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TABLE OF CONTENTS
Chapter1 Safety Notice...................................................................................................3
Chapter2 Brief Introduction ...........................................................................................4
2.1 Brief introduction of DVX-2004.........................................................................................4
2.2 Hardware Structure .............................................................................................................5
2.2.1 Back Panel ............................................................................................................5
2.2.2 Front Panel............................................................................................................5
2.2.3 Hardware: ..............................................................................................................6
2.2.4 Environmental Requirements: ............................................................................6
2.2.5 Packing List...........................................................................................................6
Chapter3 Basic Configuration .......................................................................................7
3.1 Preparation Before Operation .............................................................................................7
3.2 Before Making a Call..........................................................................................................7
3.2.1 Login IP PBX .........................................................................................................7
3.2.2 Basic Configuration..............................................................................................9
3.2.3 Time Based Rules ..............................................................................................11
3.3 Outbound Call................................................................................................................... 11
3.3.1 Trunks...................................................................................................................11
3.3.2 Custom VoIP .......................................................................................................12
3.3.3 Peer ......................................................................................................................13
3.3.4 Outbound Routes ...............................................................................................14
3.4 Inbound Call......................................................................................................................16
3.4.1 Inbound Routes...................................................................................................16
3.4.2 IVR........................................................................................................................17
3.4.3 IVR Prompts........................................................................................................18
3.4.4 Ring Groups ........................................................................................................18
3.5 Black List..........................................................................................................................19
3.5.1 Pickup Call...........................................................................................................20
3.6 On The Call.......................................................................................................................21
3.6.1 Call Parking.........................................................................................................21
3.6.2 Transfer................................................................................................................21
3.6.3 Conference..........................................................................................................22
3.6.4 Monitor .................................................................................................................24
3.7 Settings before leaving office............................................................................................24
3.7.1 Follow Me ............................................................................................................24
3.7.2 VoiceMail..............................................................................................................25
3.8 Call Queue ........................................................................................................................27
3.8.1 Create Agent .......................................................................................................27
3.8.2 Agent Registration..............................................................................................29
3.8.3 Agent Log Off ......................................................................................................30
Chapter 4 Advanced .....................................................................................................31
4.1 Options..............................................................................................................................31
4.2 VoiceMail..........................................................................................................................32
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4.3 Music Settings...................................................................................................................34
4.4 DISA .................................................................................................................................35
4.5 Paging And Intercom ........................................................................................................36
4.6 Monitor .............................................................................................................................37
4.7 Phone Book.......................................................................................................................38
4.8 Feature Codes....................................................................................................................39
Chapter 5 Status..........................................................................................................42
5.1 Record List........................................................................................................................42
5.2 Call Logs...........................................................................................................................42
5.3 Register Status...................................................................................................................43
5.4 System Info .......................................................................................................................43
Chapter 6 System ........................................................................................................44
6.1 Network And Country.......................................................................................................44
6.2 TroubleShooting................................................................................................................44
6.3 DHCP Server Settings.......................................................................................................45
6.4 DDNS & VPN...................................................................................................................45
6.4 Time Settings ....................................................................................................................46
6.5 Management......................................................................................................................48
6.6 Data Storage......................................................................................................................49
6.7 Backup ..............................................................................................................................49
6.8 Update...............................................................................................................................50
The information contained in this document is subject to change at any time without prior
notification. If you want to learn more info about our product, please visit our web
www.Dlink.com
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Chapter1 Safety Notice
Please read the following safety notices before installing or using this IP PBX. They are
crucial for a safe and reliable operation of the device.
Please use the external power supply which is included in the package. Other power
supplies may cause damage to the device, affect the performance or induce noise.
Before using the external power supply in the package, please check with residential
power voltage. Inaccurate power voltage may cause fire and damage.
Please do not damage the power cord. If power cord or plug is impaired, do not use it,
otherwise, it may cause fire or electric shock.
The plug-socket combination must be accessible at all times because it serves as the
main disconnecting device.
Do not drop, knock or shake it. Rough handling can break internal circuit boards.
Do not install the device in places where there is direct sunlight. Also do not place the
device on carpets or cushions. It may cause fire or breakdown.
Avoid exposing the device to high temperature, below -10°C or high humidity. Avoid
wetting the unit with any liquid.
Do not attempt to open it. Non-expert handling to the device could damage it. Consult
your authorized dealer for help, or else it may cause fire, electric shock or breakdown.
Do not use harsh chemicals, cleaning solvents, or strong detergents to clean it. Wipe
it with soft cloth that has been slightly dampened in a mild soap and water solution.
When lightning, do not touch power plug or phone line, it may cause an electric
shock.
Do not install this device in an ill-ventilated place.
You are in a situation that could cause bodily injury. Before you work on any
equipment, be aware of the hazard involved with electrical circuitry and be familiar
with standard practices for preventing accidents.
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Chapter2 Brief Introduction
2.1 Brief introduction of DVX-2004
The all-in-one DVX-2004 IP PBX can not only provide the traditional basic PBX
features(call hold, call forwarding, call waiting and so on), but also provide enhanced
features such as visual operator, voice mail to mail, multi-media music on hold, and auto
attendant, etc. In addition, it’s very convenient for SMEs' management and maintenance,
also easy to upgrade. SMEs can set up own phone system to improve the company
image and office efficiency.
Main Features
30 Concurrent calls
Up to 100 registers
Video Calls
Multiple Language
DID(Direct Inward Dialing Number)
Support SKYPE for SIP
Support USB disk recording
Call Monitoring
Codec: G.711-Ulaw,G.711-Alaw,G.726,G.729
GSM,SPEEX,H.261,H.263,H.263+,H.264
Caller ID/ Call Hold/ Forward/ Transfer/ Waiting/ Parking
Call Paging and Intercom
Call Queue
Black List/ Phone Book
Music On Hold
DISA(Direct Inward System Access)
Flexible Dial Plan
Ring Group/ Conference Room
Call Logs
BLF(Busy Lamp Field)
Auto Provision
Feature Codes
Follow Me
Configuration By web
Built-in SIP/IAX2 server
Build-in voice mail server
System Backup and Restore
Echo Cancelation/VAD
Support Static/DHCP/PPPOE
VPN Client(Support N2N)
DDNS Client(Support dyndns.org /no-ip.org /zoneedit.com)
Support NTP(Network Time Protocol)
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2.2 Hardware Structure
Here, we take DVX-2004G as the sample to show the interface and the indicators at the
front and back panel.
2.2.1 Back Panel
4 GSM Antennas
2 Network Interface (RJ45)
1 Power Interface (DC 12V 2A)
1 Reboot Button
2.2.2 Front Panel
Mark Function Status Description
On Power On
PWR Power Status
Off Power Off
On System working
SYS System Status
Off System Failed
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Wink Data exchanging
WAN WAN interface Status
Off No Data exchanging
Wink Data exchanging
LAN LAN Interface Status
Off No Data exchanging
Red GSM channel
G1~G4 GSM Modules Status
Off Failed
Green FXS channels
Red FXO channels
*1-4 Analog Modules Status
Off Failed
2.2.3 Hardware:
32bit embedded RISC DSP
1G Onboard Nand Flash
128M Onboard SDRAM
2.2.4 Environmental Requirements:
temperature: -10 °C -45 °C
Storage temperature: -30 °C -65 °C
humidity: 10-80% no dew
Power: AC 100~240V
2.2.5 Packing List
DVX-2004 IP PBX 1 Unit
Power Adapter 1 Unit
Quick Start Guide 1 Piece
CD 1 Piece
Product Maintenance Card 1 Piece
Antenna(Only for DVX-2004G) 4 Piece
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Chapter3 Basic Configuration
3.1 Preparation Before Operation
What kind of IP Phone can be used with DVX-2004 IP PBX?
FXS Interface
Analog Phone(normal phone)
SIP Extension
DPH-150 , DPH-400
IP Phone which support SIP/ IAX2 protocol.
3.2 Before Making a Call
3.2.1 Login IP PBX
Getting IP Address
DVX-2004 IP PBX support 3 Ways to get the IP Address: Static/ DHCP/ PPPoE
Default IP and Port of WAN & LAN:
WAN Port IP: http://192.168.1.100:9999
LAN Port IP: http://192.168.10.100:9999
LAN Supper IP: 169.254.1.254/255.255.0.0
Default configuration and function key
Web GUI username: admin
Web GUI password: admin
**11 Play the IP Address of WAN port
**12 Play the IP Address of LAN port
600 Enter into the Voicemail Box
900 Enter into the Meeting
# Blind Transfer
*2 Attended Transfer
* Disconnect Call
Login to the system
After connecting the IP PBX to the local area network, launch the web browser on a
computer which is in this local area network. Enter the IP address of the system (WAN
port IP address http://192.168.1.100:9999, LAN port IP address http://192.168.10.100:
9999). The start web page will appear like this:
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Enter Username and password (default username is admin, password is admin), then
click “login”. Once the login is successful, the home page will be displayed:
Note:
1) You have to add a network segment same with the WAN port if your PC is not at
192.168.1.XXX.
2) For safety requirement, please modify the username and password after you login.
You can modify in this page: “System”---“Management”
3) Generally, based on the default setting, if user didn’t do anything in 1 min after login,
system will reflect it’s over time. If you want to continue operating, please login again.
If username and password are right, this following page will be displayed:
Network WAN/ LAN Port IP will be displayed
Storage Total storage and used storage will be displayed
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Channels Channel information will be based on the product model
Device Info Product Model and System Version will be displayed
Common Button
Besides of the device info in the home page, the following common buttons are displayed
as well:
Log out Log out GUI
Reboot Reboot the IP PBX system
Factory Defaults Restore all settings to factory default
Activate Changes Activate the changes for your current configuration
System Menu
System Menu include the following sub menu:
Home Page Display device info
Basic Basic configuration on extension, trunks, etc
Inbound Control Configure Inbound Route, IVR and Black List, etc
Advanced Configure extension's default info, conference, etc.
Status Check record list, call logs, register status, etc here.
System Configure network, time, etc; manage call logs, back up files, etc
3.2.2 Basic Configuration
Configure Extensions
D-link IP PBX support SIP/IAX2 and analog extension, configure extension from this
page: 【Basic】----【Extensions】---【Create New User】
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Item Explanation
Search Search extension precisely or fuzzily
Show all Show all extensions
Name Extension name (English letter is supported only) eg: "Tom"
Extension Be connected to the phone eg: "888"
Password Password of SIP/IAX2 extension eg: "12u3b6"
Outbound CID Overrides the caller id when dialing out with a trunk.
VM Password Voicemail Password for this user, eg: "1234".
E-mail The e-mail address for this user, eg. "Tom@gmail.com"
Dial Plan
Please choose the Dial Plan for this user，Dial Plan is defined
under the "Outbound Routes".
Analog Phone
If this user is attached to an analog port on the system, please
choose the port number here.
Voicemail This user will have a voicemail account after choosing this option.
Can reinvite
Set up calls directly between caller and receiver, after being
connected by IP PBX system. This method is known to cause
problems with certain hardware, such as the common Cisco ATA
186.
SIP
Check this option if the User or Phone is using SIP or is a SIP
device.
IAX2
Check this option if the User or Phone is using IAX2 or is an IAX2
device.
T.38 Fax Enables T.38 fax (UDPTL) pass through on SIP to SIP calls
Agent Check this option if this User or Phone is an Call Agent.
NAT
Check this option if the User or Phone is located behind a NAT
(Network Address Translation) enabled gateway.
Pickup Group Select your pickup group.
Delete VMail
Voicemail will not be checkable by phone if you choose this option.
Messages will be sent by email only.
Note: You must configure SMTP server for this functionality.
DTMF Mode
The Dual-Tone Multi-Frequency mode to be used is specified here
and can be changed if necessary. The default is rfc2833.
Video Call Enable/Disable Video call for this extension
Permit IP
IP address and network restriction.
eg: "192.168.1.77" or "192.168.10.0/255.255.255.0"
Auto Provision
Enter the MAC address of the specified extension, the system will
automatically create the config file accordingly. (Just support
DPH150 and DPH400 IP Phone)
Codecs Configure
The allowed and disallowed codecs can be selected by clicking this
link. Default codecs are alaw, ulaw and G.729.
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Note:
1) There are 30 default extensions which number started with "8", you can add or delete
extension by your requirement.
2) As our professional suggestion, extensions don't exceed 100. If extensions were over
100, it will cause the system crashed or other problems.
3.2.3 Time Based Rules
You can set working time rule and after-working time rule, and deal with your inbound call
based on this time rule. Please set from this page: 【Inbound Control】---【Time Based
Rule】---【New Time Rule】:
New Time Rule:
Item Explanation
Rule Name Define the time rule name.
Time & Date Conditions Set time segment of Month/Date/Week.
Destination
How to deal with the inbound call in different time
segment
eg: Inbound call will be forward to IVR in working time.
3.3 Outbound Call
3.3.1 Trunks
If you want to set up outbound call to connect to PSTN(Public Switch Telephone Network)
or VoIP provider, please configure on this page: 【Basic】->【Trunks】
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D-link IP PBX support 3 kinds of trunks: Analog/GSM line, Custom VoIP, Peer.
How to add each trunk:
1) Analog/ GSM Line
Click 【Add a Dial Rule】->【Analog/GSM】
Item Explanation
Description Define description for the trunk.
Lines
Individual lines of the PBX
eg: Analog Port #3: The third analog port of the PBX.
You can configure the Analog/GSM line through D-LINK IP PBX. Same Analog line
couldn't be used in multi trunks. If you don't have available Analog/GSM trunk, you can't
set up trunk.
3.3.2 Custom VoIP
Custom VoIP allows you to create a VoIP trunk, please configure on this page: 【Add a
Trunk】->【VoIP Trunk】
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Item Explanation
Description Description for VoIP Trunk, digit or letter is allowed.
Protocol
Choose protocol for this trunk, SIP or IAX2
Dial Plan
Choose a dial plan for this trunk, define it in the submenu named
【Outbound Routes】.
Register
Check for opening register service; otherwise register service is
closed
Host Host Address provided by VoIP Provider.
Outbound proxy Outbound proxy is provided by VoIP Provider.
Proxy Port Proxy Port is provided by VoIP Provider.
Without
Authentication
If you don't use Authentication when connecting server, pls check
this option.
Username Username provided by VoIP Provider.
Password Password provided by VoIP Provider.
3.3.3 Peer
D-LINK IP PBX will be taken as a Client when you use "Peer", it's used for outbound call
by connecting to another DVX-2004 IP PBX.
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Item Explanation
Peer Name Define the Peer Name, digit or letter is allowed.
Protocol Choose protocol for this trunk, SIP or IAX2
Dial Plan
Choose a dial plan for this trunk, define it in the submenu named
【Outbound Routes】.
Host IP Address of the other DVX-2004 IP PBX
NAT
Check this option, extension user will be configured after NAT
(Network Address Translation).
Without
Authentication
If you don't use Authentication when connecting server, pls check
this option.
Username Username provided by the other DVX-2004 IP PBX.
Password Password provided by the other DVX-2004 IP PBX.
Once A trunk is added, this trunk will be displayed in the "List of Trunk". You can define the
codecs, configure advanced settings or delete this trunk from the drop downs of "Option"
3.3.4 Outbound Routes
Outbound Routes is to define what trunk is used for outbound call by extension user. If
you don't allow extension user call out, please ignore this part.
Please configure on this page: 【Basic】->【Outbound Routes】
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On this page, you can configure basic match pattern of outbound routes and create
different dial plan. Please configure by clicking【Add a Dial Rule】
Item Explanation
Rule Name Set a name for this dial rule
PIN Set
Set PIN which you need input when you dial out by this rule.
Place this call through Choose a trunk for this rule
Failover
Choose a failover trunk for using when the above chosen
trunk is not available.
Dialing Rules Define the number match pattern for dialing.
Define a custom pattern
N digit from 2 to 9
Z digit from 1 to 9
X digit from 0 to 9
. One digit or multi digits
Delete[ ]digits prefix
If deleted one digit prefix, when dial 12345, digit 2345 will be
sent.
Auto-add digit[ ] If added digit"1", when dial 12345, digit 123451 will be sent.
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3.4 Inbound Call
3.4.1 Inbound Routes
When a call from outside, you want to forward this call to an extension or IVR, this Chapter
will introduce you how to deal with the inbound calls.
Please configure on this page:【Inbound Routes】
General
When a call from a trunk (Analog/ VoIP), it could be forwarded to an extension, call queue,
conference or IVR. You can choose based on your requirement.
Analog Channel DID
If you want to direct the inbound call from a trunk (Analog) to a specified extension, call
queue, conference or IVR, please configure on this page:【Add Analog Channel】
Channel Choose Analog Port of trunk
DID Extension Select Extension, call queue, conference or IVR for DID.
VoIP Channel DID
If you want to direct the inbound call from a VoIP trunk to a specified extension, call queue,
conference or IVR, please configure on this page:【Add VoIP Channel】
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DID Number DID number calling into VoIP (This number is configured in the
advance option of VoIP trunk)
DID Extension Choose a specified extension, call queue, conference or IVR to be
directed to call.
3.4.2 IVR
IVR will improve office efficiency based on your requirement.
Please configure on this page【IVR】
Item Explanation
Name Set a name for the IVR
Extension
If you want to listen to the IVR by dialing extension, please
input an extension Number.
Please Select
Select IVR audio file, please configure in this page:
【IVR Prompts】
Repeat Loops loop times to repeat playing the IVR prompt.
Dial other Extensions
Allow caller to dial other extension besides of the ones listed
as below.
Keypress' Events Each digit will be related to the actions defined in the blank.
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3.4.3 IVR Prompts
Record or play IVR music from extension. Please configure on this page:【IVR Prompts】
Click【Record a new voice】to display the diagram as below:
File name Define a name for the recorded IVR file
Format Choose the format of the voicemail messages
in this selection box.
Extension used for recording Select an extension for recording,click【Record】
button, the selected extension will ring, then you can record IVR.
3.4.4 Ring Groups
Ring Group is a collection of extensions. When a call to a ring group, all extensions in this
ring group will ring in different way based on their different configuration, if ring time
exceeded defined time, the call will be directed to IVR or others based on your
configuration.
There isn't any data in the factory default 【Ring Groups】, please configure as below:
Click【New Ring Group】to display the diagram as below:
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Name Define a name for this ring group
Strategy Select strategy : "Ring all" or "Ring in order"
Ring Group Members Select ring group members in available channels,
click to add.
If not answered You can choose forward the call to extension, extension,
Voicemail, RingGroup, IVR or Hangup.
3.5 Black List
If some numbers need to be blocked, you can use this functionality.
Please configure in【Black List】, click【New Blacklist】to display this dialog as below:
Input caller's number in the blank, then this caller's number will be blocked when call again.
Meanwhile, extension user can add or delete the blacklist number by function key on the
phone.
Please operate as the following diagram:
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Reference Parameters and Explanation of Blacklist:
Item Explanation
*30 When the extension user (in the system) input *30 to add a
blacklist number, this number will be added to the "Black List"
*31 When the extension user input *31+ blacklist number, this number
will be deleted from the "Black List".
3.5.1 Pickup Call
If an extension user is away from his/her desk, other extension users can pickup the call
by function key on the phone. Please check the following diagram to learn:
Reference Parameters and Explanation of Pickup Calls
Item Explanation
*8
Pick up the ringing extension (in the system) at random. This can
be defined in【Feature Codes】
*7
Defined extension number must be inputted after *7. This can be
defined in【Feature Codes】.
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3.6 On The Call
3.6.1 Call Parking
If you picked up a call at your seat, but it's not convenient to talk in public, you need go to
the conference room to talk secretly. At this time, you can input 700 to park this call, the
system will tell you a parking number 701 which you can input for continuing conversation
when you go to the conference room. Please check the diagram as below to learn:
Reference Parameters and Explanation of Call Park:
Item Explanation
Extension to Dial
for Parking Calls:
Default number is 700. It can be defined in【Feature Codes】
What extension to
park calls on
Default number is 701-720.It can be defined in【Feature Codes】
How many
seconds a call can
be parked for
Default is 45 seconds. It can be defined in【Feature Codes】
3.6.2 Transfer
If an incoming call asked to speak to your colleague, you can transfer the call directly to
your colleague or transfer the call after agreed by your colleague. Please check the
diagram as below to learn:
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Reference Parameters and Explanation of Transfer:
Item Explanation
Blind Transfer Default is #, it can be defined in【Feature Codes】
Attended Transfer Default is *2, it can be defined in【Feature Codes】
Disconnect Call
Default is *, it can be used after you use function key " *2 ".
it can be defined in【Feature Codes】
Timeout for answer on
attended transfer
Default is 15 seconds, it can be defined in【Feature Codes】
3.6.3 Conference
If you wanted to create a conference room for some extension users or with external lines,
you can input conference room number 900, input conference room password 1234
(Admin's password is 2345), then enter into conference room. Please configure on this
page【Conference】:
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Item Explanation
Conference Number
The number that users call in order to access the
conference room, the default number is "900".
PIN Code Participants enter the conference room by this code.
Admin PIN Code Administrator enter the conference room by this code.
Conference DialPlan Use the dialplan when you invite the other participant.
Play hold music for first
caller
Check this option, Asterisk will play Hold Music to the first
user in a conference, until another user has joined the
same conference.
Enable caller menu
Checking this option allows a user to access the
Conference Bridge menu by pressing the * key on their
dialpad.
Announce callers
Checking this option announces to all Bridge participants,
the joining of any other participants.
Record conference Recording format is WAV。
Quiet Mode
If this option was checked, all users entering this
conference will be marked as quiet, and will be in
Listen-Only mode.
Leader Wait
Wait until the conference leader (admin user) arrives
before starting the conference.
Please check the following diagram to learn:
Go to conference:
In the conference, admin can add new participant (extension user or external number) into
the conference.
Add new participant:
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3.6.4 Monitor
Monitor the specified extension, also you can monitor in different time.
Please click【Monitor】--【New Monitor】to configure:
Item Explanation
Extension Select an extension which need to be monitored
Monitoring Time Always monitor or monitor in different time.
Monitor Settings Set inbound record and outbound record.
3.7 Settings before leaving office
3.7.1 Follow Me
If you don't want to lose any call, you can use this function.
Please click【Follow Me】---【New Follow Me】
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Item Explanation
Extension Choose an extension
Ring lasting for(s)
Default is 20 seconds, you can define it by
yourself.
Always All incoming calls will be forwarded
Busy Forward when extension is busy
Status
No answer Forward when extension not answer
Forward to an Internal
Extension
Incoming call will be forwarded to internal
extension.
Set your
Follow Me
number
Forward to an External
Extension
Incoming call will be forwarded to external
number or mobile number.
Set Internal Extension Set an internal extension to pick up the call.
Select DialPlan
Select DialPlan when forward the call to
external number.
Set External Number Set external number, like Mobile number.
3.7.2 VoiceMail
If you don't want to configure "Follow Me", you can record the message of incoming call,
and email the message to your defined mailbox.
Click【Extension】---【Extension Settings】
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【VoiceMail】must be opened and【VM Password】must be configured before using
"VoiceMail"。If no answer, when default ring time is over, the system will play and ask you
to leave your message, press # to end recording. If you configured email, your voice
message will be sent to your defined email.
Leave a message:
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Listen to the message
Note:
1) If you would like using this function, you must write correct email address in "extension
settings"
2) You need configure SMTP and Email model in【VoiceMail】, please check the details
in the following chapter【VoiceMail】
3.8 Call Queue
3.8.1 Create Agent
Check agent in the 【Extension Settings】---【Advanced Options】, then assign agent and
Ring Strategy in 【Call Queue】, please learn from the following configuration interface:
Item Explanation
Queue Number
This option defines the extension number that may be
dialed to reach this Queue.
Queue Name This option defines a name for this Queue, eg. "Sales"
Ring Strategy
RingAll -- Ring All available Agents until one
answers(default).
RoundRobin -- Take turns ringing each available Agent.
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LeastRecent -- Ring the Agent which was called least
recently.
FewestCalls -- Ring the Agent with the fewest completed
calls.
Random -- Ring a Random Agent.
RRmemory --RoundRobin with Memory, and remember
where it left off in the last ring pass.
Agents
All the users who is defined as Agent will be shown here.
Selected agent will be a member of the current Queue.
Item Explanation
Agent TimeOut(s)
This option defines the time in seconds that an Agent's
phone rings before the next Agent is rung, eg. "15"
Auto Pause Pause an Agent if they fail to answer a call.
Wrap-Up-Time(s)
After a successful call, how many seconds needed to wait
before sending another call to a potentially free agent
(Default is 0, which means No Delay).
Max Wait Time(s)
The maximum number of seconds a caller can wait in a
queue before being pulled out(empty for unlimited).
Max Callers
This option sets the maximum number of callers that may
wait in a Queue(Default is 0, Unlimited).
Join Empty
Defining this option allows callers to enter the Queue when
no Agents are available. If this option is not defined, callers
will not be able to enter Queues with no available agents.
Leave When Empty
Defining this option forces all callers to exit the Queue if New
Callers are also not able to Enter the Queue. This option
should generally be set in concert with the "Join Empty"
option.
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Auto Fill
Defining this option causes the Queue, when multiple calls
are in it at the same time, to push them to Agents
simultaneously. Thus, instead of completing one call to an
Agent at a time, the Queue will complete as many calls
simultaneously to the available Agents.
Report Hold Time
Check this option if you wish to report the caller's hold time
to the agent member before they are connected to the caller.
Frequency(s)
How often to announce queue position and estimated
holdtime(0 to Disable Announcements).
Announce Hold Time
Should we include estimated hold time in position
announcements? Either yes, no, or only once; hold time will
not be announced if <1 minute.
Repeat Frequency(s)
How often to announce a voice menu to the caller(0 to
Disable Announcements).
Announcements
Prompt
Select the 'Announcements Prompt' from IVR Prompts
3.8.2 Agent Registration
You need register for using after creating agents.
Agent Registration when hook off
Agent Registration when hook on
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3.8.3 Agent Log Off
If agent would leave and log off, none of agent will answer calls then.
Agent Log Off:
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Chapter 4 Advanced
4.1 Options
Options Include local extension settings and new extension default settings.
Click【Option】to display the diagram as below:
Item Explanation
Local Extensions Set up the digit of local extensions
Operator Extension Set up Operator Extension.
Global Ring Time Set(s) Set Ring Time for each extension.
Enable Transfer Enable transfer feature key.
Enable Music On Ringback Enable music on ringback.
Allow multiple extensions to
be assigned to one analog
phone
Allow multiple extensions to be assigned to one
analog phone.
Allow extensions to be Alpha
Numeric (SIP/IAX users)
If extension is Alpha, outside line can't call in, but
extension can call out.
NAT
Check this option if the User or Phone is located
behind a NAT (Network Address Translation)
enabled gateway.
SIP
Check this option if the User or Phone is using SIP or
is a SIP device.
IAX2
Check this option if the User or Phone is using IAX2
or is an IAX2 device.
Agent
Check this option if the User or Phone should have
Call-Waiting capability.
VM Password Voicemail Password for this user, eg: "1234".
Delete VMail Voicemail will not be checkable by phone if you
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chose this option. Messages will be sent by e-mail
only. Note:you must configure SMTP server for this
functionality.
4.2 VoiceMail
Details configuration on VoiceMail: VoiceMail Reference/ Voice Message Options/
Playback Options. If you need send message by mail to your defined mailbox, you must
configure SMTP and Email model. Click【Voicemail】to display the diagram as below:
Item Explanation
Extension for checking
messages
The number that users call in order to access their
voicemail accounts, the default number is "600".
Max greeting(seconds)
Defining this option to set a maximum time for the
greeting message.
Direct to Voicemail Defining this option to go to voicemail box directly.
Dial "0" for Operator
Callers entering the voicemail application can leave for
Operator by dialing "0".
Message Format
Choose the format of the voicemail messages in this
selection box.
Maximum Messages
Choose the maximum number of messages in this
selection box.
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Maximum message time
(min)
Choose the maximum duration of a voicemail message.
Message recording will be stopped when it's timeout.
Minimum message time
(s)
Choose the minimum duration of a voicemail message in
this selection box. Message time below this threshold will
be deleted automatically.
Say message Caller-ID
Choose this option to play Caller's ID before voicemail
message is played.
Say message duration
Choose this option to play the duration of message
before the voicemail message is played.
Play envelope
Choose this option to play envelop (including date, time
and caller ID).
Allow users to review
Choosing this option, the caller leaving the voicemail can
review their recorded message before it's submitted.
SMTP Settings:
Item Explanation
SMTP server
In order to send e-mail notifications of your voicemail. Set
the IP address or domain name of a SMTP server that
your IP PBX may connect to.
eg: mail.yourcompany.com
Port
The port number which the SMTP server running is
generally port 25. If SSL is encrypted, please use port
465 instead.
SSL/TSL Enable use SSL/TLS to send secure messages to server.
Enable SMTP
Authentication
If your SMTP server needs Authentication, please enable
SMTP Authentication, and configure the following
information.
Username Input username of your email box.
Password Input password of your email box.
Email Settings
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Item Explanation
Attach recordings to e-mail
This option defines whether or not voicemails are sent
to the Users' e-mail addresses as attachments.
Sender Name
Display the Sender name when you receive a
voicemail.
From Sender's email address
Subject Subject of the mail
Message The message pattern
4.3 Music Settings
Management for music on hold, music on ringback, music on call queue.
Click【Music Settings】 to display the diagram as below:
Music Settings:
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Please define different music file for different music folders.
Music Management:
Item Explanation
Directory Load music in the music file.
Files Display music in the music file, or you can delete it.
Enter The Music File Name
Input music file name which you want to upload.(GSM
format)
TFTP Server IP address Please enter your TFTP server IP address.
Select Music Directory
Select directory where the uploaded music file will be
saved.
4.4 DISA
A trunk call into the PBX, and call to another trunk through outbound route of the PBX. Eg:
This trunk can make international call, you are out of the office and want to contact with
your customer in foreign country, now you can dial DISA number, after PIN authentication,
you are connected to your customer, and you can speak to your customer now.
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Click【DISA】---【New DISA】to display the diagram as below:
Item Explanation
Name Give this DISA a brief name to help you identify it.
PIN The user will be prompted for this number
Response Timeout(s)
The maximum amount of time it will wait before
hanging up if the user has dialed an incomplete or
invalid number. Default is10 seconds.
Digit Timeout(s)
The maximum amount of time permitted between
digits when the user is typing in an extension. Default
is 5 seconds.
Extension for this DISA
(Optional)
If you want this DISA to be accessible by dialing an
extension, you can define an extension number for
this DISA.
Select DialPlan Set the DialPlan that calls will originate from.
4.5 Paging And Intercom
Paging And Intercom is used for calling a paging extension, all terminals which support
this function will be picked up automatically and listen, meanwhile, it supports duplex.
Click【Paging And Intercom】---【Add Paging Group】to display the diagram as below:
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Item Explanation
Paging
Extension
The number users will dial to page this group.
Description Provide a descriptive title for this Page Group.
Paging Group
Members
Selected device(s) in this page
Device List Select Device(s) to Page.
Duplex
Paging is typically one way for announcements only. Checking
this will make the paging duplex, allowing all phones in the
paging group to be able to talk and be heard by all. This makes it
like an "instant conference".
4.6 Monitor
Monitor is used for recording the defined extensions.
Click【Monitor】---【New Monitor】to display the diagram as below：
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Item Explanation
Extension Define an extension.
Monitoring Time Set monitoring time
Inbound Record Check to record inbound calls
Outbound Record Check to record outbound calls
4.7 Phone Book
If incoming call was matched with the number in the phone book, the incoming call will
display the name of matched number.
Click【Phone Book】to display the diagram as below:
Search Input contact name to search
Show All Show all contacts
Name Add contact's name, Alphabetic or numeric only.
Number Add contact's number, international phone number is supported.
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4.8 Feature Codes
Click【 Feature Codes】to display the diagram as below, you can define relevant
parameter.
Item Explanation
Extension to Dial for
Parking Calls
Set Call Parking number.
What extensions to park
calls on
What extensions to park calls on, eg: (701-720)
How many seconds a
call can be parked for
Set the call time by second, if it's time out, system will call the
previous extension again.
Pickup Extension Set Pickup Extension.
Pickup Specified
Extension
Set Pickup Specified Extension, default: dial *7+extension to
pickup the extension.
Blind Transfer Allow unattended or blind transfers. It works like this: While
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on a conversation with A, you dial the blind transfer key
sequence. The system says "Transfer" then gives you a dial
tone, while A is on hold. You dial the transferee number(B's
number) and A is put through to B immediately. Your line is
off. The caller ID displayed to B is exactly the same as the
caller ID presented to you.
Attended Transfer
Allow attended transfer or supervised transfer. It works like
this: While on conversation with A, you dial the Attended
Transfer key sequence. The system says "Transfer" then
gives you a dial tone, while A is on hold. You dial the
transferee number(B's number) and talk with B to introduce
the call, then you can hang up and A will be connected with
the B. In case B does not want to answer the call, he/she
simply hangs up and you will be back to your original
conversation.
Disconnect Call Disconnect the current transfer call(for Attended transfer).
Timeout for answer on
attended transfer
Set the answer timeout value.
Blacklist a number Add a black list number.
Remove a number from
the black list
Remove a black list number.
Invite Participant
The administrator can invite another person by pressing 0
when he/she is in the conference. When you press 0, you will
get a dialtone to enter the number of part A you also would
like to invite. After the call has been established and you talk
to B, you can press ** to direct him to the conference, or *# to
hang up the current call and return to the conference
yourself.
Create Conference
While you speak with another party you can press *0, you
and the callee are immediately transferred to conference.
Return to conference
with participant
The administrator can invite another person by pressing 0
when he/she is in the conference. When you press 0, you will
get a dialtone to enter the number of part A you also would
like to invite. After the call has been established and you talk
to B, you can press ** to direct him to the conference, or *# to
hang up the current call and return to the conference yourself
Return to conference
without participant
The administrator can invite another person by pressing 0
when he/she is in the conference. When you press 0, you will
get a dialtone to enter the number of part A you also would
like to invite. After the call has been established and you talk
to B, you can press ** to direct him to the conference, or *# to
hang up the current call and return to the conference
yourself.
Agent Login Extension
Logs the current caller into the queue as a call agent. Once
logged in, the agent can take calls with the phone off-hook;
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each call is preceded by a warning tone. Calls are ended by
pressing the "*" key.
Agent Callback Login
Extension
Extension to be dialed for the Agents to Login to the Specific
Queue.
Same as Agent Login Extension, except you do not have to
remain on the line.
Agent Logoff Extension Agent logoff from the queue.
Pause Queue Member
Extension
'Pauses' a queue member. so that the member can not
receive calls.
Unpause Queue
Member Extension
'Unpause' a queue member who is 'paused' previously. so
that the member can receive calls again.
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Chapter 5 Status
This chapter will introduce you the status of record list, call logs, system info, register
status etc.
5.1 Record List
Check the record list of defined extension or conference, you can delete the record list.
Click【Record List】---【Monitor】and【Conference】will be displayed as below:
Monitor List Interface
Conference List interface
5.2 Call Logs
Check call logs of extension by caller ID or callee ID. Click【Call Logs】 to display the
diagram as below:
Call Logs Interface
Note:
Duration in the call logs is not real charged duration, if you need billing, PSTN must
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support polarity reversal function, meanwhile, you must configure relevance parameters
of polarity reversal in trunk configuration for the IP PBX.
5.3 Register Status
Check SIP/ IAX2 User, and SIP/IAX2 Trunk status. Click【Register Status】to display the
diagram as below:
5.4 System Info
Check OS version, firmware version and memory, etc from here.
Click【System Info】to display the diagram as below:
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Chapter 6 System
This chapter will introduce you how to configure the system of D-LINK IP PBX.
6.1 Network And Country
Configure WAN/ LAN IP, and tone zone.
Click【Network And Country】to display the diagram as below:
IP Assign Support Static, DHCP and PPPoE.
Hostname The name identifying this machine on the network.
IP address Set static IP address of WAN port on this device.
Subnet mask Specify the subnet mask of the above static IP address.
Gateway Specify a gateway to reach non-local addresses.
DNS Specify a nameserver to resolve domain names.
Tone Zone Setting Define the tone zone for home country or place.
6.2 TroubleShooting
You can ping other network device through D-LINK IP PBX and track network route by
command "Traceroute" .
Click【TroubleShooting】to display the diagram as below:
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6.3 DHCP Server Settings
Click【Network Advanced】---【DHCP Server Settings】 to display the diagram as below:
DHCP Server Enable/Disable DHCP Server.
Interface Selcet DHCP service work on the which port.
Start IP Set start IP of the DHCP IP address pool.
End IP Set end IP of the DHCP IP address pool.
Lease Time Set lease time of the IP address.
TFTP Server Set TFTP server for auto provision.
6.4 DDNS & VPN
After configure DDNS, you can visit by domain remotely.
Click【DDNS】to display the diagram as below:
DDNS Settings:
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Click【VPN】to display the diagram as below:
Note:
1) DDNS supports the domain provided by Dyndns.org / no-ip.com /zoneedit.com.
2) VPN supports N2N only.
6.4 Time Settings
Click【Time Settings】to display the diagram as below:
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Item Explanation
NTP Server
Specify the NTP server that you wish to use. You may type either
the domain name or the IP address of the server, and it may be
either remote or local. The default server is pool.ntp.org. Be
aware that the PBX needs to be able to connect to a NTP server
for perfect function.
Time Zone
Select your time zone so that the system will set time based on
the time zone.
Synchronize with
current PC time
Click the button to synchronize the PBX time with the current PC
time.
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6.5 Management
Management on username, password, access permit, etc. Click【Management】to display
the diagram as below:
Change Password You can change the password of admin here (default
password is admin)
Set Language Set voice language of the system. And you can set the SIP &
Analog channel here by clicking "Show Advanced Options"
Click【Management】---【Access Permit】to display the diagram as below:
Note:
After you added a permitted IP, you can only login the system by this IP, other IP address
isn't effective to login the system.
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6.6 Data Storage
Upload the voicemail, monitor, conference, call logs, etc to the defined FTP server for
storage.
Click【Data Storage】to display the diagram as below:
Item Explanation
Enable Uploading Enable periodical FTP uploading.
Server Address Set FTP Server address(IP address or Domain).
User Name FTP account name.
Password FTP account password.
Directory Define a directory on the FTP server.
Note:
1) Upload Voicemail, Conference record, Monitor and Call logs to the defined FTP
Server automatically when flash storage is over 40%. Then the history files will be
removed out automatically.
2) NOT upload in working time by default.
6.7 Backup
Backup all the settings. Click【Backup】to display the diagram as below:
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Restore Restore your selected backup file to system.
Delete Delete your selected backup file.
Download your selected backup file to your PC. (Note: Please don't
change the backup file name)
6.8 Update
Here, you can upload firmware, IVR prompt, backup files to update the system.
Click【Update】to display the following diagram:
Extract the downloaded firmware package which includes one TFTP server and one
upgrading file.
Run TFTP server, you will see the following interface:
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Go into the "update" page, and upload firmware;
Enter the package name
Enter TFTP Server IP address,
Click Update button to finish upgrading system package after entering the TFTP Server IP.
Then system will reboot automatically.
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