Grandstream GXP2124 User Manual

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Grandstream Networks, Inc.

GXP2124 SIP Enterprise Phone

GXP2124 User Manual www.grandstream.com Firmware version: 1.0.3.19 http://esupport.grandstream.com

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GXP2124 USER MANUAL

WELCOME ................................................................................................................................... 3

INSTALLATION ............................................................................................................................ 4

QUIPMENT PACKAGING ............................................................................................................... 4

E C

ONNECTING THE GXP2124 ........................................................................................................ 4

AFETY COMPLIANCES ................................................................................................................. 5

ARRANTY .................................................................................................................................. 5

PRODUCT OVERVIEW ................................................................................................................ 6

USING THE GXP2124 SIP ENTERPRISE PHONE ..................................................................... 9

ETTING FAMILIAR WITH THE LCD ................................................................................................ 9

G M

AKING PHONE CALLS ............................................................................................................... 13

NSWERING PHONE CALLS ........................................................................................................ 16

HONE FUNCTIONS DURING A PHONE CALL ................................................................................ 16

ALL FEATURES ......................................................................................................................... 20

USTOMIZED LCD SCREEN & XML............................................................................................. 21

CON F I G U R A T I ON G U I D E ........................................................................................................... 22

ONFIGURATION VIA KEYPAD ..................................................................................................... 22

ONFIGURATION VIA WEB BROWSER ......................................................................................... 26

AVING THE CONFIGURATION CHANGES ..................................................................................... 47

EBOOTING THE PHONE REMOTELY ........................................................................................... 47

SOFTWARE UPGRADE & CUSTOMIZATION .......................................................................... 48

IRMWARE UPGRADE THROUGH TFTP/HTTP ............................................................................. 48

ONFIGURATION FILE DOWNLOAD .............................................................................................. 49

RESTORE FACTORY DEFAULT SETTING .............................................................................. 50

TABLE OF FIGURES

GXP2124 USER MANUAL

FIGURE 1: KEYPAD GUI FLOW ....................................................................................................24

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TABLE OF TABLES

GXP2124 USER MANUAL

Table 1: Equipment Packaging ...................................................................................... 4

Table 2: GXP2124 Connectors ...................................................................................... 4

Table 3: GXP2124 Feature Guide ................................................................................. 6

Table 4: GXP2124 Key Features in a Glance ................................................................ 6

Table 5: GXP2124 Hardware Specifications .................................................................. 7

Table 6: GXP2124 Technical Specifications .................................................................. 7

Table 7: LCD Display Definition ..................................................................................... 9

Table 8: LCD Icons ...................................................................................................... 11

Table 9: GXP2124 Keypad Buttons ............................................................................. 12

Table 10: GXP2124 Call Features ................................................................................. 20

Table 11: Key Pad Configuration Menu ......................................................................... 22

Table 12: Device Configuration - Status ........................................................................ 28

Table 13: Device Configuration – Settings/Basic Settings ............................................. 28

Table 14: Device Configuration – Settings /Advanced Settings ..................................... 34

Table 15: SIP Account Settings ..................................................................................... 40

GUI INTERFACE EXAMPLES

GXP2124 USER MANUAL

http://www.grandstream.com/products/gxp_series/general/documents/gxp2124_gui.zip

1. SCREENSHOT OF CONFIGURATION LOGIN PAGE

SCREENSHOT OF STATUS PAGE

SCREENSHOT OF BASIC SETTING CONFIGURATION PAGE

SCREENSHOT OF ADVANCED USER CONFIGURATION PAGE SCREENSHOT OF SIP ACCOUNT CONFIGURATION PAGE

SCREENSHOT OF CONTACTS PAGE

SCREENSHOT OF SAVED CONFIGURATION CHANGES PAGE

SCREENSHOT OF REBOOT PAGE

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Welcome

GXP2124 is a next generation enterprise grade IP phone that features 240x120 backlit graphical LCD, 4 line keys with up to 4 SIP accounts, 4 dedicated XML programmable context-sensitive soft keys, 24+4 XML programmable speed-dial/BLF extension keys, dual network ports with integrated PoE, 5-way conference, and Electronic Hook Switch (EHS). The GXP2124 delivers superior HD audio quality, rich and leading edge telephony features, personalized information and customizable application service, automated provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with most 3rd party SIP devices and leading IP PBX/Soft Switch/IMS platforms. It is an ideal solution for enterprise users looking for a high quality, feature rich multi-line IP phone with many extension keys.

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 User Manual, could void your manufacturer warranty.

Warning: Please do not use a different power adaptor with the GXP2124 as it may cause damage to the

products and void the manufacturer warranty.

 This document is subject to change without notice. The latest electronic version of this user manual is

available for download from: http://www.grandstream.com/support

 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.

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Installation

EQUIPMENT PACKAGING

Table 1: Equipment Packaging

GXP2124

Main Case Handset Phone Cord Power Adaptor Ethernet Cable Phone Stand Wall Mount Quick Start Guide

Yes Yes Yes Yes Yes Yes Yes Yes

CONNECTING THE GXP2124

Table 2: GXP2124 Connectors

PC LAN Power Jack

10/100Mbps RJ-45 ports for PC (downlink) connection. 10/100Mbps RJ-45 port for LAN (uplink) connection. Supports PoE (802.3af). 5V DC power port; UL Certified.

Headset Jack Handset Jack

RJ9, supporting EHS (Electronic Hook-Switch) with Plantronics headsets. RJ9.

To setup the GXP2124, 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.

4. The LCD will display provisioning or firmware upgrade information. 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.

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SAFETY COMPLIANCES

The GXP2124 complies with FCC/CE and various safety standards. The GXP2124 power adaptor is compliant with the UL standard. Only use the universal power adaptor provided with the GXP2124 package. The manufacturer’s warranty does not cover damages to the phone caused by unsupported power adaptors.

WARRANTY

If you purchased your GXP2124 from a reseller, please contact the company where you purchased your phone for replacement, repair or refund. If you purchased the product directly from Grandstream, contact your Grandstream Sales and Service Representative for a RMA (Return Materials Authorization) number before you return the product. Grandstream reserves the right to remedy warranty policy without prior notification.

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Product Overview

Table 3: GXP2124 Feature Guide

Features GXP2124

LCD Display

Number of Lines

Programmable Speed-dial/BLF Keys

XML Programmable Soft Keys

240x120 pixel backlit graphical LCD, 8 level grayscale

4 lines with 4 SIP accounts

24+4

Table 4: GXP2124 Key Features in a Glance

Features Benefits

Open Standards Compatible

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

Superb Audio Quality

HD wideband audio, superb full-duplex hands-free speakerphone with advanced acoustic echo cancellation and excellent double-talk performance

Network Interfaces

Dual switched auto-sensing 10/100mbps Network ports with integrated PoE

Feature Rich

Traditional voice features including caller ID, call waiting, hold, transfer, forward, block, auto answer, off-hook dial

Advanced Features

Multi-line support with dual-color LED, multi-party conferencing, line extension interface, large backlit graphic LCD, 4 navigation keys, dedicated buttons for send/redial, speakerphone, headset, transfer, conference (for up to 5 parties), mute, message, phonebook and volume, large phonebook (up to 2000 contacts) and call history (up to 500 records)

Advanced Functionality

Customized downloadable ring-tones, multi-language support, LCD content customization via XML, XML application, built-in personalized application service (e.g., local weather, stock, currency and etc.), adjustable positioning angles, wall mountable, Automatic provisioning using TR-069 or encrypted XML configuration file, TLS/SRTP/HTTPS for advanced security protection, 802.1x for media access control

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Table 5: GXP2124 Hardware Specifications

GXP2124

LAN Interface (Ethernet ports)

Graphic LCD Display

Expansion Module Support

Headset Jack

Call Appearance LED

Power over Ethernet

Universal Switching

Power Adaptor

Dimension

Weight

Temperature

Two (2) 10/100 Mbps Full/Half Duplex Ethernet Switch with LAN and PC port with auto detection

240x120 pixel No

RJ9, supporting EHS (Electronic Hook-Switch) with Plantronics headsets

Dual color (green/red)

IEEE 802.3af standard

Input: 100-240VAC 50-60 Hz Output: +5VDC, 800mA, UL certified

222mm(W) x 210mm(L) x 93mm(H)

Unit weight: 0.98KG; Package weight: 1.63KG 32 –104° F/ 0 – 40°C

Humidity

Compliance

10% – 90% (non-condensing) FCC Part 15 (CFR 47) Class B; EN55022 Class B, EN55024, EN61000-

3-2, EN61000-3-3, EN 60950-1; AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS; UL 60950 (power adapter)

Table 6: GXP2124 Technical Specifications

GXP2124

Protocols/Standards SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP,

ICMP, DNS (A record, SRV and NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, SIMPLE, 802.1x, LLDP, LDAP, TR-069, TLS, SRTP, IPV6

Network Interfaces

Graphic Display

Dual switched 10/100Mbps ports with integrated PoE Backlit 240x120 graphic LCD display with up to 8 level grayscale

Feature Keys 4 line keys with up to 4 SIP accounts, 4 XML programmable context

sensitive soft keys, 24+4 speed-dial/BLF extension keys with dual-color LED, 5 navigation/menu Keys, 9 dedicated function keys for: MUTE, PHONEBOOK, MESSAGE (with LED Indicator), HEADSET, TRANSFER, CONFERENCE, SEND/REDIAL, SPEAKERPHONE, VOLUME

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Voice Codec Support for G.723.1, G.729A/B, G.711u/a-law, G.726, G.722 (wideband),

and iLBC, in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)

Telephony Features Hold, transfer, forward, 5-way conference, busy-lamp-field (BLF), call park,

pickup, shared-call-appearance (SCA)/bridged-line-appearance (BLA), downloadable phone book (XML, LDAP, up to 2,000 items), call waiting, call log (up to 500 records), XML customization of screen, 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

Headset Jack

Base Stand

Wall Mountable

QoS

Yes, both on handset and speakerphone RJ9, supporting EHS (Electronic Hook-Switch ) with Plantronics headsets Yes, allow 2-angel positions Yes Layer 2 (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

Security User and administrator level passwords, MD5 and MD5-sess based

authentication, AES based secure configuration file, SRTP, TLS, 802.1x media access control

Multi-language English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,

Chinese, Korean, Japanese

Upgrade/Provisioning Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-

069 or encrypted XML configuration file

Power and Green Energy Efficiency

Universal power adapter included: Input: 100-240VAC 50-60Hz; Output: +5VDC, 800mA; Integrated Power-over-Ethernet (802.3af)

Max power consumption 3W (power adapter) or 3.5W (PoE)

Physical

Dimension: 222mm (W) x 210mm (H) x 93mm (D) Unit weight: 0.98KG Package weight: 1.63KG

Temperature and Humidity 32–104oF / 0–40oC, 10 - 90% (non-condensing)

Package Content

GXP2124 phone, handset with cord, base stand, universal power supply, network cable, Quick Start Guide

Compliance

FCC Part 15 (CFR 47) Class B; EN55022 Class B, EN55024, EN61000-32, EN61000-3-3, EN 60950-1; AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS; UL 60950 (power adapter)

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Using the GXP2124 SIP Enterprise Phone

ETTING FAMILIAR WITH THE LCD

GXP2124 has a dynamic and customizable screen. The screen displays differently depending on whether the phone is idle or in use (active screen).

Table 7: LCD Display Definition

Display Definition

DATE AND TIME

LOGO

NETWORK STATUS

LINE STATUS INDICATOR

SOFTKEYS - Idle Screen

Displays the current date and time. Can be synchronized with Internet time servers.

Displays company logo name. This logo name can be customized via xml screen customization.

Displays the status of the phone and network. It will indicate whether the network is down or running (IP address).

Displays the name of the account that is in use. Select another account by pressing the LINE key on the left side.

 SwitchSCR

Press this button to toggle between different idle screens: default idle screen, weather information, stock information and currency information.

 ForwardAll

Unconditionally forward the phone line (account 1) to another phone.

 MissedCalls

This option shows up unanswered calls to this phone.

 Redial

Redial the last dialed number in idle screen when there is existed call log.

 EndCall

Hang up phone when dialing out or talking.

 RefreshStock

Refresh the stock information when stock is enabled.

 ReverseCur

Display the currency information in reverse order.

 RefreshCur

Refresh the currency information when currency is enabled.

Note: If XML application is used, the softkey for XML application will show up in idle screen as configured.

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SOFTKEYS - Call Screen

The softkeys are context sensitive and will change depending on the status of the phone.

 Redial

Redial the last dialed number after offhook when there is existed call log.

 Dial

Dial the call out after offhook and entering the number.

 Hold

Put the current active call on hold.

 AnswerCall

Answer the incoming call when the phone is ringing.

 RejectCall

Reject the incoming call when the phone is ringing.

 EndCall

End the active call.

 Transfer

Transfer softkey will show up after pressing TRAN button and entering transfer target number. Press Transfer softkey to do blind transfer.



 Split

In auto-attended transfer mode, after establishing the second call, press Split to quit transfer and go back to normal talking status.



 ConfCall

Conference the active calls.

 ReConf

Re-establish the conference among the calls on hold.

Call Parking:

Please refer to GXE5024/5028 Online User Manual for more

SPECIAL SOFTKEYS (Only When Integrated with GXE5024/5028)

information.

 CallPark

When a GXP2124 dials out, the Call Park softkey will display on screen. To park the call, press the “Call Park” button.

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 PickUp

When another GXP2124 goes off-hook, the Call Pickup softkey will display on screen. To pick up the parked call, press the “Call Pickup” button.

Call Queue:

Please refer to GXE5024/5028 Online User Manual for more information.

 SignIn

Press this button to sign in to the call queue. Agent will be prompted in the LCD display to select the call queue to join. Press “menu” button on keypad to select “ok”. Once the agent completely signs in, the agent will be brought back to the main screen.

 SignOut

Press this button to sign out of the call queue. Press “menu” button on keypad to select “ok”. This will be displayed once the agent is signed in to the call queue.

STATUS ICON

Table 8: LCD Icons

Icon Definition

DND (idle): ON when “Do Not Disturb” is activated in idle screen. Forward All: All incoming calls will be forwarded to the configured number. Forward on Busy: Calls will be forwarded when phone is busy. Forward on No Answer: Calls will be forwarded if the phone does not answer.

Forward All and No Answer: Calls will be forwarded if Forward All and Forward on No Answer are enabled.

Keypad Locked: ON when the keypad is locked.

Enter Keypad Unlock Password Voice Mail: ON when there are new voice messages.

Shows the status of the phone, using icons as shown in the next table.

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Instant Message: ON when there are new instant messages.

Voice Mail and Instant Message: ON when there are both new voice messages and new instant messages.

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Network Status: Network is down. Missed Call Icon: Indicates missed call(s).

Save Call Record: Indicates phone system writing the call records into the flash. It might

take 10 to 20 seconds to finish the process. The saving interval can be configured under GXP2124 web GUI->Basic settings: Call History Flash Writing.

Waiting For Response: Please wait for the phone system to response before the keypad entry.

Handset Mode

Speaker Mode

Headset Mode Calling out: Phone is calling out.

Calling in: Phone is ringing with incoming call. Incoming Call: The current call is an incoming call. Outgoing Call: The current call is an outgoing call.

Table 9: GXP2124 Keypad Buttons

Keypad Button Definition

LINE KEYS

Call Failed: Fail to establish call.

SRTP: SRTP is used during the call.

MUTE

Call On Hold

Call Active

Conference Call Core dump: Core dump available under web GUI->Status page.

Line keys with LED, can be configured to 4 different SIP profiles. SEND/REDIAL; Enable/Disable handset mode. Transfer an ACTIVE call to another number.

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0 – 9, *, #

Press CONF button to connect Calling/Called party into conference.

Mute an active call.

Press HEADSET key to answer/hang up phone calls while using headset. It also allows user to toggle between headset and speaker.

Enter to retrieve voice mails. Brings phonebook on screen. Enable/Disable hands-free speaker. Volume up or down. Menu/OK key in the center of the navigation keys. Press to enter menu

when phone is in idle. Use it as OK key in keypad configuration. Navigation keys “Up” “Down” “Left” and “Right”.

 Press to navigate in menu options.  During the call, press “Up” “Down” to adjust volume.

Standard phone keypad; press # key to send call; press * key to for IVR functions.

Multi Purpose Keys

24 MPKs on the right side of the phone and 4 MPKs on the right side of the LCD (for BLF, Speed dial and etc).

MAKING PHONE CALLS

Handset, Speakerphone and Headset Mode

The GXP2124 allows you to make phone calls via handset, speakerphone, or headset mode. During the active calls the user can switch between the handset and the speaker by pressing the speaker key; or switch between the handset and headset by pressing the headset key. For headsets to operate, the user must plug the headset to an RJ9 port on the phone, which allows the user to pick-up, speak, or hang-up calls.

Multiple SIP Accounts and Lines

GXP2124 can support up to 4 independent SIP accounts. Each account is capable of independent SIP server, user and NAT settings. Each of the line buttons is “virtually” mapped to an individual SIP account. The name of each account is conveniently printed next to its corresponding button. In off-hook state, select an idle line and the name of the account (as configured in the web interface) is displayed on the LCD and a dial tone is heard.

For example: Configure ACCOUNT 1 and ACCOUNT 2 with Account Name as “VoIP 1”, “VoIP 2”, respectively and ensure that they are active and registered. When LINE1 is pressed, you will hear a dial tone and see “VoIP 1” on the LCD display; when LINE2 is pressed, you will hear a dial tone and see “VoIP 2” on the LCD display.

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To make a call, select the line you wish to use. The corresponding LINE LED will light up in green. User can switch lines before dialing any number by pressing the same LINE button one or more times. If you continue to press a LINE button, the selected account will circulate among the registered accounts.

For example: when LINE1 is pressed, the LCD displays “VoIP 1”; If LINE1 is pressed twice, the LCD displays “VoIP 2” and the subsequent call will be made through SIP account 2.

Incoming calls to a specific account will attempt to use its corresponding LINE if it is not in use. When the “virtually” mapped line is in use, the GXP2124 will flash the next available LINE in red.

Completing Calls

There are six ways to complete a call:

1. D

IAL: To make a phone call.

 When the phone is onhook, enter the number and press SEND key to use account 1 to dial

out. OR,

 Take Handset/SPEAKER/Headset off-hook

or press an available LINE key (activates speakerphone).

 The line will have a dial tone and the primary line (LINE1) LED is red.

If you wish, select another LINE key (alternative SIP account).

 Enter the phone number.  Press the SEND key

or press the “DIAL” softkey.

EDIAL: To redial the last dialed phone number.

2. R When redialing the phone will use the same SIP account as was used for the last call. Thus, when the third SIP account was made for the last call/call attempt, the phone will use the third account to redial.

 When the phone is onhook, press SEND key directly to redial. OR,  Take Handset/SPEAKER/Headset off-hook or

press an available LINE key (activates speakerphone), the corresponding LED will be red.

 Press the SEND button

or press the REDIAL softkey.

SING CALL HISTORY: To call a phone number in Call History.

3. U When using the call history, the phone will use the same SIP account as was used for the last call/call attempt. Thus, when returning a call made to the third SIP account, the phone will use the third SIP account return the call.

 Press the MENU button to bring up the Main Menu.  Select Call History and then “Answered Calls” “Dialed Calls”, “Missed Calls” “Transferred

Calls” or “Forwarded Calls” depending on what your needs.

 Select phone number using the arrow keys.  Press DIAL softkey to dial.

SING THE PHONEBOOK: To call a phone number from Phonebook

4. U Each entry in the phonebook can be attached to an individual SIP account. The phone will use that SIP account to make the phone call.

 Press PHONEBOOK key to bring up the phonebook.  Select phonebook entry using the arrow keys.  Press DIAL softkey to dial.

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AGING/INTERCOM:

5. P The paging/intercom function can only be used if the SERVER/PBX supports this feature and both the phones and PBX are correctly configured.

 Take the Handset/SPEAKER/Headset off-hook, or press LINE key.  Press MENU/OK key to toggle from "Dialing" to "Paging".  Dial the phone number you want to Page/Intercom  Press SEND key.

IA CALL RETURN: On the GXP2124, the Multiple Purpose Key (programmable hard key) has to be

6. V configured as Call Return under Web GUI->Basic Settings configuration. No user name and user ID has to be set on the Multiple Purpose Key for Call Return. After pressing the Call Return key, the last answered number will be dialed out.

 Take handset off hook.  Press the configured Call Return key.

NOTE:

 Dial-tone and dialed number display occurs after the handset is off-hook and the line key is selected.

The phone waits 4 seconds (by default; No key Entry Timeout) before sending and initiating the call. Press the “SEND” or “#” button to override the 4 second delay.

Speed Dial

The Multi Purpose Key buttons, located on the right-hand-side of the phone, can be configured for speed dial. Press the speed dial button to automatically call the assigned extension.

If “Speed Dial Via Active Account” is selected instead of “Speed Dial” on that Multi Purpose Key, it will act just like speed dial but based on the current active account. For instance, if the phone is offhook and account 4 is active, by pressing this key it will call the configured speed dial number using account 4.

Making Calls using IP Addresses

Direct IP Call allows two phones to talk to each other in an ad-hoc fashion without a SIP proxy. VoIP c alls can be made between two phones if:

 Both phones have public IP addresses, or  Both phones are on a same LAN/VPN using private or public IP addresses, or  Both phones can be connected through a router using public or private IP addresses (with necessary

port forwarding or DMZ).

To make a direct IP call, please follow these steps:

1. Press MENU button to bring up MAIN MENU.

2. Select “Direct IP Call” using the arrow-keys.

3. Press OK to select.

4. Input the 12-digit target IP address, for IPv4. (Please see example below)

5. Press OK softkey to initiate call.

For example: If the target IP address is 192.168.1.60 and the port is 5062 (e.g. 192.168.1.60:5062), input the following: 192*168*1*60#5062 - The “ * ” key represent the dot“.” ; The “#” key represent colon “:”. Press OK to dial out.

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Quick IP Call Mode

The GXP2124 also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only the last few digits (last octet) of the target phone’s IP-number. This is possible only if both phones are in under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP server. Controlled static IP usage is recommended.

Setting up the phone to make Quick IP calls

To enable Quick IP calls, the phone has to be setup first. This is done through the web-setup function. In the web GUI->Advanced Settings page, set the "Use Quick IP-call mode to YES. When #xxx is dialed, where x is 0-9 and xxx <=255, a direct IP call to aaa.bbb.ccc.XXX is completed. “aaa.bbb.ccc” is from the local IP address regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required (but OK).

For example:

192.168.0.2 calling 192.168.0.3 -- offhook the phone, dial #3 followed by DIAL softkey or SEND

192.168.0.2 calling 192.168.0.23 -- offhook the phone, dial #23 followed by DIAL softkey or SEND

192.168.0.2 calling 192.168.0.123 -- offhook the phone, dial #123 followed by DIAL softkey or SEND

192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3

NOTE:

 If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct IP-IP

call will also use STUN. Configure the “Use Random Port” to “NO” when completing Direct IP calls.

ANSWERING PHONE CALLS

Receiving Calls

1. Incoming single call: Phone rings with selected ring-tone. The corresponding account LINE flashes red. Answer call by taking Handset/SPEAKER/Headset off hook or pressing SPEAKER or by pressing the corresponding account LINE button.

2. Incoming multiple calls: When another call comes in while having an active call, the phone will produce a Call Waiting tone (stutter tone). Next available lines will flash red. Answer the incoming call by pressing its corresponding LINE button. The current active call will be put on hold.

3. Paging/Intercom Enabled: Phone beeps once and automatically establishes the call via SPEAKER. (PBX or Server must also supports this feature)

Do Not Disturb

Enable DND feature in MENU->Preference, the corresponding icon will be on the right hand side of the screen. When DND is enabled, the phone will not ring and send caller directly to voicemail.

HONE FUNCTIONS DURING A PHONE CALL

Call Waiting/ Call Hold

1. Hold: Place a call on hold by pressing the “HOLD” softkey during the call.

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2. Resume: Resume call by pressing the corresponding blinking LINE.

3. Multiple Calls

: Automatically place ACTIVE call on “HOLD” by selecting another available LINE to

place or receive another call. Call Waiting tone (stutter tone) audible when line is in use.

Mute

1. Press the MUTE button to enable/disable muting the microphone.

2. The LCD will show “MUTE” or "Talking to" to indicate whether the microphone is muted.

Call Transfer

GXP2124 supports both Blind and Attended transfer. Also, users could make auto-attended transfer when this feature is enabled from web GUI.

1. Blind Transfer: Press “TRANSFER” button, then dial the number and press the “SEND” button or "Transfer" softkey to complete transfer of active call.

2. Attended Transfer: Press “LINEx” button to make a call and automatically place the ACTIVE LINE on HOLD. Once the call is established, press “TRANSFER” button then press the LINE key of the waiting line to transfer the call. Hang up the call after transfer is successful.

3. Auto-Attended Transfer: To make attended transfer in an easy way, users could enable AutoAttended Transfer under Web GUI->Advanced Setting Page. To complete attended transfer using Auto-Attended Transfer mode, follow the steps below:

 GXP2124 is in an active call. Press “TRANSFER” button. It will bring up another line using

the current account and put the previous call on hold.

 Enter the number and press SEND key to establish the second call. (If pressing "Transfer"

softkey after entering the number, it will do blind transfer instead)

 After the second call is established, press “TRANSFER” button again.  Now the phone will hang up and the call will be transferred.

NOTE:

 To transfer calls across SIP domains, SIP service providers must support transfer across SIP

domains. Blind transfer will usually use the primary account SIP profile.

5-Way Conferencing

GXP2124 can host conference calls and supports up to 5-way conference calling.

1. Initiate a Conference Call

 Establish a connection with two or more parties.  Press CONF button  Choose the desired line to join the conference by pressing the corresponding LINE button

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 Repeat previous two steps for all other parties that would like to join the conference. This

can be done at any time.

2. Hold Conference:

 During the conference, pressing "Hold" softkey or a new-established call can put the current

conference call on hold.

 When the GXP2124 is on screen with all conferenced-lines on hold, press "ReConf" softkey

to re-establish the conference call.

 When the conference call is put on hold, users could also press the separate line key to talk

to each line. However, in this case, the host will have to individually re-join the held lines back into the conference by pressing CONF button and selecting the line for each line.

3. Cancel Conference:

 If after pressing the “CONF” button, a user decides not to conference anyone, press the

current active LINE button or "Cancel" softkey.

 This will resume two-way conversation with the current line.

4. End Conference:

 Press HOLD to end the conference call and put all parties on hold.  To speak with an individual party, select the corresponding blinking LINE.

GXP2124 also supports Easy Conference mode. In Easy Conference mode, users can initiate conference by calling another number when the current line is in talking or conference. Also the conference can be reestablished by pressing the ReConf softkey when the conference is on hold. Easy Conference mode can be used combined with the traditional ways to establish the 5-way conference above.

1. Initiate a Conference Call:

 Establish one call.  Press CONF button and a new line will be brought up.  Dial the number and press SEND button to establish the second call.  Press CONF button again or press the ConfCall softkey to establish the 3-way conference.

2. Join More Parties in Existed Conference:

 Establish conference call.  Press CONF button and a new line will be brought up.  Dial the number and press SEND button to establish the new call.  Press CONF button again or press the ConfCall softkey to join the new party in the existed

conference.

3. Hold Conference:

 During the conference, press HOLD button and the conference will be put on hold.

- To resume the conference, press the ReConf softkey.

- To split the conference and resume the call with each party, press the corresponding line key.

4. End Conference:

 If the users decide not to conference after establishing the second call, press EndCall

softkey instead of ConfCall softkey/CONF button. It will end the second call and the screen will show the first call/conference is on hold.

 During the conference, press EndCall softkey or hang up to end the conference.

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NOTE:

 The party that starts the conference call has to remain in the conference for its entire duration, you

can put the party on mute but it must remain in the conversation. Also, this is not applicable when the feature “Transfer on call hangup” is turned on.

 When using Easy Conference mode, press SEND button to establish the second call after entering

the number instead of using “#”.

 For the 4-way and 5-way traditional conference call, PCMU and PCMA are supported for the codec

being used in the conference.

Voice Messages (Message Waiting Indicator)

A blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Press the Message button to retrieve the message. An IVR will prompt the user through the process of message retrieval. Press a specific LINE to retrieve messages for a specific line account.

NOTE:

 Each line has a separate voicemail account. Each account requires a voicemail portal number to be

configured in the “Voicemail User ID” field.

 To check which line account has a message 1) press MENU->Voice Mails; or 2) the message

button; or 3) check each line for stutter tone; or 4) check missed calls using the menu.

Busy Lamp Field

The Multi Purpose Key buttons can be configured for Busy Lamp Field function with a specified account. When BLF is configured on one of the multi-functional buttons, the Speed Dial function will work when that line is not in use. Call Pick Up is supported when user presses a flashing BLF key.

Eventlist BLF

The Multi Purpose Key buttons can be configured for eventlist BLF (PBX server must support evenlist BLF for the phone to use it). Follow the steps below to configure the phone to use eventlist BLF.

 Under web GUI->Account setting page, configure "eventlist BLF URI" as set up in the server side.

For example, BLF12345.

 Under web GUI->Basic setting page, configure the Muti Purpose Key mode to "eventlist BLF" for the

selected account.

 For each eventlist BLF key, enter the userID to be monitored. The userID needs to be configured

under the evenlist BLF group in the server side already.

When eventlist BLF is configured on the Multi Purpose Keys, the Speed Dial function will work when that line is not in use.

Shared Call Appearance (SCA)

The GXP2124 phone supports shared call appearance by Broadsoft standard. This feature allows members of the SCA group to shared SIP lines and provides status monitoring (idle, active, progressing, hold) of the shared line. When there is an incoming call designated for the SCA group, all of the members of the group

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will be notified of an incoming call and will be able to answer the call from the phone with the SCA extension registered.

All the users that belong to the same SCA group will be notified by visual indicator when a user seizes the line and places an outgoing call, and all the users of this group will not be able to seize the line until the line goes back to an idle state or when the call is placed on hold. (With the exception of when multiple call appearances are enabled on the server side).

In the middle of the conversation, there are two types of hold: Public Hold and Private Hold. When a member of the group places the call on public hold, the other users of the SCA group will be notified of this by the redflashing button and they will be able to resume the call from their phone by pressing the line button. However, if this call is placed on private-hold, no other member of the SCA group will be able to resume that call.

To enable shared call appearance, the user would need to register the shared line account on one of the accounts on the phone. In addition, they would need to navigate to “Settings”->”Basic Settings” on the web UI and set the line to “Shared Line” with the corresponding account. If the user requires more shared call appearances, the user can configure multiple line buttons to be “shared line” buttons associated with the account.

For more information on configuring SCA to work on Broadsoft platform, please refer to the link below:

http://www.grandstream.com/products/gxp_series/general/documents/gs_sca_guide.pdf

CALL FEATURES

The GXP2124 supports traditional and advanced telephony features including caller ID, caller ID w/name, call forward/transfer/park/hold as well as intercom/paging and BLF.

Table 10: GXP2124 Call Features

Code Call Features

*30

Block Caller ID (for all subsequent calls) Offhook and dial “*30”.

*31

Send Caller ID (for all subsequent calls) Offhook and dial “*31”.

*67

Block Caller ID (per call) Offhook, dial “*67” and then enter the number to dial out.

*82

Send Caller ID (per call) Offhook, dial “*82” and then enter the number to dial out.

*70

Disable Call Waiting (per Call) Offhook, dial “*70” and then enter the number to dial out.

*71

Enable Call Waiting (per Call) Offhook, dial “*71” and then enter the number to dial out.

*72 Unconditional Call Forward

Offhook, dial “*72”. Then enter the number to forward the call. Press OK softkey or

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SEND button.

*73 Cancel Unconditional Call Forward

Offhook, dial “*73” and the phone will hang up.

*90 Busy Call Forward

Offhook, dial “*90”. Then enter the number to forward the call. Press OK softkey or SEND button.

*91 Cancel Busy Call Forward

Offhook, dial “*91” and the phone will hang up.

*92 Delayed Call Forward

Offhook, dial “*92”. Then enter the number to forward the call. Press OK softkey or SEND button.

*93 Cancel Delayed Call Forward

Offhook, dial “*93” and the phone will hang up.

CUSTOMIZED LCD SCREEN & XML

Grandstream GXP2124 Enterprise IP phone su pport s 1) XML Cu stom Screen, 2) X ML Do wnload able Ph onebo ok, and 3) XML application. Please refer to the following li nk for document ation and templates.

1) XML Custom Screen (custom idle screen logo, softkey layout and etc.)

http://www.grandstream.com/products/gxp_series/gen eral/docum ents/GXP21xx _1450_XML _Screen_Cu stomizatio n.rar

For GXP2124, please make sure the default idle screen file name is gs_idle_screen.xml.

2) XML Downloadable Phonebook

http://www.grandstream.com/products/gxp_series/general/document s/gxp_wp_xml _phonebo ok.pdf

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Configuration Guide

The GXP2124 can be configured in two ways. Firstly, using the Key Pad Configuration Menu on the phone; secondly, through embedded web-configurati on menu.

CONFIGURATION VIA KEYPAD

To enter the MENU, press the MENU/OK round button in the center of the navigation arrow keys. Navigate the menu by using the arrow keys: up/down and left/right. Press the MENU/OK button to confirm a menu selection. The phone automatically exits MENU mode with an incoming call, the phone is off-hook or the MENU mode if left idle for 60 seconds.

The menu options available are listed in table 11 a s below.

Table 11: Key Pad Configuration Menu

Menu Description

Call History Status

Phone Book

LDAP Directory Instant Messages Direct IP Call Preference

Displays histories of answered, dialed, missed, transferred and forwarded calls. Displays the network status, account status, software version, MAC address and

hardware version of GXP2124. Displays the phonebook, edit entries, search entries and downloads phonebook

XML file from the server path as configured in phone's web GUI->Advanced setting page.

Displays the LDAP directory, downloads directory and configure LDAP filter. Views received instant messages. Dials IP address for direct IP call Press MENU button to enter this sub menu including:

 Do NOT Disturb

DND (Do Not Disturb) function could be turned on or off in the “Do Not Disturb” menu.

 Forward Call

Configure call forward type (Call Forward Immediate/Call Forward Delayed) and call forward number for each account.

 Ring Tone

Choose different ring tones in the “Ring Tone” menu.

 Ring Volume

Press left/right arrow key to adjust and hear the selected ring volume. Press OK softkey to confirm the change, or Cancel softkey to exit.

 LCD Contrast

Press left/right arrow key to adjust the LCD contrast. Press OK softkey to confirm the change, or Cancel softkey to exit.

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 LCD Brightness

Press left/right arrow key to adjust the LCD brightness for active/idle screen. Press OK softkey to confirm the change, or Cancel softkey to exit.

 Download SCR XML

The phone will download the custom idle screen if available.

 Erase Custom SCR

Custom idle screen will be erased and will be replaced with default logo.

 Display Language

Users can choose English, Simplified Chinese, Traditional Chinese, Korean, Japanese, Italian, Spanish, French, German, Portuguese, Russian, Croatian, Hungarian, Polish, Slovenian, Arabic, Hebrew or Dutch which are built in the phone. Users could select Automatic for local language based on IP location if available. Also, the phone will download secondary language if available.

 Time Settings

Users can set the date and time on the phone.

Press MENU button to select the sub menu. Press left arrow button or follow the soft keys to return to the main menu.

Config

Press MENU button to display the configuration selections:

 SIP

To change SIP server settings for SIP account (SIP Proxy, Outbound Proxy, SIP User ID, SIP Auth ID, SIP Password, SIP Transport and Audio).

 Upgrade

To configure the firmware server and Config server for upgrading or provisioning the phone.

 Factory Reset

Reset the phone to factory default setting. Do not use Factory Reset unless you want to restore factory settings.

 Layer 2 QoS

Configure 802.1Q/VLAN Tag and priority value.

 Headset Type

Select GXP2124 to use normal headset or Plantronics EHS headset. Please make sure headset is plugged in the RJ9 port on the back of the phone before using the headset.

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Factory Functions

Network

Call Features

Press MENU to display the factory function items including:

 Audio Loopback

Speak into the handset or speaker. If you hear your voice from handset or speaker, the audio is working fine. Press MENU button or OK softkey to exit the audio loopback mode.

 Diagnostic Mode

All LEDs will light up. Press any key on the keypad, to display the button name in the LCD. Lift and put back the handset or press MENU button to exit the diagnostic mode.

 Keyboard Diagnostic

Press all available keys. When done, press HOOK to exit keyboard

diagnostic mode. Press left arrow button or follow the softkeys to return to the main menu. To select IP mode (DHCP/Static IP/PPPoE); to setup PPPoE, IP address,

Netmask, Gateway address, DNS Server 1 and DNS Server 2; to set up 802.1X mode (Mode/Identity/MD5 Password).

To enable/disable and configure Forward All, Forward Busy, Forward No Answer, No Answer Timeout, press Call Features and select the Account x to set the forward call features on this account.

Voice Mails

To view voice message information for each account. If voice mail user id is configured for the selected account in web GUI->Account

setting page, press MENU/OK button on the selected the account will directly dial into the voicemail.

Reboot Exit

Reboot. Exit from this menu.

In next page, Figure 1 shows the Keypad GUI flow for GXP2124.

IGURE 1: KEYPAD GUI FLOW

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Call Histor

Answered Calls Dialed Calls Missed Calls Transferred Calls Forwarded Calls Clear All Back

Call History

Status

Phone Book

LDAP Directory

Instant Message

Direct IP Call

Preference

Config

Factory Functions

Network

Call Features Voice Mails Reboot

Exit

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Phone Book

Group New Entry Search Download Phonebook XML Delete All Entries Back

LDAP Directory

View Directory Download Directory LDAP Configuration Back

Instant Message

Clear All Back

Preference

Do Not Disturb Forward Call Ring Tone Ring Volume LCD Contrast LCD Brightness Download SCR XML Erase Custom SCR Display Language Time Settings Back

Config

SIP Upgrade Factory Reset Layer 2 QoS Headset Type Back

Factory Function

Audio Loopback Diagnostic Mode Keyboard Diagnostic

Network

IP Setting PPPoE Settings IP Netmask Gateway DNS Server 1 DNS Server 2

802.1X Back

Call Features

Account 1 Account 2 Account 3

Call History Items

Delete All Entries

New Entry

First Name: Last Name Number: Acct: Confirm Add: Cancel & Return:

LDAP Configuration

Select Filter Filter Value Back

Do Not Disturb

Enable DND Disable DND Back

Ring Tone

Default Ring Ring1 Ring2 Ring 3 Back

SIP

Account SIP Proxy Outbound Proxy SIP User ID SIP Auth ID SIP Password SIP Transport Audio Save Cancel

Upgrade

Firmware Server Config Server Upgrade Via Back

Layer 2 QoS

802.1Q/VLAN Tag Priority value Reset Vlan Config Back

Account X

Forward All Forward Busy Forward No Answer No Answer Timeout

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CONFIGURATION VIA WEB BROWSER

The GXP2124 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 Microsoft’s IE, Mozilla Firefox and Google Chrome.

Access the Web Configuration Menu

To access the phone’s Web Configuration Menu

 Connect the computer to the same network as the phone  Make sure the phone is turned on and shows its IP address  Start a Web-browser on your computer  Enter the phone’s IP address in the address bar of the browser  Enter the administrator’s password to access the Web Configuration Menu

The Web-enabled computer has to be connected to the same sub-network as the phone. This can easily

be done by connecting the computer to the same hub or switch as the phone is connected to. In absence of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the phone using the PC-port on the phone.

If the phone is properly connected to a working Internet connection, the phone will display its IP address. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255. You will need this number to access the Web Configuration Menu. e.g. if the phone shows 192.168.0.60, please use “http://192.168.0.60” in the address bar your browser.

The default administrator password is “admin”; the default end-user password is “123”. Administrator level could access all the pages; end-user level could access status and basic setting pages.

NOTE:

 When changing any settings, always SUBMIT them by pressing “UPDATE” button on the bottom of

the page. If, after having submitted some changes, more settings have to be changed, press the menu option needed.

 All the options under Basic Setting and Account Setting, and most of the options under Advanced

Setting do not require reboot after submitting the changes. Under Advanced Setting, the parameters on network configuration require reboot after update.

 If you cannot log into the web GUI by using the default password, please check with the VoIP service

provider. It is most likely the VoIP service provider has provisioned the device and configured for you therefore the password has already been changed.

Definitions

This section will describe the options in the Web configuration user interface. As mentioned, a user can log in as an administrator or end-user.

Functions available for the end-user are:

 Status: Displays the network status, account status, software version and MAC address of the

phone.

 Basic: Basic preferences such as date and time settings, multi-purpose keys and LCD settings can

be set here.

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 Contacts: To view and edit phonebook, import and export phonebook XML.

Additional functions available to administrators are:

 Advanced Settings: To set advanced network settings, codec settings and XML configuration

settings.

 Account X: To configure each of the SIP accounts.

Important Settings

NAT Setting:

If the devices are kept within a private network behind a firewall, we recommend using STUN Server. The following three (3) settings are useful in the STUN Server scenario:

 STUN Server (under Advanced Settings page)

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 (under Advanced Settings page)

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 GXPs are behind the same NAT. If using a Public IP address, set this parameter to No.

 NAT Traversal (under Account Setting page)

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.

Public Mode:

The GXP2124 supports hot desking using public mode. Under public mode, users could login the GXP2124 with the SIP account user ID and password. Please following the steps below to configure the phone for public mode:

 Under web GUI->Account 1 setting page, fill up the SIP server address for account 1. Update.  Under web GUI->Advanced setting page, set Public Mode option to Yes. Update and reboot.  When phone boots up, SIP user name and Password to register to the configured SIP server in

account 1 will be required. Enter the correct account information to log in to the phone. When entering the account information, press softkey "123"/"abc" to toggle input method.

 After using the phone, go to LCD MENU->LogOut to log off the public mode.

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Table 12: Device Configuration - Status

MAC Address

IPv4 Address IPv6 Address Product Model

Part Number Software Version

System Up Time System Time Registered PPPoE Link Up

The device ID, in HEXADECIMAL format. This will be used for provisioning and is written on the label in the original box as well

as on the label located on the back panel of the device. This field shows IPv4 address of GXP2124.

This field shows IPv6 address of GXP2124 when IPv6 is used. This field contains the product model information. This field contains the product part number.

• Prog: This is the main firmware release number, which is always used for identifying the software (or firmware) upgrade version.

• Boot: Booting code version number.

• Core: Core code version number.

• Base: Base code version number.

• DSP: DSP code version number.

• Aux: Aux code version number. (For GXP2124, it shows unknown on aux version) This field shows system up time since the last reboot.

This field shows the current time on the phone system. Indicates whether accounts are registered to the related SIP server. Indicates whether the PPPoE connection is enabled (if connected to a DSL modem).

NAT type will display here too.

Service Status

• GUI: shows the GUI status - running or stopped.

• Phone: shows the phone status - running or stopped.

Core Dump

Download core dump file for troubleshooting when necessary.

Table 13: Device Configuration – Settings/Basic Settings

End User Password

This contains the password for end user to access the Web Configuration Menu. Users can enter new password here. This field is case sensitive with a maximum length of 25 alphanumeric character.

Confirm Password

Internet Protocol

IP Address

Enter the end user password again as above to confirm new password.

Select Prefer IPv4 or Prefer IPv6 for GXP2124 to obtain IP address.

The GXP2124 operates in three modes:

1. DHCP mode: The GXP2124 acquires IP address from the first DHCP server it discovers on the LAN. The DHCP option is reserved for NAT router mode. In DHCP mode, all the field values for the Static IP mode are not used.

2. PPPoE mode: Set PPPoE account ID, PPPoE password and PPPoE service name for the GXP2124 to establish PPPoE sesstion.

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3. Static IP mode: Configure IP address, Subnet Mask, Gateway, DNS Server 1, DNS Server 2 and Preferred DNS Server if Static IP mode is used.

Allow DHCP Option 120 to Override SIP Server

802.1x Mode

Line Key 1 Line Key 2 Line Key 3 Line Key 4

(line keys on the left side of LCD)

Enable DHCP Option 120 to override SIP Server. By default it's No.

This option 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-MD5. Once enabled, the user would be required to enter the following information below to be authenticated on the network:

1. Identity

2. MD5 Password

This allows the user to configure line key 1 to line key 4 and enable Shared Call Appearance for the line. Options available for Key Mode are:

1. Line

2. Shared Line

3. Speeddial

4. BLF (Busy Lamp Field) This option has to be supported on the PBX and it indicates the status of the extension. The three possible states are idle (green), busy (red), ringing (blinking red).

5. 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.

6. 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 has to support this feature.

7. Speeddial via active account This option will act just like speed dial, but based on the current active account. For instance, if the phone is offhook and account 4 is active, it will call the configured speed dial number using account 4.

8. Dial DTMF This option will allow users to enter a series of DTMF digits as configured in the UserID field for the MPK during the call. When using Dial DTMF, “Enable MPK Sending DTMF” (under Advanced Setting) has to be set to “Yes”. This option is not binding to the account.

9. Call Return The last answered calls will be dialed out. The username and UserID should be blank when using Call Return. This option is not binding to the account.

10. Transfer This option will allow user to enter the extension number to transfer with during the call.

11. Intercom This option allows user to enter the extension number to intercom.

Note:

The functions in 3, 4, 5, 6, 7, 10, 11 are connected to one of the accounts and have target username (for description purpose in web GUI) and UserID.

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Line Key 5 Line Key 6 Line Key 7 Line Key 8

(line keys on the right side of LCD)

This allows the user to configure line key 5 to line key 8, which has similar functions as Multi Purpose Keys. Options available for Key Mode are:

1. Speeddial

2. BLF (Busy Lamp Field) This option has to be supported on the PBX and it indicates the status of the extension. The three possible states are idle (green), busy (red), ringing (blinking red).

3. 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.

4. 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 has to support this feature.

5. Speeddial via active account This option will act just like speed dial, but based on the current active account. For instance, if the phone is offhook and account 4 is active, it will call the configured speed dial number using account 4.

6. Dial DTMF This option will allow users to enter a series of DTMF digits as configured in the UserID field for the MPK during the call. When using Dial DTMF, “Enable MPK Sending DTMF” (under Advanced Setting) has to be set to “Yes”. This option is not binding to the account.

7. Call Return The last answered calls will be dialed out. The username and UserID should be blank when using Call Return. This option is not binding to the account.

8. Transfer This option will allow user to enter the extension number to transfer with during the call.

9. Intercom This option allows user to enter the extension number to intercom.

Note:

The functions in 1, 2, 3, 4, 5, 10, 11 are connected to one of the accounts and have target username (for description purpose in web GUI) and UserID.

Multi Purpose Keys

These options are used to assign a function to the corresponding Multi Purpose Key. Options available are:

1. Speed Dial

2. BLF (Busy Lamp Field) This option has to be supported on the PBX and it indicates the status of the extension. The three possible states are idle (green), busy (red), ringing (blinking red).

3. 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.

5. Speed Dial Via Active Account This option will act just like speed dial, but based on the current active

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account. For instance, if the phone is offhook and account 4 is active, it will call the configured speed dial number using account 4.

7. Call Return The last answered calls will be dialed out. The username and UserID should be blank when using Call Return. This option is not binding to the account.

8. Transfer This option will allow user to enter the extension number to transfer with during the call.

9. Intercom This option allows user to enter the extension number to intercom.

Note:

The functions in 1, 2, 3, 4, 5, 8, 9 are connected to one of the accounts and have target username (for description purpose in web GUI) and UserID.

Time Zone

Self-Defined Time Zone

This parameter controls the date/time display according to the specified time zone. If “Allow DHCP Option 2 to override Time Zone setting” is checked, the time zone will

be overridden by the DHCP server. This parameter allows the users to define their own 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 which is U.S central

time. If it is positive (+) if the local time zone is 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, 3

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.

Weather Update

By default, “Enable Weather Update:” is set to “Yes”. If set to “No”, weather information will not display on the phone.

Settings to customize the display of weather via:

 City Code – Enter city code (for example, USCA0638 is the city code for Los

Angeles, CA, US.)

 Update Interval – Refresh time in minutes  Degree Unit – Select Automatic, Fahrenheit or Celsius

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Weather information is displayed on GXP2124 LCD when “Enable Weather Update” is set to “Yes” and pressing the “SwitchSCR” soft-key once.

Stock Update

Currency Update

LCD Backlight Brightness

By default, “Enable Stock Update:” is set to “Yes”. If set to “No”, stock information will not display on the phone.

Settings to customize the display of stock via:

 Stock Code – Enter stock code

Stock information is displayed in GXP2124 LCD when “Enable Stock Update” is set to “Yes” and pressing the “SwitchSCR” soft-key twice.

By default, the stock code configured on the GXP2124 is: DJI;.IXIC;INX;.FTSE;.STOXX50E;.FCHI;.N225;.BSESN;.HSI;.TWII

By default, “Enable Currency Update:” is set to “Yes”. If set to “No”, currency information will not display on the phone.

Settings to customize the display of currency via:

 Currency Code – Enter currency code

Currency information (foreign currencies to US dollar) is displayed in GXP2124 LCD when “Enable Currency Update” is set to “Yes” and pressing the “SwitchSCR” softkey three times.

By default, the stock code configured on the GXP2124 is: EUR/USD;GBP/USD;CAD/USD;AUD/USD;CNY/USD;JPY/USD

Set the LCD brightness level for idle state and active state. Range from 0 to 8 where 0 means off and 8 means the brightest. Default setting is 6 for active and 2 for idle.

LCD Contrast

Date Display Format

Set LCD contrast. Range from 0 to 20. Default is 11.

LCD date display in the idle screen. The following formats are supported:

• yyyy-mm-dd: 2011-11-17

• mm-dd-yyyy: 11-17-2011

• dd-mm-yyyy: 17-11-2011

• dddd, MMMM dd: Thursday, November 17

• MMMM dd, dddd: November 17, Thursday

Time Display

LCD time display in 12 hour or 24 hour format.

Format Disable in-call

Default is “No”. This field is used to hide the keypad input during a call.

DTMF display Disable Missed Call

Backlight Always Ring

Speaker

Default is “No”. By default, LCD backlight will light up whenever there is a missed call.

Default is "No". This options is used when Headset is used on "Toggle Headset/Speaker" mode.

If selected to "Yes", when the phone is in Headset mode, both headset and speaker

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will ring on incoming call.

HEADSET Key Mode

Headset Type

When headset is connected to GXP2124, users could use the HEADSET button in two different modes:

1. Default Mode:  When GXP2124 is in idle, press HEADSET button to offhook the phone and

make calls by using headset. Headset icon will display on the top of the call screen in dialing/talking status.

 When there is an incoming call, press HEADSET button to pick the call by

using headset.

 When there is an active call using headset, press HEADSET button to hang

up the call.

 When Speaker/Handset is 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 previous mode.

2. Toggle Headset/Speaker  When GXP2124 is in idle, press HEADSET button to switch to Headset

mode. The idle screen will display a Headset icon. In this mode, if pressing Speaker button or softkey, headset will be used by default.

 When there is an incoming call, press LINE button, “Answer” softkey or

Speaker button, headset will be used.

 When there is an active call, press HEADSET button to toggle between

Headset and Speaker.

Set headset type to Normal (for regular RJ9 headset) or Plantronics EHS (for Plantronics EHS headset).

Call History Flash Writing

Max Unsaved Log

This defines the interval to save the call history to phone's flash. By default it's 300 seconds. The valid range is 30 to 3600 seconds.

This defines the number of existed logs before they are saved to phone's flash. By default it's 200 entries. The valid range is 0 to 500.

Headset TX gain

Set headset TX gain to -6, 0 or +6. Default is 0 db.

(dB) Headset RX gain

Set headset RX gain to -6, 0 or +6. Default is 0 db.

(dB) Handset TX gain

Set handset TX gain to -6, 0 or +6. Default is 0 db.

(dB)

Advanced User configuration includes not only the end user configuration, but also advanced configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuratio n.

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Table 14: Device Configuration – Settings /Advanced Settings

Admin Password

Confirm Password

Layer 3 QoS

Layer 2 QoS

Local RTP port

Use Random Port

Keep-alive interval

Enter new administrator password. On ly the administrator can access the “Advanced Settings” and “Account Settings” page. Password field is purposely blank for security reasons after clicking update and saved. This field is case sensitive with a maximum length of 25 alphanumeric character.

Enter the end user password again as above to confirm new password. This field defines the layer 3 QoS parameter. It is the value used for IP Precedence or

Diff-Serv or MPLS. Default value is 12. This contains the value used for layer 2 802.1Q/VLAN tag and 802.1p priority value.

Default setting is 0. Note: VLAN supported equipment is required when configurin g these settings.

This parameter defines the local RTP port pair 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. The default value is 5004.

This parameter, when set to “Yes”, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple GXPs are behind the same NAT. Default is “No”.

This parameter specifies how often the GXP2124 sends a blank UDP packet to the SIP server in order to keep the NAT “pin hole” open. Default is 20 seconds.

Use NAT IP STUN Server

Firmware Upgrade and Provisioning

XML Config File Password

HTTP/HTTPS User Name

HTTP/HTTPS Password

NAT IP address used in SIP/SDP message. Default is blank. IP address or Domain name of the STUN server. STUN resolution result will display

in the STATUS page of the Web UI. Allows the user to select the following options for firmware upgrade:

 Always Check for New Firmware  Check New Firmware only when F/W pre/suffix changes  Always Skip the Firmware Check.

Firmware upgrade may take up to 10 minutes depending on network environment. Please do not interrupt the firmware upgrading process.

Note: Grandstream strongly recommends that the user upgrade firmware locally in a LAN environment if using TFTP to upgrade. Please DO NOT interrupt the upgrade process (especially the power supply) as this will damage the device.

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

The user name for the HTTP/HTTPS server.

The password for the HTTP/HTTPS server.

Upgrade Via

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This field allows the user to choose the firmware upgrade/config server path method:

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TFTP, HTTP or HTTPS.

• TFTP:

GXP2124 retrieves the new firmware files or new configuration file from the specified TFTP server path at boot time. If there is no new firmware file or configuration file, the system will start the boot process using the existing firmware or config file. If a TFTP server is configured and new firmware files are retrieved, the new downloaded image is saved into the Flash memory. (Please do NOT interrupt the TFTP upgrade process (especially the power supply) as this will damage the device.

• HTTP:

GXP2124 retrieves the new firmware files or new configuration file from the specified URL or IP for the HTTP server. For example:

provisioning.mycompany.com:6688/Grandstream/1.0.0.6 Note: “:6688” is the specific TCP port where the HTTP server is listening; Omit if

using default port 80.

• HTTPS:

GXP2124 retrieves the new firmware files or new configuration file from the specified URL or IP for the HTTP server via a secured HTTP connection. For example:

provisioning.mycompany.com

Note: HTTPS default port is 443.

Firmware Server Path

Config Server Path

Firmware File Prefix/Postfix

Config File Prefix/Postfix

Allow DHCP Option 43 and Option 66 to override server

Defines the server path for the firmware server. It can be different from the Configuration server which is used for provisioning.

For example: firmware.mycompany.com:6688/Grandstream/1.0.0.6

Defines the server path for provisioning; it can be different from the firmware server. For example:

provisioning.mycompany.com:6688/Grandstream/gxp2120 Default is blank. If configured, GXP2124 will request the firmware file with the

prefix/postfix and only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone.

Note: This setting is useful for ITSPs. End user should keep it blank. Default is blank. If configured, GXP2124 will request the config file with the

prefix/postfix and only the file with the matching encrypted prefix will be downloaded and flashed into the phone.

Note: This setting is useful for ITSPs. End user should keep it blank. Default is “Yes”. This allows device gets provisioned from the server specified path

automatically.

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Automatic Upgrade

Authenticate Conf File

Enable TR-069 ACS URL

TR-069 Username TR-069 Password Periodic Inform

Enable Periodic Inform

Interval Connection

Request Username

Default is “No”. Choose “Yes” to enable automatic HTTP upgrade and provisioning. In “Check for upgrade every” field, enter the number of minutes to check the HTTP

server for firmware upgrade or configuration changes. When set to “No”, the phone will only perform HTTP upgrade and configuration check once at boot up.

Note: This function is used by ITSP. End user should NOT touch these parameters. Default is “No”. If set to “Yes”, configuration file would be authenticated before

acceptance. End user should use default setting. Enable TR-069. Default is “No”. URL for TR-069 Auto Configuration Servers (ACS).

Enter ACS username for TR-069. Enter ACS password for TR-069. Enable periodic inform. Default is “No”. If set to YES, device will send inform packets

to the ACS. When enabling periodic inform, set up the periodic inform interval to send the inform

packets to the ACS. Enter the connection request username. This is the user name for the ACS to connect

to this device.

Connection Request Password

Authentication Method

Connection Request Port

Phonebook XML Download

Phonebook XML Server Path

Phonebook Download Interval

Remove Manuallyedited entries on Downloads

LDAP Directory

Enter the connection request password. This is the password for the ACS to connect to this device.

Select the authentication method among “No authentication”, “Basic” or “Digest”.

Enter the connection request port. This is the port for the ACS to connect to this device.

Selects the file download mode for the download server. Users can choose from TFTP/HTTP/No.

The URL/IP address of the phonebook download server.

The interval at which the phonebook will be downloaded from the download server (in minutes). The default setting is 0.

If set to “Yes”, the phone will remove the manually-edited entries in the old phonebook list before downloading the new file. The default setting is set to “Yes”.

Note:

If there is a duplicate entry (same name, same number and same index) added both manually and from phonebook xml file, only 1 entry will be saved and displayed.

IP address or domain name of LDAP script server.

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Idle Screen XML Download

Download Screen XML At Boot-up

Use custom filename

Idle Screen XML Server Path

XML Application

Offhook Auto Dial

Auto recover from abnormal

Syslog Server

Enable XML Idle Screen download via TFTP or HTTP. Select whether to “Use Custom Filename” or not, and define the “XML server path”.

Please refer to the section “customized LCD screen and XML” for more information. The phone will download the idle screen xml file at boot up if set to “Yes”. The default

setting is “No”. The phone will use custom filename specified in XML server path if set to “Yes”. The

default setting is “No”. Specify the idle screen XML server path.

Enter the xml application server path and soft key label to display on the LCD. When pressing the softkey, it will initiate the HTTP request to the HTTP server to start the application.

To configure a User ID/extension to dial automatically when the phone is taken offhook.

By default is Yes. Phone will auto recover and work normal if set to Yes.

The IP address or URL of System log server. This feature is especially useful for ITSPs.

Syslog Level

Send SIP Log

Select the syslog level for GXP2124 to report. Default is NONE. The level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are sent

based on the following events:

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

The Syslog uses USER facility. In addition to standard Syslog payload, it contains the following components: GXP_2124: [device MAC address][firmware version] error message.

For example: Nov 30 07:55:46 10.131.28.27 GXP2124_GUI:[00:0b:82:2b:3c:a4][1.0.1.108][-LCD-] IdleScreen/XmlApp is now visible

When set to “Yes”, phone will send out SIP Log to syslog server. Default setting is “No”.

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NTP server

Allow DHCP Option 42 to override NTP server

Public Mode

SSL Certificate SSL Private Key SSL Private Key

Password Distinctive Ring

Tone

This parameter defines the URI or IP address of the NTP (Network Time Protocol) serve. It is used to display the current date/time.

Default is “Yes”. This allows device gets provisioned for DHCP Option 42 from the local server automatically.

Enable to turn on hotdesking feature. To use it, fill up the SIP server address in

ccount 1 setting page first. Then set public mode to Yes, update and reboot the

phone. When phone boots up, SIP user name and Password to register to the configured SIP

server in account 1 will be required. Enter the correct account information to use the phone.

After using the phone, go to LCD MENU->LogOut to log off the public mode.

The user specified SSL certificate used for SIP over TLS. The user specified SSL private key used for SIP over TLS. User specified password to protect the private key above.

Select a Distinctive Ring Tone 1 through 3 for the caller ID. The Caller ID can be configured as a particular caller ID or Alert-Info text.

When a particular call ID is configured in the Caller ID, the selected ring tone will be used when the incoming call is from the Caller ID. System Ring Tone will be used for all other calls.

If server supports Alert-Info, the Alert-Info text can be mapped to the 3 customized ring tones. For example, if you configure the custom ring tone 1 user ID to “priority”, that ring tone will be used if we receive INVITE with Alert-Info header in the following format:

Alert-Info:;info=priority

System Ring Tone

System ring tone. Default is North American standard. Users could adjust system ring tone frequencies and cadences based on local telecom standard.

Call Progress Tones

Using these settings, users can configure ring or tone frequencies based on parameters from local telecom. By default, they are set to North American standard. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds.

Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];

- Frequencies are in Hz

- Cadence on and off are in 10ms

- ON is the period of ringing (“On time” in “ms”) while OFF is the period of silence

- In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms

and a pause of OFF ms and then repeat the pattern

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- Up to three cadences are supported

Disable Call Waiting

Disable Call Waiting Tone

Disable Direct IP Calls

Use Quick IP Call Mode

Disable Conference Enable MPK

Sending DTMF Disable DND Button

Default is “No”. If set to “Yes”, the call waiting feature will be disabled.

Default is “No”. If set to “Yes”, the call waiting tone will be disabled.

Default is “No”. If set to “Yes”, direct IP calls will be disabled.

Dial an IP address under the same LAN/VPN segment by entering the last octet in the IP address.

In the Advanced Settings page there is an option “Use Quick IP-call mode”. Default setting is “No”. When set to “Yes”, offhook the phone and dial #XXX (X is 0-9 and XXX <=255), phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet mask.

#XX or #X are also valid so leading 0 is not required (but OK). See Quick IP Call Mode section for details.

Default is “No”. If set to “Yes”, conference will be disabled. Default is No. If set to “Yes”, Multi Purpose keys can be sent as DTMF during the call.

Default is “No”. If set to “Yes”, the “DND” button on keypad will be disabled.

Disable Transfer Auto-Attended

Transfer Configuration via

Keypad Menu

Enable STAR key Keypad locking

Password to

Default is “No”. If set to “Yes”, transfer will be disabled. Default is “No”. If set to “Yes”, the phone will use attended transfer by default.

Configures the access control of configurations via the phone keypad menu. There are three modes:

 Unrestricted  Basic Settings Only

CONFIG option will not display in keypad MENU

 Constraint Mode

CONFIG, FACTORY FUNCTIONS and NETWORK options will not display in keypad MENU

Default is enabled. When the phone is in idle screen, press and hold STAR key for 4 seconds and the keypad will be locked.

To unlock the keypad, press and hold STAR key for 4 seconds. Enter the password to unlock in the prompted window, and then press MENU key to enter.

The password to lock/unlock can be configured. If users forget the password to unlock it, log into web GUI and change the password. Update and reboot the phone. When the phone boots up, press and hold STAR key for 4 seconds and enter the new password to unlock it.

Enter the password to lock the keypad in web GUI.

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lock/unlock

China Telecom Mode

Do not escape “#” as %23 in SIP URI

Display Language

To unlock the keypad, enter the password in the prompted window in the phone’s LCD screen.

Enable or disable China Telecom Mode.

Default is “No”. By default, # will be replaced as %23 in SIP URI.

Allows user to choose preferred display language in web UI and keypad UI. Currently, the phone supports these languages: Arabic, Czech, German, English, Spanish, French, Hebrew, Croatian, Hungarian, Italian, Japanese, Korean, Dutch, Polish, Portuguese, Russian, Slovenian, Simplified Chinese and Traditional Chinese.

Note: The “Automatic” setting in language refers to Grandstream’s IP2Location client which when connected to Internet would attempt to lookup a database (driven by Grandstream) with the IP address for its geographical location.

Language file postfix allows the language file to have different postfixes so the phone can request a particular file. It will append an underscore “_” plus the string in the language file postfix.

The default language file name is “gxp.txt”. If the field “Language File postfix “has “NL” string in it, then the phone will request “gxp_NL.txt” instead of “gxp.txt”.

User can only load one secondary language. Supported downloadable language: Czech, Croatian, Estonian, French, German, Italian, Polish, Portuguese, Slovak, Slovenian and Spanish.

How to set up Download Language: This is similar to updating firmware in your local network environment.

1. Get the language file gxp.txt ready. Make sure the file is using UTF-8 encoding.

2. Copy gxp.txt to the firmware server directory using your local TFTP or HTTP

server.

3. Access the advanced settings of the Web GUI, set “Display Language” to

“Download Language” and enter the server path in Firmware Server Path. Select TFTP or HTTP for firmware upgrade.

4. Update and reboot the phone.

Download Device Configuration

Download the current device configuration txt file. In the txt file, all the P values will be displayed except for the password fields.

GXP2124 has up to 4 line appearances, each with an independent SIP account. Each SIP account requires its own configuration page. Their configurations are identical.

Table 15: SIP Account Settings

Account Active Account Name SIP Server

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This field indicates whether the account is active. The default value is “Yes”. The name associated with each account – displayed on LCD. SIP Server’s IP address or Domain name provided by VoIP service provider.

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Secondary SIP Server

Outbound Proxy

SIP User ID

Authenticate ID

This field contains the URL or the IP address of a second SIP server. When this field is configured, GXP2124 will send out Registration requests and

Subscribe messages (except for message waiting) to the “SIP Server” and “Secondary SIP Server” for the same account.

When making a call, GXP2124 will use the registered primary “SIP server” first. If this primary “SIP Server” is not available, the registered “Secondary SIP Server” will be used.

If the primary “SIP Server” is not registered but “Secondary SIP Server” can be registered, GXP2124 will use the “Secondary SIP Server” directly when making a call.

Note: Please do not configure duplicate SIP Server address in “SIP server” and “Secondary SIP Server”.

IP address or Domain name of Outbound Proxy, Media Gateway, or Session Border Controller. Used for firewall or NAT penetration in different network environment. If the system detects symmetric NAT, STUN will not work. ONLY outbound proxy can provide solution for symmetric NAT.

User account information provided by VoIP service provider (ITSP); either an actual phone number or formatted like one.

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

Authenticate Password

Name DNS Mode

Primary IP

Backup IP 1 Backup IP 2 TEL URI

SIP Registration

Unregister on Reboot

SIP service subscriber’s account password for GXP2124 to register to (SIP) servers of ITSP.

SIP service subscriber’s name that is used for Caller ID display. The default is set to A Record. If users wish to locate the server by DNS SRV, users

may select SRV or NATPTR/SRV. When “Use Configured IP” option is selected, if SIP server is configured as domain name, phone will not send DNS query, but use “Primary IP” or “Secondary IP” to send sip message if at least one of them are not empty.

This option applies only if “Use Configured IP” is selected, the phone will send DNS query to the Primary IP. Insert IP address here.

Insert the first back up IP here. Insert the second back up IP here. Default is “Disabled”. If the phone has an assigned PSTN telephone number, this field

should be turned on and a “User=Phone” parameter will be attached to the “From” header in SIP request.

This parameter controls sending REGISTER messages to the proxy server. The default setting is “Yes”.

Default is “No”. If set to “Yes”, the SIP user’s registration information will be cleared on reboot.

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This parameter allows user to specify the time frequency (in minutes) that GXP2124

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refreshes its registration with the specified registrar. The default interval is 60 minutes. The maximum interval is 65,535 minutes (about 45 days).

Reregister Before Expiration

Local SIP Port

SIP Registration Failure Retry Wait Time

SIP T1 Timeout SIP T2 Interval SIP Transport SIP URI Scheme

when using TLS Use Actual

Ephemeral Port in Contact with TCP/TLS

This parameter allows user to specify the time frequency (in seconds) that GXP2124 sends out a re-registration request before the Register Expiration. By default is 0 second.

This parameter defines the local SIP port used to listen and transmit. The default value for Account X is:

- Account 1: 5060

- Account 2: 5062

- Account 3: 5064

- Account 4: 5066

Retry registration if the process failed. Default is 20 seconds.

RFC 3261 SIP T1 timer. Default is 0.5 second. RFC 3261 SIP T2 timer. Default is 4 seconds. Choose SIP Transport between UDP, TCP and TLS/TCP. Default is UDP. When TLS/TCP is used (in “SIP Transport”), select “sip:” or “sips:” in this mode.

Default is “sips:”. Enable to use actual ephemeral port in contact with TCP/TLS. Default is “No”.

Check Domain

Enable to check the domain certificate. Default is “No”.

Certificate Remove OBP from

The SIP Extension notifies the SIP server that it is behind a NAT/firewall.

Route Validate Incoming

This configuration selects whether or not the incoming messages should be validated.

Messages Support SIP

Instance ID NAT Traversal

Selects whether or not SIP Instance ID is supported.

This parameter activates the NAT traversal mechanism. It has options: No, STUN, Keep-Alive, UPnP, Auto, VPN.

If selecting STUN and a STUN server is also specified, the phone performs according to the STUN client specification. Using this mode, the embedded STUN client detects if and what type of NAT/Firewall configuration is used. If the detected NAT is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use its mapped public IP address and port in all of its SIP and SDP messages.

If selecting Keep-Alive with no specified STUN server, the GXP2124 will periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to the SIP

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server to keep the “hole” on the NAT open.

SUBSCRIBE for MWI

SUBSCRIBE for Registration

Feature Key Synchronization

PUBLISH for Presence

Proxy-Require Voice Mail UserID

Default is “No”. When set to “Yes” a SUBSCRIBE for Message Waiting Indication will be sent periodically.

Default is “No”. When set to “Yes” a SUBSCRIBE for Registration will be sent periodically.

Default is “No”. This option is to synchronize DND/Call Forward features with Broadsoft platform.

When set to “Yes”, a SUBSCRIBE will be sent out periodically to the server. Then when DND/Call Forward features (Call Forward No Answer, Unconditional Call Forward and Call Forward on Busy) are configured or changed on the phone and Broadsoft server side, those features will be synchronized on the phone side and Broadsoft server side.

Enable Presence feature.

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. When configured, user can access messages by pressing “MSG” button. This userID

is usually the VM portal access number. Note: If server side feature code is used for VM portal access number, please also

configure “Enable Call Features” to “No” under account setting page to use non-local feature code.

Send DTMF

DTMF Payload Type

Early Dial

Dial Plan Prefix Dial Plan

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

- In audio: DTMF is combined in audio signal (not very reliable with low-bit-rate

codec)

- via RTP (RFC2833)

- via SIP INFO.

Sends DTMF using RFC2833. The default is 101.

Default is “No”. Use only if proxy supports 484 responses.

Sets the prefix added to each dialed number. 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; a) xx+ - at least 2 digit numbers b) xx. - only 2 digit numbers c) ^ - exclude d) [3-5] - any digit of 3, 4, or 5 e) [147] - any digit of 1, 4, or 7 f) <2=011> - replace digit 2 with 011 when dialing g) | - the OR operand

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• Example 1: {[369]11 | 1617xxxxxxx}

Allow 311, 611, and 911 or any 10 digit numbers with leading digits 1617

• Example 2: {^1900x+ | <=1617>xxxxxxx}

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.

3. Default: Outgoing – {x+}

Allow any length of numbers. 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 dialing

7 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 - allow 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.

BLF Call-pickup Prefix

Delayed Call Forward Wait Time

Enable Call Features

Call Log

Session Expiration

Min-SE

Default is “**”. This prefix is prepended when answering call with BLF key.

Time waited before the call is forward to a number or VM. Default is 20 seconds.

Default is “Yes”. If set to “No”, Call transfer, Call Forwarding & Do-Not-Disturb are supported locally provided ITSP support those features. In addition, “ForwardAll” softkey will be hidden if call feature code is disabled for Account 1.

Enable/disable Call Log and select type of calls to log:

- Log All Calls

- Log Incoming/Outgoing only (Missed calls NOT recorded)

- Disable Call Log

The SIP Session Timer extension enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE. Once the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session is terminated.

Session Expiration is the time (in seconds) at which the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default value is 180 seconds.

Defines the minimum session expiration (in seconds). Default is 90 seconds.

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Caller Request Timer

Callee Request Timer

Force Timer

UAC Specify Refresher

UAS Specify Refresher

Force INVITE

Enable 100rel

Account Ring Tone

If set to “Yes”, the phone will use session timer when it makes outbound calls if remote party supports session timer.

If selecting “Yes”, the phone will use session timer when it receives inbound calls with session timer request.

If set to “Yes”, the phone will use session timer even if the remote party does not support this feature. If set to “No”, the session timer is enabled only when the remote party supports this feature. To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.

s a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee

or proxy server as the refresher.

s a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use

the phone as the refresher. Session Timer can be refreshed using INVITE method or UPDATE method. Select

“Yes” to use INVITE method to refresh the session timer. PRACK (Provisional Acknowledgment) method enables reliability to SIP provisional

responses (1xx series). This is required to support PSTN inter-networking. There are 4 uniquely defined ring tones:

 System Ring Tone: when selected, all calls will ring with system ring tone.  3 Customer Ring Tones: when selected, incoming calls from designated

account will play selected ring tone.

Ring Timeout Line-seize Timeout

Send Anonymous

Anonymous Call Rejection

Auto Answer

Allow Auto Answer by Call-Info

Refer-To Use Target Contact

Transfer on Conference Hangup

Check SIP User ID for Incoming

Defines how long the phone will ring when receiving a call. Default is 60 seconds. Defines how long before the line can be seized when Share Line is used. Default is

15 seconds. If this parameter is set to “Yes”, the “From” header in outgoing INVITE message will

be set to anonymous, essentially blocking the Caller ID from displaying. Default is “No”. If set to “Yes”, anonymous call will be rejected.

Default is “No”. If set to “Yes”, GXP2124 will automatically switch on speaker to answer the incoming call. Set to Intercom/Paging mode, it will answer the call based on the SIP info header from the server.

If the Call-Info header contains answer-after=0, the call be answered automatically. This fields need to be set to Yes if users would like to have the phone to be paged/intercom.

Default is “No”. If set to “Yes”, then for Attended Transfer, the “Refer-To” header uses the transferred target’s Contact header information.

Defines whether or not the call is transferred to the other party if the initiator of the conference hangs up. Default is “No”.

Check the SIP User ID in Request URI. If they don’t match, the call will be rejected.

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INVITE Preferred Vocoder

SRTP Mode

Symmetric RTP Silence

Suppression

Voice Frames per TX

GXP2124 supports up to 7 different Vocoder types including G.711(a/µ) (also known as PCMU/PCMA), G.723.1, G.729A/B, G.726-32, iLBC, G.722 (wide-band).

Configure Vocoders in a preference list that is included with the same preference order in SDP message. Enter the first Vocoder in this list by choosing the appropriate option in “Choice 1”. Similarly, enter the last Vocoder in this list by choosing the appropriate option in “Choice 8”.

Enable SRTP mode based on selection. Default is “No”. Selects whether or not symmetric RTP is supported.

This controls the silence suppression/VAD feature of the audio codec G.723 and G.729. If set to “Yes”, when silence is detected, a small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to “No”, this feature is disabled.

This field contains the number of voice frames to be transmitted in a single Ethernet packet (be advised the IS limit is based on the maximum size of Ethernet packet is 1500 byte (or 120kbps)).

When setting this value, be aware of the requested packet time (ptime, used in SDP message) is a result of configuring this parameter. This parameter is associated with the first codec in the above codec Preference List or the actual used payload type negotiated between the 2 conversation parties at run time. E.g., if the first codec is configured as G.723 and the “Voice Frames per TX” is set to 2, then the “ptime” value in the SDP message of an INVITE request will be 60ms because each G.723 voice frame contains 30ms of audio. Similarly, if this field is set to 2 and the first codec is G.729 or G.711 or G.726, then the “ptime” value in the SDP message of an INVITE request will be 20ms.

If the configured voice frames per TX exceeds the maximum allowed value, the IP phone will use and save the maximum allowed value for the corresponding first codec choice. The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20 (x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms) and 64 (x2.5ms) frames respectively.

Please be careful when editing these parameters. Adjusting these parameters will also change the dynamic jitter buffer. The GXP2124 has a patent dynamic jitter buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms.

We recommend using the default settings provided. We do not recommend adjusting these parameters if you are an average user. Incorrect settings will affect the voice quality.

No Key Entry

Default is 4 seconds. After the timeout, the phone will send out the dialed number.

Timeout Use # as Dial Key

This parameter allows users to configure the “#” key as the “Send” (or “Dial”) key. If set to “Yes”, the “#” key will immediately send the call. In this case, this key is essentially equivalent to the “(Re)Dial” key. If set to “No”, the “#” key is included as part of the dial string.

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G723 Rate G726-32 Packing

Mode iLBC Frame Size iLBC Payload Type Jitter Buffer Type Jitter Buffer Length

Eventlist BLF URI

Conference URI Special Feature

Encoding rate for G723 codec. By default, 6.3kbps rate is set. Select “ITU” or “IETF” for G726-32 packing mode.

iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required. Payload type for iLBC. Default value is 97. The valid range is between 96 and 127. Jitter buffer type: Fixed or Adaptive. Default value is Adaptive. Jitter buffer length. Default value is 300ms. The valid range is between 100ms and

800ms. If the server supports this feature, user needs to configure an "eventlist BLF" URI on

the service side (i.e.: BLF1006@myserver.com) On the GXP2124, under Account page, fill in the ""eventlist BLF" field with the URI

without the domain. (i.e.: BLF1006). Under Basic Settings, please select "eventlist BLF" in the Multi Purpose Key then choose account number, enter username and user id.

Configure the conference URI when using Broadsoft N-way calling feature. Default is Standard. Choose the selection to meet special requirements from Soft

Switch vendors.

Contacts

Under web GUI->Contacts page, users could view and edit the phone's phonebook information.

 Filter by different groups, ie., All Groups, Family, Friends, Work.  Search Phonebook entries.  Add New Contact.  Export Current phonebook XML from phone to local PC.  Import Current phonebook XML from local PC.  Click on the contact number to make a call to it. The call will be dialed out directly from GXP2124.

SAVING THE CONFIGURATION CHANGES

After the user makes a change to the configuration, press the “Update” button in the Configuration Menu. The web browser will then display a message window to confirm saved changes.

We recommend rebooting or powering cycle the IP phone after saving changes.

REBOOTING THE PHONE REMOTELY

Press the “Reboot” button at the bottom of the configuration menu to reboot the phone remotely. The web browser will then display a message window to confirm that reboot is underway. Wait about 30 seconds to log in again.

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Software Upgrade & Customization

Software (or firmware) upgrades are completed via either TFTP or HTTP. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page.

FIRMWARE UPGRADE THROUGH TFTP/HTTP

To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. “Upgrade Server” needs to be set to a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are examples of some valid URLs.

 firmware.mycompany.com:6688/Grandstream/1_0_3_19  72.172.83.110

There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu or Web Configuration Interface.

Key Pad Menu

To configure the Upgrade Server via Key Pad Menu options, select “Config” from the Main Menu, then select “Upgrade”. Under this sub Menu, user can edit Upgrade Server in either an IP address format or FQDN format. Choose “Save and use TFTP” or “Save and use HTTP” to select upgrade method. Select “Reboot” from the Main Menu to reboot the phone.

Web Configuration Interface

To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the GXP2124 IP address. Enter the admin password to access the web configuration interface. In the ADVANCED SETTINGS page, enter the Upgrade Server’s IP address or FQDN in the “Firmware Server Path” field. Select TFTP or HTTP upgrade method. Update the change by clicking the “Update” button. “Reboot” or power cycle the phone to update the new firmware.

During this stage, the LCD will display the firmware file downloading process. Please do NOT disrupt or power down the unit. If a firmware upgrade fails for any reason (e.g., TFTP/HTTP server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the phone will stop the upgrading process and re-boot using the existing firmware/software.

Firmware upgrades take 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/HTTP Server

For users who do not have a local TFTP/HTTP server, we provide a HTTP server on the public Internet for users to download the latest firmware upgrade automatically. Please check the Support/Download section of our website to obtain this HTTP server IP address: http://www.grandstream.com/support/firmware

Alternatively, download and install a free TFTP or HTTP server to the LAN to perform firmware upgrades. A free Windows version TFTP server is available:

http://support.solarwinds.net/updates/New-customerFree.cfm

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INSTRUCTIONS FOR LOCAL TFTP UPGRADE:

1. Unzip the file and put all of them under the root directory of the TFTP server.

2. The PC running the TFTP server and the GXP2124 should be in the same LAN segment.

3. Go to File -> Configure -> Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade.

4. Start the TFTP server, in the phone’s web configuration page

5. Configure the Firmware Server Path with the IP address of the PC

6. Update the change and reboot the unit

User can also choose to download the free HTTP server from http://httpd.apache.org/

or use Microsoft IIS

web server.

NOTE:

 When GXP2124 phone boots up, it will send TFTP or HTTP request to download configuration file

“cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXP2124 phone. This file is for provisioning purpose. For normal TFTP or HTTP firmware upgrades, the following error messages in a TFTP or HTTP server log can be ignored: “TFTP Error from [IP ADRESS] requesting cfg000b82023dd4 : File does not exist.Configuration File Download”

CONFIGURATION FILE DOWNLOAD

The GXP2124 can be configured via Web Interface as well as via Configuration File (binary or XML) through TFTP or HTTP/HTTPS. The “Config Server Path” is the TFTP or HTTP 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 particular field in the web configuration page. A parameter consists of a Capital letter P and 2 to 4 digit numeric numbers, i.e., P2 is associated with “Admin Password” in the ADVANCED SETTINGS page.

For a detailed parameter list, please refer to the link below to download the corresponding configuration template of the firmware.

http://www.grandstream.com/support/tools

Once the GXP2124 boots up (or re-booted), it will request a configuration file named “cfgxxxxxxxxxxxx” followed by a request for configuration XML file named “cfgxxxxxxxxxxxx.xml”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. The configurat ion file name should be in lower cases.

For more details on XML provisioning, please refer to http://www.grandstream.com/support

Managing Firmware and Configuration File Download

When “Automatic Upgrade” is set to “Yes”, a Service Provider can use P193 (Auto Check Interval, in minutes, default and minimum is 60 minutes) to have the devices periodically check for upgrades at prescheduled time intervals. By defining different intervals in P193 for different devices, a Server Provider can manage and reduce the Firmware or Provisioning Server load at any given time.

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Restore Factory Default Setting

WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone.

Please backup or print all the settings before you restoring factory default settings. We are not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.

INSTRUCTIONS FOR RESTORATION:

Step 1: Press “OK” button to bring up the keypad configuration menu, select “Config”, press “OK” to

enter submenu, select “Factory Reset”.

Step 2: A warning window will pop out to make sure a reset is requested and confirmed. Step 3: Press the “OK” confirm and the phone will reboot. Otherwise, it will exit to previous keypad menu

interface.

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Grandstream GXP2124 User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

TABLE OF CONTENTS

TABLE OF FIGURES

TABLE OF TABLES

GUI INTERFACE EXAMPLES

Welcome

Installation

EQUIPMENT PACKAGING

SAFETY COMPLIANCES

WARRANTY

Product Overview

Using the GXP2124 SIP Enterprise Phone

MAKING PHONE CALLS

Handset, Speakerphone and Headset Mode

Multiple SIP Accounts and Lines

Completing Calls

Speed Dial

Making Calls using IP Addresses

Quick IP Call Mode

Setting up the phone to make Quick IP calls

ANSWERING PHONE CALLS

Receiving Calls

Do Not Disturb

Call Waiting/ Call Hold

Mute

Call Transfer

5-Way Conferencing

Voice Messages (Message Waiting Indicator)

Busy Lamp Field

Eventlist BLF

Shared Call Appearance (SCA)

Code Call Features

CUSTOMIZED LCD SCREEN & XML

Configuration Guide

CONFIGURATION VIA KEYPAD

Access the Web Configuration Menu

Definitions

Important Settings

SAVING THE CONFIGURATION CHANGES

REBOOTING THE PHONE REMOTELY

Software Upgrade & Customization

FIRMWARE UPGRADE THROUGH TFTP/HTTP

Key Pad Menu

Web Configuration Interface

No Local TFTP/HTTP Server

CONFIGURATION FILE DOWNLOAD

Managing Firmware and Configuration File Download

Restore Factory Default Setting

INSTRUCTIONS FOR RESTORATION: