Grandstream Networks, Inc.
GXP1100/GXP1105
Small Business IP Phone
GXP1100/GXP1105 USER MANUAL
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CONFIGURATION VIA IVR MENU......................................................................................................22
CONFIGURATION VIA WEB BROWSER ...........................................................................................23
DEFINITIONS ......................................................................................................................................24
STATUS PAGE DEFINITIONS .....................................................................................................24
ACCOUNT PAGE DEFINITIONS .................................................................................................25
SETTINGS/BASIC SETTINGS PAGE..........................................................................................33
SETTINGS/ADVANCED SETTINGS PAGE.................................................................................35
NAT SETTINGS ...................................................................................................................................39
CLICK-TO-DIAL ...................................................................................................................................40
UPGRADE VIA IVR MENU..................................................................................................................42
UPGRAGE VIA WEB GUI....................................................................................................................42
NO LOCAL TFTP/HTTP SERVERS ....................................................................................................43
CONFIGURATION FILE DOWNLOAD................................................................................................44
Table of Tables
GXP1100/GXP1105 User Manual
Table 1: GXP1100/GXP1105 TECHNICAL SPECIFICATIONS.....................................................................8
Table 2: GXP1100/GXP1105 EQUIPMENT PACKAGING ..........................................................................10
Table 3: GXP1100/GXP1105 CONNECTORS ............................................................................................10
Table 4: GXP1100/GXP1105 KEYPAD DEFINITIONS ...............................................................................12
Table 5: CALL FEATURES..........................................................................................................................19
Table 6: GXP1100/GXP1105 IVR MENU ....................................................................................................22
Table of Figures
GXP1100/GXP1105 User Manual
Figure 1: GXP1100/GXP1105 Ports............................................................................................................10
Figure 2: GXP1100/GXP1105 Multi Purpose Key - 3 way Conference ......................................................17
Figure 3: Click-to-Dial..................................................................................................................................40
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GUI Interface Examples
GXP1100/GXP1105 User Manual
1. Screenshot of Configuration Login Page
2. Screenshot of Status Page
3. Screenshot of Basic Setting Configuration Page
4. Screenshot of Advanced User Configuration Page
5. Screenshot of SIP Account Configuration Page
6. Screenshot of Saved Configuration Changes Page
7. Screenshot of Reboot Page
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GNU GPL INFORMATION
GXP1100/GXP1105 firmware contains third-party software licensed under the GNU General Public
License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU
General Public License (GPL) for the exact terms and conditions of the license.
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from:
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CHANGE LOG
This section documents significant changes from previous versions of GXP1100/GXP1105 user manuals.
Only major new features or major document updates are listed here. Minor updates for corrections or
editing are not documented here.
FIRMWARE VERSION 1.0.4.23
•
•
Updated generic config file cfg.xml information. [CONFIGURATION FILE DOWNLOAD]
Added "Use Privacy Header" and "Use P-Preferred-Identity Header" options in web GUI. [ACCOUNT
•
•
Added NAT Settings information. [NAT SETTINGS]
Added Click-to-Dial feature. [CLICK-TO-DIAL]
FIRMWARE VERSION 1.0.4.9
•
•
•
•
•
•
Added instructions for connecting the phone. [CONNECTING YOUR PHONE]
Added Multi Purpose Key options VMsg, Transfer, Intercom. [SETTINGS/BASIC SETTINGS PAGE]
Added IPv6 configuration options. [SETTINGS/BASIC SETTINGS PAGE]
Added Matching Incoming Caller ID function in Account Setting. [ACCOUNT PAGE DEFINITIONS]
Added GNU GPL information. [GNU GPL INFORMATION]
Added Change Log for this user manual. [CHANGE LOG]
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WELCOME
Thank you for purchasing Grandstream GXP1100/GXP1105 Small Business IP Phone.
GXP1100/GXP1105 is a next generation small business IP phone that features up to 2 calls with 1 SIP
account, 4 programmable keys, single network port, integrated PoE (GXP1105 only). The
GXP1100/GXP1105 delivers superior HD audio quality, leading edge telephony features, automated
provisioning for easy deployment, advanced security protection for privacy, and broad interoperability with
most 3rd party SIP devices and leading SIP/NGN/IMS platforms. It is a perfect choice for small business,
lobby, and hotel applications looking for a high quality, basic IP phone with attractive cost.
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 GXP1100 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 here:
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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PRODUCT OVERVIEW
FEATURE HIGHTLIGHTS
•
•
•
•
•
Single SIP Account, up to 2 calls, 4 programmable keys
HD handset with support for wideband audio
Single 10/100Mbps network port, integrated PoE (GXP1105 only)
7 dedicated function keys for Hold, Flash/Call Waiting, Transfer, Message, Mute, Volume, Send/Redial
Automated provisioning using TR-069 or AES encrypted XML configuration file, SRTP and TLS for
advanced security and privacy protection, LLDP, IPv6
GXP1100/GXP1105 TECHNICAL SPECIFICATIONS
Table 1: GXP1100/GXP1105 TECHNICAL SPECIFICATIONS
SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A
record, SRV, NAPTR), DHCP, PPPoE, TELNET, TFTP, NTP, STUN, TR-069,
802.1x, LLDP, IPv6, TLS, SRTP
Protocols
Standards
and
Network Interfaces
Graphic Display
Single 10/100Mbps port, integrated PoE (GXP1105 only)
N/A
4 programmable keys, 7 dedicated function keys for HOLD, FLASH, TRANSFER,
MUTE, VOLUME, SEND/REDIAL and MESSAGE (with LED indicator)
Feature Keys
Voice Codec
Support for G.723.1, G.729A/B, G.711u/a, G.726-32, G.722 (wide-band), iLBC,
in-band and out-of-band DTMF (in audio, RFC2833, SIP INFO)
Hold, transfer, forward, 3-way conference, call waiting, off-hook auto dial,
Telephony Features click-to-dial, flexible dial plan, personalized music ringtones, server redundancy
and fail-over
HD Audio
Yes, HD handset with support for wideband audio
Headset Jack
Base Stand
Wall Mountable
QoS
N/A
Yes, 1 angle position available
Yes
Layer 2 (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
User and administrator level passwords, MD5 and MD5-sess based
authentication, 256-bit AES encrypted configuration file, TLS, SRTP, 802.1x media
access control
Security
Multi-language
English, German, Italian, French, Spanish, Portuguese, Russian, Croatian,
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Simplified Chinese, traditional Chinese, Korean, Japanese, and etc supported in
web configuration interface
Upgrade
and Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or
Provisioning
AES encrypted XML configuration file
Universal power adapter:
Power and Green Input: 100-240VAC 50-60Hz; Output: 5VDC, 800mA
Energy Efficiency
Integrated Power-over-Ethernet (802.3af, GXP1105 only)
Typical power consumption under 1W (power adapter) or under 1.5W (PoE)
Unit dimension: 201mm (W) x 154mm (H) x 78mm (D)
Unit weight: 0.6kg
Physical
Package weight: 1.0kg
Operating
Temperature
Humidity
and 32-104 oF / 0-40 oC, 10-90% (non-condensing)
GXP1100/GXP1105 phone, handset with cord, base stand, universal power supply,
Package Content
network cable, quick start guide
FCC Part 15 (CFR 47) Class B; EN55022 Class B, EN55024, EN61000-3-2,
EN61000-3-3, EN60950-1; AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, RoHS;
UL 60950 (power adapter)
Compliance
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INSTALLATION
EQUIPMENT PACKAGING
Table 2: GXP1100/GXP1105 EQUIPMENT PACKAGING
Main Case
Yes (1)
Yes (1)
Yes (1)
Yes (1)
Yes (1)
Yes (1)
Yes (1)
Handset
Phone Cord
Power Adaptor
Ethernet Cable
Phone Stand
Quick Start Guide
CONNECTING YOUR PHONE
Figure 1: GXP1100/GXP1105 Ports
Table 3: GXP1100/GXP1105 CONNECTORS
Handset Port
LAN Port
RJ9 handset connector port
10/100Mbps RJ-45 port connecting to Ethernet, integrated PoE (GXP1105 only)
5V DC Power connector port
Power Jack
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To set up the GXP1100/GXP1105, follow the steps below:
1. Attach the phone stand to the back of the phone where there is a slot for the phone stand;
2. Connect the handset and main phone case with the phone cord;
3. 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;
4. Connect the 5V DC output plug to the power jack on the phone; plug the power adapter into an
electrical outlet. If PoE switch is used on GXP1105 in step 3, this step could be skipped;
5. The LED on the up right corner will light up in red during the booting up/provisioning/upgrading
process. Before continuing, please wait for the LED turn off;
6. Pick up the handset and the dial tone will be heard. Press *** to use the IVR menu and enter menu
options to hear the corresponding voice prompt. For example, dial 02 in the IVR menu will hear the IP
address. You can further configure the phone using the web GUI by entering GXP1100/GXP1105's IP
address.
SAFETY COMPLIANCES
The GXP1100/GXP1105 phone complies with FCC/CE and various safety standards. The
GXP1100/GXP1105 power adapter is compliant with the UL standard. Use the universal power adapter
provided with the GXP1100/GXP1105 package only. The manufacturer’s warranty does not cover
damages to the phone caused by unsupported power adapters.
WARRANTY
If the GXP1100/GXP1105 phone was purchased from a reseller, please contact the company where the
phone was purchased for replacement, repair or refund. If the phone was purchased directly from
Grandstream, contact the Grandstream Sales and Service Representative for a RMA (Return Materials
Authorization) number before the product is returned. Grandstream reserves the right to remedy warranty
policy without prior notification.
Warning:
Use the power adapter provided with the phone. Do not use a different power adapter as this may damage
the phone. This type of damage is not covered under warranty.
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USING THE GXP1100/GXP1105
GETTING FAMILAR WITH THE KEYPAD
The following table describes the buttons used on the GXP1100/GXP1105 keypad.
Table 4: GXP1100/GXP1105 KEYPAD DEFINITIONS
Hold. Place active call on hold, or resume the call on hold.
Flash. Flash key can be used for multiple purposes.
•
•
Call waiting. Bring up a new line; or answer the second incoming call.
3-way Conference. Establish 3-way conference when FLASH key is configured
as CONF. Before using the Flash key for 3-way conference, "Enable Flash key as
CONF" option has to be set to "Yes" under web GUI->Advanced Settings.
Transfer. Transfer an active call to another number.
Message. Retrieve voicemail messages.
Programmable hard key. It can be configured for multiple purposes: Speed dial,
Dial DTMF, VMsg, Call Return, 3-way Conference, Transfer, Intercom.
Mute. Press to mute/unmute an active call.
Send. It can be used as Send or Redial.
•
•
Send. Enter the digits and then press Send to dial out the number.
Redial. Redial when there is a previously dialed call.
Volume. Press "-" or "+" to adjust the volume.
Standard phone keypad.
0 - 9, *, #
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MAKING PHONE CALLS
2 CALLS WITH 1 SIP ACCOUNT
GXP1100/GXP1105 can support up to two lines “virtually” mapped to one SIP account. By picking up the
handset, the GXP1100/GXP1105 will be in off hook state and the dial tone will be heard. To make a call,
dial out the number with the current line.
During the call, users can press the FLASH key to hold the current call and make/answer another call. If
they are 2 calls established, users can switch the two lines by pressing the FLASH key.
COMPLETING CALLS
The GXP1100/GXP1105 allows you to make phone calls after picking up the handset. There are four ways
to complete calls.
•
Dial. Enter the number and send out.
Take handset off hook. You shall hear dial tone from the handset;
Enter the number;
Press SEND key or # to dial out.
•
•
Redial. Redial the last dialed number.
Take handset off hook. You shall hear dial tone from the handset;
Press SEND key.
Speed Dial. Dial the number configured as Speed Dial on Multi Purpose Key.
Go to GXP1100/GXP1105 Web GUI->Basic Settings, configure the Multi-Purpose Key's Key Mode
as Speed Dial. Enter the Name and User ID (the number to be dialed out) for the Multi-Purpose
Key. Click on "Update" at the bottom of the Web GUI page;
Take handset off hook. You shall hear dial tone from the handset;
Press the configured Speed Dial key.
•
Call Return. Dial the last answered call.
Go to GXP1100/GXP1105 Web GUI->Basic Settings, configure the Multi Purpose Key's Key Mode
as Call Return. No Name or User ID has to be set on the Multi Purpose Key for Call Return;
Take handset off hook. You shall hear dial tone from the handset;
Press the configured Call Return key to dial out.
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Note:
•
After entering the number, the phone waits for the No Key Entry Timeout (Default timeout is 4 seconds,
configurable via Web GUI) before dialing out. Press SEND or # key to override the No Key Entry
Timeout;
•
•
If digits have been entered after handset is off hook, the SEND key will works as SEND instead of
REDIAL;
By default, # can be used as SEND to dial the number out. Users could disable it by setting "User # as
Dial Key" to "No" from Web GUI->Account page.
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 calls
can be made between two phones if:
•
•
•
Both phones have public IP addresses; or
Both phones are on the 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 the steps below:
•
•
•
•
Take handset off hook. You shall hear dial tone from the handset;
Press *** to enter the GXP1100/GXP1105 IVR menu;
Enter 47 for Direct IP Call. After hearing "Direct IP Calling", the dial tone will be heard again;
Enter the target IP address to dial (Please see example below).
For example:
If the target IP address is 192.168.1.60 and the port is 5062 (i.e., 192.168.1.60:5062), input the following:
192*168*1*60#5062. The * key represents the dot (.), the # key represents colon (:). Wait for about 4
seconds and the phone will initiate the call.
Quick IP Call Mode:
The GXP1100/GXP1105 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 address. This is possible only if both
phones are under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP
server. Controlled static IP usage is recommended.
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To enable Quick IP Call Mode, go to GXP1100/GXP1105 Web GUI->Advanced Setting page, set "Use
Quick IP Call Mode" to "Yes". Then take the handset off hook and dial #xxx where x is 0-9 and xxx<255.
Press # or SEND and a direct IP call to aaa.bbb.ccc.XXX will be completed. "aaa.bbb.ccc" is from the local
IP address regardless of subnet mask. The number #xx or #x are also valid. The leading 0 is not required
(but it's OK).
For example:
•
•
•
•
192.168.0.2 calling 192.168.0.3 -- dial #3 followed by # or “SEND”;
192.168.0.2 calling 192.168.0.23 -- dial #23 followed by # “SEND”;
192.168.0.2 calling 192.168.0.123 -- dial #123 followed by # “SEND”;
192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3.
Note:
•
•
The # will represent colon ":" in direct IP call rather than SEND key as in normal phone call;
If you have a SIP server configured, direct IP call still works. If you are using STUN, direct IP call will
also use STUN;
•
Configure the "User Random Port" to "No" when completing direct IP calls.
ANSWERING PHONE CALLS
RECEIVING CALLS
•
•
Single incoming call. Phone rings with selected ring tone. Answer call by taking handset off hook;
Multiple incoming calls. When another call comes in while having an active call, the phone will
produce a Call Waiting tone (stutter tone). Answer the incoming call by pressing the FLASH key. The
current active call will be put on hold.
DURING A PHONE CALL
CALL WAITING/CALL HOLD
•
•
•
Hold. Place a call on hold by pressing the HOLD key;
Resume. Press the HOLD key again to resume;
Multiple calls. Automatically place active call on hold or switch between two calls by pressing the
FLASH key. Call waiting tone (stutter tone) will be audible when the line is in use.
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Note:
If users hang up the current call while there is a call on hold in the other line, there will be an audible ring
tone indicating a call is on hold while your handset is put on hook. Pick up the handset so users can
resume with the call on hold.
MUTE
During an active call, press the MUTE key to mute/unmute the microphone.
CALL TRANSFER
GXP1100/GXP1105 supports Blind Transfer, Attended Transfer and Auto-Attended Transfer.
•
•
Blind Transfer.
During the first active call, press TRAN key and dial the number to transfer to;
Press SEND key or # to complete transfer of active call.
Attended Transfer.
During the first active call, press FLASH key. The first call will be put on hold;
Enter the number for the second call and establish the call;
Press TRAN key;
Press FLASH key to transfer the call.
•
Auto-Attended Transfer.
Set "Auto-Attended Transfer" to "Yes" under Web GUI->Advanced Settings page. And then click
"Update" on the bottom of the page;
Establish one call first;
During the call, press TRAN key. A new line will be brought up and the first call will be
automatically placed on hold;
Enter the number and press SEND key to establish the second call;
After the second call is established, press TRAN key again. The call will be transferred.
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Note:
•
•
To transfer calls across SIP domains, SIP service providers must support transfer across SIP domains.
In auto-attended transfer, use SEND key to dial out the second call instead of using #, even when #
could be used as SEND in normal phone calls.
3-WAY CONFERENCING
GXP1100/GXP1105 can host 3-way conference call by using Multi Purpose Key or FLASH key.
•
To use Multi-Purpose Key to establish 3-way conference call, go to GXP1100/GXP1105 Web
GUI->Settings->Basic Settings, configure the 3-way conference as the Multi Purpose Key mode. Click
"Update" on the bottom of the page. Then follow the steps below for 3-way conferencing.
Figure 2: GXP1100/GXP1105 Multi Purpose Key - 3 way Conference
1. Initiate a conference call.
Establish two active calls with two parties respectively;
Press the Multi Purpose Key previously configured as "3-way Conference" already from Web
GUI;
3-way conference will be established.
2. Split call in conference.
During the 3-way conference, press HOLD key. The conference call will be split and both calls
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will be put on hold separately;
Press HOLD key again and it will resume the 2-way conversation with the line when
establishing the conference call;
Press FLASH key to toggle between the 2 lines;
Users could re-establish conference call by pressing the Multi Purpose Key again.
3. End Conference.
Press HOLD key to split the conference call. The conference call will be ended with both calls
on hold; Or
Users could simply hang up the call to terminate the conference call.
•
To use Flash key to establish 3-way conference call, go to GXP1100/GXP1105 Web
GUI->Settings->Advanced Settings, set “Enable FLASH key as CONF” to “Yes”. Click on "Update" on
the bottom of the Web GUI page and then reboot the phone. Follow the steps below to host the 3-way
conference.
1. Initiate a conference call.
Initiate and establish two active calls with two parties from GXP1100/GXP1105;
Press the FLASH Key;
3-way conference will be established.
2. Split call in conference.
During the 3-way conference, press HOLD key. The conference call will be split and both calls
will be put on hold separately;
Press HOLD key again and it will resume the 2-way conversation with the line when
establishing the conference call;
Users could re-establish conference call by pressing the Multi-Purpose Key again.
3. End Conference.
Press HOLD key to split the conference call. The conference call will be ended with both calls
on hold; Or
Users could simply hang up the call to terminate the conference call.
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
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feature "Transfer on Conference Hangup" is turned on.
•
The option "Disable Conference" has to be set to "No" to establish conference on GXP110x.
VOICE MESSAGES (MESSAGE WAITING INDICATOR)
A blinking red MWI (Message Waiting Indicator) indicates a message is waiting. Dial into the voicemail box
to retrieve the message by entering the voice mail number of the server or pressing the MSG key (Voice
Mail User ID has to be properly configured as the voice mail number under Web GUI->Account page). An
IVR will prompt the user through the process of message retrieval.
Note:
Users can press *** to the IVR menu and then enter 86 to hear the number of new voice messages.
CALL FEATURES
The GXP1100/GXP1105 supports traditional and advanced telephony features including caller ID, caller ID
with caller Name, call forward and etc.
Table 5: CALL FEATURES
Block Caller ID (for all subsequent calls)
*30
*31
*67
•
•
Off hook the phone;
Dial *30.
Send Caller ID (for all subsequent calls)
•
•
Off hook the phone;
Dial *31.
Block Caller ID (per call)
•
•
Off hook the phone;
Dial *67 and then enter the number to dial out.
Send Caller ID (per call)
*82
*70
•
•
Off hook the phone;
Dial *82 and then enter the number to dial out.
Disable Call Waiting (per Call)
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•
•
Off hook the phone;
Dial *70 and then enter the number to dial out.
Enable Call Waiting (per Call)
*71
*72
•
•
Off hook the phone;
Dial *71 and then enter the number to dial out.
Unconditional Call Forward. To set up unconditional call forward:
•
•
•
•
•
Pick up the handset;
Dial *72. A dial tone will be heard;
Enter the forwarding number;
Press # or SEND key;
The call will hang up automatically with unconditional call forward set up.
Cancel Unconditional Call Forward. To cancel the unconditional call forward:
•
•
•
Pick up the handset;
*73
*90
*91
Dial *73. A short tone will be heard;
Wait for the call to hang up. The unconditional call forward is cancelled.
Busy Call Forward. To set up busy call forward:
•
•
•
•
Pick up the handset;
Dial *90 followed by forwarding number;
Press # or SEND key;
The call will hang up automatically with busy call forward set up.
Cancel Busy Call Forward. To cancel the busy call forward:
•
•
•
Pick up the handset;
Dial *91. A short tone will be heard;
Wait for the call to hang up. The busy call forward is cancelled.
Delayed Call Forward. To set up delayed call forward:
•
•
•
•
Pick up the handset;
*92
*93
Dial *92 followed by forwarding number;
Press # or SEND key;
The call will hang up automatically with delayed call forward set up.
Cancel Delayed Call Forward. To cancel the delayed call forward:
Pick up the handset;
•
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•
•
Dial *93. A short tone will be heard;
Wait for the call to hang up. The delayed call forward is cancelled.
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CONFIGURATION GUIDE
The GXP1100/GXP1105 can be configured via two ways:
•
•
IVR Menu using the phone's keypad;
Web GUI embedded on the phone using PC's web browser.
CONFIGURATION VIA IVR MENU
GXP1100/GXP1105 has a built-in voice prompt menu for simple device configuration. Pick up the handset
and dial *** to use the IVR menu.
Table 6: GXP1100/GXP1105 IVR MENU
Menu
Voice Prompt
Options
Press * for the next menu option.
Press # to return to the main menu.
Main Menu "Enter a Menu Option"
Enter 01 – 05, 07, 10 - 17, 47, 86 or 99 for Menu option.
01
"DHCP Mode"
"PPPoE Mode"
"Static IP Mode"
Enter 9 to toggle the selection.
If "Static IP Mode" is selected, users need configure all
the IP address information through menu 02 to 05 as
below.
If "Dynamic IP Mode" is selected, the device will retrieve
all IP address information from DHCP server
automatically after user reboots the device.
02
"IP Address" + IP address
The current WAN IP address is announced.
Enter 12-digit new IP address if in Static IP Mode.
03
04
05
07
"Subnet" + IP address
"Gateway" + IP address
"DNS Server" + IP address
"Preferred Vocoder"
Same as Menu option 02.
Same as Menu option 02.
Same as Menu option 02.
Enter 9 to go to the next selection in the list:
•
•
•
•
PCMU
PCMA
iLBC
G-726
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•
•
G-723
G-729
10
13
"MAC Address"
Announces the MAC address of the unit.
"Firmware Server IP Address"
Announces current Firmware Server IP address. Enter
12 digit new IP address.
14
15
"Configuration
Address"
Server
IP Announces current Config Server Path IP address.
Enter 12 digit new IP address.
"Upgrade Protocol"
Upgrade Protocol for firmware and configuration update.
Enter 9 to toggle between HTTP, TFTP and HTTPS.
16
17
"Firmware Version"
"Firmware Upgrade"
Firmware version information.
Firmware upgrade mode. Enter 9 to toggle among the
following three options:
•
•
•
always check
check when pre/suffix changes
never upgrade
47
"Direct IP Calling"
Enter the target IP address to make a direct IP call, after
dial tone. (See Make a Direct IP Call section)
86
99
"Voice Mail"
"RESET"
Announces number of voice mails.
Enter MAC address to restore factory default setting.
(See Restore Factory Default Setting section)
Press 9 to reboot the device.
Others
"Invalid Entry"
Automatically returns to Main Menu.
CONFIGURATION VIA WEB BROWSER
The GXP1100/GXP1105 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.
To access the GXP1100/GXP1105 Web GUI:
1. Connect the computer to the same network as the phone;
2. Make sure the phone is turned on and wait until the indicator on the top right corner turns from RED to
OFF;
3. Take the handset off hook. Enter *** and then press 02 to hear the IP address;
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4. Open a Web browser on your computer;
5. Enter the phone’s IP address in the address bar of the browser;
6. Enter the administrator’s login and password to access the Web Configuration Menu.
Note:
•
The computer has to be connected to the same sub-network as the phone. This can be easily done by
connecting the computer to the same hub or switch as the phone connected to. In absence of a
hub/switch (or free ports on the hub/switch), please connect the computer directly to the PC port on the
back of the phone;
•
If the phone is properly connected to a working Internet connection, the IP address of the phone can
be obtained from IVR Menu option 02. This address has the format: xxx.xxx.xxx.xxx, where xxx stands
for a number from 0-255. Users will need this number to access the Web GUI. For example, if the
browser;
•
The default login name for the administrator is "admin". The default administrator password is set to
"admin". The default login name for the end user is "user" while the default user password is set to
"123";
•
When changing any settings, always SUBMIT them by pressing the UPDATE button on the bottom of
the page. After submitting the changes in all the Web GUI pages, reboot the phone to have the
changes take effect.
DEFINITIONS
This section describes the options in the GXP1100/GXP1105 Web GUI. As mentioned, you can log in as
an administrator or an end user.
•
•
•
•
Status: Displays the Account status, Network status, and System Info of the phone;
Account: To configure the SIP account;
Basic Settings: To configure basic network settings, time settings, multi-purpose keys, and etc;
Advanced Settings: To configure advanced network settings, upgrading and provisioning, language
settings, call features, and etc.
STATUS PAGE DEFINITIONS
Global unique ID of device, in HEX format. The MAC address will be
used for provisioning and can be found on the label coming with original
box and on the label located on the back of the device.
MAC Address
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IPv4 Address
IPv6 Address
Product Model
Part Number
The IPv4 address obtained on the phone.
The IPv6 address obtained on the phone.
Product model of the phone.
Product part number.
•
•
•
•
boot: boot version number;
core: core version number;
base: base version number;
Software Version
prog: program version number. This is the main firmware release
number, which is always used for identifying the software system of
the phone;
•
dsp: DSP version number.
System Up Time
System Time
Registered
System up time since the last reboot.
Current system time on the phone system.
SIP account registration status.
PPPoE Link Up
Service Status
Core Dump
PPPoE connection status.
GUI and Phone service status: running or stopped.
Core dump file that could be downloaded for troubleshooting purpose.
ACCOUNT PAGE DEFINITIONS
Account Name
The name associated with the SIP account.
The URL or IP address, and port of the SIP server. This is provided by
your VoIP service provider (ITSP).
SIP Server
The URL or IP address, and port of the SIP server. This will be used
when the primary SIP server fails.
Secondary SIP Server
IP address or Domain name of the Primary Outbound Proxy, Media
Gateway, or Session Border Controller. It's used by the phone for
Firewall or NAT penetration in different network environments. If a
symmetric NAT is detected, STUN will not work and ONLY an Outbound
Proxy can provide a solution.
Outbound Proxy
SIP User ID
User account information, provided by your VoIP service provider
(ITSP). It's usually in the form of digits similar to phone number or
actually a phone number.
SIP service subscriber's Authenticate ID used for authentication. It can
be identical to or different from the SIP User ID.
Authenticate ID
Authenticate Password
The account password required for the phone to authenticate with the
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ITSP (SIP) server before the account can be registered. After it is saved,
this will appear as hidden for security purpose.
The SIP server subscriber's name (optional) that will be used for Caller
ID display.
Name
This parameter controls how the Search Appliance looks up IP
addresses for hostnames. There are four modes: A Record, SRV,
NATPTR/SRV, Use Configured IP. The default setting is "A Record". If
the user wishes to locate the server by DNS SRV, the user may select
"SRV" or "NATPTR/SRV". If "Use Configured IP" is selected, please fill
in the three fields below:
DNS Mode
•
Primary IP: The primary IP address where the phone sends DNS
query to;
•
•
Backup IP 1;
Backup IP 2.
If the phone has an assigned PSTN telephone number, this field should
be set to "User=Phone". Then a "User=Phone" parameter will be
attached to the Request-Line and "TO" header in the SIP request to
indicate the E.164 number. If set to "Enable", "Tel:" will be used instead
of "SIP:" in the SIP request. The default setting is "Disable".
Tel URI
Selects whether or not the phone will send SIP Register messages to
the proxy/server. The default setting is "Yes".
SIP Registration
If set to "Yes", the SIP user's registration information will be cleared
when the phone reboots. The SIP Contact header will contain "*" to
notify the server to unbind the connection. The default setting is "No".
Unregister On Reboot
Specifies the frequency (in minutes) in which the phone refreshes its
registration with the specified registrar. The default value is 60 minutes.
The maximum value is 64800 minutes (about 45 days).
Register Expiration
Specifies the time frequency (in seconds) that the phone sends
re-registration request before the Register Expiration. The default value
is 0.
Reregister Before Expiration
Local SIP Port
Defines the local SIP port used to listen and transmit. The default value
is 5060.
SIP Registration Failure Retry Specifies the interval to retry registration if the process is failed. The
Wait Time
default value is 20 seconds.
SIP T1 Timeout
SIP T2 interval
SIP Transport
SIP T1 Timeout. The default setting is 0.5 seconds.
SIP T2 Interval. The default setting is 4 seconds.
Determines the network protocol used for the SIP transport. Users can
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choose from TCP, UDP and TLS.
SIP URI Scheme when using Specifies if "sip:" or "sips:" will be used when TLS/TCP is selected for
TLS
SIP Transport. The default setting is "sips:".
Defines whether the actual ephemeral port in contact with TCP/TLS will
be used or not. This is used when TLS/TCP is selected for SIP Transfer.
The default setting is "No".
Use Actual Ephemeral Port in
Contact with TCP/TLS
Defines whether the domain certificates will be checked or not when
TLS/TCP is used for SIP Transport. The default setting is "No".
Check Domain Certificates
Remove OBP from route
Configures to remove outbound proxy from route. This is used for the
SIP Extension to notify the SIP server that the device is behind a
NAT/Firewall.
Defines whether the incoming messages will be validated or not. The
default setting is "No".
Validate Incoming Messages
Support SIP Instance ID
Defines whether SIP Instance ID is supported or not. The default setting
is "Yes".
This parameter configures whether the NAT traversal mechanism is
activated. Users could select the mechanism from No, STUN,
Keep-Alive, UPnP, Auto or VPN. If set to "STUN" and STUN server is
configured, the phone will route according to the STUN server. If NAT
type is Full Cone, Restricted Cone or Port-Restricted Cone, the phone
will try to use public IP addresses and port number in all the SIP&SDP
messages. The phone will send empty SDP packet to the SIP server
periodically to keep the NAT port open if it is configured to be
"Keep-Alive". Configure this to be "No" if an outbound proxy is used.
"STUN" cannot be used if the detected NAT is symmetric NAT.
NAT Traversal
When set to "Yes", a SUBSCRIBE for Message Waiting Indication will
be sent periodically. The phone supports synchronized and
non-synchronized MWI. The default setting is "No".
SUBSCRIBE for MWI
When set to "Yes", a SUBSCRIBE for Registration will be sent out
periodically. The default setting is "No".
SUBSCRIBE for Registration
Feature Key Synchronization
This feature is used for Broadsoft call feature synchronization. When it's
enabled, DND and Call Forward features can be synchronized with
Broadsoft server. The default setting is "Disabled".
A SIP Extension to notify the SIP server that the phone is behind a
NAT/Firewall. Do not configure this parameter unless this feature is
supported on the SIP server.
Proxy-Require
Voice Mail UserID
Allows you to access voice messages by pressing the MESSAGE button
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on the phone. This ID is usually the VM portal access number. For
example, in Asterisk server, 8500 could be used.
Specifies the mechanism to transmit DTMF digits. There are 3
supported modes: in audio which means DTMF is combined in the audio
signal (not very reliable with low-bit-rate codecs), via RTP (RFC2833), or
via SIP INFO.
Send DTMF
Configures the payload type for DTMF using RFC2833. The default
value is 101.
DTMF Payload Type
Selects whether or not to enable early dial. If it's set to "Yes", the SIP
proxy must support 484 response. The default setting is "No".
Early Dial
Dial Plan Prefix
Sets the prefix added to each dialed number.
A dial plan establishes the expected number and pattern of digits for a
telephone number. This parameter configures the allowed dial plan for
the phone.
Dial Plan Rules:
1. Accepted Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d;
2. Grammar: x - any digit from 0-9;
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
Dial Plan
•
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.
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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 - allows dialing special and emergency numbers 311, 411,
611 and 911.
Note:
In some cases where the user wishes to dial strings such as *123 to
activate voice mail or other applications provided by their service
provider, the * should be predefined inside the dial plan feature. An
example dial plan will be: { *x+ } which allows the user to dial * followed
by any length of numbers.
Delayed Call Forward Wait Defines the timeout (in seconds) before the call is forwarded on no
Time
answer. The default value is 20 seconds.
When enabled, call forward and other call features will be supported
locally provided ITSP support those features. The default setting is
"Yes".
Enable Call Features
Configures Call Log setting on the phone. You can log all calls, only log
incoming/outgoing calls or disable call log. The default setting is "Log All
Calls".
Call Log
The SIP Session Timer extension that enables SIP sessions to be
periodically "refreshed" via a SIP request (UPDATE, or re-INVITE). If
there is no refresh via an UPDATE or re-INVITE message, the session
will be terminated once the session interval expires. Session Expiration
is the time (in seconds) where the session is considered timed out,
provided no successful session refresh transaction occurs beforehand.
The default value is 180 seconds.
Session Expiration
The minimum session expiration (in seconds). The default value is 90
seconds.
Min-SE
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If set to "Yes" and the remote party supports session timers, the phone
will use a session timer when it makes outbound calls.
Caller Request Timer
Callee Request Timer
If set to "Yes" and the remote party supports session timers, the phone
will use a session timer when it receives inbound calls.
If Force Timer is set to "Yes", the phone will use the session timer even if
the remote party does not support this feature. If Force Timer is set to
"No", the phone will enable the session timer only when the remote party
supports this feature. To turn off the session timer, select "No".
Force Timer
As a Caller, select UAC to use the phone as the refresher; or select UAS
to use the Callee or proxy server as the refresher.
UAC Specify Refresher
UAS Specify Refresher
As a Callee, select UAC to use caller or proxy server as the refresher; or
select UAS to use the phone as the refresher.
The Session Timer can be refreshed using the INVITE method or the
UPDATE method. Select "Yes" to use the INVITE method to refresh the
session timer.
Force INVITE
The use of the PRACK (Provisional Acknowledgment) method enables
reliability to SIP provisional responses (1xx series). This is very
important in order to support PSTN internetworking. To invoke a reliable
provisional response, the 100rel tag is appended to the value of the
required header of the initial signaling messages.
Enable 100rel
Allows users to configure the ringtone for the account. Users can choose
from different ringtones from the dropdown menu.
Account Ring Tone
Specifies matching rules with number, pattern or Alert Info text. When
the incoming caller ID or Alert Info matches the rule, the phone will ring
with selected distinctive ringtone. Matching rules:
•
•
Specific caller ID number. For example, 8321123;
A defined pattern with certain length using x and + to specify, where
x could be any digit from 0 to 9. Samples:
xx+ : at least 2-digit number;
xx : only 2-digit number;
Matching Incoming Caller ID
[345]xx: 3-digit number with the leading digit of 3, 4 or 5;
[6-9]xx: 3-digit number with the leading digit from 6 to 9.
Alert Info text
•
Users could configure the matching rule as certain text (e.g., priority)
and select the custom ring tone mapped to it. The custom ring tone
will be used if the phone receives SIP INVITE with Alert-Info header
in the following format:
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Selects the distinctive ring tone for the matching rule. When the
incoming caller ID or Alert Info matches the rule, the phone will ring with
the selected ring.
Distinctive Ringtones
Defines the timeout (in seconds) for the rings on no answer. The default
setting is 60 seconds.
Ring Timeout
If set to "Yes", the "From" header in outgoing INVITE messages will be
set to anonymous, essentially blocking the Caller ID to be displayed.
Send Anonymous
Anonymous Call Rejection
If set to "Yes", anonymous calls will be rejected. The default setting is
"No".
If set to "Yes", the phone will automatically turn on the speaker phone to
Allow Auto Answer by Call-Info answer incoming calls after a short reminding beep, based on the SIP
info header sent from the server/proxy. The default setting is "No".
If set to "Yes", the "Refer-To" header uses the transferred target's
Refer-To Use Target Contact
Contact header information for attended transfer. The default setting is
"No".
Transfer
Hangup
on
Conference Defines whether or not the call is transferred to the other party if the
initiator of the conference hangs up. The default setting is "No".
If set to "Yes", SIP User ID will be checked in the Request URI of the
Check SIP User ID for
incoming INVITE
incoming INVITE. If it doesn't match the phone's SIP User ID, the call will
be rejected. The default setting is "No".
7 different vocoder types are supported on the phone, including G.711
U-law (PCMU), G.711 A-law (PCMA), G.723.1, G.729A/B, G.722 (wide
band), iLBC and G72-32. Users can configure vocoders in a preference
list that is included with the same preference order in SDP message.
Preferred Vocoder
Enables the SRTP mode based on your selection. The default setting is
"Disabled".
SRTP Mode
Defines whether symmetric RTP is supported or not. The default setting
is "No".
Symmetric RTP
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. The default setting is
"No".
Silence Suppression
Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the "ptime" value for the SDP will
change with different configurations here. This value is related to the
codec used and the actual frames transmitted during the in payload call.
Voice Frames Per TX
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For end users, it is recommended to use the default setting, as incorrect
settings may influence the audio quality.
Defines the timeout (in seconds) for no key entry. If no key is pressed
after the timeout, the digits will be sent out. The default value is 4
seconds.
No Key Entry Timeout (s)
Use # as Dial Key
Allows users to configure the "#" key as the "Send" key. If set to "Yes",
the "#" key will immediately dial out the input digits. In this case, this key
is essentially equivalent to the "Send" key. If set to "No", the "#" key is
included as part of the dialing string.
G723 Rate
Selects encoding rate for G723 codec. The default value is 5.3kbps.
Select "ITU" or "IETF" for G726-32 packing mode.
G.726-32 Packing Mode
iLBC Frame Size
Selects iLBC packet frame size. The default value is 30ms.
Specifies iLBC Payload type. The default value is 97. The valid range is
between 96 and 127.
iLBC Payload Type
Jitter Buffer Type
Jitter Buffer Length
Selects either Fixed or Adaptive based on network conditions. The
default setting is "Adaptive".
Selects Low, Medium, or High based on network conditions. The default
setting is "Medium".
Controls whether the Privacy Header will present in the SIP INVITE
message or not. The default setting is "default", which is when "Huawei
IMS" special feature is on, the Privacy Header will not show in INVITE. If
set to "Yes", the Privacy Header will always show in INVITE. If set to
"No", the Privacy Header will not show in INVITE.
Use Privacy Header
Controls whether the P-Preferred-Identity Header will present in the SIP
INVITE message or not. The default setting is "default", which is when
Use
P-Preferred-Identity "Huawei IMS" special feature is on, the P-Preferred-Identity Header will
not show in INVITE. If set to "Yes", the P-Preferred-Identity Header will
always show in INVITE. If set to "No", the P-Preferred-Identity Header
will not show in INVITE.
Header
Different soft switch vendors have special requirements. Therefore users
may need select special features to meet these requirements. Users can
Special Feature
choose from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro
or Huawei IMS depending on the server type. The default setting is
"Standard".
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SETTINGS/BASIC SETTINGS PAGE
Allows the administrator to set the password for user-level web GUI
access. This field is case sensitive with a maximum length of 30
characters.
End User Password
Confirm Password
Internet Protocol
Confirms the end user password field to be the same as above.
Selects Prefer IPv4 or Prefer IPv6.
Allows users to configure the appropriate network settings on the phone
to obtain IPv4 address. Users could select "DHCP", "Static IP" or
"PPPoE". By default, it is set to "DHCP".
IPv4 Address Type
Specifies the name of the client. This field is optional but may be
required by some Internet Service Providers.
DHCP Host name (Option 12)
DHCP Vendor Class ID
(Option 60)
Used by clients and servers to exchange vendor class ID.
Allow DHCP Option 120 to Enables DHCP Option 120 from local server to override the SIP Server
override SIP Server
PPPoE Account ID
PPPoE Password
PPPoE Service Name
IPv4 Address
on the phone. The default setting is "No".
Enter the PPPoE account ID.
Enter the PPPoE Password.
Enter the PPPoE Service Name.
Enter the IP address when static IP is used.
Enter the Subnet Mask when static IP is used.
Enter the Default Gateway when static IP is used.
Enter the DNS Server 1 when static IP is used.
Enter the DNS Server 2 when static IP is used.
Enter the Preferred DNS Server.
Subnet Mask
Gateway
DNS Server 1
DNS Server 2
Preferred DNS Server
Allows users to configure the appropriate network settings on the phone
to obtain IPv6 address. Users could select "Auto-configured" or
"Statically configured".
IPv6 Address Type
Enter the static IPv6 address when Full Static is used in "Statically
configured" IPv6 address type.
Static IPv6 Address
IPv6 Prefix Length
Enter the IPv6 prefix length when Full Static is used in "Statically
configured" IPv6 address type.
Enter the IPv6 Prefix (64 bits) when Prefix Static is used in "Statically
configured" IPv6 address type.
IPv6 Prefix
DNS Server 1
Enter the DNS Server 1 for IPv6.
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DNS Server 2
Enter the DNS Server 2 for IPv6.
Preferred DNS server
Enter the Preferred DNS Server for IPv6.
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.
802.1x mode
Identity
Enter the Identity for the 802.1x mode.
MD5 Password
Enter the MD5 Password for the 802.1x mode.
Specifies the HTTP proxy URL for the phone to send packets to. The
proxy server will act as an intermediary to route the packets to the
destination.
HTTP Proxy
Specifies the HTTPS proxy URL for the phone to send packets to. The
proxy server will act as an intermediary to route the packets to the
destination.
HTTPS Proxy
Assigns a function to the corresponding multi-purpose key. The key
mode options are:
•
Speed Dial
Enter the Speed Dial number in UserID field to be dialed.
•
Dial DTMF
Enter a series of DTMF digits in UserID field to be dialed during the
call. "Enable MPK Sending DTMF" (under Advanced Setting) has to
be set to "Yes" first.
•
•
VMsg
Enter the Voice Mail access number in the UserID field.
Multi Purpose Key X
(X: 1 - 4)
Call Return
The last answered calls can be dialed out by using Call Return. The
Name and UserID should be left blank.
•
Transfer
Enter the number in the UserID field to be transferred (blind transfer)
during the call.
•
•
Intercom
Enter the extension number in the UserID field to do the intercom.
3-way Conference
Press to establish 3-way conference. The Name and UserID should
be left blank.
Configures the date/time on the phone according to the specified time
zone.
Time Zone
This parameter allows the users to define their own time zone.
Self-Defined Time Zone
The syntax is: std offset dst [offset], start [/time], end [/time]
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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,
3rd Tuesday…)
The 3rd number indicates weekday: 0,1,2,..,6( for Sun, Mon, Tues,..,Sat)
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.
SETTINGS/ADVANCED SETTINGS PAGE
Allows users to change the admin password. The password field is
purposely hidden after clicking the Update button for security purpose.
This field is case sensitive with a maximum length of 30 characters.
Admin Password
Confirm Password
Layer 3 QoS
Confirms the admin password field to be the same as above.
Defines the Layer 3 QoS parameter. This value is used for IP
Precedence, Diff-Serv or MPLS. The default value is 12.
Layer 2 QoS 802.1Q/VLAN Assigns the VLAN Tag of the Layer 2 QoS packets. The default value is
Tag 0.
Layer 2 QoS 802.1p Priority Assigns the priority value of the Layer2 QoS packets. The default value
Value
is 0.
This parameter defines the local RTP port used to listen and transmit. It
is the base RTP port for channel 0. When configured, channel 0 will use
this port _value for RTP; channel 1 will use port_value+2 for RTP. Local
RTP port ranges from 1024 to 65400 and must be even. The default
value is 5004.
Local RTP Port
When set to "Yes", this parameter will force random generation of both
the local SIP and RTP ports. This is usually necessary when multiple
phones are behind the same full cone NAT. The default setting is "Yes"
(This parameter must be set to "No" for Direct IP Calling to work).
Use Random Port
Specifies how often the phone sends a blank UDP packet to the SIP
server in order to keep the "ping hole" on the NAT router to open. The
Keep-alive Interval
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default setting is 20 seconds.
The NAT IP address used in SIP/SDP messages. This field is blank at
the default settings. It should ONLY be used if it's required by your ITSP.
Use NAT IP
The IP address or Domain name of the STUN server. STUN resolution
results are displayed in the STATUS page of the Web GUI. Only
non-symmetric NAT routers work with STUN.
STUN Server
Specifies how firmware upgrading and provisioning request to be sent:
Always Check for New Firmware, Check New Firmware only when F/W
pre/suffix changes, Always Skip the Firmware Check.
Firmware
Upgrade
and
Provisioning
The password for encrypting the XML configuration file using OpenSSL.
This is required for the phone to decrypt the encrypted XML
configuration file.
XML Config File Password
HTTP/HTTPS User Name
HTTP/HTTPS Password
The user name for the HTTP/HTTPS server.
The password for the HTTP/HTTPS server.
Allows users to choose the firmware upgrade method: TFTP, HTTP or
HTTPS.
Upgrade Via
Defines the server path for the firmware server. It could be different from
the configuration server for provisioning.
Firmware Server Path
Config Server Path
Defines the server path for provisioning. It could be different from the
firmware server for upgrading.
Enables your ITSP to lock firmware updates. If configured, only the
firmware with the matching encrypted prefix will be downloaded and
flashed into the phone.
Firmware File Prefix
Firmware File Postfix
Config File Prefix
Enables your ITSP to lock firmware updates. If configured, only the
firmware with the matching encrypted postfix will be downloaded and
flashed into the phone.
Enables your ITSP to lock configuration updates. If configured, only the
configuration file with the matching encrypted prefix will be downloaded
and flashed into the phone.
Enables your ITSP to lock configuration updates. If configured, only the
configuration file with the matching encrypted postfix will be downloaded
and flashed into the phone.
Config File Postfix
Allow DHCP Option 43 and If DHCP option 66 is enabled on the LAN side, the TFTP server can be
Option 66 Override Server
Automatic Upgrade
redirected. The default setting is "Yes".
Enables automatic upgrade and provisioning. The default setting is "No".
Authenticates configuration file before acceptance. The default setting is
"No".
Authenticate Conf File
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Enable TR-069
ACS URL
Enables TR-069. The default setting is "No".
URL for TR-069 Auto Configuration Servers (ACS).
ACS username for TR-069.
TR-069 Username
TR-069 Password
ACS password for TR-069.
Enables periodic inform. If set to "Yes", device will send inform packets
to the ACS. The default setting is "No".
Periodic Inform Enable
Periodic Inform Interval
Sets up the periodic inform interval to send the inform packets to the
ACS.
Connection
Username
Request
The user name for the ACS to connect to the phone.
Connection Request Password The password for the ACS to connect to the phone.
Connection Request Port
CPE SSL Certificate
The port for the ACS to connect to the phone.
The Certificate File for the phone to connect to the ACS via SSL.
The Private Key for the phone to connect to the ACS via SSL.
CPE SSL Private Key
Configures a User ID/extension to dial automatically when the phone is
off hook. The phone will use the first account to dial out. The default
setting is "No".
Offhook Auto Dial
Configures whether auto recover or not when the phone is running
abnormal. The default setting is "Yes".
Auto Recover From Abnormal
Syslog Server
The URL/IP address for the syslog server.
Selects the level of logging for syslog. The default setting is None. There
are 4 levels: DEBUG, INFO, WARNING AND ERROR.
Syslog messages are sent based on the following events:
•
•
•
•
•
•
•
•
•
•
product model/version on boot up (INFO level);
NAT related info (INFO level);
sent or received SIP message (DEBUG level);
SIP message summary (INFO level);
inbound and outbound calls (INFO level);
registration status change (INFO level);
negotiated codec (INFO level);
Syslog Level
ethernet link up (INFO level);
SLIC chip exception (WARNING and ERROR levels);
memory exception (ERROR level).
Configures whether the SIP log will be included in the Syslog messages
or not. The default setting is "No".
Send SIP Log
NTP Server
Defines the URL or IP address of the NTP server. The phone may obtain
the date and time from the server.
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Defines whether DHCP Option 42 should override NTP server or not.
When enabled, DHCP Option 42 will override the NTP server if it's set
up on the LAN. The default setting is "Yes".
Allow
DHCP
Option
42
Override NTP Server
SSL Certificate
SSL Certificate used for SIP Transport in TLS/TCP.
SSL Private key used for SIP Transport in TLS/TCP.
SSL Private key password used for SIP Transport in TLS/TCP.
SSL Private Key
SSL Private Key Password
System ring tone. Default is North American standard. Users could
adjust system ring tone frequencies and cadences based on local
telecom standard.
System Ring Tone
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.
Call Progresses Tones:
Dial Tone
Frequencies should be configured with known values to avoid
uncomfortable high pitch sounds.
Message Waiting
Ring Back Tone
Call-Waiting Tone
Busy Tone
Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]];
(Frequencies are in Hz and cadence on and off are in 10ms)
ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of
silence. 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.
Up to three cadences are supported.
Reorder Tone
Disable Call-Waiting
Disables the call waiting feature. The default setting is "No".
Disables the call waiting tone when call waiting is on. The default setting
is "No".
Disable Call-Waiting Tone
Disable Direct IP Calls
Disables Direct IP Call. The default setting is "No".
When set to "Yes", users can dial an IP address under the same
LAN/VPN segment by entering the last octet in the IP address. To dial
quick IP call, 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). No
SIP server is required to make quick IP call. The default setting is "No".
Use Quick IP-Call mode
Disable Conference
Disables the Conference function. The default setting is "No".
Enables Multi Purpose Key to send DTMF during the call. The default
setting is "No".
Enable MPK sending DTMF
Enable FLASH key as CONF
If set to "Yes", FLASH key can be used to establish 3-way conference.
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The default setting is "No".
Disable Transfer
Disables the Transfer function. The default setting is "No".
If set to "Yes", the phone will use attended transfer by default. The
default setting is "No".
Auto-Attended Transfer
In-call dial number on pressing If configured, the phone will use the TRAN key to dial the number as
transfer key
DTMF during the call.
If configured, when the phone is onhook, it will go offhook after the
timeout (in seconds). The default value is 30 seconds.
Offhook timeout
Do Not Escape # as %23 in Specifies whether to replace # by %23 or not for some special situations.
SIP URI
The default setting is "No".
Disable Telnet
Display Language
Disables Telnet access. The default setting is "No".
Selects display language on the phone.
Download
Device
Click to download the device config file in .txt format.
Configuration
NAT SETTINGS
If the devices are kept within a private network behind a firewall, we recommend using STUN Server. The
following settings are useful in the STUN Server scenario:
•
STUN Server (under 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.
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CLICK-TO-DIAL
From GXP1100/GXP1105 Web GUI, users could dial out with Click-to-Dial feature
on the top menu
of the Web GUI when the account is registered. After clicking on the
icon, a new dialing window will
show as the figure below. Enter number and click on "Dial", the phone will go off hook and dial out the
number from account 1.
Figure 3: Click-to-Dial
Additionally, users could directly send the command for the phone to dial out by specifying the following
URL in PC's web browser, or in the field as required in other call modules.
http://ip_address/cgi-bin/api-make_call?phonenumber=1234&account=0&password=admin
In the above link, replace the fields with
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•
•
•
ip_address:
Phone's IP Address.
phonenumber=1234:
The number for the phone to dial out
account=0:
The account index for the phone to make call. The index is 0 for account 1, 1 for account 2, 2 for
account 3, and etc.
•
password=admin:
The admin login password of phone's Web GUI.
SAVING THE CONFIGURATION CHANGES
After users makes changes to the configuration, press the Update button on the bottom of the Web GUI
page. We recommend rebooting or powering cycle the IP phone after saving changes.
REBOOTING FROM REMOTE LOCATIONS
Press the Reboot button on the bottom of the web GUI page to reboot the phone remotely. The web
browser will then display a reboot page with message "The device is rebooting now...". Wait for about 1
minute to log in again.
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UPGRADING AND PROVISIONING
The GXP1100/GXP1105 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for
the TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP or
HTTP; the server name can be FQDN or IP address.
Examples of valid URLs:
firmware.grandstream.com
fw.ipvideotalk.com/gs
There are two ways to setup a software upgrade server: The IVR Menu or the Web Configuration Interface.
UPGRADE VIA IVR MENU
Follow the steps below to configure the Upgrade Server IP address via IVR:
•
•
Pick up the handset, press *** to access the IVR Menu;
Input menu option 15 for "Upgrading Protocol". Then press 9 to toggle between different upgrading
methods;
•
•
Press # to return to the main menu and input menu option 13 for "Firmware Server IP Address";
Input the 12-digit firmware upgrade IP address. For example, if the firmware upgrade IP address is
10.0.50.191, input 010000050191.
Then reboot the phone. The LED indicator on the top right corner will turn orange and red and then turn off
which indicates the phone has restarted. After a while the indicator will blink in red meaning the download
is in process. When upgrading is done you will see the phone restarts again. Please do not interrupt or
power cycle the phone when the upgrading process is on.
UPGRAGE VIA WEB GUI
Open a web browser on PC and enter the IP address for the GXP1100/GXP1105. Then, login with the
administrator username and password. Go to Settings->Advanced Settings page, enter the IP address or
the FQDN for the upgrade server in "Firmware Server Path" field and choose to upgrade via TFTP or
HTTP/HTTPS. Update the change by clicking the "Update" button. Then "Reboot" or power cycle the
phone to update the new firmware.
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The indicator on the top right corner will turn orange and red and then turn off which indicates the phone
has restarted. After a while the indicator will blink in red meaning the download is in process. When
download is done you will see the phone restarts again. 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
reboot 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 SERVERS
For users that would like to use remote upgrading without a local TFTP/HTTP server, Grandstream offers a
NAT-friendly HTTP server. This enables users to download the latest software upgrades for their phone via
this server. Please refer to the webpage:
Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade. A
free windows version TFTP server is available for download from :
Instructions for local firmware upgrade via TFTP:
1. Unzip the firmware files and put all of them in the root directory of the TFTP server;
2. Connect the PC running the TFTP server and the phone to the same LAN segment;
3. Launch the TFTP server and go to the File menu->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 and configure the TFTP server in the phone’s web configuration interface;
5. Configure the Firmware Server Path to the IP address of the PC;
6. Update the changes and reboot the phone.
Microsoft IIS web server.
Note:
When the phone boots up, it will send a TFTP or HTTP request to download the configuration file
"cfgxxxxxxxxxxxx" where "xxxxxxxxxxxx" is the MAC address of the phone. If it is a normal TFTP or HTTP
upgrade, the following messages “TFTP Error from [IP ADRESS] requesting cfg000b82023dd4: File does
not exist. Configuration File Download” can be ignored in the TFTP/HTTP server log.
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CONFIGURATION FILE DOWNLOAD
Grandstream SIP Devices can be configured via the Web Interface as well as via a Configuration File
(binary or XML) through TFTP or HTTP/HTTPS. The "Config Server Path" is the TFTP or HTTP/HTTPS
server path for the configuration file. It needs to be set to a valid URL, either in FQDN or IP address format.
The "Config Server Path" can be the same or different from the "Firmware Server Path".
A configuration parameter is associated with each particular field in the web configuration page. A
parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with the “Admin Password” in the Web GUI->Settings->Advanced Settings.
For a detailed parameter list, please refer to the corresponding firmware release configuration template.
When the GXP1100/GXP1105 boots up or reboots, it will issue a request to download a configuration XML
file named "cfgxxxxxxxxxxxx.xml" followed by a file named "cfgxxxxxxxxxxxx", where "xxxxxxxxxxxx" is
the MAC address of the phone, i.e., "cfg000b820102ab.xml" and "cfg000b820102ab". If the download of
"cfgxxxxxxxxxxxx.xml" file is not successful, the provision program will download a generic cfg.xml file. The
configuration file name should be in lower case letters.
For more details on XML provisioning, please refer to:
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RESTORE FACTORY DEFAULT SETTINGS
Warning:
Restoring the Factory Default Settings will delete all configuration information on the phone. Please
backup or print all the settings before you restore to the factory default settings. Grandstream is not
responsible for restoring lost parameters and cannot connect your device to your VoIP service provider.
Please follow the instructions below to reset the phone:
Pick up the handset, press *** to access the IVR menu. Enter 99 for factory reset. Then enter the MAC
address printed on the bottom of the sticker. Please use the following mapping:
0-9:
A:
0-9
22 (press the “2” key twice, “A” will show on the LCD)
B:
222
C:
D:
E:
2222
33 (press the “3” key twice, “D” will show on the LCD)
333
F:
3333
Example: if the MAC address is 000b8200e395, it should be key in as “0002228200333395”.
Note:
•
If there are digits like "22" in the MAC, you need to wait for 4 seconds to continue to key in another "2";
•
Once the MAC address is correctly input, the phone will reboot. Otherwise, it will announce “Invalid
Entry” and exit to the main menu.
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EXPERIENCING THE GXP1100/GXP1105
Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware
releases, additional features, FAQs, documentation and news on new products.
We encourage you to browse our product related documentation, FAQs and User and Developer Forum
for answers to your general questions. If you have purchased our products through a Grandstream
Certified Partner or Reseller, please contact them directly for immediate support.
Our technical support staff is trained and ready to answer all of your questions. Contact a technical support
member or submit a trouble ticket online to receive in-depth support.
Thank you again for purchasing Grandstream IP phone, it will be sure to bring convenience and color to
both your business and personal life.
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