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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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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:
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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
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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.
Associated Extension 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 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
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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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.
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.