Telco Depot TD100 SERIES, TD100-A202 User Manual

TD100 SERIES IPPBX USER MANUAL

V1.2

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TABLE OF CONTENTS

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ee ............................................................................................ - 3 -

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nn ..................................................................................... - 4 -

2.1 Brief introduction of TD100 ....................................................................................... - 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 -

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nn ................................................................................. - 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 ...................................................................................... - 12 -

3.3 Outbound Call ........................................................................................................... - 12 -

3.3.1 Trunks ........................................................................................................... - 12 -

3.3.4 Outbound Routes ........................................................................................ - 15 -

3.4 Inbound Call .............................................................................................................. - 17 -

3.4.1 Inbound Routes ........................................................................................... - 17 -

3.4.2 IVR ................................................................................................................ - 19 -

3.4.3 IVR Prompts ................................................................................................ - 19 -

3.4.4 Ring Groups ................................................................................................ - 21 -

3.5 Blacklist .................................................................................................................... - 22 -

3.5.1 Pickup Call ................................................................................................... - 23 -

3.6 On The Call ............................................................................................................... - 23 -

3.6.1 Call Parking ................................................................................................. - 23 -

3.6.2 Transfer ........................................................................................................ - 24 -

3.6.3 Conference .................................................................................................. - 24 -

3.7 Settings before leaving office .................................................................................... - 26 -

3.7.1 Follow Me ..................................................................................................... - 26 -

3.7.2 VoiceMail ...................................................................................................... - 27 -

3.8 Call Queue .................................................................................................................... - 29 -

3.8.1 Create Agent................................................................................................ - 29 -

3.8.2 Agent Registration ................................................................ ...................... - 31 -

3.8.3 Agent Log Off .............................................................................................. - 32 -

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dd ............................................................................................... - 33 -

4.1 Options ...................................................................................................................... - 33 -

4.2 VoiceMail .................................................................................................................. - 35 -

4.3 Music Settings ........................................................................................................... - 38 -

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4.4 DISA ......................................................................................................................... - 40 -

4.5 Paging And Intercom ................................................................................................. - 41 -

4.6 Call Recording .......................................................................................................... - 41 -

4.7 Phone Book ............................................................................................................... - 42 -

4.8 PIN Set ...................................................................................................................... - 43 -

4.9 Feature Codes ............................................................................................................ - 44 -

4.10 Phone Provisioning ................................................................................................. - 47 -

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ss ................................................................................................... - 48 -

5.1 Recording List ........................................................................................................... - 48 -

5.2 Call Logs ................................................................................................................... - 48 -

5.3 Register Status ........................................................................................................... - 49 -

5.4 System Info ............................................................................................................... - 49 -

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mm ................................................................................................. - 50 -

6.1 Network And Country ............................................................................................... - 50 -

6.2 TroubleShooting ........................................................................................................ - 51 -

6.3 Netword Advanced .................................................................................................... - 51 -

6.4 Time Settings ............................................................................................................ - 53 -

6.5 Management .............................................................................................................. - 54 -

6.6 Data Storage .............................................................................................................. - 56 -

6.7 Backup ...................................................................................................................... - 57 -

6.8 Upgrade ..................................................................................................................... - 58 -

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nn .............................................................................. - 60 -

7.1 How to connect the TD100 IP PBX to the Internet ................................................... - 60 -

7.2 How to combine two TD100 IP PBX in the same network ....................................... - 60 -

7.3 How to combine two IPPBX in different network .................................................... - 64 -

7.4 How to resolve problems about hearing on one side only......................................... - 66 -

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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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2.1 Brief introduction of TD100

TD100 Series 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 SIP/IAX2 registers
 Video Calls
 Phone Provisioning
 Multiple Language
 DID(Direct Inward Dialing Number)
 Support SKYPE for SIP
 Support USB disk recording(Scalable)
 Call Recording
 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)
 Configuration By web
 Built-in SIP/IAX2 server
 Build-in voice mail server
 System Backup and Restore
 Echo Cancelation/VAD
 Support Static/DHCP
 VPN Client(Support N2N/L2TP)

- 5 -

 DDNS Client(Support Dyndns.org /No-ip.com)
 Support NTP(Network Time Protocol)
 Support POE

2.2 Hardware Structure

Here, we take TD100-A202 as the sample to show the interface and the indicators.

2.2.1 Back Panel

 2 Analog Port(RJ11)
 1 Network Interface (RJ45)
 1 Power Interface (DC 12V 2A)

 1 Reboot Button

2.2.2 Front Panel

- 6 -

Mark

Function

Status

Description

PWR

Power Status
On

Power On

Off

Power Off

SYS

System Status
On

System working

Off

System Failed

WAN

WAN interface Status
Wink

Data exchanging

Off

No Data exchanging

Off

No Data exchanging

USB

USB Interface Status
on

With Mobile USB Disk

Off

Without Mobile USB Disk

Port1-Port2

Analog Modules Status
Green

FXS channels

Red

FXO channels

Off

Failed

2.2.3 Hardware:

 32bit embedded RISC DSP
 256M Onboard Nand Flash
 64M 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

 TD100 IP PBX 1 Unit
 Power Adapter 1 Unit
 Quick Start Guide 1 Piece
 Product Maintenance Card 1 Piece
 Network Cable 1 Unit

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3.1 Preparation Before Operation

What kind of IP Phone can be used with TD100 IP PBX?
FXS Interface

 Analog Phones (requires an FXS port)
 SIP Extension
 IP Phone which support SIP/ IAX2 protocol (eg: CISCO, Grandstream, etc.)

3.2 Before Making a Call

3.2.1 Login IP PBX

Getting IP Address

Series IP PBX support 3 Ways to get the IP Address: Static/ DHCP
Default IP And Port of WAN:

 WAN Port IP: http://192.168.1.100:9999

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
 *97 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. The start web page will appear like this:

- 8 -

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) Please use IE(7.0 or more higher verion) and Firefox browsers.

2) You have to add a network segment same with the WAN port if your PC is not at

192.168.1.XXX.

3) For safety requirement, please modify the username and password after you login. You
can modify in this page: “System”---“Management”

4) 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 Port IP and MAC will be displayed
 Storage Total storage and used storage will be displayed
 Channels Channel information will be based on the product model
 Device Info Product Model and System Version will be displayed

- 9 -

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

TD100 IP PBX support SIP/IAX2 and analog extension, support "Batch Add Users".
configure extension from this page: 【Basic】----【Extensions】

Extension Settings

Item

Explanation

Search

Search extension precisely or fuzzily

Show all

Show all extensions

- 10 -

Extension

Be connected to the phone eg: "888"

Name

Extension name (English letter is supported only) eg: "Tom"

Password

Support default or random password, combined by letter and
figure, eg: "12u3b6"

Caller ID

Caller's ID eg: "801"

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. "[e-mail]"

Analog Phone

If this user is attached to an analog port on the system, please
choose the port number here.

Dial Plan

Please choose the Dial Plan for this user，Dial Plan is defined
under the "Outbound Routes".

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

TD IP PBX can work with Grandstream and Yealink IP Phone on
this function. Pls select the phone manufacture and input MAC
address of the IP Phone. For more details, pls check in Part 3.10

Codecs Configure

The allowed and disallowed codecs can be selected by clicking this
link. Default codecs are alaw, ulaw and G.729.

- 11 -

Note:

1) There are 30 default extensions which number started with "8", you can add or delete

extension by your requirement.

Upload/Download Extensions

If you want to batch add users, please click【Upload/Download Extensions】to configure on
this page:

Please download the demo from【Download Extensions demo】, add extension files and
save based on the demo, choose the extension file which you wanna upload.
You can download the extension file by click【Download Extensions(.csv)】

Extensions Log

Click 【Extensions Log】to check the extensions log, you can refresh or clear the log:

- 12 -

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: 【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】

- 13 -

TD100 IP PBX support 3 kinds of trunks: Analog line, Custom VoIP, Peer.

How to add each trunk:

1) Analog

Click 【Add a Trunk】->【Analog】

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.

Prefix

The prefix will be added as default, when this trunk is used.

You can configure the Analog line through TD IP PBX. Same Analog line couldn't be used
in multiple trunks. If you don't have available Analog/GSM trunk, you can't set up trunk.

- 14 -

2) Custom VoIP

Custom VoIP allows you to create a VoIP trunk, please configure on this page:
【Add a Trunk】->【VoIP Trunk】

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.

Prefix

The prefix will be added as default, when this trunk is used.

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) Peer

TD IP PBX will be taken as a Client when you use "Peer", it's used for outbound call by
connecting to another TD100 IP PBX.

- 15 -

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 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 TD100 IP PBX.

Password

Password provided by the other TD100 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】

- 16 -

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.

Record in CDR

If you selected it, CDR will show which pin the call is outbound
through

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

- 17 -

. One digit or multiple 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.

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】

- 18 -

 Channel Choose Analog Port of trunk
 Associated 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】

 DID Number DID number calling into VoIP (This number is configured in the

advance option of VoIP trunk)

 Associated Extension Choose a specified extension, call queue, conference or

IVR to be directed to call.

DOD Settings

If you want to direct the inbound call from any trunks to a specified extension, call queue,
conference or IVR, please configure on this page:【Add DOD】

 DOD Number This number is the caller's phone number, it could be called from

analog channel or VoIP/GSM/E1/T1 Line.

 Associated Extension Choose a specified extension, call queue, conference or

IVR to be directed to call.

- 19 -

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 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.

3.4.3 IVR Prompts

Record or play IVR music from extension. Please configure on this page:【IVR Prompts】

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Click【Record a IVR Prompt】to display the diagram as below:

 File name Define a name for the recorded IVR prompt
 Format Define the format of the IVR Prompt, only

GSM/WAV(16-bit)supported

 Extension Select an extension for recording, click【Record】

button, the selected extension will ring, then you can record IVR.

If your want to listen to the recorded IVR prompt, please click【play】and input extension
number in the following diagram, click【confirm】, the extension will ring and play the IVR
prompt after hang up.

Upload Prompts

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TD PBX prompts support wav, gsm format, ulaw or alaw, and the size is limited in 15MB.

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 Blacklist

If some numbers need to be blocked, you can use this functionality.
Please configure in【Blacklist】, 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:

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".

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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】

**

Defined extension number must be inputted after **. This can be
defined in【Feature Codes】.

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 70. 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】

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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:

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. This model support 3
conference rooms. Please configure on this page【Conference】:

## +

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Item

Explanation

Extension

The number that users call in order to access the
conference room, the default number is "900".

Guest Password

Guest enter the conference room by this code.

Administrator
Password

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

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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 guest :

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

Status

Always

All incoming calls will be forwarded

Busy

Forward when extension is busy

No answer

Forward when extension not answer

Ring lasting for(s)

Default is 20 seconds, you can define it by
yourself.

Set your
Follow Me
number

Forward to an Internal
Extension

Incoming call will be forwarded to internal
extension.

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
above 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:

*97

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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.
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

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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.

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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4.1 Options

Options
Options Include local extension settings and new extension default settings.
Click【Option】to display the dialog 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.

SIP

Enable this option if the User or Phone is using SIP
or is a SIP device.

IAX2

Enable this option if the User or Phone is using IAX2
or is an IAX2 device.

Agent

Enable this option if the User or Phone is an Call
Agent.

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NAT

Enable this option if the User or Phone is located
behind a NAT (Network Address Translation)
enabled gateway.

VM Password

Voicemail Password for this user, eg: "1234".

Delete VMessage

Voicemail will not be checkable by phone if you
chose this option. Messages will be sent by e-mail
only. Note:you must configure SMTP server for this
functionality.

Global Analog Settings

Click 【Options】---【Global Analog Settings】to see the following diagram:

Item

Explaination

Caller ID Detection

For FXO trunk lines,this option causes PBX to look
for Caller ID on incoming calls

Caller ID Signalling

This option allows you to choose the type of Caller
ID signalling to use.
Bell-US-- Used in the United States;
DTMF-- Used for callerID under DTMF mode.(eg:

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Denmark, Sweden and Netherlands etc);
V23-- used in the UK;
V23-Japan-- used in Japan

Caller ID Start

This option allows you to define the start of a Caller
ID signal:
Ring-- to start when a ring is received.
Polarity-- to start when a polarity reversal is started.

CID Buffer Length

Default CID Buffer Length

FXO Mode

Select FXO Mode here

Relax DTMF

If you met trouble with DTMF detection, you can
relax the DTMF detection.

Echo Cancel

Enable/Disable the Echo Cancel function.

Echo Training

Enabling echo training will cause the PBX system to
briefly mute the channel, send an impulse, and use
the impulse response to pre-train the echo canceller
so it can start out with a much closer idea of the
actual echo. Value may be "yes", "no", or a number
of milliseconds to delay before training (default =

400)

Busy Detection

Used for detecting far end hang up or a busy signal.

Busy Count

If Busy Detection is enabled, it is also possible to
specify how many busy tones to wait for before
hanging up. The default is 4, but better results can
be achieved if set to 6 or even 8. Mind that the higher
the number, the more time that will be needed to
hang up a channel, but lower the probability that a
false detection may occur.

Call Progress

If turned on, call progress attempts to determine
answer, busy, and ringing on phone lines.

Global SIP Settings

【Global SIP Settings】is appropriated for operating by professional engineer or technician,
if you need modification, please contact with our technician support.

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 dialog as below:

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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.

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

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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.

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Email Settings

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 dialog as below:

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Music Settings:

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.

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Enter The Music File Name

Input music file name which you want to upload.(GSM/
WAV format, If it's WAV, it must be accord with PCM
16 bits, 8000HZ 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.
Click【DISA】---【New DISA】to display the dialog 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.

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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 dialog as below:

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 Call Recording

Call Recording is used for recording the defined extensions.
Click【Call Recording】---【New Call Recording】to display the dialog as below：

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Item

Explanation

Extension

Define an extension.

Call Recording 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 dialog as below:

 Search Input contact name to search
 Show All Show all contacts

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 Name Add contact's name, Alphabetic or numeric only.
 Number Add contact's number, international phone number is supported.

TD100 IP PBX also support "Batch Add Users", Click【Advanced】->【Upload/Download
Phonebook】to display the following diagram:

Download a phonebook demo from【Download Phonebook demo】, add and save
information refer to the demo content, choose the file what you want to uploaded from
【Upload Phonebook file】
You can download the phonebook file from【Download Phonebook(.csv)】

4.8 PIN Set

PIN Set will distribute one PIN Code to different extension user, if you selected PIN Set on
the Dial rule page in Outbound menu, the extension user who has the PIN code can dial
long distance call. Click【Pin Set】to show the dialog as below:

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Pic. 4.8-1 Add a PIN Set

 PIN Set Name Set the PIN Sets Name
 PIN List Enter a list of one or more PINs. One PIN per line.

4.9 Feature Codes

Click【 Feature Codes】to display the dialog as below, you can define relevant parameter.

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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 **+extension to
pickup the extension.

Blind Transfer

Allow unattended or blind transfers. It works like this: While
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

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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.

4.10 Phone Provisioning

When you need many IP Phone for using, please record the MAC, extension number, and
username of each phone according to the format (please take reference of the auto
provision script file model for details) , then, import the format file, once the phone is
connected to the local network, it will get the extension number and password
automatically.
There are two operation methods to fulfill this function, please see details as below:

Enable DHCP service

Click【System】->【Network Advanced】, enable DHCP Server in the dialog as below:

Method：

Click【 Extension】->【Creat New User】， select the relative IP Phone manufacture, and
input relative MAC in the part of Auto Provision, Save and Activate.

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This chapter will introduce you the status of record list, call logs, system info, register
status etc.

5.1 Recording List

Check the record list of defined extension or conference, you can delete the record list.
Click【Recording List】---【Extension】and【Conference】will be displayed as below:

Extension 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
dialog as below:

Call Logs Interface

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Note:

Duration in the call logs is not real charged duration, if you need billing, PSTN must 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
dialog as below:

5.4 System Info

Check OS version, firmware version and memory, etc from here.
Click【System Info】to display the dialog as below:

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This chapter will introduce you how to configure the system of TD IP PBX.

6.1 Network And Country

Configure WAN/ LAN IP, and tone zone.
Click【Network And Country】to display the dialog as below:

名称

说明

IP Assign

Static, DHCP are supported

HTTP Port

Set the http server port, default is 9999

Remote Administration

Enable/ Disable Access GUI through WAN port.

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Tone Zone Settings

Define the tone zone for home country or place.

6.2 TroubleShooting

You can ping other network device through TD IP PBX and track network route by
command "Traceroute" .
Click【TroubleShooting】to display the dialog as below:

6.3 Netword Advanced

DHCP Server Settings

TD100 Series IP PBX support DHCP , Click【Network Advanced】->【DHCP Server
Settings】 to show the following diagram:

DDNS Settings

After configure DDNS, you can visit by domain remotely.
Click【DDNS Settings】to display the dialog as below:

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VPN Settings:

A virtual private network (VPN) is a method of computer networking---typically using the
public internet---that allows users to privately share information between remote locations,
or between a remote location and a business' home network. A VPN can provide secure
information transport by authenticating users, and encrypting data to prevent unauthorized
persons from reading the information transmitted. The VPN can be used to send any kind
of network traffic securely. Series IP PBX support N2N and L2TP.

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Note:

1) DDNS supports the domain provided by Dyndns.org/ No-ip.com only.

2) VPN supports N2N/L2TP only.

6.4 Time Settings

Click【Time Settings】to display the diagram as below:

NTP

Manual Time Set

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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.

6.5 Management

Management

Click【Management】to display the diagram as below:

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 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"

Access Permit

Click【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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SIP Registered Allowed

Click【SIP Registered Allowed】---【Add Permitted IP】to define the allowed SIP user.
Input the permitted IP address---IP address and network restriction.eg: "192.168.1.77" or
"192.168.10.0/255.255.255.0"

In the following diagram, 192.168.1.100 is the allowed IP registered by SIP.

6.6 Data Storage

Upload the voicemail, call recording, conference, call logs, etc to the defined FTP server
for storage.
Click【Data Storage】to display the diagram as below:

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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

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.)

Upload Backup File

Click【Upload Backup File】to display the diagram as below:

6.8 Upgrade

 Click【WEB Upgrade】to upgrade as below

Choose the file to upload. If you enabled Restore Default Settings, the system will be
restored to default after upgrading:

 Click【TFTP Upgrade】to upgrade as below:

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Extract the downloaded firmware package which includes one TFTP server and one
upgrading file.

Run TFTP server, you will see the following interface:

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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This chapter will introduce you how to use TD IP PBX by example.

7.1 How to connect the TD100 IP PBX to the Internet

If your office access the public network through router, you can put the IP PBX behind the
router. You should connect the WAN port of the IP PBX to the LAN ports of the router, and
you can also connect HUB or Switch to the LAN port of the IP PBX to enable some PC or
IP Phone to access the public network..

7.2 How to combine two TD100 IP PBX in the same network

We start combining two IP PBX in the same network and then try to expand to different
network.
Below is the structure of how to combine two IP PBX in the same LAN:

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Register the TD100-A as an peer in TD100-B(via IAX2 trunk),so the extensions in
TD100-A can make calls to TD100-B’s extensions via this “special” trunk.
In above structure:

1. ZP302A registers to TD100-A as extension 601.

2. ZP302B registers to TD100-B as extension 801.

3. All the extensions under TD100-A are in the format 6XX.

4. All the extensions under TD100-B are in the format 8XX

5. Extensions under TD100-A can make calls to extension under TD100-B with format
8XX.

6. Extensions under TD100-B can make calls to extension underTD100-A with format
6XX.

Step 1: Set up a peer 699 in TD100-A

In the page Trunks Add a Trunk

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Peer Name: TD100B ;
Peer Username: 699 Account of this Peer
Password: 699 IAX2 Log on password
Advance Options: Select IAX protocol

Step 2: Set up an IAX trunk in TD100-B to connect to TD100-A via this TD100B Peer.

In the page Trunks--> Add a Trunk

Step 3: Set Dial Rule in TD100-B, all calls starting with 6 will be sent to TD100-A.
In the page: Outbound Routes --> Add a Dial Rule

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Step 4: Set the user 601 and Dial Plan in TD100-A.
In the page: Extensions  Dial Plan
Activate the change and apply the test:

1. Register an IP phone ZP302B to TD100-B with 801 extension.

2. Register an IP phone ZP302A to TD100-A with 601 extension.

3. 801 call 601. And you can see 601 will ring and you can pick up the call.

Above is the way to route TD100-B’s call to TD100-A,

Accordingly, if you want to call from TD100-A to TD100-B, continue as below:
Step 5: Set Dial Rule in TD100-A all calls starting with 8 will be sent to TD100-B.

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Step 6: Set the user 801 and Dial Plan in TD100-B

Activate the change and apply the test:
601 call 801, and 801 will ring and you can pick up the call.

7.3 How to combine two IPPBX in different network

The general environment for two TD100 in different locations is: two TD100 IP PBX are
both in the Internet and using the public IP.

The configuration is same as above guide(7.2 Combine two TD100 IP PBX in the same
network) , but use the public IP address as the "HOST" settings, set as below:
In the page Trunks of TD100-B--> Add a Trunk

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The general environment for two TD100 IP PBX in different location and one or both two
are behind router and using the private IP. So we need to make port forwarding in the
router and make TD100 IP PBX reach to each other.

Step 1: Set port forwarding in the router for TD100-A
For the TD100-A is behind the router, you need forward the IAX2 port in your router, so all
the packets received on the router WAN port (210.11.25.127:4569) will be forwarded to
the TD100-A (192.168.1.21:4569). Below is the setting page in a linksys router:

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Step 2: Set up the Provider Host in TD100-B

Set up the service provider and calling rule in TD100-B to make it register to TD100-A.
This method is almost the same as above, EXCEPT you need to use the 210.11.25.127
as the service provider instead of 192.168.1.21.

Step 3: Set port forwarding in the router for TD100-B

Use the same method as Step 1 to do port forwarding in router-B for TD100-B as above.

Setp4: Combine two TD100 and make calls
Accordingly, set the 601 users in TD100-A and 801 users in TD100-B, and build the
correct dial rules as above, you can make calls between two the TD100 IP PBX.

Note: You can also apply a DDNS to get one fixed domain for both TD100 IP PBX and

connect to each other rather than using the Port Forwarding in the router.

7.4 How to resolve problems about hearing on one side only

If your IP PBX is behind the Router, you should build an IP Address Map to resolve this
problem as below:
【Advance】----【Options】 ----【Global SIP Settings】 ---【NAT Support】

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 External IP Replace your external IP address as your public IP or domain
 External Host Replace your external IP address as your public IP or domain
 External Refresh Set time for refresh, default is 10
 Local Network Address Replace your local network address and mask

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Notice: The fee of your business account must be more than €50 when you use the

account first time.

1. Sign in with the business account on this page:

https://login.skype.com/account/login-form?intcmp=sign-in&return_url=https://secure.skyp
e.com/account/login

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2.When you have signed in, please click Skype Manager at the end of this page.

3.Please click the button Features.

4. Please click the Skype connect

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5. Create a SIP profile

Then you can create one sip account, you need pay for € 4.95 for one channel as
monthly rent and you need input the register information to our VoIP trunk blank, then you
can register to skype server. And you need assign money for outgoing calls, then you can
call out.

Note: Skype Channel belongs to VoIP channel, so any calls from Skype will be directed to the
same destination of VoIP.

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Then you can see the sip account information by clicking Authentications details.

<Finish, Thank You!>