GXV3380 - Administration Guide

Account

16 SIP accounts on the phone.

Number

SIP User ID for the account.

SIP Server

URL or IP address, and port of the SIP server.
Note: this parameter also supports domain string such as “grandstream”.

Status

Registration status for the SIP account.

GS Wave

Users can click “Enter” button to open GS Wave web page.

MAC Address

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

NAT Type

Type of NAT connection used by the phone.

IPv4

IPv4 Address Type

Configured address type: DHCP, Static IP or PPPoE.

IPv4 Address

IP address of the phone.

Subnet Mask

Subnet mask of the phone.

Default Gateway

Default gateway of the phone.

DNS Server 1

DNS Server 1 of the phone.

DNS Server 2

DNS Server 2 of the phone.

IPv6

IPv6 Address Type

Configured address type: DHCP, Static IP or PPPoE.

IPv6 Address

IPv6 address of the phone.

IPv6 DNS Server 1

IPv6 DNS Server 1 of the phone.

IPv6 DNS Server 2

IPv6 DNS Server 2 of the phone.

Product Model

Product model of the phone: GXV3380.

Hardware Revision

Hardware version number.

Part Number

Product part number.

Serial Number

Serial Number of the phone.

System Version

Firmware version ID. This is the main software release version, which is used to identify the software system of the phone.

Recovery Version

Recovery image version.

Boot Version

Booting code version.

Kernel Version

The kernel version.

CPE Version

The CPE version

System Up Time

System up time since the last reboot.

Account Registration

Account Active

Indicates whether the account is active. The default value for the first account is “Yes”.

Account Name

Configures the name associated with each account to be displayed on the LCD.

SIP Server

Specifies the URL or IP address, and port of the SIP server. This should be provided by VoIP service provider (ITSP).

SIP User ID

Configures user account information provided by your VoIP service provider (ITSP). It’s usually in the form of digits similar to phone number or actually a phone number.

SIP Authentication ID

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

SIP Authentication Password

Configures the account password required for the phone to authenticate with the ITSP (SIP) server before the account can be registered. After saving, it will appear as hidden for security purpose.

Display Name

Specifies the SIP server subscriber’s name (optional) that will be used for Caller ID display. The configured content will be included in the From, Contact and P-Preferred-Identity headers of SIP INVITE message.

Tel URI

Indicates E.164 number in “From” header by adding “User=Phone” parameter or using “Tel:” in SIP packets, if the phone has an assigned PSTN Number.

• Disabled: Will use “SIP User ID” information in the Request-Line and “From” header.
• User=Phone: “User=Phone” parameter will be attached to the Request-Line and “From” header in the SIP request to indicate the E.164 number. If set to “Enable”.
• Enabled: “Tel:” will be used instead of “sip:” in the SIP request.

Please consult your carrier before changing this parameter. Default is “Disabled”.

Voice Mail Access Number

Sets if the phone system allows users to access the voice messages by pressing the MESSAGE key on the phone. This ID is usually the VM portal access number. For example, in UCM6xxx IPPBX, *97 could be used.

Network Settings

Outbound Proxy

Configures the IP address or the domain name of the primary outbound proxy, media gateway or session border controller. It’s used by the phone for firewall or NAT penetration in different network environments. If a symmetric NAT is detected, STUN will not work and only an outbound proxy can provide a solution

Secondary Outbound Proxy

Sets IP address or domain name of the secondary outbound proxy, media gateway or session border controller. The phone system will try to connect the Secondary outbound proxy only if the primary outbound proxy fails.

DNS Mode

Defines which DNS service will be used to lookup IP address for SIP server’s hostname. There are three modes:

• A Record
• SRV
• NATPTR/SRV

To locate the server by DNS SRV set this option to “SRV” or “NATPTR/SRV”. Default setting is “A Record”.

Maximum Number Of SIP Request Retries

This setting configures 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 method for DNS SRV failover. There are Three DNS SRV Failover mode:

• If “Default” is selected, the primary SIP server or Outbound Proxy will always be attempted first for all REGISTER and INVITE requests.
• If “Use current server until DNS TTL” is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until DNS times out.
• If “Use current server until no response” is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until there is no response.
• 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)

This option is configured when DNS SRV failover mode “Failback follows failback expiration timer” is chosen. It 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.

Valid range: 1 – 64800

Register Before DNS SRV Fail-over

To set register before DNS SRV Fail-over. The default value is “Yes”

NAT Traversal

Specifies which NAT traversal mechanism will be enabled on the phone system. It can be selected from the dropdown list:

• NAT NO
• STUN
• Keep-alive
• UPnP
• Auto
• OpenVPN®

If the outbound proxy is configured and used, it can be set to “NAT NO”.

If set to “STUN” and STUN server is configured, the phone system will periodically send STUN message to the SUTN server to get the public IP address of its NAT environment and keep the NAT port open. STUN will not work if the NAT is symmetric type.

If set to “Keep-alive”, the phone system will send the STUN packets to maintain the connection that is first established during registration of the phone. The “Keep-alive” packets will fool the NAT device into keeping the connection open and this allows the host server to send SIP requests directly to the registered phone.

If it needs to use OpenVPN to connect host server, it needs to set it to “VPN”. If the firewall and the SIP device behind the firewall are both able to use UPNP, it can be set to “UPNP”. The both parties will negotiate to use which port to allow SIP through. The default setting is “Keep-alive”.

Proxy-Require

Adds the Proxy-Required header in the SIP message. It is used to indicate proxy-sensitive features that must be supported by the proxy. Do not configure this parameter unless this feature is supported on the SIP server.

SIP Basic Settings

SIP Registration

Allows the phone system to send SIP REGISTER messages to the proxy/server. The default setting is “Yes”.

Unregister before New Registration

Controls whether to clear SIP user’s information by sending un-register request to the proxy server.

• When set to “All”, the un-registration is performed by sending a REGISTER message with “Contact” header set to * and Expires=0 parameters to the SIP server when the phone starts pre-registration after rebooting.
• If set to “Instance”, the phone only cleans the current SIP user’s info by sending REGISTER message with “Contact” header set to concerned SIP user’s info and Expires=0 parameters to the SIP server.

The default setting is “Instance”.

Register

Expiration (m)

Configures the time period (in minutes) in which the phone refreshes its registration with the specified registrar. The default setting is 60. The maximum value is 64800 (about 45 days).

Subscribe Expiration (m)

Specifies the frequency (in minutes) in which the phone refreshes its subscription with the specified registrar. The maximum value is 64800 (about 45 days). Default value is 60.

Re-register before Expiration (s)

Specifies the time frequency (in seconds) that the phone sends re-registration request before the Register Expiration. The default setting is 0. The range is from 0 to 64,800.

Registration Retry Wait Time (s)

Configures the time period (in seconds) in which the phone will retry the registration process in the event that is failed. The default setting is 20. The maximum value is 3600 (1 hour).

Add Auth Header On RE-REGISTER

Configure if the SIP account needs to add Auth header in RE-REGISTER.

• If the option is checked, device will always add authentication header in REGISTER.
• If the option is unchecked, device will only send authentication for the first REGISTER.

Enable SIP OPTIONS Keep Alive

Enables SIP OPTIONS to track account registration status so the phone will send periodic OPTIONS message to server to track the connection status with the server. The default setting is “No”.

SIP OPTIONS Keep Alive Interval (s)

Configures the time interval when the phone sends OPTIONS message to SIP server. The default setting is 30 seconds, which means the phone will send an OPTIONS message to the server every 30 seconds. The default range is 1-64800.

SIP OPTIONS Keep Alive Maximum Tries

Configures the maximum times of sending OPTIONS message consistently from the phone to server. Phone will keep sending OPTIONS messages until it receives response from SIP server. The default setting is “3”, which means when the phone sends OPTIONS message for 3 times, and SIP server does not respond this message, the phone will send RE-REGISTER message to register again. The valid range is 3-10.

Subscribe for MWI

Configures the phone system to subscribe voice message service. If it is set to “Yes”, the phone system will periodically send SIP SUBSCRIBE message for Message Waiting Indication service. GXV3380 phone system supports both synchronized and non-synchronized MWI. The default setting is “No”.

Use Privacy Header

Determines if the Privacy header will be presented in the SIP INVITE message and if it includes the caller info in this header. If it is set to “Default”, the Privacy Header will be omitted in INVITE when “Huawei IMS” special feature is active. If set to “Yes”, it will always be presented. If set to “No”, it will always be omitted. The default setting is “Default”.

Use P-Preferred-Identity Header

Specifies if the P-Preferred-Identity Header will be presented in the SIP INVITE message. If set to “default”, the P-Preferred-Identity Header will be omitted in SIP INVITE message when “Huawei IMS” special feature is active. If set to “Yes”, the P-Preferred-Identity Header will always be presented. If set to “No”, it will be omitted. The default setting is “Default”.

Use P-Access-Network-Info Header

Enables/disables the use of P-Access-Network-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is “No”.

Use P-Emergency-Info Header

Enables/disables the use of P-Emergency-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is “No”.

Use Mac Header

• If Yes, the SIP message for register or unregister will contains MAC address in the header, and all the outgoing SIP message including REGISTER will attach the MAC address to the User-Agent header;
• If No, neither will the MAC header be included in the register or unregister message nor the MAC address be attached to the User-Agent header for any outgoing SIP message.

SIP Transport

Determines which network protocol will be used to transport the SIP message. It can be selected from TCP/UDP/TLS.

Default setting is “UDP”.

Local SIP Port

Determines the local SIP port used to listen and transmit. The default setting is 5060 for Account 1, 5062 for Account 2, 5064 for Account 3, 5066 for Account 4, 5068 for Account 5, and 5070 for Account 6. The valid range is from 5 to 65535.

SIP URI Scheme When Using TLS

Defines which SIP header, “sip” or “sips”, will be used if TLS is selected for SIP Transport. The default setting is “sip”.

Use Actual Ephemeral Port in Contact with TCP/TLS

Determines the port information in the Via header and Contact header of SIP message when the phone system use TCP or TLS. If set to No, these port numbers will use the permanent listening port on the phone. Otherwise, they will use the ephemeral port for the particular connection. The default setting is “No”.

Support SIP Instance ID

Determines if the phone system will send SIP Instance ID. The SIP instance ID is used to uniquely identify the device. If set to “Yes”, the SIP Register message Contact header will include +sip.instance tag.

The default setting is “Yes”.

SIP T1 Timeout

Defines an estimate of the round-trip time of transactions between a client and server. If no response is received in T1, the figure will increase to 2*T1 and then 4*T1. The request re-transmit retries would continue until a maximum amount of time define by T2.

The default setting is 0.5 sec.

SIP T2 Interval

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

SIP Timer D Interval

Defines the amount of time that the server transaction can remain when unreliable response (3xx-6xx) received. The valid value is 0-64 seconds.

The default value is 0.

Remove OBP from Route

Configures the phone system to remove the outbound proxy URI from the Route header. This is used for the SIP Extension to notify the SIP server that the device is behind a NAT/Firewall. If it is set to “Yes”, it will remove the Route header from SIP requests. The default setting is “No”.

Enable 100rel

Actives PRACK (Provisional Acknowledgment) method. PRACK improves the network reliability by adding an acknowledgement system to the provisional Responses (1xx). It is set to “Yes”, the phone system will response to the 1xx response from the remote party. Default is “No”.

Session Timer

Enable Session Timer

Allows the phone system to use the session timer, when set to “Yes”, it will be added in the SIP INVITE message to notify the server.

Session Expiration (s)

Configures the phone system’s SIP session timer. 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. Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand.

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

Min-SE (s)

Determines the minimum session expiration timer (in seconds) if the phone act as a timer refresher.

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

UAC Specify Refresher

Sets which party will refresh the active session if the phone makes outbound calls. If it is set to “UAC” and the remote party does not support Refresher feature, the phone system will refresh the active session.

If it is set to “UAS”, the remote party will refresh it. If it is set to “Omit”, the header will be omitted so that it can be selected by the negotiation mechanism. The default setting is “Omit”.

UAS Specify Refresher

Specifies which party will refresh the active session if the phone receives inbound calls. If it is set to “UAC”, the remote party will refresh the active session. If it is set to “UAS” and the remote party does not support refresh feature, the phone system will refresh it.

The default setting is “UAC”.

Caller Request Timer

Sets the caller party to act as refresher by force. If set to “Yes” and both party support session timers, the phone will enable the session timer feature when it makes outbound calls. The SIP INVITE will include the content “refresher=uac”. The default setting is “No”.

Callee Request Timer

Sets the callee party to act as refresher by force. If set to “Yes” and the both parties support session timers, the phone will enable the session timer feature when it receives inbound calls. The SIP 200 OK will include the content “refresher=uas”. The default setting is “No”.

Force Timer

Configures the session timer feature on the phone system by force.

• If it is set to “Yes”, the phone will use the session timer even if the remote party does not support this feature.
• If set to “No”, the phone will enable the session timer only when the remote party supports this feature. To turn off the session timer, select “No”.

The default setting is “No”.

Force INVITE

Sets the SIP message type for refresh the session. If it is set to “Yes”, the Session Timer will be refreshed by using the SIP INVITE message. Otherwise, the phone system will use the SIP UPDATE or SIP OPTIONS message. Default is “No”.

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.

Available vocoders are: PCMU / PCMA / G722 / G729A/B / G726-32 / iLBC / 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.

The default setting is “Callee”.

Use First Matching Vocoder in 200OK SDP

Configures the phone to use the first matching codec in the 200OK message. The default value is 0.

iLBC Frame Size

Sets the ILBC (Internet Low Bitrate Codec) frame size if ILBC is used. Users can select it from 20ms or 30ms. The default setting is 30ms.

G726-32 ITU Payload

Configures G726-32 payload type for ITU packing mode. Payload 2 is static and payload dynamic is dynamic.

The default setting is “2”.

G726-32 Dynamic Payload Type

Specifies the G726-32 payload type, and the valid range is 96 to 126. The default setting is “126”.

Opus Payload Type

Defines the desired value (96-126) for the payload type of the Opus codec. The default value is 123.

DTMF

Specifies the mechanism to transmit DTMF (Dual Tone Multi-Frequency) signals.

There are 3 supported modes: in audio, RFC2833, or SIP INFO.

• In audio, which means DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs);

• RFC2833, which means to specify DTMF with RTP packet. Users could know the packet is DTMF in the RTP header as well as the type of DTMF.

• SIP INFO, which uses SIP INFO to carry DTMF. The defect of this mode is that it’s easily to cause desynchronized of DTMF and media packet if the SIP and RTP messages are required to transmitted respectively.

The default setting is “RFC2833”.

DTMF Payload Type

Configures the RTP payload type that indicates the transmitted packet contains DTMF digits. Valid range is from 96 to 126. Default value is 101.

Security Settings

Check Domain Certificates

Sets the phone system to check the domain certificates if 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 system 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

It is used to configure whether to dial the number carried by Refer-to after receiving SIP REFER request actively. The default is “Disabled”.

• If it is set to “Disabled“, the phone will send error warning and stop dialing.
• If it is set to “Enabled/Force Auth“, the phone will dial the number after sending authentication, if the authentication failed, then the dialing will be stopped.
• If it is set to “Enabled“, the phone will dial up all numbers carried by SIP REFER.

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

Allow SIP Reset

It is used to configure whether to allow SIP Notification message to perform factory reset on the phone. The default setting is “No”.

Authenticate Incoming INVITE

Configures the phone system 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.

Default setting is “No”.

Advanced Call Function

Virtual Account Group

It is used to set to categorize accounts in server mode groups, the accounts in the same group will be combined as one and the account widget will display the Caller ID in the account with lowest ID. The phone can answer any incoming calls to each account in groups.

If user makes an outbound call, the phone system will use the lowest ID account by default. If the account fails or SIP INVITE message is timeout, the phone system will failover to the next account in the group with higher account ID.

If all the accounts are not available in the group, the phone system will traverse all the accounts in the group and notify the end users the session is failed.

Allow Sync Phonebook via SIP Notify

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

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. Default is “Yes”.

Basic Settings

Local RTP Port

Defines the local RTP-RTCP port pair used to listen and transmit.

The following rule is applied: N>=0

– Audio RTP port: Port_Value+10*N
– Audio RTCP port: Port_Value+10*N+1
– Video RTP port: Port_Value+10*N+2
– Video RTCP port: Port_Value+10*N+3
– FEC RTP port: Port_Value+10*N+4
– FEC RTCP port: Port_Value+10*N+5
– BFCP Protocol port: Port_Value+10*N+6
– BFCP RTP port: Port_Value+10*N+8
– BFCP RTCP port: Port_Value+10*N+9

The default value is 50040. The valid range is from 50040 to 65000.

Use Random Port

Forces the phone system 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.

Hide User Info for Video Call

Configures whether to display user information in a video call. If set to “Yes”, user information will not be displayed in the upper left corner of video area during a video call.

Enable in-call DTMF display

Enables/disables the phone system to omit the DTMF digits displaying from the LCD screen. The default setting is “Yes”.

Enable Enterprise Contacts Timeout Auto Search

Configures whether to display the matched content automatically in search of the LDAP contacts when timeout. If set to “No”, users need to click the “Search” button to search the matched contacts mentioned above. The default setting is “Yes”.

Keep-alive Interval (s)

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

STUN Server

Configures the URI of STUN (Simple Traversal of UDP for NAT) server. The phone system will send STUN Binding Request packet to the STUN server to learn the public IP address of its network. Only non-symmetric NAT routers work with STUN. The default setting is “stun.ipvideotalk.com”.

TURN Server Username

Fill in the username to validate TURN server.

TURN Server Password

Fill in the password to validate TURN 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’s required by your ITSP. The default setting is keep the box blank.

Enable Video Call Feature

Enables the video call feature on the phone.

The default setting is “Yes”.

Use Direct IP Call Mode

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

Default is “No”.

Use Paging Call Mode

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

Default is “No”.

Video Frame Rate

Configures video frame rate for SIP video call from “5 frames/second”, “15 frames/second”, “25 frames/second” and “30 frames/second”. The default setting is 15 frames/second. The video frame rate is adjustable based on network conditions. Increasing the frame rate will significantly increase the amount of data transmitted, therefore consuming more bandwidth. The video quality will be affected due to packet loss if extra bandwidth is not allocated.

Video Display Mode

Configures the video display mode to “Original proportion”, “Cut proportionally” or “Add black margin proportionally”.

• If set to “Original proportion“, the phone displays video in its original proportion.

If the video display proportion is different from the one of the phone, the phone will stretch or compress video to display it.

• If set to “Cut proportionally“, the phone will cut video to meet its own display proportion.
• If set to “Add black margin proportionally“, the phone will display video in its original proportion. If it still exists spare space, the phone will add black edge on it.

The default setting is ” Cut proportionally “.

Enable Frame Skipping in Video Decoder

Enables the phone system for frame skipping in video decoder. If it is enabled, the video decoder will skip the P frame and start decoding from the next I frame. Enabling this option will help reduce flickering in the video when the bandwidth is limited in the network environment. The default setting is “Yes”.

Multicast Paging

Paging Barge

Sets the threshold of paging calls. If the paging call’s priority is higher than the threshold, the existing call will be hold and the paging call will be answered. Otherwise, the existing call does not be affected. If it is set to Disable, any paging call will not be answered.

Default setting is “Disable”.

Paging Priority Active

Determines if a new paging call whose priority is higher than the existing paging call will be answered. If it is checked, this feature will be enabled. The default setting is disabled.

Multicast Paging Codec

Selects the codec type for the multicast paging call. This list includes PCMU, PCMA, G726-32, G722, and G729A/B, iLBC, Opus.

Enable Multicast Paging Video

Enables the video feature to establish a multicast paging call.

The default setting is disabled.

Multicast Paging Video Codec

Sets the video codec for the multicast paging call.

The default setting is “H.264”.

Multicast Paging Image Size

Sets the video image size for the multicast paging call. This list includes 1080P, 720P, 4CIF, VGA, CIF, QVGA, and QCIF. Default setting is “VGA”.

Multicast Paging Video Bit Rate

Determines the video bit rate for the multicast paging call.

The default setting is “256 kbps”.

Multicast Paging Video Frame Rate

Configures the video frame rate for the multicast paging call. This list includes “15 frames/second”, “25 frames/second”, “30 frames/second”, and “Variable frames rate”.

Multicast Paging H.264 Profile Type

Specifies the H.264 codec profile type for the multicast paging call. This list includes “Baseline Profile”, “Main Profile”, and “High Profile”.

Note: Lower profile is easier to decode, while higher profile has higher compress rate. Usually, use Baseline profile for low CPU Performance device, and choose high profile for video conference.

Multicast Paging H.264 Payload Type

Determines the H.264 codec payload type for the multicast paging call.

The default setting is “99”.

Multicast Listening

• Priority
• Listening Address
• Label

Configures the IP address and port number for monitoring multicast paging call. When the initiator initiates a call, answer the call and display listening address and tag of the monitoring target. This feature supports Video Multicast, when the initiator initiates Video Multicast, it will automatically add the number 2 on the port of the listening address. Reboot the phone to make changes take effect.

The valid IP address range is from 224.0.0.0 to 239.255.255.255. Users may also fill the label for each listening address corresponding to priority.

Preferred Internet Protocol

If IPv4 is selected, the phone will be using IPv4 addressing, otherwise, it will be using IPv6 addressing. Default is “IPv4 Only”.

Different Networks for Data and VoIP Calls

Configures whether to set up different networks for the phone data and the call. If set to “Yes”, you need to configure the data network and VoIP network respectively.

IPv4

IPv4 Address Type

Configures the appropriate network settings on the phone. Users could select from “DHCP”, “Static IP” or “PPPoE”(Point-to-point Protocol over Ethernet). By default, it is set to “DHCP”.

DHCP VLAN Override

Selects the DHCP Option VLAN mode. When set to “DHCP Option 132 and DHCP option 133”, the phone will get DHCP option 132 and 133 as VLAN ID and VLAN priority. When set to “Encapsulated in DHCP Option 43”, the phone will get values from Option 43 which encapsulate VLAN ID and VLAN priority.

Note: Please make sure the “Allow DHCP Option 43 and Option 66 to Override Server” setting under maintenance🡪upgrade is checked. The default setting is ” Encapsulated in DHCP Option 43 “.

DHCP Host name (Option 12)

Sets the name of the client in the DHCP request. It is optional but may be required by some Internet Service Providers.

Vendor Class ID

(Option 60)

Configures the vendor class ID header in the DHCP request.

The default setting is Grandstream GXV3380.

IP Address

Defines the phone’s static IP address if the static IP is used.

Subnet Mask

Determines the network’s subnet mask if the static IP is used.

Default Gateway

Defines the network’s gateway address if the static IP is used.

DNS Server 1

Configures the primary DNS IP address if the static IP is used.

DNS Server 2

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

PPPoE Account ID

Configures the PPPoE account ID if the PPPoE is used.

PPPoE Password

Sets the PPPoE password if the PPPoE is used.

Layer 2 QoS 802.1Q/VLAN Tag (Ethernet)

Assigns the VLAN Tag of the Layer 2 QoS packets for Ethernet.

The Default value is 0.

Note: When Different Networks for Data and VoIP Calls set to Yes, then this option will be applied for Data.

Layer 2 QoS 802.1p Priority Value (Ethernet)

Assigns the priority value of the Layer 2 QoS packets for Ethernet.

The Default value is 0.

Note: When Different Networks for Data and VoIP Calls set to Yes, then this option will be applied for Data.

IPv6

IPv6 Address

Configures the appropriate network settings on the phone.

Users could select from “Auto-configured” or “Statically configured”.

DNS Server 1

Configures the primary DNS IP address.

DNS Server 2

Configures the secondary DNS IP address.

Static IPv6 Address

Enter the static IPv6 address in “Statically configured” IPv6 address type.

IPv6 Prefix Length

Enter the IPv6 prefix length in “Statically configured” IPv6 address type. Default is 64.

Preferred DNS Server

Enter the Preferred DNS server.

IPv4

IPv4 Address Type

Users could select “DHCP” or “Static IP”.
• DHCP: Obtain IP address via a DHCP server in the LAN. All domain values for static IP/PPPoE are unavailable, even though the values have been saved in the flash.
• Static IP: Manually configures IP Address, Subnet Mask, Default Router’s IP Address, DNS Server 1 and DNS Server 2.

By default, it is set to “DHCP”.

IP Address

Defines the phone’s static IP address if the static IP is used.

Subnet Mask

Determines the network’s subnet mask if the static IP is used.

Default Gateway

Defines the network’s gateway address if the static IP is used.

DNS Server 1

Configures the primary DNS IP address if the static IP is used.

DNS Server 2

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

Layer 2 QoS 802.1Q/VLAN Tag (Ethernet) for VoIP Calls

When Different Networks for Data and VoIP Calls set to Yes, then this option will be applied for VoIP traffic. Assigns the VLAN Tag of the Layer 2 QoS packets for Ethernet for VoIP Calls. The Default value is 0.

Layer 2 QoS 802.1p Priority Value (Ethernet) for VoIP Calls

When Different Networks for Data and VoIP Calls set to Yes, then this option will be applied for VoIP traffic. Assigns the priority value of the Layer 2 QoS packets for Ethernet for VoIP Calls. The Default value is 0.

802.1x Mode

802.1x mode

Enables and selects the 802.1x mode for the phone system. The supported 802.1x modes are: EAP-MD5, EAP-TLS, EAP-PEAP

The default setting is “Disable”.

802.1x Identity

Enters the identity information for the selected 802.1x mode. (This setting will be displayed only if 802.1 X mode is enabled).

802.1x Secret

Enters the secret for the 802.1x mode. This option will appear when 802.1x mode is EAP-MD5 or EAP-PEAP.

CA Certificate

Uploads the CA Certificate file to the phone. (This setting will be displayed only if the 802.1 X mode is enabled)

Client Certificate

Loads the Client Certificate file to the phone. (This setting will be displayed only if the 802.1 X TLS mode is enabled)

Private Key

Loads the private key file to the phone. (This setting will be displayed only if the 802.1 X TLS mode is enabled)

Wi-Fi Basics

Wi-Fi Function

Enables/disables the Wi-Fi feature.

The default setting is “Disable”.

WiFi Band

Sets the type of WiFi Band (2.4G & 5G, 2.4G or 5G).

The default setting is 2.4G&5G.

ESSID

Allows to scan and select the available Wi-Fi networks within the range if the Wi-Fi feature is enabled. Click on “Select” to select the Wi-Fi network to connect to. The ESSID will be auto filled in the ESSID filed.

Wi-Fi Security

ESSID

Determines the ESSID of the selected Wi-Fi network. It can be auto filled by clicking the select button on the web page.

Password

Determines the password for the selected Wi-Fi network.

Security Mode for Hidden SSID

Defines the security mode used for the wireless network when the SSID is hidden. Default is “None”.

Advanced Settings

Layer 2 QoS 802.1p Priority Value (Wi-Fi)

Assigns the priority value of the Layer 2 QoS packets for Wi-Fi.

The Default value is 0.

Country Code

Configure Wi-Fi country code. The default value is “United States of America”. Note: Reboot is required to take effect.

OpenVPN® Client Key

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

OpenVPN® Cipher Method

The cipher method of OpenVPN®, must be the same cipher method used by the OpenVPN® server.

Available methods are: Blowfish, AES-128 or AES-256.

Default is “Blowfish”

OpenVPN® Username

OpenVPN® authentication username (optional).

OpenVPN® Password

OpenVPN® authentication username (optional).

Advanced Network Settings

DNS Refresh Time (m)

Configures the refresh time (in minutes) for DNS query. If set to “0”, the phone will use dns query TTL resolving from DNS server response.

DNS Failure Cache Duration (m)

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

Preferred DNS 1

Defines the preferred DNS server for the user.

Preferred DNS 2

Defines the second DNS server for the user.

Enable LLDP

Enables the LLDP (Link Layer Discovery Protocol) feature on the phone system. If it is set to “Yes”, the phone system will broadcast LLDP PDU to advertise its identity and capabilities and receive same from a physical adjacent layer 2 peer.

The default setting is “Yes”.

LLDP TX Interval (s)

Configures the interval the phone sends LLLD-MED packet.

The default setting is 30s.

Enable CDP

Configures whether to enable CDP to receive and/or transmit information from/to CDP-enabled devices. The default setting is “Yes”.

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 it is equivalent to the DSCP name constant AF31.

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 it is equivalent to the DSCP name constant EF.

Layer 3 QoS for Video

Defines the Layer 3 packet’s QoS parameters for H.264 messages in decimal pattern. This value is used for IP Precedence, Diff-Serv or MPLS.

The default setting is 34 it is equivalent to the DSCP name constant AF41.

HTTP/HTTPS

User-Agent

Sets the user-agent for phonebook and screen saver.

SIP User-Agent

Sets the user-agent for SIP. Default is “Grandstream GXV3380 $version”

“$version” is replaced by actual firmware version.

PC Port Mode

PC Port Mode

Enables and defines the PC port mode. If it is set to “Mirrored”, the traffic in the LAN port will go through PC port as well and packets can be captured by connecting a PC to the PC port. A reboot is required to take effect. The default setting is “Enabled”.

PC Port VLAN Tag

Defines the VLAN Identifier of the Layer 2 frame for PC port. This adds the VLAN tag value on the target address received from the LAN port of the phone then sends the value to the device connected to the PC port.

Note: VLAN tag value on the device connected to the PC port should be the same as the VLAN tag value assigned to the PC port here.

PC Port Priority Value

Determines the Priority Code Point within a Layer 2 frame header for PC port.

Proxy

HTTP/HTTPS Proxy Hostname

Configures the HTTP/HTTPS proxy URI of the network. Some of networks requires going through a proxy to access to the Internet. The default setting is keeping this field blank.

HTTP/HTTPS Proxy Port

Configures the HTTP/HTTPS proxy port number of the network. Some of networks requires going through a proxy to access to the Internet. The default setting is keeping this field blank.

Bypass Proxy For

Defines the specific URI that the phone can directly access to without HTTP/HTTPS proxy. If it is filled, the phone will bypass the proxy to send the packets to the specific URI. The default setting is filed blank.

Remote Control

CSTA Control

Indicates whether CSTA Control feature is enabled. Change of this configuration will need the system reboot to make it take effect.

Action URI Support

Configures whether the phone is enabled to receive and handle Action URI request.

Remote Control Pop up Window Support

Configures whether the phone is enabled to pop up allow remote control window. If set to “Yes”, when the remote console is connected to the phone, the phone will pop up a window to allow remote control or not. If set to “No”, once the remote console successfully connect s to the phone, it can directly control the phone remotely.

Action URI Allow IP List

List of allowed IP addresses from which the phone receives the Action URI. If input “any”, any remote console can access this phone.

Time Settings

Assign NTP Server Address

Defines the URL or IP address of the NTP server. The phone may obtain the current date and time information from the server. The default setting is “pool.ntp.org”.

DHCP Option 42 override NTP server

Obtains NTP server address from a DHCP server using DHCP Option 42; it will override configured NTP Server. If set to “No”, the phone will use configured NTP server to synchronize time and date even if a NTP server is provided by DHCP server. The default setting is “Yes”.

DHCP Option 2 to override Time Zone setting

Obtains time zone setting (offset) from a DHCP server using DHCP Option 2; it will override selected time zone.

If set to “No”, the phone will use selected time zone even if provided by DHCP server. The default setting is Yes.

Time Zone

Specifies the local time zone for the phone. It covers the global time zones and user can selected the specific one from the drop-down list.

Self-defined Time Zone

This allows the user to configure the customized time zone in POSIX format.

Users need to Select “Self-Defined Time Zone” from Time Zone list and configure the POSIX value in the Self-defined Time Zone field.

Time Display Format

Specifies which format will be used to display the time. It can be selected from 12-hour and 24-hour format.

Date Display Format

Determines which format will be used to display the date.

It can be selected from the drop-down list.

• Normal (YYYY/M/D): 2019/1/31
• YYYY/MM/DD: 2019/01/31
• MM/DD/YYYY: 01/31/2019
• DD/MM/YYYY: 31/01/2019

The default setting is “Normal (YYYY/M/D)”.

Language

Language

Sets the language to display on the phone’s LCD.

Select Language File

Press “Browse” to bring up a file selection menu to select the local .txt file to upload to the phone.

Web/SSH Access

Enable SSH

Allows SSH access to the phone.

The default setting is “Yes”.

SSH Port

Customizes SSH port to access to the phone. Default 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 ‘s Web UI. By default, if HTTP, the port number will be 80; if HTTPS is selected, the port number will be 443.

Web Access Control

Configures Web access control by using Whitelist or Blacklist on incoming IP addresses. Default setting is None.

Web Access Control List

Depending on the configuration set under Web Access Control Option, This setting only allow the list of IP addresses as Whitelist, or restrict the list of IP addresses as Blacklist to access Web.

Configuration via Keypad Menu

Configures access control for keypad Menu settings on the Settings interface of the phone.

• Unrestricted: configure all settings on the Settings interface;
• Basic Settings Only: The Advanced Settings option will not be displayed;
• Basic Settings & Network Settings: Only the Advanced Settings option will not be displayed
• Constraint Mode: users need to input admin user password to configure Wireless & Network and Advanced Settings.

Note: When access control for keypad is limited to “Basic Settings Only” or “Constraint Mode”, the Admin authentication will be mandatory to start Factory Reset process.

Permission to Install/Uninstall Apps

Configures the permissions for users to install/uninstall the applications.

• If set to “Allow“, the user is free to install/uninstall third-party apps.
• If set to “Require admin password“, the user need to input the correct administrator password to install/uninstall third-party apps.
• If set to “Require admin password if the app source is unknown“, the user need to input admin password only when install apps from unknown source, administrator password authentication is required when the user uninstall third-party apps.
• If set to “Not allow“, the user cannot install/uninstall third-party apps.

Default is “Allow”.

User Info Management

Current Admin Password

Enter current logged-in user’s password. This field is case sensitive. The default password is “admin”.

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 Admin Password

Enter the new Admin password again to confirm.

New User Password

Allows the administrator to set the password for user-level web GUI access. This field is case sensitive with a maximum length of 32 characters. The default password is “123”.

Confirm New User Password

Enter the new User password again to confirm.

TLS

Minimum TLS Version

Configures the minimum TLS version supported by the phone.

Maximum TLS Version

Configures the maximum TLS version supported by the phone.

Enable Weak TLS Cipher Suite

Defines the function for weak TLS cipher suites. If set to “Enable Weak TLS Cipher Suites”, allow users to encrypt data by weak TLS cipher suites. If set to “Disable Symmetric Encryption RC4/DES/3DES”, allow users to disable weak cipher DES/3DES and RC4.

SIP TLS

SIP TLS Certificate

Defines the SSL certificate used for SIP over TLS.

SIP TLS Private Key

Defines the SSL Private key used for SIP over TLS.

SIP TLS Private Key Password

Defines the SSL Private key password used for SIP over TLS.

Certificate Management

CA Certificate

Import Trusted CA Certificates

Allows to upload the CA Certificate file to phone.

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.

Custom Certificate

Import Custom Certificate

Allows to upload a Custom Certificate file to phone.

Custom Certificate

Lists trusted Custom Certificate previously uploaded. Administrator can delete a certificate from here.

LCD & LED Management

Enable Missed Call Backlight

If set to “Yes”, LCD backlight will be turned on when there is a missed call on the phone. The default setting is “Yes”.

Enable Missed Call Indicator

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when there is missed call on the phone. The default setting is “Yes”.

Enable MWI Indicator

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when there is new voicemail on the phone. The default setting is “Yes”.

Enable New Message Indicator

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when there is new message on the phone. The default setting is “Yes”.

Enable Contact Full Indicator

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when the contact storage or message storage is full. The default setting is “Yes”.

Enable Indicator When LCD is Off

If set to “Yes”, the LED indicator on the upper right corner of the phone will light up when the LCD screen is off. The default setting is “Yes”.

Peripherals Interface Management

HDMI Output Control

Defines whether to enable HDMI. If enabled, users need to set whether HDMI would synchronize with LCD.

• If set to “HDMI Display Synchronization with LCD“, HDMI device will display the same content as the phone LCD;
• If set to “Show Opposite Screen“, HDMI device displays remote video screen in a video call while other screens display synchronization with LCD. This setting will take effect in the next call. Default is “Show Opposite Screen”.

Audio Control

Echo Delay

Configures the device’s HDMI audio delay to match the audio latency of different TV sets.

RJ9 Headset TX Gain (dB)

Configures the Transmission Gain in RJ9 headset channel. It can be selected from the dropdown list. The default setting is 0dB:

• -24
• -18
• -12
• -6
• 0
• +6
• +12
• +18
• +24

3.5mm Earphone TX Gain (dB)

Set the Transmission Gain in 3.5mm earphone headset channel. It can be selected from the dropdown list. The default setting is 0dB:

• -24
• -18
• -12
• -6
• 0
• +6
• +12
• +18
• +24

Headset Type

Specifies which type of headset will be connected to the phone system. It can be selected from the dropdown list:

• Normal Headset
• Plantronics EHS

If a normal RJ11 headset is connected, it should set to “Normal Headset”. If a Plantronics EHS headset is used, it should set to “Plantronics EHS”.

Note: Reboot is required to take effect.

Enable 3.5mm Headset Control

If set to “Yes”, the headset can control the Onhook and Offhook. The default setting is “No”.

Handset TX Gain (dB)

Configures the transmission gain of the handset.

• 0
• -6
• +6

Default setting is “-6dB”.

Note: Reboot is required to take effect.

Handfree TX Gain (dB)

Configures the transmission gain for handsfree mode.

Note: Reboot is required to take effect.

Adjust Volume

Sets the phone’s volume.

Media volume

Configures the volume of media.

Alarm volume

Configures the volume of Alarm.

Enable TR-069

Sets the phone system to enable the “CPE WAN Management Protocol” (TR-069). The default setting is “Yes”.

ACS URL

Specifies URL of TR-069 ACS (e.g., http://acs.mycompany.com), or IP address. Default setting is “https://acs.gdms.cloud”

ACS User Name

Enters user name to authenticate to ACS.

ACS Password

Enters password to authenticate to ACS.

Periodic Inform Enable

Sends periodic inform packets to ACS. Default is “Yes”.

Periodic Inform Interval (s)

Configures to sends periodic “Inform” packets to ACS based on specified interval. Default setting is 86400.

Connection Request User Name

Enters user name for the ACS to connect to the phone.

Connection Request Password

Enters password for the ACS to connect to the phone.

Connection Request Port

Enters the port for the ACS to connect to the phone.

CPE Cert File

Uploads Cert File for the phone to connect to the ACS via SSL.

CPE Cert Key

Uploads Cert Key for the phone to connect to the ACS via SSL.

Firmware

Upgrade via Manually Upload

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.

Upgrade via Network

Firmware Upgrade Mode

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

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”.

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, update immediately.

Config File

Configure Manually

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 to the phone.

Notes:

•The GXV3380 supports only txt format for config file upload.

•Reboot is required to take effect.

Configure via Network

Use Grandstream GAPS

It is used to configure the download path and update mode for the configuration file server.

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

Config Update Via

Selects provisioning method: TFTP, HTTP or HTTPS. Default setting is “HTTPS”.

Config Server Path

Sets IP address or domain name of configuration server. The server hosts a copy of the configuration file to be installed on the phone.

Users can also configure variables in the provisioning server URL; The function allows users to configure a unified provisioning server URL for different phone models. Currently, the following variables are supported in the provisioning server URL:

• $PN: it is used to identify the product model name of the IP phone.
• $MAC: it is used to identify the MAC address of the IP phone.

Variables $PN and $MAC can be embedded in server URL setting in Web UI and also in DHCP Option 66.

-Example (Web UI): /192.168.0.2/$PN/$MAC

-Example (DHCP Option 66): tftp://192.168.0.2/$PN/$MAC

$PN will be replaced with phone model, e.g., GXV3380. And $MAC will be replaced with phone’s MAC address, e.g., 000b829a8ffe.

Default settings is “fm.grandstream.com/gs”.

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.

Always send HTTP Basic Authentication Information

Includes configured user name and password in HTTP request before receiving authentication challenge from the server.

Default is “No”.

Config File Prefix

Checks if configuration files are with matching prefix before downloading them.

This field enables user to store different configuration files in one directory on the provisioning server.

Config File Postfix

Checks if configuration files are with matching postfix before downloading them.

This field enables user to store different configuration files in one directory on the provisioning server.

Authenticate Conf File

Sets the phone system to authenticate configuration file before applying it. When set to “Yes”, the configuration file must include value P1 with phone system’s administration password. If it is missed or does not match the password, the phone system 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.

CUST File

GUI Customization File Download Mode

Selects download method: TFTP, HTTP or HTTPS.

Default setting is “HTTP”.

GUI Customization File URL

Sets IP address or domain name of the GUI customization file server. The server hosts a copy of the file to be installed on the phone. The Default setting is fm.grandstream.com/gs.

GUI Customization File HTTP/HTTPS User Name

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

GUI Customization File HTTP/HTTPS Password

Enters the password for the firmware HTTP/HTTPS server.

Use Configurations of Config File Server

Retrieve and download customization file with the configuration of the config file.

Provision

Automatic Upgrade

Automatic Upgrade

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

• No: The phone will only do upgrade once at boot up.
• Check every day: User needs to specify “Hour of the day (0-23)”.
• Check every week: User needs to specify “Hour of the day (0-23)” and “Day of the week (0-6)”.
• Check at a period Time: User needs to specify “Hour of the day (0-23)”

Note: Day of week is starting from Sunday.

The default setting is “No”.

Period of Time

Configures how the phone system will check the server for new firmware and configuration file downloading.

It only valid if the user selects “Check at a period of time” in the “Automatic Upgrade”.

Default setting is 10080 (namely 7 days).

Starting – Ending Hour of the Day (0-23)

Defines at which hour of the day they phone system will check the HTTP/HTTPS/TFTP server for firmware upgrades or configuration files changes.

Day of the Week

Defines which day of the week the phone system will check the HTTP/HTTPS/TFTP server for firmware upgrades or configuration files changes.

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent.

• Always Check at bootup.
• When Firmware/Config File Pre/Suffix Changes.
• Skip the Firmware Check.

The default setting is “Always Check at bootup”.

Upgrade With Prompt

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

DHCP Option

Allow DHCP Option 43, 160 and 66 to Override Server

Obtains configuration and upgrade server’s information from DHCP server using options 43, 160 and 66.

If DHCP option 43, 160 and 66 is enabled on the LAN side, the device will reset the CPE, upgrade, network VLAN tag, and priority configuration according to option 43 sent by the server. At the same time, the update mode and server path of the configuration upgrade mode will be reset according to the option 160 and 66 sent by the server.

Default setting is “Yes”.

DHCP Option 120 Override SIP Server

Configures the phone system to allow the DHCP offer message to override the Config Server Path via the Option 120 header. 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, cfg.xml, cfgMODEL.xml ,cfgMAC.xml. Default is “No”.

Config Provision

Device will download configuration files and provision by configured order.

Syslog

Syslog Protocol

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

• UDP: Syslog messages will be sent over UDP.
• SSL/TLS: Syslog messages will be sent securely over TLS connection. To upload server CA certificate, follow below steps:
• Copy CA file in SD card and plug it to the phone.
• Go to LCD menu Settings🡪Security Settings🡪Install from SD card to install the CA file.

Syslog Server

Configures the URI which the phone system will send the syslog messages to. The default setting is “log.ipvideotalk.com”.

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:

• DEBUG (Sent or received SIP messages).
• INFO (Product model/version on boot up, NAT related info, SIP message summary, Inbound and outbound calls, Registration status change, negotiated codec, Ethernet link up).
• WARNING (SLIC chip exception).
• ERROR (SLIC chip exception, Memory exception).

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

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 system.

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:

• Verbose (Default Setting)
• Debug
• Info
• Warning
• Error
• Fatal
• Silent (suppress all output)

Get Log

Displays the log file on the web page.

Debug

One-click Debugging

One-click Debugging

Capture 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

Display a list of info items that can be debugged, currently supports system logs, info log, capture package, tombstones and ANR log. The captured data can be viewed in “Debug information list”. The default is all selected.

Debug Info List

You can select the existing debugging info package or grab package. Click the “Delete” button on the right to delete the file.

View Debug Info 

You can select the existing debugging info package or grab package. Click the “Delete” button on the right to delete the file.

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. Click the “Delete” button on the right to delete this file.

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. Note: The audio data file will be saved under FileManager 🡪 Internal Storage 🡪 Recfiles folder. Users can also delete files under this folder.

Traceroute

Target Host

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

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

Press Start to send ping request to configured target host.

Press Stop to end ping running process.

NSLookup

Hostname

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.

Bootup Completed

Configures the event URL when phone boots up.

Incoming Call

Configures the event URL when phone has an incoming call.

Outgoing Call

Configures the event URL when phone has an outgoing call.

Offhook

Configures the event URL when the phone is off-hook.

Onhook

Configures the event URL when the phone is on-hook.

Missed Call

Configures the event URL when the phone has new a missed call.

Connected

Configures the event URL when a call is established.

Disconnected

Configures the event URL when a call is disconnected.

DND On

Configures the event URL when DND is enabled.

DND Off

Configures the event URL when DND is disabled.

Forward On

Configures the event URL when the forward feature is enabled on the phone.

Forward Off

Configures the event URL when the forward feature is disabled on the phone.

Blind Transfer

Configures the event URL when users transfer a call with blind transfer on the phone.

Attended Transfer

Configures the event URL when users transfer a call with attended transfer on the phone.

On Hold

Configures the event URL when users hold a call on the phone.

UnHold

Configures the event URL when users resume a call on the phone.

Log On

Configures the event URL when users log on the phone successfully.

Log Off

Configures the event URL when users log off the phone.

Register

Configures the event URL when an account in the phone is registered successfully.

Unregister

Configures the event URL when an account in the phone is unregistered.

Programmable Key

Format

Display Format

Configures the display format for the MPK. Users could select “Name”, “User ID” or “Name(User ID)”. “Name” is the one saved in phone contacts. The default setting is “Name, User ID, Key mode”.

Show Display Name from Server

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

BLF

Key Mode

The key modes are:

• Speed Dial: Press to dial the UserID when the accounts being configured as VPK.
• Busy Lamp Field: Monitor the UserID status when the accounts being configured as VPK.
• Call Transfer: Transfer the current active call to UserID when the accounts being configured as VPK.
• Call Intercom: Intercom/paging to the UserID when the accounts being configured as VPK.
• Speed Dial via Active Account: Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is offhook and account 2 is active, it will call the UserID when the accounts being configured as VPK.
• Dial DTMF: Dial the DTMF digits of the UserID when the accounts being configured as VPK during the call.
• Call Park: Configure the call park feature code to park or retrieve the call.
• Multicast Paging: For multicast sending, please fill in the display name in the Settings and fill in the sending address in the multicast address.
• Speed Conference: Quickly dial up multiple numbers to set up a meeting.
• Dial Prefix: After configured, once pressed this key, all numbers use this account will automatically add the prefix promptly

Account

Configures the SIP account when the accounts being configured as VPK.

Display Name

Configures the display name when the accounts being configured as VPK.

User ID

Configures the UserID for the corresponding VPK mode when the accounts being configured as VPK.

DTMF Content

When key mode is set to Dial DTMF it Configures the dialed DTMF content.

Address

When key mode is set to Multicast Paging, it configures the multiple broadcast address.

Conference Name

When key mode is set to Speed Conference, it sets the speed conference name.

Mute all members

When key mode is set to Speed Conference, enables/disables mute all meeting members.

Configure

When key mode is set to Speed Conference, it configures Speed Conference members list:

• Number: Enter member’s number.
• Name: Enters member’s name.
• Account: Specified the account associated.
• Operation: + to add a member, – to delete a member.

Programmable Key General Settings

General Settings

Enable LCD Turn On Automatically when BLF/SCA status changes

Configures whether the LCD should turn on automatically when BLF/SCA status changes.

Proxy Prefix

Account

Displays account name if configured, otherwise, displays “Account X” where X is the account number.

BLF Call-pick Prefix

Configures the prefix prepended to the BLF extension if the phone answers a call to the monitored party by the BLF key. Default setting is ** for each account.

Event List URI

Determines the event list BLF URI on the phone to monitor the extensions in the list with MPK keys. This feature is based on BroadSoft standard. It requires filling in the BLF ID to the box.

For example, if the server provides the URI: BLF123@myserver.com, this field should be filled with BLF123. Then the monitored extensions will be populated in the MPK app or Extension Board (if supported).

Force BLF Call-pickup by Prefix

Uses the prefix for BLF Call-pickup. The default setting is “No”.

General Settings

Sort Phonebook by

Sets which part of name, first name or last name, will be sorted in alphabetical order to display.

Default Contacts Tab

Controls the behaviors of the phonebook key. It could be set to:

• Default
• LDAP Search
• Local Phonebook
• Local Group
• Broadsoft Phonebook
• Favorites.

The default setting is “Default”, which set the phonebook key to the Contacts menu.

Emergency Call Numbers

Configures the emergency contact in logout mode. If the system is logout, guest users can dial the configured emergency contacts.

Input the number in the input box and click “Add” to add the number to the contacts list. To delete the existing ICE number, select the number in the contacts list and click “Delete”.

Import/Export Contacts

Import

Clear The Old List

Determines if the phone system will delete the previous contacts when a new contact file is imported. If set to “Yes”, the previous contacts will be removed. The default setting is “No”.

Clear Old History Mode

• If set to “Clear all”, the phone will delete all previous records before importing the new records.
• If set to “Keep Local Contacts”, the new-added local new contacts will not be deleted when importing new records.

Replace Duplicate Items

Configures the phone system to keep the original contact entries when duplicated contact entries are included in the contact file. If set to “Yes”, the phone will replace the original entries to the new one. Otherwise, the phone system will save both contact entries. The default setting is “No”.

Replace Duplicate Entries Mode

• If set to “Replace by name”, replace the records of the same name automatically when importing new records.
• If set to “Replace by number”, replace the records of the same number automatically when importing new records.

File Encoding

Specifies the encoding format for phonebook file importing. The default setting is UTF-8. It can be selected from the dropdown list:

• UTF-8
• GBK
• UTF-16
• UTF-32
• Big5
• Big5-HKSCS
• Shift-JIS
• ISO8859-1
• ISO8859-15
• Windows-1251
• EUC-KR

File Type

Sets the type format for phonebook file importing. It can be selected from the dropdown list.

• XML
• vCard

The default setting is “XML”.

Import Local File

Uploads the contact file from PC to the phone.

Export

File Encoding

Specifies the encoding format for phonebook file exporting. The default setting is UTF-8. It can be selected from the dropdown list:

• UTF-8
• GBK
• UTF-16
• UTF-32
• Big5
• Big5-HKSCS
• Shift-JIS
• ISO8859-1
• ISO8859-15
• Windows-1251
• EUC-KR

File Type

Sets the type format for phonebook file importing. It can be selected from the dropdown list.

• XML
• VCard

The default setting is “XML”.

Export

Downloads the phonebook file from the phone to PC.

Download Contacts

Clear The Old List

Sets the phone system to delete the previous contacts when a new contact file is downloaded.

If “Yes”, the previous contacts will be removed.

The default setting is “No”.

Clear Old History Mode

• If set to “Clear all”, the phone will delete all previous records before downloading the new records.
• If set to “Keep Local Contacts”, the new-added local new contacts will not be deleted when downloading new records.

Replace Duplicate Items

Keeps the original contact entries when duplicated contact entries are included in the contact file. If set to “Yes”, the phone will replace the original entries to the new one. Otherwise, the phone system will save both contact entries. The default setting is “Yes”.

Replace Duplicate Entries Mode

• If set to “Replace by name”, replace the records of the same name automatically when importing new records.
• If set to “Replace by number”, replace the records of the same number automatically when importing new records.

Download Mode

Enables the phone system to download phonebook file and select the server and protocol to download the phonebook file. It can be selected from TFTP, HTTP, and HTTPS.

The default setting is “OFF”.

File Encoding

Selects the encoding format for phonebook file download. The default setting is UTF-8. It can be selected from the dropdown list:

• UTF-8
• GBK
• UTF-16
• UTF-32
• Big5
• Big5-HKSCS
• Shift-JIS
• ISO8859-1
• ISO8859-15
• Windows-1251
• EUC-KR

Download Server

Configures the server URL to download the phonebook file.

The phone system will send a request to the server to download the phonebook file with filename phonebook.xml.

HTTP/HTTPS User Name

Configures user name for HTTP/HTTPS server to download the phonebook file.

HTTP/HTTPS Password

Specifies password for HTTP/HTTPS server to download phonebook file.

Automatic Download Interval

Determines how the phone system to send the request to the server to download the phonebook file. It can be selected from the dropdown list:

• None
• 5 Minutes
• 30 Minutes
• 1 Hour
• 2 Hour
• 4 Hour
• 6 Hour
• 8 Hour
• 12 Hour

Download Now

Starts downloading the XML phonebook to the phone immediately.

Connection Mode

Selects which protocol will be used for LDAP searching, LDAP or LDAPS.

Default is “LDAP”.

Server Address

Configures the URI of the LDAP (Lightweight Directory Access Protocol) server.

Port

Configures the LDAP server port. The default LDAP port number is 389.

Base DN

Determines the LDAP search base. This is the location in the directory where the search is requested to begin.

Example:

dc=grandstream, dc=com

ou=Boston, dc=grandstream, dc=com

User Name

Configures the bind “Username” for querying LDAP servers. Some LDAP servers allow anonymous binds in which case the setting can be left blank.

Password

Specifies the bind “Password” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

LDAP Name Attributes

Configures the “name” attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated name attributes.

Example:

cn sn description

LDAP Number Attributes

Configures the “number” attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated number attributes.

Example:

telephoneNumber

telephoneNumber Mobile

LDAP Mail Attributes

Determines the “mail” attributes of each record which are returned in the LDAP search result.

Example: mail

LDAP Name Filter

Configures the filter used for name lookups.

Examples:

(|(cn=%)(sn=%)) returns all records which has the “cn” or “sn” field starting with the entered prefix;

(!(sn=%)) returns all the records which do not have the “sn” field starting

with the entered prefix;

(&(cn=%) (telephoneNumber=*)) returns all the records with the “cn” field starting with the entered prefix and “telephoneNumber” field set.

LDAP Number Filter

Defines the filter used for number lookups.

Examples:

(|(telephoneNumber=%)(Mobile=%) returns all records which has the “telephoneNumber” or “Mobile” field starting with the entered prefix;

(&(telephoneNumber=%) (cn=*)) returns all the records with the “telephoneNumber” field starting with the entered prefix and “cn” field set.

LDAP Mail Filter

Determines the filter used for mail lookups. Example: (mail=%)

Search Field Filter

Configures to filter according to which fields when searching on LDAP. Users can choose between ‘Name Filter’, ‘Number Filter’, ‘Mail Filter’ or ‘All Filter’. The default setting is “All Filter”.

LDAP Displaying Name Attributes

Configures the entry information to be shown on phone’s LCD. Up to 3 fields can be displayed. Example: %cn %sn %telephoneNumber

Max Hits

Specifies the maximum number of results to be returned by the LDAP server. If set to 0, server will return all search results. Default setting is 50.

Search Timeout (s)

Configures the interval (in seconds) for the server to process the request and client waits for server to return. The default setting is 4 seconds.

LDAP Lookup For Dial

Sets the phone system to do the LDAP number searching when making outgoing calls. The default setting is “No”.

LDAP Lookup For Incoming Call

Sets the phone system to do LDAP number searching for incoming calls. The default setting is “No”.

LDAP Dialing Default Account

Configures the default account that being used when dialing LDAP contact. Users may choose the Account 1-6, the default setting is “Default”.

Call Recording

File name

Displays the name of the recording file

Duration

Displays the duration of the phone call

Date

Displays the date the call was recorded on

Operation

Delete, Modify or download the recording file

Normal Recording

File name

Displays the name of the recording file

Duration

Displays the duration of the phone call

Date

Displays the date the call was recorded on

Operation

Delete, Modify or download the recording file

Value-added Service (0/10)

Service Type

Users can set the service type to “Door System” to configure the door system options, or to “DTMF” to set DTMF content to send it during calls.

Door System Type

Set the door system type to “GDS” if the GDS door system is used, or set it to “Baudisch” if another door system brand is used.

Note: Each GDS door system has 2 different access passwords to control 2 doors separately named as [Related Display Name1] & [Related Display Name2] below for door 1 and 2 respectively.

System Number

Specifies the door system number which is the SIP user ID configured on door system or its IP address, if the door system is using IP call. It enables to show open door button when caller number or IP address matches with this setting. e.g:”36311″ or “192.168.124.81”.

Note: When set “Door System Type” to “Baudisch” a “configure” button will appear to allow user to configure groups of door system URL and User ID for 100 entries.

• System Number: This is used to configure the User ID of door system. Once configured, only the call from this User ID would use door system while other calls use the default mode.
• System Address: This is used to input the IP address or URL of the system in order to identify the call from door system.
  Users can set HTTP authentication credentials on the URL for Door Systems that require authentication to send HTTP stream. The URL format will be similar to the following:
  http://username:password@192.168.1.150/goform/stream?cmd=get&channel=4

Display Name

Configures the display name of the door system. When the call matches the configured system number, the name will be displayed on LCD.

Related Display Name1

Configures the name that will be displayed on LCD for door 1 when the call matches the configured GDS door system number.

Access password

Determines the door system password which should match the one configured on the used door system settings. In case the GDS is set as ‘Door System type’ parameter, the password should match the one configured on the GDS to open door 1.

Related Display Name2

Indicates the name that will be displayed on LCD for door 2 when the call matches the configured system number.

Access password

The configured password should match the one configured on the GDS to open door 2.

System Ringtone

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

DTMF Content

Set the DTMF content that is going to be sent when the DTMF button is pressed under Call screen🡪“Keypad”.

General Settings

Display Open Door Button when Calling

Configures whether display Open Door button when there is an incoming call. If set to “Yes”, you cannot open door with DTMF when preview function is disabled, the phone will hide open door button automatically.

Enable Preview

Configures whether to enable preview function or not. If set to “No”, you cannot open door with DTMF when there is an incoming call, the phone will hide open door button automatically.

Broadsoft Call Features

Feature Key Synchronization

Synchronizes the BroadSoft standard call feature. If it is enabled, the phone will send SIP SUBSCRIBE message to the server and receive SIP NOTIFY message from the server to synchronize the DND, Call Forwarding and Call Center features. The default setting is “Disable”.

Enable BroadSoft Call Park

Configures whether to send SUBSRCIRBE message to BroadSoft server to obtain Call Park notifications.

The default setting is “No”.

Conference URI

Configures the network-based conference URI (the BroadSoft Standard). If it is configured, end user needs to tap the N-way key during the conference to transfer the host to the remote media server.

Broadsoft Call Center

When enabled, Feature Key Synchronization will be enabled regardless of web settings.

Default is Disabled.

Hoteling Event

Enables BroadSoft Hoteling Event feature. Default is Disabled.

Call Center Status

When set to “Yes”, the phone will send SUBSCRIBE to the server to obtain call center status. Default is Disabled.

SCA

Enable SCA (Shared Call Appearance)

Enables/disables the Shared Call Appearance (the Broadsoft Standard) feature for this account. If it is set to “Yes”, the phone system can update and share account status with another device.

The default setting is “No”.

Enable Barge-in

Enables/disables the Barge-In feature. If it is set to “Yes”, the user could tap the SCA account to barge into an active session with another shared line. The default setting is “No”.

Auto-filling CallPark Feature Code

Sets if the phone system will add the configured feature code to park calls. If it is set to “Yes”, the “CallPark Feature Code” configured will be automatically prepended on the phone’s dial pad when parking calls. The default setting is “Yes”.

CallPark Feature Code

Configures the callpark feature code. If “Auto-filling CallPark Feature Code” is set to “Yes”, this feature code will be automatically filled in on the dial pad when parking a call.

Line Seize Timeout (s)

Defines line-seize expiration timer. For Shared Call Appearance, the phone must send a SUBSCRIBE request for the line-seize event package whenever a user attempts to take the shared line off hook. The default value is 15 seconds. The valid range is from 15 to 60.

XSI Service Settings

Authentication Type

Defines the authentication type in way of login or use SIP. If set to “Login Credentials”, please fill in Username and Password in the following options; if set to “SIP Credentials”, please fill in Username, User ID, and Password. The default setting is “SIP Credentials”.

Server

Configures the BroadWorks Xsi server URI. If the server uses HTTPS, please add the header “HTTPS” ahead of the Server URI. For instance, “https://SERVER_URI”.

Port

Configures the BroadWorks Xsi server port. The default port is 80. If the server uses HTTPS, set to 443.

Action Path

Configures Action path for BroadSoft XSI server.

BroadWorks User ID

Determines the User ID for Broadsoft XSI server.

SIP Authentication ID

Determines username for Broadsoft XSI server.

SIP Authentication Password

Determines password for Broadsoft XSI server.

Login Password

Specifies the password for the BroadWorks XSI query.

BroadSoft Directory & Call Logs Update Interval (s)

Configures the interval to obtain BroadSoft call log and directory data thus to update the matching number data on the dialing interface. Default is 1800 seconds.

BroadSoft Directory Hits

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

Associated BroadSoft Account

Configures the associated BroadSoft account when dialing BroadSoft contacts.

BroadSoft Directory Order

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

Network Directories

Group Directory

Enables/disables the BroadWorks XSI Group Directory features on the phone. The name filed is used to name the directory. If it keeps blank, the phone system will use the default name “Group” for it

Enterprise Directory

Enables/disables the BroadWorks XSI Enterprise Directory features on the phone. The name filed is used to name the directory. If it keeps blank, the phone system will use the default name “Enterprise” for it.

Group Common

Enables/disables the BroadWorks XSI Group Common features on the phone. The name filed is used to name the directory. If it keeps blank, the phone system will use the default name “Group Common” for it.

Enterprise Common

Enables/disables the BroadWorks XSI Enterprise Common features on the phone. The name filed is used to name the directory. If it keeps blank, the phone system will use the default name “Enterprise Common” for it.

Personal Directory

Enables/disables the BroadWorks XSI Personal Directory features on the phone. The name filed is used to name the directory. If it keeps blank, the phone system will use the default name “Personal Directory” for it.

Polycom Phonebook

Enables/disables the BroadWorks XSI Polycom Phonebook features on the phone. The name filed is used to name the directory. If it keeps blank, the phone system will use the default name “Polycom Phonebook” for it

Missed Call Log

Enables/disables the BroadWorks XSI Missed Call Log features on the phone. The name filed is used to name the directory. If it keeps blank, the phone system will use the default name “Missed” for it.

Placed Call Log

Enables/disables the BroadWorks XSI Placed Call Log features on the phone. The name filed used to name the directory. If it keeps blank, the phone system will use the default name “Outgoing” for it.

Received Call Log

Enables/disables the BroadWorks XSI Received Call Log features on the phone. The name filed used to name the directory. If it keeps blank, the phone system will use the default name “Incoming” for it.

Login Credentials

Server

BroadSoft IM&P server address. Usually it’s not necessary to configure and it can already be found in the BroadSoft IM&P user name

Port

Specifies the BroadSoft IM&P server port. The default port is 5222.

Username

Determines the Username for the BroadSoft IM&P query. This is usually not the same as BroadSoft account user name.

Password

Determines the password for the BroadSoft IM&P query. This is usually not the same as BroadSoft account password.

IM&P Settings

Enable BroadSoft IM&P

Enables the BroadSoft XMPP feature.

Associated BroadSoft Account

Binds the SIP account for dialing the XMPP client contacts.

Auto Login

Sets the client to be logged automatically if the application is started.

Display Non XMPP Contacts

Determines if the client app will show the non-XMPP contacts.

EUT Feature

Tx/Rx Frequency Range

2402~2480 MHz

2402~2480 MHz

2412~2472 MHz

5150~5250 MHz

5250~5350 MHz

5470~5725 MHz

Number of Channels

79

40 (37 hopping + 3 advertising channels)

13

UNII Band I:

802.11a/n-HT20-VHT20: 4 channels

802.11n/ ac -HT40-VHT40: 2 channels

802.11ac-VHT80:1 Channel

UNII Band II:

802.11a/n-HT20-VHT20: 4 channels

802.11n/ ac –HT40–VHT40: 2 channels

802.11ac-VHT80:1 Channel

UNII Band III:

802.11a/n-HT20-VHT20: 11 channels

802.11n/ac-HT40-VHT40: 5 channels

802.11ac-VHT80:2 Channels

Carrier Frequency of Each Channel

f=2402+k MHz

(k=0,1,2….,78)

f=2402+k MHz

(k=0,2,4…,39)

–

–

Antenna Type/Gain

Internal PCB Antenna /

gain 4.0 dBi

Internal PCB Antenna / gain 4.0 dBi

Internal PCB Antenna /

gain 4.0 dBi

Internal PCB Antenna /

gain 5.0 dBi

Type of Modulation

Bluetooth BR 1Mbps: GFSK

Bluetooth EDR 2Mbps: π/4-DQPSK

Bluetooth EDR 3Mbps: 8DPSK

Bluetooth LE: GFSK

802.11b: DSSS (DBPSK / DQPSK / CCK)

802.11g/n: OFDM (BPSK / QPSK / 16QAM / 64 QAM)

802.11a/n/ac: OFDM (BPSK / QPSK / 16QAM / 64QAM)

Operation temperature

0 °C ~ +40 °C

Storage temperature

-10 °C ~ +60 °C

Humidity

10 ~ 90% non-condensing

Domestic use

Industrial use Class B

WLAN

Standard

Frequency

EIRP Power (dBm)

Wi-Fi 2.4G

2.4~2.4835GHZ

18.70

Wi-Fi 5G

5.15~5.25GHz

22.06