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ZX50 SERIES IPPBX USER MANUAL Firmware Version 3.2 Document Revision 1.

2
Models ZX50-A4 ZX50-A8 ZX50-B4 ZX50-G4 - ZX50-AG42 - ZX50-AE41

Revised by Netcloud November 2012 http://www.netcloudgroup.co.uk

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 website www.zycoo.com.

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 -1ZYCOO all rights reserved V3.2

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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 -2ZYCOO all rights reserved V3.2

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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-going maintenance. 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 ZX50-A4
*

FXS 04

FXO 40

GSM

E1

BRI

A4XX

ZX50-A8

A8XX AG4204

08

80 4 2 2 4

ZX50-AG42 AG4222 ZX50-G4 ZX50-AE41 ZX50-B4


*

G4 AE41XX 04 40

B4

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

1 8 * 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 On PWR Power Status Off On SYS System Status Off Wink WAN WAN Status Off Wink LAN LAN Status Off Green G1~G4 GSM Channel Status Off SIM 1~SIM 4 SIM Card Status SIM Card Status Wink/1s Wink/5s Green *1-4 Analogue Channel Status Red Off TE1-TE4 BRI Line Status Red Green Light ALARM E1 Line Status Off E1/ PRI Line not connected Failed No access to network Access to network FXS channel connected FXO channel connected Failed BRI Line connected Calling from BRI Line E1/PRI Line connected No Data exchanging GSM channel connected No Data exchanging Data exchanging System Failed Data exchanging Power Off System working Description Power On

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Light LINK E1 Line Status Off

E1/PRI Line connected 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 LAN Port Default Username and Password Web GUI username Web GUI password SSH username SSH password Admin Admin Root Myippbx 192.168.1.100 192.168.10.100

Dial the following sequence from a connected phone to determine the IP address of the device: **11 **12 Plays back the IP Address of WAN port 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 address http://192.168.10.100:9999). The following page will be displayed:

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Enter the default username and password (default username is admin, password is admin) and click login. If login is successful then the home page will be displayed:

Note: Please use Internet Explorer version 7.0 or higher and Firefox browsers. Configuring the ZX Series IP PBX When accessing the ZX50 for example through the WAN port, then the PC used If the entered username and password are correct then the following page will be for this purpose must be in the same 192.168.1.XXX address range as the IP displayed: 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---Management page. 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 Storage Channels Device Info Common Buttons

Displays WAN/ LAN Port IP Address information Displays statistics on total and used storage Displays model dependent Channel information Displays the model number and firmware version

Certain buttons are common across the different configuration pages of the GUI. Logout Reboot Factory Defaults Activate Changes 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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Log out of the system Reboot the system Restore system to factory default Activate any changes made

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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 Status Advanced configuration settings 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 Name Password

Show all extensions Extension name Password for the extension. A secure password is generated by default

Caller ID Outbound CID VM Password E-mail

Caller's ID e.g. "801" Overrides the caller id when dialling out through a trunk. Voicemail password extension e.g. "1234". The e-mail address for this user e.g. Tom@gmail.com This 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 Can re-invite

If enabled then voicemail is activated for this extension. 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

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Agent NAT

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

Select your pickup group. Voicemail cannot be checked by phone if this option is selected. 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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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.

Upload/Download Extensions Tab If you wish to upload several extensions at the same time then the ZX Series supports 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 file and upload into the system.

Upload Extensions Download Extensions (.csv)

Upload a file containing your list of extensions. 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.
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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 Click Add a Dial Rule->Analog/GSM

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Provider Type Description Lines

Select Analog Trunk Create a name for the trunk 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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Note for UK Analogue Lines : Under the System Advanced Global 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

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

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Provider Type Description Protocol

Select VoIP Trunk. Provide a name for the trunk. 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 Username Password

If you don't use authentication when connecting to the Service Provider then check this option. Username provided by Service Provider. 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 Protocol Dial Plan

Defines the name of the Trunk Choose protocol for this trunk, options are SIP or IAX2 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 Username Password

If you don't use Authentication when connecting to the remote IPPBX then check this option. Username provided by the Remote IPPBX Password provided by the Remote IPPBX

Once a trunk has been created then it will be displayed in the List of Trunk section. 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 Routes Outbound 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 PIN Set

Defines a name for this dial rule 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 Failover

Defines the trunk to place the call through

Optionally choose a failover trunk for use if the primary trunk is unavailable..

Dialling Rules Define a custom pattern

Defines the number match pattern for dialling. Allows you to create your own number pattern. Patterns start with _ and can have any of the digits below: N Z digit from 2 to 9 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 Routes Inbound routes allow you to manage how calls originating externally are managed. This is undertaken through the Inbound 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 Select Add Analog Channel

Channel

From the Channel drop down box, select from the installed channels

Associated Extension
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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 Associated Extension

Input your DID number here. From the Associated Extension box, select the extension, queue, group or IVR that you wish the call to route to.

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.

Associated Extension

From the Associated Extension box, select the extension, queue, group or IVR that you wish the call to route to.

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 Extension

Input a name for the IVR. 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 Dial other Extensions Determines how many times the IVR should loop. 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: T is the time out condition I is 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 Format

Input a name for the IVR. 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 Strategy Ring Group Members

Input a name for the Ring Group. Select ring strategy, either "Ring all" or "Ring in order" 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.

Blacklist The 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 the blacklist.

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 Time & Date Conditions Destination Defines where the inbound rule should be routed to. This can be an extension, ring group or an IVR Defines a name for the new rule Defines the time that the rule should take effect.

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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) Enable Transfer Enable Music On Ringback Allow multiple extensions to be assigned to one analogue

Set the default ringing time for extensions. Enable the transfer feature key. Enable music on ringback. 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 VM Password If selected, new extensions will have NAT enabled. 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 FXO Mode Default CID Buffer Length 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 Echo Training Enable/Disable the Echo Cancel feature. 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 a better 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 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. 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 greeting message.

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 Password Email Settings Tab

Input username of your email box. Input password of your email box.

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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Attach recordings to email Sender Name

This option determines whether or not voicemails are sent to the Users e-mail address as an attachment. Displays the senders name when you receive a voicemail.

From Subject Message Template Variables

Sender's email address Subject of the mail The message pattern The variables defined within the diagram above can be used within the email template.

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Using your voicemail Leaving a message:

Listening to a message

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

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.

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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 Administrator Password Conference DialPlan

Guest code for entering the conference room. Administrator code for entering the conference room.

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 Quiet Mode

If selected, the conference will be recorded. If selected, all users entering this conference will be marked 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 Queue
Call 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 Ring Strategy

This option defines a name for the Queue, e.g. "Sales" 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 Wrap-Up-Time(s)

Pause an Agent if they fail to answer a call. 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 <1 minute.

Repeat Frequency(s)

How often to announce a voice menu to the caller (0 to Disable Announcements).

Announcements

Select the 'Announcements Prompt' from IVR Prompts

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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 Files

Load music into the chosen directory. 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 PIN Response Timeout(s)

Input a name for the DISA Set a PIN which the user will be prompted for. 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. Default is 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 Always Status Busy No answer Ring lasting for(s)

Choose an extension All incoming calls will be forwarded Forward when extension is busy Forward when extension does not answer Default is 20 seconds. You can adjust this value as required.

Set your Follow Me number

Forward to an Internal Extension Forward to an External Extension

Incoming call will be forwarded to internal extension. Incoming call will be forwarded to external number or mobile number. Set an internal extension to pick up the call.

Set Internal Extension

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 Description Paging Group Members Device List

The number to be called to page this group.

Provide a descriptive title for the Page Group. Selected device(s) in this page

Devices that can be selected as paging group members Paging is typically one way for announcements only. Checking

Duplex

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 Call Recording Time

Define an extension to be recorded. Defines the monitoring rules. This can either be set to record all calls, or alternatively, a scheduled recording time can be set.

Inbound Record Outbound Record

Check to record inbound calls 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 Show All

Input contact name to search. Lists all contacts

To manual add a contact; click the create contact button

Name Number

Add contact's name. 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 Download Phonebook Demo

Uploads a supported file containing names and numbers for the phone book. 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 Set
PIN 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 PIN List

Defines a name for PIN set 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 Parking Calls Call Parking What Call Parking

Sets the Call Parking number.

Defines the extension where calls are parked, e.g. (701720)

extensions to park calls on 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 Pickup Call Pickup Extension 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 transferees number (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. Transfers 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 transferees number (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 Set key sequence to Pickup Specified Extension, e.g. dial *7 plus the extension to pickup specified extension. Set key sequence to pick up call.

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revert back to you. Disconnect Call Disconnect the current transfer call (for Attended transfer). Timeout for answer on attended transfer Blacklist a number Blacklist Remove a number from the black list 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 Rooms conference, or *# to hang up the current call and return to the conference yourself. Create Conference While speaking to a party, if you press *0 then you and the callee are immediately transferred to a conference room. 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 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
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Set the answer timeout value.

Add a black list number.

Remove a black list number.

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 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. 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 Queues do not have to remain on the line. Agent Logoff Extension Pause Queue Member Extension Unpause Queue 'Unpause' a queue member who is 'paused' previously so Member Extension that the member can receive calls again. 'Pauses' a queue member so that the member cannot receive calls. Agent logoff from the queue.

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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 auto provisioning 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. Click Extension->Create New User

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Manufacturer

Select the manufacturer of the handset. Current supported handsets are Zycoo, Grandstream and Yealink

MAC Method 2

Enter the MAC address of the handset.

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 Manufacturer

Deletes all extensions prior to phone provisioning. Select the manufacturer of the handset. Current supported handsets are Zycoo, Grandstream and Yealink

MAC Please choose file to upload

Enter the MAC address of the handset. Upload your file containing the extension details.

Note: For troubleshooting purposes it is worth noting that provisioning configuration files are stored in the /persistent/tftpboot folder.

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.

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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 auto provision 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. Extension Tab

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

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.

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 useful when diagnosing connectivity issues. Sip User Status Tab Displays the status of all user extensions enabled for SIP use.

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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 LAN Port Setup Tone Zone Settings

Enable/ Disable access to the GUI from the WAN port. Set the LAN port for the system (if required). 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 Packets

Enter IP or DNS address to Ping. 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 Interface Start IP End IP Subnet Mask Gateway Primary DNS Lease Time DDNS Settings Tab

Select to enable the DNS service. Select the interface on which DNS will be enabled. Start IP address of the lease range. End IP Address of the lease range. Subnet mask to issue to DHCP client devices Default gateway address to issue to DHCP Client devices. DNS server address to issue to DHCP client devices. The maximum lease time for the issued address.

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 DDNS Server

Enable the DDNS Service. Select the DDNS Service. Supported services are dyndns.org and www.no-ip.com.

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Username Password Domain Update Time (s)

Username for the service. Password for the service. Your domain name on the service. 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 Synchronize with current PC time

Select your local time zone. Click the button to synchronize the PBX time with the current PC time.

Management
Management Tab

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Change Password Time Zone Access Permit Tab

Used to change the ZX Series password and username if required. Sets the voice language for the system.

Deny any IP attempting to

Select if you wish to further secure your system by restricting GUI 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 Server Address User Name Password Directory

Enable periodical FTP uploading.

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

FTP account name. FTP account password. 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 Delete

Restore your selected backup file to system. 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.

Upgrade
The ZX Series supports upgrade via Web interface or TFTP. Web Upgrade

Restore Defaults Settings Please choose file to upload

Select to reset the configuration back to default as part of the upgrade process. Select the firmware file from either a local or network drive.

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TFTP Upgrade

Enter the package Name TFTP Server IP Address Restore Default Settings

Enter the firmware filename.

Enter the address of your TFTP Server.

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 reboot automatically.

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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 user root and password myippbx

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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 Advanced Options Global SIP Settings. This is important 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 Basic Trunks 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 - Status Register 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 to make 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 Basic Outbound 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. In the case of IPChitChat, the ports are 5060 UDP and 10000-20000 UDP. You should only allow access to the Service Providers Servers and not open your system up to everyone. Security is important and holes in your firewall could lead to your system being hacked.

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