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VigorPhone 300
User’s Guide
Version: 1.0
Date: 05/03/2012
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Copyright Information
Copyright
Declarations
Copyright 2012 All rights reserved. This publication contains information that is
protected by copyright. No part may be reproduced, transmitted, transcribed, stored in a
retrieval system, or translated into any language without written permission from the
copyright holders.
Trademarks
The following trademarks are used in this document:
z Microsoft is a registered trademark of Microsoft Corp.
z Windows, Windows 95, 98, Me, NT, 2000, XP, Vista and Explorer are trademarks of
Microsoft Corp.
z Apple and Mac OS are registered trademarks of Apple Inc.
z Other products may be trademarks or registered trademarks of their respective
manufacturers.
Safety Instructions and Approval
Safety
Instructions
z Read the installation guide thoroughly before you set up the device.
z The router is a complicated electronic unit that may be repaired only be authorized
and qualified personnel. Do not try to open or repair the router yourself.
z Do not place the device in a damp or humid place, e.g. a bathroom.
z The device should be used in a sheltered area, within a temperature range of +5 to
+40 Celsius.
z Do not expose the device to direct sunlight or other heat sources. The housing and
electronic components may be damaged by direct sunlight or heat sources.
z Do not deploy the cable for LAN connection outdoor to prevent electronic shock
hazards.
z Keep the package out of reach of children.
z When you want to dispose of the device, please follow local regulations on
conservation of the environment.
Warranty
We warrant to the original end user (purchaser) that the device will be free from any
defects in workmanship or materials for a period of two (1) years from the date of
purchase from the dealer. Please keep your purchase receipt in a safe place as it serves
as proof of date of purchase. During the warranty period, and upon proof of purchase,
should the product have indications of failure due to faulty workmanship and/or
materials, we will, at our discretion, repair or replace the defective products or
components, without charge for either parts or labor, to whatever extent we deem
necessary tore-store the product to proper operating condition. Any replacement will
consist of a new or re-manufactured functionally equivalent product of equal value, and
will be offered solely at our discretion. This warranty will not apply if the product is
modified, misused, tampered with, dam aged by an act of God, or subjected to abnormal
working conditions. The warranty does not cover the bundled or licensed software of
other vendors. Defects which do not significantly affect the usability of the product will
not be covered by the warranty. We reserve the right to revi se the m anual and onli ne
documentation and to make changes from time to time in the contents hereof without
obligation to notify any person of such revision or changes.
Be a Registered
Owner
Web registration is preferred. You can register your device via
http://www.draytek.com.
Firmware & Tools
Updates
Due to the continuous evolution of DrayTek technology, all devices will be regularly
upgraded. Please consult the DrayTek web site for more information on newest
firmware, tools and documents.
http://www.draytek.com
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Regulatory Information
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 the 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 televisio n recept i on , whi ch can be determined by turning the equipment of f and
on, the use is encouraged to try to correct the interference by one of the following measures:
z Reorient or relocate the receiving antenna.
z Increase the separation between the equipment and receiver.
z Connect the equipment into an outlet on a circuit different form that to which the receiver is connected.
z Consult the dealer or an experienced radio/TV technician for help.
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 may accept any interference received, including interference that may cause undesired operation.
Please visit http://www.draytek.com/user/AboutRegulatory.php.
CE Notice (European Union)
The symbol indicates compliance of this equipment to the EMC Directive and the Low Voltage Directive of
the European Union. These markings indicate that this system meets the following technical standards:
z EN 55022 — “Limits and Methods of Measurement of Radio Interference Characteristics of Information
Technology Equipment.”
z EN 55024 — “Information technology equipment - Immunity characteristics - Limits and methods of
measurement.”
z EN 61000-3-2 — “Electromagnetic compatibility (EMC) - Part 3: Limits - Section 2: Limits for
harmonic current emissions (Equipment input current up to and including 16 A per phase).”
z EN 61000-3-3 — “Electromagnetic compatibility (EMC) -Part 3: Limits - Section 3: Limitation of
voltage fluctuations and flicker in low-voltage supply systems for equipment with rated current up to and
including 16 A.”
z EN 60950 — “Safety of Information Technology Equipment.”
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Table of Contents
Chapter 1 Overview ...........................................................................................................7
1.1 Package Contents..................................................................................................................7
1.2 Product Description................................................................................................................8
1.2.1 Front View....................................................................................................................................8
1.2.2 Back View....................................................................................................................................9
1.3 Setting Up the Phone ...........................................................................................................10
Chapter 2 Display Screen Configuration.......................................................................11
2.1 Memory Key .........................................................................................................................11
2.1.1 Dialing with Memory Key...........................................................................................................11
2.1.2 Edit the Memory Key .................................................................................................................12
2.1.3 Exit the Memory Key..................................................................................................................13
2.2 Do Not Disturb......................................................................................................................13
2.3 Speed Dial............................................................................................................................13
2.4 Phone Book..........................................................................................................................15
2.5 Incoming/Outgoing Call........................................................................................................17
2.6 Missed Call and Indicator Light ............................................................................................19
Chapter 3 Web Configuration .........................................................................................21
3.1 Basic.....................................................................................................................................22
3.1.1 Status.........................................................................................................................................22
3.1.2 Wizard........................................................................................................................................23
3.1.3 Call Log......................................................................................................................................29
3.1.4 MMI Set .....................................................................................................................................29
3.2 Network ................................................................................................................................30
3.2.1 WAN .........................................................................................................................................31
3.2.2 LAN.. .........................................................................................................................................34
3.2.3 QOS. .........................................................................................................................................35
3.2.4 Service Port...............................................................................................................................37
3.2.5 DHCP Server.............................................................................................................................38
3.2.6 SNTP.........................................................................................................................................40
3.3 VoIP......................................................................................................................................42
3.3.1 SIP... .........................................................................................................................................42
3.3.2 IAX2. .........................................................................................................................................48
3.3.3 STUN.........................................................................................................................................49
3.3.4 Dial Peer....................................................................................................................................50
3.4 Phone...................................................................................................................................55
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3.4.1 DSP . .........................................................................................................................................55
3.4.2 Call Service................................................................................................................................58
3.4.3 Digital Map.................................................................................................................................60
3.4.4 Phone Book...............................................................................................................................62
3.4.5 Function Key..............................................................................................................................63
3.5 Maintenance.........................................................................................................................65
3.5.1 Auto Provision............................................................................................................................65
3.5.2 Syslog........................................................................................................................................66
3.5.3 Config ........................................................................................................................................68
3.5.4 Update.......................................................................................................................................69
3.5.5 Account......................................................................................................................................70
3.5.6 Reboot.......................................................................................................................................71
3.6 Security.................................................................................................................................71
3.6.1 MMI Filter...................................................................................................................................72
3.6.2 Firewall ......................................................................................................................................72
3.6.3 NAT.. .........................................................................................................................................75
3.6.4 VPN . .........................................................................................................................................77
3.7 Logout...................................................................................................................................78
Chapter 4 Operation ........................................................................................................79
4.1 Set up VigorPhone 300 with VigorIPPBX Series..................................................................79
4.2 Answer Call ..........................................................................................................................84
4.3 Place Calls............................................................................................................................85
4.4 End Calls..............................................................................................................................86
4.5 Call Transfer.........................................................................................................................87
4.6 Call Hold...............................................................................................................................87
4.7 3-way Conference Call.........................................................................................................87
4.8 Call Records.........................................................................................................................87
4.9 Special Keys.........................................................................................................................88
4.10 Call Pickup..........................................................................................................................89
4.11 Join Call..............................................................................................................................89
4.12 Redial/Un-redial..................................................................................................................90
4.13 Click to Dial.........................................................................................................................90
Appendix A Specifications..............................................................................................91
A.1 Specification.........................................................................................................................91
A.2 Digit-character Map Table....................................................................................................92
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Chapter 5 Operation
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CChhaapptteerr 11 OOvveerrvviieeww
VigorPhone enables you to make phone calls through the IP network instead of calling through a
tradition local PSTN line.
It is workable with VigorIPPBX series for auto provision capability. To manage various calling
purposes, VigorPhone supports multi-sip registration with different accounts (up to 10) and support
G.722 codec for promoting voice quality. The simple WEB UI based configuration allows you to
operate VigorIPPBX with ease.
Read this user manual carefully to learn how to operate this product and take advantage of its features.
11..11 PPaacckkaaggee CCoonntteennttss
When unpacking the VoIP phone, ensure all the following items are present and undamaged. If anything
appears to be missing or broken, contact your dealer for a replacement.
n IP Phone
o CD
p Quick Start Guide q RJ-45 Cable
(Ethernet)
r
Phone Stand /Bracket
s
Handset
t
UK-type power adapter
USA/Taiwan-type
power adapter
EU-type power adapter
AU/NZ-type Power
Adapter
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11..22 PPrroodduucctt DDeessccrriippttiioonn
This VoIP Phone features based on SIP (RFC 3261). Please familiarize yourself with the functions of
the VoIP phone.
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Item Name Description
1 Memory key (1-6) Users could store their commonly used number in these keys, and call
for them as speed dial.
2 Display Screen Displays calls and status information.
3 Soft key 1/2/3 Keys combination, include functions such as SMS / SDial /PBook
/Answer /Conf /Enter /Save /Quit /Edit /Redial and so on.
4 Navigation Left: Checking Incoming call / Up: Checking Missed Call
Right: Checking line status / Down: Checking IP info
OK: Enter into the phone's menu
5 Release key Skip to stand-by mode.
6 Mute Press this key in calling mode, you can hear the other side, and the other
side can not hear you.
7 Envelope LED inside, if blinks remind user have new voicemail.
8 HeadSet Button Place and receive calls through an optionally connected headset.
9 Transfer Use the key to realize blind transfer or attended transfer.
10 Hold Temporarily hold the active call during the talking.
11 Volume -/+ Turn down or turn up the volume by pressing these two keys
12 Headset Jack Allow to connect another headset optionally. (Port type: 3.5mm jack)
13 Hands-free Make the phone into hands-free mode.
14 Redial
z In the hook off /hands-free mode, use the key to dial the last call
number.
z In stand-by mode, it has a function to check the OUTGOING
CALL.
15 Line1/2/3 Three SIP lines allow you to select any one to make the call, if it has
been registered.
16 Indicator light If the light blinking, indicate the phone has missed call(s).
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11..22..22 BBaacckk VViieeww
Item Name Description
1 Headset Jack Connects to an external headset.(RJ-9)
2 Handset Jack Connects to the phone.(RJ-9)
3 LAN/PC Port Connects to PC. 10/100Mbps RJ-45 port for PC (downlink)
connection. Connects to LAN cable.
4 WAN/ PoE Port 10/100Mbps RJ-45 port for LAN (uplink) connection. If you are
using Power over Ethernet (PoE), the power to the phone is supplied
when you connect the Ethernet cable. Draws power from either spare
line or signal line.
5 Power Jack Connects to AC power adapter. 5V AC power port.
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11..33 SSeettttiinngg UUpp tthhee PPhhoonnee
The following illustration shows how to connect the VoIP phone to power, LAN, WAN, and the handset
or a headset.
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CChhaapptteerr 22 DDiissppllaayy SSccrreeeenn CCoonnffiigguurraattiioonn
The menu directory enables you to setup the product configuration from Phone Settings, VoIP settings,
and Network settings. Follow these steps to access the menu and the menu items.
Below shows the LCD of VigorPhone (successful hardware connection):
22..11 MMeemmoorryy KKeeyy
Memory keys (also called function keys in the web configurator of Phone>>Function Key) can be set
with specific type, value, line and other function parameters (speed dial, push to talk, DND and so on).
You can go to Phone>>Function Key to configure the settings in details.
If you just want to edit the name and /or the number for each memory key, you can click one of the
memory keys on the IP phone to change it.
22..11..11 DDiiaalliinngg wwiitthh MMeemmoorryy KKeeyy
Simply press the memory key (1 – 6) you want and click Dial.
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22..11..22 EEddiitt tthhee MMeemmoorryy KKeeyy
1. Click one of the memory keys you want. In default, all the telephone numbers will be displayed with
Null if you haven’t created any memory key.
Button Explanation
Dial Have a phone call to the selected one.
Edit Modify the information for the selected one.
Quit Exit and return to previous page.
2. Click the soft key under Edit. The name and the number will be cleared and ask you to type new
entries.
3. In the field of Name, please type Nick; and in the field of Number, please type 668.
Button Explanation
Delete It allows you to remove the information you type.
Save Save the information you type.
Quit Exit and return to previous page.
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4. Click the soft key under Save to store the settings. Now, memory key 1 has been changed with new
name and number.
22..11..33 EExxiitt tthhee MMeemmoorryy KKeeyy
Simply press the soft key under Quit to exit the memory key and return to the home page.
22..22 DDoo NNoott DDiissttuurrbb
Simply press the soft key under DND button on the home page. The screen will be shown as below.
Now, you don’t need to worry about the incoming phone calls to interrupt your work.
22..33 SSppeeeedd DDiiaall
Speed dial means user can make calls directly without hook off or using hands-free.
1. Press the soft key under SDial to access into the configuration page. There are 12 groups that you
can set as speed dial numbers.
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2. Use the Navigation keys to move up, down, left or right to choose the one you want. In this case,
we choose #1 as an example.
Button Explanation
>> Click it to access into next entry.
Edit Modify the information for the selected one.
Quit Exit and return to previous page.
3. Next, click the soft key under Edit to display the following screen. In the field of Name, please type
Mark; and in the field of Tel, please type 667.
Button Explanation
Delete It allows you to remove the information you type.
Save Save the information you type.
Quit Exit and return to previous page.
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4. Click the soft key under Save to store the settings. Now, speed dial # 1 has been changed with new
name and number.
22..44 PPhhoonnee BBooookk
1. Press the soft key under PBook to access into the configuration page.
2. For there is no phone book created, the LCD displays the message of “List Is Empty”.
Button Explanation
Add It allows you to add a new name and telephone number to the phone
book.
Enter This button is available only when there is at least one item existed. If
not, it will be blank.
Quit Exit and return to previous page.
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3. Click the soft key under Add to display the following screen. In the field of Name, please type John;
and in the field of Tel, please type 660.
Button Explanation
Delete It allows you to remove the information you type.
Save Save the information you type.
Quit Exit and return to previous page.
4. Click the soft key under Save. When such item is created successfully, the screen will display as the
figure below.
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5. Click the soft key under Quit. You will find a new name with phone number has been created.
Button Explanation
Option It allows you to edit information, save the phone book, delete the phone
book, send a message to other people and so on.
Dial Have a phone call to the selected one.
Quit Exit and return to previous page.
22..55 IInnccoommiinngg//OOuuttggooiinngg CCaallll
Later incoming/outgoing calls will be stored temporarily and be checked from the Display Screen.
1. Press the navigation key
. You will see the incoming call records at the first. If there are many
incoming call stored, please use scroll bard on the right side of the display screen to scroll up and
down.
Button Explanation
Next Switch among the incoming call, outgoing call and missed call.
Enter This button is available only when there is at least one item existed.
Quit Exit and return to previous page.
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2. Press the soft key under Enter to access into the next page of incoming call. See the figure below.
Button Explanation
Option It allows you to check detailed information for the missed call, save the
missed call, delete the missed call, send a message to the missed call,
and so on.
Dial Call back for answering the incoming call.
Quit Exit and return to previous page.
3. Click the soft key under Dial to have a phone call to the selected incoming call.
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22..66 MMiisssseedd CCaallll aanndd IInnddiiccaattoorr LLiigghhtt
If the indicator light always blinks, it means there is a missed call that you have to process. Please do
the following:
1. Press the navigation key
to open the missed call record. All the missed calls will be shown
on the display screen. In this example, there is only one missed call.
Button Explanation
Option It allows you to check detailed information for the missed call, save the
missed call, delete the missed call, send a message to the missed call,
and so on.
Dial Call back for answering the missed call.
Quit Exit and return to previous page.
2. Click the soft key under Dial to have a phone call to the selected missed call.
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CChhaapptteerr 33 WWeebb CCoonnffiigguurraattiioonn
This chapter contains important information to help you configure the settings for your VoIP phone
from the web browser.
If your VoIP phone is using factory default, it sets LAN/PoE port as DHCP client and enable Bridge
mode for PC port. To access the web configuration menu, do the following:
1. Connect one end of the Ethernet cable provided to the LAN/PoE port of your phone to your router
with DHCP service enable.
2. Connect one end of the network cable to the PC port of your phone, connecting to your personal
computer.
3. Plug in the power of the VoIP phone. Select the Menu soft key.
4. Select Network, LAN Port Settings, and then press the Info soft key.
You should be able to see the IP address displayed on the LCD screen.
Open your browser (such as Internet Explorer, Firefox, etc.) and type in the web address of the phone.
For example, if the IP address you obtain in step 4 above is 192.168.1.2, enter the web address: http://
192.168.1.2.
The Web login front page is displayed. Enter the user name (“admin”) and the password (“admin”) and
click Login.
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After you login, move the cursor over the menu items on the left navigation bar to access the dropdown
menus.
33..11 BBaassiicc
33..11..11 SSttaattuuss
Field name Explanation
Network Shows the configuration information on WAN and LAN port, including
the connect mode of WAN port (Static, DHCP, PPPoE), MAC address,
the IP address of WAN port and LAN port, ON or OFF of DHCP mode
of LAN port.
Phone Number Shows the phone numbers provided by the SIP LINE 1-3 servers and
IAX2.
The last line shows the version number and issued date.
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Please select the proper network mode according to the network condition. VigorPhone provides three
different network settings.
Field name Explanation
Static IP Mode If your ISP server provides you the static IP address, please select this
mode, then finish Static Mode setting. If you don’t know about
parameters of Static Mode setting, please ask your ISP for them.
DHCP In this mode, you will get the information from the DHCP server
automatically; need not to input this information artificially.
PPPoE In this mode, your must input your ADSL account and password.
Static IP Mode
1. Choose Static IP Mode and click Next. You can get the following web page.
Field name Explanation
Static IP Address Input the IP address distributed to you.
Netmask Input the Netmask distributed to you.
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Gateway Input the Gateway address distributed to you.
DNS Domain Set DNS domain postfix. When the domain which you input can not be
parsed, phone will automatically add this domain to the end of the
domain which you input before and parse it again.
Primary DNS Input your primary DNS server address.
Alter DNS Input your standby DNS server address.
Back Return to the last page.
Next Get into the next page.
2. After finished the above settings, click Next to open the following page.
Field name Explanation
Display Name Set the display name.
Server Address Input your SIP server address.
Server Port Set your SIP server port.
User Name Input your SIP register account name.
Password Input your SIP register password.
Phone Number Input the phone number assigned by your VOIP service provider.
Enable Register Start to register or not by selecting it or not.
Back Return to the last page.
Next Get into the next page.
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3. After finished the above settings, click Next to open the following page.
4. Click Finish to complete the configuration.
DHCP Mode
1. Choose DHCP Mode and click Next. You can get the following web page.
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2. After finished the above settings, click Next to open the following page.
Field name Explanation
Display Name Set the display name.
Server Address Input your SIP server address.
Server Port Set your SIP server port.
User Name Input your SIP register account name.
Password Input your SIP register password.
Phone Number Input the phone number assigned by your VOIP service provider.
Enable Register Start to register or not by selecting it or not.
3. After finished the above settings, click Next to open the following page.
4. Click Finish to complete the configuration.
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PPPoE Mode
1. Choose PPPoE Mode and click Next. You can get the following web page.
2. After finished the above settings, click Next to open the following page.
Field name Explanation
PPPoE Server It will be provided by ISP.
Username Input your ADSL account.
Password Input your ADSL password.
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3. After finished the above settings, click Next to open the following page.
Field name Explanation
Display Name Set the display name.
Server Address Input your SIP server address.
Server Port Set your SIP server port.
User Name Input your SIP register account name.
Password Input your SIP register password.
Phone Number Input the phone number assigned by your VOIP service provider.
Enable Register Start to register or not by selecting it or not.
4. After finished the above settings, click Next to open the following page.
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5. Click Finish to complete the configuration.
IP Phone will save the setting automatically and reboot. After reboot, you can dial by the SIP account.
33..11..33 CCaallll LLoogg
You can query all the outgoing through this page.
Field name Explanation
Start Time Display the start time of the outgoing record.
Last Time Display the conversation time of the outgoing record.
Called Number Display the account/protocol/line of the outgoing record.
33..11..44 MMMMII SSeett
Field name Explanation
Language Set Set the language of phone, English is default.
Text Message The greeting message will display on LCD when phone is idle. It can
support 16 chars. The default chars are “VOIP PHONE”.
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Line Info In the standby screen showing the registration number of lines, when
the time is displayed as NULL is not registered.
33..22 NNeettwwoorrkk
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33..22..11 WWAANN
Please select the proper network mode according to the network condition. Vigor router provides three
different network settings (Static, DHCP and PPPoE).
WAN Status
Field name Explanation
Active IP The current IP address of the phone.
Current Netmask The current Netmask address.
MAC Address The current MAC address of the phone.
Current Gateway The current Gateway IP address.
Get MAC Time Shows the time of getting MAC address
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WAN Setting - Static
If your ISP server provides you the static IP address, please select Static and finish related setting. If
you don’t know about parameters of Static Mode setting, please ask your ISP for them.
Field name Explanation
Obtain DNS server
automatically
Select it to use DHCP mode to get DNS address, if you don’t select it,
you will use static DNS server. The default is selecting it.
IP Address Input the IP address distributed to you.
Netmask Input the Netmask distributed to you.
Gateway Input the Gateway address distributed to you.
DNS Domain Set DNS domain postfix. When the domain which you input can not be
parsed, phone will automatically add this domain to the end of the
domain which you input before and parse it again.
Primary DNS Input your primary DNS server address.
Alter DNS Input your standby DNS server address.
Apply Save the settings.
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WAN Setting - DHCP
If you uses DCHP mode, you will get the information from the DHCP server automatically. You don’t
need to input this information artificially.
Field name Explanation
Apply Save the settings.
WAN Setting - PPPoE
If you uses PPPoE mode, you need to make the following settings.
Field name Explanation
PPPoE Server It will be provided by ISP.
Username Input your ADSL account.
Password Input your ADSL password.
Apply Save the settings.
Notice:
1. Click Apply button after finished your setting. IP Phone will save the setting automatically and new
setting will take effect.
2 If you modify the IP address, the web page will not response by the old IP address. Your need input
new IP address in the address column to logon in the phone.
3. If networks ID which is DHCP server distributed is same as network ID which is used by LAN of
system, the system will use the DHCP IP to set WAN, and modify LAN’s networks ID(for
example, system will change LAN IP from 192.168.10.1 to 192.168.11.1) when it uses DHCP client
to get IP in startup. If the system uses DHCP client to get IP in running status and network ID is
also same as LAN’s, the system will refuse to accept the IP to configure WAN. So WAN’s active IP
will be 0.0.0.0.
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33..22..22 LLAANN
Field name Explanation
LAN IP Specify LAN static IP.
Netmask Specify LAN Netmask.
DHCP Service Select the DHCP server of LAN port or not. After you modify the LAN
IP address, phone will amend and adjust the DHCP Lease Table and
save the result amended automatically according to the IP address and
Netmask. You need restart the phone and the DHCP server setting will
take effect.
NAT Select NAT or not.
Bridge Mode Select Bridge Mode or not. If you select Bridge Mode, the phone will
no longer set IP address for LAN physical port. LAN and WAN will
join in the same network. Click Apply, the phone will reboot.
If you choose the bridge mode, the LAN configuration will be disabled.
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33..22..33 QQOOSS
The VOIP phone support 802.1Q/P protocol and DiffServ configuration. VLAN functionality can use
different VLAN IDs by setting signal/voice VLAN and data VLAN. The VLAN application of this
phone is very flexible.
In chart 1, there is a switch without setting VLAN. Any broadcast frame will be transmitted to the
other ports except the send port. For example, a broadcast information is sent out from port 1 then
transmitted to port 2,3 and 4.
In chart 2, red and blue circles indicate two different VLANs in the switch, and port 1 and port 2
belong to red VLAN, port 3 and port 4 belong to blue VLAN. If a broadcast frame is sent out from
port 1, the switch will transmit it to port 2, the other port in the red VLAN and not transmit it to port3
and port 4 in blue VLAN. By this means, VLAN divides the broadcast domain via restricting the
range of broadcast frame transmission.
Note: Chart 2 uses red and blue to identify the different VLANs; but in practice, VLAN uses different
VLAN IDs to identify them.
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Field name Explanation
VLAN Enable Before select it to enable VLAN, you need enable Bridge mode in LAN
configuration.
VLAN ID Check
Enable
Enable VLAN ID check by selecting it. After enable VLAN ID check,
if VLAN ID of a data package is not the same with the phone or a data
package do not have VLAN ID, the data package will be discarded.
Voice/Data VLAN
differentiated
After enable VLAN, system will set packets with different type of
VLAN ID. Undifferentiated means after using VLAN, both VoIP
packets and other data packets will use the voice VLAN ID; tag
differentiated means after using VLAN, VoIP(signal and voice) packets
will add voice VLAN ID, and other data packets will add data VLAN
ID; data untagged means after using VLAN, only VoIP packets will add
voice VLAN ID. Other data packets will not use VLAN.
DiffServ Enable Select it or not to Enable or disable DiffServ.
DiffServ Value Set DiffServ value, the common value is 0x00.
Voice 802.1P
Priority
Specify 802.1P Priority of voice/signal data package.
Data 802.1P Priority Set 802.1p of data VLAN. Non-VoIP data (such as http, telnet, ping etc)
will use this value to set VLAN package.
Voice VLAN ID Set VLAN ID of voice/signal data package.
Data VLAN ID Set 802.1q of data VLAN ID. Non-VoIP data (such as http, telnet, ping
etc) will use this value to set VLAN package.
Apply Save the settings.
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NOTICE：
1. Startup VLAN, if set Voice/Data VLAN differentiated as Undifferentiated, all packets will use
the Voice VLAN ID as the tag.
2. Startup VLAN, if set Voice/Data VLAN differentiated as tag differentiated and disables the
DiffServ, then system will not distinguish the voice and data, all packets will use the Voice
VLAN ID as the tag.
3. Startup VLAN, if set Voice/Data VLAN differentiated as tag differentiated and enables the
DiffServ, then system will distinguish the voice and data and add the VLAN ID each other.
4. Startup VLAN, if set Voice/Data VLAN differentiated as data untagged, then the packet of the
signal/voice will use the Voice VLAN ID as the tag, but the data packets will not take the
VLAN tag.
5. If Disable the VLAN, regardless to set the Voice/Data VLAN differentiated or not, all packets
will not take the VLAN tag; If enable the DiffServ, all packets will only take the DiffServ
value.
6. One must to notice, enable the VLAN ID Check Enable that is default, If enable it, the phone
will match the VLAN ID strictly. When others' VLAN ID does not match with us, the packets
will discard. Contrarily, the phone will accept the packets with the distinct VLAN ID.
7. You must gain the IP with the Static mode when you set VLAN, otherwise can't gain the IP in
the VLAN and also can not dial with point to point.
33..22..44 SSeerrvviiccee PPoorrtt
You can set the port of telnet/HTTP/RTP by this page.
Field name Explanation
HTTP Port
set web browse port, the default is 80 port，if you want to enhance
system safety，you'd better change it into non-80 standard port；
Example: The IP address is 192.168.1.70. and the port value is 8090,
the accessing address is http://192.168.1.70:8090
Telnet Port Set Telnet Port, the default is 23. You can change the value into others.
Example:
The IP address is 192.168.1.70. the telnet port value is 8023, the
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accessing address is telnet 192.168.1.70 8023
RTP Initial Port Set the RTP Initial Port. It is dynamic allocation.
RTP Port Quantity Set the maximum quantity of RTP Port, the default is 200.
Notice:
1. You need save the configuration and reboot the phone after set this page.
2. If you modify the port of Telnet and HTTP, you would better set the value more than 1024
because the port value less than 1024 is system port reserved.
3. If you set 0 for the HTTP port, it will disable HTTP service.
33..22..55 DDHHCCPP SSeerrvveerr
Field name Explanation
DHCP Leased Table IP-MAC mapping table. If the LAN port of the phone connects to a
device, this table will show the IP and MAC address of this device.
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DHCP Lease Table Shows the DHCP Lease Table, the unit of Lease time is Minute.
DHCP Lease Table
Setting
Allow to set corresponding settings for DHCP lease settings
Lease Table Name Specify the name of the lease table
Start IP Set the start IP address of the lease table
End IP Set the end IP address of the lease table, the network device connected
to LAN port will get IP address between Start IP and End IP by DHCP.
Netmask Set the Netmask of the lease table
Gateway Set the Gateway of the lease table
Lease Time Set the Lease Time of the lease table
DNS Set the default DNS server IP of the lease table; Click the Add button to
submit and add this lease table
DHCP Lease Table
Delete
Lease Table Name - Select name of lease table, click the Delete button
will delete the selected lease table from DHCP lease table.
DNS Relay Select DNS Relay. The default is enabled. Click the Apply button to
become effective.
Apply Save the settings.
Notice:
1. The size of lease table can not be larger than the quantity of C network IP address. We
recommend you to use the default lease table and not modify it.
2. If you modify the DHCP lease table, you need save the configuration and reboot.
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33..22..66 SSNNTTPP
Setting time zone and SNTP (Simple Network Time Protocol) server according to your location, you
can also manually adjust date and time in this web page.
Field name Explanation
Server Set SNTP Server IP address.
Time Zone Select the Time zone according to your location.
Time Out Set the time out, the default is 60 seconds.
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12 Hours Systems Switch the time mechanism between 12 hours and 24 hours.
Default is 24 hours mode.
SNTP Select the SNTP, and click Apply to make the SNTP Times effective.
Enable Daylight Enable daylight saving time.
Time shift (minutes) Setup the variety length.
Month Setup stat and end month.
Week Setup start and end week.
Day Setup start and end day.
Hour Setup start and end hours.
Minute Setup start and end minutes.
Manual Timeset You need specify the all items.
Apply Save the settings.
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33..33 VVooIIPP
33..33..11 SSIIPP
Set your SIP server in the following interface.
Field name Explanation
SIP Line Select Choose line to set info about SIP, there are 3 lines to choose. You can
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switch by using the Load button.
Before configuring the basic settings, you have to load one SIP line
first.
Register Status Shows if the phone has been registered the SIP server or not; or so,
show Unapplied;
Server Name Set the server name.
Server Address Input your SIP server address.
Server Port Set your SIP server port.
Account Name Input your SIP register account name.
Password Input your SIP register password.
Phone Number Input the phone number assigned by your VoIP service provider. Phone
will not register if there is no phone number configured.
Display Name Set the display name.
Proxy Server
Address
Set proxy server IP address（Usually, Register SIP Server
configuration is the same as Proxy SIP Server. But if your VoIP service
provider give different configurations between Register SIP Server and
Proxy SIP Server, you need make different settings.）
Proxy Server Port Set your Proxy SIP server port.
Proxy Username Input your Proxy SIP server account.
Proxy Password Input your Proxy SIP server password.
Domain Realm Set the sip domain if needed, otherwise this VoIP phone will use the
Register server address as sip domain automatically. (Usually it is same
with registered server and proxy server IP address).
Enable Register Start to register or not by selecting it or not.
Click Advanced Set to get more detailed settings for SIP account.
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Field name Explanation
Register Expire
Time
Set expire time of SIP server register, default is 60 seconds. If the
register time of the server requested is longer or shorter than the expire
time set, the phone will change automatically the time into the time
recommended by the server, and register again.
NAT Keep Alive
Interval
Set examining interval of the server, default is 60 seconds.
User Agent Set the user agent if have, the default is VoIP Phone 1.0.
Signal Key Set the key for signal encryption.
Media Key Set the key for RTP encryption.
Local port Set sip port of each line.
Ring type Set ring type of each line.
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Hot line Number Set hot line number of each line.
Conference Number Configure conference number in server conference.
Transfer Expire
Time
For the phone supports the transfer of certain special features server, set
interval time between sending “bye” and hanging up after the phone
transfers a call.
Enable subscribe Enable the option, the phone will receive the notification from the
server.
Enable Keep
Authentication
Enable/Disable Keep Authentication System will take the last
authentication field which is passed the authentication by server to the
request packet. It will decrease the server’s repeat authorization work, if
it is enable.
NAT Keep Alive Enable/Disable keeps NAT of SIP alive.
If some server refuse to register with too short interval time, and has no
packets sending to device in private network to keep NAT alive, user
could set this function ON. It need set the keep alive interval time less
than the NAT server’s.
Enable Via rport Enable/Disable system to support RFC3581. Via rport is special way to
realize SIP NAT.
Enable PRACK Enable or disable SIP PRACK function, suggest use the default
configuration.
Long Contact Set more parameters in contact field; connection with SEM server
Enable URI Convert Convert # to %23 when send the URI.
Dial Without
Register
Set call out by proxy without registration.
Ban Anonymous
Call
Set to ban Anonymous Call.
Enable DNS SRV Support DNS looking up with _sip.udp mode
Forward Type Select call forward mode, the default is Off.
z Off: Close down calling forward
z Busy: If the phone is busy, incoming calls will be forwarded to the
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appointed phone.
z No answer: If there is no answer, incoming calls will be forwarded to
the appointed phone.
z Always: Incoming calls will be forwarded to the appoint phone
directly.
The phone will Prompt the incoming while doing forward.
Forward Phone
Number
Appoint your forward phone number.
Server Type Select the special type of server which is encrypted, or has some unique
requirements or call flows.
DTMF Mode Select DTMF sending mode, there are three modes:
z DTMF_RELAY
z DTMF_RFC2833
z DTMF_SIP_INFO
Different VoIP Service providers may provide different modes.
RFC Protocol
Edition
Select SIP protocol version to adapt for the SIP server which uses the
same version as you select. For example, if the server is CISCO5300,
you need to change to RFC2543, else phone may not cancel call
normally. System uses RFC3261 as default.
Transport Protocol Set transport protocols, TCP or UDP.
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RFC Privacy
Edition
Set Anonymous call out safely; Support RFC3323and RFC3325.
Subscribe Expire
Time
Overtime of resending subscribe packet. Suggest using the default
configuration.
Enable Conference
number
Set to use sever conference.
MWI Number Input the number of the server's voice-mail box.
Click to Talk Set click to Talk (need practical software support).
Signal Encode Enable/Disable Signal Encrypt.
RTP Encode Enable/Disable RTP Encrypt.
Enable Session
Timer
Set Enable/Disable Session Timer, whether support RFC4028.It will
refresh the SIP sessions.
Answer With Single
Codec
Enable/Disable the function when call is incoming, phone replies SIP
message with just one codec which phone supports.
Auto TCP Set to use automatically TCP protocol to guarantee usability of
transport as message is above 1300 byte.
Enable Strict Proxy Support the special SIP server-when phone receives the packets sent
from server， phone will use the source IP address, not the address in
via field.
Enable GRUU Set to support GRUU.
Enable Display
name Quote
Set to make quotation mark to display name as the phone sends out
signal, in order to be compatible with server.
Enable user=phone It is just for satisfying the standard of SIP URI. If the SIP server or
PSTN gateway does not have any request of SIP invite, you don’t need
to enable this feature.
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33..33..22 IIAAXX22
Field name Explanation
Register Status Shows if the phone has been registered the IAX2 server or not.
IAX2 Server Addr Input your IAX2 server address.
IAX2 Server Port Set your IAX2 server port, the default is 4569.
Account Name Input your IAX2 register account name.
Account Password Input your IAX2 register password.
Phone Number Input your assigned phone number (usually it is same you’re your IAX2
account name).
Local Port
Set your local sport，the default is 4569.
Voice Mail Number Specify the voice mail’s number.
Voice Mail Text Specify the voice mail’s name.
Echo Test Number Set echo test number. If IAX2 server supports echo test, and echo test
number is non- numeric, system could set an echo test number to
replace the echo test text. So user can dial the numeric number to test
echo voice test. This function is provided with server to make endpoint
to test whether endpoint could talk through server normally.
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Echo Test Text Specify echo test text’s name.
Refresh Time Set expire time of IAX2 server register, you can set it between 60 and
3600 seconds.
Enable Register Start to register the IAX2 server or not by selecting it or not.
Enable G.729 Enable or disable code G.729 by selecting it or not
33..33..33 SSTTUUNN
By STUN server, the phone in private network could know the type of NAT and the NAT mapping IP
and port of SIP. The phone might register itself to SIP server with global IP and port to realize the
device both calling and being called in private network.
In this web page, you can configure SIP STUN.
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Field name Explanation
STUN NAT
Transverse
Shows STUN NAT Transverse estimation, true means STUN can
penetrate NAT, while False means not.
STUN Server Addr Set your SIP STUN Server IP address
STUN Server Port Set your SIP STUN Server Port
STUN Effect Time Set STUN Effective Time. If NAT server finds that a NAT mapping is
idle after time out, it will release the mapping and the system need send
a STUN packet to keep the mapping effective and alive.
Local SIP Port Set the SIP port.
Set Sip Line Enable
STUN
Choose line to set info about SIP, There are 3 lines to choose. You can
switch by using the Load button.
Use STUN Enable/Disable SIP STUN.
Apply Save the settings.
Notice: SIP STUN is used to realize SIP penetration to NAT. If your phone configures STUN Server
IP and Port (default is 3478), and enable SIP Stun, you can use the ordinary SIP Server to realize
penetration to NAT.
33..33..44 DDiiaall PPeeeerr
This functionality offers you more flexible dial rule, you can refer to the following content to know how
to use this dial rule.
z When you want to dial an IP address, the entry of IP addresses is very cumbersome, but by this
functionality, you can set number 156 to replace 192.168.1.119 here.
z When you want to dial a long distance call to Beijing, you need dial an area code 010 before local
phone number, but you can also dial number 1 instead of 010 after we make a setting according to
this dial rule. For example, you want to dial 01062213123, but you need dial only 162213123 to
realize your long distance call after you make this setting.
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z To save the memory and avoid abundant input of user, add the follow functions:
x matches any single digit that is dialed. If a user makes the above configuration, after he/she dials
11 digit numbers started with 13, the phone will send out 0 plus the dialed numbers automatically.
[ ] Specifies a range that will match digit. It may be a range, a list of ranges separated by commas,
or a list of digits.
If a user makes the above configuration, after user dials 11 digit numbers started with from 135 to
139, the phone will send out 0 plus the dialed numbers automatically.
With this setting, you can realize dialing out via different lines without switch in web interface.
Field name Explanation
Phone number There are two types of matching conditions: one is full matching, the
other is prefix matching. In the Full matching, you need input your
desired phone number in this blank, and then you need dial the phone
number to realize calling to what the phone number is mapped. In the
prefix matching, you need input your desired prefix number and T; then
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dial the prefix and a phone number to realize calling to what your prefix
number is mapped. The prefix number supports at most 30 digits.
Destination Set Destination address. This is optional configuration item. If you want
to set peer to peer call, please input destination IP address or domain
name. If you want to use this dial rule on SIP2 line, you need input
255.255.255.255 or 0.0.0.2 in it.SIP3 into 0.0.0.3.
Port Set the Signal port, the default is 5060 for SIP.
Alias Set alias. This is optional configuration item. If you don’t set Alias, it
will show no alias.
Call Mode Select different signal protocol, SIP or IAX2.
Suffix(optional) Set suffix, this is optional configuration item. It will show no suffix if
you don’t set it.
Delete Length
(optional)
Set delete length. This is optional configuration item. For example: if
the delete length is 3, the phone will delete the first 3 digits then send
out the rest digits. You can refer to examples of different alias
application to know how to set delete length.
Note: There are four types of aliases.
z Add: xxx, it means that you need dial xxx in front of phone number, which will reduce dialing
number length.
z All: xxx, it means that xxx will replace some phone number.
z Del: It means that phone will delete the number with length appointed.
z Rep: It means that phone will replace the number with length and number appointed.
You can refer to the following examples of different alias application to know more how to use different
aliases and this dial rule.
Examples of different alias application
Set by web Explanation Example
You need set phone number,
Destination, Alias and Delete Length.
Phone number is XXXT; Destination
is 255.255.255.255 (0.0.0.2) and
Alias is del.
This means any phone No. that starts
with your set phone number will be
sent via SIP2 line after the first
several digits of your dialed phone
number are deleted according to
delete length.
If you dial “93333”, the
SIP2 server will receive
“3333”
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This setting will realize speed dial
function, after you dialing the
numeric key “2”, the number after all
will be sent out.
When you dial “2”, the
SIP1 server will receive
33334444
The phone will automatically send
out alias number adding your dialed
number, if your dialed number starts
with your set phone number.
When you dial “8309“,
the SIP1 server will
receive “07558309”
You need set Phone Number, Alias
and Delete Length. Phone number is
XXXT and Alias is rep:xxx
If your dialed phone number starts
with your set phone number, the first
digits same as your set phone number
will be replaced by the alias number
specified and New phone number
will be send out.
When you dial
“0106228”, the SIP1
server will receive
“86106228”
If your dialed phone number starts
with your set phone number. The
phone will send out your dialed
phone number adding suffix number.
When you dial “147”,
the SIP1 server will
receive “1470011”
Introduction of how to set up dial-peer to implement switch between multi- SIP lines
9T mapping: If you have registered a SIP1 server and set dial-peer according to the above table, all
calls will be sent via SIP1 server when you press the numeric key “9” in front of dialing destination
phone numbers.
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8T mapping: If you have registered a Private SIP2 server and set dial-peer according to the above table，
all calls will be sent via SIP2 server when you press the numeric key “8” in front of dialing destination
phone numbers.
2T mapping: The rule of 2T means the user needs to dial the number with prefix 2 if he/she wants to
dial via IAX2 server.
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33..44 PPhhoonnee
33..44..11 DDSSPP
In this page, you can configure voice codec, input/output volume and so on.
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Field name Explanation
First Codec The fist preferential DSP codec: G.711A/u, G.722, G.723,
G.729,G.726,AMR
Second Codec The second preferential DSP codec: G.711A/u, G.722, G.723,
G.729,G.726
Third Codec The third preferential DSP codec: G.711A/u, G.722, G.723,
G.729,G.726,AMR
Forth Codec The forth preferential DSP codec: G.711A/u, G.722, G.723,
G.729,G.726,AMR
Fifth Codec The fifth preferential DSP codec: G.711A/u, G.722, G.723,
G.729,G.726,AMR
Sixth Codec The fifth preferential DSP codec: G.711A/u, G.722, G.723,
G.729,G.726,AMR
Seventh Codec
The seventh preferential DSP codec: G.711A/u, G.722, G.723,
G.729,G.726,AMR
AMR Payload
Type
AMR Payload Type.
Handdown Time Specify the least reflection time of Handdown. The default is 200ms.
Default Ring Type Set up the ring by default.
Input Volume Specify Input (MIC) Volume grade.
Output Volume Specify Output (receiver) Volume grade.
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Hands-free Volume Specify Hands-free Volume grade.
Ring Volume Specify Ring Volume grade.
G729 Payload
Length
Set G729 Payload Length.
Signal Standard Select Signal Standard.
G722 Timestamps 160/20ms or 320/20ms is available.
G723 Bit Rate 5.3kb/s or 6.3kb/s is available
VAD Select it or not to enable or disable VAD. If enable VAD, G729 Payload
length could not be set over 20ms.
DTMF Payload
Type
Set up DTMF payload type
Apply Save the settings.
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33..44..22 CCaallll SSeerrvviiccee
In this web page, you can configure Hotline, Call Transfer, Call Waiting, 3 Ways Call, Black List, white
list Limit List and so on.
Field name Explanation
Hotline
Specify Hotline number. If you set the number, you can not dial any
other numbers.
No Answer Time Specify No Answer Time
P2P IP Prefix
Set Prefix in peer to peer IP call. For example: what you want to dial is
192.168.1.119, If you define P2P IP Prefix as 192.168.1., you dial only
#119 to reach 192.168.1.119. Default is “.”. If there is no “.” Set, it
means to disable dialing IP.
Auto Answer If select it, the phone will auto answer when there is an incoming call.
Do Not Disturb
Select NO Disturb, the phone will reject any incoming call, the callers
will be reminded by busy, but any outgoing call from the phone will
work well.
Ban Outgoing
If you select Ban Outgoing to enable it, and you can not dial out any
number.
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Enable Call Transfer Enable Call Transfer by selecting it.
Enable Call Waiting Enable Call Waiting by selecting it.
Enable Three Way
Call
Enable Three Way Call
Accept Any Call
If select it, the phone will accept the call even if the called number is
not belong to the phone.
Auto Handdown
The phone will hang up and return to standby automatically at hands-
free mode
Auto Handdown
Time
Configuration automatically hang time, if it is hands-free mode, then
more than auto handdown time, the phone automatically returns to
standby mode, if the handle pattern, then more than auto handdown
time, it automatically put a dial tone.
Mute Mode
Configuring the mute mode, if the mute mode, calls LCD will flash tips,
but does not ring
XML Server Xml configuration server address and the default xml file name
Warm Line Time
Warm line set timeout to set the time line when more than warm, it will
automatically exhaled hotline number, if configured to 0, the hook
immediately exhaled hotline number.
DND Return Code
When the status of the IP phone is “DND (do not disturb)”, it will send
a message to the server based on the code selected here.
Reject Return Code
When the status of the IP phone is “Reject”, it will send a message to
the server based on the code selected here.
Busy Return Code
When the status of the IP phone is “Busy”, it will send a message to the
server based on the code selected here.
Black List
Set Add/Delete Black list. If user does not want to answer some phone
calls, add these phone numbers to the Black List, and these calls will be
rejected.
“x” and “.” are wildcard. x means matching any single digit. for
example, 4xxx expresses any number with prefix 4 which length is 4
will be forbidden to dialed out
“.” means matching any arbitrary number digit. For example, 6
expresses any number with prefix 6 will be forbidden to dialed out.
If a user wants to allow a number or a series of number incoming,
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he/she may add the number(s) to the list as the white list rule. the
configuration rule is -number, for example, -123456, or -1234xx.
Means any incoming number is forbidden except for 4119
Note: End with “.” when set up the white list
Limit List
Set Add/Delete Limit List. Please input the prefix of those phone
numbers which you forbid the phone to dial out. For example, if you
want to forbid those phones of 001 as prefix to be dialed out, you need
input 001 in the blank of limit list, and then you can not dial out any
phone number whose prefix is 001.
“x” and “.” are wildcard. x means matching any single digit. for
example, 4xxx expresses any number with prefix 4 which length is 4
will be forbidden to dialed out
“.” means matching any arbitrary number digit. For example, 6
expresses any number with prefix 6 will be forbidden to dialed out.
Notice: Black List and Limit List can record at most10 items respectively.
33..44..33 DDiiggiittaall MMaapp
This system supports 4 dial modes:
z End with “#”: dial your desired number, and then press #.
z Fixed Length: the phone will intersect the number according to your specified length.
z Time Out: After you stop dialing and waiting time out, system will send the number collected.
z User defined: you can customize digital map rules to make dialing more flexible. It is realized by
defining
the prefix of phone number and number length of dialing.
In order to keep some users' secondary dialing manner when dialing the external line with PBX, the
phone can be added a special rule to realize it so the user can dial a number as external line prefix and
get the secondary dial tone to keep dialing the external number. After finishing dialing, phone will send
the prefix and external number totally to the server.
For example, there is a rule 9, xxxxxxxx in the digital map table. After dialing 9, the phone will send the
secondary dial tone, and the user may keep going for dialing. After finished, the phone will call the
number which starts with 9; actually the number sent out is 9-digit with 9.
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Field name Explanation
End with "#" Set Enable/Disable the phone ended with “#” dial.
Fixed Length Specify the Fixed Length of phone ending with.
Time out Set the timeout of the last dial digit. The call will be sent after timeout.
Digital Rule table Set and display the user defined digital rules.
Below shows user-defined digital map rule:
[ ]: Specifies a range that will match digit. May be a range, a list of ranges separated by commas, or a
list of digits.
x: Match any single digit that is dialed.
. : Match any arbitrary number of digits including none.
Tn: Indicates an additional time out period before digits are sent of n seconds in length. n is
mandatory and can have a value of 0 to 9 seconds. Tn must be the last 2 characters of a dial plan.
If Tn is not specified, it is assumed to be T0 by default on all dial plans.
For example,
Rules Explanation
[1-8]xxx Cause extensions 1000-8999 to be dialed immediately.
9xxxxxx Cause 8 digit numbers started with 9 to be dialed immediately.
911 Cause 911 to be dialed immediately after it is entered.
99T4 Cause 99 to be dialed after 4 seconds.
9911x, T4 Cause any number started with 9911 to be dialed 4 seconds after dialing
ceases.
Notice: End with “#”, Fixed Length, Time out and Digital Map Table can be used simultaneously.
System will stop dialing and send number according to your set rules.
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33..44..44 PPhhoonnee BBooookk
You can input the name, phone number and select ring type for each name here. The maximum
capability of the phonebook is 500 items
Field name Explanation
Phonebook Table Name - Shows the name corresponding to the phone number.
Number - Shows the phone number.
Add Phone Book Name – Type the name corresponding to the phone number.
Number –Type the phone number.
Add – Click it to add a new phone entry.
Ring Type Choose one of the ring types for the incoming call.
Delete/Modify Click Modify to change the selected information and click the Delete to
delete the selected record.
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33..44..55 FFuunnccttiioonn KKeeyy
This page allows you to configure function keys (also called memory keys in IP Phone) with specific
type, value, line and other function parameters (speed dial, push to talk, DND and etc).
Field name Explanation
Interface
Configuration
Contrast - Set contrast of screen.
Luminance - Set luminance of screen.
Line Key Setting Select SIP1, SIP2, SIP3, Dial peer, or IAX2 in function key type. After
you set it, you pick up handset or hands-free, press this function key,
then you can use the corresponding IP line.
Function Key
Setting
Memory Key - Set the memory key's serial number.
Type -
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Memory Key: settings can be stored in key storage for each number, the
standby or off-hook. Selecting the function keys on the keyboard can
call this number.
DTMF：In the call, send DTMF.
Value –Set the type parameter values.
Line – Choose which lines to use this feature.
Subtype – Select the function parameters. Key Event and Memory Key
will bring about different Subtype items.
Memory keys can be configured with the following type:
Speed Dial - through the configuration of the key corresponding to the number of ways as shown below:
The User can press the F1 key to allocate this number by line1 line.
Push To Talk - you can press this key in standby to automatically answer the call and make each other:
The user can configure in accordance with the way of push to talk function. 4116 is the other number.
Then press the standby button and make it automatically answering the call 4116.
Key Event - key can be configured through certain event (e.g., DND).
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33..55 MMaaiinntteennaannccee
33..55..11 AAuuttoo PPrroovviissiioonn
Field name Explanation
Current Config
Version
Show the current config file’s version.
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Server Address Set FTP/TFTP/HTTP server IP address for auto update. The address can
be IP address or Domain name with subdirectory.
Username Set FTP server Username. System will use anonymous if username
keep blank.
Password Set FTP server Password.
Config File Name Set configuration file’s name which need to update. System will use
MAC as config file name if config file name keep blank. For example,
000102030405.。
Config Encrypt Key Input the Encrypt Key, if the configuration file is encrypted.
Protocol Type
Select the Protocol type FTP、TFTP or HTTP.
Update Interval
Time
Set update interval time, unit is hour.
Update Mode Different update modes:
1. Disable: means no update
2. Update after reboot: means update after reboot.
3. Update at time interval: means periodic update.
Enable DHCP
Option 66
This option is enabled, TFTP server address defaults to the value of
option 66
33..55..22 SSyysslloogg
Syslog is a protocol which is used to record the log messages with client/server mechanism. Syslog
server receives the messages from clients, and classifies them based on priority and type. Then these
messages will be written into log by some rules which administrator can configure. This is a better way
for log management.
There are 8 levels in debug information:
Level 0---emergency: This is highest default debug info level. You system can not work.
Level 1---alert: Your system has deadly problem.
Level 2---critical: Your system has serious problem.
Level 3---error: The error will affect your system working.
Level 4---warning: There are some potential dangers. But your system can work.
Level 5---notice: Your system works well in special condition, but you need to check its working
environment and parameter.
Level 6---info: the daily debugging info.
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Level 7---debug: the lowest debug info. Professional debugging info from R&D person.
At present, the lowest level of debug information send to Syslog is info, debug level only can be
displayed on telnet.
Field name Explanation
Server IP Set Syslog server IP address.
Server Port Set Syslog server port.
MGR Log Level/
SIP Log Level/
IAX2 Log Level
Set the level of MGR log/ Set the level of SIP log/ Set the level of IAX2
log.
Enable Syslog Select it or not to enable or disable syslog.
Apply Save the settings.
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33..55..33 CCoonnffiigg
Field name Explanation
Save Configuration You can save all changes of configurations. Click the Save button, all
changes of configuration will be saved, and be effective immediately. .
Backup Config Right clicks on “Right click here…” and select “Save Target As….”
then you will save the configuration file in .txt format
Clear Configuration A user can restore factory default configuration and reboot the phone.
If you login as Admin, the phone will reset all configurations and
restore factory default; if you login as Guest, the phone will reset all
configurations except for VoIP accounts (SIP1-2 and IAX2) and version
number.
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33..55..44 UUppddaattee
You can update your configuration with your configuration file in this web page.
Field name Explanation
Web Update Click the browse button, find out the configuration file saved before or
provided by manufacturer, download it to the phone directly, press
“Update” to save. You can also update downloaded update file, logo
picture, ring file by web.
Server Set the FTP/TFTP server address for download/upload. The address can
be IP address or Domain name with subdirectory.
Username Set the FTP server Username for download/upload.
Password Set the FTP server password for download/upload.
File name Set the name of update file or configuration file. The default name is the
MAC of the phone, such as 000102030405.
Type
Action type that system want to execute：
z Application update: download system update file
z Configuration file export: Upload the configuration file to
FTP/TFTP server, name and save it.
z Configuration file import: Download the configuration file to
phone from FTP/TFTP server. The configuration will be
effective after the phone is reset.
z Phone book export (.vcf): Upload the phonebook file to
FTP/TFTP server, name and save it.
z PhoneBook import (.vcf): Download the phonebook file to phone
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from FTP/TFTP server.
Protocol Select FTP/TFTP server.
33..55..55 AAccccoouunntt
You can add or delete user account, and change the authority of each user account in this web page.
Field name Explanation
Set Keyboard
Password
Keyboard Password - Set the password for entering the setting menu of
the phone by the phone‘s key board. The password is digit.
User Set This table shows the current user existed.
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Add User User Name - Set account user name.
User Level - Set user level, Root user has the right to modify
configuration, General can only read.
Password - Set the password..
Confirm - Confirm the password.
Submit – Save the settings.
Account Option Select the account and click the Modify to modify the selected account,
and click the Delete to delete the selected account.
General user only can add the user whose level is General.
33..55..66 RReebboooott
If you modified some configurations which need the phone’s reboot to be effective, you need click the
Reboot button. Then the phone will reboot immediately.
Notice: Before reboot, you need confirm that you have saved all configurations.
33..66 SSeeccuurriittyy
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33..66..11 MMMMII FFiilltteerr
User could make some device own IP, which is pre-specified, access to the MMI of the phone to config
and manage the phone.
Field name Explanation
MMI Filter Table MMI Filter IP Table list.
MMI Filter Table
Set
Add or delete the IP address segments that access to the phone.
Set initial IP address in the Start IP column, Set end IP address in the
End IP column, and click Add to add this IP segment. You can also
click Delete to delete the selected IP segment.
MMI Filter Table
Set
MMI Filter - Select it or not to enable or disable MMI Filter. Click
Apply to make it effective.
Do not set your visiting IP outside the MMI filter range, otherwise, you can not logon through the web.
33..66..22 FFiirreewwaallll
In this web interface, you can set up firewall to prevent unauthorized Internet users from accessing
private networks connected to the Internet (input rule), or prevent unauthorized private network devices
from accessing the Internet (output rule).
Firewall supports two types of rules: input access rule and output access rule. Each type supports at most
10 items.
Through this web page, you could set up and enable/disable firewall with input/output rules. System
could prevent unauthorized access, or access other networks set in rules for security. Firewall, is also
called access list, is a simple implementation of a Cisco-like access list (firewall). It supports two access
lists: one for filtering input packets, and the other for filtering output packets. Each kind of list could be
added 10 items.
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We will give you an instance for your reference.
Field name Explanation
In access enable Select it to Enable in_ access rule
out access enable Select it to Enable out_ access rule
Input/Output Specify current adding rule by selecting input rule or output rule.
Deny/Permit Specify current adding rule by selecting Deny rule or Permit rule.
Protocol Type Filter protocol type. You can select TCP, UDP, ICMP, or IP.
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Port Range Set the filter Port range.
Src Addr Set source address. It can be single IP address, network address,
complete address 0.0.0.0, or network address similar to *.*.*.0
Des Addr Set the destination address. It can be IP address, network address,
complete address 0.0.0.0, or network address similar to *.*.*.*
Src Mask Set the source address’ mask. For example, 255.255.255.255 means just
point to one host; 255.255.255.0 means point to a network which
network ID is C type.
Des Mask Set the destination address’ mask. For example, 255.255.255.255 means
just point to one host; 255.255.255.0 means point to a network which
network ID is C type.
Apply Save the settings.
Delete Delete the selected rule.
Click the Add button if you want to add a new output rule.
Then enable out access, and click the Apply button.
So when devices execute to ping 192.168.1.118, system will deny the request to send ICMP request to
192.168.1.118 for the out access rule. But if devices ping other devices which network ID is 192.168.1.0,
it will be normal.
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33..66..33 NNAATT
NAT is abbreviated from Net Address Translation; it’s a protocol responsible for IP address translation.
In other word, it is responsible for transforming IP and port of private network to public, also is the IP
address mapping which we usually say.
DMZ
In order to make some intranet equipments support better service for extranet, and make internal
network security more effectively, these equipments open to extranet need be separated from the other
equipments not open to extranet by the corresponding isolation method according to different demands.
We can provide the different security level protection in terms of the different resources by building a
DMZ region which can provide the network level protection for the equipments environment, reduce the
risk which is caused by providing service to distrust customer, and is the best position to put public
information .
The following chart describes the network access control of DMZ:
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Field name Explanation
Protocol Set IPSec ALG - It is an encryption technology. Select it to enable IPSec
ALG, the default is enabled.
FTP ALG - FTP is a service of connection layer which can transform
intranet IP into extranet IP when intranet IP is sending out packet.
Select it to enable FTP ALG, the default is enabled.
PPTP ALG - Select it enable PPTP ALG, the default is enabled.
NAT Table Shows the NAT TC and UDP mapping table.
NAT Table Option Transfer Type - Select the NAT mapping protocol style, TCP or UDP.
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Outside Port - Set the WAN port of the NAT mapping.
Inside IP - Set the IP address of device which is connected to LAN
interface to do NAT mapping.
Inside Port - Set the LAN port of the NAT mapping.
Notice: After finish setting, click the Add button to add new mapping
table; click the Delete button to delete the selected mapping table.
DMZ Table Shows the outside WAN port IP address and the inside LAN port IP
address.
DMZ Table Option Outside IP - Set the outside Wan port IP address of DMZ.
Inside IP- Set the inside LAN port IP address of DMZ.
Click the Add button to add new table; click the Delete button to delete
the selected mapping table.
Notice: 10M/100M adaptive means the network card, and other equipment physical consultations speed,
testing speed under bridge mode near to 100M, in order to ensure the quality of voice and
communications real-time performance, we made some sacrifices of NAT under the transmission
performance. Transmit with full capability only when system is idle, so can not guarantee that the
transmission speed reach to 100M.
33..66..44 VVPPNN
This web page provides us a safe connect mode by which we can make remote access to enterprise inner
network from public network. That is to say, you can set it to connect public networks in different areas
into inner network via a special tunnel.
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Field name Explanation
VPN IP Shows the current VPN IP address
VPN Mode Enable VPN - Select it or not to enable or disable VPN.
L2TP VPN Server Addr - Set VPN L2TP Server IP address.
VPN User Name - Set User Name access to VPN L2TP Server.
VPN Password - Set Password access to VPN L2TP Server.
33..77 LLooggoouutt
Click Logout and you will exit web page. If you want to enter it next time, you need input user name
and password again.
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CChhaapptteerr 44 OOppeerraattiioonn
44..11 SSeett uupp VViiggoorrPPhhoonnee 330000 wwiitthh VViiggoorrIIPPPPBBXX SSeerriieess
DrayTek VigorIPPBX series supports the function of auto-provisioning. VigorPhone 300 is also capable
of auto-provisioning, it can get a configuration text file from the VigorIPPBX series. The configuration
file contains SIP settings that the SIP devices can register with VigorIPPBX series.
1. Configure the extension number and password for each IP phone on VigorIPPBX 2820. You can
configure extensions from IP PBX Wizard.
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2. Click IPPBX Wizard to get the first screen as shown below:
3. Type the extension group name, group number, start number, and number of extension fields.
Click OK to save them. The new added group will be displayed on the screen. Then click Next to
access into next web page.
4. In the SIP Trunk Setup page, you can set up to six SIP profiles outside lines at one time.
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5. Type the profile name, domain/realm, proxy, account number/name, password and trunk number
fields, then click OK to save them. The new added profile will be displayed on the screen.
6. Click Next to access into office hour setup page.
7. Please specify office hours including starting point and ending point on duty day(s).Then, click
Finish to save the settings and exit the wizard.
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8. After finishing the Wizard, please go to IPPBX>Extension to configure the Extension Number
and the Password settings. Click the index number 1.
9. Type in Extension Number and Password.
10. Then connect VigorPhone to the network. Each user of VigorPhone can get the extension
number/password respectively.
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11. Access into the web configurator of VigorPhone 300 (e.g., 192.168.1.11).
12. Open VoIP and press the SIP tab to display the following page.
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13. Fill in the information according to the settings (listed in Step 1 to Step 9) configured in
VigorIPPBX series.
14. When you finished the settings, click Apply to save them. VigorPhone will try to register the
number to VigorIPPBX series.
15. Later, if Register Status display “Registered”, that means the extension number for VigorPhone
has been registered successfully.
44..22 AAnnsswweerr CCaallll
VigorPhone 300 will ring to indicate you when there is call incoming, below is the ways to answer call:
z Answer with hook off
Take handset, you can talk directly. You can just hang up to finish talk.
z
Answer with the headset button
Press the Headset key
to answer the call, press the key again to finish talk.
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z Using handset instead of hands-free during a talk
Hook off the handset when you use hands-free and want to change to use handset. Just hook on to
finish talk.
z Using headset instead of hands-free during a talk
In the hands-free calls, press the Headset key . You can use the headset to call.
After that, press the key again to hang up the call.
z Using headset instead of handset during a talk
In the handset call, the Headset key , hang up the handset to continue using the
headset call. After the call, press the key again to cut off the call.
z Using handset instead of headset during a talk
In the headset call, hook off the handset after the call, just hook on to finish talk.
44..33 PPllaaccee CCaallllss
z Using handset
Hook off (screen will show the current using line, or you could press key L1-L3 to select), after
getting dialing tone, you could begin to dial number. After finishing it, press # and the IP phone
will send the number and call the number. When you hear a ring-back tone and screen shows the
callee’s number, it shows that the person you called is ringing. If a callee answers the call, you can
begin to talk and your phone will keep showing the callee’s number and counting time. Just hang
up to finish talk.
z Using headset
Standby, press the Headset key (on screen display “Enter Number Pls”) and hear
the dialing tone, you can start dialing. After finishing it, press # or press the softkey2-Send.
IP Phone can immediately begin connecting with each other. When you hear a ring-back tone
and screen shows the callee’s number, it shows that the person you called is ringing. If callee
answers the call, you can begin to talk and your phone will keep showing callee’s number and
counting time. Just press [Headset] key to finish talk.
z Using hands-free
Press the Hands-free key （screen will show the current using line, or you could press key
L1-L3 to select), after getting dialing tone, you could begin to dial number. After finishing it, press
# and the IP phone will send the number and call the number. When you hear the ringback tone and
screen shows the callee’s number, it shows that the person you called is ringing. If the callee
answers the call, you can begin to talk and your phone will keep showing callee’s number and
counting time. Press
again to finish talk.
z Using directory
Press Soft3 (PBook) in stand-by mode, you will access to phonebook. If there are many persons
records stored in the directory, you can use navigation keys
& to select number
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or press the first character of the name for searching the person which you want to contact. Press
to forward and press to backward. Press Soft2 (Dial) to dial the current number
shown on the screen.
z Speed dial
Speed dial means user can make calls directly without hook off or using hands-free. User can dial
number in stand-by mode, but first, user need to add and edit SDial no. By pressing Soft2 (SDial)
to edit and save the number to be an SDial number. In this way, user could make a call only press
the number and Soft3 (Dial).
z Multiple-way call
If a user has 2 line calls and wants to invite the third party during the call, he/she can press Soft1
(Conf) or Soft2(Transf) “New CALL”, press Soft1(OK),enter the number ,then press Soft2(Send)
and wait for the other party to answer. When the multiple-way calls, you can press the arrow keys
to select a call.
44..44 EEnndd CCaallllss
z Hang up with handset hook on
Hook on to finish talking.
z Hang up with hands-free
Press the Hands-free key to finish talk when phone is in hands-free status.
z Hang up with headset
If you are in the headset call, press the Headset key to end the call.
z Hang up an active call with 2 calls
When there are two calls，user might use Soft1(Switch)to switch to the call you want to hang up
first. Then press Soft3 (Close) to finish talk, and phone will switch to the other call automatically.
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44..55 CCaallll TTrraannssffeerr
1. Blind Transfer
During talk, press or Soft2 (Transf), and then dial the number that you want to transfer to,
and finished by "#". Phone will transfer the current call to the third party. After finishing transfer,
the call you talk to will be hanged up.
2. Attended Transfer
During talk, press or Soft2 (Transf), then input the number that you want to transfer to
and press Soft2 (Send). After that third party answers, then press
to complete the transfer.
(You need enable call waiting and call transfer first). If there are two calls, you can just talk to one,
and keep hold to the other one. The one who is keep hold can not speak to you or hear from you.
3. Alert Transfer
During the talk, press or Soft2 (Transf) firstly, then press Soft2 (Send) after inputting the
number that you want to transfer. You are waiting for connection, now, press
or Soft2
(Transf) and the transfer will be done. (To use this feature, you need enable call waiting and call
transfer first)
44..66 CCaallll HHoolldd
During talking, user could press to hold the current call. Press again to return the call
or switch the call active.
44..77 33--wwaayy CCoonnffeerreennccee CCaallll
User can press Soft1 (Conf) to dial the line2 (press Soft1 (Answer) to answer the call directly if this call
is from line2) during talking with line1. After line2 connect, user can press Soft2 (Conf) select another
way into the three-way calling number, then press softkey1-OK to enter into conference mode. To back
to line1 from conference, please press Soft1 (Split); to end the call, please press Soft3 (Close) or press
.
44..88 CCaallll RReeccoorrddss
The IP Phone supports 100 items of missed call, 100 items of incoming call, and 100 items of dialed call.
If the records are full, the newest will replace the oldest. If phone’s power cut or reboot, call records will
be discarded.
z Missed call
Press and screen displays “Missed Call” with the number and time of missed call. User
can also use
& to browse the missed call records, or press Soft1 (Option) to
check the details of this record, then press Soft2 (EDial) again to change the current number.
Pressing Soft2 (Dial) will call this number directly if user don’t modify the number. If there is no
missed call, screen will show “List Is Empty”.
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z Incoming call
Press and screen displays “Incoming Call”, by pressing & to browse the
records; or press Soft1 (Option) to check the details of this record, then press Soft2 (EDial) again to
change the current number. Pressing Soft2 (Dial) will call this number directly if user don’t modify
the number. If there is no incoming call，screen will show “List Is Empty”.
z Dialed call
Press , and use & to browse the dialed call records; or press Soft1
(Option) to check the details of this record, then press Soft2 (EDial) again to change the current
number. Pressing Soft2 (Dial) will call this number directly if user don’t modify the number. If
there is no dialed call, screen will show “List Is Empty”.
44..99 SSppeecciiaall KKeeyyss
z SMS function
In the standby mode, press Softkey1-SMS, then press Soft1 (new) key. After inputting SMS
content, press Soft2 (send) key to input callee’s number, next, press Soft2 (OK) again to send SMS.
When user has new message, the phone will ring, there is a coin on the screen. Press softkey1-SMS,
select inbox use up/down key, and then press softkey2-OK. When a number of text messages, users
can use up/down key and press softkey2-Enter to select one to view. Press softkey2-Reply and
input message content, finally, press Soft2(Send) again to reply this message.
The phone can also
send messages by phonebook.
Note: while user browses the message numbers, new messages will be marked by “new”; when a
user edits message, press # key that to switch input method, e.g. ABC (uppercase English input),
abc (lowercase English input), 123(digit input), Korean (Korean input(if your phone’s firmware
version supports Korean). PY ( if your phone’s firmware version supports Chinese) .
z SpeedDial function
User can pre-define numbers in these keys (numeric key 0-9). Hook off, press the defined numeric
key, and then input “#”. Your pre-defined numbers will send out.
Press softkey2-SDial to set speed dial in standby,
a total of 12 numbers, users can select by memory
key. Users can delete and press # key that to switch input method.
Note:
1. First 9 numbers corresponding digit key 1-9, 10
th
number corresponding digit key 0.
2. The first 10 set of numbers in standby mode press the corresponding number key and then press
softkey3-Dial or
key to exhale, but the first 11 groups and 12 group numbers without the
corresponding number key is required to enter SDial menu to find the set of numbers by
Corresponds memory key or softkey3-Dial button to exhaled.
z Realize Secondary Dial by Dialing for only one time
When you make secondary dial in off-hook/hands-free or standby pre-input mode, press [hold]
button to postpone input, and screen display will show ^. One stands for 2 seconds. For example,
you input 123^45, the phone will send DTMF (45) 2 seconds after the phone call 123. 123^^^45
will make phone send DTMF (45) at 6 seconds interval
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z Message waiting indication
After you set it, you can pick up or hands-free, then press to listen to record in server
when you have new voice message.
z Phone book search function
In the Chinese version, users can be retrieved by the corresponding initials Chinese name, which
simplifies the steps in the phone book to find contacts.
For example, contact name is Zhang San, contact number is 123. When you enter the phonebook,
you can press 9 key to select letter z, all the numbers of beginning with z will be displayed on the
LCD. You can select the one you want to search by press up/down key.
44..1100 CCaallll PPiicckkuupp
Call pickup is implemented by simulating pickup function of IPPBX. It’s that, when A calls B, B rings
but no answer. At this moment, C can hook off and input an appointed prefix plus B’s number, pick up
A’s call and talk with A.
The following chart shows how to configure an appointed prefix in dial peer to have call pick up
function. (Configuration in Dial Peer)
*1* means appointed prefix code. After making the above configuration, C can dial *1* plus B’s phone
number to pick up A’s call. User can set prefix in random, in the case of no affecting current dialing
rules.
44..1111 JJooiinn CCaallll
When B is calling C, A can join in the existing call by inputting an appointed prefix numbers plus B or
C number, if B or C also supports join call.
The following chart shows how to configure an appointed prefix in dial peer to have join call function.
(Configuration in Dial Peer)
*2* means appointed prefix code. After making the above configuration, A can dial *2* plus B or C
number to join B and C’s call. User can set prefix in random, in the case of no affecting current dialing
rules.
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44..1122 RReeddiiaall//UUnn--rreeddiiaall
If B is in busy line when A calls B, A will get notice: busy, please hang up. If A want to connect B as
soon as B is in idle, he can use redial function at the moment and he can dials an appointed prefix
number plus B’s number to realize redial function.
What is redial function? A can’t not build a call with B when B is in busy, then A will subscribe B’s
calling mode at 60 second intervals. Once B is available, A will get reminder of rings to hook off, while
A hooks off, A will call B automatically. If at this time A is occupied temporarily and unwilling to
contact B, A also can cancel the redial function by dialing an appointed prefix plus B’s number before
making the redial function.
*3* is appointed prefix code. After making the above configuration, A can dial
*3* plus B’s phone number to make the redial function.
*4* is appointed prefix code. After configuration, A can dial *4* to cancel redial function.
User can set prefix in random, in the case of no affecting current dialing rules.
44..1133 CClliicckk ttoo DDiiaall
When the user A browses in an appointed Web page, user A can click to call user B via a link, then user
A’s phone will ring. After user A hooks off, the phone will dial to user B.
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AAppppeennddiixx AA SSppeecciiffiiccaattiioonnss
AA..11 SSppeecciiffiiccaattiioonn
A.1.1 Hardware
Item Description
Adapter
(Input/Output)
Input: 100-240V
Output: 5V 1A
WAN 10/100Base- T RJ-45 for LAN port
LAN 10/100Base- T RJ-45 for PC
Power Consumption Idle: 2.5W/Active: 2.8W
LCD Size 128x96
53.5 x 70mm
Operation Temperature
0～40℃
Relative Humidity
10～65%
CPU
Broadcom
SDRAM 16MB
Flash 4MB
Dimension(L x W x H)
11.6×8×3 in.(295×205×75mm)
Weight 0.955kg
A.1.2 Voice features
z SIP supports 3 SIP servers
z Support SIP 2.0 (RFC3261) and correlative RFCs
z Codec: G.711A/u, G.723.1 high/low, G.729a/b，G.722， G.726
z Echo cancellation: G.168 Compliance in LEC, additional acoustic echo cancellation(AEC) can
reach 96ms max filter length in hands-free mode
z Support Voice Gain Setting, VAD, CNG
z Support full duplex hands-free
z HD Voice
z SIP support SIP domain, SIP authentication(none basic, MD5), DNS name of server, Peer to Peer/
IP call
z Automatically select calling line, if one line can’t be connected, the phone can automatically
switch to other line to call.
z DTMF Relay: support SIP info，DTMF Relay，RFC2833
z SIP application: SIP Call forward/transfer（blind/attended）/hold/waiting/3 way talking/sms
/pickup /joincall /redial /unredial/multi line
z Call control features: Flexible dial map, hotline, empty calling No. reject service, black list for
reject authenticated call, limit call, no disturb, caller ID, Flexible deer peer rule.
z Support phonebook 500 records，Incoming calls / outgoing calls / missing calls. Each supports
100 records
z Support IAX2
z Phonebook supports vcard standard
z 12/24 hours time display
z Support daylight saving time
z Support path
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z Support SIP Privacy
z Support SMS
z Support WMI
z Support Speed dial
z Support XML
A.1.3 Network features
z WAN/LAN: support bridge and router model
z Support PPPoE for xDSL
z Support basic NAT and NAPT
z Support VLAN (optional: voice vlan/ data vlan)
z NAT Penetrate, Stun Penetrate
z Support DMZ
z Support VPN (L2TP) function
z Wan Port supports main DNS and secondary DNS server, can select dynamically to get DNS in
DHCP mode or statically set DNS address.
z Support DHCP client on WAN
z Support DHCP server on LAN
z QoS with DiffServ
z Network tools in telnet server: including ping, trace route, telnet client
A.1.4 Maintenance and management
z Upgrade firmware through POST mode
z Web ,telnet and keypad management
z Management with different account right
z LCD and WEB configuration can be modified into requested language, and support multi-
language dynamically shifted
z Upgrade firmware through HTTP, FTP or TFTP Telnet remote management/ upload/download
setting file
z Support Syslog
z Support Auto Provisioning (upgrade firmware or configuration file)
A.1.5 Special features
z Support 3 softkeys, 6 memory keys, Navigation key.
z RLS,Pbook,MWI,HOLD,Trans,Mute,L1-L3,Vol -/+,Redial
AA..22 DDiiggiitt--cchhaarraacctteerr MMaapp TTaabbllee
Keypad Character Keypad Character
1 @
7 P Q R S p q r s
2 A B C a b c
8 T U V t u v
3 D E F d e f
9 W X Y Z w x y z
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4 G H I g h i
*/.
5 J K L j k l
0
6 M N O m n o
#/=
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