Grandstream GXP1105 Manual
Grandstream
Telefon
GXP1105
Læs gratis den danske manual til Grandstream GXP1105 (50 sider) i kategorien Telefon. Denne vejledning er vurderet som hjælpsom af 64 personer og har en gennemsnitlig bedømmelse på 5.0 stjerner ud af 32.5 anmeldelser. Har du et spørgsmål om Grandstream GXP1105, eller vil du spørge andre brugere om produktet?
Side 1/50
GXP11 /GXP1105 USER MANUAL 00
Grandstream Networks, Inc.
GXP1100/GXP1105
Small Business IP Phone
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page 2 of 26 50
GXP11 /GXP1105 User Manual 00
Index
GUI INTERFACE EXAMPLES .................................................................... 5
GNU GPL INFORMATION .......................................................................... 6
CHANGE LOG ........................................................................................... 7
FIRMWARE VERSION 1.0.5.26 ............................................................................................................ 7
FIRMWARE VERSION 1.0.5.15 ............................................................................................................ 7
FIRMWARE VERSION 1.0.4.23 ............................................................................................................ 7
FIRMWARE VERSION 1.0.4.9 .............................................................................................................. 7
WELCOME ................................................................................................. 9
PRODUCT OVERVIEW ............................................................................ 10
FEATURE HIGHTLIGHTS ................................................................................................................... 10
GXP1100/GXP1105 TECHNICAL SPECIFICATIONS ......................................................................... 10
INSTALLATION ........................................................................................ 12
EQUIPMENT PACKAGING ................................................................................................................. 12
CONNECTING YOUR PHONE ........................................................................................................... 12
SAFETY COMPLIANCES .................................................................................................................... 13
WARRANTY ......................................................................................................................................... 14
USING THE GXP1100/GXP1105 .............................................................. 15
GETTING FAMILAR WITH THE KEYPAD ........................................................................................... 15
MAKING PHONE CALLS ..................................................................................................................... 16
2 CALLS WITH 1 SIP ACCOUNT ................................................................................................. 16
COMPLETING CALLS .................................................................................................................. 16
MAKING CALLS USING IP ADDRESSES ................................................................................... 17
ANSWERING PHONE CALLS ............................................................................................................ 18
RECEIVING CALLS ...................................................................................................................... 18
DURING A PHONE CALL .................................................................................................................... 18
CALL WAITING/CALL HOLD ....................................................................................................... 18
MUTE ............................................................................................................................................ 19
CALL TRANSFER ........................................................................................................................ 19
3-WAY CONFERENCING ............................................................................................................ 20
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page 3 of 26 50
VOICE MESSAGES (MESSAGE WAITING INDICATOR) ........................................................... 22
CALL FEATURES ................................................................................................................................ 22
CONFIGURATION GUIDE ........................................................................ 25
CONFIGURATION VIA IVR MENU ...................................................................................................... 25
CONFIGURATION VIA WEB BROWSER ........................................................................................... 26
DEFINITIONS ...................................................................................................................................... 27
STATUS PAGE DEFINITIONS ..................................................................................................... 28
ACCOUNT PAGE DEFINITIONS ................................................................................................. 29
SETTINGS PAGE DEFINITIONS ................................................................................................. 36
NETWORK PAGE DEFINITIONS ................................................................................................. 39
MAINTENANCE PAGE DEFINITIONS ......................................................................................... 40
NAT SETTINGS ................................................................................................................................... 43
CLICK- -DIAL TO ................................................................................................................................... 43
UPGRADING AND PROVISIONING ........................................................ 46
UPGRADE VIA IVR MENU .................................................................................................................. 46
UPGRAGE VIA WEB GUI .................................................................................................................... 46
NO LOCAL TFTP/HTTP SERVERS .................................................................................................... 47
CONFIGURATION FILE DOWNLOAD ................................................................................................ 48
RESTORE FACTORY DEFAULT SETTINGS ........................................... 49
EXPERIENCING THE GXP1100/GXP1105 .............................................. 50
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page 4 of 26 50
Table of Tables
GXP11 /GXP1105 User Manual 00
Table 1: GXP1100/GXP1105 TECHNICAL SPECIFICATIONS ................................................................... 10
Table 2: GXP1100/GXP1105 EQUIPMENT PACKAGING .......................................................................... 12
Table 3: GXP1100/GXP1105 CONNECTORS ............................................................................................ 12
Table 4: GXP1100/GXP1105 KEYPAD DEFINITIONS ............................................................................... 15
Table 5 CALL FEATURES : .......................................................................................................................... 22
Table 6: GXP1100/GXP1105 IVR MENU .................................................................................................... 25
Table of Figures
GXP11 /GXP1105 User Manual 00
Figure 1: GXP1100/GXP1105 Ports ............................................................................................................ 12
Figure 2: GXP1100/GXP1105 Pin-out ......................................................................................................... 13
Figure 3: GXP1100/GXP1105 Multi Purpose Key - 3 way Conference ...................................................... 20
Figure 4: Click- -Dial to .................................................................................................................................. 44
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page 6 of 26 50
GNU GPL INFORMATION
GXP1100/GXP1105 firmware contain third-party software licensed under the GNU General Public s
License (GPL). Grandstream uses software under the ecific terms of the GPL. Please see the GNU sp
General Public License (GPL) for the exact terms and conditions of the license.
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from:
http://www.grandstream.com/support/faq/gnu_gpl.
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page 7 of 26 50
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 1.0.5.26
Updated TFTP server download link for 1.0.5.26 NO LOCAL HTTP/TFTP SERVERS] , [
Added Blink message LED on ringing feature in Call Features. [CALL FEATURES] “ ”
FIRMWARE VERSION 1.0.5.15
Updated Web GUI interface examples with new screenshots for 1.0.5.15. [GUI INTERFACE
EXAMPLES]
Added pin-out information. [CONNECTING YOUR PHONE]
Updated Auto Attended Transfer information. [CALL TRANSFER]
Updated Click- -Dial feature information. [CLICK- -DIAL] To TO
Updated Web GUI options. [DEFINITIONS]
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
PAGE DEFINITIONS]
Added NAT Settings information. [NAT SETTINGS]
Added Click- -Dial feature. [CLICK- -DIAL] to TO
FIRMWARE VERSION 1.0.4.9
Added instructions for connecting the phone. [CONNECTING YOUR PHONE]
Added Multi Purpose Key options VMsg, Transfer, Intercom. [SETTINGS PAGE]
Added IPv6 configuration options. [SETTINGS PAGE]
Added Matching Incoming Caller ID function in Account Setting. [ACCOUNT PAGE DEFINITIONS]
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page 8 of 26 50
Added GNU GPL information GNU GPL INFORMATION] . [
Added Change Log for this user manual CHANGE LOG] . [
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 11 50
Simplified Chinese, traditional Chinese, Korean, Japanese, and etc supported in
web configuration interface
Upgrade and
Provisioning
Firmware upgrade via TFTP/HTTP/HTTPS, mass provisioning using TR-069 or
AES encrypted XML configuration file
Power and Green
Energy Efficiency
Universal power adapter:
Input: 100-240VAC 50-60Hz; tput: 5VDC, 800mA Ou
Integrated Power-over-Ethernet (802.3af, GXP1105 only)
Typical power consumption under 1W (power adapter) or under 1.5W (PoE)
Physical
Unit dimension: 201mm (W) x 154mm (H) x 78mm (D)
Unit weight: 0.6kg
Package weight: 1.0 kg
Operating
Temperature and
Humidity
32-104 oF / 0- 40 oC, 10-90% (non-condensing)
Package Content
GXP11 /GXP1105 phone, handset with cord, base stand, universal power supply, 00
network cable, quick start guide
Compliance
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)
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 12 50
INSTALLATION
EQUIPMENT PACKAGING
Table 2: GXP1100/GXP1105 EQUIPMENT PACKAGING
Main Case
Yes
1
Handset
Yes
1
Phone Cord
Yes
1
Power Adaptor
Yes
1
Ethernet Cable
Yes
1
Phone Stand
Yes
1
Quick Start Guide
Yes
1
CONNECTING YOUR PHONE
Figure 1: GXP1100/GXP1105 Ports
Table 3: GXP1100/GXP1105 CONNECTORS
Handset Port
RJ9 handset connector port
LAN Port
10/100Mbps RJ-45 port connecting to Ethernet, integrated PoE (GXP1105 only)
Power Jack
5V DC Power connector port
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 13 50
To set up the GXP /GXP1105, follow the steps below: 1100
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 LE on the up right corner will light up in red during the booting up/provisioning/upgrading D
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.
Please see below the pin-out information for GXP1100/GXP1105.
Figure 2: GXP1100/GXP1105 - Pin out
SAFETY COMPLIA NCES
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 pow adapter er
provided with the GXP11 /GXP1105 package only00 . The manufacturer’s warranty does not cover
damages to the phone caused by unsupported power adapters.
GXP1100/GXP1105 Power Jack
GXP1100/GXP1105 Handset Jack
GXP1100/GXP1105 Handset Plug
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 14 50
WARRANTY
If GXP1100/GXP1105 phone was purchased from a reseller, please contact the company where the the
phone was purchased for replacement, repair or refund. If the phone was purchased directly from
Grandstream, contact the Grandstream Support for a RMA (Return Materials Authorization) number before
the product is return . Grandstream reserves the right to remedy warranty policy without prior notification. ed
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.
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 15 50
USING THE GXP /GXP1 1100 105
GETTING FAMILAR WITH THE KEYPAD
T following table describes the buttons used on the GXP1100/GXP1105 keypad. he
Table 4 GXP1100/GXP1105 KEYPAD DEFINITIONS :
Hold. Place active call on hold, or resume the call on ho . ld
Flash. Flash key can be used for multiple purposes.
Call waiting. Bring up a new line; answer the second incoming call. or
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. /uPress to mute nmute 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 "- "+" to adjust the volume. " or
0 - 9, *, #
Standard phone keypad.
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 16 50
MAKING PHONE CALLS
2 CALLS WITH 1 SIP ACCOUNT
GXP1100/GXP1105 can support up to two lines “virtually” mapped to one SIP accoun By picking up the t.
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.
Enter the number and send out. Dial.
Take handset off hook. You shall hear dial tone from the handset;
Enter the number;
Press SEND key or # to dial out.
Redial the last dialed number. Redial.
Take handset off hook. You shall hear dial tone from the handset;
Press SEND key.
Dial the number configured as Speed Dial on Multi Purpose Key. Speed Dial.
Go to GXP1100/GXP1105 -> -> , configure the Web GUI Settings Programmable Keys
Multi-Purpose Key's Key Mode as Speed Dial. Enter the Description and Value (the number to be
dialed out) for the Multi-Purpose Key. Click on "Save and Apply" 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.
Dial the last answered call. Call Return.
Go to GXP1100/GXP1105 -> -> , configure the Multi Web GUI Settings Programmable Keys
Purpose Key's Key Mode as Call Return. No Value 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.
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 18 50
phones are under the same LAN/VPN. This simulates a PBX function using the CSMA/CD without a SIP
server. Controlled static IP usage is recommended.
To enable Quick IP Call Mode, go to GXP1100/GXP1105 Web GUI->Settings->Call Features, 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 follow -- ed 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 follow by # "SEND"; -- ed
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
. Phone rings with selected ring tone. Answer call by taking handset off hook;Single incoming call
. When another call comes in while having an active call, the phone will Multiple incoming calls
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
. Place a call on hold by pressing the HOLD key;Hold
. Press the HOLD key again to resume;Resume
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 19 50
. Automatically place active call on hold or switch between two calls by pressing the Multiple calls
FLASH key. Call waiting tone (stutter tone) will be audible when the line is in use.
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 handset so users can the
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 numb for the second call and establish the call;er
Press TRAN key;
Press FLASH key to transfer the call.
Auto-Attended Transfer .
Set "Auto-Attended Transfer" to "Yes" under Web GUI->Settings->Call Features. And then click
"Save and Apply" 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;
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 20 50
Enter the number and press SEND key or # to make a second call;
Press TRAN key again. The call will be transferred.
Note:
To transfer calls across SIP domains, SIP service providers must support transfer across SIP domains.
3-WAY CONFERENCING
GXP1100/GXP1105 can host 3-way conference call (PCMU/PCMA) 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->Programmable Keys, configure the 3-way conference as the Multi Purpose Key mode.
Click "Save and Apply" on the bottom of the page. Then follow the steps below for 3-way conferencing.
Figure 3: GXP1100/GXP1105 Multi Purpose Key - way Conference 3
1. . Initiate a conference call
Establish two active calls with two parties respectively;
Press the Multi Purpose Key previously configured as "3-way Conference" in 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
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 21 50
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->Call Features, set "Enable FLASH key as CONF" to "Yes" Click on "Save and Apply. "
on the bottom of the Web GUI page. 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.
N :ote
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 Conference Hangup" is turned on.
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 22 50
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->General
Settings page). An IVR will prompt the user through the process of message retrieval.
N :ote
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 :
*30
Block Caller ID (for all subsequent calls)
Off hook the phone;
Dial *30.
*31
Send Caller ID (for all subsequent calls)
Off hook the phone;
Dial *31.
*67
Block Caller ID (per call)
Off hook the phone;
Dial *67 and then enter the number to dial out.
*82
Send Caller ID (per call)
Off hook the phone;
Dial *82 and then enter the number to dial out.
*70
Disable Call Waiting (per Call)
Off hook the phone;
Dial *70 and then enter the number to dial out.
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 26 50
G- 723
G- 729
10
"MAC Address"
Announces the MAC address of the unit.
13
"Firmware Server IP Address"
Announces current Firmware Server IP address. Enter
12 digit new IP address.
14
"Configuration Server IP
Address"
Announces current Config Server Path IP address.
Enter 12 digit new IP address.
15
"Upgrade Protocol"
Upgrade Protocol for firmware and configuration update.
Enter 9 to toggle between HTTP, TFTP HTTPSand .
16
"Firmware Version"
Firmware version information.
17
"Firmware Upgrade"
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
"Voice Mail"
Announces number of voice mails.
99
"RESET"
Enter MAC address to restore factory default setting.
(See section) Restore Factory Default Setting
Press to reboot the device. 9
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 Google Chrome. and
To access th GXP /GXP1105 Web GUI: e 1100
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;
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 27 50
4. Open a Web browser on your computer;
5. address in the address bar of the browser; Enter the phone’s IP
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 can phone
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
phone has IP address 192.168.4 154, please enter "http://192.168. .154" the address bar of the 0. 40 in
browser;
The default administrator password is set to "admin". The default end user password is set to "123";
When changing any settings, always SUBMIT them by pressing the "Save" or "Save and Apply" button
on the bottom of the page. If the change is saved only but not applied, after making all the changes,
click on the "APPLY" button on top of the page to submit. After submitting the changes in all the Web
GUI pages, reboot the phone to have the changes take effect if necessary (All the options under
"Accounts" page and "Phonebook" page do not require reboot. Most of the options under "Settings"
page do not require reboot).
DEFINITIONS
This section describes the options in the GXP /GXP1105 Web GUI. As mentioned, you can log in as 1100
an administrator or an end user.
Displays the Account status, Network status, and System Info of the phone; Status:
To configure the SIP account; Account:
To configure network settings; Network:
To configure call features, ring tone, programmable keys and etc; Settings:
To configure web/Telnet access, upgrading and provisioning, language settings, Maintenance:
TR-069, security and etc.
To manage Phonebook and LDAP. Phonebook:
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 28 50
STATUS PAGE DEFINITIONS
Status -> Account Status
SIP User ID
Displays the configured SIP User ID.
SIP Server
Displays the configured SIP Server address.
SIP Registration
Displays SIP registration status YES/NO.
Status -> Network Status
MAC Address
Global unique ID of device, in HEX format. The MAC address will be used for
provisioning and can be found on the label coming with original box and on the
label located on the back of the device.
IP Setting
DHCP, Static IP or PPPoE.
IPv4 Address
The IPv4 address obtained on the phone.
IPv6 Address
The IPv6 address obtained on the phone.
Subnet Mask
The subnet mask obtained on the phone.
Gateway
The gateway address obtained on the phone.
DNS Server 1
The DNS server address 1.
DNS Server 2
The DNS server address 2.
PPPoE Link Up
PPPoE connection status.
NAT Traversal
NAT traversal status for each account.
Status -> System Info
Product Model
Product model of the phone.
Part Number
Product part number.
Software Version
Boot: boot version number;
Core: core version number;
Base: base version number;
Prog: program version number. This is the main firmware release number,
which is always used for identifying the software system of the phone;
Aux: Aux version number;
Dsp: DSP version number.
System Up Time
System up time since the last reboot.
System Time
Current system time on the phone system.
Service Status
GUI and Phone service status.
Core Dump
Core dump file that could be downloaded for troubleshooting purpose.
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 31 50
Port in Contact with
TCP/TLS
used or not. This is used when TLS/TCP is selected for SIP Transfer. The
default setting is "No".
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.
Support SIP Instance ID
Defines whether SIP Instance ID is supported or not. The default setting is
"Yes ".
SUBSCRIBE for MWI
When set to "Yes", a SUBSCRIBE for Message Waiting Indication will be sent
periodically. The phone supports synchronized and non-synchronized MWI.
The default setting is "No".
SUBSCRIBE for
Registration
When set to "Yes", a SUBSCRIBE for Registration will be sent out periodically.
The default setting is "No".
Enable 100rel
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.
Caller ID Display
When set to "Auto", the phone will look for the caller ID in the order of
P-Asserted Identity Header, Remote-Party-ID Header and From Header in the
incoming SIP INVITE. When set to "Disabled", all incoming calls are displayed
with "Unavailable". When set to "From Header", the phone will display the
caller ID based on the From Header in the incoming SIP INVITE. The default
setting is "Auto".
Use Privacy Header
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 P-Preferred-Identity
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
"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.
Account 1 -> SIP Settings -> Advanced Features
Conference URI
Configures the Conference URI when using Broadsoft N-way calling feature.
Music On Hold URI
Configures Music On Hold URI to call when a call is on hold. This feature has
to be supported on the server side.
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 32 50
Special Feature
Different soft switch vendors have special requirements. Therefore users may
need select special features to meet these requirements. Users can choose
from Standard, Nortel MCS, Broadsoft, CBCOM, RNK, Sylantro or Huawei IMS
depending on the server type. The default setting is "Standard".
Account 1 -> SIP Settings -> Session Timer
Session Expiration
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.
Min- SE
The minimum session expiration (in seconds). The default value is 90
seconds.
Caller Request Timer
If set to "Yes" and the remote party supports session timers, the phone will use
a session timer when it makes outbound calls.
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.
Force Timer
If Force Timer is set to "Yes", the phone will use the session timer even if the
remote party does not support this feature. If Force Timer is set to "No", the
phone will enable the session timer only when the remote party supports this
feature. To turn off the session timer, select "No".
UAC Specify Refresher
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.
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.
Force INVITE
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.
Account 1 -> SIP Settings -> Security Settings
Check Domain
Certificates
Defines whether the domain certificates will be checked or not when TLS/TCP
is used for SIP Transport. The default setting is "No".
Validate Incoming
Messages
Defines whether the incoming messages will be validated or not. The default
setting is "No".
Check SIP User ID for
incoming INVITE
If set to "Yes", SIP User ID will be checked in the Request URI of the incoming
INVITE. If it doesn't match the phone's SIP User ID, the call will be rejected.
The default setting is "No".
Accept Incoming SIP
from Proxy Only
When set to "Yes", the SIP address of the Request URL in the incoming SIP
message will be checked. If it doesn't match the SIP server address of the
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 33 50
account, the call will be rejected. The default setting is "No".
Authenticate Incoming
INVITE
If set to "Yes", the phone will challenge the incoming INVITE for authentication
with SIP 401 Unauthorized response. The default setting is "No".
Account 1 -> Audio Settings
Send DTMF
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.
DTMF Payload Type
Configures the payload type for DTMF using RFC2833. The default value is
101.
Preferred Vocoder
7 different vocoder types are supported on the one, including G.711 U-law ph
(PCMU), G.711 A-law (PCMA), G.723.1, G.729A/B, G.722 (wide band), iLBC
and G72- . Users can configure vocoders in a preference list that is included 32
with the same preference order in SDP message.
Use First Matching
Vocoder in 200OK SDP
When set to "Yes", the device will use the first matching vocoder in the
received 200OK SDP as the codec. The default setting is "No".
SRTP Mode
Enables the SRTP mode based on your selection. The default setting is
"Disabled ".
Symmetric RTP
Defines whether symmetric RTP is supported or not. The default setting is
"No".
Silence Suppression
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".
Voice Frames Per TX
Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the "ptime" value for the SDP will
change with different configurations here. This value is related to the codec
used and the actual frames transmitted during the in payload call. For end
users, it is recommended to use the default setting, as incorrect settings may
influence the audio quality.
G723 Rate
Selects encoding rate for G723 codec. The default value is 5.3kbps.
G.726-32 Packing Mode
Selects "ITU" or "IETF" for G726-32 packing mode.
iLBC Frame Size
Selects iLBC packet frame size. The default value is 30ms.
iLBC Payload Type
Specifies iLBC Payload type. The default value is 97. The valid range is
between 96 and 127.
Jitter Buffer Type
Selects either Fixed or Adaptive based on network conditions. The default
setting is "Adaptive ".
Jitter Buffer Length
Selects Low, Medium, or High based on network conditions. The default
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 34 50
setting is "Medium".
Account 1 -> Call Settings
Early Dial
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".
Dial Plan Prefix
Sets the prefix added to each dialed number.
Dial Plan
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. 9; Grammar: x - any digit from 0-
a) xx+ - at least 2 digit numbers
b) xx. - only 2 digit numbers
c) - exclude ^
d) 5] -[3 - 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 |
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.
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
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 36 50
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.
Ring Timeout
Defines the timeout (in seconds) for the rings on no answer. The default setting
is 60 seconds.
Send Anonymous
If set to "Yes", the "From" header in outgoing INVITE messages will be set to
anonymous, essentially blocking the Caller ID to be displayed.
Anonymous Call
Rejection
If set to "Yes", anonymous calls will be rejected. The default setting is "No".
Allow Auto Answer by
Call-Info
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep, based on the SIP info
header sent from the server/proxy. The default setting is "No".
Refer-To Use Target
Contact
If set to "Yes", the "Refer-To" header uses the transferred target's Contact
header information for attended transfer. The default setting is "No".
Transfer on Conference
Hangup
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
No Key Entry Timeout (s)
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.
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.
SETTINGS PAGE DEFINITIONS
Settings -> General Settings
Local RTP Port
This parameter defines the local RTP port used to listen and transmit. It
is the base RTP port for channel 0. When configured, channel 0 will use
this port _value for RTP; channel 1 will use port_value+2 for RTP. Local
RTP port ranges from 1024 to 65400 and must be even. The default
value is 5004.
Use Random Port
When set to "Yes", this parameter will force random generation of both
the local SIP and RTP ports. This is usually necessary when multiple
phones are behind the same full cone NAT. The default setting is "Yes"
(This parameter must be set to "No" for Direct IP Calling to work).
Keep-alive Interval
Specifies how often the phone sends a blank UDP packet to the SIP
server in order to keep the "ping hole" on the NAT router to open. The
FIRMWARE VERSION 1.0.5. GXP1100/GXP1105 USER MANUAL Page of 26 37 50
default setting is 20 seconds.
Use NAT IP
The NAT IP address used in SIP/SDP messages. This field is blank at
the default settings. It should ONLY be used if it's required by your ITSP.
STUN Server
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.
Settings -> Call Features
Off-hook Auto Dial
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".
Off-hook Timeout
If configured, when the phone is on hook, it will go off hook after the
timeout (in seconds). The default value is 30 seconds.
Disable Call Waiting
Disables the call waiting feature. The default setting is "No".
Disable Call Waiting Tone
Disables the call waiting tone when call waiting is on. The default setting
is "No".
Disable Direct IP Call
Disables Direct IP Call. The default setting is "No".
Use Quick IP Call mode
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, off hook 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".
Disable Conference
Disables the Conference function. The default setting is "No".
Enable MPK sending DTMF
Enables Multi Purpose Key to send DTMF during the call The default .
setting is "No".
Disable Transfer
Disables the Transfer function. The default setting is "No".
In-call dial number on pressing
transfer key
Configures the number for the phone to dial as DTMF during the call
using TRAN button.
Auto-Attended Transfer
If set to "Yes", the phone will use attended transfer by default. The default
setting is "No".
Do Not Escape # as %23 in
SIP URI
Specifies whether to replace # by %23 or not for some special situations.
The default setting is "No ".
Click- -Dial Feature To
Enables Click- -Dial feature. The default setting is "Disabled". To
Blink message LED on ringing
Message LED light will blink if there is an incoming call.
Call History Flash Writing:
Defines the interval (in seconds) to save the call history to phone's flash.
Produkt Specifikationer
| Mærke: | Grandstream |
| Kategori: | Telefon |
| Model: | GXP1105 |
| Bredde: | 201 mm |
| Dybde: | 78 mm |
| Højde: | 154 mm |
| Vægt: | 1000 g |
| Produktfarve: | Sort |
| Produkttype: | IP telefon |
| Berøringsskærm: | Ingen |
| Wi-Fi: | Ingen |
| Bluetooth: | Ingen |
| Relativ luftfugtighed ved drift (H-H): | 10 - 90 % |
| Ethernet LAN-porte (RJ-45): | 1 |
| Ethernet LAN: | Ja |
| Strømforbrug (typisk): | 1.5 W |
| Driftstemperatur (T-T): | 0 - 40 °C |
| Indgangsspænding for vekselstrømsadapter: | 100 - 240 V |
| Udgangsspænding for vekselstrømsadapter: | 5 V |
| Certificering: | FCC Part 15 (CFR 47) Classe B, EN55022 Classe B, EN55024, EN61000-3-2, EN61000-3-3, EN 60950-1 AS/NZS CISPR 22 Classe B, AS/NZS CISPR 24, RoHS; UL 60950 |
| Håndsæt, type: | Forbundet håndsæt |
| Højttalertelefon: | Ingen |
| Kapacitet for telefonbog: | - entries |
| DC-in-stik: | Ja |
| Servicekvalitet support (QoS): | Ja |
| Kan monteres på væggen: | Ja |
| Mikrofon mute: | Ja |
| Samtaleparkering: | Ja |
| Montering: | Bord/Væg |
| Antal linjer: | 1 Linier |
| Viderestilling af opkald: | Ja |
| Voice codecs: | G.711, G.722, G.723.1, G.726, G.729A |
| Strøm over Ethernet (PoE): | Ja |
| Antal medfølgende håndtag: | 1 stk |
| Tastatur antal taster: | 11 |
| Antal VoIP-konti: | 1 |
Har du brug for hjælp?
Hvis du har brug for hjælp til Grandstream GXP1105 stil et spørgsmål nedenfor, og andre brugere vil svare dig
Telefon Grandstream Manualer
12 Januar 2025
11 Januar 2025
16 August 2024
16 August 2024
14 August 2024
14 August 2024
14 August 2024
13 August 2024
12 August 2024
12 August 2024
Telefon Manualer
- Tecdesk
- Hamlet
- Nordmende
- Grundig
- Marmitek
- Listo
- Aastra
- 2N Telecommunications
- Linksys
- Panasonic
- Kyocera
- AudioCodes
- TriStar
- Yamaha
- Ranex
Nyeste Telefon Manualer
26 Juli 2025
26 Juli 2025
26 Juli 2025
26 Juli 2025
26 Juli 2025
26 Juli 2025
26 Juli 2025
26 Juli 2025
24 Juli 2025
24 Juli 2025