WP820 - Administration Guide

WP820

Battery status

Charging

Wi-Fi not enabled/configured

Wi-Fi signal status

Outgoing Call notification

Missed Call notification

Incoming Call notification

Voicemail notification (Urgent)

Voicemail notification (Normal)

Mute enabled icon

DND enabled icon

Bluetooth enabled icon

Bluetooth connected icon

SRTP & TLS enabled icon

Contacts

SMS

Call History

Voice Mail

Diagnosis

Settings

Applications

Status

Broadsoft XSI

Red Fast Flashing

Upgrading

Green Slow Flashing

Incoming call

Green Blinking

Charging but not full

Green Solid

Charging (full)

Red Slow Flashing

Red Solid

Network disconnected

LED OFF

Normal

Contacts

Display the list of the registered contacts and also the groups contacts with the ability of searching, adding, or editing the entries and also deleting the selected contacts. The maximum number of contacts is 1000.

Call History

Display the call history: Missed Calls, Accepted Calls, Outgoing Calls or All Calls. You can add contacts to Shared Contacts directly from call logs.

SMS

SMS stands for Short Message Service and referred to as a "text message". With a SMS, you can send a message by pressing “New” of up to 160 characters to another device or check the received ones.

Voice Mail

Diagnostics

Settings

Applications

Broadsoft XSI

Broadsoft account.

Status

Displays account status, system info, Network status and network statistics

MAC Address

Shows Device ID in hexadecimal format. This is needed by network administrators for troubleshooting. The MAC address will be used for provisioning and can be found on the label on original box and on the label located on the bottom panel of the device.

NAT Type

Indicates type of NAT for each Profile. (Based on STUN protocol.)

IPv4

Address Type

Indicates the configured address type: DHCP or Static IP

IPv4 Address

Displays assigned IP address. Example: 192.168.5.110

Subnet Mask

Displays assigned subnet mask. Example: 255.255.255.0

Default Gateway

Displays assigned default gateway. Example: 192.168.5.1

DNS Server 1

Shows assigned DNS server address 1. Example: 8.8.8.8

DNS Server 2

Shows assigned DNS server address 2. Example: 8.8.4.4

IPv6

IPv6Address Type

Indicates the configured address type: DHCP or Static IP

IPv6 Address

Displays assigned IP address.

IPv6 DNS Server 1

Shows assigned DNS server address 1.

IPv6 DNS Server 2

Shows assigned DNS server address 2.

IPv6 DUID

Shows the IPv6 DUID

Product Model

Product model of the phone.

Hardware Revision

Hardware version number.

Part Number

Product part number.

System Version

Firmware version.

Recovery Version

Recovery image version.

Boot Version

Booting code version.

Kernel Version

Kernel version.

CPE Version

CPE version.

System Up Time

System up time since last reboot.

LCD Serial Number

LCD Serial Number.

DDR Serial Number

DDR Serial Number.

Factory Serial Number

Factory Serial Number.

Preferred Vocoder

Preferred Vocoder

Lists the available and enabled Audio codecs for this account. Users can enable the specific audio codecs by moving them to the selected box and set them with a priority order from top to bottom. This configuration will be included with the same preference order in the SIP SDP message. The Supported vocoders are: PCMU, PCMA, G722, G729A/B, iLBC and Opus.

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.

Use First Matching Vocoder in 200OK SDP

Includes only the first matching vocoder in its 200OK response, otherwise it will include all matching vocoders in same order received in INVITE.

The default setting is No.

iLBC Frame Size

Specifies iLBC packet frame size (20ms or 30ms). Default is 30ms.

Opus Payload Type

Determines payload type for Opus codec. The valid range is between 96 and 126.

Default is 123.

DTMF

Specifies the mechanism to transmit DTMF (Dual Tone Multi-Frequency) signals. There are 3 supported modes:

The default setting is "RFC2833".

DTMF Payload Type

Configures the RTP payload type using RFC2833 that indicates the transmitted packet contains DTMF digits. Valid range is from 96 to 127.

Default value is 101.

Enable Audio RED with FEC

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

The default setting is "No".

Audio FEC Payload Type

Configures audio FEC payload type. The valid range is from 96 to 127.

The default value is 121.

Audio RED Payload Type

Configures audio RED payload type. The valid range is from 96 to 127.

The default value is 124

Silence Suppression

Enables the silence suppression/VAD feature. If it is 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. If set to “No”, this feature is disabled.

The default setting is “No”.

Minimum Jitter Buffer Length

Set minimum jitter buffer length based on network conditions.

The default value is 0ms.

Maximum Jitter Buffer Length

Set maximum jitter buffer length based on network conditions.

The default value is 1000ms.

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 120Kbit/s). 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 the setting is set to 2 and the first code is G. 729 or G. 711, the "ptime" value in the SDP datagram of the INVITE request is 20 milliseconds. If the "Voice Frame/TX" setting exceeds the maximum allowable value, the phone will use and save the maximum allowable value for the selection of the first encoding. It is recommended to use the default setting provided, and incorrect setting may affect the voice quality.

Preferred Video Codec

Preferred Video Codec

This parameter is to configure the preferred Video Codec used to receive and display video for SIP calls on WP820. The default setting is H.264.

Enable Video FEC

When enabled, the video sender will temporarily allocate part of the bandwidth to one data channel to send FEC data to system, thus, to improve the video quality the WP820 gets. Enabling this function will take up part of bandwidth and reduce call rate. The default setting is "Yes".

Enable RFC5168 Support

Enables/disables RFC5168 mechanism for video calls to WP820. RFC5168 allows SIP party to request the sender to refresh its video frame in H.264.The default setting is “No”.

Configures FEC payload type. The range is 96-126.

Default setting is 120.

Sets the H.264 image size. It can be selected from the dropdown list.

Note: For some network environment, the default setting “720P” might be too high that causes not receiving video or video quality issue during video call to WP820. In this case, please change “H.264 Image Size” to “VGA” or “CIF” and change “Video Bit Rate” to “384kbps” or lower. The default setting is 720P.

Video Bit Rate

Configures the bit rate for video stream received on the WP820. The default setting is 2048 kbps. The valid range is from 32 – 2048 kbps.

Note: 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, the video quality will decrease due to packet loss. For some network environment, the default setting “720P” might be too high that causes no video or video quality issue during video call to WP820. In this case, please change “H.264 Image Size” to “VGA” or “CIF” and change “Video Bit Rate” to “384kbps” or lower.

Sets the SDP bandwidth attribute. It can be selected from the drop-down list. The default setting is “Media Level”.

Note: Please do not modify this setting without knowing the session format supported by the server. Otherwise, it might cause video decoding failure.

RTP Settings

SRTP Mode

Sets if the phone will enable the SRTP (Secured RTP) mode. It can be selected from dropdown list:

The default setting is “Disable”.

SRTP Key Length

Configures all the AES (Advanced Encryption Standard) key size within SRTP. It can be selected from dropdown list:

If it is set to “AES 128&256 bit”, the phone will provide both AES 128 and 256 cipher suite for SRTP. If set to “AES 128 bit”, it only provides 128-bit cipher suite; if set to “AES 256 bit”, it only provides 256-bit cipher suite.

The default setting is “AES128&256 bit”.

Enable SRTP Key Life Time

Defines the SRTP key life time. When this option is set to be enabled, during the SRTP call, the SRTP key will be valid within 231 SIP packets, and phone will renew the SRTP key after this limitation. The default settings is “Yes”.

RTCP Keep-Alive Method

Configures the RTCP channel's keep-alive packet type. When configured as "Receiver Report," the RTCP channel will transmit "receiver report" as its keep-alive data. Conversely, if configured as "Sender Report," the RTCP channel will transmit "Sender report" as its keep-alive data.
Set to "Receiver Report", By Default.

RTP Keep-Alive Method

Configures the RTP channel keep-alive packet type. If set to "No", no data will be sent. If set to "RTP version 1", the wrong version info "1" will be carried when sending RTP data packets. If set to "RTP Packet with Silent Payload", the silent payload will be carried when sending RTP format packets.
Default value is "RTP version 1"

RTCP Destination

Configures the server address. When there is a call, the RTCP package sent from the phone will also be sent to this address.

Note: The address should contain port number.

Symmetric RTP

Configures if the phone enables the symmetric RTP mechanism.

If it is set to “Yes”, the phone will use the same socket/port for sending and receiving the RTP messages. The default setting is "No".

RTP IP Filter

Receives the RTP packets from the specified IP address and Port by communication protocol. If it is set to “IP Only”, the phone only receives the RTP packets from the specified IP address based on the communication protocol; If it is set to “IP and Port”, the phone will receive the RTP packets from the specified IP address with the specified port based on the communication protocol. The default setting is “Disable”.

RTP Timeout (s) 

Configures the RTP timeout setting; If the phone does not receive any RTP packet within the specified RTP time, the call will be automatically disconnected. Default settings is 30s and the supported range is between 0 and 600.

The corresponding P-value of this settings for Account 1 and Account 2 are:

P29068 and P29168, respectively.

Call Features

Start Video Automatically

Defines whether to enable video automatically when there is a call from contacts or dialing interface. If set to "Yes", the video call will be started automatically. If set to "No", dial the call via SIP audio.

Auto Answer

If set to "Yes", the phone will automatically turn on the speaker to answer incoming calls after a short reminder beep.

If set to "Enable Intercom/Paging", it will answer the call based on the SIP Info header sent from the server/proxy. The default setting is “No”.

Play Warning Tone for Auto Answer Intercom

When enabled, the phone will play warning tone when auto answer Intercom. By default, it is enabled.

Intercom Barging

Configures whether to answer the incoming intercom call when there is already an active call on the phone. When "Intercom Barging" is enabled, if the current active call is an intercom call, the incoming intercom call will be automatically rejected;

If the current active call is not an intercom call, 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 show up indicating the incoming intercom call without interrupting the current active call.

Send Anonymous

Sets the phone to make anonymous outgoing calls; when set to "Yes", the "From" header in outgoing INVITE messages will be set to anonymous, essentially blocking the Caller ID to be displayed. The default setting is “No”.

Intercept Anonymous Calls

Rejects the calls whose SIP INVITE message includes Anonymous information in from header when enabled. The default setting is “No”.

Call Log

Categorizes the call logs saved for this account. The default setting is “Log All”.

Enable Call Features

Configures the local call features. If set to "Yes", call features (including call forwarding, DND and etc.) will be supported locally instead of using the feature code supported on SIP server/proxy.

Mute on Answer Intercom Call

When enabled, phone will set mute feature upon answering incoming intercom calls. The corresponding P-values for Account1 and Account2 are respectively: P26071 and P26171 with possible values 0 for ‘OFF’ and 1 for ‘ON’.

Transfer on 3-way Conference Hang up

Transfers conference from hosted party when hang up, thus other parties can continue the conference without interruption. The default setting is unchecked.

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.

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

Configures the feature code to enable the DND (Do Not Disturb) feature for this account. If it is configured, the phone will dial the feature code automatically when the DND feature is enabled.

DND Call Feature Off

Configures the feature code to disable the DND (Do Not Disturb) feature for this account. If it is configured, the phone will dial the feature code automatically when the DND feature is disabled.

No Key Entry Timeout (s)

Determines the expiration timer (in seconds) for no key entry. The dialed digits will be sent out if no other digits entered within the set period. The default value is 4 seconds.

The valid range is from 1 to 15.

Ring Timeout (s)

Defines the expiration timer (in seconds) for the rings with no answer. The default setting is 60.

The valid range is from 10 to 300.

Refer-To Use Target Contact

Sets the phone to use the target’s Contact header tag to the Refer-To header in the SIP REFER message during an attended transfer.

The default setting is “No”.

RFC2543 Hold

Allows users to toggle between RFC2543 hold and RFC3261 hold.

RFC2543 hold allows user to disable the hold music sent to the other side (c=0.0.0.0 will be used in INVITE SDP for hold). RFC3261 will play the hold music to the other side.

Call Forward

Call Forward Type

Sets the Call Forwarding feature for this account.

Dial Plan

Dial Plan Prefix

Adds specified prefix to dialed number.

Disable Dial Plan

Enables/disables the Dial plan mechanism for different cases. If the specific case is checked, the Dial plan mechanism will be disabled.

The default setting is unchecking all the cases.

Dial Plan

Dial Plan Rules:

{123<611=456>} - dial 123611 will be become 123611

{123<456=0>789} – dial 123456789 will become 1230789

{<=1>x+} - dial 3000 will be become 13000

{<=611>555} – dial 555 will become 611555

{<611=>555} – dial 611555 will become 555

{123<555=>} - dial 123555 will become 123

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

•  ^1900x. – prevents dialing any number started with 1900

•  <=1617>[2-9]xxxxxx - allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code will be added automatically

•  1[2-9]xx[2-9]xxxxxx - allows dialing to any US/Canada Number with 11 digits length

•  011[2-9]x. – allows international calls starting with 011

•    [3469]11 – allows dialing special and emergency numbers 311, 411, 611 and 911

•    +x+ - allows dialing any digit with leading + sign; example: +16175669300

Note: In some cases, user wishes to dial strings such as *123 to activate voice mail or other application provided by service provider.

In this case * should be predefined inside dial plan feature.

An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of numbers.

Caller IDs

Caller ID Display

Specifies which header tag will be used from the SIP INVITE message for the Caller ID display.

The default setting is “Auto”.

Ring Tones

Account Ringtone

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

Note: User can also choose silent ring tone.

Ignore Alert-Info Header

If checked, configured default ringtone will be played by ignoring Alert-Info header.

Match Incoming Caller ID

Specifies matching rules with a number, pattern, or Alert-Info text (up to 3 matching rules). When the incoming caller ID or Alert-Info matches the rule, the phone will ring with a 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 

Selects the distinctive ring tone for the matching rule. When the incoming caller ID or Alert Info matches one of the 10 rules, the phone will ring with the associated ringtone

Security Settings

Check Domain Certificates

Defines whether the domain certificates will be checked when TLS/TCP is used for SIP Transport.

The default setting is "No".

Validate Certification Chain

Configures whether to validate certification chain when TLS/TCP is configured for SIP Transport. If this is set to “Yes”, phone will validate server against the new certificate list.

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".

Allow Unsolicited REFER

Configures whether to dial the number carried by Refer-to after received Out-of-dialog SIP REFER request actively.

The default is “Disabled”.

Only Accept SIP Requests from Known Servers

Answers the SIP request from saved servers when set to “Yes”, only the SIP requests from saved servers will be accepted; and the SIP requests from the unregistered server will be rejected.

The default setting is “No”.

Check SIP User ID for Incoming INVITE

Configures the phone to check the SIP User ID in the Request URI of the SIP INVITE message from the remote party. If it does not match the phone's SIP User ID, the call will be rejected. The default setting is “No”.

Allow SIP Reset

Configures if allow SIP Notification message to perform factory reset.

The default setting is "No".

Authenticate Incoming INVITE

Configures the phone to authenticate the SIP INVITE message from the remote party. 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 item to validate incoming INVITE, but you must enable authenticate incoming INVITE first to make it take effect. You can verify the NOTIFY information for the provision, including check- sync, resync and reboot, but only when SIP NOTIFY authentication enabled first to make it take effect.

MOH

Upload Local MOH Audio File

Loads the MOH (Music on Hold) file to the phone. Click on “Browse” button to upload the music file from local PC. The MOH audio file has to be in .wav or .mp3 format.

Note: Please be patient while the audio file is being uploaded. It could take more than 3 minutes to finish the uploading especially the file size is large. The button will show as “Processing” during the uploading. Once done, it will show as “Browse” again. Click on “Save” on the bottom of the web page and “Apply” on the top of the web page to save the change.

Enable Local MOH

Plays local MOH file if the call is being hold by the phone. Users need to upload local MOH audio file. The default setting is “No”.

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, Broadsoft, China Mobile, ZTE IMS, Mobotix, ZTE NGN, Huawei IMS, NEC or WorldStone depending on the server type. Default setting is “Standard”.

Feature Key Synchronization

This feature is used for BroadSoft/Metaswitch call feature synchronization. When set to BroadSoft/Metaswitch, DND and Call Forward features can be synchronized with BroadSoft or Metaswitch server. Default settings is “Disable”

Allow Sync Phonebook via

Allows users to synchronize XML phonebook upon receiving SIP NOTIFY message with header Event: sync-contacts.

SIP Notify 

Note: Received SIP NOTIFY will be first challenged for authentication purpose before contacting configured server to download XML phonebook. The parameters used are the ones configured at [Download Contacts]. The authentication can be done either using admin credentials (if no SIP account is configured) or using SIP account credentials.

The default setting is “Yes”.

Basic Settings

Local RTP Port

Defines local RTP port used to listen and transmit RTP packets.

Valid range is 50040 – 65000. Default is 50040.

Use Random Port

Forces the phone to use random ports for both SIP and RTP messages. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is “No”.

Note: This parameter must be set to “No” for Direct IP Calling to work.

Disable In-call DTMF Display

Disables the phone to omit displaying DTMF digits during a call on the LCD screen. The default setting is “No”.

Keep-alive Interval (s)

Specifies how often the phone will send a Binding Request packet to the SIP server in order to keep the “ping hole” on the NAT router to open. The valid range is from 10 to 160. The default setting is 20 seconds.

STUN Server

Use NAT IP

Configures the IP address for the Contact header and Connection Information in the SIP/SDP message. It should ONLY be used if it is required by your ITSP. The default setting is keep the box blank.

Guest Features

Guest Settings

Guest Login

Enable/disable guest login. Users should configure SIP domain name when using this function. Users should input SIP username and password on LCD.

Note: A reboot is required for this setting to take effect.

Guest Login PIN Code

When set "Guest Login" option to "Yes" and "Expire" on the guest login page through LCD to "Always", users should input the PIN code to login.

SIP Domain

Server Alias

The server alias is used to customize the server name.

Server List

The server list consists of all the domain names or IP addresses of the SIP servers under the selected sever alias. The registration priority of addresses in the list is from top to bottom and supports up to 2 SIP domains per alias.

Call Settings

Call Features

Enable Video Call

Enables the video call feature.

Default Video Zoom Area

In a video call, the default area which is displayed after press Zoom. The default value is 0 (full screen). Range 0-9.

Enable Transfer

Enable the transfer feature. When “unchecked”, the phone will block the TRANSFER key on the LCD screen. The default setting is “checked”.

Hold Call Before Completing Transfer

If you choose "Yes", the transfer call status will be kept as hold status after Attended transfer operation.

Enable Conference

Enable the conference. When “unchecked”, the phone will block the conference function. The default setting is “checked”.

Enable noise Shield

Configures whether to enable noise shield feature. If enabled, all the background noise will be suppressed except the sound from speaker.

Enable In-call Recording

If set to "Yes", the device can record during a call; if set to "No", this feature is hidden during a call.

Enable Call Waiting

Enables the call waiting feature. If it is unchecked, the phone will reject the second incoming call during an active session without user’s knowledge. But this missed call record will be saved to remind users. The default setting is “checked”.

Enable Call Waiting Tone

Sets the phone to play the call waiting tone if there is another incoming call.

If it is “Unchecked”, the phone will only display the indicator on the LCD screen for another incoming call. The default setting is “checked”.

Enable Proximity Sensor

If set to "Yes", the phone will automatically turn the screen off when near the proximity sensor during a call.

Note : If proximity sensor is disabled, the LCD screen will not turn off during call when handset is held near the ear.

Incoming/Outgoing Call

Always Ring Speaker

Determines if the speaker will play the ringtone if the speaker channel is not set as default channel. If set to “Yes”, the phone will force to play the ring speaker in speaker channel even if a headset is connected. The default setting is “checked”.

Auto Mute on Entry

Configures whether to mute the call on entry automatically.

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

Rejected Call Notification

Specifies whether to enable Rejected calls notification. When enabled, a missed call notification will prompt on LCD after rejecting an incoming call.

Enable DND Reminder Ring

Enables the DND reminder ring. If set to "Yes", the ring splash that indicates an incoming call when DND is enabled will be played.

Off hook Auto Dial

Configures a User ID/Extension to dial up automatically after the off-hook auto dial delay time ends when the phone is off hook or hands-free. The phone will use the first account to dial out by default.

Off hook Auto Dial Delay (s)

Defines the delay time of Off hook Timeout. Once configured Off hook Timeout, the phone will dial out automatically after the delay time ends. If not configured Off hook Timeout, there will be no delays and the phone will dial up the User ID/Extension automatically.

Off hook/On hook Timeout (s)

If configured, the phone will exit the dial-up screen when timeout after off hook or on hook. The valid range is 10-60s.

Off-cradle Pickup

Enables off hook of handset when picked up from cradle.

On-cradle Hang up

Enables on hook of handset when repositioned on cradle.

This feature is used to enable or disable audio packet drop warning.

DND Settings

DND Mode

Configures the DND mode for the phone.

Enable DND Reminder Ring

Enables the DND reminder ring. If set to "Yes", the ring splash that indicates an incoming call when DND is enabled will be played.

Advanced Settings

Number Of Accounts The Virtual Account Group Register

Configures the number of concurrent registrations with the main server and standby server(s), each account supports 1 SIP main server and 4 standby server registration(s).

Filter Characters

Sets the characters for filter when dial out numbers. Multiple characters could be set. For example, if set to "[()-]", when dial (0571)-8800-8888, the character "()-" will be filtered and dial 057188008888 directly.

The default setting is [()-]

Escape '#' as %23 in SIP URI

Determines which characters will be included in the SIP INVITE URI if end users input #. If it is set to “Yes”, the phone will replace the # by %23. Otherwise, it will include # in the SIP INVITE message.

The default setting is “checked”.

Notification Tone

Configures the notification tone for new SMS and voicemails.

Contact source priority

Configure the priority of the ID source displayed on the phone when incoming/outgoing calls. Select one ID source and click the Up/Down arrow on the right to adjust the order. Note: If the "Caller ID Display" under the account is configured as "Disabled", the caller number cannot be obtained, the phone will only display "Unavailable".

CPT Settings

Auto Config CPT by Region

Configures whether to choose Call Progress Tone automatically 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. The default setting is "No".

Configures ring or tone frequencies based on parameters from local telecom. The default value is 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.

Please refer to the document below to determine your local call progress tones: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf

Call Waiting Tone Gain

Adjusts the call waiting tone volume. Users can select "Low", "Medium" or "High". The default setting is "Low".

Default Ring Cadence

Defines the ring cadence for the phone.

The default setting is c=2000/4000.

Upload Ringtone

Upload Ringtone

Users could click on “Upload” button to upload ringtone files in .mp3/.wav/.ogg format and size less than 2M. User can upload up to 10 files.

Wallpaper Settings

Wallpaper Source

Configures the source of wallpaper, if selected:

Bluetooth

Device Name

Display name when paired with other devices.

Bluetooth Headset Mode

Enables/Disables Bluetooth headset mode. The WP820 will become a headset of the paired Bluetooth device, and can answer and dial calls for the paired device after enabled.

Bluetooth

Enables/Disables Bluetooth function. The WP820 can be found by nearby Bluetooth devices after enabled.

Bluetooth Devices

Scan available devices and complete the pairing. After enables Bluetooth function, automatically scan the available Bluetooth devices.

Wi-Fi Basics

IP Mode

Select which Internet protocol to use from dropdown list:

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

Wi-Fi Signal Warning

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

Wi-Fi Function

Enables/disables the Wi-Fi feature.

The default setting is "Disable".

Wi-Fi Band

Configures the Wi-Fi frequency band from the dropdown list:

ESSID

Allows to scan and select the available Wi-Fi networks within the range if the Wi-Fi feature is enabled. Click on "Connect" to select the Wi-Fi network and connect to. The ESSID will be auto filled in the ESSID filed, users can also click on “Details” to have more details about the connected ESSID with its status, strength, speed, frequency, and security mode. User can set the IPv4 and IPv6 type to either DHCP or Static IP when clicking on “Show Advanced Options”.

Add Network

ESSID

Determines the ESSID of the default Wi-Fi network. The default password is “wp!987@dmin” and it uses WPA2PSK security access.

Security Mode for Hidden SSID

This parameter defines the security mode used for the wireless network when the SSID is hidden. 3 Modes are available:

When WPA Enterprise is selected, users can choose one of the below EAP methods which can be configured for credential based or certificate authentication.

CA Certificate

Upload the 802.1x CA certificate to the phone; or delete existed 802.1x CA certificate from the phone.

Identity

Enter the Identity information for the 802.1x mode.

Password

Configures the hidden ESSID password.

Advanced Settings

Country Code

Configures Wi-Fi country code. The default value is "United States of America".

Note: Reboot is requested to take effect.

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.

Channel Mode

"Auto" mode is enabled by default, in this mode, devices will do fully scanning; "Customize" mode is used to specify channels, and only the selected channels will be scanned and used to make network switch and roaming decision.

Captive Portal Detection

The user has the option to activate Captive Portal Detection on their phone, which allows for automatic display of the captive portal authentication page when connected to an SSID with this feature enabled. By default, this setting is enabled.

Wi-Fi Roaming

Signal Threshold

Sets the Wi-Fi signal threshold. When the Wi-Fi signal strength of the device drops below this configured value, the device will scan for a hotspot above the threshold value and connect to it. The default setting is -70 and the valid range is [-100, -30].

Roaming Gain (dB)

This feature 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 Wi-Fi signal strength is higher than the signal threshold. Default is 600s, and valid range is from 5 to 600.

Poor Signal Scanning Interval (s)

Sets the time interval for signal scanning when the Wi-Fi signal strength is lower than the signal threshold and there is no hotspot that is higher than the current signal strength. Default is 5s, and valid range is from 5 to 600.

Enable OpenVPN®

Enables/disables OpenVPN® functionality and requires the user to have access to an OpenVPN® server.

Note: To use OpenVPN® functionalities, users must enable OpenVPN® and configure all of the settings related to OpenVPN®, including server address, port, OpenVPN® CA, certificate, and key. Additionally, the user must also set the SIP account to use "VPN" for the "NAT Traversal" (under Account à Network Settings).

OpenVPN® Mode

Simple mode only supports some basic or common parameters configuration; Professional mode supports configuration file upload in zipped format, which is totally customized by need, please refer to https://openvpn.net for more information.

Enable OpenVPN® Comp-lzo

Enables/disables the LZO compression. When LZO compression is enabled on the OpenVPN server, this option needs to be checked otherwise there will be a network connection failure. The P-value for this functionality is P8508 with two possible values 0 and 1; 0 – OFF/1 - ON. Default setting is NO.

OpenVPN® Server

Defines the URL/IP address for the OpenVPN® server.

OpenVPN® Port

Defines the network port for the OpenVPN® server. The default setting is 1194.

OpenVPN® Transport

Determines network protocol used for OpenVPN® (UDP or TCP).

The default setting is TCP.

OpenVPN® CA

OpenVPN® CA file (ca.crt) required by the OpenVPN® server for authentication purposes. Press "Upload" to upload the corresponding file to the device.

OpenVPN® Client Certificate

OpenVPN® CA file (ca.crt) required by the OpenVPN® server for authentication purposes. Press "Upload" to upload the corresponding file to the device.

OpenVPN® Client Key

OpenVPN® Client key (*.key) required by OpenVPN® server for authentication purposes. Press "Upload" to upload the corresponding file to the device.

Uploads The OpenVPN® TLS key (*.key, *.pem) required by the OpenVPN® server for authentication purposes. Press "Upload" to upload the corresponding file to the device.

OpenVPN® TLS Key Type

Select the encryption type of the OpenVPN® TLS key from the frop down menu , the options available are: 

Set to "None" by Default.

OpenVPN® Cipher Method

The cipher method of OpenVPN®, must be the same cipher method used by the OpenVPN® server. Supported methods are: Blowfish, AES-128, AES-256 and Triple-DES.

OpenVPN® Username

Configures the OpenVPN® authentication username (optional).

OpenVPN® Password

Configures the OpenVPN® authentication password (optional).

Additional Options

Configures additional options to include for the OpenVPN feature (optional).

Preferred DNS 1 Server

Defines the secondary DNS IP address if the static IP is used.

Preferred DNS 2 Server

Configures the secondary alternate DNS IP address to failover during both: the default DNS server and alternate DNS server’s outage.

IPv6 Preferred DNS Server

Defines primary choice for IPv6 DNS to handle Internet protocol mapping.

Layer 3 QoS for SIP

Defines the Layer 3 packet’s QoS parameter for SIP messages in decimal pattern. This value is used for IP Precedence, Diff-Serv or MPLS.

The default setting is 26 which is equivalent to the DSCP name constant CS6.

Layer 3 QoS for Audio

Defines the Layer 3 packet’s QoS parameter for RTP messages in decimal pattern. This value is used for IP Precedence, Diff-Serv or MPLS.

The default setting is 46 which is equivalent to the DSCP name constant CS6.

HTTP/HTTPS User-Agent

Sets the user-agent for HTTP/HTTPs Request.

SIP User-Agent

Sets the user-agent for SIP. The default setting is Grandstream WP820 $version (where $version is replaced by firmware version loaded to the phone).

Proxy

HTTP/HTTPS Proxy Hostname

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

HTTP/HTTPS Proxy Port

Specifies the HTTP/HTTPS proxy port 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

Defines 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

CSTA Control

Indicates whether CSTA control feature is enabled. This allows third party applications to control the WP820 via uaCSTA standard. Supports 3CX uaCSTA web client call control. Changes made to this parameter will require reboot to take effect.

Web/SSH Access

Enable SSH

Enable/Disable SSH access to the phone. The default setting is checked.

SSH Port

Customize the SSH port.

The default setting for SSH port is 22.

Access Method

Determines which protocol will be used to access the phone ‘s Web GUI. It can be selected from HTTP and HTTPS. The default setting is HTTP.

Port

Specifies which port to use to access the phone Web UI.

By default, if HTTP, the port number will be 80; if HTTPS is selected, the port number will be 443.

Configuration via Keypad Menu

Configures the access control for the users to configure from keypad Menu. There are four different options:

The default setting is “Unrestricted”.

User Login Timeout

Set 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 is 15 minutes.

User Info Management

Current Admin Password

Enters current logged-in user's 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

Enters 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

Enters the new user password again to confirm.

TLS

Minimum TLS Version

Configures the minimum TLS version supported by the phone. Default is 1.1.

Maximum TLS Version

Configures the maximum TLS version supported by the phone. Default is No limit.

Enable or disable weak TLS cipher suites.

SIP TLS Certificate

Specifies SSL certificate used for SIP over TLS in X.509 format. The phone has built-in private key and SSL certificate.

SIP TLS Private Key

Specifies SSL certificate key used for SIP over TLS in X.509 format. The phone has built-in private key and SSL certificate.

SIP TLS Private Key Password

Specifies SSL certificate key password used for SIP over TLS in X.509 format.

Certificate Management

CA Certificate

Import Trusted CA Certificates

Allows to upload the CA Certificate file to the phone.
A certificate authority (CA) or certification authority is an entity that stores, signs, and issues digital certificates (DigiCert).
supported file extensions are .pem, .crt, .cer, .der.

Trusted CA Certificates

Lists trusted CA certificates previously uploaded. Administrator can delete a certificate from here.

User Certificate

Add Certificate

Allows to upload & Install User Certificate file to phone.
Note: The frimware upgarde to 1.0.7.66 includes the addition of Gen2 certificate support and the capability to upgrade certificates from Gen1 to Gen2.

Certificate Application

Specifies on which application, the user certificate will be applied.

Administrator can select either “VPN and other applications” or “Wi-Fi”.

Select Certificate

Browse your computer and upload the user certificate.
supported file extensions are .pem, .crt, .cer, .der.

Certificate Name

Enter a name to identify the user certificate.

Custom Certificate

Import Custom Certificates

Allows to upload the Custom Certificate file to phone.

Custom Certificate

Lists Custom certificates previously uploaded. Administrators can delete a certificate from here.
supported file extensions are .pem, .crt, .cer, .der.

Custom Certificate

Screen Lock Function

Enables/Disables screen lock function.

Screen Lock Password

Display current screen lock password.

Lock Screen Settings

Screen Lock Function

Allows the user to configure lock screen functions on the web UI. The screen lock password was also added to provide additional security. Default Disabled.

Note: If 'Manual' selected, the device should be locked manually by pressing and holding # key.

Screen Lock Password

Add new digit password, edit current password and it displays the current screen lock password.

Sound

Silent Mode

Enables silent mode.

Vibrate Mode

Enables vibrate mode.

Charging Tone

If set to "Yes”, it will sound a tone once you start charging.

Low Battery Warning Tone

If set to "Yes”, it will sound a tone when LCD prompts low battery.

Voicemail Notification Tone

If set to "Yes”, it will sound a tone when LCD prompts unread voicemail message.

Lock Volume

If set to ”Yes”, volume adjustments on the handset become locked.

Ringtone volume

Configures the volume of ringtone.

Media volume

Configures the volume of media.

Alarm volume

Configures the volume of the alarm.

Default Ringtone

Configure default ringtone for incoming calls.

Notification Tone

Configure notification tone for new SMS, Voicemail etc.

LCD & LED Management

Enable Missed Call Indicator

When “checked", the LED indicator on the upper right corner of the phone will light up when there is missed call on the phone.

Enable MWI Indicator

When “checked", the LED indicator on the upper right corner of the phone will light up when there is new voicemail on the phone

Enable New Message Indicator

Turns on the LED indicator at device’s upper right corner when there are unread Messages. If it set to “checked”, the LED indicator will keep off if the phone has new messages. Otherwise, the LED indicator will be turned on. Default setting is "checked".

Enable Keypad Backlight

When “checked", the backlight will light up when the button is pressed.

Default setting is "checked".

Enable Incoming Call Indicator

The LED indicator on the upper right corner of the phone will light up when there is an incoming call.

Enabled by Default.

Enable Charging Indicator

When the phone is charging, the LED indicator on the upper right corner will light up.

Enabled by Default.

Enable Charging Completed Indicator

When the phone is charged, the LED indicator will illuminate in the upper right corner.

Brightness

Configures the Brightness of LCD backlight.

Timeout

Sets the timeout interval of the LCD backlight. If timeout is set to ‘never’, the screen will always stay on. The value range is between 15s and 30 minutes.

Font Size

Set the font size for the LCD display. Choices are: Standard, Small, and Large.

Gesture Control

Flip phone

Reject incoming call

If set to “Reject”, the phone will reject incoming calls if users flip the phone. If set to “Reject with SMS”, the phone will reject incoming calls when flipping the phone and send a pre-configured SMS to the caller. The default setting is “Off”.

Pick up the phone

Ringtone

If set to “Mute”, the ringtone will be muted once users pick up the phone when receiving an incoming call. If set to “Ringing Down”, the ringtone’s volume will decrease once the phone is picked up when receiving incoming calls. The default setting is “Off”.

Alarm Ringtone

If set to “Mute”, the alarm ringtone will be muted once users pick up the phone during alarm. If set to “Ringing Down”, the alarm ringtone’s volume will decrease once the phone is picked up during alarm. The default setting is “Off”.

Button Customization

Soft key

Main Screen Left Softkey

Configures the function of the left softkey when users are on the Main screen.

This can be set to “Call History”, ”Menu”, “Settings”, “Contacts”, “Voicemail”, “DND”, “LDAP”.

Main Screen Center Softkey

Configures the function of the center softkey when users are on the Main screen.

This can be set to “Call History”, ”Menu”, “Settings”, “Contacts”, “Voicemail”, “DND”, “LDAP”.

Main Screen Right Softkey

Configures the function of the right softkey when users are on the Main screen.

This can be set to “Call History”, ”Menu”, “Settings”, “Contacts”, “Voicemail”, “DND”, “LDAP”.

Call Screen Left Softkey

Configures the function of the left softkey when users are on the Call screen.

This can be set to “Option”, “Transfer”, “Hold”, “Mute”, “New Call”, “Conference”, “Record”.

Call Screen Center Softkey

Configures the function of the center softkey when users are on the Call screen.

This can be set to “Option”, “Transfer”, “Hold”, “Mute”, “New Call”, “Conference”, “Record”.

Call Screen Right Softkey

Configures the function of the right softkey when users are on the Call screen.

This can be set to “Option”, “Transfer”, “Hold”, “Mute”, “New Call”, “Conference”, “Record”.

Multi-function Button

PTT/Paging

If “PTT/Paging” is chosen to be the function for Multi-function button, a short press on the button will take users to PTT/Paging settings. Users can also press and hold the key to initiate PTT/Paging call.

SAFE/Panic Call

If “SAFE/Panic Call” is configured to be the function for Multi-function button, a short press on the key will open status page to make sure that Panic Call is enabled. Users can also press and hold the key to trigger Panic Call.

Mute

If “Mute” is chosen, users will be able to mute incoming calls ringtone by pressing on the key. Also, the key will be performing mute/unmute function during calls.

Speed Dial

Short press to open status page, long press to dial a contact.

Disable

If set to “Disabled”, Pressing the multi-function button has no function.

RADIUS for Web Access Authentication

Enable RADIUS Web Access Control

Once enabled, web login authentication will be performed through the RADIUS server. Please set the web access method to HTTPS to ensure security.
Disabled by Default

Action upon RADIUS Auth Server Error

Selects the handling method when RADIUS authentication fails. If set to "Reject Access", the web access will be denied if RADIUS authentication fails. If set to "Authenticate Locally", it will fall back to do local authentication if RADIUS authentication fails. Default is "Authenticate Locally".

RADIUS Auth Protocol

Configure RADIUS authentication protocol, the available options are:

Default value is "PAP".

RADIUS Auth Server Address 

RADIUS Auth Server Port

RADIUS Shared Secret

Configures RADIUS Shared Secret, RADIUS Shared Secret is a confidential key used to secure communications between the RADIUS client (e.g., a network device) and the RADIUS server for authentication and accounting purposes.
Example: "Secret123!"

RADIUS VSA Vendor ID

RADIUS VSA Access Level Attribute 

Firmware

Complete Upgrade

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

Upload Firmware File to Update

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

Firmware Upgrade Mode

Allows users to choose the firmware upgrade method: TFTP, HTTP, HTTPS, or Manual Upload. The default setting is “HTTPS”.

Firmware Server Path

Sets IP address or domain name of firmware server. The URL of the server that hosts the firmware release. Default is “fm.grandstream.com/gs”.

Note: starting from the firmware version 1.0.7.61, the firmware server path now does not have any input limitation, you can provide any valid type of URL.

HTTP/HTTPS User Name

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

HTTP/HTTPS Password

Enters the password for the firmware HTTP/HTTPS server.

Firmware File Prefix

Checks if firmware file is with matching prefix before downloading it. This field enables user to store different versions of firmware files in one directory on the firmware server.

Firmware File Postfix

Checks if firmware file is with matching postfix before downloading it. This field enables user to store different versions of firmware files in one directory on the firmware server.

Update Detect

Click the "Update Detect" button to check whether the firmware in the firmware server has an updated version, if so, proceed to firmware upgrade immediately.

Config File

Download Device Configuration

Downloads the phone's configuration file in text format. The config file includes all the P value parameters for phone's current settings except password for security purpose. Users can use the Grandstream configuration file generator to generate binary config file from this text file.

Upload Device Configuration

Uploads configuration file “cfgMAC.xml” or “cfgwp820.xml” to the phone.

Note: Reboot is required to take effect.

Use Grandstream GAPS

Configures the download path and update mode for the configuration file server.

Config Update Via 

Allows users to choose the config update method: TFTP, HTTP or HTTPS.
Default value is HTTP.

Config Server Path

Defines the configuration server path for provisioning. It could be different from the Firmware Server Path.
Note: starting from the firmware version 1.0.7.61, the Config server path now does not have any input limitation, you can provide any valid type of URL.

HTTP/HTTPS User Name

Configures the user name for the config HTTP/HTTPS server.

HTTP/HTTPS Password

Configures 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 phone.

Config File Postfix

Enables your ITSP to lock configuration updates.

If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone.

Authenticate Conf File

Sets the phone to authenticate configuration file before applying it.

When set to “Yes”, the configuration file must include value P1 with phone’s administration password. If it is missed or does not match the password, the phone will not apply it. The default setting is “No”.

XML Config File Password

Decrypts XML configuration file when encrypted. The password used for encrypting the XML configuration file is using OpenSSL.

Provision

Automatic Upgrade

Automatic Upgrade

Specifies when the firmware upgrade process will be initiated; there are 4 options:

Note: Day of week is starting from Sunday. The default setting is “No”.

Configures whether the phone will upgrade automatically at random time point within the configured period. This option is mainly used for multiple phones upgrade at the same time.

Firmware Upgrade and Provisioning

Defines the phone’s rules for automatic upgrade. It can be selected from:

The default setting is “Always Check at bootup”.

Upgrade without Prompt

If set to "Yes", the phone will automatically start upgrading after downloading the firmware file. Otherwise, users would need to confirm in the prompted message on the LCD screen to start upgrading process. The default setting is "Yes".

DHCP Option

Allow DHCP Option 150, 43, 160 and Option 66 to Override Server

If set to "Yes":

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 the DHCP Option fails.

Configures the phone to allow the DHCP offer message to override the Config Server Path via the Option 120 header. The default setting is "Yes".

Allow DHCP Option 242 (Avaya IP Phones)

Enables DHCP Option 242. Once enabled, the phone will use the configuration info issued by the local DHCP in Option 242 to configure proxy, transport protocol and server path. The default setting is “Yes”.

Config Provision

Download and Process All Available Config Files

By default, the device will provision the first available config in the order of cfgMAC, 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, cfgMAC.xml, cfgMODEL.xml, cfg.xml.

Config Provision

This feature’s purpose is to make the device downloads the configuration files and provision by the configured order.

PNP Feature

PNP (3CX) Auto Provision

Sets the phone system to broadcast the SIP SUBSCRIBE message during booting up to allow itself to be discovered and be configured by the SIP platform. The default setting is "Checked".

Note: This setting requires rebooting the phone to take effect.

Advanced Settings

Disable SIP NOTIFY Authentication

Disables the SIP NOTIFY Authentication on the phone. If set to “Yes”, the phone will not challenge NOTIFY with 401.

The default setting is “No”.

Validate Certification Chain

Configures whether to validate the server certificate when download the firmware/config file. If it is set to "Yes", the phone will download the firmware/config file only from the legitimate server.

The default setting is "No".

mDNS Override Server

Sets the phone to broadcast the Multicast DNS (mDNS) message during booting up to allow itself to be discovered and be configured by the SIP platform. If it is set to “User Type A’, the phone will broadcast the mDNS message “A_grandstream-cfg.local”; if it is set to “Use Type SRV”, the mDNS message will be “SRV_grandstream-cfg.local”.

The default setting is “Use Type A”.

Factory Reset

Resets the phone to the default factory setting mode.

Syslog

Syslog Protocol

Select the transport protocol over which log messages will be carried.

Syslog Level

Selects the level of logging for syslog. The default setting is "None".

There are 4 levels from the dropdown list: DEBUG, INFO, WARNING and ERROR. The following information will be included in the syslog packet:

Note: Changing syslog level does not require a reboot to take effect.

Syslog Server

Configures the URI which the phone will send the syslog messages to.

The default setting is "log.ipvideotalk.com".

Syslog Keyword Filter

Only send the syslog with keyword, multiple keywords are separated by comma. Example: set the filter keyword to “SIP” to filter SIP log.

Logcat

Clear Log

Clears the log files saved in the phone by pressing the clear button.

Log Tag

Configures the filter to display the specified process log file.

Log Priority

Selects the log priority to display. It can be selected from list below:

Get Log

Get the log files saved in the phone.

Debug

One-click Debugging

One-click Debugging

Captures the checked info in the debugging list, click "Start" to debug if including "Capture trace" item and click "Stop" to end, Click "Capture" in another situation. All retrieved files will be generated to a package, and the last package will be overwritten, while the trace file will stay remain.

Debug Info Menu

Displays a list of info items that can be debugged, currently supports system logs, info log, capture trace, tombstones, ANR log and Battery Log.

The captured data can be viewed in "Debug information list".

The default is all selected.

Debug Info List

Selects the existing debugging info package or grab package.

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 captured files are sorted in chronological order, click to download the file to the computer for analysis.

Core Dump

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

Selects the existing core dump file in the drop-down box. Users could delete the file by pressing on “Delete” button.

View Core Dump

Press “List” button to view all existing core dump files. The files are listed in chronological order, users could click the file name to download the file to the local computer.

Record

Record

Click to start capturing audio data, click the "Stop" button to end. To capture the audio data of the device can help to locate audio issues. The default is not enabled. You can record up to 1-minute audio data.

Recording List

Choose the existing audio file and click the "Delete" button on the right to delete it.

View Recording

Click on the "List" button to view. The captured audio data will be sorted by time. Click to download the data to the computer for analysis. Users can also delete files under this folder.

Screenshot

Screenshot

Click the screenshot to capture and save the current phone screen image.

Screenshot List

Selects the existing screenshot file and click the "Delete" button on the right to delete the file.

View Screenshot

Click the "List" button to view the existing screenshot file. Click the file name to open or save the screenshot to the computer.

Traceroute

Target Host

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

Press Start to send traceroute request to configured target host.

Press Stop to end traceroute running process.

Ping

Target Host

Configures the IP address or URL for the Target Host in order to ping it.

Press Start to start pinging the configured target host.

Press Stop to pinging the configured host.

NSLookup

Host Name

Enter a host name and find out the corresponding IP address. It will also do reverse name lookup and find the host name for an IP address you specify.