advertisement
1
High Definition Color PoE
IP Phone
VIP-1120PT
Copyright
Copyright
2016 by PLANET Technology Corp. All rights reserved. No part of this publication may be reproduced, transmitted, transcribed, stored in a retrieval system, or translated into any language or computer language, in any form or by any means, electronic, mechanical, magnetic, optical, chemical, manual or otherwise, without the prior written permission of PLANET.
PLANET makes no representations or warranties, either expressed or implied, with respect to the contents hereof and specifically disclaims any warranties, merchantability or fitness for any particular purpose. Any software described in this manual is sold or licensed "as it is". Should the programs prove defective following their purchase, the buyer (and not PLANET, its distributor, or its dealer) assumes the entire cost of all necessary servicing, repair, and any incidental or consequential damages resulting from any defect in the software. Further, PLANET reserves the right to revise this publication and to make changes from time to time in the contents hereof without obligation to notify any person of such revision or changes.
All brand and product names mentioned in this manual are trademarks and/or registered trademarks of their respective holders.
Federal Communication Commission Interference Statement
This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to Part 15 of FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses, and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:
1. Reorient or relocate the receiving antenna.
2. Increase the separation between the equipment and receiver.
3. Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
4. Consult the dealer or an experienced radio technician for help.
FCC Caution
To assure continued compliance, for example, use only shielded interface cables when connecting to computer or peripheral devices. Any changes or modifications not expressly approved by the party responsible for compliance could void the user’s authority to operate the equipment.
2
This device complies with Part 15 of the FCC Rules. Operation is subject to the following two conditions:
(1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation.
Federal Communication Commission (FCC) Radiation Exposure Statement
This equipment complies with FCC radiation exposure set forth for an uncontrolled environment. In order to avoid the possibility of exceeding the FCC radio frequency exposure limits, human proximity to the antenna shall not be less than 20 cm (8 inches) during normal operation.
Safety
This equipment is designed with the utmost care for the safety of those who install and use it. However, special attention must be paid to the dangers of electric shock and static electricity when working with electrical equipment. All guidelines of this and of the computer manufacture must therefore be allowed at all times to ensure the safe use of the equipment.
CE Mark Warning
This is a Class B product. In a domestic environment, this product may cause radio interference, in which case the user may be required to take adequate measures.
WEEE Regulation
To avoid the potential effects on the environment and human health as a result of the presence of hazardous substances in electrical and electronic equipment, end users of electrical and electronic equipment should understand the meaning of the crossed-out wheeled bin symbol. Do not dispose of WEEE as unsorted municipal waste; they should be collected separately.
Revision
User’s Manual of PLANET High Definition Color PoE IP Phone
Model: VIP-1120PT
Rev: 1.00 (September, 2016)
Part No. EM-VIP-1120PT_v1.0
3
Table of Contents
Using Handset/Hands-free Speaker/Headphone .................................................. 20
Introduction to the Phone User Interface ........................................................ 23
Browse and Add/Remove Contacts in Group ........................................................ 46
4
5
6
Figures
Figure 33 - Dialing a Number before Audio Channel Opened ................................... 37
Figure 34 - Dial a Number after Audio Channel Opened ........................................... 37
7
Figure 63 - Activate Call Forward and configure Call Forward Number .................... 56
Figure 64 - Set Delay Time for Call Forward on No Answer...................................... 56
Figure 66 - Web Call, Entering Number and Select Line ........................................... 61
8
Tables
Table 7 - Common Phone Feature Settings on Web ................................................. 75
9
Chapter 1. Product Introduction
1.1 Package Contents
Please read the following safety notices before installing or using this unit. They are crucial for the safe and reliable operation of the device.
The package should contain the following items:
SIP IP Phone Unit x 1
Quick Installation Guide x 1
RJ45 Cable x 1
Stand x 1
1. If any of the above items are missing, please contact your dealer immediately.
2. Using the power supply that is not the one included in the camera packet will cause damage and void the warranty for this product.
1.2 Overview
Intuitive Design Brings Quality Communication
PLANET VIP-1120PT is a two-line SIP desktop phone with color display that brings lifelike richness and voice quality to phone calls. The VIP-1120PT is a reliable communication device as it features ease of use, versatility, high-quality design and superb audio performance. Besides, its full duplex speakerphone system with HD voice can definitely make communication between two or more business parties crystal clear with no interference noise in the background as it supports the G.722 wideband codec.
10
IP Phone with Color Display
As the VIP-1120PT is an IP phone, it allows you to make calls from any location easily as long as it is online. Its 2.4-inch TFT color display with a resolution of 320 x 240 pixels offers a clear depiction of caller’s information. It complies with IEEE 802.3af PoE interface for flexible deployment and supports superior audio quality delivered by the advanced speaker and microphone system, and the digital signal processor (DSP).
Compliant with SIP 2.0
SIP phones continue to gain popularity among businesses as the preferred protocol for enhancing communication across IP networks. The VIP-1120PT supports Session Initiation Protocol 2.0 (RFC
3261) for easy integration with general voice over IP system. The VIP-1120PT is able to broadly interoperate with equipment provided by VoIP infrastructure providers, thus enabling them to provide their customers with better voice over IP services.
11
Affordable for All Businesses
The VIP-1120PT is definitely affordable for all business establishments who want flexible deployment options and expansion. It can effortlessly deliver secure toll voice quality by utilizing cutting-edge
802.1p QoS (Quality of Service) and IP ToS technology.
12
1.3 Features
Highlights
• 2.4-inch color LCD monitor
• 2-line key with color LED
• SIP identities
• Wideband G.722 HD audio
• Power over Ethernet (PoE) compliant
• QoS, TR069, auto-provisioning and multi-language
Advantageous Applications
• Supports SIP 2.0 (RFC3261)
• Inband, SIP info, RFC 2833 DTMF relay
• Soft keys and function keys programmable
• Echo cancellation: Supports G.168, and a maximum filter length of 96ms in hands-free mode
• Supports voice gain setting, voice activity detection (VAD) and comfort noise generation
(CNG)
• Full duplex hands-free speakerphone
• Hands-free headset ringing choice
• Voice codec setting for each SIP line
SIP Applications
• Call forward and transfer (blind/attended)
• Call holding and waiting
• 3-way conferencing
• Paging and intercom
• Call park, call pickup and join call
• Redial and click to dial
• Secondary dialing automatically
• Incoming calls, outgoing calls and missed calls (Each supports 300 records)
• SMS and speed dial
• Phonebook up to 500 records
Call Control Features
• Flexible dial map, hotline and empty calling no. for rejected service
• Black list for rejected authenticated calls
13
• White list and call limit
• Do not disturb (DND)
• Caller ID display
• Dial without registration
Network Features
• PPPoE/DHCP client
• 802.1 VLAN (voice VLAN/data VLAN)
• VPN (L2TP) and openVPN
• Quality of Service
Maintenance and Management
• Integrated web server provides web-based administration and configuration
• Telephone keypad configuration via display menu/navigation
• Automated provisioning and upgrade via HTTPS, HTTP, TFTP
• User authentication for configuration pages
• SNTP time synchronization
• TR069
14
1.4 Specifications
Table 1 – Specifications
VIP-1120PT Product
Hardware
Lines (Direct Numbers)
Display
Feature Keys
Network Interfaces
Connectors
Power Requirements
Weight
Dimensions (W x D x H)
Protocols and Standard
2 SIP Lines
LCD x 1: 2.4-inch (320 x 240) color LCD
Keypad: 32 keys, including
4 soft-keys
12 standard phone digit keys
5 navigation keys
2-line key with tri-color LED
5 function keys
3 volume control keys -- Up, Down, Mute (microphone)
1 hands-free feature
RJ45 Ethernet jacket x 2:
Network x 1 (802.3af PoE Class 1 enabled)
PC x 1 (Bridged Network)
HD hands-free speaker (0 ~ 7KHz) x 1
HD hands-free microphone (0 ~ 7KHz) x 1
HD handset (RJ9) x 1
IEEE 802.3af
5V 600mA
620g
196 x 164 x 60 mm
Protocols
SIP2.0 over UDP/TCP/TLS
RTP/RTCP/SRTP
STUN
DHCP
PPPoE
802.1x
L2TP (basic unencryption)
OpenVPN
15
Request for Comments (RFCs)
SNTP
FTP/TFTP
HTTP/HTTPS
TR069
354, 1321, 1350, 1769, 1889, 1890, 2131, 2132, 2616, 2617, 2661,
2833, 2976, 3261, 3262, 3263, 3264, 3265, 3268, 3311, 3489,
3711, 4346, 4566, 5630, 5865
Networking
Networking
Deployment & Maintenance
Physical: 10/100Mbps Ethernet, dual bridged port for PC bypass
IP Configuration: Static, DHCP, PPPoE
Network Access Control: 802.1x
VPN: L2TP (basic unencryption), OpenVPN
VLAN
QoS
Auto-provisioning via FTP/TFTP/HTTP/HTTPS/DHCP/OPT66/SIP
PNP/TR069
Web management portal
Web-based packet dump
Configuration backup/restore
Auto/Manual online software upgrade
Syslog
Features
Call Features
Call out, answer, reject
Mute/unmute (microphone)
Call hold, resume
Call waiting
Intercom
Caller ID display
Call forwarding (always/busy/no answer)
Call transfer (attended/unattended)
Call parking, pick-up (depending on server)
Redial
Do-not-disturb (per line/per phone)
Auto-answering (per line)
Voice message (on server)
Local 3-way conferencing
16
Phone Features
Audio Features
Environment
Operating Temperature
Operating Humidity
Emission
Hot line
Phone accessibility control by user PIN
Intelligent phonebook (up to 500 entries in total)
Remote phonebook (XML/LDAP)
Intelligent contact number matching/filtering
Call log (300 entries in total, in/out/missed)
Voice message waiting indication (VMWI)
Network time synchronization
Multi-language support in screen and web UI: English, Chinese
(Traditional/Simplified), Japanese, Russian, Italian, Turkish,
German, Dutch, Spanish, Hebrew, Polish, French, etc.
HD voice microphone/speaker (handset/hands-free, 0~7KHz frequency response)
Wideband ADC/DAC 16KHz sampling
Narrowband codec: G.711a/u, G.723.1, G.726-32K, G.729AB
Wideband codec: G.722
Full-duplex acoustic echo canceller (AEC) – a tail length of
96ms in hands-free mode
Voice activity detection (VAD), comfort noise generation (CNG), background noise estimation (BNE), noise reduction (NR)
Packet loss concealment (PLC)
Dynamic adaptive jitter buffer up to 300ms
DTMF: In-band, out-of-band – DTMF-relay (RFC2833), SIP info
0 ~ 40 degrees C
10 ~ 65% (non-condensing)
CE, FCC, RoHS
17
Chapter 2. Hardware Interface & Installation
2.1 Physical Descriptions
Figure 1 - Keypad
The above picture shows the keypad layout of the device. Each key provides its own specific function.
User should refer to the illustration in this section about the usage of each key and the description in this document about each function.
18
Some keys come with long key press. User can press and hold the key for 1.5 seconds to trigger a function.
Table 2 - Keypad Descriptions
Interface Description
1 Soft-menu Buttons
These four buttons provide different functions corresponding to the soft-menu displayed on the screen.
2
3
Line 1 ~2 Key
MWI Key
4
5
Navigation Keys
Numeric Keypad
Switch to different SIP lines.
Press this button to enter the voicemail service.
User can press the up/down navigation key to change line focus on the talking screen or move the cursor to a listed item on the screen. In some configuration or text editor screen, user can press the left/right navigation key to switch option or move cursor to the left/right.
The 12 standard telephone keys provide the same function as standard telephones, but some keys also provide special functions by long pressing the keys.
Key # - Long press to lock the phone. (Default PIN is 123)
6
7
8
9
Headset Key
Microphone Mute
Redial
Hands-free Speaker
Use the headset to call out or call in
User can mute the microphone with this button in talking mode
By pressing the ‘Redial’ button, user can redial the last dialed number
By pressing this button once, user can turn on the audio channel of hands-free speaker
19
2.2 Using Handset/Hands-free Speaker/Headphone
Using Handset
To talk over handset, user should lift the handset off the device and dial the number, or dial the number first, then lift the handset and the number will be dialed. User can switch audio channel to handset by lifting the handset when audio channel is opened in speaker or headphone.
Using Hands-free Speaker
To talk over hands-free speaker, user should press the hands-free button and then dial the number, or dial the number first and then press the hands-free button. User can switch audio channel to the speaker from handset by pressing the hands-free button when audio channel is opened in handset.
Using Headphone
To use headphone, by default, user should press the headset button, which is defined by DSS key to turn on the headphone. Like the handset and hands-free speaker, user can dial the number before or after the headphone is turned on.
Using Line Keys (Defined by DSS Key)
User can use line key to make or answer a call on specific line. If handset has been lifted, the audio channel will be opened in handset, otherwise, the audio channel will be opened in hands-free speaker or headphone.
Please connect power adapter, network, PC, handset, and headphone to the corresponding ports as described in the diagram below.
Figure 2 - Connecting to the Device
20
1
Interface
Headset
2
3
4
5
Table 3 - Interface Descriptions
Description
Connected to the headset interface
Headphone Connected to the headphone interface
Use one end of a straight Cat5 cable to directly connect to a switch/hub and the other end to the RJ45 connector for
Internet (WAN) internet access.
The WAN interface also can be connected with 802.3af PoE switch or converter for power supply
PC (LAN)
DC 5V
Use a Cat5 cable to directly connect between the RJ45 connector and a PC to maintain the existing network. structure.
Power port (This unit does not include the 5V/600mA power adapter)
Power adapter is optional for VIP-1120PT model. (This unit does not include the 5V/600mA power adapter.)
21
2.3 Desktop and Wall-mount Installation
The device supports two installation modes, desktop and wall-mount. To set up the phone on the desktop, please follow the instructions as shown in the diagrams below.
Figure 3 - Desktop Installation
To mount the device on the wall, please follow the instructions shown in the pictures below..
Figure 4 - Wall-mount Installation
22
Chapter 3. Introduction to the Phone User
Interface
3.1 Screen User Interface
Figure 5 - Screen Layout / Default Standby Screen
The screen user interface is mostly presented in the above layout except some prompt messages. The upper area is the main screen to display the device’s status and information or data for viewing or editing. The lower area is the software menu (soft-menu) buttons which will change against user’s action or device’s status.
The above picture shows the default standby screen which is also the root of the soft-menu. The default standby screen shows the greeting words and effective feature indications like voice message, missed call, auto-answering, do-not-disturb, call forward, lock state, and the network connectivity. User can get back to the default standby screen mostly by lifting and putting by the handset.
The icon illustration is described in Appendix I - Icon Illustration.
In some screens, there are more items or long text to be displayed which could not fit into the screen.
They will be arranged in a list or multiple lines with a scroll bar. If user sees a scroll bar, user can use up/down navigation buttons to scroll the list. By pressing the navigation keys a little longer, user can scroll the list or items at a faster speed.
23
Figure 6 - Vertical Scroll Bar
3.2 Web Portal
User can also use the device’s web portal to manage or operate the device. User should open the device’s web portal page by entering the device’s IP address in a browser. To get the device IP address, user could press the soft-menu button [Menus] -> [Status] or by pressing the [Down] navigation key.
Figure 7 - Check the Device’s IP Address
24
The first screen of the device’s web portal is the login page.
Figure 8 - Web Portal Login Page
Default IP Address: 172.16.0.1
Default username: admin
Default password: 123
User must enter the username and password to log in to the web portal. The default username and
password is “admin” and “123”. For details on web portal operations, please refer to Chapter 7 Web
25
Chapter 4. IP Phone Setup
In order to get the device ready for making and receiving phone calls, the device must be configured with correct network configurations and at least one of the lines must be configured with an IP telephony service.
4.1 Network Configuration
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 the device, the network parameters must be configured properly first. To configure network parameters, user should open the network configuration screen through soft-menu [Menu] -> [Settings]
-> [Advanced Settings] -> [Network] -> [Network Settings] from standby screen.
If user see a ‘Network Disconnected’ icon flashing in the middle of screen, it means the network cable is not correctly connected to the device’s network port. Please check whether the cable is connected correctly to the device and to the network switch, router, or modem.
There are three common IP configuration modes.
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 parameter manually, including IP address, subnet mask, default gateway, and DNS servers. This is usually used in an office environment or by power users.
PPPoE -- This option is often used by users who connect the device to a broadband modem or router. To establish a PPPoE connection, user should configure username and password provided by the service provider.
The device is default configured in Static mode; the default IP address is 172.16.0.1
26
4.2 Line 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 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.
To configure a line manually, user may open the line configuration screen through soft-menu button
[Menu] -> [Settings] -> [Advanced Settings] -> [Accounts] -> [SIP1] / [SIP2] -> [Basic Settings] from the standby screen.
User must enter the correct PIN code to be able to go to advanced settings to edit line configuration. (The default PIN is 123.)
The parameters and screens are listed in the diagrams below.
Figure 9 - Configure SIP Proxy Server Address
27
Figure 10 - Configure SIP Proxy Server Port
Figure 11 - Configure SIP Username
Figure 12 - Configure Authentication Name
28
Figure 13 - Configure SIP Authentication Password
Figure 14 - Configure Display Name
Figure 15 - Enable/Disable Outbound Proxy
29
Figure 16 - Enable/Disable SIP Line
Save the adjustment by pressing [OK] or [Save] when done.
For users who want to configure more options, user should use web management portal to modify or
Advanced Settings in accounts on the individual line to configure those options.
Figure 17 - Configure Advanced Line Options
30
Figure 18 - Configure SIP Realm/Domain
Figure 19 - Configure Unregistered Dial
Figure 20 - Enable/Disable Anonymous
31
Figure 21 - Configure DTMF Mode
Figure 22 - Enable/Disable STUN
Figure 23 - Configure Local Port
32
Figure 24 - Configure Ring Type
Figure 25 - Configure MWI Number
Figure 26 - Configure Pickup Number
33
Figure 27 - Configure Park Number
Figure 28 - Configure Join Call Number
Figure 29 - Enable/Disable Missed Call
34
Figure 30 - Enable/Disable Feature Sync
Figure 31 - Enable/Disable SCA
On the device, user can only configure whether STUN/Outbound Proxy is enabled to change the STUN/Outbound Proxy server address. User must edit in the device web portal. (Refer to
7.15 LINES (Global Settings and 7.12 LINES (SIP)
35
Chapter 5. Using the Phone
5.1 Making Phone Calls
Default Line
The device provides 2 line services. If both 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 change the default line, user can press the left/right navigation button to switch between two lines. To enable or disable default line, user can press
[menu] - > [Features] -> [Default Line] or configure from Web Interface (Web/PHONE
Settings/Features).
Figure 32 - Default Line
Dialing Methods
User can dial a number by,
Entering the number directly
Selecting a phone number from phonebook contacts (Refer to 5.5 Using Phonebook)
Selecting a phone number from cloud phonebook contacts (Refer to 5.9 Using Cloud
Selecting a phone number from call logs (Refer to 5.11 Call Logs)
Redialing the last dialed number
36
Dialing Number then Opening Audio
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 the [Dial] button on the soft-menu, or press the hands-free button to turn on the speaker or headphone, or lift the handset to call out with the current line, or user can press the line key to call out with a specified line.
Figure 33 - Dialing a Number before Audio Channel Opened
Opening Audio and then Dialing the Number
Another alternative is the traditional way to firstly open the audio channel by lifting the handset, turning on the hands-free speaker or headphone by pressing hands-free button, or line key, and then dial the number with one of the above methods. When the number is dialed completely, user can press the
[Dial] button or [OK] button to call out, or the number will be dialed out automatically after timeout.
Figure 34 - Dial a Number after Audio Channel Opened
37
For some users who get used to dial a number immediately by pressing # key, the user must login to the web to enable ‘Press "#" to invoke dialing’ option in page [Line] -> [Dial Plan] ->
“Basic Settings” and apply it.
Cancel Call
While calling the number, user can press the [End] button or close the audio channel by putting back the handset or press the hands-free button to end the call.
Figure 35 - Calling Remote Party
Answering Incoming Call
When there is an incoming call while the device is idle, user will be alerted to the following incoming call on the screen.
Figure 36 - Incoming Call Screen
38
User can answer the call by lifting the handset, open headphone or speaker phone by pressing the hands-free button, or the [OK] / [Answer] button. To divert the incoming call, user should press the
[Divert] button. To reject the incoming call, user should press the [Reject] button.
Talking
When the call is connected, user will see a talking mode screen as shown in the following figure.
Figure 37 - Talking Mode Screen
1. Audio Channel -- The icon reflects the current audio channel being used.
2. Current Line -- The line is being used on the call.
3. Remote Party -- The name or number of the remote party.
4. Talking time -- The time has passed since the call was established.
5. Screen Pages -- The number of screen pages. User can press the up/down navigation buttons to switch screen page.
Call Holding/Resuming
User can hold the remote party by pressing the [Hold] button and the button will be changed to
[Resume] icon. User can press the [Resume] button to resume the call.
39
Figure 38 - Call Holding Screen
Call Ended
When user finishes the call, user can put the handset back to the device to hang up the call or press the hands-free button to close the audio channel to hang up.
When the call is held or in holding state, the user must press the [Resume] button to return to call mode, or putting the handset back to the device or pressing the speaker-free button to hang up is not available.
5.2 Make/Receive Second Call
The device can support up to two concurrent 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.
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 the current screen. The device will not be ringing but playing call waiting tone in the audio channel of the current call and the LED will be flashing in green. User can accept or reject the call which is the same as normal incoming call. When the waiting call is answered, the first call will be put on hold automatically.
40
Figure 39 - Second Incoming Call Screen
Second Outgoing Call
To make a second call, user may press the [XFER] / [Conf] button to make a new call on the default line or press the line key to make new call on a specific line. Then dial the number the same way as making a phone call. Another alternative for making second call is to press the DSS Keys dialing out from the configured Keys (BLF/Speed Dial). When 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 the second dial.
Switching between Two Calls
When there are two calls established, user will see a dual call screen as shown in the following picture.
Figure 40 - Dual Calls
User can press the up/down navigation button to switch screen page, and switch call focus by pressing the [Hold] / [Resume] button.
41
Ending One Call
User may hang up the current call by closing the audio channel or press the [End] button. The device will return to single call mode in holding state.
5.3 Join/Split Two Calls (3-way Local Conference)
In the dual call mode, user can join two calls into a conference call by pressing the [Conf] button. When two calls are joined, user can split them by pressing the [Split] button.
Figure 41 - Conference Call
5.4 Call Transfer
When the user is talking with a remote party and wish to transfer the call to another remote party, there are two ways to transfer the call, attended and unattended.
Attended Transfer
The attended call transfer is also known as the ‘polite mode’ which is to dial through the other remote party, wait for the remote party to answer the call and then transfer the call.
This is the same procedure as making two concurrent calls. In the dual call mode, press the [XFER] button to transfer the first party to the second one.
Unattended/Blind Transfer
Unattended transfer is also known as ‘Fire and Forget’ mode. Instead of connecting to the second party
42
first and confirming the second call is established, user press the [XFER] button first and then dial the second party number. When the number is dialed, user can press the [XFER] button again and the first party will be transferred to the second.
This is like helping the first party to dial to the second one. However, the transfer could be successful if the second party answered it, or could be failed if the second party is busy or rejected it.
More advanced transfer configuration, please refer to 7.14 LINES (Dial Plan)
5.5 Using Phonebook
User can save contacts’ information in the phonebook and dial the contact’s phone number(s) from the phonebook. To open the phonebook, user can press the soft-menu button [Dir] in the default standby screen or keypad.
By default the phonebook is empty, user may add contact(s) to the phonebook manually or from call logs.
Figure 42 - Phonebook screen
The device can save up to a total of 500 contacts.
43
Figure 43 - Phonebook is Empty
When there are contact records in the phonebook, the contact records will be arranged in the alphabet order. User may browse the contacts with the up/down navigation keys. User may fast scroll the contact list by holding the up/down navigation keys. The record indicator tells user which contact is currently focused. User may check the contact’s information by pressing the [OK] button.
Figure 44 - Browsing Phonebook
5.6 Add/Edit/Delete Contact
To add a new contact, user should press the [Add] button to open Add Contact screen and enter the following contact information:
44
Contact Name
Tel. Number
Mobile Number
Other Number
Ring Tone
Contact Group
Figure 45 - Add New Contact
User can edit a contact by pressing the [Option] -> [Edit] button.
To delete a contact, user should move the record indicator to the position of the contact to be deleted, press the [Option] -> [Delete] button and confirm with [Yes].
5.7 Add/Edit/Delete Group
User can open the group list by pressing the [Groups] button in phonebook screen.
By default, the group list is blank. User can create his/her own groups, edit the group name, add or remove contacts in the group, and delete a group.
To add a group, press the [Add] button.
To delete a group, press the [Option] -> [Delete] button.
To edit a group, press the [Edit] button.
45
Figure 46 - Group List
5.8 Browse and Add/Remove Contacts in Group
User can browse contacts in a group by opening the group in group list with the [OK] button.
Figure 47 - Browsing Contacts in a Group
When user is browsing contacts of a group, user can also add or remove contacts in that group by pressing the [Add] button to enter the group contacts management screen. Press the [Add] button to create a new contact for the group; press the [Option] button to remove the contact from the group.
46
Figure 48 - Add/Remove Contacts in a Group
5.9 Using Cloud Phonebook
Cloud phonebook allows user to configure the device to download a phonebook from a cloud server.
This is very useful 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 for user to synchronize his/her phonebook from a personal mobile phone to the device with Cloud Phonebook Service which is to be provided publicly soon.
The cloud phonebook is ONLY temporarily downloaded to the device each time it is opened on the device to ensure the user gets the up-to-date phonebook. However, the downloading may take a couple of seconds depending on the network condition. Therefore, it is highly recommended to the users to save important contacts from cloud to local phonebook to save the time of waiting for downloading.
To open cloud phonebook list, press [Cloud Phonebook] in phonebook screen.
47
Figure 49 - Cloud Phonebook List
5.10 Open Cloud Phonebook
In cloud phonebook screen, user can open a cloud phonebook by pressing the [OK] / [Enter] button.
The device will start downloading the phonebook. If downloading failed, user will be prompted with a warning message.
Once the cloud phonebook is downloaded completely, user may browse the contact list and dial the same contact number as in local phonebook.
Figure 50 - Downloading Cloud Phonebook
48
Figure 51 - Browsing Contacts in Cloud Phonebook
5.11 Call Logs
The device can store up to 600 call log records and user can open the call logs to check all incoming, outgoing, and missed call records by pressing the soft-menu button [History].
In the call logs screen, user may browse the call logs with the up/down navigation keys.
Each call log record is presented with ‘call type’ and ‘call party number/name’. User can check further call log detail by pressing the [OK] button and dial the number with the [Dial] button, or add the call log number to phonebook by pressing [Option] -> [Add to Contact] .
User can delete a call log by pressing the [Delete] button and can clear all call logs by pressing the
[Clear] button.
Figure 52 - Call Logs
49
User can also filter call logs with specific call log type to narrow down the call log records by pressing the left/right navigation button and select one of the call log types in the soft-menu buttons.
- Missed Calls
- Received Calls/Incoming Calls
- Dialed Calls/Outgoing Calls
Figure 53 - Filter Call Log Type
5.12 Voice Message
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,
50
Figure 54 - New Voice Message Notification
To retrieve the voice messages, user must configure the voice message number first. Once the voice message number is configured, user can retrieve the voice message of a line by pressing the [Dial] button in the voice message screen.
When the device is in the default standby mode,
Check the [MWI] button.
Press the [MWI] button to open voice message configuration screen, select the line to be configured with the up/down navigation keys.
Press the [Edit] button to enable and edit the voice message number, when done, press the [OK] /
[Save] button to save the configuration.
Figure 55 - Voice Message Screen
51
Figure 56 - Configure the Voice Message Number
5.13 Do-Not-Disturb
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.
To quickly enable or disable the DND on all lines,
When the device is in the default standby mode,
Press the [DND] button to enable DND on all lines and icon will be displayed on the screen.
Press the [DND] button again to disable DND on all lines and icon will be disappeared.
Figure 57 - DND Enabled on All Lines
52
If user wishes to enable or disable DND on a specific line, user could change the DND mode in DND configurations.
Press the soft-button [Menu] till you find the [Features] item.
Enter [Features] item till you find the [DND] item.
Enter [DND] item till you find [DND item.
Press the left/right navigation key to change the DND mode or DND state on specific line. When done, press the [Save] button to save the changes.
User will see icon ‘SIP Do Not Disturb’ against the enabled line.
Figure 58 - Configure DND on Line
5.14 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.
When the device is in the default standby mode, if user wishes to enable or disable auto-answering on a specific line or change the auto-answering delay time, user could change the auto-answering configuration by adopting the following steps:
Press soft-button [Menu] till you find the [Features] item.
Enter [Features] item till you find the [Auto Answer] item.
Enter [Auto Answer] item to change the auto-answering configuration on a specific line
Press the left/right navigation button to select the auto-answering option. When done, press the
[Save] button to save the changes.
The default auto-answering delay is 5 seconds.
53
Figure 59 - Configure Auto-answering on Line1
Figure 60 - Auto-answering Enabled on Line1
5.15 Call Forward
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.
To configure call forward, when the device is in the default standby mode,
Press soft-button [Menu] till you find the [Features] item.
Enter [Features] item till you find the [Call Forward] item.
54
Press the [Call Forward] button to open call forward configuration screen and select the line to be configured with up/down navigation keys.
Press the [Enter] button to edit the call forward settings.
Select the call forward type with the up/down navigation keys. Click the [Enter] button to configure the call forward number and delay, if applicable.
Enable or Disable call forward with the left/right navigation buttons against specific lines and types.
If select ‘Enable’, browse the setting parameters with the up/down navigation keys and enter required information. When done, press the [Save] / [OK] button to save the changes.
Figure 61 - Select Line for Call Forward Configuration
Figure 62 - Select Call Forward Type
55
Figure 63 - Activate Call Forward and Configure Call Forward Number
Figure 64 - Set Delay Time for Call Forward on No Answer
56
Chapter 6. Phone Settings
6.1 Adjust Audio Volume
When the device is in the default standby mode,
Press soft-button [Menu] till you find the [Settings] item.
Enter [Settings] item till you find the [Basic Settings] item.
Enter [Basic Settings] item till you find [Voice Volume] item.
Enter [Voice Volume] item and you will find [Handset Volume], [Handsfree Volume] and [Headset
Volume] item.
Enter [Handset Volume] or [Handsfree Volume] or [Headset Volume] item, press the Left/Right navigation keys to adjust the audio volume for different mode.
Save the adjustment by pressing [OK] or [Save] when done.
6.2 Set Ring Tone Volume and Type
When the device is in the default standby mode,
Press soft-button [Menu] till you find the [Settings] item.
Enter [Settings] item till you find the [Basic Settings] item.
Enter [Basic Settings] item till you find [Ring Settings] item.
Enter [Ring Settings] item and you will find [Ring Volume] and [Ring Type] item.
Enter [Ring Volume] item and you find [Headset Volume] and [Handsfree Volume] item.
Enter [Headset Volume] or [Handsfree Volume] item, press the left/right navigation key to adjust the ring volume.
Save the adjustment by pressing [OK] when done.
Enter [Ring Type] item, press the left/right navigation keys to change the ring type.
Save the adjustment by pressing [OK] when done.
6.3 Adjust LCD Contrast
When the device is in the default standby mode,
Press soft-button [Menu] till you find the [Settings] item.
Enter [Settings] item till you find the [Basic Settings] item.
Enter [Basic Settings] item till you find [Screen Settings] item.
Enter [Basic Settings] item till you find [Contrast] item.
57
Enter [Contrast] item, press the left/right navigation keys to adjust the LCD contrast.
Save the adjustment by pressing [OK] or [Save] when done.
6.4 Set Device Time/Date
When the device is in the default standby mode,
Press soft-button [Menu] till you find the [Settings] item.
Enter [Settings] item till you find the [Basic Settings] item.
Enter [Basic Settings] item till you find [Time & Date] item.
Enter [Time & Date] item, use the up/down navigation keys to edit the time/date parameters and save the settings by pressing [OK] or [Save] when done.
Table 4 - Time Settings Parameters
Parameters
Mode
Description
Auto/Manual
Auto: Enable network time synchronization via SNTP protocol, default enabled.
Manual: User can modify data manually.
SNTP Server
Time Zone
12-hour Clock
Daylight Saving Time
Time format
SNTP server address
Select the time zone
Display the clock in 12-hour format
Enable or Disable the Daylight Saving Time
Select one of the following time formats:
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
58
Parameters Description
YYYY-MM-DD
Date (Only Mode = Manual) Edit the date of the device in the format of yyyy/mm/dd.
Time (Only Mode = Manual)
Edit the time of the device in the format of hh:mm:ss.
Hh=00~23, mm=00~59, ss=00~59
6.5 Set Device Language
When the device is in the default standby mode,
Press soft-button [Menu] till you find the [Settings] item.
Enter [Settings] item till you find the [Basic Settings] item.
Enter [Basic Settings] item till you find [Language] item.
User can change the language by using the navigation keys.
6.6 Reboot the Device
When the device is in the default standby mode,
Press soft-button [Menu] till you find the [Reboot System] item.
Press the [OK] or [Enter] button, a warning message “Reboot Now?” will be prompt to user.
Press the [Yes] or [OK] button to execute the reset command, or [No] to exit.
6.7 Reset to Factory Default
When the device is in the default standby mode,
Press soft-button [Menu] till you find the [Settings] item, and press [OK] or [Enter] button.
Enter [Advanced Settings] item, then input the device PIN (Default PIN is 123) to enter the interface.
Enter [Reset to Default] item, then a warning message “Factory Reset, Continues?” will be prompt to user.
Press the [Yes] or [OK] button to execute the reset command, or [No] to exit.
59
Chapter 7. Web Portal
7.1 Web Portal Authentication
User can log in to the device web portal to manage the device or user’s profile. User must provide correct username and password to be able to log in.
7.2 Web Call
Besides making and receiving phone calls by operating the device directly, user could also make and receive phone calls by operating the controls in the web portal.
The web call is particularly suitable for users who use headphone. User can dial out a number by clicking the [Dial] button on top of the title of the web portal page, or click a contact in the phonebook, or click a number in the call logs, to make phone call and press the [Answer] button to answer incoming call. User can click the [Hang up] button to end the call. All these operation can be done from the web portal without touching the phone.
The web call is limited to make a single call. To make or answer a second call is not allowed.
60
Figure 65 - Web Call Controls
Figure 66 - Web Call, Entering Number and Select Line
61
7.3 SYSTEM (Information)
User can get the following system information of the device on this page:
Model
Hardware Version
Software Version
Uptime
Last uptime
MEMInfo
And also summarization of network status:
Network Mode
MAC
IP
Subnet Mask
Default Gateway
Besides, summarization of SIP account status:
SIP User
SIP account status (Registered/Unapplied/Trying/Timeout)
7.4 SYSTEM (Account)
User may change his/her web authentication password on this page.
For users with Administrators privilege, the user can also manage user accounts by adding or deleting user account and assign privilege and password to new account.
There are two types of user privilege, Administrators and Users. If a user account is created as Users privilege, this account will have limited accessibility to the device and cannot change some device settings.
The user account can be used to operate the device or access the device web portal by logging in to the device or its web. User should log in to device web portal with his/her username and web password.
62
The device is shipped with a default user account. The username and password for the default account are “admin” and “123” which are shown on the bottom side of the device.
7.5 SYSTEM (Configurations)
Users with Administrators privilege can export or import the device configuration on this page and reset the device to factory default.
7.6 SYSTEM (Upgrade)
The device supports online upgrade by periodically checking the software release version on the cloud server. Meanwhile, user can download the software and upgrade the device manually when there is trouble for the device to connect to the cloud server.
7.7 SYSTEM (Auto Provision)
The Auto Provision settings help IT manager or service provider to easily deploy and manage the devices in mass volume. For details on Auto Provision, please read “VIP-1120PT Administration
Guide”.
7.8 SYSTEM (Tools)
Troubleshooting for more details.
Besides, the device also provided the screenshot feature to user, including mail screen and sub-screen.
7.9 NETWORK (Basic)
User can configure the network connection type and parameters on this page.
63
7.10 NETWORK (Advanced)
The network advanced settings is often configured by IT manager to enhance the quality of service of the device. The configurations are addressed in “VIP-1120PT Administration Guide”
7.11 NETWORK (VPN)
User may configure a VPN connection on this page. Please refer to 8.1 VPN for more details.
7.12 LINES (SIP)
The service of the line is configured on this page; choose the SIP line (SIP 1 or SIP 2) to configure.
Click the drop-down arrow to adjust configuration accounting on each line.
Table 5 - Line Configuration on Web
Parameters
Basic Settings
Description
Line Status
Display the current line status on the loading page. To get the up to date line status, user has to refresh the page manually.
Enter the username of the service account. Username
Display Name
Authentication Name
Enter the display name to be sent in a call request.
Enter the authentication name of the service account
Authentication Password
SIP Proxy Server Address
SIP Proxy Server Port
Outbound Proxy Address
Enter the authentication password of the service account
Enter the IP or FQDN address of the SIP proxy server
Enter the SIP proxy server port; default is 5060
Enter the IP or FQDN address of outbound proxy server provided by the service provider
Outbound Proxy Port
Realm
Activate
Codec Settings
Enter the outbound proxy port; default is 5060
Enter the SIP domain if requested by the service provider
Whether the service of the line should be activated
Set the priority and availability of the codecs by adding or removing them from the list.
Advanced Settings
Unconditional Call Forward
Enable unconditional call forward; all incoming calls will be forwarded to the number specified in the next field.
64
Parameters Description
Unconditional Call Forward Number Set the number of unconditional call forward.
Call Forward on Busy
Call Forward Number on Busy
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.
Call Forward on No Answer
Call Forward Number on No Answer
Call Forward Delay on No Answer
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.
Enable Hotline
Hotline Number
Hotline Delay
Enable Auto Answering
Auto Answering Delay
Subscribe For Voice Message
Voice Message Number
Voice Message Subscription Period
Enable DND
Blocking Anonymous Call
Use 182 Response for Call Waiting
Anonymous Call Standard
Dial Without Registration
User Agent
Use Quote in Display Name
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 hotline dialing number.
Set the delay for hotline before the system automatically dialed it.
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 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 Do-not-disturb; any incoming call to this line will be rejected automatically.
Reject any incoming call without presenting caller ID.
Set the device to use 182 response code at call waiting response.
Set the standard to be used for anonymous.
Set call out by proxy without registration.
Set the user agent; the default is Model with Software
Version.
Whether to add quote in display name.
65
Parameters
Ring Type
Conference Type
Server Conference Number
Transfer Timeout
Enable Long Contact
Enable Missed Call Log
Response Single Codec
Use Feature Code
Enable DND
Disable DND
Unconditional Enable Call Forward
Unconditional Disable Call Forward
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
Specific Server Type
Registration Expiration
Use VPN
Use STUN
Convert URI
DTMF Type
DTMF SIP Info Mode
Description
Set the ring tone type for the line.
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.
Set the conference room number when conference type is set to be Server.
Set the timeout of call transfer process.
Allow more parameters in contact field per RFC 3840.
If enabled, the phone will save missed calls into the call history record.
If setting enabled, the device will use single codec in response to an incoming call request.
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 line to collaborate with specific server type; this is to be address more in “VIP-1120PT Administration Guide”.
Set the SIP expiration interval.
Set the line to use VPN restrict route.
Set the line to use STUN for NAT traversal.
Convert non digit and alphabet characters to %hh hex code.
Set the DTMF type to be used for the line.
Set the SIP Info mode to send ‘*’ and ‘#’ or ‘10’ and ‘11’.
66
Parameters
Transport Protocol
SIP Version
Caller ID Header
Enable Strict Proxy
Enable user = phone
Enable SCA
Enable BLF List
Enable DNS SRV
Keep Alive Type
Keep Alive Interval
Sync Clock Time
Enable Session Timer
Session Timeout
Enable Rport
Enable PRACK
Keep Authentication
Auto TCP
Enable Feature Sync
Enable GRUU
BLF Server
BLF List Number
SIP Encryption
Description
Set the line to use TCP or UDP for SIP transmission.
Set the SIP version.
Set the Caller ID Header.
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.
Sets user = phone in SIP messages.
Enable/Disable SCA (Shared Call Appearance).
Enable/Disable BLF List.
Set the line to use DNS SRV which will resolve the FQDN in proxy server into a service list.
Set the line to use dummy UDP or SIP OPTION packet to keep NAT pinhole open.
Set the keep alive packet transmitting interval
Time Sync with server
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.
Set the line to add rport to SIP headers.
Set the line to support PRACK SIP message.
Keep the authentication parameters from previous authentication.
Using TCP protocol to guarantee usability of transport for SIP messages above 1500 bytes.
Feature Sync with server.
Supports Globally Routable User-Agent URI (GRUU).
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.
BLF List allows one BLF key to monitor the status of a group.
Multiple BLF lists are supported.
Enable SIP encryption such that SIP transmission will be encrypted
67
Parameters
SIP Encryption Key
RTP Encryption
RTP Encryption Key
Description
Set the pass phrase for SIP encryption
Enable RTP encryption such that RTP transmission will be encrypted
Set the pass phrase for RTP encryption
7.13 LINES (Dial Peer)
This functionality offers you more flexible dial rule; you can refer to the following contents to know how to use this dial rule.
Parameters
Phone number
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.
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
Port
Alias
Set Destination address. This is for IP direct.
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
Delete Length
Characters to be added at the end of the phone number. It is an optional item.
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.
68
Examples of different alias applications
This feature allows the user to create rules to make dialing easier. There are several different options for dial rules. The examples below will show how it works.
Example 1: Global Substitution
It seems like a shortcut to dial out. When user dials “32”, the dialed number will be replaced by
“833333”. But if user dials “322”, the device will still send “322” rather than “8333332”. The replacement rules should be matched globally.
Figure 67 - Global Substitution Configuration
69
Example 2: Local Substitution
To dial a long-distance call to Taipei requires dialing area code 02 before the local phone number.
Using this feature 1 can be substituted for 02. For example, to call 62213123 would only require dialing
162213123 instead of 0262213123.
Figure 68 - Local Substitution Configuration
70
Example 3: Add Prefixes
If the dialed number starts with the fixed prefix number, the phone will send out your dialed phone number adding prefix number automatically.
For example, when users dial “9312”, the device will send out “0079312”.
Figure 69 - Add Prefixes Configuration
71
Example 4: Add Suffixes
If the dialed number ends with the fixed suffix number, the phone will send out your dialed phone number adding suffix number automatically.
For example, when users dial “1383322”, the device will send out “13833220088” .
Figure 70 - Add Suffixes Configuration
72
Example 5: Deletion
If the dialed number ends with the fixed prefix number, the phone will send out your dialed phone number deleting prefix number automatically.
For example, when users dial “98322”, the device will send out “8322”
Figure 71 - Deletion Configuration
7.14 LINES (Dial Plan)
Figure 72 - Dial Plan Configuration
73
The device supports 7 dialing modes:
Press # to Send -- Dial the desired number, and press # to send it to the server.
Dial Fixed Length -- Configure the fixed length to dial out
Send after seconds -- Number will be sent to the server after the specified time.
Press # to Do Blind Transfer -- Press # after entering the target number for the transfer. The phone will transfer the current call to the third party.
Blind Transfer on Onhook -- Hang up after entering the target number for the transfer. The phone will transfer the current call to the third party.
Attended Transfer on Onhook -- Hang up after the third party answers. The phone will transfer the current call to the third party.
Attended Transfer on Conference Onhook -- Hang up during a 3-way conference call, the other two ways will make a call.
Press DSS Key to Do Blind Transfer -- When user is in the ‘XFER’ screen, user can fulfill Blind
Transfer by pressing the DSS key.
7.15 LINES (Global Settings)
Configure global settings for lines,
Table 6 - Global Settings for Lines on Web
Parameters
SIP Settings
Description
Local SIP Port
Registration Failure Retry Interval
Set the local SIP port used to send/receive SIP messages.
Set the retry interval of SIP REGISTRATION when registration failed.
STUN Settings
Server Address
Server Port
Set the STUN server address.
Set the STUN server port; default is 3478.
Binding Period
SIP Waiting Time
TLS Certification File
Set the STUN binding period which can be used to keep the NAT pinhole open.
Set the timeout of STUN binding before sending
SIP messages.
Upload or delete the TLS certification file used for encrypted SIP transmission.
74
7.16 PHONE (Features)
Configure the phone features and common Settings:
Table 7 - Common Phone Feature Settings on Web
Parameters
Enable Phone DND
Description
Configure the Phone DND.
If enable Phone DND, the phone rejects any incoming call; the caller will automatically be prompted to hang up.
Ban Outgoing
Enable Call Transfer
Semi-Attended Transfer
Enable 3-Way Conference
Enable Auto Hand Down
If you select Ban Outgoing and enable it, and you cannot dial out any number.
Enable Call Transfer by selecting it.
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 in hands-free mode.
Auto Hand Down Time
Enable Auto Redial
Auto Redial Interval
Auto Redial Times
Enable Call Waiting
Enable Call Waiting Tone
Enable Call Completion
Enable Pre-Dial
Hide DTMF
Auto Headset
Specify Auto hand down time, the phone will hang up and return to the idle automatically after Auto Hand down time in hands-free mode, and play dial tone Auto hand down time in handset mode.
Enable Auto Redial by selecting it, then the phone reminds whether redial, when the dialed line is busy or rejects.
Configure the Auto redial interval.
Configure the Auto redial times.
Enable this setting to allow user to take second incoming call during an established call. Default enabled.
Turn off this feature, and you will not hear a ‘beep’ sound in talking mode when there is another incoming call.
Enable Call Completion by selecting it; if the dialed line is busy, the sip server will inspect the dialed line status at intervals. If the dialed line is idle, the server will send notification message to inform the caller whether to redial.
To disable this feature, enter the number to open audio channel automatically.
To enable the feature, enter the number without opening audio channel.
Configure the hide DTMF mode.
Enable this feature; headset is plugged in the phone. User presses
‘answer’ key or line key to answer a call with the headset automatically.
75
Parameters
Ring From Headset
Enable Intercom
Enable Intercom Mute
Enable Intercom Tone
Enable Intercom Barge
Emergency Call Number
Enable Password Dial
Encryption Number Length
Password Dial Prefix
Restrict Active URI Source
IP
Push XML Server
Allow IP Call
Enable Multi Line
Enable Default Line
Enable Auto Switch Line
DND Response Code
Busy Response Code
Reject Response Code
Play Talking DTMF Tone
Description
Enable Ring From Handset by selecting it; the phone plays ring tone from handset.
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 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.
Configure the Emergency Call Number. Despite the keyboard is locked, you can dial the emergency call number.
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 stand 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 the prefix of the password call number.
Set the device to accept Active URI command from specific IP address.
This feature is usually used for device management and will be described in “VIP-1120PT Administration Guide”.
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.
If enabled, user can dial out with IP address.
Enable phone to make calls for 10 lines max, or disable for 2 lines max.
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.
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.
Play DTMF tone on the device when user pressed a phone digits during
76
Parameters
Play Dialing DTMF Tone
Caller ID Display Priority
Display Num With Name
Description taking; default enabled.
Play DTMF tone on the device when user presses a phone digits at dialing; default enabled.
Change caller ID display priority. The default priority is “Phonebook” >
“SIP Display Name” > “SIP URI”. User may select one of the options to change the desired caller ID display priority.
Allow user to determine the caller-id display method.
If it is checked, the caller-id will display number with name.
If the value is unchecked, the caller-id will abide by CID Priority..
Action URL
Action URL is used for device event submitting with IPPBX system. The detail is addressed in the “VIP-1120PT Administration Guide”.
Restrict Outgoing Call
The user with Administrators privilege can set the rule to restrict some numbers from dialing out. This setting is device-wide and applies to every user of the device.
User can add restricted number by entering the number in the edit box and click the “Add” button. To remove a number from the restriction list, select the number from the dropdown box and click the
“Delete” button to remove it.
7.17 PHONE (Audio)
Change the audio settings:
Parameters
First Codec
Second Codec
Third Codec
Table 8 - Audio Settings on Web
Description
The first preferential DSP codec:G.711A/U,G.722,G.723,G.729,G.726-32,
ILBC,AMR,AMR-WB
The second preferential DSP codec:
G.711A/U,G.722,G.723,G.729,G.726-32,
ILBC,AMR,AMR-WB,NONE
The third preferential DSP codec:
77
Parameters
Fourth Codec
Fifth Codec
Sixth Codec
Onhook Time
Tone Standard
Default Ring Type
Handset Volume
Headset Volume
Headset Ring Volume
Speakerphone Volume
Speakerphone Ring Volume
Headset Volume Offset
Headset Mic Offset
G.729AB Payload Length
G.723.1 Bit Rate
G.722 Timestamps
DTMF Payload Type
Enable VAD
Enable MWI Tone
Auto Answer By Headset
Description
G.711A/U,G.722,G.723,G.729,G.726-32 ,
ILBC,AMR,AMR-WB,NONE
The forth preferential DSP codec:
G.711A/U,G.722,G.723,G.729,G.726-32 ,
ILBC,AMR,AMR-WB,NONE
The fifth preferential DSP codec:
G.711A/U,G.722,G.723,G.729,G.726-32,
ILBC,AMR,AMR-WB,NONE
The sixth preferential DSP codec:
G.711A/U,G.722,G.723,G.729,G.726-32,
ILBC,AMR,AMR-WB,NONE
Configure the least reflection time of Hand down, the default is
200ms.
Set the country standard of call progress tones, including dial tone, busy tone, ring-back tone, etc.
Set the default ring type. If the caller ID of an incoming call was not configured with specific ring type, the default ring will be used.
Set the Handset volume; the value must be 1~9.
Set the Headset volume; the value must be 1~9.
Set the ring volume in the headset; the value must be 1~9.
Set the speakerphone volume; the value must be 1~9.
Set the ring volume in the speakerphone; the value must be 1~9.
This is to adjust the base volume of the headset. Please note when set the volume at the maximum level it may create noise and decrease the echo canceller.
This is to adjust the base volume of the headset Mic.
Set G729 Payload Length.
5.3kb/s or 6.3kb/s is available.
160/20ms or 320/20ms is available.
Enter the DTMF payload type; the value must be 96~127.
Enable Voice Activity Detection. When enabled, the device will suppresses the audio transmission with artificial comfort noise signal to save the bandwidth.
The phone will play MWI tone when a new MWI comes.
When this item is checked, the device will auto-answer phone calls by
78
Parameters Description headset if the auto-answer or intercom is enabled.
7.18 PHONE (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(es) without involving SIP signaling. You can also configure the phone to receive an RTP stream from pre-configured multicast listening address(es) without involving SIP signaling. You can specify up to 10 multicast listening addresses.
Table 9 - MCAST Parameters on Web
Parameters
Normal Call Priority
Enable Page Priority
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.
Name
Host: port
Multicast server name
Multicast server’s multicast IP address and port.
7.19 PHONE (Time/Date)
User can configure the device time settings on this page.
Table 10 - Time/Date Setting Parameters on Web
Parameters
Network Time Server Settings
Description
Time Synchronized via SNTP Enable time-sync through SNTP protocol.
Time Synchronized via DHCP Enable time-sync through DHCP protocol.
Primary Time Server Set primary time server address.
Secondary Time Server
Set secondary time server address; when primary server is not reachable, the device will try to connect to secondary time server to get time synchronization.
Time Zone
Resync Period
12-hour Clock
Select the time zone.
Time of re-synchronization with time server.
Set the time display in 12-hour mode.
79
Parameters
Date Format
Daylight Saving Time Settings
Location Select the user's time zone specific area
DST Set Type
Offset
Month Start
Description
Select the time/date display format.
Select automatic DST according to the preset rules of DST, or the manually input rules
The DST offset time
The DST start month
The DST start week Week Start
Weekday Start
Hour Start
The DST start weekday
The DST start hour
Minute Start
Month End
Week End
Weekday End
Hour End
The DST start minute
The DST end month
The DST end week
The DST end weekday
The DST end hour
7.20 PHONE (Advanced)
User can configure the advanced configuration settings on this page.
Screen Configuration.
Contrast
Backlight Time
LCD Menu Password Settings.
The password is 123 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 ‘ PLANET PHONE ’.
7.21 PHONEBOOK (Contacts)
User can add, delete, or edit contacts in the phonebook on this page. User can browse the phonebook and sort it by name, phones, or filter them out by group.
To add a new contact, user should enter contact’s information and press the “Add” button to add it.
80
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 the “Modify” button after finishing editing.
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 by 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 the “Add to Group” button at the bottom of the contact list. Select contacts by checking the checkbox and click “Add to Group” to add selected contacts to the group.
Similarly, user can select multiple users and add them to blacklist by clicking the “Add to Blacklist” button.
7.22 PHONEBOOK (Cloud Phonebook)
Cloud Phonebook
User may 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)
In regard to creating a cloud phonebook and setting up a cloud phonebook server, please refer to “VIP-1120PT Administration Guide”.
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.
81
To configure an 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)
In regard to creating an LDAP phonebook and setting up an LDAP phonebook server, please refer to “VIP-1120PT Administration Guide”.
7.23 PHONEBOOK (Blacklist)
By adding a number to the blacklist, user will no longer receive phone call from that number and it will be rejected automatically by the device until user deletes it from the blacklist.
User can add specific number to be blocked, or a prefix where any numbers matched the prefix will all be blocked.
7.24 PHONEBOOK (Advanced)
User may export current phonebook in xml, csv, or vcf format file and save it locally on a computer.
User can also import contacts into phonebook from an xml, csv, or vcf file.
If user repeatedly imports a same phonebook, it will create duplicated contacts in user’s phonebook.
User can add new group on this page or delete an existing one. Deleting a contact group will not delete the contacts in that group.
7.25 CALL LOGS
User can browse complete call logs on this page, order the call logs by time, caller ID, contact name, duration, or line, and can also filter the call logs by the call log types, in, out, missed, or all.
User can save a call log into his/her phonebook or add it to the blacklist.
User can also make web call by clicking on the number of a call log.
82
7.26 FUNCTION KEY
The device provides 2 users -- define DSS Keys at most. User may configure/customize each DSS key on this webpage.
Table 11 - DSS Key Setting Parameters on Web
Parameters Description
Memory Key
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
Line
Key Event
DTMF
URL
Multicast
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.
83
7.27 FUNCTION KEY (Softkey)
User can configure different functions in different screens for each softkey.
Figure 73 - Softkey Configuration
84
Chapter 8. Advanced Features
8.1 VPN
Virtual Private Network (VPN) is a technology to allow device to create a tunneling connection to a server and becomes part of the server’s network. The network transmission of the device may be routed through the VPN server.
For some users, especially enterprise users, a VPN connection might be required to be established before activate a line registration. The device supports two VPN modes, Layer 2 Transportation
Protocol (L2TP) and OpenVPN.
The VPN connection must be configured and started (or stopped) from the device web portal.
8.2 L2TP
The device only supports non-encrypted basic authentication and non-encrypted data tunneling. For users who need data encryption, please use OpenVPN instead.
To establish a L2TP connection, users should log in to the device web portal, open page [Network] ->
[VPN]. In VPN Mode, check the “Enable VPN” option and select “L2TP”, then fill in the L2TP server address, Authentication Username, and Authentication Password in the L2TP section. Press “Apply” then the device will try to connect to the L2TP server.
When the VPN connection established, the VPN IP Address should be displayed in the VPN status.
There may be some delay of the connection establishment. User may need to refresh the page to update the status.
Once the VPN is configured, the device will try to connect to the VPN automatically when the device boots up every time until user disable it. Sometimes, if the VPN connection does not established immediately, user may try to reboot the device and check if VPN connection established after reboot.
85
8.3 OpenVPN
To establish an OpenVPN connection, user should get the following authentication and configuration files from the OpenVPN hosting provider and name them as follows:
OpenVPN Configuration file: client.ovpn
CA Root Certification: ca.crt
Client Certification:
Client Key: client.crt client.key
User then upload these files to the device on the web page [Network] -> [VPN], Section OpenVPN Files.
Then user should check “Enable VPN” and select “OpenVPN” in VPN Mode and click “Apply” to enable
OpenVPN connection.
Like an L2TP connection, the connection will be established every time when system reboots until user disables it manually.
86
Chapter 9. Troubleshooting
When the device does not work properly, users may try the following methods to recover the device or gather related information.
9.1 Get Device System Information
Users may get the device system information by pressing [Settings] -> [Status]. The following basic information will be provided:
Mode
IP
Software Version
More…
User can select [More] item to get more information in details.
9.2 Upgrade to the latest software
Manufacturer will keep publishing software update to fix bugs and improve device features. The device will check for new software release on manufacturer cloud server automatically and periodically.
9.3 Reboot Device
Users may reboot the device from soft-menu, [Menu] -> [Reboot System], and confirm the action by
[Yes]. Or, simply remove the power supply and restore it again.
9.4 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 [Menu] -> [Settings] -> [Advanced settings], and then input the password to enter the interface. Then choose [Reset to Default] and press [Enter], and confirm the action by [Yes]. The device will be rebooted into a clean factory default state.
87
9.5 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 to 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 the [Stop] button on the web page when operation finishes. The network packets of the device during the period have been dumped to the saved file. User may examine the packets with a packet analyzer or send it to support.
9.6 Common Trouble Cases
Trouble Case
Device could not boot up
Table 12 - Trouble Cases
Solution
1. The device is powered by external power supply via power adapter or PoE switch. Please use standard power adapter provided or PoE switch met with the specification requirements and check if device is well connected to power source.
1. Please check if device is well connected to the network. The Device could not register to a service provider network Ethernet cable should be connected to the port, NOT the
[Network]
[PC] port. If the cable is not well connected to the network icon middle of the screen.
[WAN disconnected] will be flashing in the
2. Please check if the device has an IP address. Check the system information, if the IP displays “Negotiating…”, the device does not have an IP address. Please check if the network configurations is correct.
No Audio or Poor Audio in
Handset
1. Please check if Handset is connected to the correct Handset ( ) port, NOT Headphone ( ) port.
2. The network bandwidth and delay may be not suitable for audio call at the moment.
Poor Audio or Low Volume in 1. There are two Headphone wire sequences in the market. Please
Headphone use the Headphone provided, or consults the wire sequence if you wish to use a third-party headphone.
2. The network bandwidth and delay may be not suitable for audio call at the moment.
88
Trouble Case
Audio is chopping at far-end in Hands-free speaker mode
Solution
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.
89
Appendix I - Icon Illustration
+
-
Table 13 - Keypad Icons
Redial
Handsfree (HF) speaker
Phone lock = Long-pressed Key(#)
Mute Microphone (During Call)
Volume Down/
Volume Up
Table 14 - Status Prompt and Notification Icons
Call out
Call in
Call Hold
Network Disconnected
Keypad Locked
Missed calls
SMS
New voice message waiting
Do-Not-Disturb activated on Phone
Auto-answering activated
Call forward activated
90
Handsfree (HF) Mode
Headphone (HP) Mode
Handset (HS) Mode
Microphone Muted
Mixed type Input
Lower Case
Upper Case
Numeric Input
91
Appendix II - Text Input from Keypad
Mode
Icon
Text
Mode
Numeric
Table 15 - Look-up Table of Characters
Lower Case
Alphabets
Key
Button
1
2
8
9
0
5
6
7
*
1
2
3
4
0
*
#
6
7
8
9
3
4
5
1
2
3
4
5
6
7
8
9
0
＊
#(space)@,.:/?<>[]%! &$~+-
#
+ - * % / | @ ! : a b c d e f g h i j k l m n o p q r s t u v w x y z
0
＊
Characters Of
Each Press
#(space)@,.:/?<>[]%! &$~+-
92
Mode
Icon
Text
Mode
Upper Case
Alphabets
Mixed type
Input
8
9
0
5
6
7
*
#
1
2
3
4
0
*
#
6
7
8
9
3
4
5
Key
Button
#
1
2
Characters Of
Each Press
#
+ - * % / | @ ! :
A B C
D E F
G H I
J K L
M N O
P Q R S
T U V
WZYX
(space)
＊
#(space)@,.:/?<>[]%! &$~+-
#
1 + - * % / | @ ! :
2 a b c A B C
3 d e f D E F
4 g h I G H I
5 j k l J K L
6 m n o M N O
7 p q r s P Q R S
8 t u v T U V
9 w z y x W Z Y X
(space)
＊
#(space)@,.:/?<>[]%! &$~+-
#
93
Appendix III - LED Definition
Line Key
BLF
MWI
Type
DND
Table 16 - DSS KEY LED State
LED Light
Off
Green On
Green Blinking
Line inactive
State
Line ready (Registered)
Dialing / Ringing / Call holding
Green On
Red On
Red Blinking
Off
Red On
Off
Green On
Off
Subscribed number is idle.
Subscribed number is busy.
Subscribed number is dialing.
Subscribed number is unavailable
Enable DND
Disable DND
New voice message waiting
No new voice message
94
advertisement
* Your assessment is very important for improving the workof artificial intelligence, which forms the content of this project
