Fanvil i56A User Manual
Below you will find brief information for i56A. The i56A IP phone offers features such as PoE support, external power adapter option for flexibility, and versatile installation methods. It provides fundamental call operations like answering, making, holding and transferring calls, as well as video call capabilities. You can configure features such as call forwarding, call waiting, and set up conference calls. More advanced functionalities include BLF, call recording, and intercom.
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i56A User Manual
Software Version: 1.0.0
Release Date : 2020/10.26
Directory
I
II
III
IV
V
1 Picture
VI
VII
VIII
IX
2 Table
X
3 Safety Instruction
Please read the following safety notices before installing or using this unit. They are crucial for the safe and reliable operation of the device.
Please use the external power supply that is included in the package. Other power supply may cause damage to the phone and affect the behavior or induce noise.
Before using the external power supply in the package, please check the home power voltage. Inaccurate power voltage may cause fire and damage.
Please do not damage the power cord. If power cord or plug is impaired, do not use it because it may cause fire or electric shock.
Do not drop, knock or shake the phone. Rough handling can break internal circuit boards.
This phone is design for indoor use. Do not install the device in places where there is direct sunlight. Also do not put the device on carpets or cushions. It may cause fire or breakdown.
Avoid exposure the phone to high temperature or below 0 ℃ or high humidity.
Avoid wetting the unit with any liquid.
Do not attempt to open it. Non-expert handling of the device could damage it.
Consult your authorized dealer for help, or else it may cause fire, electric shock and breakdown.
Do not use harsh chemicals, cleaning solvents, or strong detergents to clean it.
Wipe it with a soft cloth that has been slightly dampened in a mild soap and water solution.
When lightning, do not touch power plug, it may cause an electric shock.
Do not install this phone in an ill-ventilated place. You are in a situation that could cause bodily injury. Before you work on any equipment, be aware of the hazards involved with electrical circuitry and be familiar with standard practices for preventing accidents.
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4 Overview
4.1 Overview
The i56A is a 10.1 inch indoor station, built-in Bluetooth 5.0 and 2.4G/5G Wi-Fi, integrated with Micro USB and TF interface. Based on Android 9.0 operating system, the interface operation is smooth and intelligent. It is mainly used in residential area, villa, office buildings and other places for receiving calling and communicating through the door phone and open the door remotely. It can provide reliable security assurance and the easy access control service for the users, creating a safe and comfortable living environment.
In order to help some users who are interested to read every detail of the product, this user manual is provided as a user’s reference guide. Still, the document might not be up to date with the newly release software, so please kindly download updated user manual from Fanvil website, or contact with Fanvil support if you have any question using i56A.
4.2 Product Introduction
4.2.1
i56A Physical specifications
1) Panel description
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Picture 1 -Panel description
2) Interface description
There are some interfaces on the back of the device for connecting power supply, alarms etc. The connections are as follows:
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Picture 2 -Interface Description
Table 1 - Interface Description
3) External device connection diagram
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Picture 3 -External device connection diagram
5 Install Guide
5.1 Use PoE or external Power Adapter
I56A, called as ‘the device’ hereafter, supports two power supply modes, power supply from external power adapter or over Ethernet (PoE) complied switch.
PoE power supply saves the space and cost of providing the device additional power outlet. With a PoE switch, the device can be powered through a single Ethernet cable which is also used for data transmission. By attaching UPS system to PoE switch, the device can keep working at power outage just like traditional PSTN telephone which is powered by the telephone line.
For users who do not have PoE equipment, the traditional power adaptor should be used.
If the device is connected to a PoE switch and power adapter at the same time, the power adapter will be used in priority and will switch to PoE power supply once it fails.
Please use the power adapter supplied by Fanvil and the PoE switch met the specifications to ensure the device work properly.
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5.2 Installation
（
two models
）
Picture 4 -Equipment Support
Model 1. Wall-mounted Installation
Step 1. Install wall bracket
Without 86 embedded box in the wall
A. According to the position of the cable in the wall, dig out a square hole
(height*width*depth=65.5*50.5*50mm) that can accommodate all cables.
B. Align the square hole of the wall bracket with the hole digged out before, then mark the two fixation holes through bracket on the wall.
C. Take down the bracket, using an electric drill to make the two fixation holes on the wall,then insert the two screw fixing seats provided.
D. Fix the wall bracket on the wall with two TA4*30 screws.
Picture 5 -Wall Mount 1
With 86 embedded box in the wall
A. Make sure all cables in the embedded box .
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B. Fix the wall bracket on the 86 embedded box with two PM4*16mm screws.
Picture 6 -Wall Mount 2
Step 2. Connect peripherals
A. If you need to connect other input and output devices, please connect to the host through the cable.
Step 3. Power on the device. If it is working properly, align the slot on the rear side of the panel with the pin on the wall bracket and slide the host down to complete the installation.
Picture 7 - Connecting to the Device
Model 2. Desktop Installation
Align the slot on the rear side of the panel with the pin on the desktop bracket and slide
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the bracket up to complete the installation.
Picture 8 - Desktop phone installation
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6 Appendix Table
6.1 Appendix I - Icon
Table 2 - Status Prompt and Notification Icons
Icons Description
Call out
Call in
Call Hold
Network Disconnected
SMS
DND
Call forward activated
Auto-answering activated
Hands-free (HF) Mode
Headphone (HP) Mode
Handset (HS) Mode
Mute Microphone
HD Audio
The Voice encryption of calling
Open Bluetooth
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SIP Hotspot
Connecting WIFI
Open Bluetooth
Unread voice message
USB insert tips
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Icons
Table 3 - DSSkey Icons
Description
MWI
Speed Dial
Intercom
Call Park
Call forward
Key Event/DND
Key Event/Call Hold
Key Event/Call Transfer
Key Event/Phonebook
Key Event/Redial
Key Event/Pickup
Key Event/Join
Key Event/Auto Redial On
Key Event/Auto Redial Off
Key Event/Call Forward
Key Event/Call Logs
Key Event/Flash
Key Event/
Key Event/Headset
Key Event/Release
Key Event/Lock Phone
Key Event/SMS
Key Event/Call Back
Key Event/Hide DTMF
Key Event/Power Light
Key Event/Prefix
Key Event/Hot Desking
Key Event/Agent
Key Event/End
Key Event/Disposition
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Key Event/Escalate
Key Event/Trace
Key Event/Handfree
Key Event/Answer Key
Key Event/Private Hold
URL & Action URL
BLF List
Multicast
Unfold
Collapse
Type
Line Key
BLF
DND
MWI
6.2 Appendix II –LED Definition
Presence
Off
Green On
LED Light
Green Blinking
Red Blinking
Red Blinking
Red On
Yellow Blinking
Green On
Red On
Red On
Off
Green On
Red On
Red On
Off
Red On
Off
Green Blinking
Off
Table 4 - DSS KEY LED State
LED State
Line inactive
Line ready (Registered)
Ringing
Line is trying to register
Line error (Registration failure)
Dialing/Line in use (Talking)
Call holding
Subscription number is idle.
Subscription number is busy.
Subscription number is dialing.
Subscription number is unavailable.
Subscription number is idle.
Subscription number is busy.
Subscription number is dialing.
Subscription number is unavailable.
Enable DND
Disable DND
New voice message waiting
No new voice message
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7 Introduction to the User
7.1 Idle Screen
Picture 9 -Default home screen
The image above shows the default standby screen, which is the user interface in the most of the time.
The main screen displays the frequently used function menu buttons of the device. It is also the first layer of the function menu buttons. Users can operate the phone through them
7.2 View device IP
There are two ways to check the device IP. The default WAN mode is DHCP.
method 1:
After powering on, click [Settings] >> [Common] to query.
Method 2:
After starting, pull down the notification bar to query.
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7.3 Dial
Dial:
The user can press the dialer key to dial, and the sound will be played from the speaker.
Line dial:
The user can also use the line key to specify the line to make/receive calls.
7.4 Screen Touch Instructions
The device can be configured and operated by touching the screen.
Click
The device can enter the setting and operation interface by clicking on any interface.
The device supports multi-touch.
Long Press
Long press the app icon on the standby home page, you can adjust the app location or choose to delete.
Long press the application icon in the menu interface to drag it to the main page.
Slide
The device supports sliding up and down.
Slide down the standby home page to view the network connection information, date time and other information of the device;Slide up to exit the above information interface.
Right slide can expand DSSkey, full screen display custom shortcut key information;Slide left to exit the above interface.
Drag
Long press the application icon in any interface, and you can drag it to any place.
7.5 Phone Status
The phone status includes the following information about the phone:
Network Status ：
IPv4
IP Address
Network Mode
Mac Address
The Phone Device Information ：
Phone Mode
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Hardware Version number
Software Version number
Phone Storage （ RAM and ROM ）
System Running Time
Android Version
SIP Account Information ：
SIP Account
SIP Account Status (registered / uncommitted / trying / time out)
The user can view the phone status through the phone interface and the web interface.
Phone interface ： When the phone is in standby mode, press [ Menu ] >> [ Status ] and select the option to view the corresponding information, as shown in the figure:
Picture 10 -The Phone status
WEB interface ： Refer to
to log in the phone page, enter the
[System] >> [Information] page, and check the phone status, as shown in the figure:
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Picture 11 -WEB phone status
7.6 Web Management
Phone can be configured and managed on the web page of the phone. The user needs to enter the IP address of the phone in the browser at first and open the web page of the phone. The user can check the IP address of the phone by pressing [ Menu ] >> [ Status ].
Open the browser, enter the phone IP, and log in to the phone web page. The first thing you see is the phone's login page.
Picture 12 - Landing page
Users must correctly enter the user name and password to log in to the web page. The default user name and password are "admin". For the specific details of the operation
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page, please refer to page
7.7 Network Configurations
The device supports two kinds of network connection modes: wired network connection and wireless network connection. This section describes the wired network connection.
For wireless network connection, refer to
The device relies on IP network connection to provide service. Unlike traditional phone system based on a circuit switched wire technology, IP devices are connected to each other over the network and exchange data in packet basis based on the devices’ IP address.
To enable this phone, you must first correctly configure the network configuration. To configure the network, users need to find the phone function menu button [ Settings ] >> 】
Advanced]>> [ Network ] >> [ Ethernet ].
NOTICE! If user saw a ‘WAN Disconnected’ icon flashing in the middle of screen, it means the network cable was not correctly connected to the device’s network port. Please check the cable is connected correctly to the device and to the network switch, router, or modem.
The device supports IPv4.
There are two common IP configuration modes about IPv4.
Dynamic Host Configuration Protocol (DHCP) – This is the automatic configuration mode by getting network configurations from a DHCP server. Users need not to configure any parameters manually. All configuration parameters will be getting from DHCP server and applied to the device. This is recommended for most users.
Static IP Configuration – This option allows user to configure each IP parameters manually, including IP Address, Subnet Mask, Default Gateway, and DNS servers.
This is usually used in an office environment or by power users.
The device is default configured in DHCP mode.
Please see
for detailed configuration and use.
7.8 SIP Configurations
A line must be configured properly to be able to provide telephony service. The line configuration is like a virtualized SIM card. Just like a SIM card on a mobile phone, it stores the service provider and the account information used for registration and
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authentication. When the device is applied with the configuration, it will register the device to the service provider with the server’s address and user’s authentication as stored in the configurations.
The user can conduct line configuration on the interface of the phone or the webpage, and input the corresponding information at the registered address, registered user name, registered password and SIP user, display name and registered port respectively, which are provided by the SIP server administrator.
Phone interface ： To manually configure a line, the user can press the button in the function menu [ Settings ] >> [ Advanced ]>>[ Account ] >> [ Line ] configuration, click
“OK” to save the configuration.
NOTICE! User must enter correct PIN code to be able to advanced settings to edit line configuration. (The default PIN is admin)
The parameters and screens are listed in below pictures.
Picture 13 - Phone line SIP address and account information
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Picture 14 - Phone display name and port
WEB interface ： After logging into the phone page, enter [ Line ] >> [ SIP ] and select
SIP Line for configuration, click apply to complete registration after configuration, as shown below:
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Picture 15 - Web SIP registration
8 Basic Function
8.1 Making Phone Calls
Default Line
The device provides six line services. If all lines are configured, user can make or receive phone calls on either line. If default line is configured by user, there will be a default line to be used for making outgoing call which is indicated on the top left corner. To use the default line, user can press [ Settings ] >> [ Call ] >> [ Call ] or configure from Web Interface
(Web / PHONE / Features / Basic Settings).
Picture 16 - Default line
Dialing Methods
User can dial a number by,
Entering the number from dialer
Selecting a phone number from phonebook contacts (Refer to
contacts )
Selecting a phone number from cloud phonebook contacts (Refer to
Cloud Phone Book )
Selecting a phone number from call logs (Refer to
Dial Number
To make a phone call, user can firstly dial a number by one of the above methods. When the dialed number is completed, user can press [ Dial ] button on the soft-menu to call out
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with the default line.
Picture 17 - Dial number
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Cancel Call
While calling the number, user can press to end the call with [ End ] button.
Picture 18 - Call number
8.2 Answering Calls
when the phone is idle and there is a call, the user will see the call reminder screen as below.
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Picture 19 - Answering calls
User can answer the call by press the [Answer] button. To divert the incoming call, user should press [ Divert ] button. To reject the incoming call, user should press [ Reject ] button.
8.2.1
Talking
When the call is connected, user will see a talking mode screen as the following figure.
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③
④
⑤
⑥
Number
①
②
Name
The current line
User avatar
Calls to end
Call duration
Video icon
Record icon
Picture 20 - Talking interface
Table 9 - Talking mode
Description
The line currently used by the phone.
Default display, user can customize the selection of avatar pictures.
The name or number of the person on the other end of the call.
The duration of a call after it has been established.
Click to initiate video call.
Click start recording and click end recording again.
8.2.2
Make / Receive the Second Call
The device can support up to multiple calls. When there is already a call established, user can still answer another incoming call on either lines or make a second call on either lines.
The Second Incoming Call
When there is another incoming call during talking a phone call, this call will be waiting for user to answer it. User will see the call message in the middle of current screen. The device will not be ringing but playing call waiting tone in the audio channel of the current call. User can accept or reject the call as same as normal incoming call. When the waiting call is answered, the first call will be put on hold automatically.
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Picture 21 - The second call interface
Second Outgoing Call
To make a second call, user may press [ Xfer ] / [ Conf ] button to make a new call on the default line. Then dial the number the same way as making a phone call. Another alternative for making second call is to pressing DSS Keys dial out from the configured
Keys (BLF/Speed Dial). When the user is making a second call with the above methods, the first call could be placed on hold manually first or will be put on hold automatically at second dial.
Switching between Two Calls
When there are two calls established, user will see a dual calls screen as the following picture.
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Picture 22 - Two way calling
User can switch screen page by touch the screen, and switch call focus by pressing
[ Resume ] button.
Ending One Call
User may hang up the current talking call by press [ End ] button. The device will return to single call mode in holding state.
8.3 End of the Call
After the user finishes the call, the user can press softkey [ End ] key to end the call.
8.4 Video Call
i56A supports a variety of video formats QVGA,CIF,VGA,4CIF,720P,1080p 。
The device only supports video decoding, but users can initiate video calls.
The default dialing mode is video. When the device dials, it USES video mode to call out by default. If the end device supports sending video, both sides establish video call.
The default dialing mode is voice. The above operation establishes voice call
Picture 23 - Video interface
WEB interface: enter [ Phone Settings ] >> [ Features ] >> [ Basic Settings ], and choose to configure the “Default Dial Mode” and “Default Ans Mode”.
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Picture 24 - Video Settings
8.5 Redial
Redial the last outgoing number:
Set the function key as redial key. When the phone is in standby mode, press the redial key to call out the last number dialed.
8.6 Dial-up Query
Phone is defaulted to open the dial-up inquiry function, dial-out, enter two or more
Numbers, dial the interface will automatically match call records, contacts in the number list, touch the number that you select to call out.
8.7 Auto-Answering
User may enable auto-answering feature on the device and any incoming call will be automatically answered (not including call waiting). The auto-answering can be enabled on line basis.
The user can enable the automatic answer function in the telephone interface or the webpage interface.
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Phone interface ：
Press [ Settings ] >>[ Advanced ]>> [ Account ] >> [ Line ] ;
Press the button to select the line and enter the [ Basic Settings ]. Click on/off the auto answering option and set the auto answering time. The default is 5 seconds.The icon in the upper left corner of the screen indicates that auto answer is enabled.
Picture 25 - Line 1 enables auto-answering
WEB interface ：
Log in the phone page, enter [ Line ] >> [ SIP ], select [ SIP ] >> [ Basic settings ], start auto-answering, and click apply after setting the automatic answering time.
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Picture 26 - Web page to start auto-answering
8.8 Call Back
The user can dial back the last call. If there is no call history, press the [ Callback ] button and the phone will say "can't process".
Set the callback key through the phone interface:
In standby mode, click the unfold button and long press the function key to be set, it will automatically enter the configuration interface; Type select key event type, subtype select call back, you can set the call back key name in the title input box, press [√] button to save.
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Picture 27 - Set the callback key on the phone
Set the callback key through the web interface:
Log in the phone page, enter the [ Function Key ] >> [ Function Key ] page, select the function Key, set the type as the function Key, and set the subtype as the callback, as shown in the figure:
Picture 28 - Set the callback key on the web page
8.9 Mute
You can turn on mute mode during a call and turn off the microphone so that the local voice is not heard. Normally, mute mode will be automatically turned off at the end of a call. You can also turn on mute on any screen (such as the free screen) and mute the ringtone automatically when there is an incoming call.
8.9.1
Mute the Call
During the conversation, press the mute button on the phone: turned on.
The mute will be
Red mute icon is displayed in the call interface, as shown in the figure:
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Picture 29 - Mute the call
Cancel mute: press cancel mute on the phone again. The mute icon is no longer displayed in the call screen.
8.9.2
Ringing Mute
Mute: press the mute button when the phone is ringing:
The top right corner of the phone shows the bell mute icon ， when there is an incoming call, the phone will display the incoming call interface but will not ring.
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Picture 30 - Ringing mute
Cancel ring tone mute: On the incoming call screen, press the mute button again or volume up button to cancel ring tone mute. It will no longer shows mute icon in upper right corner after cancel .The phone mute icon is off
8.10 Call Hold/Resume
The user can press the [ Hold ] button to maintain the current call, and this button will become the [ Resume ] button, and the user can press the [ resume ] button to restore the call.
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Picture 31 - Call hold interface
8.11 DND
User may enable Do-Not-Disturb (DND) feature on the device to reject incoming calls
(including call waiting). The DND can be enabled on line basis.
Enable/Disable phone all lines DND, Methods the following ：
Phone interface ： Default standby mode ，
1) In standby, press the [Do Not Disturb] button to turn on the DND function of the phone, the DND icon on the phone screen will turn red , and the phone status prompt bar will have the DND icon.
2) Press the [Do Not Disturb] button again to turn off Do Not Disturb, the DND icon on the phone screen will change back status prompt bar disappears.
, and the DND icon in the phone
Picture 32 - Enable DND
If the user wishes to enable/disable the uninterrupted function on a specific line, the user can set the uninterrupted function on the page of configuring the line.
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1) Press [ Settings ] >>[ DND ] button, Enter the [ DND ] editing interface.
2) select the line to adjust the mode and state of "do not disturb".
3) The user will see the DND icon turn red, and the sip-line has enabled the mode of "DND".
Picture 33 - DND setting interface
The user can also use the DND timer. After the setting, the DND function will be automatically turned on and the DND icon will turn red in the time range.
Picture 34 - DND timer
WEB interface ： Enter [ Phone setting ] >> [ Features ] >> [ DND settings ] ， set the
DND type (off, phone, line), and DND timing function.
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Picture 35 - DND Settings
The user turns on the DND for a specific route on the web page ： Enter [ Line ] >> [ SIP ], select a [ Line ] >> [ Basic settings ], and enable DND.
8.12 Call Forward
Picture 36 - Line DND
Call forward is also known as ‘Call Divert’ which is to divert the incoming call to a specific number based on the conditions and configurations. User can configure the call forward settings of each line.
There are three types,
Unconditional Call Forward – Forward any incoming call to the configured number.
Call Forward on Busy – When user is busy, the incoming call will be forwarded to the configured number.
Call Forward on No Answer – When user does not answer the incoming call after the configured delay time, the incoming call will be forwarded to the configured number.
Phone interface ： Default standby mode
1) Press [ Settings ] >> [ Account ] >> [ Line ] button, click any line to set up forward settings.
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Picture 37 - Select the line to set up call forwarding
2) Select the line to be set and enter the call forward settings interface
Picture 38 - Select call forward type
3) Click the slide button to select on/off.
4) Configure parameters by clicking Settings and enter the required information.
WEB interface ： Enter [ Line ] >> [ SIP ], Select a [ Line ] >> [ Basic settings ], and set the type, number and time of forwarding.
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Picture 39 - Set call forward
8.13 Call Transfer
When the user is talking with a remote party and wish to transfer the call to another remote party, there are three ways to transfer the call, blind transfer, attended transfer and Semi-Attended transfer.
Blind transfer ： Do not need to negotiate with the other side, directly transfer the call to the other side.
Semi-Attended transfer.
： When you hear the ring back, transfer the call to the other party.
Attended transfer ： When the caller answers the call, transfer the call to the caller.
Note ！ For more transfer Settings, please refer to
8.13.1
Blind transfer
During the call, the user presses the function menu button [ Transfer ] , enter the number to transfer, press the transfer key again to a third party. After the third party rings, the phone will show that the transfer is successful and hang up.
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8.13.2
Picture 40 - Transfer interface
Semi-Attended transfer
During the call, the user presses the function menu button [ transfer ] to input the number to be transferred, and then press the call button. When the third party is not answered, press the transfer on the call interface to make the semi-attendance transfer or press the end button to cancel the semi-attendance transfer.
Picture 41 - Semi-Attended transfer
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8.13.3
Attended transfer
Attendance transfer is also known as "courtesy mode", which is to transfer the call by calling the other party and waiting for the other party to answer the call.
Calling is the same procedure. In dual call mode, press the "transfer" button to transfer the first call to the second call.
Picture 42 - Attended transfer
8.14 Call Waiting
Enable call waiting: new calls can be accepted during a call.
Disable call waiting: new calls will be automatically rejected and a busy tone will be prompted.
Enable call waiting tone: when you receive a new call on the line, the tone will beep.
The user can enable/disable the call waiting function in the phone interface and the web interface.
Phone interface ： Press [ Phone Settings ] >> [ Call ] >> [ Call ], enable/disable call waiting and call waiting tone.
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Picture 43 - Call waiting setting
WEB interface: Enter [ Phone Settings ] >> [ Features ] >> [ Basic Settings ], enable/disable call waiting and call waiting tone.
Picture 44 - Web call waiting setting
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Picture 45 - Web call waiting tone setting
8.15 Conference
8.15.1
Local Conference
To conduct local conference, the user needs to log in the webpage and enter [ Line ] >>
[ SIP ] >> [ Basic settings ]. The meeting mode is set as local (the default is local mode), as shown in the figure:
Picture 46 - Local conference setting
Two ways to create a local conference ：
1 ） The device has two channels of communication. Press the conference button on the call interface. When selecting the conference number, select the other number that already exists.
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Picture 47 - Local conference （ 1 ）
2 ） If the device has a call all the way, press the conference key in the call interface, enter the number to join the meeting and press the call; After the opposite end is answered, press the conference button again to set up the local tripartite conference ：
Picture 48 - Local conference （ 2 ）
Note:During the conference, press the split button to split the conference and press the end button to end the call.
8.15.2
Network Conference
Users need server support for network conference.
Log in the web page, enter [ Line ] >> [ SIP ] >> [ Basic settings ], set the conference mode as server mode (default is local mode), set the server conference room number (please consult your system administrator), as shown in the figure ：
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Picture 49 - Network conference
Method to join a network conference ：
Call the numbers of network conference and when they enter the password then will enter the conference room.
The two phones have established common calls. Press the conference button to invite new members to the conference. Follow the voice prompt to operate.
Note: the upper limit of the number of participants in the network conference varies according to the server .
8.16 Call Park
Call Park requires server support. Consult your system administrator for support.
When you are on the call, it is not convenient to answer the phone at this time, you can press the configured park button to hold the call; After the Call Park is successful, you can resume the call by pressing the configured park button on other devices.
Set the call park button ：
Phone interface: In standby mode, long press an editable key to enter the function key setting interface. key function key type as memory and subtypes to call park, reside values for the server calls park number, set up corresponding SIP lines.
WEB interface: log in the phone page, enter the [ Function Key ] >> [ Function Key ] page, select a DSSkey, set the function key type as memory key, the subtype as call park, and the value as the call park number of the server, and set the corresponding
SIP line.
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Picture 50 - Phone set call park
Picture 51 - WEB set call park
8.17 Pick Up
Picking-up requires server support. Consult your system administrator for support.
You can use the Pick Up function to answer incoming calls from other users. The phone can pick up incoming calls by configuring DSSkey for BLF and setting the Pick Up code.
In standby mode, long press an editable key to enter the interface of function key setting.
Set the function key type as memory key and the subtype as BLF/NEW CALL, and set the corresponding SIP line. Finally fill in the grab number.
Set the line, function key type as memory key, subtype as BLF/NEW CALL, set
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57 subscription number, and pick up code
Other phones call the subscription number, and the opposite end is in the incoming ring.
Press the DSS key to pick up the phone.
The caller picks up the call and speaks to it.
WEB interface: Log in the phone webpage, enter the [ Function Key ] >>
[ Function Key ] page, select a DSSkey, set the memory key type as memory key, the subtype as BLF/NEW CALL, and set the corresponding SIP line and pick up codes.
Picture 52 - Phone pick up setting
Picture 53 - WEB pick up setting
8.18 Anonymous Call
8.18.1
Anonymous Call
The phone can set up anonymous calls to hide the calling number and the calling name.
You can see anonymity in the context of [ Settings ] >> [ Account ] >> [ Line ] >>
[ Advanced Settings ] >> [ Anonymous call edition ].
The default is none, which is off, and RFC3323 and RFC3325 are optional.
Select any one to open the anonymous call.
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Picture 54 - Enable anonymous call
On the web page [ Line ] >> [ SIP ] >> [ Advanced Settings ] can also open anonymous calls.
Setting to enable anonymous calls also corresponds to the SIP line. That is, the setting under the SIP1 page can only take effect on the SIP1 line.
Picture 55 - Enable Anonymous web page call
The following is a transcript of an anonymous call received by the phone.
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8.18.2
Picture 56 - Anonymous call log
Ban Anonymous Call
The device can be set to prohibit anonymous calls, that is anonymous calls to the number will be directly rejected.
In the phone [ Settings ] >> [ Account ] >> [ Line ] >> [ Advanced Settings ] >>
[ Ban anonymous call ], can be enable and disable.
Picture 57 - Anonymous calls are not allowed on the phone
On the web page [ Line ] >> [ SIP ] >> [ Advanced Settings ], also can disable anonymous calls.
The setup to disable anonymous calls also corresponds to the SIP line. That is, the setting under the SIP1 page can only take effect on the SIP1 line.
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Picture 58 - Page Settings blocking anonymous call
9 Advance Function
9.1 BLF (Busy Lamp Field)
9.1.1
Configure the BLF Functionality
Page interface: log in the phone page, enter the [Function key] >> [Function key] page, select a DSS key, set the function key type as memory key, choose subtype among BLF/NEW CALL, BLF/BXFER, BLF/AXFER, BLF/CONF, set BLF/DTMF value as the number which is subscribed, set the corresponding SIP line. The pickup number is provided by the server. The specific use of reference
Picture 59 - Web page configuration BLF function key
Phone interface: Click unfold, long press a function key to enter the function key
Settings interface, key function key types of memory, a subtype of BLF/NEW
CALL, BLF/BXFER, BLF/AXFER, BLF/CONF, BLF/DTMF, the values to be subscription number, and set up corresponding SIP lines.
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Picture 60 - Phone configuration BLF function key
Table 6 - BLF Function key subtype parameter list
Subtype
BLF/NEW
CALL
BLF/BXFE
R
Standby is described
Pressing the BLF key while standby to dial the subscriber number.
Pressing the BLF key while standby to dial the subscriber number.
BLF/AXFE
R
Pressing the BLF key while standby to dial the subscriber number.
BLF/Confer ence
Pressing the BLF key while standby to dial the subscriber number.
BLF/DTMF Pressing the BLF key while standby to dial the subscriber number.
Calling is described
When you press this BLF key while talking to another user, you create a new call along with the subscribed number.
When you press this BLF key while talking to another user, you blind transfer the call to the subscribed number.
When you press this BLF key while talking to another user, you attendance transfer the call to the subscribed number.
When you press this BLF key while talking to another user, you invite the subscriber number to join the meeting.
When the BLF key is pressed while talking to another user, the phone automatically sends the DTMF corresponding to the BLF key number.
9.1.2
Use the BLF Function
The BLF, also known as a "busy light field," notifies the user of the status of the
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subscribed object and is used by the server to pick up the call. BLF helps you monitor the other person's status (idle, ringing, talking, off).
BLF function ：
Monitor the status of subscribed phones.
Call the subscribed number.
Transfer calls to the subscribed number.
Pick up incoming calls from subscribed number.
1 ） Monitors the status of subscribed phones.
Configuration BLF function keys, when the subscription of the number of the state (idle, ringing, talking) is changed, the function key state of LED lights will have corresponding change, see Appendix II 6.2
to get to know each other under different status leds.
2 ） Call the subscribed number.
When the phone is in standby mode, press the configured BLF key to call out the subscribed number.
3 ） Transfer calls/calls to the subscribed number.
Refer to
subtype parameter list, the BLF key can be used for blind rotation, attention-rotation and semi-attention-rotation of the current call, and also can invite the subscribed number to join the call and send DTMF, etc.
4 ） Pickup incoming calls from subscribed phones.
When configuring BLF function key, configure the pickup number.
When the subscription number telephone rings, refer to Appendix II 6.2
will flash a red light. At this point, press the BLF button to answer the incoming call from the subscribed number.
9.2 BLF List
BLF List Key is to put the number to be subscribed into a group on the server side, and the phone uses the URL of this group to make unified subscription. The specific information, number, name and status of each number can be resolved based on notify sent from the server. The unoccupied Memory Key is then set to the BLF List Key. If the state of the subscription object changes later, the corresponding led light state will be changed.
Configure BLF List function: log in the phone page, enter the [ Line ] >> [ SIP ] >>
[ Advanced settings ] page, open the BLF List, and configure the BLF List number.
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Picture 61 - Configure the BLF List functionality
Use the BLF List function: when the configuration is completed, the phone will automatically subscribe to the contents of the BLF List group. Users can monitor, call and transfer the corresponding number by pressing the BLF List key.
Picture 62 - BLF List number display
9.3 Record
The device supports recording during a call.
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9.3.1
Local Record
When using local recording, it is necessary to start recording on the phone page
[ Application ] >> [ Manage recording ], select the local type and set the voice coding.
The webpage is as follows:
Picture 63 - WEB local recording
Local recording steps:
Open the recording on the web page, and set the recording type as local recording.
Set DSSkey type as key event and type as record in the phone/web interface.
Set up one line call and press the recording key (set DSSkey).
End the recording. End the call.
View local recording:
Enter [ Application ] >> [ Sound Recorder ]
Enter view the recording file.
Or enter the webpage [ Application ] under the [ Manage recording ] to view the recording file.
Listen to the record ：
Enter [ Application ] >> [ Sound Recorder ].
Enter view the recording file.
Select the recording file that you want to listen to, and click listen to the recording.
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9.3.2
Server Record
When using the network server to record, it is necessary to open the recording in the phone web page [ Application ] >> [ Manage recording ]. The type is selected as network, and the address and port of the recording server are filled in and the voice coding is selected. The web is as follows:
Picture 64 - Web server recording
Note: to be used with Fanvil recording software.
Please refer to the documentation for specific usage:
Call Recording Configuration and Use Description
http://www.fanvil.com.cn/Uploads/Temp/download/20180928/5badccd249853.pdf
9.3.3
SIP INFO Record
The phone is registered with a server that supports SIP INFO recording. After registering the account, check the recording module of [ Application ] >> [ Manage recording ] to open the Record Settings, and the recording type is SIP INFO.
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Picture 65 - Web SIP info recording
9.4 Agent
Agent (Agent function) of the phone can be realized: when multiple people use a device for Agent services at different times, he or she can quickly register his or her SIP account on the same server. The Agent functions of the phone can be divided into Normal and
Hotel Guest. The Hotel Guest mode requires server support.
Normal Mode ：
Configure agent function: set a DSS key as agent, press the function key or enter the
[ Settings ] >> [ Call ] >> [ More ] >> [ Agent ] to enter the agent page. The SIP server needs to be configured before the account can be configured.
Picture 66 - Configure the agent account in normal mode
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Picture 67 - Configure the proxy account-hotel Guest mode
Table 7 - Agency mode
Parameter
Normal mode
Number
User
Password
Line
CallLog
Hotel Guest mode
Number
Password
Line
CallLog
Status
Description
Set the proxy account number.
Set the proxy account number to verify the user name.
Set the proxy account number to verify the password.
Select the SIP line.
Users can choose to save all types, or delete.
Set the proxy account number.
Set the proxy account number to verify the password.
Select the SIP line.
Users can choose to save all types, or delete.
The user can select the status of the number, the optional status is: login, logout, invalid, valid, SMS.
Using agent functions:
1 ） When the phone has been configured on SIP server, fill in the correct number and user name password, click login and then the phone can be registered to the SIP server;
2 ） After registration, click logout and the phone can delete the user name and password, and log out of the SIP account.
3 ） Click Unregister and the phone retain the user name and password, and logs
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out of the SIP account.
Picture 68 - Agent logon page
9.5 Intercom
When the Intercom is enabled, it can automatically receive calls from the intercom.
Picture 69 - Web Intercom configure
Table 8 - Intercom configure
Parameter Description
Enable Intercom When intercom is enabled, the device will accept the incoming call request with a SIP header of Alert-Info instruction to automatically answer the call after specific delay.
Enable Intercom
Mute
Enable mute mode during the intercom call
Enable Intercom If the incoming call is intercom call, the phone plays the intercom tone
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Tone
Enable Intercom
Barge
Enable Intercom Barge by selecting it, the phone auto answers the intercom call during a call. If the current call is intercom call, the phone will reject the second intercom call
9.6 MCAST
This feature allows user to make some kind of broadcast call to people who are in multicast group. User can configure a multicast DSS Key on the phone, which allows user to send a Real Time Transport Protocol (RTP) stream to the pre-configured multicast address without involving SIP signaling. You can also configure the phone to receive an RTP stream from pre-configured multicast listening address without involving
SIP signaling. You can specify up to 10 multicast listening addresses.
Parameters
Normal Call Priority
Enable Page Priority
Name
Host: port
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Picture 70 - Multicast Settings Page
Table 9 - MCAST Parameters on Web
Description
Define the priority of the active call, 1 is the highest priority, 10 is the lowest.
The voice call in progress shall take precedence over all incoming paging calls.
Listened multicast server name
Listened multicast server’s multicast IP address
and port.
Multicast ：
Go to web page of [ Function Key ] >> [ Function Key ] , select the type to multicast, set the multicast address, and select the codec.
Click Apply.
Set up the name, host and port of the receiving multicast on the web page of
[ Phone Settings ] >> [ MCAST ].
Press the DSSKEY of Multicast Key which you set.
Receiver will receive multicast call and play multicast automatically.
9.7 SCA
（
Shared Call Appearance
）
Users need the support of server end to use SCA function. You can refer to http://www.fanvil.com/Uploads/Temp/download/20180920/5ba38181e4e4b.pdf
1 ） Configure on Phone
When registering with the BroadSoft server, a Fanvil Phone can register the account created previously on multiple terminals.
Picture 71 - Register BroadSoft account
After the phone registers on the BroadSoft server, a server type needs to be set.
Specifically, log in to the web page of the phone, choose [ Line ] >> [ SIP ] >> [ Advanced Settings ] and set Specific Server Type to BroadSoft, as shown in the following figure.
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Picture 72 - Set BroadSoft server
If a Fanvil phone needs to use the SCA function, enable it for the phone set.
Specifically, log in to the webpage of the phone set, choose [ Line ] >> [ SIP ] >> [ Advanced Settings ], and select Enable SCA. If
SCA is not enabled, the registered line is private line.
After an account is configured and successfully registered, you can configure DSS Keys as the lines which can enable Shared Call Appearanceas on the Function Key page to facilitate viewing the call status of the group. Each line key represents a call appearance. Understand the call status by referring to Appendix II 6.2
.
To facilitate private hold, configure keys whose DSS Key is Private Hold on the
Function Key page. Pay attention that the public hold key is the softkey [ Hold ] key
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Picture 73 - Enable SCA
during a call.
Picture 74 - Set Private Hold Function Key
After each phone registered with the BroadSoft server is configured as above, the
SCA function can be used.
2 ） LED Status
To facilitate viewing the call status of a group, configure lines whose DSS Key is SCA.
The following table describes the LEDs of lines in different states.
Table 10 - LED Status of SCA
State & Direction
Idle
Seized
Progressing (outgoing call)
Alerting (incoming call)
Active
Public Held (hold)
Held-private (private hold)
Bridge-active (Barge-in)
Bridge-held
Local Light
Off
Steady green
Steady green
Fast blinking green
Steady green
Slow blinking green
Slow blinking yellow
Steady green
Steady green
Remote Light
Off
Steady red
Steady red
Fast blinking green
Steady red
Slow blinking red
Steady red
Steady red
Steady red
3 ） Shared Call Appearance(SCA)
The following lists a couple of instances to facilitate understanding.
In the following scenarios, the manager and secretary register the same SCA account and the account is configured based on the preceding steps.
Scenario 1: When this account receives an incoming call, the phone sets of both the
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manager and the secretary will receive the call and ring. If the manager is busy, the manager can reject the call and the manager's phone set stops ringing but the secretary's phone set keeps ringing until the secretary rejects/answers the call or the call times out.
Scenario 2: When this account receives an incoming call, if the secretary answers the call first and the manager is required to answer the call, the secretary can press the
Public Hold key to hold this call and notify the manager. The manager can press the line key corresponding to the SCA to answer the call.
Scenario 3: The manager is in an important call with a customer and needs to leave for a while. If the manager does not want others to retrieve this call, the manager can press the Private Hold key.
Scenario 4: The manager is in a call with a customer and requires the secretary to join the call to make records. The secretary can press the corresponding SCA line key to barge in this call.
9.8 Message
9.8.1
MWI
（
Message Waiting Indicator
）
If the service of the lines supports voice message feature, when the user is not available to answer the call, the caller can leave a voice message on the server to the user. User will receive voice message notification from the server and device will prompt a voice message waiting icon on the standby screen.
Picture 75 - New Voice Message Notification
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To listen to a voice message, the user must first configure the voicemail number. After the voicemail number is configured, the user can retrieve the voicemail of the default line.
When the phone is in the default standby state,
The voicemail icon displays the number of unread voicemail messages.
Click the icon to view the total number of voicemail messages, or listen to the messages directly in the voicemail interface
Picture 76 - Voice message interface
Picture 77 - Configure voicemail number
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9.9 SIP Hotspot
SIP hotspot is a simple but practical function. With simple configurations, the SIP hotspot function can implement group ringing. SIP accounts can be expanded.
Set a phone as a SIP hotspot and other phones (B and C) as SIP hotspot clients. When somebody calls phone A, phone A, B, and C all ring. When any phone answers the call, other phones stop ringing. The call can be answered by only one phone. When B or C initiates a call, the SIP number registered by phone A is the calling number.
To set a SIP hotspot, register at least one SIP account.
Picture 78 - Register SIP account
Table 11 - SIP hotspot Parameters
Parameters Description
If your phone is set to “SIP hotspot server”, Device Table will display as Client
Device Table
Monitor Type
Device Table which connected to your phone.
If your phone is set to “SIP hotspot client”, Device Table will display as Server
Device Table which you can connect to.
SIP hotspot
Enable hotspot Set it to be Enable to enable the feature.
Mode
Choose hotspot, phone will be a “SIP hotspot server”; Choose Client, phone will be a “SIP hotspot Client”
Either the Multicast or Broadcast is ok. If you want to limit the broadcast packets, you’d better use broadcast. But, if client choose broadcast, the SIP hotspot phone must be broadcast.
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Monitor Address The address of broadcast, hotspot server and hotspot client must be same.
Local port
Fill in the custom hotspot communication port. The server and client ports need to be consistent
Name
Line settings
Fill in the name of the SIP hotspot, this configuration is used to distinguish different hotspots under the network to avoid connection conflicts
Set whether to associate the SIP hotspot function on the corresponding SIP line
Configure SIP hotspot server:
Picture 79 - SIP hotspot server configuration
Configure SIP hotspot client:
As a SIP hotspot client, no SIP account needs to be set. The Phone set will automatically obtain and be configured a SIP account. On the SIP Hotspot tab page, set Mode to Client.
The values of other options are the same as those of the hotspot.
Picture 80 - SIP hotspot client configuration
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As the hotspot server, the default extension number is 0. When the phone is used as the client, the extension number is increased from 1, you can view the extension number through the [ SIP Hotspot ] page.
Call extension number:
The hotspot server and the client can dial each other through the extension number.
For example, extension 1 dials extension 0.
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10 Phone Settings
10.1 Basic Settings
10.1.1
Language
The user can set the phone language through the phone interface and web interface.
Phone interface ： After resetting the factory settings, the user needs to set the language; when setting the language during standby, go to [ Settings ]
[ Language&input ] Settings, as shown in the figure.
Picture 81 - Phone language setting
Web interface: Log in to the phone webpage and set the language in the drop-down box at the top right corner of the page, as shown in the figure:
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Picture 82 - Language setting on Web page
The function box on the right side of the web interface language setting box is
“Synchronize language to phone”; if selected, the phone language will be synchronized with the webpage language. If it is not selected, it will not be synchronized.
10.1.2
Time & Date
Users can set the phone time through the phone interface and web interface.
Phone interface: When the phone is in the default standby state, press the
[ Settings ] >> [ Time & Date ] to edit parameters, as shown in the figure:
Picture 83 - Set time & date on phone
Webpage: Log in to the phone webpage and enter [ Phone Settings ] >>
[ Time/Date ] , as shown in the figure:
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Parameters
Mode
SNTP Server
Time zone
Time format
Separator
Picture 84 - Set time & date on webpage
Table 12 - Time Settings Parameters
Description
Auto/Manual
Auto: Enable network time synchronization via SNTP protocol, default enabled.
Manual: User can modify data manually.
SNTP server address
Select the time zone
Select time format from one of the followings:
1 JAN, MON
1 January, Monday
JAN 1, MON
January 1, Monday
MON, 1 JAN
Monday, 1 January
MON, JAN 1
Monday, January 1
DD-MM-YY
DD-MM-YYYY
MM-DD-YY
MM-DD-YYYY
YY-MM-DD
YYYY-MM-DD
Choose the separator between year and moth and day
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12-Hour Clock
Daylight Saving Time
Display the clock in 12-hour format
Enable or Disable the Daylight Saving Time
10.1.3
Screen
The user can adjust the brightness of phone screen in LCD in two ways.
Slide down the outgoing status bar page in standby mode. Slide down again to adjust phone brightness conveniently.
Enter the [ Settings ] >> [ Display ], and then adjust the brightness.
Picture 85 - Set screen parameters on phone
10.1.3.1
Screen Saver
When the phone is in default standby state, press the function menu [ settings ]>>
[ Display ] to enable the screen protection, as shown in the figure below:
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Picture 86 - Phone screen saver
10.1.4
Ring
When the device is in the default standby mode,
Enter[ Settings ] >> [ Sound ] >> [ Tone ] set promote tone
The prompt tone contains Settings such as caller ring, notification ring, touch prompt tone, etc.
10.1.5
Voice Volume
When the device is in the default standby mode,
Enter [ Settings ] >> [ Sound ] item till you find [ Volume ] item.
The prompt tone contains Settings such as caller ring, notification ring, touch prompt tone, etc.
10.1.6
Reboot
When the device is in the default standby mode,
Enter [ Settings ] >> [ Reboot ] item.
Click [ Reboot ] to indicate whether to restart the phone.
Press [ OK ] to restart the phone or press [ Cancel ] to exit the prompt box to return to the configuration interface.
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10.2 Phone book
10.2.1
Local contact
Users can save contact information in the phone book and dial the contact's phone number directly in the phone book.The user can open the phone book by pressing the function menu button "contacts" in the default main interface.
By default, the phone book is empty, and users can add manually or add contacts to the phone book from the call log (or cloud phone book).
NOTICE! The device can save up to total 2000 contact records.
Picture 87 - Local Phone book
When there are contact records in the phone book, the contact records will be arranged in the alphabet order. User may browse the contacts with Swipe screen. The user can directly click the contact to call, or click the exclamation mark icon on the right to view the contact details.
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10.2.1.1
Add / Edit / Delete Contact
Add a contact, click to enter the contact interface, select the first icon (contact icon, selected by default) and add the following contact information.
Contact Name
Tel. Number
Mobile Number
Other Number
Line
Ring Tone
Contact Group
Photo
Picture 88 - Add New Contact
User can edit a contact by pressing [ Edit ] button.
Delete contact: The user can enter the delete contact list by pressing the [Delete] icon, and click the select all button to select all the contacts that you want to delete, or the check box after the user selects a contact; then press the delete icon again You will be prompted to delete the selected contact.
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10.2.1.2
Add / Edit / Delete Group
By default, the group list is empty. Users can create their own group, edit group names, add or remove contacts from the group, and delete groups.
Add group. In the contact list interface, press the "group" icon to switch to the group list. Click add button again to enter the page of creating groups.
Delete groups, under groups list.
To edit the group, press edit.
Picture 89 - Group List
10.2.2
Black list
The device supports blacklist, such as the number added to the blacklist, the number of calls directly refused to the end, the end of the phone shows no incoming calls.
(Blacklisted Numbers can be called out normally)
There are multiple ways to add a number to Blacklist on the device. It can be added directly on [ Contacts ] icon >> [ Group ] icon>> [ Blacklist ].
Select any number in the phone book (both local and network) for configuration addition.
Select any number in the call log for configuration addition.
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Picture 90 - Add Blacklist
There are various ways to add number to the blacklist on web page, which can be added in the [ Phone book ] >> [ Call list ] >> [ Restricted Incoming Calls ].
Select any number in the phone book (both local and network) for configuration addition.
Select any number in the call log for configuration addition.
Picture 91 - Web Blacklist
10.2.3
Cloud Phone Book
10.2.3.1
Configure Cloud Phone book
Cloud phonebook allows user to configure the device by downloading a phonebook from a cloud server. This is convenient for office users to use the phonebook from a single source and save the effort to create and maintain the contact list individually. It is also a useful tool to synchronize his/her phonebook from a personal mobile phone to the device with Fanvil Cloud Phonebook Service and App which is to be provided publicly soon.
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NOTICE! The cloud phonebook is ONLY temporarily downloaded to the device each time when it is opened on the device to ensure the user get the latest phonebook. However, the downloading may take a couple seconds depending on the network condition. Therefore, it is highly recommended for the users to save important contacts from cloud to local phonebook for saving download time.
Open cloud phonebook list, press[ Contacts ] icon>> [ Network PhoneBook ] in phonebook screen.
TIPS! The first configuration on cloud phone should be completed on Web page by selecting [PhoneBook] >> [Cloud Contacts]. The setting of addition/deletion on device could be done after the first setting on Web page.
Picture 92 - Cloud phone book list
10.2.3.2
Downloading Cloud Phone book
In cloud phone book screen, user can open a cloud phone book by pressing the network phonebook. The device will start downloading the phone book. The user will be prompted with a warning message if the download fails,
Once the cloud phone book is downloaded completely, the user can browse the contact list and dial the contact number same as in local phonebook.
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Picture 93 - Browsing Contacts in Cloud Phone book
10.3 Call log
The device can store up to 2000 call log records and user can open the call logs to check all incoming, outgoing, and missed call records by pressing [Missed call] icon.
In the call logs screen, user may browse the call logs with up/down swipe.
Each call log record is presented with ‘call type’ and ‘call party number / name’. User can check further call log detail by pressing [ Info ] icon and dial the number with pressing the call log, or add the call log number to phonebook with pressing [ Info ] Icon >> [ Add to
Contact ].
User can delete a call log by pressing [ Delete ] button.
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Picture 94 - Call Log
Users can also filter the call records of specific call types to narrow down the scope of search records, and select a call record type by press the type icon.
- Missed Call Log
- Incoming Call Log
- Outgoing Call Log
- Forward Call Log
Picture 95 - Filter call record types
10.4 Function Key
Function key settings:
There are 5 shortcut keys displayed on the screen in standby mode, each of which can be customized by DSSkey.
Users can add/delete DSSkey pages through the webpage, and use the page switch key to switch DSSkey pages. In addition, users can also modify the settings of the corresponding keys by long pressing each shortcut key.
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Picture 96 - DSS LCD Screen Configuration
The DSS Key could be configured as followings,
Memory Key
Speed Dial/Intercom/BLF/Presence/Call Park/Call Forward (to someone)
Line
Key Event
MWI/DND/Hold/Transfer/Phonebook/Redial/Pickup/Call Forward (to specified line)/Headset/ SMS/Release
DTMF
Action URL
BLF List Key
MCAST Paging
MCAST Listening
Action URL
XML Browser
Moreover, user also can add the user-defined title for the DSS Keys, which is configured as Memory Key / Line / URL / MCAST Paging / Prefix.
NOTICE! User-defined title is up to 10 characters.
More detailed information refers to
6.2 Appendix III –LED
Definition .
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10.5 Wi-Fi
The device supports wireless Internet access and has built-in Wi-Fi without external devices.
When the device is in the default standby mode,
Pull down the menu, long press the [ WLAN ] icon,Enter [ Wi-Fi ] item.
Enable the Wi-Fi to search the current wireless network automatically.
Select to the available network, enter the user name and password to connect successfully.
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Picture 97 - WIFI settings
10.6 Headset
10.6.1
Bluetooth Headset
The device supports Bluetooth applications and can be compatible with Bluetooth headsets with CSR 4.0 chips. There is no need to use a USB Bluetooth adapter. The phone has a built-in Bluetooth and Bluetooth antenna.
The phone is in the default standby state,
Long press the function menu button [Bluetooth], click the switch to turn on Bluetooth,
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and it will automatically search for available devices.
Select the device to pair and connect.
Picture 98 - Bluetooth Settings Screen
The use of Bluetooth headset can be divided into three types: call answering; Hang up;
Bluetooth redial.
call answering
When the Bluetooth headset is connected to the phone, the incoming call can be answered by pressing the Bluetooth answer button.
Hang up
1) When talking with Bluetooth headset, you can hang up the phone by pressing the button on Bluetooth headset.
2) When there is an incoming call, double-click the answer button to reject the call.
3) When the caller is in the ringing state, press the answer button of the headset to cancel the call.
Bluetooth redial
When the Bluetooth headset is connected, double-click the answer button to redial the number dialed last time.
NOTICE! some models do not support double - click redial function. Whether this function is supported or not, you can check the instruction of the headset, or connect the Bluetooth headset to the phone, and double-click the answer button to see whether it will redial.
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10.7 Advanced
10.7.1
Line Configurations
Phone access [ settings ] >> [ Account ] >> [ Line ], select [ Register Account] to configure the SIP line on the phone.
Picture 99 - SIP address and account information
For users who want to configure more options, user should use web management portal to modify or [ More Register Settings ] in accounts on the individual line to configure those options.
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Picture 100 - Configure Advanced Line Options
10.7.2
Network Settings
10.7.2.1
Network Settings
Phone access [ Settings ] >>[ Advanced ] >>[ Network ] >> [ Ethernet ], you can configure the SIP line on the phone.
There are 2 connection mode options: DHCP, Static IP.
Picture 101 - DHCP network mode
When using DHCP mode, phone will get the IP address from DHCP server (router).
Obtain DNS Server automatically: It is enabled as default. “Enable” means phone will get DNS address from DHCP server and “disable” means not.
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Picture 102 - Static IP network mode
When using Static IP mode, user must configure the IP address manually.
IP Address ： Phone IP address.
Subnet Mask: sub mask of your LAN.
IP Gateway ： The gateway IP address. Phone could access the other network via it.
Primary DNS ： Primary DNS address. The default is 8.8.8.8, Google DNS server address.
Secondary DNS ： Secondary DNS. When primary DNS is not available, it will work.
10.7.2.2
QoS & VLAN
Access [ Settings ]>>[Advanced]>> [ Network ]>> [ Advance ]
LLDP
Link Layer Discovery Protocol. LLDP is a vendor independent link layer protocol used by network devices for advertising their identity, capabilities to neighbors on a LAN segment.
Phone could use LLDP to find the VLAN switch or other VLAN devices and use LLDP to learn feature to apply the VLAN ID from VLAN switch to phone its self.
CDP
Cisco Discovery Protocol. CDP is a not-for-profit charity that runs the global disclosure system for investors, companies, cities, states and regions to manage their environmental impacts. According to the CDP, Cisco devices could share the OS version,
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Parameters
LLDP setting
Report
Interval
Learning
QoS
QoS Mode
WAN VLAN
WAN VLAN
LAN VLAN
LAN VLAN
CDP
CDP
IP address, hardware version and so on.
Table 13 - QoS & VLAN
Description
Enable LLDP
LLDP requests interval time apply the learned VLAN ID to the phone configuration configure SIP DSCP and audio DSCP
WAN port VLAN configuration
LAN port VLAN configuration
CDP enable/disable ， CDP interval time
10.7.2.3
Web Server Type
Access [ Settings ]>>[ Advanced ] >>[ Network ]>> [ Service Port ] to configure the Web
Server mode.
Configure the Web Server mode to be HTTP or HTTPS and will be activated after the reboot. Then user could use http/https protocol to access pone web page.
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10.7.3
Picture 103 - The phone configures the web server type
Set The Secret Key
When the device is in the default standby mode,
Select [ Settings ]>> [ Advanced]>> [ Password ]
Click [ Password ] to change password.
Picture 104 - Menu password and Settings
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10.7.4
Maintenance
Phone Webpage: Login and go to [ System ] >> [ Auto provision ].
Picture 105 - Page auto provision Settings
LCD ： Enter [ Settings ] >> [Advanced] >> [ Maintain ] >> [ Auto Provision ].
Picture 106 - Phone auto provision settings
Fanvil devices support SIP PnP, DHCP options, Static provision, TR069. If all of the 4 methods are enabled, the priority from high to low as below:
PNP>DHCP>TR069> Static Provisioning
Transferring protocol: FTP 、 TFTP 、 HTTP 、 HTTPS
Details refer to Fanvil Auto Provision in
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http://www.fanvil.com/Uploads/Temp/download/20180920/5ba3816f8d5f0.pdfhttps://ww w.fanvil.com.cn/Support/download/cid/14.html
Table 14 - Auto Provision
Parameters
Basic settings
CPE Serial Number
Authentication Name
Description
Display the device SN
The user name of provision server
Authentication Password The password of provision server
Configuration File If the device configuration file is encrypted , user should add
Encryption Key the encryption key here
General Configuration File
Encryption Key
Download Fail Check
Times
Update Contact Interval
Save Auto Provision
Information
Download Common
Config enabled
If the common configuration file is encrypted, user should add the encryption key here
If there download is failed, phone will retry with the configured times.
Phone will update the phonebook with the configured interval time. If it is 0, the feature is disabled.
Save the HTTP/HTTPS/FTP user name and password. If the provision URL is kept, the information will be kept.
Whether phone will download the common configuration file.
Enable Server Digest
When the feature is enable, if the configuration of server is changed, phone will download and update.
DHCP Option
Option Value
Custom Option Value
Enable DHCP Option 120
Confiugre DHCP option, DHCP option supports DHCP custom option | DHCP option 66 | DHCP option 43, 3 methods to get the provision URL. The default is Option 66.
Custom Option value is allowed from 128 to 254. The option value must be same as server define.
Use Option120 to get the SIP server address from DHCP server.
SIP Plug and Play (PnP)
Enable SIP PnP
Whether enable PnP or not. If PnP is enable, phone will send a SIP SUBSCRIBE message with broadcast method. Any server can support the feature will respond and send a Notify with URL to phone. Phone could get the configuration file with
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Server Address
Server Port
Transport Protocol the URL.
Broadcast address. As default, it is 224.0.0.0.
PnP port
PnP protocol, TCP or UDP.
Update Interval
Static Provisioning Server
PnP message interval.
Server Address
Provisioning server address. Support both IP address and domain address.
Configuration File Name
Protocol Type
Update Interval
Update Mode
The configuration file name. If it is empty, phone will request the common file and device file which is named as its MAC address.
The file name could be a common name, $mac.cfg, $input.cfg.
The file format supports CFG/TXT/XML.
Transferring protocol type ， supports FTP 、 TFTP 、 HTTP and
HTTPS
Configuration file update interval time. As default it is 1, means phone will check the update every 1 hour.
Provision Mode.
1. Disabled.
2. Update after reboot.
3. Update after interval.
TR069
Enable TR069
ACS Server Type
ACS Server URL
ACS User
ACS Password
Enable TR069 Warning
Tone
Enable TR069 after selection
There are 2 options Serve type, common and CTC.
ACS server address
ACS server username (up to is 59 character)
ACS server password (up to is 59 character)
If TR069 is enabled, there will be a prompt tone when connecting.
TLS Version TLS version (TLS 1.0, TLS 1.1, TLS 1.2)
INFORM Sending Period INFORM signal interval time. It ranges from 1s to 999s
STUN Server Address
STUN Enable
Configure STUN server address
To enable STUN server for TR069
Table 15 - Auto Provision
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10.7.5
Firmware Upgrade
Web page ： Login phone web page, go to [ System ] >> [ Upgrade ].
Picture 107 - Web page firmware upgrade
LCD interface: go to [ Settings ] >> [Advanced] >>[Maintain] >>[ Upgrade ] .
Parameter
Upgrade server
Picture 108 - Firmware upgrade information display
Table 16 - Firmware upgrade
Description
Enable Auto Upgrade
Enable automatic upgrade, If there is a new version txt and new software firmware on the server, phone will
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Upgrade Server Address1
Upgrade Server Address2
Update Interval
Firmware Information
Current Software Version
Server Firmware Version
[Upgrade] button
New version information show a prompt upgrade message after Update Interval.
Set available upgrade server address.
Set available upgrade server address.
Set Update Interval.
description
It will show Current Software Version.
It will show Server Firmware Version.
If there is a new version txt and new software firmware on the server, the page will display version information and upgrade button will become available; Click
[Upgrade] button to upgrade the new firmware.
When there is a corresponding TXT file and version on the server side, the TXT and version information will be displayed under the new version description information.
The file requested from the server is a TXT file called vendor_model_hw10.txt.Hw
followed by the hardware version number, it will be written as hw10 if no difference on hardware. All Spaces in the filename are replaced by underline.
The URL requested by the phone is HTTP:// server address/vendor_Model_hw10
.txt
： The new version and the requested file should be placed in the download directory of the HTTP server, as shown in the figure:
TXT file format must be UTF-8
vendor_model_hw10.TXT The file format is as follows ：
Version=1.6.3 #Firmware
Firmware=xxx/xxx.z #URL ， Relative paths are supported and absolute paths are possible, distinguished by the presence of protocol headers.
BuildTime=2018.09.11 20:00
Info=TXT|XML
Xxxxx
Xxxxx
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Xxxxx
Xxxxx
After the interval of update cycle arrives, if the server has available files and versions, the phone will prompt as shown below. Click [view] to check the version information and upgrade.
10.7.6
Picture 109 - Firmware upgrade
Factory Reset
The phone is in default standby mode.
Press [ Settings ] to find [Advanced]>> [ Maintain ]>> [ Phone Reset ].
Press [ reset ] to clear all. When you select clear configuration file and clear all, the phone will restart automatically after clearing.
Picture 110 - Reset to default
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11 Web Configurations
11.1 Web Page Authentication
The user can log into the web page of the phone to manage the user's phone information and operate the phone. Users must provide the correct user name and password to log in.
11.2 System >> Information
User can get the system information of the device in this page including,
Model
Hardware Version
Software Version
Uptime
And summarization of network status,
Network Mode
MAC Address
IP
Subnet Mask
Default Gateway
Besides, summarization of SIP account status,
SIP User
SIP account status (Registered / Unapplied / Trying / Timeout )
11.3 System >> Account
On this page the user can change the password for the login page.
Users with administrator rights can also add or delete users, manage users, and set permissions and passwords for new users.
11.4 System >> Configurations
On this page, users with administrator privileges can view, export, or import the phone configuration, or restore the phone to factory Settings.
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Clear Configurations
Select the module in the configuration file to clear.
SIP: account configuration.
AUTOPROVISION: automatically upgrades the configuration
TR069:TR069 related configuration
MMI: MMI module, including authentication user information, web access protocol, etc.
DSS Key: DSS Key configuration
Clear Tables
Select the local data table to be cleared, all selected by default.
Reset Phone
The phone data will be cleared, including configuration and database tables.
11.5 System >> Upgrade
Upgrade the phone software version, customized ringtone, can also be upgraded to delete the file. Ring tone support “.wav” format.
11.6 System >> Auto Provision
The Auto Provision settings help IT manager or service provider to easily deploy and manage the devices in mass volume. For the detail of Auto Provision, please refer to this link Auto Provision Description 。 http://www.fanvil.com/Uploads/Temp/download/20180920/5ba3816f8d5f0.pdf
11.7 System >> Tools
Tools provided in this page help users to identify issues at trouble shooting. Please refer
to 13 Trouble Shooting for more detail.
11.8 System >> Reboot Phone
This page can restart the phone.
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12 Network >> Basic
This page allows users to configure network connection types and parameters.
12.1 Network >> Service Port
This page provides settings for Web page login protocol, protocol port settings and RTP port.
Parameter
Web Server Type
Web Logon Timeout
Web auto login
HTTP Port
HTTPS Port
RTP Port Range Start
RTP Port Quantity
Picture 111 - Service Port Settings
Description
Reboot to take effect after settings. Optionally, the web page login is HTTP/HTTPS.
Default as 15 minutes, the timeout will automatically exit the login page, need to login again.
After the timeout does not need to enter a user name password, will automatically login to the web page.
The default is 80. If you want system security, you can set ports other than 80.
Such as :8080, webpage login: HTTP://ip:8080
The default is 443, the same as the HTTP port.
The value range is 1025 to 65535. The value of
RTP port starts from the initial value set. For each call, the value of voice and video port is added 2.
Number of calls.
Table 17 - Service port
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12.2 Network >> Advanced
Advanced network Settings are typically configured by the IT administrator to improve
the quality of the phone service. For configuration, query the 10.7 advanced
Settings.
12.3 Line >> SIP
Configure the Line service configuration on this page.
Parameters
Register Settings
Line Status
Table 18 - Line configuration on the web page
Description
Activate
Username
Authentication User
Display Name
Authentication Password
Realm
Display the current line status at page loading.
To get the up to date line status, user has to refresh the page manually.
Whether the service of the line is activated
Enter the username of the service account.
Enter the authentication user of the service account
Enter the display name to be sent in a call request.
Enter the authentication password of the service account
Enter the SIP domain if requested by the service provider
Input server name.
Server Name
SIP Server 1
Server Address
Server Port
Transport Protocol
Enter the IP or FQDN address of the SIP server
Enter the SIP server port, default is 5060
Set up the SIP transport line using TCP or UDP or TLS.
Set SIP expiration date.
Registration Expiration
SIP Server 2
Server Address
Server Port
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Enter the IP or FQDN address of the SIP server
Enter the SIP server port, default is 5060
Transport Protocol
Registration Expiration
SIP Proxy Server Address
Proxy Server Port
Proxy User
Proxy Password
Backup Proxy Server Address
Backup Proxy Server Port
Basic Settings
Enable Auto Answering
Auto Answering Delay
Call Forward Unconditional
Call Forward Number for Unconditional
Call Forward on Busy
Call Forward Number for Busy
Call Forward on No Answer
Call Forward Number for No Answer
Call Forward Delay for No Answer
Transfer Timeout
Conference Type
Set up the SIP transport line using TCP or UDP or TLS.
Set SIP expiration date.
Enter the IP or FQDN address of the SIP proxy server.
Enter the SIP proxy server port, default is 5060.
Enter the SIP proxy user.
Enter the SIP proxy password.
Enter the IP or FQDN address of the backup proxy server.
Enter the backup proxy server port, default is
5060.
Enable auto-answering, the incoming calls will be answered automatically after the delay time
Set the delay for incoming call before the system automatically answered it
Enable unconditional call forward, all incoming calls will be forwarded to the number specified in the next field
Set the number of unconditional call forward
Enable call forward on busy, when the phone is busy, any incoming call will be forwarded to the number specified in the next field.
Set the number of call forward on busy .
Enable call forward on no answer, when an incoming call is not answered within the configured delay time, the call will be forwarded to the number specified in the next field.
Set the number of call forward on no answer.
Set the delay time of not answered call before being forwarded.
Set the timeout of call transfer process.
Set the type of call conference, Local=set up call conference by the device itself, maximum supports two remote parties, Server=set up call conference by dialing to a conference room on the server
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Server Conference Number
Subscribe For Voice Message
Voice Message Number
Voice Message Subscribe Period
Enable Hotline
Hotline Delay
Hotline Number
Dial Without Registered
Enable Missed Call Log
DTMF Type
DTMF SIP INFO Mode
Enable DND
Subscribe For Voice Message
Use VPN
Use STUN
Enable Failback
Failback Interval
Signal Failback
Signal Retry Counts
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Set the conference room number when conference type is set to be Server
Enable the device to subscribe a voice message waiting notification, if enabled, the device will receive notification from the server if there is voice message waiting on the server
Set the number for retrieving voice message
Set the interval of voice message notification subscription
Enable hotline configuration, the device will dial to the specific number immediately at audio channel opened by off-hook handset or turn on hands-free speaker or headphone
Set the delay for hotline before the system automatically dialed it
Set the hotline dialing number
Set call out by proxy without registration
If enabled, the phone will save missed calls into the call history record.
Set the DTMF type to be used for the line
Set the SIP INFO mode to send ‘*’ and ‘#’ or ‘10’ and ‘11’
Enable Do-not-disturb, any incoming call to this line will be rejected automatically
Enable the device to subscribe a voice message waiting notification, if enabled, the device will receive notification from the server if there is voice message waiting on the server
Set the line to use VPN restrict route
Set the line to use STUN for NAT traversal
Whether to switch to the primary server when it is available.
A Register message is used to periodically detect the time interval for the availability of the main Proxy.
Multiple proxy cases, whether to allow the invite/register request to also execute failback.
The number of attempts that the SIP Request
Codecs Settings
Video Codecs
Advanced Settings
Use Feature Code
Enable DND
Disable DND
Enable Call Forward Unconditional
Disable Call Forward Unconditional
Enable Call Forward on Busy
Disable Call Forward on Busy
Enable Call Forward on No Answer
Disable Call Forward on No Answer
Enable Blocking Anonymous Call
Disable Blocking Anonymous Call
Call Waiting On Code
Call Waiting Off Code
Send Anonymous On Code
Send Anonymous Off Code
SIP Encryption
RTP Encryption
Enable Session Timer
Session Timeout
Enable BLF List
BLF List Number considers proxy unavailable under multiple proxy scenarios.
Set the priority and availability of the codecs by adding or remove them from the list.
Select video code to preview video.
When this setting is enabled, the features in this section will not be handled by the device itself but by the server instead. In order to control the enabling of the features, the device will send feature code to the server by dialing the number specified in each feature code field.
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Set the feature code to dial to the server
Enable SIP encryption such that SIP transmission will be encrypted
Enable RTP encryption such that RTP transmission will be encrypted
Set the line to enable call ending by session timer refreshment. The call session will be ended if there is not new session timer event update received after the timeout period
Set the session timer timeout period
Enable/Disable BLF List
BLF List allows one BLF key to monitor the status of a group. Multiple BLF lists are
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Response Single Codec
BLF Server
Keep Alive Type
Keep Alive Interval
Keep Authentication
Blocking Anonymous Call
User Agent
Specific Server Type
SIP Version
Anonymous Call Standard
Local Port
Ring Type
Enable user=phone
Use Tel Call
Auto TCP
Enable Rport
Enable PRACK
DNS Mode
Enable Long Contact
Enable Strict Proxy
Convert URI
67 supported.
If setting enabled, the device will use single codec in response to an incoming call request
The registered server will receive the subscription package from ordinary application of BLF phone.
Please enter the BLF server, if the sever does not support subscription package, the registered server and subscription server will be separated.
Set the line to use dummy UDP or SIP OPTION packet to keep NAT pinhole opened
Set the keep alive packet transmitting interval
Keep the authentication parameters from previous authentication
Reject any incoming call without presenting caller ID
Set the user agent, the default is Model with
Software Version.
Set the line to collaborate with specific server type
Set the SIP version
Set the standard to be used for anonymous
Set the local port
Set the ring tone type for the line
Sets user=phone in SIP messages.
Set use tel call
Using TCP protocol to guarantee usability of transport for SIP messages above 1500 bytes
Set the line to add rport in SIP headers
Set the line to support PRACK SIP message
Select DNS mode, A, SRV, NAPTR
Allow more parameters in contact field per RFC
3840
Enables the use of strict routing. When the phone receives packets from the server ， it will use the source IP address, not the address in via field.
Convert not digit and alphabet characters to
Use Quote in Display Name
Enable GRUU
Sync Clock Time
Enable Inactive Hold
Caller ID Header
Use 182 Response for Call waiting
Enable Feature Sync
Enable SCA
CallPark Number
Server Expire
TLS Version uaCSTA Number
Enable Click To Talk
Enable Chgport
VQ Name
VQ Server
VQ Port
VQ HTTP/HTTPS Server
Flash mode
Flash Info Content-Type
Flash Info Content-Body
PickUp Number
JoinCall Number
Intercom Number
Unregister On Boot
Enable MAC Header
Enable Register MAC Header
%hh hex code
Whether to add quote in display name, i.e.
“Fanvil” vs Fanvil
Support Globally Routable User-Agent URI
(GRUU)
Time Sync with server
With the post-call hold capture package enabled, you can see that in the INVITE package, SDP is inactive.
Set the Caller ID Header
Set the device to use 182 response code at call waiting response
Feature Sync with server
Enable/Disable SCA (Shared Call Appearance )
Set the CallPark number.
Set the timeout to use the server.
Choose TLS Version.
Set uaCSTA Number.
With the use of special server, click to call out directly after enabling.
Whether port updates are enabled.
Open the VQ name for VQ RTCP-XR.
Open VQ server address for VQ RTCP-XR.
Open VQ port for VQ RTCP-XR.
Enable VQ server selection for VQ RTCP-XR.
Chose Flash mode ， normal or SIP info.
Set the SIP info content type.
Set the SIP info content body.
Set the scramble number when the Pickup is enabled.
Set JoinCall Number.
Set Intercom Number.
Whether to enable logout function.
Whether to open the registration of SIP package with user agent with MAC or not.
Whether to open the registration is user agent with MAC or not.
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BLF Dialog Strict Match
PTime(ms)
SIP Global Settings
Strict Branch
Enable Group
Enable RFC4475
Enable Strict UA Match
Registration Failure Retry Time
Local SIP Port
Enable uaCSTA
12.4 Line >> SIP Hotspot
Please refer to 9.9 SIP Hotspot.
12.5 Line >> Dial Plan
Whether to enable accurate matching of BLF sessions.
Set whether to bring ptime field, default no.
Set up to strictly match the Branch field.
Set open group.
Set to enable RFC4475.
Enable strict UA matching.
Set the registration failure retry time.
Modify the phone SIP port.
Set to enable the uaCSTA function.
Parameters
Press # to invoke dialing
Dial Fixed Length
Timeout dial
Picture 112 - Dial plan settings
Table 19 - Phone 7 dialing methods
Press # to Do Blind Transfer
Description
The user dials the other party's number and then adds the # number to dial out;
The number entered by the user is automatically dialed out when it reaches a fixed length
The system dials automatically after timeout
The user enters the number to be transferred and then presses the "#" key to transfer the current call to a third party
69
Blind Transfer on Onhook
Attended Transfer on Onhook
Attended Transfer on Conference Onhook
Enable E.164
Add dialing rules:
After the user enters the number, hang up the handle or turn off the hands-free function to transfer the current call to a third party.
Hang up the handle or press the hands-free button to realize the function of attention
-transfer, which can transfer the current call to a third party.
During a three-way call, hang up the handle and the remaining two parties remain on the call.
Please refer to e. 164 standard specification
Parameters
Dial rule
Picture 113 - Custom setting of dial - up rules
Table 20 - Dial - up rule configuration table
Description
There are two types of matching: Full Matching or Prefix Matching. In Full matching, the entire phone number is entered and then mapped per the Dial Peer rules.
In prefix matching, only part of the number is entered followed by T. The mapping with then take place whenever these digits are dialed.
Prefix mode supports a maximum of 30 digits.
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Note: Two different special characters are used.
x -- Matches any single digit that is dialed.
[ ] -- Specifies a range of numbers to be matched. It may be a range, a list of ranges separated by commas, or a list of digits.
Destination Set Destination address. This is for IP direct.
Port
Alias
Set the Signal port, and the default is 5060 for
SIP.
Set the Alias. This is the text to be added, replaced or deleted. It is an optional item.
Note: There are four types of aliases.
all: xxx – xxx will replace the phone number.
add: xxx – xxx will be dialed before any phone number.
del –The characters will be deleted from the phone number.
rep: xxx – xxx will be substituted for the specified characters.
Suffix Characters to be added at the end of the phone number. It is an optional item.
Length Set the number of characters to be deleted. For example, if this is set to 3, the phone will delete the first 3 digits of the phone number. It is an optional item.
This feature allows the user to create rules to make dialing easier. There are several different options for dialing rules. The examples below will show how this can be used.
Example 1 : All Substitution -- Assume that it is desired to place a direct IP call to IP address 172.168.2.208. Using this feature, 123 can be substituted for 172.168.2.208.
Picture 114 - Dial rules table (1)
Example 2 : Partial Substitution -- To dial a long distance call to Beijing requires dialing area code 010 before the local phone number. Using this feature 1 can be substituted for 010. For example, to call 62213123 would only require dialing 162213123 instead of
01062213123.
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Picture 115 - Dial rules table (2)
Example 3 : Addition -- Two examples are shown. In the first case, it is assumed that 0 must be dialed before any 11 digit number beginning with 13. In the second case, it is assumed that 0 must be dialed before any 11 digit number beginning with 135, 136, 137,
138, or 139. Two different special characters are used.
x -- Matches any single digit that is dialed.
[] -- Specifies a range of numbers to be matched. It may be a range, a list of ranges separated by commas, or a list of digits.
12.6 Line >> Action Plan
When calling to a phone, the bounded IP camera synchronously transmits video to the opposite phone (video support).
Table 21 - IP camera
Parameter
Number
Type
Direction
Line
Username
Password
URL
User Agent
Description
Auxiliary phone number (support video)
Support video display on call.
For call mode, incoming/outgoing call displays video
Set up outgoing lines.
Bind the user name of the IP camera.
Bind IP camera password.
Video streaming information.
Set user agent information
12.7 Line >> Basic Settings
Set up the register global configuration.
Table 22 - Set the line global configuration on the web page
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Parameters
STUN Settings
Server Address
Server Port
Binding Period
SIP Waiting Time
Certification File
TLS Certification File
Description
Set the STUN server address
Set the STUN server port, default is 3478
Set the STUN binding period which can be used to keep the NAT pinhole opened.
Set the timeout of STUN binding before sending
SIP messages
Upload or delete the TLS certification file used for encrypted SIP transmission.
12.8 Phone settings >> Features
Configuration phone features.
Table 23 - General function Settings
Parameters
Basic Settings
Enable Call Waiting
Enable Call Transfer
Semi-Attended Transfer
Enable 3-Way Conference
Enable Auto Onhook
Auto Onhook Time
Ring for Headset
Auto Headset
Description
Enable this setting to allow user to take second incoming call during an established call. Default enabled.
Enable Call Transfer.
Enable Semi-Attended Transfer by selecting it
Enable 3-way conference by selecting it
The phone will hang up and return to the idle automatically at hands-free mode
Specify Auto Onhook time, the phone will hang up and return to the idle automatically after Auto
Hand down time at hands-free mode, and play dial tone Auto Onhook time at handset mode
Enable Ring for Handset by selecting it, the phone plays ring tone from handset.
Enable this feature, headset plugged in the phone, user press ‘answer’ key or line key to answer a call with the headset automatically.
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Enable Silent Mode
Disable Mute for Ring
Enable Default Line
Enable Auto Switch Line
Default Ext Line
Ban Outgoing
Hide DTMF
Enable CallLog
Enable Restricted Incoming List
Enable Allowed Incoming List
Enable Restricted Outgoing List
Enable Country Code
Country Code
Area Code
Enable Number Privacy
Match Direction
Start Position
Hide Digits
Allow IP Call
P2P IP Prefix
Caller Name Priority
Emergency Call Number
Search path
LDAP Search
Emergency Call Number
Restrict Active URI Source IP
When enabled, the phone is muted, there is no ringing when calls, you can use the volume keys and mute key to unmute.
When it is enabled, you can’t mute the phone
If enabled, user can assign default SIP line for dialing out rather than SIP1.
Enable phone to select an available SIP line as default automatically
Select the default line to use for outgoing calls
If you select Ban Outgoing to enable it, and you cannot dial out any number.
Configure the hide DTMF mode.
Select whether to save the call log.
Whether to enable restricted call list.
Whether to enable the allowed call list.
Whether to enable the restricted allocation list.
Whether the country code is enabled.
Fill in the country code.
Fill in the area code.
Whether to enable number privacy.
Matching direction, there are two kinds of rules from right to left and from left to right.
Open number privacy after the start of the hidden location.
Turn on number privacy to hide the number of digits.
If enabled, user can dial out with IP address
Prefix a point-to-point IP call.
Change caller ID display priority.
Select the search path.
Select from with one LDAP for search
Configure the Emergency Call Number. Despite the keyboard is locked, you can dial the emergency call number
Set the device to accept Active URI command from specific IP address. More details please refer to this link
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Push XML Server
Enable Pre-Dial
Enable Multi Line
Line Display Format
Contact As White List Type
Block XML When Call
SIP notify
Tone Settings
Enable Holding Tone
Enable Call Waiting Tone
Play Dialing DTMF Tone
Play Talking DTMF Tone
DND Settings
DND Option
Enable DND Timer
DND Start Time
DND End Time
Intercom Settings
Enable Intercom
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Configure the Push XML Server, when phone receives request, it will determine whether to display corresponding content on the phone which sent by the specified server or not.
Disable this feature, user enter number will open audio channel automatically.
Enable the feature, user enter the number without opening audio channel.
If enabled, up to 10 simultaneous calls can exist on the phone, and if disabled, up to 2 simultaneous calls can exist on the phone.
Custom line format ： SIPn/SIPn ： xxx/xxx@SIPn
NONE/BOTH/DND White List/FWD White List
Disable XML push on call.
When enabled, the phone displays the information when it receives the relevant notify content.
When turned on, a tone plays when the call is held
When turned on, a tone plays when call waiting
Play DTMF tone on the device when user pressed a phone digits at dialing, default enabled.
Play DTMF tone on the device when user pressed a phone digits during taking, default enabled.
Select to take effect on the line or on the phone or close.
Enable DND Timer, If enabled, the DND is automatically turned on from the start time to the off time.
Set DND Start Time
Set DND End Time
When intercom is enabled, the device will accept
Enable Intercom Mute
Enable Intercom Tone
Enable Intercom Barge
Response Code Settings
DND Response Code
Busy Response Code
Reject Response Code
Password Dial Settings
Enable Password Dial
Encryption Number Length
Password Dial Prefix
Power LED
Common
SMS/MWI
Missed
Talk/Dial the incoming call request with a SIP header of
Alert-Info instruction to automatically answer the call after specific delay.
Enable mute mode during the intercom call
If the incoming call is intercom call, the phone plays the intercom tone
Enable Intercom Barge by selecting it, the phone auto answers the intercom call during a call. If the current call is intercom call, the phone will reject the second intercom call
Set the SIP response code on call rejection on
DND
Set the SIP response code on line busy
Set the SIP response code on call rejection
Enable Password Dial by selecting it, When number entered is beginning with the password prefix, the following N numbers after the password prefix will be hidden as *, N stands for the value which you enter in the Password
Length field. For example: you set the password prefix is 3, enter the Password Length is 2, then you enter the number 34567, it will display 3**67 on the phone.
Configure the Encryption Number length
Configure
number the prefix of the password call
Standby power lamp state, off when off, open is always bright red. Off by default.
The status of power lamp when there is unread short message/voice message, including off/on/slow flash/quick flash, default slow flash.
The state of the power lamp when there is a missed call, including off/on/slow flash/quick flash, the default slow flash.
In the talk/dial state, the power lamp state, off is
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Ringing
Mute
Hold/Held
Notification Popups
Display Missed Call Popup
Display MWI Popup
Display Device Connect Popup
Display SMS Popup
Display Other Popup off, on is always red bright, the default is off.
Power lamp status when there is an incoming call, including off/on/slow flash/quick flash, default flash.
Power lamp status in mute mode, including off/on/slow flash/quick flash, off by default.
The power lamp state, including off/on/slow flash/quick flash, is turned off by default when left/retained.
No incoming call popup prompt after opening, no popup prompt when closing, open by default.
Voice message popup prompt is not answered after opening, and it is opened by default if there is no popup prompt when closing.
There is a popup prompt when the WIFI adapter is connected. There is no popup prompt when the WIFI adapter is closed. It is on by default.
There is popup prompt for unread messages after opening, and there is no popup prompt when closing. It is opened by default.
When the handle is not hung back after opening, registration fails, IP acquisition fails, Tr069 connection fails and other abnormalities, there will be popup prompt when it is opened; otherwise, there will be no prompt when it is closed, and it will be opened by default.
12.9 Phone settings >> Media Settings
Change voice Settings.
Parameter
Codecs Settings
Table 24 - Voice settings
Description
Select enable or disable voice encoding:
G.711A/U,G.722,G.729,
G.726-16,G726-24,G726-32,G.726-40,
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Video codec
Video codec
Media Setting
DTMF Payload Type
Headset Mic Gain
Opus playload type
OPUS Sample Rate
ILBC Payload Type
ILBC Payload Length
Onhook Time
Enable the patting spring to generate Flash
Video bit rate
Video frame rate
Video resolution
H.264Payload Type
Display splicing frame
RTP Control Protocol(RTCP) Settings
CNAME user
CNAME host
RTP Settings
RTP keep alive
Alert Info Ring Settings
Value
Ring Type
ILBC,opus
Select to enable video encoding:H264
Enter the DTMF payload type, the value must be
96~127.
Set the earphone's radio volume gain to fit different models of earphones.
Set Opus load type, range 96~127.
Set Opus sampling rate, including opus-nb (8KHz) and opus-wb (16KHz).
Set the ILBC Payload Type, the value must be
96~127.
Set the ILBC Payload Length
Configure a minimum response time, which defaults to 200ms
Whether to turn on the plug spring to generate
Flash
Set the bit rate of video:64kbps ， 192kbps ，
256kbps ， 384kbps ， 512kbps ， 768kbps ， 1Mbps ，
1.6Mbps
， 2Mbps ， 3Mbps ， 4Mbps
Set the video frame rate ： 5fps ， 10fps ， 15fps ，
20fps ， 25fps ， 30fps
Set Video resolution ： CIF,VGA,4CIF,720P
Set the H264 Payload Type ， the value must be
96~127.
Whether to start displaying splicing frames
Set CNAME user
Set CNAME host
Hold the call and send the packet after 30s
Set the value to specify the ring type.
Type1-Type9
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12.10 Phone settings >> MCAST
This feature allows user to make some kind of broadcast call to people who are in multicast group. User can configure a multicast DSS Key on the phone, which allows user to send a Real Time Transport Protocol (RTP) stream to the pre-configured multicast address without involving SIP signaling. You can also configure the phone to receive an RTP stream from pre-configured multicast listening address without involving
SIP signaling. You can specify up to 10 multicast listening addresses.
Parameters
Normal Call Priority
Enable Page Priority
Name
Host: port
Table 25 - Multicast parameters
Description
Define the priority of the active call, 1 is the highest priority, 10 is the lowest.
The voice call in progress shall take precedence over all incoming paging calls.
Listened multicast server name
Listened multicast server’s multicast IP address and port.
12.11 Phone settings >> Action
Action URL
Note! Action urls are used for IPPBX systems to submit phone events. Please refer to Fanvil Action URL for details.
12.12 Phone settings >> Time/Date
The user can configure the time Settings of the phone on this page.
Parameters
Table 26 – Time & Date settings
Description
Network Time Server Settings
Time Synchronized via SNTP
Time Synchronized via DHCP
Primary Time Server
Secondary Time Server
Enable time-sync through SNTP protocol
Enable time-sync through DHCP protocol
Set primary time server address
Set secondary time server address, when
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Time Zone
Resync Period
12-Hour Clock
Date Format
Daylight Saving Time Settings
Local
DST Set Type
Fixed Type
Offset
Month Start
Week Start
Weekday Start
Hour Start
Minute Start
Month End
Week End
Weekday End
Hour End
Minute End
Manual Time Settings primary server is not reachable, the device will try to connect to secondary time server to get time synchronization.
Select the time zone
Time of re-synchronization with time server
Set the time display in 12-hour mode
Select the time/date display format
Choose your local, phone will set daylight saving time automatically based on the local
Choose DST Set Type, if Manual, you need to set the start time and end time.
Daylight saving time rules are based on specific dates or relative rule dates for conversion.
Display in read-only mode in automatic mode.
The offset minutes when DST started
The DST start month
The DST start week
The DST start weekday
The DST start hour
The DST start minute
The DST end month
The DST end week
The DST end weekday
The DST end hour
The DST end minute
You can set your time manually
12.13 Phone settings >> Tone
This page allows users to configure a phone prompt.
You can either select the country area or customize the area. If the area is selected, it will bring out the following information directly. If you choose to customize the area, you can modify the button tone, call back tone and other information.
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Picture 116 - Tone settings on the web
12.14 Phone settings >> Advanced
User can configure the advanced configuration settings in this page.
Screen Configuration.
Enable Energy Saving
Backlight Time
Screen Saver
LCD Menu Password Settings.
The password is admin by default.
Keyboard Lock Settings.
Configure Greeting Words
The greeting message will display on the top left corner of the LCD when the device is idle, which is limited to 16 characters. The default chars are ‘VOIP PHONE’.
12.15 Phonebook >> Contact
User can add, delete, or edit contacts in the phonebook in this page. User can browse the phonebook and sorting it by name, phones, or filter them out by group.
To add a new contact, user should enter contact’s information and press “Add” button to add it.
To edit a contact, click on the checkbox in front of the contact, the contact information will be copied to the contact edit boxes, press “Modify” button after finished editing.
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To delete one or multiple contacts, check on the checkbox in front of the contacts wished to be deleted and click the “Delete” button, or click the “Clear” button with selecting any contacts to clear the phonebook.
User can also add multiple contacts into a group by selecting the group in the dropdown options in front of “Add to Group” button at the bottom of the contact list, selecting contacts with checkbox and click “Add to Group” to add selected contacts into the group.
Similarly, user can select multiple users and add them into blacklist by click “Add to
Blacklist” button.
12.16 Phonebook >> Cloud phonebook
Cloud Phonebook
User can configure up to 8 cloud phonebooks. Each cloud phonebook must be configured with an URL where an XML phonebook is stored. The URL may be based on
HTTP/HTTPs or FTP protocol with or without authentication. If authentication is required, user must configure the username and password.
To configure a cloud phonebook, the following information should be entered,
Phonebook name (must)
Phonebook URL (must)
Access username (optional)
Access password (optional)
LDAP Settings
The cloud phonebook allows user to retrieve contact list from a LDAP Server through
LDAP protocols.
User must configure the LDAP Server information and Search Base to be able to use it on the device. If the LDAP server requests an authentication, user should also provide username and password.
To configure a LDAP phonebook, the following information should be entered,
Display Title (must)
LDAP Server Address (must)
LDAP Server Port (must)
Search Base (must)
Access username (optional)
Access password (optional)
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Note! Refer to the LDAP technical documentation before creating the LDAP phonebook and phonebook server.
Web page preview
Phone page supports preview of Internet phone directory and contacts
After setting up the XML Voip directory or LDAP,
Select [ Phone book ] >> [ Cloud phone book ] >> [ Cloud phone book ] to select the type.
Click the set XML/LDAP to download the contact for browsing.
Picture 117 - Web cloud phone book Settings
12.17 Phonebook >> Call List
Restricted Incoming Calls ：
It is similar like a blacklist. Add the number to the blacklist, and the user will no longer receive calls from the stored number until the user removes it from the list.
Users can add specific Numbers to the blacklist or add specific prefixes to the blacklist to block calls with all Numbers with this prefix.
Allowed Incoming Calls ：
When DND is enabled, the incoming call number can still be called.
Restricted Outgoing Calls ：
Adds a number that restricts outgoing calls and cannot be called until the number is removed from the table.
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12.18 Phonebook >> Web Dial
Use web pages for call, reply, and hang up operations.
12.19 Phonebook >> Advanced
Users can export the local phone book in XML, CSV, and VCF format and save it on the local computer.
Users can also import contacts into the phone book in XML, CSV, and VCF formats.
Attention! If the user imports the same phone book repeatedly, the same contact will be ignored. If the name is the same but the number is different, the contact is created again.
Users can delete groups or add new groups on this page. Deleting a contact group will not delete contacts in that group.
12.20 Call Logs
The user can browse the complete call record in this page. The call record can be sorted by time, call number, contact name or line, and the call record can be screened by call record type (incoming call, outgoing call, missed call, forward call).
The user can also save the number in the call record to his/her phone book or add it to the blacklist/whitelist.
Users can also dial the web page by clicking on the number in the call log.
Users can also download call records conditionally and save them locally.
12.21 Function Key >> Function Key
Function Key Configuration ：
One-key transfer Settings: establish new call, blind transfer, attention-transfer, one-key three-party, Play DTMF.
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DSS Key home page: None/Page1/Page2/Page3/Page4
The device provides 112 user-defined shortcuts that users can configure on a web page.
Parameters
Memory Key
Line
Key Event
DTMF
URL
Multicast
Action URL
XML browser
Table 27 - Function Key configuration
Description
BLF (NEW CALL/BXFE /AXFER) : It is used to prompt user the state of the subscribe extension, and it can also pick up the subscribed number, which help user monitor the state of subscribe extension (idle, ringing, a call). There are 3 types for one-touch
BLF transfer method.
p.s. User should enter the pick-up number for specific BLF key to fulfill the pick-up operation.
Presence : Compared to BLF, the Presence is also able to view whether the user is online.
Note: You cannot subscribe the same number for BLF and
Presence at the same time
Speed Dial : You can call the number directly which you set. This feature is convenient for you to dial the number which you frequently dialed.
Intercom : This feature allows the operator or the secretary to connect the phone quickly; it is widely used in office environments.
It can be configured as a Line Key. User is able to make a call by pressing Line Key.
User can select a key event as a shortcut to trigger.
For example: MWI / DND / Release / Headset / Hold / etc.
It allows user to dial or edit dial number easily.
Open the specific URL directly.
Configure the multicast address and audio codec. User presses the key to initiate the multicast.
The user can use a specific URL to make basic calls to the phone.
Users can set the DSS Key for specific URL download and other operations.
12.22 Function Key >> Softkey
The User Settings mode and display style, display page.
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Table 28 - Softkey configuration
Parameter
Softkey Mode
Softkey mode
Description
Disabled and More ， Default is Disabled
Softkey Style
Softkey display style Softkey Exit on Left or Right
Screen
Call Dialer
Redial/2aB/Delete/Exit/Call Back/Dial/Join/MWI/Local
Contacts/Pickup/CallLog/Missed/Clear/ln/Dialed/Pause/Next line/Prev line/Headset/Audio/Video/Remote XML/DSS Key
Conference
Desktop
Hold/Split/End/Release/Mute/DSS Key/Headset
CallLog/Menu/Local Contacts/DND/Prev Account/Next
Account/Blacklist/Call Back/CallForward/Locked/Memo/
Missed/MWI/Dialed/Reboot/Redial/Remote XML/SMS/
Headset/Status/DSS Key/ln
Divert Dialed
Ending
Predictive Dialer
Ringing
Talking
Transfer Alerting
Transfer Dialer
Trying
Waiting
Redial/2aB/Delete/Exit/Forward/Local Contacts/CallLog
/Clear/Missed/Dialed/Headset/Video/Audio/Remote XML
/DSS Key
Redial/End/Headset/Release/DSS Key
Dial/2aB/Delete/Exit/Call Back/Local Contacts/Redial
/Pickup/MWI/Join/CallLog/Release/Missed/Pause/Dialed/
Headset/Video/Audio/Remote XML/DSS Key/ln/Next line
/Prev line
Answer/Forward/Reject/Mute/Release/Headset/Video/Audio/
DSS key
Hold/Transfer/Conference/End/Mute/Release/New Call/
Local Contacts/Listen/CallLog/Next call/Prev call/
Private/Headset/Video/Audio/DSS Key
End/Transfer/Headset/Release/DSS Key
Redial/Delete/Exit/2aB/Dial/Local Contacts/Transfer/
CallLog/Clear/Missed/Dialed/Pause/Headset/Video/Audio/R emote XML/DSS Key
End/Release/Headset/DSS Key
Hold/Transfer/Conference/End/Answer/Forward/Mute/Next call/New call/Prev call/Reject/Release/Headset/Listen/
Video/Audio/DSS Key
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12.23 Function Key >> Advanced
Global key Settings
The default configuration is empty, and the global memory key function can be configured.
The configured memory key has a call path. If the global configuration is maintained, pressing the memory key again will maintain the call path. If the same configuration hung up, press the memory key again will hang up this road call.
Programmable key Settings
Please refer to the
Table 28 Softkey configuration
IP Camera List
Picture 118 -IP Camera List
12.24 Application >> Manage Recording
See
9.3 Record for details of recording.
12.25 Security >> Web Filter
The user can set up a configuration management phone that allows only machines with a certain network segment IP access.
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Picture 119 - Web Filter settings
Picture 120 - Web Filter Table
Add and remove IP segments that are accessible; Configure the starting IP address within the start IP, end the IP address within the end IP, and click [ Add ] to submit to take effect. A large network segment can be set, or it can be divided into several network segments to add. When deleting, select the initial IP of the network segment to be deleted from the drop-down menu, and then click [ Delete ] to take effect.
Enable web page filtering: configure enable/disable web page access filtering; Click the
"apply" button to take effect.
Note: if the device you are accessing is in the same network segment as the phone, please do not configure the filter segment of the web page to be outside your own network segment, otherwise you will not be able to log in the web page.
12.26 Security >> Trust Certificates
Set whether to open license certificate and general name validation, select certificate module.
You can upload and delete uploaded certificates.
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Picture 121 - Certificate of settings
12.27 Security >> Device Certificates
Select the device certificate as the default and custom certificate.
You can upload and delete uploaded certificates.
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Picture 122 - Device certificate setting
12.28 Security >> Firewall
Picture 123 - Network firewall Settings
Through this page can set whether to enable the input, output firewall, at the same time can set the firewall input and output rules, using these Settings can prevent some malicious network access, or restrict internal users access to some resources of the external network, improve security.
Firewall rule set is a simple firewall module. This feature supports two types of rules: input rules and output rules. Each rule is assigned an ordinal number, allowing up to 10 for each rule.
Considering the complexity of firewall Settings, the following is an example to illustrate:
Parameter
Enable Input Rules
Enable Output Rules
Input/Output
Deny/Permit
Protocol
Src Port Range
Src Address
Table 29 - Network Firewall
Description
Indicates that the input rule application is enabled.
Indicates that the output rule application is enabled.
To select whether the currently added rule is an input or output rule.
To select whether the current rule configuration is disabled or allowed;
There are four types of filtering protocols: TCP | UDP |
ICMP | IP.
Filter port range
Source address can be host address, network address, or all addresses 0.0.0.0; It can also be a network address similar to *.*.*.0, such as: 192.168.1.0.
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Dst Address
Src Mask
Dst Mask
The destination address can be either the specific IP address or the full address 0.0.0.0; It can also be a network address similar to *.*.*.0, such as: 192.168.1.0.
Is the source address mask. When configured as
255.255.255.255, it means that the host is specific. When set as 255.255.255.0, it means that a network segment is filtered.
Is the destination address mask. When configured as
255.255.255.255, it means the specific host. When set as
255.255.255.0, it means that a network segment is filtered.
After setting, click [ Add ] and a new item will be added in the firewall input rule, as shown in the figure below:
Picture 124 - Firewall Input rule table
Then select and click the button [ Apply ].
In this way, when the device is running: ping 192.168.1.118, the packet cannot be sent to
192.168.1.118 because the output rule is forbidden. However, other IP of the ping
192.168.1.0 network segment can still receive the response packet from the destination host normally.
Picture 125 - Delete firewall rules
Select the list you want to delete and click [ Delete ] to delete the selected list.
12.29 Device Log >> Device Log
You can grab the device log, and when you encounter an abnormal problem, you can send the log to the technician to locate the problem. See
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12.30 Security settings
Picture 126 -Input and output settings
Security settings parameter description basic settings
Ringtone
Duration
Alarm bell duration
Input &
Tamper
Server
Address
Configure the remote response server address (including the remote response server address and the alarm trigger server address). When the input port is triggered, a short message will be sent to the server, the message format is as follows:
Alarm_Info:Description=i51;SIP
User=;Mac=0c:38:3e:39:6a:b6;IP=172.16.7.189;port=Input
Input port settings
Input port Enable or disable the input port
Trigger mode
When low level trigger (closed trigger) is selected, the detection input port
(low level) closed trigger.
When the high level trigger (disconnect trigger) is selected, the detection input port (high level) disconnect trigger.
Send short message
Enable or disable the input port to send messages to the server
Dss Key
Triggered
When set to dsskey1 or dsskey2, trigger dsskey to make a call, the default is none
Support ringtone selection
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Ringtone
Table 30 -Input and Output Parameters
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13 Trouble Shooting
When the phone is not in normal use, the user can try the following methods to restore normal operation of the phone or collect relevant information and send a problem report to Fanvil technical support mailbox.
13.1 Get Device System Information
Users can get information by pressing the [ Menu ] >> [ Status ] option in the phone.The
following information will be provided:
The network information
Equipment information (model, software and hardware version), etc.
13.2 Reboot Device
Users can reboot the device from soft-menu, [ settings ] >> [ reboot ], a nd press [ Reboot ],
Or simply remove the power supply and restore it again.
13.3 Reset Device to Factory Default
Reset Device to Factory Default will erase all user’s configuration, preference, database and profiles on the device and restore the device back to the state as factory default.
To perform a factory default reset, user should press [ setting ] >>[Advanced]>>
[ maintain ]. Then choose [ Phone Reset ] and press [ Reset ]. The device will be rebooted into a clean factory default state.
13.4 Screenshot
If there is a problem with the phone, the screenshot can help the technician locate the function and identify the problem. In order to obtain screen shots, log in the phone webpage [ System ] >> [ Tools ], and you can capture the pictures of the main screen (you can capture them in the interface with problems).
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Picture 127 - Screenshot
13.5 Network Packets Capture
Sometimes it is helpful to dump the network packets of the device for issue identification.
To get the packets dump of the device, user needs to log in the device web portal, open page [ System ] >> [ Tools ] and click [ Start ] in “Network Packets Capture” section. A pop-up message will be prompt to ask user to save the capture file. User then should perform relevant operations such as activate/deactivate line or making phone calls and click [ Stop ] button in the web page when operation finished. The network packets of the device during the period have been dumped to the saved file.
Picture 128 - Web capture
User may examine the packets with a packet analyzer or send it to Fanvil support mailbox.
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13.6 Get Log Information
Log information is helpful when encountering an exception problem. In order to get the log information of the phone, the user can log in the phone web page, open the page
[ Device log ], click the [ Start ] button, follow the steps of the problem until the problem appears, and then click the [ End ] button, [ Save ] to local analysis or send the log to the technician to locate the problem.
Or use a thumb drive to export debugging log, find a thumb drive to place a text document named fv-ipphone-dump-trace.txt,
Plug in the USB port and wait for about 3 minutes.The usb flash drive automatically generates log files.
13.7 Common Trouble Cases
Table 31 - Trouble Cases
Trouble Case
Device could not boot up
Solution
1. The device is powered by external power supply via power adapter or PoE switch. Please use standard power adapter provided by Fanvil or PoE switch met with the specification requirements and check if device is well connected to power source.
2. If you saw “POST MODE” on the device screen, the device system image has been damaged. Please contact location technical support to help you restore the phone system.
Device could not register to a service provider
1. Please check if device is well connected to the network. The network Ethernet cable should be connected to the
[Network] port NOT the [PC] port. If the cable is not well connected to the network icon [WAN disconnected] will be flashing in the middle of the screen.
2. Please check if the device has an IP address. Check the system
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Audio is chopping at far-end in Hands-free speaker mode information, if the IP displays “Negotiating…”, the device does not have an IP address. Please check if the network configurations is correct.
3. If network connection is fine, please check again your line configurations. If all configurations are correct, please kindly contact your service provider to get support, or follow the instructions in “
13.5 Network Packet Capture ” to get the network
packet capture of registration process and send it to Fanvil support to analyze the issue.
This is usually due to loud volume feedback from speaker to microphone. Please lower down the speaker volume a little bit, the chopping will be gone.
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Key features
Supports PoE or external power adapter for flexible power options
Offers wall mount and desktop installation options
Provides basic call functions: answering, making, holding, & transferring calls
Supports video calls for enhanced communication
Includes advanced functions like BLF and call recording
Features intercom and multicast functionalities
Frequently asked questions
The i56A IP phone supports wall mount and desktop installation options. Refer to the installation guide section for detailed instructions.
Yes, the i56A IP phone supports video calls. Refer to the video call section for details on how to set up and use this feature.
You can configure call forwarding through the phone's settings. Refer to the call forward section of the manual for detailed instructions.
The phone supports advanced functions like BLF (Busy Lamp Field), call recording, intercom, and multicast functionalities.
The i56A phone supports local and server recording. Refer to the recording section for configurations.