ZX50 User Manual_en v32r12netcloud

7/22/2019 ZX50 User Manual_en v32r12netcloud

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ZX50 SERIES IPPBX USER MANUAL

Firmware Version 3.2Document Revision 1.2

ModelsZX50-A4ZX50-A8ZX50-B4ZX50-G4 - ZX50-AG42 - ZX50-AE41

The information contained in this document is subject to change at any time without prior notification. If you wish to

learn more about our product range then please visit our websitewww.zycoo.com.

Revised by NetcloudNovember 2012http://www.netcloudgroup.co.uk
http://www.zycoo.com/http://www.zycoo.com/http://www.zycoo.com/http://www.zycoo.com/


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ZYCOO all rights reserved V3.2- 1 -UK Version Copyright - Netcloud Ltd 2012

TABLE OF CONTENTS

Chapter1 - Important Safety Information ........................................................... - 4 -

Chapter2 - Brief Introduction .............................................................................. - 6 -

Brief introduction to the ZX Series IP PBX ..................................................................... - 6 -

Hardware Structure ....................................................................................................... - 8 -

Rear Panel .............................................................................................................. - 8 -

Front Panel .......................................................................................................... - 10 -

Hardware: ............................................................................................................ - 13 -

Environmental Requirements:............................................................................. - 13 -

Packing List .......................................................................................................... - 13 -

Chapter3 - Configuration .................................................................................. - 14 -

Default Settings ........................................................................................................... - 14 -

Logging into the System for the first time ................................................................... - 14 -

Configuring the ZX Series IP PBX ......................................................................... - 15 -

Basic Configuration .............................................................................................. - 17 -

Outbound Call .............................................................................................................. - 21 -

Trunks................................................................................................................. - 21 -

How to create a trunk:................................................................................................. - 21 -

Outbound Routes ................................................................................................ - 26 -

Inbound Control......................................................................................................... - 28 -

Inbound Routes ................................................................................................... - 28 -


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IVR...................................................................................................................... - 31 -

IVR Prompts ................................................................................................................. - 32 -

Ring Groups....................................................................................................... - 34 -

Blacklist ................................................................................................................ - 35 -

Time Based Rule............................................................................................. - 36 -

Chapter 4 - Advanced ....................................................................................... - 38 -

Options....................................................................................................................... - 38 -

Voicemail.................................................................................................................... - 42 -

Using your voicemail ........................................................................................... - 47 -

Conference........................................................................................................ - 47 -

Call Queue................................................................................................................. - 50 -

Music Settings........................................................................................................... - 53 -

DISA............................................................................................................................ - 54 -

Follow Me........................................................................................................... - 55 -

Paging And Intercom................................................................................................ - 56 -

Call Recording........................................................................................................... - 57 -

Phone Book............................................................................................................... - 58 -

Pin Set........................................................................................................................ - 60 -

Feature Codes........................................................................................................... - 61 -

Phone Provisioning................................................................................................... - 66 -

Provisioning using a third party DHCP Server ............................................................. - 68 -

Chapter 5 - Status .............................................................................................. - 70 -


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Recording List............................................................................................................ - 70 -

Call Logs.................................................................................................................... - 70 -

Register Status.......................................................................................................... - 71 -

System Info................................................................................................................ - 73 -

Chapter 6 - System ............................................................................................ - 75 -

Network & Country ..................................................................................................... - 75 -

TroubleShooting .......................................................................................................... - 76 -

Network Advanced ...................................................................................................... - 77 -

Time Settings ............................................................................................................... - 79 -

Management ............................................................................................................... - 80 -

Data Storage ................................................................................................................ - 82 -

Backup ......................................................................................................................... - 84 -

Upgrade ....................................................................................................................... - 85 -

Chapter 7 - Advanced Configuration ............................................................... - 88 -

Logging into the system through the Secure Shell (SSH) ............................................ - 88 -

Example Configurations ................................................................................... - 90 -

Setting up a VoIP Trunk ............................................................................................... - 90 -

Open Ports ........................................................................................................... - 94 -


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Chapter1 - Important Safety Information

Please read the following safety information before installing or using this IP PBX.

They are crucial for the safe and reliable operation of the device.

Please use the external power supply which is included in the package.

Alternative power supplies may cause damage to the device and adversely affect

performance or induce noise.

Before using the external power supply in the package, please check that the

power voltage is suitable for your local environment. Inaccurate power voltage

may cause fire and damage.

Please do not damage the power cord. If the 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 manner in which to disconnect the device from the power supply.

Do not drop, knock or shake the device. 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. This may result in fire or cause the

product to fail.

Avoid exposing the device to high temperatures, high humidity, or temperatures

below -10C. Ensure that the unit does not come into contact with liquids.

Do not attempt to open the casing of the device. Handling of the internal

components of the device by unqualified personnel may lead to damage and

could invalidate the product warranty. Consult your authorized dealer for

assistance. Failure to adhere to this procedure could result in damage to the unit,

or worse, may cause fire or personal injury.


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Do not use harsh chemicals, cleaning solvents, or strong detergents to clean the

device. Instead, wipe the device with a soft cloth that has been slightly

dampened in a mild soap and water solution.

During lightning storms, do not touch the power plug or phone line as it may

cause an electric shock.

Do not install this device in a poorly ventilated area.

Mishandling of electrical equipment could cause bodily injury. Before working on

any equipment, be aware of the hazard involved with electrical circuitry and

ensure that you are familiar with standard practices for preventing accidents.


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Chapter2 - Brief Introduction

Brief introduction to the ZX Series IP PBX

The ZX Series IP PBX not only provides traditional basic PBX features such as call

hold, call forwarding, call waiting, but additionally, provides a whole host of advanced

features such as visual operator, voicemail to email, music on hold, and auto

attendant etc.

It is an ideal solution for SMEs due to its ease of administration and on-goingmaintenance. This, combined with its powerful feature set makes it the perfect

solution for businesses needing to increase efficiency and raise the profile of their

company.

The ZX50 series IP PBX comprises of: ZX50-A4, ZX50-A8, ZX50-AG42, ZX50-G4,

ZX50-AE41, ZX50-B4 models. Features and structure is largely consistent across the

range with the main difference being the gateway interface. Please check the

following table for details:

Model FXS FXO GSM E1 BRI

ZX50-A4 * A4XX 04 40

ZX50-A8 * A8XX 08 80

ZX50-AG42

AG4204 4 2

AG4222 2 2 2

ZX50-G4 G4 4

ZX50-AE41* AE41XX

04 401

ZX50-B4 B4 4

Notice Models with (* ) - The first X denotes the number of FXS channels - Thesecond X denotes the number of FXO channels.


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Main Features

30 Concurrent calls/ Up to 100 registers

Video Calls

Phone Provisioning

Multiple Languages

DID (Direct Inward Dialling Number)

Supports SKYPE for SIP

Call Recording

Codecs: 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)

Web based configuration

Built-in SIP/IAX2 server

Built-in voice mail server


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System Backup and Restore

Echo Cancelation/VAD

Static and DHCP IP Addressing

PPPOE

VPN Client ( N2N and L2TP)

DDNS Client (Dyndns.org /No-ip.com)

NTP (Network Time Protocol)

Hardware Structure

The following images detail the front and rear views of all models within the ZX50

series.

Rear Panel

ZX50-A4

ZX50-A8


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ZX50-AG42

ZX50-G4

ZX50-AE41

ZX50-B4

Interface details

18 * Analogue Interface (ZX50-A4/A8/AG42/AE41)

1 x E1/PRI Interface (ZX50-AE41)

4 x BRI Interface (ZX50-B4)

4 x GSM Antennas (ZX50-G4)

2 x Network Interface (RJ45)


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1 x Power Interface (DC 12V 2A)

1 x Reset Button

Front Panel

ZX50-B4

ZX50-A8

ZX50-AG42


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ZX50-G4

ZX50-AE41

ZX50-A4


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Front Panel Details

Mark Function Status Description

PWR Power Status

On Power On

Off Power Off

SYS System Status

On System working

Off System Failed

WAN WAN Status

Wink Data exchanging

Off No Data exchanging

LAN LAN Status

Wink Data exchanging

Off No Data exchanging

G1~G4 GSM Channel StatusGreen GSM channel connected

Off Failed

SIM 1~SIM

4

SIM Card Status Wink/1s No access to network

SIM Card Status Wink/5s Access to network

*1-4Analogue Channel

Status

Green FXS channel connected

Red FXO channel connected

Off Failed

TE1-TE4 BRI Line Status Red BRI Line connected

Green Calling from BRI Line

ALARM E1 Line Status

Light E1/PRI Line connected

Off E1/ PRI Line not connected


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LINK E1 Line Status

Light E1/PRI Line connected

Off E1/PRI Line not connected

Hardware:

32bit embedded RISC DSP

1Gb On-board Nand Flash

128Mb On-board SDRAM

Environmental Requirements:

temperature: -10 C -45 C

Storage temperature: -30 C -65 C

humidity: 10-80% no dew

Power: AC 100~240V

Packing List

ZX50 IP PBX - 1 Unit

GSM Antennas (ZX50-AG42/G4)2 or 4 Units

Power Adapter - 1 Unit

CD (Quick Start Guide/ User Manual/ Pictures) - 1 Piece


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Chapter3 - Configuration

Default Settings

Default IP Address of WAN and LAN ports:WAN Port 192.168.1.100

LAN Port 192.168.10.100

Default Username and Password

Web GUI username Admin

Web GUI password Admin

SSH username Root

SSH password Myippbx

Dial the following sequence from a connected phone to determine the IP address of

the device:

**11 Plays back the IP Address of WAN port

**12 Plays back the IP Address of LAN port

Logging into the System for the first time

Logging into the system using the Web Interface

After connecting the IP PBX to your local network, launch a web browser on a

computer from with the same IP Address subnet range as the ZX Series IP PBX.

Enter the following IP address into a web browser - WAN port IP address

http://192.168.1.100:9999(this is the preferred interface to use) - LAN port IP

addresshttp://192.168.10.100:9999). The following page will be displayed:
http://192.168.1.100:9999/http://192.168.1.100:9999/http://192.168.1.100:9999/


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Enter the default username and password (default username is admin, password is

admin) and click login. Iflogin is successful then the home page will be displayed:

Configuring the ZX Series IP PBX

If the entered username and password are correct then the following page will be

displayed:

Note:

Please use Internet Explorer version 7.0 or higher and Firefox browsers.

When accessing the ZX50 for example through the WAN port, then the PC used

for this purpose must be in the same 192.168.1.XXX address range as the IP

PBX. If this is not the case then please change the address on your PC.

For security reasons, please modify the username and password after you login.

You can modify them in the System---Managementpage.

The device will automatically log out a user if no activity is detected for a period

of one minute. Please log back in if you still require access.


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Network Displays WAN/ LAN Port IP Address information

Storage Displays statistics on total and used storage

Channels Displays model dependent Channel information

Device Info Displays the model number and firmware version

Common Buttons

Certain buttons are common across the different configuration pages of the GUI.

Logout Log out of the system

Reboot Reboot the system

Factory Defaults Restore system to factory default

Activate Changes Activate any changes made

System Menu

Down the left hand side of the GUI is the System Menu. This contains the various

sections required to configure your IP PBX.

Home Page Displays summary information as


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detailed above

Basic Basic configuration of extensions, trunks

etc.

Inbound Control Configuration of inbound routes, IVR

and black list etc.

Advanced Advanced configuration settings

Status System status including register status

and call logs

System Configuration of system components

including Network

Basic Configuration

The extensions page is used to configure the phone extensions of the system. Phone

extensions can be added one at a time using the create new user button, or

alternatively, users can be added in bulk using the batch add users button

Extensions Tab

Search Search for an extension


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Show all Show all extensions

Name Extension name

Password Password for the extension. A secure password is generated

by default

Caller ID Caller's ID e.g. "801"

Outbound CID Overrides the caller id when dialling out through a trunk.

VM Password Voicemail password extension e.g. "1234".

E-mail The e-mail address for this user [email protected] is

used for voicemail to email if SMTP settings have been

configured.

Analogue Phone If this extension is attached to an analogue port on the system

then please select the port number here.

Dial Plan Please select the Dial Plan for this extension. Dial Plans are

defined under "Outbound Routes" and set the rules to control

the type of outbound calls that an extension can make.

Voicemail If enabled then voicemail is activated for this extension.

Can re-invite Sets up calls directly between the caller and receiver and

bypasses the IP PBX once connected. This method is known

to cause problems with certain hardware, such as the Cisco

ATA 186.

SIP Check this option if the User Extension or Phone is using SIP

or is a SIP device.

IAX2 Check this option if the User Extension 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
mailto:[email protected]:[email protected]:[email protected]:[email protected]


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Agent Check this option if the User or Phone is a Call Agent.

NAT Check this option if the User or Phone is located behind a

NAT (Network Address Translation) enabled gateway. This

setting is usually required if you are traversing through a

firewall

Pickup Group Select your pickup group.

Delete VMail Voicemail cannot be checked by phone if this option isselected. Messages will be sent by email only instead.

Note: You must configure an SMTP server for email

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 the ability to make a video call for this

extension

Permit IP Enter permitted IP address if you wish to add a further layer of

security to your system.

e.g. "192.168.1.77" or "192.168.10.0/255.255.255.0"

Auto Provision Zycoo IP PBX currently supports auto-provisioning of Zycoo,

Grandstream and Yealink IP Phones. Please select the phone

manufacturer and input the MAC address of the IP Phone.

If this feature is enabled then a configuration file will be written

to the /persistent/tftpboot directory.

You must be using either the systems inbuilt DHCP server or

a third party server that supports option 66.

Codecs

Configure

Configure codecs as required.


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Upload/Download Extensions Tab

If you wish to upload several extensions at the same time then the ZX Seriessupports the uploading of multiple extensions in either txt or csv format.

Download Extension Demo Please download the example file and modify using

the same format as detailed within the

demonstration file. Once complete, save the new fileand upload into the system.

Upload Extensions Upload a file containing your list of extensions.

Download Extensions (.csv) Download a copy of the extensions currently loaded

on the system.

Extensions Log Tab

Details information on the extensions uploaded as part of a bulk upload.

Note:

1) There are 30 default extensions created starting with "8". You can add or delete

extensions to meet your requirement.

2) Maximum permissible extensions for this device is 100.


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Outbound Call

Trunks

The trunks section allows you to create trunks to the PSTN (Public Switched

Telephone Network), a VoIP service provider, or another VoIP device or gateway.

How to create a trunk:

The following section details how to create a trunk for each possible category of

Analog/GSM. E1/T1 and Custom VoIP/Peer

1) Creating a trunk for an analogue line

ClickAdd a Dial Rule->Analog/GSM


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Provider Type Select Analog Trunk

Description Create a name for the trunk

Lines All Ports installed within the device will be displayed here.

Select the Ports that you wish to be used with the trunk. This

can be any number of the ports available for selection.

Prefix Input numbers that you would like prefixing to the Outbound

Caller ID


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2) Creating a trunk for E1/T1/GSM/PRI Line

E1/T1/PRI and GSM lines are configured in the exact same manner as the analogue

line example above.

3) Creating a VoIP Trunk

Creating a VoIP trunk allows you to connect the ZX Series to a VoIP service provider

or external VoIP device/gateway. You create a VoIP trunk as follows:

Add a Trunk->VoIP Trunk

Note for UK Analogue Lines :

Under the SystemAdvancedGlobal Analogue Setting

Set Caller ID Signalling to V23

Set CallerID start to polarity

Set FXO Mode to UK

If inbound Calls on analogue lines are quite then increase the inbound call volume under

the advanced properties of the trunk. Maximum value is 10 but initially try a value of 3


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Provider Type Select VoIP Trunk.

Description Provide a name for the trunk.

Protocol Choose the preferred protocol for this trunk, either SIP or

IAX2.

Dial Plan Choose a dial plan for the trunk. Dial plans provide control of

outbound calls and can be defined in the Outbound Routes

section.

Register Check if the service you are connecting to requires

registration. Do not check if the service does not require

registration.

Host Host Address of the Service Provider (either IP Address or

DNS Name)

Outbound proxy If the Service Provider offers an Outbound Proxy then the

address can be added here.


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Proxy Port The proxy port of the Service Providers Outbound Proxy can

be entered here

Prefix The prefix will be added to the start of the number for

outbound calls.

Without

Authentication

If you don't use authentication when connecting to the

Service Provider then check this option.

Username Username provided by Service Provider.

Password Password provided by Service Provider.

3) Creating a Peer Trunk

Peer trunks are generally created to make outbound calls to other IP-PBX systems

and gateways. For example, you would create a peer trunk to connect to another ZX

Series IP PBX to make calls between them.

Peer Name Defines the name of the Trunk

Protocol Choose protocol for this trunk, options are SIP or IAX2

Dial Plan Choose a dial plan for the trunk. Dial plans provide control of

outbound calls and can be defined in the Outbound Routes

section.


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Host Host Address of the Service Provider (either IP Address or

DNS Name)

NAT If checked, the trunk will be configured for NAT (Network

Address Translation).

Without

Authentication

If you don't use Authentication when connecting to the remote

IPPBX then check this option.

Username Username provided by the Remote IPPBX

Password Password provided by the Remote IPPBX

Once a trunk has been created then it will be displayed in the List of Trunksection.

Each trunk in the list has an options list which allows the user to edit the trunk, define

the codes to be used, or set advanced options.

Outbound RoutesOutbound routes allow you to control which extensions can make outbound calls and

by what route. The default setting on the ZX Series is that internal extensions cannot

make outbound calls over PSTN or VoIP trunks unless defined within a dial plan.

Please configure dial plans on this page:Basic->Outbound Routes


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To create an outbound rule you first click on add a dial rule This will display the

following popup screen.

Rule Name Defines a name for this dial rule

PIN Set If enabled, sets the PIN which you must input when you dial

out by this rule. The PIN is selected from the drop down box. If

you have not created any PIN numbers to select then you will

be prompted to do so.

Record in CDR If selected, CDR will show which pin the call was using.


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Place this call

through

Defines the trunk to place the call through

Failover Optionally choose a failover trunk for use if the primary trunk is

unavailable..

Dialling Rules Defines the number match pattern for dialling.

Define a custom

pattern

Allows you to create your own number pattern. Patterns start

with _ and can have any of the digits below:

N digit from 2 to 9

Z digit from 1 to 9

X digit from 0 to 9

. any number of additional digits

For example, if you wished to call a UK mobile number then

you could define a rule of _07XXXXXXXXX or _07.

Delete[ ]digits

prefix

The number of digits input here will be removed from the

beginning of the dialled number when passed to the trunk. E.g.

If you were to set delete digits to 2 then the number

4477956443 would pass as 77956443

Auto-add digit [ ] The digits will prefix the number e.g. If digits "0044" were to be

added, then dialling 1302638679 will actually send

00441302638679 to the trunk.

Inbound Control

Inbound RoutesInbound routes allow you to manage how calls originating externally are managed.

This is undertaken through theInbound Routesscreen.


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General Tab

This is where the default inbound calls for all inbound trunks are handled. From the

destination drop down box, select the extension, queue, group or IVR that you wish

the call to route to.

Analogue Channel DIDs Tab

If you want further control over the way you manage inbound Analogue calls then you

can achieve this through the Analogue Channel DID

SelectAdd Analog Channel

Channel From the Channel drop down box,

select from the installed channels

Associated Extension From the Associated Extension box,


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select the extension, queue, group or

IVR that you wish the call to route to.

VoIP Channel DIDs Tab

Certain VoIP service providers will pass the inbound call as a DID (sometimes

referred to as DDI) number and you can therefore handle how this call is routed.

To do this, Click the Add VoIP Channelbutton

DID Number Input your DID number here.




DODs Settings

DOD settings screen allows you to route inbound calls from inbound trunks to a

specified extension, call queue, conference or IVR.


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DOD Number This number is the caller's phone

number and can originate from an

analogue channel or VoIP/GSM/E1/T1

Line.




IVR

IVRs are designed to allow the ZX Series to present the inbound caller with a range

of options on how the call should be routed. The IVR will first play a pre-recorded

message to the caller and then wait for one of the pre-configured key events to

determine where to route the call to. An example of this would be press 1 for sales, 2

for support etc. Configuration is undertaken on the IVR page.

Name Input a name for the IVR.

Extension Input a number if you wish to call the IVR from an internal

extension e.g. 6000.

Please Select Selects the audio file to be played. These are configured


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on the IVR Prompts screen.

Repeat Loops Determines how many times the IVR should loop.

Dial other Extensions Allows the caller to dial other extensions besides the ones

listed in Key events.

Keypress' Events When the inbound caller presses a numeric value on their

phone, the corresponding Keypress Event action takes

effect.

Option are:

Disabled

Goto extension

Goto IVR

Goto Ringgroup

Goto IVR

Goto voicemail

Custom

Hang up

Play invalid

Other Values on the menu that you can set a conditional

response against are:

Tis the time out condition

Iis an invalid selection condition

IVR Prompts

The IVR prompts screen allows you to create a new IVR prompt, edit an existing

prompt, or delete an existing prompt.


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To record a new prompt, clickRecord a IVR Prompt. The screen below will

appear:

File Name Input a name for the IVR.

Format Define the format with which the prompt will be recorded.

GSM/WAV (16 bit) supported.

Extension The extension that will be used to record the prompt.

To listen to a prompt, locate the prompt within the menu and click on play. You will

then be presented with the following screen. Enter the extension on which you want

to listen to the prompt on and select confirm. The ZX Series will call the extension

and play the IVR.


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Upload Prompts Tab

Upload pre-recorded prompts from the Upload Prompts tab. Files must be one of the

following formats - wav (16bit/8000Hz/Single), GSM, Ulaw or alaw. The file size

must not exceed 15Mb.

Ring Groups

Ring Groups are a way to group a collection of extensions together for the purpose of

answering inbound calls as a collective group. The ring group can be configured to

either ring all members of the group at the same time, or alternatively, ring each

phone in a defined order.

ClickNew Ring Groupto start configuring a group.


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Name Input a name for the Ring Group.

Strategy Select ring strategy, either "Ring all" or "Ring in order"

Ring Group Members Select ring group members in available extensions

click to add

If not answered If the call isnt answered then you can choose to forward

the call to an extension, Voicemail, Ring Group, IVR or

Hang up.

BlacklistThe blacklist screen allows you to define inbound numbers that you would like to

block.

ClickNew Blacklistto display the following dialogue box:


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Blacklist Number Input the number that you wish to be blocked.

It is also possible to add numbers to the blacklist direct from your telephone handset.

The following diagram illustrates how this is achieved.

Reference Parameters and Explanation of Blacklist:

*30 *30 plus the extension number will add the number to the

blacklist.

*31 *31 plus blacklist number will delete the number from theblacklist.

Time Based Rule

Time based rules allow you to set different rules to control inbound calls at different

times and on different days. This feature is used for instance when you want inbound

calls during out of hours and holidays routing differently than during a normal working

day.

To create a new time based rule:


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Time Based Rule---New Time Rule

New Time Rule:

Rule Name Defines a name for the new rule

Time & Date

Conditions

Defines the time that the rule should take effect.

Destination Defines where the inbound rule should be routed to. This

can be an extension, ring group or an IVR


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Chapter 4 - Advanced

Options

The options screen is used to set global default settings on the ZX Series IP PBX.

ClickOptionsto display the General tab of the options screen.

Local Extensions Defines the number of digits contained within an

extension number when creating a new extension.

No restriction (varying) is an option.

Operator Extension Select an extension to use as the Operator

Extension.

Global Ring Time Set(s) Set the default ringing time for extensions.

Enable Transfer Enable the transfer feature key.

Enable Music On Ringback Enable music on ringback.

Allow multiple extensions to

be assigned to one analogue

Allow multiple extensions to be assigned to one

analogue phone.


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phone

Allow extensions to be Alpha

Numeric (SIP/IAX users)

If extensions have Alpha characters within the

extension then outside lines will be unable to call it.

The extension will however be able to make

outbound calls.

SIP If selected, new extensions will be created with SIP

protocol enabled.

IAX2 If selected, new extensions will be created with

IAX2 protocol enabled.

Agent If selected, new extensions will be created as

Agents.

NAT If selected, new extensions will have NAT enabled.

VM Password This will be the default password for Voicemail on

new extensions e.g.: "1234" if selected.

Delete VMessage Voicemail will be disabled for new extensions if

selected.

Global Analogue Settings Tab

Global settings for analogue trunks are defined on this screen.


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Caller ID Detection If checked then Caller ID will be detected on

analogue trunks.

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 caller ID under DTMF mode.(e.g.:

Denmark, Sweden and Netherlands etc.);

V23-- used in the UK;

V23-Japan-- used in Japan

Caller ID Start This option allows you to define how the Caller ID is

started.

RingCaller ID is initiated when a ring is received.


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PolarityCaller ID is initiated when a polarity

reversal is started. (This setting applies to the UK)

CID Buffer Length Default CID Buffer Length

FXO Mode This is the country specific opermode. It is

important in the UK to set this to UK as it reduces

the number of rings before the IP PBX answers the

call.

Relax DTMF If you have trouble with DTMF detection then you

can relax the DTMF detection.

Echo Cancel Enable/Disable the Echo Cancel feature.

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. In this way it enables the PBX to get abetter understanding 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. Important if you are experiencing problems

with calls not hanging up on the analogue trunks.

Busy Count If Busy Detection is enabled, it is also possible tospecify 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. Be mindful that

the higher the number, the more time that will be

required 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.


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Global SIP Settings Tab

The majority of the settings on the Global SIP settings tab have been pre-set by

Zycoo engineers for optimal performance and you should seek their guidance before

making any changes.

However, some setting should be set. If you are accessing VoIP services through a

firewall then it is beneficial to set your External IP and local Network Address within

the NAT support section.

A new security feature has been added which provides some protection against

hacking. The default behaviour is that if 10 failed sip registration (because of

incorrect password) occur within 10 minutes then the offending IP Address will be

blocked for 10 minutes.

E1/T1 Trunk Options

EI / TI trunk options are used for E1/T1/PRI lines.

Voicemail

Configuration settings for voicemail are defined within the voicemail section.

Clicking on voicemail opens the General tab as detailed below:


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Extension for checking

messages

The extension number that users must call in order to

access their voicemail accounts. The default number is

"600".

Max greeting(seconds) This option will set a maximum time for a greetingmessage.

Direct to Voicemail Define this option to go to voicemail box directly.

Therefore when checking your voicemail you will only

need to enter your password and not your extension

number.

Dial "0" for Operator Callers entering the voicemail application may leave to

speak to the Operator by dialling "0".


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Message Format Choose the format that voicemail messages will be

stored in.

Maximum Messages Choose the maximum number of messages to store per

mailbox.

Maximum message time

(min)

Choose the maximum duration of a voicemail message.

Message recording will be stopped when this threshold

has been exceeded.

Minimum message time

(s)

Choose the minimum duration of a voicemail message.

Messages that do not meet this minimum requirement

will be deleted automatically.

Say message Caller-ID Choose this option to play the Caller's ID before

voicemail message is played.

Say message duration Choose this option to play the duration of the 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 By selecting this option, the caller leaving the voicemail

can review their recorded message before it is

submitted.

SMTP Settings Tab

The Zycoo ZX Series supports voicemail to email services. To enable this feature,

you must configure an external SMTP server to relay the message. This is configured

under the SMTP Settings Tab.


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SMTP server Input the address of the external SMTP server

e.g.: mail.yourcompany.com

Port The port number which the SMTP server listens on

(generally port 25). If SSL is encrypted then please use

port 465 instead.

SSL/TSL Enable use SSL/TLS to send secure messages to

server. (some servers such as google mail require this)

Enable SMTP

Authentication

If your SMTP server requires authentication then 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 Tab

The email settings tab allows you to customise the appearance of the email message

and determine if the email will be for notification purposes only or will contain an

attachment of the recording.


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Using your voicemail

Leaving a message:

Listening to a message

Conference

Conference rooms allow you to set up conference calls for both internal and external

callers. Click on the Conference screen to display the Conference (Default) tab.

Note:

To use this feature,you must input a valid email address in "extension settings"


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Extension The number that users will call in order to access the

conference room The default number is "900".

Guest Password Guest code for entering the conference room.

Administrator

Password

Administrator code for entering the conference room.

Conference DialPlan Dialplan available to participants within the conference

room.

Play hold music for

first caller

If checked, Asterisk will play music on hold to the first

user in a conference. This will continue until another user

has joined the same conference.


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Enable caller menu Selecting this option allows a user to access the

Conference Bridge menu by pressing the * key on their

dial-pad.

Announce callers Selecting this option announces participants entering the

conference.

Record conference If selected, the conference will be recorded.

Quiet Mode If selected, all users entering this conference will bemarked as quiet and will only be able to listen.

Leader Wait Wait until the conference leader (admin user) arrives

before starting the conference.

Two further conference rooms can be defined on tab Conference 2 and tab

Conference 3.

An example of how to use the conferencing system is detailed below:

To enter a conference room:

While in the conference, an administrator can add a new participant (extension user

or external number) into the conference.

Add new participant:


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Call QueueCall Queues are a way of managing inbound calls and ensure that a call is routed to

an available agent. If no agent is available, then the caller will remain in a queue until

an agent becomes available.

Agents are configured in the Extensions screen.

Enter the Call Queue screen:

Queue Number This option defines the extension number you need to

dial to access the Queue.

Queue Name This option defines a name for the Queue, e.g. "Sales"

Ring Strategy Ring-All -- Calls all available agents until one answers

(default).

Round Robin -- Take turns calling each available Agent.


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Least Recent -- Calls the Agent which has the longest

period of time since their last call.

Fewest Calls -- Call 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 are defined as Agent will be shown

here. Selected agent will be a member of the current

Queue.

Agent TimeOut(s) This option defines the time in seconds that an Agents

phone rings before the next Agent is called, e.g. "15"

Auto Pause Pause an Agent if they fail to answer a call.

Wrap-Up-Time(s) After a successful call, this defines how many seconds are

required to wait before sending another call to a potentially

free agent (Default is 0, which means No Delay).


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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 conjunction with the "Join

Empty" option.

Auto Fill Defining this option forces calls in the Queue, where

multiple calls are queuing, to be pushed to Agents

simultaneously. Therefore, instead of completing a single

call to an Agent at a time, the Queue will complete as

many calls simultaneously to 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) Defines how often to announce queue position and

estimated hold-time to callers in the queue (0 to Disable

Announcements).

Announce Hold Time Determine whether to include estimated hold time in

position announcements? Either yes, no, or only once.

Hold time will not be announced if


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Prompt

Music Settings

The Music settings page provides management for music on hold, music on ringback

and music on call queue.

Navigate to the music management tab to upload and delete your music files

Directory Load music into the chosen directory.

Files Displays the music displayed within a directory. You

can also delete the file if required.

Select Music Directory Select directory where the uploaded music file will be

saved.

Please choose file to

upload

Input music file name which you want to upload.

(GSM/ WAV format, If WAV, then must conform to

PCM 16 bits, 8000HZ format)

Once music has been uploaded, it can be selected from the music settings tab.


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DISA

DISA allows you call into your Zycoo ZX Series IP PBX from an external trunk. You

will be prompted to enter a PIN to gain access to the system. Once access is

granted, you can make calls from the system as per the rules as defined within your

dial plan.

Select New DISA


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Name Input a name for the DISA

PIN Set a PIN which the user will be prompted for.

Response Timeout(s) The maximum amount of time the system will wait

before hanging up the user if a valid number has not

been entered. Default is10 seconds.

Digit Timeout(s) The maximum amount of time permitted between

digits when the user is typing in an extension. Defaultis 5 seconds.

Extension for this DISA

(Optional)

If you want this DISA to be accessible by dialling an

extension, you can define an extension number for

this DISA.

Select DialPlan Select the DialPlan that will be inherited.

Follow Me

The follow me feature allows you to set conditional forwarding from your extension.

This is a useful feature to set when you are out of the office and would like your calls

forwarding to a mobile device.

Select New Follow Me


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Extension Choose an extension

Status

Always All incoming calls will be forwarded

Busy Forward when extension is busy

No answer Forward when extension does not answer

Ring lasting for(s) Default is 20 seconds. You can adjust this

value as required.

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 for when forwarding the

call to an external number.

Set External Number Set external number, for instance a Mobile

number.

Paging And Intercom

Paging and intercom settings page allows you to create paging groups. Paging is

typically used to communicate one way and mainly for announcements only.

The handset needs to support paging for this feature to work.


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Paging

Extension

The number to be called to page this group.

Description Provide a descriptive title for the Page Group.

Paging Group

Members

Selected device(s) in this page

Device List Devices that can be selected as paging group members

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".

Call Recording

Call Recording is used for recording defined extensions.

Recordings are stored under the StatusRecording List


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Extension Define an extension to be recorded.

Call Recording Time Defines the monitoring rules. This can either be set to record

all calls, or alternatively, a scheduled recording time can be

set.

Inbound Record Check to record inbound calls

Outbound Record Check to record outbound calls

Phone Book

The Phone Book page allows you to define a central phone book to identify inbound

calls. If the incoming call matches the number in the phone book, the incoming call

will display the name of the matched number.

Phone Book Tab


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Search Input contact name to search.

Show All Lists all contacts

To manual add a contact; click the create contact button

Name Add contact's name.

Number Add contact's number. International phone numbers are

supported.

Upload/Download Phonebook Tab

The ZX Series supports batch uploading of phone numbers. This is undertaken on

the Upload/Download Phonebook Tab.


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Upload Phonebook

File

Uploads a supported file containing names and numbers for

the phone book.

Download

Phonebook Demo

Right click to download an example phone book file.

Manually edit the file, save and upload to the IP PBX when

complete.

Download

Phonebook

Download the phone book stored on the IP PBX in CSV

format.

Pin SetPIN Sets are a way of securing an outbound trunk from unauthorised access. Before

you can apply a PIN set to a trunk, it must first be created within this page.


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PIN Set Name Defines a name for PIN set

PIN List Enter a list of one or more PIN values (one per line).

Feature Codes

Feature codes allow you access to pbx features dialled directly from your handset.

These are defined within the feature codes page.


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Extension to

Dial for ParkingCalls

Sets the Call Parking number.

What

extensions to

park calls on

Defines the extension where calls are parked, e.g. (701-

720)

How many

seconds a call

can be parked

Defines the maximum parking time in seconds. If this

time is exceeded then the system will call the original

extension that parked the call again.


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for

CallPickup

Pickup

Extension

Set key sequence to pick up call.

Pickup

Specified

Extension

Set key sequence to Pickup Specified Extension, e.g.

dial *7 plus the extension to pickup specified extension.

Blind Transfer Defines unattended or blind transfers key sequence.

Blind transfers operate as follows:

While on a conversation with A, dial the blind transfer key

sequence. The system prompts "Transfer" then presents

a dial tone and places A on hold.

Dial the transfereesnumber (B's number) and A is put

through to B immediately. Your line will be cut off.

The caller ID that is displayed to B is exactly the same as

the caller ID presented to you.

Attended

Transfer

Defines the attended transfer or supervised transfer.

Attended transfers operate as follows:

While on a conversation with A, dial the Attended

Transfer key sequence. The system prompts "Transfer"

then presents you with a dial tone and places A on hold.

Dial the transfereesnumber (B's number) and talk with B

to introduce the call. T

You can then hang up the call and A will be connected

with the B.

In circumstances where B does not want to receive the

call, they can simply hang up and original call from A will


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revert back to you.

Disconnect Call Disconnect the current transfer call (for Attended

transfer).

Timeout for

answer on

attended

transfer

Set the answer timeout value.

Blacklist

Blacklist a

number

Add a black list number.

Remove a

number from

the black list

Remove a black list number.

ConferenceRooms

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 dial-tone to enter the number of the

participant that would like to invite.

After the call has been established and you speak to the

invited participant, you can press ** to direct them to the

conference, or *# to hang up the current call and return

to the conference yourself.

CreateConference

While speaking to a party, if you press *0 then you andthe callee are immediately transferred to a conference

room.

Return to

conference with

participant


0 when he/she is in the conference. When you press 0,

you will get a dial-tone to enter the number of the





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Return to

conference

without

participant


0 when he/she is in the conference. When you press 0,

you will get a dial-tone to enter the number of the






Queues

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; each call is preceded by a warning tone. Calls

are ended by pressing the "*" key.

Agent Callback

Login Extension

Extension to be dialled for Agents to Login to a specific

Queue.

Same function as Agent Login Extension, except that 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 cannot

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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Phone Provisioning

In situations where you are connecting many IP phone extensions to the ZX Series

IP PBX, a quick and efficient way to configure the handsets is to use the phone

provisioning feature of the system.

Currently, the supported handsets for this feature are Zycoo, Grandstream and

Yealink. You will need to record the MAC address of the phone to input into the

system.

The ZX Series supports two methods for autoprovisioning of IP Handsets and

these are detailed below:

Enable DHCP service

DHCP services are required for the phone provisioning feature. You can configure

DHCP to run on the ZX Series system.

DHCP settings can be found under System->Network Advanced

Method 1

For method 1, you configure phone provisioning settings on the user extension.

ClickExtension->Create New User


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Manufacturer Select the manufacturer of the handset. Current supported

handsets are Zycoo, Grandstream and Yealink

MAC Enter the MAC address of the handset.

Method 2

Method two is to use a script file to upload to the system. This method is particularly

useful when you have many phones to provision.

Navigate to the phone provisioning page.

First download a demonstration file from the Download Phone Provisioning demo

section and amend as necessary. Save a copy of the file.

Then on the Upload Phone Provisioning section:


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Delete All Extenion Deletes all extensions prior to phone provisioning.

Manufacturer Select the manufacturer of the handset. Current supported

handsets are Zycoo, Grandstream and Yealink

MAC Enter the MAC address of the handset.

Please choose file

to upload

Upload your file containing the extension details.

Provisioning using a third party DHCP Server

You can use an existing DHCP Server if you wish. The DHCP server must support Option 66

though as this is required to update the phone on the location of its configuration file.

The following example uses Microsoft DHCP.

First, although it will not be used, you must set the Zycoo DHCP server to reflect the network

interface you are using (WAN or LAN) and set the scope within this range. Do not enable the

service though. The following diagram shows how the service needs to configured for the

WAN interface.

Note:

For troubleshooting purposes it is worth noting that provisioning configuration files are

stored in the /persistent/tftpboot folder.


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On the Microsoft DHCP Server, Create an option 66 scope option and configure it as the ip

address of your Zycoo IP PBX.

You can now use method 1 or 2 to autoprovision your phones.


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Chapter 5 - Status

This following chapter will introduce you to the status screens, which include details

of recordings lists, call logs, system information and register status,

Recording List

When recording has been enabled for an extension or a conference, the recorded

files can be searched and managed through the Recordings List screen.

ExtensionTab

Select the extension and the date range and press go.

Conference List Tab

Recordings for Conference Rooms are searched and managed in the same way as

detailed above for extension.

Call Logs

Call logs are displayed from within the Call Logs screen. Searches can be conducted

over a specified time period and can be further filtered based on Caller ID,

Destination or Account Code.


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Call Logs Interface

Register Status

The register status screens presents a view of registration status and availability of

devices and trunks connected to the IP PBX. This functionality is particularly usefulwhen diagnosing connectivity issues.

Sip User Status Tab

Displays the status of all user extensions enabled for SIP use.

Note:

Duration in the call logs is not a real charged duration. If billing information is

required then the PSTN must support polarity reversal function. Also, you must

configure relevant parameters of polarity reversal in the trunk configuration for

the IP PBX.


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IAX2 Users Status

Displays the status of all user extensions enabled for IAX / IAX2 use.

SIP Trunk Status Tab

Displays status of trunks configured for use with SIP.


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IAX2 Trunk Status Tab

Displays status of trunks configured for use with IAX2.

System Info

System Info provides a summary of the software, status and performance of the ZX

series IP PBX

System Info Tab

Provides summary information including operating system and firmware versions,

Server uptime statistics and Hostname.

Resources Tab

Performance and capacity information is displayed within the Resources Tab.


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Chapter 6 - System

The following chapter details the System section of the ZX Series GUI.

Network & Country

Configure various Network related settings on the Network and Country page.

WAN Port Setup Set the WAN port of the system. IP Assign can be set to

Static or DHCP. If static is selected then complete the

relevant address information.

HTTP Port Defines the http server port for the ZX Series GUI. The

default is 9999

Remote Administration Enable/ Disable access to the GUI from the WAN port.

LAN Port Setup Set the LAN port for the system (if required).

Tone Zone Settings Define the telephone tone zone for your home country e.g.

UK


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TroubleShooting

The troubleshooting screen provides some basic diagnostics to help you should you

experience any connectivity issues.

Ping Tab

Ping Enter IP or DNS address to Ping.

Packets Enter the number of packets to send.


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Traceroute Tab

Traceroute Enter IP or DNS address whose route you wish to trace.

Network Advanced

Advanced network settings are configured here:

DHCP Server Settings Tab

If you wish to utilise the ZX Series IP PBX as a DHCP server then this is achieved

through the DHCP Server Settings tab.


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DHCP Service Select to enable the DNS service.

Interface Select the interface on which DNS will be enabled.

Start IP Start IP address of the lease range.

End IP End IP Address of the lease range.

Subnet Mask Subnet mask to issue to DHCP client devices

Gateway Default gateway address to issue to DHCP Client devices.

Primary DNS DNS server address to issue to DHCP client devices.

Lease Time The maximum lease time for the issued address.

DDNS Settings Tab

DDNS is used in situations where you dont have an internet facing static IP address

but you still need to know what your external address is at all times. By subscribing to

a service like dyndns.org, in combination with a supported device such as the ZX

Series IP PBX, you can use a DNS name to identify yourself on the internet and this

name will keep an updated record of you current IP Address.

DDNS Enable Enable the DDNS Service.

DDNS Server Select the DDNS Service. Supported services are

dyndns.org and www.no-ip.com.


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Username Username for the service.

Password Password for the service.

Domain Your domain name on the service.

Update Time (s) The frequency with which the device will update the

service.

VPN Settings Tab

A virtual private network (VPN) is a method of providing a secure network connection

between two devices or locations. Information can then be passed between the two

in a secure manner. The ZX Series IP PBX supports two types of VPN connections,

N2N and L2TP.

Time Settings

Set time settings for the device.


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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 which

may be local or remote. The default server is set to

pool.ntp.org. Please note that the system needs a working

connection to the named NTP server for this feature to work.

Time Zone Select your local time zone.

Synchronize with

current PC time

Click the button to synchronize the PBX time with the current

PC time.

Management

Management Tab


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Change

Password

Used to change the ZX Series password and username if

required.

Time Zone Sets the voice language for the system.

Access Permit Tab

Deny any IPattempting to

Select if you wish to further secure your system by restrictingGUI access to only IP Addresses entered. This will prevent


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access GUI

except those

listed below

access from any other IP Address,

SIP Registered Allowed Tab

For security purposes when managing SIP registrations, you can list a trusted IP

address or range of addresses that may access the ZX Series IP PBX.

Add Permitted IP Add a valid IP Address or address range of the SIP device

authorised to connect e.g. "192.168.1.77" or

"192.168.10.0/255.255.255.0"

In the following diagram, 192.168.1.100 is authorised for SIP registration.

Data Storage

The ZX Series can be configured to Upload Voicemail, Conference room recording,

Call recordings and Call logs to a defined FTP Server automatically when flash

storage is over 40%. Once uploaded, history files will be removed from the system

automatically.

Data Storage Tab


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Enable

Uploading

Enable periodical FTP uploading.

Server

Address

Set the FTP Server address (IP address or Domain).

User Name FTP account name.

Password FTP account password.

Directory Define a directory on the FTP server to store data.

Data Storage Log Tab

Displays information on the success or failure of data uploads.


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Backup

Backup Tab

Backup and restore the systems configuration from this screen.

Click Take a backup and the following pop up will be displayed.

File Name Enter a name for your backup.

Existing backup will be displayed as below:

Restore Restore your selected backup file to system.

Delete Delete your selected backup file.


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Download your selected backup file to your PC. (Note: Please

don't change the backup file name.)

Upload Backup File Tab

ClickUpload Backup Fileto display the diagram as below:

Please choose

file to upload

Select the backup file to restore.

UpgradeThe ZX Series supports upgrade via Web interface or TFTP.

Web Upgrade

Restore Defaults

Settings

Select to reset the configuration back to default as part of the

upgrade process.

Please choose

file to upload

Select the firmware file from either a local or network drive.


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TFTP Upgrade

Enter the

package Name

Enter the firmware filename.

TFTP Server IP

Address

Enter the address of your TFTP Server.

Restore Default

Settings

Select to reset the configuration back to default as part of the

upgrade process.

You will need a TFTP server in order to upgrade by this method. One such free

product is tftpd32. An example upgrade using this product is detailed below:

Run the TFTP server and the following interface will be displayed.


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Configure the Upgrade System Package as follows:

Click the Update button to complete the process. The system will rebootautomatically.


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Chapter 7 - Advanced Configuration

Logging into the system through the Secure Shell (SSH)

SSH can also be used to log into the ZX Series IP PBX and is generally used for advanced

configuration and for troubleshooting connectivity. You will require a utility tool such as

Putty to start a session. Enter the IP address of the device, select SSH and select open.

You will then need to log in using the default userroot and passwordmyippbx


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Once logged in you will see the following screen.


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Example Configurations

Setting up a VoIP Trunk

The following example details how to set up VoIP trunk to a service provider. In this

example we have used an IPChitChat SIP Trunk that has been configured to route to

a DDI.

First, ensure you have set your external IP Address and Local Network Address

under the NAT Support section at AdvancedOptionsGlobal SIP Settings. This isimportant as the registration needs to display your internet facing address to the

IPChitChat service and not your local network address.

An example of a local network address is: 192.168.1.0/255.255.255.0

Next go to BasicTrunks and Add a Trunk.

Settings for the IPChitChat Service are:


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Click Save

Now check the status of the trunk by going to - StatusRegister Status. The SIP

User Status tab shown below will give you an indication that communications and

DNS are working OK between the Zycoo unit and IPChitChat service. In this

example, the DNS name sip.ipchitchat.co.uk has been resolved to 217.14.138.127

and status is showing as OK. This indicates that all is well from a networking

perspective.

Next, move to the SIP Trunks Status tab and check if the trunk is registered.


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In this case, the trunk is showing as registered and is therefore working OK. We now

need to create an inbound rule to route any calls originating from this trunk. Because

we have set the Trunk on the IPChitChat service to route to a DDI then we need tomake the Zycoo unit aware of what DID number to expect. In this example we have

used the number 01302638888.

Go to Inbound Control - Inbound Routes. Select the VoIP/E1/T1 DIDs tab and click

the Add button.

Then enter the DID number 01302638888 and select the extension /group /IVR etc

where you wish to route the call to.

Finally, you need to create an outbound rule to make outgoing calls over the trunk.

To achieve this go to BasicOutbound Routes. Dialplan1 is displayed.

You now need to add one or more Dialling rules into the plan. Dialling rules define

how the system will route outbound calls based on the key sequence that you press.

A simple dialling rule is shown below which defines that calls starting with 0 and

having more than 6 characters will be routed through the IPChitChat trunk.


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Please note that this Dialplan must be set within an extensions settings for it to work.


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Open Ports

You will normally be separated from your VoIP service provider by a firewall and

therefore you will need to port forward the required ports for the service.

Documents

ZX50 User Manual_en v32r12netcloud