WP816/WP826 - Administration Guide

Protocol/Standards

SIP RFC3261, DNS (A record, SRV, NATPR), DHCP, SSH, NTP, STUN, LDAP, TR069, SNMP, STRP, RTP/RTCP, RTCP-XR, TFTP, SIMPLE, HTTP/HTTPS, TCP, UDP, TLS, ARP, ICMP, IPv4, IPv6, 802.1x

Voice Codecs and Capabilities

G.729A/B, G.711µ/a-law, G.726, G.722(wide-band), G.723,iLBC, Opus, in- band and out-of-band DTMF (in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC, AJB,

Headset Noise Shield

Wi-Fi

Yes, integrated dual-band Wi-Fi 6 802.11 a/b/g/n/ac/ax (2.4GHz & 5GHz). 802.11k/r/v Supported

Wi-Fi Encryption

Support WEP, WPA, WPA2, WPA3 (personal)

Graphic

1.77 inch (128x160) color LCD, 1 * Dual color MWI

Bluetooth

Yes, integrated

Peripherals

2 soft keys, navigation keys, confirm key, dial key, hang up key, speaker key, programmable key, quick access and mute key, backlight DTMF keyboard, volume keys, Push-to-Talk key, accelerometer, proximity sensor

Push-to-Talk

Customizable function button for alarm (pending) and paging

Auxiliary Ports

3.5mm headphone interface (CTIA cable sequence), Type-C charging interface (supports fast charging)

Telephony

Hold, transfer, forward, 3-way audio conference, call park, downloadable phonebook (XML, LDAP, up to 1000 items), call waiting, call log (up to 200 records), auto answer, click to dial, flexible dial plan, personalized music ringtones, server redundancy and fail-over, push-to-talk

Wall Installation

Base supports wall installation

Security

Ordinary user and administrator level passwords based on MD5 and MD5 sess authentication, SIP authentication algorithms based on SHA-256, SHA-256 sess, SHA-512-256, SHA-512-256 sess, md5 sess. AES security profile, SRTP, TLS call encryption, 802.1x media access control

HD Audio

The earpiece is broadband audio, and the speaker is narrowband audio, supporting

HAC and dual Mic.

QoS

Supports Layer 3 QoS (Tos, DiffServ, MPLS, DSCP)

Multi-language

Simplified Chinese, Traditional Chinese, English, Arabic, Catalan, Czech, German, Greek, Spanish, French, Hebrew, Croatian, Magyar, Italian, Japanese, Korean, Latvian, Dutch, Polish, Portuguese, Russian, Swedish, Slovenian, Slovak, Turkish, Ukrainian

Upgrade/Provisioning

Firmware upgrade via FTP/TFTP/FTPS/HTTP/HTTPS, mass provisioning using GDMS/TR069 or AES encrypted XML configuration file

Power & Green Energy Efficiency

Universal power adapter

Input: 100-240VAC; Output:+5VDC, 1A (5W)

1500mAh lithium-ion battery, standby time 120h, talk time 6h (laboratory data)

Physical

Handset dimensions: 135.00 x 49.00 x 15.5mm Charger cradle dimensions: 85.00 x 85.00 x 25.8mm Handset weight (not including battery): 82g Handset package weight (not including QIG): 389g

Temperature and Humidity

Operating Temperature: 0ºC to 45ºC; Operating Humidity: 10-90% (non-condensing) Storage Temperature: -20ºC to 60ºC; Storage Humidity: 10-90% (non-condensing)

Package Contents

WP816 phone, Type-C power adapter, charger base, belt clip, 1 lithium-ion battery, M3 screw, wall bracket, quick installation guide

Compliance

FCC, CE, RCM, IC

WP826

Key

Description

3.5 mm headset jack

This port allows you to connect headphones or external speakers to the phone for audio output.

LED indicator

A small light that provides visual notifications for various events like incoming calls, messages, or charging status.

Earphone

Delivers audio output.

Volume Up key

A button used to increase the volume of audio output.

Volume Down Key

A button used to decrease the volume of audio output

PTT Key

PTT (Push-to-Talk) button, to initiate PTT call.

Navigation keys

Buttons used to navigate through menus, apps, and interfaces.

Off-hook / Dial key

Initiates or answers calls when pressed, and also used to dial numbers when making outgoing calls.

1 / Voicemail key

Long pressing this key initiates a call to your voicemail service, allowing you to check for new messages or manage voicemail settings.

Standard keypad

A grid of numeric keys used for dialing phone numbers, entering text, and navigating through menus by inputting numbers or letters associated with options.

* / Symobolic key / Silent Mode

This key is used to toggle the phone's silent mode on and off, muting all incoming call and message notifications.

Handsfree / Speaker key

Pressing this key enables the phone's speakerphone function, allowing for hands-free communication during calls.

Microphone

Picks up audio earpiece and hands-free calls

Type-C headphone &charging port

This port serves a dual purpose, allowing you to connect Type-C headphones for audio output and charging the phone using a compatible Type-C cable.

Dedicated function key

Mute/Shortcut Key

A button that quickly silences incoming calls or notifications when pressed, and may also be customizable to serve as a shortcut for accessing frequently used features.

# / Input method switching / Lock key

Does one of the following:

On-hook /Power key

This key serves a dual purpose, ending calls or switching off the phone when pressed for a longer duration, and turning on the phone or waking it from sleep mode when pressed briefly.

MENU/OK key

This key serves a dual purpose, opening the menu interface or confirming selections, such as when navigating through apps or options.

Softkeys

Correspond to functions displayed on the LCD. These functions change depending on the current context.

Color LCD Screen

1.77 inch (128x160) IPS color LCD screen

Proximity sensor

The proximity sensor detects when the phone is close to the caller's face, turning off the display to prevent accidental touches and save battery life.

Noise Canceling Microphone

A microphone equipped with technology to reduce background noise, resulting in clearer audio during calls by minimizing unwanted sounds from the surrounding environment.

Key

Description

LED indicator

A small light that provides visual notifications for various events like incoming calls, messages, or charging status.

Earphone

Delivers audio output.

Volume Up key

A button used to increase the volume of audio output.

Volume Down Key

A button used to decrease the volume of audio output

PTT Key

PTT (Push-to-Talk) button, to initiate PTT call.

Navigation keys

Buttons used to navigate through menus, apps, and interfaces.

Off-hook / Dial key

Initiates or answers calls when pressed, and also used to dial numbers when making outgoing calls.

1 / Voicemail key

Long pressing this key initiates a call to your voicemail service, allowing you to check for new messages or manage voicemail settings.

Standard keypad

A grid of numeric keys used for dialing phone numbers, entering text, and navigating through menus by inputting numbers or letters associated with options.

* / Symobolic key / Silent Mode

This key is used to toggle the phone's silent mode on and off, muting all incoming call and message notifications.

Handsfree / Speaker key

Pressing this key enables the phone's speakerphone function, allowing for hands-free communication during calls.

Microphone

Picks up audio earpiece and hands-free calls

Type-C headphone &charging port

This port serves a dual purpose, allowing you to connect Type-C headphones for audio output and charging the phone using a compatible Type-C cable.

Softkeys

These buttons are assigned a specific function or shortcut, such as launching a specific app, activating a feature, or performing a predefined action.

Mute/Shortcut Key

A button that quickly silences incoming calls or notifications when pressed, and may also be customizable to serve as a shortcut for accessing frequently used features.

# / Input method switching / Lock key

Does one of the following:

On-hook /Power key

This key serves a dual purpose, ending calls or switching off the phone when pressed for a longer duration, and turning on the phone or waking it from sleep mode when pressed briefly.

MENU/OK key

This key serves a dual purpose, opening the menu interface or confirming selections, such as when navigating through apps or options.

Softkeys

Correspond to functions displayed on the LCD. These functions change depending on the current context.

Color LCD Screen

2.4 inch (240x320) IPS color LCD screen

Proximity sensor

The proximity sensor can detect when the phone is close to the caller's face, turning off the display to prevent accidental touches and save battery life.

Noise Canceling Microphone

A microphone equipped with technology to reduce background noise, resulting in clearer audio during calls by minimizing unwanted sounds from the surrounding environment.

Battery status

Charging

Wi-Fi connected on 5G band

Wi-Fi connected on 2.4G band

Wi-Fi signal status for threshold alarm

Outgoing Call notification

Missed Call notification

Rejected Call notification

Incoming Call notification

Auto-Answer feature enabled icon

Mute enabled icon

Silent mode enabled icon

Call on HD quality icon

Warning icon

DND enabled icon

Bluetooth enabled icon

Bluetooth connected icon

Call connected through Bluetooth icon

Contacts

Instant Messages

Call History

Voice Mail

Settings

Status

Status 🡪 Account Status

Account

Account index, shows a list of :

SIP User ID

Displays the configured SIP User ID for the account.

Displays the configured SIP Server address, URL or IP address, and port of the SIP server.

Operation

Displays the different types of operations that can be performed on each SIP account, including editing the account, accessing the voicemail....

Status 🡪 Network Status

WLAN MAC Address

The WLAN MAC Address is a unique identifier assigned to the Wi-Fi interface of the WP phone, facilitating network communication and device recognition within the WLAN (Wireless Local Area Network).

SSID

The SSID (Service Set Identifier) is the name of the Wi-Fi network broadcasted by the access point, allowing devices like the WP phone to identify and connect to the correct network.

BSSID

BSSID is the MAC address of the access point (AP) or router that is providing the Wi-Fi network.

Band Width

The "band" refers to the frequency range at which the Wi-Fi network operates, typically either 2.4 GHz or 5 GHz, determining the speed and coverage of the WP phone's wireless connection.

Channel

The "channel" refers to a specific frequency within the band that the Wi-Fi network uses for communication. Selecting the appropriate channel helps optimize performance and reduce interference for the WP phone's wireless connection.

RSSI

Received Signal Strength Indicator measures the signal strength of the Wi-Fi connection on the WP phone, indicating the quality of the wireless signal received from the access point.

Security Mode

Security Mode indicates the encryption method used to protect data transmitted over the Wi-Fi network on the WP phone, ensuring secure communication between devices and preventing unauthorized access.

Highest Technical Standards

Displays the Highest Wi-Fi version supported by the WP phone

Current Technical Standards

Displays the Wi-Fi version currently used by the WP phone

Country Code

Displays the country where the WP phone is deployed

IPv4 Address Type

Displays how the WP phone have had its IPv4 address assigned

IPv4 Address

Displays the IPv4 address with its corresponding subnet mask

Gateway

Displays the IP address of the gateway

IPv4 NAT Type

Displays the type of NAT used in the IPv4 network

IPv6 Address Type

Displays how the WP phone have had its IPv6 address assigned

Global Unicast Address

Displays the IPv6 Global Unicast Address

Link-Local Address

Displays the Link-Local Address

IPv6 Static Gateway

Displays the IPv6 gateway address

IPv6 DUID

Displays the DHCP Unique Identifier

IPv6 NAT Type

Displays the IPv6 NAT Type used

DNS Server

Displays the Primary and secondary DNS servers used

DNS Mode

Displays the DNS mode used by each account

NAT Traversal

Displays the NAT traversal mode used

Status 🡪 System Info

Product Model

Product model of the phone.

Part Number

Product part number.

Serial Number

Displays the serial number of the unit

Certificate Type

Displays the certificate type used

Software Version

IP Geographic Information

System Up Time

System up time since the last reboot.

System Time

Current system time on the phone system.

System Time-Zone

Displays the time zone that is configured by user

System Information

Download system information

Service Status

GUI, Phone and CPE service status.

User Space

Shows the percentage of the user space used and the status of the Database

Core Dump

Shows the status of the core dump and the core dump files generated if any. It also gives the ability to generate GUI/Phone core dump files manually.

Special Feature

OpenVPN® Support: displaying if the phone supports OpenVPN®.

Call Status

This tab displays different call informations to monitor and view the call status, information mentioned include the call quality, basic Call info , such as the extensions used , the IP address of the endpoints and the codecs used

Call Feature Status

This tab shows whether the call features for each account, such as DND, Auto Answer, Forward All, Busy Forward, and No Answer Forward, are enabled or disabled.

Account x 🡪 General Settings

Account Register

Account Active

Indicates whether the account is active.

The default setting is “No”.

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

Outbound Proxy

Defines IP address or Domain name of the Primary Outbound Proxy, Media Gateway, or Session Border Controller.

Secondary Outbound Proxy

Defines secondary outbound proxy that will be used when the primary proxy cannot be connected.

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.

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.

Account Display

When set to “Username”, the LCD will display the Username if it is not empty and when set to “User ID”, the LCD will display the User ID if it is not empty.

UCM User Password

Input UCM user login password to connect UCM user settings.

Network Settings

DNS Mode

This parameter controls how the Search Appliance looks up IP addresses for hostnames. If “Use Configured IP” is selected, please fill in Primary IP, Backup IP 1 and Backup IP 2.

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.

Primary IP

Configures the primary IP address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode.

Backup IP 1

Configures the backup IP 1 address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode.

Backup IP 2

Configures the backup IP 2 address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode.

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.

Support rport (RFC3581)

Configures to use symmetric response routing. If it is used, the "rport" field
will be added to the Via header field in the SIP Request, and the information will be extracted from the SIP 200OK Response for SIP Register to rewrite the SIP Contact information and apply it in subsequent SIP Requests.

Proxy-Require

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

Use SBC

Configures whether a SBC server is used. Note: If enabled, make sure an outbound proxy is set up.

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 on Reboot

REGISTER Expiration

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

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

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

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 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”.

OPTIONS Keep Alive Interval

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.

OPTIONS Keep Alive Max 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”.

SUBSCRIBE for Registration

When set to “Yes”, a SUBSCRIBE for Registration will be sent out 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 X-Grandstream-PBX Header

Configures to use X-Grandstream-PBX header in SIP request.

Default setting is “Yes”.

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

Configure whether the "P-Asserted-Identity Header" is present in the SIP REGISTER message.

Use P-Early-Media Header

Configure if the "P-Early-Media Header" support is enabled.

Use Zoom E911 X-switch-info SIP Header

Configure whether the "Zoom E911 X-switch-info SIP Header" is present in the SIP REGISTER message.

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”.

Enable TCP Keep-alive

Configures whether to enable TCP Keep-alive for the TCP connection between the terminal and the SIP server.

SIP Listening Mode

Configures whether or not to listen to multiple SIP protocols.

The default setting is “Transport Only”.

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.

Outbound Proxy Mode

Configures whether to put the Outbound Proxy in the Route header, or if SIP messages should always be sent to Outbound Proxy.

Default is “in route”.

Enable 100rel

When enabled, the 100rel tag is appended to the value of the Supported header of the initial signaling messages.

The default setting is “No”.

Use Route Set in Notify (Follow RFC 6665)

Configures whether to use route set in NOTIFY (follow RFC 6665).

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

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

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

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”.

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”.

Force INVITE

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

The default setting is “No”.

Account x 🡪 Codec Settings

Audio

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”.

iLBC Payload Type

Specifies iLBC payload type. Valid range is 96 to 127. Cannot be the same as Opus or DTMF payload type. Valid range is 96 to 127. The default setting is “97”.

G.726-32 Packing Mode

Selects “ITU” or “IETF” for G.726-32 packing mode.

The default setting is “ITU”.

G.726-32 Dynamic Payload Type

Specifies G.726-32 payload type. Valid range is 96 to 127. Default is 127.

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.

Send 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.

Inbound DTMF Volume

Sets the volume size when using in-audio DTMF transmission mode. The higher the parameter value, the higher the volume value. the valid range is 0-32.
The default value is 18.

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”.

Jitter Buffer Type

Selects either Fixed or Adaptive for jitter buffer type, based on network conditions. The default setting is “Adaptive”.

Jitter Buffer Length

Selects jitter buffer length from 100ms to 800ms, based on network conditions. The default setting is “300ms”.

Minimum Jitter Buffer Length

Sets the minimum jitter buffer length based on network conditions.The default value is 0ms. Works over Wi-Fi only.

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.

G.723 Rate

Selects encoding rate for G723 codec.

RTP Settings

SRTP Mode

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

The default setting is “No”.

SRTP Key Exchange

Select SRTP key exchange method, SDES(Secure Real-time Transport Protocol) or DTLS(Datagram Transport Layer Security)

SRTP Key Length

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

Default setting is AES 128&256 bit

Crypto 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 Mode

Configure RTCP port negotiation rules.

RTCP Keep-Alive Method

Configures the RTCP channel keep-alive packet type.

RTP Keep-Alive Method

Configures the RTP channel keep-alive packet type.

VQ RTCP-XR Collector Address Selection

When select "Manual", the collector address will be manually configured and transmitted using the SIP Transport protocol. When select "Auto", if the collector address is manually filled in, use this address. If not filled in, use the SIP registered address.

VQ RTCP-XR Collector Name

Configure the host name of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Address

Configure the IP address of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Port

Configure the port of the central report collector that accepts voice quality reports contained in SIP PUBLISH messages.

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 "Disable", 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.
Disabled by Default

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

General

Key as Send

Allows users to configure either the “*” or “#” keys as the “Send” key. Please make sure the dial plan is properly configured to allow dialing * and # out. The default setting is “Pound (#)”.

No Key Entry Timeout

Configures the timeout (in seconds) for no key entry. If no key is pressed after the timeout, the collected digits will be sent out. The default value is 4 seconds. The valid range is from 1 to 15.

Send Anonymous

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

Anonymous Call Rejection

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

The default setting is “No”.

Enable Call Waiting

Configures the call waiting function for this account. If set to “Default”, it will be configured according to global call waiting function. Default value is “Default”.

RFC2543 Hold

Allows users to toggle between RFC2543 hold and RFC3261 hold. RFC2543 hold (0.0.0.0) allows user to disable the hold music sent to the other side. RFC3261 (a line) will play the hold music to the other side. The default setting is “No”.

Ring Timeout

Configures the timeout (in seconds) for the phone to ring when an incoming call is not answered. Valid range is 30 to 3600. The default setting is 60.

Call Log

Configures Call Log setting on the phone.

The default setting is “Log All Calls”.

Auto Answer

Auto Answer

If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep. Default setting is “No”.

Auto Answer Numbers

Allows the user to configure specific numbers to auto answer. If not set, all numbers will be auto answered If auto answer is enabled. Up to 10 numbers can be configured.

Intercom

Play warning tone for Auto Answer Intercom

If enabled, phone will play warning tone when auto answering Intercom.

Custom Alert-Info for Auto Answer

Used exclusively to match the contents of the Alert-Info header for auto answer. The default auto answer headers will not be matched if this is defined.

Allow Auto Answer by Call-Info/Alert-Info

If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep, based on the SIP Call-Info/Alert-Info header sent from the server/proxy.

Default is “Yes”.

Allow Barging by Call-Info/Alert-Info

When enabled, the phone will automatically put the current call on hold and answer the incoming call based on the SIP Call-Info/Alert-Info header sent from the server/proxy. However, if the current call was answered based on the SIP Call-Info/Alert-Info header, then all other incoming calls with SIP Call-Info/Alert-Info headers will be rejected automatically. Default setting is “No”.

Mute on Intercom Answer

If enabled, the phone will mute the mirophone after answer an intercom call via Call-Info/Alert-Info.

Record

Record Key Default Fuction

Configures whether to turn the recording function on or off when the "Record" key is pressed for the first time in a call using this account. For example, the SIP server can be configured with automatic call recording. In this case, Record key default function should be configured as "Record off".

Call Recording On

Configures the DTMF sequence sent when pressing the Record key during a call on this account. Toggles between this value and the off code if possible; otherwise always sends this code.

Call Recording Off

Configures the DTMF sequence sent when pressing the Recording key during a call on this account when turning recording off.

Transfer 

Transfer on Conference Hangup

Configures whether the call is transferred to the other party if the conference initiator hangs up.

The default setting is “No”.

Enable Recovery on Blind Transfer

Enable recovery to the call to the transferee on failing blind transfer to the target. The default setting is “Yes”.

Notes:

Blind Transfer Wait Timeout

Configures the timeout (in seconds) when waiting for sipfrag response in blind transfer. Valid range is 30 to 300. Default setting is “30”.

Refer-To Use Target Contact

If set to “Yes”, the “Refer-To” header uses the transferred target’s Contact header information for attended transfer.

Call Forward

Enable Forward All

If set to "Yes", all calls will be forwarded to the number specified below. Disabled by Default

All To

Specifies the number to be forwarded to when enabled Forward all.

Enable Busy Forward

If set to "Yes", the call will be forwarded to the number specified below on busy. Disabled by Default

Busy To

Specifies the number to be forwarded to for Call Forward On Busy.

Enable No Answer Forward

If set to "Yes", call will be forwarded to the number specified below on no answer. Disabled by Default

No Answer To

Specifies the number to be forwarded to for Call Forward On No Answer.

No Answer Timeout (s)

Defines the timeout (in seconds) before the call is forwarded on no answer.

Enable Override Forward

If enabled, the local call forward is disabled when an incoming call comes in from the configured override forward number. Disabled by Default

Override Forward Numbers

Configures the number to override the local forward function. The Max number is 10.

Dial plan

Dial Plan Prefix

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

Bypass Dial Plan

Bypass the dial plan when dialing from one of the available items:

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.
 

Call Display

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

Ringback Tone at No Early Media

Play ringback tone when there is no receiving early media RTP packets. Disabled by Default

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”.

Trusted Domain Name List

Configure 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. This field allows numbers/letters/-/./*. It supports wildcard domain names, such as “*.mycompany.com“, and trusts any domains ending with “.mycompany.com“.

Validate Certificate Chain

Validate certification chain when TCP/TLS is configured.

The default setting is “No”.

Validate Incoming SIP Messages

Specifies if the phone will check the incoming SIP messages Caller ID and CSeq headers. If the message does not include the headers, it will be rejected. The default setting is “No”.

Omit charset=UTF-8 in MESSAGE

Omit charset=UTF-8 in MESSAGE content-type

Allow Unsolicited REFER

Configures whether to dial the number carried by Refer-to header after receiving out-of-dialog SIP REFER request actively.

If set to “Disabled“, the phone will send error warning and stop dialing.

If set to “Enabled/Force Auth“, the phone will dial the number after sending authentication. If the authentication fails, it will stop dialing.

If set to “Enabled“, the phone will dial all numbers carried by SIP REFER.

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 does not match the SIP server address of the account, the call will be rejected. The default setting is “No”.

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 does not match the phone’s SIP User ID, the call will be rejected. The default setting is “No”.

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”.

MOH

On Hold Reminder Tone

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

Music On Hold URI

Music On Hold URI to call when a call is on hold if server supports it.

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, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro, Huawei IMS, Zoom, PhonePower,UCM Call center, Zoom, and Telestra depending on the server type. The default setting is “Standard”.

Feature Key Synchronization

This feature is used for Broadsoft call feature synchronization. When it is enabled, DND, Call Forward features and Call Center Agent status can be synchronized between Broadsoft server and phone. Default is “Disabled”.

Conference URI

Configures the conference URI when using Broadsoft N-way calling feature.

PUBLISH for Presence

Enables presence feature on the phone.

The default setting is “No”.

Account x 🡪 Dial Plan

Name

Enter the name for the configured rules.

Rule

Enter the rule settings (number pattern, prefix to add …etc.).

Type

Choose the type of the rule:

Account x 🡪 Hidden Number Plan

Hidden Number Feature

If set to “Both Direction calls”, incoming and outgoing number display will be handled according to hidden number plan. The options are : Incoming Calls Only, Outgoing Calls Only, Both Direction Calls

Hidden Number Plan List

Currently, incoming and outgoing number display will be handled according to hidden number plan. Users are able to configure the hidden number rules, matching their syntax rules from top to bottom as follows:

Hidden Number Plan Test

Enter the number to test the hidden rules of the current page. After confirming, please save and apply it.​

Feature Codes

Enable Local Call Features

When enabled, Do Not Disturb, Call Forwarding and other call features can be used via the local feature codes on the phone. Otherwise, the provisioned feature codes from the server will be used. User configured feature codes will be used only if server provisioned feature codes are not provided.

Note: If the device is registered with Broadsoft account, it does not matter if local call features are enabled or disabled, once the Broadsoft account is set, special feature to Broadsoft and Feature Key Synchronization is enabled, the call feature will be handled by Broadsoft server, not by the phone.

DND

DND Call Feature On

Configures DND feature code to turn on DND.

DND Call Feature Off

Configures DND feature code to turn off DND.

Call Forward Always

On

Configures Call Forward Always feature code to activate unconditional call forwarding.

Off

Configures Call Forward Always feature code to deactivate unconditional call forwarding.

Target

The extension the call will be forwarded to.

Call Forward Busy

On

Configures Call Forward Busy feature code to activate busy call forwarding.

Off

Configures Call Forward Busy feature code to deactivate busy call forwarding.

Target

Configures the extension for the call to be forwarded to.

Call Forward No Answer

On

Configures Call Forward No Answer feature code to activate no answer call forwarding.

Off

Configures Call Forward No Answer feature code to deactivate no answer call forwarding.

Target

The extension the call will be forwarded to.

Call Forward No Answer Timeout (s)

Defines the timeout (in seconds) before the call is forwarded on no answer. valid range is 1 to 120.

Accounts 🡪 Account Swap

Swap Account Settings

Allows users to swap the two accounts that they have configured. This will Increase the flexibility of account management.

Note: Make sure to press “Start” to complete the process.

Network Settings 🡪 Wi-Fi Settings

General Settings

Wi-Fi Function

Enables / Disables the Wi-Fi on the phone. Three options are available:

Wi-Fi Band

Set the type of Wi-Fi Band whether its 2G or 5G or 5G&2G.

Country Code

Configures Wi-Fi country code.

ESSID

This parameter sets the ESSID for the Wireless network. Press “Scan” to scan for the available wireless network. Click on “Connect” and enter the authentication credentials of the Wi-Fi network to connect to. Users can connect to hidden networks by pressing on “Add Network” and configure:

Advanced Settings

Host name (Option 12)

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

Vendor Class ID (Option 60)

Used by clients and servers to exchange vendor class ID.

Wi-Fi Signal Warning

When the WiFi signal strength is lower than the threshold set by this level, the device will warning.

Roaming 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.

Poor Signal Scan 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.

VoWLAN Target Delay

Selects Low, Medium, or High based on network conditions.

Network Settings 🡪 Bluetooth Settings

Bluetooth

Enable/Disables Bluetooth

Device Name

Set the name of the local Bluetooth device

Bluetooth Device

Display a list of Bluetooth devices scanned in the surrounding environment.

Network Settings 🡪 OpenVPN® Settings

OpenVPN® Enable

Enables/Disables OpenVPN® feature. Default is “No”.

Import OpenVPN® Configuration

Imports the configuration file from the current computer. After importing, the local configuration will be overwritten and OpenVPN® function is automatically enabled.

OpenVPN® Server Address

Specify the IP address or FQDN for the OpenVPN® Server.

OpenVPN® Port

Specify the listening port of the OpenVPN® server. The valid range is 1 – 65535. The default value is “1194”.

OpenVPN® Transport

Specify the Transport Type of OpenVPN® whether UDP, TCP, UDP IPV4 Only, TCP IPV4 Only, UDP IPV6 Only, TCP IPV6 Only

The default value is “UDP”.

OpenVPN® CA

Click on “Upload” to upload the Certification Authority of OpenVPN®. For a new upload, users could click on “Delete” to erase the last certificate, and then upload a new one.

OpenVPN® Certificate

Click on “Upload” to upload OpenVPN® certificate. For a new upload, users could click on “Delete” to erase the last certificate, and then upload a new one.

OpenVPN® Client Key

Click on “Upload” to upload OpenVPN® Key.
For a new upload, users could click on “Delete” to erase the last certificate, and then upload a new one.

OpenVPN® Client Key Password

Allows user to set password for client.key file

OpenVPN® TLS Key

Uploads the OpenVPN® TLS .key file

OpenVPN® TLS Key Type

Selects the encryption type of the OpenVPN® TLS key. it can be set to : TLS-Auth, TLS-Crypt, TLS-Crypt V2

OpenVPN® Cipher Method

Specifies the Cipher method used by the OpenVPN® server. The available options are:

The default setting is “Blowfish”.

OpenVPN® Username

Configures the optional username for authentication if the OpenVPN server supports it.

OpenVPN® Password

Configures the optional password for authentication if the OpenVPN server supports it.

OpenVPN® Comp-lzo

Configures enable/disable the LZO compression. When the LZO Compression is enabled on the OpenVPN server, you must turn on it at the same time. Otherwise, the network will be abnormal. Default value is YES.

Additional Options

Additional options to be appended to the OpenVPN® config file, separated by semicolons. For example, comp-lzo no;auth SHA256

Note: Please use this option with caution. Make sure that the options are recognizable by OpenVPN® and do not unnecessarily override the other configurations above.

Network Settings 🡪 Advanced Settings

Advanced Settings

Advanced Network Settings

DNS Refresh Timer (m)

Configures the refresh time (in minutes) for DNS query. If set to "0", the phone will use the DNS query TTL from DNS server response.
the Default value is "0"

DNS Failure Cache Duration (m)

Configures the duration (in minutes) of the previous DNS cache when the DNS query fails. If set to "0", the feature will be disabled. Note: Only valid for SIP registration.
The Default value is "0"

Layer 3 QoS for SIP

Configures the Layer 3 QoS parameter for SIP. This value is used for IP Precedence, Diff-Serv or MPLS.
Default value is 26

Layer 3 QoS for RTP

Configures the Layer 3 QoS parameter for RTP. This value is used for IP Precedence, Diff-Serv or MPLS.
Default value is 46

Maximum Transmission Unit (MTU)

Configures the MTU in bytes.
Default value is 1500

Proxy

HTTP Proxy

Specifies the HTTP proxy URL for the phone to send packets to. The proxy server will act as an intermediary to route the packets to the destination.

HTTPS Proxy

Specifies the HTTPS proxy URL for the phone to send packets to. The proxy server will act as an intermediary to route the packets to the destination.

Bypass Proxy for

Configures the destination IP address where no proxy server is needed. The phone will not use a proxy server when sending packets to the specified destination IP address.

Remote Control

Action URI Support

Configures whether to enable phone to handle Action URI request.

Default is “Enabled”.

Remote Control Popup Window Support

Configures whether to enable phone to pop up Allow Remote Control window.
Enabled by Default

Action URI Allowed IP List

List of allowed IP addresses from which the phone receives the Action URI

CSTA Control

Configures whether to enable the CSTA ( Computer Supported Telecommunications Applications) Control feature.

Static DNS Cache

NAPTR

NAPTR (Naming Authority Pointer) records are used to specify rules for rewriting one type of domain name to another, typically used for handling Uniform Resource Identifiers (URIs) within the domain, when you configure NAPTR in the static DNS cache, you are specifying custom rules for how specific URIs or domain names should be resolved, the options to configure are :

SRV

SRV records are DNS records used to identify servers that provide specific services, such as email, SIP (Session Initiation Protocol) servers, or other services, Configuring SRV in the static DNS cache allows you to specify which servers should be used for particular services, helping ensure that your IP phone connects to the correct servers for specific functions, the available options to configure are:

A

A records are used to map a domain name to an IPv4 address. They are the most common type of DNS record and are used to resolve domain names to IP addresses, Configuring A records in the static DNS cache allows you to manually specify the IP addresses associated with specific domain names, ensuring that your IP phone always connects to the intended destination, the options to configure are: