Manual-Ii - Administration Guide For QX Ip Pbxs

Manual-II: Administration Guide for
QX IP PBXs
This manual is effective for all QX IP PBXs: QX20, QX50, QX200, QX500,
QX2000, QX3000 and QXISDN4+.
Please Note: This document contains confidential and proprietary information owned by Epygi Technologies, LTD. Any copying, use or disclosure of
the document or the information contained herein without the written permission of Epygi Technologies, LTD. is strictly prohibited.
Copyright © 2003-2017 Epygi Technologies, LTD. All Rights Reserved.
Manual-II: Administration Guide for QX IP PBXs
Notice to Users
This document, in whole or in part, may not be reproduced, translated or reduced to any machine-readable form without prior written approval.
Epygi provides no warranty with regard to this document or other information contained herein and hereby expressly disclaims any implied warranties of
merchantability or fitness for any particular purpose in regard to this document or such information. In no event shall Epygi be liable for any incidental,
consequential or special damages, whether based on tort, contract or otherwise, arising out of or in connection with this document or other information
contained herein or the use thereof.
Copyright and Trademarks
Copyright © 2003-2017 Epygi Technologies, LTD. All Rights Reserved. Quadro and QX are registered trademarks of Epygi Technologies, LTD. Microsoft,
Windows and the Windows logo are registered trademarks of Microsoft Corporation. All other trademarks and brand names are the property of their
respective proprietors.
Emergency 911 Calls
YOU EXPRESSLY ACKNOWLEDGE THAT EMERGENCY 911 CALLS MAY NOT FUNCTION WHEN USING QUADRO OR QX AND THAT EPYGI TECHNOLOGIES,
LTD. OR ANY AFFILIATES (AGENTS) SUBSIDIARIES, PARTNERS OR EMPLOYEES ARE NOT LIABLE FOR SUCH CALLS.
Limited Warranty
Epygi Technologies, LTD. (‘Epygi’) warrants to the original end-user purchaser every Quadro and QX to be free from physical defects in material and
workmanship under normal use for a period of one (1) year from the date of purchase (proof of purchase required) or two (2) years from the date of
purchase (proof of purchase required) for products purchased in the European Union (EU). If Epygi receives notice of such defects, Epygi will, at its
discretion, either repair or replace products that prove to be defective.
This warranty shall not apply to defects caused by (i) failure to follow Epygi’s installation, operation or maintenance instructions; (ii) external power
sources such as a power line, telephone line or connected equipment; (iii) products that have been serviced or modified by a party other than Epygi or an
authorized Epygi service center; (iv) products that have had their original manufacturer’s serial numbers altered, defaced or deleted; (v) damage due to
lightning, fire, flood or other acts of nature.
In no event shall Epygi’s liability exceed the price paid for the product from direct, indirect, special, incidental or consequential damages resulting from the
use of the product, its accompanying software or its documentation. Epygi offers no refunds for its products. Epygi makes no warranty or representation,
expressed, implied or statutory with respect to its products or the contents or use of this documentation and all accompanying software, and specifically
disclaims its quality, performance, merchantability or fitness for any particular purpose.
Return Policy
If the product proves to be defective during this warranty period, please contact the establishment where the unit was purchased. The Integrator will
provide guidance on how to return the unit in accordance with its established procedures. Epygi will provide the Return Merchandise Authorization
Number to your retailer.
Please provide a copy of your original proof of purchase. Upon receiving the defective unit, Epygi, or its service center, will use commercially reasonable
efforts to ship the repaired or a replacement unit within ten business days after receipt of the returned product. Actual delivery times may vary depending
on customer location. The Distributor is responsible for shipping and handling charges when shipping to Epygi.
European Limited Warranty
The European Limited Warranty is the same as the Limited Warranty above, except the warranty period is for two years from the date of purchase.
Extended Warranty
Extended Warranty Option
Epygi offers an extended warranty program available for purchase by end users. This option is available at the time of purchase, extending the users
original warranty for an additional three (3) years. Combined with the original warranty, the extended warranty would offer a total of five (5) years
protection for European end users and four (4) years protection for non-European end users.
Extended Warranty Statement
Epygi Technologies, LTD. extends its Limited Warranty for an additional period of three (3) years from the date of the termination of the original Limited
Warranty period (proof of purchase required).
Epygi reserves the right to revise or update its products, pricing, software, or documentation without obligation to notify any individual or entity. Please
direct all inquiries to:
Epygi Technologies, LTD.
2233 Lee Road Suite 201 Winter Park, Florida 32789
Revision 1.2 2
Administrative Council for Terminal Attachments (ACTA) Customer Information

This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA. Located on the equipment is a label that contains,
among other information, the ACTA registration number and ringer equivalence number (REN). If requested, this information must be provided to the
telephone company.
The REN is used to determine the quantity of devices which may be connected to the telephone line. Excessive REN’s on the telephone line may result in
the devices not ringing in response to an incoming call. In most, but not all areas, the sum of the REN’s should not exceed five (5.0). To be certain of the
number of devices that may be connected to the line, as determined by the total REN’s contact the telephone company to determine the maximum REN for
the calling area.
This equipment cannot be used on the telephone company-provided coin service. Connection to Party Line Service is subject to State Tariffs.
If this equipment causes harm to the telephone network, the telephone company will notify you in advance that temporary discontinuance of service may
be required. If advance notice isn’t practical, the telephone company will notify the customer as soon as possible. Also, you will be advised of your right the
file a complaint with the FCC if you believe it is necessary.
The telephone company may make changes in its facilities, equipment, operations, or procedures that could affect the operation of the equipment. If this
happens, the telephone company will provide advance notice in order for you to make the necessary modifications in order to maintain uninterrupted
service.
If trouble is experienced with this equipment, please contact EPYGI TECHNOLOGIES, LTD.
If the trouble is causing harm to the telephone network, the telephone company may request you to remove the equipment from the network until the
problem is resolved.
Electrical Safety Advisory
To reduce the risk of damaging power surges, we recommend you install an AC surge arrestor in the AC outlet from which the Quadro or QX is powered.
Industry Canada Statement
This product meets the applicable Industry Canada technical specifications.
Safety Information
Before using the Quadro or QX, please review and ensure the following safety instructions are adhered to:
 To prevent fire or shock hazard, do not expose your Quadro or QX to rain or moisture.
 To avoid electrical shock, do not open the Quadro or QX. Refer servicing to qualified personnel only.
 Never install wiring during a lightning storm.
 Never install telephone jacks in wet locations unless the jack is specified for wet locations.
 Never touch non-insulated telephone wire or terminals unless the telephone line has been disconnected at the network interface.
 Use caution when installing or modifying cable or telephone lines.
 Avoid using your Quadro or QX during an electrical storm.
 Do not use your Quadro, QX or telephone to report a gas leak in the vicinity of the leak.
 An electrical outlet should be as close as possible to the unit and easily accessible.
Emergency Services
The use of VoIP telephony is made available through IP networks such as the Internet and is dependent upon a constant source of electricity, network
availability and proper operation of the equipment. If a power outage, network disruption or equipment failure occurs, the VoIP telephony service could
be disabled. User understands that in any of those events the Quadro or QX may not be able to support 911 emergency services, and further, such services
may only be available via the user's regular telephone line or mobile lines that are not connected to the Quadro or QX. User further acknowledges that any
interruption in the supply or delivery of electricity, network availability or equipment failure is beyond Epygi's control and Epygi shall have no
responsibility for losses arising from such interruption.
Music on Hold Copyright
The default Music on Hold on the Quadro or QX is a 22 second fragment from Chopin's Nocturne Op.9 #2 performed by Marina Vardanyan and kindly
provided to Epygi Technologies, LTD. The recording is royalty free.
Compliance with Laws
You may not use the Epygi Materials for any illegal purpose or in any manner that violates applicable domestic or foreign law. You are responsible for
compliance with all domestic and foreign laws governing Voice over Internet Protocol (VoIP) calls.
Revision 1.2 3
Document Edition History
Revision Date Description Valid for Models Valid for FW

1.0 24-Mar-17 Initial Release QX IP PBXs 6.1.45 and higher
Added a new licensable feature -
1.1 16-Jun-17 QX IP PBXs 6.1.50 and higher
Calling Cost Control. Updated.
QX20, QX50, QX200,
QX500, QX2000
1.2 11-Dec-17 Updated for the new QX3000. 6.2.1 and higher
QX3000 and
QXISDN4+
Revision 1.2 4
Table of Contents
1 About Administration Guide ..........................................................................7

2 Conventions Used in this Guide ....................................................................7
3 QX’s Graphical Interface ...............................................................................9
4 Dashboard ..................................................................................................10
5 Setup Menu.................................................................................................11
5.1 Basic Setup .............................................................................................................. 12
5.2 System Security ........................................................................................................ 22
5.3 Licensed Features..................................................................................................... 23
5.4 Redundancy ............................................................................................................. 25
5.5 Language Pack ......................................................................................................... 25
6 Extensions Menu .........................................................................................27
6.1 Extensions ................................................................................................................ 28
6.2 Dialing Directories ..................................................................................................... 66
6.3 Conferences ............................................................................................................. 66
6.4 Recordings ............................................................................................................... 67
6.5 Receptionist Management ........................................................................................ 68
6.6 ACD Management .................................................................................................... 68
6.7 Authorized Phones.................................................................................................... 69
7 Interfaces Menu ...........................................................................................71
7.1 IP Lines ..................................................................................................................... 72
7.2 FXS Lines.................................................................................................................. 85
7.3 FXO Settings ............................................................................................................. 88
7.4 E1/T1 Trunk Settings ................................................................................................ 89
7.5 ISDN Trunk Settings ................................................................................................. 89
7.6 Shared PSTN Gateways ........................................................................................... 90
8 Telephony Menu ..........................................................................................91
8.1 VoIP Carrier .............................................................................................................. 92
8.2 Call Routing .............................................................................................................. 95
8.3 Call Recording Settings............................................................................................. 118
8.4 NAT Traversal ........................................................................................................... 118
8.5 RTP Settings ............................................................................................................. 122
8.6 SIP ............................................................................................................................ 123
8.7 Schedules ................................................................................................................. 126
8.8 Advanced ................................................................................................................. 126
9 Firewall Menu ..............................................................................................133
9.1 Firewall...................................................................................................................... 134
9.2 Filtering Rules ........................................................................................................... 136
9.3 Custom Services....................................................................................................... 141
9.4 IP Groups ................................................................................................................. 141
9.5 SIP IDS ..................................................................................................................... 142
10 Network Menu .............................................................................................144
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10.1 IP Routing ................................................................................................................. 145

10.2 DHCP ....................................................................................................................... 147
10.3 DNS Settings ............................................................................................................ 151
10.4 PPP/ PPTP Settings ................................................................................................. 154
10.5 SNMP Settings ......................................................................................................... 156
10.6 VLAN Settings .......................................................................................................... 156
10.7 VPN Configuration .................................................................................................... 156
10.8 OpenVPN Configuration ............................................................................................ 168
11 Status Menu ................................................................................................169
11.1 System Status .......................................................................................................... 170
11.2 Events....................................................................................................................... 176
11.3 Call History ............................................................................................................... 179
11.4 Conference History ................................................................................................... 185
11.5 Network Interfaces .................................................................................................... 185
11.6 Statistics ................................................................................................................... 186
12 Maintenance Menu ......................................................................................189
12.1 Diagnostics ............................................................................................................... 190
12.2 System Logs ............................................................................................................. 194
12.3 System Logs Settings ............................................................................................... 194
12.4 Remote Logs Settings .............................................................................................. 195
12.5 Logs Archive ............................................................................................................. 195
12.6 User Rights Management ......................................................................................... 196
12.7 Backup / Restore ...................................................................................................... 199
12.8 Firmware ................................................................................................................... 203
12.9 Reboot ...................................................................................................................... 206
12.10 Registration Form ..................................................................................................... 207
13 Appendix: Administrator Login ......................................................................208
14 Appendix: Needed Bandwidth for IP Calls .....................................................209
15 Appendix: System Default Values..................................................................210
15.1 System Settings ........................................................................................................ 210
15.2 User Extension Settings ............................................................................................ 223
16 References ..................................................................................................225
17 Appendix: Software License Agreement ........................................................226
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1 About Administration Guide

This guide is intended for administrators who need to prepare for install, configure and operate QX IP PBXs
(herein QX). In this guide, we describe the functionality and configuration of QXs with reference to other guides,
manuals and complementary resources.
This guide contains many example screen illustrations. Since QXs offer a wide variety of features and
functionality, the example screenshots shown may not appear exactly the same for your particular QX as they
appear in this manual. The example screenshots are for illustrative and explanatory purposes, and should not
be construed to represent your own unique environment.
2 Conventions Used in this Guide

Following conventions are used in this guide:
 Add – this button is used to create and add new entry.
 Edit – this button is used to modify the selected entry(s).
 Delete – this button is used to remove the selected entry(s).
 Save – this button is used to apply the changes.
 Start – this button is used to start a service, connection, etc.
 Stop – this button is used to start a service, connection, etc.
 Enable/Disable – this button is used to enable/disable the selected entry(s).
 Move Up/Move Down – are used to sort the entries in the specific table in the order they need to be
accessed.
 Generate Password – this button is used to generate a system defined strong password.
 Show Hot Desking Settings/Hide Hot Desking Settings – these links are used to show/hide the Hot
Desking settings respectively.
 Hide extensions attached to disabled IP lines / Show all extensions – these links are used to hide
extensions which are attached to disabled IP lines or show all created extensions respectively.
 Call Type – lists the available call types:
 PBX – local calls to QX extensions.
 SIP – calls via SIP.
 PSTN – calls to a legacy telephone network (N/A for QX20, QX500, QX2000 and QX3000).
 Auto – calls to a destination resolved by the Call Routing Table.
 Address (Redirect Address, Calling Address or Call to) – this field is used to define the destination
address the call will be addressed to. The address strictly depends on the call type. Thus, define an
extension number for the PBX calls, SIP address for the SIP calls, phone number for the PSTN calls,
and, finally, define a routing pattern for the Auto type calls.
 Description – this field is used to enter any optional information about the entry.
 Wildcard supported – used to mention that wildcards are allowed for the field. Go to the Allowed
Characters and Wildcards section to see the complete list of the supported characters and wildcards.
 The following options are available on the QX to select the way custom voice message will be provided:
 File – is used to upload/record the file for the message.
 RTP Channel – is used to stream the massage (hold music, ringing announcements, queue messages,
etc.) through the RTP Channels.
 Audio Line In – is used to stream the message through the Audio Line In. This option is not available on
QXISDN4+, QX2000 and QX3000.
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 Upload File/Record File – show the available methods in case if File is selected from the options
mentioned above:
 Click Choose File next to the Upload file field to open a file chooser window to upload the file.
 Click Record from Extension next to the Record file field to record a message directly on the phone.
The Recording Settings page will be opened.
Once the message has been uploaded/recorded the following links will appear:
 Download … message – used to download the uploaded/recorded message.
 Remove … message – used to remove the uploaded/recorded message or restore the default one.
Figure 1: Recording Settings page
 The Recording Settings page is used to initiate a custom voice message recording for the current
extension directly from an IP phone.
 Record from extension – lists all phone extensions that are available for recording.
Record a message as follows:
1. Select the extension from the Record from extension list.
2. Click Start Recording. The phone for the defined extension will start ringing.
3. Answer the call and follow the audio prompts to record a message.
4. Once the message has been recorded the following links will appear:
 Download Recording – is used to download the recorded message.
 Restore Default Recording – is used to remove the recorded message and restore the default one.
Note:
 The uploaded file should be either in (*.wav) or (*.mp3) format.
 The maximum duration of the uploaded file is limited to 5 minutes.
 The maximum size of the uploaded file is limited to 7.5 MB.
Revision 1.2 8
3 QX’s Graphical Interface

The following top menus and links are available on the QX Management page when logged in as an
administrator:
 Dashboard
 Setup
 Extensions
 Interfaces
 Telephony
 Firewall
 Network
 Status
 Maintenance
 Go To Extension – allows quick access to the User Settings for the selected extension.
 Pending Events – allows quick access to the system events and event settings.
 Language – available when a custom Language Pack has been installed. Is used to enable the custom
language for GUI or revert back to the default English.
 Date/Time – displays the device's current time.
 Hostname – displays the device's hostname.
 Renew WAN IP Address – will be shown if the WAN IP address for QX assigned dynamically via DHCP.
Revision 1.2 9
4 Dashboard
If you are logged in as an administrator (users: admin or localadmin), you will see the number of calls currently
active on QX. The Active Calls table includes information about the calling/called parties, call start time and
duration.
Figure 2: Dashboard menu
 The Terminate link is used to terminate the active call.

 The Start Recording link is used to manually start the recording of the corresponding call. Once the call
recording started, the link changes to Stop now and is used to manually stop the recording. The
recording can be started again if needed.
 The list of users currently logged into the system appears in the lower right corner of the page. The IP
address of the user, the time until the next automatic logout and the current version of the QX's firmware
are presented as well. The idle session timeout is set to 10 minutes. If no action is performed within 10
minutes, the user will be automatically logged out.
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5 Setup Menu
Figure 3: Setup Menu overview
Revision 1.2 11
5.1 Basic Setup

5.1.1 System (LAN)
You can login the QX WEB GUI through the LAN interface using the default IP address:
 For QX20, QX50, QX200, QX500 and QXISDN4+ – 172.30.0.1
 For QX2000 and QX3000 – 192.168.0.200
Go to the SetupBasic SetupSystem (LAN) to adjust the network parameters for the LAN interface. The
System Configuration Wizard navigate you through the following parameters and settings:
 System Configuration
 DHCP Settings for the LAN Interface
 Regional Settings and Preferences
 Emergency Codes and PSTN Access Code Settings
System Configuration
Figure 4: System Configuration section
 Host Name – set the host name for QX.

 Domain Name – set domain name which the QX belongs to.
 IP Address – set the LAN IP address.
 Subnet Mask – set the subnet mask.
Revision 1.2 12
DHCP Settings for the LAN Interface
Figure 5: DHCP Settings for the LAN Interface section
 Enable DHCP Server – enable/disable DHCP server capability on the QX.

 Dynamic IP Address Range: (from - to) – set the IP address pool.
 WINS Server – set the IP address for the WINS server.
Regional Settings and Preferences
The regional settings are important for the functionality of the QX voice subsystem.
Figure 6: Regional Settings and Preferences section
 Your Locale (location) – select the location and timezone of QX.
Revision 1.2 13
 Timezone – select the proper time zone so the QX can display correct time accordingly. TIP: The QX
supports Daylight Savings (DST) correction if it is available for the selected time zone.
 Choose System Language – select the language for system voice messages: custom or default English.
TIP: This selection is available when a custom Language Pack has been uploaded.
Emergency Codes and PSTN Access Code Settings
Figure 7: Emergency Codes and PSTN Access Code Settings section
 Emergency Codes – enter the PSTN number(s) of the emergency service(s). For each emergency code, a
routing pattern will be generated in the Call Routing Table, allowing faster and easier calls to emergency
services. TIP: Use commas to separate emergency codes in case of multiple numbers.
 PSTN Access Code – select the prefix code for accessing the PSTN line through the routing table.
Note:
 Finish the wizard and click "OK" to apply the changes made in any section of the wizard. You must
confirm the settings within 20 minutes. Otherwise the device will return back to the previous
configuration and reboot.
 It is strongly recommended to not change the factory default settings if their meanings are not fully
clear to you.
Revision 1.2 14
5.1.2 Internet (WAN)

Go to the SetupBasic SetupInternet (WAN) to configure or adjust the network parameters for the QX WAN
interface. The Internet Configuration Wizard navigates through the following basic configuration parameters and
settings:
 Uplink Configuration
 WAN Interface Protocol
 WAN Interface Configuration
 DNS Settings
Uplink Configuration
Figure 8: Uplink Configuration section
 WAN Interface Protocol – select the protocol for the WAN interface. Based on this selection the wizard's
configuration pages may differ. The following connection protocols are available:
 PPPoE
 PPTP
 Ethernet
 VLAN (TIP: This option becomes available only when VLAN is configured on the QX.)
Note: PPPoE and PPTP aren't supported on QX2000 and QX3000.
 WAN interface bandwidth settings specify the upstream and downstream speeds in Kbit/s, helping to
assure the quality of IP calls. IP call loses the voice quality if there is no available bandwidth. When
approaching the limits of a bandwidth capacity, another IP call will be declined.
 Min Data Rate – set the amount of upstream bandwidth that ought to remain for data traffic even if voice
applications use the entire available upstream bandwidth. The value selected here needs to be smaller
than the upstream bandwidth.
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PPPoE
 Keep Connection Alive – keeps the connection alive by sending control packets for the link state
verification.
 Authentication Settings – enter the authentication parameters (Username and Password) to register on
the ISP server.
 Dial Behavior – select the Dial Behavior type.
 Dial manually – if selected, a button will be displayed in the top WEB management window to switch
the connection on/off.
 Always connected – if selected, the QX will always stay connected.
 IP Address Assignment – select the IP Address assignment type for the PPPoE interface:
 Obtain an IP Address automatically – with this option QX will get an IP address dynamically.
 Use the following IP Address – set the IP address manually.
PPTP
 Obtain an IP Address automatically – with this option selected, QX will use DHCP to get an available IP
address from your local network or ISP.
 Use the following IP Address – if selected, manually provide the settings for the WAN interface.
Click Next to continue the configuration of the PPP/ PPTP settings:
 PPTP Server – enter the IP address of the PPTP server.
 Encryption – select the encryption for the traffic over the PPTP interface.
 Keep Connection Alive – keeps the connection alive by sending control packets for the link state
verification.
 Authentication Settings – enter the authentication parameters (Username and Password) to register on
the ISP server.
 Dial Behavior – select the Dial Behavior type.
 Dial manually – if selected, a button will be displayed in the top WEB management window to switch
the connection on/off.
 Always connected – if selected, the QX will always stay connected.
 IP Address Assignment – select the IP Address assignment type for the PPPoE interface:
 Obtain an IP Address automatically – with this option QX will get an IP address dynamically.
 Use the following IP Address – set the IP address manually.
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Ethernet
 Use the following IP Address – if selected, manually provide the settings for the WAN interface.
VLAN
 VLAN ID – select VLAN ID from the configured VLAN list.

Click Next to continue the configuration of the VLAN IP Configuration settings.
 Use the following IP Address – if selected, manually provide the settings for the VLAN interface.
WAN Interface Configuration
This section is used to modify the MAC address of the QX. This might be necessary if the ISP requires a
specified MAC address (e.g. for authentication).
Figure 9: WAN Interface Configuration section
 This device – selects the default MAC address of the WAN interface.
 User-defined – enter the MAC Address manually.
 MTU – select the maximum size of packet that can be sent in a packet or frame-based network such as
the Internet. QX supports packet fragmentation. TIP: The default MTU size is 1500 Bytes for Ethernet
protocol and 1400 Bytes for PPPoE.
Revision 1.2 17
DNS Settings
Figure 10: DNS Settings section
 Obtain DNS Server Address automatically – automatically configures the assignment of the name server
address from the provider party.
 Use the following DNS Server Address – is used to manually assign a name server as follows:
 Preferred DNS – enter the IP address of an external name server.
 Alternate DNS – enter the IP address of the secondary name server that will be used if the main name
server cannot be accessed.
Note:
 Finish the wizard and click "OK" to apply the changes made in any section of the wizard. You must
confirm the settings within 20 minutes. Otherwise the device will return back to the previous
configuration and reboot.
 It is strongly recommended to not change the factory default settings if their meanings are not fully clear
to you.
 The Internet Configuration Wizard is renamed to Uplink Configuration Wizard on QX2000 and QX3000.
Revision 1.2 18
5.1.3 Date and Time

The QX's Date and Time settings may be updated through the international time servers.
Figure 11: Date / Time Settings page
 Date/Time – displays the current system time.

 Enable SNTP Server – enable or disable SNTP server capability on the QX.
 Enable SNTP Client – enable or disable SNTP client on the QX. If not selected, the current system time
can be configured manually.
 Polling interval – select the time interval for the periodical synchronization between the timeserver and
QX.
The SNTP Servers table lists all defined SNTP servers. To add a new SNTP server:
1. Click Add. Define new server parameters:
 Manual – enter the SNTP server’s FQDN (Full Qualified Domain Name) or IP address.
 Predefined – select the SNTP server’s host address from the drop-down list.
2. Click Save to add the new SNTP server in the SNTP Servers table.
3. Click Move Up or Move Down to sort NTP servers in the order they need to be accessed. TIP: If the NTP
server in the first position of the SNTP Servers table does not answer, NTP server in the next position will
be attempted to reach.
Revision 1.2 19
5.1.4 E-mail (SMTP)

Simple Mail Transfer Protocol (SMTP) service allows QX to automatically generate and send alert and
notification e-mails as specified in the Event Settings.
 Enable SMTP Service – activates the SMTP service.
 SMTP Host – IP address or hostname of the SMTP server.
 E-mail Sender Address – e-mail address that is either registered on the selected SMTP server or has
permission to use the SMTP server for e-mail transmissions.
 E-mail Recipient Address – an active address to send e-mails to.
 E-mail Recipient Address (CC) – an active address to deliver e-mails’ carbon copy (CC) to.
 The server requires a secure connection (TLS) – select if the specified SMTP server requires secure
connection using TLS. If the specified SMTP server allows to use both secure and unsecure
connections, then this selection forces to establish the secure connection.
 Enable SMTP Authentication – select if the specified SMTP server requires authentication. Then enter
the Username and Password configured on the SMTP server.
Shown below is the sample e-mail settings on the QX, assuming the e-mail is using smtp.gmail.com as the
SMTP server.
Figure 12: E-mail Settings page
Once the configuration is finished, click "Send test e-mail" to send a test e-mail to the defined e-mail address to
verify the settings.
Revision 1.2 20
5.1.5 Short Text Messaging (SMS)

The SMS service allows QX to automatically generate and send alert and notification events via SMS.
 Enable SMS Service –
activates SMS service.
 Username and Password –
authentication parameters
configured on the SMS
server.
 SMS Sender Address – sms
sender's address.
 SMS Recipient Address –
sms recipient's address.
TIP: Use a space,
semicolon or a comma to
separate mobile numbers in
case of multiple recipients.
You may either use predefined
SMS gateway (Clickatell) or
define a custom service.
 Clickatell – select to use the
predefined SMS gateway.
Then enter the Clickatell
specific parameter provided
by the server in the
activated API ID field. This
parameter must be identical
on both sides.
 Custom – select to define a
custom gateway as follows:
 Resource – enter the
HTTP resource name on
the SMS gateway.
 Parameters – enter Figure 13: SMS Settings page
parameters to be
submitted to the resource
address. The value of this field represents a string with tokens (separated by percent (%) symbols) inside.
Each token indicates a value of the certain field on this page. The value depends on the SMS gateway
requirements. The tokens are the strings that have the following dependencies from the field in this page:
 %username% – indicates the username defined in the field Username.
 %password% – indicates the password defined in the field Password.
 %to% – indicates the password defined in the field SMS Recipient Address.
 %from% – indicates the password defined in the field SMS Sender Address.
 %text% – indicates the SMS text generated by QX (voice mail notification, event notification, etc.).
For example: user=%username%&password=%password%&to=%to%&from=%from%&text=%text%

 Server – IP address or hostname of the SMS gateway.
 Port – port number of the SMS gateway.
Revision 1.2 21
 Use Secure HTTP to access the SMS server via HTTPS. Then define the port number for HTTPS traffic in
the activated Secure Port field.
 Select one of the HTTP request’s methods (POST or GET) through the Request Method buttons. The QX
uses one of methods to access to the SMS gateway.
Once the configuration is finished, click " Send test SMS" to send a test SMS to the defined mobile number to
verify the settings.
5.2 System Security

The System Security Management is used to manage the QX’s global security.
Figure 14: System Security Management page
QX treats the selected security level when checking the passwords strength and when running the security
audit to get security reports. The security levels are the following:
 Low – there are no specific restrictions on the strength of the saved password. Only the critical warnings
on the Call Routing Rules to PSTN and IP-PSTN, disabled Firewall and IDS will be generated in Security
Report.
 Medium – the minimum strength of the passwords must be "moderate". The Security Report will
generate warnings on all unsecured Call Routing rules, IP Line and extension passwords, Firewall level (if
it is set below "Medium"), disabled IDS, default administrator passwords.
 High – the minimum strength of the passwords must be "strong". The Security Report will generate
warnings on the IP Line and extension passwords, disabled IDS, all unsecured Call Routing rules,
Firewall level (if it is set below "High"), default administrator passwords etc.
Revision 1.2 22
5.3 Licensed Features

5.3.1 Feature Keys
Two types of licensable feature keys are available on QX:
 Permanent keys – activate licensable features on QX permanently, without time limitation. Permanent
keys are available for all licensable features, except for these two: Epygi ACD Console and Epygi Hotel
Console.
 Time limited keys – activate or extend the operation for already activated licensable features temporarily,
for the specified period. The feature will be no longer functional after the period expiration date. The Time
limited keys are available for all licensable features.
Figure 15: Features page
 Debug – enables SSH connection towards the QX for debugging purposes.

 3PCC – enables Third Party Call Control feature on the QX. The feature allows the call controlling
applications running on a user PC to remotely initiate and handle calls on the QX and to subscribe for
certain event notifications from the QX.
 Automatic Call Distribution – enables the ACD feature (N/A for QX20 and QXISDN4+) which provides
contact center solution for queuing and automatic distribution of the calls between contact center
agents.
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 Barge-In – enables the Barge-In on the QX (N/A for QX20). The feature allows the PBX users to
participate to the third party's calls while remaining imperceptible.
 Redundancy – activates the Redundancy feature (N/A for QX20) on the QXs.
 Epygi Hotel Console – activates EHC application support (N/A for QX20) for the QX.
 Call Cost – this feature allows to limit and control the cost of calls through the routing rules on the QX.
 PMSLINK Connection – this feature is used to enable the interface for connecting to PMSLINK
middleware from char and integrate QX with PMS used in hotels.
 DCC Pro – activates Desktop Communication Console Pro-level application support for the QX.
 DCC Basic – activates Desktop Communication Console Basic-level application support for the QX.
 iQall Mobile Toggling – this feature allows users to alternate between their mobile (iPhone/Android)
running iQall application and their desk phone without the call being disconnected.
 IP Phone Expansion – enables additional IP phones support on the QX.
 Auto Dialer – activates Auto Dialer application support (N/A for QX20) for the QX.
 Audio Conference – activates the Conference feature allowing the system to act as a standalone
conference server.
 Epygi ACD Console – activates Web monitoring support for ACD processes on the QX (N/A for QX20 and
QXISDN4+).
 Call Recording – activates the Call Recording feature which is used to record PBX, SIP or PSTN calls on
the QX and save the recordings into the local recording box or upload to the remote server.
 Video Conference – activates the Video Conferencing feature.
To receive a Feature Key, register the QX device and send a corresponding request to Epygi Technical Support.
This request must include the Unique ID that is displayed in the Features page above the features list.
Enter a Feature Key as follows:
1. Click Add.
2. Enter the key in the Feature Key field.
3. Click Save. The status of the selected feature will turn to "Reboot needed".
4. Reboot QX to complete the installation. The status of the feature will turn to "Activated".
Figure 16: Adding a feature key
Note:
 Please make sure to have correct Date/Time on the device before adding the license key, otherwise you
may have issues with the applied key.
 When using Call Recording and/or ACD features on the QX50/QX200 it is advisable to use an SD
memory card to expand the system memory. Currently, the recommended SD card’s largest capacity is
32 GB.
Revision 1.2 24
 When using Call Recording and/or ACD features on the QX20/QX500 it is advisable to use a micro SD
memory card to expand the system memory. Currently, the recommended micro SD card’s largest
capacity is 64 GB.
For more information on Licensable Features, please refer to the Licensable Features on QX IP PBXs guide.
5.3.2 Free Trial

This page lists all QX features that may be activated for a trial period.
Expiration Date/Time – is used to specify the trial period. Upon expiring the specified period, the QX will reboot
and trial feature(s) will disable. TIP: The trial option can be activated on the QX only once. You cannot activate
the trial for the same or any other feature again after the first activation.
To activate trial feature:
1. Select the checkbox next to the feature.
2. Specify the needed count under the Count column (depending on the selected feature).
3. Click Save. The QX will reboot and activate the selected trial feature(s).
5.4 Redundancy
The Redundancy feature is used to increase the QX availability by using the second QX as a backup unit. This
requires two units running the same firmware version and connected to each other through Ethernet or LAN
ports, depending on the device model. The idea of redundancy is to ensure uninterrupted functionality of the
QX. The Redundancy Settings must be configured on both units. One of the units is configured as a master, the
second one as a backup.
For information on how to configure and use Redundancy feature, please refer to the Redundancy Feature on
QX IP PBXs guide.
5.5 Language Pack

All Epygi supported LPs will change the system voice messages to the custom language. Some of LPs will
change the device GUI interface and the GUI interface on supported IP phones as well.
For more information on Language Packs, please refer to the Language Packs Overview for Epygi QX Line
guide.
To upload a language pack:
1. Click Choose File to browse and select the file for the language pack.
2. Click Save to start uploading the language pack. Clicking Save will stop some vital processes on the QX,
therefore it is required to manually reboot the device even if you have cancelled the LP update procedure
on the following steps.
3. Click Yes to proceed the upload. The QX will be rebooted automatically.
4. Uploaded LP will appear in the Current language pack field. After successful upload, you will be able to:
 Change the language of the GUI session from the GUI Login page or from main menu.
 Switch the system voice messages to the custom language and change the GUI interface of some
supported IP phones. TIP: Choose the language from the Regional Settings and Preferences section of
the System Configuration Wizard to change the system voice messages and GUI language for the IP
Phones. The IP phones will be automatically rebooted to change language.
Revision 1.2 25
Figure 17: Language Pack page
After successful upload, the following links will be available:

 Remove current language pack – is used to remove the uploaded language pack.
 Upload IP Phones LP – is used to upload custom language pack files for some supported IP phones.
To upload a custom LP on the desired IP phone:
1. Click the hyperlinked IP phone vendor.
2. Click Choose File to browse a custom LP on the next appeared page.
3. Confirm LP upload to the IP phone on the next appeared page. Then reboot the IP phone to activate the
new LP. TIP: Clicking Save will stop some vital processes on the IP Phone, therefore reboot your phone
manually even if you have cancelled the language pack update procedure on the following steps.
Note: Only one custom Language Pack can be uploaded at a time. Thus, the new LP will remove the existing
one and reboot the QX. Once the QX is rebooted, the connected IP phones will reboot then.
Revision 1.2 26
6 Extensions Menu
Figure 18: Extensions Menu overview
Revision 1.2 27
6.1 Extensions
6.1.1 Extensions
Navigating to the Extensions Management page for the first time after the QX initial start or configuration restore
you will be prompted to choose the extensions length applicable to all QX default extensions.
Figure 19: Choose Extensions Length page
The following options are available:

 Leave Current Length – keep the current length of QX extensions unchanged. By default, the extension’s
length is 3 on the QX20, QX50, QX200, QX500, QXISDN4+ and is 4 on the QX2000 and QX3000. In
front of this selection, the actual configured length of extensions is displayed.
 Change Length – change the length of extensions as follows:
 Extension Length – select the length of extensions. It will be applied for all existing extensions on the
QX. The length of the extension can be 3, 4 or 5.
 Extension Prefix – define the prefix the existing extensions as well as the newly created extensions
should start with. The prefix cannot start with the digits 0 or 9.
Attention:
 By saving the settings on the Choose Extensions Length page, all existing extensions will lose the
custom voice messages and voice mails in the mailbox. The device will be rebooted. The Choose
Extensions Length page will not appear again unless the default configuration settings will not be
restored on the QX.
 QX20 is limited to 100, QX50/QXISDN4+ to 200, QX200 to 400, QX500 to 800, QX2000 to 2400 and
QX3000 to 3400 extensions in total.
Revision 1.2 28
Figure 20: Extensions Management page
The Extensions Management table consists of the following components:

 Extension – lists the numbers for extensions on the QX. These numbers are used for calling the
extensions internally.
 Display Name – is an optional name given to extension mainly to identify the extension’s owner at the
called side.
Revision 1.2 29
 Attached Line – indicates the IP or FXS line a corresponding extension is attached to. TIP: R is displayed
in this column when Remote Extension service is enabled on the extension. None is displayed when no
FXS or IP line is attached to the extension.
 SIP Address – displays the full SIP address of extension, (i.e., username@sipserver:port) when the
Registration on SIP Server is enabled. If registration is disabled, the SIP address will be displayed in the
following format: "username, Proxy: sipserver:port". If no SIP registration server or SIP server port is
defined, corresponding information will not be included in this column. If no username is defined, the
extension number will be displayed instead.
 Percentage of System Memory – indicates the memory size assigned to extension in percentage
regarding the total system memory. The actual available duration for the extension’s voice mails,
uploaded/recorded greetings and blocking messages is also displayed here. The available time duration
corresponding to the selected user space depend on the Voice Recording codec selected from the
Voice Mail page.
 External Access – indicates whether the GUI Login, 3pcc/Click2Dial login or Call Relay options are
enabled on the extension.
 Credit – indicates the available credit amount of each extension.
 Codecs – lists the short information about extension specific voice Codecs. Extension codec’s can be
accessed and modified by clicking on the link of the corresponding extension’s Codecs. The link leads to
the Extension Codecs page.
6.1.2 Add Extension

To add a new extension:
1. Click Add Extension.
 Enter the extension number.
 Select the extension type. The following types are available: Attendant, User Extension, Pickup Group,
Call Park, Paging Group and Recording Box.
2. Click Save to add the new extension to the Extension Management table.
Figure 21: Extensions Management – Add Entry page
Two types of user extensions, active and inactive, can be created on the QX.
 Active extensions are those that are attached to a line, can place and receive calls and use available
telephony services.
 Inactive extensions are those that are not attached to the line. They can use some available telephony
services, but cannot place and receive calls.
Revision 1.2 30
Note:
 Adjust the routing rules for calling extensions with custom length manually since the call routing rule(s) for
calling PBX extensions will not be adjusted automatically.
 A maximum extension length is 20 digits.
 The Recording Box extension type becomes available if the Call Recording feature is activated on the QX.
6.1.3 Add Multiple Extensions

The Add Multiple Extensions page is used to add multiple extensions to the Extensions Management table at
once.
Figure 22: Extensions Management – Add Multiple Extensions page
To add multiple extensions:

1. Select the extension type. The following types are available: Attendant, User Extension, Pickup Group,
Call Park, Paging Group and Recording Box.
2. Enter the amount of extensions.
3. Enter the number for the first extension. Based on the Quantity, the next extensions will have subsequent
numbers.
4. Enter the SIP username of the first extension. Based on the Quantity, the next extensions will have
subsequent SIP usernames.
5. Select the "Automatically attach to IP Line" option to attach extensions to IP lines.
6. Enter the number of the first IP line to be attached.
7. Enter the SIP Server address and SIP Server port. If the SIP server port is not specified, QX will access
the SIP server via the default 5060 port.
8. Select the "Registration on SIP Server" option to enable registration of the extensions on the SIP Server.
9. Click Save to add the new extensions to the Extension Management table.
Note:
Revision 1.2 31
 Adjust the routing rules for calling extensions with custom length manually since the call routing rule(s) for
PBX extensions will not be adjusted automatically.
 The Recording Box extension type becomes available if the Call Recording feature is activated on the QX.
 A maximum extension length is 20 digits.
 A maximum SIP Username length is 32 characters. The SIP Username can consist of lowercase and
uppercase alphabetic characters, digits and symbols.
6.1.4 Edit Extension

The Edit leads to the Extensions Management – Edit Entry page to editing an extension(s). When editing multiple
extensions, fields that cannot be edited for multiple records have Multiple values in the Edit Entry page. When
editing user and attendant extensions together, the Edit Entry page displays only common fields. Additionally,
"Select to modify fields" checkbox to submit changes of the corresponding settings (options), otherwise the
changes won’t be applied.
Figure 23: Extensions Management – Edit Entry page (for multiple edit operation)
Revision 1.2 32
6.1.5 User Extension

The following sections are available for configuration:
 General Settings
 SIP Settings
 SIP Advanced Settings
 Remote Settings
 Call Queue Settings
 Voice Mailbox Settings
 Class of Service Settings
 Credit Settings
 Licensing
General Settings
This section is used to uniquely identify an extension through below described parameters:
 Display Name – is the caller ID that will be displayed on the callee’s phone.
 Password – assigns a password to the extension. TIP: This password will be used for GUI login, for Call
Relay and remote access to voice mailbox.
 Attached Line – lists all free lines an extension can be attached to. Extension should be attached to a line
(either IP or FXS) to be able to make and receive calls. If there is no line attached to an extension, then it
is called Virtual Extension (herein VE). VEs can’t place/receive calls, but allowed to use a limited number
of QX telephony services, such as the call forwarding service or the voice mail service to store and
manage the messages from callers. Any VE can easily become a real extension after attaching a line and
vice versa. By default, all extensions on QX have lines attached already. Extensions cannot be detached
from the line if the Remote Extension feature is enabled on. To detach the extension from the line,
disable the Remote Extension service on the extension first.
 Use Kickback – enables the Kickback service on the extension for the call blind transfer scenario. When
an extension blindly transfers the call to other extension and if there is no answer from the called
extension, the call will automatically get back to the extension who initiated the transfer instead of getting
into the destination's voice mailbox or being disconnected.
 Allow Call Relay – enables the extension to be used to access the Call Relay service in the QX Auto
Attendant. It is recommended to define a proper and non-empty password when enabling this service in
order to protect it from an unauthorized access.
 GUI Login Allowed – enables GUI access (by extension name and password) for the current extension.
 3pcc/Click2Dial Access Allowed – enables the current extension to be used with applications based on
the QX's interface and QX's Click2Dial application.
 Show on Public Directory – if selected, allows to display the extension (Display Name, number) in the
Phone book (Directory) or Extension Directory of the QX.
 Use Parent Extension – allows the selected extension to be used as a child for a Parent Extension
selected from the Parent Extension drop-down list. When this done the Use Parent Extension checkbox
with the Parent Extension drop-down list will disappear for parent extension and Child Extension List link
will appear instead. For more information, see the Parent-Child Configuration.
Revision 1.2 33
Figure 24: User Extension – General Settings section
 Child Extension List – leads to the Child Extension page, where you can see the list of extensions(s)
defined as child extension(s) for the Parent extension.
Figure 25: Child Extension page
Revision 1.2 34
General Settings section of the Child Extension has the following components:
 Use Parent Extension – if deselected,
interrupts the Use Parent Extension
feature on the child extension.
 Parent Extension – select parent
extension for the child extension.
 Percentage of Total Memory – select the
memory space for the extension’s voice
mails and uploaded/recorded greetings
and blocking messages. When editing an
existing extension and decreasing the
memory space, the system will check the
present amount of voice mails in the
mailbox of the extension. If the required
memory size exceeds the size entered, Figure 26: Child Extension – General Settings section
the system will suggest either to remove
all voice messages from the extension’s
voice mailbox or to select a larger size so that to keep the existing voice messages in the mailbox.
 Enable Ringing Simulation – if selected, extra ring tones will be played to the caller before the voice mail
service gets activated (available on Virtual Extensions only), otherwise the voice mail service will be
activated immediately. The ring tones will be played during the timeout specified in the Ringing Simulation
Timeout.
 Edit Call Intercept Access List – leads to the Call Barge-In/Intercept Access List page to define extension(s)
allowed to intercept calls. This link will be renamed to Edit Call Barge-In / Intercept Access List after
activating the Barge-In feature.
 Allow other users to Barge-In to this extension – enables the Barge-In on the extension. Edit Call Barge-
In/Intercept Access List leads to the Call Barge-In/Intercept Access List page to define extensions allowed
to barge-in to the current extension calls or intercept calls. TIP: After activating Barge-In feature, the
extensions that are previously configured to intercept calls from the Call Intercept Access List page, will be
automatically redirected to the Call Barge-In/ Intercept Access List page along with the Barge-In options.
 Edit Watch Access List – leads to the Watch Access List page to define the extensions allowed to watch
calls.
Revision 1.2 35
Call Barge-In/Intercept Access List
This page is used to define a list of extensions that are capable to Barge-In/Intercept the current extension’s
calls and defines the appropriate permissions. The Call Barge-In / Intercept Access List page is available only if
the Barge-In is activated.
Figure 27: Call Barge-In/Intercept Access List

1. Click Add.
 Enter the extension number(s) allowed to Barge-In/Intercept the current extension’s calls.
 Select Barge-In, Intercept options, to allow the selected action only. The following options are available:
Listen-In, Whisper, Barge-In and Intercept.
2. Click Save, the new entry will be added to the Call Barge-In/Intercept Access List table.
Note: Barge-In/Call Intercept calls neither will be displayed in the Active Calls table on the Dashboard nor will be
registered in the Call History table.
Watch Access List
This page is used to define a list of extensions that are capable to watch the current extension calls and defines
the appropriate permissions.
Figure 28: Watch Access List

1. Click Add.
 Enter the extension number(s).
 Select the "Allow Presence Subscriptions" and "Allow Dialog Subscriptions" to allow subscriptions to the
current extension.
2. Click Save, the new entry will be added to the Watch Access List table.
Revision 1.2 36
SIP Settings
This section describes how to register the QX extension on a SIP server to receive external SIP calls.
Figure 29: SIP Settings section
 Username / DID Number – is the registration username or the DID number on the external server.
TIP: A maximum SIP Username length is 32 characters. The SIP Username can consist of lowercase and
uppercase alphabetic characters, digits and symbols.
 Password – is the registration password on the SIP server.
 SIP Server – is the address of the SIP server. It can be either an IP address, such as 192.168.74.15 or a
host name, such as sip.epygi.com. TIP: A maximum SIP Server length is 32 characters. The SIP Server
can consist of lowercase and uppercase alphabetic characters, digits and symbols.
 SIP Port – is the port number used to connect to the SIP server. TIP: If the SIP port is not specified, QX
will access the SIP server through the default 5060.
 SIP Registration Transport – is used to select SIP Transport (UDP, TCP and TLS) for the registration.
TIP: If the QX is located behind a NAT router, the TCP ports (for TCP and TLS) should be manually
configured from NAT Traversal – SIP Parameters page and opened on the NAT router accordingly.
 Registration on SIP Server – is used to register the current extension on the SIP server.
How it works: Upon receiving a SIP Invite message from an external server, the QX will look to match the called
number in the Username/DID Number field. If the ITSP does not require each DID to uniquely register on an
external SIP server, then only enter the DID number in the Username/DID Number field and keep the other fields
empty.
Revision 1.2 37
SIP Advanced Settings
This section describes how to configure advanced and specific SIP settings for QX extension.
Figure 30: SIP Advanced Settings section
 Authentication Username – enter an identification parameter. It should be provided by the SIP service
provider and can be requested for some SIP servers only. For others, the field should be left empty.
 Send Keep-alive Messages to Proxy – enable the SIP registration server accessibility to the verification
mechanism.
 Timeout – define the timeout between two attempts for the SIP registration server accessibility
verification. If no reply is received from the primary SIP server within this timeout, the Secondary SIP
server will be contacted. When the primary SIP server recovers, SIP packets will resume being sent to it.
 RTP Priority Level – select the level of priority (low, medium or high) of the RTP packets sent from the
extension. RTP packets with higher priority will be sent first in case of heavy traffic.
Revision 1.2 38
 Do Not Use SIP Old Hold Method – if selected, a new recommended method of call hold in SIP (the call
hold request is indicated with the "a=sendonly" media attribute, rather than with the IP address of
0.0.0.0) will be used. This checkbox must be enabled if the remote party does not recognize hold
requests initiated from the QX.
 Outbound Proxy – is the SIP server where all SIP requests and SIP messages are transferred to. Some
SIP servers use an outbound proxy to escape restrictions of NAT. If an outbound proxy is specified for
an extension then all SIP calls originating from that extension will go through that outbound proxy, i.e., all
requests will be sent to that outbound proxy.
 Secondary SIP Server – act as an alternative SIP registration server when the primary SIP Registration
Server becomes inaccessible. If the connection with the primary SIP server fails, the QX will automatically
start sending SIP messages to the Secondary SIP Server. It will switch back to the primary SIP server as
soon as the connection is reestablished.
 Host Address and Port – specify the host address and SIP port of the Outbound Proxy, Secondary SIP
Server and the Outbound Proxy for the Secondary SIP Server respectively. These settings are provided
by the SIP server providers and are used by QX to reach the selected SIP servers.
Remote Settings
This section describes how to configure Remote Extension settings for QX extension. This is an advanced
telephony feature that allows users to connect phone to the QX remotely. User needs to register an IP phone or
softphone on the QX by defining the QX global IP address and an appropriate Username/Password. The
registered phone can act fully as a phone connected locally to the QX, i.e. you can use all QX telephony
features, place and receive calls, access voice mails, etc.
 Enable Remote Extension –
activate the Remote Extension’s
service. TIP: The Remote
Extension service can be
enabled only for extensions
attached to the line (FXS or IP).
 Username and Password – enter
the identification parameters
used by the remote phone to
register it on the QX. TIP: The
Username and Password must
match on both QX and IP phone
for successful registration.
 Line Appearance – define a
number of simultaneous calls
supported by the remote phone.
 Enable RTP Proxy – if selected,
the incoming and outgoing RTP
streams to/from the remote IP
phone will be routed through the
QX, otherwise RTP packets will
move directly between peers.
 Fallback to local extension when Figure 31: Remote Settings section
not registered – if selected, the
incoming calls to the local extension will be forwarded to the remote IP phone only if it is registered.
Otherwise, when the remote IP phone is unregistered, incoming calls will be routed to the local extension
it is attached to.
Revision 1.2 39
 Symmetric RTP – must be selected when the remote extension is located behind the NAT router.
 Enable Hot Desking – enable the Hot Desking service on the current remote extension.
 Hot Desking Automatic Logout – is used to configure the Hot Desking functionality expiration on the
current extension. Following options are available:
 Never – the extension will never expire and will remain logged into the public phone.
 After – the extension will be automatically logged out from the public phone after a specified period of
time.
 At – the extension will be automatically logged out from the public phone at the specified moment (hour
and minute).
For information on how to configure and use Remote Extension service, please refer to the Remote Extension
Configuration on QX IP PBXs guide.
Call Queue Settings
This section describes how to configure the Call Queue service on QX extension allowing multiple incoming
calls to wait in the queue and be answered in the order they have been received (Figure 32). This service can be
used in the Receptionist Management as well.
 Enable Call Queue – enable or disable the Call Queue service.
 Call Queue Size – define the length of call queue. This is the maximum number of calls that will be
accepted into the queue and kept on hold while the extension user is on a call. If the queue is filled up
then the next incoming call will go to the extension’s Voice Mail, if enabled, or will be disconnected.
 Max Calls Presented to Extension – is used to define the maximum number of active calls on the line.
So, if the maximum call number is set to  and the extension user is in call then an incoming call will
go to the call queue, if the maximum call number is set to  and the extension user is in call then the
next incoming call’s alert will be played in the background (if the Call Waiting service is enabled on the
current extension) and the extension will hold the first call to answer the second one or they can be
joined for a call conference.
 Enable No Answer Redirect – if activated and configured, callers will be redirected to the specified
address after some time spent in the queue. The Prompt Repetition is used to define the number of
prompts to be played before redirection.
 ZeroOut Redirection – if activated and configured, callers dialing  during queue welcome message or
recurring prompt will be redirected to the specified address.
 Voice Mail – redirects the call to the extension’s voice mailbox.
 Call Type, Calling Address (identical for both Call Redirection and ZeroOut Redirection) – is used to
define the destination address the call will be redirected to. The address strictly depends on the call
type.
 Call Queue Welcome Message – play a voice message (default or custom) once when reaching the
extension's Call Queue.
 Call Queue Prompt – play a queue prompt after the Call Queue Welcome Message.
Note: The Call Forwarding if Busy and Voice Mail services will function once the call queue will be filled up. Thus,
these services will affect those calls that are left out of the queue.
Revision 1.2 40
Figure 32: Call Queue Settings section
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Voice Mailbox Settings
This section describes how to configure Voice Mailbox Settings on QX extension. By default, the voice mail
service (herein VMS) is active for all QX extensions and a certain percentage of memory space assigned to all
extensions. For user extensions, this is a shared memory to be used for the voice mails and custom voice
messages.
Figure 33: Voice Mailbox Settings section
 Disable Voice Mail – disable callers to reach to the extension’s voice mailbox to leave a message. The
extension user will still be able to access his voice mailbox and manage the existing messages as well as
manipulate with VMS to setup the personal settings (password, voice mail greeting and so on) from the
handset.
 Use Internal Voice Mail – enable the VMS for the extension and defines the QX internal storage as a
location for the voice messages.
 Voice Mail Configuration Wizard – if activated, prompts the extension user to provide personal
information while entering the voice mailbox first time. Click Deactivate to stop the Voice Mail
Configuration Wizard.
 Shared Mailbox – setup a mailbox sharing. The Edit Voice Mailbox Access List link leads to the Voice
Mailbox Access list page to define a list of extensions that are capable to access voice mailbox without
password authentication.
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 Use External Voice Mail – enable VMS for the current extension and defines a remote Voice Mail Server
to locate voice messages.
 Proxy Controlled Mailbox Type – keeps the recorded voice mails on the SIP proxy. When extension
accesses his mailbox by dialing , the call will be redirected to the voice mailbox on the proxy
server. It is recommended to select the Proxy Controlled Mailbox Type option if the Voice Mail Server is
combined with the SIP Proxy server.
 Proxy Controlled Mailbox Type – redirect the recorded voice mails to the defined remote Voice Mail
server. When extension accesses his mailbox by dialing , the call will be redirected to the remote
voice mail server. It is recommended to select the Independent Mailbox Type option if the Voice Mail
Server acts as a standalone location for the voice mails. It is required to enter the SIP URI of the Voice
Mail Server where voice mails of the current extension will be collected for both of the above described
options.
 Transport Protocol for SIP messages – select the transport protocol (UDP or TCP) for the transmission
of SIP messages.
 MS Exchange Server – keep recorded voice messages into one universal inbox.
 UM Auto Attendant URI – enter the SIP URI of the MS Exchange Server. When you access your
mailbox by dialing , the call will be redirected to the voice mailbox on the MS Exchange Server.
 UM Extension – enter the extension number that Unified Messaging will use when voice mail is
submitted to the user's MS Exchange Server mailbox.
Note:
 For information on how to configure and use MS Exchange Server, please refer to the Configuring MS
Exchange Server as External VM Server for QX IP PBX guide.
 Some internal voice mailbox services will become unavailable while choosing the Use External Voice Mail
option. Instead, the services of the external voice mail server will become available to the user. Please
consult with the external voice mail server administrator before enabling this option.
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Class of Service Settings
This section describes how to assign the defined classes to QX extensions. The Class of Service specifies the
User or Conference extensions that can use specific call routing rules to make a call. Extension not assigned to
a certain class of service can’t use a routing rule with Class of Service enabled.
Figure 34: Class of Service Settings section
Note: User and Conference extensions can be attached to a several class of services at the same time.
Credit Settings
The Calling Cost Control service allows to assign and manage credits to each specific extension for making
calls. The assigned credit would be used and controlled when making call through the specific ("payable") call
routing rules. The extensions not having credit can't use the routing rules with the Calling Rate Settings enabled.
The Credit Settings is used to set the credit amount for the QX's extension.
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Figure 35: Credit Settings section
 Available Credit – defines the credit that can be used by the selected extension. Once the Available
Credit expires, the call will be disconnected without prior notice. Placing new call(s) through the routing
rules with the Set Call Rate option enabled is not possible until the Available Credit is not updated, either
manually or automatically by the renewal date and amount.
 Periodic – is used to select one of the Renewal Date options:
 Daily – the defined Available Credit will be renewed every day.
 Weekly – the defined Available Credit will be renewed every week on the specified weekday.
 Monthly – the defined Available Credit will be renewed every month on the specified day.
 Renewal Amount – enter the renewal amount to be added to the Available Credit when the expiration
date of the Available Credit is reached. Leave the field empty, if you don’t need to renew the Available
Credit.
 Discard remainder before renewal – is used to discard the remainder of Available Credit before renewal
and set the Renewal Amount as the new Available Credit.
 Expires on – is used to manually define the expiration date for the Available Credit. After the Expiration
Date, the extension will not be able to make a new call through call routing rule(s) with the Set Call Rate
option enabled.
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Licensing
The Licensing section becomes available only if the corresponding licenses are activated. For more information
on how to configure and use these features, please refer to the Licensable Features on QX IP PBXs guide.
Figure 36: Licensing section
 Enable DCC Pro – allows to set the corresponding extension to be used by the DCC Pro application. TIP:
DCC Pro/Basic licenses can't be activated simultaneously for the same extension.
 Enable DCC Basic – allows to set the corresponding extension to be used by the DCC Basic level
application.
 Enable iQall Mobile Toggling – allows to allocate the iQall Mobile Toggling license to the extension.
Parent-Child Configuration
The Parent-Child extension configuration can be relevant for use in specific cases, when a number of QX users,
with analog or IP phones, wish to make and receive calls through the QX while staying invisible from outside.
The Parent-Child configuration is just for this case. It can create the appearance that many phones are
connected to the same extension. This feature can be used also with Epygi Hotel Console (EHC) licensable
feature for hotel rooms having many phones or with other applications where many phones are connected to
the same extension.
A list of extensions can be configured as Child for the selected Parent extension, and they will ring
simultaneously with the parent in case of incoming call to Parent extension. The Parent and any of Child
extension(s) can answer the call. The Child extensions are not visible from outside thus when placing an
outbound call from the Child extension(s) the Parent's caller ID and name would appear instead of the Child's.
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Note: The Parent/Child group configuration will apply some restrictions regarding the configuration settings and
inbound calls. See the list below:
 Child extension(s) will lose the SIP registration, the configured Basic and Caller ID Services.
 Child extension(s) will not be able to receive incoming calls directly and will ring only when the Parent
extension is dialed.
 If any of extensions in the Parent/Child group is busy, then entire group will be considered as busy,
therefore incoming call will follow busy state rules (busy forwarding, call queue, VMS, etc.) depending on
what is configured. If the "Call Waiting Service" is enabled on the Parent extension, then extensions of
Parent/Child group will receive the second call.
 If all extensions in the Parent/Child group are free (not busy) and are ringing, and any of them presses
"Reject" button (or somehow else declines the incoming call), then the entire group will be considered as
busy, therefore incoming call will follow busy state rules (busy forwarding, queue, VMS, etc.) depending
on what is configured.
6.1.6 Pickup Group Extension

The Call Pickup service allows to pick up calls ringing on a certain group of extensions by dialing Pickup Group
extension number.
Figure 37: Pickup Group – General Settings section
To configure Pickup Group extension:

1. Click the Edit Pickup Group link. The Pickup Group of Extension # table will be displayed.
2. Select the extension(s), whose calls should be allowed to pick up and click Enable.
3. Click the Go Back button.
4. Click the Edit Access List link. The Access List of Pickup Group extension page will be displayed.
5. Click Add.
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6. Enter the extension(s) allowed/denied permission to pick up the ringing calls on the specified group of
extensions.
7. Click Save, the new entry will be added to the Access List of Pickup Group extension table.
How it works: When call is ringing on another phone, you can pick up that call on your own phone by dialing the
number of the Pickup Group extension.
Note:
 The General Settings, SIP Settings, SIP Advanced Settings and Go To Codec Settings sections are the
same as for user extensions.
 When a caller not listed in the Access List calls the Pickup Group extension, password authorization (the
password of the Pickup Group extension) will be required to allow the call pickup.
 If a user dials the pickup extension when several extensions of the pickup group are ringing, the first
(oldest in time) call will be picked up.
6.1.7 Call Park Extension

The Call Park service allows to park a call (the call will be automatically placed on hold) then retrieve the parked
call from another phone by dialing the Call Park extension number.
Figure 38: Call Park – General Settings section
 Retrieve Timeout – define the timeout (in minutes) the parked call will stay active, i.e. the parked user will
remain on-hold.
 Customize push back number – if selected, then after the call park retrieve timeout expires, the hold
music stops playing to the parked party (user) and a new call is being placed towards the push back
number configured in the Customize push back number field. TIP: If the Customize push back number
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option is not selected, then after the call park retrieve timeout expires the call will be forwarded back to
the extension which parked the call.
To configure Call Park extension:
1. Click the Park Access List link, then click Add.
2. Enter the extension number(s) allowed to park the calls on the current Call Park extension.
3. Click Save, the new entry will be added to the Park Access List table.
4. Click the Retrieve Access List link, then click Add.
5. Enter the extension number(s) allowed to retrieve the parked calls from the current Call Park extension.
6. Click Save, the new entry will be added to the Retrieve Access List table.
How it works: To park the call, put the active call on hold and then dial  or the Call Park extension number.
The call will be parked. To retrieve the call, dial the Call Park extension number. For more information how to
park/retrieve calls on Epygi supported IP phones, please refer to the QX IP PBX Features on Epygi Supported
IP Phones guide.
Note:
 Any extension missing from the Park Access List won’t be able to park a call to the current call park
extension.
 When a caller not listed in the Retrieve Access List calls the Call Park extension, password authorization
(the password of the Call Park extension) will be required to allow retrieving the parked call.
6.1.8 Paging Group Extension

The Call Paging service (Figure 39) is used to page a group of extensions (phones) by forcing extensions to go
off-hook and opening one-way communication. The service is particularly used for announcements addressed
to a group of extensions. Service allows to page multiple extensions by dialing the Paging Group extension.
 Display Name – is the caller ID that will be displayed on the caller’s phone display.
 Password – is used to enter the password for the Paging Group extension.
 Show on Public Directory – if selected, allows to display the extension (Display Name, number) in the
Phone book (Directory) or Extension Directory of the QX.
 Edit Paging Group – leads to the Paging Group of Extension page to activate (enable) extensions to be
paged.
To configure Paging Group extension:
1. Click the Edit Paging Group link. The Paging Group of Extension # table will be displayed.
2. Select the extension(s), who can be paged and click Enable.
3. Click the Go Back button.
4. Click the Edit Access List link. The Access List of Paging Group extension page will be displayed.
5. Click Add and enter the extension(s) allowed or denied permission to dial the Paging Group.
6. Click Save, the new entry will be added to the Access List of Paging Group extension table.
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General Settings
Figure 39: Paging Group – General Settings section
How it works: When calling to the Paging Group extension, the call will be forwarded to the extensions listed in
the Paging Group table. The phones of the called extensions will automatically go off-hook (the phone speaker
automatically becomes activated) and the caller will be able to make announcement. Since the paging call
opens one-way communication, the called extensions will not be able to give an answer to the caller. To
terminate the paging call, caller should simply hang up. For more information how to use paging service on
Epygi supported IP phones, please refer to the QX IP PBX Features on Epygi Supported IP Phones guide.
Note:
 The SIP Settings, SIP Advanced Settings and Go To Codec Settings sections are the same as for user
extensions.
 When a caller not listed in the Access List table calls the Paging Group extension, password
authorization (the password of the Paging Group extension) will be required to start the call paging.
 Paging will not work if the called phone is in call.
 Paging service requires the phones called support automatic off-hook.
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6.1.9 Recording Box Extension

Calls recordings on QX can either be stored locally in the Recording Box or transferred to the remote server.
The Recording Box is used to store the recorded calls locally. The Recording Box can be accessible online from
Web Management or from handset by calling the corresponding Recording Box extension. With both options,
the user can play and delete the recorded calls.
Figure 40: General Settings section
Recording Box Settings
This section describes how to configure the Recording Box specific settings.
 Ask Password on Local Access – enable the password protection for local PBX callers when accessing
the Recording Box.
 Ask Password on Remote Access – enable the password protection for remote SIP or PSTN callers
when accessing the Recording Box.
 Play Welcome Message – enable or disable the welcome message played when accessing the
Recording Box.
 Maximum Recordings – the maximum number of recordings allowed to be stored in the Recording Box.
If this number is reached, some of call recordings should be deleted from the Recording Box, to free up
the space for new recordings.
 Single Recording Duration – the maximum recording duration for a single call. When the recording
duration expires, recording will be stopped while the call will stay active.
 Forward/Rewind Duration – select the timeout in seconds used to shift the recording playback from the
handset.
 Play announcement when starting recording – enable or disable the announcement played during the call
saying that the call recording starts. The call recording will start without notification if this option is
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disabled. You can upload/record a new announcement message, download the message to the PC or
restore the default one.
Figure 41: Recording Box Settings section
Recording Storage Settings
This section describes how to configure the recordings storage settings and is divided into two parts.
 Recording Storage Modes – offers the following recording storage options:
 FTP Mode – send the recordings to the FTP server and delete from the device immediately.
 Simple Local Mode – keep the recordings locally. When the local space is full or the maximum
recordings count is reached, the recording will be stopped then an event will be generated.
 Cyclic Local Mode – keep the recordings locally. The oldest recordings will be deleted if the local space
is full or the maximum recordings count is reached.
 Mixed Mode – keep recordings locally. The oldest recordings will be moved to the FTP server if the
local space is full or Maximum recording count is reached.
 FTP Settings – define the FTP server settings, if configured accordingly. Configure FTP settings as
follows:
 Use SFTP – enable SSH FTP (SFTP) support, which allows using secure FTP connection.
 Server Address – enter the FTP server IP address or hostname.
 Server Port – enter the FTP server port number.
 Username and Password – enter the authentication parameters.
 Directory on Server – define the location on the server where the recordings will be stored.
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 File Naming Scheme – define the naming scheme of the files to be uploaded to the FTP server. This
scheme helps to distinguish files among others and to avoid possible overwriting of the files. This field
may contain any distinctive text and also offers a list of variables:
 call_guid – unique GUID of the call
 recording_id – unique recording ID of the call
 caller_dispname – caller's display name
 caller_username – caller's username
 caller_fullname – caller's full name in the username@host[:port] format
 callee_dispname – called user's display name
 callee_username – called user's username
 callee_fullname – called user's full name in the username@host[:port] format
 duration – duration of the call
 time_hour – hour when the call recording started
 time_min – minute when the call recording started
 time_sec – second when the call recording started
 date_year – year when the call recording started
 date_month – month when the call recording started
 date_day – day when the call recording started
 extension – recording box extension
 hostname – QX hostname
 recording_id – unique recording ID of the call
Figure 42: Recording Storage Settings section
Any combination of listed variables can be used in the File Naming Scheme field.
Example for a file naming scheme: MyQX-$[caller_dispname]-$[duration]-$[time_hour]-$[ time_min] business.
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If Andrew is the caller, call recording was started at 14:10 and lasted 15 seconds then the files stored on the
FTP server for this Recording Box will have the name: MyQX-Andrew-15 sec-14-10-business.wav
Recording Box
The Recording Box can be accessible online from Web Management or from handset by calling the Recording
Box extension. With both options, the user can play and delete the recorded calls in the Recording Box.
Figure 43: Recording Box page
Note:
 When using Call Recording on the QX50/QX200 it is advisable to use an SD memory card to expand the
system memory.
 When using Call Recording on the QX20/QX500 it is advisable to use a micro SD memory card to
expand the system memory.
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6.1.10 Auto Attendant Extension

The Auto Attendant is an IVR system that replaces a receptionist and allows to distribute calls to the QX’s
extensions or services through prerecorded audio prompts. Remote access to the QX’s attendant is possible
through IP/PSTN/IP-PSTN calls, by dialing Attendant's SIP or PSTN number.
Note: The SIP Settings, SIP Advanced Settings and Go To Codec Settings sections are the same as for user
extensions.
General Settings
This section describes how to configure general settings of the Attendant.
Figure 44: Auto Attendant – General Settings section
 Display Name – is the caller ID that will be displayed on the phone when making call to attendant or from
attendant (e.g. when using callback service).
 Enable FAX forwarding – if selected, the system forwards the FAX messages to the selected extension if
incoming calls are routed to the Attendant and FAX tone is detected on the Attendant.
 Extension to forward – select the extension where the incoming FAX addressed to the Attendant will be
forwarded. The list contains only those extensions that have FAX support enabled. The FAX support can
be enabled from the Extension Codecs page. TIP: FAX forwarding is applicable only for incoming calls
from PSTN and SIP.
 Show on Public Directory – if selected, automatically includes the extension (Display Name, number) to
the Phone book (Directory) or Extension Directory of the QX.
 Percentage of Total Memory – defines the memory space allocated to AA extension for custom voice
messages.
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Attendant Settings
This section describes how to apply schedules and manage the attendant scenario(s) for the scheduled
periods. The following settings and options are available:
 Enable Schedule – select and apply preconfigured schedule to the attendant. The applied schedule
allows to configure different scenarios for scheduled periods (working hours, non-working hours and
holidays).
 Attendant Scenario – select between the auto attendant scenarios. The following scenarios are available:
 Standard scenario – available and active on the 00 attendant and newly created attendant extensions
by default.
 VXML scenario – allows to upload custom scenario file in VXML format.
 Custom scenario – allows to configure the custom scenario with the embedded scenario builder.
 ACD scenario – allows to activate special scenario for ACD agents.
Note: Enable the Schedule option and apply a schedule to the attendant, to be able to select a scenario for
each schedule period.
 Authorized Phones – leads to the Authorized Phones page. If the external SIP or PSTN caller added to
the Authorized Phones, he/she allowed to access the attendant services bypassing the authorization
procedure and use the Callback service as well.
Figure 45: Attendant Settings section
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Attendant Scenario
This section is used to configure the selected scenario.
Standard scenario
Following options will be available by selecting the Standard scenario:

 Pass Dialed Digits
through Call Routing – if
selected, sends the
dialed numbers to the
Call Routing Table.
 Enable No Input Redirect
– if activated and
configured, callers will
be redirected to the
specified address in
case if no action by
caller on the Recurring
Attendant Prompt(s).
Prompt Repetition is
used to define the
number of prompts to
be played before
redirection.
 Enable ZeroOut Redirect
– if activated and
configured, callers
dialing  during
welcome message or
recurring prompt will be
redirected to the
specified address.
Note: The routing patterns in
the Call Routing Table starting
with digit  will not work for
incoming calls to attendant if
both the ZeroOut and Pass
Dialed Digits through Call
Routing options are enabled.
The ZeroOut feature has a
higher priority. If enabled, the
system will redirect calls to
the specified destination. As a
result, calls prefixed with  Figure 46: Auto Attendant – Standard scenario
will never reach call routing.
Revision 1.2 57
 Attendant Welcome Message – allows to enable/disable and customize the Attendant welcome
message.
 Recurring Attendant Prompt – allows to customize the Attendant active recurring prompt (played after the
Welcome Message and then periodically repeated while being in the Attendant).
VXML scenario
The VXML scenario allows to upload custom scenario file and voice messages. Following options are available:
 Upload VXML scenario file – is used to upload a new scenario file. TIP: The uploaded file needs to be in
EpygiXML format and is restricted to 20 KB file size.
Figure 47: Auto Attendant – VXML scenario
 Upload VXML Scenario Voice Messages – leads to the Upload Custom Scenario Voice Messages page to
manage voice messages used in scenario. TIP: It is allowed to upload all voice messages at once. To do
this, create an archive file of the (*.tar.gz) type containing all the necessary files and upload it from the
Upload VXML Scenario Voice Messages page.
 View/Download VXML scenario – view or download the scenario file.
 Remove VXML scenario – remove the custom scenario file.
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Custom scenario
The Custom scenario allows to use the customized scenario builder.

 Create scenario – leads to the Edit Scenario page to create a new scenario. TIP: The Create scenario link
will be renamed into the Edit Scenario after creating a scenario.
 Import/Export scenario – leads to the Import/Export Scenario page to import/export the scenario file.
Figure 48: Auto Attendant – Custom scenario
 Import – is used to upload a scenario file.

 Export scenario – The Download scenario link is used to download the custom scenario. This link
appears only if there is a custom scenario available.
 Remove scenario – is used to remove the current scenario.
 View/Download VXML scenario – is used to view and download the scenario script in the VXML file
format.
Edit Scenario
The Edit Scenario page is used to modify the custom scenario through the scenario builder. This page consists
of two main sections: Main Menu and Submenus. All incoming calls to the Auto Attendant will be placed to the
Main Menu first. The Submenus are the supplementary menus which can be called from other menus. Both the
Main Menu and all Submenus can call each other. This allows to have several levels for the attendant scenario.
There are no limitations on the depth and nesting levels of menus.
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The Main Menu consists of the following sections:

 Welcome Message – is used to play a welcome message (default or custom) once when entering the
Main Menu. TIP: If the Welcome Message is not specified, then the Welcome Message for Standard
scenario will be played.
 Enable Welcome Message – activate or deactivate the welcome message (default or custom).
 Delay after message – define the break between the welcome message and recurring prompt.
Figure 49: Create scenario – Main Menu page
 Recurring Prompt – play a recurring prompt (default or custom) after the Welcome Message. TIP: If the
Recurring Prompt is not specified, then the Recurring Prompt for Standard scenario will be played.
 Play Count – displays the repetition count of the Recurring message.
 Interval – define the silence duration (in seconds) between consecutively played Recurring messages.
 User Input Options – table consists of the following components:
 Option – select the user input and configure it with some announcement and action to be taken.
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 Any input other than preconfigured in the list – configure the action taken when the caller makes a
selection other than options listed in the User Input table. If it is configured to No Action, then the timer
for No input will reset and it will be counting the No input time again.
 No input – configure the action taken when the caller doesn't enter anything during the certain period.
The No input timeout is composed of [Welcome Message duration] + Delay after message + [Recurring
Prompt duration] * Play Count + Play Count * Interval. If there is no input during that time, the action
specified for No input will take effect.
 Dial Timeout – define the timeout after user has completed the dialing before the call processing start.
The timer will start after the last digit or symbol is entered. Press the  sign to process the call
immediately.
 Incorrect number handling – leads to the Edit Incorrect Number Handling page to configure the action
taken when the user has selected a destination that resulted in a failed call, such as an invalid extension
number.
Note:
 The Incorrect number handling will be activated only if either an attempt was made to call to a non-
existing extension or to a number not matching with any "Destination Number Pattern" in the Call
Routing Table.
 The Incorrect number handling will be activated only if the call comes to the Auto Attendant from SIP,
PSTN or IP-PSTN side.
Input Option Configuration
 Add – leads to the Add Option page to configure previously unspecified inputs.
 Edit – leads to the Edit Option page to modify the actions of Input options.
The Add/Edit Option page offers the following components (Figure 50):
 Option – is used to select the user input and configure it with some announcement and action to be
taken (applicable only for User Input). The following input options are available in the list to configure the
Customized Scenario:
 Digits (from  to )
 Signs ( and )
 Announcement – is used to upload/record an announcement message for the selected User Input
option. As soon as the caller presses the preconfigured digit, the message will be played and only then
the action configured for that User Input option will be activated.
 Action – is used to configure the action which will be taken after the Announcement message
 No Action – the system continues playing the Recurring Prompt.
 Go to the following menu – leads to the specified submenu and take actions defined in that submenu.
The drop-down list allows to select a previously created submenu or create a new one by choosing the
Create New Submenu option.
 Call to the following extension – calls to the selected extension.
 Call to the following number – calls to the specified destination via the Call Routing Table.
 Call to the number dialed – sends the user inputs to the Call Routing Table (available only in case when
the "Any input other than" in the list above input is edited).
 Invoke Extensions Directory – connects the caller to the Extensions Directory.
 Terminate the call – disconnects the call.
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Figure 50: Main Menu – Add Option page
Submenus
Submenus are supplementary menus accessible from the Main Menu. Submenus allow to configure multilevel
attendant scenarios. The Submenu consists of the same sections and configuration options as the Main Menu.
The Submenus page consists of the following components:
 Add – leads to the Edit Scenario – Add menu page to define a new Menu (submenu).
 Edit – leads to the Edit Scenario page to configure the newly created submenu scenario’s settings.
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ACD scenario
The ACD scenario allows to use a special scenario for ACD agents. This scenario allows ACD agents to
change/update their status by dialing to Attendant and following voice prompts.
Note: This selection is only available if the ACD feature is previously activated.
Ringing Announcement
The Ringing Announcement section is used to play an optional custom voice message to callers instead of ring-
back tones when making calls through the auto attendant.
Note: The Attendant Ringing Announcement is played to SIP-to-Extension and PSTN-to-Extension calls only.
The announcement can also be played to SIP-Attendant-SIP and PSTN-Attendant-SIP calls if they are made by
a call routing rule with the RTP proxy enabled.
Figure 51: Auto Attendant – Ringing Announcement section
 Enable Ringing Announcement – enables/disables the Auto Attendant optional announcement message.
If selected but no custom announcement message is uploaded, the system default message will be
played to callers.
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6.1.11 Extension Codecs

To establish an IP voice communication, call participants have to use the same codec. When establishing a
communication line, this codec is negotiated. If the caller does not find an appropriate codec, the
communication does not take place. To allow communication with all IP callers, it is helpful to support as many
codecs as possible. In this case, all codecs that the system offers should be enabled in the Codecs table. On
the other hand, some codecs require quite a high transfer rate of up to 64 kbit/s. If you definitely do not want to
use these codecs, make sure they are disabled in the Codecs table. The Codecs table lists the voice and video
codecs supported by the QX. Select an entry to manage it.
The enabled codecs participate in codec negotiation at the call setup. The order of the enabled codecs is very
important. A codec placed at the top of the table is used as the preferred codec. When establishing a call, the
system will try this codec first. If the remote party does not support the preferred codec, the following codecs
will be tried out strictly in the order given in the Codecs table.
Figure 52: Extension Codecs list
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 Enable/Disable – is used to enable/disable the selected codec. Disabled codecs do not participate in the
codec negotiation, i.e. they will never be used for call setup. At least one codec must be enabled,
otherwise voice communication with an extension/attendant/conference will be impossible.
 Move Up/Down – moves the selected codec one level up/down for increasing/decreasing the codec's
priority.
 Make preferred – moves the selected codec to the top of the table, setting its priority to the highest.
Clicking Make preferred for a disabled codec will first enable the codec and then move it to the top.
 Out of Band DTMF Transport – enables the DTMF code transmission in parallel with the voice stream.
Destination received the DTMF code will play it locally if it supports the feature too. This allows to avoid
from DTMFs loss in case of heavy traffic. The feature is valuable for all codecs but it is especially
recommended for low bit rate codecs, such as G.729, G.726/16, etc.
 Enable T.38 FAX – enables the T.38 codec support of FAX transmission for incoming unified FAX
messages (fax to mailbox) and remote IP devices connected to Epygi unit via routing rules which using
the target extension user settings (UES).
 Enable Pass Through FAX – enables the G.711 codec support for incoming unified FAX messages (fax to
mailbox) and IP devices connected to the attached IP line.
If both of the above checkboxes are enabled, the T.38 codec will be used as a preferred codec for FAX
transmission. If it is not supported by the peer, the G.711 codec will be used instead. For virtual
extensions, the incoming FAX can only be stored in the extension's voice mailbox. To allow FAX to be
stored in the voice mailbox, the extension's user should not answer the incoming calls, so that they can
be forwarded to the voice mailbox. TIP: If the Enable T.38 FAX and Enable Pass Through FAX
checkboxes are disabled, no FAX transmission to the peer's voice mailbox will be possible.
 Enable Pass Through Modem – enables the modem tone detection and the G.711 codec support for the
data transmission from/to the modem attached to the line. During data transmission, the Silence
Suppression and Echo Cancellation are automatically disabled on the line. TIP: If the user extension or
attendant is intended to accept modem connections, disable the Enable T.38 FAX checkbox to allow the
system to identify the modem tones correctly, otherwise the modem connection may fail.
 Force Self Codecs Preference for Inbound Calls – allows to use your own preferred codecs (if available on
both peers).
 Secure RTP Settings – are used to configure secure voice over IP communication on the QX.
 SRTP Policy – is used to select the secure IP connection policy. The following options are available:
 Make and accept only secure calls – only the secure calls will be generated and accepted.
 Make and accept only unsecure calls – only the unsecure calls will be generated and accepted.
 Try to establish secure calls, accept anything – system will try first to establish secure call, but will fall
back to unsecure call if party doesn't accept secure calls. Both secure and unsecure incoming calls will
be accepted, as requested by remote party, with the preference given to establishing secure call.
 Make unsecure calls, accept anything – system will establish unsecure outgoing calls, but both secure
and unsecure incoming calls will be accepted as requested by remote party.
Note:
 Pay attention when configuring Auto Attendant codecs as they are used by virtual extensions for
redirecting the incoming calls.
 Video codecs are available for FXS lines also only if remote extensions are configured and used for those
lines.
 For bandwidth used by secure calls, see Needed Bandwidth for IP Calls.
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6.1.12 Bulk Import

Extension Template Management and Bulk User Extensions Importer tools are used to create and update
multiple user type extensions on the QX.
The Extension Template Management tool is for configuring the common settings, such as SIP server name,
SIP port, etc. for the extensions, while the Bulk User Extensions Importer tool for configuring the specific
settings, such as Display Name, Extension Password, etc.
For information on how to configure and use Bulk Import service, please refer to the Extensions Bulk Import on
QX IP PBXs guide.
6.2 Dialing Directories

QX provides different services allowing PBX extensions and external callers to dial the desired destinations in a
simpler way. These services are known as Dialing Directories:
 Dial by Name – allows dialing the desired extension by simply spelling the extension’s User name on the
phone keypad.
 Global Speed Dial – allows dialing the desired destination by using a preconfigured speed dial code
(shortcut number).
 Phone Book – allows to dial the desired contact by using the contact’s name from the Local Directory on
the phone.
For information on how to configure and use Dialing Directories, please refer to the Dialing Directories on QX IP
PBXs guide.
6.3 Conferences
The Epygi's conferencing is composed of the following two licensable features:
 Audio conference feature activated by installing the Audio Conference license key.
 Video conference feature activated by installing the Video Conference Server license key.
For information on how to configure and use Audio-Video conferences, please refer to the Audio-Video
Conferencing on QX IP PBXs guide.
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6.4 Recordings
The Universal Extension Recordings is used to define the voice messages universal for all extensions on QX. The
defined messages become applicable by default to all extensions on QX.
Figure 53: Universal Extension Recordings page
The Universal Extension Recordings page consists of the following components:

 Upload – is used to upload a custom message.
 Download and Remove – is used to download and/or remove the uploaded custom message.
 Percentage of System Memory – defines the memory space for universal extension recordings.
The following universal messages are available:
 Hold Music – played to the held user. Click Edit to select the way custom hold music will be provided.
 Voice Mail Regular Greeting – played when a caller reaches to the extension’s voice mailbox.
 Voice Mail Out-of-Office Greeting – played when a caller reaches to the extension’s voice mailbox if the
Out-of-office greeting is enabled.
 Incoming Call Blocking – played when a blocked user calls the extension.
 Outgoing Call Blocking – played when the extension dials a blocked destination.
 Call Queue Welcome Message – played when a caller joins the extension’s call queue.
 Call Queue Prompt – played when a caller is being held in the queue.
 Alarm Message – played when an alarm rings.
 Find me/Follow me Welcome Message – played when a user calls the extension with enabled FM/FM
service.
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Universal Extension Recordings – Hold music
The Edit Universal Extension Recordings – Hold music page is used to define the way the custom hold music
will be provided.
 Default music – enables the default music.
 File – is used to upload the custom hold music.
 RTP Channel – is used to stream the hold music through the selected RTP Channel.
 Audio Line In – is used to stream the hold music from external audio source (PC, smartphone, etc.)
through Audio Line In.
6.5 Receptionist Management

The receptionist service on the QX offers a variety of services to manipulate with multiple calls, to keep the calls
in the queue with the perspective to be answered by the receptionist and finally to be forwarded to the
corresponding destination, if needed.
For information on how to configure and use the Receptionist service, please refer to the Receptionist Service
on QX IP PBXs guide.
6.6 ACD Management

The Automatic Call Distribution is an optional feature and can be activated with a feature key. Epygi's Automatic
Call Distribution (herein ACD) feature is a complete solution for today’s call centers. ACD designed to receive
and queue high-volume inbound calls, then distribute queued calls to the available agents in call center.
Epygi's ACD Console (herein EAC) is a web application designed to support call center agents monitoring ACD
activity and performance on the QX. EAC store and format the data and produce real-time information and
statistical reports on ACD activity.
For information on how to configure and use ACD and EAC, please refer to the ACD and EAC - User Guide.
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6.7 Authorized Phones

The Authorized Phones is used to create the list of trusted external users allowed to access the QX Auto
Attendant services without authentication.
To add a new entry:
1. Click Add. The Authorized Phones – Add Entry page will be opened.
2. Select the "Enable" checkbox to activate service for the created entry.
3. Enter the caller's SIP address or PSTN number.
4. Select the Login Extension. When calling the QX’s Auto Attendant, a trusted user will automatically be
logged in as the selected extension, i.e., the extension number and password will be automatically
submitted by the system and the trusted user will directly access to the Auto Attendant services. The
SIP settings of the logged in extension will be used for making IP calls.
Figure 54: Authorized Phones – Add Entry page
5. Select the Automatically Enter Call Relay Menu checkbox. If selected allows direct access for the trusted
user to Auto Attendant Call Relay menu. If not selected, a trusted caller will be directed to the Auto
Attendant's main menu, but still will be able to reach to the Remote Access (Voice Mailbox of the
specified extension) and Call Relay services without authentication.
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6. Configure Callback Settings (optional).

 Select Enable Callback checkbox to allow the specified caller to use the Callback service.
 Specify the Call Back Destination. TIP: If the Callback Destination is left empty, the trusted caller
address will be implied as a Callback destination.
 Define Callback Response Delay (in seconds) before the Callback will be started.
How it works: When the trusted user calls the Auto Attendant, he/she will be able to use QX services as if a
PBX extension. If the CallBack service is activated the trusted user will get a call back from Auto Attendant.
Note:
 Authorized Phones will only work when the trusted caller connects to the Auto Attendant running the
Standard scenario configured.
 For more information how to configure and use Callback service, please refer to the Callback Service on
QX IP PBXs guide.
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7 Interfaces Menu
Figure 55: Interfaces Menu overview
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7.1 IP Lines
The IP Lines table lists all IP lines available on QX with specific details for each:
 Reboot – is used to reboot selected IP phone(s).
 Show disabled IP lines/Hide disabled IP lines – are used to show or hide the IP lines not activated with a
feature key.
 Enable/Disable OpenVPN – is used to enable or disable providing configuration file through OpenVPN.
Figure 56: IP Lines page
 IP Line – shows all IP lines available on the QX. Click an IP line to go the IP Line Settings page
 Attached Extension – shows the QX extension attached to the IP line. TIP: "None" is displayed if there is
no extension attached to that line.
 Click the Admin Settings icon to go the extension’s admin settings.
 Click the User Settings icon to go the extension’s user settings.
 State – shows whether the IP line is Disabled, Configured or Free.
 Details – shows the settings for the IP phone configured on the corresponding line, such as the phone
model, MAC address, used template and the authorization credentials.
 Actions – The following actions are available to manage the IP phone:
 MPK – leads to the Programmable Keys Configuration page of the phone.
 Reboot – is used to reboot the corresponding IP phone.
 Restart – is used to restart FXS Gateway (QXFXS24 or QuadroM FXS26) attached to the line.
 Web – leads to the Web configuration page of the corresponding IP phone. TIP: This link only works
from the LAN side of the QX, i.e. when the QX’s GUI is accessed from a PC located in the QX’s LAN. If
you wish to connect the GUI interface for SIP phone through the WAN, an appropriate Incoming
Traffic/Port Forwarding filtering rule should be added on the QX.
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IP Line Settings – IP Line # page (Figure 57) is used to configure the IP Line with a phone (manual or automatic
configuration).
 Inactive – if selected, changes the IP line state from Configured to Free.
 IP Phone – if selected, activates the IP line to configure with the IP phone as follows (Figure 57):
 Phone Model – select the IP phone model. Select Other if the phone model is not in the list or the
phone should be configured as a Remote Extension.
 MAC Address – enter the MAC Address of the phone.
 Line Appearance – define the number of simultaneous calls supported by the IP phone.
 Username and Password – define the authentication parameters to register the IP phone on the QX.
TIP: Set the same Username and Password as SIP registrar, SIP proxy, SIP authentication values on
the IP phone for successful registration.
 Transport – select the transport protocol for SIP messages – UDP, TCP or TLS. For TLS, you may
activate the TLS Certificates update mechanism from an IP Phone to obtain the latest certificate
generated by the QX.
 Use Template – select a preconfigured custom template for the IP phone. When the "Use default" is
selected in this list, the template selected on the IP Line Settings page will be used.
 Use Session Timer – enable the SIP session timer for the corresponding IP line. This option allows both
user agents and proxies to check and determine if the SIP session is still active.
 Symmetric RTP – must be selected when the IP phone attached to the IP Line is located behind the
NAT router.
 Use OpenVPN Settings – select this option to auto configure phone using the OpenVPN settings. The
OpenVPN service for auto configuration is available on majority of Epygi Supported IP phones.
 OpenVPN client configuration – is used to select and download OpenVPN client configuration file for the
IP phone attached to the IP Line. TIP: This option is NOT used to apply the OpenVPN configuration on
the phone.
The Hot Desking section is used to enable and configure the Hot Desking service on the IP Line as follows:
 Enable Hot Desking – enable the Hot Desking on the corresponding IP line.
 Hot Desking Automatic Logout – with this option enabled, QX will control the extension login timeout.
Once the predefined expiration time arrives, the currently logged in extension will automatically log out
and make available the public phone for other extensions. The following options are available:
 Never – the Hot Desking will never expire for the extension.
 After – extension will automatically log out from the public phone after the defined period.
 At – extension will automatically log out from the public phone at the defined moment (hour and
minute).
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Figure 57: IP Line Settings – Edit page
Hot Desking
Hot Desking originates from the definition of being the temporary physical occupant of a work station or surface
by a particular employee.
If QX has limited number of IP phones connected, but much more users wishing to make and receive calls
through the QX, some of the connected phones can be announced as public. Public phones have no static
owners; they are just connected to the IP lines. Each user that accesses the public phone should first login with
personal settings, such as the extension’s number and password of previously configured and dedicated him
virtual extension.
The Hot Desking service is used to organize the user login/logout on the public phones. Each user should have
his/her own virtual extension. The virtual extensions can be configured as needed to use all the available
Revision 1.2 74
supplementary PBX features when the user will log in from the phone with that extension. The Hot Desking
option should be enabled on the corresponding analog or IP Phones from the IP Lines or FXS Lines page
accordingly.
To access the public phone:
1. Dial  to login.
2. Enter the extension number and press .
3. Enter the extension password and press .
After successful login, the phone becomes a full featured phone connected to the QX. You can place and
receive calls and use all supplementary PBX services of the QX.
When having finished using the phone, logout.
1. Dial  to logout.
2. Enter the password of the current logged in extension and press .
When logged out, the public phone becomes available for other users.
For information on how to configure and use Hot Desking service, please refer to the Hot Desking Service on
QX IP PBXs guide.
7.1.2 IP Line Settings

The IP Line Settings is used to control the basic settings for configuring IP phones:
Figure 58: IP Line Settings page
 Enable PnP for IP lines – activate the PnP option for phones configuring with the QX. The PnP allows
Epygi supported IP phones to be automatically configured without any manual intervention in the QX and
phone settings. If selected, then connect the phone to the QX and factory reset the phone. After clean
boot-up of the phone, the QX will detect the phone settings, automatically generate the phone specific
configuration file and upload it. The phone will be then configured on the first Free IP line.
 Enable Firmware Version Control – control and manage the firmware version running on the IP Phone
configured with the QX. If the phone is running an old firmware version the system will check, then
automatically download and install the updated firmware version according to the IP phones firmware
version recommendations which are specific to each subsequent firmware version running on the QX.
 Configure IP phones from – select the network interface on the QX, where the IP phones should be
connected to. Besides LAN and WAN, this list also includes the VLAN interface if available.
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 Enable VLAN Tagging – is used to enable/disable setting the VLAN ID and priority for IP phones.
TIP: The provided IP address will always be from VLAN network. This option is enabled by default.
 Phones Default Template – select the IP phone template that will be used as default for IP Lines.
Note:
 The PnP service is available on the majority of the Epygi supported IP phones.
 Currently the Firmware Version Control service is applicable for the Mitel, Mitel (Aastra), snom and Yealink
phones.
 For QX2000 and QX3000 the Configure IP phones from list appears only if VLAN is configured.
Supported IP Phones
Below is the list of IP phones that are officially supported by Epygi and can be configured to work with the QX
using both Plug and Play (PnP) or auto configuration options.
Vendor Model SW/FW Version PnP Support

Akuvox R15(P) 15.0.5.235 Yes
Akuvox SP-R53(P) 53.0.6.115 Yes
Alcatel IP2015 (IP15) 1.0.7A-0 No
Alcatel Temporis IP100 1.0.6A-0 No
Alcatel Temporis IP150 1.0.6A-0 No
Alcatel Temporis IP200 13.60.0.89 Yes
Alcatel Temporis IP300 1.0.7B-0 No
Alcatel Temporis IP700G 1.0.7A-0 No
AudioCodes 310HD 1.6.0_build_37 No
AudioCodes 320HD 1.6.0_build_37 No
Cisco SPA303 7.4.9c Yes
Cisco SPA501G 7.4.9c Yes
Cisco SPA509G 7.4.9c Yes
Cisco SPA525G2 7.4.9c Yes
Fanvil C58/C58P 2.3.233.129 No
Fanvil C62/C62P 2.3.235.128 No
Fanvil C400 11.20.12.2.B No
Fanvil C600 11.20.12.2.B No
Fanvil F52/F52P 2.3.123.78 No
Fanvil H2/H2S 2.0.2.2776 Yes
Fanvil H3 2.0.2.2770 Yes
Fanvil H5 2.0.2.2770 Yes
Fanvil X3/X3P 1.3.511.1821 Yes
Fanvil X3S/X3G 2.0.3.3049 Yes
Fanvil X4/X4G/X4S 2.0.2.2830 Yes
Fanvil X5/X5G 1.3.511.1821 Yes
Fanvil X5S R0.7.0.1 Yes
Fanvil X6 R0.5.3 Yes
Grandstream GXP1100 1.0.8.6 Yes
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Grandstream GXP1615/1610 1.0.4.55 Yes
Grandstream GXP1625/1620 1.0.4.55 Yes
Grandstream GXP1760 1.0.0.48 No
Grandstream GXP1782/1780 1.0.0.48 No
Grandstream GXV3140 1.0.7.80 Yes
Htek UC924 2.0.4.2.24 No
Htek UC926 2.0.4.2.24 No
Mitel (Aastra) 6730 3.3.1.4305-SIP Yes
Mitel 6863 4.2.0.2023-SIP Yes
Mitel 6865 4.2.0.2023-SIP Yes
Mitel 6867 4.2.0.2023-SIP Yes
Mitel 6869 4.2.0.2023-SIP Yes
Panasonic KX-HDV130 03.004 Yes
Panasonic KX-HDV130NE, KX-HDV130X 06.101 Yes
Panasonic KX-HDV230 03.004 Yes
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Panasonic KX-HDV230NE, KX-HDV230X 06.101 Yes
Panasonic KX-TGP550T04 12.17 No
Panasonic KX-UT123 (NE/RU/X) 01.302 No
Panasonic KX-UT136 (NE/RU/X) 01.302 No
Polycom SoundPoint IP 330 3.3.5.0247 Yes
Polycom SoundStation IP 5000 3.3.5.0247 Yes
Polycom SoundStation IP 6000 3.3.5.0247 Yes
Polycom VVX 300/310 4.1.7.1210 Yes
Polycom VVX 400/410 4.1.7.1210 No
Polycom VVX 500 4.1.7.1210 No
Polycom VVX 600 4.1.7.1210 Yes
Polycom VVX 1500 3.3.5.0247 Yes
QOSIP Q7104/Q7204 1.0.3.98 No
snom 300 8.4.35 Yes
snom 320 8.4.35 Yes
snom 360 8.4.35 Yes
snom 370 8.7.5.35 Yes
snom 720 8.9.3.60 Yes
snom 760 8.9.3.60 Yes
snom 821 8.7.5.35 Yes
snom 870 8.7.5.35 Yes
snom D345 8.9.3.60 Yes
snom D375 8.9.3.60 Yes
snom D710/710 8.9.3.60 Yes
snom D715/715 8.9.3.60 Yes
snom D725 8.9.3.60 Yes
snom D745 8.9.3.60 Yes
snom D765 8.9.3.60 Yes
snom m9 9.4.7 Yes
snom MeetingPoint 8.7.5.35 Yes
snom M700 (M85/M65/M25) 03.24.0007 Yes
Spectralink KIRK Wireless Server 300 PCS14C_ No
Spectralink KIRK Wireless Server 6000 PCS14C_ No
VTech ErisStation VCS754 1.1.4.0-0 No
VTech ErisTerminal VSP600 (VSP601) 1.1.4.1-0 No
VTech ErisTerminal VSP715 1.1.4.0-0 No
VTech ErisTerminal VSP726 2.0.3.2-0 Yes
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VTech ErisTerminal VSP736 2.0.3.2-0 Yes
Yealink CP860 37.80.0.30 Yes
Yealink SIP-T19P 31.72.0.1 Yes
Yealink SIP-T19P E2 53.81.0.25 Yes
Yealink SIP-T21P E2 52.81.0.25 Yes
Yealink SIP-T23G(P) 44.81.0.25 Yes
Yealink SIP-T27G 69.81.0.25 Yes
Yealink SIP-T41S 66.81.0.25 Yes
Yealink SIP VP-T49G 51.80.0.100 Yes
Yealink SIP-T56A 58.80.0.25 Yes
Yealink SIP-T58A/V 58.80.0.25 Yes
Yealink VP-530 23.70.0.40 Yes
Yealink W52P 25.30.0.20 Yes
Table 1: Supported IP Phones
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7.1.3 Programmable Keys Configuration

The Programmable Keys Configuration is used to assign functions to the programmable keys of the IP phone.
The design of this page depends on the IP phone model. However, regardless the IP phone model, this page
contains a number of the programmable keys and Function list assigned to each of them.
Figure 59: Programmable Keys Configuration page
The following options are available in the Function list:

 Preconfigured – this option will not change anything for the key functionality. Actually, it will keep
previously configured function for that key.
 None – eliminates any functionality for the key. Actually, it disables the key.
 IP Line – allows to assign a key to the corresponding IP line. Pressing the key will provide dial tone for
making calls. The key will flash when a call is ringing to that line. The key illuminates green when the IP
line is busy with a call. TIP: Based on the phone model, the status of the BLF key and the status of the
IP Line will vary.
 Watch Ext. # – allows to watch the extension and intercept calls addressed to that extension.
 Call Park Ext. # – allows to watch the parked calls on the corresponding Call Park extension and retrieve
the parked calls.
 Shared Vmail Ext. # – allows to watch and access to the Shared Voice Mailbox.
 Schedule # – allows to watch and update the state for the specific schedule.
 Vmail – allows to access to the voice mailbox of the extension.
 DND – allows to activate/deactivate the Do Not Disturb service on the extension.
 CallFwd – allows to configure/toggle (activate/deactivate) Unconditional Call Forwarding on the
extension.
 AutoReDI – auto redials the last dialed number.
 CallBack – calls back to the last caller.
 LineInfo – plays information about the IP line.
 CallBlk – blocks the last caller.
 Record – allows to start the call recording (in case if the Manual mode for call recording is configured in
the Call Recording Settings).
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 ACD Login/Logout – allows to login/logout the corresponding ACD agent to/from all queues he/she is
involved in.
 LDAP – allows to retrieve contacts from 3-rd party LDAP server.
 URL – are basically HTTP GET Requests (often XML over HTTP) that allow the phone to interact with
web server applications. It can be used to retrieve various data from the web server.
Note: The system will ask a conformation to remotely reboot the IP phone to save changes. It is recommended
to reboot the IP phone after configuration changes on this page to make the new configuration effective on the
IP phone.
7.1.4 IP Phone Templates

Manage IP Phone Templates is used to create custom templates for the IP Phones. The templates contain a set
of configuration settings that are applied to the IP phone once it is registered on the QX. With the custom
templates, the most popular configuration settings may be adjusted accordingly. The saved custom templates
can be then configured from the IP Lines page to be used on the particular IP phone.
Figure 60: Manage IP Phone Templates page
 Add – leads to the Add Entry page to create an IP phone template as follows:
 Template Name – define the template name. This name will be visible in the Edit IP Line Settings page
when defining the template for the IP phone.
 Edit – leads to the Manage IP Phone Templates – Edit Entry page to adjust the advanced settings for
different IP phone vendors and assigned functions to the programmable keys for each phone model.
You are allowed to manage the settings for group of IP phones at once.
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7.1.5 IP Phones Logo

The IP Phones Logo is used to upload a custom logo for the IP phone. The uploaded custom logo will be visible
on the display of the IP phone.
To upload a custom logo:
1. Click the Choose File button and browse for a logo file.
2. Select the Enable Logo option.
3. Click Save to apply changes.
Figure 61: IP Phone Logo page
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7.1.6 FXS Gateways

FXS Gateway Management is used to automatically configure QXFXS24 Gateway with QX IP PBX. QXFXS24 is
an analog VoIP Gateway that allows connecting analog phones to a VoIP network. The device can be used with
QX IP PBX to emulate FXS ports. The FXS Gateway Management table lists all configured Gateways:
 Restart – is used to restart the selected FXS Gateway(s).
 Reboot – is used to reboot the selected FXS Gateway(s).
 Add – leads to the FXS Gateway Configuration Wizard to define a new FXS Gateway.
 FXS Gateway Model section allows the following to be configured:
Figure 62: FXS Gateway Model section
 Gateway Model – is used to select the FXS Gateway model from the list.
 MAC Address – enter the MAC Address of the FXS Gateway.
 Line Mapping – Add Entry section is used to assign each FXS line to IP line. System will automatically
assign the provided FXS lines to the first available IP lines on the QX. Line mapping can be manually
adjusted. FXS lines can be assigned only to inactive IP lines on the QX. If there are no enough free IP
lines, you should first deactivate the IP line from the IP Line Settings page to use it in the FXS Gateway
Configuration Wizard.
 Use OpenVPN Settings – select this option to auto configure QXFXS24 Gateway using the OpenVPN
settings.
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Figure 63: Line Mapping section
 Summary section displays all configured settings for the FXS Gateway.
Note: FXS Gateway (mapped IP lines) will be added in the IP Lines page after successful configuration. The
corresponding routing rules will be added to the Call Routing table of the FXS Gateway.
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7.2 FXS Lines

7.2.1 FXS (On-board) Line Settings
The FXS (On-board) Line Settings are used to configure on-board FXS lines, define the caller ID detection type,
configure remote party disconnect indication and select the ringer type on each of them. On-board FXS lines
are available on QX50 and QX200.
Click the hyperlinked FXS # to open the Line Settings page to configure specific settings for the selected line
(Figure 64).
Figure 64: FXS Line Settings page
 Caller ID Type – is used to send the calling party's information to the phone attached to the selected line:
 No Caller ID
 FSK, send prior to the first ring
 FSK, send between the first and second ring
 FSK, send both prior to a ring and between the first and second ring
 DTMF, send prior to the first ring
 DTMF, send between the first and the second ring
 Combined, send both DTMF prior to the first ring and FSK between the first and the second rings.
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Note: The caller ID detection method is different for various types of phones and can be found in the phone
manual.
 Enable off-hook Caller ID – is used to enable Caller ID transmission to the phone in the off-hook state
attached to a certain line. Service is applicable to the phones supporting the Call Waiting Caller ID
feature.
Remote Party Disconnect Indication parameters are used to configure the private PBX attached to the QX FXS
port.
 Enable Busy Tone Indication – is used to enable a busy tone transmission to the FXS port when the
remote party being called is disconnected.
 Busy Tone Duration – is used to select the period (in seconds) when a busy tone will be transmitted to
the FXS port.
 Enable Power Disconnect Indication – is used to enable the power cycling on the FXS line when the
remote party being called is disconnected. Power Disconnect is applied after the busy tone transmission
on the FXS line.
 Disconnect Duration – is used to select the period (in milliseconds) when the FXS line power will be
down.
 Ringer Type – is used to select the frequency of the ringer supported by the phone attached to the line.
Information can be found on the phone enclosure or in the phone's manual. Problems with the ringer
might occur if the ringer type selected here does not correspond to the one supported by the phone.
The supported ringer type can be found on the bottom of the phone, in the Ren:x.xN value where N is
the ringer type supported by the phone. For example, if N=A, the TypeA ringer type should be selected,
if N=B, the TypeB&Z ringer type should be selected.
The Hot Desking section is used to enable and configure the Hot Desking service on the FXS Line as follows:
 Enable Hot Desking – enable the Hot Desking on the corresponding FXS line.
 Hot Desking Automatic Logout – with this option enabled, QX will control the extension login timeout.
Once the predefined expiration time arrives, the currently logged in extension will automatically log out
and make available the public phone for other extensions. The following options are available:
 Never – the Hot Desking will never expire for the extension.
 After – define the period after which the extension will automatically log out from the public phone.
 At – define the certain moment (hour and minute) when the extension will automatically log out from the
public phone.
Information on the Caller ID system
Caller ID service is used to identify the caller (when performing a call or sending a voice mail) and notify the
called party about the identity of the caller. The Caller ID service is available only for phones with a display to
show that information. Two types of Caller ID notification are available on QX: FSK and DTMF.
FSK Standard
The FSK standard supports caller ID indication either with the phone handset on-hook or if the called party is
already busy with another call or operation (handset is off-hook). For internal calls, caller ID notification in FSK
can show up to two lines of identifiable parameters on the called phone’s display. The first line shows the
caller’s extension number. The second line shows the caller’s nickname (if indicated in the configuration). For
external IP calls, caller ID notification in FSK can also show up to two lines of identifiable parameters on the
called phone’s display. The first line shows the caller’s user name. The second line shows the caller’s nickname
(if indicated in configuration). If the nickname is not available and there is a display name, provided by the caller
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party, the second line will display it, otherwise the URL, in the format: username@host will be displayed. For
calls from the PSTN network, the entire caller ID message will be shown.
DTMF Standard
The DTMF standard supports caller ID indication only if the phone handset is on-hook (phone is free and ready
to accept calls). This standard also has caller ID notification conditions but they are non-configurable. Caller ID
notification in DTMF can show only one line of identifiable parameters on the called phone’s display. For internal
calls, it is the caller’s extension number. For external IP calls, it is the caller’s user name. For calls from the
PSTN network, caller ID will only display the caller’s phone number.
Note: DTMF supports only parameters consisting of digits. If any letter symbol has been used in the external
caller user name, DTMF will not display caller ID.
7.2.2 Diagnostic Loopback

The FXS Lines Loopback Settings page is used to configure the lines for voice loopback diagnostics. When
loopback is enabled on the line, any incoming calls to the corresponding line will automatically pick up on the
first ring and any voice towards the line will automatically be sent back to the caller (the caller will hear
themselves in the handset).
Figure 65: FXS Lines Loopback Settings page
 Edit – leads to FXS Lines Loopback Settings – Edit Entry page to configure the Loopback Timeout (in
seconds) for the selected FXS line(s).
 Loopback Timeout – is used to put a limit the voice loopback diagnostics duration, i.e. the caller will be
disconnected from the QX when the Loopback Timeout expires.
 Enable/Disable Loopback – is used to enable/disable the loopback service on the selected FXS line(s).
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7.3 FXO Settings

The FXO Settings is used to configure the QX's on-board FXO Lines to make PSTN calls through the on-board
FXO ports. FXO ports are available on QX50 (2 ports) and QX200 (4 ports).
Figure 66: FXO Settings page
Click the hyperlinked FXO # to open the FXO Settings – FXO # page to configure specific settings of the
selected line as follows:
 Enable Line – activate the selected
FXO line.
 Allowed Call Type – select the
allowed call directions for the FXO
line. The following options are
available:
 Both incoming and outgoing calls
will be enabled for the selected
FXO line.
 Incoming calls only (prohibiting
outgoing calls) will be enabled
for the selected FXO line.
 Outgoing calls only (prohibiting
incoming calls) will be enabled
for the selected FXO line. Figure 67: FXO Line Settings page
 Route incoming FXO Call to –
define the destination where the incoming calls will be forwarded to.
 Extension – is used to forward the calls to either PBX user extension or auto attendant extension. The
calls will be forwarded to Voice Mailbox if an inactive extension is chosen.
 Routing – is used to forward the calls to the destination defined through the Call Routing Table. Enter the
routing pattern that will be used for forwarding purposes.
 Enter a PSTN Number for the current FXO line if needed. The field value is optional and used only as an
identification parameter for the selected FXO line.
Note: The same settings and options are available for shared FXO lines.
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7.4 E1/T1 Trunk Settings

The QXs don’t have on-board E1/T1 ports. Connect QXE1T1 gateway(s) to use the shared E1/T1 trunks. The
E1/T1 Trunk Settings are used to configure the E1/T1 trunks allowing to make PSTN calls through E1/T1
trunks.
For information on how to configure and use E1/T1 trunks, please refer to the Manual-II: Administration Guide
for QX Gateways.
7.5 ISDN Trunk Settings

The Integrated Services Digital Network (ISDN) is distinguished by digital telephony and data-transport services
offered by regional telephone carriers. The ISDN Basic Rate Interface (BRI) service offers two B channels (voice
transfer) and one D channel (signaling data transfer). The BRI B-channel service operates at 64 kbit/s and is
meant to carry user data. The BRI D-channel service operates at 16 kbit/s and is meant to carry control and
signaling information, although it can support user data transmission under certain circumstances.
The QXISDN4+ has four ISDN trunks. These trunks can be configured from InterfacesISDN TrunkISDN
Trunk Settings page. The QXISDN4+ can have also shared ISDN trunks from the QXISDN4 gateway(s).
The QX20/QX50/QX200/QX500 and QX2000 don’t have own ISDN trunks. Connect QXISDN4 gateway(s) to
QX to use the shared ISDN trunks. After connection and shared mode configuration the ISDN Trunk Settings
becomes accessible on QX, allowing to configure the ISDN trunks and make/receive PSTN calls through
shared ISDN trunks. Any configuration changes applied in this page on QX will be automatically reflected on the
shared QXISDN4 gateway(s).
For information on how to configure and use ISDN trunks, please refer to the Manual-II: Administration Guide
for QX Gateways.
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7.6 Shared PSTN Gateways

The PSTN lines (FXO, E1/T1 or ISDN) of the QX gateways can be shared with QX IP PBXs. When the QX IP
PBX is configured with QX Gateways in the sharing mode to use the shared PSTN lines of the gateway, the
corresponding call routing patterns will automatically be created in the Call Routing Table on both QX IP PBX
and Gateway.
The Shared PSTN Gateways page is used to create accounts for the slave QX Gateway(s) to connect it to the
master QX for PSTN lines (FXO lines, E1/T1 and/or ISDN trunks) sharing.
Figure 68: Sharedl PSTN Gateways page
To connect QX Gateway to QX and share the PSTN Lines of the gateway:

1. Click Add and enter the following information:
 Username and Password – are used to define the authentication parameters. TIP: The Username and
Password should match on both master and slave for the successful PSTN Lines sharing.
 Click Save. The new entry will be added to the PSTN Lines Sharing table.
2. The QX will start listening connection requests from slave device.
3. Make corresponding configurations on QX gateway to establish master-slave connection. After the
slave-master connection successfully established, appropriate routing rules will be created on the Call
Routing Table for both devices (slave and master) to support PSTN line sharing.
4. Click Disconnect to disconnect the slave device from the QX. Note: The slave device will not be
reconnected automatically. You need to manually reconnect the slave device to QX from slave's WEB
GUI.
For more information on how to configure and use QX Gateways with QX IP PBXs in Share mode, please refer
to the Configuring QX Gateways with QX IP PBXs in Sharing Mode guide.
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8 Telephony Menu
Figure 69: Telephony Menu overview
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8.1 VoIP Carrier

The VoIP Carrier Wizard simplifies the configuration of the QXs with different VoIP SIP trunking services. The
wizard is for collecting the data and generating the configuration for each specific VoIP SIP trunking service on
the QX. After finishing the wizard, the extensions on the QX will be able to receive calls from the VoIP carrier SIP
trunks, as well as to place calls to the PSTN using the carrier SIP trunks.
For each configured VoIP SIP trunking service, the wizard creates a specific IP-PSTN type routing rule in the QX
Call Routing Table. By default, only PBX users can make calls through the corresponding VoIP carrier.
Additionally, a virtual extension will be automatically generated in the Extensions Management table and
registered on the VoIP Carrier's SIP server. The settings of that extension will be used to make calls towards
the created VoIP Carrier SIP Trunks.
Figure 70: Select VoIP Carrier section
The wizard composed of the following sections:

 Select VoIP Carrier section is used to select a carrier from the VoIP Carrier list. Once a carrier is found
and selected, the carrier's SIP Server and SIP Port will automatically appear on the next section of the
wizard. The Manual option selection allows to configure the VoIP Carrier settings manually from scratch.
 VoIP Carrier Settings section is used to define and configure the account from provider.
 Authentication by IP Address – if selected, deactivates the Account Name and Password fields, thus
allow skipping the IP address authentication settings. This option is intended for the VoIP carriers
requiring IP address authentication instead of account authentication and will be available if Manual has
been selected in the previous section.
 Account Name – enter the username for authentication on the carrier’s SIP server.
 Password – enter the password for authentication on the carrier’s SIP server and confirm it in the
Confirm Password field.
 SIP Server – enter the IP address or hostname for the carrier’s SIP server.
 SIP Server Port – enter the SIP server port for the carrier’s SIP server.
 Use RTP Proxy – if selected, the RTP streams between external users will be routed through the QX,
otherwise RTP packets will move directly between peers. This option is applicable only when a route is
used for calls towards a configured VoIP Carrier from a peer located outside the QX.
 Authentication Username – enter an identification parameter to reach the SIP server. It should be
provided by the SIP trunking service provider and can be requested only for certain SIP servers. For
others, the field should be left empty.
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Figure 71: VoIP Carrier Settings section
 Send Keep-alive Messages to Proxy – enables the SIP registration server accessibility to the verification
mechanism.
 Timeout – define the timeout between two attempts of SIP registration server accessibility verification.
If a reply is not received from the primary SIP server within this timeout, the secondary SIP server will
be contacted. When the primary SIP server recovers, SIP packets will continue to be sent to the
server.
 Define the Outbound Proxy, Secondary SIP Server and Outbound Proxy for Secondary SIP Server by
entering the Host Address and Port for each of them respectively. These settings are provided by the
provider and are used by the QX to reach to the selected SIP servers.
 VoIP Carrier Access Code section is used to define the routing rules for outbound/inbound calls through
VoIP carrier SIP trunks.
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Figure 72: VoIP Carrier Access Code section
 Access Code – defines the routing rule for outbound calls.

 By Prefix – is used for entering the numeric prefix that should be dialed to route call through carriers
SIP trunks. The system will route all digits matching this prefix to the carriers SIP trunks.
 By Pattern – is used to specify the pattern that should be applied to dialed digits. If an outbound call
has a destination number that matches the specified pattern, it will be completed according to the
current rule. A routing pattern may contain wildcards.
 Emergency Code – enter the emergency code supported by the specified VoIP provider. By default,
this field is filled with the information defined in the System Configuration Wizard, but this field also
allows to define the provider specific emergency codes. In case your system has both local PSTN
emergency codes and IP-PSTN codes configured, when dialing the certain emergency code, QX will
first try to reach the local PSTN allocated emergency, and if failed will dial the IP-PSTN emergency.
TIP: If the defined VoIP service is 911 compliant then you have to bind this account with the
geographical address of your device. If the provider is not 911 compliant, then the public safety agency
will not be able to determine the address automatically.
 Route Incoming Calls to – select an extension (user extension or Auto Attendant) on the QX where the
incoming calls from the configured VoIP Carrier should be routed to. There will be an unconditional
forwarding set up automatically which will care for incoming calls forwarding from the VoIP carrier to
the selected extension.
 Failover to PSTN – if selected, an additional entry will be added to the Call Routing Table to route calls
to the PSTN network through the QX on-board PSTN lines in case if the VoIP Carrier SIP trunks are not
available.
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8.2 Call Routing

8.2.1 Call Routing Table
The Call Routing Table lists the settings of all call routing records (rules) either generated by default, or added
automatically with one of the QX's system wizards: Call Routing Wizard, System Configuration Wizard or VoIP
Carrier Wizard.
Figure 73: Call Routing Table (brief view) page
 Show Brief View – if pressed, displays the most important settings of the records in the Call Routing
Table.
 Show Detailed View – if pressed, displays all settings of the records in the Call Routing Table.
 Hide disabled records/Show all records – are used to hide/show disabled records respectively.
 Enable – enables (activates) the selected records.
 Disable – disables (deactivates) the selected records.
 Add – opens the Call Routing Wizard – Add Entry page for configuring a new call routing record.
 Duplicate – duplicates the selected call routing record.
 Move Up/Move Down – moves the selected call routing record one position up/down.
 Move To – moves the selected record to specified position.
 Local Authentication – if selected, displays the Authorized Phones link for the selected routing rule.
Note: Based on the Emergency Codes and PSTN Access Codes Settings, the automatically added records in
the Call Routing Table will be marked in bold and placed in the first position of the table. Additionally, they
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cannot be modified and deleted from the Call Routing table. To remove these rules, pass through the System
Configuration Wizard and remove them from the Emergency Codes and PSTN Access Code Settings section.
For more information on how to configure and use Call Routing Rules, please refer to the Call Routing on QX IP
PBXs guide.
All calls from QX extensions, as well as some calls from external sources, are being routed in QX according to
call routing rules (records) that specify the destination based on the dialed number. When a user dials a
number, the QX matches the dialed number against the destination number patterns in call routing records.
1. If the dialed number matches only with a single pattern, then the record with respective pattern will be
used to set up the call.
2. If multiple patterns have been found to match the dialed number, the QX uses the Best Matching
Algorithm to prioritize the matching patterns.
3. Once the patterns are prioritized, the record having pattern of the highest priority will be used as a
preferred route for call setup.
The Add button starts the Call Routing Wizard for configuring a new call routing record. In general, the wizard
consists of the following sections:
 Destination Call Type
 Call Settings
 Filter on Source / Modify Caller ID
 Date / Time Settings
 Overall Call Duration Limit
 Calling Rate Settings
 Tracing / Debug Options
 Summary
Destination Call Type
This section contains the following components:

 Enable Record – this checkbox disables/enables the routing record. By default, the record is enabled.
 Destination Number Pattern – specifies a template for filtering out the calls that can be routed via
respective call routing record. If destination number of the call matches with specified pattern, then the
call can be completed via respective call routing record. The Destination Number Pattern may contain
wildcards.
 Number of Discarded Symbols – specifies the number of symbols/digits/characters that shall be removed
from the beginning of the destination number after matching it against the destination number pattern.
The field should be empty if no symbols need to be discarded.
 Prefix – specifies the symbols/digits/characters that will be added in front of the destination number after
discarding the symbols as described above. Except for single characters or character strings, the
following tags can be used for this field:
 <callerid:range> – allows to use the caller ID or its part as a prefix. For example, <callerid:1-3> indicates
that the first 3 digits of the caller ID will be considered as a prefix, <callerid:3-end> indicates that the
caller ID from its 3rd digit and up to the end will be assigned to prefix.
 <dialednum:range> – allows to use the dialed number or its part as a prefix. For example, <dialnum:1-
3> indicates that the first 3 digits of the dialed number will be as assigned to the prefix, <dialnum:1-
end> indicates that the dialed number from its 3rd digit and up to the end will be assigned to prefix.
 aaa,,,bbb – allows two-stage dialing. The aaa and bbb are the numbers to call; bbb can also be a
series of digits to inject; a comma indicates a delay of one second. For example, 11,,,11018 will call to
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11, wait until the call is established, wait for three seconds and then dial/inject 11018. The two-stage
dialing is available for FXO, ISDN, and E1/T1call types.
 Suffix – specifies the characters that will be added to destination number from the end after discarding
the symbols and adding the prefix as described above.
 Call Type – is used to select the call destination type. The following call types are available:
 PBX – local call to QX’s extension.
 PBX-Voicemail – call directly to the voice mailbox of the local PBX extension.
 PBX-Intercom – local call to PBX extensions with request to activate the Intercom service.
 SIP – calls through a SIP server.
 SIP_Tunnel – calls through an established SIP Tunnel.
 IP-PSTN – calls through the IP-PSTN provider to the global PSTN network.
 FXO – calls to the PSTN network through on-board FXO lines (available only for QX50/QX200). Or calls
to the PSTN network through available shared FXO lines (applicable for all QXs).
 ISDN – calls to the PSTN network through on-board ISDN trunks (available only for QXISDN4+). Or
calls to a PSTN network through available shared ISDN trunks (applicable for all QXs).
 E1/T1 – calls to the PSTN network through available shared E1/T1trunk(s) (applicable for all QXs).
 RTSP – Connection to RTSP server. The Number of Discarded Symbols, Prefix and Suffix fields are not
available if the RTSP call type selected.
 Metric – is used to enter a rating for the selected route in a range from 0 to 20. If no value is entered into
this field, 10 will be used as the default. If two route entries match a user’s dial string, the route with the
lower metric will be chosen.
 Enabler Key and Disabler Key – is a digital code which should be dialed from handset or the Auto
Attendant to enable or disable the routing rule. You can set the same Enabler/Disabler key for multiple
routing rules (the same key may be used as enabler for one routing rule, and as disabler for another
one). This will allow managing several routing rules with the single key.
 Require Authorization for Enabling/Disabling – is used to enable administrator’s password (Phone
Access Password) authentication when enabler/disabler keys are configured for a certain routing rule.
The service can be used locally from the handset or remotely on the Auto Attendant. When this
checkbox is selected, the password will be requested to enable/disable the certain routing rule(s).
TIP: If the password has been entered incorrectly for 3 times, no status changes will be applied to any
of the routing record(s), even to those which have no authorization enabled.
The following options give additional configuration possibilities:
 Filter on Source / Modify Caller ID – puts a limit on the routing pattern availability for selected caller(s) or
allows to modify the caller ID. This option is checked off by default.
 Date / Time Settings – allows to define a validity period(s) for the routing pattern by setting date/time
rules manually or simply assigning a working schedule.
 Overall Call Duration Limit – allows to control and limit the total calls duration for the routing pattern.
 Calling Rate Settings – is used to configure calling rate settings.
 Tracing / Debug Options – allows to enable/disable generating event notifications on the result of using
the routing rule.
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Call Settings
The content of this section strictly depends on the Call Type selected on the previous section.
Call Type – PBX
 Use RTP Proxy – if selected, RTP streams between the peers will be routed through the QX. This is
applicable when peers are located in different subnets. If not selected, the RTP streams will move
directly between peers.
 Local Authentication – if selected, the caller(s) will need to pass an authorization to make PBX calls.
 Client Code Identification – if selected, the code identification service will be activated: a caller, after
dialing the destination phone number, may optionally enter  and then an Identity Code. The Identity
Code is an arbitrary digit string entered by the user to identify a specific call or call group. The Identity
Code is sent with CDRs (Call Detail Reports) and might be used by a billing program for grouping the
calls having the same Identity Code.
 Check with 3PCC – is used to request a 3-rd party call control (3PCC) approval before placing a call
through the routing rule. If the checkbox is selected and the routing rule is used to place a call, QX sends
a request to the 3PCC application to accept or reject the specific call. The call will be placed if the
request is accepted, otherwise it will be skipped. In case of no feedback from the call controlling
application, the call will be accepted after a timeout defined in the configuration of the 3PCC application.
 Failover Reason(s) – the system will use next matching pattern(s) to establish the call if the call setup fails
due to below presented failover reasons:
 None – the system will not use next matching pattern(s) regardless of the failover.
 Busy – the system will use next matching pattern(s) if the dialed destination is busy.
 Wrong Number – the system will use next matching pattern(s) if the dialed number is wrong.
 Any – the system will use next matching pattern(s) regardless the failover reason.
Call Type – PBX-Voicemail
 Local Authentication – if selected, the caller(s) will need to pass an authorization to make PBX–Voicemail
calls.
 Voice Mail Profile – is used to define the custom Voice Mail Profile name to activate the custom voice mail
settings on the extension when the routing rule will be used. If an extension does not have a profile
specified here or the specified profile name is incorrect, the default of the extension will be used.
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Call Type – PBX Intercom
 Play audible signal before Intercom activation – if selected, the audible signal will be played while
activating Intercom service.
 Local Authentication – if selected, the caller(s) will need to pass an authorization to make PBX–Intercom
calls.
Call Type – SIP
 Use Extension Settings – is used to select the extension (also Auto Attendant) on behalf of which the call
will be placed. The SIP settings of the selected extension will be used as the caller information. If nothing
is selected from the list, the original caller information will be kept.
 Keep Original Caller ID – if selected, the called destination will receive the original caller’s information.
 Add Remote Party ID – if selected, the Remote Party ID parameter will be added in the outgoing Invite
message.
 Destination Host – is the IP address or hostname of the destination (for a direct call) or SIP server (for
calls through the SIP server). TIP: This field renamed to Modified Destination Host if the Destination
Number Pattern field (in the wizard’s first page) contains "@" symbol.
 Destination Port – is the port number of the destination or the SIP server. TIP: This field renamed
Modified Destination Port if the Destination Number Pattern field (in the wizard’s first page) contains "@"
symbol.
 Username and Password – is used to define the authentication parameters for the SIP server if needed.
 Restrict the Number of Simultaneous Calls – is used to restrict the number of simultaneous calls to the
SIP server with the same username. Allowed Call Count is used to define the number of simultaneous
calls.
 Use RTP Proxy – if selected, the RTP streams between peers will be routed through the QX. This is
applicable when the peers are both located outside the QX. If not selected, the RTP streams will move
 Single Call Duration Limit – is used to limit the duration of the call placed through the routing rule. The
single call duration will be unlimited If the checkbox is not selected. Maximum Duration is used to define
the maximum duration of the call (in seconds). The call will be disconnected without prior notice if the
maximum duration is reached.
 Local Authentication – if selected, the caller(s) will need to pass an authorization to make SIP calls.
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 Network Failure – the system will use next matching pattern(s) when system overload, network failure or
timeout expiration occurred.
 Any – stands for all failure reasons mentioned in the Failover Reason(s) group.
 Enable Failover Timeout – is used to define the period after which the call could be considered as failed
(SIP response message isn’t received). The Failover Timeout is used to define the timeout duration (in the
range from 1 to 180 seconds). The call will be established through next matching pattern(s) after the
timeout expired if the failover reason is enabled for the routing rule.
 SIP Privacy – is used to select the security level of the SIP route by means of hiding or replacing
(depending on the configuration of the SIP server) the key headers of the SIP messages used to
establish the call.
 Default Privacy – if selected, no QX specific SIP privacy will be applied, and all privacy will be relied on
the configuration of the SIP Server.
 Disable Privacy – if selected, SIP call security will be disabled, all headers of the SIP message will be
transparently visible to the destination.
 Enable Privacy – if selected, QX specific SIP privacy will be applied for the corresponding route.
Selection enables a group of checkboxes to choose the key headers to be fully or partly hidden or
replaced. Require Privacy checkbox is used to restrict the delivery of the SIP message if either of the
selected headers cannot be hidden (or replaced, depending on the configuration of the SIP server)
before being sent to the destination.
 Transport Protocol for SIP messages – is used to select the transport protocol (UDP, TCP or TLS) for
transmitting the SIP messages.
Call Type – SIP_Tunnel
 Use Extension Settings – is used to select the extension (also Auto Attendant) on behalf of which the call
will be placed. The SIP settings of the selected extension will be used as the caller information. If an entry
is not selected from this list, the original caller information will be kept.
message.
 SIP Tunnel – is used to select the previously configured SIP tunnel to route the calls through tunnel to the
remote QX device (QX IP PBXs and QX Gateways).
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 Use RTP Proxy – is applicable when a route is used for calls through QX between peers that are both
located outside the QX. RTP streams between the peers will be routed through QX if the checkbox
selected, otherwise the RTP packets will move directly between peers.
single call duration will be unlimited if the checkbox is not selected. Maximum Duration is used to define
 Local Authentication – if the checkbox selected, the caller(s) will need to pass authorization to make SIP
call through the tunnel.
 Check with 3PCC – is used to request a 3rd party call control (3PCC) approval before placing a call
 Network Failure – the system will use next matching pattern(s) if the system overload, network failure or
(depending on the configuration of the SIP server) the key headers of the SIP messages.
 Default Privacy – if selected, QX specific SIP privacy will not be applied and all privacy will rely on the
configuration of the SIP Server.
 Disable Privacy – if selected, SIP call security will be disabled and all headers of the SIP message will
be transparently visible to the destination.
 Enable Privacy – if selected, QX specific SIP privacy will be specified for the corresponding route.
replaced. Require Privacy is used to restrict the delivery of the SIP message if either of the selected
headers cannot be hidden (or replaced, depending on the configuration of the SIP server) before being
sent to the destination.
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Call Type – IP-PSTN
 Use Extension Settings – is used to select the extension (or Auto Attendant) on behalf of which the call
will be placed. The SIP settings of the selected extension will be used as the caller information. If an entry
is not selected from this list, the original caller information will be kept.
message.
 Destination Host – is the IP address or the hostname of the destination (for a direct call) or the SIP server
(for calls through the SIP server). TIP: This field renamed to Modified Destination Host if the Destination
Number Pattern field (in the wizard’s first page) contains "@" symbol.
 Destination Port – is the port number of the destination or the SIP server. TIP: This field renamed
Modified Destination Port if the Destination Number Pattern field (in the wizard’s first page) contains "@"
symbol.
 Username and Password – is used to define the authentication parameters for SIP server if needed.
SIP server with the same username. Allowed Call Count is used to define the number of simultaneous
calls.
(SIP response message isn’t received). Failover Timeout is used to define the timeout duration (in the
single call duration will be unlimited If the checkbox is not selected. Maximum Duration is used to define
 Local Authentication – if selected, the caller(s) will need to pass an authorization to make calls.
 Network Failure – the system will use next matching pattern(s) if the system overload, network failure or
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(depending on the configuration of the SIP server) the key headers of the SIP messages.
 Default Privacy – if selected, QX specific SIP privacy will not be applied and all privacy will rely on the
configuration of the SIP Server.
 Disable Privacy – if selected, SIP call security will be disabled and all headers of the SIP message will
be transparently visible to the destination.
 Enable Privacy – if selected, QX specific SIP privacy will be specified for the corresponding route.
replaced. Require Privacy checkbox is used to restrict the delivery of the SIP message if either of the
selected headers cannot be hidden (or replaced, depending on the configuration of the SIP server)
before being sent to the destination.
Call Type – RTSP
 RTSP URI – is used to define the RTSP server URI for receiving stream(s). Audio and video streams are
available depending on the RTSP server configuration.
 Username and Password – is used to define the authentication parameters for RTSP server if needed.
RTSP server. Allowed Call Count is used to define the number of simultaneous calls.
 Local Authentication – if selected, the caller(s) will need to pass an authorization to connect to the RTSP
server.
Call Type – FXO
 FXO Lines to Use – a group of radio buttons allowing to select whether a specific or any available FXO
line will be used to route the call. The following options are available:
 None – selection means no local (on-board) FXO lines will be used to route the call.
 Any Line – the call will be established through the first available local FXO line.
 Specific Line – the call will be established only through the selected local FXO line.
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If QXFXO4 gateway is connected to the QX in share mode, the following options will be available:
 Any Available Line – the call will be established through the first available on-board FXO line, then
through shared FXO lines.
 Any Line@ – the call will be established through the first available shared FXO line.
 Specific Line@ – the call will be established only through the selected shared FXO line.
 FXO Lines Load Balancing – is used to enable load balancing mechanism on the FXO lines.
 None – the system will not apply load balancing mechanism and the call will be routed through the first
available FXO line (among the selected ones).
 Round Robin – the system will apply load balancing mechanism according to internally gained statistics
of most used FXO lines, the call will be routed to the less used and currently available FXO line (among
the selected ones).
 Local Authentication – if selected, caller(s) will need to pass an authorization to make FXO calls.
 Cannot Establish Connection – the system will use next matching pattern(s) if the connection cannot be
established.
Call Type – ISDN
 Keep Original Caller ID – if selected, the called party will receive the original caller’s information (mobile
number, PSTN/SIP number, etc.) instead of extension’s information when the call(s) are forwarded.
 ISDN Trunks to Use – is used to select a specific or any available trunk to route the call(s). The following
options are available only for QXISDN4+:
 Any Trunk(User) – the calls will be established through any ISDN trunk running in User mode.
 Any Trunk(Network) – the calls will be established through any ISDN trunk running in Network mode.
 ISDN Trunk# – the calls will be established through the selected ISDN trunk.
If QXISDN4 gateway is connected to the QX in share mode, the following additional options will be available:
 Any Trunk(User)@Any – the calls will be established through the first available on-board ISDN trunk
running in User mode, then through shared ISDN trunks (running in User mode).
 Any Trunk(Network)@Any – the calls will be established through the first available on-board ISDN trunk
running in Network mode, then through shared ISDN trunks (running in Network mode).
 ISDN Trunk@ – the call will be established through the selected shared ISDN trunk.
 Any Trunk(User)@ – the calls will be established through the first available shared ISDN trunk running in
User mode.
 Any Trunk(Network)@ – the calls will be established through the first available shared ISDN trunk
running in Network mode.
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 Collect Call – is used when the calling party wants to place a call at the called party's expense. This
service is applicable only if the Collect Call service is enabled on both calling and called party's.
 Local Authentication – if selected, the caller(s) will need to pass an authorization to make ISDN calls.
established.
Attention: Additional wizard section will be available for ISDN call type to configure trunk timeslots.
 Select Timeslots – is used to select timeslot(s) which will be used for placing ISDN calls.
Call Type – E1/T1
QX IP PBXs don’t have on-board E1/T1 trunks.

 Keep Original Caller ID – if selected, the called party will receive the original caller’s information (mobile
number, PSTN/SIP number, etc.) instead of extension’s information when the call(s) are forwarded.
 E1/T1 Trunks to Use – is used to select a specific E1/T1 trunk to route the call(s). The following option is
available:
 E1/T1 Trunk1@ – the calls will be established through the selected E1/T1 trunk.
 Collect Call – is used when the calling party wants to place a call at the called party's expense. This
service is applicable only if the Collect Call service is enabled on both calling and called party's.
 Single Call Duration Limit – if selected, puts a limit on the duration of the call placed through the routing
rule, otherwise the call duration will be unlimited. Maximum Duration is used to define the maximum
duration of the call (in seconds).
 Local Authentication – if selected, the caller(s) will need to pass authorization to make E1/T1 call.
 Check with 3PCC – is used to request a 3PCC approval before placing a call with the routing rule. If the
checkbox is selected and the routing rule is used to place a call, QX sends a request to the 3rd party call
control (3PCC) to accept or reject the specific call. The call will be placed if the request is accepted,
otherwise it will be skipped.
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established.
Attention: Additional wizard section will be available for E1/T1 call type to configure trunk timeslots.
 Select Timeslots – is used to select timeslot(s) which will be used for placing E1/T1 calls.
 Up to 30 timeslots will be available for placing E1 calls regardless the trunk signaling type.
 Up to 23 timeslots will be available for placing T1 calls if the trunk signaling type is CCS.
 Up to 24 timeslots will be available for placing T1 calls if the trunk signaling type is CAS.
Radius Authentication and Authorization
RADIUS Authentication and Authorization options are available for the routing pattern regardless destination call
type if the RADIUS Client is enabled.
 RADIUS Authentication and Authorization – is used to make the caller(s) pass the authorization through
the RADIUS server to make calls.
 RADIUS Accounting – if selected, no authentication will take place, except for CDRs (call detail reports) of
the calls made through this routing record will be sent to the RADIUS server. This checkbox selection
enables the Client Code Identification checkbox. If the authentication is configured based on the caller’s
address, callers will pass the authentication automatically; otherwise they will be required to identify
themselves by a username and password.
Filter on Source / Modify Caller ID
The following components are available (Figure 74):

 Source Filter – is used to limit the routing pattern availability for selected caller(s).
 Source Number Pattern – enter the caller address for which the routing pattern will be available. The
Source Number Pattern may contain wildcards.
 Source Type – is used to select the caller source type. The following options are available:
 Any – any caller will be able to make calls regardless caller source type.
 PBX – only PBX extension(s) will be able to make calls. Attention: Additional wizard page will be
available to select defined Class of Service(s) for PBX extensions.
 SIP – only inbound SIP caller(s) will be able to make calls. To configure Source Host address (IP
address or hostname) for SIP call type, an additional wizard page will be available.
 SIP_Tunnel – only inbound callers from the selected SIP_Tunnel will be able to make calls. To select
Inbound SIP Tunnel, an additional wizard page will be available.
 FXO – only inbound FXO caller(s) will be able to make calls. To select Port ID for FXO call type, an
additional wizard page will be available.
 ISDN – only inbound ISDN caller(s) will be able to make calls. To select Port ID for ISDN call type, an
additional wizard page will be available.
 E1/T1 – only inbound E1/T1 caller(s) will be able to make calls. To select Port ID for E1/T1 call type,
an additional wizard page will be available.
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Figure 74: Filter on Source / Modify Caller ID section
 Caller ID Modification – is used to modify the Caller ID before sending them to remote party.
 Number of Discarded Symbols – enter the number of digits that should be discarded from the
beginning of the Source Number Pattern. Left the field empty if no need to discard the digits.
 Prefix – enter the symbols that will be placed in front of the Source Number Pattern. The Prefix may
contain wildcards.
 Discard Non-Numeric Symbols – is used to discard any non-numeric symbols from the Source Number
Pattern.
 Display Name – is used to replace an original caller’s ID with the custom display name. This option is
not applicable for the PBX-Voicemail call type.
 Remove Display Name – is used to remove caller IDs. This option is not applicable for PBX-Voicemail
call type.
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Date / Time Settings
This section is used to define a validity period(s) for the routing pattern:
Figure 75: Date / Time Settings section
 Typical – is used to select one of the validity periods:

 Daily – the routing pattern will be available for each day.
 Weekly – the routing pattern will be available for the selected weekday(s).
 Monthly – the routing pattern will be available for the selected day(s) in each month.
 Annually – the routing pattern will be available for the selected day(s) and month(s) for each year.
 Available Time Period – is used to define the validation time range for the routing pattern. The defined
time here will be checked against QX’s time.
 Custom – is used to manually define the validity period(s). TIP: The entered values needs to be in the
following format [MMM,MMM-MMM][DD,DD-DD][HH:mm-HH:mm]
 Schedule – is used to apply one of the configured schedules to the routing patter(s). Select the desired
schedule from the Schedule Name drop-down list.
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Overall Call Duration Limit
This section is used to limit and control the total duration of calls through the routing pattern.
Figure 76: Overall Call Duration Limit section
 Available Calling Duration – define the total duration for the calls (in minutes) through the selected routing
rule. Once the Available Calling Duration expires, the current call will be disconnected without prior
notice. Placing new calls through this rule is not possible until the Available Calling Duration is not
updated either manually or automatically by the renewal date and amount.
 Periodic – is used to select one of the Renewal Date options:
 Daily – the defined Available Calling Duration will be renewed every day.
 Weekly – the defined Available Calling Duration will be renewed every week on the specified weekday.
 Monthly – the defined Available Calling Duration will be renewed every month on the specified day.
 Renewal Amount – enter the renewal amount (in minutes) to be added to the available calling duration
when the expiration date of the Available Calling Duration is reached. Leave the field empty, if you don’t
need to renew the Available Calling Duration.
 Discard remainder before renewal – is used to discard the remainder of Available Calling Duration
before renewal and set the Renewal Amount as the new Available Calling Duration.
 Expires on – is used to define the expiration date for the Available Calling Duration. After the Expiration
Date, the routing rule becomes unavailable automatically and no new call will be possible until this field is
updated.
Note: The Overall Call Duration Limit is not applicable for PBX, PBX-Voicemail and PBX-Intercom call types.
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Calling Rate Settings
This section is used to configure calling rate settings.

 Call Completion Fee – defines the cost of a single call, regardless the call duration. The actual cost of the
call depends on the cost calculation method.
 Rate per Minute – defines the cost of one minute of call. The actual cost of the call depends on the cost
calculation method.
 Cost Calculation Method – is used to select one of the options:
 Per Second – if this method is selected, then the call cost is calculated as:
Call Cost = Call Completion Fee + CDIS/60 x CRPM
 Per Minute – if this method is selected, then the call cost is calculated as:
Call Cost = Call Completion Fee + Roundup(CDIS/60) x CRPM.
TIP: If CRPM is equal to 0, then flat fee is charged for the call.
Figure 77: Calling Rate Settings section

The calling credit assigned to extension will be charged in the following scenarios:
 Extension places a call to the destination through a payable routing rule directly.
 Extension transfers incoming call (blind or consultative) to destination through a payable routing rule.
 Extension forwards the call to destination through a payable routing rule automatically.
 Trusted caller uses the Call Relay service on auto attendant to make call to destination passing
authentication by the extension credentials (number and the password).
Note:
 The Calling Rate Settings section becomes available after Call Cost feature is activated.
 The Calling Rate Settings is not applicable for PBX, PBX-Voicemail and PBX-Intercom call types.
 The Single Call Duration Limit option cannot be used with Calling Rate Settings.
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Tracing / Debug Options
These options are used to generate event notifications on the certain execution result for the routing rule. The
events will be generated and displayed in the System Events for the following cases:
Figure 78: Tracing / Debug Options section
 In Case of Successful Call – when a call was successful established with the routing rule.
 In Case of Failover – when the call ends up due to one of the selected failover reasons.
 In Case if Call Failed to Establish – when the call executed through the routing rule failed.
Summary
The Summary section displays all configured settings for the routing pattern before applying them.
8.2.2 Call Routing
Figure 79: Call Routing page
Route all incoming SIP calls to Call Routing – if not selected, the system will first search the incoming SIP
address (Username or DID Number) in the Extensions Management table. If matching occurred, the incoming
SIP call will ring on the corresponding extension, otherwise the system will look for a matching routing rule in
the Call Routing Table. If this option is selected, the system will directly look for a matching routing rule in the
Call Routing Table and ignore the possible matches in the Extensions Management table.
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Note: Regardless of whether Route all incoming SIP calls to Call Routing is selected or not, SIP calls from
external callers will or may go to the Call Routing Table, so any unprotected routing rule can be misused. That is
why it is strongly recommended to secure the rules in the Call Routing Table by setting the filtering or
authentication options.
8.2.3 Local AAA Table

The Call Routing – Local AAA Table is used to configure and manage the local authentication database.
Figure 80: Call Routing - Local AAA Table page
To add a new AAA entry:

1. Click Add and configure the following information:
2. Select one of the Authentication methods.
 Authentication by Caller ID – set the authentication based on the caller's phone number or SIP
username (which is considered to be automatically detected).
 Authentication by Login – set the authentication based on the Username and Password provided by the
user upon login.
 Authentication by PIN – set the authentication based on the PIN Code provided by the user upon login.
3. Configure the Expiration Date and Time, if needed.
 Expires on – select to enable the Expiration Date and Time option and define the expiration date for the
configured local AAA entry.
4. Enter any Description, if needed.
5. Click Save, the new AAA entry will be added to the Local AAA table.
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Authorized Users
Caller(s) have to pass an authorization if the AAA option is enabled on the routing pattern. The caller will
automatically pass the authorization if the caller's phone number or SIP username is enabled in the Authorized
Users table, otherwise will be asked to login (enter username and password) or enter the PIN Code.
Figure 81: Authorized Users
Note: Authentication by Login cannot be combined with Authentication by PIN on the same routing rule.
Allowed Characters and Wildcards
The following is the complete list of the characters and wildcards supported in the QX system. Not all
characters and wildcards are supported for all QX options and settings. Thus, depending on the meaning of the
option some limitations can be applied.
Characters
 Numbers – 0…9
 Letters – A…Z, a…z
 Special symbols – =; +; -; $ ; / ; ~ ; _ ; – ; . ; & ; ( ) ; ' ; ! ; * ; ? ; {} ; [ ]
Note:
 The symbols (*, ?, -, ! and ,) should be prefixed with a slash (\) symbol if they are used as ordinary
characters; otherwise the system will interpret them as wildcards.
 The symbols !; { }; [ ]; – and , are used to define a range of characters and cannot be used as ordinary
characters.
Wildcards
 * – any number of any characters

 ? – any single character
 {} – a character or a string from the specified set of characters and strings
 [] – a character from the specified set of characters and strings
 Note: You can use the wildcard ? within the braces, but not *.
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The following control symbols are used to specify a set:

 Use a comma (,) to separate the elements of a set. Example: The pattern is: 9{1,3,11,a}. Numbers
matching the pattern will be: 91, 93, 911, 9a. Note: No spaces are allowed within braces.
 Use a minus sign (-) to specify a range of characters. Each successive element of the range is obtained
by increasing the previous element (the element code) by one. Example: The pattern is: 2{11-15,a-d}5
Numbers matching the pattern will be: 2115, 2125, 2135, 2145, 2155, 2a5, 2b5, 2c5, 2d5.
 Use an exclamation point (!) to exclude a character or a string from a set. Example: The pattern is: 2{11-
15,a-d,!14,!c}5. Numbers matching the pattern will be: 2115, 2125, 2135, 2145, 2155, 2a5, 2b5, 2d5.
Note: The exclamation point (!) cannot be used to exclude a range of symbols.
 Use a slash (\) before control symbols (*, ?, -, ! and ,) to use them as an ordinary character. Example:
The pattern is: 1\*[1–3]. Numbers matching the pattern will be: 1*1, 1*2, 1*3
 Use an at sign (@) to indicate full SIP address (for example: [email protected]). This pattern is mainly
used to call back users registered on the SIP server different from the one where the called party is
registered. Note: Patterns containing @ symbol will not be parsed among those that do not have @
symbol in the Call Routing Table. When calling from local extensions (the calling number for PBX
extension is sip_number@ip_address_of_QX, e.g. [email protected]), only the sip number part of
the pattern will be parsed among other entries with @ symbol in the Call Routing Table.
Best Matching Algorithm
Each call through and within a QX are made according to call routing patterns that specify a destination based
on a dialed number. When a user dials a number, the QX matches the dialed number against the existing
routing patterns.
1. If the dialed number matches only to a single pattern, this pattern will be used to set up the call.
2. If multiple patterns have been found to match the number, the QX uses the Best Matching Algorithm to
prioritize the matching patterns.
3. Once the patterns are prioritized, the pattern with the highest priority will be used as a preferred route for
call setup.
Note: The subsequent prioritized pattern will be used only if the destination specified by a pattern with higher
priority is unreachable and the corresponding Failover(s) configured.
To prioritize the matching patterns, the following criteria are sequentially applied to matching patterns. The
criteria are ordered by their priorities: Each consecutive criterion is calculated only for the patterns that take the
same value for the preceding criterion: that is Criterion 3 is calculated only for patterns that take the same value
for Criterion 1 and Criterion 2.
Criteria list
 Criterion 1 – is the presence of asterisks (*) in a pattern. The patterns without (*) have a higher priority.
 Criterion 2 – is the total number of matching digits/symbols inside and outside the braces/brackets. The
more matching digits a pattern contains, the higher its priority.
 Criterion 3 – is the number of matching digits/symbols outside the braces/brackets. The more matching
digits outside braces/brackets a pattern contains, the higher its priority. TIP: This criterion is used only if
several patterns take an equal but non-zero value for Criterion 2.
 Criterion 4 – is the total number of question marks (?) inside and outside the braces/brackets. The more
question marks a pattern contains, the higher its priority.
 Criterion 5 – is the number of question marks (?) outside braces/brackets. The more question marks
outside braces/brackets a pattern contains, the higher its priority. TIP: This criterion is used only if
several patterns take an equal but non-zero value for Criterion 4.
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 Criterion 6 – is the number of square brackets ([]). The more brackets a pattern contains, the higher its
priority.
 Criterion 7 – is the number of braces ({}). The more braces a pattern contains, the higher its priority.
 Criterion 8 – is the number of asterisks (*). The fewer asterisks a pattern contains, the higher its priority.
 Criterion 9 – is the value of the metric. The lower the metric of a pattern is, the higher its priority.
 Criterion 10 – is the position in the routing table. The higher the position of a pattern in the routing table
is, the higher its priority.
Example: The user dials 1231, the following matching patterns are found in the Call Routing Table.
Pattern Position Routing Pattern

1 *1*
2 123*
3 {11–15}3*
4 ?2?1
5 [1–3]*
6 {100–150, asd, \*\?}1
7 1[1–3]3[0–8]
8 123?
9 *2*1
10 *
Table 2: Example – The list of Patterns
Step 1: The list is sorted and the patterns with asterisks (*) are pushed back to the end of the list, due to lower
priority (Criterion 1).
Position after Step1 Routing Pattern

1 ?2?1
2 {100–150, asd, \*\?}1
3 1[1–3]3[0–8]
4 123?
5 *1*
6 123*
7 {11–15}3*
8 [1–3]*
9 *2*1
10 *
Table 3: Example – The list of Patterns after the Step 1
Step 2: The list is sorted and the patterns with the fewer number of matching digits inside and outside the
braces/brackets are pushed back to the end of the list, due to lower priority (Criterion 2). The patterns that
contain the same number of matching digits are grouped into sub-lists.
Position after Step2 Routing Pattern Matching Digits

1 1[1–3]3[0–8] 4
2 {100–150, asd, \*\?}1 4
3 123? 3
4 {11–15}3* 3
5 123* 3
6 ?2?1 2
7 *2*1 2
8 [1–3]* 1
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9 *1* 1
10 * 0
Table 4: Example – The list of Patterns after the Step 2
Step 3: Each consecutive criterion is calculated only for the patterns that take the same value for the preceding
criterion: that is Criterion 3 is calculated only for patterns that take the same value for Criterion 1 and Criterion 2.
The list is sorted and the patterns with the fewer number of matching digits outside the braces/brackets are
pushed back to the end of the list, due to lower priority (Criterion 3).

1 1[1–3]3[0–8] 2
2 {100–150, asd, \*\?}1 1
Table 5: Example – The list of the Patterns after Step 3
The Best Matching Algorithm will stop after executing Step 3 and the dialed number 1231 will pass through 1[1-
3]3[0-8] routing pattern.
Allowed SIP Addresses
Calls over IP are implemented based on Session Initiating Protocol (SIP) on the QX. When making a call to a
destination that is somewhere on the Internet, a SIP address must be provided.
SIP address needs to be entered in one of the following formats:
 "display name" <username@ipaddress:port>
 "display name" <username@ipaddress>
 username@ipaddress:port
 username@ipaddress
 username
The display name and port number are optional parameters in the SIP address. If a port is not specified, 5060
will be set up as the default one. The range of valid ports is between 1024 and 65536.
The SIP Address may contain wildcards. The following combinations can be used:
 *@ipaddress – any user from the specified SIP server
 username@* – a specified user from any SIP server
 *@* – any user from any SIP server
Note: Wildcards are allowed for called party addresses. Exceptions are addresses in the Supplementary
Addresses table that are used by Outgoing Call Blocking and Hiding Caller Information Settings services.
8.2.4 SIP Tunnel

The SIP Tunneling feature provides means for building network on Epygi QX IP PBXs (herein QX). This network
based on many "slave" QXs in satellite offices and one or more "master" QXs in the main office(s) with SIP
tunnels configured between "slave" and "master" devices. One possible scenario for using SIP Tunneling is
routing SIP calls through the remote QX device. Another scenario is building a redundant distributed PBX
system based on many slave QXs in satellite offices and two or more master QXs in the main office.
For information on how to configure and use SIP Tunnels, please refer to the SIP Tunneling Feature on QX IP
PBXs guide.
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8.2.5 Class of Service

Current implementation of the Class of Service (CoS) on the QX is used to define the permissions that PBX and
Conference extensions will have when using certain call routing rules to make a call.
The CoS provides the ability to set restrictions on the call routing rules for each extension, thus allowing to
permit or deny the extensions to use certain type of routing rules
For example, following restrictions can be applied for extensions:
1. Only Internal – internal calls to other extensions on the QX are only allowed from this extension(s). Calls
to SIP and PSTN are not allowed.
2. Only Local PSTN – calls to the local PSTN are only allowed from this extension(s).
3. Long distance IP-PSTN only – long distance IP-PSTN calls are only allowed from this extension(s).
4. International only – international calls are only allowed from this extension(s).
To realize the above defined abstract restrictions, they should be implemented as service classes on the
corresponding routing patterns in the Call Routing Table. For example, to implement the long-distance service
class, select all call routing rules on the QX that can be used for making long distance PSTN calls and assign
them to the Long-distance IP-PSTN class of service.
Pay close attention to the configuration of call routing rules on the QX. To avoid from ambiguities, do not use
the same routing rule for making calls of different classes.
Configure a CoS as follows:
1. Assign the specified CoS(s) to a certain routing rule(s).
2. Assign the specified CoS(s) to the PBX/Conference extension(s).
Note:
 CoS is applicable only for PBX source type routing rules.
 If there is no CoS assigned to the call routing rule, that rule will be generally available for any PBX
extension whether it is attached to a CoS or not.
 If the Enable Class of Service option is disabled, call routing rule(s) that are assigned to a certain CoS(s)
will be available for any PBX/Conference extension, if there are no any other filtering limitations.
Figure 82: Class of Services page
Revision 1.2 117

To create a new Class of Service and activate CoS functionality:

1. Click the Add button.
 Enter a Name for the CoS.
 Enter a Description, if needed.
2. Click Save, the newly created CoS will be added to the Class of Services table.
3. Select the Enable Class of Service option to activate the Class of Service functionality on the QX.
8.3 Call Recording Settings

The Call Recording feature allows to record all inbound and outbound calls, also those calls that pass through
the QX, keep the recordings locally or send them to the FTP server.
For information on how to configure and use Call Recording feature, please refer to the Call Recording Feature
on QX IP PBXs guide.
8.4 NAT Traversal

The NAT Traversal is divided into separate pages used to configure the General NAT Traversal Settings, SIP,
RTP and STUN parameters for NAT and the page where the NAT Exclusion table may be filled.
8.4.1 General
The General Settings page is used to select the mode the NAT Traversal will be used for the SIP traffic.
 Automatic – if selected, the system will
analyze the QX WAN IP address. If the
address is in the IP range specified for the
private networks (according to RFC), the
SIP traffic (any incoming and outgoing SIP
messages from/to QX) will be routed
through the NAT router, otherwise no SIP
traffic will be routed through the NAT
router.
 Force – if selected, all SIP traffic will be
routed through the NAT router.
 Disable – if selected, no SIP traffic will be
routed through the NAT router.
Figure 83: NAT Traversal Settings page
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8.4.2 SIP Parameters

The SIP Parameters page is used to configure the NAT specific settings for SIP and offers two independent
groups of settings.
The UDP Parameters section allows to
select the type of connection over NAT as
follows:
 Use STUN – select to automatically
discover the mapped settings for the
SIP UDP traffic over NAT. STUN
settings are configured in the STUN
Parameters page.
 Use Manual NAT Traversal – select to
manually define the mapped settings
for the SIP UDP traffic over NAT:
 Mapped Host – enter the IP address
of the mapped host for SIP UDP
traffic over NAT.
 Mapped Port – enter the port
number on the mapped host for the
SIP UDP traffic over NAT.
 Mapped TCP Host – enter the IP
address of the mapped host for SIP
TCP traffic over NAT. Figure 84: NAT Traversal – SIP Parameters page
 Mapped TCP Port – enter the port
number on the mapped host for the SIP TCP traffic over NAT.
 Mapped TLS Host – enter the IP address of the mapped host for SIP TLS traffic over NAT.
 Mapped TLS Port – enter the port number on the mapped host for the SIP TLS traffic over NAT.
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8.4.3 RTP Parameters

The RTP Parameters page is used to choose between the STUN and Manual NAT traversal connection for the
RTP traffic and define the RTP/RTCP ports for the connection over NAT.
 Use STUN – is used to automatically
discover the mapped settings for the
RTP UDP traffic over NAT. STUN
settings are configured on the STUN
Parameters page.
 Use Manual NAT Traversal – is used to
manually define the RTP/RTCP port
ranges for the RTP traffic over NAT:
 Mapped Host – is used to define the
mapped host IP address for RTP
traffic over NAT.
 Min and Max – enter the port numbers
on the mapped host for RTP and
RTSP traffic. TIP: RTP/RTCP Mapped
Port ranges should be greater than
Figure 86: NAT Traversal – RTP Parameters page
or equal to the RTP/RTCP port
ranges defined on the RTP Settings
page. Figure 85: NAT Traversal – RTP Parameters page
8.4.4 STUN Parameters

The STUN Parameters page is used to enable automatic NAT configuration through the STUN server and is
used to configure the STUN (Simple Traversal
of UDP over NAT) client on the QX as follows:
 Primary STUN Server – enter the STUN
server’s hostname or IP address.
 Primary STUN Port – enter the STUN
server port number.
 Secondary STUN Server and Secondary
STUN Port – enter the respective
parameters of the secondary STUN
server.
 Polling Interval – select the possible time
intervals between referrals to the STUN
server.
 Keep-alive Interval – define the time
interval (in seconds) for keeping NAT
mapping alive.
 NAT IP checking Interval – define the Figure 87: NAT Traversal – STUN Parameters page
interval (in seconds) between the NAT IP
checking attempts (used to distinguish the possible NAT IP address changes and to perform registration
on the new host).
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8.4.5 Exceptions
The NAT Traversal Exceptions page displays all possible IP ranges that are not included in the NAT process, but
may be accessed directly. IP addresses that are not listed in the NAT Traversal Exceptions are accessed over
NAT. For example, if a QX user needs to make SIP calls within the local network as well as outside of that
network, all local IP addresses are required to be excluded from NAT traversal settings by being listed in this
table. Otherwise, a malfunction may occur in SIP operations.
Figure 88: NAT Traversal Exceptions page
To add a new exception:

 Enter the IP Address.
 Enter the Subnet Mask. TIP: Enter 255.255.255.255 as a Subnet Mask to add only the IP address in
exception list.
2. Click Save, the new exception entry will be added to the NAT Traversal Exceptions table.
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8.5 RTP Settings

The RTP Settings page is used to configure the packet size and silence suppression for each voice codec.
The Codec Properties table lists all codecs with the packetization ranges and silence suppression associated to
each.
Figure 89: RTP Settings page
 Edit – leads to the RTP Settings – Edit Entry page to modify the selected codec settings.
 Packetization Interval – is the time interval between two RTP packets of the same stream. If this interval
is increased, the overhead is decreased, but the voice quality may deteriorate as a result. If the interval
is decreased, the network load is increased and the delay is reduced.
 Enable Silence Suppression – is used to stop RTP packet transmission in case of no voice activity. This
option helps to avoid extra traffic if the RTP stream contains no voice activity. It is activated after two
seconds of silence and restarted immediately if any audio appears.
 G.726 Standard – is used to select between packaging method of the G.726 code words into octets. If
you are experiencing problems with the voice quality when using G.726 with one of these options
selected, try switching to the next one.
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 Use ITU_T specification – if selected, the ITU I.366.2 ("AAL2 type 2 service specific convergence
sublayer for narrow-band services") type packaging of code words is used, where packing code words
into octets starts from the most significant rather than the least significant positions in the octet.
 Use IETF RFC – if selected, the IETF RFC ("RTP Profile for Audio and Video Conferences with Minimal
Control") type packaging of code words is used, where packing code words starts from the least
significant positions in the octet.
 Min and Max – enter the port numbers for RTP and RTSP traffic. TIP: RTP/RTCP Port ranges cannot
include the defined SIP ports.
 Enable RTCP Support – enables Real Time Control Protocol support and allows the RTCP packets
transmission. RTCP is used for monitoring the RTP streams and changing RTP characteristics
depending on Network conditions.
8.6 SIP
8.6.1 SIP Settings
The SIP Settings page allows to select the SIP receive UDP and TCP ports, the DNS Server configurations for
SIP and the SIP timers scheme (Figure 90).
 UDP Port – indicates the SIP UDP receive port. By default, 5060 is selected and used. The SIP UDP port
cannot be in the selected RTP/RTCP port range.
 TCP Port – indicates the SIP TCP receive port. By default, 5060 is selected and used. QX will not use
TCP protocol as a transport for SIP messages if the TCP Port field is left empty.
 TLS Port – indicates the SIP TLS receive port. By default, 5061 is selected and used. TLS port number
should be different from the TCP Port number.
 Realm – is used to define the messaging level information to be included in SIP messages sent by the
QX. This information might be used by remote side for authentication purposes.
 Enable Session Timer – enables advanced mechanisms for connection activity checking. This option
allows both user agents and proxies to determine if the SIP session is still active.
 DNS Server for SIP allows to choose between regular DNS servers configured in the DNS Settings page
and specific DNS servers for SIP traffic.
 Default – is used to apply regular DNS servers for SIP traffic.
 Specific – is used to enable SIP specific DNS servers. For this selection, both primary and secondary
SIP DNS servers should be defined in the SIP DNS 1 and SIP DNS 2 fields.
 SIP Timers is used to define the timeouts of the SIP messages retransmission.
 RFC 3261 – is used to apply standard SIP timers described in the corresponding specification.
 High Availability – is used to apply SIP timers to shorten the call establishment, registration confirmation
and registration failure procedures. This selection provides more firmness to the SIP connection but
increases the network traffic on the QX.
 Custom – is used to manually define the Registration Timeout, Registration Failure Timeout, Transaction
Duration and Session Refresh Timeout timers (in seconds).
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Figure 90: SIP Settings page
8.6.2 SIP Aliases

The Host Aliases for SIP is used to add the hostname(s) registered on remote DNS server to the Host Aliases for
SIP list. This list will be used to identify SIP packets received from remote servers where the QX is registered
with different names.
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Figure 91: Host aliases for SIP page
8.6.3 TLS Certificates

The Generate and Install New CA Root Certificate page is used to define, generate and install a new CA root
certificate for SIP TLS traffic. All fields in this page require root certificate specific information.
Figure 92: Generate and Install New CA Root Certificate page
 Generate Certificate and Install – generates a new CA root certificate based on the defined data and
installs it on the QX. The QX will reboot automatically once the new certificate is installed. You may
download the actual copy of the certificate from SIP Settings page.
 Download Current CA Root Certificate – is used to download the actual CA root certificate in the (*.crt)
format.
To ensure a secure TLS connection with the QX's defined CA root certificate, both sides should have the same
certificate installed. If the end user is an IP phone, you may activate the TLS certificate update mechanism from
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it to obtain the latest certificate generated by the QX. If the end user is a server or other device, you may
download the certificate from the QX and apply it manually on the remote side.
8.7 Schedules
The Schedules feature is designed for creating flexible weekly working schedule(s). The preconfigured
schedules then can be applied to the Call Routing and Auto Attendant. The Day/Night Switching service allows
to control and change the state of schedules manually by using the phone handset instead of going into the
GUI.
For information on how to configure and use Schedules, please refer to the Scheduling Feature on QX IP PBXs
guide.
8.8 Advanced
The Advanced group of settings is used to configure the following:
 Voice Mail
 RTP Streaming Channels
 Media Streamer
 Gain Control
 3PCC
 Radius Client
 Dial Timeout
 Call Quality Notification
8.8.1 Voice Mail

The Voice Mail Common Settings page is mainly used to select the codec for the Voice Mail recording.
Figure 93: Voice Mail Common Settings page
 Recording Codec – is used to select the codec for voice mail recording. Changing the Voice Mail
recording codec will directly affect the allocated memory size for extensions.
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 E-mail Subject for Voice Mail– is used to define a flexible subject for all e-mails sent from the QX and
carrying the voice mails.
Besides using a static text in the subject line, you may use the predefined tags to combine the needed subject:
 Hostname – is the hostname of the QX.
 Displayname – is the caller's display name. This value is not displayed for PSTN callers.
 Username – is the caller's SIP username. For PBX callers, this is the caller's extension number and for
PSTN callers, this is the caller's PSTN number.
 Full Name – is the caller's full SIP address (SIP username and the SIP server). For PBX callers, this is the
caller's extension number and for PSTN callers, this is the caller's PSTN number.
 Duration – is the voice mail duration.
 Date – is the voice mail received date.
To enter the predefined tag to the subject line, you should simply click on the corresponding tag. The following
format should be maintained to create a flexible subject:
Example: Voice mail received from $[VM_DISPNAME] $[VM_DATE].
In this example, all email subjects will contain a static text "Voice mail received from" following by the display
name of the caller and the date voice mail is received.
 FAX to E-mail Format – defines the format of the FAX document received in the extension voice mailbox
and send as an attachment to the e-mail (in case if Send new voice messages via e-mail option is
enabled for the extension). The (*.tiff) or (*.pdf) formats may be selected here.
8.8.2 RTP Streaming Channels

The RTP Streaming Channels page is used to define the channels for the broadcast RTP streaming. These
channels may be then used when configuring RTP channel streaming for music on hold (MoH), auto attendant
ringing announcement and for other custom messages.
Figure 94: RTP Streaming Channel page
To add a new RTP channel:

 Channel Name – enter the name of the RTP channel.
 Port Number – enter the broadcasting RTP port number.
 Description – enter any descriptive information, if needed.
2. Click Save, the new RTP channel will be added to the RTP Streaming Channels table.
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8.8.3 Media Streamer

Media Streamer service expand the QX audio streaming capacity. Audio files uploaded on QX can be streamed
out to specified destinations, thus giving a possibility to play music on hold and other advertisement to callers
while they are placed on hold. Audio files could be played either to remote destinations in network, or to the
extensions on the same QX.
The Media Streamer page allows to add and manage Playlists for media streamer, start and stop the audio
streaming with playlists. The configured playlists can be used to stream audio to the extensions through the
RTP Streaming Channels.
For more information on how to configure and use Media Streamer, please refer to the Customizing Voice
Messages on QX IP PBXs guide.
8.8.4 Gain Control

The Gain Control settings are used to define the Transmit and Receive gains.
The Gain Control page consists of Transmit Gain and Receive Gain drop-down lists for each line that contains
allowed gain values, which can be set up for every line.
Figure 95: Gain Control page
 For FXS lines

 Transmit Gain – defines the phone speaker volume on the call.
 Receive Gain – defines the volume of the phone microphone on the call.
 For FXO lines
 Transmit Gain – defines the level of voice transmitted from QX50/QX200 to the FXO network.
 Receive Gain – defines the volume of voice received by QX50/QX200 from the FXO network.
Revision 1.2 128

 For ISDN trunks

 Transmit Gain – defines the level of voice transmitted from QXISDN4+ to the ISDN network.
 Receive Gain – defines the volume of voice received by QXISDN4+ from the ISDN network.
 For Voice Mail
 Recording Gain – defines the volume of the phone microphone upon playing voice mails or system
messages.
 Playback Gain – defines the phone speaker volume upon playing voice mails or system messages.
 For Audio Lines
 Transmit Gain (line out) – defines the level of voice transmitted from QX to the Audio Line Out port.
 Receive Gain (line in) – defines the volume of voice received by QX from the Audio Line In port.
 Restore Default Gains – restores the default values.
8.8.5 3PCC
The 3PCC Settings are used to adjust the third-party call controlling settings. 3PCC service allows call
controlling applications to remotely initiate and handle calls on the QX and subscribe for certain event
notifications from the QX.
Figure 96: 3PCC Settings page
 Secure Connection – if selected, a secure encrypted connection will be used between the call controlling
application and QX. The Secure Connection must be set up in the same way on both sides for
successful connection.
 Request Timeout – is used to define the timeout (in seconds) during which the QX should receive a
response to the request from the call control application. If no response is received during this timeout,
QX will perform a request according default action. Let’s say the call control application is configured to
handle incoming calls on the QX. Once the incoming call occurs, the QX will try to transfer the call to the
call control application. If the call control application does not response within the mentioned timeout,
the QX will answer the call or perform an action configured for non–answered incoming calls. This setting
depends on the network conditions therefore consult with your network administrator before changing
the default value.
 Feature Key – indicates whether the feature key for the 3PCC Support is installed on the system or not.
The system will not accept connections from 3PCC applications if no key is found. The 3PCC support is
an optional feature and can be activated with a feature key from the Licensed Features page.
 WAN Port – indicates whether there is a filtering rule specified for the Call Control Access or not. If a
third-party call control application connects to the QX from the WAN interface, a filtering rule for the
corresponding host should be created on the Call Control Access page to allow the application a remote
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access. Creating a filtering rule is not required if the firewall is not setup on the QX. The field shows
Opened if there is at least one enabled filtering rule for the Call Control Access.
8.8.6 RADIUS Client

Remote Authentication Dial in User Service (RADIUS) specifies the RADIUS protocol used for authentication,
authorization and accounting, to differentiate, to secure and to account for the users. The RADIUS Server
provides the option for a caller from/through QX to pass authentication and to be able to dial a specific number.
When a RADIUS client is enabled on the QX, and according to the configuration of AAA Required option, the
RADIUS server will be used to authenticate user and/or to account for the call. This can be accomplished by
automatic detection of the caller’s number or a customized login prompt where the caller is expected to enter a
username and password.
Transactions between the client and the RADIUS server are authenticated through the use of a shared Secret
Key, which is never sent over the network. In addition, user passwords are encrypted when sent between the
client and RADIUS server to eliminate the possibility of a party viewing an unsecured network where they could
determine a user's password. If no response from the RADIUS Server is returned after the Receive Timeout
expires, the request is resent numerous times as defined in the Retry Count list. The client can also forward
requests to an alternate server(s) if the primary server is down or unreachable. An alternate server can be used
after a number of failed tries to the primary server.
Once the RADIUS server receives the request from client, it determines if the sending client is valid. A request
from a client that the RADIUS server does not recognize must be silently discarded. If the client is valid, the
RADIUS server consults a database of users to find the user whose name matches the request. The user entry
in the database contains a list of requirements (username, password, etc.) that must be met to give access to
the user. If all conditions are met, the user gets access to the QX Network.
 Enable RADIUS Client – is used to enable RADIUS client on QX. TIP: The RADIUS Client cannot be
disabled if there is at least one route with RADIUS Authentication and Authorization or RADIUS
Accounting values configured in the AAA Required drop-down list on the Call Routing Table. In order to
disable the RADIUS Client on the QX, the configured routes should be removed first.
 Primary Server – enter the IP address of the primary Radius Server.
 Secondary Server – enter the IP address of the secondary Radius Server
 NAT Station IP – enter the WAN IP address for the NAT station. If no NAT Station is specified here, QX’s
IP address will be sent to the RADIUS server.
 Secret Key – enter the secret key between the Radius client and the server.
 Retry Count – select the number of attempts authorized before canceling the registration.
 Receive Timeout – select the timeout (in seconds) between two attempts to register.
 Encoding Type – select the encoding type (PAP or CHAP) that should be unique on both the client and
the server sides for the establishment of a successful connection. Encoding type should also be
requested from the Radius Server administrator.
 Authorization Port – enter the port number on the RADIUS server where QX is to send the authentication
requests.
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 Accounting Port – enter the

port number on the RADIUS
server where QX is to send
the accounting messages.
 Enable common login for all
users in time of by phone
authentication – enable
custom settings for the callers
who passed an authorization
by phone on the QX. This
checkbox enables Username
and Password fields to enter
the custom settings that will
stand instead of the source
caller’s settings when being
delivered to the RADIUS
server.
 Authentication on Destination
RADIUS Server – enter
Username and Password to
pass authentication on the
RADIUS Server of the
destination QX. If these fields
are left empty, the original
authentication settings that
users enter for authentication
will be used.
 Username – enter an
identification username for
accounting purposes. When
no username is specified in
this field, the source username
will be used for accounting.
This field is dedicated for
accounting services only.
 Send Accounting messages –
select sending both Start and
Stop accounting messages or
only Stop accounting
message.
Figure 97: Radius Client Settings page
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8.8.7 Dial Timeout

The Dial Timeout Settings are used to adjust the timeout setting when dialing on the phone. The Routing Dial
Timeout option is used to specify a period of time after the last dialed digit that the system identifies as a
completion of dialing. If the user does not press any key within the specified timeout, the system assumes that
the dialing is completed and starts processing the dialed number. The Routing Dial Timeout setting will also be
applied to all the supported IP phones. The modified value will take effect after rebooting the IP phones.
Figure 98: Dial Timeout Settings page
8.8.8 Call Quality Notification

The Configure Call Quality Event Notification page is used to configure the policy for event notification when the
call quality is lower than the allowed level.
Figure 99: Configure Call Quality Event Notification page
 Notify when – is used to enable the call quality monitoring mechanism.

 Call Quality less than – is used to select the call quality level since when the notification will be
generated and displayed in the System Events.
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9 Firewall Menu
Figure 100: Firewal Menu overview
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9.1 Firewall
The Firewall Configuration page allows setting up the Firewall, configuring the security level and enabling the
Network Address Translation (NAT) and Intrusion Detection System (IDS) services on the QXs.
Firewall is a security service configurable through various criteria. It has three level of security policies: low,
medium and high. The Firewall allows or blocks traffic based on the policies, services and/or IP addresses.
Filtering rules will take effect only if the Firewall has been enabled and are independent from the selected firewall
security level. Additional service-based rules can be added as well.
NAT is used to connect the QX LAN members to the Internet using QX's WAN IP address. NAT also forwards
incoming packets from the WAN to the PCs or devices in the QX’s LAN. The IDS is a type of firewall. It deletes
dangerous packets or packets containing intrusion attacks, also generates a log file containing information
about the dropped packets and senders responsible for those packets. The log can be viewed on the IDS Log
page. Users can be notified about the generated logs through an email, flashing LED display notification, etc.
Note: NAT and IDS are not available for QX 2000 and QX3000.
9.1.1 Firewall and NAT

The Firewall Configuration page offers the following components:
 Enable IDS – enables the Intrusion Detection System.
 Enable NAT – enables the Network Address Translation.
 Enable Firewall – enables the firewall security service. The firewall security level has to be selected,
otherwise the firewall cannot be enabled.
The Firewall Security levels are the following:
 Low Security – everything that is not explicitly forbidden will be allowed. This security level doesn't block
anything by default. It is recommended if the device is already located behind another firewall or if every
filter has been configured correctly.
 Medium Security – traffic originating from the LAN side may pass and traffic from the WAN side will be
blocked by default. This is the recommended security level.
 High Security – everything that is not explicitly allowed will be blocked, including traffic from the LAN
side.
Figure 101: Firewall Configuration page
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9.1.2 Advanced Firewall Settings

The Advanced Firewall Settings are used to deny Ping and Portscanning operations addressed towards the
device. The QX will answer with irritating message to the Ping and Portscanning operations. The Ping and
Portscanning operations will be denied when the Firewall is enabled from the Firewall and NAT page.
Figure 102: Advanced Firewall Settings page
9.1.3 IDS Log

The IDS Logs (Intrusion Detection System) page contains information about dropped packets and the senders
responsible for those packets. The system discards dangerous packets or packets including intrusion attacks.
It generates a table with the IDS log report. The administrator can be notified about newly logged entries in
various ways (e-mail, display notification, etc.) depending on the settings in the System Events page. IDS logs
will be reported as soon as IDS is enabled from the Firewall and NAT page. The IDS Logs table is a list of new
or read IDS entries and descriptions referring to them.
Figure 103: IDS Logs page
Click on the desired entry to see it's detailed log in the IDS Detailed Logs table. The IDS Logs table is a detailed
log that shows additional information about the access protocol, IP address and port number as well as date
and time of the event.
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9.2 Filtering Rules

The Filtering Rules page allows to configure the filters for incoming and outgoing traffic. It is allowed to create
only one rule per service to prevent inaccurate configuration. You may use IP groups to include several IP
addresses for any rule. Since the filtering rules specify the operation mode of the firewall, they only take effect if
the firewall has been enabled (also NAT is enabled to use the Port Forwarding function in the Incoming
Traffic/Port Forwarding filtering rules). The filtering rules are independent from the security level, so they will
work regardless the type of selected security level.
Note:
 Applying firewall rules will prevent the establishment of new connections that violate the rules. Applying
rules does not kill existing connections that violate the rule.
 The newly created blocking filtering rules will take effect immediately only if the IP address(es) added into
the Blocked IPs.
9.2.1 View All Filtering Rules
View All table presents all configured filters, specified by their State (enabled or disabled), selected Service, type
of Action (allowed or blocked), displays Restricted IP addresses and destination of port forwarding.
Figure 104: Filtering Rules – View All page
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9.2.2 Incoming Traffic/Port Forwarding

The Incoming Traffic/Port Forwarding filtering rules are used to allow or deny incoming traffic to reach to the QX
LAN. Enable the NAT service on the QX to allow Port Forwarding in the Incoming/Forwarding filtering rules.
Figure 105: Filtering Rules – Incoming Traffic / Port Forwarding page
9.2.3 Outgoing Traffic

Outgoing Traffic rules allow or deny access to the external services for QX’s LAN users.
Figure 106: Filtering Rules – Outgoing Traffic page
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9.2.4 Management Access

Management Access rules are used to allow or deny hosts management access to the QX.
Figure 107: Filtering Rules – Management Access page
9.2.5 Call Control Access

Call Control Access rules allow or deny hosts to access Call Control interface of the QX. This is used to enable
the access from the call controlling application (DCC, HotCall Add-In, etc.) to the QX.
Figure 108: Filtering Rules – Call Control Access page
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9.2.6 SIP Access

SIP Access filtering rules are used to allow or deny access to or from SIP servers and other SIP devices in the
WAN. This filtering rule will prevent or allow incoming/outgoing SIP calls from/to specified SIP server(s) or
host(s).
Figure 109: Filtering Rules – SIP Access page
9.2.7 Blocked IPs

Blocked IP List entries are used to deny access for special hosts. Traffic to or from these hosts will be blocked
in any case, no matter what services are configured in other filters. The Blocked IP List service has a higher
priority than the Allowed IP List: if the same host is listed in both tables, it will be blocked.
Figure 110: Filtering Rules – Blocked IP List page
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9.2.8 Allowed IPs

Allowed IP List entries are used to allow trusted hosts to reach your network and vice versa. TIP: If a host also
appears in the Blocked IP List, the Blocked IP List has a higher priority, and the traffic will be blocked.
Figure 111: Filtering Rules – Allowed IP List page
To add a Filtering Rule
1. Navigate to the Filtering Rules (Incoming Traffic/Port Forwarding, Outgoing Traffic, Management Access,
SIP Access, Blocked IP List or Allowed IP List) page to add a rule.
2. Click Add on the corresponding filtering rule page.
 Select the Service to configure a rule for it.
 Select an Action to setup the rule.
 Enter the destination IP address in the Forward to IP where traffic should be transferred to if it comes
from the restricted host (Incoming Traffic/Port Forwarding rule).
 Enter a port number in the Port Translation field which will stand instead of the original port number
when incoming packet is being forwarded (Incoming Traffic/Port Forwarding rule).
 Choose the restriction type by selecting Any, Single IP, IP/Mask or Single URL and enter the required
information in the text fields or select a group.
 Enter a Description, if needed.
3. Click Save to create a rule with given parameters. The newly created filtering rule will be shown in the
corresponding Filtering Rule table and in the View All page.
4. Click Enable to activate the newly created filtering rule from the corresponding table.
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9.3 Custom Services

9.3.1 Service Pool Configuration
The Service Pool Configuration page is used to create new services with the appropriate settings (protocol type
and port range). New services can be used to add a restriction or allowance upon creating a new filtering rule.
To add a new service:
1. Click Add.
 Enter a Service Name.
 Select a Protocol type.
 Define the Port Range.
2. Click Save to add the service with
given parameters. The newly
created service will be displayed on
the Service Pool Configuration table.
Figure 112: Service Pool Configuration – Add Service page

9.4 IP Groups
9.4.1 IP Pool Configuration
The IP Pool Configuration page is used to add groups of IP addresses that have the same restriction criteria.
When adding a new filtering rule, a group can be used instead of several IP addresses. TIP: Changing a group
name will also change the references to this group, including filtering rules and member relations to the other
groups. Deleting a group will also delete any reference to the corresponding group, including filtering rules and
member relations to the other groups. The IP Pool Group Configuration page displays a list of all the added
member IP addresses for the selected group as well as allows adding/modifying members.
Figure 113: IP Pool Configuration page
Click Group name link to display an IP Pool Group Configuration page with the Members list for the current
group.
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To add a new Group with Members:

1. Click Add on the IP Pool
Configuration page.
2. Enter a Group Name and fill in the
Group Description, if needed.
3. Click Save to add the group. The
newly added group will be displayed
on the IP Pool Configuration table.
4. Open the IP Pool Group
Configuration page by clicking on
the group name.
5. Click Add on the IP Pool Group
Configuration page. A page opens
where new members may be added
to the group.
 Choose the member addition type
by selecting IP Address, IP Subnet
Figure 114: IP Pool Group Configuration – Add Member
and enter the required information
in the text fields or select A user-
defined Group.
 Enter a Member description, if needed.
6. Click Save, the new member will be added to the Current Group table.
9.5 SIP IDS

9.5.1 SIP IDS Settings
The SIP IDS Settings page includes the following components:
Figure 115: SIP IDS Settings page
Revision 1.2 142

 Enable SIP IDS – enables SIP attack prevention.

 Add the IP address into the Blocked IP List in Firewall – if selected, the system will block the SIP
attacker's IP address by adding it to the Blocked IP List of Firewall. This action will take effect if Firewall is
enabled on the QX.
 Discard SIP messages from IP address for – if selected, the system will ignore the SIP messages from
attackers IP address for the specified time period after attack detection (default period is 32 seconds).
 SIP IDS Exceptions – link leads to the Exceptions for SIP IDS page where you can specify the trusted IP
address(es) that shouldn’t be blocked.
To add a new SIP IDS exception:
1. Click the SIP IDS Exceptions link.
 Enter the IP Address.
 Enter the Mask. TIP: Enter 32 as a Mask to add only the IP address in exception list.
3. Click Save, the new exception entry will be added in the SIP IDS Exceptions table.
The Bad IP detection logic
The Bad IP detection logic is the following:

 2 failures of SIP authorization/authentication from the same IP during 250 milliseconds.
 2 messages causing Non-self-Request-URI from the same IP during 250 milliseconds.
 If there are 10 failures in a row during any period of time from the same IP, then the IP will be blocked.
Note: Any successful registration attempt from that IP will reset the counter. For example, if IP=xxx.xxx.xxx.xxx
failed to register 9 times and then successfully registered on the 10th attempt, then it resets the counter to 0.
Next time the same IP can make another 9 unsuccessful attempts before being blocked.
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10 Network Menu
Figure 116: Network Menu overview
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10.1 IP Routing
Routing is used to relay information across the Internet from a source to a destination. Along the way, at least
one intermediate node is typically encountered. Routing differs from the bridging. The main difference between
bridging and routing is that bridging operates at the OSI Data Link Layer (Level Two Media Access Control
Layer) and routing operates at OSI Network Layer (Level Three).
QX’s IP Routing service allows to route IP packets from one destination to another (or to a specified router)
through the QX or QX’s VPN. The IP Routing is used to make IP Static, IP Policy and PPTP/L2TP routes for IP
packets routing. This page consists of three tables. Entries in the tables are color coded according to the state
of the route. For example, yellow indicates disabled routes, green indicates successful routes and the red
indicates routes with an error.
10.1.1 IP Static Routes

IP Static Routes are used to forward IP packets from the Network, the QX is connected, to the specified
destination.
The IP Static Routes table displays all configured IP static routes with their parameters:
 Target State – state of the route (enabled or disabled).
 Actual State – state of the route connection (up, down or erroneous).
 Route To – subnet the incoming packets should be routed to.
 Via IP Address – router IP address incoming packets should be routed through.
Figure 117: IP Static Routes page
To add a new IP Static Route:

 Route to – enter the IP address and subnet mask of the destination the IP packet will be routed to.
 Via IP Address – enter the IP address of the router that will forward the IP packet to the specified
destination.
2. Click Save, the new route will be added to the IP Static Routes table.
3. Click Enable to activate the newly created route.
Note: The rule with the longest subnet (smallest IP range) will take effect when having two or more IP Static
routing rules with the coinciding subnets.
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10.1.2 IP Policy Routes

IP Policy Routes allow IP packets forwarding to the specified router depending on the source IP address as well
as defining the priority for the current routing rule.
The IP Policy Routes table displays all specified IP policy routes with their parameters:
 Priority – route priority.
 Route from – is where the subnet, routed packets come from.
 Via IP Address – is where the router IP address incoming packets should be routed through.
To add a new IP Policy Route:
 Priority – define a priority of the routing rule. Enter any numeric value from the 1-252 range. The lower
the number, the sooner the routing rule will take effect (higher priority).
 From – enter the packet source IP address and subnet mask of the specified destination to match with
the rule.
 Via IP Address – enter the IP address of the subsequent router to forward the IP packet to.
2. Click Save, the new route will be added to the IP Policy Routes table.
4. Click Raise Priority or Lower Priority to increase/decrease the priority of the selected policy route by one.
10.1.3 PPTP/L2TP Routes

PPTP/L2TP Routes allow IP packets forwarding through the PPTP and L2TP tunnels of the QX. VPN routes
cannot be generated if PPTP/L2TP connections do not exist on the QX.
The PPTP/L2TP Routes table displays all generated VPN routes with their parameters:
 Route to – subnet where the incoming packets should be routed.
 Via Tunnel – VPN tunnel incoming packets should be routed through.
 Tunnel State – actual state of the route tunnel (up or down).
To add a new PPTP/L2TP Route:
 Route via – select the available PPTP or L2TP connection from the drop-down list. A connection
selected from this list will be used to route the IP packet from the QX’s LAN to the peer behind the
PPTP/L2TP tunnel.
 Route to – enter the IP address range of the possible peers behind the PPTP/L2TP tunnel the IP
packets should be routed to.
2. Click Save, the new route will be added to the PPTP/L2TP Routes table.
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10.2 DHCP
The DHCP Settings are used to enable a DHCP server and control the QX user’s LAN settings. Therefore, QX
LAN users will automatically be provided with the following settings:
 IP addresses
 NTP (corresponds to the QX’s IP address)
 WINS server
 Nameserver (corresponds to the QX’s IP address)
 Domain name
10.2.1 DHCP Server

The DHCP Settings for the LAN Interface page offers the following input options:
Figure 118: DHCP Settings page for the LAN interface page
 Enable DHCP Server – activates the DHCP server on the QX. If selected, the QX will be able to assign
dynamic IP addresses to the devices in its LAN.
 Give leases only to hosts listed in the Special Devices table – if selected then the DHCP services will be
provided only to the devices listed in the Special Devices table.
 Dynamic IP Address Range (from to) – defines the range of IP addresses that will be assigned to the QX
LAN users.
 WINS Server – defines a WINS server IP address for the QX LAN users.
 DHCP Advanced Settings – leads to the DHCP Advanced Settings page to configure the QX's DHCP
server advanced options.
 Special Devices – allows to set a static IP address binding on the MAC address of the device in the QX’s
LAN. When this table is configured, the devices with defined hostnames and MAC addresses will always
get the same LAN IP address from the DHCP server. Devices not listed in this table will get dynamic LAN
IP addresses. This table is also displayed in the System Configuration Wizard.
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To add a new host:

 Hostname – enter the hostname of the device.
 MAC Address – enter the MAC address of the device.
 Static IP Address – enter a fixed IP address of the device. TIP: If you leave this field empty, the device
will get the first available IP address from range the defined in the DHCP Settings page.
2. Click Save, the new host will be added to the Special Devices table.
10.2.2 DHCP Advanced Settings
The DHCP Advanced Settings page is used to add new advanced options of the DHCP sever and modify the
existing ones. The DHCP Advanced Settings table lists DHCP server default options. All options will be sent to
the DHCP clients.
Figure 119: DHCP Advanced Settings page
To add a new DHCP option:

 Select one of the predefined DHCP Server options or define custom one.
 Predefined Options – select one of the predefined DHCP server options.
 Option Name – select DHCP server option.
 Option Value – enter the value for the selected option. Type and format of the entered value depends
on the option selected from the Option Name list.
 Custom Options – define a new DHCP server option. The following parameters must be entered for a
new option:
 Option Code – enter a code for the option. It may have values in a range from 0 to 255.
 Option Type – select the type of the option value. It may be an IP address, a Boolean or integer value,
etc.
 Option Value – enter the value of the option. This value depends on the selected Option Value Type.
2. Click Save, to add a new DHCP option to the DHCP Advanced Settings table.
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Note:
 If there are two or more values entered, they must be separated by commas.
 The changes made through the System Configuration Wizard regarding the DHCP server options will not
immediately reflect on the DHCP Advanced Settings if DHCP sever option parameters are modified, so
user will have to reconfigure changes in the DHCP Advanced Settings manually. The settings will be
changed automatically if the parameters in DHCP server options are in "bold". In this case, the DHCP
Advanced Settings will be changed automatically if you make changes through the System Configuration
Wizard.
The following DHCP Server Statements are available:
 Authoritative – enables/disables authoritative mode on the QX DHCP server. TIP: If several DHCP servers
are used on the network and the QX has to provide network parameters to IP phones only then disable
the Authoritative mode.
 Ping Check – if selected, verifies the availability of an IP address on the network before providing it to a
client. The QX will first ping an IP address retrieved from the IP pool and wait for a reply. If no reply is
received within a timeout specified in the Ping Timeout field (by default 1 sec), the retrieved IP address
will be provided to the client. Otherwise, a new IP address will be retrieved from the IP pool and the
procedure will be repeated. If not selected, the QX will provide an IP address immediately when
requested.
10.2.3 DHCP Leases

The DHCP Leases page includes a list of the leased host addresses that are part of the QX’s LAN. For these
hosts, QX acts as a server supplying them with a unique IP address. It displays a read–only table describing all
the leased IP hosts and their parameters.
Figure 120: DHCP Leases page for LAN interface
 IP Address – host IP address, assigned by the QX.

 MAC Address – host MAC address, provided by the host itself.
 Lease Start – date and time when the leased IP address has been activated.
 Lease End – date and time when the leased IP address has been or will be deactivated.
 Binding State – indicates the state of the DHCP lease.
 Hostname – hostname, provided by the host itself.
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10.2.4 DHCP Settings for the VLAN Interface

The DHCP Settings for the VLAN Interface page is used to establish virtual networks on the QX’s LAN or to
integrate the QX into the corporate network’s virtual LAN/WAN. DHCP service can be activated both on LAN or
WAN interfaces. VLAN is useful in corporate companies to divide large networks into subgroups and to have
devices like QXs and IP phones in each network separated (for example, to separate networks for data and
voice transmission). Priorities may be assigned to the interfaces for packets prioritization.
With VLAN configuration, each
virtual network will be
characterized with a VLAN ID
(tag). Packets addressed to
that network will be checked
towards the ID and if the ID
number defined in the
incoming packets matched the
corresponding network’s ID,
the packets will be accepted.
Otherwise, the packets will be
dropped. In the same way, if
the QX is integrated into the
network that uses VLAN
technology, outgoing packets
should have the ID number of
the corresponding virtual Figure 121: DHCP Settings for VLAN Interface page
network, for the remote party
to accept those packets.
The DHCP Settings for the VLAN Interface table lists all enabled VLAN interfaces created in the VLAN Settings
page and corresponding parameters (VLAN ID, IP Address Range and WINS Server).
 Enable DHCP Server – activates the DHCP server on QX for VLAN. If selected, the QX will be able to
assign dynamic IP addresses to the devices in its VLAN.
 Activate – activates the DHCP service on one of the VLAN interfaces in the list. Only one VLAN interface
can have DHCP service activated.
 Edit – is used to modify the selected VLAN interface. This page contains all the same components as the
DHCP Server page.
 VLAN Settings – leads to the VLAN Settings page to create virtual LAN/WAN interfaces.
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10.3 DNS Settings

The DNS Settings page provides the option of setting up a name server for the QX.
Figure 122: DNS Settings page
 Obtain DNS Server Address automatically (N/A on QX2000 and QX3000)– automatically configures the
assignment of the name server address from the provider party.
 Use the following DNS Server Address – is used to manually assign a name server as follows:
 Preferred DNS – enter the IP address of an external name server.
 Alternate DNS – enter the IP address of the secondary name server that will be used if the main name
server cannot be accessed.
10.3.1 DNS Server Settings

The DNS Server provides the services to the hosts in the QX’s LAN. With this service, QX returns the correct IP
address to the requested domain name, so that any device in the LAN can be accessed by its hostname or
alternative alias name. The DNS Server Settings page is used to configure DNS server settings on the QX and
define a list of aliases for the devices in the QX’s LAN.
Figure 123: DNS Server Settings page
Revision 1.2 151

 Zone – displays the QX’s host domain name as it is configured in the System Configuration Wizard.
 Time to Live (TTL) – indicates the time (in seconds) during which the DNS server will keep the resolved
names in its cache. During this time, the same address will be resolved from the cache of the DNS
server. When this timeout expires, the requested address will be resolved newly.
 Mail Exchange (MX) – indicates the mail server’s hostname. When resolving the email address, the
reference will go to the mail server defined in this field, before being sent out to the external network. The
value in this field will be used in the MX record in the DNS server on the QX.
The table on this page lists aliases for each of the device in the QX’s LAN to be resolved through the DNS
server.
To add a new host:
 IP Address – enter the IP address of the host.
 Hostname – enter the hostname of the device.
 Alias – enter up to 5 alias names by which the device will be resolved.
2. Click Save, the new host will be added to the DNS Server Settings table.
10.3.2 Dynamic DNS Settings

The Dynamic DNS (DynDNS) is a service that is used to map a dynamic IP address to a host name. This service
is used if you are connected to the Internet with a dynamic IP address (and PPP, DHCP client) and want to
allow access from the Internet to a device behind the firewall. For example, if you want to run your own WEB
server. The following options are available:
 Enable Dynamic DNS – enables the dynamic DNS service. To enable the DynDNS service on the QX, you
first have to choose a DynDNS provider and register at their website.
 Username – enter the username specified during the registration at the DynDNS provider.
 Password – enter the password specified during the registration at the DynDNS provider.
 Max Time between updates – define the interval between two updates (in hours). The values entered in
these fields should be greater than 24. Normally, whenever you set up a connection to the Internet, the
DynDNS is updated at least once in the period indicated in this field.
 Use predefined Service – enables the manual configuration of the DynDNS service.
 Service – Select the provider to be subscribed to.
 Host – enter the name of the host on the Internet.
 TZO Connection Type – enter a special parameter required by the DynDNS provider TZO.
 DHS Cloak-Title – enter a special parameter required by the DynDNS provider DHS.
 Mail Exchange – enter the address of the e-mail server the DynDNS service provider will relay e-mails
to. If this service is used, ensure that there is port forwarding configured for SMTP (port 25) to the
internal e-mail server.
 easyDNS Partner – enter a special parameter required by the DynDNS provider easyDNS.
 Create Custom HTTP GET Request – is used to switch to the custom settings of the DynDNS service.
Normally, the DynDNS provider uses HTTP get requests to map dynamic IP addresses to host names. If
the HTTP receive request is known to you, click Create Custom HTTP GET Request and enter the
appropriate value into the URL field.
 URL – is used to define the complete request to be sent to the DynDNS server. The request modifies
the name server database so that the hostname will be resolved to the new IP address.
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Figure 124: Dynamic DNS Settings page
 Basic Authentication – enables the encoding of the username and password entered in the text fields
above, and then uses the Basic Authentication method to notify the provider about the user
authentication settings.
Most of the DynDNS providers require an authentication for security. Authentication parameters can be
provided in the URL text field to be used for the HTTP get request. Select Basic Authentication if no
authentication parameters are provided.
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10.4 PPP/ PPTP Settings

The PPP/PPTP Settings (N/A on QX2000 and QX3000) are used to establish a connection over the DSL link, or
any other type of uplink, to the ISP. A connection is needed to set up and make or receive calls through PPP
over Ethernet. The connection may be configured for manual setup or always up. Once a connection has been
established between the QX and the provider, QX users will be able to make and receive calls at any time.
 PPTP Server – is used to define the IP
address of the PPTP server.
 Encryption – is used to select the
encryption for the traffic over the PPTP
interface.
 Keep Connection Alive – keeps the
connection alive by sending control
packets dedicated to the link state
verification.
 Authentication Settings – are used to
enter the authentication parameters
(Username and Password) to register on
the ISP server.
 Dial manually – if selected, a button will
be displayed in the main management
window that serves to switch the
Internet connection on/off. When
accessing the Internet, every station of
the connected LAN has to connect to
the QX first.
 Always connected – if selected then
the QX will always stay in the
connected mode. Figure 125: PPP/PPTP Settings page
 IP Address Assignment – is used to
define the IP address assignment for the PPP interface with the following options:
 Obtain an IP Address automatically – with this option selected, QX will use DHCP to get an available IP
 Use the following IP Address – manually assign an IP address to the PPP interface.
Revision 1.2 154

10.4.1 Advanced PPP Settings

The Advanced PPP Settings are used to enable/disable certain parts of the negotiation process during
connection establishment. These settings are available only if QX has a PPPoE WAN interface. Note: It is
strongly recommended to leave these switches unchanged if their meanings are not completely clear.
Figure 126: Advanced PPP Settings page
 Enable automatic PPP Restart – is used to select the time when the PPP connection will automatically be
restarted.
 LCP Echo Failures – displays the number of the LCP echo failure packets received before the PPP
connection will be considered as dead and will be restarted.
 Disable CCP (Compression Control Protocol) negotiation – select if the peer system is not working
properly. For example, if it is not accepting the requests from the PPPD (Point-to-Point Daemon) for
CCP negotiation.
 Disable magic number negotiation – select if the peer system is not working properly. If selected, PPPD
cannot detect a looped-back line.
 Disable protocol field compression negotiation in both the receive and the transmit direction – if selected,
no protocol field compression will take place.
 Disable Van Jacobson style TCP/IP header compression in both the transmit and the receive direction – if
selected, no negotiation of TCP/IP header compression will take place and the header will always be
sent uncompressed.
 Disable the connection-ID compression option in Van Jacobson style TCP/IP header compression – if
selected, PPPD will not compress the connection-ID byte from Van Jacobson and will not ask the peer
to do so.
 Disable the IPXCP and IPX protocols – select if the peer is not working properly and cannot handle
requests from PPPD for IPXCP negotiation.
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10.5 SNMP Settings

The Simple Network Management Protocol (SNMP) is an application layer protocol that facilitates the exchange
of management information between network devices and is used by network administrators to manage
network performance, find and solve network problems, and plan for network growth. The SNMP agent is
running to allow administrators to remotely manage QX’s network and the device’s configuration.
For more information on how to configure and use SNMP, please refer to the Configuring SNMP Agent on QX
IP PBXs guide.
10.6 VLAN Settings

The VLAN Settings page is used to create a new interface(s). The VLAN Settings table lists all existing virtual
interfaces on the QX.
Figure 127: VLAN Settings page
To configure a new VLAN interface:

 Enable – select to enable current virtual interface after creating it.
 Interface Type – select whether the virtual interface will be created on LAN or WAN interface.
 VLAN ID – enter the virtual network ID from the range of 0 to 4094.
 Priority – select the priority of packets in the corresponding interface. Packets with the lower priority (0)
will be delivered first.
 IP Address – enter the IP address of the virtual interface.
 Subnet Mask – enter the subnet of the virtual interface.
2. Click Save, the new interface will be added to the VLAN Settings table.
10.7 VPN Configuration

The Virtual Private Network (VPN) is established to connect two local networks (intranets) securely over the
Internet. The VPN routers manage authentication between servers and clients and handle data encryption for
the connection. Only authorized users may access the network and the data exchange cannot be intercepted.
In general, the VPN connection is similar to the Internet connection, both of them are based on the IP detection.
The VPN gateway must authenticate the IP addresses of its partners’ VPN gateways. Each time a specific VPN
is to be established, usually the same IP addresses are expected. This will not create problems if both VPN
partners have fixed WAN IP addresses. There may be circumstances reasons to prefer dynamically allocated IP
addresses. To enable devices that use a variable IP address as part of a VPN, they are turned into "Road
Warriors". For example, at this point they are able to reach their corporate network via authentication at the
company's VPN gateway device. This VPN gateway device must have a fixed IP address for Internet access.
Every VPN needs at least one VPN gateway with a fixed IP address.
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The partner devices of a VPN must have different WAN IP addresses, and if they are connected to local area
networks, these LAN’s must have different IP addresses. As all QX devices have the same default IP addresses
on delivery, at least one of them must be reconfigured in order to set a new IP address. The QX supports
several types of VPN connections such as IPSec and PPTP/L2TP.
Note:
 The VPN is not available on QX2000 and QX3000.
 It is strongly recommended not to run different types of VPN tunnels between the same endpoints
simultaneously.
10.7.1 IPSec Configuration

An IPSec connection includes authentication and encryption to protect data integrity and confidentiality. VPNs
are "virtual" in the sense that individuals can use the public Internet as a means of securely accessing an
internal network. Once the IPSec connection is established, users have access to the same network resources,
addresses, and so forth as if they were connected locally. VPNs are "private" because the data is encrypted
between two VPN gateways. Encryption makes it very difficult for anyone to intercept data and capture
sensitive information such as passwords. The QX can be set up to act as a VPN router when connected to the
Internet with a fixed IP address or as an IPSec connection Road Warrior when using dynamic IP addresses.
To establish an IPSec connection, it is required to have an operational VPN gateway on each side of the
communication line. QXs, PCs and workstations can be equipped with VPN gateways. Home offices typically
prefer dynamically allocated IP addresses.
When the QX is connected to the Internet with a fixed IP address, it will be set up to act as a VPN gateway. QX
is then prepared to establish an IPSec connection with another VPN gateway device, but also allows access to
Road Warriors. A notebook /laptop used by a traveling employee could also be a Road Warrior. Access to their
company’s intranet via an IPSec connection can be obtained regardless of their location.
The QX can also be set up to act as a Road Warrior. If a home office is connected to the Internet via QX with
Point to Point Protocol over Ethernet (PPPoE) and dynamic IP addressing, setting up the QX as a Road Warrior
will allow an IPSec connection to the corporate network.
You need to use a key to encrypt and decrypt the data transmitted via the IPSec connection. RSA is an
asymmetric key system used by the QX. It has to be available on both sides of the IPSec connection and will
generate a different pair of keys on each side, a private key and a public key. During the connection
establishment, some data is encrypted with the remote party’s public key. They can be decrypting the data
with their private key and the data encrypted there with QX’s public key can be decrypted with QX’s private
key. Since the private key is never transmitted, it stays completely unknown to everyone, thus the system
remains safe. Even if someone gets the public key, decryption cannot be possible without the private key. The
QX generates such a pair of keys automatically when it is set up. The user cannot see the private key, but must
know the public key because their IPSec connection partner will need it.
Note: A pair of keys will always be generated, a public one and a private one. The previously generated pair of
keys will become invalid as well as all existing IPSec connections that use RSA keying.
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The IPSec Configuration page consists of two sub-pages: Connection and RSA Key Management.
Connection
The Connection sub-page is used to create a new IPSec connection or manage the existing ones.
Figure 128: IPSec Configuration – Connection Settings page
The following buttons are available:

 Start – activates the selected IP Sec connection. The State will be changed to Activated or Connected
depending on the IPSec connection type.
 Stop – disconnects the selected IPSec connection. The state of the IPSec connection will be changed to
Stopped.
 Edit – leads to the IPSec Configuration Wizard to modify the parameters of the selected IPSec
connection.
 Delete – removes the selected IPSec connection(s) from the table.
 Restart All Active Connections – restarts all active IPSec connections. The State of these IPSec
connections will turn into Connected or Activated if the restart procedure has been successfully
completed.
 Add – leads to the IPSec Connection wizard to define a new IPSec connection.
The IPSec Configuration wizard composed of the following sections:
 New IPSec Connection
 IPSec Keying Properties
 Automatic Keying
 IPSec Connection Properties
 Summary
New IPSec Connection
 Connection Name – enter the name of a new IPSec connection.

 Peer type – select the remote machine type for the IPSec Connection to be established. If the list does
not include the required type of machine, choose Other.
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Figure 129: New IPSec Connection section
 VPN Network Topology – select the location of the peers participating to the VPN connection. The
following options are available:
 This device<>Peer – direct connection between the QX and peer.
 This device<>[Internet]<>Peer – connection between the QX and peer over Internet.
 This device<>NAT<>[Internet]<>Peer – connection between the QX and peer over Internet through QX
provider’s NAT.
 This device<>[Internet]<>NAT<>Peer – connection between the QX and peer over Internet through peer
provider’s NAT.
IPSec Keying Properties
The Internet Key Exchange (IKE) and Encapsulated Security Payload (ESP) parameters are used to define the
security of your IPSec tunnel.
The IKE parameters group is used to set up security association (SA) in the IPsec protocol suite.
 Encryption – is used to select encryption standard. The following standards are available:
 Triple DES – uses three DES encryptions on a single data block with three different keys to achieve a
higher security than is available from a single DES pass (block cipher algorithm with 64-bit blocks and a
56-bit key).
 AES (128 bit) cryptography scheme is a symmetric block cipher, which encrypts and decrypts 128-bit
blocks of data.
blocks of data.
blocks of data.
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Figure 130: IPSec Keying Properties section
 Authentication – is used to select authentication type:

 SHA/SHA1 (Secure Hash Algorithm) is a strong digest algorithm proposed by the US NIST (National
Institute of Standards and Technology) agency as a standard digest algorithm and is used in the Digital
Signature standard, FIPS number 186 from NIST. SHA is an improved variant of MD4 producing a 160-
bit hash. SHA and MD5 are the message digest algorithms available in IPSEC.
 MD5 (Message Digest) is a hash algorithm that makes a checksum over the messages. The checksum
is sent with the data and enables the receiver to notice whether the data has been altered.
 Diffie-Hellman Group – is used to determine the length of the base prime numbers used during the key
exchange process. The cryptographic strength of any key derived depends, in part, on the strength of
the Diffie-Hellman group, which is based upon the prime numbers. The higher is the group bit rate, the
better is encryption. If mismatched groups are specified on each peer, negotiation fails.
The ESP parameters group is used to provide origin authenticity, integrity and confidentiality protection of
packets. The same IKE encryption and authentication parameters are used.
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Automatic Keying
The Automatic Keying section is used to specify a Shared Secret password or RSA public key to secure the
IPSec Connection.
Figure 131: Automatic Keying section
 Shared Secret – is a type of password that both of the IPSec connection partners must know. The
authentication will be done with this shared secret. All encryption functions below will remain concealed.
 RSA – is used to define the public RSA key of your IPSec Connection partner.
 Local ID – is used to define the QX FQDN (Fully Qualified Domain Name) that is resolved to an IP
address, or any @-ed string that is used in the same way.
 Remote ID – is used to define the IPSec Connection partner’s FQDN (Fully Qualified Domain Name) that
is resolved to an IP address, or any @-ed string that is used in the same way.
The Local ID and Remote ID text fields may have the values in one of the formats presented below:
 IP address – example: 10.1.19.32.
 Host name – example: vpn.epygi.com. This form requires additional resources to resolve the host
name, therefore it is not recommended to use this format.
 @FQDN – example: @vpn.epygi.com. This form is considered as a string, and is not being resolved. It
is recommended to use this form for most applications.
 user@FQDN – example: [email protected]. This form is also considered as a string, and is not being
resolved. It has no advantages over the previous form.
 PFS (Perfect Forward Secrecy) – is a procedure of system key exchange, which uses a long-term key
and generates short-term keys as is required. Thus, an attacker who acquires the long-term key can
neither read previous messages that they may have captured nor read future ones.
 Use IPSec Compression – enables IPSec data compression. This option is displayed only if the IPSec-
VPN partner supports it.
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Note:
 It is not recommended to start multiple road warrior connections with the Shared Secret automatic
keying selected. For multiple road warriors to be started at the same time, it is recommended to use
RSA keying with Local ID and Remote ID fields configured.
 QX will prevent to start a connection with Shared Secret automatic keying selected if there is already a
connection with RSA automatic keying started, and vice versa.
 The Local ID and Remote ID values are mandatory for the RSA selection and are optional for Shared
Secret selection. However, it is recommended to define the Local ID and Remote ID values for multiple
road-warrior connections.
IPSec Connection Properties
Dynamic IP/Road Warrior and Static IP/ Remote Gateway buttons are used to select whether the remote QX (or
another VPN gateway device) is connected to the Internet with a dynamic IP address and is acting as a Road
Warrior, or is connected to the Internet with a fixed IP address and is acting as a VPN Gateway.
The following options is used to configure IPSec connection:
 Dynamic IP/RoadWarrior – if selected, then the Remote Gateway IP Address field will automatically
generate the value "any", to allow access independent from the sending IP address.
 Static IP/Remote Gateway – is used to enter the IP address or hostname of the remote QX (or another
VPN gateway device) in the Remote Gateway field.
 This device<>Remote Gateway – allows access from the local QX to the remote VPN gateway (local
subnet and remote subnet are not included). This includes management access. The checkbox is
disabled if the This device<>NAT<>[Internet]<>Peer or This device<>[Internet]<>NAT<>Peer option is
selected from the VPN Network Topology drop-down list on the first page of the IPSec Connection
Wizard.
 Local Subnet<>Remote Gateway – allows access from all stations connected to the local network to the
remote VPN gateway device (local QX and remote subnet are not included). The checkbox is disabled
when the This device<>[Internet]<>NAT<>Peer option is selected from the VPN Network Topology drop-
down list on the first page of the IPSec Connection Wizard.
 This device<>Remote Subnet – allows access from the local QX to all stations of the remote LAN (local
subnet and remote VPN gateway devices are not included). The checkbox is disabled when the This
device<>NAT<>[Internet]<>Peer option is selected from the VPN Network Topology drop-down list on
the first page of the IPSec Connection Wizard.
 Local Subnet<>Remote Subnet – allows access from all stations of the local network to all stations of the
remote LAN (VPN gateway devices are not included). In this case, the local and remote subnet IP
addresses and subnet masks have to be entered in the corresponding fields Local Subnet IP and
Remote Subnet IP.
 Stop connection if not successful – allows to stop the IPSec connection attempts if the partner remains
unreachable after the timeout period. If not selected, then the system will continue to try to reach the
IPSec connection partner.
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Figure 132: IPSec Connection Properties section
Note:
 It is not recommended to simultaneously start a static and a dynamic connection configured to use the
same secret key. A dynamic connection may capture the static connection peer and vice versa,
depending on which connection established first.
 The Static IP/ Remote Gateway selection is not possible if the Gateway is positioned behind NAT, since
the IP address of the remote gateway is not reachable directly in this case.
Summary
The Summary section displays all configured settings for the IPSec connection.
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RSA Key Management
The RSA Key Management sub-page is used to generate a new RSA Key. Also, this page displays the current
public RSA key and allows to send it to the IPSec connection partner.
Figure 133: RSA Key Management page
To generate a new RSA key:

1. Select one of two available RSA key lengths (1024 or 2048).
2. Click Generate, to generate the key.
3. Enter the email address and click Send to send the generated key to the partner via e-mail.
10.7.2 PPTP/L2TP Configuration

Point-to-Point Tunneling Protocol (PPTP) is used to establish a VPN over the Internet. Remote users can
access their corporate networks via any ISP that supports PPTP on its servers. PPTP encapsulates any type of
network protocol (IP, IPX, etc.) and transports it over IP. Therefore, if IP is the original protocol, IP packets ride
as encrypted messages inside PPTP packets running over the IP. PPTP is based on the Point-to-Point Protocol
(PPP) and Generic Routing Encapsulation (GRE) protocol. Encryption is performed by Microsoft's Point-to-Point
Encryption (MPPE), which is based on RC4.
Layer 2 Tunneling Protocol (L2TP) is a protocol from the IETF, which allows a PPP session to run over the
Internet, ATM, or frame relay network. L2TP does not include encryption (as does PPTP), but defaults to using
IPSec in order to provide virtual private network (VPN) connections from remote users to the corporate LAN.
Derived from Microsoft's Point-to-Point Tunneling Protocol (MPPTP) and Cisco's Layer 2 Forwarding (L2F)
technology, L2TP encapsulates PPP frames into IP packets either at the remote user's PC or at an ISP that has
an L2TP remote access concentrator (LAC). The LAC transmits the L2TP packets over the network to the L2TP
network server (LNS) at the corporate side. Large carriers also may use L2TP to offer remote POPs to smaller
ISPs. Users at the remote locations dial into the modem pool of an L2TP access concentrator, which forwards
the L2TP traffic over the Internet or private network to the L2TP servers at the ISP side, which then sends them
on to the Internet.
For PPTP and L2TP connections, two parties are required: Client and Server. The client is responsible for
establishing the connection. The server is waiting for clients; it is not able to initiate the connection itself.
Servers define the range of IP addresses that are assigned to the Server and Client hosts participating in a
Revision 1.2 164

connection. Each side is specified by the Host Name and Password. The client should know the server’s name
and password (the QX server has no password) and the server should set the client’s host name and a
password. The client and server settings have to match on both sides for successful connection establishment.
Note:
 L2TP tunnels have no data encryption mechanism.
 Only one client can be connected to the server in the same network.
The PPTP/L2TP Configuration page consists of 3 sub-pages: Connections, PPTP Server Configurations and
L2TP Server Configurations.
Connections
The Connections sub-page lists all existing connections characterized by their Connection Name, Type (PPTP or
L2TP), Client/Server mode, State, Remote Hostname IP (IP address or hostname of the connection peer) and
Status. The state of the PPTP and L2TP Connections, except for the "Stopped" state, is established as a link
that refers to the page where login/logout information about the connection status is displayed. Logs can be
useful to determine problems on PPTP or L2TP connections failure.
Figure 134: PPTP/L2TP Configuration – Connections page
 Start – initiates the selected connection(s). If it is a client connection, then this button initiates a client
activity of reaching the server.
 Stop – stops the selected connection(s). Stopping the server connection will disconnect all connected
clients and close the PPTP/L2TP tunnel.
 Add – leads to the PPTP/L2TP Connection wizard to establish a new connection.
Note: After creating a PPTP server connection, PPTP connections between devices placed on the QX LAN and
external devices will no longer be possible. The PPTP pass-through service for incoming and outgoing traffic
will be automatically disallowed once a PPTP server connection is created.
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The PPTP/L2TP Connection Wizard composed of the following sections:

 New PPTP/L2TP Connection
 PPTP Connection Properties
 Summary
New PPTP/L2TP Connection
Figure 135: New PPTP/L2TP Connection section
 Connection Name – enter connection name. The name cannot start with a digit symbol; however, it can
contain digits further in the name.
 Connection Type – select the type of the connection (PPTP or L2TP).
PPTP Connection Properties
 Peer Name – enter the connection peer name. TIP: The Peer Name must be written with Latin
characters. When creating a connection with a Windows Server, ensure that a user with the QX’s host
name and Dial-in access exists on the server. When creating a connection with a Windows Client,
ensure that the Peer Name specified on this page matches the Dial-in connection’s username.
 Password – enter the password.
 Server/Client – select whether the new connection will be a server or client. For the Client radio button
selection following information needs to be provided:
 PPTP Server (if the PPTP connection type is selected) – enter an IP address or a host name of the
PPTP server.
 L2TP Server (if the L2TP connection type is selected) – enter an IP address of the L2TP server.
 Authentication (N/A for PPTP connection) – select the authentication protocol through which the client
will communicate with the server. This section is available only if the PPTP connection type is selected
on the previous section. The MSCHAPv2 selection enables the Encryption drop-down list where the
encryption method can be selected. TIP: These authentication settings should be identically configured
on both peers for the successful connection establishment.
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Figure 136: PPTP/L2TP Connection Properties section
Summary
The Summary section displays all configured settings for the PPTP/L2TP connection.
PPTP Server Configurations
The PPTP Server Configuration sub-

page is used to configure the PPTP
server settings.
 Subnet – is used to enter the IP
address range for the PPTP
server and clients within the PPTP
tunnel. The value specified for the
subnet mask is fixed to 24 to
restrict the possible number of
clients for the PPTP connection.
TIP: The first address specified in
the PPTP Subnet will be assigned
to the PPTP server, others will be
assigned to the clients. The PPTP
server subnet must be different
from the L2TP server subnet.
 Authentication – is used to select
Figure 137: PPTP Server Configuration page
the corresponding authentication
protocol through which the client
will communicate with the server. TIP: The MSCHAPv2 selection enables Encryption drop-down list where
the encryption method can be selected.
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L2TP Server Configuration
The L2TP Server Configuration sub-page is used to configure the L2TP server settings. The L2TP Subnet is
used to enter the IP address range for the L2TP server and clients within the L2TP tunnel. The value specified
for the subnet mask is fixed to 24 to restrict the possible number of clients for the L2TP connection. TIP: The
first address specified in the L2TP Subnet will be assigned to the L2TP server, others will be assigned to the
clients. The L2TP server subnet must be different from the PPTP server subnet.
10.8 OpenVPN Configuration

OpenVPN allows secure point-to-point or site-to-site connections in routed or bridged configurations between
the QX and other devices and remote access facilities.
OpenVPN supports bidirectional authentication based on certificates, meaning that the client must authenticate
the server certificate and the server must authenticate the client certificate before mutual trust is established.
Both server and client will authenticate each other first by verifying if the presented certificate was signed by the
certificate authority (CA), then by checking the information in the now-authenticated certificate header, such as
the certificate common name or certificate type (client or server).
For information on how to configure and use OpenVPN, please refer to the OpenVPN Service on QX IP PBXs
and Auto Configuration of Epygi Supported IP Phones using OpenVPN guides.
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11 Status Menu
Figure 138: Staus Menu overview
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11.1 System Status

11.1.1 General Information
The General Information page provides the following information:
 Uptime Duration – time period the QX is running since last reboot.
 Device Hostname – displays the QX device host name.
 Firmware Version – the version of the QX's firmware and the file system.
 Language Pack – this information is presented only when a custom language pack is uploaded and
indicates the version of language pack.
Figure 139: Status – General Information page
11.1.2 Network Status

The Network Status page provides information on available network interfaces and services on the QX.
Figure 140: Status – Network Status page
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The Network Status table displays the following information:

 Interface Name – network interfaces (LAN, WAN, VLAN and etc.) available and configured on the QX.
 IP Address – IP address for the network interface.
 Subnet Mask – subnet mask for the network interface.
 Properties – MAC address for the network interface or additional information about the interface.
 Monitor – allows to watch and monitor the interface.
The Preferred DNS, Alternate DNS and Default Gateway display the corresponding settings of QX, configured in
the Internet Configuration Wizard.
The Services table displays the available services (NTP Server and Client, DHCP Server and Client, DNS,
Firewall, NAT, PPP) with their current status.
11.1.3 Lines Status

The Lines Status page displays the current status and general information for the selected Line or Trunk.
 FXS Line (available on QX50/QX200) contains the following tables:
 General Information – shows the number of attached extension, display name, the phone state and the
number of active calls.
 Caller ID Services – shows the status for the Caller ID Services (enabled or disabled) on the attached
extension.
 General Settings and Other Services – shows the settings and services configured on the attached
extension.
 IP Line contains the following tables:
 General Information – shows the number of attached extension, display name, the phone state and the
number of active calls.
 IP Line Registration – shows the IP line registration status.
 Caller ID Services – shows the status for the Caller ID Services (enabled or disabled) on the attached
extension.
 General Settings and Other Services – shows the settings and services configured on the attached
extension.
 FXO Line (available on QX50/QX200) shows the Allowed Call Type, the destination for Incoming Calls (to
Extension, Attendant or to Call Routing Table) and the state of the line (Free or Busy).
 ISDN Trunk (available on QXISDN4+) shows the status of B1 and B2 channels and the state of the trunk
(Free or Busy). The table includes a group of static and dynamic parameters. The static parameters are
always displayed. The dynamic parameters appear only whenever an event takes place on the channel.
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Figure 141: Status – Lines Status page
11.1.4 Memory Status

The Memory Status page (Figure 142) displays information on available memory size and memory allocation
among the applications and services on the QX. The Memory Size is expressed in time units calculated using a
specific codec.
The Memory Status page consists of the following sub-pages:
 General Information – shows the memory size and current memory allocation(usage) between the system
messages, voice mails, recorded calls and recorded conferences. The Databases table shows the
memory size used by different QX services.
 User Extension – shows the memory size available and currently allocated(used) to voice mails and
recorded/uploaded system voice messages for each specific user extension. The Universal Extension
Recordings shows the space used to define the system default voice messages common for all
extensions.
 Attendant – shows the memory size available and currently allocated(used) to recorded/uploaded system
voice messages for each Auto Attendant.
 Recorded Box – shows the memory size available and currently allocated(used) to recorded calls,
recorded/uploaded system messages for each specific Recording Box. Only G711 codec is used to
record calls.
 ACD Queue – shows the memory size available and allocated(used) to recorded/uploaded system
messages for each specific ACD Queue.
 Conference – shows the memory size available and allocated(used) to currently recorded conferences
and recorded/uploaded system voice messages for each Conference.
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Figure 142: Status – Memory Status page
For more information on Memory Status, please refer to the Memory Management on QX IP PBXs guide.
11.1.5 Hardware Status

The Hardware Status table shows the list of network interfaces, on-board and external devices and parts
currently available on the QX with their parameters and statuses.
Figure 143: Status – Hardware Status page
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11.1.6 SIP Registration Status

The SIP Registration Status page displays information about the QX extensions registration on SIP servers.
Information about the configured SIP Tunnels between Epygi devices is displayed here as well.
The Registration on SIP Servers table shows the following information:
 Extension – shows the extension number. The hyperlinked Extension number leads to the Extensions
Management – SIP Settings section where the SIP registration settings can be modified.
 Username/DID Number – is the registration username or the DID number on the server.
 SIP Server – indicates the address of the SIP server. It can be either an IP address or a host name.
 Registered – shows the registration status.
 Registration Time – shows the registration time.
Figure 144: Status – SIP Registration Status page
The SIP Tunnels to Slave Devices and SIP Tunnels to Master Devices tables list the SIP tunnels between local
and the remote Epygi devices. The SIP Tunnels to Slave Devices table lists those tunnels where local QX acts as
a master. The SIP Tunnels to Master Devices table lists those tunnels where local QX acts as a slave.
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11.1.7 IP Lines Registration Status

The IP Lines Registration Status provides information on IP Lines registration and used subscriptions on the QX.
The IP Lines Registration table lists the IP lines and remote extensions registered on the QX. The following
information is available:
 IP Line – shows the number of IP line. The hyperlinked Line number leads to the IP Line Settings page
where the IP Line settings can be modified.
 Extension – shows the extension number attached to the IP line.
 Username – indicates the registration username.
 Registered – shows the registration status.
 Binding IP Address – indicates the IP address of the registered device (IP phone, softphone or etc.).
 Registration Time – shows the registration time.
 Registration Expires in – shows when the registration will expire for the device.
The Subscriptions Count table shows the used and maximum allowed subscriptions on the QX. The
subscriptions are events originated by the QX's services or IP phones. The following information is available:
 Dialog(BLF) event – IP phone's BLF (Busy Lamp Field) subscriptions, used for watching the extensions,
as well as showing the states for other telephony service on the phone.
 Message Waiting Indication (MWI) event – IP phone's MWI subscriptions, used for voice mailbox status
indication on the phone.
 Presence event and Other events – used by the QX's internal services.
Figure 145: Status – IP Lines Registration Status page
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Note:
 When the allowed number of subscriptions is reached, no new subscriptions are possible. Typically, the
number of subscription should be keep reasonably below the maximum allowed number, to avoid losing
subscriptions.
 The number of Maximum allowed subscriptions can be changed from the generalconfig.cgi hidden page.
Reboot the QX, to apply the changes.
11.1.8 License Status

The License Status page provides information about the following licensable features on the QX.
 DCC Basic
 DCC Pro
 iQall Mobile Toggling
Figure 146: Status – License Status page
The License Status table shows the following information:

 License – the type of the license.
 Total – the number of available licenses.
 In Use – the number of used licenses.
 Attached Extensions – lists the extensions that are being used for the corresponding license. The
hyperlinked extension leads to the User Extension – Licensing section to activate/deactivate the
license(s).
11.2 Events
11.2.1 System Events
The System Events page lists information about system events that have occurred on the QX. When a new
event takes place, a record is added to the System Event table. Numerous circumstances may cause a certain
application on the QX to flag an event. TIP: The warning link that leads directly to the System Events page will
disappear from the management pages if the administrator has marked all new events as "read".
The System Events table is the list of new and read system events. System events have corresponding coloring
depending on the nature of the event: success (priority 1, color green), low importance failure (priority 2, color
yellow), critical failure (priority 3, color red).
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The table shows the Status of the event (new or read) as well as the name of the application the event refers to,
event description, and the date when the event was received. For example, if the event was caused by the IDS
service, the Check IDS link (N/A on QX2000 and QX3000) appears in the reference row that will lead to the IDS
Log page, or if the event has occurred due to incorrect mail sending or SIP registration, the corresponding links
will be seen in the Reference column of the table.
Figure 147: System Events list
 Current System Time – displays the current date and time on the QX.
 Mark all as read – marks newly occurred events as "read".
 Reset LED – switches off the flashing LED (if applicable) on the board. The LED notification may appear
(depending on the notification type given) in the Event Settings page when a new event occurs.
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11.2.2 Event Settings

The Event Settings page lists all possible events on the QX and allows controlling notification (action) when an
event takes place. Each entry in the events’ table has a checkbox assigned to each row. You can modify
multiple events by selecting two or more events.
Figure 148: Event Settings page
 Edit – leads to the Edit Event Settings page to modify the event action.
 Application – displays the application the event refers to. Multiple is shown here if more than one event
has been selected for the action assignment.
 Name – displays the name of the event. Multiple is shown here if more than one event has been
selected for the action assignment.
 Description – displays additional information about the event. Multiple is shown here if more than one
event has been selected for the action assignment.
 Action – is used to select event notification method:
 Display Notification – displays notification in the System Events page.
 Flash LED (N/A for QX2000) – LED flashes every second. For some events, the LED will start flashing
after a delay.
 Send Mail – an e-mail will be sent to the e-mail address specified in the E-mail (SMTP) page.
 Send SNMP Trap – a trap will be sent to the traphost(s) listed in the SNMP Trap Settings table.
 Send SMS – a SMS will be sent to the mobile number specified in the Short Text Messaging (SMS)
page.
Note:
 Actions that are not allowed for the selected event (like mail notification if the PPP link is down or the mail
server has been configured improperly) are hidden. For multiple events editing, actions that are not
appropriate for least one of the selected events will also be hidden.
 In case of an IDS (Intrusion Detection System) intrusion alert, only the first possible intrusion in each 10-
minute period will initiate an event. If the QX cannot receive an IP address from the DHCP or PPP
servers, or cannot register an extension on the SIP or Routing servers, or cannot reach an NTP server, it
raises only one event for the entire period the action has failed, but will continue to try. When the
required action is successful, the QX raises an appropriate message.
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11.3 Call History

The Call History allows to track and report the Call Detail Records (herein CDRs) for calls originated and
terminated on QX, as well as for calls passed through QX.
11.3.1 Successful, Missed and Unsuccessful Outgoing Calls

The Successful Calls, Missed Calls and Unsuccessful Outgoing Calls pages lists successful, missed and
unsuccessful outgoing calls and their parameters. The following components are available:
 Filter – allows searching for call records based on at least one of the criteria: Call Start Time, Call
Duration, Call Cost, Caller and Called parties.
 Clear Filter – is used to remove the filter.
 The Download and Download in CSV format buttons are used to download the displayed CDRs for each
page (Successful, Missed and Unsuccessful Outgoing) in the (*.log) and (*.csv) formats respectively.
Figure 149: Call History – Successful Calls page
CDRs listed in the Call History tables are characterized by the following parameters:
 Call Start Time – shows the start date and time of the call.
 Call Duration – shows the duration of the call.
 Calling Phone – shows the caller's number and display name (if available).
 Called Phone – shows the callee's number and display name (if available).
 Call Cost – shows the calculated call cost (if available).
 Details – provides the following additional information:
 Details on the call quality, audio codec used to receive and transmit packets and the call close reason.
The call close reason appears to provide more information about the call termination, such as a
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network problem, call termination by one of the parties, voice mail service activation, etc. The Details
information link leads to the RTP Statistics page where all RTP parameters of the call are shown.
 Authenticated By – shows the authentication parameters in the Local AAA Table, such as login or PIN
code used to pass the authentication when making call.
 Information about FAX statistics for the calls that have a FAX transmission handled. It only appears
when there was a FAX transmission during the call. The FAX link leads to the FAX Statistics page.
11.3.2 Call Cost

The Call Cost page shows the summarized information regarding the payable calls. The following components
are available:
 Filter – allows searching for call records based on at least one of the criteria: Timeframe, Duration,
Extension and Call Cost.
 Clear Filter – is used to remove the filter.
 The Download and Download in CSV format buttons are used to download the displayed CDRs in the
(*.log) and (*.csv) formats respectively.
The information listed in the Call Cost table are characterized by the following parameters:
 Extension – shows the extensions originating the payable calls.
 Duration – shows the total duration of all payable calls for the specific extension.
 Cost – shows the total cost of all payable calls for the specific extension.
Figure 150: Call Cost page
Click the extension's hyperlink to review the credit settings of the extension. Click the cost's hyperlink to see the
detailed information concerning the payable calls for the specific extension.
11.3.3 Settings
The Call History – Settings page (Figure 151) is used to configure specific parameters for displaying Call History.
The following options are available:
 Enable Call Reporting – is used to enable the CDR reporting and allows to select the maximal numbers
of CDRs entries to be displayed in the Call History tables respectively.
 Maximum Number of Successful/Missed/Unsuccessful Call Records – these are used to select the
maximum number of Successful, Missed and Unsuccessful Outgoing CDR entries to be displayed in
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the respective Call History tables. TIP: When the number of CDRs exceeds the numbers specified in
the Call History – Settings page, the oldest entries are being automatically deleted. To keep the call
history safe, configure and use the Archiving Settings service on the QX.
 The Download all CDRs and Download all CDRs in CSV format links are used to download the displayed
Call History in the (*.log) and (*.csv) formats respectively.
 Clear all CDRs – is used to remove all CDRs.
 CDR Parameters section provides the full list for CDR parameters on QX. You can select the specific
parameters to be excluded from the downloaded/archived call history files to make the CDR files more
compact, thus more readable. For the detailed information about the CDR parameters listed in this page,
please refer to the Call Detail Records on QX IP PBXs guide.
Figure 151: Call History – Settings page
11.3.4 Archive
The Archive page shows the Call Details Record (CDR) archived files and allows the user to download them
either in (*.log) and (*.csv) format.
The following functions are available on this page (Figure 152):
 Filter – allows to search for the specific archived CDR records in the Archive table by the record's full
name or some part of the name.
 Delete – removes the selected record(s) from in the Archive.
 Clear all Records – is used to remove all archived files.
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Figure 152: Call History – Archive page
CDRs listed in the Call History Archive table are characterized by the following specifications:
 Archive Records – shows the archived record (file) name which is actually the archiving date and time.
Click the hyperlinked [csv] or [log] to download the archived file.
 Number of Call Records – shows the number of call records in the archived file.
 External Backup Status – shows the status of the archived file backup. The following statuses are
available:
 Success – if the archived file has been successfully sent for backup (e-mail address, FTP or TFTP
server).
 Failed – if the archive file failed to be sent for backup (e-mail address, FTP or TFTP server). The Try to
send now link will appear next to this status allowing to repeat the backup process.
11.3.5 Archiving Settings

The Call History Archiving feature is used to configure the automatic archiving of the Call History. The following
options are available for archiving:
 Percentage of Total Memory allocated for Archive – defines the system memory allocated for call history
archiving.
 Enable Call History Archiving – is used to enable the service.
 File Format – is used to select the archive file format as (*.log) and (*csv).
Archiving Mode
This section is used to select the archiving mode. The following modes are available:
 Archive by Record Count – file is being archived as soon as the number of records specified in the drop-
down list is collected.
 Archive by Time Interval – file is being archived as soon as the timeframe specified in the drop-down list
is elapsed from the last archiving. If no CDRs were produced during that timeframe, archive file for that
period will not be generated.
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Figure 153: Call History – Archiving Settings page
Archiving Storage Settings
This section is used to select archiving storage and configure the backup settings.
 Archiving Storage Mode – is used to select one of the following archiving options:
 Do not send – the CDRs will be archived and kept locally only.
 Send and keep locally – the CDRs will be sent to the server and kept locally.
 Send and delete from archive – the CDRs will be sent to the server and removed from the archive.
 The following options are available for storing archived CDRs:
 Send via E-mail – allows sending the archived files via e-mail. The destination e-mail address has to be
entered in the E-mail Address field.
 Send to Server – allows sending the archived files to an external server. This selection enables the
following fields to be filled:
 Server Name – the IP address or hostname of the server.
 Server Port – the port of the server.
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 Path on Server – the path on the server.

 Send Method – the server type: TFTP or FTP. Specify the Username and Password in case of the
FTP. If these fields are left empty, anonymous authentication will be used. TIP: Select the Use SFTP
option to enable SFTP support.
 The Archive Now button is used to archive CDRs immediately.
11.3.6 RTP Statistics

The RTP Statistics page provides detailed information about the established call is provided. When QX serves
as an RTP proxy, this page displays two groups (legs) of RTP statistics. For example, when calling from an IP
Phone attached to the QX’s IP line to an external SIP destination or from one external SIP destination to
another through the QX’s Auto Attendant. Each group of parameters describes characteristics of a piece of
RTP stream composing an overall SIP session. Normally, one leg describes the RTP stream from caller to the
QX and the other leg describes the RTP stream from QX to the destination.
 Quality – indicates the call quality, which depends on RTP statistic. Below is the legend for Call Quality
definitions on the displayed RTP Statistics:
 excellent – RX Lost Packets < 1% & RX Jitter < 20
 good – RX Lost Packets < 5% & RX Jitter < 80
 satisfactory – RX Lost Packets < 10% & RX Jitter < 150
 bad – RX Lost Packets < 20% & RX Jitter < 200
 very bad – RX Lost Packets > 20% or RX Jitter > 200
 Local and Remote – indicate the two peers between which the RTP stream is transmitted. The
characteristics in the table below describes to the piece of RTP stream between these peers.
 Rx/Tx Codec – codec for received and transmitted RTP stream respectively.
 Rx/Tx Packets – is the number of RTP packets received and transmitted respectively.
 Rx/Tx Packet Size – is the size of RTP packets (payload) received and transmitted respectively.
 Rx Lost Packets – is the number of lost RTP packets for received stream.
 Rx Jitter – inter-arrival jitter is an estimate of the statistical variance of the RTP data packet inter-arrival
time, measured in timestamp units.
The inter-arrival jitter is defined to be the mean deviation (smoothed absolute value) of the difference D in packet
spacing at the receiver compared to the sender for a pair of packets. If Si is the RTP timestamp from packet i,
and Ri is the time of arrival in RTP timestamp units for packet i, then for two packets i and j, D may be
expressed as:
D(i,j) = (Rj - Ri) - (Sj - Si) = (Rj - Sj) - (Ri - Si)
J(i) = J(i-1) + (|D(i-1,i)| - J(i-1))/16, where J(i) is Rx Jitter for packet i.
For more details about Jitter calculations, please refer to the RFC1889.
 Rx Maximum Delay – is the maximum variance (absolute value) of actual arrival time of the RTP data
packet compared to estimated arrival time, measured in milliseconds. If Si is the RTP timestamp from
packet i, and Ri is the time of arrival in RTP timestamp units for packet i, then variance for packet i may
be expressed as following:
V(i) = |(Ri - R1) - (Si - S1)| = |(Ri - Si) - (R1 - S1)|
Rx Maximum Delay = max V(i) / 8
 RX Delay Increase Count – indicates the number of times the delay in jitter buffer is increased during the
call.
 RX Delay Decrease Count – indicates the number of times the delay in jitter buffer is decreased during
the call.
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 Configure Call Quality Event Notification – leads to the Call Quality Notification page to configure call
quality control notification specifics.
 Configure System Events – leads to the Event Settings page to configure the methods of notification for
each system event.
RTP Statistics is logged only when at least one of the call endpoints is located on the QX. For example, it will
not be logged when:
 Calls incoming from or addressed to the IP lines or remote extension.
 Calls from an external user are routed to another external user through QX’s routing rules.
In the first case, RTP statistics will be logged if remote extension or IP line user is calling locally to the QX’s
extension or auto attendant.
11.3.7 FAX Statistics

The FAX statistics page is accessed from the Call History page by clicking on the FAX details link in the Details
column for the calls that contain T.38 FAX transmission. This page provides information about received and
transmitted packets, lost, bad and duplicated packets. These statistics refers only to the T.38 FAX
transmission. The FAX statistics is not available for the FAX transmitted with other protocols.
11.4 Conference History

The Conference History allows to track and report the details of the conference calls that have been activated
on QX.
For more information on Audio-Video conferencing, please refer to the Audio-Video Conferencing on QX IP
PBXs guide.
11.5 Network Interfaces

The Network Interface Statistics pages display the corresponding statistics.
 LAN – current activity of the LAN (Local Area Network).
 WAN – current activity of the WAN (Wide Area Network).
 VLAN – current activity of the VLAN.
 PPTP/L2TP – current activity of the PPTP/L2TP.
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Figure 154: LAN Interface Statistics page
The table displayed here shows the number of receive and transmit events that occurred since the last resetting
of the counters by clicking the Clear button. Depending on the Watch LAN, Watch WAN, Watch VLAN, Watch
PPP link selected on the Network Status page, the LAN Interface Statistics, WAN Interface Statistics, VLAN
Interface Statistics, PPTP or L2TP statistics page will be displayed. The page is automatically refreshed every
minute. Additionally, Refresh allows to initiate manual.
11.6 Statistics
11.6.1 Network Transfer
The Transfer Statistics page shows a user-defined statistics table with the transmit/receive value (criteria),
interface type and time period. It contains the following components:
 Time range of statistic table – includes the period (in days) statistics data that is to be collected and the
corresponding diagram charts that are to be built.
 Interface drop-down list (N/A on QX2000 and QX3000) offer the values:
 LAN – show current activity of the LAN (Local Area Network).
 WAN – current activity of the WAN (Wide Area Network).
 VLAN – show current activity of the VLAN.
 PPTP/L2TP – show current activity of the PPTP/L2TP.
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Figure 155: Transfer Statistics page
 Show also as readable values – if selected, an additional table with statistics values will be displayed on
the next page.
 Receive Bytes – is the number of received bytes.
 Receive Packets – is the number of received Ethernet packets.
 Receive Errors – is the number of received packets containing errors.
 Receive Drop Errors – is the number of received packets that have been discarded.
 Receive Overrun Errors – is the number of received overrun errors that occur when the receive buffer is
not large enough to hold all incoming packets. This error usually appears due to a slow receiving system.
 Receive MultiCast Packets – is the number of received broadcast packets.
 Transmit Bytes – is the number of transmitted bytes.
 Transmit Packets – is the number of transmitted Ethernet packets.
 Transmit Errors – is the number of transmitted packets containing errors.
 Transmit Drop Errors – is the number of transmitted packets that have been discarded.
 Transmit Carrier Errors – is the number of transmit carrier errors that occur due to a defective or lost
connection on the Ethernet link.
 Transmit Collisions – is the number of transfer errors that occurred during a simultaneous packet
transmission from both sides.
 Reset Statistics – is used to reset the chart and the table (if enabled).
To see the Transfer Statistics Diagram Charts, select the desired criteria and click Show to generate the
corresponding chart and the table showing the transfer statistics values (if enabled). The letters M (millions) and
K (thousands) used in the legend of the displayed diagrams show the total number of specified criteria.
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11.6.2 PSTN Channel Usage

The PSTN Channel Usage page is used to display diagram charts for the selected onboard lines or trunks.
 Line or trunk number –
is used to select the
line(s) or trunks for
which the diagram chart
will be generated.
 Time range of statistic
table – lists the period
(in days) statistics data
that is to be collected
and the corresponding
diagram chart that is to
be built.
 Incoming Calls and
Outgoing Calls – are
used to select whether
Figure 156: FXO Channel Usage Statistics page
the FXO or ISDN
(depending on the QX
model) traffic statistics for only incoming or outgoing or for both type of calls should be displayed in the
diagram chart.
 Maximum Active Calls – is used to have the number of maximum active calls displayed in the diagram
chart. At least one of these checkboxes should be selected.
Click the Show button to generate an FXO or ISDN (depending on the QX model) channels usage diagram chart
for the selected parameters.
Note: The PSTN Channel Usage page is not applicable on QX20, QX500 and QX2000.
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12 Maintenance Menu
Figure 157: Maintenance Menu overview
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12.1 Diagnostics
The Diagnostics page gives a possibility of running Network protocol diagnostics to verify QX's connectivity and
download all system logs for possible problems recovery.
Figure 158: Diagnostics page
 Start Network Diagnostics – is used to initiate network diagnostics, i.e., to check the WAN link and IP
configuration, to verify gateway, DNS primary and secondary (if configured) servers' accessibilities.
 Start FXO Diagnostics (available only for QX50/QX200) – runs FXO diagnostic tests to determine the
optimal value for the FXO country specific regional setting (CSRS) appropriate to your PSTN provider.
Once the FXO diagnostic is complete, the recommended value should be set manually on the
"fxocfg.cgi" hidden page. Setting this value may resolve echo or poor audio quality issues on FXO lines.
 Start ISDN Diagnostics (available only for QXISDN4+) – runs ISDN diagnostics test to initiate ISDN BRI
low level diagnostic. With these tests, the ISDN physical link is checked and the Frame Synchronization
is verified.
 Download System Logs – is used to download all logs to the local PC as a (*.tar) archive file. These logs
can then be used by Epygi Technical Support to determine the problem that has occurred on your QX.
12.1.1 Security Diagnostics

The Security Diagnostics page allows running the security audit and getting the security reports.
Figure 159: Security Diagnostics page
 Start Security Audit – is used for running the security audit. The QX Security Audit is a security reporting
system, which generates the warnings regarding the QX's weaknesses relative to the selected Security
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Level. The warnings may vary depending on the selected global Security Level. The Security Audit will
detect the security related configuration issues in Firewall, IDS, IP Line passwords, Call Routing and
extension settings.
 Show Security Report – allows to display the last security audit report.
 Following useful links are available to adjust the system security:
 User Rights Management
 IP Lines
 Firewall/NAT
12.1.2 Call Capture

The Call Capture is used to capture the calls to/from onboard interfaces. You can capture calls on the following
interfaces FXS, FXO or ISDN (depending on the QX model). This page consists of two sub-pages:
 Active Calls – sub-page lists all active calls on the QX for the certain moment.
 Interfaces – sub-page lists all available interfaces on the QX.
Figure 160: Call Capture - Interfaces subpage
To start the call capture:

1. Select the checkbox next to the call, which should be captured from Active Calls sub-page or select the
available interface from Interfaces sub-page.
2. Configure the Capture Timeout, during which the call will be captured. TIP: The call capture will
automatically be stopped, when the capture timeout expires.
3. Click Start, to start capturing.
4. Click Stop, to stop capturing and download the captured file.
The captured call will be downloaded in the (*.tar) format. It contains two streams (receive and transmit) of the
captured call. These streams can be then played with an audio player application.
Note: The Call Capture duration is limited to 160 seconds.
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12.1.3 Network Capture

The Network Capture is used to capture packets for the selected network interface. The following options are
available:
 Capture on Interface – select the interface to capture packets. The Local Loopback Interface option is
used to capture the traffic within the unit.
 Stop after receiving count packet – number of packets to be captured.
 Restrict to Host – packets can be captured for only the specified IP address.
Figure 161: Network Capture page
 Capture all Packets – allows capturing all packets on the selected interface.
 Capture Protocol- specific Packets – enables restricting the capture to specific packets only (ARP, SIP,
DNS, and RTP).
To start network capture:
1. Select the Interface.
2. Configure restriction parameters, if needed.
3. Select packets to capture: all or specific ones.
4. Click Start, to start capturing.
5. Click Stop, to stop capturing and download the captured file.
Note: The Network Capture size is limited to 24 MB. This will put a limitation on the duration of captured file.
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12.1.4 Ping
Ping sends four ICMP (Internet Control
Message Protocol) requests with a default size
of 64 bytes to the destination (IP address or
host name) specified in the Ping Target. The
response times are logged, and the round-trip
time (the time required from being sent until
being received again) is measured. The
minimum and maximum round trip time and its
average as well as the percentage of lost and
of received frames results are displayed in the
lower area of the page.
To ping a target:
1. Enter the destination’s IP address or
hostname in the Ping Target field.
2. Click Start Ping. Figure 162: Ping page
3. The results of the ping will be displayed
in the Ping Output window.
12.1.5 Traceroute
Traceroute checks the Internet
connection by triggering the routers
(hops) that are passed to reach to the
defined. Trace routing gives feedback on
the routers passed by packets on the
way toward the destination and the
round-trip delay of packets to these
routers.
To traceroute a target:
1. Enter the destination’s IP address
or hostname in the Traceroute
Target field.
2. Select the Use ICMP checkbox to
send an ICMP request the ping
destination (MS Windows
standard), otherwise a UDP
request will be send (Linux
standard).
Figure 163: Traceroute page
3. Click Start Traceroute.
4. The results of the ping will be
displayed in the Traceroute Output window.
Attention: No Traceroute is possible if the Firewall level is set to "High". For the purpose of tracerouting, several
IP packets are sent out. UDP is used to send packets and ICMP is used to receive information about the
routers. In their headers, the TTL value increases from 1 to 30. When the first IP frame is received by the first
router, its IP address will be returned in its acknowledgement.
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12.2 System Logs

The System Logs page displays the generated logs on the QX. System logs are useful to determine any kind of
problems on the QX as well as to monitor the user’s access and the usage of it. On the left side of the page, a
list of main logs is displayed. Clicking on the needed link will display the most recent log lines. The number of
log lines displayed on this page is set on the System Logs Settings page.
The text field on the left side is dedicated for support personnel only and is used to search a custom log not
listed on this page. To do so, enter a required log name to the text field and click Show Custom Log.
If the user has used Logs Collection  feature code after or during (from another phone connected to the
same QX) the call, a special log file will be generated containing the details of that call and few last calls done in
the system. This log file will be internally kept in the system until the next time someone used the Logs
Collection feature code again. The collected logs will be a part of the System Logs when user downloads them
next time. This could be used to collect the logs at the exact moment when a problem happens.
12.3 System Logs Settings

The System Logs Settings page is used to adjust system logging settings.
 Enable User Logging – enables user level
logging. This logging contains brief
information about events on the QX.
 Enable Developer Logging – enables
developer high level logging. This logging
contains detailed information about events
on the QX.
 Log Lines to Show – is used to select the
maximum number of log lines to display on
the System Logs page.
 Mark all Logs – is used to set a line marker
in the logs. If you need to follow a certain
piece of log, push this button to set a
starting mark in all logs and then perform Figure 164: System Logs Settings page
the needed actions over the QX. When the
actions are done, push this button again to set an ending mark in all logs. This way you shall clearly see a
piece of log between the starting and ending marks generated during the certain actions taken over the
QX.
 Comment – is used to enter some text information which will be displayed next to the marks entered in the
logs. This comment may describe the problem captured in the following logs and may be useful for the
Technical Support.
 Download all Logs – is used to download all logs to the local PC as a (*.tar) archive file. These logs can
then be used by the Epygi Technical Support to determine the problem that has occurred on your QX.
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12.4 Remote Logs Settings

The Remote Logs Settings page (N/A for
QX2000/QX3000) is used to adjust the system logging
settings and contains the following components.
Enable Remote Logging – enables remote monitoring
of the QX’s logs. When this option is selected,
remote administrators may connect the QX with
Telnet protocol (port number 645) and access the
logs selected on this page. This is done for remote
the QX’s diagnostics and is mainly used by Epygi’s
Technical Support. To make the QX’s logs open for
remote access, appropriate Firewall level or Filtering
Rules must be created. The options below are used
to select those log types that should be accessible
remotely. Select only those logs that you wish to
have monitored remotely.
Figure 165: Remote Logs Settings page
12.5 Logs Archive

The System Logs Archive page (available only on QX2000 and QX3000) shows the archived logs table with time
period by Date. Clicking on the corresponding date will open the archived system logs table in hourly basis.
Hour shows the initiation time of the system logs. This could be used to collect the logs at the exact moment
when a problem has happened. The Unpacked size on disk shows the system logs size on disk for the
corresponding Date and Hour.
Figure 166: System Logs Archive page
 Download – is used to download the archived system logs file to the PC and opens the file-chooser
window where the saving location can be specified.
 Delete – removes the selected entry from the archived system logs table.
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12.6 User Rights Management

The User Rights service sets restrictions on the GUI access for various users, permits or denies the access to
certain Web GUI configuration pages and creates multilevel user management of the QX.
12.6.1 Users
The Users page contains a table where the Administrator and Local Administrator accounts are listed. This
page allows to modify the passwords of Administrator and Local Administrator accounts. Two levels of QX GUI
administration are available:
 admin – this is the Administrator’s account. The administrator has access to all Web GUI pages and no
one else has configuration permission to adjust this account. The administrator is responsible for
granting access to all other user groups. By default, as well as after factory reset of QX, the admin
password is set to 19.
 localadmin – this is a common sub-administrator’s account. Local Administrator has permission to
access and adjust each GUI management page. But the account of Local Administrator is disabled by
default and after each factory reset. By default, as well as after factory reset of QX, the localadmin
password is set to 19.
Figure 167: User Rights Management – Users page
To change the GUI Access Password:

1. Click the checkbox next to the admin or localadmin entry in the table and click Change Password.
2. The Change Password page appears for selected user. Select GUI Access Password tab.
 Enter the old password (by default – 19)
 Enter a new password and then re-enter it to confirm.
3. Click Save. The password has now been changed.
The Phone Access Password which is required for Administrator Login (). The Administrator Login is
used to review and modify the Auto Attendant greeting and recurring prompt, as well as the universal extension
messages.
To change the Phone Access Password:
1. Click the checkbox next to the admin entry in the table and click Change Password.
2. The Change Password page appears for selected user. Select Phone Access Password tab.
 Enter a new password and then re-enter it to confirm.
3. Click Save. The password has now been changed.
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Note:
 The GUI access password can consist of lowercase and uppercase alphabetic characters, digits and
symbols. A maximum password length is 20 characters.
 The Phone access password can consist of only digits. A maximum password length is 20 characters.
 In order to keep the Administrator’s password safe, do not write it down in public places and do not
share it with other people.
12.6.2 Roles
The Roles page contains a table where the Local Administrator and Extensions role are listed. This page allows
you to set the permissions to the GUI pages for each role in the table.
 Local Administrator – this role can have permissions to adjust each GUI page.
 Extension – this role refers to all extensions created on the QX. Permissions for an extension to access
each GUI page can be adjusted.
Figure 168: User Rights Management – Roles page
To manage the permissions for the selected role:

1. Click the hyperlinked role (Extension or Local Administrator). The Access Rights page will be opened.
2. Select the checkbox(es) next to CGI Name.
3. Click the Grant Access or Deny Access to grant/deny access to the corresponding page.
Figure 169: Access Rights – extensions page
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12.6.3 Hotel Console User

The Hotel Console User Rights Management page is used for managing the users allowed to connect to the QX
from the Epygi Hotel Console (EHC) application.
For more information, how to configure and use EHC application with QX, please refer to the Epygi Hotel
Console (EHC) - User Guide.
Figure 170: Hotel Console User Rights Management page
To add a new user:

1. Click Add and enter the following information.
 Leave the Enable checkbox selected to enable the new user immediately or unselect the checkbox to
keep the user’s status disabled.
 Username and Password – define the authentication parameters to allow user to login into the EHC
application by defined credentials.
2. Click Save, the new user will be added to the Hotel Console User Rights Management table.
3. Click the Enable and Disable button to enable/disable the selected user.
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12.7 Backup / Restore

12.7.1 Backup / Restore
The Configuration Management includes the features allowing to back up and save the QX's current
configuration, restore the configuration from backups created earlier, as well as to restore the system default
configuration. The following options are available:
Figure 171: Configuration Management page
 Backup and download current Configuration – this option is used to create a backup file with all current
configuration settings and system voice messages (default and customized). Click the Download button
to back up and download the current configuration. The file will be saved in the (*.bin) format. The
backup filename will have the following format: config_[Hostname]_[Firmware Version]_[Date/Time].bin
 Backup and download current configuration including EAC data – this option is also used to create a
backup file with all current configuration settings and system voice messages (default and customized).
Compared to the previous option the current configuration includes the EAC data, covering the EAC
Chat database, Agents Status and Call Statistics. Click the Download button to back up and download
the current configuration. The file will be saved in the (*.bin) format. The backup filename will have the
following format: config_[Hostname]_[Firmware Version]_[Date/Time].bin
Note: Voice Mails and Call Recordings are not backed up and included in the configuration file.
 Restore previously backed up configuration – this option is used to restore earlier created backup file and
replace the current configuration settings and system voice messages.
1. Click the Upload button.
2. Click Choose File to open the file chooser window and browse for the file.
3. Click Save to start configuration restore.
Note: The QX's doesn't allow to restore the earlier created backup in case it is running a firmware version lower
than the version at the time of configuration backup.
 Restore to Factory Default settings – this option is used to reset all configuration settings and restores the
device's factory default settings.
1. Click the Reset button.
2. Click Yes to proceed the factory reset procedure.
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Note: Unlike the factory reset done by pressing the Reset pin on the QX manually, this option will keep the
following data:
 The device registration with Epygi Technical Support.
 The installed license keys.
12.7.2 Automatic Backup

The Backup Configuration Management feature allows to activate and configure the automatic backup of the
current configuration and system voice messages (default and customized).
Figure 172: Automatic Backup page
The following options are available for automatic backup:

 Enable Automatic Backup – is used to enable the service.
 Include EAC Data – is used to include the EAC data, covering the EAC Chat database, Agents Status
and Call Statistics in the backup file.
 Send via Email – allows sending the backup file via e-mail. The destination e-mail address has to be
entered in the E-mail Address field.
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 Send to Server – allows sending the backup file to an external server. This selection enables the
following fields to be filled:
 Server Name – the IP address or the hostname of the server.
 Server Port – the port of the server.
 Path on Server – the path on the server.
 Send Method – the server type: TFTP or FTP. Specify the Username and Password in case of the FTP.
If these fields are left empty, anonymous authentication will be used. TIP: Select the Use SFTP option
to enable SFTP support.
 Backup Interval Selection – is used to schedule the automatic backup.
 Backup Now – is used to backup of configuration and system voice messages (default and customized)
immediately.
12.7.3 Download Legible Configuration

The Legible Configuration service allows to generate a piece of QX configuration, download it to review and
make necessary changes, then upload back to update the configuration. The downloaded legible configuration
file(s) (LCF) contain QX configuration parameters in the (*.txt) file. LCF can be edited with any text editor and
uploaded back to save the changes on the same or another QX system(s).
For information on how to configure and use Legible Configuration service, please refer to the Legible
Configuration on QX IP PBXs guide.
Figure 173: Download Legible Configuration page
The following radio buttons are used to select between a specific CGI or a group of CGIs:
 Single Page – is used to select a certain page from the list of QX’s Web management pages for which
the legible configuration can be manually managed. For example, selecting "RTP Settings" will generate
a legible configuration file with parameters present on the RTP Settings page.
 Group of Web Pages – is used to choose among the four predefined groups: Internet Connection
Settings, LAN Configuration Settings, Telephony General Settings and Extension Settings. Each of these
groups refer to all pages characterized by the selected criteria, e.g. Internet Connection Settings group
contains all parameters on the pages related to the networking and WAN configuration.
Revision 1.2 201

 Extension – is used select the settings in the generated legible configuration file to one specific
extension. For example, each of the extensions on the QX have own SIP settings or Codecs. To
download the settings for a particular extension only, you need to choose the corresponding extension
from the list. The drop-down may also have a blank selection. In that case, the legible configuration file
will contain the parameter of all available extensions on the QX (if the selected parameter applies to the
extension and not to the overall system, like RTP settings).
The following functional buttons are available:
 Start generate a legible configuration file – starts parsing the configuration structure of the device for the
defined parameters. The progress will be displayed in the window.
 Cancel generation process – stops the generation procedure. This button appears once the configuration
generation procedure has been started.
 Download generated configuration! – is used to download the generated file in the (*.txt) format. This
button appears when the legible configuration generation is finished. Necessary changes can be made in
the downloaded configuration file and then uploaded back to the system.
 View generated configuration! – is used to view the generated file directly in the browser. This button
appears when the legible configuration generation is finished.
 Restart generation! – is used to cancel the generated configuration file and start over. This button
appears when the legible configuration generation is finished.
12.7.4 Upload Legible Configuration

The Upload Legible Configuration page is used to upload a configuration file in the (*.txt) format.
Figure 174: Upload Legible Configuration page
 Choose File – is used to browse certain legible configuration file to be uploaded and updated into the
system. The file uploading progress will be displayed in the window.
Checking the Validity of a LCF
Before applying the changes specified in the LCF, QX checks the validity of the uploaded LCF. First, the QX
compares the FW version indicated in the LCF with the currently running one on the QX. If they match, the QX
will proceed checking the correctness of the specified settings similarly as it does when the user presses the
Save button to submit the changes. At any point, the QX detects a mistake  a version mismatch, the wrong
value for a setting, a wrong syntax, it will generate an error and delete the LCF without applying any change. If
no mistakes are found in an edited LCF, the QX will start to sequentially apply the changes.
Revision 1.2 202

12.8 Firmware
The Firmware section is used to update the firmware of QXs. Following options are available for updating the
current firmware:
 Upload and update firmware manually.
 Download and update firmware manually.
 Download and update firmware automatically
For more information on how to update the firmware of QX, please refer to the Firmware Update Service on
Epygi QX Line guide.
Attention:
 It is recommended to back up the configuration for emergency purposes prior to upgrading the firmware.
You can do that by clicking the Download Configuration link in the Manual Firmware Update page. The
current configuration will remain after the firmware update. Moreover, voice mails, call recordings, all
custom messages and call history will be saved during the upgrade.
 Firmware installation will take about 5 minutes. During that time, QXs will be in non-operational condition,
neither telephony nor Internet access is possible.
 You will not be automatically redirected to the Login page. To access the QX’s Web GUI, connect to an
QX again and login.
 The QX will factory reset and the system configuration will be lost while downgrading the firmware.
 After the firmware update, all IP phones attached to the QX will be restarted.
12.8.1 Manual Firmware Update

The Manual Firmware Update page is used to upload and update the QX firmware manually.
Figure 175: Manual Firmware Update page
Revision 1.2 203

The recommended manual firmware update procedure is:

1. Go to the MaintenanceFirmwareManual Firmware Update page.
2. Click the Download Configuration link to back up the current configuration (recommended).
3. Click the Choose File button to browse for image.bin file.
4. Click Save to start uploading the file.
5. Click Yes to proceed the firmware upgrade.
Note: The update process takes about 5 minutes. Normal operation will be stopped during that time.
The following information will be displayed when firmware upload finished:
 Firmware check – show the status of uploaded firmware. Status Invalid means that the uploaded
firmware is not compatible with the QX hardware version.
 Current Firmware Version/New Firmware Version – show the current/new firmware versions accordingly.
 Click Yes to proceed the update or click Discard this firmware to close the message without updating the
device.
12.8.2 Get Firmware From Server

The Manual Firmware Update from Server page is used to manually download and update the QX firmware from
the FTP Server.
Figure 176: Manual Firmware Update from Server page
The following information and functions are available in this section:
 Last Status – displays the date/time and firmware version for the last update.
 Firmware URL – is used to define the URL for the firmware on the FTP server.
 Username and Password – are used to define the authentication parameters for the FTP server.
 Save – keeps the changes before Download or Download and Update.
Revision 1.2 204

 Download – starts downloading firmware from FTP Server.

The following information will be displayed when firmware download finished:
 Firmware check – shows the status of uploaded firmware. Invalid status means the firmware is not
compatible with the QX hardware version.
 Current Firmware Version/New Firmware Version – show the current/new firmware versions accordingly.
 Update – is used to proceed the update or click Discard to close the warning message without updating
the device.
 Download and Update – is used to automatically download and update the firmware from the FTP
server.
12.8.3 Automatic Firmware Update

The Automatic Firmware Update page is used to enable and configure the automatic firmware update settings
on the QX. When this service is enabled, on the scheduled time QX will automatically check if a new firmware is
available on the server. Then, based on the preconfigured settings, will notify user or update the firmware
immediately. The following components and functions are available:
 Enable Automatic Firmware Update – is used to enable Automatic Firmware Update service.
 Server Name – enter the IP address or hostname of the server.
 Server Port – enter the port of the server.
 Update Method – select the desired update method (FTP, HTTP or HTTPS).
 Username and Password – are used to define the server authentication parameters.
Figure 177: Automatic Firmware Update page
Note: The server configuration can be done manually. The recommended and simplest method is to use the
Epygi’s public FTP server.
Revision 1.2 205

Check for updates based on one of the following options:

 Select the Check and notify option if you want QX to check for a new firmware in the server at the
scheduled time and notify.
 Select Check and update option if you want QX to check for a new firmware, automatically download
and install it on a scheduled time.
 Click Check Now to manually initiate the action selected from the Check for updates drop-down list.
To perform the automatic firmware update from Epygi's FTP server:
1. Select the Enable Automatic Firmware Update option.
2. Leave the Server Name, Server Port, Update Method, Username and Password text fields to their default
values (ftp.epygi.com, 21, ftp and anonymous respectively, use blank for password) to use Epygi's
public ftp server.
3. Select the "Check and update" option from the Check for updates drop-down list.
4. Configure the Date/Time settings.
5. Click Save.
The system will check for a new firmware at scheduled time. If there is a new firmware available, the QX will
download and update it automatically.
12.9 Reboot
The Yes, Reboot Device button is used to reboot the QX. TIP: The session with the QX will be closed, i.e., the
QX's GUI should be newly opened and a new login will be required afterwards.
Figure 178: Reboot Device page
Revision 1.2 206

12.10Registration Form
The Register Your Device in Technical Support Center page appears when administrating an unregistered QX,
and it has been created for customer support purposes. The page requires customer registration at the Epygi
Technical Support Center. It provides several links offering the following registration options:
Figure 179: Device Registration page
 Register now – leads to the Epygi Technical Support System Registration page and requires customer’s
information to submit the QX registration form.
 Remind me later – hides the registration notification in the QX until the next administrating activities.
 Don’t remind me again – hides the registration notification forever.
Revision 1.2 207

13 Appendix: Administrator Login

The Administrator Login is used to review and modify the Auto Attendant greeting and recurring prompt, as well
as the universal extension messages. Phone Access Password will be required for login.
1. Dial  to login.
2. Enter the Phone access password.
3. Follow the voice prompts to review and change system messages.
4. Dial  or hang up to logout.
System will notify about the messages that can be reviewed and modified.
Administrator Login menu
  
Review Review Review Universal Extension Messages
Attendant Attendant
Greeting Recurring
Prompt
Enter the Enter the

Attendant Attendant     
Voice Mail Incoming Call Outgoing Call Out of Office Find Me/
Number Number
Greeting Blocking Blocking message Follow Me
(in case of (in case of
message message message message
multiple AAs) multiple AAs)
      
Listen to the Listen to the Listen to the Listen to the Listen to the Listen to the Listen to the
current current current current current current current
greeting prompt message message message message message
      
Record a new Record a new Record a new Record a new Record a new Record a new Record a new
      
Restore Restore Restore Restore Restore Restore Restore
system default system default system default system default system default system default system default
      
Stop recording Stop recording Stop recording Stop recording Stop recording Stop recording Stop recording
or playback or playback or playback or playback or playback or playback or playback
Table 6: Administrator Login menu
Revision 1.2 208

14 Appendix: Needed Bandwidth for IP Calls

The bandwidth required by an IP call depends on the codecs used and these specifications are listed in the
tables below.
Packet Size (in msec)

Codecs
10 20 30 40 50 60
G.711u/G.711a 105 84 76 74 71 67
G.726-16 58 37 30 27 25 22
G.726-24 66 45 38 34 32 30
G.726-32 74 53 45 42 40 37
G.726-40 82 61 53 50 48 45
G.729a 50 29 22 19 17 15
iLBC – – 27 – – 20
G.722 105 84 76 74 71 67
G.722.1 74 53 45 42 40 37
Table 7: Required Bandwidth for Standard Packets

Codecs
10 20 30 40 50 60
G.711u/G.711a 114 89 81 76 74 72
G.726-16 66 41 33 28 26 24
G.726-24 74 49 41 36 34 32
G.726-32 82 57 49 44 42 40
G.726-40 90 65 57 52 50 48
G.729a 58 33 26 20 18 16
iLBC – – 31 – – 22
G.722 114 89 81 76 74 72
G.722.1 82 57 49 44 42 40
Table 8: Required Bandwidth for Encrypted Packets when using a SRTP

Codecs
10 20 30 40 50 60
G.711u/G.711a 148 105 90 85 80 74
G.726-16 95 59 43 38 34 29
G.726-24 108 65 52 45 41 37
G.726-32 118 74 60 53 48 45
G.726-40 124 81 66 61 56 52
G.729a 92 49 35 30 26 22
iLBC – – 41 – – 26
G.722 148 105 90 85 80 74
G.722.1 118 74 60 53 48 45
Table 9: Required Bandwidth for Encrypted Packets when using a VPN
Revision 1.2 209

15 Appendix: System Default Values

15.1 System Settings
Page/Wizard/Section Option/Parameter Default Value QX Model

Username admin
GUI Password 19
Phone Access Password 19
admin – enabled
User Rights Management Users All
localadmin – disabled
Extension – all accessible pages are
granted access
Roles
Localadmin – all accessible pages are
granted access
Hostname epygiqx All
Domain Name epygi-config.loc All
QX20/QX50/QX200/
172.30.0.1
System Configuration LAN IP Address QX500/QXISDN4+
192.168.0.200 QX2000/QX3000
QX20/QX50/QX200/
Subnet Mask 255.255.0.0
QX500/QXISDN4+
255.255.240.0 QX2000/QX3000
DHCP Settings for the LAN Interface DHCP Server Disabled All
Your locale (location) US
Regional Settings and Preferences (GMT-06:00) Central Time All
Timezone
(US&Canada)
Emergency Codes and Emergency Code 911
QX50/QX200/QXISDN4+
PSTN Access Code PSTN Access Code 9
WAN Interface Protocol Ethernet All
100.000 k/bits QX50/QX200/QXISDN4+
Upstream
1.000.000 k/bits QX20/QX500/QX2000/QX3000
Uplink Configuration
100.000 k/bits QX50/QX200/QXISDN4+
Downstream
1.000.000 k/bits QX20/QX500/QX2000/QX3000
Min Data Rate 0 All
WAN IP Configuration IP configuration of the WAN interface Obtain an IP Address automatically All
Revision 1.2 210


MAC Address This device All
WAN Interface Configuration
MTU 1500 Bytes All
Obtain DNS Server Address
DNS Settings DNS configuration All
automatically
SNTP Server Enabled All
SNTP Client Enabled All
Date / Time Settings
SNTP Server ntp1.epygi.com All
Polling Interval 6 hours All
E-mail(SMTP) Settings SMTP Service Disabled All
Short Text Messaging (SMS)
SMS Service Disabled All
Settings
System Security Management Security Level Medium All
Licensed Features – No feature is activated. All
The Redundancy feature isn’t activated
Redundancy Settings – All
by default.
The Custom Language Pack isn't
Upload Language Pack – All
uploaded.
QX20/QX50/QX200/
3
Extension Length QX500/QXISDN4+
4 QX2000/QX3000
Ext. (101-112) attached to
Extensions attachment
the IP lines (1-12) QX20
Percentage of Total Memory Ext. (101-112) – 0.1%
the FXS lines (1-2)
Extension Management QX50
the IP lines (1-16)
(Ext.101-102) – 5%
Percentage of Total Memory
(Ext.103-118) – 0.4%
the FXS lines (1-2)
QX200
the IP lines (1-24)
(Ext.101-102) – 5%
(Ext.103-126) – 0.4%
Revision 1.2 211


the IP lines (1-16) QXISDN4+
Percentage of Total Memory (Ext.101-116) – 0.4%
Extension Management
Display Name None
Password Left blank
Use Kickback Disabled
Allow Call Relay Disabled
User Extensions –General Settings All
GUI Login Allowed Disabled
3pcc/Click2Dial Access Allowed Disabled
Show on Public Directory Disabled
Use Parent Extension Disabled
Username / DID Number Same as the extension number
Password Left blank
SIP Server Left blank
User Extension – SIP Settings All
SIP Port 5060
SIP Registration Transport UDP
Registration on SIP Server Disabled
Authentication Username None
Send Keep-alive Messages
Disabled
to Proxy
User Extension – SIP Advanced RTP Priority Level Medium All
Settings
Do Not use SIP Old Hold Method Disabled
Outbound Proxy Left empty
Secondary SIP Server Left empty
Revision 1.2 212


User Extension – SIP Advanced Outbound Proxy for Secondary SIP
Left empty
Settings Server
The Remote Extension service is
User Extension – Remote Settings – All
disabled.
User Extension – Call Queue
– The Call Queue service is disabled. All
Settings
Use Internal Voice Mail Selected for all extensions
User Extension – Voice Mailbox
Configuration wizard status Activated All
Settings
Shared Mailbox Undefined
User Extension – Class of Service The Class of Service is not activated by
– All
Settings default, no available CoS to attach.
The Call Cost feature isn’t activated by
User Extension – Credit Settings – All
default.
The DCC Pro, DCC Basic and iQall
User Extension – Licensing – Toggling features aren’t activated by All
default.
G711u, G711a and G729 Enabled
Preferred Codec G711
G726-16, G726-24, G726-32, G726-
Disabled
40, iLBC, G.722, G.722.1, TDVC
H.263, H.263+ and H.264 Disabled
Out of Band DTMF Transport Enabled
User Extension – Codecs All
T.38 FAX Enabled
Pass Through FAX Enabled
Pass Through Modem Disabled
Force Self Codecs Preference for
Disabled
Inbound Calls
SRTP Policy Make unsecure calls, accept anything
Display Name Attendant
Enable FAX forwarding Disabled All
Show on Public Directory Enabled
Attendant 00 – General Settings
QX20/QX50/QX200/
5%
Percentage of System Memory QX500/QXISDN4+
0.08% QX2000/QX3000
Attendant 00 – Attendant Settings Schedule feature Disabled All
Revision 1.2 213


Attendant 00 – Attendant Settings Scenario Standard All
Pass Dialed Digits through Call
Disabled
Routing
Call Redirection Disabled
Attendant 00 – Attendant Scenario ZeroOut Redirection Disabled All
Welcome Message Enabled
Welcome Message and Recurring
Default message
Prompt
Attendant 00 – Attendant Ringing
Ringing Announcement service Disabled All
Announcement
Username / DID Number 00
Password Left blank
SIP Server Left empty
Attendant 00 – SIP Settings All
SIP Port 5060
SIP Registration Transport UDP
Registration on SIP Server Disabled
Authentication Username None
Send Keep-alive Messages to Proxy Disabled
RTP Priority Level Medium
Attendant 00 – SIP Advanced Do Not use SIP Old Hold Method Disabled
All
Settings Outbound Proxy Left empty
Secondary SIP Server Left empty
Outbound Proxy for Secondary SIP
Left empty
Server
G711u, G711a and G729 Enabled
Preferred Codec G711
G726-16, G726-24, G726-32, G726-
Disabled
40, iLBC, G.722, G.722.1, TDVC
Attendant 00 – Codecs H.263, H.263+ and H.264 Disabled All
Out of Band DTMF Transport Enabled
T.38 FAX Enabled
Pass Through FAX Enabled
Pass Through Modem Disabled
Revision 1.2 214


Force Self Codecs Preference for
Disabled
Attendant 00 – Codecs Inbound Calls All
SRTP Policy Make unsecure calls, accept anything
Dial by Name No entries
Dialing Directories Global Speed Dial No file imported All
Phone Book No file imported
The Conference feature isn’t activated
Conference Management – All
by default.
QX50/QX200/
1%
QXISDN4+
Universal Extension Recordings Percentage of System Memory
0.1% QX20/QX500
0.08% QX2000/QX3000
Receptionist Management – No entries All
The ACD feature isn’t activated by
ACD – All
default.
Authorized Phones – No entries All
IP lines 1-12 – enabled and attached to
QX20
Exts. (101-112) accordingly.
Exts. (103-118) accordingly. IP lines QX50
17-48 – disabled
Exts. (103-126) accordingly. IP lines QX200
25-200 – disabled
IP lines 1-100 – enabled and attached
QX500
IP Lines IP Lines attachment to Exts. (101-200) accordingly.
Exts. (101-116) QXISDN4+
IP lines 17-48 – disabled
to Exts. (1001-1200) accordingly. IP QX2000
lines 201-2000 – disabled
to Exts. (1001-1200) accordingly. IP QX3000
lines 201-3000 – disabled
IP Line Settings PnP for IP lines Enabled All
Revision 1.2 215


Firmware Version Control Enabled
IP Line Settings Configure IP phones from LAN All
Phones Default Template systemdefault
Systemdefault, no manually added
Manage IP Phone Templates – All
entries.
IP Phones Logo – The IP Phones Logo service is disabled. All
FXS Gateway Management – No entries All
FXS line 1-2 – enabled and attached to
FXS Lines FXS Lines attachment QX50/QX200
Exts. (101-102).
Caller ID Type Standard 2
Enable off-hook Caller ID Disabled
Busy Tone and Power Disconnect
Line Settings Disabled QX50/QX200
indications
Ringer Type Type A
Hot Desking Disabled
Loopback is disabled for all FXS lines;
FXS Lines Loopback Settings – QX50/QX200
Loopback timeout is 30.
2 FXO lines – all lines enabled,
incoming and outgoing calls are
– QX50
allowed and routed to 00 Attendant on
all lines.
FXO Settings
4 FXO lines – all lines enabled,
– QX200
all lines.
E1/T1 Trunk Settings On-board E1/T1 trunks are not
– All
available.
4 ISDN trunks – all trunks enabled,
ISDN Trunk Settings – QXISDN4+
all trunks.
Shared PSTN Gateways – No entries All
3 entries defined to call the default Auto
Call Routing Table – Attendant 00, PBX extensions and All
SIP(sip.epygi.com)
Revision 1.2 216


Route all incoming SIP calls to Call
Call Routing Disabled All
Routing
Local AAA Table – No entries All
SIP Tunnel Settings - Tunnels to
Tunnels to Slave Devices Service is disabled, no entries. All
Slave Devices
SIP Tunnel Settings - Tunnels to
Tunnels to Master Devices Service is disabled, no entries. All
Master Devices
The Class of Service is disabled, no
Class of Service – All
entries.
The Call Recording feature isn’t
Call Recording – All
activated by default.
NAT Traversal Settings NAT Traversal for SIP Automatic
NAT Traversal - SIP Parameters UDP Parameters Use STUN
NAT Traversal - RTP Parameters RTP Parameters Use STUN
Primary STUN Server stun.epygi.com
Primary STUN Port 3478
All
Secondary STUN Server Undefined
NAT Traversal - STUN Parameters Secondary STUN Port Undefined
Polling Interval 1 hour
Keep-alive Interval 120 sec.
NAT IP checking Interval 300 sec.
NAT Traversal Exceptions – No entries
Packetization Interval 20 ms.
All
Silence Suppression Yes
No packetization interval and silence
– suppression values defined for G722,
G722.1 and TDVC codec.
G726 Standard Use ITU-T specification All
RTP Settings
QX50/QX200/
6000-6255
QXISDN4+
RTP/RTCP port range 6000-6509 QX20/QX500
6000-7799 QX2000
6000–8399 QX3000
RTCP Support Disabled All
Revision 1.2 217


UDP and TCP Port 5060
TLS Port 5061
Realm Epygi
SIP Settings All
Session Timer Disabled
DNS Server for SIP Use default
SIP Timers RFC3261
Host Aliases for SIP – No entries All
No certificate is generated and
TLS Certificates – All
installed.
Company’s Schedule (Work Hours:
09:00-13:00 14:00-18:00 from Monday
Schedules – to Friday), the Do not observe Holidays All
option is disabled and no entries in the
Special Days page.
Recording Codec G711u
Voice mail received from
Voice Mail Common Settings E-mail Subject for Voice Mail All
$[VM_DISPNAME] $[VM_USERNAME]
FAX to E-mail Format TIFF
RTP Streaming Channels – No entries All
Transmit Gain: -6
FXS Lines
Receive Gain: 0
QX50/QX200
Transmit Gain: 0
FXO Lines
Receive Gain: 6
Transmit Gain: 0
Gain Control Settings ISDN Trunks QXISDN4+
Receive Gain: 0
Transmit Gain (Line out): Off QX20/QX50/
Audio Lines
Receive Gain (Line in): Off QX200/QX500
Recording Gain: 0
Voice Mail All
Playback Gain: 0
Secure Connection Disabled All
3PCC Settings
Request Timeout 10 sec. All
Revision 1.2 218


Feature Key Not added
3PCC Settings
WAN Port Not opened
RADIUS Client Settings – The Radius Client service is disabled. All
Dial Timeout Settings Routing Dial Timeout 4 sec. All
Configure Call Quality Event The Call Quality Event Notification
– All
Notification service is disabled.
IDS Enabled
NAT Enabled
Firewall Configuration All
Firewall Disabled
Firewall Level Not selected
Ping Stealth Enabled All
QX20/QX50/QX200/
Advanced Firewall Configuration Disabled
Fool Portscanner QX500/QXISDN4+
n/a QX2000/QX3000
QX20/QX50/QX200/
No entries
Incoming Traffic / Port Forwarding QX500/QXISDN4+
n/a QX2000/QX3000
QX20/QX50/QX200/
No entries
Outgoing Traffic QX500/QXISDN4+
Filtering Rules n/a QX2000/QX3000
Management Access HTTPS (all IP addresses allowed).
Call Control Access No entries
SIP Access All IP addresses allowed. All
Blocked IP List No entries
Allowed IP List No entries
Service Pool Configuration – No entries All
IP Pool Configuration – No entries All
SIP IDS Enabled
Add the IP address into the Blocked
Enabled
IP List in Firewall
SIP IDS Settings All
Discard SIP messages from IP
Enabled, set to 32 sec.
address
Exceptions for SIP IDS No entries
IP Static Routes – No entries All
Revision 1.2 219


IP Policy Routes – No entries All
PPTP/L2TP Routes – No entries All
DHCP Options
QX20/QX50/QX200/
172.30.0.1
Gateways QX500/QXISDN4+
192.168.0.200 QX2000/QX3000
QX20/QX50/QX200/
172.30.0.1
Subnet Mask QX500/QXISDN4+
192.168.0.200 QX2000/QX3000
QX20/QX50/QX200/
172.30.0.1
Domain Name Servers QX500/QXISDN4+
192.168.0.200 QX2000/QX3000
NBT Name Servers 0.0.0.0 All
DHCP Advanced Settings QX20/QX50/QX200/
172.30.0.1
NTP Servers QX500/QXISDN4+
192.168.0.200 QX2000/QX3000
Domain Name "epygi-config.loc" All
QX20/QX50/QX200/
172.30.0.1
Overload TFTP Server Name QX500/QXISDN4+
192.168.0.200 QX2000/QX3000
DHCP Server Statements
Authoritative Enabled
Ping Check Enabled All
Ping Timeout 1 sec.
Zone epygi.config.loc
Time to Live (TTL) 86400 sec.
DNS Server Settings All
Mail Exchange (MX) Undefined
Aliases No entries
Dynamic DNS Settings – The Dynamic DNS service is disabled. All
PPP / PPTP Settings – The PPP service is disabled. All
Global SNMP Settings – The SNMP service is disabled. All
SNMP Trap Settings – No entries All
VLAN Settings – No entries All
Revision 1.2 220


IPSec Configuration Connection No entries All
(n/a on QX2000/QX3000) RSA Key Management 1024-bit key is generated All
Connection No entries All
PPTP Server Configuration
Subnet 172.31.1.0/24
Authentication MSCHAPv2 All
Encryption MPPE 128-bit
L2TP Server Configuration
Subnet 172.31.2.0/24 All
PPTP/L2TP Configuration Certificate Authority and Key
(n/a on QX2000/QX3000) Organizational Unit Name MyOrganizationalUnit
Email Address [email protected]
Signature Algorithm Sha1
Key Size 1024
All
Validation Period 3650 days
– All other fields are left empty.
No certificate is generated and
–
installed.
"Display notification" for all events
except Login and Firmware Update
events. Those events have a "Do
Event Settings – All
nothing" action assigned. Additionally,
Fan Control critical and major failures
have a Flash LED action assigned.
Call Reporting Enabled
Maximum Number of Successful Call
100
Records
Maximum Number of Missed Call All
100
Call History – Settings Records
Maximum Number of Unsuccessful
100
Call Records
All CDR Parameters are included in
– All
CDR file.
Call History - Archive – No entries All
Revision 1.2 221


0%
Call History - Archiving Settings allocated for Archive All
Call History Archiving Disabled
The Conference feature isn’t activated
Conference History - Settings – All
by default.
User Logging Enabled
System Logs Settings Developer Logging Enabled All
Log Lines to show 1000
The Remote Logging service is
Remote Logs Settings – All
disabled.
The Automatic Configuration Backup
Backup Configuration Management – All
service is disabled.
QX20/QX50/QX200/
Enabled
Automatic Firmware Update QX500/QXISDN4+
Disabled QX2000/QX3000
Server Name ftp.epygi.com
Automatic Firmware Update Server Port 21
Update Method ftp
All
Username anonymous
Password Is left blank.
Check for updates Check and notify every day at 0:00
Revision 1.2 222

15.2 User Extension Settings

Maximum Voice Mail Duration 5 min.
Forward/Rewind Duration 3 sec.
Ask password before granting local
Disabled
access to Voice Mailbox
Ask password before granting remote
Enabled
access to Voice Mailbox
Play welcome message Disabled
Voice Mail Settings – General
Settings Play Voice Mail help Enabled All
Automatically play Voice Mail Enabled
Play Voice Mails count information
Disabled
message
Play date/time information message Enabled
Play beep at the end of message Enabled
Silent Voice Mail recording Disabled
Voice Mail Greeting Message Default message.
Send new Voice Mail notifications via
Disabled
E-mail
Voice Mail Settings – VM Send new Voice Mail notifications via
Disabled
Notifications SMS All
Send new Voice Mail notifications via
Disabled
phone call
Voice Mail Notification Message Default message.
Lamp indication Enabled for IP lines only.
Voice Mail Settings – VM Indication Tone indication Enabled for FXS lines only. All
Ringing indication Disabled.
Zero Out Redirect Enabled, Redirect Call Type – PBX (00)
Voice Mail Settings – VM Redirection FAX Redirection Disabled All
Automatic Fax Receiving Mode Disabled
Voice Mail Settings – Out of Office
– The Out of Office Service is disabled. All
Service
Voice Mail Profiles – No entries All
Group List – No entries All
Revision 1.2 223


Speed Calling – No entries All
Display Name None
Disabled both for incoming and
User Password Protection
outgoing calls.
Account Settings All
Enable Remote Extension Disabled
User's name for Dial by Name
Undefined
Directory
No Answer Timeout 20 sec.
Call Waiting service Enabled
Basic Services – General Settings All
Redial Interval 10 sec.
Redial Period 15 min.
Send Hold Music to Remote IP Party Disabled
Basic Services - Hold Music Settings Listen Hold Music Own_Music All
Hold Music Default message
Actual Status Service is not active
Basic Services – Do Not Disturb
Expires after 30 min. All
Settings
Send Message to Caller Enabled
Basic Services – Alarm Settings – The Alarm service is disabled. All
Basic Services – Schedule D&A – The Schedule D&A service is disabled. All
The Hot Line service is disabled on FXS
Basic Services – Hot Line – All
lines. It's not applicable on IP lines.
All services are disabled for Any
Caller ID Services – All
Address entry.
Revision 1.2 224

16 References
Refer to the below listed recourses to get more details about the configurations described in this guide:
 Manual-I: Installation Guide for QX IP PBXs
 Manual-III: User Guide for QX IP PBXs
 System Capacity of QX IP PBXs
 QX IP PBX Features on Epygi Supported IP Phones
 Licensable Features on QX IP PBXs
 Language Packs Overview for Epygi QX Line
 Audio-Video Conferencing on QX IP PBXs
 Receptionist Service on QX IP PBXs
 QX IP PBX Remote Extension Configuration
 Extensions Bulk Import on QX IP PBXs
 Auto Configuration of Epygi Supported IP Phones using OpenVPN
 Call Detail Records on QX IP PBXs
 Firmware Update Service on Epygi QX Line
 DCC – User Guide
Find the above listed documents on Epygi Support Portal.
Revision 1.2 225

17 Appendix: Software License Agreement
EPYGI TECHNOLOGIES, LTD.

Software License Agreement
THIS IS A CONTRACT.
CAREFULLY READ ALL THE TERMS AND CONDITIONS CONTAINED IN THIS AGREEMENT. USE OF
THE QUADRO HARDWARE AND OPERATIONAL SOFTWARE PROGRAM INDICATES YOUR
ACCEPTANCE OF THESE TERMS AND CONDITIONS. IF YOU DO NOT AGREE TO THESE TERMS AND
CONDITIONS, YOU MAY NOT USE THE HARDWARE OR SOFTWARE.
1. License. Epygi Technologies, LTD. (the "Licensor"), hereby grants to you a non-exclusive right to use the Quadro or QX Operational Software
program, the documentation for the software and such revisions for the software and documentation as the Licensor may make available to you from
time to time (collectively, the "Licensed Materials"). You may use the Licensed Materials only in connection with your operation of your Quadro or
QX. You may not use, copy, modify or transfer the Licensed Materials, in whole or in part, except as expressly provided for by this Agreement.
2. Ownership. By paying the purchase price for the Licensed Materials, you are entitled to use the Licensed Materials according to the terms of this
Agreement. The Licensor, however, retains sole and exclusive title to, and ownership of, the Licensed Materials, regardless of the form or media in or
on which the original Licensed Materials and other copies may exist. You acknowledge that the Licensed Materials are not your property and
understand that any and all use and/or the transfer of the Licensed Materials is subject to the terms of this Agreement.
3. Term. This license is effective until terminated. This license will terminate if you fail to comply with any terms or conditions of this Agreement or
you transfer possession of the Licensed Materials to a third party in violation of this Agreement. You agree that upon such termination, you will
return the Licensed Materials to the Licensor, at its request.
4. No Unauthorized Copying or Modification. The Licensed Materials are copyrighted and contain proprietary information and trade secrets of the
Licensor. Unauthorized copying, modification or reproduction of the Licensed Materials is expressly forbidden. Further, you may not reverse
engineer, decompile, disassemble or electronically transfer the Licensed Materials, or translate the Licensed Materials into another language under
penalty of law.
5. Transfer. You may sell your license rights in the Licensed Materials to another party that also acquires your Quadro or QX product. If you sell your
license rights in the Licensed Materials, you must at the same time transfer the documentation to the acquirer. Also, you cannot sell your license
rights in the Licensed Materials to another party unless that party also agrees to the terms and conditions of this Agreement. Except as expressly
permitted by this section, you may not transfer the Licensed Materials to a third party.
6. Protection And Security. Except as permitted under Section 5 of this Agreement, you agree not to deliver or otherwise make available the Licensed
Materials or any part thereof to any person other than the Licensor or its employees, without the prior written consent of the Licensor. You agree to
use your best efforts and take all reasonable steps to safeguard the Licensed Materials to ensure that no unauthorized person shall have access
thereto and that no unauthorized copy, publication, disclosure or distribution thereof, in whole or in part, in any form, shall be made.
7. Limited Warranty. The only warranty the Licensor makes to you in connection with this license is that the media on which the Licensed Materials
are recorded will be free from defects in materials and workmanship under normal use for a period of one (1) year from the date of purchase (the
"Warranty Period"). If you determine within the Warranty Period that the media on which the Licensed Materials are recorded are defective, the
Licensor will replace the media without charge, as long as the original media are returned to the Licensor, with satisfactory proof of purchase and
date of purchase, within the Warranty Period. This warranty is limited to you as the licensee and is not transferable. The foregoing warranty does not
extend to any Licensed Materials that have been damaged as a result of accident, misuse or abuse.
EXCEPT FOR THE LIMITED WARRANTY DESCRIBED ABOVE, THE LICENSED MATERIALS ARE PROVIDED ON AN "AS IS" BASIS. EXCEPT AS
DESCRIBED ABOVE, THE LICENSOR MAKES NO REPRESENTATIONS OR WARRANTIES THAT THE LICENSED MATERIALS ARE, OR WILL BE, FREE
FROM ERRORS, DEFECTS, OMISSIONS, INACCURACIES, FAILURES, DELAYS OR INTERRUPTIONS INCLUDING, WITHOUT LIMITATION, TO ANY
IMPLIED WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE, LACK OF VIRUSES AND ACCURACY OR COMPLETENESS OF
RESPONSES, CORRESPONDENCE TO DESCRIPTION OR NON-INFRINGEMENT. THE ENTIRE RISK ARISING OUT OF THE USE OR PERFORMANCE OF
THE LICENSED MATERIALS REMAINS WITH YOU.
8. LIMITATION OF LIABILITY AND REMEDIES. IN NO EVENT SHALL THE LICENSOR OR ANY OTHER PARTY WHO HAS BEEN INVOLVED IN THE
CREATION, PRODUCTION OR DELIVERY OF THE LICENSED MATERIALS BE LIABLE FOR ANY CONSEQUENTIAL, INCIDENTAL, DIRECT, INDIRECT,
SPECIAL, PUNITIVE OR OTHER DAMAGES, INCLUDING, WITHOUT LIMITATION, LOSS OF DATA, LOSS OF BUSINESS PROFITS, BUSINESS
INTERRUPTION, LOSS OF BUSINESS INFORMATION, OR OTHER PECUNIARY LOSS, ARISING OUT OF THE USE OF OR INABILITY TO USE THE
LICENSED MATERIALS, EVEN IF THE LICENSOR OR SUCH OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. YOU AGREE
THAT YOUR EXCLUSIVE REMEDIES, AND THE LICENSOR'S OR SUCH OTHER PARTY'S ENTIRE LIABILITY WITH RESPECT TO THE LICENSED
MATERIALS, SHALL BE AS SET FORTH HEREIN, AND IN NO EVENT SHALL THE LICENSOR'S OR SUCH OTHER PARTY'S LIABILITY FOR ANY
DAMAGES OR LOSS TO YOU EXCEED THE LICENSE FEE PAID FOR THE LICENSE MATERIALS.
The foregoing limitation, exclusion and disclaimers apply to the maximum extent permitted by applicable law.
Revision 1.2 226

9. Compliance With Laws. You may not use the Licensed Materials for any illegal purpose or in any manner that violates applicable domestic or
foreign law. You are responsible for compliance with all domestic and foreign laws governing Voice over Internet Protocol (VoIP) calls.
10. U.S. Government Restricted Rights. The Licensed Materials are provided with RESTRICTED RIGHTS. Use, duplication or disclosure by the
Government is subject to restrictions as set forth in subparagraphs (c)(1) and (2) of the Commercial Computer Software—Restricted Rights clause at
48 C.F.R. section 52.227-19, or subparagraph (c)(1)(ii) of the Rights in Technical Data and Computer Software clause at DFARS 252.227.7013, as
applicable.
11. Entire Agreement. It is understood that this Agreement, along with the Quadro or QX installation and administration manuals, constitute the
complete and exclusive agreement between you and the Licensor and supersede any proposal or prior agreement or license, oral or written, and any
other communications related to the subject matter hereof. If one or more of the provisions of this Agreement is found to be illegal or unenforceable,
this Agreement shall not be rendered inoperative but the remaining provisions shall continue in full force and effect.
12. No Waiver. Failure by either you or the Licensor to enforce any of the provisions of this Agreement or any rights with respect hereto shall in no way
be considered to be a waiver of such provisions or rights, or to in any way affect the validity of this Agreement. If one or more of the provisions
contained in this Agreement are found to be invalid or unenforceable in any respect, the validity and enforceability of the remaining provisions shall
not be affected.
13. Governing Law. This Agreement shall be governed by and construed in accordance with the laws of the state of Texas, without regard to choice of
law provisions that would cause the application of the law of another jurisdiction.
14. Attorneys' Fees. In the event of any litigation or other dispute arising as a result of or by reason of this Agreement, the prevailing party in any such
litigation or other dispute shall be entitled to, in addition to any other damages assessed, its reasonable attorneys’ fees, and all other costs and
expenses incurred in connection with settling or resolving such dispute.
If you have any questions about this Agreement, please write to Epygi at 2233 Lee Road Suite 201 Winter Park, Florida 32789 or call Epygi at (972)
692-1166.
15. Free Software. Certain software utilized in the Epygi products is free software in its original form or in its modified form. Both types of free software
are available to you free of charge for redistribution or modification under certain conditions. Permission is granted to copy, distribute and
or/modify any free software you wish to download, whether in its original or modified forms, under the GNU General Public License or Free
Documentation License, Version 1.1 or any later version published by the Free Software Foundation. BECAUSE THE FREE SOFTWARE IS LICENSED
FREE OF CHARGE, THERE IS ABSOLUTELY NO WARRANTY. Please make sure you download the GNU license from www.gnu.org . For a list of free
software go to http://www.epygi.com/about/free-software-list.
Revision 1.2 227

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Emergency 911 Calls

European Limited Warranty

Extended Warranty Option

Extended Warranty Statement

Administrative Council for Terminal Attachments (ACTA) Customer Information

Electrical Safety Advisory

Industry Canada Statement

This product meets the applicable Industry Canada technical specifications.

Music on Hold Copyright

Compliance with Laws

Document Edition History

Revision Date Description Valid for Models Valid for FW

1 About Administration Guide ..........................................................................7

10.1 IP Routing ................................................................................................................. 145

1 About Administration Guide

2 Conventions Used in this Guide

Figure 1: Recording Settings page

3 QX’s Graphical Interface

Figure 2: Dashboard menu

 The Terminate link is used to terminate the active call.

Figure 3: Setup Menu overview

5.1 Basic Setup

Figure 4: System Configuration section

 Host Name – set the host name for QX.

DHCP Settings for the LAN Interface

Figure 5: DHCP Settings for the LAN Interface section

 Enable DHCP Server – enable/disable DHCP server capability on the QX.

Regional Settings and Preferences

Figure 6: Regional Settings and Preferences section

 Your Locale (location) – select the location and timezone of QX.

Emergency Codes and PSTN Access Code Settings

Figure 7: Emergency Codes and PSTN Access Code Settings section

5.1.2 Internet (WAN)

Figure 8: Uplink Configuration section

 VLAN ID – select VLAN ID from the configured VLAN list.

WAN Interface Configuration

Figure 9: WAN Interface Configuration section

Figure 10: DNS Settings section

5.1.3 Date and Time

Figure 11: Date / Time Settings page

 Date/Time – displays the current system time.

5.1.4 E-mail (SMTP)

Figure 12: E-mail Settings page

5.1.5 Short Text Messaging (SMS)

For example: user=%username%&password=%password%&to=%to%&from=%from%&text=%text%

5.2 System Security

Figure 14: System Security Management page

5.3 Licensed Features

Figure 15: Features page

 Debug – enables SSH connection towards the QX for debugging purposes.

Figure 16: Adding a feature key

5.3.2 Free Trial

5.5 Language Pack

Figure 17: Language Pack page

After successful upload, the following links will be available:

Figure 18: Extensions Menu overview

Figure 19: Choose Extensions Length page