Brekeke PBX

Version 3

Administrator's Guide
(Basic)

Brekeke Software, Inc.

 BREKEKE SOFTWARE, INC.

Version
Brekeke PBX Version 3 Administrator’s Guide (Basic)

Copyright
This document is copyrighted by Brekeke Software, Inc.
Copyright © 2014 Brekeke Software, Inc.
This document may not be copied, reproduced, reprinted, translated, rewritten or readdressed in
whole or part without expressed, written consent from Brekeke Software, Inc.

Disclaimer
Brekeke Software, Inc. reserves the right to change any information found in this document
without any written notice to the user.

Trademark Acknowledgement
‹ LINUX is a registered trademark of Linus Torvalds in the United States and other countries.
‹ Red Hat is a registered trademark of Red Hat Software, Inc.
‹ Windows is a trademark or registered trademark of Microsoft Corporation in the United
States and other countries.
‹ Oracle and Java are registered trademarks of Oracle and/or its affiliates. Other names may
be trademarks of their respective owners.
‹ Other logos and product and service names contained in this document are the properties
of their respective owners.

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1. INTRODUCTION ...................................................................................... 7

1.1. Editions......................................................................................................................... 7

2. INSTALLATION........................................................................................ 7

2.1. System Requirements ................................................................................................. 7

2.2. Select File to Install Brekeke PBX.............................................................................. 7

2.3. Installation for Windows OS with the Executable Installer ..................................... 8

2.4. Installation for Linux OS ............................................................................................. 9

2.5. Updating Brekeke PBX.............................................................................................. 10

3. MAINTENANCE ..................................................................................... 10

3.1. Back Up / Restore ...................................................................................................... 10

3.2. Report (Multi-Tenant PBX) ........................................................................................ 11

3.3. Activating License ..................................................................................................... 11

4. UNINSTALL ........................................................................................... 11

4.1. Uninstall from Windows OS...................................................................................... 11

4.2. Uninstall from Linux OS............................................................................................ 11

5. SETTING UP BREKEKE PBX ............................................................... 12

5.1. Setting Up Phone Type.............................................................................................. 12

5.2. Setting Up Tenants (Multi-Tenant PBX) ................................................................... 12

5.3. Setting Up Administrative Users.............................................................................. 12

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5.3.1. Creating Administrative Users.................................................................................. 13

5.3.2. Setting Up Administrators’ Privilege......................................................................... 13

5.4. Setting Up User Extensions ..................................................................................... 13

5.4.1. Creating Users ......................................................................................................... 13
5.4.2. Setting Up User Extensions’ Privileges.................................................................... 14
5.4.3. Default Values of User Extensions........................................................................... 14
5.4.4. Assigning Phones to User Extensions ..................................................................... 15
5.4.5. Voicemail Settings.................................................................................................... 16
5.4.6. Voicemail Notification by Email ................................................................................ 16
5.4.7. Setting Up Call Forwarding ...................................................................................... 17
5.4.8. Setting Up No Answer Call Forwarding ................................................................... 17
5.4.9. Setting Up Busy Call Forwarding ............................................................................. 17

5.5. Message Waiting Indicator (MWI)............................................................................. 17

5.6. Setting Up Ring Groups ............................................................................................ 18

5.7. Setting Up Call Pickup Groups................................................................................. 18

5.8. Setting Up Call Hunting ............................................................................................ 19

5.9. Setting Up Auto Attendant ........................................................................................ 20

5.10. Setting Up Call Forwarding Schedule ..................................................................... 21

5.11. Setting Up Switch Plan ............................................................................................. 22

5.11.1. Sample with Switch Plan IVR Extension.................................................................. 22
5.11.2. Sample with Timer.................................................................................................... 23

5.12. Setting Up Conference Call ...................................................................................... 23

5.12.1. Creating a Conference Room .................................................................................. 23
5.12.2. Limiting Members Who Can Enter the Conference Room ...................................... 24
5.12.3. Simultaneous Calls to All of the Conference Members............................................ 24
5.12.4. Starting a Conference Call (Alternate Methods) ...................................................... 24

5.13. Setting Up Callback ................................................................................................... 24

5.14. Setting Up Confirm Call ............................................................................................ 25

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5.15. Setting Up Paging...................................................................................................... 25

5.16. Setting Up Busy Lamp Field, Presence and Shared Call Appearance................. 26

5.17. Setting Up Video ........................................................................................................ 26

5.18. Setting Up SRTP ........................................................................................................ 27

5.19. ARS Settings .............................................................................................................. 27

5.19.1. Route Settings / Route Template ............................................................................. 28
5.19.2. Global Route Variables (Brekeke PBX v3.1.x and later).......................................... 29
5.19.3. Variable Setting ........................................................................................................ 29
5.19.4. ARS Default Plug-ins ............................................................................................... 30

5.20. PSTN Access Using a VoIP Gateway ....................................................................... 31

5.20.1. VoIP Gateway Setup ................................................................................................ 31
5.20.2. ARS Route Setup..................................................................................................... 31
5.20.3. Receiving PSTN Calls.............................................................................................. 31
5.20.4. Calling PSTN Numbers............................................................................................ 32
5.20.5. Assign Values to Variables....................................................................................... 33

5.21. Connecting with Internet Telephony Service Providers (ITSPs) .......................... 33

5.21.1. Account Information for Third-Party SIP Server....................................................... 33
5.21.2. Setting ARS for ITSP using multiple accounts ......................................................... 34

5.22. ARS Outbound Route Failover................................................................................. 35

5.22.1. Usage Examples ...................................................................................................... 35
5.22.2. Settings Examples ................................................................................................... 35

5.23. ARS Route with Resource Setting (Brekeke PBX v3.1.x and later)...................... 37

5.24. ARS Route with Group Setting................................................................................. 38

5.25. DID............................................................................................................................... 40

6. SETUP ITEMS........................................................................................ 42

6.1. Start / Shutdown ........................................................................................................ 42

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6.2. Options ....................................................................................................................... 42

6.2.1. Settings .................................................................................................................... 42
6.2.2. User Access Settings ............................................................................................... 47
6.2.3. Phone Type .............................................................................................................. 47
6.2.4. Auto Sync................................................................................................................. 49
6.2.5. SA (Multi-Tenant PBX) ............................................................................................. 49
6.2.6. Advanced ................................................................................................................. 49

6.3. Voice Prompts ............................................................................................................ 49

6.3.1. System Voice Prompts ............................................................................................. 49
6.3.2. Notes for Sound Files .............................................................................................. 50

6.4. Automatic Route Selection (ARS)............................................................................ 50

6.4.1. Adding a New Route ................................................................................................ 50
6.4.2. Editing, Copying or Deleting a Route....................................................................... 50
6.4.3. Viewing an Active Route .......................................................................................... 50
6.4.4. ARS > Route Template ............................................................................................ 51

6.5. DID............................................................................................................................... 57

6.6. Call Status .................................................................................................................. 58

6.6.1. Status ....................................................................................................................... 58
6.6.2. UAs (User Agents) ................................................................................................... 58

6.7. Call Logs..................................................................................................................... 58

6.8. Notes........................................................................................................................... 59

6.9. Extensions.................................................................................................................. 59
6.9.1. System Administrator ............................................................................................... 59
6.9.2. Group Extensions..................................................................................................... 59
6.9.3. Schedule Extensions ............................................................................................... 62
6.9.4. IVR Extensions......................................................................................................... 62
6.9.5. Conference Extensions............................................................................................ 65
6.9.6. Callback Extensions................................................................................................. 66

7. SETUP ITEMS (MULTI-TENANT PBX).................................................. 66

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7.1. Tenants ....................................................................................................................... 66

7.1.1. Copy and Set Up New Tenant.................................................................................. 66
7.1.2. Settings Copied in New Tenant................................................................................ 66

7.2. Tenant Voice Prompts ............................................................................................... 67

7.3. Tenant Notes .............................................................................................................. 67

7.4. Tenant Options........................................................................................................... 67

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1. Introduction
This document explains the basic configuration of Brekeke PBX. For more advanced setting
instructions and product information, please refer to the Brekeke PBX Administrator’s Guide
(Advanced).

1.1. Editions
Brekeke PBX comes in several editions in order to meet the needs of different users:
Edition Explanation
Product trial prior to purchase with Pro Edition’s feature set.
Evaluation
This license is free of charge.
Designed for businesses and services that need sophisticated call
Commercial
management and advanced features.
Commercial - Designed as a platform that provides hosted IP-PBX service for
Multi-Tenant option Service Providers.

2. Installation
2.1. System Requirements
OS Microsoft Windows XP or later, and Linux
Java 6 or later (32-bit / 64-bit)
Java 9 Brekeke products are confirmed to run on Java provided by Oracle Sun
Microsystems.

Version 7.0.42 and later

Apache Tomcat 9 Tomcat installation is not required when Brekeke PBX is installed with the
executable installer.

Memory 512MB minimum

2.2. Select File to Install Brekeke PBX

Executable installer pbx.war (zip format)
OS Windows OS Linux OS, Windows OS
New installation
Install New installation only
Update installation
Section 2.4 (Linux)
Instructions Section 2.3 (Windows)
Section 2.5 (Update)

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2.3. Installation for Windows OS with the Executable Installer

Step 1: Installing Java SE

Install Java SE before installing the Brekeke PBX software:
1) Go to: http://www.oracle.com/technetwork/java/javase/downloads/index.html
2) Download and install the appropriate version of JRE or JDK for the type of Windows OS
you are running.

Step 2: Installing Brekeke PBX

1) Obtain the installer file from Brekeke’s website.
2) Start the installation following the installer instructions. If you check the [Run Brekeke PBX
v3.x] box at the last stage of the installation and push the [Finish] button, the Brekeke
PBX’s HTTP service will start automatically.
9 Brekeke PBX and bundled SIP server will be installed automatically.

Step 3: Starting Brekeke PBX HTTP Service

If you did not check [Run Brekeke PBX 3.x] at the last stage of the installation, start Brekeke
PBX HTTP Service by doing one of the followings:
• From the taskbar, open Brekeke PBX Service Manager and start Brekeke PBX service.
• From Control Panel / Performance and Maintenance / Administrative Tools / Services, select
and start “Brekeke PBX” service.
• Restart computer. Brekeke PBX HTTP service will start automatically.

Step 4: Starting Brekeke PBX Administration Tool (Admintool)

1) Select Start / All Programs / Brekeke PBX / Brekeke PBX Admintool.
2) Read the End User License Agreement (EULA) carefully. Only if you agree with all terms,
then copy and paste the 16-digit product ID in the given space. (Entering the same product
ID on multiple machines is not allowed.) and click on [Activate the new llicense] button.
3) If your computer is connected to the Internet, license activation will start automatically by
clicking on [Online Activation] button
Otherwise, click on [Offline Activation] button and follow the instructions shown on screen
to activate the license.
4) At the login page, enter the user ID and password.
The default setting for both the administrator user ID and the password is “sa”.
5) Click the [Start] button from the menu [Start / Shutdown]. If you see [Active] for status of

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both PBX and the bundled SIP Server, the Brekeke PBX server has started successfully. If
the Status is [Inactive], the server has not started successfully, the error should be shown.

2.4. Installation for Linux OS

Step 1: Installation of Java SE
1) Go to: http://www.oracle.com/technetwork/java/javase/downloads/index.html
2) Download and install the appropriate version of JRE or JDK for the type of OS you are
running.

Step 2: Installation of Apache Tomcat

1) Go to http://tomcat.apache.org/index.html and download the binary file of Tomcat version
7.0.42 for the type of OS you are running.
2) Set JRE or JDK installation directory for the environment variable JAVA_HOME.
3) Install downloaded Tomcat file.
9 We recommend setting autoDeploy=”false” and liveDeploy="false" in the server.xml file at Tomcat

installation directory/conf/ as below:

<Host name="localhost" appBase="webapps"
unpackWARs="true" autoDeploy="false" liveDeploy="false"
xmlValidation="false" xmlNamespaceAware="false">
or
<Host name="localhost" appBase="webapps"
unpackWARs="true" autoDeploy="false" liveDeploy="false" >

Step 3: Installation of Brekeke PBX

1) Download the file pbx.war (zip format file) from Brekeke’s website.
2) Copy the pbx.war file directly into the “webapps” directory, which is under the Tomcat
installation directory.

Step 4: Starting Tomcat

1) Start Tomcat.
2) Open a web browser and enter http://localhost:8080 in the address field. (If you chose a
port number other than the default “8080,” specify the appropriate port number in the URL.)
3) Tomcat has started successfully if the Apache Jakarta Project page is displayed.

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Step 5: Starting Brekeke PBX Administration Tool (Admintool)

1) In a web browser, enter http://localhost:8080/pbx/ in the address field. (If you chose a port
number other than default “8080,” specify the appropriate port number in the URL.)
2) Read the End User License Agreement (EULA) carefully. Only if you agree with all terms,
then copy and paste the 16-digit product ID in the given space. (Entering the same product
ID on multiple machines is not allowed.) and click on [Activate the new llicense] button.
3) If your computer is connected to the Internet, license activation will start automatically by
clicking on [Online Activation] button
Otherwise, click on [Offline Activation] button and follow the instructions shown on screen
to activate the license.
4) At the login page, enter the administrator user ID and password.
The default setting for both the administrator user ID and the password is “sa”.
5) Click the [Start] button from the menu [Start / Shutdown]. If you see [Active] as the status of
both PBX and the bundled SIP Server, the Brekeke PBX server has started successfully. If
the Status is [Inactive], the server has not started successfully, the error should be shown.

2.5. Updating Brekeke PBX

This section is for updating from an earlier version of Brekeke PBX v3.x to the current release.
Please confirm that you have downloaded the zip file (pbx.war) from Brekeke’s website.
1) If Brekeke PBX is active, click [Shutdown] from the menu [Start / Shutdown] from Brekeke
PBX Admintool
2) Open [Maintenance] > [Update Software] page
3) Click the [browse] button to select the pbx.war file that you have downloaded.
4) Click the [upload] button to upload the new file.
5) Restart your computer to complete updating Brekeke PBX.

3. Maintenance
3.1. Back Up / Restore
You can back up current configurations and messages in users’ voicemail box from Brekeke
PBX Admintool menu [Maintenance] > [Back Up]. We recommend backing up Brekeke PBX on
a regular basis. You can restore the backup data from the menu [Maintenance] > [Restore].

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3.2. Report (Multi-Tenant PBX)

Overview of all tenant setups that are hosted on the Brekeke Multi-Tenant PBX system, as well
as system tenant settings reports that can be exported in csv format.

3.3. Activating License

1) If Brekeke PBX is active, click [Shutdown] from the menu [Start / Shutdown] from Brekeke
PBX Admintool
2) Open [Maintenance] > [Activate License] page, and click on [Activate License] button
3) Read the End User License Agreement (EULA) carefully. Only if you agree with all terms
Copy and paste the 16-digit product ID to activate with new product ID.
If you are reactivating the current license, only click the [Reactivate the license] button
4) If your computer is connected to the Internet, license activation will start automatically by
clicking on [Online Activation] button
Otherwise, click on [Offline Activation] button and follow the instructions shown on screen
to activate the license.
5) After completing the activation successfully, you will see the login screen of Brekeke PBX
Admintool.

4. Uninstall
4.1. Uninstall from Windows OS
1) Kill all Java processes (java.exe) used by Brekeke PBX from Task Manager.
2) Navigate to Start / All Programs / Brekeke PBX / Uninstall Brekeke PBX. The uninstaller will
uninstall Brekeke PBX automatically.
9 If the uninstaller fails to delete the folder (C:\Program Files\Brekeke\pbx), you will need to restart the

PC and delete the folder manually.

4.2. Uninstall from Linux OS

Delete the file “pbx.war” and the folder “pbx” in the directory “webapps,” which is located under
the installation directory of Tomcat, then restart the machine.

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5. Setting Up Brekeke PBX

5.1. Setting Up Phone Type
There are three default phone types on the Brekeke PBX Admintool > [Options] > [Phone Type]
page when this is a first-time Brekeke PBX installation or when there are no other phone types
created and this is an upgrade from a previous version. System administrators can also define
new phone types with various combined settings for RTP relay, codecs, SRTP, video and
keypad commands.

Depending on the system UA’s location, codecs and Brekeke PBX feature requirements, the
user or system administrator can choose a suitable phone type for each phone ID assigned in a
Brekeke PBX user extension.

5.2. Setting Up Tenants (Multi-Tenant PBX)

For Brekeke Multi-Tenant PBX, system administrator “sa” needs to create a tenant account for
each enterprise client. Each tenant will be able to have their own internal extension system.

1) From Brekeke PBX Admintool > [PBX Admin] > [Tenants], display all current tenants
2) In this example we will create a tenant named "test".
Click on [New Tenant] button to input "test" as tenant name
3) Once the tenant is created, go to this tenant setup menu by clicking on the tenant name
4) Clicking on [Options] menu under this tenant to set tenant users and sessions limit from
[Capacity] section and enable required features for this tenant from [Features] section.
9 Lower-case letters, digits, dots and underscores are valid characters for a tenant name.

5.3. Setting Up Administrative Users

From default system administrator “sa” account, you can create other administrator accounts
and limit these administrators system access privilege.

For Brekeke Multi-Tenant PBX, default system administrator “sa” is the only administrator for
whole system. “sa” administrator needs to create a tenant administrator at the time a tenant is
created, and authorizes tenant administrators for accessing tenant DID and system notes.
By logging in as a tenant administrator from Brekeke Multi-Tenant PBX admintool, each
enterprise client administrator can define his company extensions system.

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5.3.1. Creating Administrative Users

1) Log in as system administrator “sa”
2) Go to Brekeke PBX Admintool > [PBX Admin] > [Extensions] > [Users] page.
Default administrator “sa” has already been created. (Default password is “sa”.)
For Brekeke Multi-Tenant PBX, go to Brekeke PBX Admintool > [PBX Admin] > [Tenant].
Click on the tenant name “test” and then go to this tenant [Extensions] > [Users] page.
3) Click the [Create a new user] button. When the User [Account] page opens, enter the
following settings:
[Extension] admin
[Login Password] admin
[Login Password (confirm)] admin
[User Type] Admin
4) There is no default login password. If no login password is set at the time a user is created,
the user cannot log into the account from Brekeke PBX Admintool.

5.3.2. Setting Up Administrators’ Privilege

From Brekeke PBX Admintool > [PBX Admin] > [Options] > [User Access Settings] page [Admin
Menu] section, system administrator “sa” can enable or disable the access of a system menu
from non-“sa” administrators’ account by selecting “Yes“ or “No” for the options. The non-“sa”
administrators will have the same access privilege to the enabled menu sections as that “sa”
administrator has.

5.4. Setting Up User Extensions

Administrators (system administrators or tenant administrators) can create user extensions and
assign phone IDs to each user extension so that the UAs with assigned phone ID numbers can
call each other and make use of the features in Brekeke PBX.

5.4.1. Creating Users

In this example, we will create user extensions 100-105:
1) Click Brekeke PBX Admintool > [PBX Admin] > [Extensions] > [Users].
Default administrator “sa” has already been created. (Default password is “sa”.)
For Brekeke Multi-Tenant PBX, go to Brekeke PBX Admintool > [PBX Admin] > [Tenant].
Click on the tenant name “test” and then go to this tenant [Extensions] > [Users] page.
2) Click the [Create a new user] button. When the User [Account] page opens, enter the

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following settings:
[Extension] 100
[Login Password] 100
[Login Password (confirm)] 100
[User Type] User
There is no default user login password. If no login password is set at the time a user is
created, that user cannot login to the account from Brekeke PBX Admintool.
3) Edit the user account settings as needed.
4) There is no default voicemail access PIN set at the time a user is created. Set the user’s
voicemail box PIN from the user’s [Settings] page. Please refer to the sections below for
more information about voicemail.
5) At the user’s [Phones] page, the user extension number or <tenant>_<extension>
(Multi-Tenant PBX) will be automatically set as this user’s default phone number in the first
[Phone ID] field at the time the user is created.
If the user’s phones have any numbers other than the default phone number, you can
assign these numbers to any [Phone ID] on the [Phones] page and select the proper [Type]
for each assigned phone. Please check the following sections for more details about
assigning phones to user extensions.
6) Create other user extensions 101-105 as above.

5.4.2. Setting Up User Extensions’ Privileges

From the Brekeke PBX Admintool > [PBX Admin] > [Options] > [User Access Settings] page, the
system administrator “sa” can enable or disable access to a user menu or selected fields shown
in a user’s account by selecting “Yes“ or “No.”

5.4.3. Default Values of User Extensions

The following table outlines the default values included for Brekeke PBX user extensions. If you
change these settings, the functionality of the product may differ from the examples shown in
this manual.

To change user settings, click the user’s extension number from the [PBX Admin] > [Extensions]
> [Users] list, or select a user from the [User] menu on the left-hand panel and navigate to each
page under the selected user.
For Brekeke Multi-Tenant PBX, go to Brekeke PBX Admintool > [PBX Admin] > [Tenant].
Click on the tenant name and then go to this tenant [Extensions] > [Users] page.

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Page name Setting item Details of default values

Same language as the

Account Language
administrator who created this user

Call pickup group Same group as the administrator

Settings Greeting message Default system greeting

Email notification off

- Same as user extension ID

Phone 1 > [Phone ID] - <tenant>_<extension ID>
Phones (Multi-Tenant PBX)

Type Type 1

Plan Plan 1 (Active)

All phones are checked by default.

When there is an incoming call to
Forward To > [Phones] this user, Brekeke PBX will forward
the call to all assigned phones
under this user.
Inbound
Forward To > [Call Forwarding] >
90 seconds
[Ringer Time (sec)]
Forward To > [Call Forwarding] >
[Forwarding destinations (No Forwards to user’s voicemail
answer)]
Forward To > [Call Forwarding] >
Forwards to user’s voicemail
[Forwarding destinations (Busy)]

5.4.4. Assigning Phones to User Extensions

Up to four different phone IDs can be assigned to each Brekeke PBX user extension. The
default phone ID is assigned in the [Phone 1] > [Phone ID] field and is the same as the Brekeke
PBX user extension ID from when this user was created, or is in the format of tenant name and
user extension ID (as <tenant_name>_<extension_ID>) for a tenant user in Brekeke
Multi-Tenant PBX.

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The phone ID field can be a SIP UA’s user ID, which is registered at Brekeke PBX bundled SIP
server, or a PSTN number that belongs to the user of this Brekeke user extension. Any devices
with a phone ID number assigned under a Brekeke PBX user extension can apply Brekeke PBX
features to the call.

For Brekeke Multi-Tenant PBX, we recommend to use format <tenant_name>_<extension ID>

as SIP UAs ID (phone number) used to register at Brekeke PBX bundled SIP server. This will
avoid ID confliction among different tenants and it will be easier for phone ID management.

5.4.5. Voicemail Settings

After creating the user extensions, you can set up voicemail for each of the user extensions. As
described in the section above, user 100’s [Forwarding destination (No answer)] and
[Forwarding destination (Busy)] are set to “Voicemail” by default. If devices assigned under user
100 do not answer after ringing for 90 seconds (or when user 100’s line is busy), the call will be
forwarded to voicemail. The “vm” prefix is specified under the “mediaserver_prefix” route at the
ARS settings.

♦ To leave a voice message directly, dial 07*<user extension ID>.

♦ To check voice messages from your own phones, dial “8” to reach your voicemail inbox.
♦ To check your voice messages from another user extension’s phone in the system, dial
08*<your user extension ID>.

5.4.6. Voicemail Notification by Email

Step 1: Setting Up an Email Sender
Brekeke PBX supports mail servers that provide “POP before SMTP” authentication or “SMTP”
authentication or Encrypted Connection (SSL).
To set up an Email Sender:
1) Enter the information about your mail server and email user account information at
Brekeke PBX Admintool [PBX Admin] > [Options] > [Settings] > [Email settings].
2) Set encrypted connection on or off, depending on your mail server type.
3) Restart Brekeke PBX from Admintool to apply your changes.
Step 2: Setting Up an Email Recipient
1) Navigate to the [Voicemail settings] section on a user’s [Settings] page.
2) Set [Email address*] to the recipient’s email address(es).

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3) Set [Email notification] to “on.”

4) Set [Attach WAV file to Email] to “on” or “off,” depending on whether you want to let this
user attach voice messages to email.

5.4.7. Setting Up Call Forwarding

Call Forwarding is for when a user isn’t available and you want his/her incoming calls to be
redirected to another number or to voicemail..
In this example, calls to extension 101 will be forwarded to extension 100:
1) In user extension 101’s [Other Forwarding destinations*] field, enter the user extension
number to which you want to forward the call. In this case, enter user extension “100.”
2) Make a call to extension 101. Both phones assigned under user extension 100 and 101 will
ring.

5.4.8. Setting Up No Answer Call Forwarding

To forward incoming calls to another extension, instead of using voicemail to answer calls after
ringer time out, set up a forwarding extension in the [Inbound] > [Forwarding destination (No
answer)] field.
1) In extension 101’s [Forwarding destinations (No answer)] field, enter user extension “100.”
2) Incoming calls will be forwarded to user extension 100 if user 101’s phones do not answer
before ringer time out.

5.4.9. Setting Up Busy Call Forwarding

To forward incoming calls to another user extension when all of a user’s phones are “busy,”
instead of using voicemail, set up a forwarding extension in the [Inbound] > [Forwarding
destination (Busy)] field.
1) In user extension 101’s [Forwarding destinations (Busy)] field, enter user extension “100.”
2) Calls will be forwarded to 100 if user 101’s phones return a 486 Busy response or another
error response.

5.5. Message Waiting Indicator (MWI)

If your SIP UA supports MWI with a “SUBSCRIBE” message, the Brekeke PBX MWI feature will
be activated by default. If there is a special button on your SIP UA to retrieve messages, assign
a number (default is “8”) to retrieve voicemail messages. Some types of SIP UAs can
automatically call the SIP URI that is specified in the Brekeke PBX NOTIFY packet (for MWI) to
retrieve the message without assigning the number manually.

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For those SIP UAs that do not send “SUBSCRIBE” message for MWI, you can create a new
phone type and set [MWI (NOTIFY without SUBSCRIBE)] field as on and select this phone type
from the PBX user extension [Phones] page where these SIP UAs are assigned. Then Brekeke
PBX will send voicemail notification even there is no “SUBSCRIBE” message for MWI sent from
these SIP UAs.

5.6. Setting Up Ring Groups

In this example, a Ring Group (1000) is created for all user extensions 101-105. When calls are
received at group extension 1000, all specified group extensions (101-105) will ring
simultaneously. If no one answers the call within the ringer time (ten seconds), the call will also
be forwarded to user extension 100.
1) From [Extensions] > [Groups], click [Create a new group].
2) Set as follows:
[Extension] 1000
[Type] Simultaneous Ring
[Group Extensions*] 101,102,103,104,105
[Ringer time (sec)] 10
[Forwarding destination (No answer)] 100
3) Save the settings.

5.7. Setting Up Call Pickup Groups

Call Pickup is a function that allows users to answer incoming calls to any other user extensions
by dialing a preset number. When a user extension rings, dial *<user extension ID> to answer
the call. For example, when user extension 100 rings, dialing *100 will enable you to answer the
call from any other Brekeke PBX user extension.
9 If you are using a SIP phone that has a “Call Pickup” button, please consult the manufacturer of your

SIP phone on how to use the Call Pickup feature.

Here are some other ways of using the Call Pickup feature:
♦ Answer Calls That Are Directed to a Ring Group
Calls directed to a Simultaneous Ring Group extension can be answered from any user
extension using Call Pickup. (For details on how to set up a Ring Group, please refer to the
“Setting Up Ring Groups” section.)

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Extension: 1000 (Simultaneous Ring Group)

Group Extensions* 101,102,103,104,105

Dialing *1000 (the Simultaneous Ring Group extension number) enables a user to pick up
any calls within the Ring Group. Incoming calls can still be answered by dialing *<group
extension ID>, such as *101, *102. However, using the group extension ID to pick up call
does not require you to remember each user extension ID in the group. The *<ring group
extension ID> feature even works when a call is made directly to a single extension (e.g.,
103) or when the call comes through the Auto Attendant to a extension which is set in ring
group extension.

♦ One-Touch Call Pickup for a Specified Call Pickup Group

By setting up a Call Pickup Group number (generally a ring group number) for a user
extension, you only need to dial * (Star) to pick up incoming calls to any user extensions in
that group.

You may specify a Call Pickup Group (such as “1000”) for a user extension at [Settings] >
[Call Pickup Group]. When there is an incoming call to any user extension in the
Simultaneous Ring Group extension 1000 > [Group Extensions] field (in this case,
extensions 101-105), you can answer the call by dialing * (Star) only.

5.8. Setting Up Call Hunting

1) From [Extensions] > [Groups], click [Create a new group].
2) Set as follows:
[Extension] 1001
[Type] Call Hunting
[Mode] Round-robin / Top-down
[Hunt group extensions*] 101,102,103,104,105
[Ringer time (sec)*] 5
[Waiting time in the queue (sec)] 120
[Max number of calls in the queue] 10
[Call interval (msec)] 3000
[Single attempt] no
[Forwarding destination (No Answer)] 100
3) Save the settings.

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In this example, the call will ring the user extensions in the [Hunt group extensions*] field
one-by-one with a five-second interval (Ringer time) between calling each user extension. If all
of the group extensions (101-105) are busy or do not answer, the call will be queued. If any
member becomes available within the interval set in [Waiting time in the queue (sec)], Brekeke
PBX will ring the first available extension in the group for the length of ringer time in this
extension user setting, then wait for the time of [Call interval (msec)](3000 ms) before ringing
the next available extension in the group. If all members continue to be busy after the specified
interval at [Waiting time in the queue (sec)], the call will be forwarded to the destination set in
[Forwarding destination (No answer)]. In this example, the call will be forwarded to user 100.

The music that the caller hears while waiting on hold can be changed by uploading a new sound
file in the [Sound files] > [Music on hold] field.

5.9. Setting Up Auto Attendant

The example below shows Auto Attendant for extension 1002:
1) From [PBX Admin] > [Extensions] > [IVR], click the [Create a new IVR] button.
2) Set as follows:
[Extension] 1002
[Type] Auto Attendant
[Max input digits] 3
[Max retry count] 5
[Ring timeout (sec)] 10
[Default operator] 100
[DTMF timeout (sec)] 20
[Transfer to unregistered users] disable
3) Save the settings.

In this example, the incoming call to Auto Attendant 1002 will hear the default greeting voice
prompt to input an extension.

If there is no input in 20 seconds ([DTMF timeout (sec)]), Brekeke PBX will ask the caller to
re-input the extension. After five times of no extension input ([Max retry count]), the call will be
forwarded to the destination that is set in the [Default operator] field.

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If the caller inputs an extension number, the call will transfer to the extension whose ID is the
same as the first three digits ([Max input digits]). If the transfer recipient does not answer the call
within ten seconds ([Ring timeout (sec)]), Brekeke PBX will stop ringing the input extension and
Auto Attendant will ask the caller to input a different extension number.

When [Transfer to unregistered users] is enabled, the caller can apply Brekeke PBX features
from Auto Attendant, such as accessing a voicemail box, call pickup and so on.
9 Note that a proper value must be set in the Auto Attendant > [Max input digits] field in order to allow a

caller’s input.

Optionally, an audio file for the Auto Attendant greeting can be uploaded from the Auto
Attendant > [Sound files] section.

5.10. Setting Up Call Forwarding Schedule

This feature is useful for creating call forwarding rules during specified times. Depending on the
need, call forwarding schedule can be set up by creating schedule extension or from each user
extension [Inbound] page.

In this example, incoming calls to extension 5555 during business hours are directed to the Auto
Attendant at extension 1002. After business hours, callers are scheduled to hear, “To speak with
a live operator, please call during regular business hours. Our regular business hours are
Monday through Friday, 9 a.m. to 6 p.m.”
You can use the following settings to set up such a scenario:
1) Create an Auto Attendant 1003 and upload an after business hour greeting to [Greeting
messages] under Auto Attendant 1003.
2) From [Extensions] > [Schedule], click [Create a new schedule] button.
3) Set as follows:
[Extension] 5555
[Default Forwarding Schedule] > [Call Forwarding] > [Destination] 1003
4) Click the [Add Forwarding Schedule] button.
Set [Forwarding Schedule 1] as follows:
Set the [Forward To] > [Call Forwarding] > [Forwarding Destination] field to 1002.
Click the [Conditions] tab and set the [Date/Time] section as follows:
[Term] set schedule starting and ending Date and Year.
[Days] Check 1st - 5th and Monday through Friday.

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[Time] select and check 9:00 - 18:00.

5) Save the settings.

5.11. Setting Up Switch Plan

Switch Plan or Timer can be used to change a user’s forwarding destination temporarily and
quickly by making a different forwarding schedule plan active.

In this example, incoming calls during business hours go directly to user 100’s assigned
phone(s). However, during user 100’s lunch break, you can send incoming calls directly to
voicemail. User 100 can enter DND (Do Not Disturb) mode by switching between pre-defined
inbound plans to route incoming calls to his/her voicemail temporarily during lunchtime.

The following samples show two different ways of implementation. More samples about
changing active plan with switch plan are available at Brekeke Wiki:
http://wiki.brekeke.com/wiki/Switch-plan-in-user-extension

5.11.1. Sample with Switch Plan IVR Extension

Step 1: Setting Up a Switch Plan Extension
1) From [Extensions] > [IVR], click the [Create a new IVR] button.
2) Set as follows:
[Extension] 1004
[Type] Switch Plan
[Plan number] 2
[on/off] yes
3) Save the settings.
Step 2: Setting Up Plans in User Extension
1) From the user extension 100 > [Inbound] page, click the [Add new plan] button.
“Plan 2” will show up in the plan dropdown list window.
2) Uncheck all phones listed in Plan 2 and set [Other Forwarding Destinations*] as “vm100.”
3) Save the settings.
Step 3: Enter “Do Not Disturb” Mode
1) User 100 dials the Switch Plan extension 1004 from his/her phone to enter DND mode.
At user 100 > [Inbound] page, Plan 2 will be set as active.
2) Any calls to user 100 at this time will be directed to user 100’s voicemail inbox.

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Step 4: Remove “Do Not Disturb” Mode

1) User 100 dials Switch Plan extension 1004 from his/her phone again to remove DND
mode.
At the user 100 > [Inbound] page, Plan 1 will be set as active again.
2) After exiting DND mode, any calls to user 100 will be directed to his/her assigned phone(s).

5.11.2. Sample with Timer

Step 1: Setting Up Plans in User Extensions
1) At the user extension 100 > [Inbound] page, click the [Add new plan] button.
“Plan 2” will show up in the plan’s dropdown list window.
2) Uncheck all phones listed in Plan 2 and set [Other Forwarding Destinations*] as “vm100.”
3) Save the settings.
Step 2: Setting Up Timer
1) At the user extension 100 > [Inbound] page, click [Timer 1].
2) Set Timer 1 as below:
[Term] Set Schedule starting and ending Date and Year.
[Days] Check 1st - 5th and Monday through Friday.
[Time] Select and check:
- 12:00 Plan 2
- 13:00 Plan 1
3) Save the settings.
Step 3: Enter “Do Not Disturb” Mode
Every workday at 12 pm, user 100 will enter DND mode.
According to the user 100 > [Inbound] page, Plan 2 will automatically be set as active. Any
calls to user 100 between 12 pm and 1 pm will be directed to user 100’s voicemail inbox.
Step 4: Remove “Do Not Disturb” Mode
Every workday at 1 pm, Plan 1 on the user 100 > [Inbound] page will be set as active again.
After exiting DND mode, any calls to user 100 will be directed to his/her assigned phone(s).

5.12. Setting Up Conference Call

5.12.1. Creating a Conference Room
The first step to using the Conference Call feature is to set up a Conference Room. In the
following example, extension 2000 is set up as the conference number:
9 If you are using a SIP phone that has a “Conference Call” button, please consult the manufacturer of

your SIP phone for information on how to set up the Conference Call feature.

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1) From [Extensions] > [Conference], click the [Create a new conference] button.
2) Set as follows:
[Extension] 2000
Leave all other settings as default.
3) Save the settings.
With the above settings, any user can enter the conference room by dialing 2000.

5.12.2. Limiting Members Who Can Enter the Conference Room

You can limit the members who can join the conference by specifying individual members (for
example, “101,102,103”) at the [Applies to (Caller numbers)*] field. With this setting, only users
101, 102 and 103 will be allowed to join the conference. No other users can join this conference
room.

5.12.3. Simultaneous Calls to All of the Conference Members

A conference member can convene all members of the conference room at once. For example,
set 101, 102 and 103 at [Forwarding destinations*]. By dialing 2000, all conference members
(101, 102 and 103) will be invited simultaneously.

5.12.4. Starting a Conference Call (Alternate Methods)

The Brekeke PBX Users Guide describes additional methods for starting a conference call.

5.13. Setting Up Callback

The Callback feature is a method of making low-cost international calls via a third country.
Brekeke PBX will ring a caller back when the caller dials the number of a Callback extension.

In the following example, extension 3000 is set up as a Callback extension:

1) From [Extensions] > [Callback], click the [Create a new callback] button.
2) Set as follows:
[Extension] 3000
[Ringer time (sec)] 90
[Forwarding destination (No answer)] 100
[Callback callee] 1002
3) Save the settings.

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When a caller dials a Callback extension, the caller will hear a ring tone. If the caller hangs up
before the ringer times out, Brekeke PBX will send an INVITE to the caller, who will then be
connected to the number set in the [Callback callee] field. In this example, the caller will be
connected to Auto Attendant 1002.

When a caller who dials a Callback extension does not hang up before ringer timeout, the call
will be directed to the destination set in the [Forwarding destination (No answer)] field.

5.14. Setting Up Confirm Call

1) Go to Brekeke PBX Admintool > [PBX Admin] > [Voice Prompts] and upload an audio file
named "confirmcall," which will play a voice prompt to let the call recipient press a confirm
key to establish the call.
2) Go to Brekeke PBX Admintool > [PBX Admin] > [ARS] and create a new ARS Route to use
Confirm Call, or to add the Confirm Call setting to an existing ARS outbound route.
3) At "Patterns – OUT" in the ARS Route, set the voice prompt name uploaded in step 1 to
[Confirm] field under [Deploy patterns]. You need to enclose the voice prompt name with
curly brackets "{" and "}". In this example, the [Confirm] field is set as {confirmcall}.
4) Specify the confirm key in the [Key] field next to [Confirm] field. The default key is 5.
5) Make outbound calls by applying the ARS route with confirm call setting. When the call
recipient answers the call, he/she will first hear the voice prompt "confirmcall." If the call
recipient presses the confirm key (5) from his/her phone keypad before the voice prompt
ends, the call will be established between the caller and callee. Otherwise, Brekeke PBX
will disconnect the call.
9 If the [Confirm] field is set to {confirmcall}{name:&f1}, then the callee will hear the caller’s name (if

available) or the caller’s phone number played after the voice prompt. In this case, you need to set

[From] field in the Matching patterns, e.g., sip:(.*)@

5.15. Setting Up Paging

The phones that support paging will answer incoming calls automatically without taking the
handset off-hook when these phones receive the paging information sent in the SIP header from
Brekeke PBX.

A list of SIP phones that work with the Brekeke PBX paging function, as well as its sample
configuration are available at the Brekeke Wiki:
http://wiki.brekeke.com/wiki/paging-function_phone-list

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5.16. Setting Up Busy Lamp Field, Presence and Shared Call Appearance
♦ BLF (Busy Lamp Field)
With BLF, when there is a call to a monitored phone, the corresponding key lamp on the
monitoring phone will flash and the call can be picked up from the monitoring phone.
♦ SCA (Shared Call Appearance)
With SCA, users can monitor the external line status and select an available line to place an
outbound call or to answer incoming calls.
♦ Presence
With presence, Brekeke PBX can handle presence SUBSCRIBE requests from phones and
return NOTIFY responses about the status of the monitored phones, such as "available" or
"on the phone."

A list of SIP phones that work with these functions, as well as their sample configurations are
available at the Brekeke Wiki:
http://wiki.brekeke.com/wiki/BLF-SCA-and-Presence

5.17. Setting Up Video

From Brekeke PBX v3.3.x and later, Brekeke PBX supports video calls.

Video can be enabled from several locations in Brekeke PBX: Options, ARS Settings and User
Settings. Video setting enabled in the phone type set under Users > [Phones] -> [Phone Type]
has the highest priority. When the default setting is defined in [Phone Type] > [Video] field, the
video setting under [ARS] route will be applied if the call can apply any route. When video
setting in both User [Phone Type] and ARS route is set to the default, the [Video] setting at
[Options] will be applied to the call.

The following steps show how to enable video for the phone assigned to a Brekeke PBX user
extension.

1) From Brekeke PBX Admintool > [Options] > [Phone Type] page, create a new phone type
2) At new phone type setting page, set [Video] field as on and set proper audio codecs in
[Codec Priority] field.
3) Save the new phone type and restart Brekeke PBX from admintool.
4) From a PBX user extension > [Phones] page, select the phone type created above at the

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[Type] field under the phone which supports video for the call. And save the changes.
5) The call between PBX users with video phone type or from/to external user through ARS
route with [Video] field set as "on" will have video feature.

5.18. Setting Up SRTP

From Brekeke PBX v3.2.x and later, Brekeke PBX supports SRTP.

SRTP can be enabled from several locations in Brekeke PBX: Options, ARS Settings and User
Settings. SRTP setting enabled in the phone type set under Users > [Phones] -> [Phone Type]
has the highest priority. When the default setting is defined in [Phone Type] > [SRTP] field, the
SRTP setting under [ARS] route will be applied if the call can apply any route. When SRTP
setting in both User [Phone Type] and ARS route is set to the default, the [SRTP] setting at
[Options] will be applied to the call.

5.19. ARS Settings

In Brekeke PBX v3.0.x, by clicking a route name on the [ARS] > [Running Status] page, the
details of an active ARS route can be displayed. Also, from the [ARS] > [Settings] page,
administrators can perform numerous actions on an ARS route, such as copy, delete and edit.

From Brekeke PBX v3.1.x, by clicking [View] in the ARS route list’s [Status (Reg/Route)] column,
the details of an active ARS route can be displayed. An ARS rule can also be edited by clicking
the ARS route name. If the ARS rule is a template with the [Template] field checked, you can
click [Edit Template] on the [Variable Setting] page to change the template setup and the
variables’ setups for each route under the template.

In Brekeke Multi-Tenant PBX, the tenant name needs to be set for any ARS rule that will be
used exclusively by a specific tenant. The tenant name is set in the ARS rule > [General] >
[Tenant] field.

By assigning a tenant name in ARS rules for inbound calls, Brekeke Multi-Tenant PBX will
associate inbound calls with this tenant and will route inbound calls to the proper tenant
extensions.

By assigning a tenant name in the ARS rules for outbound calls, Brekeke Multi-Tenant PBX will
look for the ARS route for this tenant, determine who the caller is and then route the call to the

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destination set for this tenant. If no tenant name is assigned in the ARS outbound route, the
route will apply to calls from any tenant.

5.19.1. Route Settings / Route Template

♦ General
Use the [General] section to enable or disable either an ARS template or any of the settings
required by Brekeke PBX features, such as [Tenant] in Multi-Tenant PBX if you want to
restrict route usage to a certain tenant, the [LineKey] and [Resource] fields for Shared Call
Appearance, the [Group] field for ARS group, the [External] field for call logs database and
for billing and so on.

♦ Registration
Register your VoIP gateway port SIP number at Brekeke PBX bundled SIP Server. Brekeke
PBX will accept calls from this gateway even if the gateway is not registered at Brekeke
PBX bundled SIP Server.

If a provider needs authentication information from a caller who makes outbound calls from
UAs to an ITSP, Brekeke PBX will send the username and password that are set in this
section to the provider in an IP address or in the domain set in the [Proxy Address] field.

♦ Patterns
Define Patterns IN and OUT to receive and send calls from/to SIP devices and services.

♦ Priority
It is useful to set priorities when there are multiple options for making calls, such as when
you have multiple PSTN gateways for outbound calls or when you subscribe to multiple
VoIP service providers. A lower number holds the higher priority.

♦ Max Sessions
Set the [Max Sessions] field to define the maximum number of sessions that can be
handled by each pattern, such as when there is a limited number of Gateway channels or
subscribed lines for your SIP services. Setting this field to “-1” specifies an unlimited
number of sessions.

For ARS routes in the same group, there is only one session counter. The session counter

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for the group will increase by one when there is a call through any ARS route in this group,
regardless of what is specified in Pattern IN or OUT. If this session counter is equal to any
pattern [Max Sessions] value that is set in the same group’s ARS routes, the next matched
session cannot apply to this pattern (IN or OUT) or to any ARS routes when all of the
patterns’ [Max Sessions] for the ARS routes in the same group are set to the same value.

5.19.2. Global Route Variables (Brekeke PBX v3.1.x and later)

Global variables can be accessed from any ARS route. The update of a global variable will affect
the ARS routes in which the related variable name is set.

From [Field Settings], administrators can set a global variable’s field type, name and description,
and can also choose whether or not a global variable will be displayed on the ARS top page.

5.19.3. Variable Setting

Brekeke PBX v3.1.x and later

From the ARS template’s [Variable Setting] page, administrators can create routes that use the
same ARS patterns template and can also set the values of the variables to be used by each
route.

♦ Common Settings
Common variables can be accessed from any route created under the same ARS template.
Updating a common variable will affect the ARS routes under the current ARS template.
From [Field Settings], administrators can set a common variable’s field type, name, initial
value and descriptions, and can also choose whether a common variable will be displayed
on the [Variable Settings] page.

♦ Route Local Settings

Create new routes and assign values to the route’s local variables defined in the ARS
template for each route, and also deploy multiple routes based on the same ARS template.
From the [Field Settings] page, administrators can change the route’s local
variables’settings and also write JavaScript code to control the variable values set in each
route.

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Sample script:
if( v1 == 3333 )
v5= "192.168.200.20";
else
v5= "192.168.200.30";

Set the above script in the [Route Local Settings] > [Field Settings] -> [Script] field. If a
route has variable v1 with value 3333, this route variable v5 will be set as “192.168.200.20”
automatically. The route variable v5 will be set as “192.168.200.30” when variable v1 is set
to any value other than 3333.

Brekeke PBX v3.0.x

The variables defined in an ARS route can be edited from the [Edit Variables] page at the
upper-right of the ARS route template.

Variables Default value

v1 User ID/Number
v2 Password
v3 - v9 Customizable fields

5.19.4. ARS Default Plug-ins

Brekeke PBX offers the default plug-ins that use Notes. The default plug-ins are contains,
lookup, and matches.

By using these plug-ins, you can search for a caller's number in a telephone directory (making
notes under [Options] > [Notes] to save data list) and handle the call as your need, such as,
decline the call, change the caller ID, or redirect the call.

For the details about the Brekeke PBX default plug-ins and how to configure ARS route to use
the plug-ins, please refer to “Developer's Guide: ARS plug-in”.

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5.20. PSTN Access Using a VoIP Gateway

Using a SIP-compliant VoIP Gateway, Brekeke PBX users can receive calls from a Public
Switched Telephone Network (PSTN) and make calls to PSTN lines.

5.20.1. VoIP Gateway Setup

Set the following at your VoIP Gateway:
SIP proxy address IP address of Brekeke PBX
Dialing number sent to Brekeke PBX PSTN line number

5.20.2. ARS Route Setup

Set up “Patterns – IN” and “Patterns – OUT” in the default Gateway ARS Route to receive and
make calls from/to the Gateway.
9 Default ARS rule “gw1” is the setup template for one-stage dialing.

9 Default ARS rule “gw2” is the setup template for two-stage dialing.

Use the following settings to register the gateway at the Brekeke PBX bundled SIP server. Many
PSTN Gateways have a short interval between sessions, during which time the line is
unavailable. You can modify the [Session interval (ms)] field setting to reflect this delay as
needed.
9 In Brekeke PBX v3.1.x, check the [Template] field to enable the [Variable Setting] page.

[Registration]
Register URI sip:&v1@127.0.0.1 Register expire (sec) 3600
Proxy address 127.0.0.1 Register update period (%) 90
User Password

5.20.3. Receiving PSTN Calls

Create Gateway ARS Route “Patterns – IN” to receive calls from a gateway.
[Patterns – IN]
Matching patterns Deploy patterns
From
To sip:&v1@ &v3

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5.20.4. Calling PSTN Numbers

One-Stage Dialing
If your VoIP Gateway supports One Stage Dialing, a Brekeke PBX user can make a PSTN direct
call by setting an ARS Route as follows:

[Patterns – OUT]
Matching patterns Deploy patterns
From
To sip:([0-9]{7,25})@ sip:$1@gw_IPaddress

In this example, Regular Expressions were used to define the Matching and Deploy patterns. A
Brekeke PBX user who dials a number that has between 7 and 25 digits will be considered as a
PSTN call. Brekeke PBX will apply the above ARS Route and the call will be sent to the
gateway.

If you use multiple VoIP Gateways for outbound calls, you can define more detailed dialing
patterns in [Matching patterns] > [To], as well as change the [Priority] field as necessary to
define the usage order of Gateways. Please note that lower numbers hold the higher priority.
You can use the [Max Sessions] field to define the total number of sessions handled by each
pattern.

Two-Stage Dialing
If your VoIP Gateway supports Two Stage Dialing, have the gateway’s PSTN port register with
Brekeke PBX bundled SIP server. Let’s suppose the gateway’s PSTN port has the SIP user
name 111. To call a PSTN number, dial the gateway’s registered PSTN port number (in this
example, it’s number 111) and then dial the destination PSTN number. Alternatively, you can
configure the DTMF setting at the ARS Route OUT pattern so that the dialed numbers will be
sent to the gateway as DTMF tones.

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The OUT pattern to send a destination number by DTMF in Two-Stage Dialing is as shown
below:
Patterns – OUT
Matching patterns Deploy patterns
From From
To sip:111(.+)@ To sip:111@gw_IPaddress
DTMF $1
9 If a delay is needed before sending a DTMF call, set the [DTMF] field as {file_name}$1. Default files

are {120ms}, {240ms}, {500ms}, {1sec} and {2sec}. Customized files can be uploaded from Brekeke

PBX Admintool > [PBX Admin] > [Voice prompts] and set [Language]: Common.

5.20.5. Assign Values to Variables

Click the [Variables] link at the upper-right corner of the ARS Route template page. At the
[Variable Setting] page, select [New Route] and assign values to the variables defined in the
above Patterns – IN and Patterns – OUT.
Route Name Gateway1 (Brekeke PBX v3.1.x and later)
Tenant A tenant name (Multi-Tenant PBX)
v1 PSTN line number set in section “VoIP Gateway Setup”
v3 Specify a Brekeke PBX extension number, such as 100.

5.21. Connecting with Internet Telephony Service Providers (ITSPs)

5.21.1. Account Information for Third-Party SIP Server

Acquiring the information shown below is necessary to connect with a third-party SIP server.
Phone number 6504106636
SIP server IP address sample_proxy.com
User ID 6504106636
Password 6636
9 Depending on the provider, there may be restrictions for connecting to services, such as available

information or equipment used to connect. Please contact your VoIP service provider for more

details.

9 Please note that we do not guarantee connection with third-party products.

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5.21.2. Setting ARS for ITSP using multiple accounts

9 In Brekeke PBX v3.1.x, check the [Template] field to enable the [Variables] setting page.

[Registration]
Field name Sample settings Explanation
Register
sip:&v1@sample_proxy.com Enter SIP URI.
URI
Can be omitted when Proxy address is
Proxy
sample_proxy.com the same as the one in the [Register URI]
address
field.
User &v1 Set value at [Variables Settings] page.
Password Set value at [Variables Settings] page.
(Brekeke &v2 This field will be displayed in text format
PBX v3.0.x) after saving.

[Patterns – IN]
In the Patterns – IN example below, Brekeke PBX user extension 100 (“&v3” value, set at the
[Edit Variables] page) is set to ring when a call comes through a third-party SIP server. Leaving
the “From” field blank carries over the Caller ID information.
Matching patterns Deploy patterns
To sip:&v1@ &v3
9 Check [Apply to Request URI instead of To] when To header sent from ITSP is different from To

defined in the ARS Route Patterns – IN.

[Patterns – OUT]
Patterns – OUT defines patterns for converting the SIP URI to match your VoIP provider’s
header format requirements. In the example below, dialed numbers with 7 to 25 digits will be
directed through the VoIP service provider. To ensure that the recipient’s caller ID display will
function, the “From header” will change according to the rules of the provider. Set [Priority] and
[Max Sessions] as you need.
Matching patterns Deploy patterns
From “&v1” <sip:&v1@sample_proxy.com>
To sip:([0-9]{7,25})@ sip:$1@sample_proxy.com
9 Some VoIP service providers restrict the connection when the FROM or TO header information is

different from their own header format.

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[Variables ]
Click the [Variables] link at the upper-right corner of the ARS Route template page. On the
[Variable Setting] page, select [New Route] and assign values to variables defined in the above
Patterns – IN and Patterns – OUT.
Route Name Route 6504106636 (Brekeke PBX v3.1.x and later)
Tenant A tenant name (Multi-Tenant PBX)
Password / v2 6636
v1 6504106636
v3 Specify a Brekeke PBX extension number, such as 100.
9 For Brekeke PBX v3.0.x, set related variables from the [Edit Variables] page.

5.22. ARS Outbound Route Failover

Utilizing the Automatic Route Selection (ARS) outbound route failover feature allows users to
create redundant telecommunications systems. If an outbound route is not available or usable,
Brekeke PBX will direct the session to an alternative route.

5.22.1. Usage Examples

♦ Brekeke PBX provides automatic failover to an alternative ITSP service in the event of a
failure at your specified ITSP service.
♦ Brekeke PBX provides automatic failover to an analog telephone session via PSTN
Gateway in the event of a failure at your specified ITSP service.
♦ Creating redundant analog telephone connections with multiple PSTN Gateways.

5.22.2. Settings Examples

The ITSP line is set for regular outbound sessions. When the ITSP line fails, the outbound
sessions will be routed through the PSTN Gateway.

In the following example, two ARS routes are created: “ITSP_A” and “MyGateway.” The route
with the highest priority, “ITSP_A,” will be used for outbound calls with a dialing numbers that
have 7 to 25 digits. If there is no response within four seconds (Response timeout: 4000 ms) for
INVITE messages or a “500-599” response is received, Brekeke PBX will continue searching for
the next route that matches the outbound session request. In this case, the next-highest
prioritized route, “MyGateway,” will be chosen as the alternative route for the session. Since the
recovery time is set to one hour (3600000 ms) in “ITSP_A,” the matching sessions will be routed
through route “MyGateway” for one hour after the failover. If “ITSP_A” is running again within an

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hour, the sessions will once again be routed through the highest priority route, “ITSP_A.”

Route name: ITSP_A

Patterns – OUT
OUT – 1 Matching patterns Deploy patterns
Priority 1 From From “xxx”<sip:xxx@itsp.com>
Max
4 To sip:([0-9]{7,25})@ To sip:$1@itsp.com
sessions

Parameters
Next route on Disable on
Yes Yes
failure registration failure
Response
4000 Error codes 500-599
timeout (ms)
Recovery
3600000 Disable on failure This route
time (ms)

♦ [Response Timeout (ms)] should be adjusted according to your environment. PSTN

Gateways and SIP servers located in the local network may not require this long of a
Response Timeout interval. For a route that requires an Internet connection or if some
other kind of delay is expected, the Response Timeout intervals should be set to a longer
time.

♦ [Disable on registration failure] is set to “yes” in Route “ITSP_A.” When registration is not
working property at the “ITSP-A” route, it will be disabled and the “MyGateway” route will
be used instead.

♦ [Disable on failure] is set as “This route,” which will disable the whole route. If there are
other IN/OUT patterns defined in this route, they will be unusable when failover happens. If
[Disable on failure] is set as “This pattern,” it will only disable the current pattern, so other
patterns in this route will still be usable.

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Route name: MyGateway

Patterns – OUT
OUT – 1 Matching patterns Deploy patterns
Priority 100 From From
Max
4 To sip:([0-9]{7,25})@ To sip:$1@GW_IPaddress
sessions

5.23. ARS Route with Resource Setting (Brekeke PBX v3.1.x and later)
When multiple ARS routes for Shared Call Appearance (SCA) are set with the same name in the
[Resource] field, Brekeke PBX SCA users will have more ARS route choices for making
outbound calls than by using the ARS settings without this [Resource] setup.

Set up the ARS template and Brekeke PBX users as described below and assign phones with
the Shared Call Appearance feature enabled to users.

Route Template:
[General]
Route name outbound
Template checked
Line key checked
Resource &v9

[Patterns – OUT]
OUT – 1 Matching patterns Deploy patterns
Priority &v1 From From
Max
&v2 To sip:([0-9]{7,25})@ To sip:$1@&v3
sessions

Variable Setting:
Route Name v1 v2 v3 v9
1 90 1 192.168.200.10 gw
2 90 -1 192.168.200.20 itsp
3 100 1 192.168.200.30 gw

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Brekeke PBX SCA Users

User: 100
[Settings] -> [Resource map]
line/1=gw/1
line/2=gw/2

User: 101
[Settings] -> [Resource map]
line/1=gw/1
line/2=itsp/1

When user 100 makes an outbound call from a phone with shared line 1, the call will apply route
“outbound*1” with resource set as “gw” and the call will be sent to the destination IP
192.168.200.10.

While user 100 is talking, if user 101 also makes an outbound call from shared line 1, the call will
apply route “outbound*3” with resource also set as “gw” and will be sent to the destination IP
192.168.200.30.

Because route “outbound*1” with resource “gw” has the [Max Sessions] variable v2 set to 1 and
one session from user 100 is using this route, Brekeke PBX looks for the next available route
with resource “gw” for the user 101 outbound call, which is route “outbound*3.”

If user 101 makes an outbound call from shared line 2, the call will apply route “outbound*2” with
resource “itsp” and will be sent to IP 192.168.200.20.

5.24. ARS Route with Group Setting

Set up the ARS template and Brekeke PBX users as described below and then assign phones
with the Shared Call Appearance feature enabled.

Route Template:
[General]
Route name outbound
Template checked
Group &v8

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Line key checked

Resource &v9

[Patterns – OUT]
OUT – 1 Matching patterns Deploy patterns
Priority &v1 From From
Max
&v2 To sip:([0-9]{7,25})@ To sip:$1@&v3
sessions

Variable Setting:
Route Name v1 v2 v3 v8 v9
1 90 1 192.168.200.10 group1 gw
2 90 -1 192.168.200.20 itsp
3 100 1 192.168.200.30 group1 gw
4 110 1 192.168.200.40 group2 gw

Brekeke PBX SCA Users

User: 100
[Settings] -> [Resource map]
line/1=gw/1
line/2=gw/2

User: 101
[Settings] -> [Resource map]
line/1=gw/1
line/2=itsp/1

If user 100 makes an outbound call from shared line 1, the call will apply route “outbound*1” with
resource set as “gw” and group as group1 and the call will be sent to the destination IP
192.168.200.10.
While user 100 is talking, if user 101 also makes an outbound call from shared line 1, the call will
apply route “outbound*4” with the resource set as “gw” and the group as group2 and the call will
be sent to the destination IP 192.168.200.40.

As explained in the above section about Max Sessions, routes in the same group use the same

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session counter. In this example, routes “outbound*1” and “outbound*3” with resource “gw” are
both in “group1” and both routes have the [Max Sessions] variable v2 set to 1. When user 100 is
making a call through route “outbound*1,” the number of sessions in group1 is 1, which will
reach the [Max Sessions] in both routes “outbound*1” and “outbound*3.” Brekeke PBX will then
look for the next available route with resource “gw” for user 101’s outbound call. This is route
“outbound*4.”

The above example explains that the [Resource] setting is used for Brekeke PBX users to look
for available routes to make outbound calls and that the [Group] setting is used to limit max
sessions through the routes of both Patterns – IN and Patterns – OUT in the same group.

9 Multiple groups can be set in an ARS route [Group] field with group names separated by commas.

9 When a session goes though a route set with multiple groups, the session counter of the related

groups will increase by 1.

5.25. DID
From the [DID] menu, administrators can access and modify the active routes setup.
For Multi-Tenant PBX, only the active DID route related to the tenant will appear in tenant
administrator accounts. Tenant administrators can modify the route fields as necessary.

The following steps show how to enable DID access for a route and to set DID access
privileges.
Step 1: Enable DID Menu
1) Log in to Brekeke PBX Admintool as system administrator (sa).
2) Go to the [Options] > [User Access Settings] page in the [Admin Menu] section.
3) Select on in [DID] menu.
Step 2: Enable a Route’s DID Access
1) Log in to Brekeke PBX Admintool as system administrator (sa).
2) Go to [ARS] and select an active ARS route template that can be accessed from a non-sa
administrator’s [DID] menu.
3) Go to the selected route template’s [Variable Setting] page, click on the [Field Settings] link
next to [Route Local Settings].
4) Select [Yes …] in the [DID] field.
[No] disables access to the routes from the DID menu in administrators’ accounts.
[Yes (Modify only)] enables access to the active routes under the route template from

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administrators’ accounts, which administrators can use to modify a route’s variable

settings.
[Yes (Modify/Add/Delete)] enables access to all routes under the route template from
administrators’ accounts, which administrators can use to modify a route’s variable settings,
to enable or disable a route and to add or delete a route.
9 The [Yes (Modify/Add/Delete)] option is not available for Brekeke Multi-Tenant PBX.

5) For Brekeke Multi-Tenant PBX, set the tenant administrator’s access privilege for each
variable to one of the following options:
[Tenant Access (List)] enables or disables displaying of the variable in the route table of
tenant administrator accounts.
[Tenant Access (Edit)], if checked, allows the variable settings to be modified from tenant
administrator accounts. If unchecked, the variable settings will not be modifiable from
tenant administrator accounts.

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6. Setup Items
6.1. Start / Shutdown
At the [Start / Shutdown] page, the system administrator can perform the following functions:
Check Brekeke PBX and its bundled SIP Server running status; check current events, such as
ARS route registration history; restart or shutdown Brekeke PBX and bundled SIP Server.

6.2. Options
The following list displays the settings under the [Options] menu, which is only available to
system administrators:

6.2.1. Settings
♦ General Settings
Name Default value Description
Auto: Brekeke PBX starts up automatically with the
Tomcat (Brekeke PBX HTTP Service).
Start up Auto
Manual: Start up manually.
Options: Auto / Manual

♦ PBX System Settings

Name Default value Description
Call pickup prefix * Prefix for picking up calls.
Port number that Brekeke PBX will use.
Port number 5052
Modify as needed to avoid port conflicts.
Max concurrent Depends on Maximum number of concurrent sessions that
sessions license Brekeke PBX can handle (cannot be modified).
Max number of Depends on Maximum number of SIP UAs that Brekeke PBX
UAs (User Agents) license can handle (cannot be modified).
Minimum port number the RTP Protocol uses for
Min RTP port 30000
sending voice data. Adjust setting as needed.
Maximum port number the RTP uses for sending
Max RTP port 49999
voice data. Adjust setting as needed.
on – RTP is handled by Brekeke PBX.
RTP relay on off – RTP is not handled by Brekeke PBX.
(Applied unless there is a different RTP relay setting

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specified at the user phone type or the ARS

Routes.)
on – need to set [RTP relay] on, enable SRTP for all
calls at Brekeke PBX system
SRTP off off – applied unless there is a different SRTP setting
specified at the user extension phone type or the
ARS route [SRTP] field
on – need to set [RTP relay] on, enable video for all
calls at Brekeke PBX system
Video off off – applied unless there is a different video setting
specified at the user extension phone type or the
ARS route [Video] field
G.711 u-law (PCMU) is used by default. When
specifying multiple payloads, separate them with a
comma (,). The following payload types can be used
at Brekeke PBX:
0 - G.711 u-law
8 - G.711 A-law
18 - G.729
Codec priority 0
98 – iLBC
If the codec priority is not set in ARS Routes or user
phone type, this setting will be applied.
From Brekeke PBX v3.3.x, Brekeke PBX supports
dynamic payload type of iLBC even only payload 98
can be set for iLBC from Brekeke PBX Admintool
[Codec priority] field.
Use the codec setting that is preferred at the remote
Use remote SIP UA. If “default” is set in [Use Remote Preferred
no
preferred codec Codec] or in ARS or user phone type, this setting
will be applied.
Max concurrent
Maximum number of concurrent sessions with call
recording 10
recording.
sessions
Use user’s name
Yes Use user account name as display name.
as Display Name

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Ringing timeout Amount of time to wait for an answer from the dialed
240000
(ms) party after ringing starts.
Maximum length of time a call can last in talking,
Talking timeout
259200000 measured by the amount of time since the last SIP
(ms)
packet was received. Value 0 signifies infinite.
Maximum number of SIP servers or Brekeke PBX
Max hop number 20
that a call can go through (hop number).
Days to keep call
90 Number of days to keep call logs.
logs
Session Timer Interval to allow UAs and SIP server to determine
0
(sec, 0=disable) whether the SIP session is still active.
Interval to send keep-alive packets to UAs during a
Session keep alive
600 call when RTP relay is set to off and session timer
(sec)
has not been used.
RTP Session Timeout value for Brekeke PBX awaiting the next
600000
Timeout (ms) RTP packet.
Enable (on) / Disable (off) for using reliable
100rel off
provisional responses (1xx series).
RFC2833 on Enable (on) / Disable (off) RFC2833 setting.
Web service security – used by Brekeke PAL and
Valid client IP
Brekeke Web Service; Set with regular expressions
pattern
of the PAL and Web Service clients IP pattern.
Java VM
Parameters passed to VM.
arguments

♦ Media Server System Settings

Name Default value Description
Port number that the Media server system uses.
Port number 5056
Modify as needed to avoid port conflicts.
Max concurrent Depends upon Maximum number of concurrent sessions for the
session limit the license Media server (cannot be modified).
G.711 u-law (PCMU) is used by default. Separate
Codec priority 0 with comma (,) when specifying multiple codecs.
Also see the PBX [Codec priority] description.
Use remote no Enable (no) / Disable (yes) on using remote codec

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preferred codec by the endpoints.

Max stored Maximum number of saved voicemail messages,
50
messages plus any recorded files for each user’s voicemail.
Message Maximum length of recording time for a voicemail
recording length 600 message. If [Message recording length (sec)] in
(sec) User setting is blank, this value will be applied.
Days to keep Number of days before unsaved messages are
unsaved 30 deleted automatically from each user’s voicemail
messages inbox.
Conversation
recording length 3600 Maximum recording length for each call.
(sec)
Conversation Yes – save recording file in user’s voicemail box.
recording file in Yes No – don’t save recording file in user’s voicemail
voicemail inbox box, but show recording files in call logs.
Days to keep
This field will be enabled when [Conversation
conversation 30
recording file in voicemail inbox] is set to No.
recording files
Minimum port number the RTP uses for sending
Min RTP port 50000
voice data.
Maximum port number the RTP uses for sending
Max RTP port 59999
voice data.
Ringing timeout Timeout value for awaiting an answer from the
240000
(ms) dialed party once the ringing starts.
Maximum length of time a call can last in talking,
Talking timeout
259200000 measured by the amount of time since the last SIP
(ms)
packet was received. Value 0 signifies infinite.
RTP Session
600000 Timeout value for awaiting the next RTP packet.
Timeout (ms)
Java VM
Parameters passed to VM.
arguments
9 ms = 0.001 second

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♦ Email Settings
Name Default value Description
The SMTP server address for sending email
SMTP Server notifications when the user receives a new
voicemail message.
SMTP port 25 SMTP server’s listening port.
SMTP Enable (on) / Disable (off) SMTP authentication
on
authentication setting.
Encrypted Enable (on) / Disable (off) Encrypted Connection
off
connection (SSL) (SSL), available since version 2.3.
Address of the POP3 server (for POP-before-SMTP
POP3 server
authentication).
POP3 port 110 POP3 server’s listening port.
Email account username for the above SMTP
User
server.
Password Email account password.
Password
Input field for confirming the above password.
(confirm)
Email address
Email notification sender’s address.
(from)
Email subject for the email notifications.
The following variables can be configured:
{from}: SIP URI who left the voicemail message
{to}: voicemail box’s SIP URI
voicemail({to})
Email subject {from-number}: number in {from}
: from {from}
{to-number}: number in {to}
{time}: time when the messaged is recorded
{recording-length}: length in time of the recorded
message
from:{from}
to:{to}
Email body for email notifications.
time:{time}
Email body Variables that can be used in this field are the same
recording
as in the [Email subject] above.
length(sec):{re
cording-length

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♦ Multi-tenant Settings (Multi-Tenant PBX)

Name Default value Description
on – G.729 codec license is shared among tenants.
G. 729 License
on off – The number of G.729 codec licenses is set on
Sharing
each tenant’s [Options] page.

♦ PAL Settings
Name Default value Description
Sends notification to Brekeke PAL or Brekeke PAL
Notification for
yes WebSocket application (or not) when a user phone
registration
has registered.
Web service security – used by Brekeke PAL
Valid WebSocket
WebSocket; Set with regular expressions of the
client IP Pattern
WebSocket clients IP pattern .

6.2.2. User Access Settings

Restricts the features and menus that are displayed and modifiable from a User or Admin
account when the user logs into Brekeke PBX Admintool.

6.2.3. Phone Type

Name Default value Description
Type 1 The default phone types that users can choose.
Type name Type 2 User extension > [Phones] page [Type] field.
Type 3 Default setting is Type 1.
Description Describe phone type.
When set to default, user phones assigned with this
phone type will apply the RTP relay setting in the
RTP relay default [Options] menu.
When set to off, Brekeke PBX will not relay RTP for
the user phones assigned with this phone type.
default - user phones assigned with this phone type
will apply the SRTP setting in the [Options] menu.
SRTP default
optional – Both RTP and SRTP calls can be made
with the user phones assigned with this phone type.

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Available with [RTP relay] on

mandatory – only SRTP call will be made with the
user phones assigned with this phone type.
Available with [RTP relay] on
off - Brekeke PBX will not enable SRTP for the calls
with the user phones assigned with this phone type
default - the user phones assigned with this phone
type will apply the video setting in the [Options]
menu.
Video default on - Brekeke PBX will enable video for the calls of
the user phones assigned with this phone type
off - Brekeke PBX will not enable video for the calls
of the user phones assigned with this phone type
The default setting will apply the [Codec Priority]
setting in the [Options] menu.
Brekeke PBX will first apply the [Codec Priority]
Codec priority
setting in the phone type assigned to user phones if
the setting in the phone type is different from the
one in the [Options] menu.
When set to default, the setting in the [Options]
menu will be applied.
Use remote Brekeke PBX will first apply the setting in the phone
default
preferred codec type assigned to user phones if the setting in the
phone type is different from the one in the [Options]
menu.
When set to on, the phones assigned with this
phone type can use Brekeke PBX keypad
Keypad commands, such as #9, #8, etc.
on
commands When set to off, the phones assigned with this
phone type cannot use Brekeke PBX keypad
commands.
When set to off, the phones assigned with this
MWI (NOTIFY
phone type can send SUBSCRIBE for MWI.
without off
When set to on, the phones assigned with this
SUBSCRIBE)
phone type cannot send SUBSCRIBE for MWI.

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Set properties that do not have corresponding fields

Properties
in Brekeke PBX Admintool.

6.2.4. Auto Sync

The [Auto Sync] menu is used for Brekeke PBX redundancy setup. This feature requires a
license upgrade.

6.2.5. SA (Multi-Tenant PBX)

SA is the only system administrator in Brekeke Multi-Tenant PBX. From the [SA] page, you can
change the SA login password and set the system language.

6.2.6. Advanced
The [Advanced] field allows you to set properties that do not have corresponding fields in
Brekeke PBX Admintool. Please refer to other manuals and tutorials regarding the type of
properties that may be edited here.

6.3. Voice Prompts

6.3.1. System Voice Prompts

Upload a customized sound file to overwrite the system default sound file, or to use as needed.
A list of Name, Language and Description will be displayed.

Name Description
Language Choose folder in which to save an uploaded file.
Name for the uploaded file in the folder.
Name If the file name is the same as the system default sound file, the
uploaded sound file will be played.
Description A memo shown on the GUI to describe the file usage.
To upload a file, click the [Browse] button. Select the file you want to
File name
upload and click [Upload]. The upload will then start.
To download a recorded sound file, click Download ( ). The file will
Download
be downloaded to your PC as a WAV file.
To delete a recorded sound file, click Delete ( ). The selected file(s)
Delete
will be deleted.

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6.3.2. Notes for Sound Files

Uploaded sound files must be formatted as below:
Sample rate 8000 Hz
Bit-depth 16-bit
Channels Mono

You may use sound-recording applications, such as Windows Microsoft Sound Recorder, to
record sound files. We recommend that you adjust the pause and sound level to suit your
needs.

6.4. Automatic Route Selection (ARS)

Brekeke PBX automatically selects the optimum call route from the preset routing options. This
feature can be used for Least Cost Routing, traffic management and load balancing of VoIP
Gateways or PBXs.

6.4.1. Adding a New Route

1) Choose [ARS] > [New Route].
2) Type the name of a route in the input field on the new popup window.
3) Click [OK] to add the route.
9 In Brekeke PBX v3.0.x, choose [ARS] > [Settings] > [New Route].

6.4.2. Editing, Copying or Deleting a Route

1) Uncheck [Hide Disabled Rules] to show all ARS routes and details.
2) Click the ARS route name to edit a route.
3) Choose the copy or delete icon to perform the desired action.
9 In Brekeke PBX v3.0.x, select [ARS] > [Settings] from the submenu.

6.4.3. Viewing an Active Route

1) Check [Hide Disabled Rules] to display active rules only.
2) Click “View” in [Status(Reg/Route)] to show details about the active route.
For the routes in an ARS template, click the ARS route template name and then “View” an
active route under the route template.
9 In Brekeke PBX v3.0.x, choose [ARS] > [Running Status]. If no ARS route is enabled, there will be no

route displayed under [Running Status]. Selecting [Settings] will display all ARS Routes.

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6.4.4. ARS > Route Template

♦ General
Name Default value Description
Route name Name of the route.
Description Description of the route.
Define if this route is a template (Brekeke PBX v3.1.x
Template
and later).
Disabled checked Disable / Enable the ARS route.
Type Type A Used for special functions.
Group ID for a group of ARS routes.
When checked, Brekeke PBX will recognize this ARS
External unchecked
route as an external line.
LineKey unchecked Check if you use Line keys (optional feature).
Set the interval period between sessions for any
Session interval
VoIP FXO Gateways that require pausing between
(ms)
sessions.
This setting takes effect when there is a registration
setup in the route.
Apply this route If set to yes, the incoming calls will apply this ARS
Yes
for incoming calls route only when the INVITE request URL is the same
as the one in the contact header of the REGISTER
request sent by this route.
Set the resource name for users with the SCA
Resource feature enabled (optional feature, Brekeke PBX
v3.1.x and later).
Tenant
Set a tenant name.
(Multi-Tenant PBX)

♦ Registration
Name Default value Description
SIP URI that is used to register Brekeke PBX at a
remote registrar server. Leave this blank when there
Register URI
is no need to register Brekeke PBX to any remote
registrar server.

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IP address of the registrar server. This field is

Proxy address optional when the proxy address is the same as the
address set in the Register URI field.
Register expire
3600 Set when REGISTER expires.
(sec)
The percentage value of the interval until re-register
Register update
90 occurs, calculated from the length specified in the
period (%)
register expire setting above.
User ID for authentication account required by
User remote registrar server. Entry is not necessary when
authentication is not used.
Password for authentication account required by
Password remote registrar server. Entry is not necessary when
authentication is not used.

♦ Pattern – IN
Default
Name Description
value
Priority 100 Lower numbers hold a higher priority.
Specify the number of sessions
(including RINGING and BYE
Max sessions -1
sessions) that are allocated to the
route.
Disabled unchecked Enable / Disable this pattern.
Specify a matching rule for the From
header using regular expressions.
Matching patterns From
When the field is left blank, all calls will
be considered as matched.
Specify a matching rule for the To
header using regular expressions.
To
When the field is left blank, all calls will
be considered as matched.
Plugin Java class name for the plug-in.
Parameters that will be used by the
Param
plug-in.

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Pattern of the value returned by the

Return
plug-in.
Apply to If checked, compares the Request URI
Request URI unchecked instead of the To header; mostly
instead of To designed for using ITSP accounts.
Apply only to
If checked, the route only applies to the
calls related to unchecked
calls related to registration.
registration
Specify replace patterns for the From
From
header using regular expressions.
Deploy patterns Specify replace patterns for the To
To
header using regular expressions.
Custom Used for special functions.
Select RTP relay ON / OFF.
If default is selected, the setting will be
the same as [Options] > [RTP relay]
Parameters RTP relay default
(unless specified at the User settings).
on – RTP is handled by PBX.
off – RTP is not handled by PBX.
If default is selected, the setting will be
the same as [Options] > [SRTP] (unless
specified at the User settings).
off – SRTP is not handled by PBX.
optional – Brekeke PBX will handle
SRTP default
both RTP and SRTP call; available with
[RTP relay] on
mandatory – only SRTP call will be
handled by Brekeke PBX; available
with [RTP relay] on

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Specify codec to be used. Use a

comma (,) when specifying multiple
payloads. The following payload types
may be used with Brekeke PBX:
0 - G.711 u-law
8 - G.711 A-law
Codec priority 18 - G.729
98 – iLBC
From Brekeke PBX v3.3.x, Brekeke
PBX supports dynamic payload type of
iLBC even only payload 98 can be set
for iLBC from Brekeke PBX Admintool
[Codec priority] field.
Enable (on) / Disable (off) for using the
Use remote
remote codec used by the endpoints.
preferred default
When set to default, the remote codec
codec
in the [Options] setting will be applied.
Block SIP Block or pass-through the SIP INFO
no
INFO (DTMF) (DTMF) from a user to the other party.
off – PBX will not handle RTCP
Send RTCP off packets.
on – PBX will handle RTCP packets.
Default: depends on the response from
the callee.
Block: remove SDP.
SDP 18x default Append: attach SDP.
If SDP is not included in the packets,
Brekeke PBX will play the Ring-Back
Tone.
If default is selected, the setting will be
the same as [Options] > [Video] (unless
Video default specified at the User settings).
on – video call is handled by PBX,
off – video is not handled by PBX.

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♦ Patterns – OUT
Default
Name Description
value
Priority 100 Lower numbers hold a higher priority.
Specify the number of sessions
(including RINGING and BYE
Max sessions -1
sessions) that are allocated to the
priority.
Disabled unchecked Enable / Disable this pattern.
Specify a matching rule for the From
header using regular expressions.
From
When the field is left blank, all calls will
be considered as matched.
Specify a matching rule for the To
header using regular expressions.
To
When the field is left blank, all calls will
be considered as matched.
Mostly designed for multiple ITSP
Matching patterns
User ^.+$ accounts, it specifies the users to which
this ARS Route applies.
Specifies the users in the class to
Class
which this ARS Route applies.
Plugin Java class name for the plug-in.
The parameters which will be used by
Param
the plug-in.
The pattern of the value returned by the
Return
plug-in.
Specify replace patterns for the From
Deploy patterns From
header using regular expressions.
Specify replace patterns for the To
To
header using regular expressions.
Destination IP address. May omit entry
Target when the destination IP address is
specified in the To header domain.

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When DTMF needs to be issued after

calling a gateway (two-stage calling),
DTMF
you can specify the DTMF string using
some part of the [To] matching pattern.
Define the voice prompt used with a
Confirm
confirm call.
Key 5 Define the confirm key entry.
Custom Used for special functions.
Select RTP relay ON / OFF.
If default is selected, this value is the
same as in [Option] menu > [RTP relay]
Parameters
RTP relay default (unless specified at the User settings).
on – RTP is handled by Brekeke PBX.
off – RTP is not handled by Brekeke
PBX.
If default is selected, the setting will be
the same as [Options] > [SRTP] (unless
specified at the User settings).
off – SRTP is not handled by PBX.
optional – Brekeke PBX hande both
SRTP default
RTP and SRTP call; available with
[RTP relay] on
mandatory – only SRTP call will be
handled by Brekeke PBX; available
with [RTP relay] on
Specify codec to be used. Use a
Codec priority comma (,) when specifying multiple
codecs.
Stop (or not) passing DTMF from a user
Block SIP
no to the other party when Brekeke PBX
INFO (DTMF)
receives DTMF.
off – Brekeke PBX will not handle
RTCP packets.
Send RTCP off
on – Brekeke PBX will handle RTCP
packets.

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Session Interval to allow both UAs and SIP

timer(sec, 0 server to determine whether the SIP
0=disable) session is still active.
Enable (on) / Disable (off) using reliable
100rel off
provisional responses (1xx series).
If default is selected, the setting will be
the same as [Options] > [Video] (unless
Video default specified at the User settings).
on – video call is handled by PBX
off – video is not handled by PBX.
Next route on Set failover for outbound sessions (or
no
failure not).
Disable on
Enable (yes) / Disable (no) this Pattern
registration no
when registration fails.
failure
Period of time when a response has not
Response
-1 been received before timeout is
timeout (ms)
activated.
Failover will be activated when
Error codes 500 specified error codes are received for
INVITE requests.
Recovery time Period of time until this pattern will be
0
(ms) reactivated.
Disable this route when using this OUT
pattern fails. Can also be set to disable
one pattern in the ARS route or multiple
Disable on
This route ARS routes with the same group ID.
failure

Options:
This route, This pattern, This group

6.5. DID
Displays a list of DID-enabled routes. Administrators can modifiy enabled route fields, such as
changing the destination extension where inbound calls to a related DID number will be sent.
For more detailed setting instructions, please refer to the DID section above.

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6.6. Call Status

The Call Status of ongoing calls is displayed under the [Call Status] menu. By specifying search
criteria, the search result will be displayed on the screen. You can view detailed information for
the selected search result.

Name Description
Total Total number of system active sessions
ID Call ID
Status Call status: In progress, Talking
UAs Users’ phone numbers in the current session

6.6.1. Status
Name Description
ID Call ID
Status Call status
Call park The number that has been parked
Conference Conference number
Start Time the call begins

6.6.2. UAs (User Agents)

Name Description
User Username
ARS Used ARS route
URI SIP URI
Connected Time the call begins
Disconnect the call. (If the user does not have the right to disconnect,
Disconnect
this option will not be displayed.)

6.7. Call Logs

Call log information is available through Brekeke PBX. By specifying a date, you can view the
call log information for that date. By default, call log information is displayed in html in the
browser. By clicking the [csv] button, you can download a log file to your local machine in .CSV
format. Individual user call logs are also available under each user.

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6.8. Notes
This menu item is used by Brekeke PBX plug-ins to access text data or to save script files for
IVR script users. You can also use it for writing some memos.

Name Description
Name Name of the note
Description Brief description of the note
Define if this note can be accessed by users, as well as what the
User access level access level is.
Select from “No Access,” “Read only” or “Read/Write.”
Note Text field where you can write your own notes

6.9. Extensions
On the left menu panel, click the [Extension] menu. Select different extension tabs and create
extensions. Click extension ID to edit this extension setting.
For user extension setting details, refer to the Brekeke PBX User Guide.

6.9.1. System Administrator

Name Default value Description
User type Admin Administrator user
User sa Default administrator username
Login password sa Default administrator login password

6.9.2. Group Extensions

♦ Simultaneous Ring Group
Name Default value Description
Extension Extension ID
Simultaneous
Type Extension type
ring
Description Extension description
User extensions’ IDs to which Brekeke PBX will
Group extensions* forward a ring group call.
Separate user extension IDs with commas.

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Amount of time phone will ring while waiting for the

recipient to answer. After the length of time set here,
the call will be transferred to the destination that is
Ringer time (sec)* 90
specified in the [Forwarding destination (No
answer)] field. If no destination is set at [Forwarding
destination (No answer)], the call will be terminated.
Forwarding
Destination to which the call will be forwarded when
destination (no
ringer timeout has occurred.
answer)

♦ Call Hunting Group

Name Default value Description
Extension Extension ID
Type Call Hunting Extension type
Description Extension description
There are two modes for call forwarding:
Round-robin – Calls will be distributed starting from
the top of the list. When a call is received, it is
forwarded to the extension following the last
Mode Round-robin
extension that received a call.
Top-down – Calls will always be distributed in the
order listed in the field, beginning with the first
number.
Hunt group Enter extension number(s) to which all calls that are
extensions* received at this extension number will be forwarded.

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Time interval Brekeke PBX waits before ringing

each destination in the [Hunt group extensions*]
field.
Multiple time intervals can be set and separated by
commas, such as, “5,15,0,10.”
With this setting, Brekeke PBX will ring the first
Ringer time (sec)* 20
destination when there is an incoming call, then wait
for five seconds before ringing the second
destination, then wait 15 seconds to ring third and
fourth destinations together (because the third ring
time is set to 0), and then wait ten seconds to ring
the final destination.
Length of time queued calls will remain on hold
Waiting time in the
0 before forwarding to the user destination set at
queue (sec)
[Forwarding destination (No answer)].
Max number of Maximum number of calls that can wait in the queue
10
calls in the queue at one time.
Amount of time to wait between dialing the available
Call interval
3000 extensions in [Hung group extensions*] while callers
(msec)
are waiting in the queue.
Enable / Disable retrying calls when an initial call
has not been answered.
When this setting is enabled, the call will be
Single attempt no
transferred to the destination set at [Forwarding
destination (No answer)] after the initial call is not
answered.
Forwarding
Destination to which the call will be forwarded when
destination (No
timeout has occurred.
Answer)

Sound Files
Name Description
Audio file that contains the music/sound that
Music on hold
will be played when the caller is on hold.

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6.9.3. Schedule Extensions

Name Default value Description
Extension Extension ID
Description Extension description
Destination number to forward all calls that are
received by this extension. Calls will be forwarded to
Forward To
the appropriate destination based on conditions
defined in the schedule.
Specify schedule information for forwarding
Conditions
incoming calls.

Name Description
Calls will be forwarded to the destination set
Default forwarding schedule* here when no other schedules can be
applied to the call.
Calls will be directed to the destination set
Forwarding schedule [n]
here when the conditions are met.
Multiple schedule plans can be created under
Plan [n] a single extension.
The active plan will be applied to calls.
Specify the schedule when a plan will be
Timer 1/Timer 2
active.

6.9.4. IVR Extensions

For the IVR type [Flow] and [Script], refer to Brekeke PBX document – Developer’s Guide: IVR

♦ Auto Attendant
Name Default value Description
Extension Extension ID
Type Auto Attendant Extension type
Description Extension description
Language English Select IVR language.
Max input digits 4 Maximum number of input digits

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Maximum number of retries when an input error has

Max retry count 5 occurred. After this number of retries has occurred,
the call will be terminated.
Length of time that the destination phone will ring
Ring timeout (sec) 30
when a call is received via Auto Attendant.
Default destination (user extension number) for an
Default operator
incoming call that has not specified a call recipient.
DTMF timeout Length of time to wait for inputting destination
20
(sec) number
Set up “Speed dial” numbers for Auto Attendant.
Example:
• 1=100 In this case, instead of dialing 100 to
reach user 100, a caller can dial 1.
• 1==100 In this case, a caller can dial either
110# to reach user 110, or dial 1# to reach the
Speed dial
user who is set as 100 in speed dial.
• Regular expressions can be used for detecting
input number pattern
^9.+$=100 In this case, all calls with inputting
number starting with 9 will be transferred to
user 100
Transfer to Enables / Disables call transfers to an unregistered
unregistered disable user.
users Options: disable / enable

Sound Files
Name Description
Greeting message that is played for the Auto
Greeting message
Attendant.
A message to prompt the caller to re-enter the
Retry message
number when an input error has occurred.
Audio file that contains the music/sound that
Music on hold
will be played when the caller is on hold.

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♦ Add / Remove Forwarding Destinations

Name Default value Description
Extension Extension ID
Add / Remove
Type forwarding Extension type
destinations
Description Extension description
Language English Select IVR language.
By calling this extension, the caller’s extension
number will be added or deleted from the [Group
Target groups*
Extensions] field for any group extensions that are
set in this field.

♦ Switch Plan
Name Default value Description
Extension Extension ID
Type Switch Plan Extension type
Description Extension description
Language English Select IVR language.
By calling this extension, the caller extension’s
Plan number 2 [Inbound] page active plan value will change to the
value set here.
When set to yes, the plan number will be set as the
active plan when a user calls this extension. The
active plan will switch back to plan 1 when the user
calls this extension again.
On/Off Yes
When set to no, the plan number will be set as the
active plan when a user calls this extension, but will
not switch back to plan 1, regardless of how many
times the user calls this extension.

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6.9.5. Conference Extensions

Name Default value Description
Extension Extension ID
Description Extension description
By specifying user extension IDs here, a user can
Forwarding
dial this extension to invite multiple attendees to a
destinations*
conference.
Number patterns of callers who can join a
conference.
Wildcard characters can be used.
Applies to (caller
* A star (*) and a question mark (?) can be used for
numbers) *
matching meta-characters. A star (*) means zero (0)
or more characters; a question mark (?) means one
character.
Yes – when call host hangs up, conference call will
end.
Exit all when host
no No – when call host hangs up, conference call will
leaves
continue if there are attendees in the conference
room.
When set to yes, only the voice of the conference
Broadcast no
call host can be heard by other participants.
When set to yes, and attendees PBX user already
have another call in conversation at the time the
Hang up existing
no conference call is initiated, Brekeke PBX will
calls
disconnect their current talking call and ring the
attendees for conference call.

Sound Files
Name Description
Audio file that contains the music/sound that
Music on hold while waiting will be played while waiting for other
attendees to join the conference.

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6.9.6. Callback Extensions

Name Default value Description
Extension Extension ID
Description Extension description
Amount of time phone will ring before directing the
Ringer time (sec) 90
call to the forwarding destination.
Forwarding
Destination where the caller will be directed when
destination (no
ringing times out.
answer)
Destination where Brekeke PBX will direct the caller
Callback callee when he/she disconnects a call before ringer time
out.

7. Setup Items (Multi-Tenant PBX)

For tenant menu details that are not listed below, please refer to the system-level explanation
above of the Tenant menu.

7.1. Tenants
From this menu, a system administrator can create a new tenant, delete a tenant and copy and
modify existing tenant settings.
9 Shutting down Brekeke Multi-Tenant PBX from Admintool is required when deleting a tenant.

7.1.1. Copy and Set Up New Tenant

1) Click the copy icon of the tenant you wish to copy.
2) At the popup window, type in the new tenant name.
3) At the next screen, edit the setup for the new tenant and save the settings.

7.1.2. Settings Copied in New Tenant

In the new tenant, the following settings will be the same as the tenant that you copied:
• Users and user descriptions
• User settings, except [Phone ID]
• Voice prompts, including users’ voice prompts and tenant system voice prompts
• Tenant options

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Settings that will not be carried over:

• Tenant descriptions
• User extensions’ Phone IDs
• Call logs
• Voicemail – all users’ mailboxes will be empty.

7.2. Tenant Voice Prompts

Upload a customized sound file under a tenant to overwrite the system default sound file or to
use as needed. The sound files uploaded at the tenant level will take effect only for the current
tenant and will have higher priority than the sound files uploaded at system [Voice Prompts]. A
list that includes Name, Language and Description will be displayed. For setup details, please
refer to system level voice prompts.

7.3. Tenant Notes

This menu item is used by Brekeke PBX plug-ins to access text data or to save script files for
IVR script users at the tenant level. You can also use this feature for writing memos. Notes
created under a tenant have a higher priority than those created under system [Notes].
For setup details, please refer to system-level Notes.

The format for accessing tenant-level notes from Brekeke PBX plug-ins is as follows:
<tenant_name>.<note_name>

7.4. Tenant Options

Tenant settings can be modified from the Options menu under each tenant.
♦ General
Name Description
Description A brief description of the tenant

♦ Capacity
Name Description
Maximum users Maximum number of user extensions allowed under this tenant
Maximum sessions Maximum number of concurrent sessions allowed for this tenant
Maximum recording Maximum number of concurrent recording sessions allowed for this
sessions tenant
Maximum G.729 Maximum number of concurrent G.729 sessions allowed for this tenant

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license when G.729 license is not shared among system

(license option required)
Maximum number of call center agents (license option required)
Maximum agents
allowed for this tenant

♦ Features
Name Description
Voicemail Enable or disable this feature for the tenant
Auto-Attendant Enable or disable this feature for the tenant
Conferencing Enable or disable this feature for the tenant
Call Recording Enable or disable this feature for the tenant
Round Robin –
Enable or disable this feature for the tenant
Topdown
Call Queue Enable or disable this feature for the tenant
PAL Enable or disable this feature for the tenant

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