User Guide for GRANDSTREAM models including: FPC-N34, GRP2602, GRP2603, GRP2604, Essential IP Phones

Grandstream Networks, Inc. - IP Phone Warehouse

15 apr 2021 —

Page | 4 Versio GRP260x Administration Guide n 1.0.1.18 Company Name: Grandstream Networks, Inc. Address: 126 Brookline Ave, 3rd Floor Boston, MA 02215, USA Tel: 1-617-5669300 Fax: 1-617-2491987 If the trouble is causing harm to the telephone network, the telephone company may request you to remove the


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Grandstream-GRP260X-Administration-Guide
Grandstream Networks, Inc.
GRP2601(P) | GRP2602(P/W) GRP2603(P) | GRP2604(P) Essential IP Phones Administration Guide

GRP2601 / GRP2601P GRP2603 / GRP2603P

GRP2602 / GRP2602P / GRP2602W GRP2604 / GRP2604P

COPYRIGHT
©2020 Grandstream Networks, Inc. http://www.grandstream.com All rights reserved. Information in this document is subject to change without notice. Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted. The latest electronic version of this guide is available for download here:
http://www.grandstream.com/support
Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc. in the United States, Europe, and other countries.

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CAUTION
Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this guide, could void your manufacturer warranty.
WARNING
Please do not use a different power adaptor with devices as it may cause damage to the products and void the manufacturer warranty.
U.S. FCC Part 68 Regulatory Information
This equipment complies with Part 68 of the FCC rules. Located on the equipment is a label that contains, among other information, the ACTA registration number and ringer equivalence number (REN). If requested, this information must be provided to the telephone company. The REN is used to determine the quantity of devices which may be connected to the telephone line. Excessive REN's on the telephone line may result in the devices not ringing in response to an incoming call. In most, but not all areas, the sum of the REN's should not exceed five (5.0). To be certain of the number of devices that may be connected to the line, as determined by the total REN's contact the telephone company to determine the maximum REN for the calling area. This equipment cannot be used on the telephone company-provided coin service. Connection to Party Line Service is subject to State Tariffs. If this equipment causes harm to the telephone network, the telephone company will notify you in advance that temporary discontinuance of service may be required. If advance notice is not practical, the telephone company will notify the customer as soon as possible. Also, you will be advised of your right the file a complaint with the FCC if you believe it is necessary. The telephone company may make changes in its facilities, equipment, operations, or procedures that could affect the operation of the equipment. If this happens, the telephone company will provide advance notice in order for you to make the necessary modifications in order to maintain uninterrupted service.
If trouble is experienced with this equipment, please contact (Agent in the US):

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Company Name: Grandstream Networks, Inc. Address: 126 Brookline Ave, 3rd Floor Boston, MA 02215, USA Tel: 1-617-5669300 Fax: 1-617-2491987
If the trouble is causing harm to the telephone network, the telephone company may request you to remove the equipment from the network until the problem is resolved. This equipment uses the following USOC jacks: RJ45C.
It is recommended that the customer install an AC surge arrester in the AC outlet to which this device is connected. This is to avoid damaging the equipment caused by local lightning strikes and other electrical surges.
Since this device has the HAC function, the earpiece is easy to absorb small, please take care to avoid scratching.

U.S. FCC Part 15 Regulatory Information
This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation.

Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment.

Note: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio
Frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:

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--Reorient or relocate the receiving antenna. --Increase the separation between the equipment and receiver. --Connect the equipment into an outlet on a circuit different from that to which the receiver is connected. --Consult the dealer or an experienced radio/TV technician for help.
This equipment complies with FCC radiation exposure limits set forth for an uncontrolled environment. This equipment should be installed and operated with minimum distance 20cm between the radiator& your body. This transmitter must not be co-located or operating in conjunction with any other antenna or transmitter.

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Caution: Exposure to Radio Frequency Radiation
This equipment complies with EU radiation exposure limits set forth for an uncontrolled environment. This equipment should be installed and operated with minimum distance of 20 cm between the radiator and your body.
CE Authentication
Hereby, Grandstream Networks, Inc. declares that the radio equipment GRP2601/GRP2601P, GRP2602/GRP2602P/GRP2602W, GRP2603/GRP2603P and GRP2604/GRP2604P are in compliance with Directive 2014/53/EU. The full text of the EU declaration of conformity is available at the following internet address: http://www.grandstream.com/support/resources/

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GNU GPL INFORMATION
GRP260X firmware contains third-party software licensed under the GNU General Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license.
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/sites/default/files/Resources/grp_gpl_color.tar.gz

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Table of Contents
DOCUMENT PURPOSE ............................................................................................... 12 CHANGE LOG .............................................................................................................. 13
Firmware Version 1.0.1.18 ...................................................................................................................... 13
GUI INTERFACE EXAMPLES ...................................................................................... 14 WELCOME ................................................................................................................... 15 PRODUCT OVERVIEW ................................................................................................ 16
Feature Highlights ................................................................................................................................... 16 Technical Specifications .......................................................................................................................... 17
GETTING STARTED..................................................................................................... 24
Equipment Packaging.............................................................................................................................. 24 GRP260X Phone Setup .......................................................................................................................... 25
Using the Phone Stand ........................................................................................................................ 25 Using the Slots for Wall Mounting........................................................................................................ 25 Connecting the GRP260X ....................................................................................................................... 26 Configuration via Keypad ........................................................................................................................ 27 Configuration via Web Browser ............................................................................................................... 32 Saving Configuration Changes................................................................................................................ 33 Rebooting from Remote Locations .......................................................................................................... 33
CONFIGURATION GUIDE ............................................................................................ 34
Status Page Definitions ........................................................................................................................... 34 Account Page Definitions ........................................................................................................................ 36 Phone Settings Page Definitions ............................................................................................................. 53 Network Settings Page Definitions .......................................................................................................... 57 Programmable keys Page Definitions ..................................................................................................... 62

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System Settings Page Definitions ........................................................................................................... 67 Maintenance Page Definitions................................................................................................................. 73 Application Page Definitions.................................................................................................................... 79 External Service Page Definitions ........................................................................................................... 83
NAT SETTINGS ............................................................................................................ 86 PACKET CAPTURE ..................................................................................................... 87 UPGRADING AND PROVISIONING ............................................................................ 88
Unified Firmware ..................................................................................................................................... 88 Firmware Upgrade................................................................................................................................... 88
Upgrade via Keypad Menu .................................................................................................................. 89 Upgrade via Web GUI.......................................................................................................................... 89 No Local TFTP/FTP/HTTP Servers ..................................................................................................... 90 Phone Provisioning.................................................................................................................................. 91 Configuration File Download................................................................................................................ 91 No Touch Provisioning ......................................................................................................................... 93
RESTORE FACTORY DEFAULT SETTING ................................................................. 95
Restore Factory Settings using LCD menu ............................................................................................. 95 Restore to Factory Default via Web GUI ................................................................................................. 95
EXPERIENCING GRP260X .......................................................................................... 96

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Table of Tables
Table 1: GRP260X Features in a Glance.................................................................................................... 16 Table 2: GRP2601/GRP2601P Technical Specifications ............................................................................ 17 Table 3: GRP2602/GRP2602P/GRP2602W Technical Specifications........................................................ 19 Table 4: GRP2603/GRP2603P Technical Specifications ............................................................................ 20 Table 5: GRP2604/GRP2604P Technical Specifications ............................................................................ 22 Table 6: Equipment Packaging ................................................................................................................... 24 Table 7: Configuration Menu ....................................................................................................................... 27 Table 8: Status Page Definitions ................................................................................................................. 34 Table 9: Account Page Definitions .............................................................................................................. 36 Table 10: Settings Page Definitions ............................................................................................................ 53 Table 11: Network Page Definitions ............................................................................................................ 57 Table 12: Programmable Keys Page Definitions ........................................................................................ 62 Table 13: System Settings Page Definitions ............................................................................................... 67 Table 14: Maintenance Page Definitions..................................................................................................... 73 Table 15: Application Page Definitions ........................................................................................................ 79 Table 16: External Service Page Definitions ............................................................................................... 83

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Table of Figures
Figure 1: GRP260X Package Content ........................................................................................................ 24 Figure 2: Phone Stand and Mounting Slots on the GRP260X.................................................................... 25 Figure 3: Tab on the Handset Cradle .......................................................................................................... 25 Figure 4: GRP260X Back Side View........................................................................................................... 27 Figure 5: GRP260x LCD settings................................................................................................................ 31 Figure 6: Packet Capture in Idle ................................................................................................................. 87 Figure 7: Packet Capture when running ..................................................................................................... 87 Figure 8: GRP260X Unified Firmware ........................................................................................................ 88 Figure 9: Config File Download................................................................................................................... 91 Figure 10: Certificates Files Download ....................................................................................................... 93

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DOCUMENT PURPOSE
This document describes how to configure GRP260X features via phone's LCD menu and Web GUI menu. The intended audiences of this document are phone administrators. To learn the basic functions of GRP2601(P)/GRP2602(P/W), GRP2603(P) and GRP2604(P), please visit http://www.grandstream.com/support to download the latest "GRP260X User Guide".
This guide covers the following topics: · Product Overview · Getting Started · Configuration Guide · NAT Settings · Packet Capture · Upgrading and Provisioning · Restore Factory Default Settings · Experiencing GRP260X

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CHANGE LOG
This section documents significant changes from previous versions of user manuals for GRP260X. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here.
Firmware Version 1.0.1.18
· This is the initial version.

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GUI INTERFACE EXAMPLES
http://www.grandstream.com/sites/default/files/Resources/GRP260X_web_gui.zip
1. Screenshot of Login Page 2. Screenshots of Status Pages 3. Screenshots of Accounts Pages 4. Screenshots of Settings Pages 5. Screenshots of Network Pages 6. Screenshots of Maintenance Pages 7. Screenshots of Phonebook Pages

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WELCOME
Thank you for purchasing Grandstream GRP260X Essential IP Phones.
Part of the GRP series of Carrier-Grade IP Phones, the GRP2601/GRP2602 is an essential 2-line model designed with zero-touch provisioning for mass deployment and easy management. It features a sleek design and a suite of next-generation features including 5-way voice conferencing to maximize productivity and dual band Wi-Fi support (GRP2602 only), EHS support for Plantronics & Jabra & Sennheiser headsets and multi-language support. The GRP series includes carrier-grade security features to provide enterpriselevel security, including secure boot, dual firmware images and encrypted data storage. For cloud provisioning and centralized management, the GRP260X is supported by Grandstream's Device Management System (GDMS), which provides a centralized interface to configure, provision, manage and monitor deployments of Grandstream endpoints. Built for the needs of desktop workers and designed for easy deployment by enterprises, service providers and other high-volume markets, the GRP2601/GRP2602 offers an easy-to-use and easy-to-deploy voice endpoint.
Part of the GRP series of Carrier-Grade IP Phones, the GRP2603/GRP2604 is an essential 3-line model designed with zero touch provisioning for mass deployment and easy management. It features a sleek design and a suite of next generation features including: 5-way voice conferencing to maximize productivity, integrated PoE (GRP2603P), full HD audio on both the speaker and handset to allow users to communicate with the utmost clarity, EHS support for Plantronics & Jabra & Sennheiser headsets and multi-language support. The GRP series includes carrier-grade security features to provide enterprise-level security, including secure boot, dual firmware images and encrypted data storage. For cloud provisioning and centralized management, the GRP2603/GRP2604 is supported by Grandstream's Device Management System (GDMS), which provides a centralized interface to configure, provision, manage and monitor deployments of Grandstream endpoints. Built for the needs of desktop workers and designed for easy deployment by enterprises, service providers and other high-volume markets, the GRP2603/GRP2604 offers an easy-to-use and easy-to-deploy voice endpoint.
The GRP260X series deliver superior HD audio quality, rich and leading-edge telephony features, protection for privacy, and broad interoperability with most 3rd party SIP devices and leading SIP/NGN/IMS platforms. GRP260X series is the perfect choice for enterprise users looking for a high quality, feature rich multi-line executive IP phone with advanced functionalities and performance.

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PRODUCT OVERVIEW

Feature Highlights

The following table contains the major features of the GRP260X phones:

Table 1: GRP260X Features in a Glance

GRP2601 GRP2601P

 4 programmable context-sensitive soft keys.  10/100M network ports.  Integrated PoE (for GRP2601P only).  5-way conference.  Electronic Hook Switch (EHS) support for Plantronics &
Jabra & Sennheiser.

GRP2602 GRP2602P GRP2602W
GRP2603 GRP2603P

 2 SIP account keys with dual-color LED  4 programmable context-sensitive soft keys.  10/100M network ports.  Integrated PoE(for GRP2602P only).  5-way conference.  Electronic Hook Switch (EHS) support for Plantronics &
Jabra & Sennheiser.  Wi-Fi support (GRP2602W only).
 3 SIP account keys with dual-color LED.  4 XML programmable context sensitive soft keys  132 x 64 backlit graphical LCD display  10/100/1000 Mbps Ethernet ports.  Integrated PoE (GRP2603P only).  5-way conference.  Electronic Hook Switch (EHS) support for Plantronics &
Jabra & Sennheiser.

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GRP2604 GRP2604P

 3 SIP account keys with dual-color LED.  4 XML programmable context sensitive soft keys  132 x 64 backlit graphical LCD display  10/100/1000 Mbps Ethernet ports.  Integrated PoE (GRP2604P only).  5-way conference.  Electronic Hook Switch (EHS) support for Plantronics &
Jabra & Sennheiser.

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 GRP260X series.
Table 2: GRP2601/GRP2601P Technical Specifications

Protocols/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, SNMP, 802.1x, TLS, SRTP, IPV6

Network Interfaces

Dual switched auto-sensing 10/100/1000 Mbps Ethernet ports, integrated PoE (GRP2601P only)

Graphic Display

132 x 48 (2.21'') LCD display

Feature Keys

4 XML programmable context sensitive soft keys, 5 (navigation, menu) keys. 8 dedicated function keys for: MESSAGE (with LED indicator), TRANSFER,HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL

Voice Codec

Support for G7.29A/B, G.711µ/a-law, G.726, G.722(wide-band), G723, iLBC, OPUS, in-band, and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics & Jabra & Sennheiser headsets)

Telephony Features

Hold, transfer, forward, 5-way conference, call park, call pickup, downloadable Phonebook (XML, LDAP, up to 2000 items), call waiting, call log(up to 800

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records), off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hotdesking, personalized music ringtones and music on hold, server redundancy and fail-over

HD audio

Yes, both on handset and full-duplex hands-free speakerphone

Base Stand

Yes, 1 angle positions available

Wall Mountable

Yes, (*wall mount sold separately)

QoS

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

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control, secure boot

Multi-language

English, Chinese, Korean, Japanese, German, Italian, French, Spanish, Portuguese, Russian, Croatian, and more

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

Power & Green Energy Efficiency

Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA; PoE: IEEE802.3af Class 1, 3.84W; IEEE802.3az (EEE) (GRP2601P Only)

Physical

Dimension: 208mm (L) x 180mm (W) x 63.4mm (H) (with handset) Unit weight:650g; Package weight:810g (860g for GRP2601)

Temperature and Humidity

Operation: 0°C to 40°C Storage: -10°C to 60°C Humidity: 10% to 90% Non-condensing

Package Content

GRP2601/2601P phone, handset with cord, base stand, universal power supply(GRP2601 only), network cable, Quick Installation Guide

Compliance

FCC: Part 15 Class B; FCC Part 68 HAC; CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 62368-1; RCM: AS/NZS CISPR32; AS/NZS 62368.1; AS/CA S004; IC: ICES-003; CS-03;

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Table 3: GRP2602/GRP2602P/GRP2602W Technical Specifications

Protocols/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, SNMP, 802.1x, TLS, SRTP, IPV6

Network Interfaces

Dual switched auto-sensing 10/100/1000 Mbps Ethernet ports, integrated PoE (GRP2602P only)

Graphic Display

132 x 48 (2.21'') backlit graphical LCD display

Wi-Fi

Yes, dual-band (GRP2602W only).

Feature Keys

2 SIP account keys with dual-color LED, 4 XML programmable context sensitive soft keys, 5 (navigation, menu) keys. 8 dedicated function keys for: MESSAGE (with LED indicator), TRANSFER, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL

Voice Codec

Support for G7.29A/B, G.711µ/a-law, G.726, G.722(wide-band),G723,iLBC, OPUS, in-band, and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics & Jabra &Sennheiser headsets)

Telephony Features

Hold, transfer, forward, 5-way conference, call park, call pickup, shared-callappearance(SCA)/bridged-line-appearance(BLA), downloadable phonebook (XML, LDAP , up to 2000 items), call waiting, call log (up to 800 records), off-hook auto dial, auto-answer, click-to-dial, flexible dial plan, hot-desking, personalized music ringtones and music on hold, server redundancy and fail-over

HD audio

Yes, HD handset and speakerphone with support for wideband audio

Base Stand

Yes, 2 angle positions available

Wall Mountable

Yes, (*wall mount sold separately)

QoS

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

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control, secure boot

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Multi-language

English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, Chinese, Korean, Japanese, and more

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

Power & Green Energy Efficiency

Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA; PoE: IEEE802.3af Class 1, 3.84W; IEEE802.3az (EEE) (GRP2602P Only)

Physical

Dimension: 208mm (L) x 180mm (W) x 63.4mm (H) (with handset) Unit weight: 670g; Package weight:830g (880g for GRP2602)

Temperature and Humidity

Operation: 0°C to 40°C Storage: -10°C to 60°C Humidity: 10% to 90% Non-condensing

Package Content

GRP2602 phone, handset with cord, base stand, universal power supply (GRP2602 only), network cable, Quick Installation Guide

Compliance

FCC: Part 15 Class B; FCC Part 68 HAC; CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 62368-1; RCM: AS/NZS CISPR32; AS/NZS 62368.1; AS/CA S004; IC: ICES-003; CS-03;

Protocols/Standards

Table 4: GRP2603/GRP2603P Technical Specifications
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, SNMP, 802.1x, TLS, SRTP, IPV6

Network Interfaces

Dual switched auto-sensing 10/100/1000 Mbps Ethernet ports, integrated PoE (GRP2603P only)

Graphic Display

132 x 64 backlit graphical LCD display

Feature Keys

3 SIP account keys with dual-color LED, 4 XML programmable context sensitive soft keys, 5 (navigation, menu) keys. 9 dedicated function keys for: MESSAGE(with LED indicator), TRANSFER, HOLD, HEADSET, MUTE,

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SEND/REDIAL, SPEAKERPHONE, VOL+, VOL-

Voice Codec

Support for G7.29A/B, G.711µ/a-law, G.726, G.722(wide-band),G723,iLBC, OPUS, in-band, and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics & Jabra &Sennheiser headsets)

Telephony Features

Hold, transfer, forward, 5-way conference, call park, call pickup, shared-callappearance(SCA)/bridged-line-appearance(BLA), downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 800 records), off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot-desking, personalized music ringtones and music on hold, server redundancy and fail-over

HD audio

Yes, HD handset and speakerphone with support for wideband audio

Base Stand

Yes, 2 angle positions available, Wall Mountable (*wall mount sold separately)

Wall Mountable

Yes, (*wall mount sold separately)

QoS

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

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control, secure boot

Multi-language

English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, Chinese, Korean, Japanese, and more

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

Power & Green Energy Efficiency

Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA;PoE: IEEE802.3af Class 1, 3.84W; IEEE802.3az (EEE) (GRP2603P Only)

Physical

Dimension: 214mm (L) x 206mm (W) x 68mm (H) (with handset) Unit weight: 780g; Package weight: 1090g for GRP2603P &1140g for GRP2603

Temperature and Humidity

Operation: 0°C to 40°C Storage: -10°C to 60°C Humidity: 10% to 90% Noncondensing

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Package Content Compliance

GRP2603 phone, handset with cord, base stand, universal power supply (GRP2603 only), network cable, Quick Installation Guide
FCC: Part 15 Class B; FCC Part 68 HAC; CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 62368-1; RCM: AS/NZS CISPR32; AS/NZS 62368.1; AS/CA S004; IC: ICES-003; CS-03;

Protocols/Standards

Table 5: GRP2604/GRP2604P Technical Specifications
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, HTTP/HTTPS, ARP, ICMP, DNS(A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, SNMP, 802.1x, TLS, SRTP, IPV6

Network Interfaces

Dual switched auto-sensing 10/100/1000 Mbps Ethernet ports, integrated PoE (GRP2603P only)

Graphic Display

132 x 64 backlit graphical LCD display

Feature Keys

3 SIP account keys with dual-color LED, 4 XML programmable context sensitive soft keys, 5 (navigation, menu) keys. 9 dedicated function keys for: MESSAGE(with LED indicator), TRANSFER, HOLD, HEADSET, MUTE, SEND/REDIAL, SPEAKERPHONE, VOL+, VOL-

Voice Codec

Support for G7.29A/B, G.711/a-law, G.726, G.722(wide-band), G723, iLBC, OPUS, in- band and out-of-band DTMF (in audio, RFC2833, SIP INFO)Support for G7.29A/B, G.711µ/a-law, G.726, G.722(wide-band),G723,iLBC, OPUS, inband and out-of-band DTMF(in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC

Auxiliary Ports

RJ9 headset jack (allowing EHS with Plantronics & Jabra &Sennheiser headsets)

Telephony Features

Hold, transfer, forward, 5-way conference, call park, call pickup, shared-callappearance(SCA)/bridged-line-appearance(BLA), downloadable phonebook (XML, LDAP, up to 2000 items), call waiting, call log (up to 800 records), off-hook auto dial, auto answer, click-to-dial, flexible dial plan, hot-desking, personalized music ringtones and music on hold, server redundancy and fail-over

HD audio

Yes, HD handset and speakerphone with support for wideband audio

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Base Stand

Yes, 2 angle positions available, Wall Mountable (*wall mount sold separately)

Wall Mountable

Yes, (*wall mount sold separately)

QoS

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

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control, secure boot

Multi-language

English, German, Italian, French, Spanish, Portuguese, Russian, Croatian, Chinese, Korean, Japanese, and more

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

Power & Green Energy Efficiency

Universal Power Supply Input 100-240VAC 50-60Hz; Output +5VDC, 600mA;PoE: IEEE802.3af Class 2, 3.84W-6.49W; IEEE802.3az (EEE) (GRP2604P Only)

Physical

Dimension: 208mm (L) x 180mm (W) x 63.4mm (H) (with handset) Unit weight: 670g; Package weight:830g (880g for GRP2602)

Temperature and Humidity

Operation: 0°C to 40°C Storage: -10°C to 60°C Humidity: 10% to 90% Noncondensing

Package Content

GRP2604 phone, handset with cord, base stand, universal power supply (GRP2604 only), network cable, Quick Installation Guide

FCC: Part 15 Class B; FCC Part 68 HAC;

Compliance

CE: EN 55032; EN 55035; EN 61000-3-2; EN 61000-3-3; EN 62368-1; RCM: AS/NZS CISPR32; AS/NZS 62368.1; AS/CA S004;

IC: ICES-003; CS-03;

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GETTING STARTED
This chapter provides basic installation instructions including the list of the packaging contents and information for obtaining the best performance with the GRP260X phone.
Equipment Packaging
Table 6: Equipment Packaging
GRP260X
· 1 x GRP260X Main Case. · 1 x Handset. · 1 x Phone Stand. · 1 x Ethernet Cable. · 1 x Power Adapter (Except for GRP260xP). · 1 x Phone cord. · 1 x Quick Installation Guide.

Figure 1: GRP260X Package Content
Note: Check the package before installation. If you find anything missing, contact your system administrator.

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GRP260X Phone Setup
The GRP260X phones can be installed on the desktop using the phone stand or attached on the wall using the slots for wall mounting.
Figure 2: Phone Stand and Mounting Slots on the GRP260X
Using the Phone Stand For installing the phone on the table with the phone stand, attach the phone stand to the bottom of the phone where there is a slot for the phone stand. (Upper half, bottom part).
Using the Slots for Wall Mounting 1. Attach the wall mount spacers to the slot for wall mount spacers on the back of the phone. 2. Attach the phone to the wall via the wall mount hole. 3. Pull out the tab from the handset cradle (See figure below). 4. Rotate the tab and plug it back into the slot with the extension up to hold the handset while the phone is mounted on the wall (See figure below).

Figure 3: Tab on the Handset Cradle
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Connecting the GRP260X
To setup the GRP260X, follow the steps below:
1. Connect the handset and main phone case with the phone cord.
2. Connect the LAN port of the phone to the RJ-45 socket of a hub/switch or a router (LAN side of the router) using the Ethernet cable.
3. Connect the 5V DC output plug to the power jack on the phone; plug the power adapter into an electrical outlet. If PoE switch is used in step 3, this step could be skipped (For GRP260xP).
4. The LCD will display Grandstream logo. Before continuing, please wait for the date/time display to show up.
5. Using the phone embedded web server or keypad configuration menu, you can further configure the phone using either a static IP or DHCP.
Using Wi-Fi (GRP2602W only): - On LCD menu, navigate to "Settings  Wi-Fi settings" and enable Wi-Fi.
- Select "Wi-Fi Network" and GRP2602W will automatically start scanning within the range.
- A list of Wi-Fi networks will be displayed. Select the desired network, and if required, enter the correct password to connect.
- Users can add and connect to a hidden network by accessing "Wi-Fi" Network" and then press on Add softkey . Then enter the Network's information. Notes: · When the GPR2602W is not connected to any network (including Ethernet and Wi-Fi), a prompt
interface will pop up to notify users about it. Users can quickly enter "Wi-Fi Network" page by pressing on the Wi-Fi softkey. · For easy deployment, The GRP2602W is out of the box is preconfigured to connect to a default SSID named wp_master with a password (WPA/WPA2 PSK) equal to wp!987@dmin, users can adapt these settings from the web UI as well to make it easier for deployment on customer site.

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Figure 4: GRP260X Back Side View

Configuration via Keypad

To configure the LCD menu using phone's keypad, follow the instructions below:
· Enter MENU options. When the phone is in idle, press the round MENU button or Menu Softkey to enter the configuration menu.

· Navigate in the menu options. Press the arrow keys UP/DOWN keys to navigate in the menu options.

· Enter/Confirm selection. Press the round MENU to enter the selected option.

· Exit. Press Return Softkey · Return to Home page.

to exit to the previous menu.

The MENU options are listed in the following table.

Table 7: Configuration Menu

Status

Displays account status, network status, software version number and Hardware · Network status. Press to enter the sub menu for IP setting information (DHCP/Static IP/PPPoE), Ipv4 address, Ipv6 address, MAC address, Subnet Mask, Gateway, DNS and NTP servers. · Account status. Shows Account registration status.

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Settings

· System Status Press to enter the sub menu for Hardware Information, Software version and IP Geographic Information.
Settings sub menu contains the following options: · Account Settings Enable/Disable SIP account, Configures Account Name, SIP server's address, SIP User ID, SIP Auth ID, SIP Password, Outbound Proxy, and Voice Mail Access Number. · Call Settings Enable/Disable DND, Enable Disable Auto Answer for SIP account, Enable/Disable Call Forward (Forward All/Busy/No Answer). · Basic Settings
o Sounds Configures account ringtone and adjusts volume settings by pressing left/right arrow key.
o Date Time Adjusts Time and Date displaying format.
o Time Zone Choose your Time Zone from the list by scrolling with UP/DOWN keys.
o Language Selects the language to be displayed on the phone's LCD. Users could select Automatic for local language based on IP location if available. By default, it is Auto.
o Keypad Lock Enables/Disables Keypad lock. Users can choose the Keypad Lock type (All Keys/Functional keys) and set up the lock password. If users enabled Keypad lock without configuring a password; They can unlock the phone by pressing on the unlock softkey.
o Headset Type Choose the headset type of the headset connect to the phone. Users could choose Normal, Plantronics EHS, Jabra EHS, Sennheiser EHS. · Advanced settings o Upgrade Check for upgrade by contacting the firmware upgrade server.

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Messages Call History

o Syslog Configures Syslog level, Transport protocol and Syslog Server's address.
o Security Enables disables Web and SSH access.
o Alternative Firmware Press softkey to switch between the dual firmware versions loaded to the phone. The phone will reboot with the chosen version.
o Factory Reset: Perform Factory reset to the phone. All device configuration and user data will be lost after factory reset. · Ethernet Settings
o Internet Protocol Selects Prefer IPv4 / Prefer IPv6 / IPv4 only or IPv6 only. The default setting is "IPv4 only".
o IPv4 Setting Selects IP mode (DHCP/Static IP/PPPoE); Configures PPPoE account ID and password; Configures static IP address, Netmask, Gateway, Preferred DNS server.
o IPv6 Setting Selects IP mode (DHCP/Static IP); Configures static IP address, IPv6 Prefix (64 bits), IPv6 Preferred DNS server. · Wi-Fi Settings (GRP2602W only)
o Wi-Fi Enables/disables Wi-Fi;
o Wi-Fi Band Choose Wi-Fi band (2G , 5G or 2G&5G).
o Wi-Fi Network Scans and displays available Wi-Fi networks.
· Instant Messages Displays received instant messages · Voicemail Displays voicemail message information in the following format: Normal/Urgent
Displays Local Call Logs: "All" Calls / "Missed" Calls/ "Dialed" Calls/ "Answered" Calls.

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Contacts Factory Functions Reboot

Contacts sub menu includes the following options: · Local Phonebook · Local Group · LDAP
User could configure phonebooks/groups options here, download phonebook XML to the phone and search and dial from the local phonebook and search and dial from LDAP phonebook.
Factory Functions sub menu includes the following options: · Audio Loopback Speak to the phone using speaker/handset/headset. If you can hear your voice, your audio is working fine. Press Menu button to exit audio loopback mode. · Keypad/Led diagnosis All LEDs will light up Press all the available keys on the phone. The LCD will display the name for the keys to be pressed to finish the keyboard diagnostic mode. Press Hook button to exit. · Certification Verification Verify the certificate loaded on the phone.
Reboots the phone.

The following picture shows the keypad MENU configuration flow:

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Figure 5: GRP260x LCD settings
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Configuration via Web Browser
The GRP260X embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Google Chrome, Mozilla Firefox, and Microsoft's IE. To access the Web GUI:
1. Connect the computer to the same network as the phone. 2. Make sure the phone is turned on and shows its IP address. You may check the IP address by
pressing the Up-arrow button when phone is at idle state. 3. Open a Web browser on your computer. 4. Enter the phone's IP address in the address bar of the browser. 5. Enter the administrator's login and password to access the Web Configuration Menu.

Notes:

· The computer must be connected to the same sub-network as the phone. This can be easily done by connecting the computer to the same hub or switch as the phone connected to. In absence of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the PC port on the back of the phone.
· If the phone is properly connected to a working Internet connection, the IP address of the phone will display in MENUStatusNetwork Status. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255. Users will need this number to access the Web GUI. For example, if the phone has IP address 192.168.40.154, please enter "http://192.168.40.154" in the address bar of the browser.
· There are two default passwords for the login page:

User Level End User Level

User user

Administrator Level

admin

Password

Web Pages Allowed

Only Status and Basic 123
Settings

Random password available on the sticker at the back of the unit.

Browse all pages

· When changing any settings, always SUBMIT them by pressing the "Save" or "Save and Apply" button
on the bottom of the page. If the change is saved only but not applied, after making all the changes,

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click on the "APPLY" button on top of the page to submit. After submitting the changes in all the Web GUI pages, reboot the phone to have the changes take effect if necessary (All the options under "Accounts" page and "Phonebook" page do not require reboot. Most of the options under "Settings" page do not require reboot).
Saving Configuration Changes
After users makes changes to the configuration, press the "Save" button will save but not apply the changes until the "Apply" button on the top of web GUI page is clicked. Or users could directly press "Save and Apply" button. We recommend rebooting or powering cycle the phone after applying all the changes.
Rebooting from Remote Locations
Press the "Reboot" button on the top right corner of the web GUI page to reboot the phone remotely. The web browser will then display a reboot message. Wait for about 1 minute to log in again.

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CONFIGURATION GUIDE

This section describes the options in the phone's Web GUI. As mentioned, you can log in as an administrator or an end user.
· Status: Displays the Account status, Network status, and System Info of the phone. · Account: To configure the SIP account. · Phone Settings: To configure phone general settings, Call Settings, ringtone, Multicast Paging. · Network Settings: To configure network settings. · Programmable keys: Configures idle and call screen softkeys, And the Multi-purpose keys settings
for the GRP2604 only. · System Settings: Configures Time and Language settings, Security Settings, Preferences, TR-069. · Maintenance: To configure web access, upgrading and provisioning, syslog, language settings, TR-
069, security , Outbound Notifications , Voice monitoring etc. · Application: Configures Web Service settings, Contacts, LDAP , Call History. · External Service: Configures GDS Settings, Call Center, BroadSoft XSI.

Status Page Definitions

Table 8: Status Page Definitions

Status  Account Status

Account

Account index. · For GRP2601/GRP2601P: 2 SIP accounts · For GRP2602/GRP2602P/GRP2602W: 4 SIP accounts · For GRP2603/GRP2603P: 6 SIP accounts · For GRP2604/GRP2604P: 6 SIP accounts

SIP User ID

Displays the configured SIP User ID for the account.

SIP Server

Displays the configured SIP Server address, URL or IP address, and port of the SIP server.

SIP Registration

Displays SIP registration status for the SIP account, it will display Yes/No with Green/Red background.

Status  Network Status

MAC Address

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

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IP Setting IPv4 Address IPv6 Address OpenVPN® IP Subnet Mask

The configured address type: DHCP, Static IP or PPPoE. The IPv4 address obtained on the phone. The IPv6 address obtained on the phone. The OpenVPN® IP obtained on the phone. The subnet mask obtained on the phone.

Gateway

The gateway address obtained on the phone.

DNS Server 1

The DNS server address 1 obtained on the phone.

DNS Server 2

The DNS server address 2 obtained on the phone.

PPPoE Link Up NAT Type NAT Traversal Status  System Info Product Model Part Number
Software Version
IP Geographic Information System Up Time

PPPoE connection status.
The type of NAT connection used by the phone. Display the status of NAT connection for each account on the phone.
Product model of the phone. Product part number. · Boot: boot version number. · Core: core version number. · Base: base version number. · Prog: program version number. This is the main firmware release
number, which is always used for identifying the software system of the phone. · Locale: locale version number. · Language: displaying language. · Time Zone: displaying time zone; System up time since the last reboot.

System Time

Current system time on the phone system.

Service Status

GUI, Phone and CPE service status.

System Information User Space

Download system information Shows the percentage of the user space used and the status of the Database

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Core Dump Special Feature

Shows the status of the core dump and the core dump files generated if any. It also gives the ability to generate GUI/Phone core dump files manually.
OpenVPN® Support: displaying if the phone supports OpenVPN®.

Account Page Definitions

Account x  General Settings

Table 9: Account Page Definitions

Account Register

Account Active Account Name SIP Server
Secondary SIP Server
SIP User ID SIP Authentication ID SIP Authentication Password Name
TEL URI

This field indicates whether the account is active. The default setting is "No". The name associated with each account to be displayed on the LCD. The URL or IP address, and port of the SIP server. This is provided by your VoIP service provider (ITSP). The URL or IP address, and port of the SIP server. When configured, phone will register to both Primary and Secondary SIP Server. If Primary SIP Server is not reachable then the phone will use Secondary SIP Server for phone services (including making/receiving calls). User account information provided by your VoIP service provider (ITSP). It is usually in the form of digits like phone number or actually a phone number. SIP service subscriber's Authenticate ID 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 ITSP (SIP) server before the account can be registered. After it is saved, this will appear as hidden for security purpose. The SIP server subscriber's name (optional) that will be used for Caller ID display. If the phone has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the Request-Line and "TO" header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request.

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Voice Mail Access Number BLF Server Account Display Network Settings Outbound Proxy Secondary Outbound Proxy
DNS Mode
DNS SRV Failover Mode

Allows users to access voice messages by pressing the MESSAGE button on the phone. This value is usually the VM portal access number. Optional server used for SUBSCRIBE requests to indicate other extensions status on the SIP server. This option allows you to configure how your SIP account label will be displayed on the phone's screen. If set to "Username", LCD account label will display the Account Name configured for this SIP account. If set to "User ID", it will then display the SIP User ID configured for this SIP account.
IP address or Domain name of the Primary Outbound Proxy, Media Gateway, or Session Border Controller. It is used by the phone for Firewall or NAT penetration in different network environments. If a symmetric NAT is detected, STUN will not work and ONLY an Outbound Proxy can provide a solution. IP address or Domain name of the Secondary Outbound Proxy which will be used when the primary proxy cannot be connected. This parameter controls how the Search Appliance looks up IP addresses for hostnames. If "Use Configured IP" is selected, please fill in Primary IP, Backup IP 1, and Backup IP 2.
· A Record · SRV · NAPTR/SRV · Use Configured IP Configure the preferred IP mode for DNS SRV. If selected "default", first IP from query result will be applied; If selected "Saved one until DNS TTL", previous IP will be applied before reaches DNS timeout; If selected "Saved one until no response", and previous IP will be applied even after DNS time out until it cannot response. · Default If the option is set with "default", it will again try to send register messages to one IP at a time, and the process repeats. · Saved one until DNS TTL If the option is set with "Saved one until DNS TTL", it will send register messages to the previously registered IP first. If no response, it will try to

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Primary IP Backup IP 1 Backup IP 2
NAT Traversal

send one at a time for each IP. This behavior lasts if DNS TTL (time-to-live) is up. · Saved one until no responses If the option is set with "Saved one until no responses", it will send register messages to the previously registered IP first, but this behavior will persist until the registered server does not respond. Configures the primary IP address where the phone sends DNS query to when "Use Configured IP" is selected for DNS mode.
Configures the backup IP 1 address where the phone sends DNS query to when "Use Configured IP" is selected for DNS mode.
Configures the backup IP 2 address where the phone sends DNS query to when "Use Configured IP" is selected for DNS mode. This parameter configures whether the NAT traversal mechanism is activated. Users could select the mechanism from No, STUN, Keep-alive, UPnP, Auto or VPN. The default setting is "Auto".
If set to "STUN" and STUN server is configured, the phone will route according to the STUN server. If NAT type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone will try to use public IP addresses and port number in all the SIP&SDP messages.

The phone will send empty SDP packet to the SIP server periodically to keep the NAT port open if it is configured to be "Keep-alive". Configure this to be "No" if an outbound proxy is used. "STUN" cannot be used if the detected NAT is symmetric NAT. Set this to "VPN" if OpenVPN is used.

A SIP Extension to notify the SIP server that the phone is behind a

Proxy-Require

NAT/Firewall. Do not configure this parameter unless this feature is supported on the SIP server.

Use SBC

Indicate whether an SBC server is used. If users want to work under SBC associated with 3CX, they should enable this feature to have better communication with the server.

Account x  SIP Settings

Basic Settings

SIP Registration

Selects whether the phone will send SIP Register messages to the proxy/server. The default setting is "Enabled".

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UNREGISTER on Reboot

Allows the SIP user's registration information to be cleared when the phone reboots. The SIP REGISTER message will contain "Expires: 0" to unbind the connection. Three options are available: The default setting is "No". · If set to "All", the SIP user's registration information will be cleared when
the phone reboots. The SIP Contact header will contain "*" to notify the server to unbind the connection. · If set to "Instance", the SIP user will be unregistered on current phone only. · If set to "No", the phone will not unregister the SIP account when rebooting.

REGISTER Expiration

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

SUBSCRIBE Expiration

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

Re-Register before Expiration

Specifies the time frequency (in seconds) that the phone sends reregistration request before the Register Expiration. The default value is 0.

Registration Retry Wait Time

Specifies the interval to retry registration if the process is failed. The valid range is 1 to 3600. The default value is 20 seconds.

Add Auth Header on Initial REGISTER

Add authorization Header on initial REGISTER from the first REGISTER

Enable OPTIONS Keep Enable OPTIONS Keep Alive to check SIP Server.

Alive

Default is "Disabled".

OPTIONS Keep Alive Interval

Time interval for OPTIONS Keep Alive feature in seconds. Default is "30" seconds.

OPTIONS Keep Alive Max Lost

Number of max lost packets for OPTIONS Keep Alive feature before the phone re-registration. Default is "3".

SUBSCRIBE for MWI

When set to "Yes", a SUBSCRIBE for Message Waiting Indication will be sent periodically. The phone supports synchronized and non-synchronized MWI. The default setting is "No".

SUBSCRIBE for Registration

When set to "Yes", a SUBSCRIBE for Registration will be sent out periodically. The default setting is "No".

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Controls whether the Privacy header will present in the SIP INVITE message or not, whether the header contains the caller info.

Use Privacy Header

· Default: The Privacy Header will show in INVITE only when "Huawei IMS" special feature is on.
· Yes: The Privacy Header will always show in INVITE.

· No: The Privacy Header will not show in INVITE. The default setting is "default". Controls whether the P-Preferred-Identity Header will present in the SIP INVITE message.

Use P-PreferredIdentity Header

· Default: The P-Preferred-Identity Header will show in INVITE unless "Huawei IMS" special feature is on.
· Yes: The P-Preferred-Identity Header will always show in INVITE.

· No: The P-Preferred-Identity Header will not show in INVITE.
The default setting is "default".

Use X-GrandstreamPBX Header

Enables / disables the use of X-Grandstream-PBX header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is "Yes".

Use P-AccessNetwork-Info Header

Enables / disables the use of P-Access-Network-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is "Yes".

Use P-Emergency-Info Header

Enables / disables the use of P-Emergency-Info header in SIP request. When disabled, the SIP message sent from the phone will not include the selected header. Default setting is "Yes".

· If Yes for REGISTER only, the sip message for register or unregister

will contains MAC address in the header, and all the outgoing SIP

messages except REGISTER message will attach the MAC address to

the User-Agent header.

Use MAC Header

· If Yes to all SIP, the sip message for register or unregister will contains MAC address in the header, and all the outgoing SIP message including REGISTER will attach the MAC address to the User-Agent header.

· If No, neither will the MAC header be included in the register or

unregister message nor the MAC address be attached to the User-Agent

header for any outgoing SIP message.

The default setting is "No".

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Add MAC in UserAgent
SIP Transport

If Yes except REGISTER, the SIP messages for register or unregister will not contain MAC address in the User-Agent header, and all the other outgoing SIP messages will attach MAC address to the User-Agent header; If Yes to ALL, all the outgoing SIP messages will attach MAC address to the User-Agent header; If No, MAC address will not be attached to the UserAgent header for any outgoing SIP message. Determines the network protocol used for the SIP transport. Users can choose from TCP, UDP and TLS. The default setting is "UDP". Determines whether to listen to multiple SIP protocols.

· Transport Only: will listen to configured transport protocol only.

· Dual: will listen to TCP when UDP is selected.

SIP Listening Mode

· Dual (Secured): will listen to TLS/TCP when UDP is selected. If TCP or TLS/TCP is selected, UDP will be listened to

· Dual (BLF Enforced): will try to enforce BLF subscriptions to use TCP protocol by adding 'transport=tcp' to the Contact header.
The default setting is "Transport Only".

Local SIP Port

Defines the local SIP port used to listen and transmit.

SIP URI Scheme when Specifies if "sip" or "sips" will be used when TLS/TCP is selected for SIP

using TLS

Transport. The default setting is "sips".

Use Actual Ephemeral Port in Contact with TCP/TLS

This option is used to control the port information in the Via header and Contact header. 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".

Support SIP Instance ID Defines whether SIP Instance ID is supported or not. Default setting is "Yes".

SIP T1 Timeout is an estimate of the round-trip time of transactions between

SIP T1 Timeout

a client and server. If no response is received the timeout is increased and request re-transmit retries would continue until a maximum amount of time

define by T2. The default setting is 0.5 seconds.

SIP T2 Timeout is the maximum retransmit time of any SIP request

SIP T2 Timeout

messages (excluding the INVITE message). The re-transmitting and doubling of T1 continues until it reaches the T2 value. Default is 4 seconds.

The Outbound proxy mode is placed in the route header when sending SIP Outbound Proxy Mode
messages, or they can be always sent to outbound proxy.

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· In route · Not in route · Always send to Default is "in route".

Enable 100rel

The use of the PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional responses (1xx series). This is very important to support PSTN internetworking. To invoke a reliable provisional response, the 100rel tag is appended to the value of the required header of the initial signaling messages. The default setting is "No".

Session Timer

Enable Session Timer

This option is used to enable or disable session timer on the phone side when server side can provide both session timer UPDATE and session audit UPDATE. The default setting is "No".

Session Expiration

The SIP Session Timer extension (in seconds) that enables SIP sessions to be periodically "refreshed" via a SIP request (UPDATE, or re-INVITE). If there is no refresh via an UPDATE or re-INVITE message, the session will be terminated once the session interval expires. Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default setting is 180. The valid range is from 90 to 64800.

Min-SE

The minimum session expiration (in seconds). The default value is 90 seconds. The valid range is from 90 to 64800.

Caller Request Timer

If set to "Yes" and the remote party supports session timers, the phone will use a session timer when it makes outbound calls. The default setting is "No".

Callee Request Timer

If set to "Yes" and the remote party supports session timers, the phone will use a session timer when it receives inbound calls. Default setting is "No".

Force Timer

If Force Timer is set to "Yes", the phone will use the session timer even if the remote party does not support this feature. If Force Timer is set to "No", the phone will enable the session timer only when the remote party supports this feature. To turn off the session timer, select "No". The default setting is "No".

As a Caller, select UAC to use the phone as the refresher; or select UAS to UAC Specify Refresher
use the Callee or proxy server as the refresher. The default setting is "UAC".

As a Callee, select UAC to use caller or proxy server as the refresher; or UAS Specify Refresher
select UAS to use the phone as the refresher. The default setting is "UAC".

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The Session Timer can be refreshed using the INVITE method or the

Force INVITE

UPDATE method. Select "Yes" to use the INVITE method to refresh the

session timer. The default setting is "No".

Account x  Codec Settings

Audio

Preferred Vocoder (Choice 1 ­ 8)

Multiple vocoder types are supported on the phone, the vocoders in the list is a higher preference. Users can configure vocoders in a preference list that is included with the same preference order in SDP message.

Configures the phone to use which codec sequence to negotiate as the

Codec Negotiation Priority

callee. When set to "Caller", the phone negotiates by SDP codec sequence from received SIP Invite. When set to "Callee", the phone negotiates by audio codec sequence on the phone. The default setting is "Callee".

Use First Matching

When it is set to "Yes", the device will use the first matching vocoder in the

Vocoder in 200OK SDP received 200OK SDP as the codec. The default setting is "No".

iLBC Frame Size

This option determines the iLBC packet frame size. Users can choose from 20ms and 30ms. The default setting is "30ms".

iLBC Payload Type

This option is used to specify iLBC payload type. Valid range is 96 to 127. The default setting is "97".

G.726-32 Packing Mode

Selects "ITU" or "IETF" for G726-32 packing mode. The default setting is "ITU".

G.726-32 Dynamic Payload Type

Specifies G.726-32 payload type. Valid range is 96 to 127. Default is 127.

Opus Payload Type

Specifies OPUS payload type. Valid range is 96 to 127. Cannot be the same as iLBC or DTMF Payload Type. Default value is 123.

This parameter specifies the mechanism to transmit DTMF digits. There are 3 supported modes:

Send DTMF

· In audio: DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs).
· RFC2833 sends DTMF with RTP packet. Users can check the RTP packet to see the DTMFs sent as well as the number pressed.

· SIP INFO uses SIP INFO to carry DTMF. Default setting is "RFC2833".

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DTMF Payload Type Enable Audio RED with FEC Audio FEC Payload Type Audio RED Payload Type Silence Suppression
Jitter Buffer Type Jitter Buffer Length
Voice Frames Per TX
G.723 Rate RTP Settings
SRTP Mode
SRTP Key Length

Configures the payload type for DTMF using RFC2833. Cannot be the same as iLBC or OPUS payload type. If set to "Yes", FEC will be enabled for audio call. The default setting is "Yes". Configures audio FEC payload type. The valid range is from 96 to 126. The default value is 121. Configures audio RED payload type. The valid range is from 96 to 126. The default value is 124. Controls the silence suppression/VAD feature of the audio codecs except forG.723 (pending) and G.729. If set to "Yes", a small quantity of RTP packets containing comfort noise will be sent during the periods of silence. If set to "No", this feature is disabled. Default setting is "No" Selects either Fixed or Adaptive for jitter buffer type, based on network conditions. The default setting is "Adaptive". Selects jitter buffer length from 100ms to 800ms, based on network conditions. The default setting is "300ms". Configures the number of voice frames transmitted per packet. When configuring this, it should be noted that the "ptime" value for the SDP will change with different configurations here. This value is related to the codec used and the actual frames transmitted during the in-payload call. For end users, it is recommended to use the default setting, as incorrect settings may influence the audio quality. The default setting is 2. Selects encoding rate for G723 codec.
Enable SRTP mode based on your selection from the drop-down menu. · No · Enabled but Not forced · Enabled and Forced · Optional
The default setting is "No". Allows users to specify the length of the SRTP calls. Available options are:
· AES 128&256 bit · AES 128 bit · AES 256 bit Default setting is AES 128&256 bit

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Crypto Lifetime

Enable or disable the crypto lifetime when using SRTP. If users set to disable this option, phone does not add the crypto lifetime to SRTP header. The default setting is "Yes".

VQ RTCP-XR Collector Configure the host name of the central report collector that accepts voice

Name

quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Configure the IP address of the central report collector that accepts voice

Address

quality reports contained in SIP PUBLISH messages.

VQ RTCP-XR Collector Configure the port of the central report collector that accepts voice quality

Port

reports contained in SIP PUBLISH messages.

Symmetric RTP

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

Configures the RTP timeout of the phone. If the phone does not receive the

RTP Timeout (s)

RTP packet within the specified RTP time, the call will be automatically disconnected. The default range is 0 and 6-600. If set to 0, the phone will not hang up the call automatically.

Account x  Call Settings

Call Features

Auto Answer

If set to "Yes", the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep. Default setting is "No".

Custom Alert-Info for Auto Answer

Allows to customize Alert-Info header for auto answer. The phone will auto answer only if matching content of the custom Alert-info header.

Allow Auto Answer by Call-Info/Alert-Info

Allows the phone to automatically turn on the speaker phone to answer incoming calls after a short reminding beep when enabled, based on the SIP Call-Info/Alert-Info header sent from the server/proxy. Default is "Yes".

Allow Barging by CallInfo/Alert-Info

When enabled, the phone will automatically put the current call on hold and answer the incoming call 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 CallInfo/Alert-Info headers will be rejected automatically. Default setting is "No".

Send Anonymous

If set to "Yes", the "From" header in outgoing INVITE messages will be set to anonymous, blocking the Caller ID to be displayed. Default is "No".

Anonymous Call Rejection

If set to "Yes", anonymous calls will be rejected. The default setting is "No".

Call Log

Configures Call Log setting on the phone. · Log All Calls

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Transfer on Conference Hangup Key as Send
Record Key Default Function
Call Recording On Call Recording Off
Enable Recovery on Blind Transfer
Blind Transfer Wait Timeout No Key Entry Timeout

· Log incoming/Outgoing Only (missed calls NOT recorded) · Disable Call Log The default setting is "Log All Calls". If set to "Yes", when the phone hangs up as the conference initiator, the conference call will be transferred to the other parties so that other parties will remain in the conference call. The default setting is "No". Allows users to configure either the "*" or "#" keys as the "Send" key. Please make sure the dial plan is properly configured to allow dialing * and # out. The default setting is "Pound (#)". Configure whether to turn the recording function on or off when the "Record" key is pressed for the first time in a call under this account, and switch between the two. For example, the SIP server can be configured with the automatic start call recording function. In this case, Record key default function needs to be configured as "Record off". Configures the DTMF sequence sent when pressing the Record key during a call on this account. Toggles between this value and the off code if possible; otherwise always sends this code. Configures the DTMF sequence sent when pressing the Recording key during a call on this account when turning recording off. Disables recovery to the call to the transferee on failing blind transfer to the target. The default setting is "Yes". Notes: 1) This feature only applies to blind transfer. 2) This feature depends on how server handles transfer. If there is any NOTIFY from server, this feature will not take effect. If server responds 4xx, phone should try to recover regardless of this option. 3) During blind transfer, after transferor received 200/202 for REFER, but there is no NOTIFY from server after 7 seconds, transferor will decide to recover the call with transferee or not depending on the options. This is the only case that this option will be applied. Defines the timeout (in seconds) for waiting SIP frag response in blind transfer. Valid range is 30 to 300. Default setting is "30". Defines the timeout (in seconds) for no key entry. If no key is pressed after the timeout, the digits will be sent out. The default value is 4 seconds. The valid range is from 1 to 15.

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Ring Timeout Refer-To Use Target Contact RFC2543 Hold Enable Call Waiting Dial plan Dial Plan Prefix Bypass Dial Plan
Dial Plan

Defines the timeout (in seconds) for the rings on no answer. The default setting is 60. The valid range is from 10 to 300. If set to "Yes", the "Refer-To" header uses the transferred target's Contact header information for attended transfer.
Allows users to toggle between RFC2543 hold and RFC3261 hold. RFC2543 hold (0.0.0.0) allows user to disable the hold music sent to the other side. RFC3261 (a line) will play the hold music to the other side. The default setting is "No". Enables / disables the call waiting feature for the current account. When set to "Default", global call feature setting will be used. Default value is "Default".
Configures the prefix added to dialed number.
Bypass the dial plan when dialing from one of the available items: Contacts ; Call History Incoming Call ; Call History Outgoing Call ; Dialing Page ; MPK ; API A dial plan establishes the expected number and pattern of digits for a telephone number. This parameter configures the allowed dial plan for the phone. Dial Plan Rules: 1. Accepted Digits: 1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d; 2. Grammar: x ­ any digit from 0-9; 3. Grammar: X ­ any character from 0-9, a-z, A-Z. a) xx+ - at least 2 digit numbers b) xx ­ only 2 digit numbers c) XX ­ two characters ( AA, Ab, 1C, f5, 68,...) d) \t\e\s\t : only string "test" will pass the dial plan check e) ^ - exclude f) [3-5] ­ any digit of 3, 4, or 5 g) [147] ­ any digit of 1, 4, or 7 h) <2=011> - replace digit 2 with 011 when dialing i) | - the OR operand · Example 1: {[369]11 | 1617xxxxxxx} Allow 311, 611, and 911 or any 11 digit numbers with leading digits 1617; · Example 2: {^1900x+ | <=1617>xxxxxxx}

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Call Display Caller ID Display
Callee ID Display Ringtone

Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit numbers; · Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} Allows any number with leading digit 1 followed by a 3-digit number, followed by any number between 2 and 9, followed by any 7-digit number OR Allows any length of numbers with leading digit 2, replacing the 2 with 011 when dialed.
Example of a simple dial plan used in a Home/Office in the US: { ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }
Explanation of example rule (reading from left to right): · ^1900x. ­ prevents dialing any number started with 1900; · <=1617>[2-9]xxxxxx ­ allows dialing to local area code (617) numbers
by dialing7 numbers and 1617 area code will be added automatically; · 1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11
digits length; · 011[2-9]x ­ allows international calls starting with 011; · [3469]11 ­ allows dialing special and emergency numbers 311, 411, 611
and 911. Note: In some cases, where the user wishes to dial strings such as *123 to activate voice mail or other applications provided by their service provider, the * should be predefined inside the dial plan feature. An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of numbers. Max length of dial plan is up to 1024 characters.
When set to "Auto", the phone will look for the caller ID in the order of PAsserted Identity Header, Remote-Party-ID Header and From Header in the incoming SIP INVITE. When set to "Disabled", all incoming calls are displayed with "Unavailable".
When set to "Auto", the phone will update the callee ID in the order of PAsserted Identity Header, Remote-Party-ID Header and To Header in the 180 Ringing. When set to "Disabled", callee id will be displayed as "Unavailable". When set to "To Header", caller id will not be updated and displayed as To Header.

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Account Ring Tone

Allows users to configure the ringtone for the account. Users can choose from different ringtones from the dropdown menu. Note: User can also choose silent ring tone.

Specifies matching rules with number, pattern, or Alert Info text (up to 10

matching rules). When the incoming caller ID or Alert Info matches the rule,

the phone will ring with selected distinctive ringtone. Matching rules:

· Specific caller ID number. For example, 8321123.

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

xx+ : at least 2-digit number.

xx : only 2-digit number.

Match Incoming Caller

[345]xx: 3-digit number with the leading digit of 3, 4 or 5.

ID

[6-9]xx: 3-digit number with the leading digit from 6 to 9.

· Alert Info text

Users could configure the matching rule as certain text (e.g., priority) and

select the custom ring tone mapped to it. The custom ring tone will be

used if the phone receives SIP INVITE with Alert-Info header in the

following format: Alert-Info: <http://127.0.0.1>; info=priority

Selects the distinctive ring tone for the matching rule. When the incoming

caller ID or Alert Info matches one of the 10 rules, the phone will ring with

the associated ringtone.

Account x  Advanced Settings

Security Settings

Check Domain Certificates

Choose whether the domain certificates will be checked or not when TLS/TCP is used for SIP Transport. The default setting is "No".

Validate Certificate Chain

Validate certification chain when TCP/TLS is configured. Default setting is "No".

Validate Incoming SIP Messages

Choose whether the incoming messages will be validated or not. The default setting is "No".

Omit charset=UTF-8 in Omit charset=UTF-8 in MESSAGE content-type
MESSAGE

Allow Unsolicited REFER

Allow unsolicited REFER to accomplish an outgoing call

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Accept Incoming SIP from Proxy Only

When set to "Yes", the SIP address of the Request URL in the incoming SIP message will be checked. If it does not match the SIP server address of the account, the call will be rejected. The default setting is "No".

Check SIP User ID for Incoming INVITE

If set to "Yes", SIP User ID will be checked in the Request URI of the incoming INVITE. If it does not match the phone's SIP User ID, the call will be rejected. The default setting is "No".

Allow SIP Reset

Allow SIP Notification message to perform factory reset

Authenticate Incoming If set to "Yes", the phone will challenge the incoming INVITE for

INVITE

authentication with SIP 401 Unauthorized response. Default setting is "No".

MOH

On Hold Reminder Tone

Plays a reminder tone when you have a call on hold

Music On Hold URI

Music On Hold URI to call when a call is on hold if server supports it.

Advanced Features

Special Feature

Different soft switch vendors have special requirements. Therefore, users may need select special features to meet these requirements. Users can choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro, Huawei IMS, PhonePower and UCM Call center depending on the server type. The default setting is "Standard".

Feature Key Synchronization

This feature is used for Broadsoft call feature synchronization. When it is enabled, DND, Call Forward features and Call Center Agent status can be synchronized between Broadsoft server and phone. Default is "Disabled".

Conference URI

Configures the conference URI when using Broadsoft N-way calling feature.

Broadsoft Call Center

When set to "Yes", a Softkey "BSCCenter" is displayed on LCD. User can access different Broadsoft Call Center agent features via this Softkey. Please note that "Feature Key Synchronization" will be enabled regardless of this setting. Default setting is "No". Note: To activate this feature, users need to change the special feature to Broadsoft and setup the Broadsoft Call Center to take effect.

Hoteling Event

Broadsoft Hoteling event feature. Default setting is "No". With "Hoteling Event" enabled, user can access the Hoteling feature option by pressing the "BSCCenter" softkey.

Call Center Status

When set to "Yes", the phone will send SUBSCRIBE to the server to obtain call center status. The default setting is "No".

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Broadsoft Executive Assistant

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

Broadsoft Call Park

When enabled, it will send SUBSCRIBE to Broadsoft server to obtain Call Park notifications. The default setting is "Disabled".

SCA

Enable SCA (Shared Call Appearance)

If set to "Yes", the Shared Call Appearance (BroadSoft Standard) will be used for the registered account.

Line-Seize Timeout

Configures the interval (in seconds) when the line-seize is considered timed out when Shared Call Appearance feature is used. Valid range is 15 to 60.

BLF (Busy Lamp Field) - GRP2604 only

Configures Presence Eventlist URI to monitor the extensions on MultiPresence Eventlist URI
Purpose Keys.

Eventlist BLF URI

Configures Eventlist BLF URI to monitor the extensions on Multi-Purpose Keys

Auto provision Eventlist

Select the type of Eventlist to get automatically provisioned onto available MPKs. Whether its BLF Eventlist or Presence Eventilist.

Configures BLF Call-pickup method:

BLF Call-pickup

· If select Auto: The phone will do either Prefix or barge in code for BLF pickup depend on which on is set.
· If select Force BLF Call-pickup by prefix: The phone will only use Prefix as BLF pickup method.

· If select Disabled: The phone will ignore both BLF pickup method, now the monitored VPK will only dial the extension if pressed

Configures the prefix prepended to the BLF extension when the phone picks BLF Call-pickup Prefix
up a call with BLF key. The default setting is **.

Call Pickup Barge-In Code
PUBLISH for Presence Account x  Dial Plan

Set feature access code of Call Pickup with Barge-In feature.
Enables presence feature on the phone. The default setting is "No".

Name

Enter the name for the configured rules.

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Rule

Enter the rule settings (number pattern, prefix to add ...etc.).

Type

Choose the type of the rule (pattern, block, dial now, prefix & second tone).

Account x  Feature codes

Enable Local Call Features

When enabled, Do Not Disturb, Call Forwarding and other call features can be used via the local feature codes on the phone. Otherwise, the provisioned feature codes from the server will be used. User configured feature codes will be used only if server provisioned feature codes are not provided. Note: If the device is registered with Broadsoft account, it does not matter if local call features are enabled or disabled, once the Broadsoft account is set, special feature to Broadsoft and Feature Key Synchronization is enabled, the call feature will be handled by Broadsoft server, not by the phone.

DND

DND Call Feature On Configures DND feature code to turn on DND.

DND Call Feature Off Configures DND feature code to turn off DND.

Call Forward Always

Configures Call Forward Always feature code to activate unconditional call On
forwarding.

Configures Call Forward Always feature code to deactivate unconditional call Off
forwarding.

Target

The extension the call will be forwarded to.

Call Forward No Answer

Configures Call Forward No Answer feature code to activate no answer call On
forwarding.

Configures Call Forward No Answer feature code to deactivate no answer Off
call forwarding.

Target

The extension the call will be forwarded to.

Call Forward No

Defines the timeout (in seconds) before the call is forwarded on no answer.

Answer Timeout (s)

valid range is 1 to 120.

Accounts  Account Swap

Swap Account Settings

Allows users to swap the two accounts that they have configured. This will Increase the flexibility of account management. Note: Make sure to press "Start" to complete the process.

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Phone Settings Page Definitions

Table 10: Settings Page Definitions
Phone Settings  General settings

Basic Settings

Local RTP Port

This parameter defines the local RTP port used to listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP; channel 1 will use port_value+2 for RTP. Local RTP port ranges from 1024 to 65400 and must be even. Default value is 5004.

Local RTP Port Range

Gives users the ability to define the parameter of the local RTP port used to listen and transmit. This parameter defines the local RTP port from 48 to 10000. This range will be adjusted if local RTP port + local RTP port range is greater than 65486. Default setting is 200.

Use Random Port

When set to "Yes", this parameter will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple phones are behind the same full cone NAT. The default setting is "No"

Keep-alive Interval STUN Server

Note: This parameter must be set to "No" for Direct IP Calling to work. Specifies how often the phone sends a blank UDP packet to the SIP server to keep the "ping hole" on the NAT router to open. The default setting is 20 seconds. The valid range is from 10 to 160. The IP address or Domain name of the STUN server. STUN resolution results are displayed in the STATUS page of the Web GUI. Only non-symmetric NAT routers work with STUN.

Use NAT IP

The NAT IP address used in SIP/SDP messages. This field is blank at the default settings. It should ONLY be used if it is required by your ITSP.

Delay Registration Test Password Strength Enable Outbound Notification Public Mode Enable Public Mode Public Mode Username Prefix

Configures specific time that the account will be registered after booting up. Only Allow password with some constraints to ensure better security. Configures whether to enable outbound notifications such as Action URL.
Configures to turn on/off public mode for hot desking feature. Configures the prefix of the username for public mode login.

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Public Mode Username Suffix

Configures the suffix of the username for public mode login.

Phone Settings  Call Settings

Key Mode

Account Mode In Calling State, each key displays the call status of the corresponding account. Click to switch to the first line under this account or select the account to initiate a new call. Line Mode In Calling State, each key controls a line, and the call line can be switched by pressing the key.

Click-To-Dial Feature

Enables Click-To-Dial feature. If this feature is enabled, user could click the green dial button on left top corner of phone's Web GUI, then choose the account and dial to the target number. The default setting is "Disabled".

Enable Call Waiting

Disables the call waiting feature. The default setting is "Yes".

Hold Call before Completing Transfer

When set to "No", phone will neither hold the current call in transfer window nor hold the call with the transfer target before referring the call in the attended transfer.

Hold Call before Adding Conferee

Configures whether to put current call on hold when adding new member to conference. If set to "Yes", the current call will be put on hold when the host presses conference or add key to invite new member. When the invited member answers the call and agrees to attend conference, the host needs to manually resume the conference with the new member added. If set to "No", the current call will not be put on hold and the invited member will join meeting automatically after answering the call.

Enable DND Feature

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

Mute Key Functions While Idle

If this feature is enabled, MUTE key will take effect in idle state and future incoming call will be answered with mute.

Enable Sending DTMF Allows certain MPKs to send DTMF in-call. This option does not affect Dial

via specific MPKs

DTMF. This option is available for the GRP2604 only.

Preferred Default Account

Select the preferred default account when off-hook/on-hook dialing. When selected account is unavailable, system will fall back to use the first available account instead.

Off-hook Auto Dial

Configures the digits to be dialed via the first account when the phone is off hook.

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Off-hook Auto Dial Delay Off-hook / On-hook Timeout Enable Live Keypad Live Keypad Expire Time Bypass Dial Plan Through Call History and Directories Do Not Escape # as %23 in SIP URI
Return Code When Refusing Incoming Call
Return Code When Enable DND
Allow Incoming Call Before Ringing User-Agent Prefix Ringtone Call Progresses Tones: · System Ring Tone

Defines the timeout (in seconds) for off-hook auto dial. Valid range is 0-30. If set to 0, it will be dialed out immediately; If set to other values, it will be dialed out after the delay. If configured, when the phone is in the off-hook or on-hook dialing state, it will go idle after the timeout (in seconds). Valid range is 10 to 60. If enable phone will automatically dials out and turns on handsfree mode as soon as a dial pad key or softkey is pressed. Set the Live DialPad expire time, interval is between 2s and 15s, default value is 5s.
Enable/Disable the dial plan check while dialing through the call history and any phonebook directories.
Specifies whether to replace # by %23 or not for some special situations. The default setting is "No". When refusing the incoming call. The phone will send the selected type of SIP message of the call. Available options are:
· Busy (486). · Temporarily Unavailable (480). · Not found (404). · Decline (603). Default setting is "Busy 486". When DND is enabled, the phone will send the selected type of SIP message. Available options are: · Busy (486). · Temporarily Unavailable (480). · Not found (404). · Decline (603). Default setting is "Temporarily Unavailable (480)". This allows incoming calls after dialed but before ringing. This can be used under custom user configuration based on need. Configure the prefix in the User-Agent header
Configures ring or tone frequencies based on parameters from local telecom. The default value is North American standard. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds.

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· Dial Tone · Second Dial Tone · Message Waiting · Ring Back Tone · Call-Waiting Tone · Busy Tone · Reorder Tone

Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]; (Frequencies are in Hz and cadence on and off are in 10ms)
ON is the period of ringing ("On time" in `ms') while OFF is the period of silence. To set a continuous ring, OFF should be zero. Otherwise, it will ring ON ms and a pause of OFF ms and then repeat the pattern. Up to three cadences are supported.

Configures the call waiting tone gain to adjust call waiting tone volume (Low, Call Waiting Tone Gain
Medium, or High). The default setting is "Low".

Lock Speaker Volume

Lock volume adjustment when the option is enabled so it cannot be changed from phone LCD. The option can be set to: "No", "Ring", "Talk" or "Both". Default setting is "No".

Multicast Paging

Multicast Paging Function

Enable or disable multicast paging

Allowed In DND Mode

Allow Multicast Paging when DND mode is enabled. Default Setting is "No".

During active call, if incoming multicast page is higher priority (1 being the

Paging Barge

highest) than this value, the call will be held, and multicast page will be played. The default setting is "Disabled".

If enabled, during a multicast page if another multicast is received with higher

Paging Priority Active

priority (1 being the highest) that one will be played instead. The default setting is "Enabled".

Multicast Channel Number

Multicast Channel Number (0-50). 0 for normal RTP packets, 1-50 for Polycom multicast format packets.

Multicast Paging Codec

The codec for sending multicast pages, there are 5 codecs could be used: G.731.1 PCMU, PCMA, G.726-32, G.729A/B, G.722 (wide band). Default setting is "PCMU".

Multicast Sender ID

Outgoing caller ID that displays to your page group recipients (for multicast channel 1 ­ 50).

Defines multicast listening addresses and labels. For example:

Multicast Listening

· "Listening Address" should match the sender's Value such as

"237.11.10.11:6767"

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· "Label" could be the description you want to use. For details, please check the "Multicast Paging User Guide" on our Website.

Network Settings Page Definitions

Table 11: Network Page Definitions
Network Settings  Ethernet Settings

Internet Protocol

Selects "IPv4 Only", "IPv6 Only", "Both, prefer IPv4" or "Both, prefer IPv6". The default setting is "IPv4 only".

IPv4 Address

IPv4 Address

Allows users to configure the appropriate network settings on the phone to obtain IPv4 address. Users could select "DHCP", "Static IP" or "PPPoE". By default, it is set to "DHCP".

IPv4 Address

Enter the IP address when static IP is used.

Subnet Mask

Enter the Subnet Mask when static IP is used for IPv4.

Gateway

Enter the Default Gateway when static IP is used for IPv4.

DNS Server 1

Enter the DNS Server 1 when static IP is used for IPv4.

DNS Server 2

Enter the DNS Server 2 when static IP is used for IPv4.

Preferred DNS Server Enters the Preferred DNS Server for Ipv4.

IPv6 Address

IPv6 Address Type

Allows users to configure the appropriate network settings on the phone to obtain IPv6 address. Users could select "Auto-configured" or "Statically configured" for the IPv6 address type.

Static IPv6 Address

Enter the static IPv6 address when Full Static is used in "Statically configured" IPv6 address type.

IPv6 Prefix Length

Enter the IPv6 prefix length when Full Static is used in "Statically configured" IPv6 address type.

DNS Server 1

Enter the DNS Server 1 for IPv6.

DNS Server 2

Enter the DNS Server 2 for IPv6.

Preferred DNS server Enter the Preferred DNS Server for IPv6.

802.1X

802.1X mode

Allows the user to enable/disable 802.1X mode on the phone. The default value is disabled. To enable 802.1X mode, this field should be set to EAP-

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MD5, users may also choose EAP-TLS, or EAP-PEAPv0/MSCHAPv2.

802.1X Identity

Enter the Identity information for the 802.1x mode. Note: Letters, digits and special characters including @ and ­ are accepted.

MD5 Password

Enter the MD5 Password for the 802.1X mode. Note: Letters, digits and special characters including @ and ­ are accepted.

802.1X CA Certificate

Uploads / deletes the 802.1X CA certificate to the phone; or delete existed 802.1X CA certificate from the phone.

802.1X Client

Uploads / deletes 802.1X Client certificate to the phone; or delete existed

Certificate

802.1X Client certificate from the phone.

Network Settings  Wi-Fi Settings (GRP2602W Only)

Enables / Disables the Wi-Fi on the phone. Three options are available:

Wi-Fi Function

· Enable: Enables Wi-Fi to connect to Wi-Fi network. · Disable: Disables Wi-Fi. User has ability to enable Wi-Fi from LCD
Menu.

· Disable & Hide Menu from LCD: Disables Wi-Fi and hides "Wi-Fi Settings" menu from phone LCD.

Wi-Fi Band

Set the type of Wi-Fi Band whether its 2G or 5G or 5G&2G.

Country Code

Configures Wi-Fi country code.

This parameter sets the ESSID for the Wireless network. Press "Scan" to scan for the available wireless network. Click on "Connect" and enter the authentication credentials of the Wi-Fi network to connect to. Users can connect to hidden networks by pressing on "Add Network" and configure:

ESSID

· ESSID: Configure the hidden ESSID name. · Security Mode: Defines the security mode used for the wireless network

when the SSID is hidden. Default is "None".

· Password: Determines the password for the selected Wi-Fi network.

· Advanced: Configures IPv4 and IPv6 modes.

Network Settings  OpenVPN® Settings

OpenVPN® Enable

Enables/Disables OpenVPN® feature. Default is "No".

OpenVPN® Server Address

Specify the IP address or FQDN for the OpenVPN® Server.

OpenVPN® Port

Specify the listening port of the OpenVPN® server. The valid range is 1 ­ 65535. The default value is "1194".

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Specify the Transport Type of OpenVPN® whether UDP or TCP. OpenVPN® Transport
The default value is "UDP".

OpenVPN® CA

Click on "Upload" to upload the Certification Authority of OpenVPN®. For a new upload, users could click on "Delete" to erase the last certificate, and then upload a new one.

OpenVPN® Certificate

Click on "Upload" to upload OpenVPN® certificate. For a new upload, users could click on "Delete" to erase the last certificate, and then upload a new one.

OpenVPN® Client Key

Click on "Upload" to upload OpenVPN® Key. For a new upload, users could click on "Delete" to erase the last certificate, and then upload a new one.

Specifies the Cipher method used by the OpenVPN® server. The available

options are:

OpenVPN® Cipher Method

· Blowfish · AES-128 · AES-256

· Triple-DES The default setting is "Blowfish".

Configures the optional username for authentication if the OpenVPN server OpenVPN® Username
supports it.

Configures the optional password for authentication if the OpenVPN server OpenVPN® Password
supports it.

Additional options to be appended to the OpenVPN® config file, separated

Additional Options

by semicolons. For example, comp-lzo no;auth SHA256 Note: Please use this option with caution. Make sure that the options are

recognizable by OpenVPN® and do not unnecessarily override the other

configurations above.

Network Settings  Advanced Settings

Advanced Network Settings

Enable LLDP

Controls the LLDP (Link Layer Discovery Protocol) service. The default setting is "Enabled".

LLDP TX Interval

Defines LLDP TX Interval (in seconds). Valid range is 1 to 3600. The default setting is "60".

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Enable CDP

Enables/Disables CDP "Cisco Discovery Protocol". The default setting is "Enabled".

Layer 3 QoS for SIP

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

Layer 3 QoS for RTP

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

Enable DHCP VLAN Enables auto configure for VLAN settings through DHCP. Disabled by default.

Enable Manual VLAN Configuration

Enables/disables manual VLAN configuration. When this option is set to Disabled, the phone will bypass VLAN configuration and only use the DHCP VLAN to configure VLAN tag and priority. Default is "Enabled".

Layer 2 QoS 802.1Q/VLAN Tag

Assigns the VLAN Tag of the Layer 2 QoS packets. The valid range is 0 ­ 4094.The default value is 0.

Layer 2 QoS 802.1p Priority Value
Maximum Transmission Unit (MTU)

Assigns the priority value of the Layer2 QoS packets. The valid range is 0 ­ 7. The default value is 0.
Defines the MTU in bytes. The valid range is 576 ­ 1500. The default value is 1500 bytes.

PC Port Mode

PC Port Mode

Configure the PC port mode. When set to "Mirrored", the traffic in the LAN port will go through PC port as well and packets can be captured by connecting a PC to the PC port. The default setting is "Enabled".

PC Port VLAN Tag

Assigns the VLAN Tag of the PC port. The valid range is 0 ­ 4094. The default value is 0.

Assigns the priority value of the PC port. The valid range is 0 ­ 7. PC Port Priority Value
The default value is 0.

Proxy

HTTP Proxy

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

HTTPS Proxy

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

Remote Control

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Action URI Support

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

Remote Control Pop up Window Support

Indicate whether the phone is enabled to pop up Allow Remote Control window.

Action URI Allowed IP List of allowed IP addresses from which the phone receives the Action URI
List

CSTA Control

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

CTI Settings

Affinity Support

Indicate whether Affinity feature is supported.

Preferred Account

Affinity target SIP account.

Network Settings  SNMP Settings

Enable SNMP

Enable/Disable SNMP service. Default is No.

Version

Choose between (Version 1, Version 2, or Version 3).

Port

Listening Port of SNMP daemon (Default 161).

Community

Name of SNMP community.

Security Level

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth. AuthUser: Users with security level authNoPriv and context name as auth. privUser : Users with security level authPriv and context name as priv.

SNMP Username

Username for SNMPv3.

Authentication Protocol

Select the Authentication Protocol: "None" or "MD5" or "SHA."

Privacy Protocol

Select the Privacy Protocol: "None" or "AES" or "DES".

Authentication Key

Enter the Authentication Key for SNMPv3.

Privacy Key

Enter the Privacy Key for SNMPv3.

SNMP Trap Version Choose the Trap version of the SNMP trap receiver.

SNMP Trap IP

IP address of trap destination.

SNMP Trap port

Port of Trap destination (Default 162)

SNMP Trap Interval

Time interval between traps (Default is 5).

SNMP Trap Community

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

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SNMP Trap Username Username for SNMPv3 Trap.

Trap Security Level

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

Trap Authentication Protocol

Select the Authentication Protocol: "None" or "MD5" or "SHA".

Trap Privacy Protocol Select the Privacy Protocol: "None" or "AES/AES128" or "DES".

Trap Authentication Enter the Trap Authentication Key. Key

Trap Privacy Key

Enter the Trap Privacy Key.

Programmable keys Page Definitions
Table 12: Programmable Keys Page Definitions
Programmable keys  Multi-Purpose Keys (GRP2604 only) Keys Settings

· Speed Dial Select the Account to dial from. And enter the Speed Dial number in the Value field to be dialed or enter the IP address to set the Direct IP call as Speed Dial.

Mode

· Busy Lamp Field (BLF) Select the Account to monitor the BLF status. Enter the extension number in the Value field to be monitored.
· Presence Watcher This option has to be supported by a presence server and it is tied to the "Do Not Disturb" status of the phone's extension.

· Eventlist BLF This option is similar to the BLF option but in this case the PBX collects the information from the phones and sends it out in one single notify message. PBX server has to support this feature.

· Speed Dial via active account

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Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is offhook and account 2 is active, it will call the configured Speed Dial number using account 2

· Dial DTMF Enter a series of DTMF digits in the Value field to be dialed during the call. "Enable MPK Sending DTMF" has to be set to "Yes" first.

· Voice Mail Select Account and enter Voice Mail access number in the Value field.

· Call Return The last answered calls can be dialed out by using Call Return. The Value field should be left blank. Also, this option is not binding to the account and the call will be returned based on the account with the last answered call.

· Transfer Select Account and enter the number in the Value field to be transferred (blind transfer) during the call.

· Call Park Select Account and enter the call park extension in the Value field to park/pick up the call.

· LDAP Search This option is to narrow the LDAP search scope. Enter the LDAP search base in the Description field. It could be the same or different from the Base in LDAP configuration under Advanced Settings. The Base in LDAP configuration will be used if the Description field is left blank. Enter the LDAP Name/Number filter in the Value field. For example: If users set MPK 1 as "LDAP Search" for "Account 1", and set filters: Description -> ou=video,ou=SZ,dc=grandstream,dc=com Value -> sn=Li Since the Base for LDAP server configuration is: "dc=grandstream,dc=com", "ou=video,ou=SZ" is added to narrow the LDAP search scope. "sn=Li" is the example to filter the last name.

· Conference

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Allow user to set their Multi-Purpose Key to "Conference" mode to trigger a conference. By setting the extension number in the value box, the users will be able to activate a 3-way conference by simply press the assigned MPK button. · Call History Select Account and enter account number in the Value field to allow configuration of call log for other extension.
· Monitored Call Park Select account from Account field and enter the call park extension in the Value field to park/pick up the call, and also monitor the parked call via Line Key's light.
· Menu Select this feature in order to display the Menu from the MPK buttons, no field dis required for configuration.
· Information Select this feature in order to display the Information popup to show the firmware version, MAC address, IP address and IP Settings from the MPK buttons, no field dis required for configuration.
· Message Select this feature in order to display the Message menu from the MPK buttons, no field dis required for configuration.
· Forward Set the MPK Button to perform call forwarding to the destination number configured on the "Value Field". During ringing press the button to perform the call forward.
· DND Press the configured key to enabled/Disable DND.
· Redial On this mode, the configured key can be used to redial numbers.
· Presence Eventlist This option is similar to the Presence Watcher option but in this case the

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PBX collects the information from the phones and sends it out in one single notify message. Note: The PBX server has to support this feature.

· Provision Select this feature in order to make the phone trigger an instant provisioning.

Account

Select the account to be associated with the configured MPK.

Value

Enter the value to be associated with the configured MPK

Label

Enter the name to be associated with the MPK.

Preview

Shows a preview of the configured MPKs label. After saving, you can print the card style in the preview. For more info about how to install the BLF paper label check the Quick Installation guide.

Basic Settings

Transfer Mode via MPK

Perform blind transfer, attended transfer, or a new call with the specific in the Value field when a user presses "Transfer" multiple-purpose key

Enable Transfer via MPK with type BLF, Speed dial, etc, will perform as transfer MPK under active Non-Transfer MPK call Programmable Keys  Idle screen softkey

Custom Idle Screen Softkey Layout

Enables/disables custom softkey layout. Default is disabled.

Custom Softkey

Press on Add Custom Softkey radio button to add/configure up to 3 custom softkeys. Supported key modes are speed dial , speed dial via active account

and voicemail.

Custom Softkey

The softkeys listed under "Enabled" tab will be displayed on the phone's idle

Layout

screen. Select the softkey from "Available" list to enable it.

Programmable Keys  Call screen softkeys

Custom Call Screen Enables/disables custom softkey layout. Default is disabled.
Softkey Layout

Enforce Softkey Layout Position

Whether to enforce the custom softkey layout position. When set to 'YES', GUI will still preserve the space if the configured softkey is unable to show. Default is disabled.

Custom Softkey

Press on Add Custom Softkey radio button to add/configure up to 3 custom softkeys. Supported key modes are speed dial , speed dial via active account

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Custom Softkey Layout

and voicemail. · Dialing State
Custom softkey layout when device is under DIALING state. Available softkeys: EndCall, Backspace, Contacts, Call History.
· Ringing State Custom softkey layout when device is under RINGING state. Available softkeys: Answer, Reject, Forward, Ring silence.
· Calling State Custom softkey layout when device is under CALLING state. Available softkeys: End Call, Conference.
· Call Connected State Custom softkey layout when device is under CALL CONNECTED state. Available softkeys: End Call, Conference , New Call, Swap , Transfer , call park, Call record, Noise shield, BS Call Center.
· On Hold State Custom softkey layout when device is under ON HOLD state. Available softkeys: End Call , Resume , New Call, Conference , Swap , Transfer, BS call center.
· Call Failed State Custom softkey layout when device is under CALL FAILED state. Available softkeys: EndCalls, Redial.
· Transfer State Custom softkey layout when device is under TRANSFER state. Available softkeys: Cancel, Backspace, Transfer , Contacts, Call History.
· Conference State Custom softkey layout when device is under CONFERENCE state. Available softkeys: Cancel, Dial , Backspace, Contacts , Call History.
· Conference Connected State Custom softkey layout when device is under CONFERENCE CONNECTED state. Available softkeys: EndCall, Conference Info , Hold , Add , Noise Shield.

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· Onhook Dialing State Custom softkey layout when device is under ONHOOK DIALING state Available softkeys: End Call , Back Space , Dial , Contacts , Call History. The softkeys listed under "Enabled" tab will be displayed on the phone's idle screen. Select the softkey from "Available" list to enable it.

System Settings Page Definitions

Table 13: System Settings Page Definitions
System Settings  Time and Language

Date and Time

NTP Server

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

Secondary NTP Server

Defines the URL or IP address of the NTP server. The phone may obtain the date and time from the server. Allow user to configure 2 NTP server domain names. GRP will loop through all the IP addresses resolved from them.

NTP Update Interval

Time interval for updating time from the NTP server. Valid time value is in between 5 to 1440 minutes. The default setting is "1440" minutes.

Allow DHCP Option 42 Override NTP Server

Defines whether DHCP Option 42 should override NTP server or not. When enabled, DHCP Option 42 will override the NTP server if it is set up on the LAN. The default setting is "No".

Time Zone

Configures the date/time used on the phone according to the specified time zone. The default setting is "Auto".

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. The default setting is enabled.

Self-Defined Time Zone

This parameter allows the users to define their own time zone, when "Time Zone" parameter is set to "Self-Defined Time Zone". The syntax is: std offset dst [offset], start [/time], end [/time] Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0 MTZ+6MDT+5 This indicates a time zone with 6 hours offset with 1 hour ahead (when daylight saving) which is U.S central time. If it is positive (+) if the local time zone is

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west of the Prime Meridian (A.K.A: International or Greenwich Meridian) and negative (-) if it is east.

M4.1.0,M11.1.0 The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec) The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd Tuesday...) The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues, ... ,Sat) Therefore, this example is the DST which starts from the First Sunday of April to the 1st Sunday of November.

Configures the date display format on the LCD. The following formats are

supported.

· yyyy-mm-dd: 2019-03-02

· mm-dd-yyyy: 03-02-2019 Date Display Format
· dd-mm-yyyy: 02-03-2019

· dddd, MMMM dd: Saturday, March 02

· MMMM dd, dddd: March 02, Saturday The default setting is yyyy-mm-dd.

Configures the time display in 12-hour or 24-hour format on the LCD. The Time Display Format
default setting is in 12-hour format.

Language

Display Language

Selects display language on the phone.

System Settings  Security Settings

SSH Access

Enable SSH

Disables SSH access. The default setting is "Yes"

SSH Public Key

Enable the device to use public key authentication as an alternative option to password authentication.

Keypad Mode

Configuration via Keypad Menu

Configures access control for keypad Menu settings. · Unrestricted: all options on LCD menu can be accessed; · Basic settings only: only options for basic setting can be displayed on LCD menu; · Constraint Mode: accessing options other than basic settings will require permission; · Locked Mode: MENU is disabled

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Web Access

HTTP Web Port

Configures the HTTP port under the HTTP web access mode. The valid range is 80 ­ 65535. The default value is "80".

HTTPS Web Port

Configures the HTTPS port under the HTTPS web access mode. The valid range is 443 ­ 65535. The default setting is "443".

Web Access Mode

Sets the protocol for web interface. · HTTPS · HTTP · Disabled · Both HTTP and HTTPS
The default setting is "HTTP".

Web Access Control Web access control by using Whitelist or Blacklist on incoming IP addresses

Web Access Control Only allow the IP address list as a whitelist or restrict the IP address list as a

List

blacklist to access the Web.

Web Session Timeout

Configures timer to logout web session during idle. The valid range is 2-60 min. The default value is 10 min

Enable User Web Access

Administrator can disable or enable user web access. The default setting is "Enabled".

Validate Server Certificates

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

Web/Restrict mode Lockout Duration

Specifies the time in minutes that the web or LCD login interface will be locked out to user after five login failures. This lockout time is used for web login, and LCD restrict mode admin login. Range is 0-60 minutes. The default setting is "5".

Web/Restrict Mode lockout Attempt Limit

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

User Info Management

User Password

New Password

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

Confirm Password Enter the new User password again to confirm.

Admin Password

Current Password

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

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New Password Confirm Password Client certificate SIP TLS Certificate

Set new password for web GUI access as Admin. This field is case sensitive. Enter the new Admin password again to confirm.
SSL Certificate used for SIP Transport in TLS/TCP.

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

SIP TLS Private Key Password

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

Custom Certificate

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

Trusted CA Certificate

Trusted CA Certificates (1 ­ 6)

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

Load CA Certificates

Phone will verify the server certificate based on the built-in, custom or both trusted certificates list. The default setting is "Default Certificates".

Keypad Lock

Enable Keypad

If set to "Yes", the keypad can be locked by pressing and holding the STAR *

Locking

key for about 4 seconds. And will also allow automatic locking.

If set to "Functional Keys", only "Functional Keys" will be locked but you are

Keypad Lock Type still allowed to make emergency calls. If set to "All Keys", all keys will be locked,

and no emergency calls can be made.

Password to Lock/Unlock

Configures the password to lock/unlock the keypad.

Keypad Lock Timer

Configures the timeout (in seconds) of idle screen for locking keypad. Valid range is 0 to 3600.

Emergency

Defines emergency call numbers. If multiple emergency call numbers are entered, they should be separated by ','.

System Settings  Preferences

Display Control

New Message LED Indicator

Configure the LED indicator mode when there is new voicemail or text message on the phone. If set to "Off", LED indicator will not light up.

Audio Control

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Headset
Headset Key Mode
Headset Type Always Ring Speaker Headset TX Gain (dB) Headset RX Gain (dB)

When headset is connected to the phone, users could use the HEADSET button in "Default Mode" or "Toggle Headset/Speaker".
· Default Mode:  When the phone is in idle, press HEADSET button to off hook the phone and make calls by using headset. Headset icon will display on the screen in dialing/talking status.
 When there is an incoming call, press HEADSET button to pick up the call using headset.
 When there is an active call using headset, press HEADSET button to hang up the call.
 When Speaker/Handset is being used in dialing/talking status, press HEADSET button to switch to headset. Press it again to hang up the call. Or press speaker/Handset to switch back to the previous mode.
· Toggle Headset/Speaker:  When the phone is in idle, press HEADSET button to switch to Headset mode. The headset icon will display on the left side of the screen.  In this mode, if pressing Speaker button or Line key to off hook the phone, headset will be used.
When there is an active call, press HEADSET button to toggle between Headset and Speaker. Selects whether the connected headset is normal RJ11 headset, Plantronics EHS, Jabra EHS, Sennheiser EHS headset. Configures to enable or disable the speaker to ring when headset is used on "Toggle Headset/Speaker" mode. If set to "Yes", when the phone is in Headset "Toggle Headset/Speaker" mode, both headset and speaker will ring on incoming call. The default setting is "No".
Configures the transmission gain of the headset. The default value is -6dB.
Configures the receiving gain of the headset. The default value is -6dB.

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Enable Headset Noise Shield

When enabled, the remote party will not hear the environmental noise during a call using the headset. Choose according to the TX loudness of the earphone. When the TX loudness of the headset is loud, please select the "Loud Headset", and when the TX loudness of the headset is soft, please select the "Thin Headset". "Moderate Headset" is selected by default.

Handset

Handset TX Gain (dB)

Configures the transmission gain of the handset.

Enable EDRC Feature

Enable EDRC feature, the remote party will not hear the environmental noise during a call

Upload Audio Parameter Mode

Developer function to upload audio parameters for different audio modes.

Upload Audio Parameter Volume

Developer function to upload audio parameters for each audio volumes

System Settings  TR-069

Enable TR-069

Enables TR-069

ACS URL

URL for TR-069 Auto Configuration Servers (ACS). Default setting is: https://acs.gdms.cloud

TR-069 Username

ACS username for TR-069.

TR-069 Password

ACS password for TR-069.

Periodic Inform Enable

Enables periodic inform. If set to "Yes", device will send inform packets to the ACS. The default setting is "No".

Periodic Inform Interval

Sets up the periodic inform interval to send the inform packets to the ACS. Default is 86400.

Connection Request
The username for the ACS to connect to the phone.
Username

Connection Request The password for the ACS to connect to the phone.
Password

Connection Request The port for the ACS to connect to the phone.
Port

CPE SSL Certificate The Cert File for the phone to connect to the ACS via SSL.

CPE SSL Private Key The Cert Key for the phone to connect to the ACS via SSL.

Start TR-069 at Random Time

When enabled, TR-069 will send out first INFORM message to server on randomized timing between 1 to 3600 seconds after phone boots up.

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Maintenance Page Definitions

Table 14: Maintenance Page Definitions
Maintenance  Upgrade and Provisioning

Firmware

Upgrade via Manually Upload

Upload Firmware File to Update

Upload and start upgrade firmware.

Upgrade via Network

Allows users to choose the firmware upgrade method via TFTP, HTTP or Firmware Upgrade via
HTTPS.

Firmware Server Path Defines the server path for the firmware server.

Firmware Server Username

The username for the firmware server.

Firmware Server Password

The password for the firmware server.

Firmware File Prefix

If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone.

Firmware file Postfix

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

Upgrade Detection

Upgrade

Press to start upgrade process.

Config File

Configure Manually

Download Device Configuration

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

Download Device Configuration (XML)

Click to download phone's configuration file in .xml format. Note: Configuration file does not include passwords or CA/Custom certificate

Download User configuration

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

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Upload Device Configuration

Uploads configuration file to phone.

Export backup Package

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

Restore from Backup package

Click to upload backup package and restore.

Configure via Network

Config Upgrade Via

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

Config Server Path

Defines the server path for provisioning.

Config Server Username

The username for the HTTP/HTTPS server.

Config Server Password

The password for the HTTP/HTTPS server.

Always Authenticate Before Challenge

Only applies to HTTP/HTTPS. If enabled, the phone will send credentials before being challenged by the server.

Config File Prefix

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

Config File Postfix

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

Authenticate Conf File Authenticates configuration file before acceptance.

XML Config File Password

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

Provision

Auto Upgrade

Automatic Upgrade

Enables automatic upgrade and provisioning. The default setting is "No".

Automatic Upgrade Check Interval (m)

Specifies the time period to check for firmware upgrade (in minutes). The default value is 10080.

Hour of the Day(0-23) Defines the hour of the day to check the HTTP/TFTP/FTP server for firmware

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Upgrade or configuration files changes. The default value is 1.

Day of the Week(0-6)

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

Randomized Automatic Upgrade

Randomized Automatic Upgrade within the range of hours of the day or Post pone the upgrade every X minute(s) by random 1 to X minute(s). The default setting is "No"

Specifies how firmware upgrading and provisioning request to be sent:

Firmware Upgrade and Provisioning

Always Check for New Firmware, Check New Firmware only when F/W pre/suffix Changes, Always Skip the Firmware Check.

The default setting is "Always Check for New Firmware".

If set to "Yes", the phone will ask the user to upgrade. If there is no response,

Firmware Upgrade Confirmation

The phone will proceed with the upgrade. If set to "No", the phone will automatically upgrade without user input. Default is Yes.

DHCP Option

Allow DHCP Option 43 and Option 66 Override Server

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

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"

Additional Override DHCP Option

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

Config Provision

Download and Process ALL Available Config Files

By default, device will provision the first available config in the order of cfgMAC, cfgMAC.xml, cfgMODEL.xml, cfg.xml and devMAC.cfg (corresponding to device specific, model specific, and global configs). If set

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User Protection
3CX Auto Provision Advanced Settings

to Yes, device will download and apply (overwrite) all available configs in the same order. When user protection is on, pvalues that user sets will not be changed by provision or provider. Phone will multicast SUBSCRIBE for provision if this feature is enabled.

Validate Hostname in Certificate

To validate the hostname in the SSL certificate

Enable SIP Notify Authentication

Device will challenge NOTIFY with 401 when set to Yes

Factory reset

Press Start to begin Factory Reset of the phone.

Maintenance  System Diagnosis

Syslog

If set to SSL/TLS, the syslog messages will be sent through secured TLS

Syslog Protocol

protocol to syslog server. Default setting is "UDP". Note: The CA certificate is required to connect with the TLS server.

Syslog Server

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

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:

Syslog Level

· 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).

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· Memory exception (ERROR level).

Syslog Keyword Filter

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

Send SIP Log

Configures whether the SIP log will be included in the syslog messages. The default setting is "No". Note: By setting Send SIP Log to Yes, the phone will still send SIP log from

syslog even when Syslog Level set to NONE.

Packet Capture

With RTP Packets

Defines whether the packet capture file contains RTP or not. The default setting is "No".

Ping

Ping

Enter Ping target's IP address or URL and click on start.

Traceroute

Traceroute

Input target's IP address or URL and click on start

Maintenance  Outbound Notification

Action URL

Setup Completed

Configures the Action URL to send when phone finishes setup process.

Registered

Configures the Action URL to send when phone successfully registers a SIP account.

Unregistered

Configures the Action URL to send when phone unregisters a SIP account.

Off-hook

Configures the Action URL to send when phone is in off-hook state.

On-hook

Configures the Action URL to send when phone is in on-hook state.

Incoming Calls

Configures the Action URL to send when phone receives an incoming call.

Outgoing Calls

Configures the Action URL to send when phone places a call.

Missed Call

Configures the Action URL to send when phone has a missed call.

Established Call

Configures the Action URL to send when phone establishes a call.

Terminated Call

Configures the Action URL to send when phone terminates a call.

Enable DND

Configures the Action URL to send when phone enables DND.

Disable DND

Configures the Action URL to send when phone disables DND.

Enable Call Forward Configures the Action URL to send when phone enables Call Forward.

Disable Call Forward Configures the Action URL to send when phone disables call forward.

Blind Transfer

Configures the Action URL to send when phone performs Blind Transfer.

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Attended Transfer Hold Call Unhold Call Destination Destination Name Protocol
Enable SSL Destination Address Port Domain Username Password From To
Extra Attribute Name
Extra Attribute Value
Destination Event Destination
Subject Message Extra Attribute Name

Configures the Action URL to send when phone performs Attended Transfer. Configures the Action URL to send when phone places a call on hold. Configures the Action URL to send when phone resumes the call on hold.
Identify the destination name. It must be unique. Configure the protocol associated with the destination server. Currently XMPP and SMTP are supported. Configure whether to use SSL to encrypt for SMTP protocol. This option is not editable for XMPP. Configure destination server address, e.g., talk.google.com. Configure destination server port, e.g., 5222. Configure the destination server domain for XMPP protocol. This option is not editable for SMTP. Configure the authorization username of the destination server. Configure the authorization user password for the destination server. Configure the sender name for SMTP protocol. This option is not editable for XMPP. Configure the receiver's address. Configure extra attribute's name reserved for protocol specific attributes such as "jid" for XMPP protocol. If "jid" is specified, username and domain will be overridden. Configure extra attribute's value reserved for protocol specific attributes such as "abc@gmail.com" for "jid" of XMPP protocol. If it is specified, username and domain will be overridden.
Configures the event, which will trigger an outbound notification. Configures the name of the destination where the outbound notification will be sent to. Configures the subject of Email notification. This option is only applicable to SMTP protocol and it is not editable for other protocols. Configures the message body or the outbound notification. Configure extra attribute's name reserved for specific attributes for a given notification in the future.

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Extra Attribute Value

Configures extra attribute's value reserved for specific attributes for a given notification in the future.

Maintenance  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.

Interval Report

VQ RTCP-XR Interval Report

When enabled, phone will send an interval quality report to the central report collector periodically throughout a call.

VQ RTCP-XR Interval Report Period

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

Alert Report

Warning Threshold for Moslq

Configure the threshold value of 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.

Critical Threshold for Moslq

Configure the threshold value of 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.

Warning Threshold for Configure the threshold value of one way delay (in milliseconds) that causes

Delay

the phone to send a warning alert quality report to the central report collector.

Critical Threshold for Delay

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

Application Page Definitions

Application  Web Service

Table 15: Application Page Definitions

Use Auto Location Service

To enable or disable auto location services on the phone. (Reboot Required)

Application  Contacts

Contacts

Add Contact

Press Add to create a new contact.

Edit

Edits the contact parameters.

Delete All Contacts

Press to delete all contacts.

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Group Management

Add Group

Specifies Group's name to add new group. More than 30 Groups supported.

Edit Group

Edits selected group.

Delete Group

Delete Selected group

Phonebook Management

Enable Phonebook XML Download

Configures to enable phonebook XML download. Users could select HTTP/HTTPS/TFTP to download the phonebook file. The default setting is "Disabled".

HTTP/HTTPS Username

The username for the HTTP/HTTPS server.

HTTP/HTTPS Password

The password for the HTTP/HTTPS server.

Phonebook XML Server Path

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

Phonebook Download Interval

Configures the phonebook download interval (in minutes). If set to 0, automatic download will be disabled. The default value is 0. Valid range is 5 to 720 minutes.

Remove Manuallyedited Entries on Download

If set to "Yes", when XML phonebook is downloaded, the entries added manually will be automatically removed. The default setting is "Yes".

Import Group Method

· When set to "Replace", existing groups will be completely replaced by imported one.
· When set to "Append", the imported groups will be attended with the current one.
The default setting is "Replace".

Sort Phonebook by

Sort phonebook based on the selection of first name or last name. The default setting is "Last Name".

Download XML Phonebook

Click on "Download" to download the XML phonebook file to local PC

Upload XML Phonebook

Click on "Upload" to upload local XML phonebook file to the phone.

Default Search Mode Application  LDAP

Configure the default phone book search mode.

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LDAP Protocol

Configures the LDAP protocol to LDAP or LDAPS. The default setting is "LDAP". LDAPS is a feature to support LDAP over TLS.

Server Address

Configures the IP address or DNS name of the LDAP server.

Port

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

Configures the LDAP search base.

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

Base DN

Example: dc=grandstream, dc=com

ou=Boston, dc=grandstream, dc=com

Username

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

Password

Configures the bind "Password" for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

Configures the filter used for number lookups.

Examples:

LDAP Number Filter

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

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

Configures the filter used for name lookups.

Examples:

(|(cn=%)(sn=%)) returns all records which has the "cn" or "sn" field starting

LDAP Name Filter

with the entered prefix; (!(sn=%)) returns all the records which do not have the "sn" field starting with

the entered prefix;

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

LDAP Version

Selects the protocol version for the phone to send the bind requests. The default setting is "Version 3".

Specifies the "name" attributes of each record which are returned in the

LDAP search result. This field allows the users to configure multiple space

separated name attributes. LDAP Name Attributes
Example:

gn

cn sn description

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Specifies the "number" attributes of each record which are returned in the LDAP search result.

LDAP Number Attributes

This field allows the users to configure multiple space separated number attributes. Example: telephoneNumber telephoneNumber Mobile

LDAP Display Name

Configures the entry information to be shown on phone's LCD. Up to 3 fields can be displayed. Example:

%cn %sn %telephoneNumber

Max. Hits

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

Search Timeout

Specifies the interval (in seconds) for the server to process the request and client waits for server to return. The default setting is 30 seconds.

Sort Results

Specifies whether the searching result is sorted or not. Default setting is "No".

LDAP Lookup

Configures to enable LDAP number searching when dialing / receiving calls.

Lookup Display Name

Configures the display name when LDAP looks up the name for incoming call or outgoing call. This field must be a subset of the LDAP Name Attributes. Example: gn cn sn description

LDAP Dialing Default Account

Configures the default account used when dialing LDAP contact

Exact Match Search

Search for exact match result. Default setting is "No".

Application  Call History

Delete

Users can select an entry, then click "Delete" to remove it from the list.

Delete All

Click on Delete All to remove all Call History stored in the phone. Note: Users could use the drop-down list to show only selected call history type (All, Answered, Dialed, Missed, and Transferred) and use navigation

keys to browse pages when many entries exist.

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External Service Page Definitions

External Service  GDS

Table 16: External Service Page Definitions

GDS

Connect to a GDS37XX and send OpenDoor request. · Service Type
Select GDS as service type. · Account
The account to be used on the phone to interact with the GDS37XX. · System Identification
A name or a number to identify the GDS37XX. · System Number
The SIP extension or the IP address of the GDS37XX depending on the deployed scenario, Peering or Registration. · Access Password The password set on the GDS37XX to unlock the door. · System Ringtone Select the system ringtone from the dropdown list to be played when there is an incoming call from the configured system number of the GDS37xx. Notes:

- When using Peering scenario, on "System Number" field of the GRP260x specify the IP address of the peered GDS37XX.

- When using Registration scenario and both GRP260x and GDS37XX are registered on the same SIP server, specify the SIP extension of the GDS37XX on "System Number" field on GXP16XX.
The "Access Password" on GRP260x should be matching "Remote PIN to Open the door" on GDS37XX. External Service  Call Center

Call Center Codes

Set the disposition code and the unavailable code for quick selection on the phone side

Wrap-up Countdown

Configure the countdown times when the agent status is Wrap-up and execute the countdown on the LCD. If set to 0, the countdown is disabled

External Service  Broadsoft XSI

Authentication Login

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Server Port XSI Action Path
XSI Authentication Type
BroadWorks User ID SIP Authentication ID SIP Authentication Password
Auto Login
Service Settings Sort Phonebook by BroadSoft Directory Update Interval (m)
Broadsoft Contacts Download Limitation
BroadSoft Contacts Search limitation
Network Directories

Broadsoft XSI server address with protocol. Port of the Broadsoft XSI server. Configure the deployment path for Broadsoft XSI Actions. If it is empty, the path "com.broadsoft.xsi-actions" will be used. Defines the authentication type to use login credentials or SIP credentials. If set to "Login Credentials", please fill in User ID and Password in the following options; If set to "SIP Credentials", please fill in user ID, Authentication ID, and Authentication Password. SIP User ID for Broadsoft XSI server. SIP Username for Broadsoft XSI server.
SIP Password for Broadsoft XSI server.
If set to "Yes", the device will automatically login in the background after booting up, so that the BS Xsi always remains logged in. LCD can get the latest Directories, can directly enter the BS Xsi user service, and trigger the update of data in the background.
Sort phonebook based on the selection of first name or last name. Configures the BroadSoft phonebook download interval (in minutes). If set to 0, automatic download will be disabled. Valid range is 5 to 4320. The maximum contacts that can be downloaded for each BroadSoft XSI server directory. The valid range is from 0 to 2000. If set to 0, the server's default contact limit will be used. If the total contact records returned by the server is larger than this limit then it will not be downloaded, and the device will be limited to remote search. The maximum remote search records that can be downloaded for the BroadSoft XSI server directory. The valid range is from 0 to 2000. If set to 0, there is no limit. If the search result total records exceed this value, it will not be downloaded, and you will need to narrow the search scope.
Enable/Disable Broadsoft Network directories. The directory types are:

Type

· Group Directory Enable/Disable and rename the BroadWorks Xsi Group Directory features on the phone. If keep the Name box blank, the phone will

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Name

use the default name "Group" for it.
· Enterprise Directory Enable/Disable and rename the BroadWorks Xsi Enterprise Directory features on the phone. If keep the Name box blank, the phone will use the default name "Enterprise" for it.
· Group Common Enable/Disable and rename the BroadWorks Xsi Group Common Directory features on the phone. If keep the Name box blank, the phone will use the default name "Group Common" for it.
· Enterprise Common Enable/Disable and rename the BroadWorks Xsi Enterprise Common Directory features on the phone. If keep the Name box blank, the phone will use default name "Enterprise Common" for it.
· Personal Directory Enable/Disable and rename the BroadWorks Xsi Personal Directory features on the phone. If keep the Name box blank, the phone will use the default name "Personal" for it.
· Missed Call Log Enable/Disable and rename the BroadWorks Xsi Missed Call Log features on the phone. If keep the Name box blank, the phone will use the default name "Missed" for it.
· Placed Call Log Enable/Disable and rename the BroadWorks Xsi Placed Call Log features on the phone. If keep the Name box blank, the phone will use the default name "Outgoing" for it.
· Received Call Log Enable/Disable and rename the BroadWorks Xsi Placed Call Log features on the phone. If keep the Name box blank, the phone will use the default name "Incoming" for it.
Defines the directory name.

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NAT SETTINGS
If the devices are kept within a private network behind a firewall, we recommend using STUN Server. The following settings are useful in the STUN Server scenario:
· STUN Server Under SettingsGeneral Settings, enter a STUN Server IP (or FQDN) that you may have, or look up a free public STUN Server on the internet and enter it on this field. If using Public IP, keep this field blank.
· Use Random Ports It is under SettingsGeneral Settings. This setting depends on your network settings. When set to "Yes", it will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple GRPs are behind the same NAT. If using a Public IP address, set this parameter to "No".
· NAT Traversal It is under Accounts XNetwork Settings. Default setting is "No". Enable the device to use NAT traversal when it is behind firewall on a private network. Select Keep-Alive, Auto, STUN (with STUN server path configured too) or other option according to the network setting.

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PACKET CAPTURE
GRP260X is embedded with packet capture function. The related options are under MaintenanceSystem Diagnosis  Packet Capture.
Figure 6: Packet Capture in Idle
User can also define whether RTP packets will be captured or not from With RTP Packets option. When the capture configuration is set, press Start button to start packet capture. The Status will become RUNNING while capturing, as showed in Figure 7: Packet Capture when running. Press Stop button to end capture. Press Download button to download capture file to local PC. The capture file is in .pcap format.

Figure 7: Packet Capture when running
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UPGRADING AND PROVISIONING
Unified Firmware
The GRP2601 / GRP2601P / GRP2602 / GRP2602P / GRP2603 / GRP2603P / GRP2604 / GRP2604P support unified firmware for all GRP260X models.

Unified Firmware
Figure 8: GRP260X Unified Firmware
Firmware Upgrade
The GRP260X series can be upgraded via TFTP / FTP / FTPS / HTTP / HTTPS by configuring the URL/IP Address for the TFTP / HTTP / HTTPS / FTP / FTPS server and selecting a download method. Configure a valid URL for TFTP, FTP/FTPS or HTTP/HTTPS, the server name can be FQDN or IP address.

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Examples of valid URLs: firmware.grandstream.com/BETA fw.mycompany.com
There are two ways to setup a software upgrade server: The LCD Keypad Menu or the Web Configuration Interface. Upgrade via Keypad Menu Follow the steps below to configure the upgrade server path via phone's keypad menu:

1. Press MENU button and navigate using Up/Down arrow to select Settings. 2. In the Settings options, select Advanced Settings and then Upgrade. 3. Enter the firmware server path and select upgrade method. The server path could be in IP address
format or FQDN format. 4. Select Start Provision option and press the "Select" Softkey. 5. A warning window will be prompt for provision confirmation. Press "YES" Softkey to start
upgrading/provisioning immediately.

When upgrading starts, the screen will show upgrading progress. When done you will see the phone restarts again. Please do not interrupt or power cycle the phone when the upgrading process is on.
Upgrade via Web GUI
Open a web browser on PC and enter the IP address of the phone. Then, login with the administrator username and password. Go to MaintenanceUpgrade and Provisioning page, enter the IP address or the FQDN for the upgrade server in "Firmware Server Path" field and choose to upgrade via TFTP or HTTP/HTTPS or FTP/FTPS. Update the change by clicking the "Save and apply" button. Then "Reboot" or power cycle the phone to update the new firmware.
When upgrading starts, the screen will show upgrading progress. When done you will see the phone restart

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again. Please do not interrupt or power cycle the phone when the upgrading process is on.
Firmware upgrading takes around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. We recommend completing firmware upgrades in a controlled LAN environment whenever possible. No Local TFTP/FTP/HTTP Servers For users that would like to use remote upgrading without a local TFTP/FTP/HTTP server, Grandstream offers a NAT-friendly HTTP server. This enables users to download the latest software upgrades for their phone via this server. Please refer to the webpage: http://www.grandstream.com/support/firmware
Alternatively, users can download a free TFTP, FTP or HTTP server and conduct a local firmware upgrade. A free window version TFTP server is available for download from: http://www.solarwinds.com/products/freetools/free_tftp_server.aspx http://tftpd32.jounin.net/.
Instructions for local firmware upgrade via TFTP: 1. Unzip the firmware files and put all of them in the root directory of the TFTP server. 2. Connect the PC running the TFTP server and the phone to the same LAN segment. 3. Launch the TFTP server and go to the File menuConfigureSecurity to change the TFTP server's
default setting from "Receive Only" to "Transmit Only" for the firmware upgrade. 4. Start the TFTP server and configure the TFTP server in the phone's web configuration interface. 5. Configure the Firmware Server Path to the IP address of the PC. 6. Update the changes and reboot the phone.
End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server.

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Phone Provisioning
Configuration File Download
Grandstream SIP Devices can be configured via the Web Interface as well as via a Configuration File (binary or XML) through TFTP, FTP/FTPS or HTTP/HTTPS. The "Config Server Path" is the TFTP, FTP/FTPS or HTTP/HTTPS server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The "Config Server Path" can be the same or different from the "Firmware Server Path".
A configuration parameter is associated with each field in the web configuration page. A parameter consists of a Capital letter P and 2 to 5-digit numeric numbers. i.e., P2 is associated with the "New Password" in the Web GUISystem SettingsSecurity Settings  User Info Management  Admin Password. For a detailed parameter list, please refer to the corresponding configuration template.
When the GRP260x series boots up or reboots, it will issue a request to download an XML file named "cfgxxxxxxxxxxxx.xml", where "xxxxxxxxxxxx" is the MAC address of the phone, i.e., "cfg000b820102ab" and "cfg000b820102ab.xml". If the download of "cfgxxxxxxxxxxxx.xml" file is not successful followed by a configuration file named "cfgxxxxxxxxxxxx", the phone will issue a request to download a specific model configuration file "cfg<model>.xml", where <model> is the phone model, i.e., "cfggrp2601.xml" for the GRP2601, "cfgrp2602" for the GRP2602, "cfgrp2603" for the GRP2603 and "cfgrp2604" for the GRP2604. If this file is not available, the phone will issue a request to download the generic "cfg.xml" file. The configuration file name should be in lower case letters, If not found, the phone will request a file named "dev[MacAddress].cfg" where "MacAddress" is the MAC address of the device, With this provision file, users are able to provision the device with both Pvalues and aliases.

Figure 9: Config File Download
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Note: (Attempt to download Config File again) When doing provision on the phone, if your first config file contains p-values listed below, phone will try to download the potential second cfg.xml file and apply the second file without rebooting. Maximum 3 extra attempts. Those P-values are: *212 -- Config upgrade via *234 -- Config prefix *235 -- Config postfix *237 -- Config upgrade Server *240 ­ Authenticate Config File *1359 ­ XML Config File Password *8463 ­ Validate Server Certificate *8467 ­ Download and process ALL Available Config Files *20713 ­ Always authenticate before challenge *22011 ­ Bypass Proxy For *22030 ­ Enable SSL host verification for provision

Note: (P-values that trigger Auto-Provision)
If the p-values listed below are changed while managing configuration on web UI or LCD, the provision process will be triggered:

* 192 -- Firmware upgrade server

* 232 -- Firmware prefix * 233 -- Firmware postfix

* 6767 -- Firmware Upgrade Via

* 6768 -- Firmware HTTP/HTTPS Username

* 6769 -- Firmware HTTP/HTTPS Password

* 237 -- Config upgrade Server * 212 -- Config upgrade via

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* 234 -- Config prefix
* 235 -- Config postfix
* 1360 -- Config HTTP/HTTPS username
* 1361 -- Config HTTP/HTTPS password.
Note: Certificates and Keys provisioning
Users can configure the phone to get all the needed certificates during boot up. Instead of putting the certificate/key content in text directly from the Web interface or uploading them manually, they can choose to provision them from the configuration file by putting the URL in the Pvalue field of each certificate and/or key. (e.g., http://ProvisionServer_address/SIP-TLS-Certificate.pem) The phone will then process the URL, search for the appropriate certificate/Key file, download it and then apply it into the phone.

Figure 10: Certificates Files Download
For more details on XML provisioning, please refer to: http://www.grandstream.com/sites/default/files/Resources/gs_provisioning_guide.pdf
No Touch Provisioning
After the phone sends, config file request to the Broadsoft provisioning server via HTTP/HTTPS, if the provisioning server responds "401 Unauthorized" asking for authentication, the phone's LCD will prompt a window for user to enter username and password. Once correct username and password are entered, the

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phone will send config file request again with authentication. Then the phone will receive the config file to download and get provisioned automatically.
Besides manually entering the username and password in LCD prompt, users can save the login credentials for provisioning process as well. The username and password configuration are under phone's web UIMaintenanceUpgrade and provisioning page: "HTTP/HTTPS Username" and "HTTP/HTTPS Password". If the saved username and password saved are correct, login window will be skipped. Otherwise, login window will be popped up to prompt users to enter correct username and password again.

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RESTORE FACTORY DEFAULT SETTING
Warning: Restoring the Factory Default Settings will delete all configuration information on the phone. Please backup or print all the settings before you restore to the factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
Restore Factory Settings using LCD menu
Please follow the instructions below to reset the phone: 1. Press MENU button to bring up the keypad configuration menu. 2. Navigate to Settings  Advanced Settings. 3. Select "Factory Reset". 4. A warning window will pop out to make sure a reset is requested and confirmed.
Press the "Yes" Softkey to confirm and the phone will reboot, or "No" Softkey to cancel the Reset.
Restore to Factory Default via Web GUI
1. Login GRP2601/GRP2602/GRP2603/GRP2604 Web GUI. 2. Navigate to Maintenance  Upgrade and provisioning  Advanced Settings  Factory reset. 3. Press on Start Button situated against Factory reset option Click "OK" to confirm and the phone will reboot, or on "Cancel" to cancel the Reset.

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EXPERIENCING GRP260X
Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation, and news on new products.
We encourage you to browse our product related documentation, FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream Certified Partner or Reseller, please contact them directly for immediate support.
Our technical support staff is trained and ready to answer all your questions. Contact a technical support member or submit a trouble ticket online to receive in-depth support.
Thank you again for purchasing Grandstream IP phone, it will be sure to bring convenience and color to both your business and personal life.

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