DP755 - Administration Guide

  • Updated on March 15, 2024

WELCOME

Thank you for purchasing Grandstream DP755 DECT Cordless VoIP Base Station and DP722/DP730 DECT Cordless HD Handsets.

The DP755 is a powerful Dect VoIP Phone Base station that pairs with up to 10 of Grandstream’s DP series DECT handsets to offer mobility to business and residential users. It supports a range of up to 400 meters with DP730 and up to 350 meters with DP722/DP720 outdoors and 50 meters indoors to give users the freedom to move around their work or home space, delivering efficient flexibility. This DECT VoIP Phone Base Station supports up to 10 handsets and 20 SIP accounts while also offering 3-way voice conferencing, full HD audio, and integrated PoE. A shared SIP account on all handsets will add seamless unified features that give users the ability to answer all calls regardless of location in real-time. The DP755 supports a variety of auto-provisioning methods and TLS/SRTP/HTTPS encryption security. When paired with Grandstream’s DP720, DP722, or DP730 handsets, the DP755 offers a powerful cordless DECT solution for any business or residential user.

The DP730 is a DECT cordless IP phone that allows users to mobilize their VoIP network throughout any business, warehouse, retail store, or residential environment. It is supported by Grandstream’s DP750, DP752, and DP755 DECT VoIP base stations and delivers a combination of mobility and top-notch telephony performance. Up to ten DP730 handsets are supported on each base station while each DP730 supports a range of up to 400 meters outdoors and 50 meters indoors from the base station. It touts a suite of robust telephony features including support for up to 10 SIP accounts and 2 concurrent calls per handset, full HD audio, a 3.5mm headset jack, push-to-talk, a speakerphone, and more. When paired with GrandStream’s DECT Base Stations, the DP730 offers a powerful cordless DECT solution for any business or residential user.

The DP722 is a DECT cordless IP phone that allows users to mobilize their VoIP network throughout any business, warehouse, retail store, or residential environment. It is supported by Grandstream’s DP750, DP752, and DP755 DECT VoIP base stations and delivers a combination of mobility and top-notch telephony performance. Up to ten DP722 handsets are supported on each base station while each DP722 supports a range of up to 350 meters outdoors and 50 meters indoors from the base station. It touts a suite of robust telephony features including support for up to 20 SIP accounts and two concurrent calls per handset, full HD audio, a 3.5mm headset jack, push-to-talk, a speakerphone, and more. When paired with Grandstream’s DECT Base Stations, the DP722 offers a powerful cordless DECT solution for any business or residential user.

PRODUCT OVERVIEW

Feature Highlights

The following tables contain the major features of the DP755 / DP730 / DP722:

  • DP755
  • 10 Handsets.

  • 20 accounts.

  • 10 Lines per handset.

  • 8 Concurrent calls.

  • PoE power support.

  • Range: Up to 400m with DP730 and up to 350m with DP722/DP720 outdoor / 50m range indoor.

DP755 Features at a Glance

  • DP730
  • DECT cordless HD.

  • 2.4 inch (240x320) color TFT LCD.

  • 500 hours standby / 40 hours talk time.

  • 15 languages embedded.

  • 10 accounts.

  • 2 concurrent calls.

  • 5 ring modes.

DP730 Features at a Glance

  • DP722
  • DECT cordless HD.

  • 1.8 inch (128x160) color TFT LCD.

  • 250 hours standby / 20 hours talk time.

  • 15 languages embedded.

  • 10 accounts.

  • 2 concurrent calls.

  • 5 ring modes.

DP722 Features at a Glance

DP755 Technical Specifications

The following table resumes all the technical specifications including the protocols/standards supported, voice codecs, telephony features, languages, and upgrade/provisioning settings for the Base station DP755.

Air Interface

Telephony standards: DECT

Frequency bands:

  • 1880 – 1900 MHz (Europe), 1920 – 1930 MHz (US)

  • 1910 – 1920 MHz (Brazil), 1786 – 1792 MHz (Korea)

  • 1893 – 1906 MHz (Japan), 1880 – 1895 MHz (Taiwan)

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

3 LED indicators: Power, Network, DECT.

Pairing/Paging button.

One 10/100 Mbps auto-sensing Ethernet port with integrated PoE

Protocols/Standards

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

Voice Codecs

G.711μ/a-law, G.723.1, G.729A/B, G.726-32, iLBC, G.722, OPUS, G.722.2/AMR-WB (special order), in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO), VAD, CNG, PLC, AJB

Telephony Features

Hold, transfer, forward, 3-way conference, push to talk, intercom, downloadable phonebook (XML, LDAP, up to 3000 entries), call waiting, call log (up to 300 records), auto answer, flexible dial plan, server redundancy and fail-over

QoS

Layer 2 QoS (802.1Q, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)

Security

User and administrator level access control, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, TLS, SRTP, HTTPS, 802.1x media access control, DECT authentication & encryption

Multi-language

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

Upgrade/ Provisioning

Firmware upgrade via HTTP/HTTPS or FTP/FTPS, mass provisioning using TR-069 or AES encrypted XML configuration file

Multiple SIP Accounts

Up to twenty (20) distinct SIP accounts per system

Each handset may map to any SIP account(s)

Each SIP account may map to any handsets(s)

Ring Group

  • Parallel Mode: All phones ring concurrently; after one phone answers, the remaining available phones can make new calls

Power & Green Energy Efficiency

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

PoE: IEEE802.3af Class 1, 0.44W–3.84W

Package Content

Base unit, Universal Power Supply, Ethernet cable, Quick Installation Guide, GPL Statement

Dimensions (H x W x D)

to be defined.

Weight

Base unit: 140g;

Universal power supply: 50g;

Package: 370g

Temperature and Humidity

Operation: -10º to 50ºC (14 to 122ºF); 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).

DP755 Technical Specifications

DP730 Technical Specifications

The following table resumes all the technical specifications including the protocols/standards supported, voice codecs, telephony features, languages, and upgrade/provisioning settings for the DP730 handsets.

Air Interface

Telephony standards: DECT

Frequency bands:

  • 1880 – 1900 MHz (Europe), 1920 – 1930 MHz (US)

  • 1910 – 1920 MHz (Brazil), 1786 – 1792 MHz (Korea)

  • 1893 – 1906 MHz (Japan), 1880 – 1895 MHz (Taiwan)

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).

DP730 Technical Specifications

DP722 Technical Specifications

The following table resumes all the technical specifications including the protocols/standards supported, voice codecs, telephony features, languages, and upgrade/provisioning settings for the DP722 handsets.

Air Interface

Telephony standards: DECT

Frequency bands:

  • 1880 – 1900 MHz (Europe), 1920 – 1930 MHz (US)

  • 1910 – 1920 MHz (Brazil), 1786 – 1792 MHz (Korea)

  • 1893 – 1906 MHz (Japan), 1880 – 1895 MHz (Taiwan)

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

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: 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).

DP722 Technical Specifications

GETTING STARTED

This chapter provides basic installation instructions including the list of the packaging contents and also information for obtaining the best performance with the DP730/DP722 DECT Cordless HD Handsets and the DP755 DECT Cordless VoIP Base Station.

Equipment Packaging

  • DP755

DP755

  • 1 Base unit

  • 1 Universal power supply 5V

  • 1 Bracket

  • 1 Ethernet cable

  • 1 Quick Installation Guide

Equipment Packaging – DP755

DP755 Package Content
  • DP730

DP730

  • 1 Handset unit

  • 1 Universal power supply 5V

  • 1 Charging station

  • 1 Handset Belt

  • 1 Rechargeable Li-ion battery

  • 1 Quick Installation Guide

Equipment Packaging – DP730

DP730 Package Content
  • DP722

DP722

  • 1 Handset unit

  • 1 Universal power supply 5V

  • 1 Charging station

  • 1 Handset Belt

  • 2 Rechargeable batteries

  • 1 Quick Installation Guide

DP722 Package Content
Note

Check the package before installation. If you find anything missing, contact your system administrator.

Connecting DP755

To set up the DP755 DECT Cordless VoIP Base Station, please follow the steps below:

DP755 Back View

You have two options for power and network connection of the base station: AC power or Power over Ethernet (PoE).

Note

For better signal range, we recommend installing DP755 with LED side facing toward the usage area. Ceiling mount is recommended for better coverage.

Connecting via AC power

  1. Connect the micro-USB connector to the related port on the base station and connect the other end of the power adapter to an electrical power outlet.
  2. Connect the supplied Ethernet cable between the Internet port on the base station and the Internet port in your network or the switch/hub device port.

Connecting via PoE

To connect the base station using PoE, you need to connect the Ethernet cable provided (or 3rd party network cable) between the Network Socket on the base station to the Ethernet port of your PoE switch/hub.

Setting up DP730/DP722 Handsets

Please follow the below steps to insert batteries into the Handsets:

  • Open the battery compartment cover.
  • For DP730: Inset the Li-ion battery with the electrodes in the bottom left corner.
  • For DP722: Insert AAA batteries with correct polarity (+ / -).
  • Close the battery compartment cover.
Note

Please charge the batteries fully before using the Handsets for the first time.

Setting up the DP730/DP722

Battery Information

DP722 Batteries Specifications

DP730 battery specifications

  • Technology: Nickel Metal Hydride (Ni-MH)

  • Size: AAA

  • Voltage: 1.2V

  • Capacity: 800mAh

  • Charging time: 12 hours from empty to full

  • Standby time: up to 250 hours

  • Talk time: up to 20 hours of active talk time

  • Technology: Li-ion

  • Nominal Voltage: 3.8V

  • Capacity: 1500mAh

  • Charging time: 12 hours from empty to full

  • Standby time: up to 500 hours

  • Talk time: up to 40 hours of active talk time

DP722/DP730 Battery Specifications

In order to get the best performance of your DP730/DP722 Handsets, we recommend using the original batteries provided in the package or batteries compliant with the above specifications.

Disclaimer

The abovementioned battery specifications can vary and depend on many factors (age of the battery, number of recharge cycles, real capacity…). The recharge cycles of the battery are limited; thus, it might need to be replaced if the battery performance is low. The number of charge cycles and battery life are affected by usage and configuration.

Important Note

Be careful when inserting the batteries into your handset to avoid any risk of short-circuit, which lead to damage your batteries and/or the handset itself. Do not use damaged batteries which can increase the risk of serious harm.

Setting up the Charge Station

Please refer to the following steps for setting up the charge station and charging the Handsets:

  1. Connect the DC plug on the power adapter to the micro-USB connector on the charge station.
  2. Connect the other end of the power adapter to an electrical power outlet.
  3. After setting up the Handsets and the charge station, place the Handsets in the charge station.
Setting up Charging Stations

DP755 LED Patterns

The DP755 has 3 LED lights on it. Please refer to the following table for the meaning of each light.

DP755 LED Patterns

LED Light

Status

Indicates Power ON/OFF.

Indicates the status of SIP account registration and network.

  • Solid ON: SIP account registered.

  • Blinking: SIP account not registered or network errorList Item 2

Indicates the status of the DECT handset registration:

  • Solid ON: Handset registered to thebase.

  • Fast Blinking (0.25s ON/0.25s OFF): Paging handset.

  • Blinking (0.5s ON/0.5s OFF): Pairing mode.

DP755 LED Patterns

DP730/DP722 Handsets Description

The LCD screen and the Keypad are the main hardware components of the DP730/DP722.

DP722 Keys Description

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.

DP730 Keys Description

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.

DP730 Keypad Keys Description

DP730/DP722 Icons Description

The following table contains a description of each icon that might be displayed on the LCD screen of the DP730/DP722 Handsets.

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

DP730/DP722 Icons Description

DP730/DP722 Handsets Menu

The Handsets has an easy-to-use menu structure. Every menu opens a list of options. To open the main menu, press “Menu” (left softkey) when the Handsets is on and in standby mode. Press the Arrow keys to navigate to the menu option you require. Then press “Select” (left softkey) or OK/Selection key to access further options or confirm the setting displayed. To go to the previous menu item, press “Back” (right softkey). You can press the Power key at any time to cancel and return to standby mode. If you do not press any key, the Handsets automatically revert to standby mode after 20 seconds.

Note

Users can navigate through the handset menu by pressing the menu number when displayed.

DP730/DP722 Menu Structure

Contacts

  • Private: Private contacts include contacts visible in the current Handsets only.

  • Global: Global phonebook contacts are the contacts shared between the Handsets subscribed to the DP755 base station.

Note: Private/Global Phonebooks will be merged on the handset.

Call History

Display the call history:

  • Missed Calls.

  • Accepted Calls.

  • Outgoing Calls.

  • All Calls.

Note: You can add contacts to Shared Contacts directly from call logs.

Registration

  • Register: Register your handset to base station.

  • Deregister: Deregister your handset from base station.

  • Select Base: Select base station.

Voice Mail

  • Play Message: Play voice mail messages received.

  • Set Voice Mail: Configure voice mail parameters.

  • Set Key 1: Configure Key 1 as VM speed dial for selected account.

Preferences

  • Outgoing Default Line: Select account to be use by default for outgoing calls.

  • Auto Answer: Enable/Disable Auto Answer. (Default is Disabled).

  • Off-Cradle Pickup: Enable/Disable Off-Cradle Pickup. If enabled, users can answer the calls by picking up the handset off-cradle. (Default is Disabled).

  • On-Cradle Hangup: Enable/Disable On-Cradle Hangup. If enabled, users can end the call by placing the handset on-cradle. (Default is Disabled).

  • Mute as DND: Enable/Disable Mute as DND. If enabled, pressing mute key on idle state will set the phone to DND mode. (Default is Enabled)

  • Disable Busy Tone: Enable/Disable Busy Tone. If set to enabled, busy tone will not be played. (Default is Disabled).

  • Disable CW Tone: Enable/Disable CW Tone. If set to enabled, Call Waiting Tone will not be played. (Default is Disabled).

  • Onhook Backlight: Enable/Disable Onhook Backlight. If enabled, pressing "Hangup" key on idle screen will switch off LCD screen. (Default is Disabled)

  • Cradle Backlight: Enable/Disable/Dim Cradle Backlight. If enabled, LCD will remain backlit when the handset is placed on-cradle/charging. If set to “Dim”, LCD brightness will be reduced when the handset is placed on-cradle/charging. (Default is Disabled)

  • SIP Account Display: Select which SIP Account information will be displayed on the screen.

  • Name Only: Display SIP Account Name only. (Default)

  • ID Only: Display SIP User ID only.

  • None: No account information will be displayed.

  • PTT (Push To Talk): Enable/Disable Push To Talk. If set to enabled, pressing and hold PTT hard/soft key, a PTT call will be initiated. Pressing the PTT hard/soft key, it will redirect you to the setting to enable or disable it. (Default is Disabled)

Customizing keys functions

Customizing keys functions:

  • : Configure Left Softkey function in idle. Function can be set as Menu, History, Contacts, Line or PTT. Default is Menu.

  • : Configure Right Softkey function. Function can be set as History, Contacts, Line or PTT Default is Contacts.

  •  : Configure Arrow UP Key function. Default is Outgoing Calls (Call History).

  •  : Configure Arrow DOWN key function. Default is Accepted Calls (Call History).

  •  : Configure Arrow LEFT key function. Default is Ringer Volume Down.

  • : Configure Arrow RIGHT key function. Default is Ringer Volume Up.

Select key and press OK button to configure function. Following functions are available for arrow keys: 1. Disabled, 2. Missed Calls, 3: Accepted Calls, 4: Outgoing Calls, 5: History, 6: Contacts, 7: Status, 8: Line, 9: Voice Mail, 10: Ringer Volume Up, 11: Ringer Volume Down, 12: Audio Volume Up, 13: Audio Volume Down, 14: Intercom.

  • Speed Dial: Assign contact numbers as speed dial.

Select a key [2], [3], [4], [5], [6], [7], [8] or [9] and press OK button. Select “Edit” to manually specify the destination number or select “From Contacts” to select a contact as speed dial destination.

Call Features

  • Do Not Disturb: Enable/disable do not disturb mode on the phone.

  • Call Forward: Configure call forward feature.

  • Call Waiting: Configure call waiting feature.

  • Paging: Configures Inter-Handsets paging feature.

Status

  • Base Status: Display Base status (Firmware, IP address, Subnet mask, Gateway, MAC Address)

  • Handset Statut: Display Handsets status (Model RF, Firmware, IPEI)

  • Line Status: Display Line status (Account name, Status)

Settings

  • Handset Name: Change the Handset name.

  • Phone Language: Select the language to be displayed on the phone’s LCD. (Default is English.)

  • Date/Time: Configure date and time on the Handsets.

  • Audio: Specify ringtones for internal/external calls, the volume, advisory tones (Keypad, Confirmation, Low battery notifications) and Vibration mode (DP730 only).

  • Display: Configure backlight, LCD timeout (Idle/Call), LCD brightness, Message Waiting Prompt and menu key timeout.

  • Gestures (DP730 only): Configure Close-to-Ear Backlight and Facedown Hangup.

  • Network Settings: Configure IP addresses and select DHCP/Static IP mode.

  • SIP Settings: Configure/View SIP accounts settings.

  • System Settings: Change Base PIN code, perform factory reset, reboot base and configure repeater mode.

  • Firmware Upgrade: Upgrade the firmware version of the Handsets.

  • Factory Functions: Diagnostic Mode (DP722) / Keypad Diagnostic (DP730)

All LEDs will light up, and the LCD will display a table listing the names of all keys in red. Press any key to diagnose; the key’s name will display in blue. After all keys are diagnosed, a prompt message (“PASS”) will display; press “Back” (right softkey) to exit.

Note: User can long press arrow UP key to exit at any time.

  • Audio Loopback: Speak to the phone using speaker/Handsets/headset. If you can hear your voice, your audio is working fine. Press “Exit” softkey to exit audio loopback mode.

  • LCD ON / OFF: Select this option to turn off LCD. Press any button to turn on LCD.

  • LCD Diagnostic: Select this option to enter LCD Diagnostic mode. Press “Next” (left softkey) to display white screen. Continue pressing the left softkey to view all remaining screens (black, blue, red, and green) and then exit. End the test early by pressing the right softkey.

  • LED Diagnostic: Enters this option and press “1” to start LED Diagnostic (you will notice that the color of the LED will be changing). Press “2” to quit. 

  • System Monitoring: Displays RSSI, battery voltage and RPN information.

  • Vibration (DP730 only): Test vibration on DP730.

  • Acceleration Sensor (DP730): Displays X, Y, Z coordinates.

  • Proximity Sensor (DP730): Test proximity sensor on DP730.

DP730/DP722 Menu Structure Definitions

CONFIGURATION GUIDE

The DP755 can be configured using:

  1. Web GUI embedded on the DP755 using PC’s web browser.
  2. LCD Configuration Menu using the paired DP730/DP722 keypad.

Via Web GUI you can configure all the functions supported by the DP755; while via paired DP730/DP722, you can access limited configuration and need the base station PIN code for some options.

Obtain DP755 Base Station IP Address via Paired DP730/DP722

DP755 is by default configured to obtain an IP address from the DHCP server where the unit is located. In order to know which IP address is assigned to your DP755, please follow the below steps using a paired DP730/DP722 Handset with your DP755 base station. Please see Register DP730/DP722 Handsets to DP755 Base Station.

  1. Press the “Menu” (left softkey) or OK button on DP730/DP722 to view the operation menu.
  2. Press Arrow (Up, Down, Left, Right) keys to move the cursor to the Status icon , then press
    “Select” (left softkey) or OK button, then select Base Status.
  3. Using Arrow keys, navigate down to view the IP address of the DP755.
Base Status

Configuration via Web Browser

The DP755 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow a user to configure the DP755 through a Web browser such as Google Chrome, Mozilla Firefox, and Microsoft’s IE.

Accessing the Web UI

  1. Connect the computer to the same network as DP755.
  2. Make sure the DP755 is booted up.
  3. You may check the DP755 IP address via a subscribed DP730/DP722 on its LCD menu at Status 🡪 Base Status 🡪 IP Address. Please see Obtain DP755 Base station IP Address via paired DP730/DP722
  4. Open the Web browser on your computer.
  5. Enter the DP755’s IP address in the address bar of the browser.
  6. Enter the administrator’s username and password to access the Web Configuration Menu.
Note

The computer must be connected to the same sub-network as the DP755. This can be easily done by connecting the computer to the same hub or switch as the DP755.

Web GUI Languages

Currently, the DP755 series web GUI supports English, Czech, German, Spanish, French, Arabic, Hebrew, Italian, Russian, Netherlands, Japanese, Polish, Chinese Simple, Chinese Tradition, Korean, Portuguese, Slovakian, Serbian, Swedish, and Turkish.

Users can select the displayed language in the web GUI login page, or at the upper right of the web GUI after logging in

DP755 Web GUI Language

Icons Bar Shortcut

Users can find the icon bar right below the main menu of every page as displayed in following screenshot:

Icons Bar Shortcut

Please refer to the following table describing the use of each icon:

Icon

Description

Refresh Button: Allows users to refresh the current page.

Subscribe Button: Allows users to open the subscription.

Paging Button: Allows users to page all the registered DP730/DP722 Handsets.

Saving the Configuration Changes

After users make changes to the configuration, pressing the Save button will save but not apply the changes until the Apply button on the top of the web GUI page is pressed. Users can instead directly press the Save and Apply button. We recommend rebooting or powering cycle the phone after applying all the changes.

Web UI Access Level Management

There are two default passwords for the login page:

User Level

Username

Password

Web Pages Allowed

End User Level

user

123

Only Status, Settings, and Maintenance

Administrator Level

admin

A random password is available on the sticker at the back of the unit.

All pages

The password is case-sensitive with a maximum length of 25 characters.
Note: When accessing the web GUI with the end-user level, the “Advanced Settings” page will be hidden.

When changing any settings, always SUBMIT them by pressing the Save or Save and Apply button at the bottom of the page. If using the Save button, after making all the changes, click on the Apply button on top of the page to submit. After submitting the changes in all the Web GUI pages, reboot DP755 to have the changes take effect if necessary; most of the options under the Settings page require a reboot, but options under the Accounts and Phonebook pages do not.

Changing User Level Password

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings.
  3. Go to Maintenance 🡪 Web/SSH Access.
  4. In the Web/SSH Access page, locate the User Password section:
    • Type in your new user password in the New Password field.
    • Type in again the same entered password in Confirm Password field.
  5. Press Save and Apply to save your new setting.
Changing User Level Password
Note

DO NOT USE the same password for both user and admin accounts.

Changing Admin Level Password

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings.
  3. Go to Maintenance 🡪 Web/SSH Access.
  4. In the Web/SSH Access page, locate the Admin Password section:
    • Type in your new Admin Password in the New Password field.
    • Type in again the same entered password in Confirm Password field.
  5. Press Save and Apply to save your new setting.
Changing Admin Level Password
Note

DO NOT USE the same password for both user and admin accounts.

Changing HTTP / HTTPS Web Access Port

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings.
  3. Go to Maintenance 🡪 Security Settings 🡪 Web/SSH.
  4. In the Web/SSH Settings page, locate HTTP / HTTPS Web Port field and change it to your desired/new HTTP / HTTPS port.
    Note: By default, the HTTP port is 80 and HTTPS is 443.
  5. Select the Web Access Mode depending on the desired protocol (HTTP or HTTPS).
  6. Press Save and Apply to save your new setting.
Note

A reboot is required for this change to take effect.

Note

Selecting “Disabled” for Web Access Mode will disable web UI access.

Web Access Port

Web Configuration Definitions

This section describes the options in the DP755 Web UI. As mentioned, you can log in as an administrator or an end user.

  • Status: Display system info, network status, base and repeater status, account status, and line options.
  • Accounts: Configure the accounts with general settings, network settings, SIP settings, audio settings, call settings, ring tones, and more.
  • DECT: Configure DECT general settings, Account Assignments, and Handsets line settings.
  • Settings: Configure ring tones and system features.
  • Network: Configures the network settings such as OpenVPN Settings and SNMP Settings.
  • Maintenance: Configure networks, upgrade and provisioning, web/SSH access, TR-069, security settings, date and time, and syslog.
  • Phonebook: Manage phonebooks: global (XML or LDAP) and private (XML).

Status Page Definitions

Account Status

Account

Displays list of configured accounts’ names, from Account 1 to Account 10.

SIP User ID

Displays list of SIP user id registered.

SIP Server

Displays list of SIP Server.

SIP Registration

Shows the status of SIP registration. If the SIP account is successfully registered, it will display “YES” with green background. If the SIP account is not registered, it will display “NO” with red background.

Ringing Mode

Displays the HS mode configured for each account.

HS status table

Illustrates both Handsets and SIP accounts statuses. Each column is dedicated to one HS; each row shows the status of the account on that HS:

  • Gray: HS is not configured to use this account.

  • Green: HS is idle on this account.

  • Green Blinking: HS is using this account.

  • Red/Orange Blinking: HS is ringing on this account.Green Blinking: HS is using this account.

  • Brown: The line is configured, but the handset is not subscribed.

For example, if accounts 1, 3 and 4 are assigned to HS3 with account 3 in use, the column for HS3 will have cell 3 with red icon, cells 1 and 4 with green icon, and cells 2 and 5 with gray icon.

DECT Base Status

Base Station Name

Displays name of base station. Default is DP755_[last 6 digits of MAC address].

Base DECT FW Version

Shows firmware version of base DECT.

Base DECT RF Region

Indicates region of base DECT RF.

Base DECT RFPI Address

Specifies DECT RFPI (Radio Fixed Part Identity) address which is a unique identity for the base.

Global Functions

Displays Global Information about upgrading handsets.

Handset

Displays Handset index.

Name

Displays Handsets names

IPEI

Indicates IPEI number of each Handsets; this is the unique identity for the Handsets. If the Handsets is in range, the IPEI will be displayed with a green background, otherwise, it will be displayed with a red background.

Illustrates battery status for each handset; it can be either:

  •  Fully charged : 

  •  Not Fully charged : 

  • Low, needs to be charged or replaced:

  • Charging:

Page

Sends paging request to corresponding Handsets, which will receive incoming ring tone and “Paging” will be displayed on their LCD screens; this function helps you locate the Handsets.

Unsubscribe

Unsubscribes corresponding handset from DECT base station.

HS Firmware

Indicates Handsets’ firmware version number.

Upgrade

Shows Handsets upgrade status or trigger handset upgrade process.

Line Options

Account

Account index.

SIP User ID

Displays the configured SIP User ID for the account.

DND

DND status of the account. Default No.

Forward

The unconditional forward number.

Busy Forward

The forward number for Call Forward Busy.

Delayed Forward

The no answer delayed forward number.

Network Status

MAC Address

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

IP Setting

Indicates used IP address mode: DHCP, Static IP or PPPoE.

IPv4 Address

Displays assigned IPv4 address.
Example: 192.168.5.110

IPv6 Address

Displays assigned IPv6 address.
Example: 0:0:0:0:0:ffff:c0a8:056e

OpenVPN® IP

Displays OpenVPN® IP address.

Subnet Mask

Displays assigned subnet mask.
Example: 255.255.255.0

Gateway

Displays assigned default gateway.
Example: 192.168.5.1

Primary DNS

Shows assigned Primary DNS server address. Example: 8.8.8.8

Secondary DNS

Shows assigned Secondary DNS server address. Example: 8.8.4.4

PPPoE Link Up

Indicates PPPoE connection status.

NAT Type

Displays the NAT type enabled.

NAT Traversal

Indicates the type of NAT for each account ( 20 accounts in total).

System Info

Product Model

Displays product model info. Default is DP755.

Part Number

Shows product part number. Example: 9610006512A (last 2 digits show HW version, in this example 12A for HW version 1.2A)

Certficate Type

Displays the Certificate type installed on the device.

Software Version

Boot: Specifies Boot version.
Core: Specifies Core version.
Prog: Specifies Prog version, This is the main firmware release number, which is always used for identifying the software system of the DP755.
Handset: Specifies Handset firmware version.

IP Geographic Information

  • City: Displays the city where the DP755 is located

  • Language: Displays the language of the unit.

  • Time Zone: Displays the Time Zone that the DP755 is on.

Special Feature

Displays wether the OpenVPN® support is enabled or not.

System Up Time

Indicates system uptime since last reboot.

System Time

Shows actual time and date according to your configuration.

Service Status

Reveals status of VoIP applications.

System Information

Gives the option to download System information

User Space

Shows User sapce used and the database status

Core Dump

Generates core dump by killing programs gs_cmbs and gs_phone.

Status Page Definitions

Accounts Page Definitions

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.

DNS SRV Failover mode

This feature is sued to configure the preferred IP mode for DNS SRV. If selected “default”, first IP from query result will be applied; If selected “Saved one until DNS TTL”, previous IP will be applied before reaches DNS timeout; If selected “Saved one until no response”, previous IP will be applied even after DNS time out until it cannot response.

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").


  • Disabled: SRTP is not used at all. Voice communication will be transmitted without encryption or authentication, which may pose a security risk.

  • Enabled and forced:  Ensures that SRTP is always used for voice communication. The base station enforces the use of SRTP and requires connected devices to support it. If a device doesn't support SRTP, the call may be rejected or the communication won't be established.

  • Follow SIP Transport: When "Follow SIP Transport" is chosen, the SRTP protocol follows the transport protocol specified in the SIP signaling messages to determine when to start and stop the encryption and decryption of the RTP traffic.

  • Optional: This option allows SRTP to be used, but it is not mandatory. The base station can negotiate with the devices to determine whether to enable SRTP. If SRTP is agreed on, the communication will be encrypted and authenticated. If not, the communication will proceed without SRTP.

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.

Accounts Page Definitions

DECT Page Definitions

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.

DECT Page Definitions

Settings Page Definitions

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:

  • Busy(486)

  • Temporarily unavailable(480)

  • Not Found(404)

  • Decline(603)

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:

  • Available

  • Transmit

  • Subscribe

  • Join channel

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.

Settings Page Definitions

Network Page Definitions

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 🡪 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:

  • Version 1

  • Version 2

  • Version 3

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

Network Page Definitions

Maintenance Page Definitions

User Password

New Password

Set new password for web GUI access as User. This field is case sensitive.

Confirm Password

Enter the new User password again to confirm.

Admin Password

Current Password

The current admin password is required for setting a new admin password.

New Password

Set new password for web GUI access as Admin. This field is case sensitive

Confirm Password

Enter the new Admin password again to confirm.

Upgrade Firmware

Allows users to upload the firmware file locally by pressing Start, after selecting the correct firmware file from the local storage, the phone will start the firmware upgrade automatically.

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent: Always Check for New Firmware, Check New Firmware only when F/W pre/suffix changes, Always Skip the Firmware Check.
The default setting is “Always Check for New Firmware”.

Always Authenticate Before Challenge

Only applies to HTTP/HTTPS. If enabled, the phone will send credentials before being challenged by the server. The default setting is “No”.

Disable Firmware Upgrade Confirmation

Disables the Firmware Upgrade confirmation popup.
Set to “No” by Default.

Validate Hostname in Certificate

To validate the hostname in the SSL certificate

Allow DHCP Option 43 and Option 66 Override Server

The default setting is “Yes”. DHCP option 66 originally was only designed for TFTP servers. Later on, it was extended to support an HTTP URL. WP phones support both TFTP and HTTP servers via option 66. Users can also use the DHCP option 43 vendor-specific option to do this.
DHCP option 43 approach has priorities. The phone is allowed to fall back to the original server path configured in case the server from option 66 fails.

Additional Override DHCP Option

When enabled, users could select Option 150 or Option 160 to override the firmware server instead of using the configured firmware server path or the server from option 43 and option 66 in the local network. Please note this option will be effective only when option “Allow DHCP Option 43 and Option 66 to Override Server” is enabled. The default setting is “None”.

Allow DHCP Option 120 to override SIP Server

Enables DHCP Option 120 from local server to override the SIP Server on the phone. The default setting is “No”.

3CX Auto Provision

The phone will multicast SUBSCRIBE for provision if this feature is enabled.
The default setting is “Yes”.

Automatic Upgrade

Enables automatic upgrade and provisioning, the options can be :

  •  Yes, check for upgrade every 10080 minute(s)

  • Yes, check for upgrade every day

  • Yes, check for upgrade every week

  • No

Set to "No" by Default.

Randomized Automatic Upgrade

Randomized Automatic Upgrade within the range of hours of the day or postpone the upgrade every X minute(s) by random 1 to X minute(s).

Hour of the Day (0-23)

Defines the hour of the day to check the HTTP/TFTP/FTP server for firmware upgrades or configuration files changes. The default value is 1.

Day of the Week (0-6)

Defines the day of the week to check HTTP/TFTP/FTP server for firmware upgrades or configuration files changes. The default value is 1.

Disable SIP NOTIFY Authentication

The device will not challenge NOTIFY with 401 when set to “Yes”.
The default setting is “No”.

Config

Config Upgrade Via

Allows users to choose the config upgrade method: TFTP, FTP, FTPS, HTTP or HTTPS. The default setting is “HTTPS”.

Config Server Path

Defines the server path for provisioning.

Config Server Username

The username for the config server.

Config Server Password

The password for the config server.

Config File Prefix

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

Config File Postfix

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

XML Config File Password

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

Enable Handset Config Upgrade

Enable handset config upgrade for handset related settings.
Disabled by Default.

Handset Config File Prefix

If configured, only the handset configuration file with the matching encrypted prefix will be downloaded and flashed into the device. If the file is located in a subdirectory on the provisioning server, the directory should be included here; for example "handset/ipei_".

Handset Config File Postfix

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

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. Default setting is “No”.

Download Device Configuration

Click to download the phone’s configuration file in .txt format.
Note: Configuration backup file does not include passwords or CA/Custom certificate

Download Device Configuration (XML)

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

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 this option is enabled, the phone will inverse the downloading process to cfg.xml > cfgMAC.bin > cfgMAC.xml.
The following files will override the files that have already been loaded and processed.

Download User configuration

This allows users to download part of the configuration that does not include any personal settings like Username and Passwords. Also, it will include all the changes manually made by user from web UI, or config file uploaded from “Upload Device Configuration”, but not include the changes from the server provision via TFTP/FTP/FTPS/HTTP/HTTPS.

Upload Device Configuration

Uploads configuration file to phone.

Export backup Package

Export backup package which contains device configuration along with personal data.

Restore from Backup package

Click to upload backup package and restore.

Firmware

Firmware Upgrade Via

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

Firmware Server Path

Defines the server path for the firmware server.

Firmware Server Username

The username for the firmware server.

Firmware Server Password

The password for the firmware server.

Firmware File Prefix

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

Firmware File Postfix

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

HS Firmware

Handset firmware

Upload Handset Firmware. Reboot the device after uploading to use the new firmware.

Automatic Upgrade

Enables automatic upgrade and provisioning.

Syslog

Syslog Protocol

If set to SSL/TLS, the Syslog messages will be sent through secured TLS protocol to Syslog server.
The default setting is UDP.
Note: The CA certificate is required to connect with the TLS server.

Syslog Server

The URL or IP address of the syslog server for the phone to send syslog to.
Note: By adding a port number to the Syslog server field (i.e. 172.18.1.1:1000), the phone will send Syslog to the corresponding port of that IP.

Syslog Level

 Selects the level of logging for syslog.

The default setting is “None”. There are 4 levels: DEBUG, INFO, WARNING and ERROR.

Syslog messages are sent based on the following events:

Product model/version on boot up (INFO level).
NAT related info (INFO level).
sent or received SIP message (DEBUG level).
SIP message summary (INFO level).
inbound and outbound calls (INFO level).
registration status change (INFO level).
negotiated codec (INFO level).
Ethernet link up (INFO level).
SLIC chip exception (WARNING and ERROR levels).
Memory exception (ERROR level).

Syslog Keyword Filtering

Syslog will be filtered based on keywords provided. If you enter multiple keywords, it should be separated by ‘,’. Please note that no spaces are allowed.

Send SIP Log

Configures whether the SIP log will be included in the syslog messages. The default setting is “No”.
Note: By setting Send SIP Log to Yes, the phone will still send SIP log from syslog even when Syslog Level is set to NONE.

Syslog Capture

Status

 Shows the status of the capture, weather it is "stopped" or capturing, you have the possibility to strat the capture, stop the capture, and download it.

Capture Mode

Sets the capture mode. Either set to Timed mode or continuous.

  • Timed Mode: When a new capture is running, the previous files are deleted. Capture Timer is optional, if Internal Storage is selected, the maximum Capture Timer limit is 30 minutes.

  • Continuous Mode: This mode allows device to capture logs continuously during the days set under Continuous Capture Days option.

Capture Timer

If Capture Mode is set to “Timed” this field will appear to specify how long to capture syslog in minutes. 0 is unlimited. Internal capture has a 30-minute maximum limit.

Log File Rotation

Rotation is always enabled when capturing internally.

Log File Rotation will maintain a fixed maximum limit of the file size based on the Max Log File Size and Max Log Files configured. Old logs will be deleted when rotated.

Max Log File Size

 The maximum log file size used when rotation is enabled

Max Log Files

The number of log files used when rotation is enabled

TR-069

Enable TR-069

Sets the phone to enable the “CPE WAN Management Protocol” (TR-069). The default setting is “Yes”.
Note: Once you enable or disable TR-069 and click “save,” a confirmation pop-up will appear, asking you to confirm your desire to reboot the device.

ACS URL

Specifies URL of TR-069 ACS (e.g., http://acs.mycompany.com), or IP address.
The default setting is https://acs.dgms.cloud

TR-069 Username

Specifies the username to authenticate to ACS.

TR-069 Password

Specifies the password to authenticate to ACS.

Periodic Inform Enable

When enabled, periodic information packets to the ACS server will be sent.
The default setting is “Yes”

Periodic Inform Interval

Configures periodic inform intervals to send the inform packets to TR-069 Auto Configuration Server.
The default setting is 86400

Connection Request Username

Specifies the username for the ACS to connect to the phone.

Connection Request Password

Specifies the password for the ACS to connect to the phone.

Connection Request Port

The port for the ACS to connect to the phone.

CPE SSL Certificate

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

CPE SSL Private Key

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

Randomized TR069 Startup

When enabled, TR069 will send out first INFORM message to server on randomized timing between 1 to 3600 seconds after phone boots up.

Security Settings

Validate Server Certificates

After enabling this feature, the phone will validate the server’s certificate. If the server that our phone tries to register on is not on our list, it will not allow the server to access the phone.

SIP TLS Certificate

SSL Certificate used for SIP Transport in TLS/TCP.

SIP TLS Private Key

SSL Private key used for SIP Transport in TLS/TCP.

SIP TLS Private Key Password

SSL Private key password used for SIP Transport in TLS/TCP.

Custom Certificate

The uploaded custom certificate will be used for SSL/TLS communication instead of the WP phone default certificate.

Web Access Mode

Sets the protocol for web interface. The default setting is “HTTP”.

Enable User Web Access

Administrator can disable or enable user web access. Default is Enabled.

HTTP Web Port

Configures the HTTP port under the HTTP web access mode.

HTTPS Web Port

Configures the HTTPS port under the HTTPS web access mode. Default setting is “443”.

Disable SSH

Disables SSH access. The default setting is “No”.

SSH Public Key

This option allows you to use authentication keys for SSH access. The public key should be loaded to the phone’s web UI while the private key should be used on the SSH tool side.
Note: This will allow upcoming SSH access without a password.

Web Session Timeout

Configures timer to logout web session during idle. Default is 10 min. Range is 2-60 min.

Web Access Attempt Limit

Configures attempt limit before lockout. Default is 5. Range is 1-10.

Minimum TLS Version

Allows users to choose the minimum TLS version for HTTPS provisioning.
Note: Minimum TLS version should be less or equal to the Maximum TLS version. The default setting is TLS 1.1

Maximum TLS Version

Allows users to choose the maximum TLS version for HTTPS provisioning.
Set to unlimited.

Trusted CA Certificates

Trusted CA Certificates

Allows to upload and delete up to 6 CA Certificates files to the phone.
Note: Users can either upload the file directly from the web or they can choose to provision it from their cfg.xml file.

Load CA Certificates

Users are able to specify which certificate they are going to use:
All Certificates: (Default) Both built-in and uploaded Certificates.
Default Certificates: Built-in Certificates.
Custom Certificates: Uploaded Certificates;

Packet Capture

With RTP Packets

Choose whether the packet capture file contains RTP or not.
Set to no by default.

With Secret Key Information

Allows users to make packet capture including the secret key to decrypt the captured TLS packets. Default value is No.

Factory Reset

Force Reboot

Allows for manual restarts, resolving issues by power cycling the system, enhancing overall performance and stability.

Configure Web UI Button

Reset Type

Specifies the type of reset to perform via the web UI button below, the options are:

  • Full Factory Reset.

  • NVRAM Settings only.

  • DECT Settings only.

By default, it is set to Full Factory Reset.

Perform Selected Reset

Executes the type of reset chosen.

Configure Hardware Button

Reset Type

Specifies the type of reset to perform via the web UI button below, the options are:

  • Full Factory Reset.

  • NVRAM Settings only.

  • DECT Settings only.

By default, it is set to Full Factory Reset.

Tools

Provision

Makes the phone trigger an instant provisioning.

Ping

Makes the phone ping an URL to check if it has access to it.

Traceroute

Checks the route packets take to the specified URL.

Maintenance Page Definitions

Phonebook Page Definitions

Global Phonebook XML Settings

Global Phonebook Type

Selects type of global phonebook to use.

  • If set to XML, DP755 will use the configuration in the Global Phonebook XML Settings page.

  • If set to LDAP, DP755 will use the configuration in the Global Phonebook LDAP Settings page.

  • If set to XSI, DP755 will use the configuration in the Global Phonebook XSI Settings page.

Automatic XML Phonebook Download

Enable Automatic XML Phonebook Download

Sends periodic requests to download XML Phonebook via HTTP, HTTPS, or TFTP.

HTTP/HTTPS User Name

Enters user name to authenticate with HTTP/HTTPS server.

HTTP/HTTPS Password

Enters password to authenticate with HTTP/HTTPS server.

Phonebook XML Server Path

Indicates server path to download XML phonebook file. This field could be IP address or URL, with up to 256 characters.

Phonebook Download Interval

Sets interval to send XML phonebook download requests (in minutes). If set to 0, automatic download is disabled. Valid range is 5 to 720. Default is 5 minutes.

Manual XML Phonebook Management

Import XML Phonebook

Upload: Uploads manually the global XML phonebook file to the base station.
Delete: Clears global XML phonebook file in the base station.

Export XML Phonebook

Downloads global XML phonebook from the base station in .xml format.

Global Phonebook LDAP Settings

Global Phonebook Type

Selects type of global phonebook to use.
If set to XML, DP755 will use the configuration in Global Phonebook XML Settings page. If set to LDAP, DP752 will use configuration in Global Phonebook LDAP Settings page.

LDAP Phonebook Settings

LDAP protocol

Chooses LDAP or LDAPS (LDAP over TLS) protocol. Default is LDAP.

Server Address

Configures IP address or domain name of the LDAP server.

Port

Determines LDAP server port. Default is 389.

Base

Indicates 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

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

Password

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

LDAP Filter

 LDAP filter to limit which contacts are fetched from the server. LDAP statement to limit which contacts are fetched from the server. Statement must be in parenthesis.

LDAP Version

Selects LDAP protocol version to send bind requests. Default is Version 3.

First Name Attribute

Defines first name attributes of each record to be returned in the LDAP search result.
This field allows users to configure multiple space-separated name attributes.
Example: gn
cn sn description

Last Name Attribute

Defines the last name attributes of each record to be returned in the LDAP search result.
This field allows users to configure multiple space-separated name attributes.
Example: gn cn sn description

Work Number Attribute

Specifies which LDAP attribute represent the contact’s work number. Must be in number attributes on LDAP server.

Home Number Attribute

 Specifies which LDAP attribute represent the contact’s home number. Must be in number attributes on LDAP server.

Mobile Number Attribute

Specifies which LDAP attribute represent the contact’s mobile number. Must be in number attributes on LDAP server.

Max. Hits

Specifies a maximum number of results to be returned by the LDAP server. If set to 0, the server will return all search results. The valid range is 1 to 3000.
The default is 500.

Search Timeout

Sets interval (in seconds) for the server to process the request and return search results to the client. Default is 30 seconds.

LDAP Contact Download Interval

Configures the download interval (in minutes). The base station will cache the results for the handsets to access. Valid range is 0 to 1440. If set to 0, will instead query LDAP whenever a handset accesses the contact list menu.
Default value is 5 minutes.

Global Phonebook Broadsoft XSI Settings

Broadsoft XSI
Note:
The broadsoft XSI settings can be applied independtly on all the 20 accounts supported by the DP755 base station.

XSI

Server

Configure 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

Configure the BroadWorks Xsi server port. The default port is 80. If the server uses HTTPS, please configure 443.

XSI Actions Path

configure the deployment path for Broadsoft XSI Actions. If it is empty, the path "com.broadsoft.xsi-actions" will be used.

Broadsoft Contact Download Interval

Configures the broadsoft phonebook download interval (in minutes). If set to 0, automatic download will be disabled. Valid range is 5 to 4320.
Default is 4320 Minutes.

XSI Authentication Type

This feature allows users to choose the type of authentification that will be used to access the XSI information from the handset, there are three types of authentifications:

  • Through Login credentials: Uses the log in Username and password for authentification

  • Through SIP credentials: Uses the SIP User ID, SIP Auth ID, and SIP Password for authentification

  • Match SIP Account: Uses the credentials used in the SIP account of the DP755.

Login Credentials

Login Username

Configure the Username for the BroadWorks XSI server.

Login Password

Configure the password for the BroadWorks XSI server.

SIP Credentials

SIP UserName

Configure SIP Username for the BroadWorks XSI server.

SIP User ID

Configure SIP User ID for the BroadWorks XSI server.

SIP Password

Configure SIP Password for the BroadWorks XSI server.

XSI Account Assignment

This setting can be applied individually for the 10 handsets supported by DP755.

Handset

Displays the handset on which the XSI setting will be applied

XSI Account

Sets the SIP account that will be used on the specific handset

Call Log Type

Sets which call logs will be displayed on the handset, there are two options:

  • Use XSI Call logs

  • Use Base Station Call Logs

Network Directories

Enable/Disable and choose the name of the following types:
Group Directory; Enterprise Directory; Group Common; Enterprise Common; Personal Directory.

Private Phonebook Settings

Upload XML Phonebook 1-10

Phonebook Name

Defines private phonebook name.

Import XML Phonebook

Upload: Uploads manually a private XML phonebook file to the base station.
Delete: Clears private XML phonebook file in the base station

Export XML Phonebook

Downloads private XML phonebook from the base station in .xml format.

Phonebook Page Definitions

Change Base Station Admin PIN code

For security reasons, advanced settings in the DP755 base station cannot be accessed from DP730/DP722 Handsets except if an Admin PIN code is provided. By default, the Admin PIN code is 0000.

We strongly recommend changing your Admin PIN code following below steps:

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings.
  3. Go to the DECT 🡪 General Settings tab.
  4. Enter your new Admin PIN Code (only digits accepted) in the appropriate field.
  5. Press Save and Apply to save your settings.

Register DP730/DP722 Handsets to DP755 Base Station

  1. On DP755 Base station, press and hold the Radio/Page button for 4 seconds until the Radio icon starts blinking to start the subscription process. Or Access web UI, and press Subscribe icon to Open Subscription.
  2. On DP730/DP722, press “Subscribe” softkey is available on the main screen or access Menu 🡪 Registration 🡪 Register while the DP755 Radio icon is blinking.
    Note: “Subscribe” softkey appears only if DP730/DP722 is not registered to any DP755 base station.
  3. Select BaseX (X=1-4) corresponding to the desired base station DP755, then press Subscribe.
  4. The DP730/DP722 will search for nearby base stations and will display the RFPI code and Base station name of the discovered DP755.
  5. Press Subscribe to pair with the displayed DP755.
  6. The DP730/DP722 will display Easy Pairing on the LCD and play an audible buzz when successful. Then it will return to the home screen, displaying the Handsets name and number assigned by the registered base station.
Registration Process
Note

DP755 supports 10 handsets registration and 16 simultaneous calls on 8 handsets.

Using DP730/DP722 with Multiple DP755 Base Stations

DP730/DP722 is able to be registered to up four different DP755 base stations.

Registering DP730/DP722 to an additional DP755 base station

Considering DP730/DP722 is previously registered to an initial base station, please follow below steps to register a Handsets to an additional base station:

  1. Press Menu (left softkey or the selection key) to bring up operation menu.
  2. Use arrow keys to reach Registration.
  3. Select Register.
  4. Navigate to an unsubscribed base using arrow keys, and click on Subscribe.
  5. Make sure that the subscription is opened on the new base station.
Multiple Base Stations Registration

Switching Between Different Base Stations

  1. Press “Menu” (left softkey or the selection key) to bring up the operation menu.
  2. Use arrow keys to reach Registration.
  3. Navigate to Select Base using the arrow keys.
  4. Select the desired base station and press Select.
Switching Between Base Stations

Unregister the DP730/DP722

  • Using DP730/DP722 Handsets:
  1. On DP730/DP722, press “Menu” (left softkey or the selection key) to bring up the operation menu.
  2. Press the arrow keys to move the cursor to Registration, then press “Select” (left softkey).
  3. Navigate to Deregister.
  4. Select the Handsets to be unregistered and press “Deregister” (left softkey).
  5. Enter the system PIN code (default: 0000).
  6. Press “Done” (left softkey) to confirm or “Back” (right softkey) to cancel.
  • Using DP755 Base Station UI:
  1. Access DP755 Web Interface.
  2. Go to Status 🡪 DECT Base Status.
  3. Locate the Handsets to unregister and press the “Unsubscribe” button.
Unregister DP730/DP722 from DP755 web UI

Locating DP730/DP722 Handsets from DP755 Base station

In some situations, you may have a DP730/DP722 Handsets incorrectly positioned and you don’t know its current location. You can locate a DP730/DP722 Handsets from his registered DP755 base station using the below steps:

Locate via DP755 Web UI

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings and navigate to the Status 🡪 DECT Base Status tab.
  3. Choose which Handsets to locate and press the corresponding Page button.
  4. A paging call will be received on the selected DP730/DP722 Handsets.
    Note: If you press Page All icon, all registered DP730/DP722 Handsets will be receiving the paging call.
  5. Once located, you can press End Softkey to end the paging call.
Locate Handsets via Web UI

Locate via DP755 Base station

  1. On the DP755 Base station front side, press Radio/Page button.
  2. All registered Handsets will receive Paging call.
  3. Once located, you can end the paging calling by pressing any key on the Handsets or by pressing again Radio/Page button.

Register a SIP Account

DP755 supports up to 20 SIP accounts, 10 Handsets. Each Handset can be configured with up to 20 accounts. Please be aware that line settings will be affected by DID settings (hunting group settings) in “DECT 🡪 SIP Account Settings”.

Register Account via Web User Interface

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings and navigate to the Accounts tab and select an account to use.
    Note: DP755 supports up to 20 accounts. An account is a set of settings including general settings, network settings, SIP settings, audio settings, call settings, and ring tones, etc.
  3. In General, Settings, set the following:
    • Account Active to Yes.
    • SIP Server field with your SIP server IP address or FQDN.
    • Failover SIP Server with your Failover SIP Server IP address or FQDN. Leave empty if not available.
    • Prefer Primary SIP Server to No or Yes depending on your configuration. Set to No if no Failover SIP Server is defined. If “Yes”, account will register to Primary SIP Server when failover registration expires.
    • Outbound Proxy with your Outbound Proxy IP Address or FQDN. Leave empty if not available.
    • SIP User ID User account information, provided by VoIP service provider (ITSP). Usually in the form of a digit similar to a phone number or actually a phone number.
    • Authenticate ID: SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or different from SIP User ID.
    • Authenticate Password: SIP service subscriber’s account password to register to SIP server of ITSP. For security reasons, the password will field will be shown as empty.
    • Name: Any name to identify this specific user.
  4. Press Save and Apply to save your configuration.
SIP Settings

5. Go to DECT 🡪 ACCOUNT ASSIGNEMENT

6. Configure your SIP details in the desired account:

  • Ringing Mode: Select the corresponding Ringing mode of the assigned account, which handset will ring when extension 1070 is called, selecting parallel will mean all selected handsets will be ringing at the same time.

7. Press Save and Apply to save your configuration.

Account Status

After applying your configuration, your phone will register to your SIP Server.

You can verify if your DECT phone has registered with your SIP server from your DP755 web interface under Status 🡪 Account Status (a green background with Yes under the SIP Registration column for the corresponding account indicates the account(s) has been successfully registered).

Accoun registered

Multiple Lines and Hunting Groups

The DP755 Base Station has the ability to assign 10 lines to each registered DP730/DP722 Handsets (Up to 10 Handsets) to receive/make calls.

When a Handset has many lines configured, users can select specific lines for outgoing calls using the Outgoing Default Line feature.

For incoming calls, users can choose either to redirect them to a specific Handset or to many using the Hunting Group feature so as to have the same phone number and incoming calls will be distributed in a Linear, Circular, or Parallel manner among the Handsets active in that Hunting Group. The number of hunting groups is limited by the number of SIP accounts registered to the base station (up to 20 accounts).

The hunting group feature is mainly used in office, warehouse, and call center environments to distribute incoming calls in the best way depending on the type of hunt group.

In order to configure hunting groups for DP730/DP722 Handsets registered to the Base, users need first to register SIP accounts on DP755 Base Station Account Settings and then assign accounts accordingly as lines for DP730/DP722 Account Assignment.

Account Assignment


This section will describe how to assign an account for each DP730/DP722 Handsets for making calls.

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings.
  3. Go to DECT 🡪 Account Assignment and assign to Handsets the SIP accounts already configured. Each Handset can be configured to use up to 20 SIP accounts.
Handsets Line Settings

After applying your configuration, the Account Status page will display the status of Handsets along with the accounts status. Each column shows one HS; each row shows if the account is assigned to an HS.

For example: If account 1 is assigned to the HS1, and HS10, the column of both handsets will be marked in brown color as shown in the screenshot below

Account Status

Outgoing Default Line

When a Handset is configured with more than one line, users can change the default outgoing line on DP730/DP722 Handsets using the keypad Menu 🡪 Preferences 🡪 Outgoing Default Line.

Outgoing Default Line

Hunting Groups

DP755 supports parallel hunting groups as described below:
In the examples below, we consider that all Handsets are in the same hunting group.

  • Parallel: In this mode, all phones ring concurrently. If one phone answers, the remaining available phones can make new outgoing calls.

This section will describe how to configure hunting groups for incoming calls:

The below steps are considering that SIP accounts were previously registered.

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings.
  3. Go to DECT 🡪 Account Assignement.
  4. Set Ringing Mode depending on your needs to configure your hunting groups.
  5. Press Save and Apply to save your settings.

Example:

In the example below Account 1 (1070) is assigned to HS1, HS2, HS3, and HS4, and the hunting group (HS Mode) is set to Parallel, so incoming calls to that account will make all handsets ring at the same time.

Hunting Group configuration

Configuration via Keypad

To configure the LCD menu using DP730/DP722’s keypad, follow the instructions below:

  • Register the DP730/DP722 to DP755. Please see Register DP730/DP722 Handsets to DP755 Base Station;
  • Enter/Confirm/ selection: Press the left softkey, right softkey, on-hook key or OK/Select key to enter the selected option, back to the last layer, or exit;
  • Exit: Press “right softkey” to exit to the previous menu;
  • Return to the Home page: Press the “On-hook” key to exit the main menu.
  • The DP730/DP722 automatically exits to main mode with an incoming call, when the phone is off the hook or left idle for more than 20 seconds.
  • When the phone is idle, pressing the DOWN navigation key can enter the Outgoing call log.

Please refer to DP730/DP722 Handsets Menu Structure for more details.

Call Features

The DP755/DP730/DP722 supports traditional and advanced telephony features including caller ID, caller ID with caller Name, call forward and etc.

*30

Block Caller ID (for all subsequent calls)

Off hook the phone;

Dial *30.

*31

Send Caller ID (for all subsequent calls)

Off hook the phone;

Dial *31.

*67

Call with Caller ID Blocked (per call)

Off hook the phone;

Dial *67 and then enter the number to dial out.

*82

Call with Caller ID Enabled (per call)

Off hook the phone;

Dial *82 and then enter the number to dial out.

*72

Unconditional Call Forward. To set up unconditional call forward:

Off hook the phone;

Dial *72 and then enter the number to forward the call;

Press the OK softkey or SEND key.

*73

Cancel Unconditional Call Forward. To cancel the unconditional call forward:

Off hook the phone;

Dial *73;

*90

Busy Call Forward. To set up a busy call forward:

Off hook the phone;

Dial *90 and then enter the number to forward the call;

Press the OK softkey or SEND key.

*91

Cancel Busy Call Forward. To cancel the busy call forward:

Off hook the phone;

Dial *91;

*92

Delayed Call Forward. To set up a delayed call forward:

Off hook the phone;

Dial *92 and then enter the number to forward the call;

Press the OK softkey or SEND key.

*93

Cancel Delayed Call Forward. To cancel the delayed call forward:

Off hook the phone;

Dial *93;

DP755 Phonebook Management

DP755/DP730/DP722 support Private and Global Phonebooks; both phonebook types can be used at same time:

Private Phonebook

A private phonebook allows you to manage your contacts on each registered handset; each handset can have his own private phonebook with his own contacts. DP755 supports up to 10 private phonebooks.

A private phonebook can be assigned to one or more handsets registered to the base.

The following steps explain how to upload your private phonebook and assign it to a specific Handsets:

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings and go to Phonebook 🡪 Private Phonebook Settings.
Private Phonebook Settings
  1. In the Private XML Phonebook X section (X from 1 to 10):
    1. Enter Phonebook X Name (default value is PB1 for first Handset, PB2 for second Handset, etc.).
    2. Press the Upload button to Import XML Phonebook X.
    3. Browse your computer files and select your desired phonebook.xml file.
    4. Press Save and Apply to save your settings.
  2. Go to the DECT 🡪 General Settings tab.
  3. In the Handsets Settings section, select your Handsets Phonebook to assign it to a specific Handsets as shown below where PB1 is assigned to HS1, PB2 is assigned to HS2

You can assign the same Private Phonebook to more than one Handsets.

For example, we can assign Handsets Phonebook named PB1 to HS1 and HS2.

Any change in PB1 contacts will be applied to both HS1 and HS2 private phonebooks.

  1. Press Save and Apply to save your configuration.

After applying your configuration, your DP730/DP722 Handsets will display uploaded phonebook contacts. You can access your private phonebook by pressing Contacts on your DP730/DP722 Handsets. You can press Option Softkey in order to view, Create New Contact or Edit Dial if adding changes on a contact number is required before dialing.

Global Phonebook

Global phonebook allows you to manage contacts and use them in all registered Handsets. The contacts can be imported either via XML or via LDAP. Follow the steps below to upload your shared phonebook:

Global Phonebook via XML

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings.
  3. Go to Phonebook 🡪 Global Phonebook XML Settings tab.
  4. Set Global Phonebook Type to XML (in this case, LDAP phonebook will not be available).
Global Phonebook XML Settings

5. There are two methods to import/download your XML Phonebook:

  • Automatic XML Phonebook Download
    For this method, you need to use a TFTP or HTTP, or HTTPS server and make your phonebook.xml file available on your preferred server.
    1. Set Enable Automatic XML Phonebook Download to Enabled, use TFTP/HTTP or HTTPS depending on your server.
    2. If using HTTP or HTTPS server and User Name and Password are required to connect to the server, set HTTP/HTTPS User Name and HTTP/HTTPS Password fields with appropriate values.
    3. Configure Phonebook XML Server Path field. This field could be an IP address or URL, with up to 256 characters. The phone will request a file named phonebook.xml from the provided directory. Example: server_URL/directory
    4. Configure the Phonebook Download Interval (in minutes) to periodically contact your server to download a new phonebook file version if available. If set to 0, automatic download will be disabled. The valid range is 5 to 720.
Automatic XML Phonebook Download
  • Manual XML Phonebook Management
    1. Press Upload in Import XML Phonebook.
    2. Browse your files and select your phonebook.xml file.
Manual XML Phonebook Management

XML Phonebook file format

<?xml version="1.0" encoding="UTF-8"?>
<AddressBook>
   <Contact>
        <FirstName>First name</FirstName>      
        <LastName>Last name</LastName>         
        <Ringtone>Ringtone ID (default 0)</Ringtone>
        <Phone type="Home">
            <phonenumber>Home phone number</phonenumber>
       </Phone>
        <Phone type="Work">
            <phonenumber>Work phone number</phonenumber>
       </Phone>
        <Phone type="Mobile">
            <phonenumber>Mobile phone number</phonenumber>
        </Phone>
   </Contact> 
</AddressBook>

Object

Position

Type

Values

Comments

AddressBook

Root element

Mandatory

Root element of the XML document

Contact

Child element

Mandatory

Each contact is an entry

LastName

Child element

At least one of them present

String

Last name of the contact

FirstName

Child element

String

First name of the contact

Phone

Child element

Mandatory

Phone number

PhoneNumber

Child element

At least one present

Int

Type=”Home” or Type=”Work” or Type=”Mobile”

XML Phonebook Example:

<?xml version="1.0" encoding="UTF-8"?>
<AddressBook>
   <Contact>
        <FirstName>John</FirstName>
        <LastName>Doe</LastName>
        <Ringtone>0</Ringtone>
        <Phone type="Home">
            <phonenumber>1000</phonenumber>
       </Phone>
        <Phone type="Work">
            <phonenumber>1001</phonenumber>
       </Phone>
        <Phone type="Mobile">
            <phonenumber>1002</phonenumber>
        </Phone>
   </Contact>
   <Contact>
        <FirstName>Alice</FirstName>
        <LastName>Beck</LastName>
        <Ringtone>0</Ringtone>
        <Phone type="Home">
            <phonenumber>2000</phonenumber>
       </Phone>
        <Phone type="Work">
            <phonenumber>2001</phonenumber>
       </Phone>
        <Phone type="Mobile">
            <phonenumber>2002</phonenumber>
        </Phone>
   </Contact> 
 </AddressBook>

Global Phonebook via LDAP

  1. Access the Web GUI of your DP755 using the admin’s username and password.
  2. Press Login to access your settings.
  3. Go to Phonebook 🡪 Global Phonebook LDAP Settings tab.
  4. Set Global Phonebook Type to LDAP (in this case, XML phonebook will not be available).
Global Phonebook LDAP Settings

5. Under LDAP Phonebook Settings, set your LDAP parameters to connect to your LDAP server.
Refer to [Phonebook Page Definitions] for parameters explanation.

6. Press Save and Apply to save your configuration.

  • Example of configuration:
LDAP protocol: LDAP
Server Address:  192.168.1.100
Port: 389
Base: dc=pbx,dc=com
User Name:
Password:
LDAP Filter: (mobile=%)(sn=%)
LDAP Version: Version 3
First Name Attribute: sn
Last Name Attribute: cn
Work Name Attribute:
Home Name Attribute:
Mobile Number Attribute: mobile
Max. Hits: 500
Search Timeout: 30

After applying your configuration, your global phonebook will be synchronized with all registered Handsets and contacts will be displayed on your DP730/DP722 Handsets screens.

DP755 ASSOCIATION WITH DP760 DECT REPEATER

Important Note

  • DP760 can relay up to 2 concurrent calls.
  • After pressing the Page/Reset button for more than 2 seconds on the DP760, it will enter AUTO region mode, in this mode, the three LEDs on right side keep quickly blinking, then the DP760 will search the base signal in the current environment to auto associate with it and then auto switch to the same region (EU, US or BR) of the base station.
  • If you have a DP760 that has FW before 1.0.3.34, you will need upgrade it to 1.0.3.34 first, then do a factory reset. After that, the unit will support the Auto-Region feature and it will enter Auto-Region mode.
  • DP755 does NOT include the DP760 model firmware, it can only work as an extended range for the DP755 but is not part of the base station’s firmware releases.