DP755 - Administration Guide

Air Interface

Telephony standards: DECT

Frequency bands:

Number of channels: 10 (Europe), 5 (US, Brazil or Japan), 3 (Korea), 8 (Taiwan)

Range: up to 400 meters outdoor and 50 meters indoor
 

Peripherals

2.4 inch (240×320) color TFT LCD

27 keys including 3 soft keys, 5 navigation/ menu keys, 4 dedicated function keys for SEND, POWER/END, SPEAKERPHONE, MUTE, 3 side keys including 2 volume (up and down) and 1 Push-to-Talk key

3-color MWI LED

3.5mm headset jack

Proximity and accelerometer sensors

Backlit keypad

Removable belt clip

Micro-USB port for alternative charging and non-battery operation

Protocols/Standards

Hearing Aid Compatibility (HAC) compliant

Voice Codecs

G.722 codec for HD audio and G.726 codec for narrow band audio (G.711μ/a-law, G.723.1, G.729A/B, iLBC and OPUS are supported via companion DECT base station DP755), AEC, AGC, Ambient noise reduction on microphone capturing and advanced noise suppression on audio playing

Telephony Features

Hold, transfer, forward, 3-way conference, push-to-talk, intercom, call park, call pickup, downloadable phonebook, call waiting, call log, auto answer, click-to-dial, flexible dial plan

HD Audio

Yes, in both Handsets and Speakerphone modes

Security

DECT authentication & encryption

Multi-language

Chinese Simple, Chinese Tradition, Czech, Danish, Dutch, English, Estonian, Finnish, French, German, Hebrew, Hungarian, Japanese, Korean, Norwegian, Polish, Portuguese, Romanian, Spanish, Turkish.

Upgrade/ Provisioning

Software Upgrade Over-The-Air (SUOTA), handsets provisioning Over-The-Air

Multiple Line Access

Each handset may access up to 10 lines

Power & Green Energy Efficiency

Universal Power Supply Input AC 100-240V 50/60Hz; Output 5VDC 1A; Micro-USB connection;

Rechargeable Li-ion battery (500 hours of standby time and 40 hours of talk time)

Package Content

Handset unit, universal power supply, charger cradle, belt clip, 1 battery, Quick Installation Guide

Dimensions (H x W x D)

Handset: 168.5 x 52.5 x 21.8mm;

Charger cradle: 76 x 73 x 81mm

Weight

Handset: 180g;

Charger cradle: 78g;

Universal power supply: 50g;

Package: 465g

Temperature and Humidity

Operation: -10º to 50ºC (14 to 122ºF); Charging: 0 to 45ºC (32 to 113ºF); Storage: -20º to 60ºC (-4 to 140ºF); Humidity: 10% to 90% non-condensing

Compliance

FCC: FCC Part 15B; FCC Part 15D; SAR (FCC 47 CFR Part2.1093; IEEE 1528; IEC 62209-2); FCC Part68 HAC; FCC ID

CE：EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 60950-1;EN 301 489-1/-6; EN 301 406; EN 50332-2; SAR(EN50360;EN50566;EN 50663;EN62209-1; EN62209-2; EN 62479); RED NB Cert

RCM：AS/NZS CISPR32; AS/NZS 60950.1;AS/CA S004;AS/ACIF S040. ANATEL, EAC, UL(adapter).

Air Interface

Telephony standards: DECT

Frequency bands:

Number of channels: 10 (Europe), 5 (US, Brazil or Japan), 3 (Korea), 8 (Taiwan)

Range: up to 350 meters outdoor and 50 meters indoor
 

Peripherals

1.8 inch (128×160) color TFT LCD

23 keys including 2 softkeys, 5 navigation / menu keys, 4 dedicated function keys for SEND, POWER/END, SPEAKERPHONE, MUTE

3-color MWI LED

3.5mm headset jack

Removable belt clip

Micro-USB port for alternative charging and non-battery operation

Protocols/Standards

Hearing Aid Compatibility (HAC) compliant

Voice Codecs

Telephony Features

Hold, transfer, forward, 3-way conference, push-to-talk, intercom, call park, call pickup, downloadable phonebook, call waiting, call log, auto answer, click-to-dial, flexible dial plan

HD Audio

Yes, in both Handsets and Speakerphone modes

Security

DECT authentication & encryption

Multi-language

Chinese Simple, Chinese Tradition, Czech, Danish, Dutch, English, Estonian, Finnish, French, German, Hebrew, Hungarian, Japanese, Korean, Norwegian, Polish, Portuguese, Romanian, Spanish, Turkish.

Upgrade/ Provisioning

Software Upgrade Over-The-Air (SUOTA), handsets provisioning Over-The-Air

Multiple Line Access

Each handset may access up to 10 lines

Power & Green Energy Efficiency

Universal Power Supply Input AC 100-240V 50/60Hz; Output 5VDC 1A; Micro-USB connection;

Rechargeable Li-ion battery (500 hours of standby time and 40 hours of talk time)

Package Content

Handset unit, universal power supply, charger cradle, belt clip, 1 battery, Quick Installation Guide

Dimensions (H x W x D)

Handset: 158 x 50 x 28.1mm;

Charger cradle: 81.15 x 75.89 x 36.36mm

Weight

Handset: 110g;

Charger cradle: 44g;

Universal power supply: 50g;

Package: 328g

Temperature and Humidity

Operation: -10º to 50ºC (14 to 122ºF); Charging: 0 to 45ºC (32 to 113ºF); Storage: -20º to 60ºC (-4 to 140ºF); Humidity: 10% to 90% non-condensing

Compliance

FCC: FCC Part 15B; FCC Part 15D; SAR (FCC 47 CFR Part2.1093; IEEE 1528; IEC 62209-2); FCC Part68 HAC; FCC ID

CE：EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 60950-1;EN 301 489-1/-6; EN 301 406; EN 50332-2; SAR(EN50360;EN50566;EN 50663;EN62209-1; EN62209-2; EN 62479); RED NB Cert

RCM：AS/NZS CISPR32; AS/NZS 60950.1;AS/CA S004;AS/ACIF S040. ANATEL, EAC, UL(adapter).

DP730

DP722

LED Light

Status

Indicates Power ON/OFF.

Indicates the status of SIP account registration and network.

Indicates the status of the DECT handset registration:

Key

Description

1

Earphone

Delivers audio output.

2

LED Indication

Red: Charging. Green: Charge completed.

Blinking: Missed call(s) or Voice Mail received.

3,5

Left and right softkeys

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

4

LCD display

Shows call information, handset status icons, prompt messages, etc.

6

4 Arrow key combination

Permits navigation of the cursor through the displayed menu options.

7

Men/Ok key

Selects the option chosen by the cursor. (Enters the main menu from the home screen.)

8

Off-hook / Dial key

Enters dialing mode, or dials number entered.

9

On-hook / Power key

Terminates calls or turns the handset on / off.

10

Alphanumeric Keypad

Provides the digits, letters, and special characters in context-sensitive applications. For the + sign, press and hold key 0.

11

# / Lock key

Locks keypad against unintentional entries when keep pressing #.

Press and hold the

# key for approximately 2 seconds to lock the keys. Press Unlock Softkey

and then # to unlock the keys.

12

Mute key

Activates or deactivates the mute feature.

13

Hands-free / Speaker key

Switches between handset and hands-free/speaker modes.

14

Microphone

Picks up audio earpiece and hands-free calls.

Key

Description

1

Proximity sensor

The proximity sensor can detect and measure gravitational acceleration, tilt, vibration, altitude changes, and static position.

2

Earphone

Delivers audio output.

3,4

Volume up / Down Keys

Configure the handset and ringtone volume.

5

PTT Key

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

6

Hands-free / Speaker key

Switches between Handset and Hands-free / Speaker modes.

7

Arrow key combination (Up, Down, Left, Right)

Allows navigation of the cursor through the displayed menu options.

8

Off-hook / Dial key

Enters dialing mode, or dials number entered.

9

Alphanumeric Keypad

Provides the digits, letters, and special characters in context-sensitive applications. For the + sign, press and hold key 0.

10

* / Silent Mode key

Activates or deactivates the silent mode (no ringtone heard during an incoming call) when keep pressing on * in idle screen.

11

LED indicator

1 dual-color LED indicator indicating power, call, battery, message waiting…

12

3.5 mm headset jack

Phone connector for the headphones/headsets.

13

Color LCD Screen

2.4-inch (240×320) TFT color LCD

14

Softkeys

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

15

Mute

Mute the microphone during the conversation.

16

Menu/OK key

Selects the option chosen by the cursor or enter the main menu from the home screen.

17

On-hook or Power key

Terminates calls or turns the handset on / off.

18

# / Lock key

Locks keypad against unintentional entries when keep pressing #.

Press and hold the

# key for approximately 2 seconds to lock the keys. Press Unlock Softkey

and then # to unlock the keys.

19

Microphone

Picks up audio earpiece and hands-free calls.

Battery status

Not equipped with a battery

Battery status

Battery empty

Battery status

Battery low

Battery status

Battery normal

Battery status

Battery full

Battery status

Charging

Signal status

Not subscribed

Signal status

Not in range

Signal status

Signal very low

Signal status

Signal low

Signal status

Signal normal

Signal status

Signal good

Signal status

Signal very good

Microphone MUTE Status

OFF – Not muted

ON – Muted

Speaker status

OFF – The speaker

is inactivated ON – The speaker

is activated

Headset icon

Missed Call icon

Voicemail icon

Ringtone status

OFF – Ringtone off (Silent mode)

ON – Ringtone on

Keypad Lock status

OFF – Keypad unlock

ON – Keypad locked

DND Status.

OFF – Do Not Disturb disabled

ON – Do Not Disturb enabled

Call waiting

Information

Account not registered

Account Registered

Error message

Handset number

Incoming Call notification

Outgoing Call notification

Missed Call notification

Voicemail notification

Contacts

Call History

Registration

Voice Mail

Preferences

Shortcut

Call Features

Status

Settings

General Settings

Account Active

Activates or deactivates SIP Account.

Account Name

Determines the name of the account, this account name can also be used in Handsets config provisioning for validation.

SIP Server

Configures SIP server IP address or domain name provided by VoIP service provider. This is the primary SIP server used to send/receive SIP messages from/to DP755.

Secondary SIP Server

Specifies failover SIP server IP address or domain name provided by VoIP service provider. This server will be used if the primary SIP server becomes unavailable.

Outbound Proxy

Specifies the IP address or domain name of an outbound proxy, a media gateway, or a session border controller. Used by DP755 for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work, and only the outbound proxy can correct the problem.

Backup Outbound Proxy

IP address or Domain name of the Secondary Outbound Proxy which will be used when the primary proxy cannot be connected.

SIP User ID

User account information, provided by your VoIP service provider.

Authenticate ID

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

Authenticate Password

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

Name

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

Voice Mail Access Number

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

Network Settings

DNS Mode

Selects DNS mode to use for the client to look up server. Default is A Record.
A Record: resolves IP Address of target according to domain name.
SRV: DNS SRV resource records indicate how to find services for various protocols.
NAPTR/SRV: Naming Authority Pointer according to RFC 2915.
Use Configured IP: If selected, please fill in Primary IP, Backup IP 1 and Backup IP 2 to be used for server look up.

Maximum Number of SIP Request Retries

Specifies the maximum number of retries on SIP Requests.
Default value is 2.

DNS SRV Failover mode

Configures the preferred method for DNS SRV failover.
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 "Use current server until failback timer expires" is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until the failback timer expires.

Failback Timer

Specifies the time interval (in minutes) that the device should continue to send SIP requests (REGISTER or INVITE) to the failover IP. Once it expires, the SIP requests will be sent to the preferred IP. The default value is 60 minutes, and the max value is 45 days.

Register before DNS SRV failover

When the DNS SRV Failover Mode is enabled, you can also choose to “Register before DNS SRV failover” that can waive the 3 failed tries, or still try 3 times then use the failover DNS.

Primary IP

Specifies primary IP address where the base sends DNS query to, when “Use Configured IP” is selected for DNS mode.

Backup IP 1

Specifies backup IP 1 address where the base sends DNS query to, when “Primary IP” is not responding.

Backup IP 2

Specifies backup IP 2 address where the base sends DNS query to, when “Backup IP 1” is not responding.

NAT Traversal

Enables/disables NAT traversal mechanism. If activated (by choosing “STUN”) and a STUN server is also specified (Maintenance 🡪 Network Settings 🡪 STUN Settings); the base performs according to STUN client specification.
Under this mode, embedded STUN client will detect if and what type of firewall/NAT is being used. If detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the base will use its mapped public IP address and port in all of its SIP and SDP messages. If NAT Traversal field is set to “Keep Alive”, the base will periodically (every 20 seconds) send a blank UDP packet (with no payload data) to SIP proxy to keep the “ping hole” on the NAT open.

Proxy-Require

Determines a SIP Extension to notify the SIP server that the base is behind a NAT/Firewall.

SIP Settings

Basic Settings

TEL URI

Indicates E.164 number in the “From” header by adding “User=Phone” parameter or using “Tel:” in SIP packets, if the base has an assigned PSTN Number.
Disabled: Will use “SIP User ID” information in the Request-Line and “From” header.
Disabled: Will use “SIP User ID” information in the Request-Line and “From” header.
Enabled: “Tel:” will be used instead of “sip:” in the SIP request.
Please consult your carrier before changing this parameter. The default is Disabled.

SIP Registration

Controls whether to send SIP REGISTER messages to the proxy server. Device may not be able to make/receives calls if disabled. Default is Yes.

Unregister on Reboot

Controls whether to clear SIP user’s information by sending an un-register request to the proxy server. The un-registration is performed by sending a REGISTER message with the “Contact” header set to * and Expires=0 parameters to the SIP server. This will unregister all SIP accounts under the concerned account. The default value is "No".

Register Expiration

Specifies the frequency (in minutes) when the phone refreshes its registration with the specified registrar. The maximum value is 64800 (about 45 days).
The default value is 60 minutes.

Subscribe Expiration

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

Reregister Before Expiration

Sends re-register request after specific time (in seconds) to renew registration before the previous registration session expires.

Enable OPTIONS Keep Alive

Enables OPTIONS Keep Alive, to check SIP server.
Default is No.

OPTIONS Keep Alive Interval

Time interval for OPTIONS Keep Alive feature in seconds. Range of values is 1–64800. Default is 30.

OPTIONS Keep Alive Max Lost

A maximum number of lost packets for the OPTIONS Keep Alive feature before the phone sends a re-registration. Range of values 3-10.
The default is 3.

Enable TCP Keep Alive

Ensures continuous monitoring of the connection with connected devices, promptly detecting interruptions and allowing for quick re-establishment of the connection for reliable communication.
Enabled by Default

Local SIP Port

The parameter specifies the local ports that the base station uses for sending and receiving SIP packets. By default, Account 1 uses port 5060, Account 2 uses 5062, Account 3 uses 5064, and Account 4 uses 5066, with increments of two for each subsequent account up to Account 20.

SIP Registration Failure Retry Wait Time

Sends re-register request after specific time (in seconds) when registration process fails. Maximum interval is 3600 seconds (1 hour). Default is 20 seconds.

SIP T1 Timeout

Defines T1 timeout value. It is an estimate of the round-trip time between the client and server transactions. For example, the base station will attempt to send a request to a SIP server. The time it takes between sending out the request to the point of getting a response is the SIP T1 timer. If no response is received the timeout is increased to (2*T1) and then (4*T1). Request re-transmit retries would continue until a maximum amount of time defined by T2. Default is 0.5 seconds.

SIP T2 Timeout

Identifies maximum retransmission interval for non-INVITE requests and INVITE responses. Retransmitting and doubling of T1 continues until it reaches T2 value. Default is 4 seconds.

SIP Transport

Selects transport protocol for SIP packets; UDP or TCP or TLS. Make sure your SIP server or network environment supports SIP over the selected transport method. Default is UDP.

SIP Listening Mode

Determines whether or not to listen to multiple SIP protocols. Dual will listen to TCP when UDP is selected while Dual (Secured) will listen to TLS/TCP when UDP is selected. If TCP or TLS/TCP is selected, UDP will be listened to.
Set to Transport Only by Default.

SIP URI Scheme When Using TLS

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

Use Actual Ephemeral Port in Contact TCP/TLS

Defines whether the actual ephemeral port in contact with TCP/TLS will be used when TLS/TCP is selected for SIP Transport. 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 connection.
The default setting is “No”.

Outbound Proxy Mode

In Outbound Proxy mode, SIP messages can include the outbound proxy in the route header, or they can be directly sent to the outbound proxy without the route header.
the options to set are: in route, not in route, and always send to.
Set to in-route by default.

Support SIP Instance ID

Adds “SIP Instance ID” attribute to “Contact” header in REGISTER request as defined in IETF SIP outbound draft. Default is Yes.

SUBSCRIBE for MWI

Sends periodic “SUBSCRIBE” requests (depends on “Register Expiration” parameter) for message waiting indication service. Default is No.

SUBSCRIBE for Registration

When set to "Yes", a SUBSCRIBE for Registration will be sent out periodically.

Enable 100rel

Appends “100rel” attribute to the “required” header of the initial signaling messages. Default is No.

Callee ID Display

When the phone is set to 'Auto,' the callee ID in the 180 Ringing will be updated in the order of P-Asserted Identity Header, Remote-Party-ID Header, and To Header. If set to 'Disabled,' the callee ID will show as 'Unavailable.' Choosing 'To Header' will keep the caller ID unchanged and display it as per the To Header

Caller ID Display

When configured as "Auto," the phone will search for the caller ID in the order of P-Asserted Identity Header, Remote-Party-ID Header, and From Header in the incoming SIP INVITE. If set to "Disabled," all incoming calls will be shown as "Unavailable."

Add Auth Header On Initial REGISTER

Adds “Authentication” header with blank “nonce” attribute in the initial SIP REGISTER request. Default is No.

Allow SIP Reset

Allows to reset the devices directly through SIP Notify. If “Allow SIP Reset” is set to “YES”, then the base receives the NOTIFY from the SIP server with Event: reset, the base should perform a factory reset after the authentication.

The authentication in this case can be either with:
The admin password if no SIP account is configured on the base.
With the credentials of the SIP account if configured on the base.

By default, it is set to "No".

Ignore Alert-Info Header

This option is used to configure default ringtone. If set to “Yes”, configured default ringtone will be played. The default setting is No.

Custom SIP Headers

Use Privacy Header

Controls whether the Privacy Header will be present in SIP INVITE message. Default is Default.

Use P-Preferred-Identity Header

Controls whether PPI Header will be present in SIP INVITE message. Default is Default.

Use P-Access-Network-Info Header

Use P-Access-Network-Info header in SIP request.
Enabled by Default.

Use P-Emergency-Info Header

Use P-Emergency-Info header in SIP request.
Enabled by Default.

Use MAC Header

When set to "Only for REGISTER," the MAC header will only be included in SIP messages for registration and unregistration. If set to "Yes to All SIP," the MAC header will be included in all outgoing SIP messages. When set to "No," the MAC header will not be present in any outgoing SIP message.

Add MAC in User-Agent

When set to "Yes except REGISTER," the phone's MAC address will be added to the User-Agent header in all outgoing SIP messages, except for REGISTER and UNREGISTER. If set to "Yes to All SIP," the phone's MAC address will be included in the User-Agent header of all outgoing SIP messages. When set to "No," the phone's MAC address will not appear in the User-Agent header of any outgoing SIP messages.

Advanced Features

PUBLISH for Presence

Enables Presence feature on the phone.
Disabled by Default.

Omit charset=UTF-8 in MESSAGE

Determines whether the base station sends SIP MESSAGE requests without including the "charset=UTF-8" declaration (when enabled) or includes it (when disabled) for specifying the character encoding of the text in the message.
Disabled by Default.

Feature Key Synchronization

When enabled, call features like DND, call forward, call waiting will be synchronized between the server and the phone. It will use NOTIFY to send the status in XML content to server and accept the NOTIFY from the server. This feature following the Broadsoft and MetaSwitch standard. Any server following the same standard will be compatible with this feature.

Special Feature

Selects Soft switch vendors’ special mode. Examples of vendors: Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro, Huawei IMS, Phonepower, and UCM Call Center.
The default is Standard.

Session Timer

Enable Session Timer

Enables/Disables the Session Timer Support. Default is Yes.

Session Expiration

Enables periodic refresh of SIP session via a SIP request (UPDATE, or re-INVITE). When the session interval expires and there is no refresh via an UPDATE or re-INVITE message, the session will be terminated. Session Expiration is the time at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. Default is 180 seconds.

Min-SE

Defines Minimum session expiration (in seconds).
Default is 90 seconds.

Caller Request Timer

Uses session timer when making outbound calls if remote party supports it.
Default is No.

Callee Request Timer

Uses session timer when receiving inbound calls with session timer request.
Default is No.

Force Timer

It uses a session timer even if the remote party does not support this feature. Selecting “No” will enable session timer only when the remote party supports it.
The default is No.
To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.

UAC Specify Refresher

Specifies which end will act as refresher for outgoing calls:
UAC: The base station acts as the refresher.
UAS: Callee or proxy server act as the refresher.
Default is Omit.

UAS Specify Refresher

Specifies which end will act as a refresher for incoming calls:
UAS: The base station serves as the refresher.
UAC: Callee or proxy server act as the refresher.
The default is Omit.

Force INVITE

Uses INVITE message to refresh the session timer. Default is No.

Security Settings

Check Domain Certificates

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

Validate Certificate Chain

Validates certificate chain when TLS/TCP is configured.
Disabled by Default.

Validate Incoming Messages

Defines whether incoming messages will be validated or not. Default is No.

Check SIP User ID for Incoming INVITE

Checks SIP User ID in the Request URI of incoming INVITE; if it doesn’t match the base SIP User ID, the call will be rejected. Direct IP calling will also be disabled. Default is No.

Accept Incoming SIP from Proxy Only

Checks SIP address of the Request URI in the incoming SIP message; if it doesn’t match SIP server address of the account, the call will be rejected. Default is No.

Authenticate Incoming INVITE

Challenges the incoming INVITE for authentication with SIP 401 Unauthorized message. Default is No.

Audio Settings

Preferred Vocoder- Choice x

Configures vocoders in a preference list (up to 8 preferred vocoders) that will be included with same order in SDP message. Vocoder types are G.711 A-/U-law, G.722, G.726-32, G.723, G.729, iLBC and OPUS

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. Default is No.

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".

Disable Multiple m line in SDP

If enabled, the phone always responds to 1 m line in SDP regardless multiple m lines are offered.
Disabled by Default.

SRTP Mode

Selects the SRTP mode to use (“Disabled”, “Enabled but not forced”, “Enabled and forced”,  "Follow SIP Transport", or "Optional").

The default is Disabled.

SRTP Key Length

The cipher method / key length to use if SRTP is enabled.

Crypto Life Time

Adds crypto lifetime header to SRTP packets. The default is Yes.

Symmetric RTP

Defines whether symmetric RTP is supported or not. The default setting is "No".

Silence Suppression

Allows detection of the absence of audio and conserves bandwidth by preventing the transmission of “silent packets” over the network. The default is No.

Jitter Buffer Type

Selects either Fixed or Adaptive based on network conditions.
Set to Adaptive by Default.

Jitter Buffer Length

Selects Low, Medium, or High based on network conditions.
Set to 300ms by Default

Voice Frames per TX

Transmits a specific number of voice frames per packet. Default is 2; increases to 10/20/32/64 for G711/G726/G723/other codecs respectively.

G726-32 Packing Mode

Defines G726-32 packing mode (“ITU” or “IETF”). Default is ITU.

iLBC Frame Size

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

iLBC Payload type

Determines payload type for iLBC. The valid range is between 96 and 127.
Default is 97.

OPUS Payload Type

Determines OPUS payload type. The valid range is between 96 and 127.
Default is 123.

DTMF Payload Type

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

Send DTMF

Specifies the mechanism to transmit DTMF digits.
you can choose to do it in-audio, via RTP(RFC2833), or via SIP INFO

Call Settings

Dial Plan

Dial Plan Rules:

Accept Digits : +,1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d ;
Grammar: x – any digit from 0-9;xx+ – at least 2-digit number;
xx – exactly 2-digit number;
^ – exclude;
. – wildcard, matches one or more characters
[3-5] – any digit of 3, 4, or 5;
[147] – any digit 1, 4, or 7;
<2=011> – replace digit 2 with 011 when dialing
<=1> – add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed
| – or
Example 1: {[369]11 | 1617xxxxxxx} –
Allow 311, 611, 911, and any 10-digit

numbers of leading digits 1617

Example 2: {^1900x+ | <=1617>xxxxxxx} –
Block any number with leading digits 1900 and add prefix 1617 for any dialed 7-digit numbers

Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –
Allow any length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9; If leading digit is 2, replace leading digit 2 with 011 before dialing.

Default: Outgoing – {x+}
Example of a simple dial plan used in a Home/Office in the US:

{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 | +x+ }

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

^1900x. – prevents dialing any number starting with 1900
<=1617>[2-9]xxxxxx – allows dialing to local area code (617) numbers by dialing 7 numbers and the 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 – allow dialing special and emergency numbers 311, 411, 611 and 911
+x+ – allow dialing any digit with leading + sign; example: +16175669300
Note: In some cases, the user wishes to dial strings such as *123 to activate voice mail or other application provided by the service provider. In this case * should be predefined inside the dial plan feature. An example dial plan will be { *x+ } which allows the user to dial * followed by any length of numbers.

Bypass Dial plan

Enable/Disable the dial plan check while making outgoing calls.
The default setting is “No”.

On Hold Reminder Tone

Supports to disable or enable “On Hold Reminder Tone” to play a reminder tone when a call is on hold.

Call Log

Configure the level of call logs or disable the call log.
By default, it is set to Log all calls.

Send Anonymous

Sets “From”, “Privacy” and “P_Asserted_Identity” headers in outgoing INVITE message to “anonymous”, blocking caller ID. Default is No.

Anonymous Call Rejection

Rejects incoming calls with anonymous caller ID with “486 Busy here” message. Default is No.

Refer-To Use Target Contact

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

Transfer on Conference Hangup

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

Blind Transfer Wait Timeout

Defines the timeout (in seconds) for waiting SIP frag response in blind transfer. Valid range is 30 to 300.

Key As Send

Pressing selected key will immediately dial out.
Pound "#" is the default selection.

RFC2543 Hold

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

Match Incoming Caller ID

Specifies matching rules with number, pattern or Alert Info text. When the incoming caller ID or Alert Info matches the rule, the phone will ring with selected distinctive ringtone.

Matching rules:

• Specific caller ID number. For example, 8321123;

• A defined pattern with certain length using x and + to specify, where x could be any digit from 0 to 9. Samples:

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.
• Alert-Info text

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 ringtone for the matching rule. When the incoming caller ID or Alert Info matches the rule, the phone will ring with the selected ring.
Note: Alert info can also be input as a string.

Ring Timeout

Stops ringing when incoming call is not answered within a specific period of time. Default is 60 seconds.

Intercom Settings

Allow Auto Answer by Call-Info/Alert-Info

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

Allow Barging by Call-Info/Alert-Info

When enabled, the phone will automatically put the current call on hold and answer the incoming calls based on the SIP Call-Info/Alert-Info header sent from the server/proxy. However, if the current call was answered based on the SIP Call-Info/Alert-Info header, then all other incoming calls with SIP Call-Info/Alert-Info headers will be rejected automatically.
Disabled by Default.

Custom Alert-Info for Auto Answer

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

Feature Codes

Enable Local Call Features

When enabled, Do No Disturb, Call Forwarding and other call features can be used via the local feature codes on the phone. Otherwise, the provisioned feature codes from the server will be used. User-configured feature codes will be used only if server-provisioned feature codes are not provided.
Enabled By Default

Account Swap

Swap Account Settings

Swap configurations between two accounts.

General Settings

Base Station Name

Displays the name of the base station. The default is DP755[last 6 digits of MAC address].

Admin PIN Code

Configures admin PIN code for authentication. Default is 0000

Enable Repeater Mode

Enables the base station repeater mode to associate with available repeaters. Once enabled the base station starts searching for nearby repeaters and opens a subscription to associate with the available repeaters. This option requires rebooting the base station to take effect. The default is No.

Enable Repeater Management

Enables base station network management of discovered and paired repeaters.
Once enabled, users need first to reboot the base station to take effect, then login
on the web UI and browser to Status page, a new tab “DECT Repeater Status” will
be available to display discovered and paired devices and also allowing users to
associate / dissociate repeaters and also access their web GUI. Default is No.

DECT PTT Silence  Timer

Sets timeout for PTT call (in minutes) if no handset unmutes. If set to 0, this timer
will be disabled.

Collapse Call History

Enables collapsing of like call logs into a single entry for display on handset. Calls will only be combined if of the same type (missed, incoming accepted, or outgoing) and with the same remote party.

Clear Call Logs

Deletes call history logs of all handsets from base station

Account Assignment

SIP User ID

displays the SIP User ID assigned to the account number.

Ringing Mode

Specifies the ringing mode for the account, allowing either parallel ringing or ringing only on the selected handset.

HS1-HS10

Within the handset matrix, you can designate specific handsets to ring for each SIP extension. When the ringing mode is set to Parallel, up to 10 handsets can ring simultaneously for a single extension.

Handset Settings (1-10)

Handset Name

Displays the handset name.

Enable Auto Answer

Enables / disables auto answer of incoming calls to handset. Default setting is No.

Enable Mute for Auto Answer

Enables/disables auto mute right after the call been answer, this can be configured by either the handset GUI or Web UI.

Enable Off hook on Cradle Pickup

Enables / disables off hook of handset when picked up from cradle. Default setting is No.

Enable on a hook on  Cradle Reposition

Enables / disables on hook of handset when repositioned on cradle. Default setting
is No.

Disable Conference

Enables / disables the conference option on this handset. Default setting is No.

Disable Transfer

Enables / disables transfer option on this handset. Default setting is No.

Disable Busy Tone on Remote Disconnect

Enables / disables the busy tone heard in the handset when call is disconnected
remotely.

Disable Call Waiting Tone

Disables playing call waiting tone during active call when receiving a second incoming call. The CWCID will still be displayed. Default is No.

Play warning tone for Auto Answer
Intercom

Allows to play a warning tone for auto answer for protect the privacy. This setting can be configured by either the handset GUI or the Web UI.

No Key Entry Timeout

Initiates the call within this time interval if no additional key entry during dialing stage. Default is 4 seconds.

Custom Ringtone

Assigns custom ringtone to specific handset from the ringtones available on the
base station. It takes up to 10 ringtone files which have be named as ring1.bin to
ring10.bin, and you can assign one ringtone to each handset. Default is Disabled.

Time Format

Set the displayed Time Format on handsets to 12 hours or 24 hours. Default is 12hr.

Date Format

Set the displayed Date Format on handsets.

Handset Phonebook

Selects the phonebook for the handset , you can select from the 10 XML phonebooks available under Phonebook => Private Phonebook Setting.

Offhook Auto-dial

Enables automatic dialing of a predefined number when a handset goes off-hook (lifted from its cradle) without any user input.

Off-hook Auto Dial Delay

Sets the time delay before initiating automatic dialing when a handset goes off-hook (lifted from its cradle) without user input.

Settings => General Settings

Local RTP Port

This parameter defines the local RTP port used to listen and transmit. The valid range is 1024 to 65400 and it must be even.
Default is 5004

Local RTP Port Range

This parameter defines the range of local RTP ports from 24 to 10000.
Default is 200

Use Random Port

When set to "Yes", this parameter will force random generation of both the local SIP and RTP ports.
Enabled by Default.

Keep-Alive Interval

Specifies how often the phone sends a blank UDP packet to the SIP server in order to keep the "ping hole" on the NAT router open.
The default is 70 minutes.

Use NAT IP

The NAT IP address used in SIP/SDP messages. It should ONLY be used if required by your ITSP.

STUN server

The IP address or Domain name of the STUN server. Only non-symmetric NAT routers work with STUN.

Delay Registration

It configs the specific time that the account will be registered after booting up.
Default is 0.

Test Password Strength

Only Allow password with some constraints to ensure better security.
Disabled by Default.

Settings => External Service

Order

Displays the order of the service. (1 – 10)

Service Type

Specifies the service’s type. Two options are available: None or GDS.
The default setting is None.

Note: The DP755 supports up to 10 GDS items.
For more details, refer to Facility Access Systems

Account

Specifies the account on which the service will be applied.

System Identification

Specifies the name to identify the service.

System Number

Specifies the system number, in case the service type option is set to GDS, the system number is the SIP user ID configured on GDS37xx, or the IP address of the GDS37xx itself if it’s using IP call.

Access Password

Determines the access password, in case the service type option is set to GDS, the access password is the one configured on “Remote PIN to Open the Door” field on GDS37xx settings.

Settings => Call Features

Disable Direct IP Call

Enables/Disables Direct IP Call feature.
Disabled by Default

Enable DND Feature

Enables/Disables DND Call feature.

Enabled by Default.
If set to "No", a user cannot turn on Do Not Disturb feature via MUTE key, or menu on LCD

Do Not Escape '#' as %23 in SIP URI

Replaces # by %23 for some special situations.

Return Code When Refusing Incoming Call

When refusing the incoming call, the phone will send the selected type of SIP message to the call.

Return Code When Enable DND

When DND is enabled, the phone will send the selected type of SIP message.
the options are:

By default, it is set to Temporarily unavailable(480)

User-Agent Prefix

Configures the prefix in the User-Agent header.

Settings => PTT/Multicast

PTT multicast address

Defines the multicast address used for the Push-to-talk communication. the Multicast address should contain the IP Address and the port number.

PTT Config

PTT

Enables/Disables the PTT feature.

Default Channel

Sets the default channel for PTT. When pressing and holding the PTT button, PTT will be initiated using the default channel. Default channel is "channel 1"

Priority Channel

Sets priority channel for PTT. PTT received on priority channel will take precedence over active PTT on normal channel. The valid range is 0-25, and default value is 24th chaneel

Emergency Channel

Sets emergency channel for PTT. Emergency channel has the highest priority. PTT using emergency channel will take precedence over PTT on priority or normal channel. Please note PTT to emergency channel will not be rejected even when device has enabled DND. the default chaneel is channel 25.

Caller ID

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

PTime (ms)

Sets payload size for PTT in miliseconds, the default value is 30ms.

Audio Codec

Sets audio codec for PTT. Default is PCMU

Channel Config (1-25)

The user can manually configure the options that will be included in each channel individually, the options are:

Settings => Preferences

Date and Time

NTP Server

Defines the URL or IP address of the NTP server. The phone may obtain the date and time from the server.

Secondary NTP Server

Defines the URL or IP address of the secondary NTP server. The phone may obtain the date and time from the server.

NTP Update Interval

The time interval for updating time from the NTP server. Valid time value is in between 5 to 1440 minutes.
Default is 1440 minutes.

Allow DHCP Option 42 to override NTP server

When enabled, DHCP Option 42 will override the NTP server if it's set up on the LAN.

Allow DHCP Option 2 to Override Time Zone Setting

Allows device to get provisioned for Time Zone from DHCP Option 2 in the local server.
Enabled by Default.

Time Zone

Configures the date/time used on the phone according to the specified time zone.
Enabled by Default.

Self-Defined Time Zone

This parameter allows the users to define their own time zone. For syntax and examples, please refer to user manual.

Ringtone

System Ring Cadence

Sets ring cadences for all incoming calls.
Syntax: c=on1/off1-on2/off2-on3/off3;) Default is set to c=2000/4000; (US standards) on1 is the period of ringing (“On time” in “ms”) while off1 is the period of silence. Up to three cadences are supported.

Call Progress Tones

Configures tone frequencies according to user preference. By default, the tones are set to North American frequencies. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds. 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 repeats the pattern.

• “Dial tone”

• “Ring back tone”

• “Busy tone”

• “Call-Waiting tone”

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

Settings => Voice Monitoring

Session Report

VQ RTCP-XR Session Report

When enabled, phone will send a session quality report to the central report collector at the end of each call.
Disabled by Default.

Interval Report

VQ RTCP-XR Interval Report

When enabled, phone will send a session quality report to the central report collector at the end of each call.
Disabled by Default.

VQ RTCP-XR Interval Report Period

Configure the interval (in seconds) of phone sending an interval quality report to the central report collector periodically throughout a call.
Default is 20 seconds

Alert Report

Warning Threshold for Moslq

Configure the threshold value of the listening MOS score (MOS-LQ) multiplied by 10. The threshold value of MOS-LQ causes the phone to send a warning alert quality report to the central report collector.
The default is 0.

Critical Threshold for Moslq

Configure the threshold value of the listening MOS score (MOS-LQ) multiplied by 10. The threshold value of MOS-LQ causes the phone to send a critical alert quality report to the central report collector.
The default is 0.

Warning Threshold for Delay

Configure the threshold value of the one-way delay (in milliseconds) that causes the phone to send a warning alert quality report to the central report collector.
The default is 0.

Critical Threshold for Delay

Configure the threshold value of one-way delay (in milliseconds) that causes the phone to send a critical alert quality report to the central report collector.
The default is 0.

Network Settings – Basic Settings

Internet Protocol

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

IPv4 Address

Select IP address mode (DHCP, Static IP, or PPPoE) for DP755 Base Station.

Hostname (Option 12)

Specifies the name of the client. The name may or may not be qualified with the local domain name. This field is optional but may be required by ISP.

Vendor Class ID (Option 60)

Exchanges vendor class IDs by clients and servers to convey particular configuration or other identification information about a client. The default is Grandstream DP755.

DNS Server 1

Preferred DNS Server

DNS Server 2

Enter DNS Server 2 when static IP is used.

Preferred DNS Server

Specifies preferred DNS server to use when DHCP, PPPoE, or Static mode is set.

IPv6 Address

The IPv6 address that is obtained on the phone.
it can be set to be auto-configured (DHCP), or statically configured.

Full Static

Defines the Static IPv6 Address, and the IPv6 Prefix length

Prefix Static

Defines the IPv6 Prefix (64 bits).

DNS Server 1

Enter DNS Server 1 when static IP is used in IPv6 format.

DNS Server 2

Enter DNS Server 2 when static IP is used in IPv6 format.

Preferred DNS Server

Specifies preferred DNS server to use when DHCP, PPPoE or Static mode is set in IPv6 format.

Network Settings – Advanced Settings

802.1X Mode

Enables/Disables 802.1X mode. To enable this mode, you should select EAP-MD5, EAP-TLS, or EAP-PEAPv0/MSCHAPv2. The default is disabled.

802.1X Identity

Configures the identity for 802.1X mode.

MD5 Password

Determines the MD5 password for 802.1X mode.

802.1X CA Certificate

Uploads / deletes the 802.1X CA certificates.

802.1X Client Certificate

Uploads / Deletes the 802.1X Client Certificates.

HTTP Proxy

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

HTTPS Proxy

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

Bypass Proxy For

Enter host names that do not require a proxy to reach. Those names should be separated by commas.

Layer 3 QoS for SIP

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

Layer 3 QoS for RTP

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

Enable DHCP VLAN

Enable auto-configure for VLAN settings through DHCP.
Disabled by Default.

Enable Manual VLAN Configuration

Assigns the priority value of the Layer 2 QoS packets. Valid range is 0 to 7.
Enabled by Default.

Layer 2 QoS 802.1Q/VLAN Tag

Sets layer 2 QoS 802.1Q/VLAN tag. Default is 0.

Layer 2 QoS 802.1p Priority Value

Sets layer 2 QoS 802.1p priority value for SIP signaling.

Enable CDP

Enable/Disable the CDP (Cisco Discovery Protocol).
Enabled by Default.

Enable LLDP

Activates LLDP (Link Layer Discovery Protocol).
Enabled by Default.
The default is 0.

LLDP TX Interval

Defines LLDP TX Interval (in seconds). Valid range is 1 to 3600.
Default value is 60.

Maximum Transmission Unit (MTU)

Defines the MTU in bytes.
Default is 1500.

Network Settings – Management Virtual IP Address Settings

Management Virtual IP address

Enable or disable the management virtual IP address.
Disabled by default.

Management Access

Allow management of the device through virtual IP address or both actual IP and virtual IP addresses, you can choose one of the two options:

Set to "Management Virtual IP Address only" by Default

SNMP Through Management Interface

Enable or disable snmp through the management interface.
Disabled by default.

TR069 Through Management Interface

Enable or disable tr069 through the management interface.
Disabled by default.

SYSLOG Through Management Interface

Enable or disable syslog through the management interface.
Disabled by default.

Layer 2 QoS 802.1Q/VLAN Tag

Assigns the VLAN Tag of the Layer 2 QoS packets.

Layer 2 QoS 802.1p Priority Value

Assigns the priority value of the Layer 2 QoS packets. Valid range is 0 to 7.

IP Address Mode

The IPv4 address obtained on the phone using which method: DHCP or Static IP assignement.

Static IP Settings

IPv4 Address

Enter the IP address when static IP is used.

Subnet Mask

Enter the Subnet Mask when static IP is used.

Gateway

Enter the Default Gateway when static IP is used.

DNS Server 1

Enter DNS Server 1 when static IP is used.

DNS Server 2

Enter DNS Server 2 when static IP is used.

Network 🡪 Open VPN® Settings

OpenVPN® Enable

Enables/Disables the OpenVPN® feature.
Default settings is No.

OpenVPN® Server Address

Configures the address of the OpenVPN® server.

OpenVPN® Port

Defines the port of the OpenVPN® server. Default is 1194.

OpenVPN® Transport

Determines network protocol UDP or TCP used for OpenVPN®. Default is UDP.

OpenVPN® CA

Uploads the OpenVPN® CA.

OpenVPN® Certificate

Uploads the OpenVPN® Certificate.

OpenVPN® Client Key

Uploads the OpenVPN® Client Key.

OpenVPN® Cipher Method

Must be the same cipher method used by the OpenVPN® server

OpenVPN® Username

OpenVPN® authentication username (optional)

OpenVPN® Password

OpenVPN® authentication password (optional)

Additional Options

Additional options are to be appended to the OpenVPN® config file, separated by a semicolon. For example comp-lzo no; auth SHA256
Note: Please use it with caution. Make sure that the options are recognizable by OpenVPN® and do not unnecessarily override the other configurations above.

Network => SNMP Settings

Enable SNMP

Enables/Disables the SNMP feature. Default settings is No

Version

Version SNMP version, the available options are:

Port

SNMP port. Default is 161.

Community

Configures the Name of SNMP trap community.

SNMP Trap Version

SNMP Trap Version. Default is Trap Version 2.

SNMP Trap IP

IP address of the SNMP trap receiver.

SNMP Trap Port

Port of the SNMP trap receiver. Default is 162.

SNMP Trap Interval

The interval between each trap sent to the trap receiver. Default is 60.

SNMP Trap Community

Community string associated to the trap. It must match the community string of the trap receiver.

SNMP Username

Username for SNMP.

Security Level

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.
authUser: Users with security level authNoPriv and context name as auth.
privUser: Users with security level authPriv and context name as priv.
Default is NoAuthUser.

Authentication Protocol

Select the Authentication Protocol: “None” or “MD5” or “SHA”. Default is None.

Privacy Protocol

Select the Privacy Protocol: “None” or “DES” or “AES”. Default is None.

Authentication Key

Enter the Authentication Key

Privacy Key

Enter the Privacy Key.

SNMP Trap Username

User name for SNMP Trap.

Trap Security Level

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.
authUser: Users with security level authNoPriv and context name as auth.
privUser: Users with security level authPriv and context name as priv.
Default is NoAuthUser.

Trap Authentication Protocol

Select the Authentication Protocol: “None” or “MD5” or “SHA”. Default is None.

Trap Privacy Protocol

Select the Privacy Protocol: “None” or “DES” or “AES”. Default is None.

Trap Authentication Key

Enter the Trap Authentication Key.

Trap Privacy Key

Enter the Trap Privacy Key