DP755 Cordless VoIP Base Station

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Product Information

Specifications

Air Interface| 3 LED indicators: Power, Network, DECT. Pairing/Paging button.
One 10/100 Mbps auto-sensing Ethernet port with integrated PoE
---|---
Peripherals| Base unit, Universal Power Supply, Ethernet cable, Quick
Installation Guide, GPL Statement
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.711/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)

Product Usage Instructions

Setting Up the Base Station DP755:

1. Connect the base station to power using the universal power
   supply.

2. Connect the base station to your network using the Ethernet
   cable.

3. Follow the Quick Installation Guide for initial setup
   steps.

Pairing Handsets DP722/DP730:

1. Ensure the handsets are charged.
2. Press the Pairing/Paging button on the base station.
3. Follow the on-screen instructions on the handsets to complete
   pairing.

Making Calls:

1. Select the desired SIP account on the handset.
2. Dial the number using the keypad.
3. To end a call or switch between calls, use the handset’s
   controls.

FAQ

Q: How many concurrent calls are supported by DP755?

A: DP755 supports up to 8 concurrent calls.

Q: Can DP730 and DP722 handsets be used interchangeably with

DP755?

A: Yes, both DP730 and DP722 handsets are compatible with DP755
base station.

Q: How far is the outdoor range supported by DP755?

A: The outdoor range is up to 400m with DP730 and up to 350m
with DP722/DP720.

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Grandstream Networks, Inc.
DP755 $GPLQLVWUDWLRQ*XLGH

DP755 – Administration Guide
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

DP730

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

DP722

DECT cordless HD. 2.4 inch (240×320) 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

DECT cordless HD. 1.8 inch (128×160) 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
Peripherals Protocols/Standards Voice Codecs Telephony Features QoS Security Multi-language Upgrade/ Provisioning Multiple SIP Accounts

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
3 LED indicators: Power, Network, DECT. Pairing/Paging button. One 10/100 Mbps auto-sensing Ethernet port with integrated PoE
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
G.711/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
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, exible dial plan, server redundancy and fail-over
Layer 2 QoS (802.1Q, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
User and administrator level access control, MD5 and MD5-sess based authentication, 256-bit AES encrypted con guration le, TLS, SRTP, HTTPS, 802.1x media access control, DECT authentication & encryption
Chinese Simple, Chinese Tradition, Czech, Danish, Dutch, English, Estonian, Finnish, French, German, Hebrew, Hungarian, Japanese, Korean, Norwegian, Portuguese, Romanian, Spanish, Swedish, Turkish.
Firmware upgrade via HTTP/HTTPS or FTP/FTPS, mass provisioning using TR-069 or AES encrypted XML con guration le
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 Power & Green Energy E ciency Package Content Dimensions (H x W x D) Weight
Temperature and Humidity
Compliance

Parallel Mode: All phones ring concurrently; after one phone answers, the remaining available phones can make new calls
Universal Power Supply Input AC 100-240V 50/60Hz; Output 5VDC, 1A; Micro-USB connection; PoE: IEEE802.3af Class 1, 0.44W­3.84W
Base unit, Universal Power Supply, Ethernet cable, Quick Installation Guide, GPL Statement
to be de ned.
Base unit: 140g; Universal power supply: 50g; Package: 370g
Operation: -10º to 50ºC (14 to 122ºF); Storage: -20º to 60ºC (-4 to 140ºF); Humidity: 10% to 90% non-condensing
FCC: FCC Part 15B; FCC Part 15D; SAR (FCC 47 CFR Part2.1093; IEEE 1528; IEC 62209-2); FCC Part68 HAC; FCC ID CEEN 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 RCMAS/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
Protocols/Standards Voice Codecs Telephony Features HD Audio Security Multi- language Upgrade/ Provisioning Multiple Line Access Power & Green Energy E ciency Package Content Dimensions (H x W x D)
Weight
Temperature and Humidity
Compliance

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-toTalk 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
Hearing Aid Compatibility (HAC) compliant
G.722 codec for HD audio and G.726 codec for narrow band audio (G.711/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
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, exible dial plan
Yes, in both Handsets and Speakerphone modes
DECT authentication & encryption
Chinese Simple, Chinese Tradition, Czech, Danish, Dutch, English, Estonian, Finnish, French, German, Hebrew, Hungarian, Japanese, Korean, Norwegian, Polish, Portuguese, Romanian, Spanish, Turkish.
Software Upgrade Over-The-Air (SUOTA), handsets provisioning Over-The-Air
Each handset may access up to 10 lines
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)
Handset unit, universal power supply, charger cradle, belt clip, 1 battery, Quick Installation Guide
Handset: 168.5 x 52.5 x 21.8mm; Charger cradle: 76 x 73 x 81mm
Handset: 180g; Charger cradle: 78g; Universal power supply: 50g; Package: 465g
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
FCC: FCC Part 15B; FCC Part 15D; SAR (FCC 47 CFR Part2.1093; IEEE 1528; IEC 62209-2); FCC Part68 HAC; FCC ID CEEN 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 RCMAS/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 Voice Codecs Telephony Features HD Audio Security Multi- language Upgrade/ Provisioning Multiple Line Access Power & Green Energy E ciency Package Content Dimensions (H x W x D)
Weight
Temperature and Humidity
Compliance

Hearing Aid Compatibility (HAC) compliant
G.722 codec for HD audio and G.726 codec for narrow band audio (G.711/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
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, exible dial plan
Yes, in both Handsets and Speakerphone modes
DECT authentication & encryption
Chinese Simple, Chinese Tradition, Czech, Danish, Dutch, English, Estonian, Finnish, French, German, Hebrew, Hungarian, Japanese, Korean, Norwegian, Polish, Portuguese, Romanian, Spanish, Turkish.
Software Upgrade Over-The-Air (SUOTA), handsets provisioning Over-The-Air
Each handset may access up to 10 lines
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)
Handset unit, universal power supply, charger cradle, belt clip, 1 battery, Quick Installation Guide
Handset: 158 x 50 x 28.1mm; Charger cradle: 81.15 x 75.89 x 36.36mm
Handset: 110g; Charger cradle: 44g; Universal power supply: 50g; Package: 328g
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
FCC: FCC Part 15B; FCC Part 15D; SAR (FCC 47 CFR Part2.1093; IEEE 1528; IEC 62209-2); FCC Part68 HAC; FCC ID CEEN 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 RCMAS/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

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

DP755 Package Content Equipment Packaging ­ DP730

DP722 DP722

1 Handset unit 1 Universal power supply 5V 1 Charging station 1 Handset Belt 2 Rechargeable batteries 1 Quick Installation Guide

DP730 Package Content

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.

Battery Information
DP722 Batteries Speci cations
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

Setting up the DP730/DP722
DP730 battery speci cations
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.

LED Light

DP755 LED Patterns
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.

Key

1

Earphone

2

LED Indication

3,5

Left and right softkeys

4

LCD display

6

4 Arrow key combination

7

Men/Ok key

8

Off-hook / Dial key

9

On-hook / Power key

10

Alphanumeric Keypad

11

/ Lock key

12

Mute key

13

Hands-free / Speaker key

14

Microphone

DP722 Keys Description
Description Delivers audio output. Red: Charging. Green: Charge completed. Blinking: Missed call(s) or Voice Mail received. Correspond to functions displayed on the LCD. These functions change depending on the current context. Shows call information, handset status icons, prompt messages, etc. Permits navigation of the cursor through the displayed menu options. Selects the option chosen by the cursor. (Enters the main menu from the home screen.) Enters dialing mode, or dials number entered. Terminates calls or turns the handset on / off. Provides the digits, letters, and special characters in context-sensitive applications. For the + sign, press and hold key 0. 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. Activates or deactivates the mute feature. Switches between handset and hands-free/speaker modes. Picks up audio earpiece and hands-free calls.

Key

1

Proximity sensor

2

Earphone

3,4

Volume up / Down Keys

DP730 Keys Description
Description The proximity sensor can detect and measure gravitational acceleration, tilt, vibration, altitude changes, and static position. Delivers audio output. Con gure the handset and ringtone volume.

5

PTT Key

6

Hands-free / Speaker key

7

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

8

Off-hook / Dial key

9

Alphanumeric Keypad

10

• / Silent Mode key

11

LED indicator

12

3.5 mm headset jack

13

Color LCD Screen

14

Softkeys

15

Mute

16

Menu/OK key

17

On-hook or Power key

18

/ Lock key

19

Microphone

PTT (Push-to-Talk) button, to initiate a PTT call. Switches between Handset and Hands-free / Speaker modes.
Allows navigation of the cursor through the displayed menu options. Enters dialing mode, or dials number entered.
Provides the digits, letters, and special characters in context-sensitive applications. For the + sign, press and hold key 0.
Activates or deactivates the silent mode (no ringtone heard during an incoming call) when keep pressing on * in idle screen.
1 dual-color LED indicator indicating power, call, battery, message waiting… Phone connector for the headphones/headsets.
2.4-inch (240×320) TFT color LCD Correspond to functions displayed on the LCD. These functions change depending on the current context.
Mute the microphone during the conversation. Selects the option chosen by the cursor or enter the main menu from the home screen.
Terminates calls or turns the handset on / off. 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.
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 noti cation
Outgoing Call noti cation
Missed Call noti cation
Voicemail noti cation
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.

Contacts Call History Registration Voice Mail
Preferences
Customizing keys functions

DP730/DP722 Menu Structure
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.
Display the call history:
1. Missed Calls. 2. Accepted Calls. 3. Outgoing Calls. 4. All Calls. Note: You can add contacts to Shared Contacts directly from call logs.
Register: Register your handset to base station. Deregister: Deregister your handset from base station. Select Base: Select base station.
Play Message: Play voice mail messages received. Set Voice Mail: Con gure voice mail parameters. Set Key 1: Con gure Key 1 as VM speed dial for selected account.
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:
L : Con gure Left Softkey function in idle. Function can be set as Menu, History, Contacts, Line or PTT. Default is Menu. R : Con gure Right Softkey function. Function can be set as History, Contacts, Line or PTT Default is Contacts. : Con gure Arrow UP Key function. Default is Outgoing Calls (Call History). : Con gure Arrow DOWN key function. Default is Accepted Calls (Call History). : Con gure Arrow LEFT key function. Default is Ringer Volume Down.

Call Features

: Con gure Arrow RIGHT key function. Default is Ringer Volume Up. Select key and press OK button to con gure 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.
Do Not Disturb: Enable/disable do not disturb mode on the phone. Call Forward: Con gure call forward feature. Call Waiting: Con gure call waiting feature. Paging: Con gures 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: Con gure date and time on the Handsets. Audio & Vibration: Specify ringtones for internal/external calls, the volume, advisory tones (Keypad, Con rmation, Low battery noti cations), Handset TX Gain (-6dB, 0dB, +6dB), and Vibration
mode (DP730 only). Display: Con gure backlight, LCD timeout (Idle/Call), LCD brightness, Message Waiting Prompt and menu key timeout. Gestures (DP730 only): Con gure Close-to-Ear Backlight and Facedown Hangup. Network Settings: Con gure IP addresses and select DHCP/Static IP mode. SIP Settings: Con gure/View SIP accounts settings. System Settings: Change Base PIN code, perform factory reset, reboot base and con gure repeater mode. Firmware Upgrade: Upgrade the rmware 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 ne. 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 “Select” (left softkey) or OK button, then select Base Status.
3. Using Arrow keys, navigate down to view the IP address of the DP755.

, then press

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:

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

Icons Bar Shortcut

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 End User Level Administrator Level

Username Password

user

123

admin

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

Web Pages Allowed Only Status, Settings, and Maintenance 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.

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 User Level Password

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

Changing Admin Level Password

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 SIP User ID SIP Server SIP Registration Ringing Mode
HS status table
DECT Base Status Base Station Name Base DECT FW Version Base DECT RF Region Base DECT RFPI Address Global Functions Handset

Displays list of con gured accounts’ names, from Account 1 to Account 10.
Displays list of SIP user id registered.
Displays list of SIP Server.
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.
Displays the HS mode con gured for each account.
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 con gured 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 con gured, 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.
Displays name of base station. Default is DP755_[last 6 digits of MAC address].
Shows rmware version of base DECT.
Indicates region of base DECT RF.
Speci es DECT RFPI (Radio Fixed Part Identity) address which is a unique identity for the base.
Displays Global Information about upgrading handsets.
Displays Handset index.

Name IPEI
Page Unsubscribe HS Firmware Upgrade Line Options Account SIP User ID DND Forward Busy Forward Delayed Forward Network Status MAC Address IP Setting IPv4 Address IPv6 Address OpenVPN® IP Subnet Mask Gateway Primary DNS Secondary DNS PPPoE Link Up NAT Type NAT Traversal System Info Product Model

Displays Handsets names 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:
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. Unsubscribes corresponding handset from DECT base station. Indicates Handsets’ rmware version number. Shows Handsets upgrade status or trigger handset upgrade process.
Account index. Displays the con gured SIP User ID for the account. DND status of the account. Default No. The unconditional forward number. The forward number for Call Forward Busy. The no answer delayed forward number.
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. Indicates used IP address mode: DHCP, Static IP or PPPoE. Displays assigned IPv4 address. Example: 192.168.5.110 Displays assigned IPv6 address. Example: 0:0:0:0:0:ffff:c0a8:056e Displays OpenVPN® IP address. Displays assigned subnet mask. Example: 255.255.255.0 Displays assigned default gateway. Example: 192.168.5.1 Shows assigned Primary DNS server address. Example: 8.8.8.8 Shows assigned Secondary DNS server address. Example: 8.8.4.4 Indicates PPPoE connection status. Displays the NAT type enabled. Indicates the type of NAT for each account ( 20 accounts in total).
Displays product model info. Default is DP755.

Part Number Cert cate Type
Software Version

Shows product part number. Example: 9610006512A (last 2 digits show HW version, in this example 12A for HW version 1.2A)
Displays the Certi cate type installed on the device.
Boot: Speci es Boot version. Core: Speci es Core version. Prog: Speci es Prog version, This is the main rmware release number, which is always used for identifying the software system of the DP755. Handset: Speci es Handset rmware version.

IP Geographic Information
Special Feature System Up Time System Time Service Status System Information User Space Core Dump

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.
Displays wether the OpenVPN® support is enabled or not. Indicates system uptime since last reboot. Shows actual time and date according to your con guration. Reveals status of VoIP applications. Gives the option to download System information Shows User sapce used and the database status Generates core dump by killing programs gs_cmbs and gs_phone.
Status Page Definitions

Accounts Page Definitions

General Settings Account Active Account Name SIP Server Secondary SIP Server Outbound Proxy Backup Outbound Proxy SIP User ID Authenticate ID Authenticate Password Name Voice Mail Access Number Network Settings
DNS Mode
Maximum Number of SIP Request Retries
DNS SRV Failover mode
Failback Timer Register before DNS SRV failover Primary IP Backup IP 1

Activates or deactivates SIP Account.
Determines the name of the account, this account name can also be used in Handsets con g provisioning for validation.
Con gures 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.
Speci es 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.
Speci es the IP address or domain name of an outbound proxy, a media gateway, or a session border controller. Used by DP755 for rewall 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.
IP address or Domain name of the Secondary Outbound Proxy which will be used when the primary proxy cannot be connected.
User account information, provided by your VoIP service provider.
SIP service subscriber’s Authenticate ID is used for authentication. It can be identical to or different from the SIP User ID.
The account password required for the phone to authenticate with the SIP server before the account can be registered.
The SIP server subscriber’s name (optional) that will be used for Caller ID display (e.g., John Doe).
Allows users to access voice messages by pressing the MESSAGE button on the phone. This value is usually the VM portal access number.
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 nd services for various protocols. NAPTR/SRV: Naming Authority Pointer according to RFC 2915. Use Con gured IP: If selected, please ll in Primary IP, Backup IP 1 and Backup IP 2 to be used for server look up.
Speci es the maximum number of retries on SIP Requests. Default value is 2.
Con gures the preferred method for DNS SRV failover. If “Default” is selected, the primary SIP server or Outbound Proxy will always be attempted rst for all REGISTER and INVITE requests. If “Use current server until DNS TTL” is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until DNS times out. If “Use current server until no response” is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until there is no response. If “Use current server until failback timer expires” is selected, the device will send all SIP messages to the current failover SIP server or Outbound Proxy until the failback timer expires.
Speci es the time interval (in minutes) that the device should continue to send SIP requests (REGISTER or INVITE) to the failover IP. Once it expires, the SIP requests will be sent to the preferred IP. The default value is 60 minutes, and the max value is 45 days.
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.
Speci es primary IP address where the base sends DNS query to, when “Use Con gured IP” is selected for DNS mode.
Speci es backup IP 1 address where the base sends DNS query to, when “Primary IP” is not responding.

Backup IP 2
NAT Traversal
Proxy-Require SIP Settings Basic Settings
TEL URI
SIP Registration Unregister on Reboot Register Expiration Subscribe Expiration Reregister Before Expiration Enable OPTIONS Keep Alive OPTIONS Keep Alive Interval OPTIONS Keep Alive Max Lost
Enable TCP Keep Alive
Local SIP Port SIP Registration Failure Retry Wait Time SIP T1 Timeout SIP T2 Timeout SIP Transport SIP Listening Mode SIP URI Scheme When Using TLS Use Actual Ephemeral Port in Contact TCP/TLS
Outbound Proxy Mode Support SIP Instance ID SUBSCRIBE for MWI SUBSCRIBE for Registration Enable 100rel Callee ID Display

Speci es backup IP 2 address where the base sends DNS query to, when “Backup IP 1” is not responding.
Enables/disables NAT traversal mechanism. If activated (by choosing “STUN”) and a STUN server is also speci ed (Maintenance Network Settings STUN Settings); the base performs according to STUN client speci cation. Under this mode, embedded STUN client will detect if and what type of rewall/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 eld 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.
Determines a SIP Extension to notify the SIP server that the base is behind a NAT/Firewall.
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.
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.
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”.
Speci es the frequency (in minutes) when the phone refreshes its registration with the speci ed registrar. The maximum value is 64800 (about 45 days). The default value is 60 minutes.
Speci es the frequency (in minutes) when the phone refreshes its subscription with the speci ed registrar. The maximum value is 64800 (about 45 days). The default value is 60 minutes.
Sends re-register request after speci c time (in seconds) to renew registration before the previous registration session expires.
Enables OPTIONS Keep Alive, to check SIP server. Default is No.
Time interval for OPTIONS Keep Alive feature in seconds. Range of values is 1­64800. Default is 30.
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.
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
The parameter speci es 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.
Sends re-register request after speci c time (in seconds) when registration process fails. Maximum interval is 3600 seconds (1 hour). Default is 20 seconds.
De nes 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 (2T1) and then (4*T1). Request re- transmit retries would continue until a maximum amount of time de ned by T2. Default is 0.5 seconds.
Identi es 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.
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.
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.
Speci es if “sip” or “sips” will be used when TLS/TCP is selected for SIP Transport. The default setting is “sips”.
De nes 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”.
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.
Adds “SIP Instance ID” attribute to “Contact” header in REGISTER request as de ned in IETF SIP outbound draft. Default is Yes.
Sends periodic “SUBSCRIBE” requests (depends on “Register Expiration” parameter) for message waiting indication service. Default is No.
When set to “Yes”, a SUBSCRIBE for Registration will be sent out periodically.
Appends “100rel” attribute to the “required” header of the initial signaling messages. Default is No.
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 con gured 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 con gured on the base. With the credentials of the SIP account if con gured on the base. By default, it is set to “No”.

Ignore Alert-Info Header

This option is used to con gure default ringtone. If set to “Yes”, con gured 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

De nes 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

Speci es 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

Speci es 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 Certi cates

De nes whether the domain certi cates will be checked when TLS/TCP is used for SIP Transport.

Validate Certi cate Chain Validate Incoming Messages Check SIP User ID for Incoming INVITE Accept Incoming SIP from Proxy Only Authenticate Incoming INVITE Audio Settings Preferred Vocoder- Choice x Use First Matching Vocoder in 200OK SDP Codec Negotiation Priority Disable Multiple m line in SDP
SRTP Mode
SRTP Key Length Crypto Life Time Symmetric RTP Silence Suppression Jitter Buffer Type Jitter Buffer Length Voice Frames per TX G726-32 Packing Mode iLBC Frame Size iLBC Payload type OPUS Payload Type DTMF Payload Type Send DTMF Call Settings Dial Plan

Disabled by Default
Validates certi cate chain when TLS/TCP is con gured. Disabled by Default.
De nes whether incoming messages will be validated or not. Default is No.
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.
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.
Challenges the incoming INVITE for authentication with SIP 401 Unauthorized message. Default is No.
Con gures 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
Includes only the rst matching vocoder in its 200OK response, otherwise it will include all matching vocoders in same order received in INVITE. Default is No.
Con gures 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”.
If enabled, the phone always responds to 1 m line in SDP regardless multiple m lines are offered. Disabled by Default.
Selects the SRTP mode to use (“Disabled”, “Enabled but not forced”, “Enabled and forced”, “Follow SIP Transport”, or “Optional”).
1. Disabled: SRTP is not used at all. Voice communication will be transmitted without encryption or authentication, which may pose a security risk. 2. 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. 3. Follow SIP Transport: When “Follow SIP Transport” is chosen, the SRTP protocol follows the transport protocol speci ed in the SIP signaling messages to determine when to start and
stop the encryption and decryption of the RTP tra c. 4. 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.
The cipher method / key length to use if SRTP is enabled.
Adds crypto lifetime header to SRTP packets. The default is Yes.
De nes whether symmetric RTP is supported or not. The default setting is “No”.
Allows detection of the absence of audio and conserves bandwidth by preventing the transmission of “silent packets” over the network. The default is No.
Selects either Fixed or Adaptive based on network conditions. Set to Adaptive by Default.
Selects Low, Medium, or High based on network conditions. Set to 300ms by Default
Transmits a speci c number of voice frames per packet. Default is 2; increases to 10/20/32/64 for G711/G726/G723/other codecs respectively.
De nes G726-32 packing mode (“ITU” or “IETF”). Default is ITU.
Speci es iLBC packet frame size (20ms or 30ms). Default is 20ms.
Determines payload type for iLBC. The valid range is between 96 and 127. Default is 97.
Determines OPUS payload type. The valid range is between 96 and 127. Default is 123.
Con gures the payload type for DTMF using RFC2833. Cannot be the same as iLBC or OPUS payload type
Speci es the mechanism to transmit DTMF digits. you can choose to do it in- audio, via RTP(RFC2833), or via SIP INFO
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 pre x 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/O ce 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 prede ned 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

Con gure 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

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

Blind Transfer Wait Timeout

De nes 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

Speci es 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: · Speci c caller ID number. For example, 8321123; · A de ned 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 con gure 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 speci c period of time. Default is 60 seconds.

Intercom Settings

Allow Auto Answer by CallInfo/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 de ned.

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-con gured feature codes will be used only if server-provisioned feature codes are not provided. Enabled By Default

Account Swap

Swap Account Settings

Swap con gurations between two accounts.

Accounts Page Definitions

DECT Page Definitions

General Settings

Base Station Name Admin PIN Code Enable Repeater Mode
Enable Repeater Management
DECT PTT Silence Timer Collapse Call History Clear Call Logs Account Assignment SIP User ID Ringing Mode HS1-HS10

Displays the name of the base station. The default is DP755[last 6 digits of MAC address].
Con gures admin PIN code for authentication. Default is 0000
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.
Enables base station network management of discovered and paired repeaters. Once enabled, users need rst 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.
Sets timeout for PTT call (in minutes) if no handset unmutes. If set to 0, this timer will be disabled.
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.
Deletes call history logs of all handsets from base station
displays the SIP User ID assigned to the account number.
Speci es the ringing mode for the account, allowing either parallel ringing or ringing only on the selected handset.
Within the handset matrix, you can designate speci c 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 con gured 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 con gured 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 speci c handset from the ringtones available on the base station. It takes up to 10 ringtone les 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 prede ned 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 Local RTP Port Range

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

Speci es 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 con gs the speci c 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

Speci es 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

Speci es the account on which the service will be applied.

System Identi cation

Speci es the name to identify the service.

System Number

Speci es the system number, in case the service type option is set to GDS, the system number is the SIP user ID con gured 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 con gured on “Remote PIN to Open the Door” eld 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 Pre x

Con gures the pre x in the User-Agent header.

Settings => PTT/Multicast

PTT multicast address

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

PTT Con g

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 Con g (1-25)

The user can manually con gure 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

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

Secondary NTP Server

De nes 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

Con gures the date/time used on the phone according to the speci ed time zone. Enabled by Default.

Self-De ned Time Zone

This parameter allows the users to de ne 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

Con gures tone frequencies according to user preference. By default, the tones are set to North American frequencies. Frequencies should be con gured 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/ les/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

Con gure 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

Con gure 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

Con gure 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

Con gure 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

Con gure 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 IPv4 Address

Selects which Internet protocol to use. When both IPv4 and IPv6 are enabled, phone attempts to use preferred protocol rst and switches to the other choice if it fails. Set to IPv4 Only by Default.
Select IP address mode (DHCP, Static IP, or PPPoE) for DP755 Base Station.

Hostname (Option 12)

Speci es the name of the client. The name may or may not be quali ed with the local domain name. This eld is optional but may be required by ISP.

Vendor Class ID (Option 60)

Exchanges vendor class IDs by clients and servers to convey particular con guration or other identi cation 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

Speci es 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-con gured (DHCP), or statically con gured.

Full Static

De nes the Static IPv6 Address, and the IPv6 Pre x length

Pre x Static

De nes the IPv6 Pre x (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

Speci es 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

Con gures the identity for 802.1X mode.

MD5 Password

Determines the MD5 password for 802.1X mode.

802.1X CA Certi cate

Uploads / deletes the 802.1X CA certi cates.

802.1X Client Certi cate

Uploads / Deletes the 802.1X Client Certi cates.

HTTP Proxy

Speci es 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

Speci es 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

De nes 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

De nes 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-con gure for VLAN settings through DHCP. Disabled by Default.

Enable Manual VLAN Con guration

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

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

Maximum Transmission Unit (MTU)

De nes the MTU in bytes. Default is 1500.

Network Settings ­ Management Virtual IP Address Settings

Management Virtual IP address

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

Management Access

Allow management of the device through virtual IP address or both actual IP and virtual IP addresses, you can choose one of the two options:
1. Management Virtual IP Address only: In this scenario, the device is accessible for management purposes (con gurations, monitoring, updates, etc.) solely through a virtual IP address. This means that regardless of the physical IP address of the device, users interact with it using the virtual IP address. This approach is often used for enhancing exibility and redundancy in network management.
2. Both actual IP and virtual IP addtesses: This option allows management of the device through both its actual (physical) IP address and a virtual IP address. Users have the exibility to choose either the physical IP address or the virtual one to access and manage the device. Having both options can be useful in scenarios where redundancy and failover mechanisms are crucial. It ensures that even if the physical IP address becomes inaccessible or changes, administrators can still manage the device through the virtual IP address.

SNMP Through Management Interface TR069 Through Management Interface SYSLOG Through Management Interface Layer 2 QoS 802.1Q/VLAN Tag Layer 2 QoS 802.1p Priority Value IP Address Mode Static IP Settings IPv4 Address Subnet Mask Gateway DNS Server 1 DNS Server 2 Network Open VPN® Settings
OpenVPN® Enable
OpenVPN® Server Address OpenVPN® Port OpenVPN® Transport OpenVPN® CA OpenVPN® Certi cate OpenVPN® Client Key OpenVPN® Cipher Method OpenVPN® Username OpenVPN® Password Additional Options
Network => SNMP Settings Enable SNMP
Version
Port Community SNMP Trap Version SNMP Trap IP SNMP Trap Port SNMP Trap Interval SNMP Trap Community SNMP Username

Set to “Management Virtual IP Address only” by Default Enable or disable snmp through the management interface. Disabled by default. Enable or disable tr069 through the management interface. Disabled by default. Enable or disable syslog through the management interface. Disabled by default. Assigns the VLAN Tag of the Layer 2 QoS packets. Assigns the priority value of the Layer 2 QoS packets. Valid range is 0 to 7. The IPv4 address obtained on the phone using which method: DHCP or Static IP assignement.
Enter the IP address when static IP is used. Enter the Subnet Mask when static IP is used. Enter the Default Gateway when static IP is used. Enter DNS Server 1 when static IP is used. Enter DNS Server 2 when static IP is used.
Enables/Disables the OpenVPN® feature. Default settings is No. Con gures the address of the OpenVPN® server. De nes the port of the OpenVPN® server. Default is 1194. Determines network protocol UDP or TCP used for OpenVPN®. Default is UDP. Uploads the OpenVPN® CA. Uploads the OpenVPN® Certi cate. Uploads the OpenVPN® Client Key. Must be the same cipher method used by the OpenVPN® server OpenVPN® authentication username (optional) OpenVPN® authentication password (optional) Additional options are to be appended to the OpenVPN® con g le, 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 con gurations above.
Enables/Disables the SNMP feature. Default settings is No Version SNMP version, the available options are: Version 1 Version 2 Version 3 SNMP port. Default is 161. Con gures the Name of SNMP trap community. SNMP Trap Version. Default is Trap Version 2. IP address of the SNMP trap receiver. Port of the SNMP trap receiver. Default is 162. The interval between each trap sent to the trap receiver. Default is 60. Community string associated to the trap. It must match the community string of the trap receiver. Username for SNMP.

Security Level
Authentication Protocol Privacy Protocol Authentication Key Privacy Key SNMP Trap Username
Trap Security Level
Trap Authentication Protocol Trap Privacy Protocol Trap Authentication Key Trap Privacy Key

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.

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

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

Enter the Authentication Key

Enter the Privacy Key.

User name for SNMP Trap.

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.

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

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

Enter the Trap Authentication 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 eld is case sensitive.

Con rm Password

Enter the new User password again to con rm.

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 eld is case sensitive

Con rm Password

Enter the new Admin password again to con rm.

Upgrade Firmware

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

Firmware Upgrade and Provisioning

Speci es how rmware upgrading and provisioning request to be sent: Always Check for New Firmware, Check New Firmware only when F/W pre/su x 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 Con rmation

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

Validate Hostname in Certi cate

To validate the hostname in the SSL certi cate

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-speci c option to do this. DHCP option 43 approach has priorities. The phone is allowed to fall back to the original server path con gured 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 rmware server instead of using the con gured rmware 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)

De nes the hour of the day to check the HTTP/TFTP/FTP server for rmware upgrades or con guration les changes. The default value is 1.

Day of the Week (0-6)

De nes the day of the week to check HTTP/TFTP/FTP server for rmware upgrades or con guration les 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”.

Con g

Con g Upgrade Via

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

Con g Server Path

De nes the server path for provisioning. Note: you can de ne the server path prepended with the protocol used, For example when using HTTPS as the upgarde method, you can set the con g server path to: https://server_address

Con g Server Username

The username for the con g server.

Con g Server Password

The password for the con g server.

Con g File Pre x

Enables your ITSP to lock con guration updates. If con gured, only the con guration le with the matching encrypted pre x will be downloaded and ashed into the phone.

Con g File Post x

Enables your ITSP to lock con guration updates. If con gured, only the con guration le with the matching encrypted post x will be downloaded and ashed into the phone.

XML Con g File Password

The password for encrypting XML con guration le using OpenSSL. This is required for the phone to decrypt the encrypted XML con guration le.

Enable Handset Con g Upgrade

Enable handset con g upgrade for handset related settings. Disabled by Default.

Handset Con g File Pre x

If con gured, only the handset con guration le with the matching encrypted pre x will be downloaded and ashed into the device. If the le is located in a subdirectory on the provisioning server, the directory should be included here; for example “handset/ipei_”.

Handset Con g File Post x

If con gured, only the handset con guration le with the matching encrypted post x will be downloaded and ashed into the device.

Authenticate Conf File

Sets the phone system to authenticate con guration le before applying it. When set to “Yes”, the con guration le 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 Con guration

Click to download the phone’s con guration le in .txt format. Note: Con guration backup le does not include passwords or CA/Custom certi cate

Download Device Con guration (XML)

Click to download the device con guration le in .xml format.

Download and Process All Available Con g Files

By default, the device will provision the rst available con g in the order of cfgMAC, cfgMAC.xml, cfgMODEL.xml, and cfg.xml (corresponding to device-speci c, model-speci c, and global con gs). If this option is enabled, the phone will inverse the downloading process to cfg.xml > cfgMAC.bin > cfgMAC.xml. The following les will override the les that have already been loaded and processed.

Download User con guration

This allows users to download part of the con guration 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 con g le uploaded from “Upload Device Con guration”, but not include the changes from the server provision via TFTP/FTP/FTPS/HTTP/HTTPS.

Upload Device Con guration

Uploads con guration le to phone.

Export backup Package

Export backup package which contains device con guration along with personal data.

Restore from Backup package

Click to upload backup package and restore.

Firmware

Firmware Upgrade Via

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

Firmware Server Path

De nes the server path for the rmware server. Note: you can de ne the server path prepended with the protocol used, For example when using HTTPS as the upgarde method, you can set the rmware server path to: https://server_address

Firmware Server Username

The username for the rmware server.

Firmware Server Password

The password for the rmware server.

Firmware File Pre x

Enables your ITSP to lock rmware updates. If con gured, only the rmware with the matching encrypted pre x will be downloaded and ashed into the phone.

Firmware File Post x

Enables your ITSP to lock rmware updates. If con gured, only the rmware with the matching encrypted post x will be downloaded and ashed into the phone.

HS Firmware

Handset rmware

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

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.

Syslog Server
Syslog Level
Syslog Keyword Filtering Send SIP Log Syslog Capture Status Capture Mode
Capture Timer Log File Rotation Max Log File Size Max Log Files TR-069 Enable TR-069 ACS URL TR-069 Username TR-069 Password Periodic Inform Enable Periodic Inform Interval Connection Request Username Connection Request Password Connection Request Port CPE SSL Certi cate CPE SSL Private Key Randomized TR069 Startup Security Settings Validate Server Certi cates SIP TLS Certi cate SIP TLS Private Key SIP TLS Private Key Password

Note: The CA certi cate is required to connect with the TLS 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 eld (i.e. 172.18.1.1:1000), the phone will send Syslog to the corresponding port of that IP. Administrators can save the syslog settings parameters across a factory reset, by assigning value 1 to the P-value P82307 from the device con guration le, this is because this
con guartion parameter is not available on the webUI.
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 will be ltered based on keywords provided. If you enter multiple keywords, it should be separated by `,’. Please note that no spaces are allowed.
Con gures 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.
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.
Sets the capture mode. Either set to Timed mode or continuous. Timed Mode: When a new capture is running, the previous les 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.
If Capture Mode is set to “Timed” this eld will appear to specify how long to capture syslog in minutes. 0 is unlimited. Internal capture has a 30-minute maximum limit.
Rotation is always enabled when capturing internally. Log File Rotation will maintain a xed maximum limit of the le size based on the Max Log File Size and Max Log Files con gured. Old logs will be deleted when rotated.
The maximum log le size used when rotation is enabled
The number of log les used when rotation is enabled
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 con rmation pop-up will appear, asking you to con rm your desire to reboot the device.
Speci es URL of TR-069 ACS (e.g., http://acs.mycompany.com), or IP address. The default setting is https://acs.dgms.cloud
Speci es the username to authenticate to ACS.
Speci es the password to authenticate to ACS.
When enabled, periodic information packets to the ACS server will be sent. The default setting is “Yes”
Con gures periodic inform intervals to send the inform packets to TR-069 Auto Con guration Server. The default setting is 86400
Speci es the username for the ACS to connect to the phone.
Speci es the password for the ACS to connect to the phone.
The port for the ACS to connect to the phone.
Uploads Cert File for the phone to connect to the ACS via SSL.
Uploads Cert Key for the phone to connect to the ACS via SSL.
When enabled, TR069 will send out rst INFORM message to server on randomized timing between 1 to 3600 seconds after phone boots up.
After enabling this feature, the phone will validate the server’s certi cate. 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.
SSL Certi cate used for SIP Transport in TLS/TCP.
SSL Private key used for SIP Transport in TLS/TCP.
SSL Private key password used for SIP Transport in TLS/TCP.

Custom Certi cate Web Access Mode Enable User Web Access HTTP Web Port HTTPS Web Port Disable SSH SSH Public Key Web Session Timeout Web Access Attempt Limit Minimum TLS Version Maximum TLS Version Trusted CA Certi cates Trusted CA Certi cates
Load CA Certi cates
Packet Capture With RTP Packets With Secret Key Information Factory Reset Force Reboot Con gure Web UI Button
Reset Type
Perform Selected Reset Con gure Hardware Button
Reset Type
Tools Provision Ping Traceroute

The uploaded custom certi cate will be used for SSL/TLS communication instead of the WP phone default certi cate.
Sets the protocol for web interface. The default setting is “HTTP”.
Administrator can disable or enable user web access. The default value is Disabled.
Con gures the HTTP port under the HTTP web access mode.
Con gures the HTTPS port under the HTTPS web access mode. Default setting is “443”.
Disables SSH access. The default setting is “No”.
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.
Con gures timer to logout web session during idle. Default is 10 min. Range is 2-60 min.
Con gures attempt limit before lockout. Default is 5. Range is 1-10.
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
Allows users to choose the maximum TLS version for HTTPS provisioning. Set to unlimited.

Allows to upload and delete up to 6 CA Certi cates les to the phone. Note: Users can either upload the le directly from the web or they can choose to provision it from their cfg.xml le.
Users are able to specify which certi cate they are going to use: All Certi cates: (Default) Both built-in and uploaded Certi cates. Default Certi cates: Built-in Certi cates. Custom Certi cates: Uploaded Certi cates;

Choose whether the packet capture le contains RTP or not. Set to no by default.
Allows users to make packet capture including the secret key to decrypt the captured TLS packets. Default value is No.

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

Speci es 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.
Executes the type of reset chosen.

Speci es 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.

Makes the phone trigger an instant provisioning.

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

Checks the route packets take to the speci ed 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 con guration in the Global Phonebook XML Settings page. If set to LDAP, DP755 will use the con guration in the Global Phonebook LDAP Settings page. If set to XSI, DP755 will use the con guration 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 le. This eld 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 le to the base station. Delete: Clears global XML phonebook le 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 con guration in Global Phonebook XML Settings page. If set to LDAP, DP752 will use con guration in Global Phonebook LDAP Settings page.

LDAP Phonebook Settings

LDAP protocol

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

Server Address

Con gures 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 eld can be left blank if the LDAP server allows anonymous binds.

LDAP Filter

LDAP lter 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

De nes rst name attributes of each record to be returned in the LDAP search result. This eld allows users to con gure multiple space-separated name attributes. Example: gn cn sn description

Last Name Attribute

De nes the last name attributes of each record to be returned in the LDAP search result. This eld allows users to con gure multiple space-separated name attributes. Example: gn cn sn description

Work Number Attribute

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

Home Number Attribute

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

Mobile Number Attribute

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

Max. Hits

Speci es 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

Con gures 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

Con gure 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 XSI Actions Path Broadsoft Contact Download Interval
XSI Authentication Type
Login Credentials Login Username Login Password SIP Credentials SIP UserName SIP User ID SIP Password XSI Account Assignment Handset XSI Account
Call Log Type
Network Directories Private Phonebook Settings Upload XML Phonebook 1-10 Phonebook Name Import XML Phonebook Export XML Phonebook

Con gure the BroadWorks Xsi server port. The default port is 80. If the server uses HTTPS, please con gure 443. con gure the deployment path for Broadsoft XSI Actions. If it is empty, the path “com.broadsoft.xsi-actions” will be used. Con gures 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. This feature allows users to choose the type of authenti cation that will be used to access the XSI information from the handset, there are three types of authenti cations: Through Login credentials: Uses the log in Username and password for authenti cation Through SIP credentials: Uses the SIP User ID, SIP Auth ID, and SIP Password for authenti cation Match SIP Account: Uses the credentials used in the SIP account of the DP755.
Con gure the Username for the BroadWorks XSI server. Con gure the password for the BroadWorks XSI server.
Con gure SIP Username for the BroadWorks XSI server. Con gure SIP User ID for the BroadWorks XSI server. Con gure SIP Password for the BroadWorks XSI server. This setting can be applied individually for the 10 handsets supported by DP755. Displays the handset on which the XSI setting will be applied Sets the SIP account that will be used on the speci c handset Sets which call logs will be displayed on the handset, there are two options: Use XSI Call logs Use Base Station Call Logs Enable/Disable and choose the name of the following types: Group Directory; Enterprise Directory; Group Common; Enterprise Common; Personal Directory.
De nes private phonebook name. Upload: Uploads manually a private XML phonebook le to the base station. Delete: Clears private XML phonebook le in the base station 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.

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

Registration Process

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.

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.

Multiple Base Stations Registration

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.

5. Go to DECT ACCOUNT ASSIGNEMENT

SIP Settings

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.

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

Account Status

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.

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.

Outgoing Default Line

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.

Block Caller ID (for all subsequent calls)

*30

Off hook the phone;

Dial *30.

Send Caller ID (for all subsequent calls)

*31

Off hook the phone;

Dial *31.

Call with Caller ID Blocked (per call)

*67

Off hook the phone;

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

Call with Caller ID Enabled (per call)

*82

Off hook the phone;

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

Unconditional Call Forward. To set up unconditional call forward: Off hook the phone; 72 Dial 72 and then enter the number to forward the call; Press the OK softkey or SEND key.

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

*73

Off hook the phone;

Dial *73;

Busy Call Forward. To set up a busy call forward: Off hook the phone; 90 Dial 90 and then enter the number to forward the call; Press the OK softkey or SEND key.

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

*91

Off hook the phone;

Dial *91;

Delayed Call Forward. To set up a delayed call forward: Off hook the phone; 92 Dial 92 and then enter the number to forward the call; Press the OK softkey or SEND key.

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

*93

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
3. 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.
4. Go to the DECT General Settings tab. 5. 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.
6. 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).

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

Global Phonebook XML Settings

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.

Manual XML Phonebook Management 1. Press Upload in Import XML Phonebook. 2. Browse your files and select your phonebook.xml file.
XML Phonebook file format

Automatic XML Phonebook Download Manual XML Phonebook Management

<?xml version=”1.0″ encoding=”UTF-8″?>

Object AddressBook Contact LastName FirstName Phone PhoneNumber

Position Root element Child element Child element Child element Child element Child element

XML Phonebook Example:
<?xml version=”1.0″ encoding=”UTF-8″?>

Type Mandatory Mandatory At least one of them present String Mandatory At least one present

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).

Values ­ ­ String First name of the contact ­ Int

Comments Root element of the XML document Each contact is an entry Last name of the contact
Phone number Type=”Home” or Type=”Work” or Type=”Mobile”

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.
Enabling Repeater Mode on DP755
Before associating the DP760 DECT Repeater to your DP755 Base Station, you should first enable the repeater mode on your base station. Please refer following steps to enable the repeater mode on the DP755 base station:
1. Access the Web GUI of your DP755 using the admin’s username and password. 2. Navigate to DECT General Settings and set Enable Repeater Mode to Yes.

3. Press Save and Apply, then Reboot the base to apply the new settings.

Enable Repeater Mode

DP760 DECT Repeater Association
After enabling the repeater mode on DP755 Base Station, you can easily associate it with your DP760 DECT Repeater using one of the following methods:
Auto Association Manual Association Use DP755 Repeater Management to associate DP760

Auto Association
To auto-associate your DP760 DECT Repeater with DP755 Base Station:
1. Power on the DP760 DECT Repeater. 2. After the DP760 finishes booting, the association LED will be blinking to indicate that the repeater is searching for nearby base stations.
Note: If all the LEDs remain on, it indicates that the DP760 Repeater has successfully associated with a DP755 Base and is ready for use. You may need to dissociate it before continuing (Please refer to [DP760 DECT REPEATER DISSOCIATION]). 3. Open a subscription on your DP755 Base Station using one of the following methods:
1. Rebooting the Base station.

2. Long press on page/subscribe

button on DP755 for 4 seconds.

3. Access DP755 Web GUI and press subscribe button

.

4. The DP760 DECT Repeater will automatically pair with DP755. Upon successful association with the DP755 Base station all LEDs will turn ON and the repeater status will be updated as following.

Repeater Status ­ Associated (Auto Association)
Notes For auto association, the option Use manually configured RFPI under DP760 Web GUI should be disabled (by default disabled). The RF auto association requires the repeater to be close to the base during the pairing process to work properly (this is a standard DECT repeater pairing limitation).

Manual Association
To manually associate your DP760 DECT Repeater to your DP755 Base Station:
1. Access the Web GUI of your DP760 using the admin’s username and password. 2. Navigate to DECT General Settings and set Use Manually Configured RFPI to Yes. 3. On the Manually Configured Base RFPI field, configure your DP755 Base Station RFPI address.
Note: The RFPI Address of your DP755 Base Station is available under the Web GUI Status DECT Status Base DECT RFPI Address. 4. Click Save and Apply to confirm the new settings. 5. Open a subscription on your DP755 Base Station using one of following methods:
Rebooting the Base station

Long press on page/subscribe

button of DP755 for 7 seconds

Access DP755 Web GUI and press subscribe button

.

6. The DP760 DECT Repeater will automatically pair with DP755. If the association is correctly associated, all the LEDs will remain solid on and the repeater status will be updated as following.

Note Make sure that Repeater Mode is enabled on your DP755 Base Station.

Repeater Status ­ Associated (Manual Association)

Use DP755 Repeater Management to Associate DP760
Enabling Repeater Management Mode The following steps illustrate how to enable and use the Repeater Management mode on DP755:
1. Access the Web GUI of your DP755 using the admin’s username and password. 2. Navigate to DECT General Settings and set Enable Repeater Management to Yes. 3. Press Save and Apply, then reboot the unit to apply the new settings.

Associating DP760 using Repeater Management Mode on DP755

Repeater Associated Status

The repeater management mode displays discovered and paired devices using their name, IP and Mac addresses and also gives the possibility of associating the DP755 Base Station with DP760 DECT Repeater.

Please refer to following steps in order to associate DP760 with DP755 using repeater management mode:

1. Access the Web GUI of your DP755 using the admin’s username and password. 2. Navigate to Status DECT Repeater Status and select the repeater station from Discovered Devices. 3. Click the Link button to associate the base and repeater stations.

Note: The DP760 Repeater stations associated with DP755 Base Station are displayed and availab

References

• base.by Выставлен на продажу
• Grandstream Firmware
• Welcome! - The Apache HTTP Server Project
• Логойск / LOG.BY
• DP Handsets Provisioning Guide - Documentation Center
• documentation.grandstream.com/knowledge-base/dp755-administration-guide/?hkb-redirect&nonce=fbb0b6a5bd&check=13ou7&redirect=helpdesk.grandstream.com&otype=ht_kb_article&oid=71228&source=widget
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