sensitive with a maximum length of 32 characters.

Confirm New User Password

Enter the new User password again to confirm.

Certificate Management

Trusted CA Certificates

Tusted CA certificates ensure secure communication by verifying the authenticity of SSL/TLS connections. They validate the identity of parties involved, safeguarding against unauthorized access or data breaches. Properly configured, they establish a foundation of trust for encrypted communication, enhancing overall system security, the WP856 supports contains two types of CA certificates:

User Certificates

User certificates serve as digital credentials verifying user identities, granting access based on permissions, and facilitating secure interactions with the panel’s functionalities, bolstering overall system security, you can add a user certificate in PEM, PFX, or P12 format, by clicking the icon “Add”, then defining a label, and an optional password.

Enable TR-069

Enables TR-069 feature.

ACS URL

Defines the URL for TR-069 Auto Configuration Servers (ACS).

ACS Username

Defines the ACS username for TR-069.

ACS Password

Defines the ACS password for TR-069.

Enable Periodic Inform

Enables periodic inform. If set to “Yes”, device will send inform packets to the ACS.

Periodic Inform Interval (s)

Sets up the periodic inform interval in seconds to send the inform packets to the ACS.

Connection Request Username

The user name for the ACS to connect to the device. It should match the configuration in the ACS.

Connection Request Password

The password for the ACS to connect to the device. It should match the configuration in the ACS.

Connection Request Port

The port for the request sent from the ACS to the device. It should not be occupied by other protocol used on the device. For example, it cannot be 5060 or 5004 which are already used for SIP protocol.

CPE CA Certificate

Select the CA certificate for server verification.

CPE User Certificate

Select the user certificate to be used for mutual server authentication. If the private key is included, upload it with the user certificate.

Firmware

Complete Upgrade

If enabled, all files will be replaced except user data.

Disabled by Default

Upload Firmware File to Update

Allows users to load the local firmware to the device to update the firmware.

This setting requires a reboot

Firmware Upgrade Mode

Allows users to choose the firmware upgrade method:  HTTP, HTTPS.

Firmware Server Path

Defines the server path for the firmware server. It could be different from the Config Server Path which is for provisioning.

HTTP/HTTPS Username

Defines the user name for the firmware HTTP/HTTPS server.

HTTP/HTTPS Password

Defines the password for the firmware HTTP/HTTPS server.

Firmware File Prefix

Enables your ITSP to lock firmware updates. Only matching encrypted postfix and(or) suffix, will the firmware be downloaded and flashed into the device.

Firmware File Postfix

Enables your ITSP to lock firmware updates. If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the device.

Firmware Upgrade

 
Firmware Updates: Click the “Update Detect” button to check whether the firmware in the firmware server has an updated version. If so, update immediately.

Config File

Download Device Configuration

Click to download the device configuration file in .txt format.

Upload Device Configuration

Upload configuration file to the device.

Use Grandstream GAPS

It is used to configure the download path and update mode for the configuration file server. If set to “Yes”, the device will set the download path of the configuration file to fm.grandstream.com/gs by default, and use HTTPS protocol to connect to the server; If set to “No”, users can manually configure the path and update mode for the configuration file server.

Config Upgrade Mode

Allows users to choose the config upgrade method:  HTTP or HTTPS.

Config Server Path

Defines the server path for provisioning. It could be different from the Firmware Server Path.

HTTP/HTTPS User Name

The user name for the config HTTP/HTTPS server.

HTTP/HTTPS Password

The password for the config HTTP/HTTPS server.

Config File Prefix

Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted prefix will be downloaded and flashed into the device.

Config File Postfix

Enables your ITSP to lock configuration updates. If configured, only the configuration file with the matching encrypted prefix will be downloaded and flashed into the device.

Authenticate Conf File

Authenticate the configuration file before the device accepts the file.

XML Config File Password

The password for encrypting the XML configuration file using OpenSSL. This is required for the device to decrypt the encrypted XML configuration file.

Provision

Automatic Upgrade

Set automatic upgrade every intervals/day/week. The device will send request to upgrade automatically according to the setup time.

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent.

Upgrade with Prompt

If set to “Yes”, the device will pop up a prompt after downloading the firmware files to confirm whether start upgrading . Otherwise, the device will automatically start upgrading process.

Allow DHCP Option 43, 160 and 66 Override Server

If set to “Yes”, the device will reset the CPE, upgrade, network VLAN tag and priority configuration according to option 43 sent by the server. At the same time, the upgrade mode and server path of the configuration upgrade mode will be reset according to option 160 and 66 sent by the server. If set to “Prefer, fallback when failed”, the device can fallback to use the configured provisioning server under its Firmware and Config server path in case the server from DHCP Option fails.

DHCP Option 120 Override SIP Server

If set to “Yes”, the device will enable DHCP Option 120 from local server to override the SIP Server on the device.

Allow DHCP Option 242 (Avaya IP Phones)

Once enabled, the device will use the configuration info issued by the local DHCP in Option 242 to configure proxy, transport protocol and server path.

Download and Process All Available Config Files

By default, the device will provision the first available config in the order of cfgMAC.xml, cfgMODEL.xml, and cfg.xml(corresponding to device specific, model specific, and global configs). If set to “Yes”, the device will download and apply (override) all available configs in the order of cfgMAC.xml, cfgMODEL.xml, cfg.xml.

Config Provision

The device will download the configuration files and provision by the configured order.

PNP (3CX) Auto Provision

If enabled, the device will send SUBSCRIBE request for automatic assigned URL to the multicast address in LAN when bootup to accomplish automatic configuration of SIP account, It requires 3CX server support.

 Advanced Settings

Send HTTP Basic Authentication by Default

Determine whether to send basic HTTP authentication information to the server by default when using wget to download firmware or config file. If set to “Yes”, send HTTP/HTTPS user name and password no matter the server needs authentication or not. If set to “No”, only send HTTP/HTTPS user name and password when the server needs authentication.

Enable SIP NOTIFY Authentication

Device will challenge NOTIFY with 401 when set to “Yes”.

Validate Server Certificates

Configures whether to validate the server certificate when downloading the firmware/config file. If set to “Yes”, the device downloads firmware/configuration files only from servers validated by CA certificates.

Validate Hostname in Certificate

This option ensures that the hostname of the server you’re connecting to matches the one specified in the server’s SSL/TLS certificate. When enabled, this feature adds an extra layer of security by preventing connections to servers that present certificates with mismatched hostnames.

Disabled by default.

CA Certificate

Select the CA certificate for server verification.

User Certificate

Select the user certificate to be used for mutual server authentication. If the private key is included, upload it with the user certificate.

mDNS Override Server

Enable mDNS to override config/firmware server settings on the phone., available options are: Disabled, Use Type A DNS caching, Use Type SRV DNS caching.

Default is Disabled

Factory Reset

Restore to factory default settings.

Note: Please backup the data to avoid data loss.

Safe Mode

Configures enable/disable safe mode. If enabled, the device will enter safe mode after rebooting, which will help remote troubleshooting when an abnormal situation occurs.

Note: Once entering safe mode, only the system applications will be up and running, all widgets and 3rd party apps will be disabled.

Syslog

Syslog Protocol

Configure sending syslog through UDP or secured SSL/TLS protocol to syslog server.

The default value is “UDP”

Syslog Server

The IP address or URL for the System log server.

The default value is “log.ipvideotalk.com”

Syslog Level

Selects the level of logging for syslog. There are 5 levels: None, DEBUG, INFO, WARNING and ERROR. each level will have specific information displayed on the syslog traces.

The default value is “None”

Syslog Keyword Filter

Only send the syslog with keyword. Multiple keywords are separated by comma. E.g.: set the filter keyword to “SIP” to filter SIP log.

Debug

One-click Debugging

Capture the checked items in the debugging list. If “Capture trace” is selected, click “Start” to start capture and click “Stop” to end. Otherwise, click “Capture” to download. All log files will be generated in a TAR package and the trace file as PCAP.

Debug Info Menu

Display a list of info items that can be debugged. Currently system logs, info log, capture trace, tombstones, ANR log are supported. The captured data can be viewed in “Debug information list”. By default all items are selected.

Debug Info List

Select an existing debug file and click the “Delete” button on the right to delete the file.

View Debug Info

Click “List” to view the existing debugging info package or trace file. The files are listed in chronological order. Click the file to download to computer.

Enable Core Dump Generation

Configures whether to generate and save the core dump file when the program crashes. The default setting is “No”.

Core Dump List

Select the existing core dump file in the drop-down box. Users could click the “Delete” button on the right to delete the file.

View Core Dump

Click “List” to view all existing core dump files. The files are listed in chronological order. Click the file to download to computer.

Capture in Monitor Mode

Capture in Monitor Mode

Enable the Wi-Fi monitoring mode to capture network data in the environment. In this mode, Wi-Fi cannot be connected and accounts cannot be registered. Please choose carefully.

Monitor Mode Channel

When in Wi-Fi monitoring mode, select the monitoring channel.

Monitor Mode Bandwidth

When in Wi-Fi monitoring mode, select the monitoring bandwidth.

Monitor Info List

Select an existing capture file and click the “Delete” button on the right to delete the file.

View Monitor Info

Click “List” to view the existing monitor capture files. The files are listed in chronological order. Click the file to download to computer.

Traceroute

Target Host

The IP address or URL for the Target Host of the Traceroute.

Ping

Target Host

The IP address or URL for the Target Host of the Ping.

NSLookup

Host Name

Enter a host name to look up the corresponding IP address. This feature can also do reverse name lookup and find the host name for the entered IP address.

Bootup Completed

Set the URL for events on device web GUI. When the corresponding event occurs on the device, the device will send the configured URL to SIP server. The dynamic variables in the URL will be replaced by the actual values of the device before sending to SIP server, in order to achieve the purpose of events notification. Here are the standards:

When the corresponding event occurs on the device, the device will send the MAC address and phone number to server address 192.168.40.207.

Incoming Call

Outgoing Call

Missed Call

Connected

Disconnected

DND On

DND Off

Forward On

Forward Off

Blind Transfer

Attended Transfer

On Hold

Unhold

Log On

Log Off

Register

Unregister

Display Format

Configures the display format for the Programmable Key. Users could select “Display Name”, “Number” , “Display Name,Number” or “Display Name,Number,Key mode”.

Show Display Name From Server

If selected, the display name on the server will replace the name users configured.

Key Mode

The key modes are:

Account

Configures the SIP account when the accounts are configured as Programmable Key.

Name

Configures the display name when the accounts are configured as Programmable Key.

User ID

Configures the number for the corresponding Programmable Key mode when the accounts are configured as Programmable Key.

Account

Value-added Services – Value-added Services

Service Type – Door System

The WP856 supports adding up to 50 door control devices, including GDS37xx Door control models and third-party variants. To add a door control device, simply click the “add” icon and proceed to specify the following fields:

Service Type – DTMF

General Settings

Display Open Door Button when Calling

Configures whether to display Open Door button when there is an incoming call. If enabled, the phone will hide the Open Door button and will not allow open door with DTMF if the preview function is disabled.

Enable Preview 

Configures whether to enable preview function or not.

Value-added Services – BroadSoft Directories

XSI Service Settings

Authentication Type 

Defines the authentication type to use for Broadsoft XSI. If set to “Login Credentials”, please fill in the user ID and password in the following fields. If set to “SIP Credentials”, please fill in user ID, auth ID, and password.

Server

Defines BroadSoft XSI server address with protocol.

Port

Port of the BroadSoft XSI server

Action Path

Defines Action Path for BroadSoft XSI server

BroadWorks User ID 

Defines the User ID for BroadSoft XSI server

Login Password

Password for BroadSoft XSI server.

SIP Authentication ID

When using SIP credentials as the authentification type, you can define SIP Authentification ID

SIP Authentication Password

When using SIP credentials as the authentification type, you can define SIP Authentification Password

BroadSoft Directory & Call Logs Update Interval (s)

Configures the interval (in seconds) to retrieve BroadSoft call log and directory data.

BroadSoft Directory Hits

The maximum hits returned from the BroadSoft XSI server directory. The valid range is from 1 to 1000. If set to blank, server’s default value will be used

Associated BroadSoft Account 

Configures the associated BroadSoft account when dialing BroadSoft contacts.

BroadSoft Directory Order 

Defines the BroadSoft directory order displayed on LCD. Select one item and click the Up/Down arrow on the right to adjust the order.

Network Directories

Enables /Disables BroadSoft Network Directories and assigns name to them:

All directories are enabled by default

Value-added Services – Monitor

Device Name

A custom name used to identify the monitored RTSP device (for example, an IP camera or door station).

Connection Type

Defines how the RTSP stream is transmitted. Three options are supported:

Default connetion type is RTSP (TCP).

RTSP URL

The complete RTSP stream address of the monitored device (e.g., rtsp://192.168.1.100:554/stream1).

RTSP Username

The username required to access the RTSP stream if authentication is enabled.

RTSP Password

The password associated with the RTSP username for authentication.